An Introduction to Computer Networks

Release 1.0

Peter L Dordal

February 04, 2015

CONTENTS

0 Preface
0.1 Classroom Use . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
0.2 Progress Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
0.3 Technical considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1 An Overview of Networks
1.1 Layers . . . . . . . . . . . . .
1.2 Bandwidth and Throughput . .
1.3 Packets . . . . . . . . . . . . .
1.4 Datagram Forwarding . . . . .
1.5 Topology . . . . . . . . . . . .
1.6 Routing Loops . . . . . . . . .
1.7 Congestion . . . . . . . . . . .
1.8 Packets Again . . . . . . . . .
1.9 LANs and Ethernet . . . . . .
1.10 IP - Internet Protocol . . . . .
1.11 DNS . . . . . . . . . . . . .
1.12 Transport . . . . . . . . . . .
1.13 Firewalls . . . . . . . . . . .
1.14 Network Address Translation
1.15 IETF and OSI . . . . . . . . .
1.16 Berkeley Unix . . . . . . . .
1.17 Epilog . . . . . . . . . . . . .
1.18 Exercises . . . . . . . . . . .
2 Ethernet
2.1 10-Mbps classic Ethernet
2.2 100 Mbps (Fast) Ethernet
2.3 Gigabit Ethernet . . . . .
2.4 Ethernet Switches . . . .
2.5 Spanning Tree Algorithm
2.6 Virtual LAN (VLAN) . .
2.7 Epilog . . . . . . . . . .
2.8 Exercises . . . . . . . . .
3 Other LANs

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3
3
5
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i

3.1 Virtual Private Network . . . . . . .

3.2 Carrier Ethernet . . . . . . . . . . .
3.3 Wi-Fi . . . . . . . . . . . . . . . . .
3.4 WiMAX . . . . . . . . . . . . . . .
3.5 Fixed Wireless . . . . . . . . . . . .
3.6 Token Ring . . . . . . . . . . . . . .
3.7 Virtual Circuits . . . . . . . . . . . .
3.8 Asynchronous Transfer Mode: ATM
3.9 Epilog . . . . . . . . . . . . . . . .
3.10 Exercises . . . . . . . . . . . . . .

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55
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80
80

4 Links
4.1 Encoding and Framing . . .
4.2 Time-Division Multiplexing
4.3 Epilog . . . . . . . . . . .
4.4 Exercises . . . . . . . . . .

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85
85
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93
93

5 Packets
5.1 Packet Delay . . . . . .
5.2 Packet Delay Variability
5.3 Packet Size . . . . . . .
5.4 Error Detection . . . . .
5.5 Epilog . . . . . . . . .
5.6 Exercises . . . . . . . .

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95
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98
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106

6 Abstract Sliding Windows

6.1 Building Reliable Transport: Stop-and-Wait
6.2 Sliding Windows . . . . . . . . . . . . . . .
6.3 Linear Bottlenecks . . . . . . . . . . . . . .
6.4 Epilog . . . . . . . . . . . . . . . . . . . .
6.5 Exercises . . . . . . . . . . . . . . . . . . .

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109
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7 IP version 4
7.1 The IPv4 Header . . . . . . . . . . . . . . . .
7.2 Interfaces . . . . . . . . . . . . . . . . . . . .
7.3 Special Addresses . . . . . . . . . . . . . . .
7.4 Fragmentation . . . . . . . . . . . . . . . . .
7.5 The Classless IP Delivery Algorithm . . . . .
7.6 IP Subnets . . . . . . . . . . . . . . . . . . .
7.7 Address Resolution Protocol: ARP . . . . . .
7.8 Dynamic Host Configuration Protocol (DHCP)
7.9 Internet Control Message Protocol . . . . . .
7.10 Unnumbered Interfaces . . . . . . . . . . . .
7.11 Mobile IP . . . . . . . . . . . . . . . . . . .
7.12 Epilog . . . . . . . . . . . . . . . . . . . . .
7.13 Exercises . . . . . . . . . . . . . . . . . . .

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8 IP version 6

ii

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153

8.1 The IPv6 Header . . . . . . . . . .

8.2 IPv6 addresses . . . . . . . . . . .
8.3 IPv6 Multicast . . . . . . . . . . .
8.4 IPv6 Extension Headers . . . . . .
8.5 Network Prefixes . . . . . . . . . .
8.6 Neighbor Discovery . . . . . . . .
8.7 IPv6 Host Address Assignment . .
8.8 Globally Exposed Addresses . . .
8.9 ICMPv6 . . . . . . . . . . . . . .
8.10 Routerless Connection Examples
8.11 IPv6-to-IPv4 connectivity . . . .
8.12 Epilog . . . . . . . . . . . . . . .
8.13 Exercises . . . . . . . . . . . . .

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9 Routing-Update Algorithms
9.1 Distance-Vector Routing-Update Algorithm
9.2 Distance-Vector Slow-Convergence Problem
9.3 Observations on Minimizing Route Cost . .
9.4 Loop-Free Distance Vector Algorithms . . .
9.5 Link-State Routing-Update Algorithm . . .
9.6 Routing on Other Attributes . . . . . . . . .
9.7 Epilog . . . . . . . . . . . . . . . . . . . .
9.8 Exercises . . . . . . . . . . . . . . . . . . .

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10 Large-Scale IP Routing
10.1 Classless Internet Domain Routing: CIDR
10.2 Hierarchical Routing . . . . . . . . . . .
10.3 Legacy Routing . . . . . . . . . . . . . .
10.4 Provider-Based Routing . . . . . . . . .
10.5 Geographical Routing . . . . . . . . . .
10.6 Border Gateway Protocol, BGP . . . . .
10.7 Epilog . . . . . . . . . . . . . . . . . . .
10.8 Exercises . . . . . . . . . . . . . . . . .

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11 UDP Transport
11.1 User Datagram Protocol UDP . . .
11.2 Fundamental Transport Issues . . . .
11.3 Trivial File Transport Protocol, TFTP
11.4 Remote Procedure Call (RPC) . . . .
11.5 Epilog . . . . . . . . . . . . . . . . .
11.6 Exercises . . . . . . . . . . . . . . .

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12 TCP Transport
12.1 The End-to-End Principle . . .
12.2 TCP Header . . . . . . . . . . .
12.3 TCP Connection Establishment
12.4 TCP and WireShark . . . . . .
12.5 TCP simplex-talk . . . . . . . .

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iii

12.6 TCP state diagram . . . . . . . . . .

12.7 TCP Old Duplicates . . . . . . . . .
12.8 TIMEWAIT . . . . . . . . . . . . . .
12.9 The Three-Way Handshake Revisited
12.10 Anomalous TCP scenarios . . . . .
12.11 TCP Faster Opening . . . . . . . . .
12.12 Path MTU Discovery . . . . . . . .
12.13 TCP Sliding Windows . . . . . . . .
12.14 TCP Delayed ACKs . . . . . . . . .
12.15 Nagle Algorithm . . . . . . . . . .
12.16 TCP Flow Control . . . . . . . . . .
12.17 TCP Timeout and Retransmission .
12.18 KeepAlive . . . . . . . . . . . . . .
12.19 TCP timers . . . . . . . . . . . . .
12.20 Epilog . . . . . . . . . . . . . . . .
12.21 Exercises . . . . . . . . . . . . . .

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13 TCP Reno and Congestion Management

13.1 Basics of TCP Congestion Management
13.2 Slow Start . . . . . . . . . . . . . . . .
13.3 TCP Tahoe and Fast Retransmit . . . .
13.4 TCP Reno and Fast Recovery . . . . .
13.5 TCP NewReno . . . . . . . . . . . . .
13.6 SACK TCP . . . . . . . . . . . . . . .
13.7 TCP and Bottleneck Link Utilization .
13.8 Single Packet Losses . . . . . . . . . .
13.9 TCP Assumptions and Scalability . . .
13.10 TCP Parameters . . . . . . . . . . . .
13.11 Epilog . . . . . . . . . . . . . . . . .
13.12 Exercises . . . . . . . . . . . . . . .

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14 Dynamics of TCP Reno

14.1 A First Look At Queuing . . . . . . . .
14.2 Bottleneck Links with Competition . .
14.3 TCP Fairness with Synchronized Losses
14.4 Notions of Fairness . . . . . . . . . . .
14.5 TCP Reno loss rate versus cwnd . . . .
14.6 TCP Friendliness . . . . . . . . . . . .
14.7 AIMD Revisited . . . . . . . . . . . .
14.8 Active Queue Management . . . . . . .
14.9 The High-Bandwidth TCP Problem . .
14.10 The Lossy-Link TCP Problem . . . .
14.11 The Satellite-Link TCP Problem . . .
14.12 Epilog . . . . . . . . . . . . . . . . .
14.13 Exercises . . . . . . . . . . . . . . .

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315
315
315
315

15 Newer TCP Implementations

323
15.1 High-Bandwidth Desiderata . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 324

iv

15.2 RTTs . . . . .
15.3 Highspeed TCP
15.4 TCP Vegas . .
15.5 FAST TCP . .
15.6 TCP Westwood
15.7 TCP Veno . . .
15.8 TCP Hybla . .
15.9 TCP Illinois . .
15.10 H-TCP . . . .
15.11 TCP CUBIC .
15.12 Epilog . . . .
15.13 Exercises . .

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324
325
326
328
330
332
333
334
335
336
340
341

16 Network Simulations: ns-2

16.1 The ns-2 simulator . . . . . . . . . . . . .
16.2 A Single TCP Sender . . . . . . . . . . . .
16.3 Two TCP Senders Competing . . . . . . .
16.4 TCP Loss Events and Synchronized Losses
16.5 TCP Reno versus TCP Vegas . . . . . . . .
16.6 Wireless Simulation . . . . . . . . . . . .
16.7 Epilog . . . . . . . . . . . . . . . . . . . .
16.8 Exercises . . . . . . . . . . . . . . . . . .

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347
347
349
361
376
387
389
395
395

17 The ns-3 Network Simulator

17.1 Installing and Running ns-3
17.2 A Single TCP Sender . . . .
17.3 Wireless . . . . . . . . . . .
17.4 Exercises . . . . . . . . . .

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399
399
400
408
414

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415
415
416
416
417
418
430
431
434
440
444
446
448
450
452
452

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18 Queuing and Scheduling

18.1 Queuing and Real-Time Traffic . . . . . . .
18.2 Traffic Management . . . . . . . . . . . . .
18.3 Priority Queuing . . . . . . . . . . . . . . .
18.4 Queuing Disciplines . . . . . . . . . . . . .
18.5 Fair Queuing . . . . . . . . . . . . . . . . .
18.6 Applications of Fair Queuing . . . . . . . . .
18.7 Hierarchical Queuing . . . . . . . . . . . . .
18.8 Hierarchical Weighted Fair Queuing . . . . .
18.9 Token Bucket Filters . . . . . . . . . . . . .
18.10 Applications of Token Bucket . . . . . . . .
18.11 Token Bucket Queue Utilization . . . . . .
18.12 Hierarchical Token Bucket . . . . . . . . .
18.13 Fair Queuing / Token Bucket combinations .
18.14 Epilog . . . . . . . . . . . . . . . . . . . .
18.15 Exercises . . . . . . . . . . . . . . . . . .

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19 Quality of Service
457
19.1 Net Neutrality . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 458

19.2 Where the Wild Queues Are . . . . . . . .

19.3 Real-time Traffic . . . . . . . . . . . . . .
19.4 Integrated Services / RSVP . . . . . . . . .
19.5 Global IP Multicast . . . . . . . . . . . . .
19.6 RSVP . . . . . . . . . . . . . . . . . . . .
19.7 Differentiated Services . . . . . . . . . . .
19.8 RED with In and Out . . . . . . . . . . . .
19.9 NSIS . . . . . . . . . . . . . . . . . . . .
19.10 Comcast Congestion-Management System
19.11 Real-time Transport Protocol (RTP) . . .
19.12 Multi-Protocol Label Switching (MPLS) .
19.13 Epilog . . . . . . . . . . . . . . . . . . .
19.14 Exercises . . . . . . . . . . . . . . . . .
20 Network Management and SNMP
20.1 Network Architecture . . . . . . . . .
20.2 SNMP Basics . . . . . . . . . . . . .
20.3 SNMP Naming and OIDs . . . . . .
20.4 MIBs . . . . . . . . . . . . . . . . .
20.5 SNMPv1 Data Types . . . . . . . . .
20.6 ASN.1 Syntax and SNMP . . . . . .
20.7 SNMP Tables . . . . . . . . . . . . .
20.8 SNMP Operations . . . . . . . . . .
20.9 MIB Browsing . . . . . . . . . . . .
20.10 MIB-2 . . . . . . . . . . . . . . . .
20.11 SNMPv1 communities and security
20.12 SNMP and ASN.1 Encoding . . . .
20.13 SNMPv2 . . . . . . . . . . . . . . .
20.14 Table Row Creation . . . . . . . . .
20.15 SNMPv3 . . . . . . . . . . . . . . .
20.16 Exercises . . . . . . . . . . . . . .
21 Security
21.1 Code-Execution Intrusion . . . .
21.2 Stack Buffer Overflow . . . . . .
21.3 Heap Buffer Overflow . . . . . .
21.4 Network Intrusion Detection . . .
21.5 Secure Hashes . . . . . . . . . .
21.6 Shared-Key Encryption . . . . . .
21.7 Diffie-Hellman-Merkle Exchange
21.8 Public-Key Encryption . . . . . .
21.9 SSH and TLS . . . . . . . . . . .
21.10 Exercises . . . . . . . . . . . .

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458
459
461
462
466
468
472
472
473
474
478
480
481

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483
485
485
486
488
489
490
491
495
500
501
509
511
513
524
532
541

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545
546
547
555
561
562
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573
575
578
587

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Bibliography

591

Indices and tables

593

vi

Bibliography

595

Index

601

vii

viii

An Introduction to Computer Networks, Release 1.0

Peter L Dordal
Department of Computer Science
Loyola University Chicago

Contents:

CONTENTS

An Introduction to Computer Networks, Release 1.0

CONTENTS

0 PREFACE

No man but a blockhead ever wrote, except for money. - Samuel Johnson
The textbook world is changing. On the one hand, open source software and creative-commons licensing
have been great successes; on the other hand, unauthorized PDFs of popular textbooks are widely available,
and it is time to consider flowing with rather than fighting the tide. Hence this open textbook, released for
free under the Creative Commons license described below. Mene, mene, tekel pharsin.
Perhaps the last straw, for me, was patent 8195571 for a roundabout method to force students to purchase
textbooks. (A simpler strategy might be to include the price of the book in the course.) At some point,
faculty have to be advocates for their students rather than, well, Hirudinea.
This is not to say that I have anything against for-profit publishing. It is just that this particular book does not
and will not belong to that category. In this it is in good company: there is Wikipedia, there is Gnu/Linux,
and there is an increasing number of other free online textbooks out there. The market inefficiencies of
traditional publishing are sobering: the return to authors of advanced textbooks is at best modest, and costs
to users are quite high.
This text is released under the Creative Commons license Attribution-NonCommercial-NoDerivs; this text
is like a conventional book, in other words, except that it is free. You may copy the work and distribute it
to others, but reuse requires attribution. Creation of derivative works eg modifying chapters or creating
additional chapters and distributing them as part of this work also requires permission.
The work may not be used for commercial purposes without permission. Permission is likely to be granted
for use and distribution of all or part of the work in for-profit and commercial training programs, provided
there is no direct charge to recipients for the work and provided the free nature of the work is made clear to
recipients (eg by including this preface). However, such permission must always be requested. Alternatively,
participants in commercial programs may be instructed to download the work individually.
The official book website (potentially subject to change) is intronetworks.cs.luc.edu. The book is available
there as online html, as a zipped archive of html files, in .pdf format, and in other formats as may prove
useful.

0.1 Classroom Use

This book is meant as a serious and more-or-less thorough text for an introductory college or graduate course
in computer networks, carefully researched, with consistent notation and style, and complete with diagrams
and exercises. My intent is to create a text that covers to a reasonable extent why the Internet is the way it is,
to avoid the endless dreary focus on TLAs (Three-Letter Acronyms), and to remain not too mathematical.
For the last, I have avoided calculus, linear algebra, and, for that matter, quadratic terms (though some
inequalities do sneak in at times). That said, the book includes a large number of back-of-the-envelope
calculations in settings as concrete as I could make them illustrating various networking concepts.
Overall, I tried to find a happy medium between practical matters and underlying principles. My goal has
been to create a book that is useful to a broad audience, including those interested in network management,
in high-performance networking, in software development, or just in how the Internet is put together.

An Introduction to Computer Networks, Release 1.0

The book can also be used as a networks supplement or companion to other resources for a variety of
other courses that overlap to some greater or lesser degree with networking. At Loyola, earlier versions of
this material have been used coupled with a second textbook in courses in computer security, network
management, telecommunications, and even introduction-to-computing courses for non-majors. Another
possibility is an alternative or nontraditional presentation of networking itself. It is when used in concert
with other works, in particular, that this books being free is of marked advantage.
Finally, I hope the book may also be useful as a reference work. To this end, I have attempted to ensure that
the indexing and cross-referencing is sufficient to support the drop-in reader. Similarly, obscure notation is
kept to a minimum.
Much is sometimes made, in the world of networking textbooks, about top-down versus bottom-up sequencing. This book is not really either, although the chapters are mostly numbered in bottom-up fashion.
Instead, the first chapter provides a relatively complete overview of the LAN, IP and transport network layers
(along with a few other things), allowing subsequent chapters to refer to all network layers without forward
reference, and, more importantly, allowing the chapters to be covered in a variety of different orders. As a
practical matter, when I use this text to teach Loyolas Introduction to Computer Networks course, I cover
the IP and TCP material more or less in parallel.
A distinctive feature of the book is the extensive coverage of TCP: TCP dynamics, newer versions of TCP
such as TCP Cubic, and a chapter on using the ns-2 simulator to explore actual TCP behavior. This has
its roots in a longstanding goal to find better ways to present competition and congestion in the classroom.
Another feature is the detailed chapter on queuing disciplines.
One thing this book makes little attempt to cover in detail is the application layer; the token example included is SNMP. While SNMP actually makes a pretty good example of a self-contained application, my
recommendation to instructors who wish to cover more familiar examples is to combine this text with the
appropriate application documentation.
For those interested in using the book for a traditional networks course, I with some trepidation offer the
following set of core material. In solidarity with those who prefer alternatives to a bottom-up ordering, I
emphasize that this represents a set and not a sequence.
1 An Overview of Networks
Selected sections from 2 Ethernet, particularly switched Ethernet
Selected sections from 3.3 Wi-Fi
Selected sections from 5 Packets
6 Abstract Sliding Windows
7 IP version 4 and/or 8 IP version 6
Selected sections from 9 Routing-Update Algorithms and 10 Large-Scale IP Routing
11 UDP Transport
12 TCP Transport
13 TCP Reno and Congestion Management
With some care in the topic-selection details, the above can be covered in one semester along with a survey
of selected important network applications, or the basics of network programming, or the introductory con-

0 Preface

An Introduction to Computer Networks, Release 1.0

figuration of switches and routers, or coverage of additional material from this book, or some other set of
additional topics. Of course, non-traditional networks courses may focus on a quite different sets of topics.

Peter Dordal
Shabbona, Illinois

0.2 Progress Notes

Edition 1.0 was declared complete as of March 2014.
The intronetworks.cs.luc.edu website carries both edition 1.0 and also the current 1.x edition, x>0. My plan
is to make sure, as far as is possible, that the two editions are classroom-compatible: existing exercises
will not be renumbered (though new exercises may be introduced with fractional numbers), and section
renumbering will be avoided to the extent practical.
As of November 2014 chapters on network management and security have been added; this completes the
set of chapters originally envisioned.
At this point I am actively seeking reviewers either for style or for technical accuracy.

0.3 Technical considerations

The book was prepared in reStructuredText using the linux Sphinx package, which can produce multiple
formats from the same source. That said, the primary format is html.
In order to read this book online in html format you will need the appropriate unicode character set installed.
At a minimum, the following characters must display properly:

=
The characters above should look roughly as they do in the following image:

If no available browser displays these properly, I recommend the pdf or epub formats. Generally Firefox
handles the necessary characters out of the box, as does Internet Explorer, but Chrome does not.
The diagrams in the body of the text are now all in bitmap .png format, although a few diagrams rendered
with line-drawing characters still appear in the exercises. I would prefer to use the vector-graphics .svg
format, but as of January 2014 most browsers do not appear to support zooming in on .svg images, which is
really the whole point.

0.2 Progress Notes

An Introduction to Computer Networks, Release 1.0

0 Preface

1 AN OVERVIEW OF NETWORKS

Somewhere there might be a field of interest in which the order of presentation of topics is well agreed upon.
Computer networking is not it.
There are many interconnections in the field of networking, as in most technical fields, and it is difficult
to find an order of presentation that does not involve endless forward references to future chapters; this
is true even if as is done here a largely bottom-up ordering is followed. I have therefore taken here a
different approach: this first chapter is a summary of the essentials LANs, IP and TCP across the board,
and later chapters expand on the material here.
Local Area Networks, or LANs, are the physical networks that provide the connection between machines
within, say, a home, school or corporation. LANs are, as the name says, local; it is the IP, or Internet
Protocol, layer that provides an abstraction for connecting multiple LANs into, well, the Internet. Finally,
TCP deals with transport and connections and actually sending user data.
This chapter also contains some important other material. The section on datagram forwarding, central
to packet-based switching and routing, is essential. This chapter also discusses packets generally, congestion, and sliding windows, but those topics are revisited in later chapters. Firewalls and network address
translation are also covered here and not elsewhere.

1.1 Layers
These three topics LANs, IP and TCP are often called layers; they constitute the Link layer, the Internetwork layer, and the Transport layer respectively. Together with the Application layer (the software you use),
these form the four-layer model for networks. A layer, in this context, corresponds strongly to the idea
of a programming interface or library (though some of the layers are not accessible to ordinary users): an
application hands off a chunk of data to the TCP library, which in turn makes calls to the IP library, which
in turn calls the LAN layer for actual delivery.
The LAN layer is in charge of actual delivery of packets, using LAN-layer-supplied addresses. It is often
conceptually subdivided into the physical layer dealing with, eg, the analog electrical, optical or radio
signaling mechanisms involved, and above that an abstracted logical LAN layer that describes all the
digital that is, non-analog operations on packets; see 2.1.2 The LAN Layer. The physical layer is
generally of direct concern only to those designing LAN hardware; the kernel software interface to the LAN
corresponds to the logical LAN layer. This LAN physical/logical division gives us the Internet five-layer
model. This is less a formal hierarchy as an ad hoc classification method. We will return to this below in
1.15 IETF and OSI.

1.2 Bandwidth and Throughput

Any one network connection eg at the LAN layer has a data rate: the rate at which bits are transmitted.
In some LANs (eg Wi-Fi) the data rate can vary with time. Throughput refers to the overall effective

An Introduction to Computer Networks, Release 1.0

transmission rate, taking into account things like transmission overhead, protocol inefficiencies and perhaps
even competing traffic. It is generally measured at a higher network layer than the data rate.
The term bandwidth can be used to refer to either of these, though we here try to use it mostly as a synonym
for data rate. The term comes from radio transmission, where the width of the frequency band available is
proportional, all else being equal, to the data rate that can be achieved.
In discussions about TCP, the term goodput is sometimes used to refer to what might also be called
application-layer throughput: the amount of usable data delivered to the receiving application. Specifically, retransmitted data is counted only once when calculating goodput but might be counted twice under
some interpretations of throughput.
Data rates are generally measured in kilobits per second (Kbps) or megabits per second (Mbps); in the
context of data rates, a kilobit is 103 bits (not 210 ) and a megabit is 106 bits. The use of the lowercase b
means bits; data rates expressed in terms of bytes often use an upper-case B.

1.3 Packets
Packets are modest-sized buffers of data, transmitted as a unit through some shared set of links. Of necessity,
packets need to be prefixed with a header containing delivery information. In the common case known as
datagram forwarding, the header contains a destination address; headers in networks using so-called
virtual-circuit forwarding contain instead an identifier for the connection. Almost all networking today
(and for the past 50 years) is packet-based, although we will later look briefly at some circuit-switched
options for voice telephony.

At the LAN layer, packets can be viewed as the imposition of a buffer (and addressing) structure on top
of low-level serial lines; additional layers then impose additional structure. Informally, packets are often
referred to as frames at the LAN layer, and as segments at the Transport layer.
The maximum packet size supported by a given LAN (eg Ethernet, Token Ring or ATM) is an intrinsic
attribute of that LAN. Ethernet allows a maximum of 1500 bytes of data. By comparison, TCP/IP packets
originally often held only 512 bytes of data, while early Token Ring packets could contain up to 4KB of
data. While there are proponents of very large packet sizes, larger even than 64KB, at the other extreme the
ATM (Asynchronous Transfer Mode) protocol uses 48 bytes of data per packet, and there are good reasons
for believing in modest packet sizes.
One potential issue is how to forward packets from a large-packet LAN to (or through) a small-packet LAN;
in later chapters we will look at how the IP (or Internet Protocol) layer addresses this.

1 An Overview of Networks

An Introduction to Computer Networks, Release 1.0

Generally each layer adds its own header. Ethernet headers are typically 14 bytes, IP headers 20 bytes, and
TCP headers 20 bytes. If a TCP connection sends 512 bytes of data per packet, then the headers amount to
10% of the total, a not-unreasonable overhead. For one common Voice-over-IP option, packets contain 160
bytes of data and 54 bytes of headers, making the header about 25% of the total. Compressing the 160 bytes
of audio, however, may bring the data portion down to 20 bytes, meaning that the headers are now 73% of
the total; see 19.11.4 RTP and VoIP.
In datagram-forwarding networks the appropriate header will contain the address of the destination and
perhaps other delivery information. Internal nodes of the network called routers or switches will then make
sure that the packet is delivered to the requested destination.
The concept of packets and packet switching was first introduced by Paul Baran in 1962 ([PB62]). Barans
primary concern was with network survivability in the event of node failure; existing centrally switched
protocols were vulnerable to central failure. In 1964, Donald Davies independently developed many of the
same concepts; it was Davies who coined the term packet.
It is perhaps worth noting that packets are buffers built of 8-bit bytes, and all hardware today agrees what
a byte is (hardware agrees by convention on the order in which the bits of a byte are to be transmitted).
8-bit bytes are universal now, but it was not always so. Perhaps the last great non-byte-oriented hardware
platform, which did indeed overlap with the Internet era broadly construed, was the DEC-10, which had a
36-bit word size; a word could hold five 7-bit ASCII characters. The early Internet specifications introduced
the term octet (an 8-bit byte) and required that packets be sequences of octets; non-octet-oriented hosts had
to be able to convert. Thus was chaos averted. Note that there are still byte-oriented data issues; as one
example, binary integers can be represented as a sequence of bytes in either big-endian or little-endian byte
order. RFC 1700 specifies that Internet protocols use big-endian byte order, therefore sometimes called
network byte order.

1.4 Datagram Forwarding

In the datagram-forwarding model of packet delivery, packet headers contain a destination address. It is up
to the intervening switches or routers to look at this address and get the packet to the correct destination.
In the diagram below, switch S1 has interfaces 0, 1 and 2, and S2 has interfaces 0,1,2,3. If A is to send a
packet P to B, S1 must know that P must be forwarded out interface 2 and S2 must know P must be forwarded
out interface 3. In datagram forwarding this is achieved by providing each switch with a forwarding table of
destination,next_hop pairs. When a packet arrives, the switch looks up the destination address (presumed
globally unique) in this table, and finds the next_hop information: the address to which or interface by
which the packet should be forwarded in order to bring it one step closer to its final destination. In the
network below, a complete forwarding table for S1 (using interface numbers as next_hop values) would be:
S1
destination
A
C
B
D
E

next_hop
0
1
2
2
2

1.4 Datagram Forwarding

An Introduction to Computer Networks, Release 1.0

The table for S2 might be as follows, where we have consolidated destinations A and C for visual simplicity.
S2
destination
A,C
D
E
B

next_hop
0
1
2
3

Alternatively, we could replace the interface information with next-node, or neighbor, information, as all
the links above are point-to-point and so each interface connects to a unique neighbor. In that case, S1s
table might be written as follows (with consolidation of the entries for B, D and E):
S1
destination
A
C
B,D,E

next_hop
A
C
S2

A central feature of datagram forwarding is that each packet is forwarded in isolation; the switches involved do not have any awareness of any higher-layer logical connections established between endpoints.
This is also called stateless forwarding, in that the forwarding tables have no per-connection state. RFC
1122 put it this way (in the context of IP-layer datagram forwarding):
To improve robustness of the communication system, gateways are designed to be stateless,
forwarding each IP datagram independently of other datagrams. As a result, redundant paths can
be exploited to provide robust service in spite of failures of intervening gateways and networks.
Datagram forwarding is sometimes allowed to use other information beyond the destination address. In
theory, IP routing can be done based on the destination address and some quality-of-service information,
allowing, for example, different routing to the same destination for high-bandwidth bulk traffic and for lowlatency real-time traffic. In practice, many ISPs ignore quality-of-service information in the IP header, and
route only based on the destination.
By convention, switching devices acting at the LAN layer and forwarding packets based on the LAN address
are called switches (or, in earlier days, bridges), while such devices acting at the IP layer and forwarding
on the IP address are called routers. Datagram forwarding is used both by Ethernet switches and by IP
routers, though the destinations in Ethernet forwarding tables are individual nodes while the destinations in
IP routers are entire networks (that is, sets of nodes).

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In IP routers within end-user sites it is common for a forwarding table to include a catchall default entry,
matching any IP address that is nonlocal and so needs to be routed out into the Internet at large. Unlike the
consolidated entries for B, D and E in the table above for S1, which likely would have to be implemented as
actual separate entries, a default entry is a single record representing where to forward the packet if no other
destination match is found. Here is a forwarding table for S1, above, with a default entry replacing the last
three entries:
S1
destination
A
C
default

next_hop
0
1
2

Default entries make sense only when we can tell by looking at an address that it does not represent a
nearby node. This is common in IP networks because an IP address encodes the destination network, and
routers generally know all the local networks. It is however rare in Ethernets, because there is generally
no correlation between Ethernet addresses and locality. If S1 above were an Ethernet switch, and it had
some means of knowing that interfaces 0 and 1 connected directly to individual hosts, not switches and S1
knew the addresses of these hosts then making interface 2 a default route would make sense. In practice,
however, Ethernet switches do not know what kind of device connects to a given interface.

1.5 Topology
In the network diagrammed in the previous section, there are no loops; graph theorists might describe this
by saying the network graph is acyclic, or is a tree. In a loop-free network there is a unique path between
any pair of nodes. The forwarding-table algorithm has only to make sure that every destination appears in
the forwarding tables; the issue of choosing between alternative paths does not arise.
However, if there are no loops then there is no redundancy: any broken link will result in partitioning the
network into two pieces that cannot communicate. All else being equal (which it is not, but never mind for
now), redundancy is a good thing. However, once we start including redundancy, we have to make decisions
among the multiple paths to a destination. Consider, for a moment, the following network:

Should S1 list S2 or S3 as the next_hop to B? Both paths A S1 S2 S4 B and A S1 S3 S4 B get there.

There is no right answer. Even if one path is faster than the other, taking the slower path is not exactly
wrong (especially if the slower path is, say, less expensive). Some sort of protocol must exist to provide a
mechanism by which S1 can make the choice (though this mechanism might be as simple as choosing to
route via the first path discovered to the given destination). We also want protocols to make sure that, if S1
reaches B via S2 and the S2 S4 link fails, then S1 will switch over to the still-working S1 S3 S4 B route.
As we shall see, many LANs (in particular Ethernet) prefer tree networks with no redundancy, while IP
has complex protocols in support of redundancy.

1.5 Topology

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1.6 Routing Loops

A potential drawback to datagram forwarding is the possibility of a routing loop: a set of entries in the
forwarding tables that cause some packets to circulate endlessly. For example, in the previous picture we
would have a routing loop if, for (nonexistent) destination C, S1 forwarded to S2, S2 forwarded to S4,
S4 forwarded to S3, and S3 forwarded to S1. A packet sent to C would not only not be delivered, but
in circling endlessly it might easily consume a large majority of the bandwidth. Routing loops typically
arise because the creation of the forwarding tables is often distributed, and there is no global authority to
detect inconsistencies. Even when there is such an authority, temporary routing loops can be created due to
notification delays.
Routing loops can also occur in networks where the underlying link topology is loop-free; for example, in
the previous diagram we could, again for destination C, have S1 forward to S2 and S2 forward back to S1.
We will refer to such a case as a linear routing loop.
All datagram-forwarding protocols need some way of detecting and avoiding routing loops. Ethernet, for
example, avoids nonlinear routing loops by disallowing loops in the underlying network topology, and avoids
linear routing loops by not having switches forward a packet back out the interface by which it arrived. IP
provides for a one-byte Time to Live (TTL) field in the IP header; it is set by the sender and decremented
by 1 at each router; a packet is discarded if its TTL reaches 0. This limits the number of times a wayward
packet can be forwarded to the initial TTL value, typically 64.
In datagram routing, a switch is responsible only for the next hop to the ultimate destination; if a switch
has a complete path in mind, there is no guarantee that the next_hop switch or any other downstream switch
to agree to forward along that path. Misunderstandings can potentially lead to routing loops. Consider this
network:

D might feel that the best path to B is DECB (perhaps because it believes the AD link is to be avoided).
If E similarly decides the best path to B is EDAB, and if D and E both choose their next_hop for B
based on these best paths, then a linear routing loop is formed: D routes to B via E and E routes to B via D.
Although each of D and E have identified a usable path, that path is not in fact followed. Moral: successful
datagram routing requires cooperation and a consistent view of the network.

1.7 Congestion
Switches introduce the possibility of congestion: packets arriving faster than they can be sent out. This can
happen with just two interfaces, if the inbound interface has a higher bandwidth than the outbound interface;
another common source of congestion is traffic arriving on multiple inputs and all destined for the same
output.

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Whatever the reason, if packets are arriving for a given outbound interface faster than they can be sent, a
queue will form for that interface. Once that queue is full, packets will be dropped. The most common
strategy (though not the only one) is to drop any packets that arrive when the queue is full.
The term congestion may refer either to the point where the queue is just beginning to build up, or to the
point where the queue is full and packets are lost. In their paper [CJ89], Chiu and Jain refer to the first point
as the knee; this is where the slope of the load v throughput graph flattens. They refer to the second point as
the cliff; this is where packet losses may lead to a precipitous decline in throughput. Other authors use the
term contention for knee-congestion.
In the Internet, most packet losses are due to congestion. This is not because congestion is especially bad
(though it can be, at times), but rather that other types of losses (eg due to packet corruption) are insignificant
by comparison.
When to Upgrade?
Deciding when a network really does have insufficient bandwidth is not a technical issue but an economic one. The number of customers may increase, the cost of bandwidth may decrease or customers
may simply be willing to pay more to have data transfers complete in less time; customers here can
be external or in-house. Monitoring of links and routers for congestion can, however, help determine
exactly what parts of the network would most benefit from upgrade.
We emphasize that the presence of congestion does not mean that a network has a shortage of bandwidth.
Bulk-traffic senders (though not real-time senders) attempt to send as fast as possible, and congestion is
simply the networks feedback that the maximum transmission rate has been reached.
Congestion is a sign of a problem in real-time networks, which we will consider in 19 Quality of Service.
In these networks losses due to congestion must generally be kept to an absolute minimum; one way to
achieve this is to limit the acceptance of new connections unless sufficient resources are available.

1.8 Packets Again

Perhaps the core justification for packets, Barans concerns about node failure notwithstanding, is that the
same link can carry, at different times, different packets representing traffic to different destinations and from
different senders. Thus, packets are the key to supporting shared transmission lines; that is, they support
the multiplexing of multiple communications channels over a single cable. The alternative of a separate
physical line between every pair of machines grows prohibitively complex very quickly (though virtual
circuits between every pair of machines in a datacenter are not uncommon; see 3.7 Virtual Circuits).
From this shared-medium perspective, an important packet feature is the maximum packet size, as this represents the maximum time a sender can send before other senders get a chance. The alternative of unbounded
packet sizes would lead to prolonged network unavailability for everyone else if someone downloaded a
large file in a single 1 Gigabit packet. Another drawback to large packets is that, if the packet is corrupted,
the entire packet must be retransmitted; see 5.3.1 Error Rates and Packet Size.
When a router or switch receives a packet, it (generally) reads in the entire packet before looking at the
header to decide to what next node to forward it. This is known as store-and-forward, and introduces
a forwarding delay equal to the time needed to read in the entire packet. For individual packets this

1.8 Packets Again

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forwarding delay is hard to avoid (though some switches do implement cut-through switching to begin
forwarding a packet before it has fully arrived), but if one is sending a long train of packets then by keeping
multiple packets en route at the same time one can essentially eliminate the significance of the forwarding
delay; see 5.3 Packet Size.
Total packet delay from sender to receiver is the sum of the following:
Bandwidth delay, ie sending 1000 Bytes at 20 Bytes/millisecond will take 50 ms. This is a per-link
delay.
Propagation delay due to the speed of light. For example, if you start sending a packet right now
on a 5000-km cable across the US with a propagation speed of 200 m/sec (= 200 km/ms, about 2/3
the speed of light in vacuum), the first bit will not arrive at the destination until 25 ms later. The
bandwidth delay then determines how much after that the entire packet will take to arrive.
Store-and-forward delay, equal to the sum of the bandwidth delays out of each router along the path
Queuing delay, or waiting in line at busy routers. At bad moments this can exceed 1 sec, though that
is rare. Generally it is less than 10 ms and often is less than 1 ms. Queuing delay is the only delay
component amenable to reduction through careful engineering.
See 5.1 Packet Delay for more details.

1.9 LANs and Ethernet

A local-area network, or LAN, is a system consisting of
physical links that are, ultimately, serial lines
common interfacing hardware connecting the hosts to the links
protocols to make everything work together
We will explicitly assume that every LAN node is able to communicate with every other LAN node. Sometimes this will require the cooperation of intermediate nodes acting as switches.
Far and away the most common type of LAN is Ethernet, originally described in a 1976 paper by Metcalfe
and Boggs [MB76]. Ethernets popularity is due to low cost more than anything else, though the primary
reason Ethernet cost is low is that high demand has led to manufacturing economies of scale.
The original Ethernet had a bandwidth of 10 Mbps (megabits per second; we will use lower-case b for
bits and upper-case B for bytes), though nowadays most Ethernet operates at 100 Mbps and gigabit (1000
Mbps) Ethernet (and faster) is widely used in server rooms. (By comparison, as of this writing the data transfer rate to a typical faster hard disk is about 1000 Mbps.) Wireless (wi-fi) LANs are gaining popularity,
and in some settings have supplanted wired Ethernet to end-users.
Many early Ethernet installations were unswitched; each host simply tapped in to one long primary cable
that wound through the building (or floor). In principle, two stations could then transmit at the same time,
rendering the data unintelligible; this was called a collision. Ethernet has several design features intended
to minimize the bandwidth wasted on collisions: stations, before transmitting, check to be sure the line is
idle, they monitor the line while transmitting to detect collisions during the transmission, and, if a collision
is detected, they execute a random backoff strategy to avoid an immediate recollision. See 2.1.3 The Slot

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Time and Collisions. While Ethernet collisions definitely reduce throughput, in the larger view they should
perhaps be thought of as a part of a remarkably inexpensive shared-access mediation protocol.
In unswitched Ethernets every packet is received by every host and it is up to the network card in each host
to determine if the arriving packet is addressed to that host. It is almost always possible to configure the
card to forward all arriving packets to the attached host; this poses a security threat and password sniffers
that surreptitiously collected passwords via such eavesdropping used to be common.
Password Sniffing
In the fall of 1994 at Loyola University I remotely changed the root password on several CS-department
unix machines at the other end of campus, using telnet. I told no one. Within two hours, someone else
logged into one of these machines, using the new password, from a host in Europe. Password sniffing
was the likely culprit.
Two months later was the so-called Christmas Day Attack (12.9.1 ISNs and spoofing). One of the
hosts used to launch this attack was Loyolas hacked apollo.it.luc.edu. It is unclear the degree to which
password sniffing played a role in that exploit.
Due to both privacy and efficiency concerns, almost all Ethernets today are fully switched; this ensures that
each packet is delivered only to the host to which it is addressed. One advantage of switching is that it
effectively eliminates most Ethernet collisions; while in principle it replaces them with a queuing issue, in
practice Ethernet switch queues so seldom fill up that they are almost invisible even to network managers
(unlike IP router queues). Switching also prevents host-based eavesdropping, though arguably a better
solution to this problem is encryption. Perhaps the more significant tradeoff with switches, historically, was
that Once Upon A Time they were expensive and unreliable; tapping directly into a common cable was dirt
cheap.
Ethernet addresses are six bytes long. Each Ethernet card (or network interface) is assigned a (supposedly)
unique address at the time of manufacture; this address is burned into the cards ROM and is called the cards
physical address or hardware address or MAC (Media Access Control) address. The first three bytes of
the physical address have been assigned to the manufacturer; the subsequent three bytes are a serial number
assigned by that manufacturer.
By comparison, IP addresses are assigned administratively by the local site. The basic advantage of having
addresses in hardware is that hosts automatically know their own addresses on startup; no manual configuration or server query is necessary. It is not unusual for a site to have a large number of identically configured
workstations, for which all network differences derive ultimately from each workstations unique Ethernet
address.
The network interface continually monitors all arriving packets; if it sees any packet containing a destination
address that matches its own physical address, it grabs the packet and forwards it to the attached CPU (via a
CPU interrupt).
Ethernet also has a designated broadcast address. A host sending to the broadcast address has its packet
received by every other host on the network; if a switch receives a broadcast packet on one port, it forwards
the packet out every other port. This broadcast mechanism allows host A to contact host B when A does
not yet know Bs physical address; typical broadcast queries have forms such as Will the designated server
please answer or (from the ARP protocol) will the host with the given IP address please tell me your
physical address.

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Traffic addressed to a particular host that is, not broadcast is said to be unicast.
Because Ethernet addresses are assigned by the hardware, knowing an address does not provide any direct
indication of where that address is located on the network. In switched Ethernet, the switches must thus have
a forwarding-table record for each individual Ethernet address on the network; for extremely large networks
this ultimately becomes unwieldy. Consider the analogous situation with postal addresses: Ethernet is
somewhat like attempting to deliver mail using social-security numbers as addresses, where each postal
worker is provided with a large catalog listing each persons SSN together with their physical location. Real
postal mail is, of course, addressed hierarchically using ever-more-precise specifiers: state, city, zipcode,
street address, and name / room#. Ethernet, in other words, does not scale well to large sizes.
Switched Ethernet works quite well, however, for networks with up to 10,000-100,000 nodes. Forwarding
tables with size in that range are straightforward to manage.
To forward packets correctly, switches must know where all active destination addresses in the LAN are
located; Ethernet switches do this by a passive learning algorithm. (IP routers, by comparison, use active
protocols.) Typically a host physical address is entered into a switchs forwarding table when a packet from
that host is first received; the switch notes the packets arrival interface and source address and assumes
that the same interface is to be used to deliver packets back to that sender. If a given destination address
has not yet been seen, and thus is not in the forwarding table, Ethernet switches still have the backup
delivery option of forwarding to everyone, by treating the destination address like the broadcast address,
and allowing the host Ethernet cards to sort it out. Since this broadcast-like process is not generally used
for more than one packet (after that, the switches will have learned the correct forwarding-table entries), the
risk of eavesdropping is minimal.
The host,interface forwarding table is often easier to think of as host,next_hop, where the next_hop node
is whatever switch or host is at the immediate other end of the link connecting to the given interface. In a
fully switched network where each link connects only two interfaces, the two perspectives are equivalent.

1.10 IP - Internet Protocol

To solve the scaling problem with Ethernet, and to allow support for other types of LANs and point-to-point
links as well, the Internet Protocol was developed. Perhaps the central issue in the design of IP was to
support universal connectivity (everyone can connect to everyone else) in such a way as to allow scaling to
enormous size (in 2013 there appear to be around ~109 nodes, although IP should work to 1010 nodes or
more), without resulting in unmanageably large forwarding tables (currently the largest tables have about
300,000 entries.)
In the early days, IP networks were considered to be internetworks of basic networks (LANs); nowadays
users generally ignore LANs and think of the Internet as one large (virtual) network.
To support universal connectivity, IP provides a global mechanism for addressing and routing, so that
packets can actually be delivered from any host to any other host. IP addresses (for the most-common
version 4, which we denote IPv4) are 4 bytes (32 bits), and are part of the IP header that generally follows
the Ethernet header. The Ethernet header only stays with a packet for one hop; the IP header stays with the
packet for its entire journey across the Internet.
An essential feature of IPv4 (and IPv6) addresses is that they can be divided into a network part (a prefix)
and a host part (the remainder). The legacy mechanism for designating the IPv4 network and host
address portions was to make the division according to the first few bits:
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first few bits

0
10
110

first byte
0-127
128-191
192-223

network bits
8
16
24

host bits
24
16
8

name
class A
class B
class C

application
a few very large networks
institution-sized networks
sized for smaller entities

For example, the original IP address allocation for Loyola University Chicago was 147.126.0.0, a class B.
In binary, 147 is 10010011. The network/host division point is not carried within the IP header; in fact,
nowadays the division into network and host is dynamic, and can be made at different positions in the
address at different levels of the network.
IP addresses, unlike Ethernet addresses, are administratively assigned. Once upon a time, you would get
your Class B network prefix from the Internet Assigned Numbers Authority, or IANA (they now delegate
this task), and then you would in turn assign the host portion in a way that was appropriate for your local
site. As a result of this administrative assignment, an IP address usually serves not just as an endpoint
identifier but also as a locator, containing embedded location information.
The Class A/B/C definition above was spelled out in 1981 in RFC 791, which introduced IP. Class D was
added in 1986 by RFC 988; class D addresses must begin with the bits 1110. These addresses are for
multicast, that is, sending an IP packet to every member of a set of recipients (ideally without actually
transmitting it more than once on any one link).
The network portion of an IP address is sometimes called the network number or network address or
network prefix; as we shall see below, most forwarding decisions are made using only the network portion.
It is commonly denoted by setting the host bits to zero and ending the resultant address with a slash followed
by the number of network bits in the address: eg 12.0.0.0/8 or 147.126.0.0/16. Note that 12.0.0.0/8 and
12.0.0.0/9 represent different things; in the latter, the second byte of any host address extending the network
address is constrained to begin with a 0-bit. An anonymous block of IP addresses might be referred to only
by the slash and following digit, eg we need a /22 block to accommodate all our customers.
All hosts with the same network address (same network bits) must be located together on the same LAN; as
we shall see below, if two hosts share the same network address then they will assume they can reach each
other directly via the underlying LAN, and if they cannot then connectivity fails. A consequence of this rule
is that outside of the site only the network bits need to be looked at to route a packet to the site.
Each individual LAN technology has a maximum packet size it supports; for example, Ethernet has a maximum packet size of about 1500 bytes but the once-competing Token Ring had a maximum of 4 KB. Today
the world has largely standardized on Ethernet and almost entirely standardized on Ethernet packet-size limits, but this was not the case when IP was introduced and there was real concern that two hosts on separate
large-packet networks might try to exchange packets too large for some small-packet intermediate network
to carry.
Therefore, in addition to routing and addressing, the decision was made that IP must also support fragmentation: the division of large packets into multiple smaller ones (in other contexts this may also be called
segmentation). The IP approach is not very efficient, and IP hosts go to considerable lengths to avoid fragmentation. IP does require that packets of up to 576 bytes be supported, and so a common legacy strategy
was for a host to limit a packet to at most 512 user-data bytes whenever the packet was to be sent via a
router; packets addressed to another host on the same LAN could of course use a larger packet size. Despite
its limited use, however, fragmentation is essential conceptually, in order for IP to be able to support large
packets without knowing anything about the intervening networks.
IP is a best effort system; there are no IP-layer acknowledgments or retransmissions. We ship the packet
off, and hope it gets there. Most of the time, it does.
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Architecturally, this best-effort model represents what is known as connectionless networking: the IP layer
does not maintain information about endpoint-to-endpoint connections, and simply forwards packets like a
giant LAN. Responsibility for creating and maintaining connections is left for the next layer up, the TCP
layer. Connectionless networking is not the only way to do things: the alternative could have been some
form connection-oriented internetworking, in which routers do maintain state information about individual
connections. Later, in 3.7 Virtual Circuits, we will examine how virtual-circuit networking can be used to
implement a connection-oriented approach; virtual-circuit switching is the primary alternative to datagram
switching.
Connectionless (IP-style) and connection-oriented networking each have advantages. Connectionless networking is conceptually more reliable: if routers do not hold connection state, then they cannot lose connection state. The path taken by the packets in some higher-level connection can easily be dynamically rerouted.
Finally, connectionless networking makes it hard for providers to bill by the connection; once upon a time
(in the era of dollar-a-minute phone calls) this was a source of mild astonishment to many new users. The
primary advantage of connection-oriented networking, however, is that the routers are then much better positioned to accept reservations and to make quality-of-service guarantees. This remains something of a
sore point in the current Internet: if you want to use Voice-over-IP, or VoIP, telephones, or if you want to
engage in video conferencing, your packets will be treated by the Internet core just the same as if they were
low-priority file transfers. There is no priority service option.
Perhaps the most common form of IP packet loss is router queue overflows, representing network congestion.
Packet losses due to packet corruption are rare (eg less than one in 104 ; perhaps much less). But in a
connectionless world a large number of hosts can simultaneously decide to send traffic through one router,
in which case queue overflows are hard to avoid.

1.10.1 IP Forwarding
IP routers use datagram forwarding, described in 1.4 Datagram Forwarding above, to deliver packets, but
the destination values listed in the forwarding tables are network prefixes representing entire LANs
instead of individual hosts. The goal of IP forwarding, then, becomes delivery to the correct LAN; a separate
process is used to deliver to the final host once the final LAN has been reached.
The entire point, in fact, of having a network/host division within IP addresses is so that routers need to list
only the network prefixes of the destination addresses in their IP forwarding tables. This strategy is the key
to IP scalability: it saves large amounts of forwarding-table space, it saves time as smaller tables allow faster
lookup, and it saves the bandwidth that would be needed for routers to keep track of individual addresses.
To get an idea of the forwarding-table space savings, there are currently (2013) around a billion hosts on the
Internet, but only 300,000 or so networks listed in top-level forwarding tables. When network prefixes are
used as forwarding-table destinations, matching an actual packet address to a forwarding-table entry is no
longer a matter of simple equality comparison; routers must compare appropriate prefixes.
IP forwarding tables are sometimes also referred to as routing tables; in this book, however, we make
at least a token effort to use forwarding to refer to the packet forwarding process, and routing to refer
to mechanisms by which the forwarding tables are maintained and updated. (If we were to be completely
consistent here, we would use the term forwarding loop rather than routing loop.)
Now let us look at a simple example of how IP forwarding (or routing) works. We will assume that all
network nodes are either hosts user machines, with a single network connection or routers, which do
packet-forwarding only. Routers are not directly visible to users, and always have at least two different

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network interfaces representing different networks that the router is connecting. (Machines can be both
hosts and routers, but this introduces complications.)
Suppose A is the sending host, sending a packet to a destination host D. The IP header of the packet will
contain Ds IP address in the destination address field (it will also contain As own address as the source
address). The first step is for A to determine whether D is on the same LAN as itself or not; that is, whether
D is local. This is done by looking at the network part of the destination address, which we will denote by
Dnet . If this net address is the same as As (that is, if it is equal numerically to Anet ), then A figures D is on
the same LAN as itself, and can use direct LAN delivery. It looks up the appropriate physical address for D
(probably with the ARP protocol, 7.7 Address Resolution Protocol: ARP), attaches a LAN header to the
packet in front of the IP header, and sends the packet straight to D via the LAN.
If, however, Anet and Dnet do not match D is non-local then A looks up a router to use. Most ordinary
hosts use only one router for all non-local packet deliveries, making this choice very simple. A then forwards
the packet to the router, again using direct delivery over the LAN. The IP destination address in the packet
remains D in this case, although the LAN destination address will be that of the router.
When the router receives the packet, it strips off the LAN header but leaves the IP header with the IP
destination address. It extracts the destination D, and then looks at Dnet . The router first checks to see
if any of its network interfaces are on the same LAN as D; recall that the router connects to at least one
additional network besides the one for A. If the answer is yes, then the router uses direct LAN delivery to the
destination, as above. If, on the other hand, Dnet is not a LAN to which the router is connected directly, then
the router consults its internal forwarding table. This consists of a list of networks each with an associated
next_hop address. These net,next_hop tables compare with switched-Ethernets host,next_hop tables;
the former type will be smaller because there are many fewer nets than hosts. The next_hop addresses in the
table are chosen so that the router can always reach them via direct LAN delivery via one of its interfaces;
generally they are other routers. The router looks up Dnet in the table, finds the next_hop address, and uses
direct LAN delivery to get the packet to that next_hop machine. The packets IP header remains essentially
unchanged, although the router most likely attaches an entirely new LAN header.
The packet continues being forwarded like this, from router to router, until it finally arrives at a router that
is connected to Dnet ; it is then delivered by that final router directly to D, using the LAN.
To make this concrete, consider the following diagram:

With Ethernet-style forwarding, R2 would have to maintain entries for each of A,B,C,D,E,F. With IP forwarding, R2 has just two entries to maintain in its forwarding table: 200.0.0/24 and 200.0.1/24. If A sends
to D, at 200.0.1.37, it puts this address into the IP header, notes that 200.0.0 = 200.0.1, and thus concludes
D is not a local delivery. A therefore sends the packet to its router R1, using LAN delivery. R1 looks up the
destination network 200.0.1 in its forwarding table and forwards the packet to R2, which in turn forwards it
to R3. R3 now sees that it is connected directly to the destination network 200.0.1, and delivers the packet
via the LAN to D, by looking up Ds physical address.
In this diagram, IP addresses for the ends of the R1R2 and R2R3 links are not shown. They could be
assigned global IP addresses, but they could also use private IP addresses. Assuming these links are

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point-to-point links, they might not actually need IP addresses at all; we return to this in 7.10 Unnumbered
Interfaces.
One can think of the network-prefix bits as analogous to the zip code on postal mail, and the host bits
as analogous to the street address. The internal parts of the post office get a letter to the right zip code,
and then an individual letter carrier gets it to the right address. Alternatively, one can think of the network
bits as like the area code of a phone number, and the host bits as like the rest of the digits. Newer protocols that support different net/host division points at different places in the network sometimes called
hierarchical routing allow support for addressing schemes that correspond to, say, zip/street/user, or
areacode/exchange/subscriber.
The Invertebrate Internet
Once upon a time, each leaf node connected through its provider to the backbone, and traffic between
any two nodes (or at least any two nodes not sharing a provider) passed through the backbone. The
backbone still carries a lot of traffic, but it is now also common for large providers such as Google
to connect (or peer) directly with large residential ISPs such as Comcast. See, for example, peering.google.com.
We will refer to the Internet backbone as those IP routers that specialize in large-scale routing on the
commercial Internet, and which generally have forwarding-table entries covering all public IP addresses;
note that this is essentially a business definition rather than a technical one. We can revise the table-size
claim of the previous paragraph to state that, while there are many private IP networks, there are about
300,000 visible to the backbone. A forwarding table of 300,000 entries is quite feasible; a table a hundred
times larger is not, let alone a thousand times larger.
IP routers at non-backbone sites generally know all locally assigned network prefixes, eg 200.0.0/24 and
200.0.1/24 above. If a destination does not match any locally assigned network prefix, the packet needs
to be routed out into the Internet at large; for typical non-backbone sites this almost always this means
the packet is sent to the ISP that provides Internet connectivity. Generally the local routers will contain a
catchall default entry covering all nonlocal networks; this means that the router needs an explicit entry only
for locally assigned networks. This greatly reduces the forwarding-table size. The Internet backbone can be
approximately described, in fact, as those routers that do not have a default entry.
For most purposes, the Internet can be seen as a combination of end-user LANs together with point-to-point
links joining these LANs to the backbone, point-to-point links also tie the backbone together. Both LANs
and point-to-point links appear in the diagram above.
Just how routers build their destnet,next_hop forwarding tables is a major topic itself, which we cover in
9 Routing-Update Algorithms. Unlike Ethernet, IP routers do not have a broadcast delivery mechanism
as a fallback, so the tables must be constructed in advance. (There is a limited form of IP broadcast, but it
is basically intended for reaching the local LAN only, and does not help at all with delivery in the event that
the network is unknown.)
Most forwarding-table-construction algorithms used on a set of routers under common management fall into
either the distance-vector or the link-state category. In the distance-vector approach, often used at smaller
sites, routers exchange information with their immediately neighboring routers; tables are built up this
way through a sequence of such periodic exchanges. In the link-state approach, routers rapidly propagate
information about the state of each link; all routers in the organization receive this link-state information and
each one uses it to build and maintain a map of the entire network. The forwarding table is then constructed
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(sometimes on demand) from this map.

Unrelated organizations exchange information through the Border Gateway Protocol, BGP (10 Large-Scale
IP Routing), which allows for a fusion of technical information (which sites are reachable at all, and through
where) with policy information representing legal or commercial agreements: which outside routers are
preferred, whose traffic we will carry even if it isnt to one of our customers, etc.
Internet 2 is a consortium of research sites with very-high-bandwidth internal interconnections. Before
Internet 2, Loyolas largest router table consisted of maybe a dozen internal routes, plus one default route
to the outside Internet. After Internet 2, the default route still pointed to the commercial Internet, but the
master forwarding table now had to have an entry for every Internet-2 site so this traffic would take the
Internet-2 path. See 7 IP version 4, exercise 5.

1.10.2 The Future of IPv4

As mentioned earlier, allocation of blocks of IP addresses is the responsibility of the Internet Assigned
Numbers Authority. IANA long ago delegated the job of allocating network prefixes to individual sites; they
limited themselves to handing out /8 blocks (class A blocks) to the five regional registries, which are
ARIN North America
RIPE Europe, the Middle East and parts of Asia
APNIC East Asia and the Pacific
AfriNIC most of Africa
LACNIC Central and South America
As of the end of January 2011, the IANA finally ran out of /8 blocks. There is a table at
http://www.iana.org/assignments/ipv4-address-space/ipv4-address-space.xml of all IANA assignments of /8
blocks; examination of the table shows all have now been allocated.
A few months after the IANA pool ran out, Microsoft purchased 666,624 IP addresses (2604 Class-C blocks)
in a Nortel bankruptcy auction for $7.5 million. By a year later, IP-address prices appeared to have retreated
only slightly. It is possible that the market for IPv4 address blocks will continue to develop; alternatively,
this turn of events may accelerate implementation of IPv6, which has 128-bit addresses.
An IPv4 address price in the range of $10 is unlikely to have much impact in residential Internet access,
where annual connection fees are often $600. Large organizations already use NAT extensively, leading to
the need for only a small number of globally visible addresses. The IPv4 address shortage does not even
seem to have affected wireless networking. It does, however, lead to inefficient routing tables, as a site that
once had a single /20 address block and thus a single backbone forwarding-table entry might now be
spread over more than a hundred /27 blocks and concomitant forwarding entries.

1.11 DNS
IP addresses are hard to remember (nearly impossible in IPv6). The domain name system, or DNS, comes
to the rescue by creating a way to convert hierarchical text names to IP addresses. Thus, for example, one can
type www.luc.edu instead of 147.126.1.230. Virtually all Internet software uses the same basic library
calls to convert DNS names to actual addresses.
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One thing DNS makes possible is changing a websites IP address while leaving the name alone. This
allows moving a site to a new provider, for example, without requiring users to learn anything new. It
is also possible to have several different DNS names resolve to the same IP address, and through some
modest trickery have the http (web) server at that IP address handle the different DNS names as completely
different websites.
DNS is hierarchical and distributed; indeed, it is the classic example of a widely distributed database.
In looking up www.cs.luc.edu three different DNS servers may be queried: for cs.luc.edu, for
luc.edu, and for .edu. Searching a hierarchy can be cumbersome, so DNS search results are normally
cached locally. If a name is not found in the cache, the lookup may take a couple seconds. The DNS
hierarchy need have nothing to do with the IP-address hierarchy.
Besides address lookups, DNS also supports a few other kinds of searches. The best known is probably
reverse DNS, which takes an IP address and returns a name. This is slightly complicated by the fact that
one IP address may be associated with multiple DNS names, so DNS must either return a list, or return one
name that has been designated the canonical name.

1.12 Transport
Think about what types of communications one might want over the Internet:
Interactive communications such as via ssh or telnet, with long idle times between short bursts
Bulk file transfers
Request/reply operations, eg to query a database or to make DNS requests
Real-time voice traffic, at (without compression) 8KB/sec, with constraints on the variation in delivery time (known as jitter; see 19.11.3 RTP Control Protocol for a specific numeric interpretation)
Real-time video traffic. Even with substantial compression, video generally requires much more
bandwidth than voice
While separate protocols might be used for each of these, the Internet has standardized on the Transmission
Control Protocol, or TCP, for the first three (though there are periodic calls for a new protocol addressing
the third item above), and TCP is sometimes pressed into service for the last two. TCP is thus the most
common transport layer for application data.
The IP layer is not well-suited to transport. IP routing is a best-effort mechanism, which means packets
can and do get lost sometimes. Data that does arrive can arrive out of order. The sender has to manage
division into packets; that is, buffering. Finally, IP only supports sending to a specific host; normally, one
wants to send to a given application running on that host. Email and web traffic, or two different web
sessions, should not be commingled!
TCP extends IP with the following features:
reliability: TCP numbers each packet, and keeps track of which are lost and retransmits them after a
timeout, and holds early-arriving out-of-order packets for delivery at the correct time. Every arriving
data packet is acknowledged by the receiver; timeout and retransmission occurs when an acknowledgment isnt received by the sender within a given time.

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connection-orientation: Once a TCP connection is made, an application sends data simply by writing
to that connection. No further application-level addressing is needed.
stream-orientation: The application can write 1 byte at a time, or 100KB at a time; TCP will buffer
and/or divide up the data into appropriate sized packets.
port numbers: these provide a way to specify the receiving application for the data, and also to
identify the sending application.
throughput management: TCP attempts to maximize throughput, while at the same time not contributing unnecessarily to network congestion.
TCP endpoints are of the form host,port; these pairs are known as socket addresses, or sometimes as just
sockets though the latter refers more properly to the operating-system objects that receive the data sent to
the socket addresses. Servers (or, more precisely, server applications) listen for connections to sockets they
have opened; the client is then any endpoint that initiates a connection to a server.
When you enter a host name in a web browser, it opens a TCP connection to the servers port 80 (the standard
web-traffic port), that is, to the server socket with socket-address server,80. If you have several browser
tabs open, each might connect to the same server socket, but the connections are distinguishable by virtue
of using separate ports (and thus having separate socket addresses) on the client end (that is, your end).
A busy server may have thousands of connections to its port 80 (the web port) and hundreds of connections
to port 25 (the email port). Web and email traffic are kept separate by virtue of the different ports used. All
those clients to the same port, though, are kept separate because each comes from a unique host,port pair. A
TCP connection is determined by the host,port socket address at each end; traffic on different connections
does not intermingle. That is, there may be multiple independent connections to www.luc.edu,80. This is
somewhat analogous to certain business telephone numbers of the operators are standing by type, which
support multiple callers at the same time to the same number. Each call is answered by a different operator
(corresponding to a different cpu process), and different calls do not overhear each other.
TCP uses the sliding-windows algorithm, 6 Abstract Sliding Windows, to keep multiple packets en route
at any one time. The window size represents the number of packets simultaneously en route; if the window
size is 10, for example, then at any one time 10 packets are out there (perhaps 5 data packets and 5 returning
acknowledgments). As each acknowledgment arrives, the window slides forward and the data packet 10
packets ahead is sent. For example, consider the moment when the ten packets 20-29 are in transit. When
ACK[20] is received, Data[30] is sent, and so now packets 21-30 are in transit. When ACK[21] is received,
Data[31] is sent, so packets 22-31 are in transit.
Sliding windows minimizes the effect of store-and-forward delays, and propagation delays, as these then
only count once for the entire windowful and not once per packet. Sliding windows also provides an automatic, if partial, brake on congestion: the queue at any switch or router along the way cannot exceed the
window size. In this it compares favorably with constant-rate transmission, which, if the available bandwidth falls below the transmission rate, always leads to a significant percentage of dropped packets. Of
course, if the window size is too large, a sliding-windows sender may also experience dropped packets.
The ideal window size, at least from a throughput perspective, is such that it takes one round-trip time to send
an entire window, so that the next ACK will always be arriving just as the sender has finished transmitting the
window. Determining this ideal size, however, is difficult; for one thing, the ideal size varies with network
load. As a result, TCP approximates the ideal size. The most common TCP strategy that of so-called TCP
Reno is that the window size is slowly raised until packet loss occurs, which TCP takes as a sign that it
has reached the limit of available network resources. At that point the window size is reduced to half its
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previous value, and the slow climb resumes. The effect is a sawtooth graph of window size with time,
which oscillates (more or less) around the optimal window size. For an idealized sawtooth graph, see
13.1.1 The Steady State; for some real (simulation-created) sawtooth graphs see 16.4.1 Some TCP Reno
cwnd graphs.
While this window-size-optimization strategy has its roots in attempting to maximize the available bandwidth, it also has the effect of greatly limiting the number of packet-loss events. As a result, TCP has come
to be the Internet protocol charged with reducing (or at least managing) congestion on the Internet, and
relatedly with ensuring fairness of bandwidth allocations to competing connections. Core Internet routers
at least in the classical case essentially have no role in enforcing congestion or fairness restrictions at all.
The Internet, in other words, places responsibility for congestion avoidance cooperatively into the hands of
end users. While cheating is possible, this cooperative approach has worked remarkably well.
While TCP is ubiquitous, the real-time performance of TCP is not always consistent: if a packet is lost,
the receiving TCP host will not turn over anything further to the receiving application until the lost packet
has been retransmitted successfully; this is often called head-of-line blocking. This is a serious problem
for sound and video applications, which can discretely handle modest losses but which have much more
difficulty with sudden large delays. A few lost packets ideally should mean just a few brief voice dropouts
(pretty common on cell phones) or flicker/snow on the video screen (or just reuse of the previous frame);
both of these are better than pausing completely.
The basic alternative to TCP is known as UDP, for User Datagram Protocol. UDP, like TCP, provides port
numbers to support delivery to multiple endpoints within the receiving host, in effect to a specific process on
the host. As with TCP, a UDP socket consists of a host,port pair. UDP also includes, like TCP, a checksum
over the data. However, UDP omits the other TCP features: there is no connection setup, no lost-packet
detection, no automatic timeout/retransmission, and the application must manage its own packetization.
The Real-time Transport Protocol, or RTP, sits above UDP and adds some additional support for voice and
video applications.

1.13 Firewalls
One problem with having a program on your machine listening on an open TCP port is that someone may
connect and then, using some flaw in the software on your end, do something malicious to your machine.
Damage can range from the unintended downloading of personal data to compromise and takeover of your
entire machine, making it a distributor of viruses and worms or a steppingstone in later break-ins of other
machines.
A strategy known as buffer overflow has been the basis for a great many total-compromise attacks. The idea
is to identify a point in a server program where it fills a memory buffer with network-supplied data without
careful length checking; almost any call to the C library function gets(buf) will suffice. The attacker
then crafts an oversized input string which, when read by the server and stored in memory, overflows the
buffer and overwrites subsequent portions of memory, typically containing the stack-frame pointers. The
usual goal is to arrange things so that when the server reaches the end of the currently executing function,
control is returned not to the calling function but instead to the attackers own payload code located within
the string.
A firewall is a program to block connections deemed potentially risky, eg those originating from outside
the site. Generally ordinary workstations do not ever need to accept connections from the Internet; client

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machines instead initiate connections to (better-protected) servers. So blocking incoming connections works
pretty well; when necessary (eg for games) certain ports can be selectively unblocked.
The original firewalls were routers. Incoming traffic to servers was often blocked unless it was sent to one
of a modest number of open ports; for non-servers, typically all inbound connections were blocked. This
allowed internal machines to operate reasonably safely, though being unable to accept incoming connections
is sometimes inconvenient. Nowadays per-machine firewalls in addition to router-based firewalls are
common: you can configure your machine not to accept inbound connections to most (or all) ports regardless
of whether software on your machine requests such a connection. Outbound connections can, in many cases,
also be prevented.

1.14 Network Address Translation

A related way of providing firewall protection for ordinary machines is Network Address Translation, or
NAT. This technique also conserves IP addresses. Instead of assigning each host at a site a real IP address,
just one address is assigned to a border router. Internal machines get internal IP addresses, typically of the
form or 192.168.x.y or 10.x.y.z (addresses beginning with 192.168.0.0/16 or with 10.0.0.0/8 cannot be used
on the Internet). Inbound connections to the internal machines are banned, as with some firewalls. When
an internal machine wants to connect to the outside, the NAT router intercepts the connection, allocates a
new port, and forwards the connection on as if it came from that new port and its own IP address. The
remote machine responds, and the NAT router remembers the connection and forwards data to the correct
internal host, rewriting the address and port fields of the incoming packets. Done properly, NAT improves
the security of a site. It also allows multiple machines to share a single IP address, and so is popular with
home users who have a single broadband connection but who wish to use multiple computers. A typical
NAT router sold for residential or small-office use is commonly simply called a router, or (somewhat
more precisely) a residential gateway.
Suppose the NAT router is NR, and internal hosts A and B each connect from port 3000 to port 80 on
external hosts C and D, respectively. Here is what NRs NAT table might look like. No column for NRs IP
address is given, as it is fixed.
remote host
C
D

remote port
80
80

outside source port

3000
3000

inside host
A
B

inside port
3000
3000

A packet to C from A,3000 would be rewritten by NR so that the source was NR,3000. A packet from
C,80 addressed to NR,3000 would be rewritten and forwarded to A,3000. Similarly, a packet from
D,80 addressed to NR,3000 would be rewritten and forwarded to B,3000; the NAT table takes into
account the sending socket address as well as the destination.
Now suppose B opens a connection to C,80, also from inside port 3000. This time NR must remap the
port number, because that is the only way to distinguish between packets from C,80 to A and to B. The
new table is
remote host
C
D
C

remote port
80
80
80

outside source port

3000
3000
3001

1.14 Network Address Translation

inside host
A
B
B

inside port
3000
3000
3000

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Typically NR would not create TCP connections between itself and C,80 and D,80; the NAT table does
forwarding but the endpoints of the connection are still at the inside hosts. However, NR might very well
monitor the TCP connections to know when they have closed, and so can be removed from the table.
It is common for Voice-over-IP (VoIP) telephony using the SIP protocol (RFC 3261) to prefer to use UDP
port 5060 at both ends. If a VoIP server is outside the NAT router (which must be the case as the server
must generally be publicly visible) and a telephone is inside, likely port 5060 will pass through without
remapping, though the telephone will have to initiate the connection. But if there are two phones inside, one
of them will appear to be connecting to the server from an alternative port.
VoIP systems run into a much more serious problem with NAT, however. A call ultimately between two
phones is typically first negotiated between the phones respective VoIP servers. Once the call is set up, the
servers would prefer to step out of the loop, and have the phones exchange voice packets directly. The SIP
protocol was designed to handle this by having each phone report to its respective server the UDP socket
(IP address,port pair) it intends to use for the voice exchange; the servers then report these phone sockets
to each other, and from there to the opposite phones. This socket information is rendered incorrect by NAT,
however, certainly the IP address and quite likely the port as well. If only one of the phones is behind a
NAT firewall, it can initiate the voice connection to the other phone, but the other phone will see the voice
packets arriving from a different socket than promised and will likely not recognize them as part of the call.
If both phones are behind NAT firewalls, they will not be able to connect to one another at all. The common
solution is for the VoIP server of a phone behind a NAT firewall to remain in the communications path,
forwarding packets to its hidden partner.
If a site wants to make it possible to allow connections to hosts behind a NAT router or other firewall, one
option is tunneling. This is the creation of a virtual LAN link that runs on top of a TCP connection
between the end user and one of the sites servers; the end user can thus appear to be on one of the organizations internal LANs; see 3.1 Virtual Private Network. Another option is to open up a specific port: in
essence, a static NAT-table entry is made connecting a specific port on the NAT router to a specific internal
host and port (usually the same port). For example, all UDP packets to port 5060 on the NAT router might
be forwarded to port 5060 on internal host A, even in the absence of any prior packet exchange.
NAT routers work very well when the communications model is of client-side TCP connections, originating
from the inside and with public outside servers as destination. The NAT model works less well for peerto-peer networking, where your computer and a friends, each behind a different NAT router, wish to
establish a connection. NAT routers also often have trouble with UDP protocols, due to the tendency for
such protocols to have the public server reply from a different port than the one originally contacted. For
example, if host A behind a NAT router attempts to use TFTP (11.3 Trivial File Transport Protocol, TFTP),
and sends a packet to port 69 of public server C, then C is likely to reply from some new port, say 3000,
and this reply is likely to be dropped by the NAT router as there will be no entry there yet for traffic from
C,3000.

1.15 IETF and OSI

The Internet protocols discussed above are defined by the Internet Engineering Task Force, or IETF (under
the aegis of the Internet Architecture Board, or IAB, in turn under the aegis of the Internet Society,
ISOC). The IETF publishes Request For Comment or RFC documents that contain all the formal Internet
standards; these are available at http://www.ietf.org/rfc.html (note that, by the time a document appears
here, the actual comment-requesting period is generally long since closed). The five-layer model is closely

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associated with the IETF, though is not an official standard.

RFC standards sometimes allow modest flexibility. With this in mind, RFC 2119 declares official understandings for the words MUST and SHOULD. A feature labeled with MUST is an absolute requirement
for the specification, while the term SHOULD is used when
there may exist valid reasons in particular circumstances to ignore a particular item, but the
full implications must be understood and carefully weighed before choosing a different course.
The original ARPANET network was developed by the US governments Defense Advanced Research
Projects Agency, or DARPA; it went online in 1969. The National Science Foundation began NSFNet in
1986; this largely replaced ARPANET. In 1991, operation of the NSFNet backbone was turned over to
ANSNet, a private corporation. The ISOC was founded in 1992 as the NSF continued to retreat from the
networking business.
Hallmarks of the IETF design approach were David Clarks declaration
We reject: kings, presidents and voting.
We believe in: rough consensus and running code.
and RFC Editor Jon Postels aphorism
Be liberal in what you accept, and conservative in what you send.
There is a persistent though false notion that the distributed-routing architecture of IP was due to a US
Department of Defense mandate that the original ARPAnet be able to survive a nuclear attack. In fact, the
developers of IP seemed unconcerned with this. However, Paul Baran did write, in his 1962 paper outlining
the concept of packet switching, that
If [the number of stations] is made sufficiently large, it can be shown that highly survivable
system structures can be built even in the thermonuclear era.
In 1977 the International Organization for Standardization, or ISO, founded the Open Systems Interconnection project, or OSI, a process for creation of new network standards. OSI represented an attempt at the
creation of networking standards independent of any individual government.
The OSI project is today perhaps best known for its seven-layer networking model: between Transport and
Application were inserted the Session and Presentation layers. The Session layer was to handle sessions
between applications (including the graceful closing of Transport-layer connections, something included
in TCP, and the re-establishment of broken Transport-layer connections, which TCP could sorely use),
and the Presentation layer was to handle things like defining universal data formats (eg for binary numeric
data, or for non-ASCII character sets), and eventually came to include compression and encryption as well.
However, most of the features at this level have ended up generally being implemented in the Application
layer of the IETF model with no inconvenience. Indeed, there are few examples in the TCP/IP world of data
format, compression or encryption being handled as a sublayer of the application, in which the sublayer
is responsible for connections to the Transport layer (SSL/TLS might be an example for encryption; applications read and write data directly to the SSL endpoint). Instead, applications usually read and write data
directly from/to the TCP layer, and then invoke libraries to handle things like data conversion, compression
and encryption.
OSI has its own version of IP and TCP. The IP equivalent is CLNP, the ConnectionLess Network Protocol,
although OSI also defines a connection-oriented protocol CMNS. The TCP equivalent is TP4; OSI also
defines TP0 through TP3 but those are for connection-oriented networks.

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It seems clear that the primary reasons the OSI protocols failed in the marketplace were their ponderous
bureaucracy for protocol management, their principle that protocols be completed before implementation
began, and their insistence on rigid adherence to the specifications to the point of non-interoperability. In
contrast, the IETF had (and still has) a two working implementations rule for a protocol to become a
Draft Standard. From RFC 2026:
A specification from which at least two independent and interoperable implementations from different
code bases have been developed, and for which sufficient successful operational experience has been
obtained, may be elevated to the Draft Standard level. [emphasis added]
This rule has often facilitated the discovery of protocol design weaknesses early enough that the problems
could be fixed. The OSI approach is a striking failure for the waterfall design model, when competing
with the IETFs cyclic prototyping model. However, it is worth noting that the IETF has similarly been
unable to keep up with rapid changes in html, particularly at the browser end; the OSI mistakes were mostly
evident only in retrospect.
Trying to fit protocols into specific layers is often both futile and irrelevant. By one perspective, the RealTime Protocol RTP lives at the Transport layer, but just above the UDP layer; others have put RTP into the
Application layer. Parts of the RTP protocol resemble the Session and Presentation layers. A key component
of the IP protocol is the set of various router-update protocols; some of these freely use higher-level layers.
Similarly, tunneling might be considered to be a Link-layer protocol, but tunnels are often created and
maintained at the Application layer.
A sometimes-more-successful approach to understanding layers is to view them instead as parts of a
protocol graph. Thus, in the following diagram we have two protocols at the transport layer (UDP and
RTP), and one protocol (ARP) not easily assigned to a layer.

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1.16 Berkeley Unix

Though not officially tied to the IETF, the Berkeley Unix releases became de facto reference implementations for most of the TCP/IP protocols. 4.1BSD (BSD for Berkeley Software Distribution) was released in
1981, 4.2BSD in 1983, 4.3BSD in 1986, 4.3BSD-Tahoe in 1988, 4.3BSD-Reno in 1990, and 4.4BSD in
1994. Descendants today include FreeBSD, OpenBSD and NetBSD. The TCP implementations TCP Tahoe
and TCP Reno (13 TCP Reno and Congestion Management) took their names from the corresponding
4.3BSD releases.

1.17 Epilog
This completes our tour of the basics. In the remaining chapters we will expand on the material here.

1.18 Exercises
1. Give forwarding tables for each of the switches S1-S4 in the following network with destinations A, B,
C, D. For the next_hop column, give the neighbor on the appropriate link rather than the interface number.
A

S1

S2

S3

S4

2. Give forwarding tables for each of the switches S1-S4 in the following network with destinations A, B,
C, D. Again, use the neighbor form of next_hop rather than the interface form. Try to keep the route to
each destination as short as possible. What decision has to be made in this exercise that did not arise in the
preceding exercise?
A

S1

S2

S4

S3

3. Consider the following arrangement of switches and destinations. Give forwarding tables (in neighbor
form) for S1-S4 that include default forwarding entries; the default entries should point toward S5. Eliminate all table entries that are implied by the default entry (that is, if the default entry is to S3, eliminate all
other entries for which the next hop is S3).
A

S1
D

S3

S2

S4

S5

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4. Four switches are arranged as below. The destinations are S1 through S4 themselves.
S1

S2

S4

S3

(a). Give the forwarding tables for S1 through S4 assuming packets to adjacent nodes are sent along the
connecting link, and packets to diagonally opposite nodes are sent clockwise.
(b). Give the forwarding tables for S1 through S4 assuming the S1S4 link is not used at all, not even for
S1S4 traffic.

5. Suppose we have switches S1 through S4; the forwarding-table destinations are the switches themselves. The table
S2: S1,S1 S3,S3 S4,S3
S3: S1,S2 S2,S2 S4,S4
From the above we can conclude that S2 must be directly connected to both S1 and S3 as its table lists them
as next_hops; similarly, S3 must be directly connected to S2 and S4.

(a). Must S1 and S4 be directly connected? If so, explain; if not, give a network in which there is no direct
link between them, consistent with the tables above.
(b). Now suppose S3s table is changed to the following. In this case must S1 and S4 be directly
connected? Why or why not?
S3: S1,S4 S2,S2 S4,S4
While the table for S4 is not given, you may assume that forwarding does work correctly. However, you
should not assume that paths are the shortest possible; in particular, you should not assume that each switch
will always reach its directly connected neighbors by using the direct connection.
6. (a) Suppose a network is as follows, with the only path from A to C passing through B:
...

...

Explain why a single routing loop cannot include both A and C.

(b). Suppose a routing loop follows the path A S1 S2 ... Sn A, where none of the Si are equal to
A. Show that all the Si must be distinct. (A corollary of this is that any routing loop created by datagramforwarding either involves forwarding back and forth between a pair of adjacent switches, or else involves
an actual graph cycle in the network topology; linear loops of length greater than 1 are impossible.)
7. Consider the following arrangement of switches:
S1

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S2

S5

S11

S3

S6

S12

Suppose S1-S6 have the forwarding tables below. For each destination A,B,C,D,E,F, suppose a packet is
sent to the destination from S1. Give the switches it passes through, including the initial switch S1, up until
the final switch S10-S12.
S1: (A,S4), (B,S2), (C,S4), (D,S2), (E,S2), (F,S4)
S2: (A,S5), (B,S5), (D,S5), (E,S3), (F,S3)
S3: (B,S6), (C,S2), (E,S6), (F,S6)
S4: (A,S10), (C,S5), (E,S10), (F,S5)
S5: (A,S6), (B,S11), (C,S6), (D,S6), (E,S4), (F,S2)
S6: (A,S3), (B,S12), (C,S12), (D,S12), (E,S5), (F,S12)
8. In the previous exercise, the routes taken by packets A-D are reasonably direct, but the routes for E and F
are rather circuitous.
Some routing applications assign weights to different links, and attempt to choose a path with the lowest
total link weight.

(a). Assign weights to the seven links S1S2, S2S3, S1S4, S2S5, S3S6, S4S5 and S5S6 so that
destination Es route in the previous exercise becomes the optimum (lowest total link weight) path.
(b). Assign (different!) weights to the seven links that make destination Fs route in the previous exercise
optimal.

Hint: you can do this by assigning a weight of 1 to all links except to one or two bad links; the bad links
get a weight of 10. In each of (a) and (b) above, the route taken will be the route that avoids all the bad
links. You must treat (a) entirely differently from (b); there is no assignment of weights that can account for
both routes.
9. Suppose we have the following three Class C IP networks, joined by routers R1R4. Give the forwarding
table for each router. For networks directly connected to a router (eg 200.0.1/24 and R1), include the network
in the table but list the next hop as direct.
R1

R4

R3

200.0.1/24

R2

200.0.2/24

200.0.3/24

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2 ETHERNET

We now turn to a deeper analysis of the ubiquitous Ethernet LAN protocol. Current user-level Ethernet today
(2013) is usually 100 Mbps, with Gigabit Ethernet standard in server rooms and backbones, but because
Ethernet speed scales in odd ways, we will start with the 10 Mbps formulation. While the 10 Mbps speed is
obsolete, and while even the Ethernet collision mechanism is largely obsolete, collision management itself
continues to play a significant role in wireless networks.

2.1 10-Mbps classic Ethernet

The original Ethernet specification was the 1976 paper of Metcalfe and Boggs, [MB76]. The data rate was
10 megabits per second, and all connections were made with coaxial cable instead of todays twisted pair. In
its original form, an Ethernet was a broadcast bus, which meant that all packets were, at least at the physical
layer, broadcast onto the shared medium and could be seen, theoretically, by all other nodes. If two nodes
transmitted at the same time, there was a collision; proper handling of collisions was an important part of
the access-mediation strategy for the shared medium. Data was transmitted using Manchester encoding; see
4.1.3 Manchester.
The linear bus structure could be modified with repeaters (below), into an arbitrary tree structure, though
loops remain something of a problem even with todays Ethernet.
Whenever two stations transmitted at the same time, the signals would collide, and interfere with one another; both transmissions would fail as a result. In order to minimize collision loss, each station implemented
the following:
1. Before transmission, wait for the line to become quiet
2. While transmitting, continually monitor the line for signs that a collision has occurred; if a collision
happens, then cease transmitting
3. If a collision occurs, use a backoff-and-retransmit strategy
These properties can be summarized with the CSMA/CD acronym: Carrier Sense, Multiple Access, Collision Detect. (The term carrier sense was used by Metcalfe and Boggs as a synonym for signal sense;
there is no literal carrier frequency to be sensed.) It should be emphasized that collisions are a normal event
in Ethernet, well-handled by the mechanisms above.
IEEE 802 Network Standards
The IEEE network standards all begin with 802: 802.3 is Ethernet, 802.11 is Wi-Fi, 802.16 is WiMAX,
and there are many others. One sometimes encounters the claim that 802 represents the date of an early
meeting: February 1980. However, the IEEE has a continuous stream of standards (with occasional
gaps): 799: Handling and Disposal of Transformer PCBs, 800: D-C Aircraft Rotating Machines, 803:
Recommended Practice for Unique Identification in Power Plants, etc.

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Classic Ethernet came in version 1 [1980, DEC-Intel-Xerox], version 2 [1982, DIX], and IEEE 802.3. There
are some minor electrical differences between these, and one rather substantial packet-format difference. In
addition to these, the Berkeley Unix trailing-headers packet format was used for a while.
There were three physical formats for 10 Mbps Ethernet cable: thick coax (10BASE-5), thin coax (10BASE2), and, last to arrive, twisted pair (10BASE-T). Thick coax was the original; economics drove the successive
development of the later two. The cheaper twisted-pair cabling eventually almost entirely displaced coax, at
least for host connections.
The original specification included support for repeaters, which were in effect signal amplifiers although
they might attempt to clean up a noisy signal. Repeaters processed each bit individually and did no buffering.
In the telecom world, a repeater might be called a digital regenerator. A repeater with more than two ports
was commonly called a hub; hubs allowed branching and thus much more complex topologies.
Bridges later known as switches came along a short time later. While repeaters act at the bit layer,
a switch reads in and forwards an entire packet as a unit, and the destination address is likely consulted
to determine to where the packet is forwarded. Originally, switches were seen as providing interconnection (bridging) between separate Ethernets, but later a switched Ethernet was seen as one large virtual
Ethernet. We return to switching below in 2.4 Ethernet Switches.
Hubs propagate collisions; switches do not. If the signal representing a collision were to arrive at one port of
a hub, it would, like any other signal, be retransmitted out all other ports. If a switch were to detect a collision
one one port, no other ports would be involved; only packets received successfully are ever retransmitted
out other ports.
In coaxial-cable installations, one long run of coax snaked around the computer room or suite of offices;
each computer connected somewhere along the cable. Thin coax allowed the use of T-connectors to attach
hosts; connections were made to thick coax via taps, often literally drilled into the coax central conductor.
In a standalone installation one run of coax might be the entire Ethernet; otherwise, somewhere a repeater
would be attached to allow connection to somewhere else.
Twisted-pair does not allow mid-cable attachment; it is only used for point-to-point links between hosts,
switches and hubs. In a twisted-pair installation, each cable runs between the computer location and a
central wiring closest (generally much more convenient than trying to snake coax all around the building).
Originally each cable in the wiring closet plugged into a hub; nowadays the hub has likely been replaced by
a switch.
There is still a role for hubs today when one wants to monitor the Ethernet signal from A to B (eg for
intrusion detection analysis), although some switches now also support a form of monitoring.
All three cable formats could interconnect, although only through repeaters and hubs, and all used the same
10 Mbps transmission speed. While twisted-pair cable is still used by 100 Mbps Ethernet, it generally needs
to be a higher-performance version known as Category 5, versus the 10 Mbps Category 3.
Here is the format of a typical Ethernet packet (DIX specification):

The destination and source addresses are 48-bit quantities; the type is 16 bits, the data length is variable up
to a maximum of 1500 bytes, and the final CRC checksum is 32 bits. The checksum is added by the Ethernet
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hardware, never by the host software. There is also a preamble, not shown: a block of 1 bits followed by a
0, in the front of the packet, for synchronization. The type field identifies the next higher protocol layer; a
few common type values are 0x0800 = IP, 0x8137 = IPX, 0x0806 = ARP.
The IEEE 802.3 specification replaced the type field by the length field, though this change never caught on.
The two formats can be distinguished as long as the type values used are larger than the maximum Ethernet
length of 1500 (or 0x05dc); the type values given in the previous paragraph all meet this condition.
Each Ethernet card has a (hopefully unique) physical address in ROM; by default any packet sent to this
address will be received by the board and passed up to the host system. Packets addressed to other physical
addresses will be seen by the card, but ignored (by default). All Ethernet devices also agree on a broadcast
address of all 1s: a packet sent to the broadcast address will be delivered to all attached hosts.
It is sometimes possible to change the physical address of a given card in software. It is almost universally
possible to put a given card into promiscuous mode, meaning that all packets on the network, no matter
what the destination address, are delivered to the attached host. This mode was originally intended for
diagnostic purposes but became best known for the security breach it opens: it was once not unusual to find
a host with network board in promiscuous mode and with a process collecting the first 100 bytes (presumably
including userid and password) of every telnet connection.

2.1.1 Ethernet Multicast

Another category of Ethernet addresses is multicast, used to transmit to a set of stations; streaming video
to multiple simultaneous viewers might use Ethernet multicast. The lowest-order bit in the first byte of
an address indicates whether the address is physical or multicast. To receive packets addressed to a given
multicast address, the host must inform its network interface that it wishes to do so; once this is done, any
arriving packets addressed to that multicast address are forwarded to the host. The set of subscribers to a
given multicast address may be called a multicast group. While higher-level protocols might prefer that the
subscribing host also notifies some other host, eg the sender, this is not required, although that might be the
easiest way to learn the multicast address involved. If several hosts subscribe to the same multicast address,
then each will receive a copy of each multicast packet transmitted.
If switches (below) are involved, they must normally forward multicast packets on all outbound links, exactly
as they do for broadcast packets; switches have no obvious way of telling where multicast subscribers might
be. To avoid this, some switches do try to engage in some form of multicast filtering, sometimes by snooping
on higher-layer multicast protocols. Multicast Ethernet is seldom used by IPv4, but plays a larger role in
IPv6 configuration.
The second-to-lowest-order bit of the Ethernet address indicates, in the case of physical addresses, whether
the address is believed to be globally unique or if it is only locally unique; this is known as the Universal/Local bit. For real Ethernet physical addresses, the multicast and universal/local bits of the first byte
should both be 0.
When (global) Ethernet IDs are assigned by the manufacturer, the first three bytes serve to indicate the
manufacturer. They are allocated by the IEEE, and are officially known as organizationally unique identifiers. These can be looked up at any of several sites on the Internet to identify the manufacturer associated
with any given Ethernet address; the official IEEE site is standards.ieee.org/develop/regauth/oui/public.html
(OUIs must be entered here without colons).

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As long as the manufacturer involved is diligent in assigning the second three bytes, every manufacturerprovided Ethernet address should be globally unique. Lapses, however, are not unheard of.

2.1.2 The LAN Layer

The LAN layer, at its upper end, supplies to the network layer a mechanism for addressing a packet and
sending it from one station to another. At its lower end, it handles interactions with the physical layer. The
LAN layer covers packet addressing, delivery and receipt, forwarding, error detection, collision detection
and collision-related retransmission attempts.
In IEEE protocols, the LAN layer is divided into the media access control, or MAC, sublayer and a
higher logical link control, or LLC, sublayer for higher-level flow-control functions that today have moved
largely to the transport layer. For example, the HDLC protocol (4.1.5.1 HDLC) supports sliding windows
(6.2 Sliding Windows) as an option, as did the early X.25 protocol. ATM, 3.8 Asynchronous Transfer
Mode: ATM, also supports some higher-level functions, though not sliding windows.
Because the LLC layer is so often insignificant, and because the most well-known LAN-layer functions are
in fact part of the MAC sublayer, it is common to identify the LAN layer with its MAC sublayer, especially
for IEEE protocols where the MAC layer has official standing. In particular, LAN-layer addresses are
perhaps most often called MAC addresses.
Generally speaking, much of the operation of the LAN/MAC layer takes place in the network card. Host
systems (including drivers) are, for example, generally oblivious to collisions (although they may query the
card for collision statistics). In some cases, eg with Wi-Fi rate scaling (3.3.3 Dynamic Rate Scaling), the
host-system driver may get involved.

2.1.3 The Slot Time and Collisions

The diameter of an Ethernet is the maximum distance between any pair of stations. The actual total length
of cable can be much greater than this, if, for example, the topology is a star configuration. The maximum
allowed diameter, measured in bits, is limited to 232 (a sample budget for this is below). This makes
the round-trip-time 464 bits. As each station involved in a collision discovers it, it transmits a special jam
signal of up to 48 bits. These 48 jam bits bring the total above to 512 bits, or 64 bytes. The time to send
these 512 bits is the slot time of an Ethernet; time intervals on Ethernet are often described in bit times but
in conventional time units the slot time is 51.2 sec.
The value of the slot time determines several subsequent aspects of Ethernet. If a station has transmitted
for one slot time, then no collision can occur (unless there is a hardware error) for the remainder of that
packet. This is because one slot time is enough time for any other station to have realized that the first
station has started transmitting, so after that time they will wait for the first station to finish. Thus, after one
slot time a station is said to have acquired the network. The slot time is also used as the basic interval for
retransmission scheduling, below.
Conversely, a collision can be received, in principle, at any point up until the end of the slot time. As a result,
Ethernet has a minimum packet size, equal to the slot time, ie 64 bytes (or 46 bytes in the data portion). A
station transmitting a packet this size is assured that if a collision were to occur, the sender would detect it
(and be able to apply the retransmission algorithm, below). Smaller packets might collide and yet the sender
not know it, ultimately leading to greatly reduced throughput.

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If we need to send less than 46 bytes of data (for example, a 40-byte TCP ACK packet), the Ethernet packet
must be padded out to the minimum length. As a result, all protocols running on top of Ethernet need to
provide some way to specify the actual data length, as it cannot be inferred from the received packet size.
As a specific example of a collision occurring as late as possible, consider the diagram below. A and B are
5 units apart, and the bandwidth is 1 byte/unit. A begins sending helloworld at T=0; B starts sending just
as As message arrives, at T=5. B has listened before transmitting, but As signal was not yet evident. A
doesnt discover the collision until 10 units have elapsed, which is twice the distance.

Here are typical maximum values for the delay in 10 Mbps Ethernet due to various components. These
are taken from the Digital-Intel-Xerox (DIX) standard of 1982, except that point-to-point link cable is
replaced by standard cable. The DIX specification allows 1500m of coax with two repeaters and 1000m
of point-to-point cable; the table below shows 2500m of coax and four repeaters, following the later IEEE
802.3 Ethernet specification. Some of the more obscure delays have been eliminated. Entries are one-way
delay times, in bits. The maximum path may have four repeaters, and ten transceivers (simple electronic
devices between the coax cable and the NI cards), each with its drop cable (two transceivers per repeater,
plus one at each endpoint).
Ethernet delay budget
item
coax
transceiver cables
transceivers
repeaters
encoders

length
2500M
500M

delay, in bits
110 bits
25 bits
40 bits, max 10 units
25 bits, max 4 units
20 bits, max 10 units

explanation (c = speed of light)

23 meters/bit (.77c)
19.5 meters/bit (.65c)
4 bits each
6+ bits each (DIX 7.6.4.1)
2 bits each (for signal generation)

The total here is 220 bits; in a full accounting it would be 232. Some of the numbers shown are a little high,
but there are also signal rise time delays, sense delays, and timer delays that have been omitted. It works out
fairly closely.
Implicit in the delay budget table above is the length of a bit. The speed of propagation in copper is about
0.77c, where c=3108 m/sec = 300 m/sec is the speed of light in vacuum. So, in 0.1 microseconds (the

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time to send one bit at 10 Mbps), the signal propagates approximately 0.77c10-7 = 23 meters.
Ethernet packets also have a maximum packet size, of 1500 bytes. This limit is primarily for the sake
of fairness, so one station cannot unduly monopolize the cable (and also so stations can reserve buffers
guaranteed to hold an entire packet). At one time hardware vendors often marketed their own incompatible
extensions to Ethernet which enlarged the maximum packet size to as much as 4KB. There is no technical
reason, actually, not to do this, except compatibility.
The signal loss in any single segment of cable is limited to 8.5 db, or about 14% of original strength.
Repeaters will restore the signal to its original strength. The reason for the per-segment length restriction
is that Ethernet collision detection requires a strict limit on how much the remote signal can be allowed to
lose strength. It is possible for a station to detect and reliably read very weak remote signals, but not at the
same time that it is transmitting locally. This is exactly what must be done, though, for collision detection
to work: remote signals must arrive with sufficient strength to be heard even while the receiving station is
itself transmitting. The per-segment limit, then, has nothing to do with the overall length limit; the latter is
set only to ensure that a sender is guaranteed of detecting a collision, even if it sends the minimum-sized
packet.

2.1.4 Exponential Backoff Algorithm

Whenever there is a collision the exponential backoff algorithm operating at the MAC layer is used
to determine when each station will retry its transmission. Backoff here is called exponential because the
range from which the backoff value is chosen is doubled after every successive collision involving the same
packet. Here is the full Ethernet transmission algorithm, including backoff and retransmissions:
1. Listen before transmitting (carrier detect)
2. If line is busy, wait for sender to stop and then wait an additional 9.6 microseconds (96 bits). One
consequence of this is that there is always a 96-bit gap between packets, so packets do not run together.
3. Transmit while simultaneously monitoring for collisions
4. If a collision does occur, send the jam signal, and choose a backoff time as follows: For transmission
N, 1N10 (N=0 represents the original attempt), choose k randomly with 0 k < 2N . Wait k slot
times (k51.2 sec). Then check if the line is idle, waiting if necessary for someone else to finish,
and then retry step 3. For 11N15, choose k randomly with 0 k < 1024 (= 210 )
5. If we reach N=16 (16 transmission attempts), give up.
If an Ethernet sender does not reach step 5, there is a very high probability that the packet was delivered
successfully.
Exponential backoff means that if two hosts have waited for a third to finish and transmit simultaneously,
and collide, then when N=1 they have a 50% chance of recollision; when N=2 there is a 25% chance, etc.
When N10 the maximum wait is 52 milliseconds; without this cutoff the maximum wait at N=15 would
be 1.5 seconds. As indicated above in the minimum-packet-size discussion, this retransmission strategy
assumes that the sender is able to detect the collision while it is still sending, so it knows that the packet
must be resent.
In the following diagram is an example of several stations attempting to transmit all at once, and using the
above transmission/backoff algorithm to sort out who actually gets to acquire the channel. We assume we
have five prospective senders A1, A2, A3, A4 and A5, all waiting for a sixth station to finish. We will
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assume that collision detection always takes one slot time (it will take much less for nodes closer together)
and that the slot start-times for each station are synchronized; this allows us to measure time in slots. A solid
arrow at the start of a slot means that sender began transmission in that slot; a red X signifies a collision. If
a collision occurs, the backoff value k is shown underneath. A dashed line shows the station waiting k slots
for its next attempt.

At T=0 we assume the transmitting station finishes, and all the Ai transmit and collide. At T=1, then, each
of the Ai has discovered the collision; each chooses a random k<2. Let us assume that A1 chooses k=1, A2
chooses k=1, A3 chooses k=0, A4 chooses k=0, and A5 chooses k=1.
Those stations choosing k=0 will retransmit immediately, at T=1. This means A3 and A4 collide again, and
at T=2 they now choose random k<4. We will Assume A3 chooses k=3 and A4 chooses k=0; A3 will try
again at T=2+3=5 while A4 will try again at T=2, that is, now.
At T=2, we now have the original A1, A2, and A5 transmitting for the second time, while A4 trying again
for the third time. They collide. Let us suppose A1 chooses k=2, A2 chooses k=1, A5 chooses k=3, and A4
chooses k=6 (A4 is choosing k<8 at random). Their scheduled transmission attempt times are now A1 at
T=3+2=5, A2 at T=4, A5 at T=6, and A4 at T=9.
At T=3, nobody attempts to transmit. But at T=4, A2 is the only station to transmit, and so successfully
seizes the channel. By the time T=5 rolls around, A1 and A3 will check the channel, that is, listen first, and
wait for A2 to finish. At T=9, A4 will check the channel again, and also begin waiting for A2 to finish.
A maximum of 1024 hosts is allowed on an Ethernet. This number apparently comes from the maximum
range for the backoff time as 0 k < 1024. If there are 1024 hosts simultaneously trying to send, then,
once the backoff range has reached k<1024 (N=10), we have a good chance that one station will succeed in
seizing the channel, that is; the minimum value of all the random ks chosen will be unique.
This backoff algorithm is not fair, in the sense that the longer a station has been waiting to send, the lower
its priority sinks. Newly transmitting stations with N=0 need not delay at all. The Ethernet capture effect,
below, illustrates this unfairness.

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2.1.5 Capture effect

The capture effect is a scenario illustrating the potential lack of fairness in the exponential backoff algorithm.
The unswitched Ethernet must be fully busy, in that each of two senders always has a packet ready to
transmit.
Let A and B be two such busy nodes, simultaneously starting to transmit their first packets. They collide.
Suppose A wins, and sends. When A is finished, B tries to transmit again. But A has a second packet, and
so A tries too. A chooses a backoff k<2 (that is, between 0 and 1 inclusive), but since B is on its second
attempt it must choose k<4. This means A is favored to win. Suppose it does.
After that transmission is finished, A and B try yet again: A on its first attempt for its third packet, and B
on its third attempt for its first packet. Now A again chooses k<2 but B must choose k<8; this time A is
much more likely to win. Each time B fails to win a given backoff, its probability of winning the next one
is reduced by about 1/2. It is quite possible, and does occur in practice, for B to lose all the backoffs until
it reaches the maximum of N=16 attempts; once it has lost the first three or four this is in fact quite likely.
At this point B simply discards the packet and goes on to the next one with N reset to 1 and k chosen from
{0,1}.
The capture effect can be fixed with appropriate modification of the backoff algorithm; the Binary Logarithmic Arbitration Method (BLAM) was proposed in [MM94]. The BLAM algorithm was considered for the
then-nascent 100 Mbps Fast Ethernet standard. But in the end a hardware strategy won out: Fast Ethernet
supports full-duplex mode which is collision-free (see 2.2 100 Mbps (Fast) Ethernet, below). While
full-duplex mode is not required for using Fast Ethernet, it was assumed that any sites concerned enough
about performance to be worried about the capture effect would opt for full-duplex.

2.1.6 Hubs and topology

Ethernet hubs (multiport repeaters) change the topology, but not the fundamental constraints. Hubs allow
much more branching; typically, each station in the office now has its own link to the wiring closet. Loops
are still forbidden. Before inexpensive switches were widely available, 10BASE-T (twisted pair Ethernet)
used hubs heavily; with twisted pair, a device can only connect to the endpoint of the wire. Thus, typically,
each host is connected directly to a hub. The maximum diameter of an Ethernet consisting of multiple
segments, joined by hubs, is constrained by the round-trip-time, and the need to detect collisions before the
sender has completed sending, as before. However, twisted-pair links are required to be much shorter, about
100 meters.

2.1.7 Errors
Packets can have bits flipped or garbled by electrical noise on the cable; estimates of the frequency with
which this occurs range from 1 in 104 to 1 in 106 . Bit errors are not uniformly likely; when they occur,
they are likely to occur in bursts. Packets can also be lost in hubs, although this appears less likely. Packets
can be lost due to collisions only if the sending host makes 16 unsuccessful transmission attempts and gives
up. Ethernet packets contain a 32-bit CRC error-detecting code (see 5.4.1 Cyclical Redundancy Check:
CRC) to detect bit errors. Packets can also be misaddressed by the sending host, or, most likely of all, they
can arrive at the receiving host at a point when the receiver has no free buffers and thus be dropped by a
higher-layer protocol.

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2.1.8 CSMA persistence

A carrier-sense/multiple-access transmission strategy is said to be nonpersistent if, when the line is busy,
the sender waits a randomly selected time. A strategy is p-persistent if, after waiting for the line to clear,
the sender sends with probability p1. Ethernet uses 1-persistence. A consequence of 1-persistence is that,
if more than one station is waiting for line to clear, then when the line does clear a collision is certain.
However, Ethernet then gracefully handles the resulting collision via the usual exponential backoff. If N
stations are waiting to transmit, the time required for one station to win the backoff is linear in N.
When we consider the Wi-Fi collision-handling mechanisms in 3.3 Wi-Fi, we will see that collisions cannot
be handled quite as cheaply: for one thing, there is no way to detect a collision in progress, so the entire
packet-transmission time is wasted. In the Wi-Fi case, p-persistence is used with p<1.
An Ethernet broadcast storm was said to occur when there were too many transmission attempts, and most
of the available bandwidth was tied up in collisions. A properly functioning classic Ethernet had an effective
bandwidth of as much as 50-80% of the nominal 10Mbps capacity, but attempts to transmit more than this
typically resulted in successfully transmitting a good deal less.

2.1.9 Analysis of Classic Ethernet

How much time does Ethernet waste on collisions? A paradoxical attribute of Ethernet is that raising the
transmission-attempt rate on a busy segment can reduce the actual throughput. More transmission attempts
can lead to longer contention intervals between packets, as senders use the transmission backoff algorithm
to attempt to acquire the channel. What effective throughput can be achieved?
It is convenient to refer to the time between packet transmissions as the contention interval even if there is
no actual contention, even if the network is idle. Thus, a timeline for Ethernet always consists of alternating
packet transmissions and contention intervals:

As a first look at contention intervals, assume that there are N stations waiting to transmit at the start of the
interval. It turns out that, if all follow the exponential backoff algorithm, we can expect O(N) slot times
before one station successfully acquires the channel; thus, Ethernets are happiest when N is small and there
are only a few stations simultaneously transmitting. However, multiple stations are not necessarily a severe
problem. Often the number of slot times needed turns out to be about N/2, and slot times are short. If N=20,
then N/2 is 10 slot times, or 640 bytes. However, one packet time might be 1500 bytes. If packet intervals
are 1500 bytes and contention intervals are 640 byes, this gives an overall throughput of 1500/(640+1500)
= 70% of capacity. In practice, this seems to be a reasonable upper limit for the throughput of classic
shared-media Ethernet.

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2.1.9.1 The ALOHA models

Another approach to analyzing the Ethernet contention interval is by using the ALOHA model that was a
precursor to Ethernet. In the ALOHA model, stations transmit packets without listening first for a quiet line
or monitoring the transmission for collisions (this models the situation of several ground stations transmitting
to a satellite; the ground stations are presumed unable to see one another). Similarly, during the Ethernet
contention interval, stations transmit one-slot packets under what are effectively the same conditions (we
return to this below).
The ALOHA model yields very similar throughput values to the O(N) model of the previous section. We
make, however, a rather artificial assumption: that there are a very large number of active senders, each
transmitting at a very low rate.
To model the success rate of ALOHA, assume all the packets are the same size and let T be the time to send
one (fixed-size) packet; T represents the Aloha slot time. We will find the transmission rate that optimizes
throughput.
The core assumption of this model is that that a large number N of hosts are transmitting, each at a relatively
low rate of s packets/slot. Denote by G the average number of transmission attempts per slot; we then have
G = Ns. We will derive an expression for S, the average rate of successful transmissions per slot, in terms of
G.
If two packets overlap during transmissions, both are lost. Thus, a successful transmission requires everyone
else quiet for an interval of 2T: if a sender succeeds in the interval from t to t+T, then no other node can have
tried to begin transmission in the interval tT to t+T. The probability of one station transmitting during an
interval of time T is G = Ns; the probability of the remaining N1 stations all quiet for an interval of 2T is
(1s)2(N1) . The probability of a successful transmission is thus
S = Ns*(1s)2(N1)
= G(1G/N)2N
Math Warning
Finding the limit of G(1G/N)2N and finding the maximum of Ge-2G realistically requires a little background in calculus. However, these are not central to applying the model.
As N gets large, the second line approaches Ge-2G . The function S = G e-2G has a maximum at G=1/2,
S=1/2e. The rate G=1/2 means that, on average, a transmission is attempted every other slot; this yields the
maximum successful-transmission throughput of 1/2e. In other words, at this maximum attempt rate G=1/2,
we expect about 2e1 slot times worth of contention between successful transmissions. What happens to
the remaining GS unsuccessful attempts is not addressed by this model; presumably some higher-level
mechanism (eg backoff) leads to retransmissions.
A given throughput S<1/2e may be achieved at either of two values for G; that is, a given success rate may
be due to a comparable attempt rate or else due to a very high attempt rate with a similarly high failure rate.

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2.1.9.2 ALOHA and Ethernet

The relevance of the Aloha model to Ethernet is that during one Ethernet slot time there is no way to detect
collisions (they havent reached the sender yet!) and so the Ethernet contention phase resembles ALOHA
with an Aloha slot time T of 51.2 microseconds. Once an Ethernet sender succeeds, however, it continues
with a full packet transmission, which is presumably many times longer than T.
The average length of the contention interval, at the maximum throughput calculated above, is 2e1 slot
times (from ALOHA); recall that our model here supposed many senders sending at very low individual
rates. This is the minimum contention interval; with lower loads the contention interval is longer due to
greater idle times and with higher loads the contention interval is longer due to more collisions.
Finally, let P be the time to send an entire packet in units of T; ie the average packet size in units of T. P is
thus the length of the packet phase in the diagram above. The contention phase has length 2e1, so the
total time to send one packet (contention+packet time) is 2e1+P. The useful fraction of this is, of course, P,
so the effective maximum throughput is P/(2e1+P).
At 10Mbps, T=51.2 microseconds is 512 bits, or 64 bytes. For P=128 bytes = 2*64, the effective bandwidth
becomes 2/(2e-1+2), or 31%. For P=512 bytes=8*64, the effective bandwidth is 8/(2e+7), or 64%. For
P=1500 bytes, the model here calculates an effective bandwidth of 80%.
These numbers are quite similar to our earlier values based on a small number of stations sending constantly.

2.2 100 Mbps (Fast) Ethernet

In all the analysis here of 10 Mbps Ethernet, what happens when the bandwidth is increased to 100 Mbps,
as is done in the so-called Fast Ethernet standard? If the network physical diameter remains the same, then
the round-trip time will be the same in microseconds but will be 10-fold larger measured in bits; this might
mean a minimum packet size of 640 bytes instead of 64 bytes. (Actually, the minimum packet size might
be somewhat smaller, partly because the jam signal doesnt have to speed up at all, and partly because
some of the numbers in the 10 Mbps delay budget above were larger than necessary, but it would still be
large enough that a substantial amount of bandwidth would be consumed by padding.) The designers of Fast
Ethernet felt this was impractical.
However, Fast Ethernet was developed at a time (~1995) when reliable switches (below) were widely available, and longer networks could be formed by chaining together shorter ones with switches. So instead
of increasing the minimum packet size, the decision was made to ensure collision detectability by reducing
the network diameter instead. The network diameter chosen was a little over 400 meters, with reductions to
account for the presence of hubs. At 2.3 meters/bit, 400 meters is 174 bits, for a round-trip of 350 bits.
This 400-meter number, however, may be misleading: by far the most popular Fast Ethernet standard is
100BASE-TX which uses twisted-pair copper wire (so-called Category 5, or better), and in which any individual cable segment is limited to 100 meters. The maximum 100BASE-TX network diameter allowing
for hubs is just over 200 meters. The 400-meter distance does apply to optical-fiber-based 100BASE-FX
in half-duplex mode, but this is not common.
The 100BASE-TX network-diameter limit of 200 meters might seem small; it amounts in many cases to
a single hub with multiple 100-meter cable segments radiating from it. In practice, however, such star
configurations can easily be joined with switches. As we will see below in 2.4 Ethernet Switches, switches
partition an Ethernet into separate collision domains; the network-diameter rules apply to each domain
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separately but not to the aggregated whole. In a fully switched (that is, no hubs) 100BASE-TX LAN, each
collision domain is simply a single twisted-pair link, subject to the 100-meter maximum length.
Fast Ethernet also introduced the concept of full-duplex Ethernet: two twisted pairs could be used, one
for each direction. Full-duplex Ethernet is limited to paths not involving hubs, that is, to single station-tostation links, where a station is either a host or a switch. Because such a link has only two potential senders,
and each sender has its own transmit line, full-duplex Ethernet is collision-free.
Fast Ethernet uses 4B/5B encoding, covered in 4.1.4 4B/5B.
Fast Ethernet 100BASE-TX does not particularly support links between buildings, due to the networkdiameter limitation. However, fiber-optic point-to-point links are quite effective here, provided full-duplex
is used to avoid collisions. We mentioned above that the coax-based 100BASE-FX standard allowed a
maximum half-duplex run of 400 meters, but 100BASE-FX is much more likely to use full duplex, where
the maximum cable length rises to 2,000 meters.

2.3 Gigabit Ethernet

If we continue to maintain the same slot time but raise the transmission rate to 1000 Mbps, the network
diameter would now be 20-40 meters. Instead of that, Gigabit Ethernet moved to a 4096-bit (512-byte) slot
time, at least for the twisted-pair versions. Short frames need to be padded, but this padding is done by the
hardware. Gigabit Ethernet 1000Base-T uses so-called PAM-5 encoding, below, which supports a special
pad pattern (or symbol) that cannot appear in the data. The hardware pads the frame with these special
patterns, and the receiver can thus infer the unpadded length as set by the host operating system.
However, the Gigabit Ethernet slot time is largely irrelevant, as full-duplex (bidirectional) operation is almost
always supported. Combined with the restriction that each length of cable is a station-to-station link (that is,
hubs are no longer allowed), this means that collisions simply do not occur and the network diameter is no
longer a concern.
There are actually multiple Gigabit Ethernet standards (as there are for Fast Ethernet). The different standards apply to different cabling situations. There are full-duplex optical-fiber formulations good for many
miles (eg 1000Base-LX10), and even a version with a 25-meter maximum cable length (1000Base-CX),
which would in theory make the original 512-bit slot practical.
The most common gigabit Ethernet over copper wire is 1000BASE-T (sometimes incorrectly referred to
as 100BASE-TX. While there exists a TX, it requires Category 6 cable and is thus seldom used; many
devices labeled TX are in fact 1000BASE-T). For 1000BASE-T, all four twisted pairs in the cable are used.
Each pair transmits at 250 Mbps, and each pair is bidirectional, thus supporting full-duplex communication.
Bidirectional communication on a single wire pair takes some careful echo cancellation at each end, using
a circuit known as a hybrid that in effect allows detection of the incoming signal by filtering out the
outbound signal.
On any one cable pair, there are five signaling levels. These are used to transmit two-bit symbols
(4.1.4 4B/5B) at a rate of 125 symbols/sec, for a data rate of 250 bits/sec. Two-bit symbols in theory only require four signaling levels; the fifth symbol allows for some redundancy which is used for error
detection and correction, for avoiding long runs of identical symbols, and for supporting a special pad symbol, as mentioned above. The encoding is known as 5-level pulse-amplitude modulation, or PAM-5. The
target bit error rate (BER) for 1000BASE-T is 10-10 , meaning that the packet error rate is less than 1 in 106 .

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In developing faster Ethernet speeds, economics plays at least as important a role as technology. As new
speeds reach the market, the earliest adopters often must take pains to buy cards, switches and cable known
to work together; this in effect amounts to installing a proprietary LAN. The real benefit of Ethernet,
however, is arguably that it is standardized, at least eventually, and thus a site can mix and match its cards
and devices. Having a given Ethernet standard support existing cable is even more important economically;
the costs of replacing cable often dwarf the costs of the electronics.

2.4 Ethernet Switches

Switches join separate physical Ethernets (or Ethernets and token rings). A switch has two or more Ethernet
interfaces; when a packet is received on one interface it is retransmitted on one or more other interfaces.
Only valid packets are forwarded; collisions are not propagated. The term collision domain is sometimes
used to describe the region of an Ethernet in between switches; a given collision propagates only within its
collision domain. All the collision-detection rules, including the rules for maximum network diameter, apply
only to collision domains, and not to the larger virtual Ethernets created by stringing collision domains
together with switches.
As we shall see below, a switched Ethernet offers much more resistance to eavesdropping than a nonswitched (eg hub-based) Ethernet.
Like simpler unswitched Ethernets, the topology for a switched Ethernet is in principle required to be loopfree, although in practice, most switches support the spanning-tree loop-detection protocol and algorithm,
below, which automatically prunes the network topology to make it loop-free.
And while a switch does not propagate collisions, it must maintain a queue for each outbound interface in
case it needs to forward a packet at a moment when the interface is busy; on occasion packets are lost when
this queue overflows.
Ethernet switches use datagram forwarding as described in 1.4 Datagram Forwarding. They start out with
empty forwarding tables, and build them through a learning process. If a switch does not have an entry for
a particular destination, it will fall back to broadcast: it will forward the packet out every interface other
than the one on which the packet arrived. The availability of fallback-to-broadcast is what makes it possible
for Ethernet switches to learn their forwarding tables without any switch-to-switch communication.
A switch learns address locations as follows: for each interface, the switch maintains a table of physical
addresses that have appeared as source addresses in packets arriving via that interface. The switch thus
knows that to reach these addresses, if one of them later shows up as a destination address, the packet needs
to be sent only via that interface. Specifically, when a packet arrives on interface I with source address S
and destination unicast address D, the switch enters S,I into its forwarding table.
To actually deliver the packet, the switch also looks up D in the forwarding table. If there is an entry D,J
with J=I that is, D is known to be reached via interface J then the switch forwards the packet out interface
J. If J=I, that is, the packet has arrived on the same interfaces by which the destination is reached, then the
packet does not get forwarded at all; it presumably arrived at interface I only because that interface was
connected to a shared Ethernet segment that also either contained D or contained another switch that would
bring the packet closer to D. If there is no entry for D, the switch must forward the packet out all interfaces
J with J=I; this represents the fallback to broadcast. As time goes on, this fallback to broadcast is needed
less and less often.

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If the destination address D is the broadcast address, or, for many switches, a multicast address, broadcast
is required.

In the diagram above, each switchs tables are indicated by listing near each interface the destinations known
to be reachable by that interface. The entries shown are the result of the following packets:
A sends to B; all switches learn where A is
B sends to A; this packet goes directly to A; only S3, S2 and S1 learn where B is
C sends to B; S4 does not know where B is so this packet goes to S5; S2 does know where B is so the
packet does not go to S1.
Switches do not automatically discover directly connected neighbors; S1 does not learn about A until A
transmits a packet.
Once all the switches have learned where all (or most of) the hosts are, packet routing becomes optimal. At
this point packets are never sent on links unnecessarily; a packet from A to B only travels those links that
lie along the (unique) path from A to B. (Paths must be unique because switched Ethernet networks cannot
have loops, at least not active ones. If a loop existed, then a packet sent to an unknown destination would be
forwarded around the loop endlessly.)
Switches have an additional advantage in that traffic that does not flow where it does not need to flow is
much harder to eavesdrop on. On an unswitched Ethernet, one host configured to receive all packets can
eavesdrop on all traffic. Early Ethernets were notorious for allowing one unscrupulous station to capture,
for instance, all passwords in use on the network. On a fully switched Ethernet, a host physically only sees
the traffic actually addressed to it; other traffic remains inaccessible.
Typical switches have room for table with 104 - 106 entries, though maxing out at 105 entries may be more
common; this is usually enough to learn about all hosts in even a relatively large organization. A switched
Ethernet can fail when total traffic becomes excessive, but excessive total traffic would drown any network
(although other network mechanisms might support higher bandwidth). The main limitations specific to
switching are the requirement that the topology must be loop-free (thus disallowing duplicate paths which
might otherwise provide redundancy), and that all broadcast traffic must always be forwarded everywhere.
As a switched Ethernet grows, broadcast traffic comprises a larger and larger percentage of the total traffic,
and the organization must at some point move to a routing architecture (eg as in 7.6 IP Subnets).
One of the differences between an inexpensive Ethernet switch and a pricier one is the degree of internal
parallelism it can support. If three packets arrive simultaneously on ports 1, 2 and 3, and are destined for
respective ports 4, 5 and 6, can the switch actually transmit the packets simultaneously? A simple switch
likely has a single CPU and a single memory bus, both of which can introduce transmission bottlenecks.
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For commodity five-port switches, at most two simultaneous transmissions can occur; such switches can
generally handle that degree of parallelism. It becomes harder as the number of ports increases, but at some
point the need to support full parallel operation can be questioned; in many settings the majority of traffic
involves one or two server or router ports. If a high degree of parallelism is in fact required, there are various
architectures known as switch fabrics that can be used; these typically involve multiple simple processor
elements.

2.5 Spanning Tree Algorithm

In theory, if you form a loop with Ethernet switches, any packet with destination not already present in the
forwarding tables will circulate endlessly; naive switches will actually do this.
In practice, however, loops allow a form of redundancy if one link breaks there is still 100% connectivity
and so are desirable. As a result, Ethernet switches have incorporated a switch-to-switch protocol to
construct a subset of the switch-connections graph that has no loops and yet allows reachability of every
host, known as a spanning tree. The switch-connections graph is the graph with nodes consisting of both
switches and of the unswitched Ethernet segments and isolated individual hosts connected to the switches.
Multi-host Ethernet segments are most often created via Ethernet hubs (repeaters). Edges in the graph
represent switch-segment and switch-switch connections; each edge attaches to its switch via a particular,
numbered interface. The goal is to disable redundant (cyclical) paths while remaining able to deliver to any
segment. The algorithm is due to Radia Perlman, [RP85].
Once the spanning tree is built, all packets are sent only via edges in the tree, which, as a tree, has no loops.
Switch ports (that is, edges) that are not part of the tree are not used at all, even if they would represent
the most efficient path for that particular destination. If a given segment connects to two switches that
both connect to the root node, the switch with the shorter path to the root is used, if possible; in the event
of ties, the switch with the smaller ID is used. The simplest measure of path cost is the number of hops,
though current implementations generally use a cost factor inversely proportional to the bandwidth (so larger
bandwidth has lower cost). Some switches permit other configuration here. The process is dynamic, so if
an outage occurs then the spanning tree is recomputed. If the outage should partition the network into two
pieces, both pieces will build spanning trees.
All switches send out regular messages on all interfaces called bridge protocol data units, or BPDUs (or
Hello messages). These are sent to the Ethernet multicast address 01:80:c2:00:00:00, from the Ethernet
physical address of the interface. (Note that Ethernet switches do not otherwise need a unique physical
address for each interface.) The BPDUs contain
The switch ID
the ID of the node the switch believes is the root
the path cost to that root
These messages are recognized by switches and are not forwarded naively. Bridges process each message,
looking for
a switch with a lower ID (thus becoming the new root)
a shorter path to the existing root
an equal-length path to the existing root, but via a switch or port with a lower ID (the tie-breaker rule)

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When a switch sees a new root candidate, it sends BPDUs on all interfaces, indicating the distance. The
switch includes the interface leading towards the root.
Once this process is complete, each switch knows
its own path to the root
which of its ports any further-out switches will be using to reach the root
for each port, its directly connected neighboring switches
Now the switch can prune some (or all!) of its interfaces. It disables all interfaces that are not enabled by
the following rules:
1. It enables the port via which it reaches the root
2. It enables any of its ports that further-out switches use to reach the root
3. If a remaining port connects to a segment to which other segment-neighbor switches connect as well,
the port is enabled if the switch has the minimum cost to the root among those segment-neighbors, or,
if a tie, the smallest ID among those neighbors, or, if two ports are tied, the port with the smaller ID.
4. If a port has no directly connected switch-neighbors, it presumably connects to a host or segment, and
the port is enabled.
Rules 1 and 2 construct the spanning tree; if S3 reaches the root via S2, then Rule 1 makes sure S3s port
towards S2 is open, and Rule 2 makes sure S2s corresponding port towards S3 is open. Rule 3 ensures that
each network segment that connects to multiple switches gets a unique path to the root: if S2 and S3 are
segment-neighbors each connected to segment N, then S2 enables its port to N and S3 does not (because
2<3). The primary concern here is to create a path for any host nodes on segment N; S2 and S3 will create
their own paths via Rules 1 and 2. Rule 4 ensures that any stub segments retain connectivity; these would
include all hosts directly connected to switch ports.

2.5.1 Example 1: Switches Only

We can simplify the situation somewhat if we assume that the network is fully switched: each switch port
connects to another switch or to a (single-interface) host; that is, no repeater hubs (or coax segments!) are
in use. In this case we can dispense with Rule 3 entirely.
Any switch ports directly connected to a host can be identified because they are silent; the switch never
receives any BPDU messages on these interfaces because hosts do not send these. All these host port ends
up enabled via Rule 4. Here is our sample network, where the switch numbers (eg 5 for S5) represent their
IDs; no hosts are shown and interface numbers are omitted.

S1 has the lowest ID, and so becomes the root. S2 and S4 are directly connected, so they will enable the
interfaces by which they reach S1 (Rule 1) while S1 will enable its interfaces by which S2 and S4 reach it
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(Rule 2).
S3 has a unique lowest-cost route to S1, and so again by Rule 1 it will enable its interface to S2, while by
Rule 2 S2 will enable its interface to S3.
S5 has two choices; it hears of equal-cost paths to the root from both S2 and S4. It picks the lower-numbered
neighbor S2; the interface to S4 will never be enabled. Similarly, S4 will never enable its interface to S5.
Similarly, S6 has two choices; it selects S3.
After these links are enabled (strictly speaking it is interfaces that are enabled, not links, but in all cases here
either both interfaces of a link will be enabled or neither), the network in effect becomes:

2.5.2 Example 2: Switches and Segments

As an example involving switches that may join via unswitched Ethernet segments, consider the following
network; S1, S2 and S3, for example, are all segment-neighbors via their common segment B. As before, the
switch numbers represent their IDs. The letters in the clouds represent network segments; these clouds may
include multiple hosts. Note that switches have no way to detect these hosts; only (as above) other switches.

Eventually, all switches discover S1 is the root (because 1 is the smallest of {1,2,3,4,5,6}). S2, S3 and S4
are one (unique) hop away; S5, S6 and S7 are two hops away.

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Algorhyme
I think that I shall never see
a graph more lovely than a tree.
A tree whose crucial property
is loop-free connectivity.
A tree that must be sure to span
so packet can reach every LAN.
First, the root must be selected.
By ID, it is elected.
Least-cost paths from root are traced.
In the tree, these paths are placed.
A mesh is made by folks like me,
then bridges find a spanning tree.
Radia Perlman
For the switches one hop from the root, Rule 1 enables S2s port 1, S3s port 1, and S4s port 1. Rule 2
enables the corresponding ports on S1: ports 1, 5 and 4 respectively. Without the spanning-tree algorithm
S2 could reach S1 via port 2 as well as port 1, but port 1 has a smaller number.
S5 has two equal-cost paths to the root: S5S4S1 and S5S3S1. S3 is the switch with the
lower ID; its port 2 is enabled and S5 port 2 is enabled.
S6 and S7 reach the root through S2 and S3 respectively; we enable S6 port 1, S2 port 3, S7 port 2 and S3
port 3.
The ports still disabled at this point are S1 ports 2 and 3, S2 port 2, S4 ports 2 and 3, S5 port 1, S6 port 2
and S7 port 1.
Now we get to Rule 3, dealing with how segments (and thus their hosts) connect to the root. Applying Rule
3,
We do not enable S2 port 2, because the network (B) has a direct connection to the root, S1
We do enable S4 port 3, because S4 and S5 connect that way and S4 is closer to the root. This enables
connectivity of network D. We do not enable S5 port 1.
S6 and S7 are tied for the path-length to the root. But S6 has smaller ID, so it enables port 2. S7s
port 1 is not enabled.
Finally, Rule 4 enables S4 port 2, and thus connectivity for host J. It also enables S1 port 2; network F has
two connections to S1 and port 2 is the lower-numbered connection.
All this port-enabling is done using only the data collected during the root-discovery phase; there is no
additional negotiation. The BPDU exchanges continue, however, so as to detect any changes in the topology.
If a link is disabled, it is not used even in cases where it would be more efficient to use it. That is, traffic
from F to B is sent via B1, D, and B5; it never goes through B7. IP routing, on the other hand, uses the

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shortest path. To put it another way, all spanning-tree Ethernet traffic goes through the root node, or along
a path to or from the root node.
The traditional (IEEE 802.1D) spanning-tree protocol is relatively slow; the need to go through the treebuilding phase means that after switches are first turned on no normal traffic can be forwarded for ~30
seconds. Faster, revised protocols have been proposed to reduce this problem.
Another issue with the spanning-tree algorithm is that a rogue switch can announce an ID of 0, thus likely
becoming the new root; this leaves that switch well-positioned to eavesdrop on a considerable fraction of
the traffic. One of the goals of the Cisco Root Guard feature is to prevent this; another goal of this and
related features is to put the spanning-tree topology under some degree of administrative control. One likely
wants the root switch, for example, to be geographically at least somewhat centered.

2.6 Virtual LAN (VLAN)

What do you do when you have different people in different places who are logically tied together? For
example, for a while the Loyola University CS department was split, due to construction, between two
buildings.
One approach is to continue to keep LANs local, and use IP routing between different subnets. However,
it is often convenient (printers are one reason) to configure workgroups onto a single virtual LAN, or
VLAN. A VLAN looks like a single LAN, usually a single Ethernet LAN, in that all VLAN members will
see broadcast packets sent by other members and the VLAN will ultimately be considered to be a single IP
subnet (7.6 IP Subnets). Different VLANs are ultimately connected together, but likely only by passing
through a single, central IP router.
VLANs can be visualized and designed by using the concept of coloring. We logically assign all nodes on
the same VLAN the same color, and switches forward packets accordingly. That is, if S1 connects to red
machines R1 and R2 and blue machines B1 and B2, and R1 sends a broadcast packet, then it goes to R2 but
not to B1 or B2. Switches must, of course, be told the color of each of their ports.

In the diagram above, S1 and S3 each have both red and blue ports. The switch network S1-S4 will deliver
traffic only when the source and destination ports are the same color. Red packets can be forwarded to the
blue VLAN only by passing through the router R, entering Rs red port and leaving its blue port. R may
apply firewall rules to restrict redblue traffic.

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When the source and destination ports are on the same switch, nothing needs to be added to the packet; the
switch can keep track of the color of each of its ports. However, switch-to-switch traffic must be additionally
tagged to indicate the source. Consider, for example, switch S1 above sending packets to S3 which has nodes
R3 (red) and B3 (blue). Traffic between S1 and S3 must be tagged with the color, so that S3 will know to
what ports it may be delivered. The IEEE 802.1Q protocol is typically used for this packet-tagging; a 32-bit
color tag is inserted into the Ethernet header after the source address and before the type field. The first
16 bits of this field is 0x8100, which becomes the new Ethernet type field and which identifies the frame as
tagged.
Double-tagging is possible; this would allow an ISP to have one level of tagging and its customers to have
another level.

2.7 Epilog
Ethernet dominates the LAN layer, but is not one single LAN protocol: it comes in a variety of speeds
and flavors. Higher-speed Ethernet seems to be moving towards fragmenting into a range of physical-layer
options for different types of cable, but all based on switches and point-to-point linking; different Ethernet
types can be interconnected only with switches. Once Ethernet finally abandons physical links that are
bi-directional (half-duplex links), it will be collision-free and thus will no longer need a minimum packet
size.
Other wired networks have largely disappeared (or have been renamed Ethernet). Wireless networks,
however, are here to stay, and for the time being at least have inherited the original Ethernets collisionmanagement concerns.

2.8 Exercises
1. Simulate the contention period of five Ethernet stations that all attempt to transmit at T=0 (presumably
when some sixth station has finished transmitting), in the style of the diagram in 2.1.4 Exponential Backoff
Algorithm. Assume that time is measured in slot times, and that exactly one slot time is needed to detect a
collision (so that if two stations transmit at T=1 and collide, and one of them chooses a backoff time k=0,
then that station will transmit again at T=2). Use coin flips or some other source of randomness.
2. Suppose we have Ethernet switches S1 through S3 arranged as below. All forwarding tables are initially
empty.
S1

S2

S3

(a). If A sends to B, which switches see this packet?

(b). If B then replies to A, which switches see this packet?
(c). If C then sends to B, which switches see this packet?
(d). If C then sends to D, which switches see this packet?

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3. Suppose we have the Ethernet switches S1 through S4 arranged as below. All forwarding tables are
empty; each switch uses the learning algorithm of 2.4 Ethernet Switches.
B
S4
A

S1

S2

S3

Now suppose the following packet transmissions take place:

A sends to B
B sends to A
C sends to B
D sends to A
For each switch, list what source addresses (eg A,B,C,D) it has seen (and thus what nodes it has learned the
location of).
4. In the switched-Ethernet network below, find two packet transmissions so that, when a third transmission
AD occurs, the packet is delivered to B (that is, it is forwarded out all ports of S2), but is not similarly
delivered to C. All forwarding tables are initially empty, and each switch uses the learning algorithm of
2.4 Ethernet Switches.

S1

S2

S3

Hint: Destination D must be in S3s forwarding table, but must not be in S2s.
5. Given the Ethernet network with learning switches below, with (disjoint) unspecified parts represented
by ?, explain why it is impossible for a packet sent from A to B to be forwarded by S1 only to S2, but to be
forwarded by S2 out all of S2s other ports.

?
|
S1

?
|
S2

6. In the diagram of 2.4 Ethernet Switches, suppose node D is connected to S5, and, with the tables as
shown below the diagram, D sends to B.

(a). Which switches will see this packet, and thus learn about D?
(b). Which of the switches in part (a) do not already know where B is and will use fallback-to-broadcase
(ie, will forward the packet out all non-arrival interfaces)?

7. Suppose two Ethernet switches are connected in a loop as follows; S1 and S2 have their interfaces 1 and
2 labeled. These switches do not use the spanning-tree algorithm.
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Suppose A attempts to send a packet to destination B, which is unknown. S1 will therefore forward the
packet out interfaces 1 and 2. What happens then? How long will As packet circulate?
8. The following network is like that of 2.5.1 Example 1: Switches Only, except that the switches are
numbered differently. Again, the ID of switch Sn is n, so S1 will be the root. Which links end up pruned
by the spanning-tree algorithm, and why?
S1

S4

S6

S3

S5

S2

9. Suppose you want to develop a new protocol so that Ethernet switches participating in a VLAN all keep
track of the VLAN color associated with every destination. Assume that each switch knows which of its
ports (interfaces) connect to other switches and which may connect to hosts, and in the latter case knows the
color assigned to that port.

(a). Suggest a way by which switches might propagate this destination-color information to other switches.
(b). What happens if a port formerly reserved for connection to another switch is now used for a host?

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3 OTHER LANS

In the wired era, one could get along quite well with nothing but Ethernet and the occasional long-haul pointto-point link joining different sites. However, there are important alternatives out there. Some, like token
ring, are mostly of historical importance; others, like virtual circuits, are of great conceptual importance but
so far of only modest day-to-day significance.
And then there is wireless. It would be difficult to imagine contemporary laptop networking, let alone mobile
devices, without it. In both homes and offices, Wi-Fi connectivity is the norm. A return to being tethered by
wires is almost unthinkable.

3.1 Virtual Private Network

Suppose you want to connect to your workplace network from home. Your workplace, however, has a
security policy that does not allow outside IP addresses to access essential internal resources. How do you
proceed, without leasing a dedicated telecommunications line to your workplace?
A virtual private network, or VPN, provides a solution; it supports creation of virtual links that join farflung nodes via the Internet. Your home computer creates an ordinary Internet connection (TCP or UDP)
to a workplace VPN server (IP-layer packet encapsulation can also be used; see 7.11 Mobile IP). Each
end of the connection is associated with a software-created virtual network interface; each of the two
virtual interfaces is assigned an IP address. When a packet is to be sent along the virtual link, it is actually
encapsulated and sent along the original Internet connection to the VPN server, wending its way through the
commodity Internet; this process is called tunneling. To all intents and purposes, the virtual link behaves
like any other physical link.
Tunneled packets are usually encrypted as well as encapsulated, though that is a separate issue. One example
of a tunneling protocol is to treat a TCP home-workplace connection as a serial line and send packets over
it back-to-back, using PPP with HDLC; see 4.1.5.1 HDLC and RFC 1661.
At the workplace side, the virtual network interface in the VPN server is attached to a router or switch; at
the home users end, the virtual network interface can now be assigned an internal workplace IP address.
The home computer is now, for all intents and purposes, part of the internal workplace network.
In the diagram below, the users regular Internet connection is via hardware interface eth0. A connection is
established to Site As VPN server; a virtual interface tun0 is created on the users machine which appears
to be a direct link to the VPN server. The tun0 interface is assigned a Site-A IP address. Packets sent via
the tun0 interface in fact travel over the original connection via eth0 and the Internet.

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After the VPN is set up, the home hosts tun0 interface appears to be locally connected to Site A, and thus
the home host is allowed to connect to the private area within Site A. The home hosts forwarding table will
be configured so that traffic to Site As private addresses is routed via interface tun0.
VPNs are also commonly used to connect entire remote offices to headquarters. In this case the remote-office
end of the tunnel will be at that offices local router, and the tunnel will carry traffic for all the workstations
in the remote office.
To improve security, it is common for the residential (or remote-office) end of the VPN connection to use
the VPN connection as the default route for all traffic except that needed to maintain the VPN itself. This
may require a so-called host-specific forwarding-table entry at the residential end to allow the packets that
carry the VPN tunnel traffic to be routed correctly via eth0. This routing strategy means that potential
intruders cannot access the residential host and thus the workplace internal network through the original
residential Internet access. A consequence is that if the home worker downloads a large file from a nonworkplace site, it will travel first to the workplace, then back out to the Internet via the VPN connection, and
finally arrive at the home.

3.2 Carrier Ethernet

Carrier Ethernet is a leased-line point-to-point link between two sites, where the subscriber interface at each
end of the line looks like Ethernet (in some flavor). The physical path in between sites, however, need not
have anything to do with Ethernet; it may be implemented however the carrier wishes. In particular, it will
be (or at least appear to be) full-duplex, it will be collision-free, and its length may far exceed the maximum
permitted by any IEEE Ethernet standard.
Bandwidth can be purchased in whatever increments the carrier has implemented. The point of carrier
Ethernet is to provide a layer of abstraction between the customers, who need only install a commodity
Ethernet interface, and the provider, who can upgrade the link implementation at will without requiring
change at the customer end.
In a sense, carrier Ethernet is similar to the widespread practice of provisioning residential DSL and cable
routers with an Ethernet interface for customer interconnection; again, the actual link technologies may not
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look anything like Ethernet, but the interface will.

A carrier Ethernet connection looks like a virtual VPN link, but runs on top of the providers internal network
rather than the Internet at large. Carrier Ethernet connections often provide the primary Internet connectivity
for one endpoint, unlike Internet VPNs which assume both endpoints already have full Internet connectivity.

3.3 Wi-Fi
Wi-Fi is a trademark denoting any of several IEEE wireless-networking protocols in the 802.11 family,
specifically 802.11a, 802.11b, 802.11g, 802.11n, and 802.11ac. Like classic Ethernet, Wi-Fi must deal
with collisions; unlike Ethernet, however, Wi-Fi is unable to detect collisions in progress, complicating the
backoff and retransmission algorithms. Wi-Fi is designed to interoperate freely with Ethernet at the logical
LAN layer; that is, Ethernet and Wi-Fi traffic can be freely switched from the wired side to the wireless side.
Band Width
To radio engineers, band width means the frequency range used by a signal, not data rate; in keeping
with this we will in this section and 3.4 WiMAX use the term data rate instead of bandwidth. We
will use the terms channel width or width of the frequency band for the frequency range. All else
being equal, the data rate achievable with a radio signal is proportional to the channel width.
Generally, Wi-Fi uses the 2.4 GHz ISM (Industrial, Scientific and Medical) band used also by microwave
ovens, though 802.11a uses a 5 GHz band, 802.11n supports that as an option and the new 802.11ac has
returned to using 5 GHz exclusively. The 5 GHz band has reduced ability to penetrate walls, often resulting
in a lower effective range. Wi-Fi radio spectrum is usually unlicensed, meaning that no special permission
is needed to transmit but also that others may be trying to use the same frequency band simultaneously; the
availability of unlicensed channels in the 5 GHz band continues to evolve.
The table below summarizes the different Wi-Fi versions. All bit rates assume a single spatial stream;
channel widths are nominal.
IEEE name
802.11a
802.11b
802.11g
802.11n
802.11ac

maximum bit rate

54 Mbps
11 Mbps
54 Mbps
65-150 Mbps
78-867 Mbps

frequency
5 GHz
2.4 GHz
2.4 GHz
2.4/5 GHz
5 GHz

channel width
20 MHz
20 MHz
20 MHz
20-40 MHz
20-160 MHz

The maximum bit rate is seldom achieved in practice. The effective bit rate must take into account, at a
minimum, the time spent in the collision-handling mechanism. More significantly, all the Wi-Fi variants
above use dynamic rate scaling, below; the bit rate is reduced up to tenfold (or more) in environments with
higher error rates, which can be due to distance, obstructions, competing transmissions or radio noise. All
this means that, as a practical matter, getting 150 Mbps out of 802.11n requires optimum circumstances; in
particular, no competing senders and unimpeded line-of-sight transmission. 802.11n lower-end performance
can be as little as 10 Mbps, though 40-100 Mbps (for a 40 MHz channel) may be more typical.
The 2.4 GHz ISM band is divided by international agreement into up to 14 officially designated channels,
each about 5 MHz wide, though in the United States use may be limited to the first 11 channels. The 5 GHz

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band is similarly divided into 5 MHz channels. One Wi-Fi sender, however, needs several of these official
channels; the typical 2.4 GHz 802.11g transmitter uses an actual frequency range of up to 22 MHz, or up
to five channels. As a result, to avoid signal overlap Wi-Fi use in the 2.4 GHz band is often restricted to
official channels 1, 6 and 11. The end result is that unrelated Wi-Fi transmitters can and do interact with and
interfere with each other.
The United States requires users of the 5 GHz band to avoid interfering with weather and military applications in the same frequency range. Once that is implemented, however, there are more 5 MHz channels at
this frequency than in the 2.4 GHz ISM band, which is one of the reasons 802.11ac can run faster (below).
Wi-Fi designers can improve speed through a variety of techniques, including
improved radio modulation techniques
improved error-correcting codes
smaller guard intervals between symbols
increasing the channel width
allowing multiple spatial streams via multiple antennas
The first two in this list seem by now to be largely tapped out; the third reduces the range but may increase
the data rate by 11%.
The largest speed increases are obtained by increasing the number of 5 MHz channels used. For example,
the 65 Mbps bit rate above for 802.11n is for a nominal frequency range of 20 MHz, comparable to that
of 802.11g. However, in areas with minimal competition from other signals, 802.11n supports using a 40
MHz frequency band; the bit rate then goes up to 135 Mbps (150 Mbps with a smaller guard interval). This
amounts to using two of the three available 2.4 GHz Wi-Fi bands. Similarly, the wide range in 802.11ac
bit rates reflects support for using channel widths ranging from 20 MHz up to 160 MHz (32 5-MHz official
channels).
For all the categories in the table above, additional bits are used for error-correcting codes. For 802.11g
operating at 54 Mbps, for example, the actual raw bit rate is (4/3)54 = 72 Mbps, sent in symbols consisting
of six bits as a unit.

3.3.1 Multiple Spatial Streams

The latest innovation in improving Wi-Fi data rates is to support multiple spatial streams, through an antenna
technique known as multiple-input-multiple output, or MIMO. To use N streams, both sender and receiver
must have N antennas; all the antennas use the same frequency channels but each transmitter antenna sends
a different data stream. While the antennas are each more-or-less omnidirectional, careful signal analysis
at the receiver can in principle recover each data stream. In practice, overall data-rate improvement over a
single antenna can be considerably less than N.
The 802.11n standard allows for up to four spatial streams, for a theoretical maximum bit rate of 600 Mbps.
802.11ac allows for up to eight spatial streams, for an even-more-theoretical maximum of close to 7 Gbps.
MIMO support is sometimes described with an ABC notation, eg 332, where A and B are the number
of transmitting and receiving antennas and C min(A,B) is the number of spatial streams.

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3.3.2 Wi-Fi and Collisions

We looked extensively at the 10 Mbps Ethernet collision-handling mechanisms in 2.1 10-Mbps classic
Ethernet, only to conclude that with switches and full-duplex links, Ethernet collisions are rapidly becoming
a thing of the past. Wi-Fi, however, has brought collisions back from obscurity. While there is a largelycollision-free mode for Wi-Fi operation (3.3.7 Wi-Fi Polling Mode), it is not commonly used, and collision
management has a significant impact on ordinary Wi-Fi performance.
Wi-Fi, like almost any other wireless protocol, simply cannot detect collisions in progress. This has to do
with the relative signal strength of the remote signal at the local transmitter. Along a wire-based Ethernet
the remote signal might be as weak as 1/100 of the transmitted signal but that 1% received signal is still
detectable during transmission. However, with radio the remote signal might easily be as little as 1/100,000
of the transmitted signal, and it is simply overwhelmed during transmission. Thus, while the Ethernet
algorithm was known as CSMA/CD, where CD stood for Collision Detection, Wi-Fi uses CSMA/CA, where
CA stands for Collision Avoidance.
The basic collision-avoidance algorithm is as follows. There are three parameters applicable here (values
are for 802.11b/g in the 2.4 GHz band); the value we call IFS is more formally known as DIFS (D for
distributed; see 3.3.7 Wi-Fi Polling Mode).
slot time: 20 sec
IFS, the normal InterFrame Spacing: 50 sec
SIFS, the short IFS: 10 sec
A sender wanting to send a new data packet waits the IFS time after first sensing the medium to see if it is
idle. If no other traffic is seen in this interval, the station may then transmit immediately. However, if other
traffic is sensed, the sender waits for the other traffic to finish and then for one IFS time after. If the station
had to wait for other traffic, even briefly, it must then do an exponential backoff even for its first transmission
attempt; other stations, after all, are likely also waiting, and avoiding an initial collision is strongly preferred.
The initial backoff is to choose a random k<25 = 32 (recall that classic Ethernet in effect chooses an initial
backoff of k<20 = 1; ie k=0). The prospective sender then waits k slot times. While waiting, the sender
continues to monitor for other traffic; if any other transmission is detected, then the sender suspends the
backoff-wait clock. The clock resumes when the other transmission has completed and one followup idle
interval of length IFS has elapsed. Under this rule, the backoff-wait clock can expire only when the clock is
running and therefore the medium is idle, at which point the transmission begins.
If there is a collision, the station retries, after doubling the backoff range to 64, then 128, 256, 512, 1024
and again 1024. If these seven attempts all fail, the packet is discarded and the sender starts over.
In one slot time, radio signals move 6,000 meters; the Wi-Fi slot time unlike that for Ethernet has nothing
to do with the physical diameter of the network. As with Ethernet, though, the Wi-Fi slot time represents
the fundamental unit for backoff intervals.
Finally, we note that, unlike Ethernet collisions, Wi-Fi collisions are a local phenomenon: if A and B
transmit simultaneously, a collision occurs at node C only if the signals of A and B are both strong enough
at C to interfere with one another. It is possible that a collision occurs at station C midway between A and
B, but not at station D that is close to A. We return to this in 3.3.2.2 Hidden-Node Problem.
Because Wi-Fi cannot detect collisions directly, the protocol adds link-layer ACK packets (unrelated to
the later TCP ACK), at least for unicast transmission. Once a station has received a packet addressed to it,
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it waits for time SIFS and sends the ACK; at the instant when the end of the SIFS interval is reached, the
receiver will be the only station authorized to send. Any other stations waiting the longer IFS period will see
the ACK before the IFS time has elapsed and will thus not interfere with the ACK; similarly, any stations
with a running backoff-wait clock will continue to have that clock suspended.
3.3.2.1 Wi-Fi RTS/CTS
Wi-Fi stations optionally also use a request-to-send/clear-to-send (RTS/CTS) protocol. Usually this is
used only for larger packets; often, the RTS/CTS threshold (the size of the largest packet not sent using RTS/CTS) is set (as part of the Access Point configuration) to be the maximum packet size, effectively
disabling this feature. The idea here is that a large packet that is involved in a collision represents a significant waste of potential throughput; for large packets, we should ask first.
The RTS packet which is small is sent through the normal procedure outlined above; this packet includes
the identity of the destination and the size of the data packet the station desires to transmit. The destination
station then replies with CTS after the SIFS wait period, effectively preventing any other transmission after
the RTS. The CTS packet also contains the data-packet size. The original sender then waits for SIFS after
receiving the CTS, and sends the packet. If all other stations can hear both the RTS and CTS messages, then
once the RTS and CTS are sent successfully no collisions should occur during packet transmission, again
because the only idle times are of length SIFS and other stations should be waiting for time IFS.
3.3.2.2 Hidden-Node Problem
Consider the diagram below. Each station has a 100-meter range. Stations A and B are 150 meters apart and
so cannot hear one another at all; each is 75 meters from C. If A is transmitting and B senses the medium in
preparation for its own transmission, as part of collision avoidance, then B will conclude that the medium is
idle and will go ahead and send.

However, C is within range of both A and B. If A and B transmit simultaneously, then from Cs perspective
a collision occurs. C receives nothing usable. We will call this a hidden-node collision as the senders A
and B are hidden from one another; the general scenario is known as the hidden-node problem.
Note that node D receives only As signal, and so no collision occurs at D.

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The hidden-node problem can also occur if A and B cannot receive one anothers transmissions due to a
physical obstruction such as a radio-impermeable wall:

One of the rationales for the RTS/CTS protocol is the prevention of hidden-node collisions. Imagine that,
instead of transmitting its data packet, A sends an RTS packet, and C responds with CTS. B has not heard
the RTS packet from A, but does hear the CTS from C. A will begin transmitting after a SIFS interval, but B
will not hear As transmission. However, B will still wait, because the CTS packet contained the data-packet
size and thus, implicitly, the length of time all other stations should remain idle. Because RTS packets are
quite short, they are much less likely to be involved in collisions themselves than data packets.
3.3.2.3 Wi-Fi Fragmentation
Conceptually related to RTS/CTS is Wi-Fi fragmentation. If error rates or collision rates are high, a sender
can send a large packet as multiple fragments, each receiving its own link-layer ACK. As we shall see in
5.3.1 Error Rates and Packet Size, if bit-error rates are high then sending several smaller packets often
leads to fewer total transmitted bytes than sending the same data as one large packet.
Wi-Fi packet fragments are reassembled by the receiving node, which may or may not be the final destination.
As with the RTS/CTS threshold, the fragmentation threshold is often set to the size of the maximum packet.
Adjusting the values of these thresholds is seldom necessary, though might be appropriate if monitoring
revealed high collision or error rates. Unfortunately, it is essentially impossible for an individual station
to distinguish between reception errors caused by collisions and reception errors caused by other forms of
noise, and so it is hard to use reception statistics to distinguish between a need for RTS/CTS and a need for
fragmentation.

3.3.3 Dynamic Rate Scaling

Wi-Fi senders, if they detect transmission problems, are able to reduce their transmission bit rate in a process
known as rate scaling or rate control. The idea is that lower bit rates will have fewer noise-related errors,
and so as the error rate becomes unacceptably high perhaps due to increased distance the sender should
fall back to a lower bit rate. For 802.11g, the standard rates are 54, 48, 36, 24, 18, 12, 9 and 6 Mbps. Senders
attempt to find the transmission rate that maximizes throughput; for example, 36 Mbps with a packet loss
rate of 25% has an effective throughput of 36 75% = 27 Mbps, and so is better than 24 Mbps with no
losses.
Senders may update their bit rate on a per-packet basis; senders may also choose different bit rates for
different recipients. For example, if a sender sends a packet and receives no confirming link-layer ACK, the

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sender may fall back to the next lower bit rate. The actual bit-rate-selection algorithm lives in the particular
Wi-Fi driver in use; different nodes in a network may use different algorithms.
The earliest rate-scaling algorithm was Automatic Rate Fallback, or ARF, [KM97]. The rate decreases after
two consecutive transmission failures (that is, the link-layer ACK is not received), and increases after ten
transmission successes.
A significant problem for rate scaling is that a packet loss may be due either to low-level random noise
(white noise, or thermal noise) or to a collision (which is also a form of noise, but less random); only in the
first case is a lower transmission rate likely to be helpful. If a larger number of collisions is experienced, the
longer packet-transmission times caused by the lower bit rate may increase the frequency of hidden-node
collisions. In fact, a higher transmission rate (leading to shorter transmission times) may help; enabling the
RTS/CTS protocol may also help.
Signal Strength
Most Wi-Fi drivers report the received signal strength. Newer drivers use the IEEE Received Channel
Power Indicator convention; the RCPI is an 8-bit integer proportional to the absolute power received
by the antenna as measured in decibel-milliwatts (dBm). Wi-Fi values range from -10 dBm to -90 dBm
and below. For comparison, the light from the star Polaris delivers about -97 dBm to one eye on a good
night; Venus typically delivers about -73 dBm. A GPS satellite might deliver -127 dBm to your phone.
(Inspired by Wikipedia on DBm.)
A variety of newer rate-scaling algorithms have been proposed; see [JB05] for a summary. One, ReceiverBased Auto Rate (RBAR, [HVB01]), attempts to incorporate the signal-to-noise ratio into the calculation of
the transmission rate. This avoids the confusion introduced by collisions. Unfortunately, while the signalto-noise ratio has a strong theoretical correlation with the transmission bit-error rate, most Wi-Fi radios
will report to the host system the received signal strength. This is not the same as the signal-to-noise ratio,
which is harder to measure. As a result, the RBAR approach has not been quite as effective in practice as
might be hoped.
The Collision-Aware Rate Adaptation algorithm (CARA, [KKCQ06]) attempts (among other things) to infer
that a packet was lost to a collision rather than noise if, after one SIFS interval following the end of the packet
transmission, no link-layer ACK has been received and the channel is still busy. This will detect collisions,
of course, only with longer packets.
Because the actual data in a Wi-Fi packet may be sent at a rate not every participant is close enough to
receive correctly, every Wi-Fi transmission begins with a brief preamble at the minimum bit rate. Link-layer
ACKs, too, are sent at the minimum bit rate.

3.3.4 Wi-Fi Configurations

There are two common Wi-Fi configurations: infrastructure and ad hoc. The latter includes individual
Wi-Fi-equipped nodes communicating informally; for example, two laptops can set up an ad hoc connection
to transfer data at a meeting. Ad hoc connections are often used for very simple networks not providing
Internet connectivity. Complex ad hoc networks are, however, certainly possible; see 3.3.8 MANETs.
The more-common infrastructure configuration involves designated access points to which individual WiFi stations must associate before general communication can begin. The association process carried
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out by an exchange of special management packets may be restricted to stations with hardware (LAN)
addresses on a predetermined list, or to stations with valid cryptographic credentials. Stations may regularly
re-associate to their Access Point, especially if they wish to communicate some status update.
Access Points
Generally, a Wi-Fi access point has special features; Wi-Fi-enabled station devices like phones and
workstations do not act as access points. However, it may be possible to for a station device to become
an access point if the access-point mode is supported by the underlying radio hardware and if suitable
drivers can be found. Under linux, the hostapd package is one option.
Stations in an infrastructure network communicate directly only with their access point. If B and C share
access point A, and B wishes to send a packet to C, then B first forwards the packet to A and A then forwards
it to C. While this introduces a degree of inefficiency, it does mean that the access point and its associated
nodes automatically act as a true LAN: every node can reach every other node. In an ad hoc network, by
comparison, it is quite common for two nodes to be able to reach each other only by forwarding through an
intermediate third node; this is in fact exactly the hidden-node scenario.
Finally, Wi-Fi is by design completely interoperable with Ethernet; if station A is associated with access
point AP, and AP also connects via (cabled) Ethernet to station B, then if A wants to send a packet to B it
sends it using AP as the Wi-Fi destination but with B also included in the header as the actual destination.
Once it receives the packet by wireless, AP acts as an Ethernet switch and forwards the packet to B.
While this forwarding is transparent to senders, the Ethernet and Wi-Fi LAN header formats are entirely
different.

The above diagram illustrates an Ethernet header and the Wi-Fi header for a typical data packet not using
Wi-Fi quality-of-service features. The Ethernet type field usually moves to an IEEE Logical Link Control
header in the Wi-Fi region labeled data. The receiver and transmitter addresses are the MAC addresses
of the nodes receiving and transmitting the (unicast) packet; these may each be different from the ultimate
destination and source addresses. In infrastructure mode one of the receiver or transmitter addresses is the
access point; in typical situations either the receiver is the destination or the sender is the transmitter.

3.3.5 Wi-Fi Roaming

Wi-Fi access points are generally identified by their SSID, an administratively defined string such as
linksys or loyola. These are periodically broadcast by the access point in special beacon packets (though

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for pseudo-security reasons beacon packets can be suppressed). Large installations can create roaming access among multiple access points by assigning all the access points the same SSID. An individual station
will stay with the access point with which it originally associated until the signal strength falls below a certain level, at which point it will seek out other access points with the same SSID and with a stronger signal.
In this way, a large area can be carpeted with multiple Wi-Fi access points, so as to look like one large Wi-Fi
domain.
In order for this to work, traffic to wireless node B must find Bs current access point AP. This is done in
much the same way as, in a wired Ethernet, traffic finds a laptop that has been unplugged, carried to a new
building, and plugged in again. The distribution network is the underlying wired network (eg Ethernet) to
which all the access points connect. If the distribution network is a switched Ethernet supporting the usual
learning mechanism (2.4 Ethernet Switches), then Wi-Fi location update is straightforward. Suppose B
is a wireless node that has been exchanging packets via the distribution network with C (perhaps a router
connecting B to the Internet). When B moves to a new access point, all it has to do is send any packet over
the LAN to C, and the Ethernet switches involved will then learn the route through the switched Ethernet
from C to Bs current AP, and thus to B.
This process may leave other switches not currently communicating with B still holding in their forwarding tables the old location for B. This is not terribly serious, but can be avoided entirely if, after moving, B
sends out an Ethernet broadcast packet.
Ad hoc networks also have SSIDs; these are generated pseudorandomly at startup. Ad hoc networks have
beacon packets as well; all nodes participate in the regular transmission of these via a distributed algorithm.

3.3.6 Wi-Fi Security

Wi-Fi traffic is visible to anyone nearby with an appropriate receiver; this eavesdropping zone can be expanded by use of a larger antenna. Because of this, Wi-Fi security is important, and Wi-Fi supports several
types of traffic encryption.
The original Wired-Equivalent Privacy, or WEP, encryption standard, contained a fatal (and now-classic)
flaw. Bytes of the key could be could be broken that is, guessed sequentially. Knowing bytes 0 through
i1 would allow an attacker to guess byte i with a relatively small amount of data, and so on through the
entire key. See 21.6.8 Wi-Fi WEP Encryption Failure for details.
WEP has been replaced with Wi-Fi Protected Access, or WPA; the current version is WPA2. WPA2 encryption is believed to be quite secure, although there was a vulnerability in the associated Wi-Fi Protected
Setup protocol.
Key management, however, can be a problem. Many smaller sites use pre-shared key mode, known as
WPA-Personal, in which a single key is entered into the Access Point (ideally not over the air) and into
each connecting station. The key is usually a hash of a somewhat longer passphrase. The use of a common
key for multiple stations makes changing the key, or revoking the key for a particular user, difficult.
A more secure approach is WPA-Enterprise; this allows each station to have a separate key, and also for
each station key to be recognized by all access points. It is part of the IEEE 802.1X security framework
(technically so is WPA-Personal, though a much smaller part). In this model, the client node is known as
the supplicant, the Access Point is known as the authenticator, and there is a third system involved known
as the authentication server. One authentication server can support multiple access points.

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To begin the association process, the supplicant contacts the authenticator using the Extensible Authentication Protocol, or EAP, with what amounts to a request to associate to that access point. EAP is a generic
message framework meant to support multiple specific types of authentication; see RFC 3748 and RFC
5247. The EAP request is forwarded to an authentication server, which may exchange (via the authenticator) several challenge/response messages with the supplicant. EAP is usually used in conjunction with
the RADIUS (Remote Authentication Dial-In User Service) protocol (RFC 2865), which is a specific (but
flexible) authentication-server protocol. WPA-Enterprise is sometimes known as 802.1X mode, EAP mode
or RADIUS mode.
One peculiarity of EAP is that EAP communication takes place before the supplicant is given an IP address
(in fact before the supplicant has completed associating itself to the access point); thus, a mechanism must
be provided to support exchange of EAP packets between supplicant and authenticator. This mechanism
is known as EAPOL, for EAP Over LAN. EAP messages between the authenticator and the authentication
server, on the other hand, can travel via IP; in fact, sites may choose to have the authentication server hosted
remotely.
Once the authentication server (eg RADIUS server) is set up, specific per-user authentication methods can
be entered. This can amount to username,password pairs, or some form of security certificate, or often
both. The authentication server will generally allow different encryption protocols to be used for different
supplicants, thus allowing for the possibility that there is not a common protocol supported by all stations.
When this authentication strategy is used, the access point no longer needs to know anything about what
authentication protocol is actually used; it is simply the middleman forwarding EAP packets between the
supplicant and the authentication server. The access point allows the supplicant to associate into the network
once it receives permission to do so from the authentication server.

3.3.7 Wi-Fi Polling Mode

Wi-Fi also includes a polled mechanism, where one station (the Access Point) determines which stations
are allowed to send. While it is not often used, it has the potential to greatly reduce collisions, or even
eliminate them entirely. This mechanism is known as Point Coordination Function, or PCF, versus the
collision-oriented mechanism which is then known as Distributed Coordination Function. The PCF name
refers to the fact that in this mode it is the Access Point that is in charge of coordinating which stations get
to send when.
The PCF option offers the potential for regular traffic to receive improved throughput due to fewer collisions.
However, it is often seen as intended for real-time Wi-Fi traffic, such as voice calls over Wi-Fi.
The idea behind PCF is to schedule, at regular intervals, a contention-free period, or CFP. During this
period, the Access Point may
send Data packets to any receiver
send Poll packets to any receiver, allowing that receiver to reply with its own data packet
send a combination of the two above (not necessarily to the same receiver)
send management packets, including a special packet marking the end of the CFP
None of these operations can result in a collision (unless an unrelated but overlapping Wi-Fi domain is
involved).

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Stations receiving data from the Access Point send the usual ACK after a SIFS interval. A data packet from
the Access Point addressed to station B may also carry, piggybacked in the Wi-Fi header, a Poll request to
another station C; this saves a transmission. Polled stations that send data will receive an ACK from the
Access Point; this ACK may be combined in the same packet with the Poll request to the next station.
At the end of the CFP, the regular contention period or CP resumes, with the usual CSMA/CA strategy.
The time interval between the start times of consecutive CFP periods is typically 100 ms, short enough to
allow some real-time traffic to be supported.
During the CFP, all stations normally wait only the Short IFS, SIFS, between transmissions. This works
because normally there is only one station designated to respond: the Access Point or the polled station.
However, if a station is polled and has nothing to send, the Access Point waits for time interval PIFS (PCF
Inter-Frame Spacing), of length midway between SIFS and IFS above (our previous IFS should now really
be known as DIFS, for DCF IFS). At the expiration of the PIFS, any non-Access-Point station that happens
to be unaware of the CFP will continue to wait the full DIFS, and thus will not transmit. An example of
such a CFP-unaware station might be one that is part of an entirely different but overlapping Wi-Fi network.
The Access Point generally maintains a polling list of stations that wish to be polled during the CFP. Stations
request inclusion on this list by an indication when they associate or (more likely) reassociate to the Access
Point. A polled station with nothing to send simply remains quiet.
PCF mode is not supported by many lower-end Wi-Fi routers, and often goes unused even when it is available. Note that PCF mode is collision-free, so long as no other Wi-Fi access points are active and within
range. While the standard has some provisions for attempting to deal with the presence of other Wi-Fi
networks, these provisions are somewhat imperfect; at a minimum, they are not always supported by other
access points. The end result is that polling is not quite as useful as it might be.

3.3.8 MANETs
The MANET acronym stands for mobile ad hoc network; in practice, the term generally applies to ad hoc
wireless networks of sufficient complexity that some internal routing mechanism is needed to enable full
connectivity. The term mesh network is also used. While MANETs can use any wireless mechanism, we
will assume here that Wi-Fi is used.
MANET nodes communicate by radio signals with a finite range, as in the diagram below.

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Each nodes radio range is represented by a circle centered about that node. In general, two MANET nodes
may be able to communicate only by relaying packets through intermediate nodes, as is the case for nodes
A and G in the diagram above.
In the field, the radio range of each node may not be very circular, due to among other things signal reflection and blocking from obstructions. An additional complication arises when the nodes (or even just
obstructions) are moving in real time (hence the mobile of MANET); this means that a working route may
stop working a short time later. For this reason, and others, routing within MANETs is a good deal more
complex than routing in an Ethernet. A switched Ethernet, for example, is required to be loop-free, so there
is never a choice among multiple alternative routes.
Note that, without successful LAN-layer routing, a MANET does not have full node-to-node connectivity
and thus does not meet the definition of a LAN given in 1.9 LANs and Ethernet. With either LAN-layer or
IP-layer routing, one or more MANET nodes may serve as gateways to the Internet.
Note also that MANETs in general do not support broadcast, unless the forwarding of broadcast messages
throughout the MANET is built in to the routing mechanism. This can complicate the assignment of IP
addresses; the common IPv4 mechanism we will describe in 7.8 Dynamic Host Configuration Protocol
(DHCP) relies on broadcast and so usually needs some adaptation.
Finally, we observe that while MANETs are of great theoretical interest, their practical impact has been
modest; they are almost unknown, for example, in corporate environments. They appear most useful in
emergency situations, rural settings, and settings where the conventional infrastructure network has failed
or been disabled.
3.3.8.1 Routing in MANETs
Routing in MANETs can be done either at the LAN layer, using physical addresses, or at the IP layer with
some minor bending (below) of the rules.

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Either way, nodes must find out about the existence of other nodes, and appropriate routes must then be
selected. Route selection can use any of the mechanisms we describe later in 9 Routing-Update Algorithms.
Routing at the LAN layer is much like routing by Ethernet switches; each node will construct an appropriate
forwarding table. Unlike Ethernet, however, there may be multiple paths to a destination, direct connectivity
between any particular pair of nodes may come and go, and negotiation may be required even to determine
which MANET nodes will serve as forwarders.
Routing at the IP layer involves the same issues, but at least IP-layer routing-update algorithms have always
been able to handle multiple paths. There are some minor issues, however. When we initially presented
IP forwarding in 1.10 IP - Internet Protocol, we assumed that routers made their decisions by looking
only at the network prefix of the address; if another node had the same network prefix it was assumed to be
reachable directly via the LAN. This model usually fails badly in MANETs, where direct reachability has
nothing to do with addresses. At least within the MANET, then, a modified forwarding algorithm must be
used where every address is looked up in the forwarding table. One simple way to implement this is to have
the forwarding tables contain only host-specific entries as were discussed in 3.1 Virtual Private Network.
Multiple routing algorithms have been proposed for MANETs. Performance of a given algorithm may
depend on the following factors:
The size of the network
Whether some nodes have agreed to serve as routers
The degree of node mobility, especially of routing-node mobility if applicable
Whether the nodes are under common administration, and thus may agree to defer their own transmission interests to the common good
per-node storage and power availability

3.4 WiMAX
WiMAX is a wireless network technology standardized by IEEE 802.16. It supports both stationary subscribers (802.16d) and mobile subscribers (802.16e). The stationary-subscriber version is often used to
provide residential Internet connectivity, in both urban and rural areas. The mobile version is sometimes
referred to as a fourth generation or 4G networking technology; its similar primary competitor is known
as LTE. WiMAX is used in many mobile devices, from smartphones to traditional laptops with wireless
cards installed.
As in the sidebar at the start of 3.3 Wi-Fi we will use the term data rate for what is commonly called
bandwidth to avoid confusion with the radio-specific meaning of the latter term.
WiMAX can use unlicensed frequencies, like Wi-Fi, but its primary use is over licensed radio spectrum.
WiMAX also supports a number of options for the width of its frequency band; the wider the band, the
higher the data rate. Wider bands also allow the opportunity for multiple independent frequency channels.
Downlink (base station to subscriber) data rates can be well over 100 Mbps (uplink rates are usually smaller).
Like Wi-Fi, WiMAX subscriber stations connect to a central access point, though the WiMAX standard
prefers the term base station which we will use henceforth. Stationary-subscriber WiMAX, however, operates on a much larger scale. The coverage radius of a WiMAX base station can be tens of kilometers

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if larger antennas are provided, versus less (sometimes much less) than 100 meters for Wi-Fi; mobilesubscriber WiMAX might have a radius of one or two kilometers. Large-radius base stations are typically
mounted in towers. Subscriber stations are not generally expected to be able to hear other stations; they
interact only with the base station. As WiMAX distances increase, the data rate is reduced.
As with Wi-Fi, the central contention problem is how to schedule transmissions of subscriber stations so
they do not overlap; that is, collide. The base station has no difficulty broadcasting transmissions to multiple
different stations sequentially; it is the transmissions of those stations that must be coordinated. Once a
station completes the network entry process to connect to a base station (below), it is assigned regular
(though not necessarily periodic) transmission slots. These transmission slots may vary in size over time;
the base station may regularly issue new transmission schedules.
The centralized assignment of transmission intervals superficially resembles Wi-Fi PCF mode (3.3.7 WiFi Polling Mode); however, assignment is not done through polling, as propagation delays are too large
(below). Instead, each WiMAX subscriber station is told in effect that it may transmit starting at an assigned
time T and for an assigned length L. The station synchronizes its clock with that of the base station as part
of the network entry process.
Because of the long distances involved, synchronization and transmission protocols must take account of
speed-of-light delays. The round-trip delay across 30 km is 200 sec which is ten times larger than the basic
Wi-Fi SIFS interval; at 160 Mbps, this is the time needed to send 4 KB. If a station is to transmit so that its
message arrives at the base station at a certain time, it must actually begin transmission early by an amount
equal to the one-way station-to-base propagation delay; a special ranging mechanism allows stations to
figure out this delay.
A subscriber station begins the network-entry connection process to a base station by listening for the base
stations transmissions (which may be organized into multiple channels); these message streams contain
regular management messages containing, among other things, information about available data rates in
each direction.
Also included in the base stations message stream is information about start times for ranging intervals.
The station waits for one of these intervals and sends a range-request message to the base station. These
ranging intervals are open to all stations attempting network entry, and if another station transmits at the
same time there will be a collision. However, network entry is only done once (for a given base station) and
so the likelihood of a collision in any one ranging interval is small. An Ethernet/Wi-Fi-like exponentialbackoff process is used if a collision does occur. Ranging intervals are the only times when collisions can
occur; afterwards, all station transmissions are scheduled by the base station.
If there is no collision, the base station responds, and the station now knows the propagation delay and thus
can determine when to transmit so that its data arrives at the base station exactly at a specified time. The
station also determines its transmission signal strength from this ranging process.
Finally, and perhaps most importantly, the station receives from the base station its first timeslot for a
scheduled transmission. These timeslot assignments are included in regular uplink-map packets broadcast
by the base station. Each stations timeslot includes both a start time and a total length; lengths are in the
range of 2 to 20 ms. Future timeslots will be allocated as necessary by the base station, in future uplink-map
packets. Scheduled timeslots may be periodic (as is would be appropriate for voice) or may occur at varying
intervals. WiMAX stations may request any of several quality-of-service levels and the base station may
take these requests into account when determining the schedule. The base station also creates a downlink
schedule, but this does not need to be communicated to the subscriber stations; the base station simply uses
it to decide what to broadcast when to the stations. When scheduling the timeslots, the base station may also
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take into account availability of multiple transmission channels and of directional antennas.
Through the uplink-map schedules and individual ranging, each station transmits so that one transmission
finishes arriving just before the next transmission begins arriving, as seen from the perspective of the base
station. Only minimal guard intervals need be included between consecutive transmissions. Two (or
more) consecutive transmissions may in fact be in the air simultaneously, as far-away stations need to
begin transmitting early so their signals will arrive at the base station at the expected time. The following
diagram illustrates this for stations separated by relatively large physical distances.

Mobile stations will need to update their ranging information regularly, but this can be done through future
scheduled transmissions. The distance to the base station is used not only for the mobile stations transmission timing, but also to determine its power level; signals from each mobile station, no matter where located,
should arrive at the base station with about the same power.
When a station has data to send, it includes in its next scheduled transmission a request for a longer transmission interval; if the request is granted, the station may send the data (or at least some of the data) in its
next scheduled transmission slot. When a station is done transmitting, its timeslot shrinks to the minimum,
and may be scheduled less frequently as well, but it does not disappear. Stations without data to send remain
connected to the base station by sending empty messages during these slots.

3.5 Fixed Wireless

This category includes all wireless-service-provider systems where the subscribers location does not
change. Often, but not always, the subscriber will have an outdoor antenna for improved reception and
range. Fixed-wireless systems can involve relay through satellites, or can be terrestrial.

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3.5.1 Terrestrial Wireless

Terrestrial wireless also called terrestrial broadband or fixed-wireless broadband involves direct (nonsatellite) radio communication between subscribers and a central access point. Access points are usually
tower-mounted and serve multiple subscribers, though single-subscriber point-to-point microwave links
also exist. A multi-subscriber access point may serve an area with radius up to several tens of miles, depending on the technology, though more common ranges are under ten miles. WiMAX 802.16d is one form
of terrestrial wireless, but there are several others. Frequencies may be either licensed or unlicensed. Unlicensed frequency bands are available at around 900 MHz, 2.4 GHz, and 5 GHz. Nominally all three bands
require that line-of-sight transmission be used, though that requirement becomes stricter as the frequency
increases. Lower frequencies tend to be better at seeing through trees and other obstructions.

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Trees vs Signal

Photo of the author attempting to improve his 2.4 GHz terrestrial-wireless signal via tree trimming.
Terrestrial fixed wireless was originally popularized for rural areas, where residential density is too low for
economical cable connections. However, some fixed-wireless ISPs now operate in urban areas, often using
WiMAX. One advantage of terrestrial fixed-wireless in remote areas is that the antennas covers a much
smaller geographical area than a satellite, generally meaning that there is more data bandwidth available per
user and the cost per megabyte is much lower.
Outdoor subscriber antennas often use a parabolic dish to improve reception; sizes range from 10 to 50 cm
in diameter. The size of the dish may depend on the distance to the central tower.
While there are standardized fixed-wireless systems, such as WiMAX, there are also a number of proprietary alternatives, including systems from Trango and Canopy. Fixed-wireless systems might, in fact, be
considered one of the last bastions of proprietary LAN protocols. This lack of standardization is due to a
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variety of factors; two primary ones are the relatively modest overall demand for this service and the the fact
that most antennas need to be professionally installed by the ISP to ensure that they are properly mounted,
aligned, grounded and protected from lightning.

3.5.2 Satellite Internet

An extreme case of fixed wireless is satellite Internet, in which signals pass through a satellite in geosynchronous orbit (35,786 km above the earths surface). Residential customers have parabolic antennas typically from 70 to 100 cm in diameter, larger than those used for terrestrial wireless but smaller than the dish
antennas used at access points. Transmitter power is typically 1-2 watts, remarkably low for a signal that
travels 35,786 km.
The primary problem associated with satellite Internet is very long RTTs. The the speed-of-light round-trip
propagation delay is about 500 ms to which must be added queuing delays for the often-backlogged access
point (my own personal experience suggested that RTTs of close to 1,000 ms were the norm). These long
delays affect real-time traffic such as VoIP and gaming, but as we shall see in 14.11 The Satellite-Link TCP
Problem bulk TCP transfers also perform poorly with very long RTTs. To provide partial compensation for
the TCP issue, many satellite ISPs provide some sort of acceleration for bulk downloads: a web page, for
example, would be downloaded rapidly by the access point and streamed to the satellite and back down to
the user via a proprietary mechanism. Acceleration, however, cannot help interactive connections such as
VPNs.
Another common feature of satellite Internet is a low daily utilization cap, typically in the hundreds of
megabytes. Utilization caps are directly tied to the cost of maintaining satellites, but also to the fact that one
satellite covers a great deal of ground, and so its available capacity is shared by a large number of users.

3.6 Token Ring

A significant part of the previous chapter was devoted to classic Ethernets collision mechanism for supporting shared media access. After that, it may come as a surprise that there is a simple multiple-access
mechanism that is not only collision-free, but which supports fairness in the sense that if N stations wish to
send then each will receive 1/N of the opportunities.
That method is Token Ring. Actual implementations come in several forms, from Fiber-Distributed Data
Interface (FDDI) to so-called IBM Token Ring. The central idea is that stations are connected in a ring:

Packets will be transmitted in one direction (clockwise in the ring above). Stations in effect forward most
packets around the ring, although they can also remove a packet. (It is perhaps more accurate to think

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of the forwarding as representing the default cable connectivity; non-forwarding represents the stations
momentarily breaking that connectivity.)
When the network is idle, all stations agree to forward a special, small packet known as a token. When a
station, say A, wishes to transmit, it must first wait for the token to arrive at A. Instead of forwarding the
token, A then transmits its own packet; this travels around the network and is then removed by A. At that
point (or in some cases at the point when A finishes transmitting its data packet) A then forwards the token.
In a small ring network, the ring circumference may be a small fraction of one packet. Ring networks
become large at the point when some packets may be entirely in transit on the ring. Slightly different
solutions apply in each case. (It is also possible that the physical ring exists only within the token-ring
switch, and that stations are connected to that switch using the usual point-to-point wiring.)
If all stations have packets to send, then we will have something like the following:
A waits for the token
A sends a packet
A sends the token to B
B sends a packet
B sends the token to C
C sends a packet
C sends the token to D
...
All stations get an equal number of chances to transmit, and no bandwidth is wasted on collisions.
One problem with token ring is that when stations are powered off it is essential that the packets continue
forwarding; this is usually addressed by having the default circuit configuration be to keep the loop closed.
Another issue is that some station has to watch out in case the token disappears, or in case a duplicate token
appears.
Because of fairness and the lack of collisions, IBM Token Ring was once considered to be the premium LAN
mechanism. As such, a premium price was charged (there was also the matter of licensing fees). But due
to a combination of lower hardware costs and higher bitrates (even taking collisions into account), Ethernet
eventually won out.
There was also a much earlier collision-free hybrid of 10 Mbps Ethernet and Token Ring known as Token
Bus: an Ethernet physical network (often linear) was used with a token-ring-like protocol layer above
that. Stations were physically connected to the (linear) Ethernet but were assigned identifiers that logically
arranged them in a (virtual) ring. Each station had to wait for the token and only then could transmit a
packet; after that it would send the token on to the next station in the virtual ring. As with real Token
Ring, some mechanisms need to be in place to monitor for token loss.
Token Bus Ethernet never caught on. The additional software complexity was no doubt part of the problem,
but perhaps the real issue was that it was not necessary.

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3.7 Virtual Circuits

Before we can get to our final LAN example, ATM, we need to detour briefly through virtual circuits.
Virtual circuits are The Road Not Taken by IP.
Virtual-circuit switching is an alternative to datagram switching, which was introduced in Chapter 1. In
datagram switching, routers know the next_hop to each destination, and packets are addressed by destination. In virtual-circuit switching, routers know about end-to-end connections, and packets are addressed
by a connection ID.
Before any packets can be sent, a connection needs to be established first. For that connection, the route is
computed and then each link along the path is assigned a connection ID, traditionally called the VCI, for
Virtual Circuit Identifier. In most cases, VCIs are only locally unique; that is, the same connection may use
a different VCI on each link. The lack of global uniqueness makes VCI allocation much simpler. Although
the VCI keeps changing along a path, the VCI can still be thought of as identifying the connection. To send
a packet, the host marks the packet with the VCI assigned to the hostrouter1 link.
Packets arrive at (and depart from) switches via one of several ports, which we will assume are numbered
beginning at 0. Switches maintain a connection table indexed by VCI,port pairs; unlike a forwarding
table, the connection table has a record of every connection through that switch at that particular moment.
As a packet arrives, its inbound VCIin and inbound portin are looked up in this table; this yields an outbound
VCIout ,portout pair. The VCI field of the packet is then rewritten to VCIout , and the packet is sent via
portout .
Note that typically there is no source address information included in the packet (although the sender can
be identified from the connection, which can be identified from the VCI at any point along the connection).
Packets are identified by connection, not destination. Any node along the path (including the endpoints) can
in principle look up the connection and figure out the endpoints.
Note also that each switch must rewrite the VCI. Datagram switches never rewrite addresses (though they
do update hopcount/TTL fields). The advantage to this rewriting is that VCIs need be unique only for a
given link, greatly simplifying the naming. Datagram switches also do not make use of a packets arrival
interface.
As an example, consider the network below. Switch ports are numbered 0,1,2,3. Two paths are drawn in,
one from A to F in red and one from B to D in green; each link is labeled with its VCI number in the same
color.
We will construct virtual-circuit connections between
A and F (shown above in red)
A and E
A and C
B and D (shown above in green)
A and F again (a separate connection)
The following VCIs have been chosen for these connections. The choices are made more or less randomly
here, but in accordance with the requirement that they be unique to each link. Because links are generally

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taken to be bidirectional, a VCI used from S1 to S3 cannot be reused from S3 to S1 until the first connection
closes.
A to F: A 4

S1 6

S2 4 S4 8

S5 5 F; this path goes from S1 to S4 via S2

A to E: A 5 S1 6 S3 3 S4 8 E; this path goes, for no particular reason, from S1 to S4

via S3, the opposite corner of the square
A to C: A 6

S1 7

S3 3

B to D: B 4

S3 8 S1 7

A to F: A 7

S1 8

C
S2 8

S2 5 S4 9

D
S5 2 F

One may verify that on any one link no two different paths use the same VCI.
We now construct the actual VCI,port tables for the switches S1-S4, from the above; the table for S5 is left
as an exercise. Note that either the VCIin ,portin or the VCIout ,portout can be used as the key; we cannot
have the same pair in both the in columns and the out columns. It may help to display the port numbers for
each switch, as in the upper numbers in following diagram of the above red connection from A to F (lower
numbers are the VCIs):

Switch S1:
VCIin
4
5
6
8
7

portin
0
0
0
1
0

VCIout
6
6
7
7
8

portout
2
1
1
2
2

connection
AF #1
AE
AC
BD
AF #2

Switch S2:

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VCIin
6
7
8

portin
0
0
0

VCIout
4
8
5

portout
1
2
1

connection
AF #1
BD
AF #2

VCIout
3
3
8

portout
2
1
3

connection
AE
AC
BD

VCIout
8
8
9

portout
2
1
2

connection
AF #1
AE
AF #2

Switch S3:
VCIin
6
7
4

portin
3
3
0

Switch S4:
VCIin
4
3
5

portin
3
0
3

The namespace for VCIs is small, and compact (eg contiguous). Typically the VCI and port bitfields can be
concatenated to produce a VCI,Port composite value small enough that it is suitable for use as an array
index. VCIs work best as local identifiers. IP addresses, on the other hand, need to be globally unique, and
thus are often rather sparsely distributed.
Virtual-circuit switching offers the following advantages:
connections can get quality-of-service guarantees, because the switches are aware of connections and
can reserve capacity at the time the connection is made
headers are smaller, allowing faster throughput
headers are small enough to allow efficient support for the very small packet sizes that are optimal for
voice connections. ATM packets, for instance, have 48 bytes of data; see below.
Datagram forwarding, on the other hand, offers these advantages:
Routers have less state information to manage.
Router crashes and partial connection state loss are not a problem.
If a router or link is disabled, rerouting is easy and does not affect any connection state. (As mentioned
in Chapter 1, this was Paul Barans primary concern in his 1962 paper introducing packet switching.)
Per-connection billing is very difficult.
The last point above may once have been quite important; in the era when the ARPANET was being developed, typical daytime long-distance rates were on the order of $1/minute. It is unlikely that early TCP/IP
protocol development would have been as fertile as it was had participants needed to justify per-minute
billing costs for every project.
It is certainly possible to do virtual-circuit switching with globally unique VCIs say the concatenation of
source and destination IP addresses and port numbers. The IP-based RSVP protocol (19.6 RSVP) does
exactly this. However, the fast-lookup and small-header advantages of a compact namespace are then lost.
Note that virtual-circuit switching does not suffer from the problem of idle channels still consuming resources, which is an issue with circuits using time-division multiplexing (eg shared T1 lines)

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3.8 Asynchronous Transfer Mode: ATM

ATM is a LAN mechanism intended to accommodate real-time traffic as well as bulk data transfer. It was
particularly intended to support voice. A significant source of delay in voice traffic is the packet fill time: at
DS0 speeds, a voice packet fills at 8 bytes/ms. If we are sending 1KB packets, this means voice is delayed
by about 1/8 second, meaning in turn that when one person stops speaking, the earliest they can hear the
others response is 1/4 second later. Voice delay also can introduce an annoying echo. When voice is sent
over IP (VoIP), one common method is to send 160 bytes every 20 ms.
ATM took this small-packet strategy even further: packets have 48 bytes of data, plus 5 bytes of header. Such
small packets are often called cells. To manage such a small header, virtual-circuit routing is a necessity. IP
packets of such small size would likely consume more than 50% of the bandwidth on headers, if the LAN
header were included.
Aside from reduced voice fill-time, other benefits to small cells are reduced store-and-forward delay and
minimal queuing delay, at least for high-priority traffic. Prioritizing traffic and giving precedence to highpriority traffic is standard, but high-priority traffic is never allowed to interrupt transmission already begun
of a low-priority packet. If you have a high-priority voice cell, and someone else has a 1500-byte packet
just started, your cell has to wait about 30 cell times, because 1500 bytes is about 30 cells. However, if their
low-priority traffic is instead made up of 30 cells, you have only to wait for their first cell to finish; the delay
is 1/30 as much.
ATM also made the decision to require fixed-size cells. The penalty for one partially used cell among many
is small. Having a fixed cell size simplifies hardware design, and, in theory, allows it easier to design for
parallelism.
Unfortunately, ATM also chose to mandate no cell reordering. This means cells can use a smaller sequencenumber field, but also makes parallel switches much harder to build. A typical parallel switch design might
involve forwarding incoming cells to any of several input queues; the queues would then handle the VCI
lookups in parallel and forward the cells to the appropriate output queues. With such an architecture, avoiding reordering is difficult. It is not clear to what extent the no-reordering decision was related to the later
decline of ATM in the marketplace.
ATM cells have 48 bytes of data and a 5-byte header. The header contains up to 28 bits of VCI information,
three type bits, one cell-loss priority, or CLP, bit, and an 8-bit checksum over the header only. The
VCI is divided into 8-12 bits of Virtual Path Identifier and 16 bits of Virtual Channel Identifier, the latter
supposedly for customer use to separate out multiple connections between two endpoints. Forwarding is by
full switching only, and there is no mechanism for physical (LAN) broadcast.

3.8.1 ATM Segmentation and Reassembly

Due to the small packet size, ATM defines its own mechanisms for segmentation and reassembly of larger
packets. Thus, individual ATM links in an IP network are quite practical. These mechanisms are called
ATM Adaptation Layers, and there are four of them: AALs 1, 2, 3/4 and 5 (AAL 3 and AAL 4 were once
separate layers, which merged). AALs 1 and 2 are used only for voice-type traffic; we will not consider
them further.
The ATM segmentation-and-reassembly mechanism defined here is intended to apply only to large data; no
cells are ever further subdivided. Furthermore, segmentation is always applied at the point where the data

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enters the network; reassembly is done at exit from the ATM path. IPv4 fragmentation, on the other hand,
applies conceptually to IP packets, and may be performed by routers within the network.
For AAL 3/4, we first define a high-level wrapper for an IP packet, called the CS-PDU (Convergence
Sublayer - Protocol Data Unit). This prefixes 32 bits on the front and another 32 bits (plus padding) on the
rear. We then chop this into as many 44-byte chunks as are needed; each chunk goes into a 48-byte ATM
payload, along with the following 32 bits worth of additional header/trailer:
2-bit type field:
10: begin new CS-PDU
00: continue CS-PDU
01: end of CS-PDU
11: single-segment CS-PDU
4-bit sequence number, 0-15, good for catching up to 15 dropped cells
10-bit MessageID field
CRC-10 checksum.
We now have a total of 9 bytes of header for 44 bytes of data; this is more than 20% overhead. This did not
sit well with the IP-over-ATM community (such as it was), and so AAL 5 was developed.
AAL 5 moved the checksum to the CS-PDU and increased it to 32 bits from 10 bits. The MID field was
discarded, as no one used it, anyway (if you wanted to send several different types of messages, you simply
created several virtual circuits). A bit from the ATM header was taken over and used to indicate:
1: start of new CS-PDU
0: continuation of an existing CS-PDU
The CS-PDU is now chopped into 48-byte chunks, which are then used as the entire body of each ATM
cell. With 5 bytes of header for 48 bytes of data, overhead is down to 10%. Errors are detected by the
CS-PDU CRC-32. This also detects lost cells (impossible with a per-cell CRC!), as we no longer have any
cell sequence number.
For both AAL3/4 and AAL5, reassembly is simply a matter of stringing together consecutive cells in order
of arrival, starting a new CS-PDU whenever the appropriate bits indicate this. For AAL3/4 the receiver
has to strip off the 4-byte AAL3/4 headers; for AAL5 the receiver has to verify the CRC-32 checksum once
all cells are received. Different cells from different virtual circuits can be jumbled together on the ATM
backbone, but on any one virtual circuit the cells from one higher-level packet must be sent one right after
the other.
A typical IP packet divides into about 20 cells. For AAL 3/4, this means a total of 200 bits devoted to CRC
codes, versus only 32 bits for AAL 5. It might seem that AAL 3/4 would be more reliable because of this,
but, paradoxically, it was not! The reason for this is that errors are rare, and so we typically have one or at
most two per CS-PDU. Suppose we have only a single error, ie a single cluster of corrupted bits small enough
that it is likely confined to a single cell. In AAL 3/4 the CRC-10 checksum will fail to detect that error (that
is, the checksum of the corrupted packet will by chance happen to equal the checksum of the original packet)
with probability 1/210 . The AAL 5 CRC-32 checksum, however, will fail to detect the error with probability
1/232 . Even if there are enough errors that two cells are corrupted, the two CRC-10s together will fail to

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detect the error with probability 1/220 ; the CRC-32 is better. AAL 3/4 is more reliable only when we have
errors in at least four cells, at which point we might do better to switch to an error-correcting code.
Moral: one checksum over the entire message is often better than multiple shorter checksums over parts of
the message.

3.9 Epilog
Along with a few niche protocols, we have focused primarily here on wireless and on virtual circuits. Wireless, of course, is enormously important: it is the enabler for mobile devices, and has largely replaced
traditional Ethernet for home and office workstations.
While it is sometimes tempting (in the IP world at least) to write off ATM as a niche technology, virtual
circuits are a serious conceptual alternative to datagram forwarding. As we shall see in 19 Quality of
Service, IP has problems handling real-time traffic, and virtual circuits offer a solution. The Internet has
so far embraced only small steps towards virtual circuits (such as MPLS, 19.12 Multi-Protocol Label
Switching (MPLS)), but they remain a tantalizing strategy.

3.10 Exercises
1. Suppose remote host A uses a VPN connection to connect to host B, with IP address 200.0.0.7. As
normal Internet connection is via device eth0 with IP address 12.1.2.3; As VPN connection is via device
ppp0 with IP address 10.0.0.44. Whenever A wants to send a packet via ppp0, it is encapsulated and
forwarded over the connection to B at 200.0.0.7.

(a). Suppose As IP forwarding table is set up so that all traffic to 200.0.0.7 uses eth0 and all traffic to
anywhere else uses ppp0. What happens if an intruder M attempts to open a connection to A at 12.1.2.3?
What route will packets from A to M take?
(b). Suppose As IP forwarding table is (mis)configured so that all outbound traffic uses ppp0. Describe
what will happen when A tries to send a packet.

2. Suppose remote host A wishes to use a TCP-based VPN connection to connect to host B, with IP address
200.0.0.7. However, the VPN software is not available for host A. Host A is, however, able to run that
software on a virtual machine V hosted by A; A and V have respective IP addresses 10.0.0.1 and 10.0.0.2
on the virtual network connecting them. V reaches the outside world through network address translation
(1.14 Network Address Translation), with A acting as Vs NAT router. When V runs the VPN software,
it forwards packets addressed to B the usual way, through A using NAT. Traffic to any other destination it
encapsulates over the VPN.
Can A configure its IP forwarding table so that it can make use of the VPN? If not, why not? If so, how? (If
you prefer, you may assume V is a physical host connecting to a second interface on A; A still acts as Vs
NAT router.)

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3. Token Bus was a proprietary Ethernet-based network. It worked like Token Ring in that a small token
packet was sent from one station to the next in agreed-upon order, and a station could transmit only when it
had just received the token.

(a). If the data rate is 10 Mbps and the token is 64 bytes long (the 10-Mbps Ethernet minimum packet size),
how long does it take on average to send a packet on an idle network with 40 stations? Ignore the
propagation delay and the gap Ethernet requires between packets.
(b). Repeat part (a) assuming the tokens are only 16 bytes long.
(c). Sketch a protocol by which stations can sort themselves out to decide the order of token transmission;
that is, an order of the stations S0 ... Sn-1 where station Si sends the token to station S(i+1) mod n .

4. The IEEE 802.11 standard states transmission of the ACK frame shall commence after a SIFS period,
without regard to the busy/idle state of the medium; that is, the ACK sender does not listen first for an idle
network. Give a scenario in which the Wi-Fi ACK frame would fail to be delivered in the absence of this
rule, but succeed with it. Hint: this is another example of the hidden-node problem, 3.3.2.2 Hidden-Node
Problem.
5. Suppose the average contention interval in a Wi-Fi network (802.11g) is 64 SlotTimes. The average
packet size is 1 KB, and the data rate is 54 Mbps. At that data rate, it takes about (81000)/54 = 148 sec
to transmit a packet.

(a). How long is the average contention interval, in sec?

(b). What fraction of the total potential bandwidth is lost to contention? (See 2.1.9 Analysis of Classic
Ethernet for a similar example).

6. WiMAX subscriber stations are not expected to hear one another at all. For Wi-Fi non-access-point
stations in an infrastructure (access-point) setting, on the other hand, listening to other non-access-point
transmissions is encouraged.

(a). List some ways in which Wi-Fi non-access-point stations in an infrastructure (access-point) network do
sometimes respond to packets sent by other non-access-point stations. The responses need not be in the
form of transmissions.
(b). Explain why Wi-Fi stations cannot be required to respond as in part (a).

7. Suppose WiMAX subscriber stations can be moving, at speeds of up to 33 meters/sec (the maximum
allowed under 802.16e).

(a). How much earlier (or later) can one subscriber packet arrive? Assume that the ranging process updates
the stations propagation delay once a minute. The speed of light is about 300 meters/sec.

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(b). With 5000 senders per second, how much time out of each second must be spent on guard intervals
accommodating the early/late arrivals above? You will need to double the time from part (a), as the base
station cannot tell whether the signal from a moving subscriber will arrive earlier or later.

8. [SM90] contained a proposal for sending IP packets over ATM as N cells as in AAL-5, followed by one
cell containing the XOR of all the previous cells. This way, the receiver can recover from the loss of any
one cell. Suppose N=20 here; with the SM90 mechanism, each packet would require 21 cells to transmit;
that is, we always send 5% more. Suppose the cell loss-rate is p (presumably very small). If we send 20
cells without the SM90 mechanism, we have a probability of about 20p that any one cell will be lost, and we
will have to retransmit the entire 20 again. This gives an average retransmission amount of about 20p extra
packets. For what value of p do the with-SM90 and the without-SM90 approaches involve about the same
total number of cell transmissions?
9. In the example in 3.7 Virtual Circuits, give the VCI table for switch S5.
10. Suppose we have the following network:
A

S1

S2

S3

S4

The virtual-circuit switching tables are below. Ports are identified by the node at the other end. Identify all
the connections. Give the path for each connection and the VCI on each link of the path.
Switch S1:
VCIin
1
2
3

portin
A
A
A

VCIout
2
2
3

portout
S3
S2
S2

VCIout
1
3
4

portout
B
S4
S4

VCIout
2
2

portout
S4
C

VCIout
2
3
1

portout
S2
S3
D

Switch S2:
VCIin
2
2
3

portin
S4
S1
S1

Switch S3:
VCIin
2
3

portin
S1
S4

Switch S4:
VCIin
2
3
4

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11. Suppose we have the following network:

A

S1

S2

S3

S4

Give virtual-circuit switching tables for the following connections. Route via a shortest path.

(a). AD
(b). CB, via S4
(c). BD
(d). AD, via whichever of S2 or S3 was not used in part (a)

12. Below is a set of switches S1 through S4. Define VCI-table entries so the virtual circuit from A to B
follows the path
A S1 S2 S4 S3 S1 S2 S4 S3 B
That is, each switch is visited twice.
A

S1

S2

S3

S4

3.10 Exercises

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4 LINKS

At the lowest (logical) level, network links look like serial lines. In this chapter we address how packet
structures are built on top of serial lines, via encoding and framing. Encoding determines how bits and
bytes are represented on a serial line; framing allows the receiver to identify the beginnings and endings of
packets.
We then conclude with the high-speed serial lines offered by the telecommunications industry, T-carrier and
SONET, upon which almost all long-haul point-to-point links that tie the Internet together are based.

4.1 Encoding and Framing

A typical serial line is ultimately a stream of bits, not bytes. How do we identify byte boundaries? This is
made slightly more complicated by the fact that, beneath the logical level of the serial line, we generally
have to avoid transmitting long runs of identical bits, because the receiver may simply lose count; this is
the clock synchronization problem (sometimes called the clock recovery problem). This means that, one
way or another, we cannot always just send the desired bits sequentially; for example, extra bits are often
inserted to break up long runs. Exactly how we do this is the encoding mechanism.
Once we have settled the transmission of bits, the next step is to determine how the receiver identifies the start
of each new packet. Ethernet packets are separated by physical gaps, but for most other link mechanisms
packets are sent end-to-end, with no breaks. How we tell when one packet stops and the next begins is the
framing problem.
To summarize:
encoding: correctly recognizing all the bits in a stream
framing: recognizing packet boundaries
These are related, though not the same.
For long (multi-kilometer) electrical serial lines, among other things we in addition want the average voltage
to be zero; that is, we want no DC component. We will mostly concern ourselves here, however, only with
lines short enough for this not to be a major concern.

4.1.1 NRZ
NRZ (Non-Return to Zero) is perhaps the simplest encoding; it corresponds to direct bit-by-bit transmission
of the 0s and 1s in the data. We have two signal levels, lo and hi, we set the signal to one or the other
of these depending on whether the data bit is 0 or 1, as in the diagram below. Note that in the diagram the
signal bits have been aligned with the start of the pulse representing that signal value.

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NRZ replaces an earlier RZ (Return to Zero) encoding, in which hi and lo corresponded to +1 and -1, and
between each pair of pulses corresponding to consecutive bits there was a brief return to the 0 level.
One drawback to NRZ is that we cannot distinguish between 0-bits and a signal that is simply idle. However,
the more serious problem is the lack of synchronization: during long runs of 0s or long runs of 1s, the
receiver can lose count, eg if the receivers clock is running a little fast or slow. The receivers clock can
and does resynchronize whenever there is a transition from one level to the other. However, suppose bits
are sent at one per s, the sender sends 5 1-bits in a row, and the receivers clock is running 10% fast. The
signal sent is a 5-s hi pulse, but when the pulse ends the receivers clock reads 5.5 s due to the clock
speedup. Should this represent 5 1-bits or 6 1-bits?

4.1.2 NRZI
An alternative that helps here (though not obviously at first) is NRZI, or NRZ Inverted. In this encoding,
we represent a 0-bit as no change, and a 1-bit as a transition from lo to hi or hi to lo:

Now there is a signal transition aligned above every 1-bit; a 0-bit is represented by the lack of a transition.
This solves the synchronization problem for runs of 1-bits, but does nothing to address runs of 0-bits.
However, NRZI can be combined with techniques to minimize runs of 0-bits, such as 4B/5B (below).

4.1.3 Manchester
Manchester encoding sends the data stream using NRZI, with the addition of a clock transition between
each pair of consecutive data bits. This means that the signaling rate is now double the data rate, eg 20
MHz for 10Mbps Ethernet (which does use Manchester encoding). The signaling is as if we doubled the
bandwidth and inserted a 1-bit between each pair of consecutive data bits, removing this extra bit at the
receiver:

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All these transitions mean that the longest the clock has to count is 1 bit-time; clock synchronization is
essentially solved, at the expense of the doubled signaling rate.

4.1.4 4B/5B
In 4B/5B encoding, for each 4-bit nybble of data we actually transmit a designated 5-bit symbol, or code,
selected to have enough 1-bits. A symbol in this sense is a digital or analog transmission unit that decodes
to a set of data bits; the data bits are not transmitted individually.
Specifically, every 5-bit symbol used by 4B/5B has at most one leading 0-bit and at most two trailing 0-bits.
The 5-bit symbols corresponding to the data are then sent with NRZI, where runs of 1s are safe. Note that
the worst-case run of 0-bits has length three. Note also that the signaling rate here is 1.25 times the data
rate. 4B/5B is used in 100-Mbps Ethernet, 2.2 100 Mbps (Fast) Ethernet. The mapping between 4-bit data
values and 5-bit symbols is fixed by the 4B/5B standard:
data
0000
0001
0010
0011
0100
0101
0110
0111
1000
1001
1010

symbol
11110
01001
10100
10101
01010
01011
01110
01111
10010
10011
10110

data
1011
1100
1101
1110
1111
IDLE
HALT
START
END
RESET
DEAD

symbol
10111
11010
11011
11100
11101
11111
00100
10001
01101
00111
00000

There are more than sixteen possible symbols; this allows for some symbols to be used for signaling rather
than data. IDLE, HALT, START, END and RESET are shown above, though there are others. These can be
used to include control and status information without fear of confusion with the data. Some combinations
of control symbols do lead to up to four 0-bits in sequence; HALT and RESET have two leading 0-bits.
10-Mbps and 100-Mbps Ethernet pads short packets up to the minimum packet size with 0-bytes, meaning
that the next protocol layer has to be able to distinguish between padding and actual 0-byte data. Although
100-Mbps Ethernet uses 4B/5B encoding, it does not make use of special non-data symbols for packet
padding. Gigabit Ethernet uses PAM-5 encoding (2.3 Gigabit Ethernet), and does use special non-data
symbols (inserted by the hardware) to pad packets; there is thus no ambiguity at the receiving end as to
where the data bytes ended.
The choice of 5-bit symbols for 4B/5B is in principle arbitrary; note however that for data from 0100 to
1101 we simply insert a 1 in the fourth position, and in the last two we insert a 0 in the fourth position. The
4.1 Encoding and Framing

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first four symbols (those with the most zeroes) follow no obvious pattern, though.

4.1.5 Framing
How does a receiver tell when one packet stops and the next one begins, to keep them from running together?
We have already seen the following techniques for addressing this framing problem: determining where
packets end:
Interpacket gaps (as in Ethernet)
4B/5B and special bit patterns
Putting a length field in the header would also work, in principle, but seems not to be widely used. One
problem with this technique is that restoring order after desynchronization can be difficult.
There is considerable overlap of framing with encoding; for example, the existence of non-data bit patterns
in 4B/5B is due to an attempt to solve the encoding problem; these special patterns can also be used as
unambiguous frame delimiters.
4.1.5.1 HDLC
HDLC (High-level Data Link Control) is a general link-level packet format used for a number of applications, including Point-to-Point Protocol (PPP) (which in turn is used for PPPoE PPP over Ethernet which
is how a great many Internet subscribers connect to their ISP), and Frame Relay, still used as the low-level
protocol for delivering IP packets to many sites via telecommunications lines. HDLC supports the following
two methods for frame separation:
HDLC over asynchronous links: byte stuffing
HDLC over synchronous links: bit stuffing
The basic encapsulation format for HDLC packets is to begin and end each frame with the byte 0x7E, or, in
binary, 0111 1110. The problem is that this byte may occur in the data as well; we must make sure we dont
misinterpret such a data byte as the end of the frame.
Asynchronous serial lines are those with some sort of start/stop indication, typically between bytes; such
lines tend to be slower. Over this kind of line, HDLC uses the byte 0x7D as an escape character. Any
data bytes of 0x7D and 0x7E are escaped by preceding them with an additional 0x7D. (Actually, they are
transmitted as 0x7D followed by (original_byte xor 0x20).) This strategy is fundamentally the same as that
used by C-programming-language character strings: the string delimiter is and the escape character is \.
Any occurrences of or \ within the string are escaped by preceding them with \.
Over synchronous serial lines (typically faster than asynchronous), HDLC generally uses bit stuffing. The
underlying bit encoding involves, say, the reverse of NRZI, in which transitions denote 0-bits and lack of
transitions denote 1-bits. This means that long runs of 1s are now the problem and runs of 0s are safe.
Whenever five consecutive 1-bits appear in the data, eg 011111, a 0-bit is then inserted, or stuffed, by the
transmitting hardware (regardless of whether or not the next data bit is also a 1). The HDLC frame byte of
0x7E = 0111 1110 thus can never appear as encoded data, because it contains six 1-bits in a row. If we had
0x7E in the data, it would be transmitted as 0111 11010.
The HDLC receiver knows that
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six 1-bits in a row marks the end of the packet

when five 1-bits in a row are seen, followed by a 0-bit, the 0-bit is removed
Example:
Data:
011110 0111110 01111110
Sent as: 011110 01111100 011111010 (stuffed bits in bold)
Note that bit stuffing is used by HDLC to solve two unrelated problems: the synchronization problem where
long runs of the same bit cause the receiver to lose count, and the framing problem, where the transmitted
bit pattern 0111 1110 now represents a flag that can never be mistaken for a data byte.
4.1.5.2 B8ZS
While insertion of an occasional extra bit or byte is no problem for data delivery, it is anathema to voice
engineers; extra bits upset the precise 64 Kbps DS-0 rate. As a result, long telecom lines prefer encodings
that, like 4B/5B, do not introduce timing fluctuations. Very long (electrical) lines also tend to require
encodings that guarantee a long-term average voltage level of 0 (versus 0.5 if half the bits are 1 v and half
are 0 v in NRZ); that is, the signal must have no DC component.
The AMI (Alternate Mark Inversion) technique eliminates the DC component by using three voltage levels,
nominally +1, 0 and -1; this ternary encoding is also known as bipolar. Zero bits are encoded by the 0
voltage, while 1-bits take on alternating values of +1 and -1 volts. Thus, the bits 011101 might be encoded
as 0,+1,-1,+1,0,-1, or, more compactly, 0++0. Over a long run, the +1s and the 1s cancel out.
Plain AMI still has synchronization problems with long runs of 0-bits. The solution used on North American
T1 lines (1.544 Mbps) is known as B8ZS, for bipolar with 8-zero substitution. The sender replaces any run
of 8 zero bits with a special bit-pattern, either 000+0+ or 000+0+. To decide which, the sender checks
to see if the previous 1-bit sent was +1 or 1; if the former, the first pattern is substituted, if the latter then the
second pattern is substituted. Either way, this leads to two instances of violation of the rule that consecutive
1-bits have opposite sign. For example, if the previous bit were +, the receiver sees

This double-violation is the clue to the receiver that the special pattern is to be removed and replaced with
the original eight 0-bits.

4.2 Time-Division Multiplexing

Classical circuit switching means a separate wire for each connection. This is still in common use for
residential telephone connections: each subscriber has a dedicated wire to the Central Office. But a separate
physical line for each connection is not a solution that scales well.

4.2 Time-Division Multiplexing

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Once upon a time it was not uncommon to link computers with serial lines, rather than packet networks.
This was most often done for file transfers, but telnet logins were also done this way. The problem with this
approach is that the line had to be dedicated to one application (or one user) at a time.
Packet switching naturally implements multiplexing (sharing) on links; the demultiplexer is the destination
address. Port numbers allow demultiplexing of multiple streams to same destination host.
There are other ways for multiple channels to share a single wire. One approach is frequency-division
multiplexing, or putting each channel on a different carrier frequency. Analog cable television did this.
Some fiber-optic protocols also do this, calling it wavelength-division multiplexing.
But perhaps the most pervasive alternative to packets is the voice telephone systems time division multiplexing, or TDM, sometimes prefixed with the adjective synchronous. The idea is that we decide on a
number of channels, N, and the length of a timeslice, T, and allow each sender to send over the channel for
time T, with the senders taking turns in round-robin style. Each sender gets to send for time T at regular
intervals of NT, thus receiving 1/N of the total bandwidth. The timeslices consume no bandwidth on headers
or addresses, although sometimes there is a small amount of space dedicated to maintaining synchronization
between the two endpoints. Here is a diagram of sending with N=8:

Note, however, that if a sender has nothing to send, its timeslice cannot be used by another sender. Because
so much data traffic is bursty, involving considerable idle periods, TDM has traditionally been rejected for
data networks.

4.2.1 T-Carrier Lines

TDM, however, works extremely well for voice networks. It continues to work when the timeslice T is
small, when packet-based approaches fail because the header overhead becomes unacceptable. Consider for
a moment the telecom Digital Signal hierarchy. A single digitized voice line in North America is one 8-bit
sample every 1/8,000 second, or 64 Kbps; this is known as a DS0 channel. A T1 line the lowest level of
the T-carrier hierarchy and known at the logical level as a DS1 line represents 24 DS0 lines multiplexed
via TDM, where each channel sends a single byte at a time. Thus, every 1/8,000 of a second a T1 line carries
24 bytes of user data, one byte per channel (plus one bit for framing), for a total of 193 bits. This gives a
raw line speed of 1.544 Mbps.
Note that the per-channel frame size here is a single byte. There is no efficient way to send single-byte
packets. The advantage to the single-byte approach is that it greatly reduces the latency across the line. The
biggest source of delay in packet-based digital voice lines is the packet fill time at the senders end: the
sender generates voice data at a rate of 8 bytes/ms, and a packet cannot be sent until it is full. For a 1KB
packet, thats about a quarter second. For standard Voice-over-IP or VoIP channels, RTP is used with 160
bytes of data sent every 20 ms; for ATM, a 48-byte packet is sent every 6 ms. But the fill-time delay for a
call sent over a T1 line is 0.125 ms, which is negligible (to be fair, 6 ms and even 20 ms turn out to be pretty
negligible in terms of call quality). The T1 one-byte-at-a-time strategy also means that T1 multiplexers
need to do essentially no buffering, which might have been important back in 1962 when T-carrier was
introduced.
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The next most common T-carrier / Digital Signal line is perhaps T3/DS3; this represents the TDM multiplexing of 28 DS1 signals. The problem is that some individual DS1s may run a little slow, so an elaborate pulse
stuffing protocol has been developed. This allows extra bits to be inserted at specific points, if necessary, in
such a way that the original component T1s can be exactly recovered even if there are clock irregularities.
The pulse-stuffing solution did not scale well, and so T-carrier levels past T3 were very rarely used.
While T-carrier was originally intended as a way of bundling together multiple DS0 channels on a single
high-speed line, it also allows providers to offer leased digital point-to-point links with data rates in almost
any multiple of the DS0 rate.

4.2.2 SONET
SONET stands for Synchronous Optical NETwork; it is the telecommunications industrys standard mechanism for very-high-speed TDM over optical fiber. While there is now flexibility regarding the the optical
part, the synchronous part is taken quite seriously indeed, and SONET senders and receivers all use very
precisely synchronized clocks (often atomic). The actual bit encoding is NRZI.
Due to the frame structure, below, the longest possible run of 0-bits is ~250 bits (~30 bytes), but is usually
much less. Accurate reception of 250 0-bits requires a clock accurate to within (at a minimum) one part in
500, which is generally within reach. This mechanism solves most of the clock-synchronization problem,
though SONET also has a resynchronization protocol in case the receiver gets lost.
The primary reason for SONETs accurate clocking, however, is not the clock-synchronization problem as
we have been using the term, but rather the problem of demultiplexing and remultiplexing multiple component bitstreams in a setting in which some of the streams may run slow. One of the primary design goals for
SONET was to allow such multiplexing without the need for pulse stuffing, as is used in the Digital Signal
hierarchy. SONET tributary streams are in effect not allowed to run slow (although SONET does provide
for occasional very small byte slips, below). Furthermore, as multiple SONET streams are demultiplexed
at a switching center and then remultiplexed into new SONET streams, synchronization means that none of
the streams falls behind or gets ahead.
The basic SONET format is known as STS-1. Data is organized as a 9x90 byte grid. The first 3 bytes of
each row (that is, the first three columns) form the frame header. Frames are not addressed; SONET is a
point-to-point protocol and a node sends a continuous sequence of frames to each of its neighbors. When the
frames reach their destination, in principle they need to be fully demultiplexed for the data to be forwarded
on. In practice, there are some shortcuts to full demultiplexing.

4.2 Time-Division Multiplexing

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The actual bytes sent are scrambled: the data is XORed with a standard, fixed pseudorandom pattern before
transmission. This introduces many 1-bits, on which clock resynchronization can occur, with a high degree
of probability.
There are two other special columns in a frame, each guaranteed to contain at least one 1-bit, so the maximum run of data bytes is limited to ~30; this is thus the longest run of possible 0s.
The first two bytes of each frame are 0xF628. SONETs frame-synchronization check is based on verifying
these byte values at the start of each frame. If the receiver is ever desynchronized, it begins a frame resynchronization procedure: the receiver searches for those 0xF628 bytes at regular 810-byte (6480-bit)
spacing. After a few frames with 0xF628 in the right place, the receiver is very sure it is looking at the
synchronization bytes and not at a data-byte position. Note that there is no evident byte boundary to a
SONET frame, so the receiver must check for 0xF628 beginning at every bit position.
SONET frames are transmitted at a rate of 8,000 frames/second. This is the canonical byte sampling rate
for standard voice-grade (DS0, or 64 Kbps) lines. Indeed, the classic application of SONET is to transmit
multiple DS0 voice calls using TDM: within a frame, each data byte position is given over to one voice
channel. The same byte position in consecutive frames constitutes one byte every 1/8000 seconds. The
basic STS-1 data rate of 51.84 Mbps is exactly 810 bytes/frame 8 bits/byte 8000 frames/sec.
To a customer who has leased a SONET-based channel to transmit data, a SONET link looks like a very fast
bitstream. There are several standard ways of encoding data packets over SONET. One is to encapsulate the
data as ATM cells, and then embed the cells contiguously in the bitstream. Another is to send IP packets
encoded in the bitstream using HDLC-like bit stuffing, which means that the SONET bytes and the IP bytes
may no longer correspond. The advantage of HDLC encoding is that it makes SONET re-synchronization
vanishingly infrequent. Most IP backbone traffic today travels over SONET links.
Within the 990-byte STS-1 frame, the payload envelope is the 987 region nominally following the three
header columns; this payload region has its own three reserved columns meaning that there are 84 columns
(984 bytes) available for data. This 987-byte payload envelope can float within the physical 990byte frame; that is, if the input frames are running slow then the output physical frames can be transmitted
at the correct rate by letting the payload frames slip backwards, one byte at a time. Similarly, if the input
frames are arriving slightly too fast, they can slip forwards by up to one byte at a time; the extra byte is
stored in a reserved location in the three header columns of the 990 physical frame.
Faster SONET streams are made by multiplexing slower ones. The next step up is STS-3, an STS-3 frame
is three STS-1 frames, for 9270 bytes. STS-3 (or, more properly, the physical layer for STS-3) is also
called OC-3, for Optical Carrier. Beyond STS-3, faster lines are multiplexed combinations of four of the
next-slowest lines. Here are some of the higher levels:
STS
STS-1
STS-3
STS-12
STS-48
STS-192
STS-768

STM
STM-0
STM-1
STM-4
STM-16
STM-64
STM-256

bandwidth
51.84 Mbps
155.52 Mbps
622.08 Mbps (=12*51.84, exactly)
2.488 Gbps
9.953 Gbps
39.8 Gbps

Faster SONET lines have been defined, but a simpler way to achieve very high data rates over optical fiber is
to use wavelength-division multiplexing (that is, frequency-division multiplexing at optical frequencies);
this means we have separate SONET channels at different wavelengths of light.

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SONET provides a wide variety of leasing options at various bandwidths. High-volume customers can lease
an entire STS-1 or larger unit. Alternatively, the 84 columns of an STS-1 frame can be divided into seven
virtual tributary groups, each of twelve columns; these groups can be leased individually or in multiples,
or be further divided into as few as three columns (which works out to be just over the T1 data rate).
4.2.2.1 Other Optical Fiber

Optical Fiber and Lightning

One of the advantages of of optical fiber (particularly at mountaintop observatories) is its resistance to
damage and interference from lightning. Some fiber-optic cables do, however, have metal jackets to
add strength or to resist animals; lightning resistance must be researched carefully.
SONET is primarily, though not exclusively, used by telecommunications carriers. There are also multiple
fiber-optic alternatives for smaller-scale operations that generally do not involve SONETs rigid synchronization mechanisms and voice-oriented framing. These are often part of the Ethernet family although they
may have little except their bitrate in common with one another or with Ethernet over copper wire. Another
standard that supports optical fiber links is so-called Fibre Channel, though that too also now supports
copper.
Distances supported by fiber-optic Ethernet range from hundreds of meters to tens of kilometers. Generally,
the long-haul links require the use of full duplex to avoid the collision-detection (slot time) requirement that
a sender continue to transmit for one full Ethernet-segment RTT, but full duplex is common at high Ethernet
speeds even for short links. For 100 Mbps Ethernet, fiber-optic standards include 100BASE-FX, 100BASESX, 100BASE-LX and 100BASE-BX. The latter supports distances up to 40 km; the limiting factor is the
need for signal regeneration. 1000-Gbit Ethernet optical standards include 1000BASE-SX, 1000BASE-LX,
1000BASE-BX and 1000BASE-EX; the latter again supports up to 40 km. These forms of Ethernet are
often deployed in residential fiber Internet connections, although in some cases the last hundred meters or
so may still involve copper.
At speeds of 10 Gbps, long-range optical fiber alternatives include 10GBASE-ER and 10GBASE-ZR, supportking 40 and 80 km respectively; 10GBASE-ZR is based on SONET STM-64 standards. It is worth
noting that fiber is often used for short links as well at 1 Gbps and 10 Gbps speeds; at 10 Gbps some copperlink standards support distances of only a few meters. As of 2014, several 10-Gbps Ethernet physical-layer
standards come from industry consortiums rather than the IEEE.

4.3 Epilog
This completes our discussion of common physical links. Perhaps the main takeaway point is that transmitting bits over any distance is not quite as simple as it may appear; simple NRZ transmission is not effective.

4.4 Exercises
1. What is encoded by the following NRZI signal? The first two bits are shown.

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2. Argue that sending 4 0-bits via NRZI requires a clock accurate to within 1 part in 8. Assume that the
receiver resynchronizes its clock whenever a 1-bit transition is received, but that otherwise it attempts to
sample a bit in the middle of the bits timeslot.
3.(a) What bits are encoded by the following Manchester-encoded sequence?

(b). Why is there no ambiguity as to whether the first transition is a clock transition or a data (1-bit)
transition?
(c). Give an example of a signal pattern consisting of an NRZI encoding of 0-bits and 1-bits that does not
contain two consecutive 0-bits and which is not a valid Manchester encoding of data. Such a pattern could
thus could be used as a special non-data marker.

4. What three ASCII letters (bytes) are encoded by the following 4B/5B pattern? (Be careful about uppercase
vs lowercase.)
010110101001110101010111111110
5.(a) Suppose a device is forwarding SONET STS-1 frames. How much clock drift, as a percentage, on
the incoming line would mean that the output payload envelopes must slip backwards by one byte per three
physical frames? | (b). In 4.2.2 SONET it was claimed that sending 250 0-bits required a clock accurate to
within 1 part in 500. Describe how a SONET clock might meet the requirement of part (a) above, and yet
fail at this second requirement. (Hint: in part (a) the requirement is a long-term average).

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5 PACKETS

In this chapter we address a few abstract questions about packets, and take a close look at transmission
times. We also consider how big packets should be, and how to detect transmission errors. These issues are
independent of any particular set of protocols.

5.1 Packet Delay

There are several contributing sources to the delay encountered in transmitting a packet. On a LAN, the most
significant is usually what we will call bandwidth delay: the time needed for a sender to get the packet onto
the wire. This is simply the packet size divided by the bandwidth, after everything has been converted to
common units (either all bits or all bytes). For a 1500-byte packet on 100 Mbps Ethernet, the bandwidth
delay is 12,000 bits / (100 bits/sec) = 120 sec.
There is also propagation delay, relating to the propagation of the bits at the speed of light (for the transmission medium in question). This delay is the distance divided by the speed of light; for 1,000 m of Ethernet
cable, with a signal propagation speed of about 230 m/sec, the propagation delay is about 4.3 sec. That is,
if we start transmitting the 1500 byte packet of the previous paragraph at time T=0, then the first bit arrives
at a destination 1,000 m away at T = 4.3 sec, and the last bit is transmitted at 120 sec, and the last bit
arrives at T = 124.3 sec.
Minimizing Delay
Back in the last century, gamers were sometimes known to take advantage of players with slow (as in
dialup) links; an opponent could be eliminated literally before he or she could respond. As an updated
take on this, some financial-trading firms have set up microwave-relay links between trading centers,
say New York and Chicago, in order to reduce delay. In computerized trading, milliseconds count. A
direct line of sight from New York to Chicago which we round off to 1200 km takes about 4 ms
in air, where signals propagate at essentially the speed of light c = 300 km/ms. But fiber is slower;
even an absolutely straight run would take 6 ms at glass fibers propagation speed of 200 km/ms. In
the presence of high-speed trading, this 2 ms savings is of considerable financial significance.
Bandwidth delay, in other words, tends to dominate within a LAN.
But as networks get larger, propagation delay begins to dominate. This also happens as networks get faster:
bandwidth delay goes down, but propagation delay remains unchanged.
An important difference between bandwidth delay and propagation delay is that bandwidth delay is proportional to the amount of data sent while propagation delay is not. If we send two packets back-to-back, then
the bandwidth delay is doubled but the propagation delay counts only once.
The introduction of switches leads to store-and-forward delay, that is, the time spent reading in the entire
packet before any of it can be retransmitted. Store-and-forward delay can also be viewed as an additional
bandwidth delay for the second link.

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Finally, a switch may or may not also introduce queuing delay; this will often depend on competing traffic.
We will look at this in more detail in 14 Dynamics of TCP Reno, but for now note that a steady queuing delay
(eg due to a more-or-less constant average queue utilization) looks to each sender more like propagation
delay than bandwidth delay, in that if two packets are sent back-to-back and arrive that way at the queue,
then the pair will experience only a single queuing delay.

5.1.1 Delay examples

Case 1: A

Propagation delay is 40 sec

Bandwidth is 1 byte/sec (1 mB/sec, 8 Mbit/sec)
Packet size is 200 bytes (200 sec bandwidth delay)
Then the total one-way transmit time is 240 sec = 200 sec + 40 sec
Case 2: A

Like the previous example except that the propagation delay is increased to 4 ms
The total transmit time is now 4200 sec = 200 sec + 4000 sec
Case 3: A

We now have two links, each with propagation delay 40 sec; bandwidth and packet size as in
Case 1
The total transmit time for one 200-byte packet is now 480 sec = 240 + 240. There are two propagation
delays of 40 sec each; A introduces a bandwidth delay of 200 sec and R introduces a store-and-forward
delay (or second bandwidth delay) of 200 sec.
Case 4: A

The same as 3, but with data sent as two 100-byte packets

The total transmit time is now 380 sec = 3x100 + 2x40. There are still two propagation delays, but there is
only 3/4 as much bandwidth delay because the transmission of the first 100 bytes on the second link overlaps
with the transmission of the second 100 bytes on the first link.

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These ladder diagrams represent the full transmission; a snapshot state of the transmission at any one
instant can be obtained by drawing a horizontal line. In the middle, case 3, diagram, for example, at no
instant are both links active. Note that sending two smaller packets is faster than one large packet. We
expand on this important point below.
Now let us consider the situation when the propagation delay is the most significant component. The crosscontinental US roundtrip delay is typically around 50-100 ms (propagation speed 200 km/ms in cable, 5,00010,000 km cable route, or about 3-6000 miles); we will use 100 ms in the examples here. At 1.0 Mbit, 100ms
is about 12KB, or eight full-sized Ethernet packets. At this bandwidth, we would have four packets and four
returning ACKs strung out along the path. At 1.0 Gbit, in 100ms we can send 12,000 KB, or 800 Ethernet
packets, before the first ACK returns.
At most non-LAN scales, the delay is typically simplified to the round-trip time, or RTT: the time between
sending a packet and receiving a response.
Different delay scenarios have implications for protocols: if a network is bandwidth-limited then protocols
are easier to design. Extra RTTs do not cost much, so we can build in a considerable amount of back-andforth exchange. However, if a network is delay-limited, the protocol designer must focus on minimizing
extra RTTs. As an extreme case, consider wireless transmission to the moon (0.3 sec RTT), or to Jupiter (1
hour RTT).
At my home I formerly had satellite Internet service, which had a roundtrip propagation delay of ~600 ms.
This is remarkably high when compared to purely terrestrial links.
When dealing with reasonably high-bandwidth large-scale networks (eg the Internet), to good approximation most of the non-queuing delay is propagation, and so bandwidth and total delay are effectively
independent. Only when propagation delay is small are the two interrelated. Because propagation delay
dominates at this scale, we can often make simplifications when diagramming. In the illustration below, A
sends a data packet to B and receives a small ACK in return. In (a), we show the data packet traversing
several switches; in (b) we show the data packet as if it were sent along one long unswitched link, and in (c)
we introduce the idealization that bandwidth delay (and thus the width of the packet line) no longer matters.
(Most later ladder diagrams in this book are of this type.)

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The bandwidth delay product (usually involving round-trip delay, or RTT), represents how much we can
send before we hear anything back, or how much is pending in the network at any one time if we send
continuously. Note that, if we use RTT instead of one-way time, then half the pending packets will be
returning ACKs. Here are a few values
RTT
1 ms
100 ms
100 ms

bandwidth
10 Mbps
1.5 Mbps
600 Mbps

bandwidth delay
1.2 KB
20 KB
8,000 KB

5.2 Packet Delay Variability

For many links, the bandwidth delay and the propagation delay are rigidly fixed quantities, the former by the
bandwidth and the latter by the speed of light. This leaves queuing delay as the major source of variability.
This state of affairs lets us define RTTnoLoad to be the time it takes to transmit a packet from A to B, and
receive an acknowledgment back, with no queuing delay.
While this is often a reasonable approximation, it is not necessarily true that RTTnoLoad is always a fixed
quantity. There are several possible causes for RTT variability. On Ethernet and Wi-Fi networks there is an
initial contention period before transmission actually begins. Although this delay is related to waiting for
other senders, it is not exactly queuing delay, and a packet may encounter considerable delay here even if it
ends up being the first to be sent. For Wi-Fi in particular, the uncertainty introduced by collisions into packet
delivery times even with no other senders competing can complicate higher-level delay measurements.
It is also possible that different packets are routed via slightly different paths, leading to (hopefully) minor
variations in travel time, or are handled differently by different queues of a parallel-processing switch.
A links bandwidth, too, can vary dynamically. Imagine, for example, a T1 link comprised of the usual
24 DS0 channels, in which all channels not currently in use by voice calls are consolidated into a single
data channel. With eight callers, the data bandwidth would be cut by a third from 24 DS0 to 16 DS0.
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Alternatively, perhaps routers are allowed to reserve a varying amount of bandwidth for high-priority traffic,
depending on demand, and so the bandwidth allocated to the best-effort traffic can vary. Perceived link
bandwidth can also vary over time if packets are compressed at the link layer, and some packets are able to
be compressed more than others.
Finally, if mobile nodes are involved, then the distance and thus the propagation delay can change. This can
be quite significant if one is communicating with a wireless device that is being taken on a cross-continental
road trip.
Despite these sources of fluctuation, we will usually assume that RTTnoLoad is fixed and well-defined, especially when we wish to focus on the queuing component of delay.

5.3 Packet Size

How big should packets be? Should they be large (eg 64 KB) or small (eg 48 bytes)?
The Ethernet answer to this question had to do with equitable sharing of the line: large packets would not
allow other senders timely access to transmit. In any network, this issue remains a concern.
On the other hand, large packets waste a smaller percentage of bandwidth on headers. However, in most of
the cases we will consider, this percentage does not exceed 10% (the VoIP/RTP example in 1.3 Packets is
an exception).
It turns out that if store-and-forward switches are involved, smaller packets have much better throughput.
The links on either side of the switch can be in use simultaneously, as in Case 4 of 5.1.1 Delay examples.
This is a very real effect, and has put a damper on interest in support for IP jumbograms. The ATM
protocol (intended for both voice and data) pushes this to an extreme, with packets with only 48 bytes of
data and 5 bytes of header.
As an example of this, consider a path from A to B with four switches and five links:
A

R1

R2

R3

R4

Suppose we send either one big packet or five smaller packets. The relative times from A to B are illustrated
in the following figure:

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The point is that we can take advantage of parallelism: while the R4B link above is handling packet 1,
the R3R4 link is handling packet 2 and the R2R3 link is handling packet 3 and so on. The five smaller
packets would have five times the header capacity, but as long as headers are small relative to the data, this
is not a significant issue.
The sliding-windows algorithm, used by TCP, uses this idea as a continuous process: the sender sends a
continual stream of packets which travel link-by-link so that, in the full-capacity case, all links may be in
use at all times.

5.3.1 Error Rates and Packet Size

Packet size is also influenced, to a modest degree, by the transmission error rate. For relatively high error
rates, it turns out to be better to send smaller packets, because when an error does occur then the entire
packet containing it is lost.
For example, suppose that 1 bit in 10,000 is corrupted, at random, so the probability that a single bit is
transmitted correctly is 0.9999 (this is much higher than the error rates encountered on real networks). For a
1000-bit packet, the probability that every bit in the packet is transmitted correctly is (0.9999)1000 , or about
90%. For a 10,000-bit packet the probability is (0.9999)10,000 37%. For 20,000-bit packets, the success
rate is below 14%.
Now suppose we have 1,000,000 bits to send, either as 1000-bit packets or as 20,000-bit packets. Nominally
this would require 1,000 of the smaller packets, but because of the 90% packet-success rate we will need to
retransmit 10% of these, or 100 packets. Some of the retransmissions may also be lost; the total number of
packets we expect to need to send is about 1,000/90% 1,111, for a total of 1,111,000 bits sent. Next, let
us try this with the 20,000-bit packets. Here the success rate is so poor that each packet needs to be sent on

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average seven times; lossless transmission would require 50 packets but we in fact need 750 = 350 packets,
or 7,000,000 bits.
Moral: choose the packet size small enough that most packets do not encounter errors.
To be fair, very large packets can be sent reliably on most cable links (eg TDM and SONET). Wireless,
however, is more of a problem.

5.3.2 Packet Size and Real-Time Traffic

There is one other concern regarding excessive packet size. As we shall see in 19 Quality of Service, it
is common to commingle bulk traffic on the same links with real-time traffic. It is straightforward to give
priority to the real-time traffic in such a mix, meaning that a router does not begin forwarding a bulk-traffic
packet if there are any real-time packets waiting (we do need to be sure in this case that real-time traffic
will not amount to so much as to starve the bulk traffic). However, once a bulk-traffic packet has begun
transmission, it is impractical to interrupt it.
Therefore, one component of any maximum-delay bound for real-time traffic is the transmission time for
the largest bulk-traffic packet; we will call this the largest-packet delay. As a practical matter, most IPv4
packets are limited to the maximum Ethernet packet size of 1500 bytes, but IPv6 has an option for so-called
jumbograms up to 2 MB in size. Transmitting one such packet on a 100 Mbps link takes about 1/6 of a
second, which is likely too large for happy coexistence with real-time traffic.

5.4 Error Detection

The basic strategy for packet error detection is to add some extra bits formally known as an error-detection
code that will allow the receiver to determine if the packet has been corrupted in transit. A corrupted packet
will then be discarded by the receiver; higher layers do not distinguish between lost packets and those never
received. While packets lost due to bit errors occur much less frequently than packets lost due to queue
overflows, it is essential that data be received accurately.
Intermittent packet errors generally fall into two categories: low-frequency bit errors due to things like
cosmic rays, and interference errors, typically generated by nearby electrical equipment. Errors of the latter
type generally occur in bursts, with multiple bad bits per packet. Occasionally, a malfunctioning network
device will introduce bursty errors as well.
Networks v Refrigerators
At Loyola we once had a workstation used as a mainframe terminal that kept losing its connection.
We eventually noticed that the connection dropped every time the office refrigerator kicked on. Sure
enough, the cable ran directly behind the fridge; rerouting it solved the problem.
The simplest error-detection mechanism is a single parity bit; this will catch all one-bit errors. There is,
however, no straightforward generalization to N bits! That is, there is no N-bit error code that catches all
N-bit errors; see exercise 9.
The so-called Internet checksum, used by IP, TCP and UDP, is formed by taking the ones-complement sum
of the 16-bit words of the message. Ones-complement is an alternative way of representing signed integers
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in binary; if one adds two positive integers and the sum does not overflow the hardware word size, then
ones-complement and the now-universal twos-complement are identical. To form the ones-complement sum
of 16-bit words A and B, first take the ordinary twos-complement sum A+B. Then, if there is an overflow
bit, add it back in as low-order bit. Thus, if the word size is 4 bits, the ones-complement sum of 0101 and
0011 is 1000 (no overflow). Now suppose we want the ones-complement sum of 0101 and 1100. First we
take the exact sum and get 1|0001, where the leftmost 1 is an overflow bit past the 4-bit wordsize. Because
of this overflow, we add this bit back in, and get 0010.
The ones-complement numeric representation has two forms for zero: 0000 and 1111 (it is straightforward
to verify that any 4-bit quantity plus 1111 yields the original quantity; in twos-complement notation 1111
represents -1, and an overflow is guaranteed, so adding back the overflow bit cancels the -1 and leaves us
with the original number). It is a fact that the ones-complement sum is never 0000 unless all bits of all the
summands are 0; if the summands add up to zero by coincidence, then the actual binary representation will
be 1111. This means that we can use 0000 in the checksum to represent checksum not calculated, which
the UDP protocol used to permit.
Ones-complement
Long ago, before Loyola had any Internet connectivity, I wrote a primitive UDP/IP stack to allow me
to use the Ethernet to back up one machine that did not have TCP/IP to another machine that did. We
used private IP addresses of the form 10.0.0.x. I set as many header fields to zero as I could. I paid
no attention to how to implement ones-complement addition; I simply used twos-complement, for the
IP header only, and did not use a UDP checksum at all. Hey, it worked.
Then we got a real Class B address block 147.126.0.0/16, and changed IP addresses. My software no
longer worked. It turned out that, in the original version, the IP header bytes were all small enough
that when I added up the 16-bit words there were no carries, and so ones-complement was the same as
twos-complement. With the new addresses, this was no longer true. As soon as I figured out how to
implement ones-complement addition properly, my backups worked again.
There is another way to look at the (16-bit) ones-complement sum: it is in fact the remainder upon dividing
the message (seen as a very long binary number) by 216 - 1. This is similar to the decimal casting out nines
rule: if we add up the digits of a base-10 number, and repeat the process until we get a single digit, then that
digit is the remainder upon dividing the original number by 10-1 = 9. The analogy here is that the message
is looked at as a very large number written in base-216 , where the digits are the 16-bit words. The process
of repeatedly adding up the digits until we get a single digit amounts to taking the ones-complement
sum of the words.
A weakness of any error-detecting code based on sums is that transposing words leads to the same sum, and
the error is not detected. In particular, if a message is fragmented and the fragments are reassembled in the
wrong order, the ones-complement sum will likely not detect it.
While some error-detecting codes are better than others at detecting certain kinds of systematic errors (for
example, CRC, below, is usually better than the Internet checksum at detecting transposition errors), ultimately the effectiveness of an error-detecting code depends on its length. Suppose a packet P1 is corrupted
randomly into P2, but still has its original N-bit error code EC(P1). This N-bit code will fail to detect the
error that has occurred if EC(P2) is, by chance, equal to EC(P1). The probability that two random N-bit
codes will match is 1/2N (though a small random change in P1 might not lead to a uniformly distributed
random change in EC(P1); see the tail end of the CRC section below).

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This does not mean, however, that one packet in 2N will be received incorrectly, as most packets are errorfree. If we use a 16-bit error code, and only 1 packet in 100,000 is actually corrupted, then the rate at which
corrupted packets will sneak by is only 1 in 100,000 65536, or about one in 6 109 . If packets are 1500
bytes, you have a good chance (90+%) of accurately transferring a terabyte, and a 37% chance (1/e) at ten
terabytes.

5.4.1 Cyclical Redundancy Check: CRC

The CRC error code is based on long division of polynomials, which tends to sound complicated but which
in fact reduces all the long-division addition/subtraction operations to the much-simpler XOR operation.
We treat the message, in binary, as a giant polynomial m(X), using the bits of the message as successive
coefficients (eg 10011011 = X7 + X4 + X3 + X + 1). We standardize a divisor polynomial p(X) of degree 32
(at least for CRC-32 codes); the full specification of a given CRC code requires giving this polynomial. We
divide m(X) by p(X); our checksum is the remainder r(X), of maximum degree 31 (that is, 32 bits).
This is a reasonably secure hash against real-world network corruption, in that it is very hard for systematic errors to result in the same hash code. However, CRC is not secure against intentional corruption;
given msg1, there are straightforward mathematical means for tweaking the last bytes of msg2 so that so
CRC(msg1) = CRC(msg2)
As an example of CRC, suppose that the CRC divisor is 1011 (making this a CRC-3 code) and the message
is 0110 0001 1110. Here is the division:
1110 0110
1011

0110 0001 1100

101 1
011 10
10 11
01 010
1 011
0 0011 11
10 11
01 000
1 011
0 0110

The remainder, at the bottom, is 110; this is the CRC code.

CRC is easily implemented in hardware, using bit-shifting registers. Fast software implementations are also
possible, usually involving handling the bits one byte at a time, with a precomputed lookup table with 256
entries.
If we randomly change enough bits in packet P1 to create P2, then CRC(P1) and CRC(P2) are effectively
independent random variables, with probability of a match 1 in 2N where N is the CRC length. However, if
we change just a few bits then the change is not so random. In particular, for many CRC codes (that is, for
many choices of the underlying polynomial p(X)), changing up to three bits in P1 to create a new message
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P2 guarantees that CRC(P1) = CRC(P2). For the Internet checksum, this is not guaranteed even if we know
only two bits were changed.
Finally, there are also secure hashes, such as MD-5 and SHA-1 and their successors. Nobody knows
(or admits to knowing) how to produce two messages with same hash here. However, these secure-hash
codes are generally not used in network error-correction as they take considerable time to compute; they are
generally used only for secure authentication and other higher-level functions.

5.4.2 Error-Correcting Codes

If a link is noisy, we can add an error-correction code (also called forward error correction) that allows the
receiver in many cases to figure out which bits are corrupted, and fix them. This has the effect of improving
the bit error rate at a cost of reducing throughput. Error-correcting codes tend to involve many more bits than
are needed for error detection. Typically, if a communications technology proves to have an unacceptably
high bit-error rate (such as wireless), the next step is to introduce an error-correcting code to the protocol.
This generally reduces the virtual bit-error rate (that is, the error rate as corrected) to acceptable levels.
Perhaps the easiest error-correcting code to visualize is 2-D parity, for which we need O(N1/2 ) additional
bits. We take NN data bits and arrange them into a square; we then compute the parity for every column,
for every row, and for the entire square; this is 2N+1 extra bits. Here is a diagram with N=4, and with even
parity; the column-parity bits (in blue) are in the bottom (fifth) row and the row-parity bits (also in blue)
are in the rightmost (fifth) column. The parity bit for the entire 44 data square is the light-blue bit in the
bottom right corner.

Now suppose one bit is corrupted; for simplicity, assume it is one of the data bits. Then exactly one columnparity bit will be incorrect, and exactly one row-parity bit will be incorrect. These two incorrect bits mark
the column and row of the incorrect data bit, which we can then flip to the correct state.
We can make N large, but an essential requirement here is that there be only a single corrupted bit per square.
We are thus likely either to keep N small, or to choose a different code entirely that allows correction of
multiple bits. Either way, the addition of error-correcting codes can easily increase the size of a packet
significantly; some codes double or even triple the total number of bits sent.
The Hamming code is another popular error-correction code; it adds O(log N) additional bits, though if N is
large enough for this to be a material improvement over the O(N1/2 ) performance of 2-D parity then errors
must be very infrequent. If we have 8 data bits, let us number the bit positions 0 through 7. We then write
each bits position as a binary value between 000 and 111; we will call these the position bits of the given
data bit. We now add four code bits as follows:
1. a parity bit over all 8 data bits

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2. a parity bit over those data bits for which the first digit of the position bits is 1 (these are positions 4,
5, 6 and 7)
3. a parity bit over those data bits for which the second digit of the position bits is 1 (these are positions
010, 011, 110 and 111, or 2, 3, 6 and 7)
4. a parity bit over those data bits for which the third digit of the position bits is 1 (these are positions
001, 011, 101, 111, or 1, 3, 5 and 7)
We can tell whether or not an error has occurred by the first code bit; the remaining three code bits then tell
us the respective three position bits of the incorrect bit. For example, if the #2 code bit above is correct, then
the first digit of the position bits is 0; otherwise it is one. With all three position bits, we have identified the
incorrect data bit.
As a concrete example, suppose the data word is 10110010. The four code bits are thus
1. 0, the (even) parity bit over all eight bits
2. 1, the parity bit over the second half, 10110010
3. 1, the parity bit over the bold bits: 10110010
4. 1, the parity bit over these bold bits: 10110010
If the received data+code is now 10111010 0111, with the bold bit flipped, then the fact that the first code
bit is wrong tells the receiver there was an error. The second code bit is also wrong, so the first bit of the
position bits must be 1. The third code bit is right, so the second bit of the position bits must be 0. The
fourth code bit is also right, so the third bit of the position bits is 0. The position bits are thus binary 100, or
4, and so the receiver knows that the incorrect bit is in position 4 (counting from 0) and can be flipped to the
correct state.

5.5 Epilog
The issues presented here are perhaps not very glamorous, and often play a supporting, behind-the-scenes
role in protocol design. Nonetheless, their influence is pervasive; we may even think of them as part of the
underlying physics of the Internet.
As the early Internet became faster, for example, and propagation delay became the dominant limiting
factor, protocols were often revised to limit the number of back-and-forth exchanges. A classic example is
the Simple Mail Transport Protocol (SMTP), amended by RFC 1854 to allow multiple commands to be sent
together called pipelining instead of individually.
While there have been periodic calls for large-packet support in IPv4, and IPv6 protocols exist for jumbograms in excess of a megabyte, these are very seldom used, due to the store-and-forward costs of large
packets as described in 5.3 Packet Size.
Almost every LAN-level protocol, from Ethernet to Wi-Fi to point-to-point links, incorporates an errordetecting code chosen to reflect the underlying transportation reliability. Ethernet includes a 32-bit CRC
code, for example, while Wi-Fi includes extensive error-correcting codes due to the noisier wireless environment. The Wi-Fi fragmentation option (3.3.2.3 Wi-Fi Fragmentation) is directly tied to 5.3.1 Error
Rates and Packet Size.

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5.6 Exercises
1. Suppose a link has a propagation delay of 20 sec and a bandwidth of 2 bytes/sec.

(a). How long would it take to transmit a 600-byte packet over such a link?
(b). How long would it take to transmit the 600-byte packet over two such links, with a store-and-forward
switch in between?

2. Suppose the path from A to B has a single switch S in between: A

delay of 60 sec and a bandwidth of 2 bytes/sec.

B. Each link has a propagation

(a). How long would it take to send a single 600-byte packet from A to B?
(b). How long would it take to send two back-to-back 300-byte packets from A to B?
(c). How long would it take to send three back-to-back 200-byte packets from A to B?

3. Repeat parts (a) and (b) of the previous exercise, except change the per-link propagation delay from 60
sec to 600 sec.
4. Again suppose the path from A to B has a single switch S in between: A
width and propagation delays are as follows:
link
A S
S B

bandwidth
5 bytes/sec
3 bytes/sec

B. The per-link band-

propagation delay
24 sec
13 sec

(a). How long would it take to send a single 600-byte packet from A to B?
(b). How long would it take to send two back-to-back 300-byte packets from A to B? Note that, because
the S B link is slower, packet 2 arrives at S from A well before S has finished transmitting packet 1 to B.

5. Suppose in the previous exercise, the AS link has the smaller bandwidth of 3 bytes/sec and the SB
link has the larger bandwidth of 5 bytes/sec. Now how long does it take to send two back-to-back 300-byte
packets from A to B?
6. Suppose we have five links, A R1
R2 R3
R4
bytes/ms. Assume we model the per-link propagation delay as 0.

B. Each link has a bandwidth of 100

(a). How long would it take a single 1500-byte packet to go from A to B?

(b). How long would it take five consecutive 300-byte packets to go from A to B?

The diagram in 5.3 Packet Size may help.

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7. Suppose there are N equal-bandwidth links on the path between A and B, as in the diagram below, and
we wish to send M consecutive packets.
S1

...

SN-1

Let BD be the bandwidth delay of a single packet on a single link, and let PD be the propagation delay on a
single link. Show that the total bandwidth delay is (M+N-1)BD, and the total propagation delay is NPD.
Hint: When does the Mth packet leave A? What is its total transit time to B? Why do no packets have to wait
at any Si for the completion of the transmission of an earlier packet?
8. Repeat the analysis in 5.3.1 Error Rates and Packet Size to compare the probable total number of bytes
that need to be sent to transmit 107 bytes using

(a). 1,000-byte packets

(b). 10,000-byte packets

Assume the bit error rate is 1 in 16 105 , making the error rate per byte about 1 in 2 105 .
9. In the text it is claimed there is no N-bit error code that catches all N-bit errors for N2 (for N=1, a
parity bit works). Prove this claim for N=2. Hint: pick a length M, and consider all M-bit messages with a
single 1-bit. Any such message can be converted to any other with a 2-bit error. Show, using the Pigeonhole
Principle, that for large enough M two messages m1 and m2 must have the same error code, that is, e(m1 ) =
e(m2 ). If this occurs, then the error code fails to detect the error that converted m1 into m2 .
10. In the description in the text of the Internet checksum, the overflow bit was added back in after each
ones-complement addition. Show that the same final result will be obtained if we add up the 16-bit words
using 32-bit twos-complement arithmetic (the normal arithmetic on all modern hardware), and then add the
upper 16 bits of the sum to the lower 16 bits. (If there is an overflow at this last step, we have to add that
back in as well.)
11. Suppose a message is 110010101. Calculate the CRC-3 checksum using the polynomial X3 + 1, that is,
find the 3-bit remainder using divisor 1001.
12. The CRC algorithm presented above requires that we process one bit at a time. It is possible to do the
algorithm N bits at a time (eg N=8), with a precomputed lookup table of size 2N . Complete the steps in the
following description of this strategy for N=3 and polynomial X3 + X + 1, or 1011.
13. Consider the following set of bits sent with 2-D even parity; the data bits are in the 44 upper-left block
and the parity bits are in the rightmost column and bottom row. Which bit is corrupted?
1

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14. (a) Show that 2-D parity can detect any three errors.
(b). Find four errors that cannot be detected by 2-D parity.
(c). Show that that 2-D parity cannot correct all two-bit errors. Hint: put both bits in the same row or
column.

15. Each of the following 8-bit messages with 4-bit Hamming code contains a single error. Correct the
message.

(a). 10100010 0111

(b). 10111110 1011

16. (a) What happens in 2-D parity if the corrupted bit is in the parity column or parity row?
(b). In the following 8-bit message with 4-bit Hamming code, there is an error in the code portion. How
can this be determined?
1001 1110 0100

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In this chapter we take a general look at how to build reliable data-transport layers on top of unreliable
lower layers. This is achieved through a retransmit-on-timeout policy; that is, if a packet is transmitted
and there is no acknowledgment received during the timeout interval then the packet is resent. As a class,
protocols where one side implements retransmit-on-timeout are known as ARQ protocols, for Automatic
Repeat reQuest.
In addition to reliability, we also want to keep as many packets in transit as the network can support. The
strategy used to achieve this is known as sliding windows. It turns out that the sliding-windows algorithm
is also the key to managing congestion; we return to this in 13 TCP Reno and Congestion Management.
The End-to-End principle, 12.1 The End-to-End Principle, suggests that trying to achieve a reliable transport layer by building reliability into a lower layer is a misguided approach; that is, implementing reliability
at the endpoints of a connection as is described here is in fact the correct mechanism.

6.1 Building Reliable Transport: Stop-and-Wait

Retransmit-on-timeout generally requires sequence numbering for the packets, though if a network path is
guaranteed not to reorder packets then it is safe to allow the sequence numbers to wrap around surprisingly
quickly (for stop-and-wait, a single-bit sequence number will work). However, as the no-reordering hypothesis does not apply to the Internet at large, we will assume conventional numbering. Data[N] will be the Nth
data packet, acknowledged by ACK[N].
In the stop-and-wait version of retransmit-on-timeout, the sender sends only one outstanding packet at a
time. If there is no response, the packet may be retransmitted, but the sender does not send Data[N+1] until
it has received ACK[N]. Of course, the receiving side will not send ACK[N] until it has received Data[N];
each side has only one packet in play at a time. In the absence of packet loss, this leads to the following:

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6.1.1 Packet Loss

Lost packets, however, are a reality. The left half of the diagram below illustrates a lost Data packet, where
the sender is the host sending Data and the Receiver is the host sending ACKs. The receiver is not aware of
the loss; it sees Data[N] as simply slow to arrive.

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The right half of the diagram, by comparison, illustrates the case of a lost ACK. The receiver has received
a duplicate Data[N]. We have assumed here that the receiver has implemented a retransmit-on-duplicate
strategy, and so its response upon receipt of the duplicate Data[N] is to retransmit ACK[N].
As a final example, note that it is possible for ACK[N] to have been delayed (or, similarly, for the first
Data[N] to have been delayed) longer than the timeout interval. Not every packet that times out is actually
lost!

In this case we see that, after sending Data[N], receiving a delayed ACK[N] (rather than the expected
ACK[N+1]) must be considered a normal event.
In principle, either side can implement retransmit-on-timeout if nothing is received. Either side can also
implement retransmit-on-duplicate; this was done by the receiver in the second example above but not by
the sender in the third example (the sender received a second ACK[N] but did not retransmit Data[N+1]).

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At least one side must implement retransmit-on-timeout; otherwise a lost packet leads to deadlock as the
sender and the receiver both wait forever. The other side must implement at least one of retransmit-onduplicate or retransmit-on-timeout; usually the former alone. If both sides implement retransmit-on-timeout
with different timeout values, generally the protocol will still work.

6.1.2 Sorcerers Apprentice Bug

Sorcerers Apprentice
The Sorcerers Apprentice bug is named for the legend in which the apprentice casts a spell
on a broom to carry water, one bucket at a time. When the basin is full, the apprentice
chops the broom in half, only to find both halves carrying water. See Disneys Fantasia,
http://www.youtube.com/watch?v=XChxLGnIwCU, at around T = 5:35.
A strange thing happens if one side implements retransmit-on-timeout but both sides implement retransmiton-duplicate, as can happen if the implementer takes the naive view that retransmitting on duplicates is
safer; the moral here is that too much redundancy can be the Wrong Thing. Let us imagine that an
implementation uses this strategy, and that the initial ACK[3] is delayed until after Data[3] is retransmitted
on timeout. In the following diagram, the only packet retransmitted due to timeout is the second Data[3]; all
the other duplications are due to the retransmit-on-duplicate strategy.

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All packets are sent twice from Data[3] on. The transfer completes normally, but takes double the normal
bandwidth.

6.1.3 Flow Control

Stop-and-wait also provides a simple form of flow control to prevent data from arriving at the receiver faster
than it can be handled. Assuming the time needed to process a received packet is less than one RTT, the
stop-and-wait mechanism will prevent data from arriving too fast. If the processing time is slightly larger
than RTT, all the receiver has to do is to wait to send ACK[N] until Data[N] has not only arrived but also
been processed, and the receiver is ready for Data[N+1].
For modest per-packet processing delays this works quite well, but if the processing delays are long it
introduces a new problem: Data[N] may time out and be retransmitted even though it has successfully been
received; the receiver cannot send an ACK until it has finished processing. One approach is to have two kinds
of ACKs: ACKWAIT [N] meaning that Data[N] has arrived but the receiver is not yet ready for Data[N+1],
and ACKGO [N] meaning that the sender may now send Data[N+1]. The receiver will send ACKWAIT [N]
when Data[N] arrives, and ACKGO [N] when it is done processing it.
Presumably we want the sender not to time out and retransmit Data[N] after ACKWAIT [N] is received, as
a retransmission would be unnecessary. This introduces a new problem: if the subsequent ACKGO [N] is
lost and neither side times out, the connection is deadlocked. The sender is waiting for ACKGO [N], which
is lost, and the receiver is waiting for Data[N+1], which the sender will not send until the lost ACKGO [N]
arrives. One solution is for the receiver to switch to a timeout model, perhaps until Data[N+1] is received.
TCP has a fix to the flow-control problem involving sender-side polling; see 12.16 TCP Flow Control.

6.2 Sliding Windows

Stop-and-wait is reliable but it is not very efficient (unless the path involves neither intermediate switches
nor significant propagation delay; that is, the path involves a single LAN link). Most links along a multi-hop
stop-and-wait path will be idle most of the time. During a file transfer, ideally we would like zero idleness
(at least along the slowest link; see 6.3 Linear Bottlenecks).
We can improve overall throughput by allowing the sender to continue to transmit, sending Data[N+1]
(and beyond) without waiting for ACK[N]. We cannot, however, allow the sender get too far ahead of the
returning ACKs. Packets sent too fast, as we shall see, simply end up waiting in queues, or, worse, dropped
from queues. If the links of the network have sufficient bandwidth, packets may also be dropped at the
receiving end.
Now that, say, Data[3] and Data[4] may be simultaneously in transit, we have to revisit what ACK[4] means:
does it mean that the receiver has received only Data[4], or does it mean both Data[3] and Data[4] have
arrived? We will assume the latter, that is, ACKs are cumulative: ACK[N] cannot be sent until Data[K] has
arrived for all KN.
The sender picks a window size, winsize. The basic idea of sliding windows is that the sender is allowed
to send this many packets before waiting for an ACK. More specifically, the sender keeps a state variable
last_ACKed, representing the last packet for which it has received an ACK from the other end; if data
packets are numbered starting from 1 then initially last_ACKed = 0.

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Window Size
In this chapter we will assume winsize does not change. TCP, however, varies winsize up and down
with the goal of making it as large as possible without introducing congestion; we will return to this in
13 TCP Reno and Congestion Management.
At any instant, the sender may send packets numbered last_ACKed + 1 through last_ACKed + winsize; this
packet range is known as the window. Generally, if the first link in the path is not the slowest one, the sender
will most of the time have sent all these.
If ACK[N] arrives with N > last_ACKed (typically N = last_ACKed+1), then the window slides forward; we
set last_ACKed = N. This also increments the upper edge of the window, and frees the sender to send more
packets. For example, with winsize = 4 and last_ACKed = 10, the window is [11,12,13,14]. If ACK[11]
arrives, the window slides forward to [12,13,14,15], freeing the sender to send Data[15]. If instead ACK[13]
arrives, then the window slides forward to [14,15,16,17] (recall that ACKs are cumulative), and three more
packets become eligible to be sent. If there is no packet reordering and no packet losses (and every packet
is ACKed individually) then the window will slide forward in units of one packet at a time; the next arriving
ACK will always be ACK[last_ACKed+1].
Note that the rate at which ACKs are returned will always be exactly equal to the rate at which the slowest
link is delivering packets. That is, if the slowest link (the bottleneck link) is delivering a packet every
50 ms, then the receiver will receive those packets every 50 ms and the ACKs will return at a rate of
one every 50 ms. Thus, new packets will be sent at an average rate exactly matching the delivery rate;
this is the sliding-windows self-clocking property. Self-clocking has the effect of reducing congestion by
automatically reducing the senders rate whenever the available fraction of the bottleneck bandwidth is
reduced.
Here is a video of sliding windows in action, with winsize = 5. The second link, RB, has a capacity of five
packets in transit either way; the AR link has a capacity of one packet in transit either way.
The packets in the very first RTT represent connection setup. This particular video also demonstrates TCP
Slow Start: in the first data-packet RTT, two packets are sent, and in the second data RTT four packets
are sent. The full window size of five is reached in the third data RTT. For the rest of the connection, at
any moment (except those instants where packets have just been received) there are five packets in flight,
either being transmitted on a link as data, or being transmitted as an ACK, or sitting in a queue (this last
does not happen in the video).

6.2.1 Bandwidth Delay

As indicated previously (5.1 Packet Delay), the bandwidth RTT product represents the amount of data
that can be sent before the first response is received. This is generally the optimum size for the window size.
There is, however, one catch: if a sender chooses winsize larger than the bandwidth RTT product, then the
RTT simply grows due to queuing delays to the point that bandwidth RTT matches the chosen winsize.
That is, a connections own traffic can inflate RTTactual to well above RTTnoLoad ; see 6.3.1.3 Case 3: winsize
= 6 below for an example. For this reason, a sender is often more interested in bandwidth RTTnoLoad , or,
at the very least, the RTT before the senders own packets had begun contributing to congestion.
We will sometimes refer to the bandwidth RTTnoLoad product as the transit capacity of the route. As
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will become clearer below, a winsize smaller than this means underutilization of the network, while a larger
winsize means each packet spends time waiting in a queue somewhere.
Below are simplified diagrams for sliding windows with window sizes of 1, 4 and 6, each with a path
bandwidth of 6 packets/RTT (so bandwidth RTT = 6 packets). The diagram shows the initial packets sent
as a burst; these then would be spread out as they pass through the bottleneck link so that, after the first
burst, packet spacing is uniform. (Real sliding-windows protocols such as TCP generally attempt to avoid
such initial bursts.)

With winsize=1 we send 1 packet per RTT; with winsize=4 we always average 4 packets per RTT. To put
this another way, the three window sizes lead to bottle-neck link utilizations of 1/6, 4/6 and 6/6 = 100%,
respectively.
While it is tempting to envision setting winsize to bandwidth RTT, in practice this can be complicated;
neither bandwidth nor RTT is constant. Available bandwidth can fluctuate in the presence of competing
traffic. As for RTT, if a sender sets winsize too large then the RTT is simply inflated to the point that
bandwidth RTT matches winsize; that is, a connections own traffic can inflate RTTactual to well above
RTTnoLoad . This happens even in the absence of competing traffic.
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6.2.2 The Receiver Side

Perhaps surprisingly, sliding windows can work pretty well with the receiver assuming that winsize=1, even
if the sender is in fact using a much larger value. Each of the receivers in the diagrams above receives
Data[N] and responds with ACK[N]; the only difference with the larger sender winsize is that the Data[N]
arrive faster.
If we are using the sliding-windows algorithm over single links, we may assume packets are never reordered,
and a receiver winsize of 1 works quite well. Once switches are introduced, however, life becomes more
complicated (though some links may do link-level sliding-windows for per-link throughput optimization).
If packet reordering occurs, it is common for the receiver to use the same winsize as the sender. This
means that the receiver must be prepared to buffer a full window full of packets. If the window is
[11,12,13,14,15,16], for example, and Data[11] is delayed, then the receiver may have to buffer Data[12]
through Data[16].
Like the sender, the receiver will also maintain the state variable last_ACKed, though it will not be
completely synchronized with the senders version. At any instant, the receiver is willing to accept
Data[lastACKed+1] through Data[lastACKed+winsize]. For any but the first of these, the receiver must
buffer the arriving packet. If Data[lastACKed+1] arrives, then the receiver should consult its buffers and
send back the largest cumulative ACK it can for the data received; for example, if the window is [11-16]
and Data[12], Data[13] and Data[15] are in the buffers, then on arrival of Data[11] the correct response is
ACK[13]. Data[11] fills the gap, and the receiver has now received everything up through Data[13]. The
new receive window is [14-19], and as soon as the ACK[13] reaches the sender that will be the new send
window as well.

6.2.3 Loss Recovery Under Sliding Windows

Suppose winsize = 4 and packet 5 is lost. It is quite possible that packets 6, 7, and 8 may have been received.
However, the only (cumulative) acknowledgment that can be sent back is ACK[4]; the sender does not
know how much of the windowful made it through. Because of the possibility that only Data[5] (or more
generally Data[lastACKed+1]) is lost, and because losses are usually associated with congestion, when we
most especially do not wish to overburden the network, the sender will usually retransmit only the first lost
packet, eg packet 5. If packets 6, 7, and 8 were also lost, then after retransmission of Data[5] the sender
will receive ACK[5], and can assume that Data[6] now needs to be sent. However, if packets 6-8 did make
it through, then after retransmission the sender will receive back ACK[8], and so will know 6-8 do not need
retransmission and that the next packet to send is Data[9].
Normally Data[6] through Data[8] would time out shortly after Data[5] times out. After the first timeout, however, sliding windows protocols generally suppress further timeout/retransmission responses until
recovery is more-or-less complete.
Once a full timeout has occurred, usually the sliding-windows process itself has ground to a halt, in that there
are usually no packets remaining in flight. This is sometimes described as pipeline drain. After recovery,
the sliding-windows process will have to start up again. Most implementations of TCP, as we shall see later,
implement a mechanism (fast recovery) for early detection of packet loss, before the pipeline has fully
drained.

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6.3 Linear Bottlenecks

Consider the simple network path shown below, with bandwidths shown in packets/ms. The minimum
bandwidth, or path bandwidth, is 3 packets/ms.

The slow links are R2R3 and R3R4. We will refer to the slowest link as the bottleneck link; if there are
(as here) ties for the slowest link, then the first such link is the bottleneck. The bottleneck link is where the
queue will form. If traffic is sent at a rate of 4 packets/ms from A to B, it will pile up in an ever-increasing
queue at R2. Traffic will not pile up at R3; it arrives at R3 at the same rate by which it departs.
Furthermore, if sliding windows is used (rather than a fixed-rate sender), traffic will eventually not queue up
at any router other than R2: data cannot reach B faster than the 3 packets/ms rate, and so B will not return
ACKs faster than this rate, and so A will eventually not send data faster than this rate. At this 3 packets/ms
rate, traffic will not pile up at R1 (or R3 or R4).
There is a significant advantage in speaking in terms of winsize rather than transmission rate. If A sends to
B at any rate greater than 3 packets/ms, then the situation is unstable as the bottleneck queue grows without
bound and there is no convergence to a steady state. There is no analogous instability, however, if A uses
sliding windows, even if the winsize chosen is quite large (although a large-enough winsize will overflow
the bottleneck queue). If a sender specifies a sending window size rather than a rate, then the network will
converge to a steady state in relatively short order; if a queue develops it will be steadily replenished at the
same rate that packets depart, and so will be of fixed size.

6.3.1 Simple fixed-window-size analysis

We will analyze the effect of window size on overall throughput and on RTT. Consider the following network
path, with bandwidths now labeled in packets/second.

We will assume that in the backward BA direction, all connections are infinitely fast, meaning zero
delay; this is often a good approximation because ACK packets are what travel in that direction and they
are negligibly small. In the AB direction, we will assume that the AR1 link is infinitely fast, but
the other four each have a bandwidth of 1 packet/second (and no propagation-delay component). This
makes the R1R2 link the bottleneck link; any queue will now form at R1. The path bandwidth is 1
packet/second, and the RTT is 4 seconds.
As a roughly equivalent alternative example, we might use the following:

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with the following assumptions: the CS1 link is infinitely fast (zero delay), S1S2 and S2D each
take 1.0 sec bandwidth delay (so two packets take 2.0 sec, per link, etc), and ACKs also have a 1.0 sec
bandwidth delay in the reverse direction.
In both scenarios, if we send one packet, it takes 4.0 seconds for the ACK to return, on an idle network. This
means that the no-load delay, RTTnoLoad , is 4.0 seconds.
(These models will change significantly if we replace the 1 packet/sec bandwidth delay with a 1-second
propagation delay; in the former case, 2 packets take 2 seconds, while in the latter, 2 packets take 1 second.
See exercise 4.)
We assume a single connection is made; ie there is no competition. Bandwidth delay here is 4 packets (1
packet/sec 4 sec RTT)
6.3.1.1 Case 1: winsize = 2
In this case winsize < bandwidthdelay (where delay = RTT). The table below shows what is sent by A and
each of R1-R4 for each second. Every packet is acknowledged 4 seconds after it is sent; that is, RTTactual
= 4 sec, equal to RTTnoLoad ; this will remain true as the winsize changes by small amounts (eg to 1 or 3).
Throughput is proportional to winsize: when winsize = 2, throughput is 2 packets in 4 seconds, or 2/4 =
1/2 packet/sec. During each second, two of the routers R1-R4 are idle. The overall path will have less than
100% utilization.
Time
T
0
1
2
3
4
5
6
7
8
9

A
sends
1,2

3
4

5
6

R1
queues
2

R1
sends
1
2

3
4

5
6

R2
sends
1
2

3
4

R3
sends

1
2

3
4

R4
sends

1
2

3
4

B
ACKs

1
2

3
4

Note the brief pile-up at R1 (the bottleneck link!) on startup. However, in the steady state, there is no
queuing. Real sliding-windows protocols generally have some way of minimizing this initial pileup.
6.3.1.2 Case 2: winsize = 4
When winsize=4, at each second all four slow links are busy. There is again an initial burst leading to a brief
surge in the queue; RTTactual for Data[4] is 7 seconds. However, RTTactual for every subsequent packet is 4
seconds, and there are no queuing delays (and nothing in the queue) after T=2. The steady-state connection
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throughput is 4 packets in 4 seconds, ie 1 packet/second. Note that overall path throughput now equals the
bottleneck-link bandwidth, so this is the best possible throughput.
T
0
1
2
3
4
5
6
7
8

A sends
1,2,3,4

R1 queues
2,3,4
3,4
4

5
6
7
8
9

R1 sends
1
2
3
4
5
6
7
8
9

R2 sends

R3 sends

R4 sends

B ACKs

1
2
3
4
5
6
7
8

1
2
3
4
5
6
7

1
2
3
4
5
6

1
2
3
4
5

At T=4, R1 has just finished sending Data[4] as Data[5] arrives from A; R1 can begin sending packet 5
immediately. No queue will develop.
Case 2 is the congestion knee of Chiu and Jain [CJ89], defined here in 1.7 Congestion.
6.3.1.3 Case 3: winsize = 6
T
0
1
2
3
4
5
6
7
8
9
10

A sends
1,2,3,4,5,6

7
8
9
10
11
12
13

R1 queues
2,3,4,5,6
3,4,5,6
4,5,6
5,6
6,7
7,8
8,9
9,10
10,11
11,12
12,13

R1 sends
1
2
3
4
5
6
7
8
9
10
11

R2 sends

R3 sends

R4 sends

B ACKs

1
2
3
4
5
6
7
8
9
10

1
2
3
4
5
6
7
8
9

1
2
3
4
5
6
7
8

1
2
3
4
5
6
7

Note that packet 7 is sent at T=4 and the acknowledgment is received at T=10, for an RTT of 6.0 seconds. All
later packets have the same RTTactual . That is, the RTT has risen from RTTnoLoad = 4 seconds to 6 seconds.
Note that we continue to send one windowful each RTT; that is, the throughput is still winsize/RTT, but RTT
is now 6 seconds.
One might initially conjecture that if winsize is greater than the bandwidthRTTnoLoad product, then the
entire window cannot be in transit at one time. In fact this is not the case; the sender does usually have the
entire window sent and in transit, but RTT has been inflated so it appears to the sender that winsize equals
the bandwidthRTT product.
In general, whenever winsize > bandwidthRTTnoLoad , what happens is that the extra packets pile up at
a router somewhere along the path (specifically, at the router in front of the bottleneck link). RTTactual is
inflated by queuing delay to winsize/bandwidth, where bandwidth is that of the bottleneck link; this means
winsize = bandwidthRTTactual . Total throughput is equal to that bandwidth. Of the 6 seconds of RTTactual
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RTTnoLoad =4. The other two seconds, therefore, must be spent in a queue; there is no other place for packets
wait. Looking at the table, we see that each second there are indeed two packets in the queue at R1.
If the bottleneck link is the very first link, packets may begin returning before the sender has sent the entire
windowful. In this case we may argue that the full windowful has at least been queued by the sender, and
thus has in this sense been sent. Suppose the network, for example, is

where, as before, each link transports 1 packet/sec from A to B and is infinitely fast in the reverse direction.
Then, if A sets winsize = 6, a queue of 2 packets will form at A.

6.3.2 RTT Calculations

We can make some quantitative observations of sliding windows behavior, and about queue utilization. First,
we note that RTTnoLoad is the physical travel time (subject to the limitations addressed in 5.2 Packet
Delay Variability); any time in excess of RTTnoLoad is spent waiting in a queue somewhere. Therefore, the
following holds regardless of competing traffic, and even for isolated packets:
1. queue_time = RTTactual RTTnoLoad
When the bottleneck link is saturated, that is, is always busy, the number of packets actually in transit (not
queued) somewhere along the path will always be bandwidth RTTnoLoad .
Second, we always send one windowful per actual RTT, assuming no losses and each packet is individually
acknowledged. This is perhaps best understood by consulting the diagrams above, but here is a simple nonvisual argument: if we send Data[N] at time TD , and ACK[N] arrives at time TA , then RTT = TA TD , by
definition. At time TA the sender is allowed to send Data[N+winsize], so during the RTT interval TD T
< TA the sender must have sent Data[N] through Data[N+winsize-1]; that is, winsize many packets in time
RTT. Therefore (whether or not there is competing traffic) we always have
2. throughput = winsize/RTTactual
where throughput is the rate at which the connection is sending packets.
In the sliding windows steady state, where throughput and RTTactual are reasonably constant, the average
number of packets in the queue is just throughputqueue_time (where throughput is measured in packets/sec):
3. queue_usage = throughput (RTTactual RTTnoLoad )
= winsize (1 RTTnoLoad /RTTactual )
To give a little more detail making the averaging perhaps clearer, each packet spends time (RTTactual
RTTnoLoad ) in the queue, from equation 1 above. The total time spent by a windowful of packets is winsize
(RTTactual RTTnoLoad ), and dividing this by RTTactual thus gives the average number of packets in the
queue over the RTT interval in question.
In the presence of competing traffic, the throughput referred to above is simply the connections current
share of the total bandwidth. It is the value we get if we measure the rate of returning ACKs. If there is

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no competing traffic and winsize is below the congestion knee winsize < bandwidth RTTnoLoad then
winsize is the limiting factor in throughput. Finally, if there is no competition and winsize bandwidth
RTTnoLoad then the connection is using 100% of the capacity of the bottleneck link and throughput is equal
to the bottleneck-link physical bandwidth. To put this another way,
4. RTTactual = winsize/bottleneck_bandwidth
queue_usage = winsize bandwidth RTTnoLoad
Dividing the first equation by RTTnoLoad , and noting that bandwidth RTTnoLoad = winsize - queue_usage
= transit_capacity, we get
5. RTTactual /RTTnoLoad = winsize/transit_capacity = (transit_capacity + queue_usage) / transit_capacity
Regardless of the value of winsize, in the steady state the sender never sends faster than the bottleneck
bandwidth. This is because the bottleneck bandwidth determines the rate of packets arriving at the far
end, which in turn determines the rate of ACKs arriving back at the sender, which in turn determines the
continued sending rate. This illustrates the self-clocking nature of sliding windows.
We will return in 14 Dynamics of TCP Reno to the issue of bandwidth in the presence of competing traffic.
For now, suppose a sliding-windows sender has winsize > bandwidth RTTnoLoad , leading as above to a
fixed amount of queue usage, and no competition. Then another connection starts up and competes for the
bottleneck link. The first connections effective bandwidth will thus decrease. This means that bandwidth
RTTnoLoad will decrease, and hence the connections queue usage will increase.

6.3.3 Graphs at the Congestion Knee

Consider the following graphs of winsize versus
1. throughput
2. delay
3. queue utilization

The critical winsize value is equal to bandwidth RTTnoLoad ; this is known as the congestion knee. For
winsize below this, we have:
throughput is proportional to winsize

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delay is constant
queue utilization in the steady state is zero
For winsize larger than the knee, we have
throughput is constant (equal to the bottleneck bandwidth)
delay increases linearly with winsize
queue utilization increases linearly with winsize
Ideally, winsize will be at the critical knee. However, the exact value varies with time: available bandwidth
changes due to the starting and stopping of competing traffic, and RTT changes due to queuing. Standard
TCP makes an effort to stay well above the knee much of the time, presumably on the theory that maximizing
throughput is more important than minimizing queue use.
The power of a connection is defined to be throughput/RTT. For sliding windows below the knee, RTT is
constant and power is proportional to the window size. For sliding windows above the knee, throughput is
constant and delay is proportional to winsize; power is thus proportional to 1/winsize. Here is a graph, akin
to those above, of winsize versus power:

6.3.4 Simple Packet-Based Sliding-Windows Implementation

Here is a pseudocode outline of the receiver side of a sliding-windows implementation. Winsize is W; we
abbreviate lastACKed by LA. Thus, the next packet expected is LA+1 and the window is LA+1 ... LA+W.
We have a pool EA of early arrivals: packets buffered for future use. When packet Data[M] arrives:
if MLA or M>LA+W, ignore the packet
if M>LA+1, put the packet into EA.
if M==LA+1,
output the packet (that is, Data[LA+1])
LA = LA+1 (slide window forward by 1)
while (Data[LA+1] is in EA) {
output Data[LA+1]
LA = LA+1

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}
send ACK[LA]
There are a couple details omitted.
A possible implementation of EA is as an array of packet objects, of size W. We always put packet Data[K]
into position K % W.
At any point between packet arrivals, Data[LA+1] is not in EA, but some later packets may be present.
For the sender side, we begin by sending a full windowful of packets Data[1] through Data[W], and setting
LA=0. When ACK[M] arrives:
if MLA or M>LA+W, ignore the packet
otherwise:
set K = LA+1
set LA = M, the new bottom edge of the window
for (i=K; iLA; i++) send Data[i]
Note that new ACKs may arrive while we are in the loop at that last line. We assume here that the sender
stolidly sends what it may send and only after that does it start to process arriving ACKs. Some implementations may take a more asynchronous approach, perhaps with one thread processing arriving ACKs and
incrementing LA and another thread sending everything it is allowed to send.

6.4 Epilog
This completes our discussion of the sliding-windows algorithm in the abstract setting. We will return to
concrete implementations of this in 11.3.2 TFTP Stop-and-Wait (stop-and-wait) and in 12.13 TCP Sliding
Windows; the latter is one of the most important mechanisms on the Internet.

6.5 Exercises
1. Sketch a ladder diagram for stop-and-wait if Data[3] is lost the first time it is sent. Continue the diagram
to the point where Data[4] is successfully transmitted. Assume an RTT of 1 second, no sender timeout (but
the sender retransmits on duplicate), and a receiver timeout of 2 seconds.
2. Suppose a stop-and-wait receiver has an implementation flaw. When Data[1] arrives, ACK[1] and ACK[2]
are sent, separated by a brief interval; after that, the receiver transmits ACK[N+1] when Data[N] arrives,
rather than the correct ACK[N].

(a). Assume the sender responds to each ACK as it arrives. What is the first ACK[N] that it will be able to
determine is incorrect? Assume no packets are lost.
(b). Is there anything the transmitter can do to detect this receiver problem earlier?

3.
A

Create a table as in 6.3.1

Simple fixed-window-size analysis for the original
R1 R2
R3 R4
B network with winsize = 8. As in the text examples, assume 1

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packet/sec bandwidth delay for the R1R2, R2R3, R3R4 and R4B links. The AR link and
all reverse links (from B to A) are infinitely fast. Carry out the table for 10 seconds.
B. The A
4. Create a table as in 6.3.1 Simple fixed-window-size analysis for a network A R1 R2
R1 ink is infinitely fast; the R1R2 and R2B each have a 1-second propagation delay, in each direction,
and zero bandwidth delay (that is, one packet takes 1.0 sec to travel from R1 to R2; two packets also take
1.0 sec to travel from R1 to R2). Assume winsize=6. Carry out the table for 8 seconds. Note that with zero
bandwidth delay, multiple packets sent together will remain together until the destination.
5. Suppose RTTnoLoad = 4 seconds and the bottleneck bandwidth is 1 packet every 2 seconds.

(a). What window size is needed to remain just at the knee of congestion?
(b). Suppose winsize=6. How many packets are in the queue, at the steady state, and what is RTTactual ?

6. Create a table as in 6.3.1 Simple fixed-window-size analysis for a network A R1

R2 R3 B.
The AR1 link is infinitely fast. The R1R2 and R3B links have a bandwidth delay of 1 packet/second
with no additional propagation delay. The R2R3 link has a bandwidth delay of 1 packet / 2 seconds, and no
propagation delay. The reverse BA direction (for ACKs) is infinitely fast. Assume winsize = 6. Carry
out the table for 10 seconds. Note that in this exercise you will need to show the queue for both R1 and
R2. (If you carry out the pattern at least partially until T=18, you can verify that RTTactual for packet 8 is as
calculated in the previous exercise. You will need more than 10 packets, but fewer than 16; the use of hex
labels A, B, C for packets 10, 11, 12 is a convenient notation.)
Hint: The column for R2 sends (or, more literally, R2 is in the process of sending) should look like this:
T
0
1
2
3
4
5
6
...

R2 sends
1
1
2
2
3
3
...

7. Argue that, if A sends to B using sliding windows, and in the path from A to B the slowest link is not the
first link out of A, then eventually A will have the entire window outstanding (except at the instant just after
each new ACK comes in).
8. Suppose RTTnoLoad is 50 ms and the available bandwidth is 2,000 packets/sec. Sliding windows is used
for transmission.

(a). What window size is needed to remain just at the knee of congestion?
(b). If RTTactual rises to 60 ms (due to use of a larger winsize), how many packets are in a queue at any one
time?
(c). What value of winsize would lead to RTTactual = 60 ms?
(d). What value of winsize would make RTTactual rise to 100 ms?
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9. Suppose winsize=4 in a sliding-windows connection, and assume that while packets may be lost, they
are never reordered (that is, if two packets P1 and P2 are sent in that order, and both arrive, then they arrive
in that order). Show that if Data[8] is in the receivers window (meaning that everything up through Data[4]
has been received and acknowledged), then it is no longer possible for even a late Data[0] to arrive at the
receiver. (A consequence of the general principle here is that in the absence of reordering we can replace
the packet sequence number with (sequence_number) mod (2winsize+1) without ambiguity.)
10. Suppose winsize=4 in a sliding-windows connection, and assume as in the previous exercise that while
packets may be lost, they are never reordered. Give an example in which Data[8] is in the receivers window
(so the receiver has presumably sent ACK[4]), and yet Data[1] legitimately arrives. (Thus, the late-packet
bound in the previous exercise is the best possible.)
11. Suppose the network is A R1
R2 B, where the AR1 ink is infinitely fast and the R1R2
link has a bandwidth of 1 packet/second each way, for an RTTnoLoad of 2 seconds. Suppose also that A
begins sending with winsize = 6. By the analysis in 6.3.1.3 Case 3: winsize = 6, RTT should rise to
winsize/bandwidth = 6 seconds. Give the RTTs of the first eight packets. How long does it take for RTT to
rise to 6 seconds?

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7 IP VERSION 4

There are multiple LAN protocols below the IP layer and multiple transport protocols above, but IP itself
stands alone. The Internet is essentially the IP Internet. If you want to run your own LAN somewhere or if
you want to run your own transport protocol, the Internet backbone will still work for you. But if you want
to change the IP layer, you may encounter difficulty. (Just talk to the IPv6 people, or the IP-multicasting or
IP-reservations groups.)
IP is, in effect, a universal routing and addressing protocol. The two are developed together; every node
has an IP address and every router knows how to handle IP addresses. IP was originally seen as a way to
interconnect multiple LANs, but it may make more sense now to view IP as a virtual LAN overlaying all
the physical LANs.
A crucial aspect of IP is its scalability. As the Internet has grown to ~109 hosts, the forwarding tables are
not much larger than 105 (perhaps now 105.5 ). Ethernet, in comparison, scales poorly.
Furthermore, IP, unlike Ethernet, offers excellent support for multiple redundant links. If the network
below were an IP network, each node would communicate with each immediate neighbor via their shared
direct link. If, on the other hand, this were an Ethernet network with the spanning-tree algorithm, then one
of the four links would simply be disabled completely.

The IP network service model is to act like a LAN. That is, there are no acknowledgments; delivery is
generally described as best-effort. This design choice is perhaps surprising, but it has also been quite fruitful.
Currently the Internet uses (almost exclusively) IP version 4, with its 32-bit address size. As the Internet has
run out of new large blocks of IPv4 addresses (1.10 IP - Internet Protocol), there is increasing pressure to
convert to IPv6, with its 128-bit address size. Progress has been slow, however, and delaying tactics such
as IPv4-address markets and NAT have allowed IPv4 to continue. Aside from the major change in address
structure, there are relatively few differences in the routing models of IPv4 and IPv6. We will study IPv4 in
this chapter and IPv6 in the following.
If you want to provide a universal service for delivering any packet anywhere, what else do you need besides
routing and addressing? Every network (LAN) needs to be able to carry any packet. The protocols spell
out the use of octets (bytes), so the only possible compatibility issue is that a packet is too large for a given
network. IP handles this by supporting fragmentation: a network may break a too-large packet up into
units it can transport successfully. While IP fragmentation is inefficient and clumsy, it does guarantee that
any packet can potentially be delivered to any node.

7.1 The IPv4 Header

The IPv4 Header needs to contain the following information:

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destination and source addresses

indication of ipv4 versus ipv6
a Time To Live (TTL) value, to prevent infinite routing loops
a field indicating what comes next in the packet (eg TCP v UDP)
fields supporting fragmentation and reassembly.
Here is how it is all laid out in the header. Each row is 32 bits wide.

The IP header, and basics of IP protocol operation, were defined in RFC 791; some minor changes have
since occurred. Most of these changes were documented in RFC 1122, though the DS field was defined in
RFC 2474 and the ECN bits were first proposed in RFC 2481.
The Version field is, for IPv4, the number 4: 0100. The IHL field represents the total IP Header Length, in
32-bit words; an IP header can thus be at most 15 words long. The base header takes up five words, so the
IP Options can consist of at most ten words. If one looks at IPv4 packets using a packet-capture tool that
displays the packets in hex, the first byte will most often be 0x45.
The Differentiated Services (DS) field is used by the Differentiated Services suite to specify preferential handling for designated packets, eg those involved in VoIP or other real-time protocols. The Explicit
Congestion Notification bits are there to allow routers experiencing congestion to mark packets, thus indicating to the sender that the transmission rate should be reduced. We will address these in 14.8.2 Explicit
Congestion Notification (ECN). These two fields together replace the old 8-bit Type of Service field.
The Total Length field is present because an IP packet may be smaller than the minimum LAN packet
size (see Exercise 1) or larger than the maximum (if the IP packet has been fragmented over several LAN
packets. The IP packet length, in other words, cannot be inferred from the LAN-level packet size. Because
the Total Length field is 16 bits, the maximum IP packet size is 216 bytes. This is probably much too large,
even if fragmentation were not something to be avoided.
The second word of the header is devoted to fragmentation, discussed below.
The Time-to-Live (TTL) field is decremented by 1 at each router; if it reaches 0, the packet is discarded. A
typical initial value is 64; it must be larger than the total number of hops in the path. In most cases, a value of
32 would work. The TTL field is there to prevent routing loops always a serious problem should they occur
from consuming resources indefinitely. Later we will look at various IP routing-table update protocols and
how they minimize the risk of routing loops; they do not, however, eliminate it. By comparison, Ethernet
headers have no TTL field, but Ethernet also disallows cycles in the underlying topology.
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The Protocol field contains a value to indicate if the body of the IP packet represents a TCP packet or a
UDP packet, or, in unusual cases, something else altogether.
The Header Checksum field is the Internet checksum applied to the header only, not the body. Its only
purpose is to allow the discarding of packets with corrupted headers. When the TTL value is decremented
the router must update the header checksum. This can be done algebraically by adding a 1 in the correct
place to compensate, but it is not hard simply to re-sum the 8 halfwords of the average header.
The Source and Destination Address fields contain, of course, the IP addresses. These would be updated
only by NAT firewalls.
One option is the Record Route option, in which routers are to insert their own IP address into the IP
header option area. Unfortunately, with only ten words available, there is not enough space to record most
longer routes (but see 7.9.1 Traceroute and Time Exceeded, below). Another option, now deprecated as
security risk, is to support source routing. The sender would insert into the IP header option area a list of IP
addresses; the packet would be routed to pass through each of those IP addresses in turn. With strict source
routing, the IP addresses had to represent adjacent neighbors; no router could be used if its IP address were
not on the list. With loose source routing, the listed addresses did not have to represent adjacent neighbors
and ordinary IP routing was used to get from one listed IP address to the next. Both forms are essentially
never used, again for security reasons: if a packet has been source-routed, it may have been routed outside
of the at-least-somewhat trusted zone of the Internet backbone.

7.2 Interfaces
IP addresses (IPv4 and IPv6) are, strictly speaking, assigned not to hosts or nodes, but to interfaces. In the
most common case, where each node has a single LAN interface, this is a distinction without a difference. In
a room full of workstations each with a single Ethernet interface eth0 (or perhaps Ethernet adapter
Local Area Connection), we might as well view the IP address assigned to the interface as assigned
to the workstation itself.
Each of those workstations, however, likely also has a loopback interface (at least conceptually), providing
a way to deliver IP packets to other processes on the same machine. On many systems, the name localhost
resolves to the IPv4 address 127.0.0.1 (although the IPv6 address ::1 is also used). Delivering packets to
the localhost address is simply a form of interprocess communication; a functionally similar alternative is
named pipes. Loopback delivery avoids the need to use the LAN at all, or even the need to have a LAN.
For simple client/server testing, it is often convenient to have both client and server on the same machine,
in which case the loopback interface is convenient and fast. On unix-based machines the loopback interface
represents a genuine logical interface, commonly named lo. On Windows systems the interface may not
represent an actual entity, but this is of practical concern only to those interested in sniffing all loopback
traffic; packets sent to the loopback address are still delivered as expected.
Workstations often have special other interfaces as well. Most recent versions of Microsoft Windows have
a Teredo Tunneling pseudo-interface and an Automatic Tunneling pseudo-interface; these are both intended
(when activated) to support IPv6 connectivity when the local ISP supports only IPv4. The Teredo protocol
is documented in RFC 4380.
When VPN connections are created, as in 3.1 Virtual Private Network, each end of the logical connection
typically terminates at a virtual interface (one of these is labeled tun0 in the diagram of 3.1 Virtual Private

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Network). These virtual interfaces appear, to the systems involved, to be attached to a point-to-point link
that leads to the other end.
When a computer hosts a virtual machine, there is almost always a virtual network to connect the host and
virtual systems. The host will have a virtual interface to connect to the virtual network. The host may act as
a NAT router for the virtual machine, hiding that virtual machine behind its own IP address, or it may act
as an Ethernet switch, in which case the virtual machine will need an additional public IP address.
Whats My IP Address?
This simple-seeming question is in fact not very easy to answer, if by my IP address one means the IP
address assigned to the interface that connects directly to the Internet. One strategy is to find the address
of the default router, and then iterate through all interfaces (eg with the Java NetworkInterface
class) to find an IP address with a matching network prefix. Unfortunately, finding the default router
is hard to do in an OS-independent way, and even then this approach can fail if the Wi-Fi and Ethernet
interfaces both are assigned IP addresses on the same network, but only one is actually connected.
Many workstations have both an Ethernet interface and a Wi-Fi interface. Both of these can be used simultaneously (with different IP addresses assigned to each), either on the same IP network or on different IP
networks.
Routers always have at least two interfaces on two separate LANs. Generally this means a separate IP
address for each interface, though some point-to-point interfaces can be used without being assigned an IP
address (7.10 Unnumbered Interfaces).
Finally, it is usually possible to assign multiple IP addresses to a single interface. Sometimes this is done
to allow two IP networks to share a single LAN; the interface might be assigned one IP address for each
IP network. Other times a single interface is assigned multiple IP addresses that are on the same LAN; this
is often done so that one physical machine can act as a server (eg a web server) for multiple distinct IP
addresses corresponding to multiple distinct domain names.
While it is important to be at least vaguely aware of all these special cases, we emphasize again that in most
ordinary contexts each end-user workstation has one IP address that corresponds to a LAN connection.

7.3 Special Addresses

A few IP addresses represent special cases.
While the standard IPv4 loopback address is 127.0.0.1, any IP address beginning with 127 can serve as a
loopback address. Logically they all represent the current host. Most hosts are configured to resolve the
name localhost to 127.0.0.1. However, any loopback address eg. 127.255.33.57 should work, eg with
ping.
Private addresses are IP addresses intended only for site internal use, eg either behind a NAT firewall or
intended to have no Internet connectivity at all. If a packet shows up at any non-private router (eg at an
ISP router), with a private IP address as either source or destination address, the packet should be dropped.
Three standard private-address blocks have been defined:
10.0.0.0/8

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172.16.0.0/12
192.168.0.0/16
The last block is the one from which addresses are most commonly allocated by DHCP servers
(7.8.1 DHCP and the Small Office) built into NAT routers.
Broadcast addresses are a special form of IP address intended to be used in conjunction with LAN-layer
broadcast. The most common forms are broadcast to this network, consisting of all 1-bits, and broadcast
to network D, consisting of Ds network-address bits followed by all 1-bits for the host bits. If you try to
send a packet to the broadcast address of a remote network D, the odds are that some router involved will
refuse to forward it, and the odds are even higher that, once the packet arrives at a router actually on network
D, that router will refuse to broadcast it. Even addressing a broadcast to ones own network will fail if the
underlying LAN does not support LAN-level broadcast (eg ATM).
The highly influential early Unix implementation Berkeley 4.2 BSD used 0-bits for the broadcast bits, instead of 1s. As a result, to this day host bits cannot be all 1-bits or all 0-bits in order to avoid confusion with
the IP broadcast address. One consequence of this is that a Class C network has 254 usable host addresses,
not 256.

7.3.1 Multicast addresses

Finally, IP multicast addresses remain as the last remnant of the Class A/B/C strategy: multicast addresses
are Class D, with first byte beginning 1110 (meaning that the first byte is, in decimal, 224-239). Multicasting
means delivering to a specified set of addresses, preferably by some mechanism more efficient than sending
to each address individually. A reasonable goal of multicast would be that no more than one copy of the
multicast packet traverses any given link.
Support for IP multicast requires considerable participation by the backbone routers involved. For example,
if hosts A, B and C each connect to different interfaces of router R1, and A wishes to send a multicast packet
to B and C, then it is up to R1 to receive the packet, figure out that B and C are the intended recipients, and
forward the packet twice, once for Bs interface and once for Cs. R1 must also keep track of what hosts have
joined the multicast group and what hosts have left. Due to this degree of router participation, backbone
router support for multicasting has not been entirely forthcoming. A discussion of IP multicasting appears
in 19 Quality of Service.

7.4 Fragmentation
If you are trying to interconnect two LANs (as IP does), what else might be needed besides Routing and
Addressing? IP explicitly assumes all packets are composed on 8-bit bytes (something not universally true
in the early days of IP; to this day the RFCs refer to octets to emphasize this requirement). IP also defines
bit-order within a byte, and it is left to the networking hardware to translate properly. Neither byte size nor
bit order, therefore, can interfere with packet forwarding.
There is one more feature IP must provide, however, if the goal is universal connectivity: it must accommodate networks for which the maximum packet size, or Maximum Transfer Unit, MTU, is smaller than
the packet that needs forwarding. Otherwise, if we were using IP to join Token Ring (MTU = 4KB, at least

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originally) to Ethernet (MTU = 1500B), the token-ring packets might be too large to deliver to the Ethernet side, or to traverse an Ethernet backbone en route to another Token Ring. (Token Ring, in its day, did
commonly offer a configuration option to allow Ethernet interoperability.)
So, IP must support fragmentation, and thus also reassembly. There are two major strategies here: per-link
fragmentation and reassembly, where the reassembly is done at the opposite end of the link (as in ATM), and
path fragmentation and reassembly, where reassembly is done at the far end of the path. The latter approach
is what is taken by IP, partly because intermediate routers are too busy to do reassembly (this is as true today
as it was thirty years ago), and partly because IP fragmentation is seen as the strategy of last resort.
An IP sender is supposed to use a different value for the IDENT field for different packets, at least up until
the field wraps around. When an IP datagram is fragmented, the fragments keep the same IDENT field, so
this field in effect indicates which fragments belong to the same packet.
After fragmentation, the Fragment Offset field marks the start position of the data portion of this fragment
within the data portion of the original IP packet. Note that the start position can be a number up to 216 , the
maximum IP packet length, but the FragOffset field has only 13 bits. This is handled by requiring the data
portions of fragments to have sizes a multiple of 8 (three bits), and left-shifting the FragOffset value by 3
bits before using it.
As an example, consider the following network, where MTUs are excluding the LAN header:

Suppose A addresses a packet of 1500 bytes to B, and sends it via the LAN to the first router R1. The packet
contains 20 bytes of IP header and 1480 of data.
R1 fragments the original packet into two packets of sizes 20+976 = 996 and 20+504=544. Having 980
bytes of payload in the first fragment would fit, but violates the rule that the sizes of the data portions be
divisible by 8. The first fragment packet has FragOffset = 0; the second has FragOffset = 976.
R2 refragments the first fragment into three packets as follows:
first: size = 20+376=396, FragOffset = 0
second: size = 20+376=396, FragOffset = 376
third: size = 20+224 = 244 (note 376+376+224=976), FragOffset = 752.
R2 refragments the second fragment into two:
first: size = 20+376 = 396, FragOffset = 976+0 = 976
second: size = 20+128 = 148, FragOffset = 976+376=1352
R3 then sends the fragments on to B, without reassembly.
Note that it would have been slightly more efficient to have fragmented into four fragments of sizes 376,
376, 376, and 352 in the beginning. Note also that the packet format is designed to handle fragments of
different sizes easily. The algorithm is based on multiple fragmentation with reassembly only at the final
destination.
Each fragment has its IP-header Total Length field set to the length of that fragment.

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We have not yet discussed the three flag bits. The first bit is reserved, and must be 0. The second bit is the
Dont Fragment bit. If it is set to 1 by the sender then a router must not fragment the packet and must
drop it instead; see 12.12 Path MTU Discovery for an application of this. The third bit is set to 1 for all
fragments except the final one (this bit is thus set to 0 if no fragmentation has occurred). The third bit tells
the receiver where the fragments stop.
The receiver must take the arriving fragments and reassemble them into a whole packet. The fragments
may not arrive in order unlike in ATM networks and may have unrelated packets interspersed. The
reassembler must identify when different arriving packets are fragments of the same original, and must
figure out how to reassemble the fragments in the correct order; both these problems were essentially trivial
for ATM.
Fragments are considered to belong to the same packet if they have the same IDENT field and also the same
source and destination addresses and same protocol.
As all fragment sizes are a multiple of 8 bytes, the receiver can keep track of whether all fragments have
been received with a bitmap in which each bit represents one 8-byte fragment chunk. A 1 KB packet could
have up to 128 such chunks; the bitmap would thus be 16 bytes.
If a fragment arrives that is part of a new (and fragmented) packet, a buffer is allocated. While the receiver
cannot know the final size of the buffer, it can usually make a reasonable guess. Because of the FragOffset
field, the fragment can then be stored in the buffer in the appropriate position. A new bitmap is also allocated,
and a reassembly timer is started.
As subsequent fragments arrive, not necessarily in order, they too can be placed in the proper buffer in the
proper position, and the appropriate bits in the bitmap are set to 1.
If the bitmap shows that all fragments have arrived, the packet is sent on up as a completed IP packet. If, on
the other hand, the reassembly timer expires, then all the pieces received so far are discarded.
TCP connections usually engage in Path MTU Discovery, and figure out the largest packet size they can
send that will not entail fragmentation (12.12 Path MTU Discovery). But it is not unusual, for example,
for UDP protocols to use fragmentation, especially over the short haul. In the Network File Sharing (NFS)
protocol, for example, UDP is used to carry 8KB disk blocks. These are often sent as a single 8+ KB IP
packet, fragmented over Ethernet to five full packets and a fraction. Fragmentation works reasonably well
here because most of the time the packets do not leave the Ethernet they started on. Note that this is an
example of fragmentation done by the sender, not by an intermediate router.
Finally, any given IP link may provide its own link-layer fragmentation and reassembly; we saw in
3.8.1 ATM Segmentation and Reassembly that ATM does just this. Such link-layer mechanisms are, however, generally invisible to the IP layer.

7.5 The Classless IP Delivery Algorithm

Recall from Chapter 1 that any IP address can be divided into a net portion IPnet and a host portion IPhost ;
the division point was determined by whether the IP address was a Class A, a Class B, or a Class C. We
also hinted in Chapter 1 that the division point was not always so clear-cut; we now present the delivery
algorithm, for both hosts and routers, that does not assume a globally predeclared division point of the input
IP address into net and host portions. We will, for the time being, punt on the question of forwarding-table

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lookup and assume there is a lookup() method available that, when given a destination address, returns
the next_hop neighbor.
Instead of class-based divisions, we will assume that each of the IP addresses assigned to a nodes interfaces
is configured with an associated length of the network prefix; following the slash notation of 1.10 IP Internet Protocol, if B is an address and the prefix length is k = kB then the prefix itself is B/k. As usual, an
ordinary host may have only one IP interface, while a router will always have multiple interfaces.
Let D be the given IP destination address; we want to decide if D is local or nonlocal. The host or router
involved may have multiple IP interfaces, but for each interface the length of the network portion of the
address will be known. For each network address B/k assigned to one of the hosts interfaces, we compare
the first k bits of B and D; that is, we ask if D matches B/k.
If one of these comparisons yields a match, delivery is local; the host delivers the packet to its final
destination via the LAN connected to the corresponding interface. This means looking up the LAN
address of the destination, if applicable, and sending the packet to that destination via the interface.
If there is no match, delivery is nonlocal, and the host passes D to the lookup() routine of the
forwarding table and sends to the associated next_hop (which must represent a physically connected
neighbor). It is now up to lookup() routine to make any necessary determinations as to how D
might be split into Dnet and Dhost .
The forwarding table is, abstractly, a set of network addresses now also with lengths each of the form
B/k, with an associated next_hop destination for each. The lookup() routine will, in principle, compare
D with each table entry B/k, looking for a match (that is, equality of the first k bits). As with the interfaces
check above, the net/host division point (that is, k) will come from the table entry; it will not be inferred
from D or from any other information borne by the packet. There is, in fact, no place in the IP header to
store a net/host division point, and furthermore different routers along the path may use different values of k
with the same destination address D. In 10 Large-Scale IP Routing we will see that in some cases multiple
matches in the forwarding table may exist; the longest-match rule will be introduced to pick the best match.
Here is a simple example for a router with immediate neighbors A-E:
destination
10.3.0.0/16
10.4.1.0/24
10.4.2.0/24
10.4.3.0/24
10.3.37.0/24

next_hop
A
B
C
D
E

The IP addresses 10.3.67.101 and 10.3.59.131 both route to A. The addresses 10.4.1.101, 10.4.2.157 and
10.4.3.233 route to B, C and D respectively. Finally, 10.3.37.103 matches both A and E, but the E match is
longer so the packet is routed that way.
The forwarding table may also contain a default entry for the next_hop, which it may return in cases when
the destination D does not match any known network. We take the view here that returning such a default
entry is a valid result of the routing-table lookup() operation, rather than a third option to the algorithm
above; one approach is for the default entry to be the next_hop corresponding to the destination 0.0.0.0/0,
which does indeed match everything (use of this would definitely require the above longest-match rule,
though).
Default routes are hugely important in keeping leaf forwarding tables small. Even backbone routers some-

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times expend considerable effort to keep the network address prefixes in their forwarding tables as short as
possible, through consolidation.
Routers may also be configured to allow passing quality-of-service information to the lookup() method,
as mentioned in Chapter 1, to support different routing paths for different kinds of traffic (eg bulk file-transfer
versus real-time).
For a modest exception to the local-delivery rule described here, see below in 7.10 Unnumbered Interfaces.

7.6 IP Subnets
Subnets were the first step away from Class A/B/C routing: a large network (eg a class A or B) could be
divided into smaller IP networks called subnets. Consider, for example, a typical Class B network such
as Loyola Universitys (originally 147.126.0.0/16); the underlying assumption is that any packet can be
delivered via the underlying LAN to any internal host. This would require a rather large LAN, and would
require that a single physical LAN be used throughout the site. What if our site has more than one physical
LAN? Or is really too big for one physical LAN? It did not take long for the IP world to run into this
problem.
Subnets were first proposed in RFC 917, and became official with RFC 950.
Getting a separate IP network prefix for each subnet is bad for routers: the backbone forwarding tables now
must have an entry for every subnet instead of just for every site. What is needed is a way for a site to appear
to the outside world as a single IP network, but for further IP-layer routing to be supported inside the site.
This is what subnets accomplish.
Subnets introduce hierarchical routing: first we route to the primary network, then inside that site we route
to the subnet, and finally the last hop delivers to the host.
Routing with subnets involves in effect moving the IPnet division line rightward. (Later, when we consider
CIDR, we will see the complementary case of moving the division line to the left.) For now, observe that
moving the line rightward within a site does not affect the outside world at all; outside routers are not even
aware of site-internal subnetting.
In the following diagram, the outside world directs traffic addressed to 147.126.0.0/16 to the router R. Internally, however, the site is divided into subnets. The idea is that traffic from 147.126.1.0/24 to 147.126.2.0/24
is routed, not switched; the two LANs involved may not even be compatible. Most of the subnets shown
are of size /24, meaning that the third byte of the IP address has become part of the network portion of the
subnets address; one /20 subnet is also shown. RFC 950 would have disallowed the subnet with third byte
0, but having 0 for the subnet bits generally does work.

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What we want is for the internal routing to be based on the extended network prefixes shown, while externally continuing to use only the single routing entry for 147.126.0.0/16.
To implement subnets, we divide the sites IP network into some combination of physical LANs the subnets
, and assign each a subnet address: an IP network address which has the sites IP network address as prefix.
To put this more concretely, suppose the sites IP network address is A, and consists of n network bits (so
the site address may be written with the slash notation as A/n); in the diagram above, A/n = 147.126.0.0/16.
A subnet address is an IP network address B/k such that:
The address B/k is within the site: the first n bits of B are the same as A/ns
B/k extends A/n: kn
An example B/k in the diagram above is 147.126.1.0/24. (There is a slight simplification here in that subnet
addresses do not absolutely have to be prefixes; see below.)
We now have to figure out how packets will be routed to the correct subnet. For incoming packets we could
set up some proprietary protocol at the entry router to handle this. However, the more complicated situation
is all those existing internal hosts that, under the class A/B/C strategy, would still believe they can deliver
via the LAN to any site host, when in fact they can now only do that for hosts on their own subnet. We need
a more general solution.
We proceed as follows. For each subnet address B/k, we create a subnet mask for B consisting of k 1-bits
followed by enough 0-bits to make a total of 32. We then make sure that every host and router in the site
knows the subnet mask for every one of its interfaces. Hosts usually find their subnet mask the same way
they find their IP address (by static configuration if necessary, but more likely via DHCP, below).
Hosts and routers now apply the IP delivery algorithm of the previous section, with the proviso that, if a
subnet mask for an interface is present, then the subnet mask is used to determine the number of address bits
rather than the Class A/B/C mechanism. That is, we determine whether a packet addressed to destination
D is deliverable locally via an interface with subnet address B/k and corresponding mask M by comparing
D&M with B&M, where & represents bitwise AND; if the two match, the packet is local. This will generally

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involve a match of more bits than if we used the Class A/B/C strategy to determine the network portion of
addresses D and B.
As stated previously, given an address D with no other context, we will not be able to determine the network/host division point in general (eg for outbound packets). However, that division point is not in fact
what we need. All that is needed is a way to tell if a given destination host address D belongs to the current
subnet, say B; that is, we need to compare the first k bits of D and B where k is the (known) length of B.
In the diagram above, the subnet mask for the /24 subnets would be 255.255.255.0; bitwise ANDing any IP
address with the mask is the same as extracting the first 24 bits of the IP address, that is, the subnet portion.
The mask for the /20 subnet would be 255.255.240.0 (240 in binary is 1111 0000).
In the diagram above none of the subnets overlaps or conflicts: the subnets 147.126.0.0/24 and
147.126.1.0/24 are disjoint. It takes a little more effort to realize that 147.126.16.0/20 does not overlap
with the others, but note that an IP address matches this network prefix only if the first four bits of the third
byte are 0001, so the third byte itself ranges from decimal 32 to decimal 63 = binary 0001 1111.
Note also that if host A = 147.126.0.1 wishes to send to destination D = 147.126.1.1, and A is not subnetaware, then delivery will fail: A will infer that the interface is a Class B, and therefore compare the first
two bytes of A and D, and, finding a match, will attempt direct LAN delivery. But direct delivery is now
likely impossible, as the subnets are not joined by a switch. Only with the subnet mask will A realize that
its network is 147.126.0.0/24 while Ds is 147.126.1.0/24 and that these are not the same. A would still be
able to send packets to its own subnet. In fact A would still be able to send packets to the outside world:
it would realize that the destination in that case does not match 147.126.0.0/16 and will thus forward to its
router. Hosts on other subnets would be the only unreachable ones.
Properly, the subnet address is the entire prefix, eg 147.126.65.0/24. However, it is often convenient to
identify the subnet address with just those bits that represent the extension of the site IP-network address;
we might thus say casually that the subnet address here is 65.
The class-based IP-address strategy allowed any host anywhere on the Internet to properly separate any
address into its net and host portions. With subnets, this division point is now allowed to vary; for example,
the address 147.126.65.48 divides into 147.126 | 65.48 outside of Loyola, but into 147.126.65 | 48 inside.
This means that the net-host division is no longer an absolute property of addresses, but rather something
that depends on where the packet is on its journey.
Technically, we also need the requirement that given any two subnet addresses of different, disjoint subnets,
neither is a proper prefix of the other. This guarantees that if A is an IP address and B is a subnet address
with mask M (so B = B&M), then A&M = B implies A does not match any other subnet. Regardless of
the net/host division rules, we cannot possibly allow subnet 147.126.16.0/20 to represent one LAN while
147.126.16.0/24 represents another; the second subnet address block is a subset of the first. (We can,
and sometimes do, allow the first LAN to correspond to everything in 147.126.16.0/20 that is not also in
147.126.16.0/24; this is the longest-match rule.)
The strategy above is actually a slight simplification of what the subnet mechanism actually allows: subnet
address bits do not in fact have to be contiguous, and masks do not have to be a series of 1-bits followed by
0-bits. The mask can be any bit-mask; the subnet address bits are by definition those where there is a 1 in
the mask bits. For example, we could at a Class-B site use the fourth byte as the subnet address, and the
third byte as the host address. The subnet mask would then be 255.255.0.255. While this generality was
once sometimes useful in dealing with legacy IP addresses that could not easily be changed, life is simpler
when the subnet bits precede the host bits.

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7.6.1 Subnet Example

As an example of having different subnet masks on different interfaces, let us consider the division of a
class-C network into subnets of size 70, 40, 25, and 20. The subnet addresses will of necessity have different
lengths, as there is not room for four subnets each able to hold 70 hosts.
A: size 70
B: size 40
C: size 25
D: size 20
Because of the different subnet-address lengths, division of a local IP address LA into net versus host on
subnets cannot be done in isolation, without looking at the host bits. However, that division is not in fact
what we need. All that is needed is a way to tell if the local address LA belongs to a given subnet, say B;
that is, we need to compare the first n bits of LA and B, where n is the length of Bs subnet mask. We do
this by comparing LA&M to B&M, where M is the mask corresponding to n. LA&M is not necessarily the
same as LAnet , if LA actually belongs to one of the other subnets. However, if LA&M = B&M, then LA
must belong subnet B, in which case LA&M is in fact LAnet .
We will assume that the sites IP network address is 200.0.0.0/24. The first three bytes of each subnet address
must match 200.0.0. Only some of the bits of the fourth byte will be part of the subnet address, so we will
switch to binary for the last byte, and use both the /n notation (for total number of subnet bits) and also add
a vertical bar | to mark the separation between subnet and host.
Example: 200.0.0.10 | 00 0000 / 26
Note that this means that the 0-bit following the 1-bit in the fourth byte is significant in that for a subnet
to match, it must match this 0-bit exactly. The remaining six 0-bits are part of the host portion.
To allocate our four subnet addresses above, we start by figuring out just how many host bits we need in
each subnet. Subnet sizes are always powers of 2, so we round up the subnets to the appropriate size. For
subnet A, this means we need 7 host bits to accommodate 27 = 128 hosts, and so we have a single bit in the
fourth byte to devote to the subnet address. Similarly, for B we will need 6 host bits and will have 2 subnet
bits, and for C and D we will need 5 host bits each and will have 8-5=3 subnet bits.
We now start choosing non-overlapping subnet addresses. We have one bit in the fourth byte to choose for
As subnet; rather arbitrarily, let us choose this bit to be 1. This means that every other subnet address must
have a 0 in the first bit position of the fourth byte, or we would have ambiguity.
Now for Bs subnet address. We have two bits to work with, and the first bit must be 0. Let us choose
the second bit to be 0 as well. If the fourth byte begins 00, the packet is part of subnet B, and the subnet
addresses for C and D must therefore not begin 00.
Finally, we choose subnet addresses for C and D to be 010 and 011, respectively. We thus have
subnet
A
B
C
D

138

size
128
64
32
32

address bits in fourth byte

1
00
010
011

host bits in 4th byte

7
6
5
5

decimal range
128-255
0-63
64-95
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As desired, none of the subnet addresses in the third column is a prefix of any other subnet address.
The end result of all of this is that routing is now hierarchical: we route on the site IP address to get to a
site, and then route on the subnet address within the site.

7.6.2 Links between subnets

Suppose the Loyola CS department subnet (147.126.65.0/24) and a department at some other site, we will
say 147.100.100.0/24, install a private link. How does this affect routing?
Each department router would add an entry for the other subnet, routing along the private link. Traffic
addressed to the other subnet would take the private link. All other traffic would go to the default router.
Traffic from the remote department to 147.126.64.0/24 would take the long route, and Loyola traffic to
147.100.101.0/24 would take the long route.
Subnet anecdote
A long time ago I was responsible for two hosts, abel and borel. One day I was informed that machines
in computer lab 1 at the other end of campus could not reach borel, though they could reach abel.
Machines in lab 2, adjacent to lab 1, however, could reach both borel and abel just fine. What was the
difference?
It turned out that borel had a bad (/16 instead of /24) subnet mask, and so it was attempting local
delivery to the labs. This should have meant it could reach neither of the labs, as both labs were on
a different subnet from my machines; I was still perplexed. After considerably more investigation,
it turned out that between abel/borel and the lab building was a bridge-router: a hybrid device that
properly routed subnet traffic, but which also forwarded Ethernet packets directly, the latter feature
apparently for the purpose of backwards compatibility. Lab 2 was connected directly to the bridgerouter and thus appeared to be on the same LAN as borel, despite the apparently different subnet; lab
1 was connected to its own router R1 which in turn connected to the bridge-router. Lab 1 was thus, at
the LAN level, isolated from abel and borel.
Moral 1: Switching and routing are both great ideas, alone. But switching mixed with routing is not.
Moral 2: Test thoroughly! The reason the problem wasnt noticed earlier was that previously borel
communicated only with other hosts on the same subnet and with hosts outside the university entirely.
Both of these worked with the bad subnet mask; it is different-subnet local hosts that are the problem.
How would nearby subnets at either endpoint decide whether to use the private link? Classical link-state or
distance-vector theory (9 Routing-Update Algorithms) requires that they be able to compare the privatelink route with the going-around-the-long-way route. But this requires a global picture of relative routing
costs, which, as we shall see, almost certainly does not exist. The two departments are in different routing
domains; if neighboring subnets at either end want to use the private link, then manual configuration is likely
the only option.
..index:: single: switching, vs subnets single: subnets, vs switching

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7.6.3 Subnets versus Switching

A frequent network design question is whether to have many small subnets or to instead have just a few (or
even only one) larger subnet. With multiple small subnets, IP routing would be used to interconnect them;
the use of larger subnets would replace much of that routing with LAN-layer communication, likely Ethernet switching. Debates on this route-versus-switch question have gone back and forth in the networking
community, with one aphorism summarizing a common view:
Switch when you can, route when you must
This aphorism reflects the idea that switching is faster, cheaper and easier to configure, and that subnet
boundaries should be drawn only where necessary.
Ethernet switching equipment is indeed generally cheaper than routing equipment, for the same overall
level of features and reliability. And switching requires no configuration at all, while to implement subnets
not only must the subnets be created by hand but one must also set up and configure the routing-update
protocols. However, the price difference between switching and routing is not always significant in the big
picture, and the configuration involved is often straightforward.
Somewhere along the way, however, switching has acquired a reputation often deserved for being faster
than routing. It is true that routers have more to do than switches: they must decrement TTL, update the
header checksum, and attach a new LAN header. But these things are relatively minor: a larger reason many
routers are slower than switches may simply be that they are inevitably asked to serve as firewalls. This
means deep inspection of every packet, eg comparing every packet to each of a large number of firewall
rules. The firewall may also be asked to keep track of connection state. All this drives down the forwarding
rate, as measured in packets-per-second. The industry has come up with the term Layer 3 Switch to
describe routers that in effect do not support all the usual firewall bells and whistles; such streamlined
routers may also be able to do most of the hard work in specialized hardware, another source of speedup. (It
should be no surprise that Layer 3 switching is usually marketed in terms of the hardware speedup, not in
terms of the reduced flexibility.)
Switching scales remarkably well, but it does have limitations. First, broadcast packets must be forwarded
throughout a switched network; they do not, however, pass to different subnets. Second, LAN networks do
not like redundant links (that is, loops); while one can rely on the spanning-tree algorithm to eliminate these,
that algorithm too becomes less efficient at larger scales.

7.7 Address Resolution Protocol: ARP

If a host or router A finds that the destination IP address D = DIP matches the network address of one
of its interfaces, it is to deliver the packet via the LAN. This means looking up the LAN address DLAN
corresponding to DIP .
One approach would be via a special server, but the spirit of early IP development was to avoid such servers,
for both cost and reliability issues. Instead, the Address Resolution Protocol (ARP) is used. This is our first
protocol that takes advantage of the existence of LAN-level broadcast; on LANs without physical broadcast
(such as ATM), some other mechanism (usually involving a server) must be used.
The basic idea of ARP is that the host A sends out a broadcast ARP query or who-has DIP ? request, which
includes As own IP and LAN addresses. All hosts on the LAN receive this message. The host for whom the
message is intended, D, will recognize that it should reply, and will return an ARP reply or is-at message
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containing DLAN . Because the original request contained ALAN , Ds response can be sent directly to A, that
is, unicast.

Additionally, all hosts maintain an ARP cache, consisting of IP,LAN address pairs for other hosts on the
network. After the exchange above, A has DIP ,DLAN in its table; anticipating that A will soon send it a
packet to which it needs to respond, D also puts AIP ,ALAN into its cache.
ARP-cache entries eventually expire. The timeout interval used to be on the order of 10 minutes, but linux
systems now use a much smaller timeout (~30 seconds observed in 2012). Somewhere along the line, and
probably related to this shortened timeout interval, repeat ARP queries about a timed-out entry are first sent
unicast, not broadcast, to the previous Ethernet address on record. This cuts down on the total amount of
broadcast traffic; LAN broadcasts are, of course, still needed for new hosts. The ARP cache on a linux
system can be examined with the command ip -s neigh; the corresponding windows command is arp
-a.
The above protocol is sufficient, but there is one further point. When A sends its broadcast who-has D?
ARP query, all other hosts C check their own cache for an entry for A. If there is such an entry (that is, if
AIP is found there), then the value for ALAN is updated with the value taken from the ARP message; if there
is no pre-existing entry then no action is taken. This update process serves to avoid stale ARP-cache entries,
which can arise is if a host has had its Ethernet card replaced.

7.7.1 ARP Finer Points

Most hosts today implement self-ARP, or gratuitous ARP, on startup (or wakeup): when station A starts up
it sends out an ARP query for itself : who-has A?. Two things are gained from this: first, all stations that
had A in their cache are now updated with As most current ALAN address, in case there was a change, and
second, if an answer is received, then presumably some other host on the network has the same IP address
as A.
Self-ARP is thus the traditional IPv4 mechanism for duplicate address detection. Unfortunately, it does
not always work as well as might be hoped; often only a single self-ARP query is sent, and if a reply is
received then frequently the only response is to log an error message; the host may even continue using
the duplicate address! If the duplicate address was received via DHCP, below, then the host is supposed to
notify its DHCP server of the error and request a different IPv4 address.
RFC 5227 has defined an improved mechanism known as Address Conflict Detection, or ACD. A host
using ACD sends out three ARP queries for its new IP address, spaced over a few seconds and leaving the
ARP field for the senders IP address filled with zeroes. This last step means that any other host with that
IP address in its cache will ignore the packet, rather than update its cache. If the original host receives no
replies, it then sends out two more ARP queries for its new address, this time with the ARP field for the
senders IP address filled in with the new address; this is the stage at which other hosts on the network will

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make any necessary cache updates. Finally, ACD requires that hosts that do detect a duplicate address must
discontinue using it.
It is also possible for other stations to answer an ARP query on behalf of the actual destination D; this is
called proxy ARP. An early common scenario for this was when host C on a LAN had a modem connected
to a serial port. In theory a host D dialing in to this modem should be on a different subnet, but that requires
allocation of a new subnet. Instead, many sites chose a simpler arrangement. A host that dialed in to Cs
serial port might be assigned IP address DIP , from the same subnet as C. C would be configured to route
packets to D; that is, packets arriving from the serial line would be forwarded to the LAN interface, and
packets sent to CLAN addressed to DIP would be forwarded to D. But we also have to handle ARP, and as
D is not actually on the LAN it will not receive broadcast ARP queries. Instead, C would be configured to
answer on behalf of D, replying with DIP ,CLAN . This generally worked quite well.
Proxy ARP is also used in Mobile IP, for the so-called home agent to intercept traffic addressed to the
home address of a mobile device and then forward it (eg via tunneling) to that device. See 7.11 Mobile
IP.
One delicate aspect of the ARP protocol is that stations are required to respond to a broadcast query. In the
absence of proxies this theoretically should work quite well: there should be only one respondent. However,
there were anecdotes from the Elder Days of networking when a broadcast ARP query would trigger an
avalanche of responses. The protocol-design moral here is that determining who is to respond to a broadcast
message should be done with great care. (RFC 1122 section 3.2.2 addresses this same point in the context
of responding to broadcast ICMP messages.)
ARP-query implementations also need to include a timeout and some queues, so that queries can be resent
if lost and so that a burst of packets does not lead to a burst of queries. A naive ARP algorithm without these
might be:
To send a packet to destination DIP , see if DIP is in the ARP cache. If it is, address the packet
to DLAN ; if not, send an ARP query for D
To see the problem with this approach, imagine that a 32KB packet arrives at the IP layer, to be sent over
Ethernet. It will be fragmented into 22 fragments (assuming an Ethernet MTU of 1500 bytes), all sent at
once. The naive algorithm above will likely send an ARP query for each of these. What we need instead is
something like the following:
To send a packet to destination DIP :
If DIP is in the ARP cache, send to DLAN and return
If not, see if an ARP query for DIP is pending.
If it is, put the current packet in a queue for D.
If there is no pending ARP query for DIP , start one,
again putting the current packet in the (new) queue for D
We also need:
If an ARP query for some CIP times out, resend it (up to a point)
If an ARP query for CIP is answered, send off any packets in Cs queue

7.7.2 ARP security

Suppose A wants to log in to secure server S, using a password. How can B (for Bad) impersonate S?
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Here is an ARP-based strategy, sometimes known as ARP Spoofing. First, B makes sure the real S is down,
either by waiting until scheduled downtime or by launching a denial-of-service attack against S.
When A tries to connect, it will begin with an ARP who-has S?. All B has to do is answer, S is-at B.
There is a trivial way to do this: B simply needs to set its own IP address to that of S.
A will connect, and may be convinced to give its password to B. B now simply responds with something
plausible like backup in progress; try later, and meanwhile use As credentials against the real S.
This works even if the communications channel A uses is encrypted! If A is using the SSH protocol
(21.9.1 SSH), then A will get a message that the other sides key has changed (B will present its own
SSH key, not Ss). Unfortunately, many users (and even some IT departments) do not recognize this as a
serious problem. Some organizations especially schools and universities use personal workstations with
frozen configuration, so that the filesystem is reset to its original state on every reboot. Such systems may
be resistant to viruses, but in these environments the user at A will always get a message to the effect that
Ss credentials are not known.

7.7.3 ARP Failover

Suppose you have two front-line servers, A and B (B for Backup), and you want B to be able to step in if A
freezes. There are a number of ways of achieving this, but one of the simplest is known as ARP Failover.
First, we set AIP = BIP , but for the time being B does not use the network so this duplication is not a problem.
Then, once B gets the message that A is down, it sends out an ARP query for AIP , including BLAN as the
source LAN address. The gateway router, which previously would have had AIP ,ALAN in its ARP cache,
updates this to AIP ,BLAN , and packets that had formerly been sent to A will now go to B. As long as B is
trafficking in stateless operations (eg html), B can pick up right where A left off.

7.7.4 Detecting Sniffers

Finally, here is an interesting use of ARP to detect Ethernet password sniffers (generally not quite the issue
it once was, due to encryption and switching). To find out if a particular host A is in promiscuous mode,
send an ARP who-has A? query. Address it not to the broadcast Ethernet address, though, but to some
nonexistent Ethernet address.
If promiscuous mode is off, As network interface will ignore the packet. If promiscuous mode is on, As
network interface will pass the ARP request to A itself, which is likely then to answer it.
Alas, linux kernels reject at the ARP-software level ARP queries to physical Ethernet addresses other than
their own. However, they do respond to faked Ethernet multicast addresses, such as ff:ff:ff:00:00:00 or
ff:ff:ff:ff:ff:fe.

7.8 Dynamic Host Configuration Protocol (DHCP)

DHCP is the most common mechanism by which hosts are assigned their IP addresses. DHCP started as a
protocol known as Reverse ARP (RARP), which evolved into BOOTP and then into its present form. It is
documented in RFC 2131.

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Recall that ARP is based on the idea of someone broadcasting an ARP query for a host, containing the hosts
IP address, and the host answering it with its LAN address. DHCP involves a host, at startup, broadcasting a
query containing its own LAN address, and having a server reply telling the host what IP address is assigned
to it. The DHCP response is likely to contain several other essential startup options as well, including:
IP address
subnet mask
default router
DNS Server
These four items are a pretty standard minimal network configuration.
Default Routers and DHCP
If you lose your default router, you cannot communicate. Here is something that used to happen to me,
courtesy of DHCP:
1. I am connected to the Internet via Ethernet, and my default router is via my Ethernet interface
2. I connect to my institutions wireless network.
3. Their DHCP server sends me a new default router on the wireless network. However, this default
router will only allow access to a tiny private network, because I have neglected to complete the
Wi-Fi network registration process.
4. I therefore disconnect from the wireless network, and my wireless-interface default router goes
away. However, my system does not automatically revert to my Ethernet default-router entry;
DHCP does not work that way. As a result, I will have no router at all until the next scheduled
DHCP lease renegotiation, and must fix things manually.
The DHCP server has a range of IP addresses to hand out, and maintains a database of which IP address has
been assigned to which LAN address. Reservations can either be permanent or dynamic; if the latter, hosts
typically renew their DHCP reservation periodically (typically one to several times a day).

7.8.1 DHCP and the Small Office

If you have a large network, with multiple subnets, a certain amount of manual configuration is inevitable.
What about, however, a home or small office, with a single line from an ISP? A combination of NAT
(1.14 Network Address Translation) and DHCP has made autoconfiguration close to a reality.
The typical home/small-office router is in fact a NAT firewall coupled with a switch (usually also coupled
with a Wi-Fi access point); we will refer to this as a NAT router. One specially designated port, the external
port, connects to the ISPs line, and uses DHCP to obtain an IP address for that port. The other, internal,
ports are connected together by an Ethernet switch; these ports as a group are connected to the external port
using NAT translation. If wireless is supported, the wireless side is connected directly to the internal ports.
Isolated from the Internet, the internal ports can thus be assigned an arbitrary non-public IP address block,
eg 192.168.0.0/24. The NAT router contains a DCHP server, usually enabled by default, that will hand out
IP addresses to everything connecting from the internal side.
Generally this works seamlessly. However, if a second NAT router is also connected to the network (sometimes attempted to extend Wi-Fi range, in lieu of a commercial Wi-Fi repeater), one then has two operating
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DHCP servers on the same subnet. This often results in chaos, as two different hosts may be assigned the
same IP address, or a hosts IP address may suddenly change if it gets a new IP address from the other server.
Disabling one of the DHCP servers fixes this.
While omnipresent DHCP servers have made IP autoconfiguration work out of the box in many cases, in
the era in which IP was designed the need for such servers would have been seen as a significant drawback in
terms of expense and reliability. IPv6 has an autoconfiguration strategy (8.7.2 Stateless Autoconfiguration
(SLAAC)) that does not require DHCP, though DHCPv6 may well end up displacing it.

7.8.2 DHCP and Routers

It is often desired, for larger sites, to have only one or two DHCP servers, but to have them support multiple
subnets. Classical DHCP relies on broadcast, which isnt forwarded by routers, and even if it were, the
DHCP server would have no way of knowing on what subnet the host in question was actually located.
This is generally addressed by DHCP Relay (sometimes still known by the older name BOOTP Relay).
The router (or, sometimes, some other node on the subnet) receives the DHCP broadcast message from a
host, and notes the subnet address of the arrival interface. The router then relays the DHCP request, together
with this subnet address, to the designated DHCP Server; this relayed message is sent directly (unicast), not
broadcast. Because the subnet address is included, the DHCP server can figure out the correct IP address to
assign.
This feature has to be specially enabled on the router.

7.9 Internet Control Message Protocol

The Internet Control Message Protocol, or ICMP, is a protocol for sending IP-layer error and status messages. It is, like IP, host-to-host, and so they are never delivered to a specific port, even if they are sent
in response to an error related to something sent from that port. In other words, individual UDP and TCP
connections do not receive ICMP messages, even when it would be helpful to get them.
Here are the ICMP types, with subtypes listed in the description. The list has been pruned to include only
the most common values.

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Type
Echo Request
Echo Reply

Destination Unreachable

Source Quench
Redirect Message
Router Solicitation
Time Exceeded
Bad IP Header or Parameter

Description
ping queries
ping responses
Destination network unreachable
Destination host unreachable
Destination port unreachable
Fragmentation required but DF flag set
Network administratively prohibited
Congestion control
Redirect datagram for the network
Redirect datagram for the host
Redirect for TOS and network
Redirect for TOS and host
Router discovery/selection/solicitation
TTL expired in transit
Fragment reassembly time exceeded
Pointer indicates the error
Missing a required option
Bad length

ICMP is perhaps best known for Echo Request/Reply, on which the ping tool is based. Ping remains very
useful for network troubleshooting: if you can ping a host, then the network is reachable, and any problems
are higher up the protocol chain. Unfortunately, ping replies are blocked by default by many firewalls, on
the theory that revealing even the existence of computers is a security risk. While this may be an appropriate
decision, it does significantly impair the utility of ping. Most routers do still pass ping requests, but some
site routers block them.
Source Quench was used to signal that congestion has been encountered. A router that drops a packet due to
congestion experience was encouraged to send ICMP Source Quench to the originating host. Generally the
TCP layer would handle these appropriately (by reducing the overall sending rate), but UDP applications
never receive them. ICMP Source Quench did not quite work out as intended, and was formally deprecated
by RFC 6633. (Routers can inform TCP connections of impending congestion by using the ECN bits.)
The Destination Unreachable type has a large number of subtypes:
Network unreachable: some router had no entry for forwarding the packet, and no default route
Host unreachable: the packet reached a router that was on the same LAN as the host, but the host
failed to respond to ARP queries
Port unreachable: the packet was sent to a UDP port on a given host, but that port was not open.
TCP, on the other hand, deals with this situation by replying to the connecting endpoint with a reset
packet. Unfortunately, the UDP Port Unreachable message is sent to the host, not to the application on
that host that sent the undeliverable packet, and so is close to useless as a practical way for applications
to be informed when packets cannot be delivered.
Fragmentation required but DF flag set: a packet arrived at a router and was too big to be forwarded
without fragmentation. However, the Dont Fragment bit in the IP header was set, forbidding fragmentation. Later we will see how TCP uses this option as part of Path MTU Discovery, the process
of finding the largest packet we can send to a specific destination without fragmentation. The basic

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idea is that we set the DF bit on some of the packets we send; if we get back this message, that packet
was too big.
Administratively Prohibited: this is sent by a router that knows it can reach the network in question,
but has configured to drop the packet and send back Administratively Prohibited messages. A router
can also be configured to blackhole messages: to drop the packet and send back nothing.

7.9.1 Traceroute and Time Exceeded

The traceroute program uses ICMP Time Exceeded messages. A packet is sent to the destination (often
UDP to an unused port), with the TTL set to 1. The first router the packet reaches decrements the TTL to
0, drops it, and returns an ICMP Time Exceeded message. The sender now knows the first router on the
chain. The second packet is sent with TTL set to 2, and the second router on the path will be the one to
return ICMP Time Exceeded. This continues until finally the remote host returns something, likely ICMP
Port Unreachable.
Many routers no longer respond with ICMP Time Exceeded messages when they drop packets. For the
distance value corresponding to such a router, traceroute reports ***.
Traceroute assumes the path does not change. This is not always the case, although in practice it is seldom
an issue.
Route Efficiency
Once upon a time (~2001), traceroute showed that traffic from my home to the office, both in the
Chicago area, went through the MAE-EAST Internet exchange point, outside of Washington DC. That
inefficient route was later fixed. A situation like this is typically caused by two higher-level providers
who did not negotiate sufficient Internet exchange points.
Traceroute to a nonexistent site works up to the point when the packet reaches the Internet backbone: the
first router which does not have a default route. At that point the packet is not routed further (and an ICMP
Destination Network Unreachable should be returned).

7.9.2 Redirects
Most non-router hosts start up with an IP forwarding table consisting of a single (default) router, discovered
along with their IP address through DHCP. ICMP Redirect messages help hosts learn of other useful routers.
Here is a classic example:

A is configured so that its default router is R1. It addresses a packet to B, and sends it to R1. R1 receives
the packet, and forwards it to R2. However, R1 also notices that R2 and A are on the same network, and
so A could have sent the packet to R2 directly. So R1 sends an appropriate ICMP redirect message to A
(Redirect Datagram for the Network), and A adds a route to B via R2 to its own forwarding table.
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7.9.3 Router Solicitation

These ICMP messages are used by some router protocols to identify immediate neighbors. When we look
at routing-update algorithms, below, these are where the process starts.

7.10 Unnumbered Interfaces

We mentioned in 1.10 IP - Internet Protocol and 7.2 Interfaces that some devices allow the use of pointto-point IP links without assigning IP addresses to the interfaces at the ends of the link. Such IP interfaces
are referred to as unnumbered; they generally make sense only on routers. It is a firm requirement that the
node (ie router) at each endpoint of such a link has at least one other interface that does have an IP address;
otherwise, the node in question would be anonymous.
The diagram below shows a link L joining routers R1 and R2, which are connected to subnets 200.0.0.0/24
and 201.1.1.0/24 respectively. The endpoint interfaces of L, both labeled link0, are unnumbered.

The endpoints of L could always be assigned private IP addresses (7.3 Special Addresses), such as 10.0.0.1
and 10.0.0.2. To do this we would need to create a subnet; because the host bits cannot be all 0s or all 1s,
the minimum subnet size is four (eg 10.0.0.0/30). Furthermore, the routing protocols to be introduced in
9 Routing-Update Algorithms will distribute information about the subnet throughout the organization or
routing domain, meaning care must be taken to ensure that each links subnet is unique. Use of unnumbered links avoids this.
If R1 were to originate a packet to be sent to (or forwarded via) R2, the standard strategy is for it to treat
its link0 interface as if it shared the IP address of its Ethernet interface eth0, that is, 200.0.0.1; R2
would do likewise. This still leaves R1 and R2 violating the IP local-delivery rule of 7.5 The Classless
IP Delivery Algorithm; R1 is expected to deliver packets via local delivery to 201.1.1.1 but has no interface
that is assigned an IP address on the destination subnet 201.1.1.0/24. The necessary dispensation, however,
is granted by RFC 1812. All that is necessary by way of configuration is that R1 be told R2 is a directly
connected neighbor reachable via its link0 interface. On linux systems this might be done with the ip
route command on R1 as follows:
ip route
The linux ip route command illustrated here was tested on a virtual point-to-point link created with
ssh and pppd; the link interface name was in fact ppp0. While the command appeared to work as
advertised, it was only possible to create the link if endpoint IP addresses were assigned at the time
of creation; these were then removed with ip route del and then re-assigned with the command
shown here.

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ip route add 201.1.1.1 dev link0

Because L is a point-to-point link, there is no destination LAN address and thus no ARP query.

7.11 Mobile IP
In the original IPv4 model, there was a strong if implicit assumption that each IP host would stay put. One
role of an IP address is simply as a unique endpoint identifier, but another role is as a locator: some prefix
of the address (eg the network part, in the class-A/B/C strategy, or the provider prefix) represents something
about where the host is physically located. Thus, if a host moves far enough, it may need a new address.
When laptops are moved from site to site, it is common for them to receive a new IP address at each
location, eg via DHCP as the laptop connects to the local Wi-Fi. But what if we wish to support devices like
smartphones that may remain active and communicating while moving for thousands of miles? Changing
IP addresses requires changing TCP connections; life (and application development) might be simpler if a
device had a single, unchanging IP address.
One option, commonly used with smartphones connected to some so-called 3G networks, is to treat the
phones data network as a giant wireless LAN. The phones IP address need not change as it moves within
this LAN, and it is up to the phone provider to figure out how to manage LAN-level routing, much as is
done in 3.3.5 Wi-Fi Roaming.
But Mobile IP is another option, documented in RFC 5944. In this scheme, a mobile host has a permanent
home address and, while roaming about, will also have a temporary care-of address, which changes from
place to place. The care-of address might be, for example, an IP address assigned by a local Wi-Fi network,
and which in the absence of Mobile IP would be the IP address for the mobile host. (This kind of care-of
address is known as co-located; the care-of address can also be associated with some other device known
as a foreign agent in the vicinity of the mobile host.) The goal of Mobile IP is to make sure that the mobile
host is always reachable via its home address.
To maintain connectivity to the home address, a Mobile IP host needs to have a home agent back on the
home network; the job of the home agent is to maintain an IP tunnel that always connects to the devices
current care-of address. Packets arriving at the home network addressed to the home address will be forwarded to the mobile device over this tunnel by the home agent. Similarly, if the mobile device wishes to
send packets from its home address that is, with the home address as IP source address it can use the
tunnel to forward the packet to the home agent.
The home agent may use proxy ARP (7.7.1 ARP Finer Points) to declare itself to be the appropriate
destination on the home LAN for packets addressed to the home (IP) address; it is then straightforward for
the home agent to forward the packets.
An agent discovery process is used for the mobile host to decide whether it is mobile or not; if it is, it then
needs to notify its home agent of its current care-of address.
There are several forms of packet encapsulation that can be used for Mobile IP tunneling, but the default one
is IP-in-IP encapsulation, defined in RFC 2003. In this process, the entire original IP packet (with header
addressed to the home address) is used as data for a new IP packet, with a new IP header (the outer header)
addressed to the care-of address.

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A special value in the IP-header Protocol field indicates that IP-in-IP tunneling was used, so the receiver
knows to forward the packet on using the information in the inner header. The MTU of the tunnel will be
the original MTU of the path to the care-of address, minus the size of the outer header.

7.12 Epilog
At this point we have concluded the basic mechanics of IPv4. Still to come is a discussion of how IP
routers build their forwarding tables. This turns out to be a complex topic, divided into routing within single
organizations and ISPs 9 Routing-Update Algorithms and routing between organizations 10 LargeScale IP Routing.
But before that, in the next chapter, we compare IPv4 with IPv6, now twenty years old but still seeing very
limited adoption. The biggest issue fixed by IPv6 is IPv4s lack of address space, but there are also several
other less dramatic improvements.

7.13 Exercises
1. Suppose an Ethernet packet represents a TCP acknowledgment; that is, the packet contains an IP header
and a 20-byte TCP header but nothing else. Is the IP packet here smaller than the Ethernet minimum-packet
size, and, if so, by how much?
2. How can a receiving host tell if an arriving IP packet is unfragmented?
3. How long will it take the IDENT field of the IP header to wrap around, if the sender host A sends a stream
of packets to host B as fast as possible? Assume the packet size is 1500 bytes and the bandwidth is 600
Mbps.
4. The following diagram has routers A, B, C, D and E; E is the border router connecting the site to
the Internet. All router-to-router connections are via Ethernet-LAN /24 subnets with addresses of the form
200.0.x. Give forwarding tables for each of A, B, C and D. Each table should include each of the listed
subnets and also a default entry that routes traffic toward router E.
200.0.5

200.0.6

200.0.7

200.0.8

Internet

200.0.9
C

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200.0.10

5. (This exercise is an attempt at modeling Internet-2 routing.) Suppose sites S1 ... Sn each have a single
connection to the standard Internet, and each site Si has a single IP address block Ai . Each sites connection
to the Internet is through a single router Ri ; each Ri s default route points towards the standard Internet. The
sites also maintain a separate, higher-speed network among themselves; each site has a single link to this
separate network, also through Ri . Describe what the forwarding tables on each Ri will have to look like so
that traffic from one Si to another will always use the separate higher-speed network.
6. For each IP network prefix given (with length), identify which of the subsequent IP addresses are part of
the same subnet.

(a). 10.0.130.0/23: 10.0.130.23, 10.0.129.1, 10.0.131.12, 10.0.132.7

(b). 10.0.132.0/22: 10.0.130.23, 10.0.135.1, 10.0.134.12, 10.0.136.7
(c). 10.0.64.0/18: 10.0.65.13, 10.0.32.4, 10.0.127.3, 10.0.128.4

7. Suppose that the subnet bits below for the following five subnets A-E all come from the beginning of the
fourth byte of the IP address; that is, these are subnets of a /24 block.
A: 00
B: 01
C: 110
D: 111
E: 1010

(a). What are the sizes of each subnet, and the corresponding decimal ranges? Count the addresses with
host bits all 0s or with host bits all 1s as part of the subnet.
(b). How many IP addresses in the class-C block do not belong to any of the subnets A, B, C, D and E?

8. In 7.7 Address Resolution Protocol: ARP it was stated that, in newer implementations, repeat ARP
queries about a timed out entry are first sent unicast, in order to reduce broadcast traffic. Why is this unicast
approach likely to succeed most of the time? Can you give an example of a situation in which the unicast
query would fail, but a followup broadcast query would succeed?
9. Suppose A broadcasts an ARP query who-has B?, receives Bs response, and proceeds to send B a
regular IP packet. If B now wishes to reply, why is it likely that A will already be present in Bs ARP cache?
Identify a circumstance under which this can fail.
10. Suppose A broadcasts an ARP request who-has B, but inadvertently lists the physical address of
another machine C instead of its own (that is, As ARP query has IPsrc = A, but LANsrc = C). What will
happen? Will A receive a reply? Will any other hosts on the LAN be able to send to A? What entries will
be made in the ARP caches on A, B and C?
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What has been learned from experience with IPv4? First and foremost, more than 32 bits are needed for
addresses; the primary motive in developing IPv6 was the specter of running out of IPv4 addresses (something which, at the highest level, has already happened; see the discussion at the end of 1.10 IP - Internet
Protocol). Another important issue is that IPv4 requires a modest amount of effort at configuration; IPv6
was supposed to improve this.
By 1990 the IETF was actively interested in proposals to replace IPv4. A working group for the so-called
IP next generation, or IPng, was created in 1993 to select the new version; RFC 1550 was this groups
formal solicitation of proposals. In July 1994 the IPng directors voted to accept a modified version of the
Simple Internet Protocol, or SIP (unrelated to the Session Initiation Protocol), as the basis for IPv6.
SIP addresses were originally to be 64 bits in length, but in the month leading up to adoption this was
increased to 128. 64 bits would probably have been enough, but the problem is less the actual number than
the simplicity with which addresses can be allocated; the more bits, the easier this becomes, as sites can be
given relatively large address blocks without fear of waste. A secondary consideration in the 64-to-128 leap
was the potential to accommodate now-obsolete CLNP addresses, which were up to 160 bits in length (but
compressible).
IPv6 has to some extent returned to the idea of a fixed division between network and host portions: in most
ordinary-host cases, the first 64 bits of the address is the network portion (including any subnet portion)
and the remaining 64 bits represent the host portion. While there are some configuration alternatives here,
and while the IETF occasionally revisits the issue, at the present time the 64/64 split seems here to stay.
Routing, however, can, as in IPv4, be done on different prefixes of the address at different points of the
network. Thus, it is misleading to think of IPv6 as a return to Class A/B/C address allocation.
IPv6 is now twenty years old, and yet usage remains quite modest. However, the shortage in IPv4 addresses
has begun to loom ominously; IPv6 adoption rates may rise quickly if IPv4 addresses begin to climb in
price.

8.1 The IPv6 Header

The IPv6 fixed header looks like the following; at 40 bytes, it is twice the size of the IPv4 header. The
fixed header is intended to support only what every packet needs: there is no support for fragmentation, no
header checksum, and no option fields. However, the concept of extension headers has been introduced
to support some of these as options; some IPv6 extension headers are described below. Some fixed-header
header fields have been renamed from their IPv4 analogues: the IPv4 TTL is now the IPv6 Hop_Limit (still
decremented by each router with the packet discarded when it reaches 0), and the IPv4 DS field has become
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The Flow Label is new. RFC 2460 states that it

may be used by a source to label sequences of packets for which it requests special handling by
the IPv6 routers, such as non-default quality of service or real-time service.
Senders not actually taking advantage of any quality-of-service options are supposed to set the Flow Label
to zero.
When used, the Flow Label represents a sender-computed hash of the source and destination addresses, and
perhaps the traffic class. Routers can use this field as a way to look up quickly any priority or reservation
state for the packet. All packets belonging to the same flow should have the same Routing Extension header,
below. Note that the Flow Label will in general not include any information about the source and destination
port numbers, except that only some of the connections between a pair of hosts may make use of this field.
A flow, as the term is used here, is one-way; the return traffic belongs to a different flow. Historically,
the term flow has been used at various other scales: a single TCP connection, multiple related TCP
connections, or even all traffic from a particular subnet (eg the computer-lab flow).

8.2 IPv6 addresses

IPv6 addresses are written in eight groups of four hex digits, separated by colons, and with leading 0s
optionally removed, eg
FEDC:BA98:1654:310:FEDC:BA98:7654:3210
If an address contains a long run of 0s, as would be the case if the IPv6 address had an embedded IPv4
address, then when writing the address the string :: may be used to represent however many blocks of
0000 as are needed to create an address of the correct length; to avoid ambiguity this can be used only once.
Also, embedded IPv4 addresses may continue to use the . separator:
::FFFF:147.126.65.141

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The above is an example of the standard IPv6 format for representing IPv4 addresses. A separate representation of IPv4 addresses, with the FFFF block replaced by 0-bits, is used for tunneling IPv6 traffic over
IPv4. The IPv6 loopback address is ::1 (that is, 127 0-bits followed by a 1-bit).
Network address prefixes may be written with the / notation, as in IPv4:
12AB:0:0:CD30::/60
RFC 3513 suggested that initial IPv6 unicast-address allocation be initially limited to addresses beginning
with the bits 001, that is, the 2000::/3 block (20 in binary is 0010 0000).
Generally speaking, IPv6 addresses consist of a 64-bit network prefix (perhaps including subnet bits) and a
64-bit host identifier. See 8.5 Network Prefixes.

8.2.1 Host identifier

If desired, the second 64 bits of an IPv6 address can be a host identifier derived from the LAN address.
To create the host identifier from a 48-bit Ethernet address, for example, 0xFFFE is inserted between the
first three bytes and the last three bytes, to get 64 bits in all. The seventh bit of the first byte (the Ethernet
universal/local flag) is then set to 1. The result is officially known as the Modified EUI-64 Identifier,
where EUI stands for Extended Unique Identifier; details can be found in RFC 4291. For example, for a
host with Ethernet address 00:a0:cc:24:b0:e4, the EUI-64 address would be 02a0:ccff:fe24:b0e4 (note the
leading 00 becomes 02 when the seventh bit is turned on).

8.2.2 Link-local addresses

IPv6 defines link-local addresses, intended to be used only on a single LAN (and never routed). These begin
with the link-local prefix of the ten bits 1111 1110 10 followed by 54 more zero bits; that is, FE80::/64. The
final 64 bits are the host identifier for the link interface in question, above. The Ethernet link-local address of
the machine in the previous paragraph with Ethernet address 00:a0:cc:24:b0:e4 is fe80::2a0:ccff:fe24:b0e4.
When sending to a link-local address, one must include somewhere an identification of which link to use.
IPv4 addresses, on the other hand, almost always implicitly identify the link by virtue of the network prefix.
See 8.10.1 ping6 and 8.10.2 TCP connections with link-local addresses for examples of sending to linklocal addresses.
Once the link-local address is created, it must pass the duplicate-address detection test before being used;
see 8.7.1 Duplicate Address Detection.

8.2.3 Anycast addresses

IPv6 also introduces anycast addresses. An anycast address might be assigned to each of a set of routers (in
addition to each routers own unicast address); a packet addressed to this anycast address would be delivered
to only one member of this set. Note that this is quite different from multicast addresses; a packet addressed
to the latter is delivered to every member of the set.
It is up to the local routing infrastructure to decide which member of the anycast group would receive the
packet; normally it would be sent to the closest member. This allows hosts to send to any of a set of
routers, rather than to their designated individual default router.
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8.3 IPv6 Multicast

IPv6 has moved away from LAN-layer broadcast, instead providing a wide range of LAN-layer multicast
groups. (Note that LAN-layer multicast is straightforward; it is general IP-layer multicast (19.5 Global
IP Multicast) that is problematic. See 2.1.1 Ethernet Multicast for the Ethernet implementation.) This
switch to multicast is intended to limit broadcast traffic in general, though many switches still propagate
LAN multicast traffic everywhere, like broadcast.
An IPv6 multicast address is one beginning with the eight bits 1111 1111; numerous specific such addresses,
and even classes of addresses, have been defined. For actual delivery, IPv6 multicast addresses correspond
to LAN-layer (eg Ethernet) multicast addresses through a well-defined static correspondence; specifically,
if x, y, z and w are the last four bytes of the IPv6 multicast address, in hex, then the corresponding Ethernet
multicast address is 33:33:x:y:z:w (RFC 2464). A typical IPv6 host will need to join (that is, subscribe to)
several Ethernet multicast groups.
The IPv6 multicast address with the broadest scope is all-nodes, with address FF02::1; the corresponding
Ethernet multicast address is 33:33:00:00:00:01. This essentially corresponds to IPv4s LAN broadcast,
though the use of LAN multicast here means that non-IPv6 hosts should not see packets sent to this address.
Another important IPv6 multicast address is FF02::2, the all-routers address. This is meant to be used to
reach all routers, and routers only; ordinary hosts do not subscribe.
Generally speaking, IPv6 nodes on Ethernets send LAN-layer Multicast Listener Discovery (MLD) messages to multicast groups they wish to start using; these messages allow multicast-aware Ethernet switches
to optimize forwarding so that only those hosts that have subscribed to the multicast group in question
will receive the messages. Otherwise switches are supposed to treat multicast like broadcast; worse, some
switches may simply fail to forward multicast packets to destinations that have not explicitly opted to join
the group.

8.4 IPv6 Extension Headers

In IPv4, the IP header contained a Protocol field to identify the next header; usually UDP or TCP. All IPv4
options were contained in the IP header itself. IPv6 has replaced this with a scheme for allowing an arbitrary
chain of IPv6 headers. The IPv6 Next Header field can indicate that the following header is UDP or TCP,
but can also indicate one of several IPv6 options. These optional, or extension, headers include:
Hop-by-Hop options header
Destination options header
Routing header
Fragment header
Authentication header
Mobility header
Encapsulated Security Payload header
These extension headers must be processed in order; the recommended order for inclusion is as above. Most
of them are intended for processing only at the destination host; the routing header is an exception.

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8.4.1 Hop-by-Hop Options Header

This consists of a set of type,value pairs which are intended to be processed by each router on the path. A
tag in the type field indicates what a router should do if it does not understand the option: drop the packet,
or continue processing the rest of the options. The only Hop-by-Hop options provided by RFC 2460 were
for padding, so as to set the alignment of later headers.
RFC 2675 later defined a Hop-by-Hop option to support IPv6 jumbograms: datagrams larger than 65,535
bytes. The need for such large packets remains unclear, in light of 5.3 Packet Size. IPv6 jumbograms are
not meant to be used if the underlying LAN does not have an MTU larger than 65,535 bytes; the LAN world
is not currently moving in this direction.

8.4.2 Destination Options Header

This is very similar to the Hop-by-Hop Options header. It again consists of a set of type,value pairs,
and the original specification only defined options for padding. The Destination header is intended to be
processed at the destination, before turning over the packet to the transport layer.

8.4.3 Routing Header

The routing header contains a list of IPv6 addresses through which the packet should be routed. These
do not have to be contiguous: if the list to be visited en route to destination D is R1,R2,...,Rn, then the
IPv6 option header contains R2,R3,...,Rn,D and the initial destination address is R1; R1 then updates this
header to R1,R3,...,Rn,D (that is, the old destination R1 and the current next-router R2 are swapped), and
sends the packet on to R2. This continues on until Rn addresses the packet to the final destination D. The
header contains a Segments Left pointer indicating the next address to be processed, so that when the packet
arrives at D the Routing Header contains the routing list R1,R3,...,Rn This is, in general principle, very
much like IPv4 Loose Source routing. Note, however, that routers between the listed routers R1...Rn do not
need to examine this header; they process the packet based only on its current destination address.

8.4.4 Fragment Header

IPv6 supports limited IPv4-style fragmentation via the Fragment Header. This header contains a 13-bit
Fragment Offset field, which contains as in IPv4 the 13 high-order bits of the actual 16-bit offset of the
fragment. This header also contains a 32-bit Identification field; all fragments of the same packet must carry
the same value in this field.
IPv6 fragmentation is done only by the original sender; routers along the way are not allowed to fragment
or re-fragment a packet. Sender fragmentation would occur if, for example, the sender had an 8KB IPv6
packet to send via UDP, and needed to fragment it to accommodate the 1500-byte Ethernet MTU.
If a packet needs to be fragmented, the sender first identifies the unfragmentable part, consisting of the
IPv6 fixed header and any extension headers that must accompany each fragment (these would include
Hop-by-Hop and Routing headers). These unfragmentable headers are then attached to each fragment.
IPv6 also requires that every link on the Internet have an MTU of at least 1280 bytes beyond the LAN
header; link-layer fragmentation and reassembly can be used to meet this MTU requirement (which is what
ATM links do).
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Generally speaking, fragmentation should be avoided at the application layer when possible. UDP-based
applications that attempt to transmit filesystem-sized (usually 8 KB) blocks of data remain persistent users
of fragmentation.

8.5 Network Prefixes

We have been assuming that an IPv6 address, at least as seen by a host, is composed of a 64-bit network
prefix and a 64-bit host portion. As of this writing this is a requirement; RFC 4291 (IPv6 Addressing
Architecture) states:
For all unicast addresses, except those that start with the binary value 000, Interface IDs are
required to be 64 bits long....
This /64 requirement is occasionally revisited by the IETF, but we will assume here that it remains in force.
This is a departure from IPv4, where the host/subnet division point has depended, since the development of
subnets, on local configuration.
Note that while the net/host division point is fixed, routers may still use CIDR and may still route on any
shorter prefix than /64.
A typical IPv6 site may involve a variety of specialized network prefixes, including the link-local prefix
and prefixes for anycast and multicast addresses. Private IPv6 address prefixes, corresponding to IPv4s
10.0.0.0/8, may also be in use.
The most important class of 64-bit network prefixes, however, are those supplied by a provider or other
address-numbering entity, and which represent the first half of globally routable IPv6 addresses. These are
the prefixes that will be visible to the outside world.
IPv6 customers will typically be assigned a relatively large block of addresses, eg /48 or /56. The former
allows 6448 = 16 bits for local subnet specification within a 64-bit network prefix; the latter allows 8
subnet bits. These subnet bits are as in IPv4 supplied through router configuration. The closest IPv6
analogue to the IPv4 subnet mask is that all network prefixes are supplied to hosts with an associated length,
although that length will often always be 64 bits.
Many sites will have only a single externally visible address block. However, some sites may be multihomed
and thus have multiple independent address blocks.
Sites may also have private unique local address prefixes, corresponding to IPv4 private address blocks like
192.168.0.0/16 and 10.0.0.0/8. They are officially called Unique Local Unicast Addresses and are defined
in RFC 4193; these replace an earlier site-local address plan formally deprecated in RFC 3879. The first 8
bits of such a prefix are 1111 1101 (FD00::/8); the related prefix 1111 1100 (FC00::/8) is reserved for future
use. The last 16 bits of the 64-bit prefix represent the subnet ID, and are assigned either administratively or
via autoconfiguration. The 40 bits in between, from bit 8 up to bit 48, represent the Global ID. A site is to
set the Global ID to a pseudorandom value.
The resultant prefix is almost certainly globally unique, although it is not supposed to be routed and a site
would generally not admit any packets from the outside world addressed to a destination with the Global
ID as prefix. One rationale for choosing unique Global IDs for each site is to accommodate potential later
mergers of organizations without the need for renumbering; this has been a chronic problem for sites using
private IPv4 address blocks. Another justification is to accommodate VPN connections from other sites. For

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example, if I use IPv4 block 10.0.0.0/8 at home, and connect using VPN to a site also using 10.0.0.0/8, it is
possible that my printer will have the same IPv4 address as their application server.

8.6 Neighbor Discovery

IPv6 Neighbor Discovery, or ND, is a protocol that replaces (and combines) several IPv4 tools, most notably
ARP, ICMP redirects and most non-address-assignment parts of DHCP. The original specification for ND is
in RFC 2461, later updated by RFC 4861. ND provides the following services:
Finding a local hosts LAN address, given its IPv6 address
Finding the local router(s)
Finding the set of network address prefixes that can be reached via local delivery (IPv6 allows there
to be more than one)
Determining that some neighbors are now unreachable
Detecting duplicate IPv6 addresses

8.6.1 Router Advertisement

IPv6 routers periodically send Router Advertisement packets; these are also sent in response to Router
Solicitation requests sent by hosts to the all-routers multicast group. Router Advertisement packets serve to
identify the routers; this process is sometimes called Router Discovery. These RA packets also contain a list
of all network address prefixes in use on the LAN. These packets may contain other important information
about the LAN as well, such as an agreed-on MTU; much of the information included in RA packets is, in
IPv4, sent via DHCP.
These IPv6 router messages represent a marked change from IPv4, in which routers need not send anything
besides forwarded packets, and in which hosts typically discover their default routers via DHCP. IPv6 routers
may even interact directly with ordinary hosts, if the SLAAC configuration mechanism (below) is used. To
become an IPv4 router, a node need only enable IPv4 forwarding in the kernel. To become an IPv6 router,
by comparison, in addition to forwarding a node MUST (RFC 4294) also run software to support Router
Advertisement; the linux radvd package is one option here.

8.6.2 Prefix Discovery

Closely related to Router Discovery is the Prefix Discovery process by which hosts learn what IPv6
network-address prefixes, above, are valid on the network. It is also where hosts learn which prefixes are
considered to be local to the hosts LAN, and thus reachable at the LAN layer instead of requiring router
assistance for delivery. IPv6, in other words, does not limit determination of whether delivery is local to the
IPv4 mechanism of having a node check a destination address against each of the network-address prefixes
assigned to the nodes interfaces.
Even IPv4 allows two IPv4 network prefixes to share the same LAN (eg a private one 10.1.2.0/24 and a
public one 147.126.65.0/24), but a consequence of IPv4 routing is that two such subnets can only reach one

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another via a router on the LAN, even though they should in principle be able to communicate directly. IPv6
drops this restriction.
The Router Advertisement packets sent by the router should contain a complete list of valid network-address
prefixes, as the Prefix Information option. In simple cases this list may contain a single externally routable
64-bit prefix. If a particular LAN is part of multiple (overlapping) physical subnets, the prefix list will
contain an entry for each subnet; these 64-bit prefixes will themselves likely have a common prefix of length
N<64. For multihomed sites the prefix list may contain multiple unrelated prefixes corresponding to the
different address blocks. Finally, private and local IPv6 address prefixes may also be included.
Each prefix will have an associated lifetime; nodes receiving a prefix from an RA packet are to use it only
for the duration of this lifetime. On expiration (and likely much sooner) a node must obtain a newer RA
packet with a newer prefix list. The rationale for inclusion of the prefix lifetime is ultimately to allow sites
to easily renumber; that is, to change providers and switch to a new network-address prefix provided by a
new router. Each prefix is also tagged with a bit indicating whether it can be used for autoconfiguration, as
in 8.7.2 Stateless Autoconfiguration (SLAAC) below.

8.6.3 Neighbor Solicitation

Neighbor Solicitation messages are the IPv6 analogues of IPv4 ARP requests. These are essentially queries
of the form who has IPv6 address X? While ARP requests were broadcast, IPv6 Neighbor Solicitation
messages are sent to the solicited-node multicast address, which at the LAN layer usually represents a
rather small multicast group. This address is FF02::0001:x.y.z.w, where x, y, z and w are the low-order 32
bits of the IPv6 address the sender is trying to look up. Each IPv6 host on the LAN will need to subscribe to
all the solicited-node multicast addresses corresponding to its own IPv6 addresses (normally this is not too
many).
Neighbor Solicitation messages are repeated regularly, but followup verifications are initially sent to the
unicast LAN address on file (this is common practice with ARP implementations, but is optional). Unlike
with ARP, other hosts on the LAN are not expected to eavesdrop on the initial Neighbor Solicitation message.
The target hosts response to a Neighbor Solicitation message is called Neighbor Advertisement; a host
may also send these unsolicited if it believes its LAN address may have changed.
The analogue of Proxy ARP is still permitted, in that a node may send Neighbor Advertisements on behalf
of another. The most likely reason for this is that the node receiving proxy services is a mobile host
temporarily remote from the home LAN. Neighbor Advertisements sent as proxies have a flag to indicate
that, if the real target does speak up, the proxy advertisement should be ignored.
Hosts may also use the Authentication Header or the Encapsulated Security Payload Header to supply digital
signatures for ND packets. If a node is statically configured to require such checks, then the IPv6 analogue
of ARP spoofing (7.7.2 ARP security) can be prevented.
Finally, IPv4 ICMP Redirect messages have also been moved in IPv6 to the Neighbor Discovery protocol.

8.7 IPv6 Host Address Assignment

IPv6 provides two competing ways for hosts to obtain their full IP addresses. One is DHCPv6, based
on IPv4s DHCP (7.8 Dynamic Host Configuration Protocol (DHCP)). In addition to DHCPv6, IPv6
also supports StateLess Address AutoConfiguration, or SLAAC. The original idea behind SLAAC was
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to support complete plug-and-play network setup: hosts on a completely isolated LAN could talk to one
another out of the box, and if a router was introduced connecting the LAN to the Internet, then hosts would
be able to determine unique, routable addresses from information available from the router.
In the early days of IPv6 development, in fact, DHCPv6 may have been intended only for address assignments to routers and servers, with SLAAC meant for ordinary hosts. In that era, it was still common for
IPv4 addresses to be assigned statically, via per-host configuration files. RFC 4862 states that SLAAC is
to be used when a site is not particularly concerned with the exact addresses hosts use, so long as they are
unique and properly routable.
SLAAC and DHCPv6 evolved to some degree in parallel. While SLAAC solves the autoconfiguration problem quite neatly, at this point DHCPv6 solves it just as effectively, and provides for greater administrative
control. For this reason, SLAAC may end up less widely deployed. On the other hand, SLAAC gives hosts
greater control over their IPv6 addresses, and so may end up offering hosts a greater degree of privacy by
allowing endpoint management of the use of private and temporary addresses (below).
When a host first begins the Neighbor Discovery process, it receives a Router Advertisement packet. In this
packet are two special bits: the M (managed) bit and the O (other configuration) bit. The M bit is set to
indicate that DHCPv6 is available on the network for address assignment. The O bit is set to indicate that
DHCPv6 is able to provide additional configuration information (eg the name of the DNS server) to hosts
that are using SLAAC to obtain their addresses.

8.7.1 Duplicate Address Detection

Whenever an IPv6 host obtains a unicast address a link-local address, an address created via SLAAC,
an address received via DHCPv6 or a manually configured address it goes through a duplicate-address
detection (DAD) process. The host sends one or more Neighbor Solicitation messages (that is, like an ARP
query), as in 8.6 Neighbor Discovery, asking if any other host has this address; if anyone answers, then
the address is a duplicate. As with IPv4 ACD (7.7.1 ARP Finer Points), but not as with the original IPv4
self-ARP, the source-IP-address field of this NS message is set to a special unspecified value; this allows
other hosts to recognize it as a DAD query.
Because this NS process may take some time, and because addresses are in fact almost always unique, RFC
4429 defines an optimistic DAD mechanism. This allows limited use of an address before the DAD process
completes; in the meantime, the address is marked as optimistic.
Outside the optimistic-DAD interval, a host is not allowed to use an IPv6 address if the DAD process has
failed. RFC 4862 in fact goes further: if a host with an established address receives a DAD query for that
address, indicating that some other host wants to use that address, then the original host should discontinue
use of the address.

8.7.2 Stateless Autoconfiguration (SLAAC)

To obtain an address via SLAAC, defined in RFC 4862, a host first generates its link-local address, as above
in 8.2.2 Link-local addresses, appending the standard 64-bit link-local prefix to its 64-bit host identifier
(8.2.1 Host identifier) derived from the LAN address. The host must then ensure that its newly configured
link-local address is in fact unique; it uses DAD (above) to verify this. Assuming no duplicate is found, then
at this point the host can talk to any other hosts on the same LAN, eg to figure out where the printers are.

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The next step is to see if there is a router available. The host sends a Router Solicitation (RS) message
to the all-routers multicast address. A router if present should answer with a Router Advertisement
(RA) message that also contains a Prefix Information option; that is, a list of IPv6 network-address prefixes
(8.6.2 Prefix Discovery). The RA message will mark with a flag those prefixes eligible for use with
SLAAC; if no prefixes are so marked, then SLAAC should not be used. All prefixes will also be marked
with a lifetime, indicating how long the host may continue to use the prefix; once the prefix expires, the host
must obtain a new one via a new RA message.
The host chooses an appropriate prefix, stores the prefix-lifetime information, and, in the original version
of SLAAC, appends the prefix to the front of its host identifier to create what should now be a routable
address. The prefix length plus the host-identifier length must equal 128 bits; in the most common case each
is 64 bits. The address so formed must now be verified through the duplicate-address-detection mechanism
above.
An address generated in this way will, because of the embedded host identifier, uniquely identify the host
for all time. This includes identifying the host even when it is connected to a new network and is given
a different network prefix. Therefore, RFC 4941 defines a set of privacy extensions to SLAAC: optional
mechanisms for the generation of alternative host identifiers, based on pseudorandom generation using the
original LAN-address-based host identifier as a seed value. The probability of two hosts accidentally
choosing the same host identifier in this manner is very small; the Neighbor Solicitation mechanism with
DAD must, however, still be used to verify that the address is in fact unique. DHCPv6 also provides an
option for temporary address assignments, also to improve privacy, but one of the potential advantages of
SLAAC is that this process is entirely under the control of the end system.
Regularly (eg every few hours, or less) changing the host portion of an IPv6 address will make external
tracking of a host slightly more difficult. However, for a residential site with only a handful of hosts, a
considerable degree of tracking may be obtained simply by using the common 64-bit prefix.
In theory, if another host B on the LAN wishes to contact host A with a SLAAC-configured address containing the original host identifier, and B knows As IPv6 address AIPv6 , then B might extract As LAN address
from the low-order bits of AIPv6 . This was never actually allowed, however, even before the RFC 4941
privacy options, as there is no way for B to know that As address was generated via SLAAC at all. B would
always find As LAN address through the usual process of IPv6 Neighbor Solicitation.
A host using SLAAC may receive multiple network prefixes, and thus generate for itself multiple addresses.
RFC 6724 defines a process for a host to determine, when it wishes to connect to destination address D,
which of its own multiple addresses to use. For example, if D is a site-local address, not globally visible,
then the host will likely want to use an address that is also site-local. RFC 6724 also includes mechanisms
to allow a host with a permanent public address (eg corresponding to a DNS entry) to prefer alternative
temporary or privacy addresses for outbound connections.
At the end of the SLAAC process, the host knows its IPv6 address (or set of addresses) and its default router.
In IPv4, these would have been learned through DHCP along with the identity of the hosts DNS server; one
concern with SLAAC is that there is no obvious way for a host to find its DNS server. One strategy is to fall
back on DHCPv6 for this. However, RFC 6106 now defines a process by which IPv6 routers can include
DNS-server information in the RA packets they send to hosts as part of the SLAAC process; this completes
the final step of the autoconfiguration process.
How to get DNS names for SLAAC-configured IPv6 hosts into the DNS servers is an entirely separate
issue. One approach is simply not to give DNS names to such hosts. In the NAT-router model for IPv4
autoconfiguration, hosts on the inward side of the NAT router similarly do not have DNS names (although
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they are also not reachable directly, while SLAAC IPv6 hosts would be reachable). If DNS names are needed
for hosts, then a site might choose DHCPv6 for address assignment instead of SLAAC. It is also possible to
figure out the addresses SLAAC would use (by identifying the host-identifier bits) and then creating DNS
entries for these hosts. Hosts can also use Dynamic DNS (RFC 2136) to update their own DNS records.

8.7.3 DHCPv6
The job of the DHCPv6 server is to tell an inquiring host its network prefix(es) and also supply a 64-bit host-identifier.
Hosts begin the process by sending a DHCPv6 request to the
All_DHCP_Relay_Agents_and_Servers multicast IPv6 address FF02::1:2 (versus the broadcast address for
IPv4). As with DHCPv4, the job of a relay agent is to tag a DHCP request with the correct current subnet, and then to forward it to the actual DCHPv6 server. This allows the DHCP server to be on a different subnet from the requester. Note that the use of multicast does nothing to diminish the need for
relay agents; use of the multicast group does not necessarily identify a requesters subnet. In fact, the
All_DHCP_Relay_Agents_and_Servers multicast address scope is limited to the current link; relay agents
then forward to the actual DHCP server using the site-scoped address All_DHCP_Servers.
Hosts using SLAAC to obtain their address can still use a special Information-Request form of DHCPv6 to
obtain their DNS server and any other static DHCPv6 information.
Clients may ask for temporary addresses. These are identified as such in the DHCPv6 request, and are
handled much like permanent address requests, except that the client may ask for a new temporary address
only a short time later. When the client does so, a different temporary address will be returned; a repeated
request for a permanent address, on the other hand, would usually return the same address as before.
When the DHCPv6 server returns a temporary address, it may of course keep a log of this address. The
absence of such a log is one reason SLAAC may provide a greater degree of privacy. Another concern
is that the DHCPv6 temporary-address sequence might have a flaw that would allow a remote observer to
infer a relationship between different temporary addresses; with SLAAC, a host is responsible itself for the
security of its temporary-address sequence and is not called upon to trust an external entity.
A DHCPv6 response contains a list (perhaps of length 1) of IPv6 addresses. Each separate address has an
expiration date. The client must send a new request before the expiration of any address it is actually using;
unlike for DHCPv4, there is no separate address lease lifetime.
In DHCPv4, the host portion of addresses typically comes from address pools representing small ranges
of integers such as 64-254; these values are generally allocated consecutively. A DHCPv6 server, on the
other hand, should take advantage of the enormous range (264 ) of possible host portions by allocating values
more sparsely, through the use of pseudorandomness. This makes it very difficult for an outsider who knows
one of a sites host addresses to guess the addresses of other hosts. Some DHCPv6 servers, however, do not
yet support this; such servers make the SLAAC approach more attractive.

8.7.4 Manual Configuration

Like IPv4 addresses, IPv6 addresses can also be set up purely by manual configuration. In theory, this would
be done only in the absence of an IPv6 router, in which case a unique-local prefix (8.5 Network Prefixes)
would likely be appropriate. See 8.10.3 Manual address configuration for one example. While it might

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be convenient to distribute only the /64 prefix via manual configuration, and have SLAAC supply the loworder 64 bits, this option is not described in the SLAAC RFCs and seems not to be available in common
implementations.

8.8 Globally Exposed Addresses

Perhaps the most striking difference between a contemporary IPv4 network and an IPv6 network is that on
the former, many hosts are likely to be hidden behind a NAT router (1.14 Network Address Translation).
On an IPv6 network, on the other hand, every host is likely to be globally visible to the IPv6 world (though
NAT may still be used to allow connectivity to legacy IPv4 servers).
Legacy IPv4 NAT routers provide a measure of each of privacy, security and nuisance. Privacy in IPv6 can
be handled, as above, through private or temporary addresses.
The degree of security provided via NAT is largely if not entirely due to the fact that all connections must
be initiated from the inside; no packet from the outside is allowed through the NAT firewall unless it is a
response to a packet sent from the inside. This feature, however, can also be implemented via a conventional firewall (IPv4 or IPv6); one might need to configure the firewall with a list of inside addresses (or
subnets) meant to be globally visible. Furthermore, given such a conventional firewall, it would then be
straightforward to modify it so as to support limited and regulated connections from the outside world as
desired; an analogous modification of a NAT router is quite difficult. (It remains true, however, that as of
this writing consumer-grade IPv6 firewalls of the type described here do not really exist.) Finally, one of
the major reasons for hiding IPv4 addresses is that with IPv4 it is easy to map a /24 subnet by pinging or
otherwise probing each of the 254 possible hosts; such mapping may reveal internal structure. In IPv6 such
mapping is impractical as a /64 subnet has 264 18 quintillion hosts.
As for nuisance, NAT has always broken protocols that involve negotiation of new connections (eg TFTP, or
SIP, used by VoIP); IPv6 should make these much easier to manage.

8.9 ICMPv6
RFC 4443 defines an updated version of the ICMP protocol. It includes an IPv6 version of ICMP Echo
Request / Echo Reply, upon which the ping command is based. It also handles the error conditions below;
this list is somewhat cleaner than the corresponding ICMPv4 list:
Destination Unreachable
In this case, one of the following numeric codes is returned:
0. No route to destination, returned when a router has no next_hop entry.
1. Communication with destination administratively prohibited, returned when a router has a
next_hop entry, but declines to use it for policy reasons. Codes 5 and 6 are special cases; these
more-specific codes are returned when appropriate.
2. Beyond scope of source address, returned when a router is, for example, asked to route a packet
to a global address, but the return address is site-local. In IPv4, when a host with a private address
attempts to connect to a global address, NAT is almost always involved.

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3. Address unreachable, a catchall category for routing failure not covered by any other message. An
example is if the packet was successfully routed to the last_hop router, but Neighbor Discovery failed
to find a LAN address corresponding to the IPv6 address.
4. Port unreachable, returned when, as in ICMPv4, the destination host does not have the requested
UDP port open.
5. Source address failed ingress/egress policy, see code 1.
6. Reject route to destination, see code 1.
Packet Too Big
This is like ICMPv4s Fragmentation Required but DontFragment flag sent; IPv6 however has no routerbased fragmentation.
Time Exceeded
This is used for cases where the Hop Limit was exceeded, and also where source-based fragmentation was
used and the fragment-reassembly timer expired.
Parameter Problem
This is used when there is a malformed entry in the IPv6 header, eg an unrecognized Next Header value.

8.10 Routerless Connection Examples

Most IPv6 networks require at least one IPv6 router; both SLAAC and DHCPv6 configuration requires
this. However, link-local addresses are quite serviceable, as long as one remembers that the interface must
be specified; manually configured addresses are another option. Here are some examples. One practical
problem with link-local addresses is that application documentation describing how to include a specification
of the interface is sometimes sparse.

8.10.1 ping6
We will start with the linux version of ping6, the IPv6 analogue of the familiar ping command. It is
used to send ICMPv6 Echo Requests. The ping6 command supports an option to specify the interface (-I
eth0); as noted above, this is mandatory when sending to link-local addresses.
ping6 ::1: This allows me to ping my own loopback address.
ping6 -I eth0 ff02::1: This pings the all-nodes multicast group on interface eth0. I get these answers:
64 bytes from fe80::3e97:eff:fe2c:2beb (this is the host I am pinging from)
64 bytes from fe80::2a0:ccff:fe24:b0e4 (another linux host)
My VoIP phone on the same subnet but apparently supporting IPv4 only remains mute.
ping6 -I eth0 fe80::6267:20ff:fe72:8960: This pings the link-local address of the other linux host answering
the previous query. Note the ff:fe in the host identifier. Also note the flipped seventh bit of the two bytes
02a0; the other linux host has Ethernet address 00:a0:cc:24:b0:e4.

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8.10.2 TCP connections with link-local addresses

The next step is to create a TCP connection. Some commands, like ping6 above, may provide for a way of
specifying the interface as an option. Failing that, many linux systems allow appending %interface to the
address (which unfortunately usually is required to be in numeric form). The following, for example, allows
me to connect using ssh to the other linux host above:
ssh fe80::2a0:ccff:fe24:b0e4%eth0
That the ssh service was listening for IPv6 connections can be verified on that host by
netstat -a | grep -i tcp6
That the ssh connection actually used IPv6 can be verified by, say, use of a network sniffer like WireShark
(for which the filter expression ip.version == 6 may be useful).
If the connection fails, but ssh works for IPv4 connections and shows as listening in the tcp6 list from the
netstat command, a blocked port is a likely suspect. In this case the commands ufw and ip6tables
--list may be useful.

8.10.3 Manual address configuration

The use of manually configured addresses, 8.7.4 Manual Configuration, is also possible; this avoids the
need to specify an interface. The first step is to pick a unique-local prefix, eg fd37:dead:beef:cafe::0/64 (note
that this particular prefix does not meet the randomness rules for unique-local address prefixes). Low-order
bits can then be added and addresses assigned manually; on linux this is done with:
host1: ip -6 address add fd37:dead:beef:cafe::1 dev eth0
host2: ip -6 address add fd37:dead:beef:cafe::2 dev eth0
Now on host1 the command
ssh fd37:dead:beef:cafe::2
should work, again assuming ssh is listening for IPv6 connections. Because the address here is not linklocal, /etc/host entries may also be created, eg the following on host1:
fd37:dead:beef:cafe::2 host2-6
Now to connect all that should be needed is
ssh host2-6

8.10.4 Node Information Messages

In addition to ICMPv6, IPv6 also has Node Information (NI) Messages, defined in RFC 4620. One form of
NI query allows a host to be asked directly for its name; this is accomplished in IPv4 via DNS reverse-name
lookups. Other NI queries allow a host to be asked for its other IPv6 addresses, or for its IPv4 addresses.

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8.11 IPv6-to-IPv4 connectivity

What happens if you switch to IPv6 completely, perhaps because your ISP has run out of IPv4 addresses?
Ideally you will only need to talk to IPv6 servers. For example, the DNS name microsoft.com corresponds to an IPv4 address, but also to an IPv6 address. If there is not IPv6 connectivity between you and
the IPv6 server you are trying to reach, tunneling of IPv6 traffic over IPv4 can be used.
But what do you do if you have only an IPv6 address and want to reach an IPv4-only server? IPv4 and IPv6
cannot directly interoperate; while IPv6 could have been modified to support an interoperability feature, a
change of such magnitude in IPv4 is politically and economically impossible. Thus, you will have to find
some sort of translator, such as an IPv6-to-IPv4 NAT router. RFC 2766 defines an IPv6-to-IPv4 flavor of
NAT that also includes any necessary translation of the higher transport layer as well. The NAT translator
will have an IPv4 address (otherwise it cannot talk to other IPv4 nodes), but that address can be shared
among multiple IPv6 clients.

8.12 Epilog
IPv4 has run out of large address blocks, as of 2011. IPv6 has reached a mature level of development. Most
common operating systems provide excellent IPv6 support.
Yet conversion has been slow. Many ISPs still provide limited (to nonexistent) support, and inexpensive
IPv6 firewalls to replace the ubiquitous consumer-grade NAT routers do not really exist. Time will tell how
all this evolves. However, while IPv6 has now been around for twenty years, top-level IPv4 address blocks
disappeared just three years ago. It is quite possible that this will prove to be just the catalyst IPv6 needs.

8.13 Exercises
1. Each IPv6 address is associated with a specific solicited-node multicast address. Explain why, on a
typical Ethernet, if the original IPv6 host address was obtained via SLAAC then the LAN multicast group
corresponding to the hosts solicited-node multicast addresses is likely to be small, in many cases consisting
of one host only. (Packet delivery to small LAN multicast groups can be much more efficient than delivery
to large multicast groups.)
(b). What steps might a DHCPv6 server take to ensure that, for the IPv6 addresses it hands out, the LAN
multicast groups corresponding to the host addresses solicited-node multicast addresses will be small?
2. If an attacker sends a large number of probe packets via IPv4, you can block them by blocking the
attackers IP address. Now suppose the attacker uses IPv6 to launch the probes; for each probe, the attacker
changes the low-order 64 bits of the address. Can these probes be blocked efficiently? If so, what do you
have to block? Might you also be blocking other users?
3. Suppose someone tried to implement ping6 so that, if the address was a link-local address and no interface
was specified, the ICMPv6 Echo Request was sent out all non-loopback interfaces. Could the end result be
different than conventional ping6 with the correct interface supplied? If so, how likely is this?
4. Create an IPv6 ssh connection as in 8.10 Routerless Connection Examples. Examine the connections
packets using WireShark or the equivalent. Does the TCP handshake (12.3 TCP Connection Establishment)
look any different over IPv6?
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5. Create an IPv6 ssh connection using manually configured addresses as in 8.10.3 Manual address
configuration. Again use WireShark or the equivalent to monitor the connection. Is DAD (8.7.1 Duplicate
Address Detection) used?

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9 ROUTING-UPDATE ALGORITHMS

How do IP routers build and maintain their forwarding tables?

Ethernet bridges can always fall back on broadcast, so they can afford to build their forwarding tables
incrementally, putting a host into the forwarding table only when that host is first seen as a sender. For
IP, there is no fallback delivery mechanism: forwarding tables must be built before delivery can succeed.
While manual table construction is possible, it is not practical.
In the literature it is common to refer to router-table construction as routing algorithms. We will avoid
that term, however, to avoid confusion with the fundamental datagram-forwarding algorithm; instead, we
will call these routing-update algorithms.
The two classes of algorithms we will consider here are distance-vector and link-state. Both assume that
consistent information is available as to the cost of each link (eg that the two routers at opposite ends of each
link know this cost). This requirement classifies these algorithms as interior routing-update algorithms: the
routers involved are internal to a larger organization or other common administrative regime that has an
established policy on how to assign link weights. The set of routers following a common policy is known as
a routing domain or (from the BGP protocol) an autonomous system. The simplest link-weight strategy
is to give each link a cost of 1; link costs can also be based on bandwidth, propagation delay, financial cost,
or administrative preference value.
In the following chapter we will look at the Border Gateway Protocol, or BGP, in which no link-cost calculations are made, because links may be between unrelated organizations which have no agreement in place
on determining link cost.
Generally, all these algorithms apply to IPv6 as well as IPv4, though specific protocols of course may need
modification.
Finally, we should point out that from the early days of the Internet, routing was allowed to depend not just
on the destination, but also on the quality of service (QoS) requested; thus, forwarding table entries are
strictly speaking not destination, next_hop but rather destination, QoS, next_hop. Originally, the Type
of Service field in the IPv4 header (7.1 The IPv4 Header) could be used to specify QoS (often then called
ToS). Packets could request low delay, high throughput or high reliability, and could be routed accordingly.
In practice, the Type of Service field was rarely used, and was eventually taken over by the DS field and
ECN bits. The first three bits of the Type of Service field, known as the precedence bits, remain available,
however, and can still be used for QoS routing purposes (see the Class Selector PHB of 19.7 Differentiated
Services for examples of these bits). See also RFC 2386.
In much of the following, we are going to ignore QoS information, and assume that routing decisions
are based only on the destination. See, however, the first paragraph of 9.5 Link-State Routing-Update
Algorithm, and also 9.6 Routing on Other Attributes.

9.1 Distance-Vector Routing-Update Algorithm

Distance-Vector is the simplest routing-update algorithm, used by the Routing Information Protocol, or RIP.
On unix-based systems the process in charge of this is often called routed (pronounced route-d).

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Routers identify their router neighbors (through some sort of neighbor-discovery mechanism), and
add a third column to their forwarding tables for cost; table entries are thus of the form
destination,next_hop,cost. The simplest case is to assign a cost of 1 to each link (the hopcount metric); it is also possible to assign more complex numbers.
Each router then reports the destination,cost portion of its table to its neighboring routers at regular
intervals (these table portions are the vectors of the algorithm name). It does not matter if neighbors
exchange reports at the same time, or even at the same rate.
Each router also monitors its continued connectivity to each neighbor; if neighbor N becomes unreachable
then its reachability cost is set to infinity.
Actual destinations in IP would be networks attached to routers; one router might be directly connected
to several such destinations. In the following, however, we will identify all a routers directly connected
networks with the router itself. That is, we will build forwarding tables to reach every router. While
it is possible that one destination network might be reachable by two or more routers, thus breaking our
identification of a router with its set of attached networks, in practice this is of little concern. See exercise 4
for an example in which networks are not identified with adjacent routers.

9.1.1 Distance-Vector Update Rules

Let A be a router receiving a report D,cD from neighbor N at distance cN . Note that this means A can
reach D via N with cost c = cD + cN . A updates its own table according to the following three rules:
1. New destination: D is a previously unknown destination. A adds D,N,cD +cN
2. Lower cost: D is a known destination with entry D,M,c, but c > cD +cN . A updates the entry for D
to D,N,cD +cN . It is possible that M=N, meaning that N is now reporting a distance decrease to D.
3. Next_hop increase: A has a previous entry D,N,c, but c < cD +cN . A updates the entry for D to
D,N,cD +cN . N is As next_hop to D, and N is now reporting a distance increase.
The first two rules are for new destinations and a shorter path to existing destinations. In these cases, the cost
to each destination monotonically decreases (at least if we consider all unreachable destinations as being at
distance ). Convergence is automatic, as the costs cannot decrease forever.
The third rule, however, introduces the possibility of instability, as a cost may also go up. It represents the
bad-news case, in that neighbor N has learned that some link failure has driven up its own cost to reach D,
and is now passing that bad news on to A, which routes to D via N.
The next_hop-increase case only passes bad news along; the very first cost increase must always come from
a router discovering that a neighbor N is unreachable, and thus updating its cost to N to . Similarly, if
router A learns of a next_hop increase to destination D from neighbor B, then we can follow the next_hops
back until we reach a router C which is either the originator of the cost= report, or which has learned of
an alternative route through one of the first two rules.

9.1.2 Example 1
For our first example, no links will break and thus only the first two rules above will be used. We will start
out with the network below with empty forwarding tables; all link costs are 1.

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After initial neighbor discovery, here are the forwarding tables. Each node has entries only for its directly
connected neighbors:
A: B,B,1 C,C,1 D,D,1
B: A,A,1 C,C,1
C: A,A,1 B,B,1 E,E,1
D: A,A,1 E,E,1
E: C,C,1 D,D,1
Now let D report to A; it sends records A,1 and E,1. A ignores Ds A,1 record, but E,1 represents a
new destination; A therefore adds E,D,2 to its table. Similarly, let A now report to D, sending B,1 C,1
D,1 E,2 (the last is the record we just added). D ignores As records D,1 and E,2 but As records
B,1 and C,1 cause D to create entries B,A,2 and C,A,2. A and Ds tables are now, in fact, complete.
Now suppose C reports to B; this gives B an entry E,C,2. If C also reports to E, then Es table will have
A,C,2 and B,C,2. The tables are now:
A: B,B,1 C,C,1 D,D,1 E,D,2
B: A,A,1 C,C,1 E,C,2
C: A,A,1 B,B,1 E,E,1
D: A,A,1 E,E,1 B,A,2 C,A,2
E: C,C,1 D,D,1 A,C,2 B,C,2
We have two missing entries: B and C do not know how to reach D. If A reports to B and C, the tables will
be complete; B and C will each reach D via A at cost 2. However, the following sequence of reports might
also have occurred:
E reports to C, causing C to add D,E,2
C reports to B, causing B to add D,C,3
In this case we have 100% reachability but B routes to D via the longer-than-necessary path BCED.
However, one more report will fix this: suppose A reports to B. B will received D,1 from A, and will
update its entry D,C,3 to D,A,2.
Note that A routes to E via D while E routes to A via C; this asymmetry was due to indeterminateness in the
order of initial table exchanges.
If all link weights are 1, and if each pair of neighbors exchange tables once before any pair starts a second
exchange, then the above process will discover the routes in order of length, ie the shortest paths will be the
first to be discovered. This is not, however, a particularly important consideration.

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9.1.3 Example 2
The next example illustrates link weights other than 1. The first route discovered between A and B is the
direct route with cost 8; eventually we discover the longer ACDB route with cost 2+1+3=6.

The initial tables are these:

A: C,C,2 B,B,8
B: A,A,8 D,D,3
C: A,A,2 D,D,1
D: B,B,3 C,C,1
After A and C exchange, A has D,C,3 and C has B,A,10. After C and D exchange, C updates its B,A,10
entry to B,D,4 and D adds A,C,3; D receives Cs report of B,10 but ignores it. Now finally suppose
B and D exchange. D ignores Bs route to A, as it has a better one. B, however, gets Ds report A,3 and
updates its entry for A to A,D,6. At this point the tables are as follows:
A: C,C,2 B,B,8 D,C,3
B: A,D,6 D,D,3
C: A,A,2 D,D,1 B,D,4
D: B,B,3 C,C,1 A,C,3
We have two more things to fix before we are done: A has an inefficient route to B, and B has no route to
C. The first will be fixed when C reports B,4 to A; A will replace its route to B with B,C,6. The second
will be fixed when D reports to B; if A reports to B first then B will temporarily add the inefficient route
C,A,10; this will change to C,D,4 when Ds report to B arrives. If we look only at the AB route, B
discovers the lower-cost route to A once, first, C reports to D and, second, after that, D reports to B; a similar
sequence leads to As discovering the lower-cost route.

9.1.4 Example 3
Our third example will illustrate how the algorithm proceeds when a link breaks. We return to the first
diagram, with all tables completed, and then suppose the DE link breaks. This is the bad-news case: a
link has broken, and is no longer available; this will bring the third rule into play.

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We shall assume, as above, that A reaches E via D, but we will here assume contrary to Example 1 that
C reaches D via A (see exercise 3.5 for the original case).
Initially, upon discovering the break, D and E update their tables to E,-, and D,-, respectively
(whether or not they actually enter into their tables is implementation-dependent; we may consider this
as equivalent to removing their entries for one another; the - as next_hop indicates there is no next_hop).
Eventually D and E will report the break to their respective neighbors A and C. A will apply the bad-news
rule above and update its entry for E to E,-,. We have assumed that C, however, routes to D via A, and
so it will ignore Es report.
We will suppose that the next steps are for C to report to E and to A. When C reports its route D,2 to E,
E will add the entry D,C,3, and will again be able to reach D. When C reports to A, A will add the route
E,C,2. The final step will be when A next reports to D, and D will have E,A,3. Connectivity is restored.

9.1.5 Example 4
The previous examples have had a global perspective in that we looked at the entire network. In the next
example, we look at how one specific router, R, responds when it receives a distance-vector report from its
neighbor S. Neither R nor S nor we have any idea of what the entire network looks like. Suppose Rs table
is initially as follows, and the SR link has cost 1:
destination
A
B
C
D

next_hop
S
T
S
U

cost
3
4
5
6

S now sends R the following report, containing only destinations and its costs:
destination
A
B
C
D
E

cost
2
3
5
4
2

R then updates its table as follows:

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destination
A
B
C
D
E

next_hop
S
T
S
S
S

cost
3
4
6
5
3

reason
No change; S probably sent this report before
No change; Rs cost via S is tied with Rs cost via T
Next_hop increase
Lower-cost route via S
New destination

Whatever Ss cost to a destination, Rs cost to that destination via S is one greater.

9.2 Distance-Vector Slow-Convergence Problem

There is a significant problem with Distance-Vector table updates in the presence of broken links. Not only
can routing loops form, but the loops can persist indefinitely! As an example, suppose we have the following
arrangement, with all links having cost 1:

Now suppose the DA link breaks:

If A immediately reports to B that D is no longer reachable (distance = ), then all is well. However, it is
possible that B reports to A first, telling A that it has a route to D, with cost 2, which B still believes it has.
This means A now installs the entry D,B,3. At this point we have what we called in 1.6 Routing Loops a
linear routing loop: if a packet is addressed to D, A will forward it to B and B will forward it back to A.
Worse, this loop will be with us a while. At some point A will report D,3 to B, at which point B will
update its entry to D,A,4. Then B will report D,4 to A, and As entry will be D,B,5, etc. This process
is known as slow convergence to infinity. If A and B each report to the other once a minute, it will take
2,000 years for the costs to overflow an ordinary 32-bit integer.

9.2.1 Slow-Convergence Fixes

The simplest fix to this problem is to use a small value for infinity. Most flavors of the RIP protocol use
infinity=16, with updates every 30 seconds. The drawback to so small an infinity is that no path through
the network can be longer than this; this makes paths with weighted link costs difficult. Cisco IGRP uses a
variable value for infinity up to a maximum of 256; the default infinity is 100.
There are several well-known other fixes:

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9.2.1.1 Split Horizon

Under split horizon, if A uses N as its next_hop for destination D, then A simply does not report to N that
it can reach D; that is, in preparing its report to N it first deletes all entries that have N as next_hop. In the
example above, split horizon would mean B would never report to A about the reachability of D because A
is Bs next_hop to D.
Split horizon prevents all linear routing loops. However, there are other topologies where it cannot prevent
loops. One is the following:

Suppose the A-D link breaks, and A updates to D,-,. A then reports D, to B, which updates its
table to D,-,. But then, before A can also report D, to C, C reports D,2 to B. B then updates to
D,C,3, and reports D,3 back to A; neither this nor the previous report violates split-horizon. Now As
entry is D,B,4. Eventually A will report to C, at which point Cs entry becomes D,A,5, and the numbers
keep increasing as the reports circulate counterclockwise. The actual routing proceeds in the other direction,
clockwise.
Split horizon often also includes poison reverse: if A uses N as its next_hop to D, then A in fact reports
D, to N, which is a more definitive statement that A cannot reach D by itself. However, coming up with
a scenario where poison reverse actually affects the outcome is not trivial.
9.2.1.2 Triggered Updates
In the original example, if A was first to report to B then the loop resolved immediately; the loop occurred
if B was first to report to A. Nominally each outcome has probability 50%. Triggered updates means that
any router should report immediately to its neighbors whenever it detects any change for the worse. If A
reports first to B in the first example, the problem goes away. Similarly, in the second example, if A reports
to both B and C before B or C report to one another, the problem goes away. There remains, however, a
small window where B could send its report to A just as A has discovered the problem, before A can report
to B.
9.2.1.3 Hold Down
Hold down is sort of a receiver-side version of triggered updates: the receiver does not use new alternative
routes for a period of time (perhaps two router-update cycles) following discovery of unreachability. This
gives time for bad news to arrive. In the first example, it would mean that when A received Bs report D,2,
it would set this aside. It would then report D, to B as usual, at which point B would now report D,
back to A, at which point Bs earlier report D,2 would be discarded. A significant drawback of hold down
is that legitimate new routes are also delayed by the hold-down period.

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These mechanisms for preventing slow convergence are, in the real world, quite effective. The Routing
Information Protocol (RIP, RFC 2453) implements all but hold-down, and has been widely adopted at
smaller installations.
However, the potential for routing loops and the limited value for infinity led to the development of alternatives. One alternative is the link-state strategy, 9.5 Link-State Routing-Update Algorithm. Another
alternative is Ciscos Enhanced Interior Gateway Routing Protocol, or EIGRP, 9.4.2 EIGRP. While part of
the distance-vector family, EIGRP is provably loop-free, though to achieve this it must sometimes suspend
forwarding to some destinations while tables are in flux.

9.3 Observations on Minimizing Route Cost

Does distance-vector routing actually achieve minimum costs? For that matter, does each packet incur the
cost its sender expects? Suppose node A has a forwarding entry D,B,c, meaning that A forwards packets
to destination D via next_hop B, and expects the total cost to be c. If A sends a packet to D, and we follow
it on the actual path it takes, must the total link cost be c? If so, we will say that the network has accurate
costs.
The answer to the accurate-costs question, as it turns out, is yes for the distance-vector algorithm, if we
follow the rules carefully, and the network is stable (meaning that no routing reports are changing, or, more
concretely, that every update report now circulating is based on the current network state); a proof is below.
However, if there is a routing loop, the answer is of course no: the actual cost is now infinite. The answer
would also be no if As neighbor B has just switched to using a longer route to D than it last reported to A.
It turns out, however, that we seek the shortest route not because we are particularly trying to save money
on transit costs; a route 50% longer would generally work just fine. (AT&T, back when they were the
Phone Company, once ran a series of print advertisements claiming longer routes as a feature: if the direct
path was congested, they could still complete your call by routing you the long way round.) However, we
are guaranteed that if all routers seek the shortest route and if the network is stable then all paths are
loop-free, because in this case the network will have accurate costs.
Here is a simple example illustrating the importance of global cost-minimization in preventing loops. Suppose we have a network like this one:

Now suppose that A and B use distance-vector but are allowed to choose the shortest route to within 10%. A
would get a report from C that D could be reached with cost 1, for a total cost of 21. The forwarding entry
via C would be D,C,21. Similarly, A would get a report from B that D could be reached with cost 21, for
a total cost of 22: D,B,22. Similarly, B has choices D,C,21 and D,A,22.
If A and B both choose the minimal route, no loop forms. But if A and B both use the 10%-overage rule,
they would be allowed to choose the other route: A could choose D,B,22 and B could choose D,A,22.

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If this happened, we would have a routing loop: A would forward packets for D to B, and B would forward
them right back to A.
As we apply distance-vector routing, each router independently builds its tables. A router might have some
notion of the path its packets would take to their destination; for example, in the case above A might believe
that with forwarding entry D,B,22 its packets would take the path ABCD (though in distance-vector
routing, routers do not particularly worry about the big picture). Consider again the accurate-cost question
above. This fails in the 10%-overage example, because the actual path is now infinite.
We now prove that, in distance-vector routing, the network will have accurate costs, provided
each router selects what it believes to be the shortest path to the final destination, and
the network is stable, meaning that further dissemination of any reports would not result in changes
To see this, suppose the actual route taken by some packet from source to destination, as determined by
application of the distributed algorithm, is longer than the cost calculated by the source. Choose an example
of such a path with the fewest number of links, among all such paths in the network. Let S be the source,
D the destination, and k the number of links in the actual path P. Let Ss forwarding entry for D be D,N,c,
where N is Ss next_hop neighbor. To have obtained this route through the distance-vector algorithm, S must
have received report D,cD from N, where we also have the cost of the SN link as cN and c = cD + cN . If
we follow a packet from N to D, it must take the same path P with the first link deleted; this sub-path has
length k-1 and so, by our hypothesis that k was the length of the shortest path with non-accurate costs, the
cost from N to D is cD . But this means that the cost along path P, from S to D via N, must be cD + cN = c,
contradicting our selection of P as a path longer than its advertised cost.
There is one final observation to make about route costs: any cost-minimization can occur only within
a single routing domain, where full information about all links is available. If a path traverses multiple
routing domains, each separate routing domain may calculate the optimum path traversing that domain. But
these local minimums do not necessarily add up to a globally minimal path length, particularly when
one domain calculates the minimum cost from one of its routers only to the other domain rather than to a
router within that other domain. Here is a simple example. Routers BR1 and BR2 are the border routers
connecting the domain LD to the left of the vertical dotted line with domain RD to the right. From A to B,
LD will choose the shortest path to RD (not to B, because LD is not likely to have information about links
within RD). This is the path of length 3 through BR2. But this leads to a total path length of 3+8=11 from
A to B; the global minimum path length, however, is 4+1=5, through BR1.

In this example, domains LD and RD join at two points. For a route across two domains joined at only a
single point, the domain-local shortest paths do add up to the globally shortest path.

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9.4 Loop-Free Distance Vector Algorithms

It is possible for routing-update algorithms based on the distance-vector idea to eliminate routing loops
and thus the slow-convergence problem entirely. We present brief descriptions of two such algorithms.

9.4.1 DSDV
DSDV, or Destination-Sequenced Distance Vector, was proposed in [PB94]. It avoids routing loops by
the introduction of sequence numbers: each router will always prefer routes with the most recent sequence
number, and bad-news information will always have a lower sequence number then the next cycle of corrected information.
DSDV was originally proposed for MANETs (3.3.8 MANETs) and has some additional features for traffic
minimization that, for simplicity, we ignore here. It is perhaps best suited for wired networks and for small,
relatively stable MANETs.
DSDV forwarding tables contain entries for every other reachable node in the system. One successor of
DSDV, Ad Hoc On-Demand Distance Vector routing or AODV, allows forwarding tables to contain only
those destinations in active use; a mechanism is provided for discovery of routes to newly active destinations.
See [PR99] and RFC 3561.
Under DSDV, each forwarding table entry contains, in addition to the destination, cost and next_hop, the current sequence number for that destination. When neighboring nodes exchange their distance-vector reachability reports, the reports include these per-destination sequence numbers.
When a router R receives a report from neighbor N for destination D, and the report contains a sequence
number larger than the sequence number for D currently in Rs forwarding table, then R always updates to
use the new information. The three cost-minimization rules of 9.1.1 Distance-Vector Update Rules above
are used only when the incoming and existing sequence numbers are equal.
Each time a router R sends a report to its neighbors, it includes a new value for its own sequence number,
which it always increments by 2. This number is then entered into each neighbors forwarding-table entry for
R, and is then propagated throughout the network via continuing report exchanges. Any sequence number
originating this way will be even, and whenever another nodes forwarding-table sequence number for R is
even, then its cost for R will be finite.
Infinite-cost reports are generated in the usual way when former neighbors discover they can no longer reach
one another; however, in this case each node increments the sequence number for its former neighbor by 1,
thus generating an odd value. Any forwarding-table entry with infinite cost will thus always have an odd
sequence number. If A and B are neighbors, and As current sequence number is s, and the AB link breaks,
then B will start reporting A at distance with sequence number s+1 while A will start reporting its own
new sequence number s+2. Any other node now receiving a report originating with B (with sequence number
s+1) will mark A as having cost , but will obtain a valid route to A upon receiving a report originating
from A with new (and larger) sequence number s+2.
The triggered-update mechanism is used: if a node receives a report with some destinations newly marked
with infinite cost, it will in turn forward this information immediately to its other neighbors, and so on. This
is, however, not essential; bad and good reports are distinguished by sequence number, not by relative
arrival time.

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It is now straightforward to verify that the slow-convergence problem is solved. After a link break, if there is
some alternative path from router R to destination D, then R will eventually receive Ds latest even sequence
number, which will be greater than any sequence number associated with any report listing D as unreachable.
If, on the other hand, the break partitioned the network and there is no longer any path to D from R, then the
highest sequence number circulating in Rs half of the original network will be odd and the associated table
entries will all list D at cost . One way or another, the network will quickly settle down to a state where
every destinations reachability is accurately described.
In fact, a stronger statement is true: not even transient routing loops are created. We outline a proof. First,
whenever router R has next_hop N for a destination D, then Ns sequence number for D must be greater
than or equal to Rs, as R must have obtained its current route to D from one of Ns reports. A consequence
is that all routers participating in a loop for destination D must have the same (even) sequence number s for
D throughout. This means that the loop would have been created if only the reports with sequence number
s were circulating. As we noted in 9.1.1 Distance-Vector Update Rules, any application of the next_hopincrease rule must trace back to a broken link, and thus must involve an odd sequence number. Thus, the
loop must have formed from the sequence-number-s reports by the application of the first two rules only.
But this violates the claim in Exercise 10.
There is one drawback to DSDV: nodes may sometimes briefly switch to routes that are longer than optimum
(though still correct). This is because a router is required to use the route with the newest sequence number,
even if that route is longer than the existing route. If A and B are two neighbors of router R, and B is closer
to destination D but slower to report, then every time Ds sequence number is incremented R will receive
As longer route first, and switch to using it, and Bs shorter route shortly thereafter.
DSDV implementations usually address this by having each router R keep track of the time interval between
the first arrival at R of a new route to a destination D with a given sequence number, and the arrival of the
best route with that sequence number. During this interval following the arrival of the first report with a new
sequence number, R will use the new route, but will refrain from including the route in the reports it sends
to its neighbors, anticipating that a better route will soon arrive.
This works best when the hopcount cost metric is being used, because in this case the best route is likely
to arrive first (as the news had to travel the fewest hops), and at the very least will arrive soon after the first
route. However, if the networks cost metric is unrelated to the hop count, then the time interval between
first-route and best-route arrivals can involve multiple update cycles, and can be substantial.

9.4.2 EIGRP
EIGRP, or the Enhanced Interior Gateway Routing Protocol, is a once-proprietary Cisco distance-vector
protocol that was released as an Internet Draft in February 2013. As with DSDV, it eliminates the risk of
routing loops, even ephemeral ones. It is based on the distributed update algorithm (DUAL) of [JG93].
EIGRP is an actual protocol; we present here only the general algorithm. Our discussion follows [CH99].
Each router R keeps a list of neighbor routers NR , as with any distance-vector algorithm. Each R also
maintains a data structure known (somewhat misleadingly) as its topology table. It contains, for each
destination D and each N in NR , an indication of whether N has reported the ability to reach D and, if so, the
reported cost c(D,N). The router also keeps, for each N in NR , the cost cN of the link from R to N. Finally,
the forwarding-table entry for any destination can be marked passive, meaning safe to use, or active,
meaning updates are in process and the route is temporarily unavailable.

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Initially, we expect that for each router R and each destination D, Rs next_hop to D in its forwarding table
is the neighbor N for which the following total cost is a minimum:
c(D,N) + cN
Now suppose R receives a distance-vector report from neighbor N1 that it can reach D with cost c(D,N1 ).
This is processed in the usual distance-vector way, unless it represents an increased cost and N1 is Rs
next_hop to D; this is the third case in 9.1.1 Distance-Vector Update Rules. In this case, let C be Rs
current cost to D, and let us say that neighbor N of R is a feasible next_hop (feasible successor in Ciscos
terminology) if Ns cost to D (that is, c(D,N)) is strictly less than C. R then updates its route to D to use the
feasible neighbor N for which c(D,N) + cN is a minimum. Note that this may not in fact be the shortest path;
it is possible that there is another neighbor M for which c(D,M)+cM is smaller, but c(D,M)C. However,
because Ns path to D is loop-free, and because c(D,N) < C, this new path through N must also be loop-free;
this is sometimes summarized by the statement one cannot create a loop by adopting a shorter route.
If no neighbor N of R is feasible which would be the case in the DAB example of 9.2 Distance-Vector
Slow-Convergence Problem, then R invokes the DUAL algorithm. This is sometimes called a diffusion
algorithm as it invokes a diffusion-like spread of table changes proceeding away from R.
Let C in this case denote the new cost from R to D as based on N1 s report. R marks destination D as
active (which suppresses forwarding to D) and sends a special query to each of its neighbors, in the form
of a distance-vector report indicating that its cost to D has now increased to C. The algorithm terminates
when all Rs neighbors reply back with their own distance-vector reports; at that point R marks its entry for
D as passive again.
Some neighbors may be able to process Rs report without further diffusion to other nodes, remain passive,
and reply back to R immediately. However, other neighbors may, like R, now become active and continue
the DUAL algorithm. In the process, R may receive other queries that elicit its distance-vector report; as
long as R is active it will report its cost to D as C. We omit the argument that this process and thus the
network must eventually converge.

9.5 Link-State Routing-Update Algorithm

Link-state routing is an alternative to distance-vector. It is often though certainly not always considered
to be the routing-update algorithm class of choice for networks that are sufficiently large, such as those of
ISPs.
In distance-vector routing, each node knows a bare minimum of network topology: it knows nothing about
links beyond those to its immediate neighbors. In the link-state approach, each node keeps a maximum
amount of network information: a full map of all nodes and all links. Routes are then computed locally
from this map, using the shortest-path-first algorithm. The existence of this map allows, in theory, the
calculation of different routes for different quality-of-service requirements. The map also allows calculation
of a new route as soon as news of the failure of the existing route arrives; distance-vector protocols on the
other hand must wait for news of a new route after an existing route fails.
Link-state protocols distribute network map information through a modified form of broadcast of the status
of each individual link. Whenever either side of a link notices the link has died (or if a node notices that
a new link has become available), it sends out link-state packets (LSPs) that flood the network. This
broadcast process is called reliable flooding; in general broadcast protocols work poorly with networks that
have even small amounts of topological looping (that is, redundant paths). Link-state protocols must ensure
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that every router sees every LSP, and also that no LSPs circulate repeatedly. (The acronym LSP is used by
a link-state implementation known as IS-IS; the preferred acronym used by the Open Shortest Path First
(OSPF) implementation is LSA, where A is for advertisement.) LSPs are sent immediately upon link-state
changes, like triggered updates in distance-vector protocols except there is no race between bad news
and good news.
It is possible for ephemeral routing loops to exist; for example, if one router has received a LSP but another
has not, they may have an inconsistent view of the network and thus route to one another. However, as soon
as the LSP has reached all routers involved, the loop should vanish. There are no race conditions, as with
distance-vector routing, that can lead to persistent routing loops.
The link-state flooding algorithm avoids the usual problems of broadcast in the presence of loops by having
each node keep a database of all LSP messages. The originator of each LSP includes its identity, information
about the link that has changed status, and also a sequence number. Other routers need only keep in their
databases the LSP packet with the largest sequence number; older LSPs can be discarded. When a router
receives a LSP, it first checks its database to see if that LSP is old, or is current but has been received before;
in these cases, no further action is taken. If, however, an LSP arrives with a sequence number not seen
before, then in typical broadcast fashion the LSP is retransmitted over all links except the arrival interface.
As an example, consider the following arrangement of routers:

Suppose the AE link status changes. A sends LSPs to C and B. Both these will forward the LSPs to D;
suppose Bs arrives first. Then D will forward the LSP to C; the LSP traveling CD and the LSP traveling
DC might even cross on the wire. D will ignore the second LSP copy that it receives from C and C will
ignore the second copy it receives from D.
It is important that LSP sequence numbers not wrap around. (Protocols that do allow a numeric field to wrap
around usually have a clear-cut idea of the active range that can be used to conclude that the numbering has
wrapped rather than restarted; this is harder to do in the link-state context.) OSPF uses lollipop sequencenumbering here: sequence numbers begin at -231 and increment to 231 -1. At this point they wrap around
back to 0. Thus, as long as a sequence number is less than zero, it is guaranteed unique; at the same time,
routing will not cease if more than 231 updates are needed. Other link-state implementations use 64-bit
sequence numbers.
Actual link-state implementations often give link-state records a maximum lifetime; entries must be periodically renewed.

9.5.1 Shortest-Path-First Algorithm

The next step is to compute routes from the network map, using the shortest-path-first algorithm. Here is
our example network:

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The shortest path from A to D is A-B-C-D, which has cost 3+4+2=9.

We build the set R of all shortest-path routes iteratively. Initially, R contains only the 0-length route to the
start node; one new destination and route is added to R at each stage of the iteration. At each stage we have
a current node, representing the node most recently added to R. The initial current node is our starting
node, in this case, A.
We will also maintain a set T, for tentative, of routes to other destinations. This is also initialized to empty.
At each stage, we find all nodes which are immediate neighbors of the current node and which do not
already have routes in the set R. For each such node N, we calculate the length of the route from the start
node to N that goes through the current node. We see if this is our first route to N, or if the route improves
on any route to N already in T; if so, we add or update the route in T accordingly.
At the end of this process, we choose the shortest path in T, and move the route and destination node to R.
The destination node of this shortest path becomes the next current node. Ties can be resolved arbitrarily,
but note that, as with distance-vector routing, we must choose the minimum or else the accurate-costs
property will fail.
We repeat this process until all nodes have routes in the set R.
For the example above, we start with current = A and R = {A,A,0}. The set T will be {B,B,3, C,C,10,
D,D,11}. The shortest entry is B,B,3, so we move that to R and continue with current = B.
For the next stage, the neighbors of B without routes in R are C and D; the routes from A to these through B
are C,B,7 and D,B,12. The former is an improvement on the existing T entry C,C,10 and so replaces
it; the latter is not an improvement over D,D,11. T is now {C,B,7, D,D,11}. The shortest route in T is
that to C, so we move this node and route to R and set C to be current.
For the next stage, D is the only non-R neighbor; the path from A to D via C has entry D,B,9, an improvement over the existing D,D,11 in T. The only entry in T is now D,B,9; this is the shortest and thus we
move it to R.
We now have routes in R to all nodes, and are done.
Here is another example, again with links labeled with costs:

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We start with current = A. At the end of the first stage, B,B,3 is moved into R, T is {D,D,12}, and
current is B. The second stage adds C,B,5 to T, and then moves this to R; current then becomes C. The
third stage introduces the route (from A) D,B,10; this is an improvement over D,D,12 and so replaces it
in T; at the end of the stage this route to D is moved to R.
A link-state source node S computes the entire path to a destination D. But as far as the actual path that a
packet sent by S will take to D, S has direct control only as far as the first hop N. While the accurate-cost
rule we considered in distance-vector routing will still hold, the actual path taken by the packet may differ
from the path computed at the source, in the presence of alternative paths of the same length. For example,
S may calculate a path SNAD, and yet a packet may take path SNBD, so long as the NAD and
NBD paths have the same length.
Link-state routing allows calculation of routes on demand (results are then cached), or larger-scale calculation. Link-state also allows routes calculated with quality-of-service taken into account, via straightforward
extension of the algorithm above.

9.6 Routing on Other Attributes

There is sometimes a desire to route on attributes other than the destination, or the destination and QoS
bits. This kind of routing is decidedly nonstandard, and is generally limited to the imposition of very local
routing policies. It is, however, often available.
On linux systems this is part of the Linux Advanced Routing facility, often grouped with some advanced
queuing features known as Traffic Control; the combination is referred to as LARTC.
As a simple example of what can be done, suppose a site has two links L1 and L2 to the Internet, where L1
is faster and L2 serves more as a backup; in such a configuration the default route to the Internet will often
be via L1. A site may wish to route some outbound traffic via L2, however, for any of the following reasons:
the traffic may involve protocols deemed lower in priority (eg email)
the traffic may be real-time traffic that can benefit from reduced competition on L2
the traffic may come from lower-priority senders; eg some customers within the site may be relegated
to using L2 because they are paying less
In the first two cases, routing might be based on the destination port numbers; in the third, it might be based
on the source IP address.
Note that nothing can be done in the inbound direction unless L1 and L2 lead to the same ISP, and even
there any special routing would be at the discretion of that ISP.
The trick with LARTC is to be compatible with existing routing-update protocols; this would be a problem if
the kernel forwarding table simply added columns for other packet attributes that neighboring non-LARTC
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routers knew nothing about. Instead, the forwarding table is split up into multiple dest, next_hop (or dest,
QoS, next_hop) tables. One of these tables is the main table, and is the table that is updated by routingupdate protocols interacting with neighbors. Before a packet is forwarded, administratively supplied rules
are consulted to determine which table to apply; these rules are allowed to consult other packet attributes.
The collection of tables and rules is known as the routing policy database.
As a simple example, in the situation above the main table would have an entry default, L1 (more precisely,
it would have the IP address of the far end of the L1 link instead of L1 itself). There would also be another
table, perhaps named slow, with a single entry default, L2. If a rule is created to have a packet routed
using the slow table, then that packet will be forwarded via L2. Here is one such linux rule, applying to
traffic from host 10.0.0.17:
ip rule add from 10.0.0.17 table slow

Now suppose we want to route traffic to port 25 (the SMTP port) via L2. This is harder; linux provides no
support here for routing based on port numbers. However, we can instead use the iptables mechanism to
mark all packets destined for port 25, and then create a routing-policy rule to have such marked traffic
use the slow table. The mark is known as the forwarding mark, or fwmark; its value is 0 by default. The
fwmark is not actually part of the packet; it is associated with the packet only while the latter remains
within the kernel.
iptables --table mangle --append PREROUTING \\
--protocol tcp --source-port 25 --jump MARK --set-mark 1
ip rule add fwmark 1 table slow

Consult the applicable man pages for further details.

The iptables mechanism can also be used to set the IP precedence bits, so that a single standard IP forwarding
table can be used, though support for the IP precedence bits is limited.

9.7 Epilog
At this point we have concluded the basics of IP routing, involving routing within large (relatively) homogeneous organizations such as multi-site corporations or Internet Service Providers. Every router involved
must agree to run the same protocol, and must agree to a uniform assignment of link costs.
At the very largest scales, these requirements are impractical. The next chapter is devoted to this issue of
very-large-scale IP routing, eg on the global Internet.

9.8 Exercises
1. Suppose the network is as follows, where distance-vector routing update is used. Each link has cost 1,
and each router has entries in its forwarding table only for its immediate neighbors (so As table contains
B,B,1, D,D,1 and Bs table contains A,A,1, C,C,1).
A

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(a). Suppose each node creates a report from its initial configuration and sends that to each of its neighbors.
What will each nodes forwarding table be after this set of exchanges? The exchanges, in other words, are
all conducted simultaneously; each node first sends out its own report and then processes the reports
arriving from its two neighbors.
(b). What will each nodes table be after the simultaneous-and-parallel exchange process of part (a) is
repeated a second time?
Hint: you do not have to go through each exchange in detail; the only information added by an exchange is
additional reachability information.

2. Now suppose the configuration of routers has the link weights shown below.
A

2
D

C
12

(a). As in the previous exercise, give each nodes forwarding table after each node exchanges with its
immediate neighbors simultaneously and in parallel.
(b). How many iterations of such parallel exchanges will it take before C learns to reach F via B; that is,
before it creates the entry F,B,11? Count the answer to part (a) as the first iteration.

3. Suppose a router R has the following distance-vector table:

destination
A
B
C
D
E

cost
5
6
7
8
9

next hop
R1
R1
R2
R2
R3

R now receives the following report from R1; the cost of the RR1 link is 1.
destination
A
B
C
D
E
F

9.8 Exercises

cost
4
7
7
6
8
8

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Give Rs updated table after it processes R1s report. For each entry that changes, give a brief explanation,
in the style of 9.1.5 Example 4.
3.5. At the start of Example 3 (9.1.4 Example 3), we changed Cs routing table so that it reached D via A
instead of via E: Cs entry D,E,2 was changed to D,A,2. This meant that C had a valid route to D at the
start.
How might the scenario of Example 3 play out if Cs table had not been altered? Give a sequence of reports
that leads to correct routing between D and E.
4. In the following exercise, A-D are routers and the attached networks N1-N6, which are the ultimate
destinations, are shown explicitly. Routers still exchange distance-vector reports with neighboring routers,
as usual. If a router has a direct connection to a network, you may report the next_hop as direct, eg, from
As table, N1,direct,0
A

N1

N3
N6

N5

N4
N2

(a). Give the initial tables for A through D, before any distance-vector exchanges.
(b). Give the tables after each router A-D exchanges with its immediate neighbors simultaneously and in
parallel.
(c). At the end of (b), what networks are not known by what routers?

5. Suppose A, B, C, D and E are connected as follows. Each link has cost 1, and so each forwarding table
is uniquely determined; Bs table is A,A,1, C,C,1, D,A,2, E,C,2. Distance-vector routing update is
used.

Now suppose the DE link fails, and so D updates its entry for E to E,-,.

(a). Give As table after D reports E, to A

(b). Give Bs table after A reports to B
(c). Give As table after B reports to A; note that B has an entry E,C,2
(d). Give Ds table after A reports to D.

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6. Consider the network in 9.2.1.1 Split Horizon:, using distance-vector routing updates.

D,A,2

D,A,2

Suppose the DA link breaks and then

A reports D, to B (as before)
C reports D,2 to B (as before)
A now reports D, to C (instead of B reporting D,3 to A)

(a). What reports (a pair should suffice) will lead to the formation of the routing loop?
(b). What (single) report will eliminate the possibility of the routing loop?

7. Suppose the network of 9.2 Distance-Vector Slow-Convergence Problem is changed to the following.
Distance-vector update is used; again, the AD link breaks.
D,D,1
A

D,E,2
B

(a). Explain why Bs report back to A, after A reports D,-,, is now valid.
(b). Explain why hold down (9.2.1.3 Hold Down) will delay the use of the new route ABED.

8. Suppose the routers are A, B, C, D, E and F, and all link costs are 1. The distance-vector forwarding
tables for A and F are below. Give the network with the fewest links that is consistent with these tables.
Hint: any destination reached at cost 1 is directly connected; if X reaches Y via Z at cost 2, then Z and Y
must be directly connected.
As table
dest
B
C
D
E
F

cost
1
1
2
2
3

next_hop
B
C
C
C
B

Fs table

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dest
A
B
C
D
E

cost
3
2
2
1
1

next_hop
E
D
D
D
E

9. (a) Suppose routers A and B somehow end up with respective forwarding-table entries D,B,n and
D,A,m, thus creating a routing loop. Explain why the loop may be removed more quickly if A and B both
use poison reverse with split horizon, versus if A and B use split horizon only.
(b). Suppose the network looks like the following. The AB link is extremely slow.
A
C

Suppose A and B send reports to each other advertising their routes to D, and immediately afterwards the
CD link breaks and C reports to A and B that D is unreachable. After those unreachability reports are
processed, A and Bs reports sent to each other before the break finally arrive. Explain why the network is
now in the state described in part (a).
10. Suppose the distance-vector algorithm is run on a network and no links break (so by the last paragraph
of 9.1.1 Distance-Vector Update Rules the next_hop-increase rule is never applied).

(a). Prove that whenever A is Bs next_hop to destination D, then As cost to D is strictly less than Bs.
Hint: assume that if this claim is true, then it remains true after any application of the rules in
9.1.1 Distance-Vector Update Rules. If the lower-cost rule is applied to B after receiving a report from A,
resulting in a change to Bs cost to D, then one needs to show As cost is less than Bs, and also Bs new
cost is less than that of any neighbor C that uses B as its next_hop to D.
(b). Use (a) to prove that no routing loops ever form.

11. Give a scenario illustrating how a (very temporary!) routing loop might form in link-state routing.
12. Use the Shortest Path First algorithm to find the shortest path from A to E in the network below. Show
the sets R and T, and the node current, after each step.

13. Suppose you take a laptop, plug it into an Ethernet LAN, and connect to the same LAN via Wi-Fi. From
laptop to LAN there are now two routes. Which route will be preferred? How can you tell which way traffic
is flowing? How can you configure your OS to prefer one path or another?

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In the previous chapter we considered two classes of routing-update algorithms: distance-vector and linkstate. Each of these approaches requires that participating routers have agreed not just to a common protocol,
but also to a common understanding of how link costs are to be assigned. We will address this further below
in 10.6 Border Gateway Protocol, BGP, but the basic problem is that if one site prefers the hop-count
approach, assigning every link a cost of 1, while another site prefers to assign link costs in proportion to
their bandwidth, then path cost comparisons between the two sites simply cannot be done. In general, we
cannot even translate costs from one site to another, because the paths themselves depend on the cost
assignment strategy.
The term routing domain is used to refer to a set of routers under common administration, using a common
link-cost assignment. Another term for this is autonomous system. While use of a common routingupdate protocol within the routing domain is not an absolute requirement for example, some subnets may
internally use distance-vector while the sites backbone routers use link-state we can assume that all
routers have a uniform view of the sites topology and cost metrics.
One of the things included in the term large-scale IP routing is the coordination of routing between multiple routing domains. Even in the earliest Internet there were multiple routing domains, if for no other
reason than that how to measure link costs was (and still is) too unsettled to set in stone. However, another
component of large-scale routing is support for hierarchical routing, above the level of subnets; we turn to
this next.

10.1 Classless Internet Domain Routing: CIDR

CIDR is the mechanism for supporting hierarchical routing in the Internet backbone. Subnetting moves the
network/host division line further rightwards; CIDR allows moving it to the left as well. With subnetting,
the revised division line is visible only within the organization that owns the IP network address; subnetting
is not visible outside. CIDR allows aggregation of IP address blocks in a way that is visible to the Internet
backbone.
When CIDR was introduced in 1993, the following were some of the justifications for it, all relating to the
increasing size of the backbone IP forwarding tables, and expressed in terms of the then-current Class A/B/C
mechanism:
The Internet is running out of Class B addresses (this happened in the mid-1990s)
There are too many Class Cs (the most numerous) for backbone forwarding tables to be efficient
Eventually IANA (the Internet Assigned Numbers Authority) will run out of IP addresses (this happened in 2011)
Assigning non-CIDRed multiple Class Cs in lieu of a single Class B would have helped with the first point
in the list above, but made the second point worse.
Ironically, the current (2013) very tight market for IP address blocks is likely to lead to larger and larger
backbone IP forwarding tables, as sites are forced to use multiple small address blocks instead of one large
block.
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By the year 2000, CIDR had essentially eliminated the Class A/B/C mechanism from the backbone Internet,
and had more-or-less completely changed how backbone routing worked. You purchased an address block
from a provider or some other IP address allocator, and it could be whatever size you needed, from /27 to
/15.
What CIDR enabled is IP routing based on an address prefix of any length; the Class A/B/C mechanism of
course used fixed prefix lengths of 8, 16 and 24 bits. Furthermore, CIDR allows different routers, at different
levels of the backbone, to route on prefixes of different lengths.
CIDR was formally introduced by RFC 1518 and RFC 1519. For a while there were strategies in place to
support compatibility with non-CIDR-aware routers; these are now obsolete. In particular, it is no longer
appropriate for large-scale routers to fall back on the Class A/B/C mechanism in the absence of CIDR
information; if the latter is missing, the routing should fail.
The basic strategy of CIDR is to consolidate multiple networks going to the same destination into a single
entry. Suppose a router has four class Cs all to the same destination:
200.7.0.0/24 foo
200.7.1.0/24 foo
200.7.2.0/24 foo
200.7.3.0/24 foo
The router can replace all these with the single entry
200.7.0.0/22 foo
It does not matter here if foo represents a single ultimate destination or if it represents four sites that just
happen to be routed to the same next_hop.
It is worth looking closely at the arithmetic to see why the single entry uses /22. This means that the first 22
bits must match 200.7.0.0; this is all of the first and second bytes and the first six bits of the third byte. Let
us look at the third byte of the network addresses above in binary:
200.7.000000 00.0/24 foo
200.7.000000 01.0/24 foo
200.7.000000 10.0/24 foo
200.7.000000 11.0/24 foo
The /24 means that the network addresses stop at the end of the third byte. The four entries above cover
every possible combination of the last two bits of the third byte; for an address to match one of the entries
above it suffices to begin 200.7 and then to have 0-bits as the first six bits of the third byte. This is another
way of saying the address must match 200.7.0.0/22.
Most implementations actually use a bitmask, eg FF.FF.FC.00 (in hex) rather than the number 22; note 0xFC
= 1111 1100 with 6 leading 1-bits, so FF.FF.FC.00 has 8+8+6=22 1-bits followed by 10 0-bits.
The IP delivery algorithm of 7.5 The Classless IP Delivery Algorithm still works with CIDR, with the
understanding that the routers forwarding table can now have a network-prefix length associated with any
entry. Given a destination D, we search the forwarding table for network-prefix destinations B/k until we
find a match; that is, equality of the first k bits. In terms of masks, given a destination D and a list of table
entries prefix,mask = B[i],M[i], we search for i such that (D & M[i]) = B[i].
It is possible to have multiple matches, and responsibility for avoiding this is much too distributed to be

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declared illegal by IETF mandate. Instead, CIDR introduced the longest-match rule: if destination D
matches both B1 /k1 and B2 /k2 , with k1 < k2 , then the longer match B2 /k2 match is to be used. (Note that if
D matches two distinct entries B1 /k1 and B2 /k2 then either k1 < k2 or k2 < k1 ).

10.2 Hierarchical Routing

Strictly speaking, CIDR is simply a mechanism for routing to IP address blocks of any prefix length; that is,
for setting the network/host division point to an arbitrary place within the 32-bit IP address.
However, by making this network/host division point variable, CIDR introduced support for routing on
different prefix lengths at different places in the backbone routing infrastructure. For example, top-level
routers might route on /8 or /9 prefixes, while intermediate routers might route based on prefixes of length
14. This feature of routing on fewer bits at one point in the Internet and more bits at another point is exactly
what is meant by hierarchical routing.
We earlier saw hierarchical routing in the context of subnets: traffic might first be routed to a class-B site
147.126.0.0/16, and then, within that site, to subnets such as 147.126.1.0/24, 147.126.2.0/24, etc. But with
CIDR the hierarchy can be much more flexible: the top level of the hierarchy can be much larger than the
customer level, lower levels need not be administratively controlled by the higher levels (as is the case
with subnets), and more than two levels can be used.
CIDR is an address-block-allocation mechanism; it does not directly speak to the kinds of policy we might
wish to implement with it. Here are three possible applications; the latter two involve hierarchical routing:
Application 1 (legacy): CIDR allows IANA to allocate multiple blocks of Class C, or fragments of a
Class A, to a single customer, so as to require only a single forwarding-table entry
Application 2 (legacy): CIDR allows opportunistic aggregation of routes: a router that sees the four
200.7.x.0/24 routes above in its table may consolidate them into a single entry.
Application 3 (current): CIDR allows huge provider blocks, with suballocation by the provider. This
is known as provider-based routing.
Application 4 (hypothetical): CIDR allows huge regional blocks, with suballocation within the region, somewhat like the original scheme for US phone numbers with area codes. This is known as
geographical routing.
Hierarchical routing does introduce one new wrinkle: the routes chosen may no longer be globally optimal.
Suppose, for example, we are trying to route to prefix 200.20.0.0/16. Suppose at the top level that traffic
to 200.0.0.0/8 is routed to router R1, and R1 then routes traffic to 200.20.0.0/16 to R2. A packet sent
to 200.20.1.2 by an independent router R3 would always pass through R1, even if there were a shorter
path R3R4R2 that bypassed R1. This is addressed in further detail below in 10.4.3 Provider-Based
Hierarchical Routing.
We can avoid this by declaring that CIDR will only be used when it turns out that, in the language of the
example of the preceding paragraph, the best path to reach 200.20.0.0/16 is always through R1. But this is
seldom the case, and is even less likely to remain the case as new links are added. Such a policy also defeats
much of the potential benefit of CIDR at reducing router forwarding-table size by supporting the creation
of arbitrary-sized address blocks and then routing to them as a single unit. The Internet backbone might
be much happier if all routers simply had to maintain a single entry 200.0.0.0/8, R1, versus 256 entries
200.x.0.0/16, R1 for all but one value of x.
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10.3 Legacy Routing

Back in the days of NSFNet, the Internet backbone was a single routing domain. While most customers did
not connect directly to the backbone, the intervening providers tended to be relatively compact, geographically that is, regional and often had a single primary routing-exchange point with the backbone. IP
addresses were allocated to subscribers directly by the IANA, and the backbone forwarding tables contained
entries for every site, even the Class Cs.
Because the NSFNet backbone and the regional providers did not necessarily share link-cost information,
routes were even at this early point not necessarily globally optimal; compromises and approximations were
made. However, in the NSFNet model routers generally did find a reasonable approximation to the shortest
path to each site referenced by the backbone tables. While the legacy backbone routing domain was not
all-encompassing, if there were differences between two routes, at least the backbone portions the longest
components would be identical.

10.4 Provider-Based Routing

In provider-based routing, large CIDR blocks are allocated to large-scale providers. The different providers
each know how to route to one another. Subscribers (usually) obtain their IP addresses from within their
providers blocks; thus, traffic from the outside is routed first to the provider, and then, within the providers
routing domain, to the subscriber. We may even have a hierarchy of providers, so packets would be routed
first to the large-scale provider, and eventually to the local provider. There may no longer be a central
backbone; instead, multiple providers may each build parallel transcontinental networks.
Here is a simpler example, in which providers have unique paths to one another. Suppose we have providers
P0, P1 and P2, with customers as follows:
P0: customers A,B,C
P1: customers D,E
P2: customers F,G
We will also assume that each provider has an IP address block as follows:
P0: 200.0.0.0/8
P1: 201.0.0.0/8
P2: 202.0.0.0/8
Let us now allocate addresses to the customers:
A: 200.0.0.0/16
B: 200.1.0.0/16
C: 200.2.16.0/20 (16 = 0001 0000)

D: 201.0.0.0/16
E: 201.1.0.0/16

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F: 202.0.0.0/16
G: 202.1.0.0/16

The routing model is that packets are first routed to the appropriate provider, and then to the customer.
While this model may not in general guarantee the shortest end-to-end path, it does in this case because
each provider has a single point of interconnection to the others. Here is the network diagram:

With this diagram, P0s forwarding table looks something like this:
destination
200.0.0.0/16
200.1.0.0/16
200.2.16.0/20
201.0.0.0/8
202.0.0.0/8

next_hop
A
B
C
P1
P2

That is, P0s table consists of

one entry for each of P0s own customers
one entry for each other provider
If we had 1,000,000 customers divided equally among 100 providers, then each providers table would have
only 10,099 entries: 10,000 for its own customers and 99 for the other providers. Without CIDR, each
providers forwarding table would have 1,000,000 entries.
Even if we have some additional secondary links, that is, additional links that do not create alternative paths between providers, the routing remains relatively straightforward. Shown here are the private
customer-to-customer links CD and EF; these are likely used only by the customers they connect. Two
customers are multi-homed; that is, they have connections to alternative providers: AP1 and EP2. Typically, though, while A and E may use these secondary links all they want for outbound traffic, their respective
inbound traffic would still go through their primary providers P0 and P1 respectively.

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10.4.1 Internet Exchange Points

The long links joining providers in these diagrams are somewhat misleading; providers do not always like
sharing long links and the attendant problems of sharing responsibility for failures. Instead, providers often connect to one another at Internet eXchange Points or IXPs; the link P0
P1 might actually be
P0
IXP P1, where P0 owns the left-hand link and P1 the right-hand. IXPs can either be third-party
sites open to all providers, or private exchange points. The term Metropolitan Area Exchange, or MAE,
appears in the names of the IXPs MAE-East, originally near Washington DC, and MAE-West, originally in
San Jose, California; each of these is now actually a set of IXPs. MAE in this context is now a trademark.

10.4.2 CIDR and Staying Out of Jail

Suppose we want to change providers. One way we can do this is to accept a new IP-address block from the
new provider, and change all our IP addresses. The paper Renumbering: Threat or Menace [LKCT96] was
frequently cited at least in the early days of CIDR as an intimation that such renumbering was inevitably
a Bad Thing. In principle, therefore, we would like to allow at least the option of keeping our IP address
allocation while changing providers.
An address-allocation standard that did not allow changing of providers might even be a violation of the
US Sherman Antitrust Act; see American Society of Mechanical Engineers v Hydrolevel Corporation, 456
US 556 (1982). The IETF thus had the added incentive of wanting to stay out of jail, when writing the
CIDR standard so as to allow portability between providers (actually, antitrust violations usually involve
civil penalties).
The CIDR longest-match rule turns out to be exactly what we (and the IETF) need. Suppose, in the diagrams
above, that customer C wants to move from P0 to P1, and does not want to renumber. What routing changes
need to be made? One solution is for P0 to add a route 200.2.16.0/20, P1 that routes all of Cs traffic to
P1; P1 will then forward that traffic on to C. P1s table will be as follows, and P1 will use the longest-match
rule to distinguish traffic for its new customer C from traffic bound for P0.
destination
200.0.0.0/8
202.0.0.0/8
201.0.0.0/16
201.1.0.0/16
200.2.16.0/20

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P2
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This does work, but all Cs inbound traffic except for that originating in P1 will now be routed through
Cs ex-provider P0, which as an ex-provider may not be on the best of terms with C. Also, the routing is
inefficient: Cs traffic from P2 is routed P2P0P1 instead of the more direct P2P1.
A better solution is for all providers other than P1 to add the route 200.2.16.0/20, P1. While traffic to
200.0.0.0/8 otherwise goes to P0, this particular sub-block is instead routed by each provider to P1. The
important case here is P2, as a stand-in for all other providers and their routers: P2 routes 200.0.0.0/8 traffic
to P0 except for the block 200.2.16.0/20, which goes to P1.
Having every other provider in the world need to add an entry for C is going to cost some money, and, one
way or another, C will be the one to pay. But at least there is a choice: C can consent to renumbering (which
is not difficult if they have been diligent in using DHCP and perhaps NAT too), or they can pay to keep their
old address block.
As for the second diagram above, with the various private links (shown as dashed lines), it is likely that the
longest-match rule is not needed for these links to work. As private link to P1 might only mean that
A can send outbound traffic via P1
P1 forwards As traffic to A via the private link
P2, in other words, is still free to route to A via P0. P1 may not advertise its route to A to anyone else.

10.4.3 Provider-Based Hierarchical Routing

With provider-based routing, the route taken may no longer be end-to-end optimal; we have replaced the
problem of finding an optimal route from A to B with the two problems of finding an optimal route from
A to Bs provider, and then from that provider entry point to B; the second strategy may not yield the same
result. This second strategy mirrors the two-step routing process of first routing on the address bits that
identify the provider, and then routing on the address bits including the subscriber portion.
Consider the following example, in which providers P1 and P2 have three interconnection links (r1s1,
r2s2 and r3s3), each with cost 1. We assume that P1s costs happen to be comparable with P2s costs.

The globally shortest path between A and B is via the r2s2 crossover, with total length 6+1+5=12. However,
traffic from A to B will be routed by P1 to its closest crossover to P2, namely the r3s3 link. The total path is
2+1+8+5=16. Traffic from B to A will be routed by P2 via the r1s1 crossover, for a length of 2+1+7+6=16.

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This routing strategy is sometimes called hot-potato routing; each provider tries to get rid of any traffic (the
potatoes) as quickly as possible, by routing to the closest exit point.
Not only are the paths taken inefficient, but the AB and BA paths are now asymmetric. This can be
a problem if forward and reverse timings are critical, or if one of P1 or P2 has significantly more bandwidth
or less congestion than the other. In practice, however, route asymmetry is of little consequence.
As for the route inefficiency itself, this also is not necessarily a significant problem; the primary reason
routing-update algorithms focus on the shortest path is to guarantee that all computed paths are loop-free.
As long as each half of a path is loop-free, and the halves do not intersect except at their common midpoint,
these paths too will be loop-free.
The BGP MED value (10.6.5.3 MULTI_EXIT_DISC) offers an optional mechanism for P1 to agree that
AB traffic should take the r1s1 crossover. This might be desired if P1s network were better and
customer A was willing to pay extra to keep its traffic within P1s network as long as possible.

10.5 Geographical Routing

The classical alternative to provider-based routing is geographical routing; the archetypal model for this is
the telephone area code system. A call from anywhere in the US to Loyola Universitys main switchboard,
773-274-3000, would traditionally be routed first to the 773 area code in Chicago. From there the call would
be routed to the north-side 274 exchange, and from there to subscriber 3000. A similar strategy can be used
for IP routing.
Geographical addressing has some advantages. Figuring out a good route to a destination is usually straightforward, and close to optimal in terms of the path physical distance. Changing providers never involves
renumbering (though moving may). And approximate IP address geolocation (determining a hosts location
from its IP address) is automatic.
Geographical routing has some minor technical problems. First, routing may be inefficient between immediate neighbors A and B that happen to be split by a boundary for larger geographical areas; the path might
go from A to the center of As region to the center of Bs region and then to B. Another problem is that some
larger sites (eg large corporations) are themselves geographically distributed; if efficiency is the goal, each
office of such a site would need a separate IP address block appropriate for its physical location.
But the real issue with geographical routing is apparently the business question of who carries the traffic.
The provider-based model has a very natural answer to this: every link is owned by a specific provider. For
geographical IP routing, my local provider might know at once from the prefix that a packet of mine is to be
delivered from Chicago to San Francisco, but who will carry it there? My provider might have to enter into
different traffic contracts for multiple different regions. If different local providers make different arrangements for long-haul packet delivery, the routing efficiency (at least in terms of table size) of geographical
routing is likely lost. Finally, there is no natural answer for who should own those long inter-region links. It
may be useful to recall that the present area-code system was created when the US telephone system was an
AT&T monopoly, and the question of who carried traffic did not exist.
That said, the top five Regional Internet Registries represent geographical regions (usually continents), and
provider-based addressing is below that level. That is, the IANA handed out address blocks to the geographical RIRs, and the RIRs then allocated address blocks to providers.
At the intercontinental level, geography does matter: some physical link paths are genuinely more expensive

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than other (shorter) paths. It is much easier to string terrestrial cable than undersea cable. However, within
a continent physical distance does not always matter as much as might be supposed. Furthermore, a large
geographically spread-out provider can always divide up its address blocks by region, allowing internal
geographical routing to the correct region.
Here is a diagram of IP address allocation as of 2006: http://xkcd.com/195.

10.6 Border Gateway Protocol, BGP

In 9 Routing-Update Algorithms, we considered interior routing-update protocols. For both DistanceVector and Link State methods, the per-link cost played an essential role: by trying to minimize the cost, we
were assured that no routing loops would be present in a stable network.
But when systems under different administration (eg two large ISPs) talk to each other, comparing metrics
simply does not work. Each sides metrics may be based on any of the following:
hopcount
bandwidth
cost
congestion
One providers metric may even use larger numbers for better routes (though if this is done then total path
costs, or preference, cannot be obtained simply by adding the per-link values). Any attempt at comparison
of values from the different sides is a comparison of apples and oranges.
The Border Gateway Protocol, or BGP, is assigned the job of handling exchange of routing information
between neighboring independent organizations; this is sometimes called exterior routing. The current
version is BGP-4, documented in RFC 4271. The BGP term for a routing domain under coordinated administration, and using one consistent link-cost metric throughout, is Autonomous System, or AS. That said,
all that is strictly required is that all BGP routers within an AS have the same consistent view of routing, and
in fact some Autonomous Systems do run multiple routing protocols and may even use different metrics at
different points. As indicated above, BGP does not support the exchange of link-cost information between
Autonomous Systems.
BGP also has a second goal, in addition to the purely technical problem of finding routes in the absence of
cost information: BGP also provides support for policy-based routing; that is, for making routing decisions
based on managerial or administrative input (perhaps regarding who is paying what for the traffic carried).
Every non-leaf site (and some large leaf sites) has one or more BGP speakers: the routers that run BGP.
If there is more than one, they must remain coordinated with one another so as to present a consistent view
of the sites connections and advertisements; this coordination process is sometimes called internal BGP
to distinguish it from the communication with neighboring Autonomous Systems. The latter process is then
known as external BGP.
The BGP speakers of a site are often not the busy border routers that connect directly to the neighboring AS,
though they are usually located near them and are often on the same subnet. Each interconnection point with
a neighboring AS generally needs its own BGP speaker. Connections between BGP speakers of neighboring
Autonomous Systems sometimes called BGP peers are generally configured administratively; they are
not subject to a neighbor discovery process like that used by most interior routers.
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The BGP speakers must maintain a database of all routes received, not just of the routes actually used.
However, the speakers exchange with neighbors only the routes they (and thus their AS) use themselves;
this is a firm BGP rule.
The current BGP standard is RFC 4271.

10.6.1 AS-paths
At its most basic level, BGP involves the exchange of lists of reachable destinations, like distance-vector
routing without the distance information. But that strategy, alone, cannot avoid routing loops. BGP solves
the loop problem by having routers exchange, not just destination information, but also the entire path used
to reach each destination. Paths including each router would be too cumbersome; instead, BGP abbreviates
the path to the list of ASs traversed; this is called the AS-path. This allows routers to make sure their routes
do not traverse any AS more than once, and thus do not have loops.
As an example of this, consider the network below, in which we consider Autonomous Systems also to be
destinations. Initially, we will assume that each AS discovers its immediate neighbors. AS3 and AS5 will
then each advertise to AS4 their routes to AS2, but AS4 will have no reason at this level to prefer one route
to the other (BGP does use the shortest AS-path as part of its tie-breaking rule, but, before falling back on
that rule, AS4 is likely to have a commercial preference for which of AS3 and AS5 it uses to reach AS2).

Also, AS2 will advertise to AS3 its route to reach AS1; that advertisement will contain the AS-path
AS2,AS1. Similarly, AS3 will advertise this route to AS4 and then AS4 will advertise it to AS5. When
AS5 in turn advertises this AS1-route to AS2, it will include the entire AS-path AS5,AS4,AS3,AS2,AS1,
and AS2 would know not to use this route because it would see that it is a member of the AS-path. Thus,
BGP is spared the kind of slow-convergence problem that traditional distance-vector approaches were subject to.
It is theoretically possible that the shortest path (in the sense, say, of the hopcount metric) from one host to
another traverses some AS twice. If so, BGP will not allow this route.
AS-paths potentially add considerably to the size of the AS database. The number of paths a site must keep
track of is proportional to the number of ASs, because there will be one AS-path to each destination AS.
(Actually, an AS may have to record many times that many AS-paths, as an AS may hear of AS-paths that
it elects not to use.) Typically there are several thousand ASs in the world. Let A be the number of ASs.
Typically the average length of an AS-path is about log(A), although this depends on connectivity. The
amount of memory required by BGP is
CAlog(A) + KN,
where C and K are constants.
The other major goal of BGP is to allow some degree of administrative input to what, for interior routing,

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is largely a technical calculation (though an interior-routing administrator can set link costs). BGP is the
interface between large ISPs, and can be used to implement contractual agreements made regarding which
ISPs will carry other ISPs traffic. If ISP2 tells ISP1 it has a good route to destination D, but ISP1 chooses
not to send traffic to ISP2, BGP can be used to implement this.
Despite the exchange of AS-path information, temporary routing loops may still exist. This is because BGP
may first decide to use a route and only then export the new AS-path; the AS on the other side may realize
there is a problem as soon as the AS-path is received but by then the loop will have at least briefly been in
existence. See the first example below in 10.6.8 Examples of BGP Instability.
BGPs predecessor was EGP, which guaranteed loop-free routes by allowing only a single route to any AS,
thus forcing the Internet into a tree topology, at least at the level of Autonomous Systems. The AS graph
could contain no cycles or alternative routes, and hence there could be no redundancy provided by alternative
paths. EGP also thus avoided having to make decisions as to the preferred path; there was never more than
one choice. EGP was sometimes described as a reachability protocol; its only concern was whether a given
network was reachable.

10.6.2 AS-Paths and Route Aggregation

There is some conflict between the goal of reporting precise AS-paths to each destination, and of consolidating as many address prefixes as possible into a single prefix (single CIDR block). Consider the following
network:

Suppose AS2 has paths

path=AS2, destination 200.0.0/23
path=AS2,AS3, destination 200.0.2/24
path=AS2,AS4, destination 200.0.3/24
If AS2 wants to optimize address-block aggregation using CIDR, it may prefer to aggregate the three destinations into the single block 200.0.0/22. In this case there would be two options for how AS2 reports its
routes to AS1:
Option 1: report 200.0.0/22 with path AS2. But this ignores the ASs AS3 and AS4! These
are legitimately part of the AS-paths to some of the destinations within the block 200.0.0/22; loop
detection could conceivably now fail.
Option 2: report 200.0.0/22 with path AS2,AS3,AS4, which is not a real path but which does
include all the ASs involved. This ensures that the loop-detection algorithm works, but artificially
inflates the length of the AS-path, which is used for certain tie-breaking decisions.
As neither of these options is desirable, the concept of the AS-set was introduced. A list of Autonomous
Systems traversed in order now becomes an AS-sequence. In the example above, AS2 can thus report net
200.0.0/22 with
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AS-sequence=AS2
AS-set={AS3,AS4}
AS2 thus both achieves the desired aggregation and also accurately reports the AS-path length.
The AS-path can in general be an arbitrary list of AS-sequence and AS-set parts, but in cases of simple
aggregation such as the example here, there will be one AS-sequence followed by one AS-set.
RFC 6472 now recommends against using AS-sets entirely, and recommends that aggregation as above be
avoided.

10.6.3 Transit Traffic

It is helpful to distinguish between two kinds of traffic, as seen from a given AS. Local traffic is traffic that
either originates or terminates at that AS; this is traffic that belongs to that AS. At leaf sites (that is, sites
that connect only to their ISP and not to other sites), all traffic is local.
The other kind of traffic is transit traffic; the AS is forwarding it along on behalf of some nonlocal party.
For ISPs, most traffic is transit traffic. A large almost-leaf site might also carry a small amount of transit
traffic for one particular related (but autonomous!) organization.
The decision as to whether to carry transit traffic is a classic example of an administrative choice, implemented by BGPs support for policy-based routing.

10.6.4 BGP Filtering and Policy Routing

As stated above, one of the goals of BGP is to support policy routing; that is, routing based on managerial
or administrative concerns in addition to technical ones. A BGP speaker may be aware of multiple routes to
a destination. To choose the one route that we will use, it may combine a mixture of optimization rules and
policy rules. Some examples of policy rules might be:
do not use AS13 as we have an adversarial relationship with them
do not allow transit traffic
BGP implements policy through filtering rules that is, rules that allow rejection of certain routes at three
different stages:
1. Import filtering is applied to the lists of routes a BGP speaker receives from its neighbors.
2. Best-path selection is then applied as that BGP speaker chooses which of the routes accepted by the
first step it will actually use.
3. Export filtering is done to decide what routes from the previous step a BGP speaker will actually
advertise. A BGP speaker can only advertise paths it uses, but does not have to advertise every such
path.
While there are standard default rules for all these (accept everything imported, use simple tie-breakers,
export everything), a site will usually implement at least some policy rules through this filtering process (eg
prefer routes through the ISP we have a contract with).

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As an example of import filtering, a site might elect to ignore all routes from a particular neighbor, or to
ignore all routes whose AS-path contains a particular AS, or to ignore temporarily all routes from a neighbor
that has demonstrated too much recent route instability (that is, rapidly changing routes). Import filtering
can also be done in the best-path-selection stage. Finally, while it is not commonly useful, import filtering
can involve rather strange criteria; for example, in 10.6.8 Examples of BGP Instability we will consider
examples where AS1 prefers routes with AS-path AS3,AS2 to the strictly shorter path AS2.
The next stage is best-path selection, for which the first step is to eliminate AS-paths with loops. Even if the
neighbors have been diligent in not advertising paths with loops, an AS will still need to reject routes that
contain itself in the associated AS-path.
The next step in the best-path-selection stage, generally the most important in BGP configuration, is to assign
a local_preference, or weight, to each route received. An AS may have policies that add a certain amount to
the local_preference for routes that use a certain AS, etc. Very commonly, larger sites will have preferences
based on contractual arrangements with particular neighbors. Provider ASs, for example, will in general
prefer routes that go through their customers, as these are cheaper. A smaller ISP that connects to two
or more larger ones might be paying to route almost all its outbound traffic through a particular one of the
two; its local_preference values will then implement this choice. After BGP calculates the local_preference
value for every route, the routes with the best local_preference are then selected.
Domains are free to choose their local_preference rules however they wish. Some choices may lead to
instability, below, so domains are encouraged to set their rules in accordance with some standard principles,
also below.
In the event of ties two routes to the same destination with the same local_preference a first tie-breaker
rule is to prefer the route with the shorter AS-path. While this superficially resembles a shortest-path algorithm, the real work should have been done in administratively assigning local_preference values.
Local_preference values are communicated internally via the LOCAL_PREF path attribute, below. They
are not shared with other Autonomous Systems.
The final significant step of the route-selection phase is to apply the Multi_Exit_Discriminator value; we
postpone this until below. A site may very well choose to ignore this value entirely. There may then be
additional trivial tie-breaker rules; note that if a tie-breaker rule assigns significant traffic to one AS over
another, then it may have significant economic consequences and shouldnt be considered trivial. If this
situation is detected, it would probably be addressed in the local-preferences phase.
After the best-path-selection stage is complete, the BGP speaker has now selected the routes it will use. The
final stage is to decide what rules will be exported to which neighbors. Only routes the BGP speaker will
use that is, routes that have made it to this point can be exported; a site cannot route to destination D
through AS1 but export a route claiming D can be reached through AS2.
It is at the export-filtering stage that an AS can enforce no-transit rules. If it does not wish to carry transit
traffic to destination D, it will not advertise D to any of its AS-neighbors.
The export stage can lead to anomalies. Suppose, for example, that AS1 reaches D and AS5 via AS2, and
announces this to AS4.

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Later AS1 switches to reaching D via AS3, but A is forbidden by policy to announce AS3-paths to AS4.
Then A must simply withdraw the announcement to AS4 that it could reach D at all, even though the route
to D via AS2 is still there.

10.6.5 BGP Path attributes

BGP supports the inclusion of various path attributes when exchanging routing information. Attributes
exchanged with neighbors can be transitive or non-transitive; the difference is that if a neighbor AS does
not recognize a received path attribute then it should pass it along anyway if it is marked transitive, but not
otherwise. Some path attributes are entirely local, that is, internal to the AS of origin. Other flags are used
to indicate whether recognition of a path attribute is required or optional, and whether recognition can be
partial or must be complete.
The AS-path itself is perhaps the most fundamental path attribute. Here are a few other common attributes:
10.6.5.1 NEXT_HOP
This mandatory external attribute allows BGP speaker B1 of AS1 to inform its BGP peer B2 of AS2 what
actual router to use to reach a given destination. If B1, B2 and AS1s actual border router R1 are all on the
same subnet, B1 will include R1s IP address as its NEXT_HOP attribute. If B1 is not on the same subnet
as B2, it may not know R1s IP address; in this case it may include its own IP address as the NEXT_HOP
attribute. Routers on AS2s side will then look up the immediate next hop they would use as the first step
to reach B1, and forward traffic there. This should either be R1 or should lead to R1, which will then route
the traffic properly (not necessarily on to B1).

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10.6.5.2 LOCAL_PREF
If one BGP speaker in an AS has been configured with local_preference values, used in the best-pathselection phase above, it uses the LOCAL_PREF path attribute to share those preferences with all other
BGP speakers at a site.
10.6.5.3 MULTI_EXIT_DISC
The Multi-Exit Discriminator, or MED, attribute allows one AS to learn something of the internal structure
of another AS, should it elect to do so. Using the MED information provided by a neighbor has the potential
to cause an AS to incur higher costs, as it may end up carrying traffic for longer distances internally; MED
values received from a neighboring AS are therefore only recognized when there is an explicit administrative
decision to do so.
Specifically, if an autonomous system AS1 has multiple links to neighbor AS2, then AS1 can, when advertising an internal destination D to AS2, have each of its BGP speakers provide associated MED values so
that AS2 can know which link AS1 would prefer that AS2 use to reach D. This allows AS2 to route traffic
to D so that it is carried primarily by AS2 rather than by AS1. The alternative is for AS2 to use only the
closest gateway to AS1, which means traffic is likely carried primarily by AS1.
MED values are considered late in the best-path-selection process; in this sense the use of MED values is a
tie-breaker when two routes have the same local_preference.
As an example, consider the following network (from 10.4.3 Provider-Based Hierarchical Routing, with
providers now replaced by Autonomous Systems); the numeric values on links are their relative costs. We
will assume that border routers R1, R2 and R3 are also AS1s BGP speakers.

In the absence of the MED, AS1 will send traffic from A to B via the R3S3 link, and AS2 will return the
traffic via S1R1. These are the links that are closest to R and S, respectively, representing AS1 and AS2s
desire to hand off the outbound traffic as quickly as possible.
However, AS1s R1, R2 and R3 can provide MED values to AS2 when advertising destination A, indicating
a preference for AS2AS1 traffic to use the rightmost link:
R1: destination A has MED 200
R2: destination A has MED 150

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R3: destination A has MED 100

If this is done, and AS2 abides by this information, then AS2 will route traffic from B to A via the S3R3
link; that is, via the link with the lowest MED value. Note the importance of fact that AS2 is allowed to
ignore the MED; use of it may shift costs from AS1 to AS2!
The relative order of the MED values for R1 and R2 is irrelevant, unless the R3S3 link becomes disabled,
in which case the numeric MED values above would mean that AS2 should then prefer the R2S2 link for
reaching A.
There is no way to use MED values to cause AB traffic to take the R2S2 link; that link is the minimal-cost
link only in the global sense, and the only way to achieve global cost minimization is for the two ASs to
agree to use a common distance metric and a common metric-based routing algorithm, in effect becoming
one AS. While AS1 does provide different numeric MED values for the three cross-links, they are used only
in ranking precedence, not as numeric measures of cost (though they are sometimes derived from that).
As a hypothetical example of why a site might use the MED, suppose site B in the diagram above wants
its customers to experience high-performance downloads. It contracts with AS2, which advertises superior
quality in the downloads experienced by users connecting to servers in its domain. In order to achieve
this superior quality, it builds a particularly robust network S1SS2S3. It then agrees to accept MED
information from other providers so that it can keep outbound traffic in its own network as long as possible,
instead of handing it off to other networks of lower quality. AS2 would thus want BA traffic to travel via
S3 to maximize the portion of the path under its own control.
In the example above, the MED values are used to decide between multiple routes to the same destination
that all pass through the same AS, namely AS1. Some BGP implementations allow the use of MED values to
decide between different routes through different neighbor ASs. The different neighbors must all have the
same local_preference values. For example, AS2 might connect to AS3 and AS4 and receive the following
BGP information:
AS3: destination A has MED 200
AS4: destination A has MED 100
Assuming AS2 assigns the same local_preference to AS3 and AS4, it might be configured to use these MED values as
On Cisco routers, the always-compare-med command is used to create this behavior.
MED values are not intended to be used to communicate routing preferences to non-neighboring ASs.
Additional information on the use of MED values can be found in RFC 4451.
10.6.5.4 COMMUNITY
This is simply a tag to attach to routes. Routes can have multiple tags corresponding to membership in multiple communities. Some communities are defined globally; for example, NO_EXPORT and
NO_ADVERTISE. A route marked with one of these two communities will not be shared further. Other
communities may be relevant only to a particular AS.
The importance of communities is that they allow one AS to place some of its routes into specific categories
when advertising them to another AS; the categories must have been created and recognized by the receiving
AS. The receiving AS is not obligated to honor the community memberships, of course, but doing so has

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the effect of allowing the original AS to configure itself without involving the receiving AS in the process. Communities are often used, for example, by (large) customers of an ISP to request specific routing
treatment.
A customer would have to find out from the provider what communities the provider defines, and what their
numeric codes are. At that point the customer can place itself into the providers community at will.
Here are some of the community values once supported by a no-longer-extant ISP that we shall call AS1.
The full community value would have included AS1s AS-number.
value
90
100
105
110
990
991

action
set local_preference used by AS1 to 90
set local_preference used by AS1 to 100, the default
set local_preference used by AS1 to 105
set local_preference used by AS1 to 110
the route will not leave AS1s domain; equivalent to NO_EXPORT
route will only be exported to AS1s other customers

10.6.6 BGP Policy and Transit Traffic

Perhaps the most common source of simple policy decisions is whether a site wants to accept transit traffic.
As a first example, let us consider the case of configuring a private link, such as the dashed link1 below
between friendly but unaffiliated sites A and B:

If A and B fully advertise link1, by exporting to their respective ISPs routes to each other, then ISP1 (paid
by A) may end up carrying much of Bs traffic or ISP2 (paid by B) may end up carrying much of As traffic.
Economically, these options are not desirable unless fully agreed to by both parties. The primary issue here
is the use of the ISP1A link by B, and the ISP2B link by A; use of the shared link1 might be a secondary
issue depending on the relative bandwidths and A and Bs understandings of appropriate uses for link1.
Three common options A and B might agree to regarding link1 are no-transit, backup, and load-balancing.
For the no-transit option, A and B simply do not export the route to their respective ISPs at all. This is done
via export filtering. If ISP1 does not know A can reach B, it will not send any of Bs traffic to A.
For the backup option, the intent is that traffic to A will normally arrive via ISP1, but if the ISP1 link is down
then As traffic will be allowed to travel through ISP2 and B. To achieve this, A and B can export their link1route to each other, but arrange for ISP1 and ISP2 respectively to assign this route a low local_preference
value. As long as ISP1 hears of a route to B from its upstream provider, it will reach B that way, and will
not advertise the existence of the link1 route to B; ditto ISP2. However, if the ISP2 route to B fails, then As
upstream provider will stop advertising any route to B, and so ISP1 will begin to use the link1 route to B

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and begin advertising it to the Internet. The link1 route will be the primary route to B until ISP2s service is
restored.
A and B must convince their respective ISPs to assign the link1 route a low local_preference; they cannot
mandate this directly. However, if their ISPs recognize community attributes that, as above, allow customers
to influence their local_preference value, then A and B can use this to create the desired local_preference.
For outbound traffic, A and B will need a way to send through one another if their own ISP link is down.
One approach is to consider their default-route path (eg to 0.0.0.0/0) to be a concrete destination within BGP.
ISP1 advertises this to A, using As interior routing protocol, but so does B, and A has configured things so
Bs route has a higher cost. Then A will route to 0.0.0.0/0 through ISP1 that is, will use ISP1 as its default
route as long as it is available, and will switch to B when it is not.
For inbound load balancing, there is no easy fix, in that if ISP1 and ISP2 both export routes to A, then A has
lost all control over how other sites will prefer one to the other. A may be able to make one path artificially
appear more expensive, and keep tweaking this cost until the inbound loads are comparable. Outbound
load-balancing is up to A and B.
Another basic policy question is which of the two available paths site (or regional AS) A uses to reach site
D, in the following diagram. B and C are Autonomous Systems.

How can A express preference for B over C, assuming B and C both advertise to A their routes to D?
Generally A will use a local_preference setting to make the carrier decision for AD traffic, though it is
D that makes the decision for the DA traffic. It is possible (though not customary) for one of the transit
providers to advertise to A that it can reach D, but not advertise to D that it can reach A.
Here is a similar diagram, showing two transit-providing Autonomous Systems B and C connecting at
Internet exchange points IXP1 and IXP2.

B and C each have routers within each IXP. B would probably like to make sure C does not attempt to save
on its long-haul transit costs by forwarding AD traffic over to B at IXP1, and DA traffic over to
B at IXP2. B can avoid this problem by not advertising to C that it can reach A and D. In general, transit
providers are often quite careful about advertising reachability to any other AS for whom they do not intend
to provide transit service, because to do so may implicitly mean getting stuck with that traffic.
If B and C were both to try try to get away with this, a routing loop would be created within IXP1! But in
that case in Bs next advertisement to C at IXP1, B would state that it reaches D via AS-path C (or C,D

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if D were a full-fledged AS), and C would do similarly; the loop would not continue for long.

10.6.7 BGP Relationships

Arbitrarily complex policies may be created through BGP, and, as we shall see in the following section,
convergence to a stable set of routes is not guaranteed. However, most AS policies are determined by a few
simple AS business relationships, as identified in [LG01].
Customer-Provider: A provider contracts to transit traffic from the customer to the rest of the
Internet. A customer may have multiple providers, for backup or for more efficient routing, but
a customer does not accept transit traffic between the providers.
Siblings: Siblings are ISPs that have a connection between them that they intend as a backup
route or for internal traffic. Two siblings may or may not have the same upstream ISP provider
as parent. The siblings do not use their connection for transit traffic except when one of their
primary links is down.
Peers: Peers are two providers that agree to exchange all their customer traffic with each other;
thus, peers do use their connection for transit traffic. Generally the idea is for the interconnection to be seen as equally valuable by both parties (eg because the parties exchange comparable
volumes of traffic); in such a case there would likely be no exchange of cash, but if the volume
flow is significantly asymmetric then compensation can certainly be negotiated.
These business relationships can be described in terms of or even inferred from what routes are accepted
[LG01]. Every AS has, first of all, its customer routes, that is, the routes of its direct customers, and its
customers customers, etc. These might be referred to as the ISPs own routes. Other routes are learned from
peers or, sometimes, providers. Essentially every AS exports its customer routes to all its AS neighbors; the
issue is with provider routes and peer routes. Here are the rules of [LG01]:
Customers do not export to their providers routes they have learned from peers or other providers.
This would make them into transit providers, which is not their job.
Providers do export provider/peer routes to their customers (though a provider may just provide a
single consolidated default route to a single-homed customer).
Peers do not export peer or provider routes to other peers; that is, if ISP P1 has peers P2 and P3, and
P1 learns of a route to destination D through P2, then P1 does not export that route to P3. Instead,
P3 would be expected to have its own relationship with P2, or be a customer of a provider that had a
relationship with P2. Peers do not (usually) provide transit services to third parties.
Siblings do export provider/peer routes to each other, though likely with an understanding about
preference values. If S1 and S2 are two siblings with a mutual-backup relationship, and S1 loses its
normal connectivity and must rely on the S1S2 link, then S2 must know all about S1s routes.
All ASs do export their own routes, and their customers routes, to customers, siblings, peers and
providers.
There may be more complicated variations: if a regional provider is a customer of a large transit backbone,
then the backbone might only announce routes listed in transit agreement (rather than all routes, as above).
There is a supposition here that the regional provider has multiple connections, and has contracted with that
particular transit backbone only for certain routes.

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Following these rules creates a simplified BGP world. Special cases for special situations have the potential
to introduce non-convergence or instability.
The so-called tier-1 providers are those that are not customers of anyone; these represent the top-level
backbone providers. Each tier-1 AS must, as a rule, peer with every other tier-1 AS.
A consequence of the use of the above classification and attendant export rules is the no-valley theorem
[LG01]: if every AS has BGP policies consistent with the scheme above, then when we consider the full ASpath from A to B, there is at most one peer-peer link. Those to the left of the peer-peer link are (moving from
left to right) either customerprovider links or siblingsibling links; that is, they are non-downwards
(ie upwards or level). To the right of the peer-peer link, we see providercustomer or siblingsibling
links; that is, these are non-upwards. If there is no peer-peer link, then we can still divide the AS-path into
a non-downwards first part and a non-upwards second part.
The above constraints are not quite sufficient to guarantee convergence of the BGP system to a stable set
of routes. To ensure convergence in the case without sibling relationships, it is shown in [GR01] that the
following simple local_preference rule suffices:
If AS1 gets two routes r1 and r2 to a destination D, and the first AS of the r1 route is a customer
of AS1, and the first AS of r2 is not, then r1 will be assigned a higher local_preference value
than r2.
More complex rules exist that allow for cases when the local_preference values can be equal; one such rule
states that strict inequality is only required when r2 is a provider route. Other straightforward rules handle
the case of sibling relationships, eg by requiring that siblings have local_preference rules consistent with the
use of their shared connection only for backup.
As a practical matter, unstable BGP arrangements appear rare on the Internet; most actual relationships and
configurations are consistent with the rules above.

10.6.8 Examples of BGP Instability

What if the normal rules regarding BGP preferences are not followed? It turns out that BGP allows
genuinely unstable situations to occur; this is a consequence of allowing each AS a completely independent
hand in selecting preference functions. Here are two simple examples, from [GR01].
Example 1: A stable state exists, but convergence to it is not guaranteed. Consider the following network
arrangement:

We assume AS1 prefers AS-paths to destination D in the following order:

AS2,AS0, AS0

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That is, AS2,AS0 is preferred to the direct path AS0 (one way to express this preference might be
prefer routes for which the AS-PATH begins with AS2; perhaps the AS1AS0 link is more expensive).
Similarly, we assume AS2 prefers paths to D in the order AS1,AS0, AS0. Both AS1 and AS2 start out
using path AS0; they advertise this to each other. As each receives the others advertisement, they apply
their preference order and therefore each switches to routing Ds traffic to the other; that is, AS1 switches
to the route with AS-path AS2,AS0 and AS2 switches to AS1,AS0. This, of course, causes a routing
loop! However, as soon as they export these paths to one another, they will detect the loop in the AS-path
and reject the new route, and so both will switch back to AS0 as soon as they announce to each other the
change in what they use.
This oscillation may continue indefinitely, as long as both AS1 and AS2 switch away from AS0 at the
same moment. If, however, AS1 switches to AS2,AS0 while AS2 continues to use AS0, then AS2 is
stuck and the situation is stable. In practice, therefore, eventual convergence to a stable state is likely.
AS1 and AS2 might choose not to export their D-route to each other to avoid this instability.
Example 2: No stable state exists. This example is from [VGE00]. Assume that the destination D is attached
to AS0, and that AS0 in turn connects to AS1, AS2 and AS3 as in the following diagram:

AS1-AS3 each have a direct route to AS0, but we assume each prefers the AS-path that takes their clockwise
neighbor; that is, AS1 prefers AS3,AS0 to AS0; AS3 prefers AS2,AS0 to AS0, and AS2 prefers
AS1,AS0 to AS0. This is a peculiar, but legal, example of input filtering.
Suppose all adopt AS0, and advertise this, and AS1 is the first to look at the incoming advertisements.
AS1 switches to the route AS3,AS0, and announces this.
At this point, AS2 sees that AS1 uses AS3,AS0; if AS2 switches to AS1 then its path would be
AS1,AS3,AS0 rather than AS1,AS0 and so it does not make the switch.
But AS3 does switch: it prefers AS2,AS0 and this is still available. Once it makes this switch, and
advertises it, AS1 sees that the route it had been using, AS3,AS0, has become AS3,AS1,AS0. At this
point AS1 switches back to AS0.
Now AS2 can switch to using AS1,AS0, and does so. After that, AS3 finds it is now using AS2,AS1,AS0
and it switches back to AS0. This allows AS1 to switch to the longer route, and then AS2 switches back
to the direct route, and then AS3 gets the longer route, then AS2 again, etc, forever rotating clockwise.

10.7 Epilog
CIDR was a deceptively simple idea. At first glance it is a straightforward extension of the subnet concept,
moving the net/host division point to the left as well as to the right. But it has ushered in true hierarchical
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routing, most often provider-based. While CIDR was originally offered as a solution to some early crises in
IPv4 address-space allocation, it has been adopted into the core of IPv6 routing as well.
Interior routing using either distance-vector or link-state protocols is neat and mathematical. Exterior
routing with BGP is messy and arbitrary. Perhaps the most surprising thing about BGP is that the Internet
works as well as it does, given the complexity of provider interconnections. The business side of routing
almost never has an impact on ordinary users. To an extent, BGP works well because providers voluntarily limit the complexity of their filtering preferences, but that seems to be largely because the business
relationships of real-world ISPs do not seem to require complex filtering.

10.8 Exercises
1. Consider the following IP forwarding table that uses CIDR. IP address bytes are in hexadecimal, so each
hex digit corresponds to four address bits.
destination
81.30.0.0/12
81.3c.0.0/16
81.3c.50.0/20
81.40.0.0/12
81.44.0.0/14

next_hop
A
B
C
D
E

For each of the following IP addresses, indicate to what destination it is forwarded.

(i) 81.3b.15.49
(ii) 81.3c.56.14
(iii) 81.3c.85.2e
(iv) 81.4a.35.29
(v) 81.47.21.97
(vi) 81.43.01.c0
2. Consider the following IP forwarding table, using CIDR. As in exercise 1, IP address bytes are in hexadecimal.
destination
00.0.0.0/2
40.0.0.0/2
80.0.0.0/2
C0.0.0.0/2

next_hop
A
B
C
D

(a). To what next_hop would each of the following be routed? 63.b1.82.15, 9e.00.15.01, de.ad.be.ef
(b). Explain why every IP address is routed somewhere, even though there is no default entry.

3. Give an IPv4 forwarding table using CIDR that will route all Class A addresses to next_hop A, all
Class B addresses to next_hop B, and all Class C addresses to next_hop C.

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4. Suppose a router using CIDR has the following entries. Address bytes are in decimal except for the third
byte, which is in binary.
destination
37.119.0000 0000.0/18
37.119.0100 0000.0/18
37.119.1000 0000.0/18
37.119.1100 0000.0/18

next_hop
A
A
A
B

These four entries cannot be consolidated into a single /16 entry, because they dont all go to the same
next_hop. How could they be consolidated into two entries?

5. Suppose P, Q and R are ISPs with respective CIDR address blocks (with bytes in decimal) 51.0.0.0/8, 52.0.0.0/8 and
A: 51.10.0.0/16
B: 51.23.0.0/16
Q has customers C and D and assigns them address blocks as follows:
C: 52.14.0.0/16
D: 52.15.0.0/16

(a). Give forwarding tables for P, Q and R assuming they connect to each other and to each of their own
customers.
(b). Now suppose A switches from provider P to provider Q, and takes its address block with it. Give the
forwarding tables for P, Q and R; the longest-match rule will be needed to resolve conflicts.
(c). Now suppose in addition to A switching from P to Q, C switches from provider Q to provider R. Give
the forwarding tables.

6. Suppose P, Q and R are ISPs as in the previous problem. P and R do not connect directly; they route traffic
to one another via Q. In addition, customer B is multi-homed and has a secondary connection to provider R;
customer D is also multi-homed and has a secondary connection to provider P. R and P use these secondary
connections to send to B and D respectively; however, these secondary connections are not advertised to
other providers. Give forwarding tables for P, Q and R.
7. Consider the following network of providers P-S, all using BGP. The providers are the horizontal lines;
each provider is its own AS.

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(a). What routes to network NS will P receive, assuming there is no export filtering? For each route, list the
AS-path.
(b). What routes to network NQ will P receive? For each route, list the AS-path.
(c). Suppose R uses export filtering so as not to advertise to P any of its routes except those that involve S
in their AS-path. What routes to network NR will P receive, with AS-paths?

8. Consider the following network of Autonomous Systems AS1 through AS6, which double as destinations.
When AS1 advertises itself to AS2, for example, the AS-path it provides is AS1.
AS1

AS2

AS4

AS5

AS3
:
:
:
AS6

(a). If neither AS3 nor AS6 exports their AS3AS6 link to their neighbors AS2 and AS5 to the left, what
routes will AS2 receive to reach AS5? Specify routes by AS-path.
(b). What routes will AS2 receive to reach AS6?
(c). Suppose AS3 exports to AS2 its link to AS6, but AS6 continues not to export the AS3AS6 link to
AS5. How will AS5 now reach AS2? How will AS2 now reach AS6? Assume that there are no local
preferences in use in BGP best-path selection, and that the shortest AS-path wins.

9. Suppose that Internet routing in the US used geographical routing, and the first 12 bits of every IP
address represent a geographical area similar in size to a telephone area code. Megacorp gets the prefix
12.34.0.0/16, based geographically in Chicago, and allocates subnets from this prefix to its offices in all 50
states. Megacorp routes all its internal traffic over its own network.

(a). Assuming all Megacorp traffic must enter and exit in Chicago, what is the route of traffic to and from
the San Diego office to a client also in San Diego?

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(b). Now suppose each office has its own link to a local ISP, but still uses its 12.34.0.0/16 IP addresses.
Now what is the route of traffic between the San Diego office and its neighbor?
(c). Suppose Megacorp gives up and gets a separate geographical prefix for each office. What must it do to
ensure that its internal traffic is still routed over its own network?

10. Suppose we try to use BGPs strategy of exchanging destinations plus paths as an interior routing-update
strategy, perhaps replacing distance-vector routing. No costs or hop-counts are used, but routers attach to
each destination a list of the routers used to reach that destination. Routers can also have route preferences,
such as prefer my link to B whenever possible.
(a). Consider the network of 9.2 Distance-Vector Slow-Convergence Problem:
D

The DA link breaks, and B offers A what it thinks is its own route to D. Explain how exchanging path
information prevents a routing loop here.
(b). Suppose the network is as below, and initially each router knows about itself and its immediately
adjacent neighbors. What sequence of router announcements can lead to A reaching F via
ADEBCF, and what individual router preferences would be necessary? (Initially, for example,
A would reach B directly; what preference might make it prefer ADEB?)

(c). Explain why this method is equivalent to using the hopcount metric with either distance-vector or
link-state routing, if routers are not allowed to have preferences and if the router-path length is used as a
tie-breaker.

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11 UDP TRANSPORT

The standard transport protocols riding above the IP layer are TCP and UDP. As we saw in Chapter 1, UDP
provides simple datagram delivery to remote sockets, that is, to host,port pairs. TCP provides a much
richer functionality for sending data, but requires that the remote socket first be connected. In this chapter,
we start with the much-simpler UDP, including the UDP-based Trivial File Transfer Protocol.
We also review some fundamental issues any transport protocol must address, such as lost final packets and
packets arriving late enough to be subject to misinterpretation upon arrival. These fundamental issues will
be equally applicable to TCP connections.

11.1 User Datagram Protocol UDP

RFC 1122 refers to UDP as almost a null protocol; while that is something of a harsh assessment, UDP
is indeed fairly basic. The two features it adds beyond the IP layer are port numbers and a checksum on
the data. The UDP header consists of the following:

The port numbers are what makes UDP into a real transport protocol: with them, an application can now
connect to an individual server process (that is, the process owning the port number in question), rather
than simply to a host.
UDP is unreliable, in that there is no UDP-layer attempt at timeouts, acknowledgment and retransmission;
applications written for UDP must implement these. As with TCP, a UDP host,port pair is known as a
socket (though UDP ports are considered a separate namespace from TCP ports). UDP is also unconnected,
or stateless; if an application has opened a port on a host, any other host on the Internet may deliver packets
to that host,port socket without preliminary negotiation.
UDP packets use the 16-bit Internet checksum (5.4 Error Detection) on the data. While it is seldom done
now, the checksum can be disabled and the field set to the all-0-bits value, which never occurs as an actual
ones-complement sum.
UDP packets can be dropped due to queue overflows either at an intervening router or at the receiving host.
When the latter happens, it means that packets are arriving faster than the receiver can process them. Higherlevel protocols that define ACK packets (eg UDP-based RPC, below) typically include some form of flow
control to prevent this.
UDP is popular for local transport, confined to one LAN. In this setting it is common to use UDP as the
transport basis for a Remote Procedure Call, or RPC, protocol. The conceptual idea behind RPC is that
one host invokes a procedure on another host; the parameters and the return value are transported back and
forth by UDP. We will consider RPC in greater detail below, in 11.4 Remote Procedure Call (RPC); for

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now, the point of UDP is that on a local LAN we can fall back on rather simple mechanisms for timeout and
retransmission.
UDP is well-suited for request-reply semantics beyond RPC; one can use TCP to send a message and get
a reply, but there is the additional overhead of setting up and tearing down a connection. DNS uses UDP,
largely for this reason. However, if there is any chance that a sequence of request-reply operations will be
performed in short order then TCP may be worth the overhead.
UDP is also popular for real-time transport; the issue here is head-of-line blocking. If a TCP packet is lost,
then the receiving host queues any later data until the lost data is retransmitted successfully, which can take
several RTTs; there is no option for the receiving application to request different behavior. UDP, on the
other hand, gives the receiving application the freedom simply to ignore lost packets. This approach is very
successful for voice and video, where small losses simply degrade the received signal slightly, but where
larger delays are intolerable. This is the reason the Real-time Transport Protocol, or RTP, is built on top
of UDP rather than TCP. It is common for VoIP telephone calls to use RTP and UDP.

11.1.1 QUIC
Sometimes UDP is used simply because it allows new or experimental protocols to run entirely as user-space
applications; no kernel updates are required, as would be the case with TCP changes. Google has created
a protocol named QUIC (Quick UDP Internet Connections, chromium.org/quic) in this category, though
QUIC also takes advantage of UDPs freedom from head-of-line blocking. For example, one of QUICs
goals includes supporting multiplexed streams in a single connection (eg for the multiple components of a
web page). A lost packet blocks its own stream until it is retransmitted, but the other streams can continue
without waiting. Because QUIC supports error-correcting codes (5.4.2 Error-Correcting Codes), a lost
packet might not require any waiting at all; this is another feature that would be difficult to add to TCP.
QUIC also eliminates the extra RTT needed for setting up a TCP connection.
QUIC provides support for advanced congestion control, currently (2014) including a UDP analog of TCP
CUBIC (15.11 TCP CUBIC). QUIC does this at the application layer but new congestion-control mechanisms within TCP often require client operating-system changes even when the mechanism lives primarily
at the server end. QUIC represents a promising approach to using UDPs flexibility to support innovative
or experimental transport-layer features. The downside of QUIC is its nonstandard programming interface,
but note that Google can achieve widespread web utilization of QUIC simply by distributing the client side
in its Chrome browser.

11.1.2 UDP Simplex-Talk

One of the early standard examples for socket programming is simplex-talk. The client side reads lines of
text from the users terminal and sends them over the network to the server; the server then displays them on
its terminal. Simplex here refers to the one-way nature of the flow; duplex talk is the basis for Instant
Messaging, or IM. Even at this simple level we have some details to attend to regarding the data protocol:
we assume here that the lines are sent with a trailing end-of-line marker. In a world where different OSs
use different end-of-line marks, including them in the transmitted data can be problematic. However, when
we get to the TCP version, if arriving packets are queued for any reason then the embedded end-of-line
character will be the only thing to separate the arriving data into lines.
As with almost every Internet protocol, the server side must select a port number, which with the servers IP

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address will form the socket address to which clients connect. Clients must discover that port number or
have it written into their application code. Clients too will have a port number, but it is largely invisible.
On the server side, simplex-talk must do the following:
ask for a designated port number
create a socket, the sending/receiving endpoint
bind the socket to the socket address, if this is not done at the point of socket creation
receive packets sent to the socket
for each packet received, print its sender and its content
The client side has a similar list:
look up the servers IP address, using DNS
create an anonymous socket; we dont care what the clients port number is
read a line from the terminal, and send it to the socket address server_IP,port
11.1.2.1 The Server
We will start with the server side, presented here in Java. We will use port 5432; the socket-creation and
port-binding operations are combined into the single operation new DatagramSocket(destport).
Once created, this socket will receive packets from any host that addresses a packet to it; there is no need
for preliminary connection. We also need a DatagramPacket object that contains the packet data and
source IP_address,port for arriving packets. The server application does not acknowledge anything sent to
it, or in fact send any response at all.
The server application needs no parameters; it just starts. (That said, we could make the port number a
parameter, to allow easy change. The port we use here, 5432, has also been adopted by PostgreSQL for TCP
connections.) The server accepts both IPv4 and IPv6 connections; we return to this below.
Though it plays no role in the protocol, we will also have the server time out every 15 seconds and display
a message, just to show how this is done; implementations of real protocols essentially always must arrange
when attempting to receive a packet to time out after a certain interval with no response. The file below is
at udp_stalks.java.
/* simplex-talk server, UDP version */
import java.net.*;
import java.io.*;
public class stalks {
static public int destport = 5432;
static public int bufsize = 512;
static public final int timeout = 15000; // time in milliseconds
static public void main(String args[]) {
DatagramSocket s;
// UDP uses DatagramSockets

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try {
s = new DatagramSocket(destport);
}
catch (SocketException se) {
System.err.println("cannot create socket with port " + destport);
return;
}
try {
s.setSoTimeout(timeout);
// set timeout in milliseconds
} catch (SocketException se) {
System.err.println("socket exception: timeout not set!");
}
// create DatagramPacket object for receiving data:
DatagramPacket msg = new DatagramPacket(new byte[bufsize], bufsize);
while(true) { // read loop
try {
msg.setLength(bufsize); // max received packet size
s.receive(msg);
// the actual receive operation
System.err.println("message from <" +
msg.getAddress().getHostAddress() + "," + msg.getPort() + ">");
} catch (SocketTimeoutException ste) {
// receive() timed out
System.err.println("Response timed out!");
continue;
} catch (IOException ioe) {
// should never happen!
System.err.println("Bad receive");
break;
}
String str = new String(msg.getData(), 0, msg.getLength());
System.out.print(str);
// newline must be part of str
}
s.close();
} // end of main
}

11.1.2.2 UDP and IP addresses

The line s = new DatagramSocket(destport) creates a DatagramSocket object bound to the
given port. If a host has multiple IP addresses, packets sent to that port to any of those IP addresses will
be delivered to the socket, including localhost (and in fact all IPv4 addresses between 127.0.0.1 and
127.255.255.255) and the subnet broadcast address (eg 192.168.1.255). If a client attempts to connect to the
subnet broadcast address, multiple servers may receive the packet (in this we are perhaps fortunate that the
stalk server does not reply).
Alternatively, we could have used
s = new DatagramSocket(int port, InetAddress local_addr)

in which case only packets sent to the host and port through the hosts specific IP address local_addr
would be delivered. It does not matter here whether IP forwarding on the host has been enabled. In the
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original C socket library, this binding of a port to (usually) a server socket was done with the bind() call.
To allow connections via any of the hosts IP addresses, the special IP address INADDR_ANY is passed to
bind().
When a host has multiple IP addresses, the standard socket library does not provide a way to find out to
which these an arriving UDP packet was actually sent. Normally, however, this is not a major difficulty. If
a host has only one interface on an actual network (ie not counting loopback), and only one IP address for
that interface, then any remote clients must send to that interface and address. Replies (if any, which there
are not with stalk) will also come from that address.
Multiple interfaces do not necessarily create an ambiguity either; the easiest such case to experiment with
involves use of the loopback and Ethernet interfaces (though one would need to use an application that,
unlike stalk, sends replies). If these interfaces have respective IPv4 addresses 127.0.0.1 and 192.168.1.1,
and the client is run on the same machine, then connections to the server application sent to 127.0.0.1 will
be answered from 127.0.0.1, and connections sent to 192.168.1.1 will be answered from 192.168.1.1. The
IP layer sees these as different subnets, and fills in the IP source-address field according to the appropriate
subnet. The same applies if multiple Ethernet interfaces are involved, or if a single Ethernet interface is
assigned IP addresses for two different subnets, eg 192.168.1.1 and 192.168.2.1.
Life is slightly more complicated if a single interface is assigned multiple IP addresses on the same subnet,
eg 192.168.1.1 and 192.168.1.2. Regardless of which address a client sends its request to, the servers reply
will generally always come from one designated address for that subnet, eg 192.168.1.1. Thus, it is possible
that a legitimate UDP reply will come from a different IP address than that to which the initial request was
sent.
If this behavior is not desired, one approach is to create multiple server sockets, and to bind each of the
hosts network IP addresses to a different server socket.
11.1.2.3 The Client
Next is the Java client version udp_stalkc.java. The client any client must provide the name of the
host to which it wishes to send; as with the port number this can be hard-coded into the application but is
more commonly specified by the user. The version here uses host localhost as a default but accepts any
other hostname as a command-line argument. The call to InetAddress.getByName(desthost)
invokes the DNS system, which looks up name desthost and, if successful, returns an IP address.
(InetAddress.getByName() also accepts addresses in numeric form, eg 127.0.0.1, in which case
DNS is not necessary.) When we create the socket we do not designate a port in the call to new
DatagramSocket(); this means any port will do for the client. When we create the DatagramPacket
object, the first parameter is a zero-length array as the actual data array will be provided within the loop.
A certain degree of messiness is introduced by the need to create a BufferedReader object to handle
terminal input.
// simplex-talk CLIENT in java, UDP version
import java.net.*;
import java.io.*;
public class stalkc {
static public BufferedReader bin;

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static public int destport = 5432;

static public int bufsize = 512;
static public void main(String args[]) {
String desthost = "localhost";
if (args.length >= 1) desthost = args[0];
bin = new BufferedReader(new InputStreamReader(System.in));
InetAddress dest;
System.err.print("Looking up address of " + desthost + "...");
try {
dest = InetAddress.getByName(desthost);
// DNS query
}
catch (UnknownHostException uhe) {
System.err.println("unknown host: " + desthost);
return;
}
System.err.println(" got it!");
DatagramSocket s;
try {
s = new DatagramSocket();
}
catch(IOException ioe) {
System.err.println("socket could not be created");
return;
}
System.err.println("Our own port is " + s.getLocalPort());
DatagramPacket msg = new DatagramPacket(new byte[0], 0, dest, destport);
while (true) {
String buf;
int slen;
try {
buf = bin.readLine();
}
catch (IOException ioe) {
System.err.println("readLine() failed");
return;
}
if (buf == null) break;

// user typed EOF character

buf = buf + "\n";

// append newline character
slen = buf.length();
byte[] bbuf = buf.getBytes();
msg.setData(bbuf);
msg.setLength(slen);

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try {
s.send(msg);
}
catch (IOException ioe) {
System.err.println("send() failed");
return;
}
} // while
s.close();
}
}

The default value of desthost here is localhost; this is convenient when running the client and the
server on the same machine, in separate terminal windows.
Like the server, the client works with both IPv4 and IPv6. The InetAddress object dest in the server
code above can hold either IPv4 or IPv6 addresses; InetAddress is the base class with child classes
Inet4Address and Inet6Address. If the client and server can communicate at all via IPv6 and if the
value of desthost supplied to the client is an IPv6-only name, then dest will be an Inet6Address
object and IPv6 will be used. For example, if the client is invoked from the command line with java
stalkc ip6-localhost, and the name ip6-localhost resolves to the IPv6 loopback address ::1,
the client will connect to an stalk server on the same host using IPv6 (and the loopback interface). If greater
IPv4-versus-IPv6 control is desired, one can to replace the getByName() call with getAllByName(),
which returns an array of all addresses (InetAddress[]) associated with the given name. One can
then find the IPv6 addresses by searching this array for addresses addr for which addr instanceof
Inet6Address.
Finally, here is a simple python version of the client, udp_stalkc.py.
#!/usr/bin/python3
from socket import *
from sys import argv
portnum = 5432
def talk():
rhost = "localhost"
if len(argv) > 1:
rhost = argv[1]
print("Looking up address of " + rhost + "...", end="")
try:
dest = gethostbyname(rhost)
except (GAIerror, herror) as mesg:
# GAIerror: error in gethostbyname()
errno,errstr=mesg.args
print("\n
", errstr);
return;
print("got it: " + dest)
addr=(dest, portnum)
# a socket address
s = socket(AF_INET, SOCK_DGRAM)
s.settimeout(1.5)
# we dont actually need to set timeout here
while True:
buf = input("> ")

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if len(buf) == 0: return
# an empty line exits
s.sendto(bytes(buf + "\n", ascii), addr)
talk()

Why not C?
While C is arguably the most popular language for network programming, it does not support IP
addresses and other network objects as first-class types, and so we omit it here. But see 21.2.2 An
Actual Stack-Overflow Example for a TCP-based C version of an stalk-like program.
To experiment with these on a single host, start the server in one window and one or more clients in other
windows. One can then try the following:
have two clients simultaneously running, and sending alternating messages to the same server
invoke the client with the external IP address of the server in dotted-decimal, eg 10.0.0.3 (note that
localhost is 127.0.0.1)
run the java and python clients simultaneously, sending to the same server
run the server on a different host (eg a virtual host or a neighboring machine)
invoke the client with a nonexistent hostname
Note that, depending on the DNS server, the last one may not actually fail. When asked for the DNS name
of a nonexistent host such as zxqzx.org, many ISPs will return the IP address of a host running a web server
hosting an error/search/advertising page (usually their own). This makes some modicum of sense when
attention is restricted to web searches, but is annoying if it is not, as it means non-web applications have no
easy way to identify nonexistent hosts.
Simplex-talk will work if the server is on the public side of a NAT firewall. No server-side packets need to
be delivered to the client! But if the other direction works, something is very wrong with the firewall.

11.1.3 Binary Data

In the stalk example above, the client sent strings to the server. However, what if we are implementing a
protocol that requires us to send binary data? Or designing such a protocol? The client and server will now
have to agree on how the data is to be encoded.
As an example, suppose the client is to send to the server a list of 32-bit integers, organized as follows.
The length of the list is to occupy the first two bytes; the remainder of the packet contains the consecutive
integers themselves, four bytes each, as in the diagram:

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The client needs to create the byte array organized as above, and the server needs to extract the values. (The
inclusion of the list length as a short int is not really necessary, as the receiver will be able to infer the
list length from the packet size, but we want to be able to illustrate the encoding of both int and short
int values.)
The protocol also needs to define how the integers themselves are laid out. There are two common ways to
represent a 32-bit integer as a sequence of four bytes. Consider the integer 0x01020304 = 12563 + 22562
+ 3256 + 4. This can be encoded as the byte sequence [1,2,3,4], known as big-endian encoding, or as
[4,3,2,1], known as little-endian encoding; the former was used by early IBM mainframes and the latter is
used by most Intel processors. (We are assuming here that both architectures represent signed integers using
twos-complement; this is now universal but was not always.)
To send 32-bit integers over the network, it is certainly possible to tag the data as big-endian or little-endian,
or for the endpoints to negotiate the encoding. However, by far the most common approach on the Internet
at least below the application layer is to follow the convention of RFC 1700 and use big-endian encoding
exclusively; big-endian encoding has since come to be known as network byte order.
How one converts from host byte order to network byte order is language-dependent. It must always be
done, even on big-endian architectures, as code may be recompiled on a different architecture later.
In Java the byte-order conversion is generally combined with the process of conversion from int to
byte[]. The client will use a DataOutputStream class to support the writing of the binary values to an output stream, through methods such as writeInt() and writeShort(), together with a
ByteArrayOutputStream class to support the conversion of the output stream to type byte[]. The
code below assumes the list of integers is initially in an ArrayList<Integer> named theNums.
ByteArrayOutputStream baos = new ByteArrayOutputStream();
DataOutputStream dos = new DataOutputStream(baos);
try {
dos.writeShort(theNums.size());
for (int n : theNums) {
dos.writeInt(n);
}
} catch (IOException ioe) { /* exception handling */ }
byte[] bbuf = baos.toByteArray();
msg.setData(bbuf);

// msg is the DatagramPacket object to be sent

The server then needs to to the reverse; again, msg is the arriving DatagramPacket. The code below simply
calculates the sum of the 32-bit integers in msg:
ByteArrayInputStream bais = new ByteArrayInputStream(msg.getData(), 0, msg.getLength());
DataInputStream dis = new DataInputStream(bais);
int sum = 0;
try {
int count = dis.readShort();
for (int i=0; i<count; i++) {
sum += dis.readInt();
}
} catch (IOException ioe) { /* more exception handling */ }

A version of simplex-talk for lists of integers can be found in client saddc.java and server sadds.java. The
client reads from the command line a list of character-encoded integers (separated by whitespace), constructs
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the binary encoding as above, and sends them to the server; the server prints their sum. Port 5434 is used;
this can be changed if necessary.
In the C language, we can simply allocate a char[] of the appropriate size and write the network-byteorder values directly into it. Conversion to network byte order and back is done with the following library
calls:
htonl: host-to-network conversion for long (32-bit) integers
ntohl: network-to-host conversion for long integers
htons: host-to-network conversion for short (16-bit) integers
ntohs: network-to-host conversion for short integers
A certain amount of casting between int * and char * is also necessary.
In general, the designer of a protocol needs to select an unambiguous format for all binary data; protocoldefining RFCs always include such format details. This can be a particular issue for floating-point data,
for which two formats can have the same endianness but still differ, eg in normalization or the size of the
exponent field. Formats for structured data, such as arrays, must also be spelled out; in the example above
the list size was indicated by a length field but other options are possible.
The example above illustrates fixed-field-width encoding. Another possible option, using variable-length
encoding, is ASN.1 using the Basic Encoding Rules (20.6 ASN.1 Syntax and SNMP); fixed-field encoding
sometimes becomes cumbersome as data becomes more hierarchical.
At the application layer, the use of non-binary encodings is common, though binary encodings continue to
remain common as well. Two popular formats using human-readable unicode strings for data encoding are
ASN.1 with its XML Encoding Rules and JSON. While the latter format originated with JavaScript, it is
now widely supported by many other languages.

11.2 Fundamental Transport Issues

As we turn to actual transport protocols, including eventually TCP, we will encounter the following standard
problematic cases that must be addressed if the integrity of the data is to be ensured.
Old duplicate packets: These packets can be either internal from an earlier point in the same connection
instance or external from a previous instance of the connection. For the internal case, the receiver must
make sure that it does not accept an earlier duplicated and delayed packet as current data (if the earlier packet
was not duplicated, in most cases the transfer would not have advanced). Usually internal old duplicates are
prevented by numbering the data, either by block or by byte. However, if the numbering field is allowed to
wrap around, an old and a new packet may have the same number.
For the external case, the connection is closed and then reopened a short time later, using the same port
numbers. (A connection is typically defined by its endpoint socket addresses; thus, we refer to reopening
the connection even if the second instance is completely unrelated. Two separate instances of a connection between the same socket addresses are sometimes known as separate incarnations of the connection.)
Somehow a delayed copy of a packet from the first instance (or incarnation) of the connection arrives while
the second instance is in progress. This old duplicate must not be accepted, incorrectly, as valid data, as that
would corrupt the second transfer. (An illustration of this, in the context of TCP, appears in 12.7 TCP Old
Duplicates.)
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Both these scenarios assume that the old duplicate was sent earlier, but was somehow delayed in transit
for an extended period of time, while later packets were delivered normally. Exactly how this might occur
remains unclear; perhaps the least far-fetched scenario is the following:
A first copy of the old duplicate was sent
A routing error occurs; the packet is stuck in a routing loop
An alternative path between the original hosts is restored, and the packet is retransmitted successfully
Some time later, the packet stuck in the routing loop is released, and reaches its final destination
Another scenario involves a link in the path that supplies link-layer acknowledgment: the packet was sent
once across the link, the link-layer ACK was lost, and so the packet was sent again. Some mechanism is
still needed to delay one of the copies.
Most solutions to the old-duplicate problem assume some cap on just how late an old duplicate can be. In
practical terms, TCP officially once took this time limit to be 60 seconds, but implementations now usually
take it to be 30 seconds. Other protocols often implicitly adopt the TCP limit. Once upon a time, IP routers
were expected to decrement a packets TTL field by 1 for each second the router held the packet in its queue;
in such a world, IP packets cannot be more than 255 seconds old.
It is also possible to prevent external old duplicates by including a connection count parameter in the
transport or application header. For each consecutive connection, the connection count is incremented by (at
least) 1. A separate connection-count value must be maintained by each side; if a connection-count value
is ever lost, a suitable backup mechanism based on delay might be used. As an example, see 12.11 TCP
Faster Opening.
Lost final ACK: Most packets will be acknowledged. The final packet (typically but not necessarily an
ACK) cannot itself be acknowledged, as then it would not be the final packet. Somebody has to go last. This
leaves some uncertainty on the part of the sender: did the last packet make it through, or not?
Duplicated connection request: How do we distinguish between two different connection requests and a
single request that was retransmitted? Does it matter?
Reboots: What if one side reboots while the other side is still sending data? How will the other side detect
this? Are there any scenarios that could lead to corrupted data?

11.3 Trivial File Transport Protocol, TFTP

As an actual protocol based on UDP, we consider the Trivial File Transport Protocol, TFTP. While TFTP
supports clients sending files to the server, we will restrict attention to the more common case where the
client requests a file from the server.
Although TFTP is a very simple protocol, it addresses all the fundamental transport issues listed above, to
at least some degree.
TFTP, documented first in RFC 783 and updated in RFC 1350, has five packet types:
Read ReQuest, RRQ, containing the filename and a text/binary indication
Write ReQuest, WRQ
Data, containing a 16-bit block number and up to 512 bytes of data
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ACK, containing a 16-bit block number

Error, for certain designated errors. All errors other than Unknown Transfer ID are cause for termination.
Data block numbering begins at 1; we will denote the packet with the Nth block of data as Data[N]. Acknowledgments contain the block number of the block being acknowledged; thus, ACK[N] acknowledges
Data[N]. All blocks of data contain 512 bytes except the final block, which is identified as the final block
by virtue of containing less than 512 bytes of data. If the file size was divisible by 512, the final block will
contain 0 bytes of data.
Because TFTP uses UDP it must take care of packetization itself, and thus must fix a block size small enough
to be transmitted successfully everywhere.
In the absence of packet loss or other errors, TFTP file requests proceed as follows.
1. The client sends a RRQ to server port 69, from client port c_port
2. The server obtains a new port, s_port, from the OS
3. The server sends Data[1] from s_port
4. The client receives Data[1], and thus learns the value of s_port
5. The client sends ACK[1] (and all future ACKs) to the servers s_port
6. The server sends Data[2], etc, each time waiting for the client ACK[N] before sending Data[N+1]
7. The transfer process stops when the server sends its final block, of size less than 512 bytes, and the
client sends the corresponding ACK
8. An optional but recommended step of server-side dallying is used to address the lost-final-ACK issue

11.3.1 Port Changes

In the above, the server changes to a new port s_port when answering. While this change plays a modest
role in the reliability of the protocol, below, it also makes the implementers life much easier. When the
server creates the new port, it is assured that the only packets that will arrive at that port are those related
to the original client request; other client requests will be associated with other server ports. The server can
create a new process for this new port, and that process will be concerned with only a single transfer even if
multiple parallel transfers are taking place.
Random Ports?
RFC 1350 states The TIDs [port numbers] chosen for a connection should be randomly chosen, so
that the probability that the same number is chosen twice in immediate succession is very low. A
literal interpretation is that an implementation should choose a random 16-bit number and ask for that
as its TID. But the author knows of no implementation that actually does this; all seem to create sockets
(eg with Javas DatagramSocket()) and accept the port number assigned to the new socket by the
operating system. That port number will not be random in the statistical sense, but will be very likely
different from any recently used port. Does this represent noncompliance with the RFC? The author
has no idea.

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If the server answered all requests from port 69, it would have to distinguish among multiple concurrent
transfers by looking at the client socket address; each client transfer would have its own state information
including block number, open file, and the time of the last successful packet. This considerably complicates
the implementation.
This port-change rule does break TFTP when the server is on the public side of a NAT firewall. When
the client sends an RRQ to port 69, the NAT firewall will now allow the server to respond from port 69.
However, the servers response from s_port is generally blocked, and so the client never receives Data[1].

11.3.2 TFTP Stop-and-Wait

TFTP uses a very straightforward implementation of stop-and-wait (6.1 Building Reliable Transport: Stopand-Wait). Acknowledgment packets contain the block number of the data packet being acknowledged; that
is, ACK[N] acknowledges Data[N].
In the original RFC 783 specification, TFTP was vulnerable to the Sorcerers Apprentice bug (6.1.2 Sorcerers Apprentice Bug). Correcting this problem was the justification for updating the protocol in RFC
1350, eleven years later. The omnibus hosts-requirements document RFC 1123 (referenced by RFC 1350)
describes the necessary change this way:
Implementations MUST contain the fix for this problem: the sender (ie, the side originating the
DATA packets) must never resend the current DATA packet on receipt of a duplicate ACK.

11.3.3 Lost Final ACK

The receiver, after receiving the final DATA packet and sending the final ACK, might exit. But if it does so,
and the final ACK is lost, the sender will continue to timeout and retransmit the final DATA packet until it
gives up; it will never receive confirmation that the transfer succeeded.
TFTP addresses this by recommending that the receiver enter into a DALLY state when it has sent the
final ACK. In this state, it responds only to received duplicates of the final DATA packet; its response is to
retransmit the final ACK. While one lost final ACK is possible, multiple such losses are unlikely; sooner or
later the sender will receive the final ACK and may then exit.
The dally interval should be at least twice the senders timeout interval. Note that the receiver has no direct
way to determine this value.
The TCP analogue of dallying is the TIMEWAIT state, though TIMEWAIT also has another role.

11.3.4 Duplicated Connection Request

Suppose the first RRQ is delayed. The client times out and retransmits it.
One approach would be for the server to recognize that the second RRQ is a duplicate, perhaps by noting
that it is from the same client socket address and contains the same filename. In practice, however, this
would significantly complicate the design of a TFTP implementation, because having the server create a
new process for each RRQ that arrived would no longer be possible.
So TFTP allows the server to start two sender processes, from two ports s_port1 and s_port2. Both will
send Data[1] to the receiver. The receiver is expected to latch on to the port of the first Data[1] packet
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received, recording its source port. The second Data[1] now appears to be from an incorrect port; the TFTP
specification requires that a receiver reply to any packets from an unknown port by sending an ERROR
packet with the code Unknown Transfer ID (where Transfer ID means port number). Were it not for
this duplicate-RRQ scenario, packets from an unknown port could probably be simply ignored.
What this means in practice is that the first of the two sender processes above will successfully connect to
the receiver, and the second will receive the Unknown Transfer ID message and will exit.
A more unfortunate case related to this is below, example 4 under TFTP Scenarios.

11.3.5 TFTP States

The TFTP specification is relatively informal; more recent protocols are often described using finite-state
terminology. In each allowable state, the specification spells out the appropriate response to all packets.
Above we defined a DALLYING state, for the receiver only, with a specific response to arriving Data[N]
packets. There are two other important conceptual states for TFTP receivers, which we might call UNLATCHED and ESTABLISHED.
When the receiver-client first sends RRQ, it does not know the port number from which the sender will send
packets. We will call this state UNLATCHED, as the receiver has not latched on to the correct port. In
this state, the receiver waits until it receives a packet from the sender that looks like a Data[1] packet; that
is, it is from the senders IP address, it has a plausible length, it is a DATA packet, and its block number is
1. When this packet is received, the receiver records s_port, and enters the ESTABLISHED state.
Once in the ESTABLISHED state, the receiver verifies for all packets that the source port number is s_port.
If a packet arrives from some other port, the receiver sends back to its source an ERROR packet with
Unknown Transfer ID, but continues with the original transfer.
Here is an outline, in java, of what part of the TFTP receiver source code might look like; the code here
handles the ESTABLISHED state. Somewhat atypically, the code here times out and retransmits ACK
packets if no new data is received in the interval TIMEOUT; generally timeouts are implemented only at the
TFTP sender side. Error processing is minimal, though error responses are sent in response to packets from
the wrong port as described in the previous section. For most of the other error conditions checked for, there
is no defined TFTP response.
The variables state, sendtime, TIMEOUT, thePacket, theAddress, thePort, blocknum and
expected_block would need to have been previously declared and initialized; sendtime represents
the time the most recent ACK response was sent. Several helper functions, such as getTFTPOpcode()
and write_the_data(), would have to be defined. The remote port thePort would be initialized at
the time of entry to the ESTABLISHED state; this is the port from which a packet must have been sent if it
is to be considered valid. The loop here transitions to the DALLY state when a packet marking the end of
the data has been received.
// TFTP code for ESTABLISHED state
while (state == ESTABLISHED) {
// check elapsed time
if (System.currentTimeMillis() > sendtime + TIMEOUT) {
retransmit_most_recent_ACK()
sendtime = System.currentTimeMillis()

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// receive the next packet

try {
s.receive(thePacket);
}
catch (SocketTimeoutException stoe) { continue; }
catch (IOException ioe) { System.exit(1); }

// try again
// other errors

if (thePacket.getAddress() != theAddress) continue;

if (thePacket.getPort() != thePort) {
send_error_packet(...);
// Unknown Transfer ID; see text
continue;
}
if (thePacket.getLength() < TFTP_HDR_SIZE) continue;
// TFTP_HDR_SIZE = 4
opcode = thePacket.getData().getTFTPOpcode()
blocknum = thePacket.getData().getTFTPBlock()
if (opcode != DATA) continue;
if (blocknum != expected_block) continue;
write_the_data(...);
expected_block ++;
send_ACK(...);
// and save it too for possible retransmission
sendtime = System.currentTimeMillis();
datasize = thePacket.getLength() - TFTP_HDR_SIZE;
if (datasize < MAX_DATA_SIZE) state = DALLY; // MAX_DATA_SIZE = 512
}

Note that the check for elapsed time is quite separate from the check for the
SocketTimeoutException. It is possible for the receiver to receive a steady stream of wrong
packets, so that it never encounters a SocketTimeoutException, and yet no good packet arrives
and so the receiver must still arrange (as above) for a timeout and retransmission.

11.3.6 TFTP scenarios

1. Duplicated RRQ: This was addressed above. Usually two child processes will start on the server. The
one that the client receives Data[1] from first is the one the client will latch on to; the other will be sent an
ERROR packet.
2. Lost final ACK: This is addressed with the DALLYING state.
3. Old duplicate: From the same connection, this is addressed by not allowing the 16-bit sequence number
to wrap around. This limits the maximum TFTP transfer to 65,535 blocks, or 32 megabytes.
For external old duplicates, involving an earlier instance of the connection, the only way this can happen is
if both sides choose the same port number for both instances. If either side chooses a new port, this problem
is prevented. If ports are chosen at random as in the sidebar above, the probability that both sides will chose
the same ports for the subsequent connection is around 1/232 ; if ports are assigned by the operating system,
there is an implicit assumption that the OS will not reissue the same port twice in rapid succession. Note that
this issue represents a second, more fundamental and less pragmatic, reason for having the server choose a
new port for each transfer.
After enough time, port numbers will eventually be recycled, but we will assume old duplicates have a
limited lifetime.

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4. Getting a different file than requested: Suppose the client sends RRQ(foo), but transmission is
delayed. In the meantime, the client reboots or aborts, and then sends RRQ(bar). This second RRQ is
lost, but the server sends Data[1] for foo.
At this point the client believes it is receiving file bar, but is in fact receiving file foo.
In practical terms, this scenario seems to be of limited importance, though diskless workstations often did
use TFTP to request their boot image file when restarting.
If the sender reboots, the transfer simply halts.
5. Malicious flooding: A malicious application aware that client C is about to request a file might send
repeated copies of bad Data[1] to likely ports on C. When C does request a file (eg if it requests a boot
image upon starting up, from port 1024), it may receive the malicious file instead of what it asked for.
This is a consequence of the server handoff from port 69 to a new port. Because the malicious application
must guess the clients port number, this scenario too appears to be of limited importance.

11.3.7 TFTP Throughput

On a single physical Ethernet, the TFTP sender and receiver would alternate using the channel, with very
little turnaround time; the effective throughput would be close to optimal.
As soon as the store-and-forward delays of switches and routers are introduced, though, stop-and-wait becomes a performance bottleneck. Suppose for a moment that the path from sender A to receiver B passes
through two switches: AS1S2B, and that on all three links only the bandwidth delay is significant.
Because ACK packets are so much smaller than DATA packets, we can effectively ignore the ACK travel
time from B to A.
With these assumptions, the throughput is about a third of the underlying bandwidth. This is because only
one of the three links can be active at any given time; the other two must be idle. We could improve
throughput threefold by allowing A to send three packets at a time:
packet 1 from A to S1
packet 2 from A to S1 while packet 1 goes from S1 to S2
packet 3 from A to S1 while packet 2 goes from S1 to S2 and packet 1 goes from S2 to B
This amounts to sliding windows with a winsize of three. TFTP does not support this; in the next chapter
we study TCP, which does.

11.4 Remote Procedure Call (RPC)

A very different communications model, usually but not always implemented over UDP, is that of Remote
Procedure Call, or RPC. The name comes from the idea that a procedure call is being made over the
network; host A packages up a request, with parameters, and sends it to host B, which returns a reply. The
term request/reply protocol is also used for this. The side making the request is known as the client, and
the other side the server.
One common example is that of DNS: a host sends a DNS lookup request to its DNS server, and receives a
reply. Other examples include password verification, system information retrieval, database queries and file
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I/O (below). RPC is also quite successful as the mechanism for interprocess communication within CPU
clusters, perhaps its most time-sensitive application.
While TCP can be used for processes like these, this adds the overhead of creating and tearing down a
connection; in many cases, the RPC exchange consists of nothing further beyond the request and reply and
so the TCP overhead would be nontrivial. RPC over UDP is particularly well suited for transactions where
both endpoints are quite likely on the same LAN, or are otherwise situated so that losses due to congestion
are negligible.
The drawback to UDP is that the RPC layer must then supply its own acknowledgment protocol. This is
not terribly difficult; usually the reply serves to acknowledge the request, and all that is needed is another
ACK after that. If the protocol is run over a LAN, it is reasonable to use a static timeout period, perhaps
somewhere in the range of 0.5 to 1.0 seconds.

Nonetheless, there are some niceties that early RPC implementations sometimes ignored, leading to a complicated history; see 11.4.2 Sun RPC below.
It is essential that requests and replies be numbered (or otherwise identified), so that the client can determine
which reply matches which request. This also means that the reply can serve to acknowledge the request;
if reply[N] is not received; the requester retransmits request[N]. This can happen either if request[N] never
arrived, or if it was reply[N] that got lost:

When the server creates reply[N] and sends it to the client, it must also keep a cached copy of the reply, until
such time as ACK[N] is received.

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After sending reply[N], the server may receive ACK[N], indicating all is well, or may receive request[N]
again, indicating that reply[N] was lost, or may experience a timeout, indicating that either reply[N] or
ACK[N] was lost. In the latter two cases, the server should retransmit reply[N] and wait again for ACK[N].

11.4.1 Network File Sharing

In terms of total packet volume, the application making the greatest use of early RPC was Suns Network
File Sharing, or NFS; this allowed for a filesystem on the server to be made available to clients. When the
client opened a file, the server would send back a file handle that typically included the files identifying
inode number. For read() operations, the request would contain the block number for the data to be
read, and the corresponding reply would contain the data itself; blocks were generally 8 KB in size. For
write() operations, the request would contain the block of data to be written together with the block
number; the reply would contain an acknowledgment that it was received.
Usually an 8 KB block of data would be sent as a single UDP/IP packet, using IP fragmentation for transmission over Ethernet.

11.4.2 Sun RPC

The original simple model above is quite serviceable. However, in the RPC implementation developed by
Sun Microsystems and documented in RFC 1831 (and officially known as Open Network Computing, or
ONC, RPC), the final acknowledgment was omitted. As there are relatively few packet losses on a LAN,
this was not quite as serious as it might sound, but it did have a major consequence: the server could now
not afford to cache replies, as it would never receive an indication that it was ok to delete them. Therefore,
the request was re-executed upon receipt of a second request[N], as in the right-hand lost reply diagram
above.
This was often described as at-least-once semantics: if a client sent a request, and eventually received
a reply, the client could be sure that the request was executed at least once, but if a reply got lost then
the request might be transmitted more than once. Applications, therefore, had to be aware that this was a
possibility.
It turned out that for many requests, duplicate execution was not a problem. A request that has the same result
(and same side effects on the server) whether executed once or executed twice is known as idempotent.
While a request to read or write the next block of a file is not idempotent, a request to read or write block
37 (or any other specific block) is idempotent. Most data queries are also idempotent; a second query
simply returns the same data as the first. Even file open() operations are idempotent, or at least can be
implemented as such: if a file is opened the second time, the file handle is simply returned a second time.
Alas, there do exist fundamentally non-idempotent operations. File locking is one, or any form of exclusive
file open. Creating a directory is another, because the operation must fail if the directory already exists. Even
opening a file is not idempotent if the server is expected to keep track of how many open() operations have
been called, in order to determine if a file is still in use.
So why did Sun RPC take this route? One major advantage of at-least-once semantics is that it allowed the
server to be stateless. The server would not need to maintain any RPC state, because without the final ACK
there is no server RPC state to be maintained; for idempotent operations the server would generally not have
to maintain any application state either. The practical consequence of this was that a server could crash and,
because there was no state to be lost, could pick up right where it left off upon restarting.
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Statelessness Inaction
Back when the Loyola CS department used Sun NFS extensively, server crashes would bring people
calmly out of their offices to find out what had happened; client-workstation processes doing I/O would
have locked up. Everyone would mill about in the hall until the server was rebooted, at which point
they would return to their work and were almost always able to pick up where they left off. If the server
had not been stateless, users would have been quite a bit less happy.
It is, of course, also possible to build recovery mechanisms into stateful protocols.
The lack of file-locking and other non-idempotent I/O operations, along with the rise of cheap clientworkstation storage (and, for that matter, more-reliable servers), eventually led to the decline of NFS over
RPC, though it has not disappeared. NFS can, if desired, also be run (statefully!) over TCP.

11.4.3 Serialization
In some RPC systems, even those with explicit ACKs, requests are executed serially by the server. Serial
execution is automatic if request[N+1] serves as an implicit ACK[N]. This is a problem for file I/O operations, as physical disk drives are generally most efficient when the I/O operations can be reordered to suit
the geometry of the disk. Disk drives commonly use the elevator algorithm to process requests: the read
head moves from low-numbered tracks outwards to high-numbered tracks, pausing at each track for which
there is an I/O request. Waiting for the Nth read to complete before asking the disk to start the N+1th one is
slow.
The best solution here is to allow multiple outstanding requests and out-of-order replies.

11.4.4 Refinements
One basic network-level improvement to RPC concerns the avoidance of IP-level fragmentation. While
fragmentation is not a major performance problem on a single LAN, it may have difficulties over longer
distances. One possible refinement is an RPC-level large-message protocol, that fragments at the RPC layer
and which supports a mechanism for retransmission, if necessary, only of those fragments that are actually
lost.
Another optimization might address the possibility that the server reboots. If a client really wants to be sure
that its request is executed only once, it needs to be sure that the server did not reboot between the original
request and the clients retransmission following a timeout. One way to achieve this is for the server to
maintain a reboot counter, written to the disk and incremented after each restart, and then to include the
value of the reboot counter in each reply. Requests contain the clients expected value for the server reboot
counter; if at the server end there is not a match, the client is notified of the potential error. Full recovery
from what may have been a partially executed request, however, requires some form of application-layer
journal log like that used for database servers.

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11.5 Epilog
UDP does not get as much attention as TCP, but between avoidance of connection-setup overhead, avoidance
of head-of-line blocking and high LAN performance, it holds its own.
We also use UDP here to illustrate fundamental transport issues, both abstractly and for the specific protocol
TFTP. We will revisit these fundamental issues extensively in the next chapter in the context of TCP; these
issues played a major role in TCPs design.

11.6 Exercises
1. Perform the UDP simplex-talk experiments discussed at the end of 11.1.2
multiple clients have simultaneous sessions with the same server?

UDP Simplex-Talk. Can

2. What would happen in TFTP if both sides implemented retransmit-on-timeout and neither side implemented retransmit-on-duplicate? Assume the actual transfer time is negligible. Assume Data[3] is sent but
the first instance is lost. Consider these cases:
sender timeout = receiver timeout = 2 seconds
sender timeout = 1 second, receiver timeout = 3 seconds
sender timeout = 3 seconds, receiver timeout = 1 second
3. In the previous problem, how do things change if ACK[3] is the packet that is lost?
4. Spell out plausible responses for a TFTP receiver upon receipt of a Data[N] packet for each of the TFTP
states UNLATCHED, ESTABLISHED, and DALLYING proposed in 11.3.5 TFTP States. Your answer
may depend on N and the packet size. Indicate the events that cause a transition from one state to the next.
Example: upon receipt of an ERROR packet, TFTP would in all three states exit.
5.
In the TFTP-receiver code in 11.3.5
TFTP States, explain why we must check
thePacket.getLength() before extracting the opcode and block number.
6. Assume both the TFTP sender and the TFTP receiver implement retransmit-on-timeout but not retransmiton-duplicate. Outline a TFTP scenario in which the TFTP receiver of 11.3.5 TFTP States sets a socket
timeout interval but never encounters a hard timeout that is, a SocketTimeoutException and
yet must timeout and retransmit. Hint: the only way to avoid a hard timeout is constantly to receive some
packet before the timeout timer expires.
7. In 11.3.6 TFTP scenarios, under Old duplicate, we claimed that if either side changed ports, the
old-duplicate problem would not occur.

(a). If the client changes its port number on a subsequent connection, but the server does not, what prevents
the old-duplicate problem?
(b). If the server changes its port number on a subsequent connection, but the client does not, what prevents
the old-duplicate problem?

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8. In the simple RPC protocol at the beginning of 11.4 Remote Procedure Call (RPC), suppose that the
server sends reply[N] and experiences a timeout, receiving nothing back from the client. In the text we
suggested that most likely this meant ACK[N] was lost. Give another loss scenario, involving the loss of
two packets. Assume the client and the server have the same timeout interval.
9. Suppose a Sun RPC read() request ends up executing twice. Unfortunately, in between successive
read() operations the block of data is updated by another process, so different data is returned. Is this a
failure of idempotence? Why or why not?
10. Outline an RPC protocol in which multiple requests can be outstanding, and replies can be sent in
any order. Assume that requests are numbered, and that ACK[N] acknowledges reply[N]. Should ACKs be
cumulative? If not, what should happen if an ACK is lost?

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12 TCP TRANSPORT

The standard transport protocols riding above the IP layer are TCP and UDP. As we saw in 11 UDP
Transport, UDP provides simple datagram delivery to remote sockets, that is, to host,port pairs. TCP
provides a much richer functionality for sending data to (connected) sockets.
TCP is quite different in several dimensions from UDP. TCP is stream-oriented, meaning that the application can write data in very small or very large amounts and the TCP layer will take care of appropriate
packetization. TCP is connection-oriented, meaning that a connection must be established before the beginning of any data transfer. TCP is reliable, in that TCP uses sequence numbers to ensure the correct order
of delivery and a timeout/retransmission mechanism to make sure no data is lost short of massive network
failure. Finally, TCP automatically uses the sliding windows algorithm to achieve throughput relatively
close to the maximum available.
These features mean that TCP is very well suited for the transfer of large files. The two endpoints open a
connection, the file data is written by one end into the connection and read by the other end, and the features
above ensure that the file will be received correctly. TCP also works quite well for interactive applications
where each side is sending and receiving streams of small packets. Examples of this include ssh or telnet,
where packets are exchanged on each keystroke, and database connections that may carry many queries per
second. TCP even works reasonably well for request/reply protocols, where one side sends a message,
the other side responds, and the connection is closed. The drawback here, however, is the overhead of
setting up a new connection for each request; a better application-protocol design might be to allow multiple
request/reply pairs over a single TCP connection.
Note that the connection-orientation and reliability of TCP represent abstract features built on top of the IP
layer which supports neither of them.
The connection-oriented nature of TCP warrants further explanation. With UDP, if a server opens a socket
(the OS object, with corresponding socket address), then any client on the Internet can send to that socket,
via its socket address. Any UDP application, therefore, must be prepared to check the source address of each
packet that arrives. With TCP, all data arriving at a connected socket must come from the other endpoint of
the connection. When a server S initially opens a socket s, that socket is unconnected; it is said to be in
the LISTEN state. While it still has a socket address consisting of its host and port, a LISTENing socket
will never receive data directly. If a client C somewhere on the Internet wishes to send data to s, it must first
establish a connection, which will be defined by the socketpair consisting of the socket addresses at both C
and S. As part of this connection process, a new connected child socket sC will be created; it is sC that will
receive any data sent from C. Usually, S will also create a new thread or process to handle communication
with sC . Typically the server S will have multiple connected children of s, and, for each one, a process
attached to it.
If C1 and C2 both connect to s, two connected sockets at S will be created, s1 and s2 , and likely two separate
processes. When a packet arrives at S addressed to the socket address of s, the source socket address will
also be examined to determine whether the data is part of the C1S or the C2S connection, and thus whether
a read on s1 or on s2 , respectively, will see the data.
If S is acting as an ssh server, the LISTENing socket listens on port 22, and the connected child sockets
correspond to the separate user login connections; the process on each child socket represents the login
process of that user, and may run for hours or days.

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In Chapter 1 we likened TCP sockets to telephone connections, with the server like one high-volume phone
number 800-BUY-NOWW. The unconnected socket corresponds to the number everyone dials; the connected sockets correspond to the actual calls. (This analogy breaks down, however, if one looks closely at
the way such multi-operator phone lines are actually configured: each typically does have its own number.)

12.1 The End-to-End Principle

The End-to-End Principle is spelled out in [SRC84]; it states in effect that transport issues are the responsibility of the endpoints in question and thus should not be delegated to the core network. This idea has been
very influential in TCP design.
Two issues falling under this category are data corruption and congestion. For the first, even though essentially all links on the Internet have link-layer checksums to protect against data corruption, TCP still adds
its own checksum (in part because of a history of data errors introduced within routers). For the latter, TCP
is today essentially the only layer that addresses congestion management.
Saltzer, Reed and Clark categorized functions that were subject to the End-to-End principle this way:
The function in question can completely and correctly be implemented only with the knowledge
and help of the application standing at the end points of the communication system. Therefore,
providing that questioned function as a feature of the communication system itself is not possible. (Sometimes an incomplete version of the function provided by the communication system
may be useful as a performance enhancement.)
This does not mean that the backbone Internet should not concern itself with congestion; it means that
backbone congestion-management mechanisms should not completely replace end-to-end congestion management.

12.2 TCP Header

Below is a diagram of the TCP header. As with UDP, source and destination ports are 16 bits. The checksum
prevents delivery of corrupted data. The Data Offset is for specifying the number of words of Options.

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The sequence and acknowledgment numbers are for numbering the data, at the byte level. This allows
TCP to send 1024-byte blocks of data, incrementing the sequence number by 1024 between successive
packets, or to send 1-byte telnet packets, incrementing the sequence number by 1 each time. There is no
distinction between DATA and ACK packets; all packets carrying data from A to B also carry the most
current acknowledgment of data sent from B to A. Many TCP applications are largely unidirectional, in
which case the sender would include essentially the same acknowledgment number in each packet while the
receiver would include essentially the same sequence number.
It is traditional to refer to the data portion of TCP packets as segments.
The value of the sequence number, in relative terms, is the position of the first byte of the packet in the data
stream, or the position of what would be the first byte in the case that no data was sent. The value of the
acknowledgment number, again in relative terms, represents the byte position for the next byte expected.
Thus, if a packet contains 1024 bytes of data and the first byte is number 1, then that would be the sequence
number. The data bytes would be positions 1-1024, and the ACK returned would have acknowledgment
number 1025.
The sequence and acknowledgment numbers, as sent, represent these relative values plus an Initial Sequence Number, or ISN, that is fixed for the lifetime of the connection. Each direction of a connection has
its own ISN; see below.
TCP acknowledgments are cumulative: when an endpoint sends a packet with an acknowledgment number
of N, it is acknowledging receipt of all data bytes numbered less than N. Standard TCP provides no mechanism for acknowledging receipt of packets 1, 2, 3 and 5; the highest cumulative acknowledgment that could
be sent in that situation would be to acknowledge packet 3.
The TCP header defines six important flag bits; the brief definitions here are expanded upon in the sequel:
SYN: for SYNchronize; marks packets that are part of the new-connection handshake
ACK: indicates that the header Acknowledgment field is valid; that is, all but the first packet
FIN: for FINish; marks packets involved in the connection closing
PSH: for PuSH; marks non-full packets that should be delivered promptly at the far end
RST: for ReSeT; indicates various error conditions
URG: for URGent; part of a now-seldom-used mechanism for high-priority data
CWR and ECE: part of the Explicit Congestion Notification mechanism, 14.8.2 Explicit Congestion
Notification (ECN)

12.3 TCP Connection Establishment

TCP connections are established via an exchange known as the three-way handshake. If A is the client and
B is the LISTENing server, then the handshake proceeds as follows:
A sends B a packet with the SYN bit set (a SYN packet)
B responds with a SYN packet of its own; the ACK bit is now also set
A responds to Bs SYN with its own ACK

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Normally, the three-way handshake is triggered by an applications request to connect; data can be sent only
after the handshake completes. This means a one-RTT delay before any data can be sent. The original TCP
standard RFC 793 does allow data to be sent with the first SYN packet, as part of the handshake, but such
data cannot be released to the remote-endpoint application until the handshake completes. Most traditional
TCP programming interfaces offer no support for this early-data option.
There are recurrent calls for TCP to support earlier data in a more useful manner, so as to achieve request/reply turnaround comparable to that with RPC (11.4 Remote Procedure Call (RPC)). We return to
this in 12.11 TCP Faster Opening.
To close the connection, a superficially similar exchange involving FIN packets may occur:
A sends B a packet with the FIN bit set (a FIN packet), announcing that it has finished sending data
B sends A an ACK of the FIN
When B is also ready to cease sending, it sends its own FIN to A
A sends B an ACK of the FIN; this is the final packet in the exchange

The FIN handshake is really more like two separate two-way FIN/ACK handshakes. If B is ready to close
immediately, it may send its FIN along with its ACK of As FIN, as is shown in the above diagram at the
left. In theory this is rare, however, as the ACK of As FIN is generated by the kernel but Bs FIN cannot
be sent until Bs process is scheduled to run on the CPU. On the other hand, it is possible for B to send a
considerable amount of data back to A after A sends its FIN, as is shown at the right. The FIN is, in effect, a
promise not to send any more, but that side of the connection must still be prepared to receive data. A good
example of this occurs when A is sending a stream of data to B to be sorted; A sends FIN to indicate that it
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is done sending, and only then does B sort the data and begin sending it back to A. This can be generated
with the command, on A, cat thefile | ssh B sort.
In the following table, relative sequence numbers are used, which is to say that sequence numbers begin
with 0 on each side. The SEQ numbers in bold on the A side correspond to the ACK numbers in bold on
the B side; they both count data flowing from A to B.
1
2
3
4
5
6
7
8
9
10
11,12
13
14
15

A sends
SYN, seq=0

B sends
SYN+ACK, seq=0, ack=1 (expecting)

ACK, seq=1, ack=1 (ACK of SYN)

abc, seq=1, ack=1
ACK, seq=1, ack=4
defg, seq=4, ack=1
seq=1, ack=8
foobar, seq=8, ack=1
seq=1, ack=14, hello
seq=14, ack=6, goodbye
seq=21, ack=6, FIN

seq=6, ack=21 ;; ACK of goodbye, crossing packets

seq=6, ack=22 ;; ACK of FIN
seq=6, ack=22, FIN

seq=22, ack=7 ;; ACK of FIN

(We will see below that this table is slightly idealized, in that real sequence numbers do not start at 0.)
Here is the ladder diagram corresponding to this connection:

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In terms of the sequence and acknowledgment numbers, SYNs count as 1 byte, as do FINs. Thus, the SYN
counts as sequence number 0, and the first byte of data (the a of abc) counts as sequence number 1.
Similarly, the ack=21 sent by the B side is the acknowledgment of goodbye, while the ack=22 is the
acknowledgment of As subsequent FIN.
Whenever B sends ACN=n, A follows by sending more data with SEQ=n.
TCP does not in fact transport relative sequence numbers, that is, sequence numbers as transmitted do not
begin at 0. Instead, each side chooses its Initial Sequence Number, or ISN, and sends that in its initial
SYN. The third ACK of the three-way handshake is an acknowledgment that the server sides SYN response
was received correctly. All further sequence numbers sent are the ISN chosen by that side plus the relative
sequence number (that is, the sequence number as if numbering did begin at 0). If A chose ISNA =1000, we
would add 1000 to all the bold entries above: A would send SYN(seq=1000), B would reply with ISNB and
ack=1001, and the last two lines would involve ack=1022 and seq=1022 respectively. Similarly, if B chose
ISNB =7000, then we would add 7000 to all the seq values in the B sends column and all the ack values in
the A sends column. The table above up to the point B sends goodbye, with actual sequence numbers
instead of relative sequence numbers, is below:

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1
2
3
4
5
6
7
8
9
10

A, ISN=1000
SYN, seq=1000

B, ISN=7000
SYN+ACK, seq=7000, ack=1001

ACK, seq=1001, ack=7001

abc, seq=1001, ack=7001
ACK, seq=7001, ack=1004
defg, seq=1004, ack=7001
seq=7001, ack=1008
foobar, seq=1008, ack=7001
seq=7001, ack=1014, hello
seq=1014, ack=7006, goodbye

If B had not been LISTENing at the port to which A sent its SYN, its response would have been RST
(reset), meaning in this context connection refused. Similarly, if A sent data to B before the SYN
packet, the response would have been RST. Finally, either side can abort the connection at any time by
sending RST.
If A sends a series of small packets to B, then B has the option of assembling them into a full-sized I/O buffer
before releasing them to the receiving application. However, if A sets the PSH bit on each packet, then B
should release each packet immediately to the receiving application. In Berkeley Unix and most (if not all)
BSD-derived socket-library implementations, there is in fact no way to set the PSH bit; it is set automatically
for each write. (But even this is not guaranteed as the sender may leave the bit off or consolidate several
PuSHed writes into one packet; this makes using the PSH bit as a record separator difficult. In the program
written to generate the WireShark packet trace, below, most of the time the strings abc, defg, etc were
PuSHed separately but occasionally they were consolidated into one packet.)
As for the URG bit, imagine an ssh connection, in which A has sent a large amount of data to B, which
is momentarily stalled processing it. The user at A wishes to abort the connection by sending the interrupt
character CNTL-C. Under normal processing, the application at B would have to finish processing all the
pending data before getting to the CNTL-C; however, the use of the URG bit can enable immediate asynchronous delivery of the CNTL-C. The bit is set, and the TCP headers Urgent Pointer field points to the
CNTL-C far ahead in the normal data stream. The receiver then skips processing the arriving data stream in
first-come-first-served order and processes the urgent data first. For this to work, the receiving process must
have signed up to receive an asynchronous signal when urgent data arrives.
The urgent data does appear in the ordinary TCP data stream, and it is up to the protocol to determine the
length of the urgent data substring, and what to do with the unread, buffered data sent ahead of the urgent
data. For the CNTL-C example, the urgent data consists of a single character, and the earlier data is simply
discarded.

12.4 TCP and WireShark

Below is a screenshot of the WireShark program displaying a tcpdump capture intended to represent the
TCP exchange above. Both hosts involved in the packet exchange were linux systems. Side A uses socket
address 10.0.0.3,45815 and side B (the server) uses 10.0.0.1,54321.
WireShark is displaying relative sequence numbers. The first three packets correspond to the three-way
handshake, and packet 4 is the first data packet. Every data packet has the flags [PSH, ACK] displayed.

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The data in the packet can be inferred from the WireShark Len field, as each of the data strings sent has a
different length.

The packets are numbered the same as in the table above up through packet 8, containing the string foobar.
At that point the table shows B replying by a combined ACK plus the string hello; in fact, TCP sent the
ACK alone and then the string hello; these are WireShark packets 9 and 10 (note packet 10 has Len=5).
Wireshark packet 11 is then a standalone ACK from A to B, acknowledging the hello. WireShark packet
12 (the packet highlighted) then corresponds to table packet 10, and contains goodbye (Len=7); this string
can be seen at the right side of the bottom pane.
The table view shows As FIN (packet 11) crossing with Bs ACK of goodbye (packet 12). In the WireShark view, As FIN is packet 13, and is sent about 0.01 seconds after goodbye; B then ACKs them both
with packet 14. That is, the table-view packet 12 does not exist in the WireShark view.
Packets 11, 13, 14 and 15 in the table and 13, 14, 15 and 16 in the WireShark screen dump correspond to
the connection closing. The program that generated the exchange at Bs side had to include a sleep delay
of 40 ms between detecting the closed connection (that is, reading As FIN) and closing its own connection
(and sending its own FIN); otherwise the ACK of As FIN traveled in the same packet with Bs FIN.
The ISN for A here was 551144795 and Bs ISN was 1366676578. The actual pcap packet-capture file is at
demo_tcp_connection.pcap. The hardware involved used TCP checksum offloading to have the networkinterface card do the actual checksum calculations; as a result, the checksums are wrong in the actual capture
file. WireShark has an option to disable the reporting of this.

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12.5 TCP simplex-talk

Here is a Java version of the simplex-talk server for TCP. The main while loop has the call ss.accept()
at the start; ss is the ServerSocket object and is in LISTEN state. This call blocks until an incoming
connection is established, at which point it returns the connected child socket. Connections will be accepted
from all IP addresses of the server host, eg the normal IP address and also the loopback address 127.0.0.1.
Unlike the UDP case (11.1.2.2 UDP and IP addresses), the server response packets will always be sent
from the same server IP address that the client first used to reach the server.
A server application can process these connected children either serially or in parallel. The stalk version
here can handle both situations, either one connection at a time (THREADING = false), or by creating a
new thread for each connection (THREADING = true). In the former mode, if a second client connection
is made while the first is connected, then data can be sent on the second connection but it sits in limbo
until the first connection closes, at which point control returns to the ss.accept() call, the second
connection is processed, and the second connections data suddenly appears. The main loop is within the
line_talker() call, and does not return to ss.accept() until the connection has closed.
In the latter mode, the main loop spends almost all its time waiting in ss.accept(); when this returns a
child connection we immediately spawn a new thread to handle it, allowing the parent process to go back
to ss.accept(). This allows the program to accept multiple concurrent client connections, like the UDP
version.
The code here serves as a very basic example of the creation of Java threads. The inner class Talker has
a run() method, needed to implement the runnable interface. To start a new thread, we create a new
Talker instance; the start() call then begins Talker.run(), which runs for as long as the client
keeps the connection open. The file here is tcp_stalks.java.
/* THREADED simplex-talk TCP server */
/* can handle multiple CONCURRENT client connections */
/* newline is to be included at client side */
import java.net.*;
import java.io.*;
public class tstalks {
static public int destport = 5431;
static public int bufsize = 512;
static public boolean THREADING = true;
static public void main(String args[]) {
ServerSocket ss;
Socket s;
try {
ss = new ServerSocket(destport);
} catch (IOException ioe) {
System.err.println("cant create server socket");
return;
}
System.err.println("server starting on port " + ss.getLocalPort());
while(true) { // accept loop

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try {
s = ss.accept();
} catch (IOException ioe) {
System.err.println("Cant accept");
break;
}
if (THREADING) {
Talker talk = new Talker(s);
(new Thread(talk)).start();
} else {
line_talker(s);
}
} // accept loop
} // end of main

public static void line_talker(Socket s) {

int port = s.getPort();
InputStream istr;
try { istr = s.getInputStream(); }
catch (IOException ioe) {
System.err.println("cannot get input stream");
// most likely cause: s w
return;
}
System.err.println("New connection from <" +
s.getInetAddress().getHostAddress() + "," + s.getPort() + ">");
byte[] buf = new byte[bufsize];
int len;
while (true) {
// while not done reading the socket
try {
len = istr.read(buf, 0, bufsize);
}
catch (SocketTimeoutException ste) {
System.out.println("socket timeout");
continue;
}
catch (IOException ioe) {
System.err.println("bad read");
break;
// probably a socket ABORT; treat as a close
}
if (len == -1) break;
// other end closed gracefully
String str = new String(buf, 0, len);
System.out.print("" + port + ": " + str); // str should contain newline
} //while reading from s

try {istr.close();}
catch (IOException ioe) {System.err.println("bad stream close");return;}
try {s.close();}
catch (IOException ioe) {System.err.println("bad socket close");return;}
System.err.println("socket to port " + port + " closed");
} // line_talker

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static class Talker implements Runnable {

private Socket _s;
public Talker (Socket s) {
_s = s;
}

public void run() {

line_talker(_s);
} // run
// class Talker

Here is the client tcp_stalkc.java. As with the UDP version, the default host to connect to is localhost.
We first call InetAddress.getByName() to perform the DNS lookup. Part of the construction of the
Socket object is the connection to the desired dest and destport.
// TCP simplex-talk CLIENT in java
import java.net.*;
import java.io.*;
public class stalkc {
static public BufferedReader bin;
static public int destport = 5431;
static public void main(String args[]) {
String desthost = "localhost";
if (args.length >= 1) desthost = args[0];
bin = new BufferedReader(new InputStreamReader(System.in));
InetAddress dest;
System.err.print("Looking up address of " + desthost + "...");
try {
dest = InetAddress.getByName(desthost);
}
catch (UnknownHostException uhe) {
System.err.println("unknown host: " + desthost);
return;
}
System.err.println(" got it!");
System.err.println("connecting to port " + destport);
Socket s;
try {
s = new Socket(dest, destport);
}
catch(IOException ioe) {
System.err.println("cannot connect to <" + desthost + "," + destport + ">");
return;
}

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OutputStream sout;
try {
sout = s.getOutputStream();
}
catch (IOException ioe) {
System.err.println("I/O failure!");
return;
}
//============================================================
while (true) {
String buf;
try {
buf = bin.readLine();
}
catch (IOException ioe) {
System.err.println("readLine() failed");
return;
}
if (buf == null) break;
// user typed EOF character
buf = buf + "\n";
// protocol requires sender includes \n
byte[] bbuf = buf.getBytes();
try {
sout.write(bbuf);
}
catch (IOException ioe) {
System.err.println("write() failed");
return;
}
} // while
}
}

Here are some things to try with THREADING=false:

start up two clients while the server is running. Type some lines into both. Then exit the first client.
start up the client before the server.
start up the server, and then the client. Type some text to the client. Kill the server. What happens to
the client? (It may take a couple lines of input)
start the server, then the client. Kill the server and restart it. Now what happens to the client?
With THREADING=true, try connecting multiple clients simultaneously to the server. How does this behave
differently from the first example above?

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12.6 TCP state diagram

A formal definition of TCP involves the state diagram, with conditions for transferring from one state to
another, and responses to all packets from each state. The state diagram originally appeared in RFC 793; the
following diagram came from http://commons.wikimedia.org/wiki/File:Tcp_state_diagram_fixed.svg. The
blue arrows indicate the sequence of state transitions typically followed by the server; the brown arrows
represent the client. Arrows are labeled with event / action; that is, we move from LISTEN to SYN_RECD
upon receipt of a SYN packet; the action is to respond with SYN+ACK.
In general, the state-diagram approach to protocol design has proven very effective, and is used for most
protocols.

Here is the ladder diagram for the 14-packet connection described above, this time labeled with TCP states.

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The reader who is implementing TCP is encouraged to consult RFC 793 and updates. For the rest of us,
below are a few general observations about opening and closing connections.
Either side may elect to close the connection (just as either party to a telephone call may elect to hang up).
The first side to send a FIN takes the Active CLOSE path; the other side takes the Passive CLOSE path.
Although it essentially never occurs in practice, it is possible for each side to send the other a SYN, requesting a connection, simultaneously (that is, the SYNs cross on the wire). The telephony analogue occurs
when each party dials the other simultaneously. On traditional land-lines, each party then gets a busy signal.
On cell phones, your mileage may vary. With TCP, a single connection is created. With OSI TP4, two
connections are created. The OSI approach is not possible in TCP, as a connection is determined only by
the socketpair involved; if there is only one socketpair then there can be only one connection.
A simultaneous close having both sides simultaneously send each other FINs is a little more likely,
though still not very. Each side would move to state FIN_WAIT_1. Then, upon receiving each others FIN
packets, each side would move to CLOSING, and then to TIMEWAIT.
A TCP endpoint is half-closed if it has sent its FIN (thus promising not to send any more data) and is
waiting for the other sides FIN. A TCP endpoint is half-open if it is in the ESTABLISHED state, but in the
meantime the other side has rebooted. As soon as the ESTABLISHED side sends a packet, the other side
will respond with RST and the connection will be fully closed.
The normal close is

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In this scenario, A has moved from ESTABLISHED to FIN_WAIT_1 to FIN_WAIT_2 to TIMEWAIT (below) to CLOSED. B moves from ESTABLISHED to CLOSE_WAIT to LAST_ACK to CLOSED. All this
essentially amounts to two separate two-way closure handshakes.
However, it is possible for Bs ACK and FIN to be combined. In this case, A moves directly from
FIN_WAIT_1 to TIMEWAIT. In order for this to happen, when As FIN arrives at B, the socket-owning
process at B must immediately wake up, recognize that A has closed its end, and immediately close its own
end as well. This generates Bs FIN; all this must happen before Bs TCP layer sends the ACK of As FIN.
If the TCP layer adopts a policy of immediately sending ACKs upon receipt of any packet, this will never
happen, as the FIN will arrive well before Bs process can be scheduled to do anything. However, if B delays
its ACKs slightly, then it is possible for Bs ACK and FIN to be sent together.
Although this is not evident from the state diagram, the per-state response rules of TCP require that in the
ESTABLISHED state, if the receiver sends an ACK outside the current sliding window, then the correct
response is to reply with ones own current ACK. This includes the case where the receiver acknowledges
data not yet sent.
It is possible to view connection states under either linux or Windows with netstat -a. Most states are
ephemeral, exceptions being LISTEN, ESTABLISHED, TIMEWAIT, and CLOSE_WAIT. One sometimes
sees large numbers of connections in CLOSE_WAIT, meaning that the remote endpoint has closed the
connection and sent its FIN, but the process at your end has not executed close() on its socket. Often
this represents a programming error; alternatively, the process at the local end is still working on something.
Given a local port number p in state CLOSE_WAIT on a linux system, the (privileged) command lsof -i
:p will identify the process using port p.

12.7 TCP Old Duplicates

Conceptually, perhaps the most serious threat facing the integrity of TCP data is external old duplicates
(11.2 Fundamental Transport Issues), that is, very late packets from a previous instance of the connection.
Suppose a TCP connection is opened between A and B. One packet from A to B is duplicated and unduly
delayed, with sequence number N. The connection is closed, and then another instance is reopened, that is, a
connection is created using the same ports. At some point in the second connection, when an arriving packet
with seq=N would be acceptable at B, the old duplicate shows up. Later, of course, B is likely to receive a
seq=N packet from the new instance of the connection, but that packet will be seen by B as a duplicate (even
though the data does not match), and (we will assume) be ignored.

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For TCP, it is the actual sequence numbers, rather than the relative sequence numbers, that would have to
match up. The diagram above ignores that.
As with TFTP, coming up with a possible scenario accounting for the generation of such a late packet is not
easy. Nonetheless, many of the design details of TCP represent attempts to minimize this risk.
Solutions to the old-duplicates problem generally involve setting an upper bound on the lifetime of any
packet, the MSL, as we shall see in the next section. T/TCP (12.11 TCP Faster Opening) introduced a
connection-count field for this.
TCP is also vulnerable to sequence-number wraparound: arrival of an old duplicates from the same instance
of the connection. However, if we take the MSL to be 60 seconds, sequence-number wrap requires sending
232 bytes in 60 seconds, which requires a data-transfer rate in excess of 500 Mbps. TCP offers a fix for this
(Protection Against Wrapped Segments, or PAWS), but it was introduced relatively late; we return to this in
12.10 Anomalous TCP scenarios.

12.8 TIMEWAIT
The TIMEWAIT state is entered by whichever side initiates the connection close; in the event of a simultaneous close, both sides enter TIMEWAIT. It is to last for a time 2MSL, where MSL = Maximum Segment
Lifetime is an agreed-upon value for the maximum lifetime on the Internet of an IP packet. Traditionally MSL was taken to be 60 seconds, but more modern implementations often assume 30 seconds (for a
TIMEWAIT period of 60 seconds).
One function of TIMEWAIT is to solve the external-old-duplicates problem. TIMEWAIT requires that
between closing and reopening a connection, a long enough interval must pass that any packets from the
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first instance will disappear. After the expiration of the TIMEWAIT interval, an old duplicate cannot arrive.
A second function of TIMEWAIT is to address the lost-final-ACK problem (11.2 Fundamental Transport
Issues). If host A sends its final ACK to host B and this is lost, then B will eventually retransmit its final
packet, which will be its FIN. As long as A remains in state TIMEWAIT, it can appropriately reply to a
retransmitted FIN from B with a duplicate final ACK.
TIMEWAIT only blocks reconnections for which both sides reuse the same port they used before. If A
connects to B and closes the connection, A is free to connect again to B using a different port at As end.
Conceptually, a host may have many old connections to the same port simultaneously in TIMEWAIT; the
host must thus maintain for each of its ports a list of all the remote IP_address,port sockets currently in
TIMEWAIT for that port. If a host is connecting as a client, this list likely will amount to a list of recently
used ports; no port is likely to have been used twice within the TIMEWAIT interval. If a host is a server,
however, accepting connections on a standardized port, and happens to be the side that initiates the active
close and thus later goes into TIMEWAIT, then its TIMEWAIT list for that port can grow quite long.
Generally, busy servers prefer to be free from these bookkeeping requirements of TIMEWAIT, so many
protocols are designed so that it is the client that initiates the active close. In the original HTTP protocol,
version 1.0, the server sent back the data stream requested by the http GET message, and indicated the end
of this stream by closing the connection. In HTTP 1.1 this was fixed so that the client initiated the close;
this required a new mechanism by which the server could indicate I am done sending this file. HTTP 1.1
also used this new mechanism to allow the server to send back multiple files over one connection.
In an environment in which many short-lived connections are made from host A to the same port on server
B, port exhaustion having all ports tied up in TIMEWAIT is a theoretical possibility. If A makes 1000
connections per second, then after 60 seconds it has gone through 60,000 available ports, and there are
essentially none left. While this rate is high, early Berkeley-Unix TCP implementations often made only
about 4,000 ports available to clients; with a 120-second TIMEWAIT interval, port exhaustion would occur
with only 33 connections per second.
If you use ssh to connect to a server and then issue the netstat -a command on your own host (or, more
conveniently, netstat -a |grep -i tcp), you should see your connection in ESTABLISHED state.
If you close your connection and check again, your connection should be in TIMEWAIT.

12.9 The Three-Way Handshake Revisited

As stated earlier in 12.3 TCP Connection Establishment, both sides choose an ISN; actual sequence numbers are the sum of the senders ISN and the relative sequence number. There are two original reasons for this
mechanism, and one later one (12.9.1 ISNs and spoofing). The original TCP specification, as clarified in
RFC 1122, called for the ISN to be determined by a special clock, incremented by 1 every 4 microseconds.
The most basic reason for using ISNs is to detect duplicate SYNs. Suppose A initiates a connection to B
by sending a SYN packet. B replies with SYN+ACK, but this is lost. A then times out and retransmits its
SYN. B now receives As second SYN while in state SYN_RECEIVED. Does this represent an entirely new
request (perhaps A has suddenly restarted), or is it a duplicate? If A uses the clock-driven ISN strategy, B
can tell (almost certainly) whether As second SYN is new or a duplicate: only in the latter case will the ISN
values in the two SYNs match.
While there is no danger to data integrity if A sends a SYN, restarts, and sends the SYN again as part

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of a reopening the same connection, the arrival of a second SYN with a new ISN means that the original
connection cannot proceed, because that ISN is now wrong.
The clock-driven ISN also originally added a second layer of protection against external old duplicates.
Suppose that A opens a connection to B, and chooses a clock-based ISN N1 . A then transfers M bytes
of data, closed the connection, and reopens it with ISN N2 . If N1 + M < N2 , then the old-duplicates
problem cannot occur: all of the absolute sequence numbers used in the first instance of the connection
are less than or equal to N1 + M, and all of the absolute sequence numbers used in the second instance
will be greater than N2 . In fact, early Berkeley-Unix implementations of the socket library often allowed
a second connection meeting this ISN requirement to be reopened before TIMEWAIT would have expired;
this potentially addressed the problem of port exhaustion. Of course, if the first instance of the connection
transferred data faster than the ISN clock rate, that is at more than 250,000 bytes/sec, then N1 + M would
be greater than N2 , and TIMEWAIT would have to be enforced. But in the era in which TCP was first
developed, sustained transfers exceeding 250,000 bytes/sec were not common.
The three-way handshake was extensively analyzed by Dalal and Sunshine in [DS78]. The authors noted
that with a two-way handshake, the second side receives no confirmation that its ISN was correctly received.
The authors also observed that a four-way handshake in which the ACK of ISNA is sent separately from
ISNB , as in the diagram below could fail if one side restarted.

For this failure to occur, assume that after sending the SYN in line 1, with ISNA1 , A restarts. The ACK
in line 2 is either ignored or not received. B now sends its SYN in line 3, but A interprets this as a new
connection request; it will respond after line 4 by sending a fifth, SYN packet containing a different ISNA2 .
For B the connection is now ESTABLISHED, and if B acknowledges this fifth packet but fails to update its
record of As ISN, the connection will fail as A and B would have different notions of ISNA .

12.9.1 ISNs and spoofing

The clock-based ISN proved to have a significant weakness: it often allowed an attacker to guess the ISN a
remote host might use. It did not help any that an early version of Berkeley Unix, instead of incrementing the
ISN 250,000 times a second, incremented it once a second, by 250,000 (plus something for each connection).
By guessing the ISN a remote host would choose, an attacker might be able to mimic a local, trusted host,
and thus gain privileged access.
Specifically, suppose host A trusts its neighbor B, and executes with privileged status commands sent by
B; this situation was typical in the era of the rhost command. A authenticates these commands because
the connection comes from Bs IP address. The bad guy, M, wants to send packets to A so as to pretend

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to be B, and thus get a privileged command invoked. The connection only needs to be started; if the ruse
is discovered after the command is executed, it is too late. M can easily send a SYN packet to A with
Bs IP address in the source-IP field; M can probably temporarily disable B too, so that As SYN-ACK
response, which is sent to B, goes unnoticed. What is harder is for M to figure out how to guess how to ACK
ISNA . But if A generates ISNs with a slowly incrementing clock, M can guess the pattern of the clock with
previous connection attempts, and can thus guess ISNA with a considerable degree of accuracy. So M sends
SYN to A with B as source, A sends SYN-ACK to B containing ISNA , and M guesses this value and sends
ACK(ISNA +1) to A, again with B listed in the IP header as source, followed by a single-packet command.
This IP-spoofing technique was first described by Robert T Morris in [RTM85]; Morris went on to launch
the Internet Worm of 1988 using unrelated attacks. The IP-spoofing technique was used in the 1994 Christmas Day attack against UCSD, launched from Loyolas own apollo.it.luc.edu; the attack was associated with
Kevin Mitnick though apparently not actually carried out by him. Mitnick was arrested a few months later.
RFC 1948, in May 1996, introduced a technique for introducing a degree of randomization in ISN selection,
while still ensuring that the same ISN would not be used twice in a row for the same connection. The ISN
is to be the sum of the 4-s clock, C(t), and a secure hash of the connection information as follows:
ISN = C(t) + hash(local_addr, local_port, remote_addr, remote_port, key)
The key value is a random value chosen by the host on startup. While M, above, can poll A for its current
ISN, and can probably guess the hash function and the first four parameters above, without knowing the key
it cannot determine (or easily guess) the ISN value A would have sent to B. Legitimate connections between
A and B, on the other hand, see the ISN increasing at the 4-s rate.

12.10 Anomalous TCP scenarios

TCP, like any transport protocol, must address the transport issues in 11.2 Fundamental Transport Issues.
As we saw above, TCP addresses the Duplicate Connection Request (Duplicate SYN) issue by noting
whether the ISN has changed. This is handled at the kernel level by TCP, versus TFTPs application-level
(and rather desultory) approach to handing Duplicate RRQs.
TCP addresses Loss of Final ACK through TIMEWAIT: as long as the TIMEWAIT period has not expired,
if the final ACK is lost and the other side resends its final FIN, TCP will still be able to reissue that final
ACK. TIMEWAIT in this sense serves a similar function to TFTPs DALLY state.
External Old Duplicates, arriving as part of a previous instance of the connection, are prevented by TIMEWAIT, and may also be prevented by the use of a clock-driven ISN.
Internal Old Duplicates, from the same instance of the connection, that is, sequence number wraparound, is
only an issue for bandwidths exceeding 500 Mbps: only at bandwidths above that can 4 GB be sent in one
60-second MSL. In modern TCP implementations this is addressed by PAWS: Protection Against Wrapped
Segments (RFC 1323). PAWS adds a 32-bit timestamp option to the TCP header. The granularity of the
timestamp clock is left unspecified; one tick must be small enough that sequence numbers cannot wrap in
that interval (eg less than 3 seconds for 10,000 Mbps), and large enough that the timestamps cannot wrap
in time MSL. This is normally easy to arrange. An old duplicate due to sequence-number wraparound can
now be rejected as having an old timestamp.
Reboots are a potential problem as the host presumably has no record of what aborted connections need to
remain in TIMEWAIT. TCP addresses this on paper by requiring hosts to implement Quiet Time on Startup:
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no new connections are to be accepted for 1*MSL. No known implementations actually do this; instead,
they assume that the restarting process itself will take at least one MSL. This is no longer as certain as it
once was, but serious consequences have not ensued.

12.11 TCP Faster Opening

If a client wants to connect to a server, send a request and receive an immediate reply, TCP mandates one
full RTT for the three-way handshake before data can be delivered. This makes TCP one RTT slower than
UDP-based request-reply protocols. There have been periodic calls to allow TCP clients to include data with
the first SYN packet and have it be delivered immediately upon arrival this is known as accelerated open.
If there will be a series of requests and replies, the simplest fix is to pipeline all the requests and replies over
one persistent connection; the one-RTT delay then applies only to the first request. If the pipeline connection
is idle for a long-enough interval, it may be closed, and then reopened later if necessary.
An early accelerated-open proposal was T/TCP, or TCP for Transactions, specified in RFC 1644. T/TCP
introduced a connection count TCP option, called CC; each participant would include a 32-bit CC value
in its SYN; each participants own CC values were to be monotonically increasing. Accelerated open was
allowed if the server side had the clients previous CC in a cache, and the new CC value was strictly greater
than this cached value. This ensured that the new SYN was not a duplicate of an older SYN.
Unfortunately, this also bypasses the modest authentication of the clients IP address provided by the full
three-way handshake, worsening the spoofing problem of 12.9.1 ISNs and spoofing. If malicious host M
wants to pretend to be B when sending a privileged request to A, all M has to do is send a single SYN+Data
packet with an extremely large value for CC. Generally, the accelerated open succeeded as long as the CC
value presented was larger that the value A had cached for B; it did not have to be larger by exactly 1.
The recent TCP Fast Open proposal, described in Internet Draft draft-ietf-tcpm-fastopen-05.txt, involves a
secure cookie sent by the client as a TCP option; if a SYN+Data packet has a valid cookie, then the client
has proven its identity and the data may be released immediately to the receiving application. Cookies are
cryptographically secure, and are requested ahead of time from the server.
Because cookies have an expiration date and must be requested ahead of time, TCP Fast Open is not fundamentally faster from the connection-pipeline option, except that holding a TCP connection open uses
more resources than simply storing a cookie. The likely application for TCP Fast Open is in accessing web
servers. Web clients and servers already keep a persistent connection open for a while, but often a while
here amounts only to several seconds; TCP Fast Open cookies could remain active for much longer.

12.12 Path MTU Discovery

TCP connections are more efficient if they can keep large packets flowing between the endpoints. Once upon
a time, TCP endpoints included just 512 bytes of data in each packet that was not destined for local delivery,
to avoid fragmentation. TCP endpoints now typically engage in Path MTU Discovery which almost always
allows them to send larger packets; backbone ISPs are now usually able to carry 1500-byte packets. The
Path MTU is the largest packet size that can be sent along a path without fragmentation.
The strategy is to send an initial data packet with the IP DONT_FRAG bit set. If the ICMP message
Frag_Required/DONT_FRAG_Set comes back, or if the packet times out, the sender tries a smaller size. If
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the sender receives a TCP ACK for the packet, on the other hand, indicating that it made it through to the
other end, it might try a larger size. Usually, the size range of 512-1500 bytes is covered by less than a dozen
discrete values; the point is not to find the exact Path MTU but to determine a reasonable approximation
rapidly.

12.13 TCP Sliding Windows

TCP implements sliding windows, in order to improve throughput. Window sizes are measured in terms
of bytes rather than packets; this leaves TCP free to packetize the data in whatever segment size it elects.
In the initial three-way handshake, each side specifies the maximum window size it is willing to accept, in
the Window Size field of the TCP header. This 16-bit field can only go to 64 KB, and a 1 Gbps 100
ms bandwidthdelay product is 12 MB; as a result, there is a TCP Window Scale option that can also be
negotiated in the opening handshake. The scale option specifies a power of 2 that is to be multiplied by the
actual Window Size value. In the WireShark example above, the client specified a Window Size field of
5888 (= 4 1472) in the third packet, but with a Window Scale value of 26 = 64 in the first packet, for an
effective window size of 64 5888 = 256 segments of 1472 bytes. The server side specified a window size
of 5792 and a scaling factor of 25 = 32.
TCP may either transmit a bulk stream of data, using sliding windows fully, or it may send slowly generated
interactive data; in the latter case, TCP may never have even one full segment outstanding.
In the following chapter we will see that a sender frequently reduces the actual TCP window size, in order
to avoid congestion; the window size included in the TCP header is known as the Advertised Window Size.
On startup, TCP does not send a full window all at once; it uses a mechanism called slow start.

12.14 TCP Delayed ACKs

TCP receivers are allowed briefly to delay their ACK responses to new data. This offers perhaps the most
benefit for interactive applications that exchange small packets, such as ssh and telnet. If A sends a data
packet to B and expects an immediate response, delaying Bs ACK allows the receiving application on B
time to wake up and generate that application-level response, which can then be sent together with Bs ACK.
Without delayed ACKs, the kernel layer on B may send its ACK before the receiving application on B has
even been scheduled to run. If response packets are small, that doubles the total traffic. The maximum ACK
delay is 500 ms, according to RFC 1122 and RFC 2581.
For bulk traffic, delayed ACKs simply mean that the ACK traffic volume is reduced. Because ACKs are
cumulative, one ACK from the receiver can in principle acknowledge multiple data packets from the sender.
Unfortunately, acknowledging too many data packets with one ACK can interfere with the self-clocking
aspect of sliding windows; the arrival of that ACK will then trigger a burst of additional data packets, which
would otherwise have been transmitted at regular intervals. Because of this, the RFCs above specify that an
ACK be sent, at a minimum, for every other data packet. For a discussion of how the sender should respond
to delayed ACKs, see 13.2.1 Per-ACK Responses.

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Bandwidth Conservation
Delayed ACKs and the Nagle algorithm both originated in a bygone era, when bandwidth was in much
shorter supply than it is today. In RFC 896, John Nagle writes (in 1984) In general, we have not been
able to afford the luxury of excess long-haul bandwidth that the ARPANET possesses, and our longhaul links are heavily loaded during peak periods. Transit times of several seconds are thus common
in our network. Today, it is unlikely that extra small packets would cause significant problems.

12.15 Nagle Algorithm

Like delayed ACKs, the Nagle algorithm (RFC 896) also attempts to improve the behavior of interactive
small-packet applications. It specifies that a TCP endpoint generating small data segments should queue
them until either it accumulates a full segments worth or receives an ACK for the previous batch of small
segments. If the full-segment threshold is not reached, this means that only one (consolidated) segment will
be sent per RTT.
As an example, suppose A wishes to send to B packets containing consecutive letters, starting with a.
The application on A generates these every 100 ms, but the RTT is 501 ms. At T=0, A transmits a. The
application on A continues to generate b, c, d, e and f at times 100 ms through 500 ms, but A
does not send them immediately. At T=501 ms, ACK(a) arrives; at this point A transmits its backlogged
bcdef. The ACK for this arrives at T=1002, by which point A has queued ghijk. The end result is that
A sends a fifth as many packets as it would without the Nagle algorithm. If these letters are generated by a
user typing them with telnet, and the ACKs also include the echoed responses, then if the user pauses the
echoed responses will very soon catch up.
The Nagle algorithm does not always interact well with delayed ACKs, or with user expectations. It can
usually be disabled on a per-connection basis. See also exercise 10.

12.16 TCP Flow Control

A TCP receiver may reduce the Window Size value of an open connection, thus informing the sender to
switch to a smaller window size. This has nothing to do with congestion management but is instead related
to flow control. This reduction appears in the ACKs sent back by the receiver. A given ACK is not supposed
to reduce the window size by more than the size of the data it is acknowledging; in other words, the upper
end of the window, lastACKed+winsize, is never supposed to get smaller. This means that no data sent by the
sender will ever be retroactively invalidated by becoming beyond the upper edge of the window. A window
might shrink from the byte range [20,000..28,000] to [22,000..28,000] but never to [20,000..26,000].
If a TCP receiver uses this technique to shrink the window size to 0, this means that the sender may not send
data. This would be done by the receiver to acknowledge that, yes, the data was received, but that more may
not yet be sent. This corresponds to the ACKWAIT suggested in 6.1.3 Flow Control. Eventually, when the
receiver is ready to receive data, it will send an ACK increasing the window size again.
If the TCP sender has its window size reduced to 0, and the ACK from the receiver increasing the window
is lost, then the connection would be deadlocked. TCP has a special feature specifically to avoid this: if
the window size is reduced to zero, the sender sends dataless packets to the receiver, at regular intervals.

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Each of these polling packets elicits the receivers current ACK; the end result is that the sender will
receive the eventual window-enlargement announcement reliably. These polling packets are regulated by
the so-called persist timer.

12.17 TCP Timeout and Retransmission

For TCP to work well for both intra-server-room and trans-global connections, the timeout value must
adapt. TCP manages this by maintaining a running estimate of the RTT, EstRTT. In the original version,
TCP then set TimeOut = 2EstRTT (in the literature, the TCP TimeOut value is often known as RTO,
for Retransmission TimeOut). EstRTT itself was a running average of periodically measured SampleRTT
values, according to
EstRTT = EstRTT + (1-)SampleRTT
for a fixed , 0<<1. Typical values of might be =1/2 or =7/8. For close to 1 this is very conservative
in that EstRTT is slow to change. For close to 0, EstRTT is very volatile.
There is a potential RTT measurement ambiguity: if a packet is sent twice, the ACK received could be in
response to the first transmission or the second. The Karn/Partridge algorithm resolves this: on packet loss
(and retransmission), the sender
Doubles Timeout
Stops recording SampleRTT
Uses the doubled Timeout as EstRTT when things resume
Setting TimeOut = 2EstRTT proved too short during congestion periods and too long other times. Jacobson
and Karels ([JK88]) introduced a way of calculating the TimeOut value based on the statistical variability of
EstRTT. After each SampleRTT value was collected, the sender would also update EstDeviation according
to
SampleDev = | SampleRTT EstRTT |
EstDeviation = EstDeviation + (1-)SampleDev
for a fixed , 0<<1. Timeout was then set to EstRTT + 4EstDeviation. EstDeviation is an estimate of the
so-called mean deviation; 4 mean deviations corresponds (for normally distributed data) to about 5 standard
deviations. If the SampleRTT values were normally distributed (which they are not), this would mean that
the chance that a non-lost packet would arrive outside the TimeOut period is vanishingly small.
Keeping track of when packets time out is usually handled by putting a record for each packet sent into
a timer list. Each record contains the packets timeout time, and the list is kept sorted by these times.
Periodically, eg every 100 ms, the list is inspected and all packets with expired timeout are then retransmitted.
When an ACK arrives, the corresponding packet timeout record is removed from the list. Note that this
approach means that a packets timeout processing may be slightly late.

12.18 KeepAlive
There is no reason that a TCP connection should not be idle for a long period of time; ssh/telnet connections,
for example, might go unused for days. However, there is the turned-off-at-night problem: a workstation
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might telnet into a server, and then be shut off (not shut down gracefully) at the end of the day. The
connection would now be half-open, but the server would not generate any traffic and so might never detect
this; the connection itself would continue to tie up resources.
KeepAlive in action
One evening long ago, when dialed up (yes, that long ago) into the Internet, my phone line disconnected
while I was typing an email message in an ssh window. I dutifully reconnected, expecting to find my
message in the file dead.letter, which is what would have happened had I been disconnected while
using the even-older tty dialup. Alas, nothing was there. I reconstructed my email as best I could and
logged off.
The next morning, there was my lost email in a file dead.letter, dated two hours after the initial crash!
What had happened, apparently, was that the original ssh connection on the server side just hung there,
half-open. Then, after two hours, KeepAlive kicked in, and aborted the connection. At that point ssh
sent my mail program the HangUp signal, and the mail program wrote out what it had in dead.letter.
To avoid this, TCP supports an optional KeepAlive mechanism: each side polls the other with a dataless
packet. The original RFC 1122 KeepAlive timeout was 2 hours, but this could be reduced to 15 minutes. If
a connection failed the KeepAlive test, it would be closed.
Supposedly, some TCP implementations are not exactly RFC 1122-compliant: either KeepAlives are enabled by default, or the KeepAlive interval is much smaller than called for in the specification.

12.19 TCP timers

To summarize, TCP maintains the following four kinds of timers. All of them can be maintained by a single
timer list, above.
TimeOut: a per-segment timer; TimeOut values vary widely
2MSL TIMEWAIT: a per-connection timer
Persist: the timer used to poll the receiving end when winsize = 0
KeepAlive, above

12.20 Epilog
At this point we have covered the basic mechanics of TCP, but have one important topic remaining: how
TCP manages its window size so as to limit congestion. That will be the focus of the next three chapters.

12.21 Exercises
1. Experiment with the TCP version of simplex-talk. How does the server respond differently with
threading enabled and without, if two simultaneous attempts to connect are made?

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2. Trace the states visited if nodes A and B attempt to create a TCP connection by simultaneously sending
each other SYN packets, that then cross in the network. Draw the ladder diagram.
3. When two nodes A and B simultaneously attempt to connect to one another using the OSI TP4 protocol,
two bidirectional network connections are created (rather than one, as with TCP). If TCP had instead chosen
the TP4 semantics here, what would have to be added to the TCP header? Hint: if a packet from A,port1
arrives at B,port2, how would we tell to which of the two possible connections it belongs?
4. Simultaneous connection initiations are rare, but simultaneous connection termination is relatively common. How do two TCP nodes negotiate the simultaneous sending of FIN packets to one another? Draw the
ladder diagram. Which node goes into TIMEWAIT state?

5. (a) Suppose you see multiple connections on your workstation in state FIN_WAIT_1. What is likely
going on? Whose fault is it?
(b). What might be going on if you see connections languishing in state FIN_WAIT_2?

6. Suppose that, after downloading a file, a user workstation is unplugged from the network. The workstation
may or may not have first sent a FIN to start closing the connection.

(a). Suppose the receiver has not sent the first FIN. According to the transitions shown in the TCP state
diagram, what TCP states could be reached by the server?
(b). Now suppose the receiver has sent its FIN before being unplugged. Again, what TCP states could be
reached by the server?

(Eventually, the server connection should transition to CLOSED due to repeated timeouts, but this is not
shown in the state diagram.)
7. Suppose A and B create a TCP connection with ISNA =20,000 and ISNB =5,000. A sends three 1000-byte
packets (Data1, Data2 and Data3 below), and B ACKs each. Then B sends a 1000-byte packet DataB to A
and terminates the connection with a FIN. In the table below, fill in the SEQ and ACK fields for each packet
shown.
A sends
SYN, ISNA =20,000

B sends
SYN, ISNB =5,000, ACK=______

ACK, SEQ=______, ACK=______

Data1, SEQ=______, ACK=______
ACK, SEQ=______, ACK=______
Data2, SEQ=______, ACK=______
ACK, SEQ=______, ACK=______
Data3, SEQ=______, ACK=______
ACK, SEQ=______, ACK=______
DataB, SEQ=______, ACK=______
ACK, SEQ=_____, ACK=______
FIN, SEQ=______, ACK=______
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8. Suppose you are watching a video on a site like YouTube, where there is a progress bar showing how
much of the video has been downloaded (and which hopefully stays comfortably head of the second progress
bar showing your position in viewing the video). Occasionally, the download-progress bar jumps ahead by
a modest but significant amount, instantaneously fast (much faster than the bandwidth alone could account
for). At the TCP layer, what is going on to cause this jump? Hint: what does a TCP receiver perceive when
a packet is lost and retransmitted?
9. Suppose you are creating software for a streaming-video site. You want to limit the video read-ahead
the gap between how much has been downloaded and how much the viewer has actually watched to 100
KB; the server should pause in sending when necessary to enforce this. An ordinary TCP connection will
simply transfer data as fast as possible. What support, if any, do you need the receiving (client) application
layer to provide, in order to enable this? What must the server-side application then do?
10. A user moves the computer mouse and sees the mouse-cursors position updated on the screen. Suppose
the mouse-position updates are being transmitted over a TCP connection with a relatively long RTT. The
user attempts to move the cursor to a specific point. How will the user perceive the mouses motion

(a). with the Nagle algorithm

(b). without the Nagle algorithm

11. Suppose you have fallen in with a group that wants to add to TCP a feature so that, if A and B1 are
connected, then B1 can hand off its connection to a different host B2; the end result is that A and B2 are
connected and A has received an uninterrupted stream of data. Either A or B1 can initiate the handoff.

(a). Suppose B1 is the host to send the final FIN (or HANDOFF) packet to A. How would you handle
appropriate analogues of the TIMEWAIT state for host B1? Does the fact that A is continuing the
connection, just not with B1, matter?
(b). Now suppose A is the party to send the final FIN/HANDOFF, to B1. What changes to TIMEWAIT
would have to be made at As end? Note that A may potentially hand off the connection again and again, eg
to B3, B4 and then B5.

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13 TCP RENO AND CONGESTION MANAGEMENT

This chapter addresses how TCP manages congestion, both for the connections own benefit (to improve its
throughput) and for the benefit of other connections as well (which may result in our connection reducing its
own throughput). Early work on congestion culminated in 1990 with the flavor of TCP known as TCP Reno.
The congestion-management mechanisms of TCP Reno remain the dominant approach on the Internet today,
though alternative TCPs are an active area of research and we will consider a few of them in 15 Newer
TCP Implementations.
The central TCP mechanism here is for a connection to adjust its window size. A smaller winsize means
fewer packets are out in the Internet at any one time, and less traffic means less congestion. All TCPs reduce
winsize when congestion is apparent, and increase it when it is not. The trick is in figuring out when and
by how much to make these winsize changes. Many of the improvements to TCP have come from mining
more and more information from the ACK stream.
Recall Chiu and Jains definition from 1.7 Congestion that the knee of congestion occurs when the queue
first starts to grow, and the cliff of congestion occurs when packets start being dropped. Congestion can
be managed at either point, though dropped packets can be a significant waste of resources. Some newer
TCP strategies attempt to take action at the congestion knee, but TCP Reno is a cliff-based strategy: packets
must be lost before the sender reduces the window size.
In 19 Quality of Service we will consider some router-centric alternatives to TCP for Internet congestion
management. However, for the most part these have not been widely adopted, and TCP is all that stands in
the way of Internet congestive collapse.
The first question one might ask about TCP congestion management is just how did it get this job? A TCP
sender is expected to monitor its transmission rate so as to cooperate with other senders to reduce overall
congestion among the routers. While part of the goal of every TCP node is good, stable performance for
its own connections, this emphasis on end-user cooperation introduces the prospect of cheating: a host
might be tempted to maximize the throughput of its own connections at the expense of others. Putting
TCP nodes in charge of congestion among the core routers is to some degree like putting the foxes in
charge of the henhouse. More accurately, such an arrangement has the potential to lead to the Tragedy of
the Commons. Multiple TCP senders share a common resource the Internet backbone and while the
backbone is most efficient if every sender cooperates, each individual sender can improve its own situation
by sending faster than allowed. Indeed, one of the arguments used by virtual-circuit routing adherents is that
it provides support for the implementation of a wide range of congestion-management options under control
of a central authority.
Nonetheless, TCP has been quite successful at distributed congestion management. In part this has been
because system vendors do have an incentive to take the big-picture view, and in the past it has been quite
difficult for individual users to replace their TCP stacks with rogue versions. Another factor contributing to
TCPs success here is that most bad TCP behavior requires cooperation at the server end, and most server
managers have an incentive to behave cooperatively. Servers generally want to distribute bandwidth fairly
among their multiple clients, and theoretically at least a servers ISP could penalize misbehavior. So far,
at least, the TCP approach has worked remarkably well.

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13.1 Basics of TCP Congestion Management

TCPs congestion management is window-based; that is, TCP adjusts its window size to adapt to congestion. The window size can be thought of as the number of packets out there in the network; more precisely,
it represents the number of packets and ACKs either in transit or enqueued. An alternative approach often
used for real-time systems is rate-based congestion management, which runs into an unfortunate difficulty
if the sending rate momentarily happens to exceed the available rate.
In the very earliest days of TCP, the window size for a TCP connection came from the AdvertisedWindow
value suggested by the receiver, essentially representing how many packet buffers it could allocate. This
value is often quite large, to accommodate large bandwidthdelay products, and so is often reduced out of
concern for congestion. When winsize is adjusted downwards for this reason, it is generally referred to as
the Congestion Window, or cwnd (a variable name first appearing in Berkeley Unix). Strictly speaking,
winsize = min(cwnd, AdvertisedWindow).
If TCP is sending over an idle network, the per-packet RTT will be RTTnoLoad , the travel time with no
queuing delays. As we saw in 6.3.2 RTT Calculations, (RTTRTTnoLoad ) is the time each packet spends
in the queue. The path bandwidth is winsize/RTT, and so the number of packets in queues is winsize
(RTTRTTnoLoad ) / RTT. Usually all the queued packets are at the router at the head of the bottleneck link.
Note that the sender can calculate this number (assuming we can estimate RTTnoLoad ; the most common
approach is to assume that the smallest RTT measured corresponds to RTTnoLoad ).
TCPs self-clocking (ie that new transmissions are paced by returning ACKs) guarantees that, again assuming an otherwise idle network, the queue will build only at the bottleneck router. Self-clocking means that
the rate of packet transmissions is equal to the available bandwidth of the bottleneck link. There are some
spikes when a burst of packets is sent (eg when the sender increases its window size), but in the steady state
self-clocking means that packets accumulate only at the bottleneck.
We will return to the case of the non-otherwise-idle network in the next chapter, in 14.2 Bottleneck Links
with Competition.
The optimum window size for a TCP connection would be bandwidth RTTnoLoad . With this window
size, the sender has exactly filled the transit capacity along the path to its destination, and has used none of
the queue capacity.
Actually, TCP Reno does not do this.
Instead, TCP Reno does the following:
guesses at a reasonable initial window size, using a form of polling
slowly increases the window size if no losses occur, on the theory that maximum available throughput
may not yet have been reached
rapidly decreases the window size otherwise, on the theory that if losses occur then drastic action is
needed
In practice, this usually leaves TCPs window size well above the theoretical optimum.
One interpretation of TCPs approach is that there is a time-varying ceiling on the number of packets the
network can accept. Each sender tries to stay near but just below this level. Occasionally a sender will
overshoot and a packet will be dropped somewhere, but this just teaches the sender a little more about where
the network ceiling is. More formally, this ceiling represents the largest cwnd that does not lead to packet

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loss, ie the cwnd that at that particular moment completely fills but does not overflow the bottleneck queue.
We have reached the ceiling when the queue is full.
In Chiu and Jains terminology, the far side of the ceiling is the cliff, at which point packets are lost. TCP
tries to stay above the knee, which is the point when the queue first begins to be persistently utilized, thus
keeping the queue at least partially occupied; whenever it sends too much and falls off the cliff, it retreats.
The ceiling concept is often useful, but not necessarily as precise as it might sound. If we have reached the
ceiling by gradually expanding the sliding-windows window size, then winsize will be as large as possible.
But if the sender suddenly releases a burst of packets, the queue may fill and we will have reached the ceiling
without fully utilizing the transit capacity. Another source of ceiling ambiguity is that the bottleneck link
may be shared with other connections, in which case the ceiling represents our connections particular share,
which may fluctuate greatly with time. Finally, at the point when the ceiling is reached, the queue is full
and so there are a considerable number of packets waiting in the queue; it is not possible for a sender to pull
back instantaneously.
It is time to acknowledge the existence of different versions of TCP, each incorporating different congestionmanagement algorithms. The two we will start with are TCP Tahoe (1988) and TCP Reno (1990); the
names Tahoe and Reno were originally the codenames of the Berkeley Unix distributions that included
these respective TCP implementations. The ideas behind TCP Tahoe came from a 1988 paper by Jacobson
and Karels [JK88]; TCP Reno then refined this a couple years later. TCP Reno is still in widespread use
over twenty years later, and is still the undisputed TCP reference implementation, although some modest
improvements (NewReno, SACK) have crept in.
A common theme to the development of improved implementations of TCP is for one end of the connection
(usually the sender) to extract greater and greater amounts of information from the packet flow. For example,
TCP Tahoe introduced the idea that duplicate ACKs likely mean a lost packet; TCP Reno introduced the
idea that returning duplicate ACKs are associated with packets that have successfully been transmitted but
follow a loss. TCP Vegas (15.4 TCP Vegas) introduced the fine-grained measurement of RTT, to detect
when RTT > RTTnoLoad .
It is often helpful to think of a TCP sender as having breaks between successive windowfuls; that is, the
sender sends cwnd packets, is briefly idle, and then sends another cwnd packets. The successive windowfuls of packets are often called flights. The existence of any separation between flights is, however, not
guaranteed.

13.1.1 The Steady State

We will begin with the state in which TCP has established a reasonable guess for cwnd, comfortably below the Advertised Window Size, and which largely appears to be working. TCP then engages in some
fine-tuning. This TCP steady state is usually referred to as the congestion avoidance phase, though all
phases of the process are ultimately directed towards avoidance of congestion. The central strategy (which
we expand on below) is that when a packet is lost, cwnd should decrease rapidly, but otherwise should
increase slowly. As TCP finishes each windowful of packets, it notes whether a loss occurred. The
cwnd-adjustment rule introduced by TCP Tahoe and [JK88] is the following:
if there were no losses in the previous windowful, cwnd = cwnd +1
if packets were lost, cwnd = cwnd/2

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We are informally measuring cwnd in units of full packets; strictly speaking, cwnd is measured in bytes
and is incremented by the maximum TCP segment size.
This strategy here is known as Additive Increase, Multiplicative Decrease, or AIMD; cwnd = cwnd+1 is
the additive increase and cwnd = cwnd/2 is the multiplicative decrease. Typically, setting cwnd=cwnd/2
is a medium-term goal; in fact, TCP Tahoe briefly sets cwnd=1 in the immediate aftermath of an actual
timeout. With no losses, TCP will send successive windowfuls of, say, 20, 21, 22, 23, 24, .... This amounts
to conservative probing of the network and, in particular, of the queue at the bottleneck router. TCP tries
larger cwnd values because the absence of loss means the current cwnd is below the network ceiling; that
is, the queue at the bottleneck router is not yet overfull.
If a loss occurs (including multiple losses in a single windowful), TCPs response is to cut the window size
in half. (As we will see, TCP Tahoe actually handles this in a somewhat roundabout way.) Informally, the
idea is that the sender needs to respond aggressively to congestion. More precisely, lost packets mean the
queue of the bottleneck router has filled, and the sender needs to dial back to a level that will allow the queue
to clear. If we assume that the transit capacity is roughly equal to the queue capacity (say each is equal to
N), then we overflow the queue and drop packets when cwnd = 2N, and so cwnd = cwnd/2 leaves us with
cwnd = N, which just fills the transit capacity and leaves the queue empty. (When the sender sets cwnd=N,
the actual number of packets in transit takes at least one RTT to fall from 2N to N.)
Of course, assuming any relationship between transit capacity and queue capacity is highly speculative. On
a 5,000 km fiber-optic link with a bandwidth of 1 Gbps, the round-trip transit capacity would be about 6
MB. That is much larger than the likely size of any router queue. A more contemporary model of a typical
long-haul high-bandwidth TCP path might be that the queue size is a small fraction of the bandwidthdelay
product; we return to this in 13.7 TCP and Bottleneck Link Utilization.
Note that if TCP experiences a packet loss, and there is an actual timeout, then the sliding-window pipe
has drained. No packets are in flight. No self-clocking can govern new transmissions. Sliding windows
therefore needs to restart from scratch.
The congestion-avoidance algorithm leads to the classic TCP sawtooth graph, where the peaks are at
the points where the slowly rising cwnd crossed above the network ceiling. We emphasize that the
TCP sawtooth is specific to TCP Reno and related TCP implementations that share Renos additiveincrease/multiplicative-decrease mechanism.

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During periods of no loss, TCPs cwnd increases linearly; when a loss occurs, TCP sets cwnd = cwnd/2.
This diagram is an idealization as when a loss occurs it takes the sender some time to discover it, perhaps as
much as the TimeOut interval.
The fluctuation shown here in the red ceiling curve is somewhat arbitrary. If there are only one or two
other competing senders, the ceiling variation may be quite dramatic, but with many concurrent senders the
variations may be smoothed out.
For some TCP sawtooth graphs created through actual simulation, see 16.2.1 Graph of cwnd v time and
16.4.1 Some TCP Reno cwnd graphs.
13.1.1.1 A first look at fairness
The transit capacity of the path is more-or-less unvarying, as is the physical capacity of the queue at the
bottleneck router. However, these capacities are also shared with other connections, which may come and
go with time. This is why the ceiling does vary in real terms. If two other connections share a path with
total capacity 60 packets, the fairest allocation might be for each connection to get about 20 packets as its
share. If one of those other connections terminates, the two remaining ones might each rise to 30 packets.
And if instead a fourth connection joins the mix, then after equilibrium is reached each connection might
hope for a fair share of 15 packets.
Will this kind of fair allocation actually happen? Or might we end up with one connection getting 90% of
the bandwidth while two others each get 5%?
Chiu and Jain [CJ89] showed that the additive-increase/multiplicative-decrease algorithm does indeed converge to roughly equal bandwidth sharing when two connections have a common bottleneck link, provided
also that
both connections have the same RTT
during any given RTT, either both connections experience a packet loss, or neither connection does

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To see this, let cwnd1 and cwnd2 be the connections congestion-window sizes, and consider the quantity
cwnd1 cwnd2. For any RTT in which there is no loss, cwnd1 and cwnd2 both increment by 1, and so
cwnd1 cwnd2 stays the same. If there is a loss, then both are cut in half and so cwnd1 cwnd2 is also
cut in half. Thus, over time, the original value of cwnd1 cwnd2 is repeatedly cut in half (during each
RTT in which losses occur) until it dwindles to inconsequentiality, at which point cwnd1 cwnd2.
Graphical and tabular versions of this same argument are in the next chapter, in 14.3 TCP Fairness with
Synchronized Losses.
The second bulleted hypothesis above we may call the synchronized-loss hypothesis. While it is very
reasonable to suppose that the two connections will experience the same number of losses as a long-term
average, it is a much stronger statement to suppose that all loss events are shared by both connections. This
behavior may not occur in real life and has been the subject of some debate; see [GV02]. We return to this
point in 16.3 Two TCP Senders Competing. Fortunately, equal-RTT fairness still holds if each connection
is equally likely to experience a packet loss: both connections will have the same loss rate, and so, as we
shall see in 14.5 TCP Reno loss rate versus cwnd, will have the same cwnd. However, convergence to
fairness may take rather much longer. In 14.3 TCP Fairness with Synchronized Losses we also look at some
alternative hypotheses for the unequal-RTT case.

13.2 Slow Start

How do we make that initial guess as to the network capacity? What value of cwnd should we begin with?
And even if we have a good target for cwnd, how do we avoid flooding the network sending an initial burst
of packets?
The answer is known as slow start. If you are trying to guess a number in a fixed range, you are likely to
use binary search. Not knowing the range for the network ceiling, a good strategy is to guess cwnd=1 (or
cwnd=2) at first and keep doubling until you have gone too far. Then revert to the previous guess, which is
known to have worked. At this point you are guaranteed to be within 50% of the true capacity.
The actual slow-start mechanism is to increment cwnd by 1 for each ACK received. This seems linear,
but that is misleading: after we send a windowful of packets (cwnd many), we have received cwnd ACKs
and so have incremented cwnd-many times, and so have set cwnd to (cwnd+cwnd) = 2cwnd. In other
words, cwnd=cwnd2 after each windowful is the same as cwnd+=1 after each packet.
Assuming packets travel together in windowfuls, all this means cwnd doubles each RTT during slow start;
this is possibly the only place in the computer science literature where exponential growth is described as
slow. It is indeed slower, however, than the alternative of sending an entire windowful at once.
Here is a diagram of slow start in action. This diagram makes the implicit assumption that the no-load RTT
is large enough to hold well more than the 8 packets of the maximum window size shown.

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For a different case, with a much smaller RTT, see 13.2.3 Slow-Start Multiple Drop Example.
Eventually the bottleneck queue gets full, and drops a packet. Let us suppose this is after N RTTs, so
cwnd=2N . Then during the previous RTT, cwnd=2N-1 worked successfully, so we go back to that previous
value by setting cwnd = cwnd/2.

13.2.1 Per-ACK Responses

During slow start, incrementing cwnd by one per ACK received is equivalent to doubling cwnd after each
windowful. We can find a similar equivalence for the congestion-avoidance phase, above.
During congestion avoidance, cwnd is incremented by 1 after each windowful. To formulate this as a perACK increase, we spread this increment of 1 over the entire windowful, which of course has size cwnd.
This amounts to the following upon each ACK received:
cwnd = cwnd + 1/cwnd
This is a slight approximation, because cwnd keeps changing, but it works well in practice. Because TCP
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actually measures cwnd in bytes, floating-point arithmetic is normally not required; see exercise 13. An
exact equivalent to the per-windowful incrementing strategy is cwnd = cwnd + 1/cwnd0 , where cwnd0 is
the value of cwnd at the start of that particular windowful. Another, simpler, approach is to use cwnd +=
1/cwnd, and to keep the fractional part recorded, but to use floor(cwnd) (the integer part of cwnd) when
actually sending packets.
Most actual implementations keep track of cwnd in bytes, in which case using integer arithmetic is sufficient
until cwnd becomes quite large.
If delayed ACKs are implemented (12.14 TCP Delayed ACKs), then in bulk transfers one arriving ACK
actually acknowledges two packets. RFC 3465 permits a TCP receiver to increment cwnd by 2/cwnd in
that situation, which is the response consistent with incrementing cwnd by 1 for each windowful.

13.2.2 Threshold Slow Start

Sometimes TCP uses slow start even when it knows the working network capacity. After a packet loss and
timeout, TCP knows that a new cwnd of cwndold /2 should work. If cwnd had been 100, TCP halves it to
50. The problem, however, is that after timeout there are no returning ACKs to self-clock the continuing
transmission, and we do not want to dump 50 packets on the network all at once. So in restarting the flow
TCP uses what might be called threshold slow start: it uses slow-start, but stops when cwnd reaches the
target. Specifically, on packet loss we set the variable ssthresh to cwnd/2; this is our new target for
cwnd. We set cwnd itself to 1, and switch to the slow-start mode (cwnd += 1 for each ACK). However,
as soon as cwnd reaches ssthresh, we switch to the congestion-avoidance mode (cwnd += 1/cwnd for
each ACK). Note that the transition from threshold slow start to congestion avoidance is completely natural,
and easy to implement.
TCP will use threshold slow-start whenever it is restarting from a pipe drain; that is, every time slow-start
is needed after its very first use. (If a connection has simply been idle, non-threshold slow start is typically
used when traffic starts up again.)
Threshold slow-start can be seen as an attempt at combining rapid window expansion with self-clocking.
By comparison, we might refer to the initial, non-threshold slow start as unbounded slow start. Note that
unbounded slow start serves a fundamentally different purpose initial probing to determine the network
ceiling to within 50% than threshold slow start.
Here is the TCP sawtooth diagram above, modified to show timeouts and slow start. The first two packet
losses are displayed as coarse timeouts; the rest are displayed as if Fast Retransmit, below, were used.

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RFC 2581 allows slow start to begin with cwnd=2.

13.2.3 Slow-Start Multiple Drop Example

Slow start has the potential to cause multiple dropped packets at the bottleneck link; packet losses continue
for quite some time because the TCP sender is slow to discover them. The network topology is as follows,
where the AR link is infinitely fast and the RB link has a bandwidth in the RB direction of 1 packet/ms.

Assume that R has a queue capacity of 100, not including the packet it is currently forwarding to B,
and that ACKs travel instantly from B back to A. In this and later examples we will continue to use the
Data[N]/ACK[N] terminology of 6.2 Sliding Windows, beginning with N=1; TCP numbering is not done
quite this way but the distinction is inconsequential.
When A uses slow-start here, the successive windowfuls will almost immediately begin to overlap. A will
send one packet at T=0; it will be delivered at T=1. The ACK will travel instantly to A, at which point A
will send two packets. From this point on, ACKs will arrive regularly at A at a rate of one per second. Here
is a brief chart:

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Time
0
1
2
3
4
5
..
N

A receives

R sends
Data[1]
Data[2]
3
4
5
6

Rs queue

ACK[1]
ACK[2]
ACK[3]
ACK[4]
ACK[5]

A sends
Data[1]
Data[2],Data[3]
4,5
6,7
8,9
10,11

ACK[N]

2N,2N+1

N+1

N+2 .. 2N+1

Data[3]
4,5
5..7
6..9
7..11

At T=N, Rs queue contains N packets. At T=100, Rs queue is full. Data[200], sent at T=100, will be
delivered and acknowledged at T=200, giving it an RTT of 100. At T=101, R receives Data[202] and
Data[203] and drops the latter one. Unfortunately, As timeout interval must of course be greater than the
RTT, and so A will not detect the loss until, at an absolute minimum, T=200. At that point, A has sent
packets up through Data[401], and the 100 packets Data[203], Data[205], ..., Data[401] have all been lost.
In other words, at the point when A first receives the news of one lost packet, in fact at least 100 packets
have already been lost.
Fortunately, unbounded slow start generally occurs only once per connection.

13.2.4 Summary of TCP so far

So far we have the following features:
Unbounded slow start at the beginning
Congestion avoidance with AIMD once some semblance of a steady state is reached
Threshold slow start after each loss
Each threshold slow start transitioning naturally to congestion avoidance
Here is a table expressing the slow-start and congestion-avoidance phases in terms of manipulating cwnd.
phase
slow start
cong avoid

cwnd change, loss

per window
cwnd/2
cwnd/2

cwnd change, no loss

per window per ACK
cwnd *= 2 cwnd += 1
cwnd +=1
cwnd += 1/cwnd

The problem TCP often faces, in both slow-start and congestion-avoidance phases, is that when a packet is
lost the sender will not detect this until much later (at least until the bottleneck routers current queue has
been sent); by then, it may be too late to avoid further losses.

13.3 TCP Tahoe and Fast Retransmit

TCP Tahoe has one more important feature. Recall that TCP ACKs are cumulative; if packets 1 and 2 have
been received and now Data[4] arrives, but not yet Data[3], all the receiver can (and must!) do is to send
back another ACK[2]. Thus, from the senders perspective, if we send packets 1,2,3,4,5,6 and get back
ACK[1], ACK[2], ACK[2], ACK[2], ACK[2], we can infer two things:

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Data[3] got lost, which is why we are stuck on ACK[2]

Data 4,5 and 6 probably did make it through, and triggered the three duplicate ACK[2]s (the three
ACK[2]s following the first ACK[2]).
The Fast Retransmit strategy is to resend Data[N] when we have received three dupACKs for Data[N-1];
that is, four ACK[N-1]s in all. Because this represents a packet loss, we also set ssthresh = cwnd/2,
set cwnd=1, and begin the threshold-slow-start phase. The effect of this is typically to reduce the delay
associated with the lost packet from that of a full timeout, typically 2RTT, to just a little over a single RTT.
The lost packet is now discovered before the TCP pipeline has drained. However, at the end of the next RTT,
when the ACK of the retransmitted packet will return, the TCP pipeline will have drained, hence the need
for slow start.
TCP Tahoe included all the features discussed so far: the cwnd+=1 and cwnd=cwnd/2 responses, slow
start and Fast Retransmit.
Fast Retransmit waits for the third dupACK to allow for the possibility of moderate packet reordering.
Suppose packets 1 through 6 are sent, but they arrive in the order 1,3,4,2,6,5, perhaps due to a router along
the way with an architecture that is strongly parallelized. Then the ACKs that would be sent back would be
as follows:
Received
Data[1]
Data[3]
Data[4]
Data[2]
Data[6]
Data[5]

Response
ACK[1]
ACK[1]
ACK[1]
ACK[4]
ACK[4]
ACK[6]

Waiting for the third dupACK is in most cases a successful compromise between responsiveness to lost
packets and reasonable evidence that the data packet in question is actually lost.
However, a router that does more substantial delivery reordering would wreck havoc on connections using
Fast Retransmit. In particular, consider the router R in the diagram below; when sending packets to B it
might in principle wish to alternate on a packet-by-packet basis between the path via R1 and the path via R2.
This would be a mistake; if the R1 and R2 paths had different propagation delays then this strategy would
introduce major packet reordering. R should send all the packets belonging to any one TCP connection via
a single path.

In the real world, routers generally go to considerable lengths to accommodate Fast Retransmit; in particular,
use of multiple paths for a single TCP connection is almost universally frowned upon. Some actual data on
packet reordering can be found in [VP97]; the author suggests that a switch to retransmission on the second
dupACK would be risky.

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13.4 TCP Reno and Fast Recovery

Fast Retransmit requires a sender to set cwnd=1 because the pipe has drained and there are no arriving
ACKs to pace transmission. Fast Recovery is a technique that often allows the sender to avoid draining the
pipe, and to move from cwnd to cwnd/2 in the space of a single RTT. TCP Reno is TCP Tahoe with the
addition of Fast Recovery.
The idea is to use the arriving dupACKs to pace retransmission. We set cwnd=cwnd/2, and then to figure
out how many dupACKs we have to wait for. Initially, at least, we will assume that only one packet is lost.
Let cwnd = N, and suppose we have sent packets 0 through N and packet 1 is lost (we send Data[N] only
after ACK[0] has arrived at the sender). We will then get N-1 dupACK[0]s representing packets 2 through
N.
During the recovery process, we will ignore cwnd and instead use the concept of Estimated FlightSize, or
EFS, which is the senders best guess at the number of outstanding packets. Under normal circumstances,
EFS is the same as cwnd, at least between packet departures and arrivals.
At the point of the third dupACK, the sender calculates as follows: EFS had been cwnd = N. However,
one of the packets has been lost, making it N1. Three dupACKs have arrived, representing three later
packets no longer in flight, so EFS is now N4. Fast Retransmit had the sender retransmit the packet that
was inferred as lost, so EFS increments by 1, to N3. The sender expects at this point to receive N4 more
dupACKs, plus one new ACK for the retransmission of the packet that was lost. This last ACK will be for
the entire original windowful.
The new target for cwnd is N/2. So, we wait for N/2 3 more dupACKs to arrive, at which point EFS
is N3(N/23) = N/2. After this point the sender will resume sending new packets; it will send one new
packet for each of the N/2 subsequently arriving dupACKs.
After the last of the dupACKs will come the ACK corresponding to the retransmission of the lost packet; it
will actually be a cumulative ACK for all the later received packets as well. At this point the sender declares
cwnd = N/2, and resumes with sliding windows. As EFS was already N/2, and there are no lost packets
outstanding, the sender has exactly one full windowful outstanding for the new value of cwnd. That is, we
are right where we are supposed to be.
Here is a detailed diagram illustrating Fast Recovery. We have cwnd=10 initially, and Data[10] is lost.

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Data[9] elicits the initial ACK[9], and the nine packets Data[11] through Data[19] each elicit a dupACK[9].
We denote the dupACK[9] elicited by Data[N] by dupACK[9]/N; these are shown along the upper right.
Unless SACK TCP (below) is used, the sender will have no way to determine N or to tell these dupACKs
apart. When dupACK[9]/13 (the third dupACK) arrives at the sender, the sender uses Fast Recovery to infer
that Data[10] was lost and retransmits it. At this point EFS = 7: the sender has sent the original batch of 10
data packets, plus Data[19], and received one ACK and three dupACKs, for a total of 10+1-1-3 = 7. The
sender has also inferred that Data[10] is lost (EFS = 1) but then retransmitted it (EFS += 1). Six more
dupACK[9]s are on the way.
EFS is decremented for each subsequent dupACK arrival; after we get two more dupACK[9]s, EFS is 5.
The next dupACK[9] (dupACK[9]/16) reduces EFS to 4 and so allows us transmit Data[20] (which promptly
bumps EFS back up to 5). We have
receive
dupACK[9]/16
dupACK[9]/17
dupACK[9]/18
dupACK[9]/19

send
Data[20]
Data[21]
Data[22]
Data[23]

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We emphasize again that the TCP sender does not see the numbers 16 through 19 in the receive column
above; it determines when to begin transmission by counting dupACK[9] arrivals.
Working Backwards
Figuring out when a fast-recovery sender should resume transmissions of new data is error-prone.
Perhaps the simplest approach is to work backwards from the retransmitted lost packet: it should
trigger at the receiver an ACK for the entire original windowful. When Data[10] above was lost, the
stuck window was Data[10]-Data[19]. The retransmitted Data[10] thus triggers ACK[19]; when
ACK[19] arrives, cwnd should be 10/2 = 5 so Data[24] should be sent. That in turn means the four
packets Data[20] through Data[23] must have been sent earlier, via Fast Recovery. There are 101 = 9
dupACKs, so to send on the last four we must start with the sixth. The diagram above indeed shows
new Fast Recovery transmissions beginning with the sixth dupACK.
The next thing to arrive at the sender side is the ACK[19] elicited by the retransmitted Data[10]; at the point
Data[10] arrives at the receiver, Data[11] through Data[19] have already arrived and so the cumulative-ACK
response is ACK[19]. The sender responds to ACK[19] with Data[24], and the transition to cwnd=5 is now
complete.
During sliding windows without losses, a sender will send cwnd packets per RTT. If a coarse timeout
occurs, typically it is not discovered until after at least one complete RTT of link idleness; there are additional
underutilized RTTs during the slow-start phase. It is worth examining the Fast Recovery sequence shown
in the illustration from the perspective of underutilized bandwidth. The diagram shows three round-trip
times, as seen from the sender side. During the first RTT, the ten packets Data[9]-Data[18] are sent. The
second RTT begins with the sending of Data[19] and continues through sending Data[22], along with the
retransmitted Data[10]. The third RTT begins with sending Data[23], and includes through Data[27]. In
terms of recovery efficiency, the RTTs send 9, 5 and 5 packets respectively (we have counted Data[10]
twice); this is remarkably close to the ideal of reducing cwnd to 5 instantaneously.
The reason we cannot use cwnd directly in the formulation of Fast Recovery is that, until the lost Data[10]
is acknowledged, the window is frozen at Data[10]-Data[19]. The original RFC 2001 description of Fast
Recovery described retransmission in terms of cwnd inflation and deflation. Inflation would begin at the
point the sender resumed transmitting new packets, at which point cwnd would be incremented for each
dupACK; in the diagram above, cwnd would finish at 15. When the ACK[20] elicited by the lost packet
finally arrived, the recovery phase would end and cwnd would immediately deflate to 5. For a diagram
illustrating cwnd inflation and deflation, see 17.2.1 Running the Script.

13.5 TCP NewReno

TCP NewReno, described in [JH96] and RFC 2582, introduced a modest tweak to Fast Recovery which
greatly improves handling of the case when two or more packets are lost in a windowful. If two data packets
are lost and the first is retransmitted, the receiver will acknowledge data up to just before the second packet,
and then continue sending dupACKs of this until the second lost packet is also retransmitted. These ACKs
of data up to just before the second packet are sometimes called partial ACKs, because retransmission of
the first lost packet did not result in an ACK of all the outstanding data. NewReno uses these partial ACKs
as evidence to retransmit later lost packets, and also to keep pacing the Fast Recovery process.

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In the diagram below, packets 1 and 4 are lost in a window 0..11 of size 12. Initially the sender will get
dupACK[0]s; the first 11 ACKs (dashed lines from right to left) are ACK[0] and 10 dupACK[0]s. When
packet 1 is successfully retransmitted on receipt of the third dupACK[0], the receivers response will be
ACK[3] (the heavy dashed line). This is the first partial ACK (a full ACK would have been ACK[12]). On
receipt of any partial ACK during the Fast Recovery process, TCP NewReno assumes that the immediately
following data packet was lost and retransmits it immediately; the sender does not wait for three dupACKs
because if the following data packet had not been lost, no instances of the partial ACK would ever have been
generated, even if packet reordering had occurred.
The TCP NewReno sender response here is, in effect, to treat each partial ACK as a dupACK[0], except that
the sender also retransmits the data packet that based upon receipt of the partial ACK it is able to infer
is lost. NewReno continues pacing Fast Recovery by whatever ACKs arrive, whether they are the original
dupACKs or later partial ACKs or dupACKs.

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When the receivers first ACK[3] arrives at the sender, NewReno infers that Data[4] was lost and resends it;
this is the second heavy data line. No dupACK[3]s need arrive; as mentioned above, the sender can infer
from the single ACK[3] that Data[4] is lost. The sender also responds as if another dupACK[0] had arrived,
and sends Data[17].
The arrival of ACK[3] signals a reduction in the EFS by 2: one for the inference that Data[4] was lost, and
one as if another dupACK[0] had arrived; the two transmissions in response (of Data[4] and Data[17]) bring
EFS back to where it was. At the point when Data[16] is sent the actual (not estimated) flightsize is 5, not 6,

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because there is one less dupACK[0] due to the loss of Data[4]. However, once NewReno resends Data[4]
and then sends Data[17], the actual flightsize is back up to 6.
There are four more dupACK[3]s that arrive. NewReno keeps sending new data on receipt of each of these;
these are Data[18] through Data[21].
The receivers response to the retransmitted Data[4] is to send ACK[16]; this is the cumulative of all the
data received up to that moment. At the point this ACK arrives back at the sender, it had just sent Data[21]
in response to the fourth dupACK[3]; its response to ACK[16] is to send the next data packet, Data[22].
The sender is now back to normal sliding windows, with a cwnd of 6. Similarly, the Data[17] immediately
following the retransmitted Data[4] elicits an ACK[17] (this is the first Data[N] to elicit an exactly matching ACK[N] since the losses began), and the corresponding response to the ACK[17] is to continue with
Data[23].
As with the previous Fast Recovery example, we consider the number of packets sent per RTT; the diagram
shows four RTTs as seen from the sender side.
RTT
first
second
third
fourth

First packet
Data[0]
Data[12]
Data[16]
Data[21]

Packets sent
Data[0]-Data[11]
Data[12]-Data[15], Data[1]
Data[16]-Data[20], Data[4]
Data[21]-Data[26]

count
12
5
6
6

Again, after the loss is detected we segue to the new cwnd of 6 with only a single missed packet (in the
second RTT). NewReno is, however, only able to send one retransmitted packet per RTT.
Note that TCP Newreno, like TCPs Tahoe and Reno, is a sender-side innovation; the receiver does not have
to do anything special. The next TCP flavor, SACK TCP, requires receiver-side modification.

13.6 SACK TCP

SACK TCP is TCP with Selective ACKs, so the sender does not just guess from dupACKs what has gotten
through. The receiver can send an ACK that says:
All packets up through 1000 have been received
All packets up through 1050 have been received except for 1001, 1022, and 1035.
Note that SACK TCP involves protocol modification at both ends of the connection. Most TCP implementations now support SACK TCP.
Specifically, SACK TCP includes the following information in its acknowledgments; the additional data in
the ACKs is included as a TCP Option.
The latest cumulative ACK
The three most recent blocks of consecutive packets received.
Thus, if we have lost 1001, 1022, 1035, and now 1051, and the highest received is 1060, the ACK might
say:
All packets up through 1000 have been received
1060-1052 have been received
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1050-1036 have been received

1034-1023 have been received
If the sender has been paying close attention to the previous SACKs received, it likely already knows that
1002 through 1021 have been received.
In practice, selective ACKs provide at best a modest performance improvement in most situations; TCP
NewReno does rather well, in moderate-loss environments. The paper [FF96] compares Tahoe, Reno,
NewReno and SACK TCP, in situations involving from one to four packet losses in a single RTT. While
Classic Reno performed poorly with two packet losses in a single RTT and extremely poorly with three
losses, the three-loss NewReno and SACK TCP scenarios played out remarkably similarly. Only when connections experienced four losses in a single RTT did SACK TCPs performance start to pull slightly ahead
of that of NewReno.

13.7 TCP and Bottleneck Link Utilization

Consider a TCP Reno sender with no competing traffic. As cwnd saws up and down, what happens to
throughput? Do those halvings of cwnd result in at least a dip in throughput? The answer depends to some
extent on the size of the queue ahead of the bottleneck link, relative to the transit capacity of the path. As
was discussed in 6.3.2 RTT Calculations, when cwnd is less than the transit capacity, the link is less than
100% utilized and the queue is empty. When cwnd is more than the transit capacity, the link is saturated
(100% utilized) and the queue has about (cwnd transit_capacity) packets in it.
The diagram below shows two TCP Reno teeth; in the first, the queue capacity exceeds the path transit
capacity and in the second the queue capacity is a much smaller fraction of the total.

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In the first diagram, the bottleneck link is always 100% utilized, even at the left edge of the teeth. In
the second the interval between loss events (the left and right margins of the tooth) is divided into a linkunsaturated phase and a queue-filling phase. In the unsaturated phase, the bottleneck link utilization is
less than 100% and the queue is empty; in the later phase, the link is saturated and the queue begins to fill.
Consider again the idealized network below, with an RB bandwidth of 1 packet/ms.

We first consider the queue transit case. Assume that the total RTTnoLoad delay is 20 ms, mostly due to
propagation delay; this makes the bandwidthdelay product 20 packets. The question for consideration is
to what extent TCP Reno, once slow-start is over, sometimes leaves the RB link idle.
The RB link will be saturated at all times provided A always keeps 20 packets in transit, that is, we always
have cwnd 20 (6.3.2 RTT Calculations). If cwndmin = 20, then cwndmax = 2cwndmin = 40. For
this to be the maximum, the queue capacity must be at least 19, so that the path can accommodate 39
packets without loss: 20 packets in transit + 19 packets in the queue. In general, TCP Reno never leaves the
bottleneck link idle as long as the queue capacity in front of that link is at least as large as the path round-trip
transit capacity.
Now suppose instead that the queue size is 9. Packet loss will occur when cwnd reaches 30, and so
cwndmin = 15. Qualitatively this case is represented by the second diagram above, though the queue-tonetwork_ceiling proportion illustrated there is more like 1:8 than 1:3. There are now periods when the RB
link is idle. During RTT intervals when cwnd=15, throughput will be 75% of the maximum and the RB
link will be idle 25% of the time.
However, cwnd will be 15 just for the first RTT following the loss. After 5 RTTs, cwnd will be back to 20,
and the link will be saturated. So we have 5 RTTs with an average cwnd of 17.5, where the link is 17.5/20
= 87.5% saturated, followed by 10 RTTs where the link is 100% saturated. The long-term average here is
95.8% utilization of the bottleneck link. This is not bad at all, given that using 10% of the link bandwidth
on packet headers is almost universally considered reasonable. Furthermore, at the point when cwnd drops
after a loss to cwndmin =15, the queue must have been full. It may take one or two RTTs for the queue to
drain; during this time, link utilization will be even higher.
If most or all of the time the bottleneck link is saturated, as in the first diagram, it may help to consider
the average queue size. Let the queue capacity be Cqueue and the transit capacity be Ctransit , with Cqueue >
Ctransit . Then cwnd will vary from a maximum of Cqueue +Ctransit to a minimum of what works out to be
(Cqueue -Ctransit )/2 + Ctransit . We would expect an average queue size about halfway between these, less the
Ctransit term: 3/4Cqueue - 1/4Ctransit . If Cqueue =Ctransit , the expected average queue size should be about
Cqueue /2.
The second case, with queue capacity less than transit capacity, is arguably the more common situation,
and becoming more so as time goes on and bandwidths increase. This is almost always the case that
applies to high-bandwidthdelay connections, where the queue size is typically much smaller than the
bandwidthdelay product. Cisco routers, for example, generally are configured to use queue sizes considerably less than 100, regardless of bandwidths involved, while the bandwidthdelay product for a 50ms RTT
with 1 Gbps bandwidth is ~6MB, or 4000 packets of 1500 bytes each. In this case the tooth is divided into
a large link-unsaturated phase and a small queue-filling phase.

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The worst case for TCP link utilization is if the queue size is close to zero. Using again a bandwidthdelay
product 20 of packets, we can see that cwndmax will be 20 (or 21), and so cwndmin will be 10. Link
utilization therefore ranges from a low of 10/20 = 50% to a high 100%, over 10 RTTs; the average utilization
is 75%. While this is not ideal, and while some TCP variants have attempted to improve this figure, 75%
link utilization is not all that bad, and can again be compared with the 10% of the bandwidth consumed as
packet headers. (Note that a literally zero-sized queue may not work at all well with slow start, because the
sender will regularly send packets two at a time.)
We will return to this point in 16.2.6 Single-sender Throughput Experiments and (for two senders)
16.3.10.2 Higher bandwidth and link utilization, using the ns simulator to get experimental data. See
also exercise 12.

13.8 Single Packet Losses

Again assuming no competition on the bottleneck link, the TCP Reno additive-increase policy has a simple
consequence: at the end of each tooth, only a single packet will be lost.
To see this, let A be the sender, R be the bottleneck router, and B be the receiver:

Let T be the bandwidth delay at R, so that packets leaving R are spaced at least time T apart. A will
therefore transmit packets T time units apart, except for those times when cwnd has just been incremented
and A sends a pair of packets back-to-back. Let us call the second packet of such a back-to-back pair the
extra packet. To simplify the argument slightly, we will assume that the two packets of a pair arrive at R
essentially simultaneously.
Only an extra packet can result in an increase in queue utilization; every other packet arrives after an interval
T from the previous packet, giving R enough time to remove a packet from its queue.
A consequence of this is that cwnd will reach the sum of the transit capacity and the queue capacity without
R dropping a packet. (This is not necessarily the case if a cwnd this large were sent as a single burst.)
Let C be this combined capacity, and assume cwnd has reached C. When A executes its next cwnd += 1
additive increase, it will as usual send a pair of back-to-back packets. The second of this pair the extra
is doomed; it will be dropped when it reaches the bottleneck router.
At this point there are C = cwnd 1 packets outstanding, all spaced at time intervals of T. Sliding windows
will continue normally until the ACK of the packet just before the lost packet arrives back at A. After this
point, A will receive only dupACKs. A has received C = cwnd1 ACKs since the last increment to cwnd,
but must receive C+1 = cwnd ACKs in order to increment cwnd again. This will not happen, as no more
new ACKs will arrive until the lost packet is transmitted.
Following this, cwnd is reduced and the next sawtooth begins; the only packet that is lost is the extra
packet of the previous flight.
See 16.2.3 Single Losses for experimental confirmation, and exercise 14.

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13.9 TCP Assumptions and Scalability

In the TCP design portrayed above, several embedded assumptions have been made. Perhaps the most important is that every loss is treated as evidence of congestion. As we shall see in the next chapter, this fails
for high-bandwidth TCP (when rare random losses become significant); it also fails for TCP over wireless
(either Wi-Fi or other), where lost packets are much more common than over Ethernet. See 14.9 The
High-Bandwidth TCP Problem and 14.10 The Lossy-Link TCP Problem.
The TCP cwnd-increment strategy to increment cwnd by 1 for each RTT has some assumptions of
scale. This mechanism works well for cross-continent RTTs on the order of 100 ms, and for cwnd in the
low hundreds. But if cwnd = 2000, then it takes 100 RTTs perhaps 20 seconds for cwnd to grow 10%;
linear increase becomes proportionally quite slow. Also, if the RTT is very long, the cwnd increase is slow.
The absolute set-by-the-speed-of-light minimum RTT for satellite Internet is 480 ms, and typical satelliteInternet RTTs are close to 1000 ms. Such long RTTs also lead to slow cwnd growth; furthermore, as we
shall see below, such long RTTs mean that these TCP connections compete poorly with other connections.
See 14.11 The Satellite-Link TCP Problem.
Another implicit assumption is that if we have a lot of data to transfer, we will send all of it in one single
connection rather than divide it among multiple connections. The web http protocol violates this routinely,
though. With multiple short connections, cwnd may never properly converge to the steady state for any of
them; TCP Reno does not support carrying over what has been learned about cwnd from one connection
to the next. A related issue occurs when a connection alternates between relatively idle periods and full-on
data transfer; most TCPs set cwnd=1 and return to slow start when sending resumes after an idle period.
Finally, TCPs Fast Retransmit assumes that routers do not significantly reorder packets.

13.10 TCP Parameters

In TCP Renos Additive Increase, Multiplicative Decrease strategy, the increase increment is 1.0 and the
decrease factor is 1/2. It is natural to ask if these values have some especial significance, or what are the
consequences if they are changed.
Neither of these values plays much of a role in determining the average value of cwnd, at least in the short
term; this is largely dictated by the path capacity, including the queue size of the bottleneck router. It seems
clear that the exact value of the increase increment has no bearing on congestion; the per-RTT increase is
too small to have a major effect here. The decrease factor of 1/2 may play a role in responding promptly to
incipient congestion, in that it reduces cwnd sharply at the first sign of lost packets. However, as we shall see
in 15.4 TCP Vegas, TCP Vegas in its normal mode manages quite successfully with an Additive Decrease
strategy, decrementing cwnd by 1 at the point it detects approaching congestion (to be sure, it detects this
well before packet loss), and, by some measures, responds better to congestion than TCP Reno. In other
words, not only is the exact value of the AIMD decrease factor not critical for congestion management, but
multiplicative decrease itself is not mandatory.
There are two informal justifications in [JK88] for a decrease factor of 1/2. The first is in slow start: if at
the Nth RTT it is found that cwnd = 2N is too big, the sender falls back to cwnd/2 = 2N-1 , which is known
to have worked without losses the previous RTT. However, a change here in the decrease policy might best
be addressed with a concomitant change to slow start; alternatively, the reduction factor of 1/2 might be left
still to apply to unbounded slow start, while a new factor of might apply to threshold slow start.

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The second justification for the reduction factor of 1/2 applies directly to the congestion avoidance phase;
written in 1988, it is quite remarkable to the modern reader:
If the connection is steady-state running and a packet is dropped, its probably because a new
connection started up and took some of your bandwidth.... [I]ts probable that there are now
exactly two conversations sharing the bandwidth. I.e., you should reduce your window by half
because the bandwidth available to you has been reduced by half. [JK88], D
Today, busy routers may have thousands of simultaneous connections. To be sure, Jacobson and Karels
go on to state, if there are more than two connections sharing the bandwidth, halving your window is
conservative and being conservative at high traffic intensities is probably wise. This advice remains apt
today.
But while they do not play a large role in setting cwnd or in avoiding congestive collapse, it turns out that
these increase-increment and decrease-factor values of 1 and 1/2 respectively play a great role in fairness:
making sure competing connections get the bandwidth allocation they should get. We will return to this
in 14.3 TCP Fairness with Synchronized Losses, and also 14.7 AIMD Revisited.

13.11 Epilog
TCP Renos core congestion algorithm is based on algorithms in Jacobson and Karels 1988 paper [JK88],
now twenty-five years old, although NewReno and SACK have been almost universally added to the standard
Reno implementation.
There are also broad changes in TCP usage patterns. Twenty years ago, the vast majority of all TCP traffic
represented downloads from major servers. Today, over half of all Internet TCP traffic is peer-to-peer
rather than server-to-client. The rise in online video streaming creates new demands for excellent TCP
real-time performance.
In the next chapter we will examine the dynamic behavior of TCP Reno, focusing in particular on fairness
between competing connections, and on other problems faced by TCP Reno senders. Then, in 15 Newer
TCP Implementations, we will survey some attempts to address these problems.

13.12 Exercises
1. Consider the following network, with each link other than the first having a bandwidth delay of 1
packet/second. Assume ACKs travel instantly from B to R (and thus to A). Assume there are no propagation delays, so the RTTnoLoad is 4; the bandwidthRTT product is then 4 packets. If A uses sliding
windows with a window size of 6, the queue at R1 will eventually have size 2.

Suppose A uses threshold slow start with ssthresh = 6, and with cwnd initially 1. Complete the table
below until two rows after cwnd = 6; for these final two rows, A will send only one new packet for each
ACK received. How big will the queue at R1 grow?
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T
0
1
2
3
4
5
6
7
8

A sends
1

R1 queues

R1 sends
1

B receives/ACKs

cwnd
1

2,3

2
3

2
2

4,5

Note that if, instead of using slow start, A simply sends the initial windowful of 6 packets all at once, then
the queue at R1 will initially hold 6-1 = 5 packets.
2. Consider the following network from 13.2.3 Slow-Start Multiple Drop Example, with links labeled with
bandwidths in packets/ms. Assume ACKs travel instantly from B to R (and thus to A).

A begins sending to B using slow start, beginning with Data[1] at T=0. Write out all packet deliveries
assuming Rs queue size is 5, up until the first dupACK triggered by the arrival at B of a packet that followed
a packet that was lost.
3. Repeat the previous problem, except assume Rs queue size is 2. Assume no retransmission mechanism
is used at all (no timeouts, no fast retransmit), and that A sends new data only when it receives new ACKs
(dupACKs, in other words, do not trigger new data transmissions). With these assumptions, new data
transmissions will eventually cease; continue the table until all transmitted data packets are received by B.
4. Suppose a connection starts with cwnd=1 and increments cwnd by 1 each RTT with no loss, and sets
cwnd to cwnd/2, rounding down, on each RTT with at least one loss. Lost packets are not retransmitted,
and propagation delays dominate so each windowful is sent more or less together. Packets 5, 13, 14, 23 and
30 are lost. What is the window size each RTT, up until the first 40 packets are sent? What packets are sent
each RTT? Hint: in the first RTT, Data[1] is sent. There is no loss, so in the second RTT cwnd = 2 and
Data[2] and Data[3] are sent.
5. Suppose TCP Reno is used to transfer a large file over a path with a bandwidth of one packet per 10 sec
and an RTT of 80 ms. Assume the receiver places no limits on window size. Note the bandwidthdelay
product is 8,000 packets.

(a). How many RTTs will it take for the window size to first reach ~8,000 packets, assuming unbounded
slow start is used and there are no packet losses?
(b). Approximately how many packets will have been sent and acknowledged by that point?
(c). What fraction of the total bandwidth will have been used up to that point? Hint: the total bandwidth is
8,000 packets per RTT.

6. (a) Repeat the diagram in 13.4 TCP Reno and Fast Recovery, done there with cwnd=10, for a window
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size of 8. Assume as before that the lost packet is Data[10]. There will be seven dupACK[9]s, which it
may be convenient to tag as dupACK[9]/11 through dupACK[9]/17. Be sure to indicate clearly when
sending resumes.
(b). Suppose you try to do this with a window size of 6. Is this window size big enough for Fast Recovery
still to work? If so, at what dupACK[9]/N does new data transmission begin? If not, what goes wrong?

7. Suppose the window size is 100, and Data[1001] is lost. There will be 99 dupACK[1000]s sent, which
we may denote as dupACK[1000]/1002 through dupACK[1000]/1100. TCP Reno is used.

(a). At which dupACK[1000]/N does the sender start sending new data?
(b). When the retransmitted data[1001] arrives at the receiver, what ACK is sent in response?
(c). When the acknowledgment in (b) arrives back at the sender, what data packet is sent?
Hint: express EFS in terms of dupACK[1000]/N, for N>1004.

8. Suppose the window size is 40, and Data[1001] is lost. Packet 1000 will be ACKed normally. Packets
1001-1040 will be sent, and 1002-1040 will each trigger a duplicate ACK[1000].

(a). What actual data packets trigger the first three dupACKs? (The first ACK[1000] is triggered by
Data[1000]; dont count this one as a duplicate.)
(b). After the third dupACK[1000] has been received and the lost data[1001] has been retransmitted, how
many packets/ACKs should the sender estimate as in flight?

When the retransmitted Data[1001] arrives at the receiver, ACK[1040] will be sent back.

(c). What is the first Data[N] sent for which the response is ACK[N], for N>1000?
(d). What is the first N for which Data[N+20] is sent in response to ACK[N] (this represents the point when
the connection is back to normal sliding windows, with a window size of 20)?

9. Suppose slow-start is modified so that, on each arriving ACK, three new packets are sent rather than two;
cwnd will now triple after each RTT.

(a). For each arriving ACK, by how much must cwnd now be incremented?
(b). Suppose a path has mostly propagation delay. Progressively larger windowfuls are sent, until a cwnd
is reached where a packet loss occurs. What window size can the sender be reasonably sure does work,
based on earlier experience?

10. Suppose in the example of 13.5 TCP NewReno, Data[4] had not been lost.

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(a). When Data[1] is received, what ACK would be sent in response?

(b). At what point in the diagram is the sender able to resume ordinary sliding windows with cwnd = 6?

11. Suppose in the example of 13.5 TCP NewReno, Data[1] and Data[2] had been lost, but not Data[4].

(a). The third dupACK[0] is sent in response to what Data[N]?

(b). When the retransmitted Data[1] reaches the receiver, ACK[1] is the response. When this ACK[1]
reaches the sender, which Data packets are sent in response?

12. Suppose two TCP connections have the same RTT and share a bottleneck link, on which there is no other
traffic. The size of the bottleneck queue is negligible when compared to the bandwidth RTTnoLoad product.
Loss events occur at regular intervals, and are completely synchronized. Show that the two connections
together will use 75% of the total bottleneck-link capacity, as in 13.7 TCP and Bottleneck Link Utilization
(there done for a single connection).
See also Exercise 18 of the next chapter.
13. In 13.2.1 Per-ACK Responses we stated that the per-ACK response of a TCP sender was to increment
cwnd as follows:
cwnd = cwnd + 1/cwnd

(a). What is the corresponding formulation if the window size is in fact measured in bytes rather than
packets? Let SMSS denote the senders maximum segment size, and let bwnd = SMSScwnd denote the
congestion window as measured in bytes.
(b). What is the appropriate formulation if delayed ACKs are used (12.14 TCP Delayed ACKs) and we
still want cwnd to be incremented by 1 for each windowful?

14. In 13.8 Single Packet Losses we simplified the argument slightly by assuming that when A sent a pair
of packets, they arrived at R essentially simultaneously.
Give a scenario in which it is not the extra packet (the second of the pair) that is lost, but the packet that
follows it. Hint: see 16.3.4.1 Single-sender phase effects.

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14 DYNAMICS OF TCP RENO

In this chapter we introduce, first and foremost, the possibility that there are other TCP connections out
there competing with us for throughput. In 6.3 Linear Bottlenecks (and in 13.7 TCP and Bottleneck Link
Utilization) we looked at the performance of TCP through an uncontested bottleneck; now we allow for
competition.
We also look more carefully at the long-term behavior of TCP Reno (and Reno-like) connections, as the
value of cwnd increases and decreases according to the TCP sawtooth. In particular we analyze the average cwnd; recall that the average cwnd divided by the RTT is the connections average throughput (we
momentarily ignore here the fact that RTT is not constant, but the error this introduces is usually small).
A few of the ideas presented here apply as well to non-Reno connections as well. Some non-Reno TCP
alternatives are presented in the following chapter; the section on TCP Friendliness below addresses how to
extend TCP Renos competitive behavior even to UDP.
We also consider some router-based mechanisms such as RED and ECN that take advantage of TCP Renos
behavior to provide better overall performance.
The chapter closes with a summary of the central real-world performance problems faced by TCP today;
this then serves to introduce the proposed TCP fixes in the following chapter.

14.1 A First Look At Queuing

In what order do we transmit the packets in a routers outbound-interface queue? The conventional answer
is in the order of arrival; technically, this is FIFO (First-In, First-Out) queuing. What happens to a packet
that arrives at a router whose queue for the desired outbound interface is full? The conventional answer is
that it is dropped; technically, this is known as tail-drop.
While FIFO tail-drop remains very important, there are alternatives. In an admittedly entirely different
context (the IPv6 equivalent of ARP), RFC 4681 states, When a queue overflows, the new arrival SHOULD
replace the oldest entry. This might be called head drop; it is not used for router queues.
An alternative drop-policy mechanism that has been considered for router queues is random drop. Under
this policy, if a packet arrives but the destination queue is full, with N packets waiting, then one of the N+1
packets in all the N waiting plus the new arrival is chosen at random for dropping. The most recent
arrival has thus a very good chance of gaining an initial place in the queue, but also a reasonable chance of
being dropped later on.
While random drop is seldom if ever put to production use its original form, it does resolve a peculiar
synchronization problem related to TCPs natural periodicity that can lead to starvation for one connection.
This situation known as phase effects will be revisited in 16.3.4 Phase Effects. Mathematically,
random-drop queuing is sometimes more tractable than tail-drop because a packets loss probability has
little dependence on arrival-time race conditions with other packets.

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14.1.1 Priority Queuing

A quite different alternative to FIFO is priority queuing. We will consider this in more detail in 18.3 Priority Queuing, but the basic idea is straightforward: whenever the router is ready to send the next packet, it
looks first to see if it has any higher-priority packets to send; lower-priority packets are sent only when there
is no waiting higher-priority traffic. This can, of course, lead to complete starvation for the lower-priority
traffic, but often there are bandwidth constraints on the higher-priority traffic (eg that it amounts to less than
10% of the total available bandwidth) such that starvation does not occur.
In an environment of mixed real-time and bulk traffic, it is natural to use priority queuing to give the realtime traffic priority service, by assignment of such traffic to the higher-priority queue. This works quite well
as long as, say, the real-time traffic is less than some fixed fraction of the total; we will return to this in
19 Quality of Service.

14.2 Bottleneck Links with Competition

So far we have been ignoring the fact that there are other TCP connections out there. A single connection
in isolation needs not to overrun its bottleneck router and drop packets, at least not too often. However,
once there are other connections present, then each individual TCP connection also needs to consider how
to maximize its share of the aggregate bandwidth.
Consider a simple network path, with bandwidths shown in packets/ms. The minimum bandwidth, or path
bandwidth, is 3 packets/ms.

14.2.1 Example 1: linear bottleneck

Below is the example we considered in 6.3 Linear Bottlenecks; bandwidths are shown in packets/ms.

The bottleneck link for AB traffic is at R2, and the queue will form at R2s outbound interface.
We claimed earlier that if the sender uses sliding windows with a fixed window size, then the network will
converge to a steady state in relatively short order. This is also true if multiple senders are involved; however,
a mathematical proof of convergence may be more difficult.

14.2.2 Example 2: router competition

The bottleneck-link concept is a useful one for understanding congestion due to a single connection. However, if there are multiple senders in competition for a link, the situation is more complicated. Consider the
following diagram, in which links are labeled with bandwidths in packets/ms:

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For a moment, assume R3 uses priority queuing, with the BC path given priority over AC. If Bs
flow to C is fixed at 3 packets/ms, then As share of the R3C link will be 1 packet/ms, and As bottleneck
will be at R3. However, if Bs total flow rate drops to 1 packet/ms, then the R3C link will have 3 packets/ms
available, and the bottleneck for the AC path will become the 2 packet/ms R1R3 link.
Now let us switch to the more-realistic FIFO queuing at R3. If Bs flow is 3 packets/ms and As is 1
packet/ms, then the R3C link will be saturated, but just barely: if each connection sticks to these rates, no
queue will develop at R3. However, it is no longer accurate to describe the 1 packet/ms as As share: if A
wishes to send more, it will begin to compete with B. At first, the queue at R3 will grow; eventually, it is
quite possible that Bs total flow rate might drop because B is losing to A in the competition for R3s queue.
This latter effect is very real.
In general, if two connections share a bottleneck link, they are competing for the bandwidth of that link.
That bandwidth share, however, is precisely dictated by the queue share as of a short while before. R3s
fixed rate of 4 packets/ms means one packet every 250 s. If R3 has a queue of 100 packets, and in that
queue there are 37 packets from A and 63 packets from B, then over the next 25 ms (= 100 250 s) R3s
traffic to C will consist of those 37 packets from A and the 63 from B. Thus the competition between A and
B for R3C bandwidth is first fought as a competition for R3s queue space. This is important enough to
state as as rule:
Queue-Competition Rule: in the steady state, if a connection utilizes fraction 1 of a FIFO
routers queue, then that connection has a share of of the routers total outbound bandwidth.
Below is a picture of R3s queue and outbound link; the queue contains four packets from A and eight from
B. The link, too, contains packets in this same ratio; presumably packets from B are consistently arriving
twice as fast as packets from A.

In the steady state here, A and B will use four and eight packets, respectively, of R3s queue capacity. As
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acknowledgments return, each sender will replenish the queue accordingly. However, it is not in As longterm interest to settle for a queue utilization at R3 of four packets; A may want to take steps that will lead in
this setting to a gradual increase of its queue share.
Although we started the discussion above with fixed packet-sending rates for A and B, in general this leads
to instability. If A and Bs combined rates add up to more than 4 packets/ms, R3s queue will grow without
bound. It is much better to have A and B use sliding windows, and give them each fixed window sizes; in
this case, as we shall see, a stable equilibrium is soon reached. Any combination of window sizes is legal
regardless of the available bandwidth; the queue utilization (and, if necessary, the loss rate) will vary as
necessary to adapt to the actual bandwidth.
If there are several competing flows, then a given connection may have multiple bottlenecks, in the sense
that there are several routers on the path experiencing queue buildups. In the steady state, however, we can
still identify the link (or first link) with minimum bandwidth; we can call this link the bottleneck. Note that
the bottleneck link in this sense can change with the senders winsize and with competing traffic.

14.2.3 Example 3: competition and queue utilization

In the next diagram, the bottleneck RC link has a normalized bandwidth of 1 packet per ms (or, more
abstractly, one packet per unit time). The bandwidths of the AR and BR links do not matter, except they
are greater than 1 packet per ms. Each link is labeled with the propagation delay, measured in the same
time unit as the bandwidth; the delay thus represents the number of packets the link can be transporting at
the same time, if sent at the bottleneck rate.

The network layout here, with the shared RC link as the bottleneck, is sometimes known as the singlebell
topology. A perhaps-more-common alternative is the dumbbell topology of 14.3 TCP Fairness with
Synchronized Losses, though the two are equivalent for our purposes.
Suppose A and B each send to C using sliding windows, each with fixed values of winsize wA and wB .
Suppose further that these winsize values are large enough to saturate the RC link. How big will the queue
be at R? And how will the bandwidth divide between the AC and BC flows?
For the two-competing-connections example above, assume we have reached the steady state. Let denote
the fraction of the bandwidth that the AC connection receives, and let = 1- denote the fraction that
the BC connection gets; because of our normalization choice for the RC bandwidth, and are the
respective throughputs. From the Queue-Competition Rule above, these bandwidth proportions must agree
with the queue proportions; if Q denotes the combined queue utilization of both connections, then that queue
will have about Q packets from the AC flow and about Q packets from the BC flow.
We worked out the queue usage precisely in 6.3.2 RTT Calculations for a single flow; we derived there the
following:

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queue_usage = winsize throughput RTTnoLoad

where we have here used throughput instead of bandwidth to emphasize that this is the dynamic share
rather than the physical transmission capacity.
This equation remains true for each separate flow in the present case, where the RTTnoLoad for the AC
connection is 2(dA +d) (the factor of 2 is to account for the round-trip) and the RTTnoLoad for the BC
connection is 2(dB +d). We thus have
Q = wA 2(dA +d)
Q = wB 2(dB +d)
or, alternatively,
[Q + 2d + 2dA ] = wA
[Q + 2d + 2dB ] = wB
If we add the first pair of equations above, we can obtain the combined queue utilization:
Q = wA + wB 2d 2(dA +dB )
The last term here, 2(dA +dB ), represents the number of As packets in flight on the AR link plus the
number of Bs packets in flight on the BR link.
We can solve these equations exactly for , and Q in terms of the known quantities, but the algebraic
solution is not particularly illuminating. Instead, we examine a few more-tractable special cases.
14.2.3.1 The equal-delays case
We consider first the special case of equal delays: dA = dB = d. In this case the term (dA +dB ) simplifies
to d, and thus we have Q = wA + wB 2d 2d. Furthermore, if we divide corresponding sides of the
second pair of equations above, we get / = wA /wB ; that is, the bandwidth (and thus the queue utilization)
divides in exact accordance to the window-size proportions.
If, however, dA is larger than dB , then a greater fraction of the AC packets will be in transit, and so fewer
will be in the queue at R, and so will be somewhat smaller and somewhat larger.
14.2.3.2 The equal-windows case
If we assume equal winsize values instead, wA = wB = w, then we get
/ = [Q + 2d + 2dB ] / [Q + 2d + 2dA ]
The bandwidth ratio here is biased against the larger of dA or dB . That is, if dA > dB , then more of As
packets will be in transit, and thus fewer will be in Rs queue, and so A will have a smaller fraction of the
the bandwidth. This bias is, however, not quite proportional: if we assume dA is double dB and dB = d =
Q/2, then / = 3/4, and A gets 3/7 of the bandwidth to Bs 4/7.
Still assuming wA = wB = w, let us decrease w to the point where the link is just saturated, but Q=0. At this
point / = [d+dB ]/[d +dA ]; that is, bandwidth divides according to the respective RTTnoLoad values. As w
rises, additional queue capacity is used and / will move closer to 1.

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14.2.3.3 The fixed-wB case

Finally, let us consider what happens if wB is fixed at a large-enough value to create a queue at R from the
BC traffic alone, while wA then increases from zero to a point much larger than wB . Denote the number
of Bs packets in Rs queue by QB ; with wA = 0 we have =1 and Q = QB = wB 2(dB +d) = throughput
(RTT RTTnoLoad ).
As wA begins to increase from zero, the competition will decrease Bs throughput. We have =
wA /[Q+2d+2dA ]; small changes in wA will not lead to much change in Q, and even less in Q+2d+2dA ,
and so will initially be approximately proportional to wA .
For Bs part, increased competition from A (increased wA ) will always decrease Bs share of the bottleneck
RC link; this link is saturated and every packet of As in transit there must take away one slot on that link
for a packet of Bs. This in turn means that Bs bandwidth must decrease as wA rises. As Bs bandwidth
decreases, QB = Q = wB 2(dB +d) must increase; another way to put this is as the transit capacity falls,
the queue utilization rises. For QB = Q to increase while decreases, Q must be increasing faster than
is decreasing.
Finally, we can conclude that as wA gets large and 0, the limiting value for Bs queue utilization QB at R
will be the entire windowful wB , up from its starting value (when wA =0) of wB 2(dB +d). If dB +d had been
small relative to wB , then QB s increase will be modest, and it may be appropriate to consider QB relatively
constant.
14.2.3.4 The iterative solution
Given d, dA , dB , wA and wB , one way to solve for , and Q is to proceed iteratively. Suppose an initial
, is given, as the respective fractions of packets in the queue at R. Over the next period of time,
and must (by the Queue Rule) become the bandwidth ratios. If the AC connection has bandwidth
(recall that the RC connection has bandwidth 1.0, in packets per unit time, so a bandwidth fraction of
means an actual bandwidth of ), then the number of packets in bidirectional transit will be 2(dA +d),
and so the number of AC packets in Rs queue will be QA = wA 2(dA +d); similarly for QB . At that
point we will have new = QA /(QA +QB ). Starting with an appropriate guess for and iterating new a
few times, if the sequence converges then it will converge to the steady-state solution. Convergence is not
guaranteed, however, and is dependent on the initial guess for . One guess that often leads to convergence
is wA /(wA +wB ).

14.2.4 Example 4: cross traffic and RTT variation

In the following diagram, let us consider what happens to the AB traffic when the CD link ramps up.
Bandwidths shown are expressed as packets/ms and all queues are FIFO. We will assume that propagation
delays are small enough that only an inconsequential number of packets from C to D can be simultaneously
in transit at the bottleneck rate of 5 packets/ms. All senders will use sliding windows.

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Let us suppose the AB link is idle, and the CD connection begins sending with a window size chosen
so as to create a queue of 30 of Cs packets at R1 (if propagation delays are such that two packets can be in
transit each direction, we would achieve this with winsize=34).
Now imagine A begins sending. If A sends a single packet, is not shut out even though the R1R2 link is
100% busy. As packet will simply have to wait at R1 behind the 30 packets from C; the waiting time in the
queue will be 30 packets (5 packets/ms) = 6 ms. If we change the winsize of the CD connection, the
delay for As packets will be directly proportional to the number of Cs packets in R1s queue.
To most intents and purposes, the CD flow here has increased the RTT of the AB flow by 6 ms. As
long as As contribution to R1s queue is small relative to Cs, the delay at R1 for As packets looks more
like propagation delay than bandwidth delay, because if A sends two back-to-back packets they will likely
be enqueued consecutively at R1 and thus be subject to a single 6 ms queuing delay. By varying the CD
window size, we can, within limits, increase or decrease the RTT for the AB flow.
Let us return to the fixed CD window size denoted wC chosen to yield a queue of 30 of Cs packets
at R1. As A increases its own window size from, say, 1 to 5, the CD throughput will decrease slightly,
but Cs contribution to R1s queue will remain dominant.
As in the argument at the end of 14.2.3.3 The fixed-wB case, small propagation delays mean that wC will
not be much larger than 30. As wA climbs from zero to infinity, Cs contribution to R1s queue rises from 30
to at most wC , and so the 6ms delay for AB packets remains relatively constant even as As winsize rises
to the point that As contribution to R1s queue far outweighed Cs. (As we will argue in the next paragraphs,
this can actually happen only if the R2R3 bandwidth is increased). Each packet from A arriving at R1 will,
on average, face 30 or so of Cs packets ahead of it, along with anywhere from many fewer to many more
of As packets.
If As window size is 1, its one packet at a time will wait 6 ms in the queue at R1. If As window size
is greater than 1 but remains small, so that A contributes only a small proportion of R1s queue, then As
packets will wait only at R1. Initially, as As winsize increases, the queue at R1 grows but all other queues
remain empty.
However, if As winsize grows large enough that its packets consume 40% of R1s queue in the steady state,
then this situation changes. At the point when A has 40% of R1s queue, by the Queue Competition Rule it
will also have a 40% share of the R1R2 links bandwidth, that is, 40% 5 = 2 packets/ms. Because the
R2R3 link has a bandwidth of 2 packets/ms, the AB throughput can never grow beyond this. If the CD
contribution to R1s queue is held constant at 30 packets, then this point is reached when As contribution to
R1s queue is 20 packets.
Because As proportional contribution to R1s queue cannot increase further, any additional increase to As
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winsize must result in those packets now being enqueued at R2.

We have now reached a situation where As packets are queuing up at both R1 and at R2, contrary to the
single-sender principle that packets can queue at only one router. Note, however, that for any fixed value of
As winsize, a small-enough increase in As winsize will result in either that increase going entirely to R1s
queue or entirely to R2s queue. Specifically, if wA represents As winsize at the point when A has 40%
of R1s queue (a little above 20 packets if propagation delays are small), then for winsize < wA any queue
growth will be at R1 while for winsize > wA any queue growth will be at R2. In a sense the bottleneck link
switches from R1R2 to R2R3 at the point winsize = wA .
In the graph below, As contribution to R1s queue is plotted in green and As contribution to R2s queue is
in blue. It may be instructive to compare this graph with the third graph in 6.3.3 Graphs at the Congestion
Knee, which illustrates a single connection with a single bottleneck.

In Exercise 5 we consider some minor changes needed if propagation delay is not inconsequential.

14.2.5 Example 5: dynamic bottlenecks

The next example has two links offering potential competition to the AB flow: CD and EF.
Either of these could send traffic so as to throttle (or at least compete with) the AB traffic. Either of
these could choose a window size so as to build up a persistent queue at R1 or R3; a persistent queue of 20
packets would mean that AB traffic would wait 4 ms in the queue.

Despite situations like this, we will usually use the term bottleneck link as if it were a precisely defined
concept. In Examples 2, 3 and 4 above, a better term might be competitive link; for Example 5 we perhaps
should say competitive links.

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14.2.6 Packet Pairs

One approach for a sender to attempt to measure the physical bandwidth of the bottleneck link is the packetpairs technique: the sender repeatedly sends a pair of packets P1 and P2 to the receiver, one right after the
other. The receiver records the time difference between the arrivals.
Sooner or later, we would expect that P1 and P2 would arrive consecutively at the bottleneck router R, and
be put into the queue next to each other. They would then be sent one right after the other on the bottleneck
link; if T is the time difference in arrival at the far end of the link, the physical bandwidth is size(P1)/T. At
least some of the time, the packets will remain spaced by time T for the rest of their journey.
The theory is that the receiver can measure the different arrival-time differences for the different packet
pairs, and look for the minimum time difference. Often, this will be the time difference introduced by the
bandwidth delay on the bottleneck link, as in the previous paragraph, and so the ultimate receiver will be
able to infer that the bottleneck physical bandwidth is size(P1)/T.
Two things can mar this analysis. First, packets may be reordered; P2 might arrive before P1. Second, P1
and P2 can arrive together at the bottleneck router and be sent consecutively, but then, later in the network,
the two packets can arrive at a second router R2 with a (transient) queue large enough that P2 arrives while
P1 is in R2s queue. If P1 and P2 are consecutive in R2s queue, then the ultimate arrival-time difference is
likely to reflect R2s (higher) outbound bandwidth rather than Rs.
Additional analysis of the problems with the packet-pair technique can be found in [VP97], along with a
proposal for an improved technique known as packet bunch mode.

14.3 TCP Fairness with Synchronized Losses

This brings us to the question of just what is a fair division of bandwidth. A starting place is to assume
that fair means equal, though, as we shall see below, the question does not end there.
For the moment, consider again two competing TCP connections: Connection 1 (in blue) from A to C and
Connection 2 (in green) from B to D, through the same bottleneck router R, and with the same RTT. The
router R will use tail-drop queuing.

The layout illustrated here, with the shared link somewhere in the middle of each path, is sometimes known
as the dumbbell topology.
For the time being, we will also continue to assume the synchronized-loss hypothesis: that in any one
RTT either both connections experience a loss or neither does. (This assumption is suspect; we explore
it further in 14.3.3 TCP RTT bias and in 16.3 Two TCP Senders Competing). This was the model
reviewed previously in 13.1.1.1 A first look at fairness; we argued there that in any RTT without a loss, the
expression (cwnd1 - cwnd2 ) remained the same (both cwnds incremented by 1), while in any RTT with a
loss the expression (cwnd1 - cwnd2 ) decreased by a factor of 2 (both cwnds decreased by factors of 2).
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Here is a graphical version of the same argument, as originally introduced in [CJ89]. We plot cwnd1 on the
x-axis and cwnd2 on the y-axis. An additive increase of both (in equal amounts) moves the point (x,y) =
(cwnd1 ,cwnd2 ) along the line parallel to the 45 line y=x; equal multiplicative decreases of both moves the
point (x,y) along a line straight back towards the origin. If the maximum network capacity is Max, then a
loss occurs whenever x+y exceeds Max, that is, the point (x,y) crosses the line x+y=Max.

Beginning at the initial state, additive increase moves the state at a 45 angle up to the line x+y=Max, in
small increments denoted by the small arrowheads. At this point a loss would occur, and the state jumps
back halfway towards the origin. The state then moves at 45 incrementally back to the line x+y=Max, and
continues to zigzag slowly towards the equal-shares line y=x.
Any attempt to increase cwnd faster than linear will mean that the increase phase is not parallel to the line
y=x, but in fact veers away from it. This will slow down the process of convergence to equal shares.
Finally, here is a timeline version of the argument. We will assume that the AC path capacity, the BD path
capacity and Rs queue size all add up to 24 packets, and that in any RTT in which cwnd1 + cwnd2 > 24,
both connections experience a packet loss. We also assume that, initially, the first connection has cwnd=20,
and the second has cwnd=1.
T
0
1
2

298

AC
20
21
22

BD
1
2
3

total cwnd is 25; packet loss

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T
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
...
32
33

Table 1 continued from previous page

AC BD
11
1
12
2
13
3
14
4
15
5
16
6
17
7
18
8
second packet loss
9
4
10
5
11
6
12
7
13
8
14
9
15
10
third packet loss
7
5
8
6
9
7
10
8
11
9
12
10
13
11
14
12
fourth loss
7
6
cwnds are quite close
13
6

12
6

loss
cwnds are equal

So far, fairness seems to be winning.

14.3.1 Example 2: Faster additive increase

Here is the same kind of timeline again with the synchronized-loss hypothesis but with the additiveincrease increment changed from 1 to 2 for the BD connection (but not for AC); both connections start
with cwnd=1. Again, we assume a loss occurs when cwnd1 + cwnd2 > 24

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T
0
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15

AC
1
2
3
4
5
6
7
8
9
4
5
6
7
8
9
4

BD
1
3
5
7
9
11
13
15
17
8
10
12
14
16
18
9

first packet loss

second loss
essentially where we were at T=9

The effect here is that the second connections average cwnd, and thus its throughput, is double that of
the first connection. Thus, changes to the additive-increase increment lead to very significant changes in
fairness.

14.3.2 Example 3: Longer RTT

For the next example, we will return to standard TCP Reno, with an increase increment of 1. But here
we assume that the RTT of the AC connection is double that of the BD connection, perhaps because of
additional delay in the AR link. The longer RTT means that the first connection sends packet flights only
when T is even. Here is the timeline, where we allow the first connection a hefty head-start. As before, we
assume a loss occurs when cwnd1 + cwnd2 > 24.
T
0
1
2
3
4
5
6
7
8
9
10
11
12
13
14

AC
20
21
22
11
12
13
14
15

BD
1
2
3
4
5
2
3
4
5
6
7
8
9
10
11

first loss

second loss
Continued on next page

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T
15
16
17
18
20
22
24
26
28
30
32
34
35
36
38
40
42
44
45
46

Table 2 continued from previous page

AC BD
5
7
6
7
8
8
BD has caught up
9
10
from here on only even values for T shown
10
12
11
14
third loss
5
8
BD is now ahead
6
10
7
12
8
14
9
16
fourth loss
8
4
9
5
11
6
13
7
15
8
17
fifth loss
8
4
9
exactly where we were at T=36

The interval 36T<46 represents the steady state here; the first connections average cwnd is 6 while the
second connections average is (8+9+...+16+17)/10 = 12.5. Worse, the first connection sends a windowful
only half as often. In the interval 36T<46 the first connection sends 4+5+6+7+8 = 30 packets; the second
connection sends 125. The cost of the first connections longer RTT is quadratic; in general, as we argue
more formally below, if the first connection has RTT = > 1 relative to the seconds, then its bandwidth will
be reduced by a factor of 1/2 .
Is this fair?
Early thinking was that there was something to fix here; see [F91] and [FJ92], 3.3 where the Constant-Rate
window-increase algorithm is discussed. A more recent attempt to address this problem is TCP Hybla,
[CF04]; discussed later in 15.8 TCP Hybla.
Alternatively, we may simply define TCP Renos bandwidth allocation as fair, at least in some contexts.
This approach is particularly common when the issue at hand is making sure other TCP implementations
and non-TCP flows compete for bandwidth in roughly the same way that TCP Reno does. While TCP
Renos strategy is now understood to be greedy in some respects, fixing it in the Internet at large is
generally recognized as a very difficult option.

14.3.3 TCP RTT bias

Let us consider more carefully the way TCP allocates bandwidth between two connections sharing a bottleneck link with relative RTTs of 1 and >1. We claimed above that the slower connections bandwidth will
be reduced by a factor of 1/2 ; we will now show this under some assumptions. First, uncontroversially,

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we will assume FIFO droptail queuing at the bottleneck router, and also that the network ceiling (and hence
cwnd at the point of loss) is sufficiently large. We will also assume, for simplicity, that the network
ceiling C is constant.
We need one more assumption: that most loss events are experienced by both connections. This is the
synchronized losses hypothesis, and is the most debatable; we will explore it further in the next section.
But first, here is the general argument with this assumption.
Let connection 1 be the faster connection, and assume a steady state has been reached. Both connections
experience loss when cwnd1 +cwnd2 C, because of the synchronized-loss hypothesis. Let c1 and c2
denote the respective window sizes at the point just before the loss. Both cwnd values are then halved.
Let N be the number of RTTs for connection 1 before the network ceiling is reached again. During this
time c1 increases by N; c2 increases by approximately N/ if N is reasonably large. Each of these increases
represents half the corresponding cwnd; we thus have c1 /2 = N and c2 /2 = N/. Taking ratios of respective
sides, we get c1 /c2 = N/(N/) = , and from that we can solve to get c1 = C/(1+) and c2 = C/(1+).
To get the relative bandwidths, we have to count packets sent during the interval between losses. Both
connections have cwnd averaging about 3/4 of the maximum value; that is, the average cwnds are 3/4 c1
and 3/4 c2 respectively. Connection 1 has N RTTs and so sends about 3/4 c1 N packets. Connection 2, with
its slower RTT, has only about N/ RTTs (again we use the assumption that N is reasonably large), and so
sends about 3/4 c2 N/ packets. The ratio of these is c1 /(c2 /) = 2 . Connection 1 sends fraction 2 /(1+2 )
of the packets; connection 2 sends fraction 1/(1+2 ).

14.3.4 Synchronized-Loss Hypothesis

The synchronized-loss hypothesis is based on the idea that, if the queue is full, late-arriving packets from
each connection will find it so, and be dropped. Once the queue becomes full, in other words, it stays full
for long enough for each connection to experience a packet loss.
That said, it is certainly possible to come up with hypothetical situations where losses are not synchronized.
Recall that a TCP Reno connections cwnd is incremented by only 1 each RTT; losses generally occur when
this single extra packet generated by the increment to cwnd arrives to find a full queue. Generally speaking,
packets are leaving the queue about as fast as they are arriving; actual overfull-queue instants may be rare.
It is certainly conceivable that, at least some of the time, one connection would overflow the queue by one
packet, and halve its cwnd, in a short enough time interval that the other connection misses the queue-full
moment entirely. Alternatively, if queue overflows lead to effectively random selection of lost packets (as
would certainly be true for random-drop queuing, and might be true for tail-drop if there were sufficient
randomness in packet arrival times), then there is a finite probability that all the lost packets at a given loss
event come from the same connection.
The synchronized-loss hypothesis is still valid if either or both connection experiences more than one packet
loss, within a single RTT; the hypothesis fails only when one connection experiences no losses.
We will return to possible failure of the synchronized-loss hypothesis in 14.5.2 Unsynchronized TCP
Losses. In 16.3 Two TCP Senders Competing we will consider some TCP Reno simulations in which
actual measurement does not entirely agree with the synchronized-loss model. Two problems will emerge.
The first is that when two connections compete in isolation, a form of synchronization known as phase
effects (16.3.4 Phase Effects) can introduce a persistent perhaps-unexpected bias. The second is that the
longer-RTT connection often does manage to miss out on the full-queue moment entirely, as discussed above

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in the second paragraph of this section. This results in a larger cwnd than the synchronized-loss hypothesis
would predict.

14.3.5 Loss Synchronization

The synchronized-loss hypothesis assumes all losses are synchronized. There is another side to this phenomenon that is an issue even if only some reasonable fraction of loss events are synchronized: synchronized
losses may represent a collective inefficiency in the use of bandwidth. In the immediate aftermath of a synchronized loss, it is very likely that the bottleneck link will go underutilized, as (at least) two connections
using it have just cut their sending rate in half. Better utilization would be achieved if the loss events could
be staggered, so that at the point when connection 1 experiences a loss, connection 2 is only halfway to its
next loss. For an example, see exercise 18.
This loss synchronization is a very real effect on the Internet, even if losses are not necessarily all synchronized. A major contributing factor to synchronization is the relatively slow response of all parties involved
to packet loss. In the diagram above at 14.3 TCP Fairness with Synchronized Losses, if A increments its
cwnd leading to an overflow at R, the AR link is likely still full of packets, and Rs queue remains full,
and so there is a reasonable likelihood that sender B will also experience a loss, even if its cwnd was not
particularly high, simply because its packets arrived at the wrong instant. Congestion, unfortunately, takes
time to clear.

14.3.6 Extreme RTT Ratios

What happens to TCP fairness if one TCP connection has a 100-fold-larger RTT than another? The short
answer is that the shorter connection may get 10,000 times the throughput. The longer answer is that this
isnt quite as easy to set up as one might imagine. For the arguments above, it is necessary for the two
connections to have a common bottleneck link:

In the diagram above, the AC connection wants its cwnd to be about 200 ms 10 packets/ms = 2,000
packets; it is competing for the RC link with the BD connection which is happy with a cwnd of 22. If Rs
queue capacity is also about 20, then with most of the bandwidth the BC connection will experience a loss
about every 20 RTTs, which is to say every 22 ms. If the AC link shares even a modest fraction of those
losses, it is indeed in trouble.
However, the AC cwnd cannot fall below 1.0; to test the 10,000-fold hypothesis taking this constraint into
account we would have to scale up the numbers on the BC link so the transit capacity there was at least
10,000. This would mean a 400 Gbps RC bandwidth, or else an unrealistically large AR delay.
As a second issue, realistically the AC link is much more likely to have its bottleneck somewhere in the
middle of its long path. In a typical real scenario along the lines of that diagrammed above, B, C and R are
all local to a site, and bandwidth of long-haul paths is almost always less than the local LAN bandwidth
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within a site. If the AR path has a 1 packet/ms bottleneck somewhere, then it may be less likely to be as
dramatically affected by BC traffic.
A few actual simulations using the methods of 16.3 Two TCP Senders Competing resulted in an average
cwnd for the AC connection of between 1 and 2, versus a BC cwnd of 20-25, regardless of whether the
two links shared a bottleneck or if the AC link had its bottleneck somewhere along the AR path. This may
suggest that the AC connection was indeed saved by the 1.0 cwnd minimum.

14.4 Notions of Fairness

There are several definitions for fair allocation of bandwidth among flows sharing a bottleneck link. One
is equal-shares fairness; another is what we might call TCP-Reno fairness: to divide the bandwidth the
way TCP Reno would. There are additional approaches to deciding what constitutes a fair allocation of
bandwidth.

14.4.1 Max-Min Fairness

A natural generalization of equal-shares fairness to the case where some flows may be capped is maxmin fairness, in which no flow bandwidth can be increased without decreasing some smaller flow rate.
Alternatively, we maximize the bandwidth of the smallest-capacity flow, and then, with that flow fixed,
maximize the flow with the next-smallest bandwidth, etc. A more intuitive explanation is that we distribute
bandwidth in tiny increments equally among the flows, until the bandwidth is exhausted (meaning we have
divided it equally), or one flow reaches its externally imposed bandwidth cap. At this point we continue
incrementing among the remaining flows; any time we encounter a flows external cap we are done with it.
As an example, consider the following, where we have connections AD, BD and CD, and where the
AR link has a bandwidth of 200 Kbps and all other links are 1000 Kbps. Starting from zero, we increment
the allocations of each of the three connections until we get to 200 Kbps per connection, at which point the
AD connection has maxed out the capacity of the AR link. We then continue allocating the remaining
400 Kbps equally between BD and CD, so they each end up with 400 Kbps.

As another example, known as the parking-lot topology, suppose we have the following network:

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There are four connections: one from A to D covering all three links, and three single-link connections AB,
BC and CD. Each link has the same bandwidth. If bandwidth allocations are incrementally distributed
among the four connections, then the first point at which any link bandwidth is maxed out occurs when all
four connections each have 50% of the link bandwidth; max-min fairness here means that each connection
has an equal share.

14.4.2 Proportional Fairness

A bandwidth allocation of rates r1 ,r2 ,...,rN for N connections satisfies proportional fairness if it is a
legal allocation of bandwidth, and for any other allocation s1 ,s2 ,...,sN , the aggregate proportional change
satisfies
(r1 s1 )/s1 + (r2 s2 )/s2 + ... + (rN sN )/sN < 0
Alternatively, proportional fairness means that the sum log(r1 )+log(r2 )+...+log(rN ) is minimized. If the
connections share only the bottleneck link, proportional fairness is achieved with equal shares. However,
consider the following two-stage parking-lot network:

Suppose the AB and BC links have bandwidth 1 unit, and we have three connections AB, BC and AC.
Then a proportionally fair solution is to give the AC link a bandwidth of 1/3 and each of the AB and BC
links a bandwidth of 2/3 (so each link has a total bandwidth of 1). For any change b in the bandwidth for
the AC link, the AB and BC links each change by -b. Equilibrium is achieved at the point where a 1%
reduction in the AC link results in two 0.5% increases, that is, the bandwidths are divided in proportion
1:2. Mathematically, if x is the throughput of the AC connection, we are minimizing log(x) + 2log(1-x).
Proportional fairness partially addresses the problem of TCP Renos bias against long-RTT connections;
specifically, TCPs bias here is still not proportionally fair, but TCPs response is closer to proportional
fairness than it is to max-min fairness.

14.5 TCP Reno loss rate versus cwnd

It turns out that we can express a connections average cwnd in terms of the packet loss rate, p, eg p =
10-4 = one packet lost in 10,000. The relationship comes by assuming that all packet losses are because
the network ceiling was reached. We will also assume that, when the network ceiling is reached, only one
packet is lost, although we can dispense with this by counting a cluster of related losses (within, say, one
RTT) as a single loss event.
Let C represent the network ceiling so that when cwnd reaches C a packet loss occurs. While the assumption that C is constant represents a very stable network, the general case is not terribly different. Then
cwnd varies between C/2 and C, with packet drops occurring whenever cwnd = C is reached. Let N = C/2.
Then between two consecutive packet loss events, that is, over one tooth of the TCP connection, a total of
N+(N+1)+ ... +2N packets are sent in N+1 flights; this sum can be expressed algebraically as 3/2 N(N+1)
1.5 N2 . The loss rate is thus one packet out of every 1.5 N2 , and the loss rate p is 1/(1.5 N2 ).

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The average cwnd in this scenario is 3/2 N (that is, the average of N=cwndmin and 2N=cwndmax ). If we
let M = 3/2 N represent the average cwnd, cwndmean , we can express the above loss rate in terms of M: the
number of packets between losses is 2/3 M2 , and so p=3/2 M-2 .
Now let us solve this for M=cwndmean in terms of p; we get M2 = 3/2 p-1 and thus
M = cwndmean = 1.225 p-1/2
where 1.225 is the square root of 3/2. Seen in this form, a given network loss rate sets the window size; this
loss rate is ultimately be tied to the network capacity. If we are interested in the maximum cwnd instead of
the mean, we multiply the above by 4/3.
From the above, the bandwidth available to a connection is now as follows (though RTT may not be constant):

bandwidth = cwnd/RTT = 1.225/(RTT p)

In [PFTK98] the authors consider a TCP Reno model that takes into account the measured frequency of
coarse timeouts (in addition to fast-recovery responses leading to cwnd halving), and develop a related
formula with additional terms.

14.5.1 Irregular teeth

In the preceding, we assumed that all teeth were the same size. What if they are not? In [OKM96], this
problem was considered under the assumption that every packet faces the same (small) loss probability (and
so the intervals between packet losses are exponentially distributed). In this model, it turns out that the
above formula still holds except the constant changes from 1.225 to 1.309833.
To understand how irregular teeth lead to a bigger constant, imagine sending a large number K of packets
which encounter n losses. If the losses are regularly spaced, then the TCP graph will have n equally sized
teeth, each with K/n packets. But if the n losses are randomly distributed, some teeth will be larger and some
will be smaller. The average tooth height will be the same as in the regularly-spaced case (see exercise 13).
However, the number of packets in any one tooth is generally related to the square of the height of that tooth,
and so larger teeth will count disproportionately more. Thus, the random distribution will have a higher total
number of packets delivered and thus a higher mean cwnd.
See also exercise 17, for a simple simulation that generates a numeric estimate for the constant 1.309833.
Note that losses at uniformly distributed random intervals may not be an ideal model for TCP either; in
the presence of congestion, loss events are far from statistical independence. In particular, immediately
following one loss another loss is unlikely to occur until the queue has time to fill up.

14.5.2 Unsynchronized TCP Losses

In 14.3.3 TCP RTT bias we considered a model in which all loss events are fully synchronized; that is,
whenever the queue becomes full, both TCP Reno connections always experience packet loss. In that model,
if RTT2 /RTT1 = then cwnd1 /cwnd2 = and bandwidth1 /bandwidth2 = 2 , where cwnd1 and cwnd2 are
the respective average values for cwnd.
What happens if loss events for two connections do not have such a neat one-to-one correspondence? We will
derive the ratio of loss events (or, more precisely, TCP loss responses) for connection 1 versus connection 2

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in terms of the bandwidth and RTT ratios, without using the synchronized-loss hypothesis.
Note that we are comparing the total number of loss events (or loss responses) here the total number of TCP
Reno teeth over a large time interval, and not the relative per-packet loss probabilities. One connection
might have numerically more losses than a second connection but, by dint of a smaller RTT, send more
packets between its losses than the other connection and thus have fewer losses per packet.
Let losscount1 and losscount2 be the number of loss responses for each connection over a long time interval
T. The ith connections per-packet loss probability is pi = losscounti /(bandwidthi T) = (losscounti
RTTi )/(cwndi T). But by the result of 14.5 TCP Reno loss rate versus cwnd, we also have cwndi =

k/ pi , or pi = k2 /cwndi 2 . Equating, we get

pi = k2 /cwndi 2 = (losscounti RTTi ) / (cwndi T)
and so
losscounti = k2 T / (cwndi RTTi )
Dividing and canceling, we get
losscount1 /losscount2 = (cwnd2 /cwnd1 ) (RTT2 /RTT1 )
We will make use of this in 16.4.2.2 Relative loss rates.
We can go just a little further with this: let denote the losscount ratio above:
= (cwnd2 /cwnd1 ) (RTT2 /RTT1 )
Therefore, as RTT2 /RTT1 = , we must have cwnd2 /cwnd1 = / and thus
bandwidth1 /bandwidth2 = (cwnd1 /cwnd2 ) (RTT2 /RTT1 ) = 2 /.
Note that if =, that is, if the longer-RTT connection has fewer loss events in exact inverse proportion to
the RTT, then bandwidth1 /bandwidth2 = = RTT2 /RTT1 , and also cwnd1 /cwnd2 = 1.

14.6 TCP Friendliness

Suppose we are sending packets using a non-TCP real-time protocol. How are we to manage congestion? In
particular, how are we to manage congestion in a way that treats other connections particularly TCP Reno
connections fairly?
For example, suppose we are sending interactive audio data in a congested environment. Because of the
real-time nature of the data, we cannot wait for lost-packet recovery, and so must use UDP rather than TCP.
We might further suppose that we can modify the encoding so as to reduce the sending rate as necessary
that is, that we are using adaptive encoding but that we would prefer in the absence of congestion to keep
the sending rate at the high end. We might also want a relatively uniform rate of sending; the TCP sawtooth
leads to periodic variations in throughput that we may wish to avoid.
Our application may not be windows-based, but we can still monitor the number of packets it has in flight
on the network at any one time; if the packets are small, we can count bytes instead. We can use this count
instead of the TCP cwnd.
We will say that a given communications strategy is TCP Friendly if the number of packets on the network
at any one time is approximately equal to the TCP Reno cwndmean for the prevailing packet loss rate p. Note
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that assuming losses are independent events, which is definitely not quite right but which is often Close
Enough in a long-enough time interval, all connections sharing a common bottleneck can be expected to
experience approximately the same packet loss rate.
The point of TCP Friendliness is to regulate the number of the non-Reno connections outstanding packets
in the presence of competition with TCP Reno, so as to achieve a degree of fairness. In the absence of
competition, the number of any connections outstanding packets will be bounded by the transit capacity
plus capacity of the bottleneck queue. Some non-Reno protocols (eg TCP Vegas, 15.4 TCP Vegas, or
constant-rate traffic, 14.6.2 RTP) may in the absence of competition have a loss rate of zero, simply
because they never overflow the queue.
Another way to approach TCP Friendliness is to start by defining Reno Fairness to be the bandwidth
allocations that TCP Reno assigns in the face of competition. TCP Friendliness then simply means that the
given non-Reno connection will get its Reno-Fair share not more, not less.
We will return to TCP Friendliness in the context of general AIMD in 14.7 AIMD Revisited.

14.6.1 TFRC
TFRC, or TCP-Friendly Rate Control, RFC 3448, uses the loss rate experienced, p, and the formulas above
to calculate a sending rate. It then allows sending at that rate; that is, TFRC is rate-based rather than windowbased. As the loss rate increases, the sending rate is adjusted downwards, and so on. However, adjustments
are done more smoothly than with TCP.
From RFC 5348:
TFRC is designed to be reasonably fair when competing for bandwidth with TCP flows, where
we call a flow reasonably fair if its sending rate is generally within a factor of two of the
sending rate of a TCP flow under the same conditions. [emphasis added; a factor of two might
not be considered close enough in some cases.]
The penalty of having smoother throughput than TCP while competing fairly for bandwidth is that TFRC
responds more slowly than TCP to changes in available bandwidth.
TFRC senders include in each packet a sequence number, a timestamp, and an estimated RTT.
The TFRC receiver is charged with sending back feedback packets, which serve as (partial) acknowledgments, and also include a receiver-calculated value for the loss rate over the previous RTT. The response
packets also include information on the current actual RTT, which the sender can use to update its estimated
RTT. The TFRC receiver might send back only one such packet per RTT.
The actual response protocol has several parts, but if the loss rate increases, then the primary feedback

mechanism is to calculate a new (lower) sending rate, using some variant of the cwnd = k/ p formula, and
then shift to that new rate. The rate would be cut in half only if the loss rate p quadrupled.
Newer versions of TFRC have a various features for responding more promptly to an unusually sudden
problem, but in normal use the calculated sending rate is used most of the time.

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14.6.2 RTP
The Real-Time Protocol, or RTP, is sometimes (though not always) coupled with TFRC. RTP is a UDPbased protocol for streaming time-sensitive data.
Some RTP features include:
The sender establishes a rate (rather than a window size) for sending packets
The receiver returns periodic summaries of loss rates
ACKs are relatively infrequent
RTP is suitable for multicast use; a very limited ACK rate is important when every packet sent might
have hundreds of recipients
The sender adjusts its cwnd-equivalent up or down based on the loss rate and the TCP-friendly

cwnd=k/ p rule
Usually some sort of stability rule is incorporated to avoid sudden changes in rate
As a common RTP example, a typical VoIP connection using a DS0 (64 Kbps) rate might send one packet
every 20 ms, containing 160 bytes of voice data, plus headers.
For a combination of RTP and TFRC to be useful, the underlying application must be rate-adaptive, so that
the application can still function when the available rate is reduced. This is often not the case for simple
VoIP encodings; see 19.11.4 RTP and VoIP.
We will return to RTP in 19.11 Real-time Transport Protocol (RTP).

14.7 AIMD Revisited

TCP Tahoe chose an increase increment of 1 on no losses, and a decrease factor of 1/2 otherwise.
Another approach to TCP Friendliness is to retain TCPs additive-increase, multiplicative-decrease strategy,
but to change the numbers. Suppose we denote by AIMD(,) the strategy of incrementing the window size
by after a window of no losses, and multiplying the window size by (1-)<1 on loss (so =0.1 means the
window is reduced by 10%). TCP Reno is thus AIMD(1,0.5).
Any AIMD(,) protocol also follows a sawtooth, where the slanted top to the tooth has slope . All
combinations of >0 and 0<<1 are possible. The dimensions of one tooth of the sawtooth are somewhat
constrained by and . Let h be the maximum height of the tooth and let w be the width (as measured in
RTTs). Then, if the losses occur at regular intervals, the height of the tooth at the left (low) edge is (1-)h
and the total vertical difference is h. This vertical difference must also be w, and so we get w = h,
or h/w = /; these values are labeled on the rightmost teeth in the diagram below. These equations mean
that the proportions of the tooth (h to w) are determined by and . Finally, the mean height of the tooth is
(1-/2)h.
We are primarily interested in AIMD(,) cases which are TCP Friendly (14.6 TCP Friendliness). TCP
friendliness means that an AIMD(,) connection with the same loss rate as TCP Reno will have the same
mean cwnd. Each tooth of the sawtooth represents one loss. The number of packets sent per tooth is, using
h and w as in the previous paragraph, (1-/2)hw.

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Geometrically, the number of packets sent per tooth is the area of the tooth, so two connections with the
same per-packet loss rate will have teeth with the same area. TCP Friendliness means that two connections
will have the same mean cwnd and thus the same tooth height. If the teeth of two connections have the
same area and the same height, they must have the same width (in RTTs), and thus that the rates of loss per
unit time must be equal, not just the rates of loss per number of packets.
The diagram below shows a TCP Reno tooth (blue) together with some unfriendly AIMD(,) teeth on the
left (red) and two friendly teeth on the right (green), the second friendly tooth is superimposed on the Reno
tooth.

The additional dashed lines within the central Reno tooth demonstrate the Reno 112 dimensions, and
show that the horizontal dashed line, representing cwndmean , is at height 3/2 w, where w is, as before, the
width.
In the rightmost green tooth, superimposed on the Reno tooth, we can see that h = (3/2)w + (/2)w. We
already know h = (/)w; setting these expressions equal, canceling the w and multiplying by 2 we get
(3+) = 2/, or = 2/(3+). Solving for we get
= 3/(2-)
or 1.5 for small . As the reduction factor 1- gets closer to 1, the protocol can remain TCP-friendly
by appropriately reducing ; eg AIMD(1/5, 1/8).
Having a small means that a connection does not have sudden bandwidth drops when losses occur; this can
be important for applications that rely on a regular rate of data transfer (such as voice). Such applications
are sometimes said to be slowly responsive, in contrast to TCPs cwnd = cwnd/2 rapid response.

14.7.1 AIMD and Convergence to Fairness

While TCP-friendly AIMD(,) protocols will converge to fairness when competing with TCP Reno (with
equal RTTs), a consequence of decreasing is that fairness may take longer to arrive; here is an example.
We will assume, as above in 14.3.3 TCP RTT bias, that loss events for the two competing connections are
synchronized. Recall that for two same-RTT TCP Reno connections (that is, AIMD(,) where =1/2),
if the initial difference in the connections respective cwnds is D, then D is reduced by half on each loss
event.
Now suppose we have two AIMD(,) connections with some other value of , and again with a difference
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D in their cwnd values. The two connections will each increase cwnd by each RTT, and so when losses
are not occurring D will remain constant. At loss events, D will be reduced by a factor of 1-. If =1/4,
corresponding to =3/7, then at each loss event D will be reduced only to 3/4 D, and the half-life of D
will be almost twice as large. The two connections will still converge to fairness as D0, but it will take
twice as long.

14.8 Active Queue Management

Active Queue Management means that routers take some active steps to manage their queues. Generally this
means either marking packets or dropping them. All routers drop packets, but this falls into the category
of active management when packets are dropped before the queue has run completely out of space. Queue
management can be done at the congestion knee, when queues just start to build (and when marking is
more appropriate), or as the queue starts to become full and approaches the cliff.

14.8.1 DECbit
In the congestion-avoidance technique proposed in [RJ90], routers encountering early signs of congestion
marked the packets they forwarded; senders used these markings to adjust their window size. The system
became known as DECbit in reference to the authors employer and was implemented in DECnet (closely
related to the OSI protocol suite), though apparently there was never a TCP/IP implementation. The idea
behind DECbit eventually made it into TCP/IP in the form of ECN, below, but while ECN like TCPs
other congestion responses applies control near the congestion cliff, DECbit proposed introducing control
when congestion was still minimal, just above the congestion knee.
The DECbit mechanism allowed routers to set a designated congestion bit. This would be set in the data
packet being forwarded, but the status of this bit would be echoed back in the corresponding ACK (otherwise
the sender would never hear about the congestion).
DECbit routers defined congestion as an average queue size greater than 1.0; that is, congestion meant that
the connection was just past the knee. Routers would set the congestion bit whenever this average-queue
condition was met.
The target for DECbit senders would then be to have 50% of packets marked as congested. If fewer than
50% of packets were marked, cwnd would be incremented by 1; if more than 50% were marked, then cwnd
would be decreased by a factor of 0.875. Note this is very different from the TCP approach in that DECbit
begins marking packets at the congestion knee while TCP Reno responds only to packet losses which
occur just over the cliff.
A consequence of this knee-based mechanism is that DECbit shoots for very limited queue utilization,
unlike TCP Reno. At a congested router, a DECbit connection would attempt to keep about 1.0 packets
in the routers queue, while a TCP Reno connection might fill the remainder of the queue. Thus, DECbit
would in principle compete poorly with any connection where the sender ignored the marked packets and
simply tried to keep cwnd as large as possible. As we will see in 15.4 TCP Vegas, TCP Vegas also strives
for limited queue utilization; in 16.5 TCP Reno versus TCP Vegas we investigate through simulation how
fairly TCP Vegas competes with TCP Reno.

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14.8.2 Explicit Congestion Notification (ECN)

ECN is the TCP/IP equivalent of DECbit, though the actual mechanics are quite different. The current
version is specified in RFC 3168, modifying an earlier version in RFC 2481. The IP header contains a twobit ECN field, consisting of the ECN-Capable Transport (ECT) bit and the Congestion Experienced (CE)
bit; these are shown in 7.1 The IPv4 Header. The ECT bit is set by a sender to indicate to routers that it is
able to use the ECN mechanism. (These are actually the older RFC 2481 names for the bits, but they will
serve our purposes here.) The TCP header contains an additional two bits: the ECN-Echo bit (ECE) and the
Congestion Window Reduced (CWR) bit; these are shown in the fourth row in 12.2 TCP Header.
Routers set the CE bit in the IP header when they might otherwise drop the packet (or possibly when the
queue is at least half full, or in lieu of a RED drop, below). As in DECbit, receivers echo the CE status back
to the sender in the ECE bit of the next ACK; the reason for using the ECE bit is that this bit belongs to the
TCP header and thus the TCP layer can be assured of control of it.
TCP senders treat ACKs with the ECE bit set the same as if a loss occurred: cwnd is cut in half. Because
there is no actual loss, the arriving ACKs can still pace continued sliding-windows sending. The Fast
Recovery mechanism is not needed.
When the TCP sender has responded to an ECE bit, it sets the CWR bit. Once the receiver has received a
packet with the CE bit set in the IP layer, it sets the ECE bit in all subsequent ACKs until it receives a data
packet with the CWR bit set. This provides for reliable communication of the congestion information, and
helps the sender respond just once to multiple packet losses within a single windowful.
Note that the initial packet marking is done at the IP layer, but the generation of the marked ACK and the
sender response to marked packets is at the TCP layer (the same is true of DECbit though the layers have
different names).
Only a packet that would otherwise have been dropped has its CE bit set; the router does not mark all
waiting packets once its queue reaches a certain threshold. Any marked packet must, as usual, wait in the
queue for its turn to be forwarded. The sender finds out about the congestion after one full RTT, versus
one full RTT plus four packet transmission times for Fast Retransmit. A much earlier, legacy strategy
was to require routers, upon dropping a packet, to immediately send back to the sender an ICMP Source
Quench packet. This is a faster way (the fastest possible way) to notify a sender of a loss. It was never
widely implemented, however, and was officially deprecated by RFC 6633.
Because ECN congestion is treated the same way as packet drops, ECN competes fairly with TCP Reno.

14.8.3 RED gateways

Standard routers drop packets only when the queue is full; senders have no overt indication before then
that the cliff is looming. The idea behind Random Early Detection (RED) gateways, introduced in [FJ93],
is that the router is allowed to drop an occasional packet much earlier, say when the queue is only half full.
These early packet drops provide a signal to senders that they should slow down; we will call them signaling
losses. While packets are indeed lost, they are dropped in such a manner that usually only one packet per
windowful (per connection) will be lost. Classic TCP Reno, in particular, behaves poorly with multiple
losses per window and RED is able to avoid such multiple losses.
RED is, in a technical sense, its own queuing discipline; we address these further in 18 Queuing and
Scheduling. However, it is often more helpful to think of RED as a technique that an otherwise-FIFO router

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can use to improve the performance of TCP traffic through it.

Tuning RED has taken some thought. An earlier version known as Early Random Drop (ERD) gateways
simply introduced a small uniform drop probability p, eg p=0.01, once the queue had reached a certain
threshold. This addresses the TCP Reno issue reasonably well, except that dropping with a uniform probability p leads to a surprisingly high rate of multiple drops in a cluster, or of long stretches with no drops.
More uniformity was needed, but drops at regular intervals are too uniform.
The actual RED algorithm does two things. First, the base drop probability pbase rises steadily from a
minimum queue threshold qmin to a maximum queue threshold qmax (these might be 40% and 80% respectively of the absolute queue capacity); at the maximum threshold, the drop probability is still quite small.
The base probability pbase increases linearly in this range according to the following formula, where pmax is
the maximum RED-drop probability; the value for pmax proposed in [FJ93] was 0.02.
pbase = pmax (avg_queuesize qmin )/(qmax qmin )
Second, as time passes after a RED drop, the actual drop probability pactual begins to rise, according to the
next formula:
pactual = pbase / (1 countpbase )
Here, count is the number of packets sent since the last RED drop. With count=0 we have pactual = pbase ,
but pactual rises from then on with a RED drop guaranteed within the next 1/pbase packets. This provides a
mechanism by which RED drops are uniformly enough spaced that it is unlikely two will occur in the same
window of the same connection, and yet random enough that it is unlikely that the RED drops will remain
synchronized with a single connection, thus targeting it unfairly.
One potential RED drawback is that the choice of the various parameters is decidedly ad hoc, and it is not
clear how to set them so that TCP connections with both small and large bandwidthdelay products are
handled appropriately. The probability pbase should, for example, be roughly 1/winsize, but winsize for TCP
connections can vary by several orders of magnitude. Nonetheless, RED gateways have worked quite well
in practice.
In 19.8 RED with In and Out we will look at an application of RED to quality-of-service guarantees.

14.9 The High-Bandwidth TCP Problem

The TCP Reno algorithm has a serious consequence for high-bandwidth connections: the cwnd needed
implies a very small unrealistically small packet-loss rate p. Noise losses (losses not due to congestion)
are not frequent but no longer negligible; these keep the window significantly smaller than it should be. The
following table, from RFC 3649, is based on an RTT of 0.1 seconds and a packet size of 1500 bytes, for
various throughputs. The cwnd values represent the bandwidthRTT products.
TCP Throughput (Mbps)
1
10
100
1000
10,000

RTTs between losses

5.5
55
555
5555
55555

14.9 The High-Bandwidth TCP Problem

cwnd
8.3
83
833
8333
83333

Packet Loss Rate P

0.02
0.0002
2 10-6
2 10-8
2 10-10

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Note the very small loss probability needed to support 10 Gbps; this works out to a bit error rate of less than
2 10-14 . For fiber optic data links, alas, a physical bit error rate of 10-13 is often considered acceptable;
there is thus no way to support the window size of the final row above.
Here is a similar table, expressing cwnd in terms of the packet loss rate:
Packet Loss Rate P
10-2
10-3
10-4
10-5
10-6
10-7
10-8
10-9
10-10

cwnd
12
38
120
379
1,200
3,795
12,000
37,948
120,000

RTTs between losses

8
25
80
252
800
2,530
8,000
25,298
80,000

The above two tables indicate that large window sizes require extremely small drop rates. This is the highbandwidth-TCP problem: how do we maintain a large window when a path has a large bandwidthdelay
product? The primary issue is that non-congestive (noise) packet losses bring the window size down, potentially far below where it could be. A secondary issue is that, even if such random drops are not significant,
the increase of cwnd to a reasonable level can be quite slow. If the network ceiling were about 2,000 packets, then the normal sawtooth return to the ceiling after a loss would take 1,000 RTTs. This is slow, but the
sender would still average 75% throughput, as we saw in 13.7 TCP and Bottleneck Link Utilization. Perhaps more seriously, if the network ceiling were to double to 4,000 packets due to decreases in competing
traffic, it would take the sender an additional 2,000 RTTs to reach the point where the link was saturated.
In the following diagram, the network ceiling and the ideal TCP sawtooth are shown in green. The ideal
TCP sawtooth should range between 50% and 100% of the ceiling; in the diagram, noise or non-congestive
losses occur at the red xs, bringing down the throughput to a much lower average level.

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14.10 The Lossy-Link TCP Problem

Closely related to the high-bandwidth problem is the lossy-link problem, where one link on the path has a
relatively high non-congestive-loss rate; the classic example of such a link is Wi-Fi. If TCP is used on a
path with a 1.0% loss rate, then 14.5 TCP Reno loss rate versus cwnd indicates that the sender can expect
an average cwnd of only about 12, no matter how high the bandwidthdelay product is.
The only difference between the lossy-link problem and the high-bandwidth problem is one of scale; the
lossy-link problem involves unusually large values of p while the high-bandwidth problem involves circumstances where p is quite low but not low enough. For a given non-congestive loss rate p, if the

bandwidthdelay product is much in excess of 1.22/ p then the sender will be unable to maintain a cwnd
close to the network ceiling.

14.11 The Satellite-Link TCP Problem

A third TCP problem, only partially related to the previous two, is that encountered by TCP users with very
long RTTs. The most dramatic example of this involves satellite Internet links (3.5.2 Satellite Internet).
Communication each way involves routing the signal through a satellite in geosynchronous orbit; a round
trip involves four up-or-down trips of ~36,000 km each and thus has a propagation delay of about 500ms.
If we take the per-user bandwidth to be 1 Mbps (satellite ISPs usually provide quite limited bandwidth,
though peak bandwidths can be higher), then the bandwidthdelay product is about 40 packets. This is not
especially high, even when typical queuing delays of another ~500ms are included, but the fact that it takes
many seconds to reach even a moderate cwnd is an annoyance for many applications. Most ISPs provide an
acceleration mechanism when they can identify a TCP connection as a file download; this usually involves
transferring the file over the satellite portion of the path using a proprietary protocol. However, this is not
much use to those using TCP connections that involve multiple bidirectional exchanges; eg those using VPN
connections.

14.12 Epilog
TCP Renos core congestion algorithm is based on algorithms in Jacobson and Karels 1988 paper [JK88],
now twenty-five years old. There are concerns both that TCP Reno uses too much bandwidth (the greediness
issue) and that it does not use enough (the high-bandwidth-TCP problem).
In the next chapter we consider alternative versions of TCP that attempt to solve some of the above problems
associated with TCP Reno.

14.13 Exercises
1. Consider the following network, where the bandwidths marked are all in packets/ms. C is sending to D
using sliding windows and A and B are idle.
C
100

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100

R1

R2

100

100
D

Suppose the propagation delay on the 100 packet/ms links is 1 ms, and the propagation delay on the R1R2
link is 2 ms. The RTTnoLoad for the CD path is thus about 8 ms, for a bandwidthdelay product of 40
packets. If C uses winsize = 50, then the queue at R1 will have size 10.
Now suppose A starts sending to B using sliding windows, also with winsize = 50. What will be the size of
the queue at R1?
Hint: by symmetry, the queue will be equally divided between As packets and Cs, and A and C will each
see a throughput of 2.5 packets/ms. RTTnoLoad , however, does not change.
2. In the previous exercise, give the average number of data packets in transit on each link:

(a). for the original case in which C is the only sender, with winsize = 50 (the only active links here are
CR1, R1R2 and R2D).
(b). for the new case in which B is also sending, also with winsize = 50. In this case all links are active.

Each link will also have an equal number of ACK packets in transit in the reverse direction.
3. Consider the CD path from the diagram of 14.2.4 Example 4: cross traffic and RTT variation:
C

100

R1

R2

100

Link numbers are bandwidths in packets/ms. Assume C is the only sender.

(a). Give propagation delays for the links CR1 and R2D so that there will be an average of 5 packets in
transit on the CR1 and R2D links, in each direction, if C uses a winsize sufficient to saturate the
bottleneck R1R2 link.
(b). Give propagation delays for all three links so that, when C uses a winsize equal to the round-trip transit
capacity, there are 5 packets each way on the CR1 link, 10 on the R1R2 link, and 20 on the R2D link.

4. Suppose we have the network layout below of 14.2.4 Example 4: cross traffic and RTT variation, except
that the R1R2 bandwidth is 6 packets/ms and the R2R3 bandwidth is 3. Suppose also that A and C have
settled upon window sizes so that each contributes 30 packets to R1s queue and thus each has 50% of the
bandwidth. R2 will then be sending 3 packets/ms to R3 and so will have no queue.
Now As winsize is incremented by 10, initially, at least, leading to A contributing more than 50% of R1s
queue. When the steady state is reached, how will these extra 10 packets be distributed between R1 and R2?
Hint: As As winsize increases, As overall throughput cannot rise due to the bandwidth restriction of the
R2R3 link.

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C
100
A

100

R1

R2

R3

100

100
D

Diagram for exercises 4 and 5

5. Suppose we have the network layout above similar to 14.2.4 Example 4: cross traffic and RTT variation
for which we are given that the round-trip CD RTTnoLoad is 5 ms. The round-trip AB RTTnoLoad may be
different.
The R1R2 bandwidth is 6 packets/ms, so with A idle the CD The R2R3 bandwidth is 3 packets/ms.

(a). Suppose that A and C have window sizes such that, with both transmitting, each has 30 packets in the
queue at R1. What is Cs winsize? Hint: Cs bandwidth is now 3 packets/ms.
(b). Now suppose Cs winsize, with A idle, is 60. In this case the CD transit capacity would be 5 ms 6
packets/ms = 30 packets, and so C would have 6030 = 30 packets in R1s queue. A then begins sending,
with a winsize chosen so that A and Cs contributions to R1s queue are equal; Cs winsize remains at 60.
What will be Cs (and thus As) queue usage at R1? Hint: find the transit capacity for a bandwidth of 3
packets/ms.
(c). Suppose the AB RTTnoLoad is 10 ms. If Cs winsize is 60, find the winsize for A that makes A and Cs
contributions to R1s queue equal.

6. One way to address the reduced bandwidth TCP Reno gives to long-RTT connections is for all connections to use an increase increment of RTT2 instead of 1; that is, everyone uses AIMD(RTT2 ,1/2) instead of
AIMD(1,1/2) (or AIMD(kRTT2 ,1/2), where k is an arbitrary scaling factor that applies to everyone).

(a). Construct a table in the style of of 14.3.2 Example 3: Longer RTT above, showing the result of two
connections using this strategy, where one connection has RTT = 1 and the other has RTT = 2. Start the
connections with cwnd=RTT2 , and assume a loss occurs when cwnd1 + cwnd2 > 24.
(b). Explain why this strategy might not be desirable if one connection is over a direct LAN with an RTT of
1 ms, while the second connection has a very long path and an RTT of 1.0 sec.

7. For each value or below, find the other value so that AIMD(,) is TCP-friendly.

(a). = 1/5
(b). = 2/9
(c). = 1/5

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Then pick the pair that has the smallest , and draw a sawtooth diagram that is approximately proportional:
should be the slope of the linear increase, and should be the decrease fraction at the end of each tooth.
8. Suppose two TCP flows compete. The first flow uses AIMD(1 , 1 ) and the second uses AIMD(2 , 2 );
neither flow is necessarily TCP-Reno-friendly. The two connections, however, compete fairly with one
another; that is, they have the same average packet-loss rates. Show that 1 / 1 = (2- 2 )/(2- 1 ) 2 / 2 .
Assume regular losses, and use the methods of 14.7 AIMD Revisited.
9. Suppose two 1KB packets are sent as part of a packet-pair probe, and the minimum time measured
between arrivals is 5 ms. What is the estimated bottleneck bandwidth?
10. Consider the following three causes of a 1-second network delay between A and B. In all cases, assume
ACKs travel instantly from B back to A.
(i) An intermediate router with a 1-second-per-packet bandwidth delay; all other bandwidth
delays negligible
(ii) An intermediate link with a 1-second propagation delay; all bandwidth delays negligible
(iii) An intermediate router with a 100-ms-per-packet bandwidth delay, and a steadily
replenished queue of 10 packets, from another source (as in the diagram in 14.2.4 Example 4:
cross traffic and RTT variation).
How might a sender distinguish between these three cases? Hint: consider packet pairs, or some variant.
11. Consider again the three-link parking-lot network from 14.4.1 Max-Min Fairness:

(a). Suppose we have two end-to-end connections, in addition to one single-link connection for each link.
Find the max-min-fair allocation.
(b). Suppose we have a single end-to-end connection, and one BC and CD connection, but two AB
connections. Find the max-min-fair allocation.

12. Consider the two-link parking-lot network:

A

Suppose there are two AC connections, one AB connection and one AC connection. Find the allocation
that is proportionally fair.
13. Suppose we use TCP Reno to send K packets over R RTT intervals. The transmission experiences n
not-necessarily-uniform loss events; the TCP cwnd graph thus has n sawtooth peaks of heights N1 through
Nn . At the start of the graph, cwnd = A, and at the end of the graph, cwnd = B. Show that the sum N1 +
... + Nn is 2(R+A-B), and in particular the average tooth height is independent of the distribution of the loss
events.

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14. Suppose TCP Reno has regularly spaced sawtooth peaks of the same height, but the packet losses come
in pairs, with just enough separation that both losses in a pair are counted separately. N is large enough that
the spacing between the two losses is negligible. The net effect is that each large-scale tooth ranges from

height N/4 to N. As in 14.5 TCP Reno loss rate versus cwnd, cwndmean = K/ p for some constant K. Find
the constant. Note that the loss rate here is p = 2/(number of packets sent in one tooth).

15. As in the previous exercise, suppose a TCP transmission has large-scale teeth of height N. Between
each pair of consecutive large teeth, however, there are K-1 additional losses resulting in K-1 additional tiny
teeth; N is large enough that these tiny teeth can be ignored. A non-Reno variant of TCP is used, so that
between these tiny teeth cwnd is assumed not to be cut in half; during the course of these tiny teeth cwnd
does not change much at all. The large-scale tooth has width N/2 and height ranging from N/2 to N, and

there are K losses per large-scale tooth. Find the ratio cwnd/ p, in terms of K. When K=1 your answer
should reduce to that derived in 14.5 TCP Reno loss rate versus cwnd.

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16. Suppose a TCP Reno tooth starts with cwnd = c, and contains N packets. Let w be the width of the
tooth, in RTTs as usual. Show that w = (c2 + 2N)1/2 c. Hint: the maximum height of the tooth will be c+w,
and so the average height will be c + w/2. Find an equation relating c, w and N, and solve for w using the
quadratic formula.
17. Suppose in a TCP Reno run each packet is equally likely to be lost; the number of packets N in each
tooth will therefore be distributed exponentially. That is, N = -k log(X), where X is a uniformly distributed
random number in the range 0<X<1 (k, which does not really matter here, is the mean interval between
losses). Write a simple program that simulates such a TCP Reno run. At the end of the simulation, output

an estimate of the constant C in the formula cwndmean = C/ p. You should get a value of about 1.31, as in
the formula in 14.5.1 Irregular teeth.
Hint: There is no need to simulate packet transmissions; we simply create a series of teeth of random size,
and maintain running totals of the number of packets sent, the number of RTT intervals needed to send them,
and the number of loss events (that is, teeth). After each loss event (each tooth), we update:
total_packets += packets sent in this tooth
RTT_intervals += RTT intervals in this tooth
loss_events += 1 (one tooth = one loss event)
If a loss event marking the end of one tooth occurs at a specific value of cwnd, the next tooth begins at
height c = cwnd/2. If N is the random value for the number of packets in this tooth, then by the previous
exercise the tooth width in RTTs is w = (c2 + 2N)1/2 c; the next peak (that is, loss event) therefore occurs
when cwnd = c+w. Update the totals as above and go on to the next tooth. It should be possible to run this
simulation for 1 million teeth in modest time.
18. Suppose two TCP connections have the same RTT and share a bottleneck link, for which there is no other
competition. The size of the bottleneck queue is negligible when compared to the bandwidth RTTnoLoad
product. Loss events occur at regular intervals.
In Exercise 12 of the previous chapter, you were to show that if losses are synchronized then the two
connections together will use 75% of the total bottleneck-link capacity
Now assume the two TCP connections have no losses in common, and, in fact, alternate losses at regular
intervals as in the following diagram.

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Both connections have a maximum cwnd of C. When Connection 1 experiences a loss, Connection 2 will
have cwnd = 75% of C, and vice-versa.

(a). What is the combined transit capacity of the paths, in terms of C? (Because the queue size is
negligible, the transit capacity is approximately the sum of the cwnds at the point of loss.)
(b). Find the bottleneck-link utilization. Hint: it should be at least 85%.

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Since the rise of TCP Reno, several TCP alternatives to Reno have been developed; each attempts to address
some perceived shortcoming of Reno. While many of them are very specific attempts to address the highbandwidth problem we considered in 14.9 The High-Bandwidth TCP Problem, some focus primarily or
entirely on other TCP Reno foibles. One such issue is TCP Renos greediness in terms of queue utilization;
another is the lossy-link problem (14.10 The Lossy-Link TCP Problem) experienced by, say, Wi-Fi users.
Generally speaking, a TCP implementation can respond to congestion at the cliff that is, it can respond
to packet losses or can respond to congestion at the knee that is, it can detect the increase in RTT
associated with the filling of the queue. These strategies are sometimes referred to as loss-based and delaybased, respectively; the latter term because of the rise in RTT. TCP implementers can tweak both the loss
response the multiplicative decrease of TCP Reno and also the way TCP increases its cwnd in the
absence of loss. There is a rich variety of options available.
The concept of monitoring the RTT to avoid congestion at the knee was first introduced in TCP Vegas
(15.4 TCP Vegas). One striking feature of TCP Vegas is that, in the absence of competition, the queue
may never fill, and thus there may not be any congestive losses. The TCP sawtooth, in other words, is not
inevitable.
When losses do occur, most of the mechanisms reviewed here continue to use the TCP NewReno recovery
strategy. As most of the implementations here are relatively recent, the senders can generally expect that the
receiving end will support SACK TCP, which allows more rapid recovery from multiple losses.
On linux systems, the TCP congestion-control mechanism can be set by writing an appropriate string to
/proc/sys/net/ipv4/tcp_congestion_control; the options on my system as of this writing
are
highspeed
htcp
hybla
illinois
vegas
veno
westwood
bic
cubic
We review several of these below. TCP Cubic is currently (2013) the default linux congestion-control
implementation; TCP Bic was a precursor.

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15.1 High-Bandwidth Desiderata

One goal of all TCP implementations that attempt to fix the high-bandwidth problem is to be unfair to TCP
Reno: the whole point is to allow cwnd to increase more aggressively than is permitted by Reno. Beyond
that, let us review what else a TCP version should do.
First is the backwards-compatibility constraint: any new TCP should exhibit reasonable fairness with TCP
Reno at lower bandwidthdelay products. In particular, it should not ever have a significantly lower cwnd
than a competing TCP Reno would get. But also it should not take bandwidth unfairly from a TCP Reno
connection: the above comment about unfairness to Reno notwithstanding, the new TCP, when competing
with TCP Reno, should leave the Reno connection with about the same bandwidth it would have if it were
competing with another Reno connection. This is possible because at higher bandwidthdelay products
TCP Reno does not efficiently use the available bandwidth; the new TCP should to the extent possible
restrict itself to consuming this previously unavailable bandwidth rather than eating significantly into the
bandwidth of a competing TCP Reno connection.
There is also the self-fairness issue: multiple connections using the new TCP should receive similar bandwidth allocations, at least with similar RTTs. For dissimilar RTTs, the bandwidth proportions should ideally
be no worse than they would be under TCP Reno.
Ideally, we also want relatively rapid convergence to fairness; fairness is something of a hollow promise
if only connections transferring more than a gigabyte will benefit from it. For TCP Reno, two connections
halve the difference in their respective cwnds at each shared loss event; as we saw in 14.7.1 AIMD and
Convergence to Fairness, slower convergence is possible.
It is harder to hope for fairness between competing new implementations. However, at the very least, if new
implementations tcp1 and tcp2 are competing, then neither should get less than TCP Reno would get.
Some new TCPs make use of careful RTT measurements, and, as we shall see below, such measurements
are subject to a considerable degree of noise. Any new TCP implementation should be reasonably robust in
the face of inaccuracies in RTT measurement; a modest or transient measurement error should not make the
protocol behave badly, in either the direction of low cwnd or of high.
Finally, a new TCP should ideally try to avoid clusters of multiple losses at each loss event. Such multiple
losses, for example, are a problem for TCP NewReno without SACK: as we have seen, it takes one RTT
to retransmit each lost packet. Even with SACK, multiple losses complicate recovery. Yet if a new TCP
increments cwnd by an amount N>1 after each RTT, then there is potential for the network ceiling to be
exceeded by N within one RTT, making a cluster of N losses reasonably likely to occur. These losses are
likely distributed among all connections, not just the new-TCP one.
All TCPs addressing the high-bandwidth issue will need a cwnd-increment N that is fairly large, at least
some of the time, apparently conflicting with this no-multiple-losses ideal. One trick is to reduce N when
packet loss appears to be imminent. TCP Illinois and TCP Cubic do have mechanisms in place to reduce
multiple losses.

15.2 RTTs
The exact performance of some of the faster TCPs we consider for that matter, the exact performance of
TCP Reno is influenced by the RTT. This may affect individual TCP performance and also competition

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between different TCPs. For reference, here are a few typical RTTs from Chicago to various other places:
US West Coast: 50-100 ms
Europe: 100-150 ms
Southeast Asia: 100-200 ms

15.3 Highspeed TCP

An early proposed fix for the high-bandwidth-TCP problem is HighSpeed TCP, documented in RFC 3649
(Floyd, 2003). Highspeed TCP is sometimes called HS-TCP, but we use the longer name here to avoid
confusion with the entirely unrelated H-TCP, below.
For each loss-free RTT, Highspeed TCP allows a cwnd increment by more than 1.0, at least once cwnd is
large enough. If the cwnd-increment value is N = N(cwnd), this is equivalent to having N parallel TCP
Reno connections. Here are the cwnd-increment values in terms of cwnd:
cwnd
1
10
100
1,000
10,000
100,000

N(cwnd)
1.0
1.0
1.4
3.6
9.2
23.0

The formula for N(cwnd) is largely empirical; an algebraic expression for it is

N(cwnd) = max(1.0, 0.23cwnd0.4 )
The second term in the max() above begins to dominate when cwnd = 38 or so.
It may be helpful to view Highspeed TCP in terms of the cwnd graph between losses. For ordinary TCP,
the graph increases linearly. For Highspeed TCP, the graph is convex (lying above its tangent). This means
that there is an increase in the rate of cwnd increase, as time goes on (up to the point of packet loss).

15.3 Highspeed TCP

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This might be an appropriate time to point out that in TCP Reno, the cwnd-versus-time graph between
losses is actually slightly concave (lying below its tangent). We do get a strictly linear graph if we plot
cwnd as a function of the count of elapsed RTTs, but the RTTs are themselves slowly increasing as a
function of time once the queue starts filling up. At that point, the cwnd-versus-time graph bends slightly
down. If the bottleneck queue capacity matches the total path transit capacity, the RTTs for a full queue are
about double the RTTs for an empty queue.
In general, when Highspeed-TCP competes with a new TCP Reno flow it is N times as aggressive, and grabs
N times the bandwidth, where N = N(cwnd). In this it behaves very much like N separate TCP flows, or,
more precisely, N separate TCP flows that have all their loss events completely synchronized.

15.4 TCP Vegas

TCP Vegas, introduced in [BP95], is the only new TCP version we consider here that dates from the previous century. The goal was not directly to address the high-bandwidth problem, but rather to improve TCP
throughput generally; indeed, in 1995 the high-bandwidth problem had not yet surfaced as a practical concern. The goal of TCP Vegas is essentially to eliminate congestive losses, and to try to keep the bottleneck
link 100% utilized at all times, thus improving on TCP Renos typical sawtooth-average bottleneck-link
utilization of 75% (13.7 TCP and Bottleneck Link Utilization).
TCP Vegas achieves this improvement by, like DECbit, recognizing TCP congestion at the knee, that is,
at the point where the bottleneck link becomes saturated and further cwnd increases simply result in RTT
increases. A TCP Vegas sender alone or in competition only with other TCP Vegas connections will seldom
if ever approach the cliff where packet losses occur.
To accomplish this, no special router cooperation or even receiver cooperation is necessary. Instead, the
sender uses careful monitoring of the RTT to keep track of the number of extra packets (ie packets sitting
in queues) it has injected into the network. In the absence of competition, the RTT will remain constant,
equal to RTTnoLoad , until cwnd has increased to the point when the bottleneck link has become saturated and
the queue begins to fill (6.3.2 RTT Calculations). By monitoring the bandwidth as well, a TCP sender can
even determine the actual number of packets in the bottleneck queue, as bandwidth (RTT RTTnoLoad ).
TCP Vegas uses this information to attempt to maintain at all times a small but positive number of packets
in the bottleneck queue.
A TCP sender can easily measure available bandwidth; the simplest measurement is cwnd/RTT (measured
in packets/sec). Let us denote this bandwidth estimate by BWE; for the time being we will accept BWE as
accurate, though see 15.6.1 ACK Compression and Westwood+ below. TCP Vegas estimates RTTnoLoad by
the minimum RTT (RTTmin ) encountered during the connection. The ideal cwnd that just saturates the
bottleneck link is BWERTTnoLoad . Note that BWE will be much more volatile than RTTmin ; the latter will
typically reach its final value early in the connection, while BWE will fluctuate up and down with congestion
(which will also act on RTT, but by increasing it).
As in 6.3.2 RTT Calculations, any TCP sender can estimate queue utilization as
queue_use = cwnd BWERTTnoLoad = cwnd (1 RTTnoLoad /RTTactual )
TCP Vegas then adjusts cwnd regularly to maintain the following:
queue_use
which is the same as
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BWERTTnoLoad + cwnd BWERTTnoLoad +

Typically = 2-3 packets and = 4-6 packets. We increment cwnd by 1 if cwnd falls below the lower
limit (eg if BWE has increased). Similarly, we decrement cwnd by 1 if BWE drops and cwnd exceeds
BWERTTnoLoad + . These adjustments are conceptually done once per RTT. Typically a TCP Vegas
sender would also set cwnd = cwnd/2 if a packet were actually lost, though this does not necessarily
happen nearly as often as with TCP Reno.
TCP Vegas achieves its goal quite well. If one monitors the number of packets in queues, through real
measurement or in simulation, the number does indeed stay between and . In the absence of competition
from TCP Reno, a single TCP Vegas connection will never experience congestive packet loss. This is a
remarkable achievement.
The use of returning ACKs to determine BWE is subject to errors due to ACK compression, 15.6.1 ACK
Compression and Westwood+. This is generally not a major problem with TCP Vegas, however.

15.4.1 TCP Vegas versus TCP Reno

Despite its striking ability to avoid congestive losses in the absence of competition, TCP Vegas encounters
a potentially serious fairness problem when competing with TCP Reno, at least for the case when queue
capacity exceeds or is close to the transit capacity (13.7 TCP and Bottleneck Link Utilization). TCP Vegas
will try to minimize its queue use, while TCP Reno happily fills the queue. And whoever has more packets
in the queue has a greater share of bandwidth.
To make this precise, suppose we have two TCP connections sharing a bottleneck router R, the first using
TCP Vegas and the second using TCP Reno. Suppose further that both connections have a path transit
capacity of 10 packets, and Rs queue capacity is 40 packets. If =3 and =5, TCP Vegas might keep an
average of four packets in the queue. Unfortunately, TCP Reno then gobbles up most of the rest of the
queue space, as follows. There are 40-4 = 36 spaces left in the queue after TCP Vegas takes its quota, and
10 in the TCP Reno connections path, for a total of 46. This represents the TCP Reno connections network
ceiling, and is the point at which TCP Reno halves cwnd; therefore cwnd will vary from 23 to 46 with an
average of about 34. Of these 34 packets, if 10 are in transit then 24 are in Rs queue. If on average R has 24
packets from the Reno connection and 4 from the Vegas connection, then the bandwidth available to these
connections will also be in this same 6:1 proportion. The TCP Vegas connection will get 1/7 the bandwidth,
because it occupies 1/7 the queue, and the TCP Reno connection will take the other 6/7.
To put it another way, TCP Vegas is potentially too civil to compete with TCP Reno.
Even worse, Renos aggressive queue filling will eventually force the TCP Vegas cwnd to decrease; see
Exercise 4 below.
This Vegas-Reno fairness problem is most significant when the queue size is an appreciable fraction of
the path transit capacity. During periods when the queue is empty, TCPs Vegas and Reno increase cwnd
at the same rate, so when the queue size is small compared to the path capacity, as is the case for most
high-bandwidth paths, TCP Vegas and TCP Reno are much closer to being fair.
In 16.5 TCP Reno versus TCP Vegas we compare TCP Vegas with TCP Reno in actual simulation. With
a transit capacity of 220 packets and a queue capacity of 10 packets, TCPs Vegas and Reno receive almost
exactly the same bandwidth.

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Note that if the bottleneck router used Fair Queuing (to be introduced in 18.5 Fair Queuing) on a perconnection basis, then the TCP Reno connections queue greediness would not be of any benefit, and both
connections would get similar shares of bandwidth with the TCP Vegas connection experiencing lower delay.
Let us next consider how TCP Vegas behaves when there is an increase in RTT due to the kind of cross
traffic shown in 14.2.4 Example 4: cross traffic and RTT variation and again in the diagram below. Let
AB be the TCP Vegas connection and assume that its queue-size target is 4 packets (eg =3, =5). We will
also assume that the RTTnoLoad for the AB path is about 5ms and the RTT for the CD path is also low. As
before, the link labels represent bandwidths in packets/ms, meaning that the round-trip AB transit capacity
is 10 packets.

Initially, in the absence of CD traffic, the AB connection will send at a rate of 2 packets/ms (the R2R3
bottleneck), and maintain a queue of four packets at R2. Because the RTT transit capacity is 10 packets, this
will be achieved with a window size of 10+4 = 14.
Now let the CD traffic start up, with a winsize so as to keep about four times as many packets in R1s queue
as A, once the new steady-state is reached. If all four of the AB connections queue packets end up now
at R1 rather than R2, then C would need to contribute at least 16 packets. These 16 packets will add a delay
of about 16/5 3ms; the AB path will see a more-or-less-fixed 3ms increase in RTT. A will also see a
decrease in bandwidth due to competition; with C consuming 80% of R1s queue, As share wll fall to 20%
and thus its bandwidth will fall to 20% of the R1R2 link bandwidth, that is, 1 packet/ms. Denote this new
value by BWEnew . TCP Vegas will attempt to decrease cwnd so that
cwnd BWEnew RTTnoLoad + 4
As estimate of RTTnoLoad , as RTTmin , will not change; the RTT has gotten larger, not smaller. So we have
BWEnew RTTnoLoad 1 packet/ms 5 ms = 5 packets; adding the 4 reserved for the queue, the new value
of cwnd is now about 9, down from 14.
On the one hand, this new value of cwnd does represent 5 packets now in transit, plus 4 in R1s queue; this
is indeed the correct response. But note that this division into transit and queue packets is an average. The
actual physical AB round-trip transit capacity remains about 10 packets, meaning that if the new packets
were all appropriately spaced then none of them might be in any queue.

15.5 FAST TCP

FAST TCP is closely related to TCP Vegas; the idea is to keep the fixed-queue-utilization feature of TCP
Vegas to the extent possible, but to provide overall improved performance, in particular in the face of com-

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petition with TCP Reno. Details can be found in [JWL04] and [WJLH06]. FAST TCP is patented; see
patent 7,974,195.
As with TCP Vegas, the sender estimates RTTnoLoad as RTTmin . At regular short fixed intervals (eg 20ms)
cwnd is updated via the following weighted average:
cwndnew = (1-)cwnd + ((RTTnoLoad /RTT)cwnd + )
where is a constant between 0 and 1 determining how volatile the cwnd update is (1 is the most
volatile) and is a fixed constant, which, as we will verify shortly, represents the number of packets the
sender tries to keep in the bottleneck queue, as in TCP Vegas. Note that the cwnd update frequency is not
tied to the RTT.
If RTT is constant for multiple consecutive update intervals, and is larger than RTTnoLoad , the above will
converge to a constant cwnd, in which case we can solve for it. Convergence implies cwndnew = cwnd =
((RTTnoLoad /RTT)cwnd + ), and from there we get cwnd(RTTRTTnoLoad )/RTT = . As we saw in
6.3.2 RTT Calculations, cwnd/RTT is the throughput, and so = throughput (RTTRTTnoLoad ) is then
the number of packets in the queue. In other words, FAST TCP, when it reaches a steady state, leaves
packets in the queue. As long as this is the case, the queue will not overflow (assuming is less than the
queue capacity).
Whenever the queue is not full, though, we have RTT = RTTnoLoad , in which case FAST TCPs cwnd-update
strategy reduces to cwndnew = cwnd + . For =0.5 and =10, this increments cwnd by 5. Furthermore,
FAST TCP performs this increment at a specific rate independent of the RTT, eg every 20ms; for long-haul
links this is much less than the RTT. FAST TCP will, in other words, increase cwnd very aggressively until
the point when queuing delays occur and RTT begins to increase.
When FAST TCP is competing with TCP Reno, it does not directly address the queue-utilization competition
problem experienced by TCP Vegas. FAST TCP will try to limit its queue utilization to ; TCP Reno,
however, will continue to increase its cwnd until the queue is full. Once the queue begins to fill, TCP Reno
will pull ahead of FAST TCP just as it did with TCP Vegas. However, FAST TCP does not reduce its cwnd
in the face of TCP Reno competition as quickly as TCP Vegas.
Additionally, FAST TCP can often offset this Reno-competition problem in other ways as well. First, the
value of can be increased from the value of around 4 packets originally proposed for TCP Vegas; in
[TWHL05] the value =30 is suggested. Second, for high bandwidthdelay products, the queue-filling
phase of a TCP Reno sawtooth (see 13.7 TCP and Bottleneck Link Utilization) becomes relatively smaller.
In the earlier link-unsaturated phase of each sawtooth, TCP Reno increases cwnd by 1 each RTT. As noted
above, however, FAST TCP is allowed to increase cwnd much more rapidly in this earlier phase, and so
FAST TCP can get substantially ahead of TCP Reno. It may fall back somewhat during the queue-filling
phase, but overall the FAST and Reno flows may compete reasonably fairly.

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The diagram above illustrates a FAST TCP graph of cwnd versus time, in blue; it is superimposed over one
sawtooth of TCP Reno with the same network ceiling. Note that cwnd rises rapidly when it is below the
path transit capacity, and then levels off sharply.

15.6 TCP Westwood

TCP Westwood represents an attempt to use the RTT-monitoring strategies of TCP Vegas to address the
high-bandwidth problem; recall that the issue there is to distinguish between congestive and non-congestive
losses. TCP Westwood can also be viewed as a refinement of TCP Renos cwnd=cwnd/2 strategy, which is
a greater drop than necessary if the queue capacity at the bottleneck router is less than the transit capacity.
As in TCP Vegas, the sender keeps a continuous estimate of bandwidth, BWE, and estimates RTTnoLoad by
RTTmin . The minimum window size to keep the bottleneck link busy is, again as in TCP Vegas, BWE
RTTnoLoad . In TCP Vegas, BWE was calculated as cwnd/RTT; we will continue to use this for the time
being but in fact TCP Westwood has used a wide variety of other algorithms, some of which are discussed
in the following subsection, to infer the available average bandwidth from the returning ACKs.
The core TCP Westwood innovation is to, on loss, reduce cwnd as follows:
cwnd = max(cwnd/2, BWERTTnoLoad ) if cwnd > BWERTTnoLoad
no change, if cwnd BWERTTnoLoad
The product BWERTTnoLoad represents what the sender believes is its current share of the transit capacity of the path. This product represents how many packets can be in transit (rather than in queues) at
the current bandwidth BWE. The RTTnoLoad estimate as RTTmin is relatively constant, but BWE may be
markedly reduced in the presence of competing traffic. A TCP Westwood sender never drops cwnd below
what it believes to be the current transit capacity for the path.

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Consider again the classic TCP Reno sawtooth behavior:

cwnd alternates between cwndmin and cwndmax = 2cwndmin .
cwndmax transit_capacity + queue_capacity (or at least the senders share of these)
As we saw in 13.7 TCP and Bottleneck Link Utilization, if transit_capacity < cwndmin , then Reno does
a reasonably good job keeping the bottleneck link saturated. However, if transit_capacity > cwndmin , then
when Reno drops to cwndmin , the bottleneck link is not saturated until cwnd climbs to transit_capacity. For
high-speed networks, this latter case is the more likely one.
Westwood, on the other hand, would in that situation reduce cwnd only to transit_capacity, a smaller reduction. Thus TCP Westwood potentially better handles a wide range of router queue capacities. For bottleneck
routers where the queue capacity is small compared to the transit capacity, TCP Westwood would have a
higher, finer-pitched sawtooth than TCP Reno: the teeth would oscillate between the network ceiling (=
queue+transit) and the transit_capacity, versus Renos oscillation between the network ceiling and half the
ceiling.
In the event of a non-congestive (noise-related) packet loss, if it happens that cwnd is less than transit_capacity then TCP Westwood does not reduce the window size at all. That is, non-congestive losses
with cwnd < transit_capacity have no effect. When cwnd > transit_capacity, losses reduce cwnd only to
transit_capacity, and thus the link stays saturated.
In the large-cwnd, high-bandwidth case, non-congestive packet losses can easily lower the TCP Reno cwnd
to well below what is necessary to keep the bottleneck link saturated. In TCP Westwood, on the other hand,
the average cwnd may be lower than it would be without the non-congestive losses, but it will be high
enough to keep the bottleneck link saturated.
TCP Westwood uses BWERTTnoLoad as a floor for reducing cwnd. TCP Vegas shoots to have the actual
cwnd be just a few packets above this.
TCP Westwood is not any more aggressive than TCP Reno at increasing cwnd in no-loss situations. So
while it handles the non-congestive-loss part of the high-bandwidth TCP problem very well, it does not
particularly improve the ability of a sender to take advantage of a sudden large rise in the network ceiling.
TCP Westwood is also potentially very effective at addressing the lossy-link problem, as most noncongestive losses would result in no change to cwnd.

15.6.1 ACK Compression and Westwood+

So far, we have been assuming that ACKs never encounter queuing delays. They in fact will not, if they
are traveling in the reverse direction from all data packets. But while this scenario covers any single-sender
model and also systems of two or more competing senders, real networks have more complicated traffic
patterns, and returning ACKs from an AB data flow can indeed experience queuing delays if there is
third-party traffic along some link in the BA path.
Delay in the delivery of ACKs, leading to clustering of their arrival, is known as ACK compression; see
[ZSC91] and [JM92] for examples. ACK compression causes two problems. First, arriving clusters of ACKs
trigger corresponding bursts of data transmissions in sliding-windows senders; the end result is an uneven
data-transmission rate. Normally the bottleneck-router queue can absorb an occasional burst; however, if
the queue is nearly full such bursts can lead to premature or otherwise unexpected losses.

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The second problem with late-arriving ACKs is that they can lead to inaccurate or fluctuating measurements
of bandwidth, upon which both TCP Vegas and TCP Westwood depend. For example, if bandwidth is
estimated as cwnd/RTT, late-arriving ACKs can lead to inaccurate calculation of RTT. The original TCP
Westwood strategy was to estimate bandwidth from the spacing between consecutive ACKs, much as is
done with the packet-pairs technique (14.2.6 Packet Pairs) but smoothed with a suitable running average.
This strategy turned out to be particularly vulnerable to ACK-compression errors.
For TCP Vegas, ACK compression means that occasionally the senders cwnd may fail to be decremented
by 1; this does not appear to be a significant impact, perhaps because cwnd is changed by at most 1 each
RTT. For Westwood, however, if ACK compression happens to be occurring at the instant of a packet loss,
then a resultant transient overestimation of BWE may mean that the new post-loss cwnd is too large; at a
point when cwnd was supposed to fall to the transit capacity, it may fail to do so. This means that the sender
has essentially taken a congestion loss to be non-congestive, and ignored it. The influence of this ignored
loss will persist through the much-too-high value of cwnd until the following loss event.
To fix these problems, TCP Westwood has been amended to Westwood+, by increasing the time interval over
which bandwidth measurements are made and by inclusion of an averaging mechanism in the calculation of
BWE. Too much smoothing, however, will lead to an inaccurate BWE just as surely as too little.
Suitable smoothing mechanisms are given in [FGMPC02] and [GM03]; the latter paper in particular examines several smoothing algorithms in terms of their resistance to aliasing effects: the tendency for intermittent measurement of a periodic signal (the returning ACKs) to lead to much greater inaccuracy than might
initially be expected. One smoothing filter suggested by [GM03] is to measure BWE only over entire RTTs,
and then to keep a cumulative running average as follows, where BWMk is the measured bandwidth over
the kth RTT:
BWEk = BWEk-1 + (1)BWMk
A suggested value of is 0.9.

15.7 TCP Veno

TCP Veno ([FL03]) is a synthesis of TCP Vegas and TCP Reno, which attempts to use the RTT-monitoring
ideas of TCP Vegas while at the same time remaining about as aggressive as TCP Reno in using queue
capacity. TCP Veno has generally been presented as an option to address TCPs lossy-link problem, rather
than the high-bandwidth problem per se.
A TCP Veno sender estimates the number N of packets likely in the bottleneck queue as Nqueue = cwnd BWERTTnoLoad , like TCP Vegas. TCP Veno then modifies the TCP Reno congestion-avoidance rule as
follows, where the parameter , representing the queue-utilization value at which TCP Veno slows down,
might be around 5.
if Nqueue <, cwnd = cwnd + 1 each RTT
if Nqueue , cwnd = cwnd + 0.5 each RTT
The above strategy makes cwnd growth less aggressive once link saturation is reached, but does continue
to increase cwnd (half as fast as TCP Reno) until the queue is full and congestive losses occur.
When a packet loss does occur, TCP Veno uses its current value of Nqueue to attempt to distinguish between
non-congestive and congestive loss, as follows:

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if Nqueue <, the loss is probably not due to congestion; set cwnd = (4/5)cwnd
if Nqueue , the loss probably is due to congestion; set cwnd = (1/2)cwnd as usual
The idea here is that most router queues will have a total capacity much larger than , so a loss with fewer
than likely does not represent a queue overflow. Note that, by comparison, TCP Westwood leaves cwnd
unchanged if it thinks the loss is not due to congestion, and its threshold for making that determination is
Nqueue =0.
If TCP Veno encounters a series of non-congestive losses, the above rules make it behave like AIMD(1,0.8).
Per the analysis in 14.7 AIMD Revisited, this is equivalent to AIMD(2,0.5); this means TCP Veno will
be about twice as aggressive as TCP Reno in recovering from non-congestive losses. This would provide
a definite improvement in lossy-link situations with modest bandwidthdelay products, but may not be
enough to make a major dent in the high-bandwidth problem.

15.8 TCP Hybla

TCP Hybla ([CF04]) has one very specific focus: to address the TCP satellite problem (3.5.2 Satellite
Internet) of very long RTTs. TCP Hybla selects a more-or-less arbitrary reference RTT, called RTT0 , and
attempts to scale TCP Reno so as to behave like a TCP Reno connection with an RTT of RTT0 . In the paper
[CF04] the authors suggest RTT0 = 25ms.
Suppose a TCP Reno connection has at a loss event at time t0 reduced cwnd to cwndmin . TCP Reno will
then increment cwnd by 1 for each RTT, until the next loss event. This Reno behavior can be equivalently
expressed in terms of the current time t as follows:
cwnd = (tt0 )/RTT + cwndmin
What TCP Hybla does is to use the above formula after replacing the actual RTT (or RTTnoLoad ) with RTT0 .
Equivalently, TCP Hybla defines the ratio of the two RTTs as = RTT/RTT0 , and then after each windowful
(each time interval of length RTT) increments cwnd by 2 instead of by 1. In the event that RTT < RTT0 ,
is set to 1, so that short-RTT connections are not penalized.
Because cwnd now increases each RTT by 2 , which can be relatively large, there is a good chance that
when the network ceiling is reached there will be a burst of losses of size ~2 . Therefore, TCP Hybla
strongly recommends that the receiving end support SACK TCP, so as to allow faster recovery from multiple
packet losses. Another recommended feature is the use of TCP Timestamps; this is a standard TCP option
that allows the sender to include its own timestamp in each data packet. The receiver is to echo back the
timestamp in the corresponding ACK, thus allowing more accurate measurement by the receiver of the
actual RTT. Finally, to further avoid having these relatively large increments to cwnd result in multiple
packet losses, TCP Hybla recommends some form of pacing to smooth out the actual transmission times.
Rather than sending out four packets upon receipt of an ACK, for example, we might estimate the time T to
the next transmission batch (eg when the next ACK arrives) and send the packets at intervals of T/4.
TCP Hybla applies a similar -fold scaling mechanism to threshold slow start, when a value for ssthresh
is known. But the initial unbounded slow start is much more difficult to scale. Scaling at that point would
mean repeatedly doubling cwnd and sending out flights of packets, before any ACKs have been received;
this would likely soon lead to congestive collapse.

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15.9 TCP Illinois

The general idea behind TCP Illinois, described in [LBS06], is to use the usual AIMD(,) strategy but to
have = (RTT) be a decreasing function of the current RTT, rather than a constant. When the queue is
empty and RTT is equal to RTTnoLoad , then will be large, and cwnd will increase rapidly. Once RTT starts
to increase, however, will decrease rapidly, and the cwnd growth will level off. This leads to the same
kind of concave cwnd graph as we saw above in FAST TCP; a consequence of this is that for most of the
time between consecutive loss events cwnd is large enough to keep the bottleneck link close to saturated,
and so to keep throughput high.
The actual () function is not of RTT, but rather of delay, defined to be RTT RTTnoLoad . As with TCP
Vegas, RTTnoLoad is estimated by RTTmin . As a connection progresses, the sender maintains continually
updated values not only for RTTmin but also for RTTmax . The sender then sets delaymax to be RTTmax
RTTmin .
We are now ready to define (delay). We first specify the highest value of , max , and the lowest, min . In
[LBS06] these are 10.0 and 0.1 respectively; in my linux 3.5 kernel they are 10.0 and 0.3. We also define
delaythresh to be 0.01delaymax (the 0.01 is another tunable parameter). We then define (delay) as follows
(delay) = max if delay delaythresh
(delay) = k1 /(delay+k2 ) if delaythresh delay delaymax
where k1 and k2 are chosen so that, for the lower formula, (delaythresh ) = max and (delaymax ) = min . In
case there is a sudden spike in delay, delaymax is updated before the above is evaluated, so we always have
delay delaymax . Here is a graph:

Whenever RTT = RTTnoLoad , delay=0 and so (delay) = max . However, as soon as queuing delay just
barely starts to begin, we will have delay > delaythresh and so (delay) begins to fall rather precipitously
to min . The value of (delay) is always positive, though, so cwnd will continue to increase (unlike TCP
Vegas) until a congestive loss eventually occurs. However, at that point the change in cwnd is very small,
which minimizes the probability that multiple packets will be lost.
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Note that, as with FAST TCP, the increase in delay is used to trigger the reduction in .
TCP Illinois also supports having be a decreasing function of delay, so that (small_delay) might be
0.2 while (larger_delay) might match TCP Renos 0.5. However, the authors of [LBS06] explain that
the adaptation of as a function of average queuing delay is only relevant in networks where there are
non-congestion-related losses, such as wireless networks or extremely high speed networks.

15.10 H-TCP
H-TCP, or TCP-Hamilton, is described in [LSL05]. Like Highspeed-TCP it primarily allows for faster
growth of cwnd; unlike Highspeed-TCP, the cwnd increment is determined not by the size of cwnd but
by the elapsed time since the previous loss event. The threshold for accelerated cwnd growth is generally
set to be 1.0 seconds after the most recent loss event. Using an RTT of 50 ms, that amounts to 20 RTTs,
suggesting that when cwndmin is less than 20 then H-TCP behaves very much like TCP Reno.
The specific H-TCP acceleration rule first defines a time threshold tL . If t is the elapsed time in seconds
since the previous loss event, then for ttL the per-RTT window-increment is 1. However, for t>tL we
define
(t) = 1 + 10(ttL ) + (ttL )2 /4
We then increment cwnd by (t) after each RTT, or, equivalently, by (t)/cwnd after each received ACK.
At t=tL +1 seconds (nominally 2 seconds), is 12. The quadratic term dominates the linear term when ttL
> 40. If RTT = 50 ms, that is 800 RTTs.
Even if cwnd is very large, growth is at the same rate as for TCP Reno until t>tL ; one consequence of this
is that, at least in the first second after a loss event, H-TCP competes fairly with TCP Reno, in the sense that
both increase cwnd at the same absolute rate. H-TCP starts from scratch after each packet loss, and does
not re-enter its high-speed mode, even if cwnd is large, until after time tL .
A full H-TCP implementation also adjusts the multiplicative factor as follows (the paper [LSL05] uses
to represent what we denote by 1). The RTT is monitored, as with TCP Vegas. However, the RTT
increase is not used for per-packet or per-RTT adjustments; instead, these measurements are used after each
loss event to update so as to have
1 = RTTmin /RTTmax
The value 1 is capped at a maximum of 0.8, and at a minimum of 0.5. To see where the ratio above comes
from, first note that RTTmin is the usual stand-in for RTTnoLoad , and RTTmax is, of course, the RTT when the
bottleneck queue is full. Therefore, by the reasoning in 6.3.2 RTT Calculations, equation 5, 1 is the ratio
transit_capacity / (transit_capacity + queue_capacity). At a congestion event involving a single uncontested
flow we have cwnd = transit_capacity + queue_capacity, and so after reducing cwnd to (1)cwnd, we
have cwndnew = transit_capacity, and hence (as in 13.7 TCP and Bottleneck Link Utilization) the bottleneck
link will remain 100% utilized after a loss. The cap on 1 of 0.8 means that if the queue capacity is smaller
than a quarter of the transit capacity then the bottleneck link will experience some idle moments.
When is changed, H-TCP also adjusts to = 2(t) so as to improve fairness with other H-TCP
connections with different current values of .

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15.11 TCP CUBIC

TCP Cubic attempts, like Highspeed TCP, to solve the problem of efficient TCP transport when
bandwidthdelay is large. TCP Cubic allows very fast window expansion; however, it also makes attempts
to slow the growth of cwnd sharply as cwnd approaches the current network ceiling, and to treat other TCP
connections fairly. Part of TCP Cubics strategy to achieve this is for the window-growth function to slow
down (become concave) as the previous network ceiling is approached, and then to increase rapidly again
(become convex) if this ceiling is surpassed without losses. This concave-then-convex behavior mimics the
graph of the cubic polynomial cwnd = t3 , hence the name (TCP Cubic also improves an earlier TCP version
known as TCP BIC).

As mentioned above, TCP Cubic is currently (2013) the default linux congestion-control implementation.
TCP Cubic is documented in [HRX08]. TCP Cubic is not described in an RFC, but there is an Internet Draft
http://tools.ietf.org/id/draft-rhee-tcpm-cubic-02.txt.
TCP Cubic has a number of interrelated features, in an attempt to address several TCP issues:
Reduction in RTT bias
TCP Friendliness when most appropriate
Rapid recovery of cwnd following its decrease due to a loss event, maximizing throughput
Optimization for an unchanged network ceiling (corresponding to cwndmax )
Rapid expansion of cwnd when a raised network ceiling is detected
The eponymous cubic polynomial y=x3 , appropriately shifted and scaled, is used to determine changes in
cwnd. No special algebraic properties of this polynomial are used; the point is that the curve, while steadily
increasing, is first concave and then convex; the authors of [HRX08] write [t]he choice for a cubic function
is incidental and out of convenience. This y=x3 polynomial has an inflection point at x=0 where the tangent
line is horizontal; this is the point where the graph changes from concave to convex.
We start with the basic outline of TCP Cubic and then consider some of the bells and whistles. We assume
a loss has just occurred, and let Wmax denote the value of cwnd at the point when the loss was discovered.
TCP Cubic then sets cwnd to 0.8Wmax ; that is, TCP Cubic uses = 0.2. The corresponding for TCPFriendly AIMD(,) would be =1/3, but TCP Cubic uses this only in its TCP-Friendly adjustment,
below.

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We now define a cubic polynomial W(t), a shifted and scaled version of w=t3 . The parameter t represents
the elapsed time since the most recent loss, in seconds. At time t>0 we set cwnd = W(t). The polynomial
W(t), and thus the cwnd rate of increase, as in TCP Hybla, is no longer tied to the connections RTT; this is
done to reduce if not eliminate the RTT bias that is so deeply ingrained in TCP Reno.
We want the function W(t) to pass through the point representing the cwnd just after the loss, that is, t,W
= 0,0.8Wmax . We also want the inflection point to lie on the horizontal line y=Wmax . To fully determine
the curve, it is at this point sufficient to specify the value of t at this inflection point; that is, how far
horizontally W(t) must be stretched. This horizontal distance from t=0 to the inflection point is represented
by the constant K in the following equation; W(t) returns to its pre-loss value Wmax at t=K. C is a second
constant.
W(t) = C(tK)3 + Wmax
It suffices algebraically to specify either C or K; the two constants are related by the equation obtained by
plugging in t=0. K changes with each loss event, but it turns out that the value of C can be constant, not only
for any one connection but for all TCP Cubic connections. TCP Cubic specifies for C the ad hoc value 0.4;
we can then set t=0 and, with a bit of algebra, solve to obtain
K = (Wmax /2)1/3 seconds
If Wmax = 250, for example, K=5; if RTT = 100 ms, this is 50 RTTs.
When each ACK arrives, TCP Cubic records the arrival time t, calculates W(t), and sets cwnd = W(t). At
the next packet loss the parameters of W(t) are updated.
If the network ceiling does not change, the next packet loss will occur when cwnd again reaches the same
Wmax ; that is, at time t=K after the previous loss. As t approaches K and the value of cwnd approaches
Wmax , the curve W(t) flattens out, so cwnd increases slowly.
This concavity of the cubic curve, increasing rapidly but flattening near Wmax , achieves two things. First,
throughput is boosted by keeping cwnd close to the available path transit capacity. In 13.7 TCP and
Bottleneck Link Utilization we argued that if the path transit capacity is large compared to the bottleneck
queue capacity (and this is the case for which TCP Cubic was designed), then TCP Reno averages 75%
utilization of the available bandwidth. The bandwidth utilization increases linearly from 50% just after a
loss event to 100% just before the next loss. In TCP Cubic, the initial rapid rise in cwnd following a loss
means that the average will be much closer to 100%. Another important advantage of the flattening is that
when cwnd is finally incremented to the point of loss, it likely is just over the network ceiling; the connection
has an excellent chance that only one or two packets are lost rather than a large burst. This facilitates the
NewReno Fast Recovery algorithm, which TCP Cubic still uses if the receiver does not support SACK TCP.
Once t>K, W(t) becomes convex, and in fact begins to increase rapidly. In this region, cwnd > Wmax , and
so the sender knows that the network ceiling has increased since the previous loss. The TCP Cubic strategy
here is to probe aggressively for additional capacity, increasing cwnd very rapidly until the new network
ceiling is encountered. The cubic increase function is in fact quite aggressive when compared to any of the
other TCP variants discussed here, and time will tell what strategy works best. As an example in which
the TCP Cubic approach seems to pay off, let us suppose the current network ceiling is 2,000 packets, and
then (because competing connections have ended) increases to 3,000. TCP Reno would take 1,000 RTTs for
cwnd to reach the new ceiling, starting from 2,000; if one RTT is 50 ms that is 50 seconds. To find the time
t-K that TCP Cubic will need to increase cwnd from 2,000 to 3,000, we solve 3000 = W(t) = C(tK)3 +
2000, which works out to t-K 13.57 seconds (recall 2000 = W(K) here).
The constant C=0.4 is determined empirically. The cubic inflection point occurs at t = K = (Wmax /C)1/3 .
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A larger C reduces the time K between the a loss event and the next inflection point, and thus the time
between consecutive losses. If Wmax = 2000, we get K=10 seconds when =0.2 and C=0.4. If the RTT were
50 ms, 10 seconds would be 200 RTTs.
For TCP Reno, on the other hand, the interval between adjacent losses is Wmax /2 RTTs. If we assume a
specific value for the RTT, we can compare the Reno and Cubic time intervals between losses; for an RTT
of 50 ms we get
Wmax
2000
250
54

Reno
50 sec
6.2 sec
1.35 sec

Cubic
10 sec
5 sec
3 sec

For smaller RTTs, the basic TCP Cubic strategy above runs the risk of being at a competitive disadvantage
compared to TCP Reno. For this reason, TCP Cubic makes a TCP-Friendly adjustment in the windowsize calculation: on each arriving ACK, cwnd is set to the maximum of W(t) and the window size that TCP
Reno would compute. The TCP Reno calculation can be based on an actual count of incoming ACKs, or be
based on the formula (1-)Wmax + t/RTT.
Note that this adjustment is only half-friendly: it guarantees that TCP Cubic will not choose a window
size smaller than TCP Renos, but places no restraints on the choice of a larger window size.
A consequence of the TCP-Friendly adjustment is that, on networks with modest bandwidthdelay products, TCP Cubic behaves exactly like TCP Reno.
TCP Cubic also has a provision to detect if a given Wmax is lower than the previous value, suggesting
increasing congestion; in this situation, cwnd is lowered by an additional factor of 1/2. This is known as
fast convergence, and helps TCP Cubic adapt more quickly to reductions in available bandwidth.
The following graph is taken from [RX05], and shows TCP Cubic connections competing with each other
and with TCP Reno.

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The diagram shows four connections, all with the same RTT. Two are TCP Cubic and two are TCP Reno.
The red connection, cubic-1, was established and with a maximum cwnd of about 4000 packets when the
other three connections started. Over the course of 200 seconds the two TCP Cubic connections reach a fair
equilibrium; the two TCP Reno connections reach a reasonably fair equilibrium with one another, but it is
much lower than that of the TCP Cubic connections.
On the other hand, here is a graph from [LSM07], showing the result of competition between two flows using
an earlier version of TCP Cubic over a low-speed connection. One connection has an RTT of 160ms and the
other has an RTT a tenth that. The bottleneck bandwidth is 1 Mbit/sec, meaning that the bandwidthdelay
product for the 160ms connection is 13-20 packets (depending on the packet size used).

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Note that the longer-RTT connection (the solid line) is almost completely starved, once the shorter-RTT
connection starts up at T=100. This is admittedly an extreme case, and there have been more recent fixes to
TCP Cubic, but it does serve as an example of the need for testing a wide variety of competition scenarios.

15.12 Epilog
TCP Renos core congestion algorithm is based on algorithms in Jacobson and Karels 1988 paper [JK88],
now twenty-five years old. There are concerns both that TCP Reno uses too much bandwidth (the greediness
issue) and that it does not use enough (the high-bandwidth-TCP problem).
There are also broad changes in TCP usage patterns. Twenty years ago, the vast majority of all TCP traffic
represented downloads from major servers. Today, over half of all Internet TCP traffic is peer-to-peer
rather than server-to-client. The rise in online video streaming creates new demands for excellent TCP
real-time performance.
So which TCP version to use? That depends on circumstances; some of the TCPs above are primarily
intended for relatively specific environments; for example, TCP Hybla for satellite links and TCP Veno
for mobile devices (including wireless laptops). If the sending and receiving hosts are under common
management, and especially if intervening traffic patterns are relatively stable, one can simply make sure
the receiver has what it needs for optimum performance (eg SACK TCP) and run a few simple experiments
to find what works best.
That leaves the question of what TCP to use on a server that is serving up large volumes of data, perhaps to a
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range of disparate hosts and with a wide variety of competing-traffic scenarios. Experimentation works here
too, but likely with a much larger number of trials. There is also the possibility that one eventually finds a
solution that works well, only to discover that it succeeds at the expense of other, competing traffic. These
issues suggest a need for continued research into how to update and improve TCP, and Internet congestionmanagement generally.
Finally, while most new TCPs are designed to hold their own in a Reno world, there is some question that
perhaps we would all be better off with a radical rather than incremental change. Might TCP Vegas be a
better choice, if only the queue-grabbing greediness of TCP Reno could be restrained? Questions like these
are today entirely hypothetical, but it is not impossible to envision an Internet backbone that implemented
non-FIFO queuing mechanisms (18 Queuing and Scheduling) that fundamentally changed the rules of the
game.

15.13 Exercises
1. How would TCP Vegas respond if it estimated RTTnoLoad = 100ms, with a bandwidth of 1 packet/ms,
and then due to a routing change the RTTnoLoad increased to 200ms without changing the bandwidth? What
cwnd would be chosen? Assume no competition from other senders.
2. Suppose a TCP Vegas connection from A to B passes through a bottleneck router R. The RTTnoLoad is 50
ms and the bottleneck bandwidth is 1 packet/ms.

(a). If the connection keeps 4 packets in the queue (eg =3, =5), what will RTTactual be? What value of
cwnd will the connection choose? What will be the value of BWE?
(b). Now suppose a competing (non-Vegas) connection keeps 6 packets in the queue to the Vegas
connections 4, eventually meaning that the other connection will have 60% of the bandwidth. What will be
the Vegas connections steady-state values for RTTactual , cwnd and BWE?

3. Suppose a TCP Vegas connection has R as its bottleneck router. The transit capacity is M, and the queue
utilization is currently Q>0 (meaning that the transit path is 100% utilized, although not necessarily by the
TCP Vegas packets). The current TCP Vegas cwnd is cwndV . Show that the number of packets TCP Vegas
calculates are in the queue, queue_use, is
queue_use = cwndV Q/(Q+M)
4. Suppose that at time T=0 a TCP Vegas connection and a TCP Reno connection share the same path, and
each has 100 packets in the bottleneck queue, exactly filling the transit capacity of 200. TCP Vegas uses
=1, =2. By the previous exercise, in any RTT with cwndV TCP Vegas packets and cwndR TCP Reno
packets in flight and cwndV +cwndR >200, Nqueue is cwndV /(cwndV +cwndR ) multiplied by the total queue
utilization cwndV +cwndR 200.
Continue the following table, where T is measured in RTTs, up through the next two RTTs where cwndV
is not decremented; that is, find the next two rows where the TCP Vegas queue share is less than 2. (After
each of these RTTs, cwndV is not decremented.) This can be done either with a spreadsheet or by simple
algebra. Note that the TCP Reno cwndR will always increment.

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T
0
1
2
3
4
5
6

cwndV
100
101
102
101
101
100
99

cwndR
100
101
102
103
104
105
106

TCP Vegas queue share

0
1
2
(101/204)x4 = 1.980 <
Vegas has (101/205)5 = 2.463 packets in queue
Vegas has (100/205)5 = 2.435 packets in queue
(99/205)5 = 2.439

This exercise attempts to explain the linear decrease in the TCP Vegas graph in the diagram in 16.5 TCP
Reno versus TCP Vegas. Competition with TCP Reno means not only that cwndV stops increasing, but in
fact it decreases by 1 most RTTs.
5. Suppose that, as in the previous exercise, a FAST TCP connection and a TCP Reno connection share
the same path, and at T=0 each has 100 packets in the bottleneck queue, exactly filling the transit capacity
of 200. The FAST TCP parameter is 0.5. The FAST TCP and TCP Reno connections have respective
cwnds of cwndF and cwndR . You may use the fact that, as long as the queue is nonempty, RTT/RTTnoLoad
= (cwndF +cwndR )/200.
Find the value of cwndF at T=40, where T is counted in units of 20 ms until T = 40, using =4, =10 and
=30. Assume RTT 20 ms as well. Use of a spreadsheet is recommended. The table here uses =10.
T
0
1
2
3
4

cwndF
100
105
108.47
110.77
112.20

cwndR
100
101
102
103
104

6. Suppose A sends to B as in the layout below. The packet size is 1 KB and the bandwidth of the bottleneck
RB link is 1 packet / 10 ms; returning ACKs are thus normally spaced 10 ms apart. The RTTnoLoad for the
AB path is 200 ms.
A

However, large amounts of traffic are also being sent from C to A; the bottleneck link for that path is RA
with bandwidth 1 KB / 5 ms. The queue at R for the RA link has a capacity of 40 KB. ACKs are 50 bytes.

(a). What is the maximum possible arrival time difference on the AB path for ACK[0] and ACK[20], if
there are no queuing delays at R in the AB direction? ACK[0] should be forwarded immediately by R;
ACK[20] should have to wait for 40 KB at R
(b). What is the minimum possible arrival time difference for the same ACK[0] and ACK[20]?

7. Suppose a TCP Veno and a TCP Reno connection compete along the same path; there is no other traffic.
Both start at the same time with cwnds of 50; the total transit capacity is 160. Both share the next loss
event. The bottleneck routers queue capacity is 60 packets; sometimes the queue fills and at other times it

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is empty. TCP Venos parameter is zero, meaning that it shifts to a slower cwnd increment as soon as the
queue just begins filling.

(a). In how many RTTs will the queue begin filling?

(b). At the point the queue is completely filled, how much larger will the Reno cwnd be than the Veno
cwnd?

8. Suppose two connections use TCP Hybla. They do not compete. The first connection has an RTT of 100
ms, and the second has an RTT of 1000 ms. Both start with cwndmin = 0 (literally meaning that nothing is
sent the first RTT).

(a). How many packets are sent by each connection in four RTTs (involving three cwnd increases)?
(b). How many packets are sent by each connection in four seconds? Recall 1+2+3+...+N = N(N+1)/2.

9. Suppose that at time T=0 a TCP Illinois connection and a TCP Reno connection share the same path,
and each has 100 packets in the bottleneck queue, exactly filling the transit capacity of 200. The respective
cwnds are cwndI and cwndR . The bottleneck queue capacity is 100.
Find the value of cwndI at T=50, where T is the number of elapsed RTTs. At this point cwndR is, of course,
150.
T
0
1
2

cwndI
100
101
?

cwndR
100
101
102

You may assume that the delay, RTT RTTnoLoad , is proportional to queue_utilization = cwndI +cwndR
200. Using this expression to represent delay, delaymax = 100 and so delaythresh = 1. When calculating
(delay), assume max = 10 and min = 0.1.
10. Assume that a TCP connection has an RTT of 50 ms, and the time between loss events is 10 seconds.

(a). For a TCP Reno connection, what is the bandwidthdelay product?

(b). For an H-TCP connection, what is the bandwidthdelay product?

11. For each of the values of Wmax below, find the change in TCP Cubics cwnd over one 100 ms RTT at
each of the following points:
i. Immediately after the previous loss event, when t = 0.
ii. At the midpoint of the tooth, when t=K/2
iii. At the point when cwnd has returned to Wmax , at t=K

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(a). Wmax = 250 (making K=5)

(b). Wmax = 2000 (making K=10)

12. Suppose a TCP Reno connection is competing with a TCP Cubic connection. There is no other traffic.
All losses are synchronized. In this setting, once the steady state is reached, the cwnd graphs for one tooth
will look like this:

Let c be the maximum cwnd of the TCP Cubic connection (c=Wmax ) and let r be the maximum of the TCP
Reno connection. Let M be the network ceiling, so a loss occurs when c+r reaches M. The width of the
tooth for TCP Reno is (r/2)RTT, where RTT is measured in seconds; the width of the TCP Cubic tooth is
(c/2)1/3 . For the examples here, ignore the TCP-Friendly feature of TCP Cubic.

(a). If M = 200 and RTT = 50 ms = 0.05 sec, show that at the steady state r 130.4 and c = Mr 69.6.
(b). Find equilibrium r and c (to the nearest integer) for M=1000 and RTT = 50 ms. Hint: use of a
spreadsheet or scripting language makes trial-and-error quite practical.
(c). Find equilibrium r and c for M = 1000 and RTT = 100 ms.

13. Suppose a TCP Westwood connection has the path A R1 R2

B. The R1R2 bandwidth is 1
packet/ms, and RTTnoLoad is 200 ms. The queue capacity at R1 is unlimited. At T=10, with cwnd = 300 so
the queue at R1 has 100 AB packets, a second C
R1 R2
D connection starts up, sending 1 packet
every 2 ms at a higher priority than the A B connection (so that the bottleneck link share of the bandwidth
for the A B connection immediately falls to 1 packet / 2 ms). At that same time, a single AB packet is
lost.

(a). Suppose A responds to the loss using the original BWE of 1 packet/ms. How will A update its cwnd?

(b). How would A update its cwnd in response to the loss if it used a BWE of 1 packet / 2 ms?
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(c). Suppose A calculates BWE as cwnd/RTT, where RTT is measured by monitoring one packet at a time;
A waits until the packet returns before measuring a new RTT. The current RTT-measuring packet happens
to be the one sent just before the packet that was eventually lost. What value for BWE will A have at the
time the loss is detected? Assume A discovers the loss through Fast Retransmit, just over one RTT later.
Between the idea
And the reality
Between the motion
And the act
Falls the Shadow
TS Eliot, The Hollow Men
Try to leave out the part that readers tend to skip.
Elmore Leonard, 10 Rules for Writing

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16 NETWORK SIMULATIONS: NS-2

In previous chapters, especially 14 Dynamics of TCP Reno, we have at times made simplifying assumptions
about TCP Reno traffic. In the present chapter we will look at actual TCP behavior, through simulation,
enabling us to explore the accuracy of some of these assumptions. The primary goal is to provide comparison
between idealized TCP behavior and the often-messier real behavior; a secondary goal is perhaps to shed
light on some of the issues involved in simulation. For example, in the discussion in 16.3 Two TCP Senders
Competing of competition between TCP Reno connections with different RTTs, we address several technical
issues that affect the relevance of simulation results.
Parts of this chapter may serve as a primer on using the ns-2 simulator, though a primer focused on the goal
of illuminating some of the basic operation and theory of TCP through experimentation. However, some of
the outcomes described may be of interest even to those not planning on designing their own simulations.
Simulation is almost universally used in the literature when comparing different TCP flavors for effective
throughput (for example, the graphs excerpted in 15.11 TCP CUBIC were created through simulation).
We begin this chapter by looking at a single connection and analyzing how well the TCP sawtooth utilizes
the bottleneck link. We then turn to competition between two TCP senders. The primary competition
example here is between TCP Reno connections with different RTTs. This example allows us to explore
the synchronized-loss hypothesis (14.3.4 Synchronized-Loss Hypothesis) and to introduce phase effects,
transient queue overflows, and other unanticipated TCP behavior. We also introduce some elements of
designing simulation experiments. The second example compares TCP Reno and TCP Vegas. We close
with a straightforward example of a wireless simulation.

16.1 The ns-2 simulator

The tool used for much research-level network simulations is ns, for network simulator and originally developed at the Information Sciences Institute. The ns simulator grew out of the REAL simulator developed
by Srinivasan Keshav [SK88]; later development work was done by the Network Research Group at the
Lawrence Berkeley National Laboratory.
We will describe in this chapter the ns-2 simulator, hosted at www.isi.edu/nsnam/ns. There is now also
an ns-3 simulator, available at nsnam.org. Because ns-3 is not backwards-compatible with ns-2 and the
programming interface has changed considerably, we take the position that ns-3 is an entirely different
package, though one likely someday to supercede ns-2 entirely. While ns-3 will eventually also be included
in this book, its use is arguably quite a bit more complicated for beginners, and the particular simulation
examples presented here are well-suited to ns-2. While ns-3 supports more complex and realistic modeling,
and is the tool of choice for serious research, this added complexity comes at a price in terms of configuration
and programming. The standard ns-2 tracefile format is also quite easy to work with.
Research-level use of ns-2 often involves building new modules in C++, and compiling them in to the
system. For our purposes here, the stock ns-2 distribution is sufficient. The simplest way to install ns-2 is
probably with the allinone distribution, which does still require compiling but comes with a very simple
install script. (As of 2014, the ns-allinone-2.35 version appeared to compile only with gcc/g++ no more
recent than version 4.6.)

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The native environment for ns-2 (and ns-3) is linux. Perhaps the simplest approach for Windows users is to
install a linux virtual machine, and then install ns-2 under that. It is also possible to compile ns-2 under the
Cygwin system; an older version of ns-2 may still be available as a Cygwin binary.
To create an ns-2 simulation, we need to do the following (in addition to a modest amount of standard
housekeeping).
define the network topology, including all nodes, links and router queuing rules
create some TCP (or UDP) connections, called Agents, and attach them to nodes
create some Applications usually FTP for bulk transfer or telnet for intermittent random packet
generation and attach them to the agents
start the simulation
Once started, the simulation runs for the designated amount of time, driven by the packets generated by
the Application objects. As the simulated applications generate packets for transmission, the ns-2 system
calculates when these packets arrive and depart from each node, and generates simulated acknowledgment
packets as appropriate. Unless delays are explicitly introduced, node responses such as forwarding a
packet or sending an ACK are instantaneous. That is, if a node begins sending a simulated packet from
node N1 to N2 at time T=1.000 over a link with bandwidth 60 ms per packet and with propagation delay
200 ms, then at time T=1.260 N2 will have received the packet. N2 will then respond at that same instant, if
a response is indicated, eg by enqueuing the packet or by forwarding it if the queue is empty.
Ns-2 does not necessarily require assigning IP addresses to every node, though this is possible in more
elaborate simulations.
Advanced use of ns-2 (and ns-3) often involves the introduction of randomization; for example, we will in
16.3 Two TCP Senders Competing introduce both random sliding-windows delays and traffic generators
that release packets at random times. While it is possible to seed the random-number generator so that
different runs of the same experiment yield different outcomes, we will not do this here, so the randomnumber generator will always produce the same sequence. A consequence is that the same ns-2 script
should yield exactly the same result each time it is run.

16.1.1 Using ns-2

The scripting interface for ns-2 uses the language Tcl, pronounced tickle; more precisely it is object-Tcl,
or OTcl. For simple use, learning the general Tcl syntax is not necessary; one can proceed quite successfully
by modifying standard examples.
The network simulation is defined in a Tcl file, perhaps sim.tcl; to run the simulation one then types
ns sim.tcl
The result of running the ns-2 simulation will be to create some files, and perhaps print some output. The
most common files created are the ns-2 trace file perhaps sim.tr which contains a record for every
packet arrival, departure and queue event, and a file sim.nam for the network animator, nam, that allows
visual display of the packets in motion (the ns-3 version of nam is known as NetAnim). The sliding-windows
video in 6.2 Sliding Windows was created with nam.
Within Tcl, variables can be assigned using the set command. Expressions in [ ...
Numeric expressions must also use the expr command:
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set foo [expr $foo + 1]

As in unix-style shell scripting, the value of a variable X is $X; the name X (without the $) is used when
setting the value (in Perl and PHP, on the other hand, many variable names begin with $, which is included
both when evaluating and setting the variable). Comments are on lines beginning with the # character.
Comments can not be appended to a line that contains a statement (although it is possible first to start a new
logical line with ;).
Objects in the simulation are generally created using built-in constructors; the constructor in the line below
is the part in the square brackets (recall that the brackets must enclose an expression to be evaluated):
set tcp0 [new Agent/TCP/Reno]

Object attributes can then be assigned values; for example, the following sets the data portion of the packets
in TCP connection tcp0 to 960 bytes:
$tcp0 set packetSize_ 960

Object attributes are retrieved using set without a value; the following assigns variable ack0 the current
value of the ack_ field of tcp0:
set ack0 [$tcp0 set ack_]

The goodput of a TCP connection is, properly, the number of application bytes received. This differs from
the throughput the total bytes sent in two ways: the latter includes both packet headers and retransmitted
packets. The ack0 value above includes no retransmissions; we will occasionally refer to it as goodput
in this sense.

16.2 A Single TCP Sender

For our first script we demonstrate a single sender sending through a router. Here is the topology we will
build, with the delays and bandwidths:

The smaller bandwidth on the RB link makes it the bottleneck. The default TCP packet size in ns2 is 1000 bytes, so the bottleneck bandwidth is nominally 100 packets/sec or 0.1 packets/ms. The
bandwidthRTTnoLoad product is 0.1 packets/ms 120 ms = 12 packets. Actually, the default size of 1000
bytes refers to the data segment, and there are an additional 40 bytes of TCP and IP header. We therefore
set packetSize_ to 960 so the actual transmitted size is 1000 bytes; this makes the bottleneck bandwidth
exactly 100 packets/sec.
We want the router R to have a queue capacity of 6 packets, plus the one currently being transmitted; we
set queue-limit = 7 for this. We create a TCP connection between A and B, create an ftp sender on top
that, and run the simulation for 20 seconds. The nodes A, B and R are named; the links are not.

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The ns-2 default maximum window size is 20; we increase that to 100 with $tcp0 set window_ 100;
otherwise we will see an artificial cap on the cwnd growth (in the next section we will increase this to
65000).
The script itself is in a file basic1.tcl, with the 1 here signifying a single sender.
# basic1.tcl simulation: A---R---B
#Create a simulator object
set ns [new Simulator]
#Open the nam file basic1.nam and the variable-trace file basic1.tr
set namfile [open basic1.nam w]
$ns namtrace-all $namfile
set tracefile [open basic1.tr w]
$ns trace-all $tracefile
#Define a finish procedure
proc finish {} {
global ns namfile tracefile
$ns flush-trace
close $namfile
close $tracefile
exit 0
}
#Create the network nodes
set A [$ns node]
set R [$ns node]
set B [$ns node]
#Create a duplex link between the nodes
$ns duplex-link $A $R 10Mb 10ms DropTail
$ns duplex-link $R $B 800Kb 50ms DropTail
# The queue size at $R is to be 7, including the packet being sent
$ns queue-limit $R $B 7
# some hints for nam
# color packets of flow 0 red
$ns color 0 Red
$ns duplex-link-op $A $R orient right
$ns duplex-link-op $R $B orient right
$ns duplex-link-op $R $B queuePos 0.5
# Create a TCP sending agent and attach it to A
set tcp0 [new Agent/TCP/Reno]
# We make our one-and-only flow be flow 0
$tcp0 set class_ 0
$tcp0 set window_ 100
$tcp0 set packetSize_ 960
$ns attach-agent $A $tcp0

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# Lets trace some variables

$tcp0 attach $tracefile
$tcp0 tracevar cwnd_
$tcp0 tracevar ssthresh_
$tcp0 tracevar ack_
$tcp0 tracevar maxseq_
#Create a TCP receive agent (a traffic sink) and attach it to B
set end0 [new Agent/TCPSink]
$ns attach-agent $B $end0
#Connect the traffic source with the traffic sink
$ns connect $tcp0 $end0
#Schedule the connection data flow; start sending data at T=0, stop at T=10.0
set myftp [new Application/FTP]
$myftp attach-agent $tcp0
$ns at 0.0 "$myftp start"
$ns at 10.0 "finish"
#Run the simulation
$ns run

After running this script, there is no command-line output (because we did not ask for any); however, the
files basic1.tr and basic1.nam are created. Perhaps the simplest thing to do at this point is to view the
animation with nam, using the command nam basic1.nam.
In the animation we can see slow start at the beginning, as first one, then two, then four and then eight
packets are sent. A little past T=0.7, we can see a string of packet losses. This is visible in the animation
as a tumbling series of red squares from the top of Rs queue. After that, the TCP sawtooth takes over;
we alternate between the cwnd linear-increase phase (congestion avoidance), packet loss, and threshold
slow start. During the linear-increase phase the bottleneck link is at first incompletely utilized; once the
bottleneck link is saturated the router queue begins to build.

16.2.1 Graph of cwnd v time

Here is a graph of cwnd versus time, prepared (see below) from data in the trace file basic1.tr:

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Slow start is at the left edge. Unbounded slow start runs until about T=0.75, when a timeout occurs; bounded
slow start runs from about T=1.2 to T=1.8. After that, all losses have been handled with fast recovery (we
can tell this because cwnd does not drop below half its previous peak). The first three teeth following slow
start have heights (cwnd peak values) of 20.931, 20.934 and 20.934 respectively; when the simulation is
extended to 1000 seconds all subsequent peaks have exactly the same height, cwnd = 20.935. The spacing
between the peaks is also constant, 1.946 seconds.
Because cwnd is incremented by ns-2 after each arriving ACK as described in 13.2.1 Per-ACK Responses,
during the linear-increase phase there are a great many data points jammed together; the bunching effect is
made stronger by the choice here of a large-enough dot size to make the slow-start points clearly visible.
This gives the appearance of continuous line segments. Upon close examination, these line segments are
slightly concave, as discussed in 15.3 Highspeed TCP, due to the increase in RTT as the queue fills.
Individual flights of packets can just be made out at the lower-left end of each tooth, especially the first.

16.2.2 The Trace File

To examine the simulation (or, for that matter, the animation) more quantitatively, we turn to a more detailed
analysis of the trace file, which contains records for all packet events plus (because it was requested)
variable-trace information for cwnd_, ssthresh_, ack_ and maxseq_; these were the variables for
which we requested traces in the basic1.tcl file above.
The bulk of the trace-file lines are event records; three sample records are below. (These are in the default event-record format for point-to-point links; ns-2 has other event-record formats for wireless. See
use-newtrace in 16.6 Wireless Simulation below.)
r 0.58616 0 1 tcp 1000 ------- 0 0.0 2.0 28 43
+ 0.58616 1 2 tcp 1000 ------- 0 0.0 2.0 28 43
d 0.58616 1 2 tcp 1000 ------- 0 0.0 2.0 28 43

The twelve event-record fields are as follows:

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1. r for received, d for dropped, + for enqueued, - for dequeued. Every arriving packet is enqueued,
even if it is immediately dequeued. The third packet above was the first dropped packet in the entire
simulation.
2. the time, in seconds.
3. the number of the sending node, in the order of node definition and starting at 0. If the first field was
+, - or d, this is the number of the node doing the enqueuing, dequeuing or dropping. Events
beginning with - represent this node sending the packet.
4. the number of the destination node. If the first field was r, this record represents the packets arrival
at this node.
5. the protocol.
6. the packet size, 960 bytes of data (as we requested) plus 20 of TCP header and 20 of IP header.
7. some TCP flags, here represented as ------- because none of the flags are set. Flags include E
and N for ECN and A for reduction in the advertised winsize.
8. the flow ID. Here we have only one: flow 0. This value can be set via the fid_ variable in the Tcl
source file; an example appears in the two-sender version below. The same flow ID is used for both
directions of a TCP connection.
9. the source node (0.0), in form (node . connectionID). ConnectionID numbers here are simply an
abstraction for connection endpoints; while they superficially resemble port numbers, the node in
question need not even simulate IP, and each connection has a unique connectionID at each end.
ConnectionID numbers start at 0.
10. the destination node (2.0), again with connectionID.
11. the packet sequence number as a TCP packet, starting from 0.
12. a packet identifier uniquely identifying this packet throughout the simulation; when a packet is forwarded on a new link it keeps its old sequence number but gets a new packet identifier.
The three trace lines above represent the arrival of packet 28 at R, the enqueuing of packet 28, and then the
dropping of the packet. All these happen at the same instant.
Mixed in with the event records are variable-trace records, indicating a particular variable has been
changed. Here are two examples from t=0.3833:
0.38333
0.38333

0
0

0
0

2
2

0
0

ack_ 3
cwnd_ 5.000

The format of these lines is

1. time
2. source node of the flow
3. source port (as above, an abstract connection endpoint, not a simulated TCP port)
4. destination node of the flow
5. destination port
6. name of the traced variable
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7. value of the traced variable

The two variable-trace records above are from the instant when the variable cwnd_ was set to 5. It was
initially 1 at the start of the simulation, and was incremented upon arrival of each of ack0, ack1, ack2 and
ack3. The first line shows the ack counter reaching 3 (that is, the arrival of ack3); the second line shows the
resultant change in cwnd_.
The graph above of cwnd v time was made by selecting out these cwnd_ lines and plotting the first field
(time) and the last. (Recall that during the linear-increase phase cwnd is incremented by 1.0/cwnd with
each arriving new ACK.)
The last ack in the tracefile is
9.98029 0 0 2 0 ack_ 808
Since ACKs started with number 0, this means we sent 809 packets successfully. The theoretical bandwidth
was 100 packets/sec 10 sec = 1000 packets, so this is about an 81% goodput. Use of the ack_ value this
way tells us how much data was actually delivered. An alternative statistic is the final value of maxseq_
which represents the number of distinct packets sent; the last maxseq_ line is
9.99029 0 0 2 0 maxseq_ 829
As can be seen from the cwnd-v-time graph above, slow start ends around T=2.0. If we measure goodput
from then until the end, we do a little better than 81%. The first data packet sent after T=2.0 is at 2.043184;
it is data packet 72. 737 packets are sent from packet 72 until packet 808 at the end; 737 packets in 8.0
seconds is a goodput of 92%.
It is not necessary to use the tracefile to get the final values of TCP variables such as ack_ and maxseq_;
they can be printed from within the Tcl scripts finish() procedure. The following example illustrates
this, where ack_ and maxseq_ come from the connection tcp0. The global line lists global variables
that are to be made available within the body of the procedure; tcp0 must be among these.
proc finish {} {
global ns nf f tcp0
$ns flush-trace
close $namfile
close $tracefile
set lastACK [$tcp0 set ack_]
set lastSEQ [$tcp0 set maxseq_]
puts stdout "final ack: $lastACK, final seq num: $lastSEQ"
exit 0
}

For TCP sending agents, useful member variables to set include:

class_: the identifying number of a flow
window_: the maximum window size; the default is much too small.
packetSize_: we set this to 960 above so the total packet size would be 1000.
Useful member variables either to trace or to print at the simulations end include:
maxseq_: the number of the last packet sent, starting at 1 for data packets
ack_: the number of the last ACK received

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cwnd_: the current value of the congestion window

nrexmitpack_: the number of retransmitted packets
To get a count of the data actually received, we need to look at the TCPsink object, $end0 above. There is no
packet counter here, but we can retrieve the value bytes_ representing the total number of bytes received.
This will include 40 bytes from the threeway handshake which can either be ignored or subtracted:
set ACKed [expr round ( [$end0 set bytes_] / 1000.0)]

This is a slightly better estimate of goodput. In very long simulations, however, this (or any other) byte
count will wrap around long before any of the packet counters wrap around.
In the example above every packet event was traced, a consequence of the line
$ns trace-all $trace

We could instead have asked only to trace particular links. For example, the following line would request
tracing for the bottleneck (RB) link:
$ns trace-queue $R $B $trace

This is often useful when the overall volume of tracing is large, and we are interested in the bottleneck link
only. In long simulations, full tracing can increase the runtime 10-fold; limiting tracing only to what is
actually needed can be quite important.

16.2.3 Single Losses

By examining the basic1.tr file above for packet-drop records, we can confirm that only a single drop occurs
at the end of each tooth, as was argued in 13.8 Single Packet Losses. After slow start finishes at around
T=2, the next few drops are at T=3.963408, T=5.909568 and T=7.855728. The first of these drops is of
Data[254], as is shown by the following record:
d 3.963408 1 2 tcp 1000 ------- 0 0.0 2.0 254 531

Like most real implementations, the ns-2 implementation of TCP increments cwnd (cwnd_ in the tracefile) by 1/cwnd on each new ACK (13.2.1 Per-ACK Responses). An additional packet is sent by A whenever cwnd is increased this way past another whole number; that is, whenever floor(cwnd) increases. At
T=3.95181, cwnd_ was incremented to 20.001, triggering the double transmission of Data[253] and the
doomed Data[254]. At this point the RTT is around 190 ms.
The loss of Data[254] is discovered by Fast Retransmit when the third dupACK[253] arrives. The first
ACK[253] arrives at A at T=4.141808, and the dupACKs arrive every 10 ms, clocked by the 10 ms/packet
transmission rate of R. Thus, A detects the loss at T=4.171808; at this time we see cwnd_ reduced by half
to 10.465; the tracefile times for variables are only to 5 decimal places, so this is recorded as
4.17181

cwnd_ 10.465

That represents an elapsed time from when Data[254] was dropped of 207.7 ms, more than one RTT. As
described in 13.8 Single Packet Losses, however, A stopped incrementing cwnd_ when the first ACK[253]
arrived at T=4.141808. The value of cwnd_ at that point is only 20.931, not quite large enough to trigger
transmission of another back-to-back pair and thus eventually a second packet drop.
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16.2.4 Reading the Tracefile in Python

Deeper analysis of ns-2 data typically involves running some sort of script on the tracefiles; we will mostly
use the Python (python3) language for this, although the awk language is also traditional. The following is
the programmer interface to a simple module (library) nstrace.py:
nsopen(filename): opens the tracefile
isEvent(): returns true if the current line is a normal ns-2 event record
isVar(): returns true if the current line is an ns-2 variable-trace record
isEOF(): returns true if there are no more tracefile lines
getEvent(): returns a twelve-element tuple of the ns-2 event-trace values, each cast to the correct
type. The ninth and tenth values, which are node.port pairs in the tracefile, are returned as (node port)
sub-tuples.
getVar(): returns a seven-element tuple of ns-2 variable-trace values
skipline(): skips the current line (useful if we are interested only in event records, or only in variabletrace records, and want to ignore the other type of record)
We will first make use of this in 16.2.6.1 Link utilization measurement; see also 16.4 TCP Loss Events
and Synchronized Losses. The nstrace.py file above includes regular-expression checks to verify that each
tracefile line has the correct format, but, as these are slow, they are disabled by default. Enabling these
checks is potentially useful, however, if some wireless trace records are also included.

16.2.5 The nam Animation

Let us now re-examine the nam animation, in light of what can be found in the trace file.
At T=0.120864, the first 1000-byte data packet is sent (at T=0 a 40-byte SYN packet is sent); the actual
packet identification number is 1 so we will refer to it as Data[1]. At this point cwnd_ = 2, so Data[2] is
enqueued at this same time, and sent at T=0.121664 (the delay exactly matches the AR links bandwidth
of 8000 bits in 0.0008 sec). The first loss occurs at T=0.58616, of Data[28]; at T=0.59616 Data[30] is lost.
(Data[29] was not lost because R was able to send a packet and thus make room).
From T=.707392 to T=.777392 we begin a string of losses: packets 42, 44, 46, 48, 50, 52, 54 and 56.
At T=0.76579 the first ACK[27] makes it back to A. The first dupACK[27] arrives at T=0.77576; another
arrives at T=0.78576 (10 ms later, exactly the bottleneck per-packet time!) and the third dupACK arrives at
T=0.79576. At this point, Data[28] is retransmitted and cwnd is halved from 29 to 14.5.
At T=0.985792, the sender receives ACK[29]. DupACK[29]s are received at T=1.077024, T=1.087024,
T=1.097024 and T=1.107024. Alas, this is TCP Reno, in Fast Recovery mode, and it is not implementing
Fast Retransmit while Fast Recovery is going on (TCP NewReno in effect fixes this). Therefore, the connection experiences a hard timeout at T=1.22579; the last previous event was at T=1.107024. At this point
ssthresh is set to 7 and cwnd drops to 1. Slow start is used up to ssthresh = 7, at which point the sender
switches to the linear-increase phase.

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16.2.6 Single-sender Throughput Experiments

According to the theoretical analysis in 13.7 TCP and Bottleneck Link Utilization, a queue size of close
to zero should yield about a 75% bottleneck utilization, a queue size such that the mean cwnd equals the
transit capacity should yield about 87.5%, and a queue size equal to the transit capacity should yield close
to 100%. We now test this.
We first increase the per-link propagation times in the basic1.tcl simulation above to 50 and 100 ms:
$ns duplex-link $A $R 10Mb 50ms DropTail
$ns duplex-link $R $B 800Kb 100ms DropTail

The bottleneck link here is 800 Kb, or 100 Kbps, or 10 ms/packet, so these propagation-delay changes mean
a round-trip transit capacity of 30 packets (31 if we include the bandwidth delay at R). In the table below,
we run the simulation while varying the queue-limit parameter from 3 to 30. The simulations run for
1000 seconds, to minimize the effect of slow start. Tracing is disabled to reduce runtimes. The received
column gives the number of distinct packets received by B; if the link utilization were 100% then in 1,000
seconds B would receive 100,000 packets.
queue_limit
3
4
5
8
10
12
16
20
22
26
30

received
79767
80903
83313
87169
89320
91382
94570
97261
98028
99041
99567

utilization %, RB
79.8
80.9
83.3
87.2
89.3
91.4
94.6
97.3
98.0
99.0
99.6

In ns-2, every arriving packet is first enqueued, even if it is immediately dequeued, and so queue-limit
cannot actually be zero. A queue-limit of 1 or 2 gives very poor results, probably because of problems
with slow start. The run here with queue-limit = 3 is not too far out of line with the 75% predicted by
theory for a queue-limit close to zero. When queue-limit is 10, then cwnd will range from 20 to
40, and the link-unsaturated and queue-filling phases should be of equal length. This leads to a theoretical
link utilization of about (75%+100%)/2 = 87.5%; our measurement here of 89.3% is in good agreement. As
queue-limit continues to increase, the link utilization rapidly approaches 100%, again as expected.
16.2.6.1 Link utilization measurement
In the experiment above we estimated the utilization of the RB link by the number of distinct packets
arriving at B. But packet duplicate transmissions sometimes occur as well (see 16.2.6.4 Packets that are
delivered twice); these are part of the RB link utilization but are hard to estimate (nominally, most packets
retransmitted by A are dropped by R, but not all).
If desired, we can get an exact value of the RB link utilization through analysis of the ns-2 trace file. In
this file R is node 1 and B is node 2 and our flow is flow 0; we look for all lines of the form

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- time 1 2 tcp 1000 ------- 0 0.0 2.0 x y

that is, with field1 = -, field3 = 1, field4 = 2, field6 = 1000 and field8 = 0 (if we do not check field6=1000
then we count one extra packet, a simulated SYN). We then simply count these lines; here is a simple script
to do this in Python using the nstrace module above:
#!/usr/bin/python3
import nstrace
import sys
def link_count(filename):
SEND_NODE = 1
DEST_NODE = 2
FLOW = 0
count = 0

nstrace.nsopen(filename)
while not nstrace.isEOF():
if nstrace.isEvent():
(event, time, sendnode, dest, dummy, size, dummy, flow, dummy, dummy, dummy, dum
if (event == "-" and sendnode == SEND_NODE and dest == DEST_NODE and size >= 100
count += 1
else:
nstrace.skipline()
print ("packet count:", count);
link_count(sys.argv[1])

For completeness, here is the same program implemented in the Awk scripting language.
BEGIN {count=0; SEND_NODE=1; DEST_NODE=2; FLOW=0}
$1 == "-" { if ($3 == SEND_NODE && $4 == DEST_NODE && $6 >= 1000 && $8 == FLOW) {
count++;
}
}
END
{print count;}

16.2.6.2 ns-2 command-line parameters

Experiments where we vary one parameter, eg queue-limit, are facilitated by passing in the parameter
value on the command line. For example, in the basic1.tcl file we can include the following:
set queuesize $argv
...
$ns queue-limit $R $B $queuesize

Then, from the command line, we can run this as follows:

ns basic1.tcl 5

If we want to run this simulation with parameters ranging from 0 to 10, a simple shell script is

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queue=0
while [ $queue -le 10 ]
do
ns basic1.tcl $queue
queue=$(expr $queue + 1)
done

If we want to pass multiple parameters on the command line, we use lindex to separate out arguments
from the $argv string; the first argument is at position 0 (in bash and awk scripts, by comparison, the
first argument is $1). For two optional parameters, the first representing queuesize and the second
representing endtime, we would use
if { $argc >= 1 } {
set queuesize [expr [lindex $argv 0]]
}
if { $argc >= 2 } {
set endtime [expr [lindex $argv 1]]
}

16.2.6.3 Queue utilization

In our previous experiment we examined link utilization when queue-limit was smaller than the
bandwidthdelay product. Now suppose queue-limit is greater than the bandwidthdelay product,
so the bottleneck link is essentially never idle. We calculated in 13.7 TCP and Bottleneck Link Utilization
what we might expect as an average queue utilization. If the transit capacity is 30 packets and the queue
capacity is 40 then cwndmax would be 70 and cwndmin would be 35; the queue utilization would vary from
70-30 = 40 down to 35-30 = 5, averaging around (40+5)/2 = 22.5.
Let us run this as an ns-2 experiment. As before, we set the AR and RB propagation delays to 50 ms
and 100 ms respectively, making the RTTnoLoad 300 ms, for about 30 packets in transit. We also set the
queue-limit value to 40. The simulation runs for 1000 seconds, enough, as it turns out, for about 50
TCP sawteeth.
At the end of the run, the following Python script maintains a running time-weighted average of the queue
size. Because the queue capacity exceeds the total transit capacity, the queue is seldom empty.
#!/usr/bin/python3
import nstrace
import sys

def queuesize(filename):
QUEUE_NODE = 1
nstrace.nsopen(filename)
sum = 0.0
size= 0
prevtime=0
while not nstrace.isEOF():
if nstrace.isEvent():
# counting regular trace lines
(event, time, sendnode, dnode, proto, dummy, dummy, flow, dummy, dummy, seqno, p
if (sendnode != QUEUE_NODE): continue
if (event == "r"): continue

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sum += size * (time -prevtime)

prevtime = time
if (event==d): size -= 1
elif (event=="-"): size -= 1
elif (event=="+"): size += 1
else:
nstrace.skipline()
print("avg queue=", sum/time)
queuesize(sys.argv[1])

The answer we get for the average queue size is about 23.76, which is in good agreement with our theoretical
value of 22.5.
16.2.6.4 Packets that are delivered twice
Every dropped TCP packet is ultimately transmitted twice, but classical TCP theory suggests that relatively
few packets are actually delivered twice. This is pretty much true once the TCP sawtooth phase is reached,
but can fail rather badly during slow start.
The following Python script will count packets delivered two or more times. It uses a dictionary, COUNTS,
which is indexed by sequence numbers.
#!/usr/bin/python3
import nstrace
import sys

def dup_counter(filename):
SEND_NODE = 1
DEST_NODE = 2
FLOW = 0
count = 0
COUNTS = {}
nstrace.nsopen(filename)
while not nstrace.isEOF():
if nstrace.isEvent():
(event, time, sendnode, dest, dummy, size, dummy, flow, dummy, dummy, seqno, dum
if (event == "r" and dest == DEST_NODE and size >= 1000 and flow == FLOW):
if (seqno in COUNTS):
COUNTS[seqno] += 1
else:
COUNTS[seqno] = 1
else:
nstrace.skipline()
for seqno in sorted(COUNTS.keys()):
if (COUNTS[seqno] > 1): print(seqno, COUNTS[seqno])
dup_counter(sys.argv[1])

When run on the basic1.tr file above, it finds 13 packets delivered twice, with TCP sequence numbers 43,
45, 47, 49, 51, 53, 55, 57, 58, 59, 60, 61 and 62. These are sent the second time between T=1.437824 and
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T=1.952752; the first transmissions are at times between T=0.83536 and T=1.046592. If we look at our
cwnd-v-time graph above, we see that these first transmissions occurred during the gap between the end of
the unbounded slow-start phase and the beginning of threshold-slow-start leading up to the TCP sawtooth.
Slow start, in other words, is messy.
16.2.6.5 Loss rate versus cwnd: part 1
If we run the basic1.tcl simulation above until time 1000, there are 94915 packets acknowledged and 512
loss events. This yields a loss rate of p = 512/94915 = 0.00539, and so by the formula of 14.5 TCP Reno

loss rate versus cwnd we should expect the average cwnd to be about 1.225/ p 16.7. The true average
cwnd is the number of packets sent divided by the elapsed time in RTTs, but as RTTs are not constant here
(they get significantly longer as the queue fills), we turn to an approximation. From 16.2.1 Graph of cwnd
v time we saw that the peak cwnd was 20.935; the mean cwnd should thus be about 3/4 of this, or 15.7.
While not perfect, agreement here is quite reasonable.
See also 16.4.3 Loss rate versus cwnd: part 2.

16.3 Two TCP Senders Competing

Now let us create a simulation in which two TCP Reno senders compete for the bottleneck link, and see how
fair an allocation each gets. According to the analysis in 14.3 TCP Fairness with Synchronized Losses,
this is really a test of the synchronized-loss hypothesis, and so we will also examine the ns-2 trace files for
losses and loss responses. We will start with classic TCP Reno, but eventually also consider SACK TCP.
Note that, in terms of packet losses in the immediate vicinity of any one queue-filling event, we can expect
TCP Reno and SACK TCP to behave identically; they differ only in how they respond to losses.
The initial topology will be as follows (though we will very soon raise the bandwidths tenfold, though not
the propagation delays):

Broadly speaking, the simulations here will demonstrate that the longer-delay BD connection receives less
bandwidth than the AD connection, but not quite so much less as was predicted in 14.3 TCP Fairness with
Synchronized Losses. The synchronized-loss hypothesis increasingly fails as the BR delay increases, in that
the BD connection begins to escape some of the packet-loss events experienced by the AD connection.
We admit at the outset that we will not, however, obtain a quantitative answer to the question of bandwidth
allocation. In fact, as we shall see, we run into some difficulties even formulating the proper question. In
the course of developing the simulation, we encounter several potential problems:

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1. The two senders can become synchronized in an unfortunate manner

2. When we resolve the previous issue by introducing randomness, the bandwidth division is sensitive
to the method selected
3. As Rs queue fills, the RTT may increase significantly, thus undermining RTT-based measurements
(16.3.9 The RTT Problem)
4. Transient queue spikes may introduce unexpected losses
5. Coarse timeouts may introduce additional unexpected losses
The experiments and analyses below divide into two broad categories. In the first category, we make use
only of the final goodput measurements for the two connections. We consider the first two points of the
list above in 16.3.4 Phase Effects, and the third in 16.3.9 The RTT Problem and 16.3.10 Raising the
Bandwidth. The first category concludes with some simple loss modeling in 16.3.10.1 Possible models.
In the second category, beginning at 16.4 TCP Loss Events and Synchronized Losses, we make use of
the ns-2 tracefiles to extract information about packet losses and the extent to which they are synchronized.
Examples related to points four and five of the list above are presented in 16.4.1 Some TCP Reno cwnd
graphs. The second category concludes with 16.4.2 SACK TCP and Avoiding Loss Anomalies, in which
we demonstrate that SACK TCP is, in terms of loss and recovery, much better behaved than TCP Reno.

16.3.1 The Tcl Script

Below is a simplified version of the ns-2 script for our simulation; the full version is at basic2.tcl. The most
important variable is the additional one-way delay on the BR link, here called delayB. Other defined
variables are queuesize (for Rs queue_limit), bottleneckBW (for the RD bandwidth), endtime
(the length of the simulation), and overhead (for introducing some small degree of randomization, below).
As with basic1.tcl, we set the packet size to 1000 bytes total (960 bytes TCP portion), and increase the
advertised window size to 65000 (so it is never the limiting factor).
We have made the delayB value be a command-line parameter to the Tcl file, so we can easily experiment with changing it (in the full version linked to above, overhead, bottleneckBW, endtime and
queuesize are also parameters). The one-way propagation delay for the AD path is 10 ms + 100 ms =
110 ms, making the RTT 220 ms plus the bandwidth delays. At the bandwidths above, the bandwidth delay
for data packets adds an additional 11 ms; ACKs contribute an almost-negligible 0.44 ms. We return to the
script variable RTTNL, intended to approximate RTTnoLoad , below.
With endtime=300, the theoretical maximum number of data packets that can be delivered is 30,000. If
bottleneckBW = 0.8 Mbps (100 packets/sec) then the RD link can hold ten RD data packets in
transit, plus another ten DR ACKs.
In the finish() procedure we have added code to print out the number of packets received by D for each
connection; we could also extract this from the trace file.
To gain better control over printing, we have used the format command, which works something like Cs
sprintf. It returns a string containing spliced-in numeric values replacing the corresponding %d or %f
tokens in the control string; this returned string can then be printed with puts.
The full version linked to above also contains some nam directives, support for command-line arguments,
and arranges to name any tracefiles with the same base filename as the Tcl file.

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# NS basic2.tcl example of two TCPs competing on the same link.

# Create a simulator object
set ns [new Simulator]
#Open the trace file
set trace [open basic2.tr w]
$ns trace-all $trace
############## some globals (modify as desired) ##############
# queuesize on bottleneck link
set queuesize 20
# default run time, in seconds
set endtime 300
# "overhead" of D>0 introduces a uniformly randomized delay d, 0dD; 0 turns it off.
set overhead 0
# delay on the A--R link, in ms
set basedelay 10
# ADDITIONAL delay on the B--R link, in ms
set delayB 0
# bandwidth on the bottleneck link, either 0.8 or 8.0 Mbit
set bottleneckBW 0.8
# estimated no-load RTT for the first flow, in ms
set RTTNL 220
############## arrange for output ##############

set outstr [format "parameters: delayB=%f overhead=%f bottleneckBW=%f" $delayB $overhead $b

puts stdout $outstr

# Define a finish procedure that prints out progress for each connection
proc finish {} {
global ns tcp0 tcp1 end0 end1 queuesize trace delayB overhead RTTNL
set ack0 [$tcp0 set ack_]
set ack1 [$tcp1 set ack_]
# counts of packets *received*
set recv0 [expr round ( [$end0 set bytes_] / 1000.0)]
set recv1 [expr round ( [$end1 set bytes_] / 1000.0)]
# see numbers below in topology-creation section
set rttratio [expr (2.0*$delayB+$RTTNL)/$RTTNL]
# actual ratio of throughputs fast/slow; the 1.0 forces floating point
set actualratio [expr 1.0*$recv0/$recv1]
# theoretical ratio fast/slow with squaring; see text for discussion of ratio1 and
set rttratio2 [expr $rttratio*$rttratio]
set ratio1 [expr $actualratio/$rttratio]
set ratio2 [expr $actualratio/$rttratio2]
set outstr [format "%f %f %d %d %f %f %f %f %f" $delayB $overhead $recv0 $recv1 $rt
puts stdout $outstr
$ns flush-trace
close $trace
exit 0
}

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############### create network topology ##############

# A
#
\
#
\
#
R---D (Destination)
#
/
#
/
# B
#Create four nodes
set A [$ns node]
set B [$ns node]
set R [$ns node]
set D [$ns node]
set fastbw [expr $bottleneckBW * 10]
#Create links between the nodes; propdelay on B--R link is 10+$delayB ms
$ns duplex-link $A $R ${fastbw}Mb ${basedelay}ms DropTail
$ns duplex-link $B $R ${fastbw}Mb [expr $basedelay + $delayB]ms DropTail
# this last link is the bottleneck; 1000 bytes at 0.80Mbps => 10 ms/packet
# A--D one-way delay is thus 110 ms prop + 11 ms bandwidth
# the values 0.8Mb, 100ms are from Floyd & Jacobson
$ns duplex-link $R $D ${bottleneckBW}Mb 100ms DropTail
$ns queue-limit $R $D $queuesize
############## create and connect TCP agents, and start ##############
Agent/TCP set window_ 65000
Agent/TCP set packetSize_ 960
Agent/TCP set overhead_ $overhead
#Create a TCP agent and attach it to node A, the delayed path
set tcp0 [new Agent/TCP/Reno]
$tcp0 set class_ 0
# set the flowid here, used as field 8 in the trace
$tcp0 set fid_ 0
$tcp0 attach $trace
$tcp0 tracevar cwnd_
$tcp0 tracevar ack_
$ns attach-agent $A $tcp0
set tcp1 [new Agent/TCP/Reno]
$tcp1 set class_ 1
$tcp1 set fid_ 1
$tcp1 attach $trace
$tcp1 tracevar cwnd_
$tcp1 tracevar ack_
$ns attach-agent $B $tcp1
set end0 [new Agent/TCPSink]

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$ns attach-agent $D $end0

set end1 [new Agent/TCPSink]
$ns attach-agent $D $end1
#Connect the traffic source with the traffic sink
$ns connect $tcp0 $end0
$ns connect $tcp1 $end1
#Schedule the connection data flow
set ftp0 [new Application/FTP]
$ftp0 attach-agent $tcp0
set ftp1 [new Application/FTP]
$ftp1 attach-agent $tcp1
$ns at 0.0 "$ftp0 start"
$ns at 0.0 "$ftp1 start"
$ns at $endtime "finish"
#Run the simulation
$ns run

16.3.2 Equal Delays

We first try this out by running the simulation with equal delays on both AR and RB. The following values
are printed out (arranged here vertically to allow annotation)
value
0.000000
0.000000
14863
14771
1.000000
1.000000
1.006228

variable
meaning
delayB
Additional BR propagation delay, compared to AR delay
overhead
overhead; a value of 0 means this is effectively disabled
recv0
Count of cumulative AD packets received at D (that is, goodput)
recv1
Count of cumulative BD packets received at D (again, goodput)
rttratio
RTT_ratio: BD/AD (long/short)
rttratio2 The square of the previous value
actualratioActual ratio of AD/BD goodput, that is, 14863/14771 (note change in order
versus RTT_ratio)
1.006228 ratio1
actual_ratio/RTT_ratio
1.006228 ratio2
actual_ratio/RTT_ratio2
The one-way AD propagation delay is 110 ms; the bandwidth delays as noted above amount to 11.44 ms,
10 ms of which is on the RD link. This makes the AD RTTnoLoad about 230 ms. The BD delay is, for
the time being, the same, as delayB = 0. We set RTTNL = 220, and calculate the RTT ratio (within Tcl, in
the finish() procedure) as (2delayB + RTTNL)/RTTNL. We really should use RTTNL=230 instead of 220
here, but 220 will be closer when we later change bottleneckBW to 8.0 Mbit/sec rather than 0.8, below.
Either way, the difference is modest.
Note that the above RTT calculations are for when the queue at R is empty; when the queue contains 20
packets this adds another 200 ms to the AD and BD times (the reverse direction is unchanged). This
may make a rather large difference to the RTT ratio, and we will address it below, but does not matter yet
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because the propagation delays so far are identical.

In the model of 14.3.3 TCP RTT bias we explored a model in which we expect that ratio2, above, would
be about 1.0. The final paragraph of 14.5.2 Unsynchronized TCP Losses hints at a possible model (the =
case) in which ratio1 would be about 1.0. We will soon be in a position to test these theories experimentally.
Note that the order of B and A in the goodput and RTT ratios is reversed, to reflect the expected inverse
relationship.
In the 300-second run here, 14863+14771 = 29634 packets are sent. This means that the bottleneck link is
98.8% utilized.
In 16.4.1 Some TCP Reno cwnd graphs we will introduce a script (teeth.py) to count and analyze the
teeth of the TCP sawtooth. Applying this now, we find there are 67 loss events total, and thus 67 teeth,
and in every loss event each flow loses exactly one packet. This is remarkably exact conformance to the
synchronized-loss hypothesis of 14.3.3 TCP RTT bias. So far.

16.3.3 Unequal Delays

We now begin increasing the additional BR delay (delayB). Some preliminary data are in the table below,
and point to a significant problem: goodput ratios in the last column here are not varying smoothly. The
value of 0 ms in the first row means that the BR delay is equal to the AR delay; the value of 110 ms in
the last row means that the BD RTTnoLoad is double the AD RTTnoLoad . The column labeled (RTT ratio)2
is the expected goodput ratio, according to the model of 14.3 TCP Fairness with Synchronized Losses; the
actual AD/BD goodput ratio is in the final column.
delayB
0
5
23
24
25
26
35
36
40
41
45
60
61
85
110

RTT ratio
1.000
1.045
1.209
1.218
1.227
1.236
1.318
1.327
1.364
1.373
1.409
1.545
1.555
1.773
2.000

AD goodput
14863
4229
22142
17683
14958
24034
16932
25790
20005
24977
18437
18891
25834
20463
22624

BD goodput
14771
24545
6879
9842
13754
5137
11395
3603
8580
4215
10211
9891
3135
8206
5941

(RTT ratio)2
1.000
1.093
1.462
1.484
1.506
1.529
1.738
1.762
1.860
1.884
1.986
2.388
2.417
3.143
4.000

goodput ratio
1.006
0.172
3.219
1.797
1.088
4.679
1.486
7.158
2.332
5.926
1.806
1.910
8.241
2.494
3.808

For a few rows, such as the first and the last, agreement between the last two columns is quite good. However,
there are some decidedly anomalous cases in between (particularly the numbers in bold). As delayB
changes from 35 to 36, the goodput ratio jumps from 1.486 to 7.158. Similar dramatic changes in goodput
appear as delayB ranges through the sets {23, 24, 25, 26}, {40, 41, 45}, and {60, 61}. These values were,
admittedly, specially chosen by trial and error to illustrate relatively discontinuous behavior of the goodput
ratio, but, still, what is going on?

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16.3.4 Phase Effects

This erratic behavior in the goodput ratio in the table above turns out to be due to what are called phase
effects in [FJ92]; transmissions of the two TCP connections become precisely synchronized in some way
that involves a persistent negative bias against one of them. What is happening is that a race condition
occurs for the last remaining queue vacancy, and one connection consistently loses this race the majority of
the time.
16.3.4.1 Single-sender phase effects
We begin by taking a more detailed look at the bottleneck queue when no competition is involved. Consider
a single sender A using a fixed window size to send to destination B through bottleneck router R (so the
topology is ARB), and suppose the window size is large enough that Rs queue is not empty. For the sake
of definiteness, assume Rs bandwidth delay is 10 ms/packet; R will send packets every 10 ms, and an ACK
will arrive back at A every 10 ms, and A will transmit every 10 ms.
Now imagine that we have an output meter that reports the percentage that has been transmitted of the packet
R is currently sending; it will go from 0% to 100% in 10 ms, and then back to 0% for the next packet.
Our first observation is that at each instant when a packet from A fully arrives at R, R is always at exactly
the same point in forwarding some earlier packet on towards B; the output meter always reads the same
percentage. This percentage is called the phase of the connection, sometimes denoted .
We can determine as follows. Consider the total elapsed time for the following:
R finishes transmitting a packet and it arrives at B
its ACK makes it back to A
the data packet triggered by that ACK fully arrives at R
In the absence of other congestion, this R-to-R time includes no variable queuing delays, and so is constant.
The output-meter percentage above is determined by this elapsed time. If for example the R-to-R time is
83 ms, and Data[N] leaves R at T=0, then Data[N+winsize] (sent by A upon arrival of ACK[N]) will arrive
at R when R has completed sending packets up through Data[N+8] (80 ms) and is 3 ms into transmitting
Data[N+9]. We can get this last as a percentage by dividing 83 by Rs 10-ms bandwidth delay and taking
the fractional part (in this case, 30%). Because the elapsed R-to-R time here is simply RTTnoLoad minus the
bandwidth delay at R, we can also compute the phase as
= fractional_part(RTTnoLoad (Rs bandwidth delay)).
If we ratchet up the winsize until the queue becomes full when each new packet arrives, then the phase
percentage represents the fraction of the time the queue has a vacancy. In the scenario above, if we start the
clock at T=0 when R has finished transmitting a packet, then the queue has a vacancy until T=3 when a new
packet from A arrives. The queue is then full until T=10, when R starts transmitting the next packet in its
queue.
Finally, even in the presence of competition through R, the phase of a single connection remains constant
provided there are no queuing delays along the bidirectional AB path except at R itself, and there only in
the forward direction towards B. Other traffic sent through R can only add delay in integral multiples of Rs
bandwidth delay, and so cannot affect the AB phase.

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16.3.4.2 Two-sender phase effects

In the present simulation, we can by the remark in the previous paragraph calculate the phase for each
sender; let these be A and B . The significance of phase to competition is that whenever A and B send
packets that happen to arrive at R in the same 10-ms interval while R is forwarding some other packet, if the
queue has only one vacancy then the connection with the smaller phase will always win it.
As a concrete example, suppose that the respective RTTnoLoad s of A and B are 221 and 263 ms. Then As
phase is 0.1 (fractional_part(22110)) and Bs is 0.3. The important thing is not that As packets take less
time, but that in the event of a near-tie As packet must arrive first at R. Imagine that R forwards a B packet
and then, four packets (40 ms) later, forwards an A packet. The ACKs elicited by these packets will cause
new packets to be sent by A and B; As packet will arrive first at R followed 2 ms later by Bs packet. Of
course, R is not likely to send an A packet four packets after every B packet, but when it does so, the arrival
order is predetermined (in As favor) rather than random.
Now consider what happens when a packet is dropped. If there is a single vacancy at R, and packets from
A and B arrive in a near tie as above, then it will always be Bs packet that is dropped. The occasional
packet-pair sent by A or B as part of the expansion of cwnd will be the ultimate cause of loss events, but
the phase effect has introduced a persistent degree of bias in As favor.
We can visualize phase effects with ns-2 by letting delayB range over, say, 0 to 50 in small increments, and
plotting the corresponding values of ratio2 (above). Classically we expect ratio2 to be close to 1.00. In the
graph below, the blue curve represents the goodput ratio; it shows a marked (though not perfect) periodicity
with period 5 ms.

The orange curve represents (RTT_ratio)2 ; according to 14.3 TCP Fairness with Synchronized Losses we
would expect the blue and orange curves to be about the same. When the blue curve is high, the slower BD
connection is proportionately at an unexpected disadvantage. Seldom do phase effects work in favor of the
BD connection, because As phase here is quite small (0.144, based on As exact RTTnoLoad of 231.44 ms).
(If we change the AR propagation delay (basedelay) to 12 ms, making As phase 0.544, the blue curve
oscillates somewhat more evenly both above and below the orange curve, but still with approximately the
same amplitude.)
Recall that a 5 ms change in delayB corresponds to a 10 ms change in the AD connections RTT, equal to
router Rs transmission time. What is happening here is that as the BD connections RTT increases through
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a range of 10 ms, it cycles through from phase-effect neutrality to phase-effect deficit and back.

16.3.5 Minimizing Phase Effects

In the real world, the kind of precise transmission synchronization that leads to phase effects is seldom
evident, though perhaps this has less to do with rarity and more with the fact that head-to-head TCP competitions are difficult to observe intimately. Usually, however, there seems to be sufficient other traffic present
to disrupt the synchronization. How can we break this synchronization in simulations? One way or another,
we must inject some degree of randomization into the bulk TCP flows.
Techniques introduced in [FJ92] to break synchronization in ns-2 simulations were random telnet traffic
involving smaller packets sent according to a given random distribution and the use of random-drop
queues (not included in the standard ns-2 distribution). The second, of course, means we are no longer
simulating FIFO queues.
A third way of addressing phase effects is to make use of the ns-2 overhead variable, which introduces
some modest randomization in packet-departure times at the TCP sender. Because this technique is simpler,
we will start with it. One difference between the use of overhead and telnet traffic is that the latter has
the effect of introducing delays at all nodes of the network that carry the traffic, not just at the TCP sources.

16.3.6 Phase Effects and overhead

For our first attempt at introducing phase-effect-avoiding randomization in the competing TCP flows, we will
start with ns-2s TCP overhead attribute. This is equal to 0 by default and is measured in units of seconds.
If overhead > 0, then the TCP source introduces a uniformly distributed random delay of between 0 and
overhead seconds whenever an ACK arrives and the source is allowed to send a new packet. Because
the distribution is uniform, the average delay so introduced is thus overhead/2. To introduce an average
delay of 10 ms, therefore, one sets overhead = 0.02. Packets are always sent in order; if packet 2 is
assigned a small overhead delay and packet 1 a large overhead delay, then packet 2 waits until packet
1 has been sent. For this reason, it is a good idea to keep the average overhead delay no more than the
average packet interval (here 10 ms).
The following graph shows four curves representing overhead values of 0, 0.005, 0.01 and 0.02 (that is,
5 ms, 10 ms and 20 ms). For each curve, ratio1 (not the actual goodput ratio and not ratio2) is plotted as
a function of delayB as the latter ranges from 25 to 55 ms. The simulations run for 300 seconds, and
bottleneckBW = 0.8. (We will return to the choice of ratio1 here in 16.3.9 The RTT Problem; the
corresponding ratio2 graph is however quite similar, at least in terms of oscillatory behavior.)

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The dark-blue curve for overhead = 0 is wildly erratic due to phase effects; the light-blue curve for
overhead = 0.005 has about half the oscillation. Even the light-green overhead = 0.01 curve exhibits
some wiggling; it is not until overhead = 0.02 for the darker green curve that the graph really settles
down. We conclude that the latter two values for overhead are quite effective at mitigating phase effects.
One crude way to quantify the degree of graph oscillation is by calculating the mean deviation; the respective
deviation values for the curves above are are 1.286, 0.638, 0.136 and 0.090.
Recall that the time to send one packet on the bottleneck link is 0.01 seconds, and that the average delay
introduced by overhead d is d/2; thus, when overhead is 0.02 each connection would, if acting alone,
have an average sender delay equal to the bottleneck-link delay (though overhead delay is like propagation
delay, and so a high overhead will not prevent queue buildup).
Compared to the 10-ms-per-packet RD transmission time, average delays of 5 and 10 ms per flow
(overhead of 0.01 and 0.02 respectively) may not seem disproportionate. They are, however, quite large
when compared to the 1.0 ms bandwidth delay on the AR and BR legs. Generally, if the goal is to reduce phase effects then overhead should be comparable to the bottleneck-router transmission rate. Using
overhead > 0 does increase the RTT, but in this case not considerably.
We conclude that using overhead to break the synchronization that leads to phase effects appears to have
worked, at least in the sense that with the value of overhead = 0.02 the goodput ratio increases more-orless monotonically with increasing delayB.
The problem with using overhead this way is that it does not correspond to any physical network delay
or other phenomenon. Its use here represents a decidedly ad hoc strategy to introduce enough randomization
that phase effects disappear.

16.3.7 Phase Effects and telnet traffic

We can also introduce anti-phase-effect randomization by making use of the ns-2 telnet application to generate low-to-moderate levels of random traffic. This requires an additional two Agent/TCP objects, representing AD and BD telnet connections, to carry the traffic; this telnet traffic will then introduce slight delays

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in the corresponding bulk (ftp) traffic. The size of the telnet packets sent is determined by the TCP agents
usual packetSize_ attribute.
For each telnet connection we create an Application/Telnet object and set its attribute interval_; in
the script fragment below this is set to tninterval. This represents the average packet spacing in seconds;
transmissions are then scheduled according to an exponential random distribution with interval_ as its
mean.
Actual (simulated) transmissions, however, are also constrained by the telnet connections sliding window.
It is quite possible that the telnet application releases a new packet for transmission, but it cannot yet be sent
because the telnet TCP connections sliding window is momentarily frozen, waiting for the next ACK. If
the telnet packets encounter congestion and the interval_ is small enough then the sender may have a
backlog of telnet packets in its outbound queue that are waiting for the sliding window to advance enough
to permit their departure.
set
$ns
set
$ns

tcp10 [new Agent/TCP]

attach-agent $A $tcp10
tcp11 [new Agent/TCP]
attach-agent $B $tcp11

set
set
$ns
$ns

end10 [new Agent/TCPSink]

end11 [new Agent/TCPSink]
attach-agent $D $end10
attach-agent $D $end11

set telnet0 [new Application/Telnet]

set telnet1 [new Application/Telnet]
set tninterval 0.001

;# see text for discussion

$telnet0 set interval_ $tninterval

$tcp10 set packetSize_ 210
$telnet1 set interval_ $tninterval
$tcp11 set packetSize_ 210
$telnet0 attach-agent $tcp10
$telnet1 attach-agent $tcp11

Real telnet packets are most often quite small; in the simulations here we use an uncharacteristically large
size of 210 bytes, leading to a total packet size of 250 bytes after the 40-byte simulated TCP/IP header is
attached. We denote the latter number by actualSize. See exercise 9.
The bandwidth theoretically consumed by the telnet connection is simply actualSize/$tninterval;
the actual bandwidth may be lower if the telnet packets are encountering congestion as noted above. It is
convenient to define an attribute tndensity that denotes the fraction of the RD links bandwith that the
Telnet application will be allowed to use, eg 2%. In this case we have
$tninterval = actualSize / ($tndensity * $bottleneckBW)

For example, if actualSize = 250 bytes, and $bottleneckBW corresponds to 1000 bytes every 10 ms,
then the telnet connection could saturate the link if its packets were spaced 2.5 ms apart. However, if we
want the telnet connection to use up to 5% of the bottleneck link, then $tninterval should be 2.5/0.05
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= 50 ms (converted for ns-2 to 0.05 sec).

The first question we need to address is whether telnet traffic can sufficiently dampen phase-effect oscillations, and, if so, at what densities. The following graph is the telnet version of that above in 16.3.6 Phase
Effects and overhead; bottleneckBW is still 0.8 but the simulations now run for 3000 seconds. The telnet
total packet size is 250 bytes. The given telnet density percentages apply to each of the AD and BD telnet
connections; the total telnet density is thus double the value shown.

As we hoped, the oscillation does indeed substantially flatten out as the telnet density increases from 0 (dark
blue) to 1% (bright green); the mean deviation falls from 1.36 to 0.084. So far the use of telnet is very
promising.
Unfortunately, if we continue to increase the telnet density past 1%, the oscillation increases again, as can
be seen in the next graph:

The first two curves, plotted in green, correspond to the good densities of the previous graph, with the
vertical axis stretched by a factor of two. As densities increase, however, phase-effect oscillation returns,
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and the curves converge towards the heavier red curve at the top.
What appears to be happening is that, beyond a density of 1% or so, the limiting factor in telnet transmission
becomes the telnet sliding window rather than the random traffic generation, as mentioned in the third
paragraph of this section. Once the sliding window becomes the limiting factor on telnet packet transmission,
the telnet connections behave much like and become synchronized with their corresponding bulk-traffic
ftp connections. At that point their ability to moderate phase effects is greatly diminished, as actual packet
departures no longer have anything to do with the exponential random distribution that generates the packets.
Despite this last issue, the fact that small levels of random traffic can lead to large reductions in phase effects
can be taken as evidence that, in the real world, where other traffic sources are ubiquitous, phase effects will
seldom be a problem.

16.3.8 overhead versus telnet

The next step is to compare the effect on the original bulk-traffic flows of overhead randomization and
telnet randomization. The graphs below plot ratio1 as a function of delayB as the latter ranges from 0 to
400 in increments of 5. The bottleneckBW is 0.8 Mbps, the queue capacity at R is 20, and the run-time
is 3000 seconds.

The four reddish-hued curves represent the result of using telnet with a packet size of 250, at densities
ranging from 0.5% to 5%. These may be compared with the three green curves representing the use of
overhead, with values 0.01, 0.015 and 0.02. While telnet with a density of 1% is in excellent agreement
with the use of overhead, it is also apparent that smaller telnet densities give a larger ratio1 while larger
densities give a smaller. This raises the awkward possibility that the exact mechanism by which we introduce
randomization may have a material effect on the fairness ratios that we ultimately observe. There is no
right answer here; different randomization sources or levels may simply lead to differing fairness results.
The advantage of using telnet for randomization is that it represents an actual network phenomenon, unlike
overhead. The drawback to using telnet is that the effect on the bulk-traffic goodput ratio is, as the graph
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above shows, somewhat sensitive to the exact value chosen for the telnet density.
In the remainder of this chapter, we will continue to use the overhead model, for simplicity, though we
do not claim this is a universally appropriate approach.

16.3.9 The RTT Problem

In all three of the preceding graphs (16.3.6 Phase Effects and overhead, 16.3.7 Phase Effects and telnet
traffic and 16.3.8 overhead versus telnet), the green curves on the graphs appear to show that, once sufficient
randomization has been introduced to disrupt phase effects, ratio1 converges to 1.0. This, however, is in fact
an artifact, due to the second flaw in our simulated network: RTT is not very constant. While RTTnoLoad
for the AD link is about 220 ms, queuing delays at R (with queuesize = 20) can almost double that by
adding up to 20 10 ms = 200 ms. This means that the computed values for RTTratio are too large, and
the computed values for ratio1 are thus too small. While one approach to address this problem is to keep
careful track of RTTactual , a simpler strategy is to create a simulated network in which the queuing delay is
small compared to the propagation delay and so the RTT is relatively constant. We turn to this next.

16.3.10 Raising the Bandwidth

In modern high-bandwidth networks, queuing delays tend to be small compared to the propagation delay;
see 13.7 TCP and Bottleneck Link Utilization. To simulate such a network here, we simply increase the
bandwidth of all the links tenfold, while leaving the existing propagation delays the same. We achieve this by
setting bottleneckBW = 8.0 instead of 0.8. This makes the AD RTTnoLoad equal to about 221 ms; queuing
delays can now amount to at most an additional 20 ms. The value of overhead also needs to be scaled
down by a factor of 10, to 0.002 sec, to reflect an average delay in the same range as the bottleneck-link
packet transmission time.
Here is a graph of results for bottleneckBW = 8.0, time = 3000, overhead = 0.002 and queuesize = 20.
We still use 220 ms as the estimated RTT, though the actual RTT will range from 221 ms to 241 ms. The
delayB parameter runs from 0 to 400 in steps of 1.0.

The first observation to make is that ratio1 is generally too large and ratio2 is generally too small, when

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compared to 1.0. In other words, neither is an especially good fit. This appears to be a fairly general
phenomenon, in both simulation and the real world: TCP Reno throughput ratios tend to be somewhere
between the corresponding RTT ratio and the square of the RTT ratio.
The synchronized-loss hypothesis led to the prediction (14.3 TCP Fairness with Synchronized Losses) that
the goodput ratio would be close to RTT_ratio2 . As this conclusion appears to fail, the hypothesis too must
fail, at least to a degree: it must be the case that not all losses are shared.
Throughout the graph we can observe a fair amount of noise variation. Most of this variation appears
unrelated to the 5 ms period we would expect for phase effects (as in the graph at 16.3.4.2 Two-sender
phase effects). However, it is important to increment delayB in amounts much smaller than 5 ms in order
to rule this out, hence the increment of 1.0 here.
There are strong peaks at delayB = 110, 220 and 330. These delayB values correspond to increasing
the RTT by integral multiples 2, 3 and 4 respectively, and the peaks are presumably related to some kind of
higher-level phase effect.
16.3.10.1 Possible models
If the synchronized-loss fairness model fails, with what do we replace it? Here are two ad hoc options. First,
we can try to fit a curve of the form
goodput_ratio = K(RTT_ratio)
to the above data. If we do this, the value for the exponent comes out to about 1.58, sort of a compromise
between ratio2 (=2) and ratio1 (=1), although the value of the exponent here is somewhat sensitive to the
details of the simulation.
An entirely different curve to fit to the data, based on the appearance in the graph that ratio2 0.5 past 120,
is
goodput_ratio = 1/2 (RTT_ratio)2
We do not, however, possess for either of these formulas a model for the relative losses in the two primary
TCP connections that is precise enough to offer an explanation of the formula (though see the final paragraph
of 16.4.2.2 Relative loss rates).
16.3.10.2 Higher bandwidth and link utilization
One consequence of raising the bottleneck bandwidth is that total link utilization drops, for delayB = 0, to
80% of the bandwidth of the bottleneck link, from 98%; this is in keeping with the analysis of 13.7 TCP
and Bottleneck Link Utilization. The transit capacity is 220 packets and another 20 can be in the queue at
R; thus an ideal sawtooth would oscillate between 120 and 240 packets. We do have two senders here, but
when delayB = 0 most losses are synchronized, meaning the two together behave like one sender with an
additive-increase value of 2. As cwnd varies linearly from 120 to 240, it spends 5/6 of the time below the
transit capacity of 220 packets during which period the average cwnd is (120+220)/2 = 170 and 1/6 of
the time with the path 100% utilized; the weighted average estimating total goodput is thus (5/6)170/220
+ (1/6)1 = 81%.
When delayB = 400, combined TCP Reno goodput falls to about 51% of the bandwidth of the bottleneck
link. This low utilization, however, is indeed related to loss and timeouts; the corresponding combined good16.3 Two TCP Senders Competing

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put percentage for SACK TCP (which as we shall see in 16.4.2 SACK TCP and Avoiding Loss Anomalies
is much better behaved) is 68%.

16.4 TCP Loss Events and Synchronized Losses

If the synchronized-loss model is not entirely accurate, as we concluded from the graph above, what does
happen with packet losses when the queue fills?
At this point we shift focus from analyzing goodput ratios to analyzing the underlying loss events, using the
ns-2 tracefile. We will look at the synchronization of loss events between different connections and at how
many individual packet losses may be involved in a single TCP loss response.
One of the conclusions we will reach is that TCP Renos response to queue overflows in the face of competition is often quite messy, versus the single-loss behavior in the absence of competition as described above in
16.2.3 Single Losses. If we are trying to come up with a packet-loss model to replace the synchronized-loss
hypothesis, it turns out that we would do better to switch to SACK TCP, though use of SACK TCP will not
change the goodput ratios much at all.
Packets are dropped when the queue fills. It may be the case that only a single packet is dropped; it may
be the case that multiple packets are dropped from each of multiple connections. We will refer to the set of
packets dropped as a drop cluster. After a packet is dropped, TCP Reno discovers this just over one RTT
after it was sent, through Fast Retransmit, and then responds by halving cwnd, through Fast Recovery.
TCP Reno retransmits only one lost packet per RTT; TCP NewReno does the same but SACK TCP may
retransmit multiple lost packets together. If enough packets were lost from a TCP Reno/NewReno connection, not all of them may be retransmitted by the point the retransmission-timeout timer expires (typically
1-2 seconds), resulting in a coarse timeout. At that point, TCP abandons its Fast-Recovery process, even if
it had been progressing steadily.
Eventually the TCP senders that have experienced packet loss reduce their cwnd, and thus the queue utilization drops. At that point we would classically not expect more losses until the senders involved have had
time to grow their cwnds through the additive-increase process to the point of again filling the queue. As
we shall see in 16.4.1.3 Transient queue peaks, however, this classical expectation is not entirely correct.
It is possible that the packet losses associated with one full-queue period are spread out over sufficient time
(more than one RTT, at a minimum) that TCP Reno responds to them separately and halves cwnd more than
once in rapid succession. We will refer to this as a loss-response cluster, or sometimes as a tooth cluster.
In the ns-2 simulator, counting individual lost packets and TCP loss responses is straightforward enough.
For TCP Reno, there are only two kinds of loss responses: Fast Recovery, in which cwnd is halved, and
coarse timeout, in which cwnd is set to 1.
Counting clusters, however, is more subjective; we need to decide when two drops or responses are close
enough to one another that they should be counted as part of the same cluster. We use the notion of granularity here: two or more losses separated by less time than the granularity time interval are counted as a
single event. We can also use granularity to decide when two loss responses in different connections are to
be considered parts of the same event, or to tie loss responses to packet losses.
The appropriate length of the granularity interval is not as clear-cut as might be hoped. In some cases a
couple RTTs may be sufficient, but note that the RTTnoLoad of the BD connection above ranges from 0.22

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sec to over 1.0 sec as delayB increases from 0 to 400 ms. In order to cover coarse timeouts, a granularity
of from two to three seconds often seems to work well for packet drops.

If we are trying to count losses to estimate the loss rate as in the formula cwnd = 1.225/ p as in 14.5 TCP
Reno loss rate versus cwnd, then we should count every loss response separately; the argument in 14.5 TCP
Reno loss rate versus cwnd depended on counting all loss responses. The difference between one fastrecovery response and two in rapid succession is that in the latter case cwnd is halved twice, to about a
quarter of its original value.
However, if we are interested in whether or not losses between two connections are synchronized, we need
again to make use of granularity to make sure two close losses are counted as one. In this setting, a
granularity of one to two seconds is often sufficient.

16.4.1 Some TCP Reno cwnd graphs

We next turn to some examples of actual TCP behavior, and present a few hopefully representative cwnd
graphs. In each figure, the red line represents the longer-path (BD) flow and the green line represents the
shorter (AD) flow. The graphs are of our usual simulation with bottleneckBW = 8.0 and overhead
= 0.002, and run for 300 seconds.
ns-2 sensitivity
Some of the graphs here were prepared with an earlier version of the basic2.tcl simulation script above
in which the nodes A and B were reversed, which means that packet-arrival ties at R may be resolved
in a different order. That is enough sometimes to lead to noticeably different sets of packet losses.
While the immediate goal is to illuminate some of the above loss-clustering issues above, the graphs serve
as well to illustrate the general behavior of TCP Reno competition and its variability. We will also use one
of the graphs to explore the idea of transient queue spikes.
In each case we can count teeth visually or via a Python script; see 16.4.2.1 Counting teeth in Python. In
the latter case we must use an appropriate granularity interval (eg 2.0 seconds) if we want the count to agree
even approximately with the visual count. Many times the tooth count is quite dependent on the exact value
of the granularity.

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16.4.1.1 delayB = 0

We start with the equal-RTTs graph, that is, delayB = 0. In this figure the teeth (loss events) are almost
completely synchronized; the only unsynchronized loss events are the green flows losses at T=100 and
at about T=205. The two cwnd graphs, though, do not exactly move in lockstep. The red flow has three
coarse timeouts (where cwnd drops to 0), at about T=30, T=125 and T=145; the green flow has seven coarse
timeouts.
The red graph gets a little ahead of the green in the interval 50-100, despite synchronized losses there. Just
before T=50 the green graph has a fast-recovery response followed by a coarse timeout; the next three green
losses also involve coarse timeouts. Despite perfect loss synchronization in the range from T=40 to T=90,
the green graph ends up set back for a while because its three loss events all involve coarse timeouts while
none of the red graphs do.
16.4.1.2 delayB = 25

In this delayB = 25 graph, respective packet losses for the red and green flows are marked along the top,
and the cwnd graphs are superimposed over a graph of the averaged queue utilization in gold. The time
scale for queue-utilization averaging is about one RTT here. The queue graph is scaled vertically (by a
factor of 8) so the queue values (maximum 20) are numerically comparable to the cwnd values (the transit
capacity is about 230).
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There is one large red tooth from about T=40 to T=90 that corresponds to three green teeth; from about
T=220 to T=275 there is one red tooth corresponding to two green teeth. Aside from these two points,
representing three isolated green-only losses, the red and green teeth appear quite well synchronized.
We also see evidence of loss-response clusters. At around T=143 the large green tooth peaks; halfway
down there is a little notch ending at T=145 that represents a fast-recovery loss event interpreted by TCP
as distinct. There is actually a third event, as well, representing a coarse timeout shortly after T=145, and
then a fourth fast-recovery event at about T=152 which we will examine shortly. At T80, the red tooth has
a fast-recovery loss event followed very shortly by a coarse timeout; this happens for both flows at about
T=220.
Overall, the red path has 11 teeth if we use a tooth-counting granularity of 3 seconds, and 8 teeth if the
granularity is 5 seconds. We do not count anything that happens in the slow-start phase, generally before
T=3.0, nor do we count the tooth at T=300 when the graph ends.
The slope of the green teeth is slightly greater than the slope of the red teeth, representing the longer RTT
for the red connection.
As for the queue graph, there is perhaps more noise than expected, but generally the right edges of the
teeth the TCP loss responses are very well aligned with peaks representing the queue filling up. Recall
that the transit capacity for flow1 here is about 230 packets and the queue capacity is about 20; we therefore
in general expect the sum of the two cwnds to range between 125 and 250, and that the queue should mostly
remain empty until the sum reached 230. We can indeed confirm visually that in most cases the tooth peaks
do indeed add up to about 250.
16.4.1.3 Transient queue peaks
The loss in the graph above at around T=152 is a little peculiar; this is fully 10 seconds after the primary
tooth peak that preceded it. Let us zoom in on it a little more closely, this time without any averaging of the
queue-utilization graph.

There are two very sharp, narrow peaks in the queue graph, just before T=146 and just after T=152, each
causing packet drops for both flows. Neither peak, especially the latter one, appears to have much to do with
the gradual queue-filling and loss event at T=143. Neither one of these transient queue peaks is associated
with the sum of the cwnds approaching the maximum of about 250; at the T146 peak the sum of the
cwnds is about 22+97 = 119 and at T152 the sum is about 21+42 = 63. ACKs for either sender cannot

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return from the destination D faster than the bottleneck-link rate of 1 packet/ms; this might suggest new
packets from senders A and B cannot arrive at R faster than a combined rate of 1 packet/ms. What is going
on?
What is happening is that each flow sends a flight of about 20 packets, spaced 1 ms apart, but coincidentally
timed so they begin arriving at R at the same moment. The runup in the queue near T=152 occurs from
T=152.100 to the first drop at T=152.121. During this 21 ms interval, a flight of 20 packets arrive from
node A (flow 0), and a flight of 19 packets arrive from node B (flow 1). These 39 packets in 21 ms means
the queue utilization at R must increase by 39-21 = 18, which is sufficient to overflow the queue as it was
not quite empty beforehand. The ACK flights that triggered these data-packet flights were indeed spaced 1
ms apart, consistent with the bottleneck link, but the ACKs (and the data-packet flights that triggered those
ACKs) passed through R at quite different times, because of the 25-ms difference in propagation delay on
the AR and BR links.
Transient queue peaks like this complicate any theory of relative throughput based on gradually filling the
queue. Fortunately, while transient peaks themselves are quite common (as can be seen from the zoomed
graph above), only occasionally do they amount to enough to overflow the queue. And, in the examples
created here, the majority of transient overflow peaks (including the one analyzed above) are within a few
seconds of a TCP coarse timeout response and may have some relationship to that timeout.
16.4.1.4 delayB = 61

For this graph, note that the red and green loss events at T=130 are not quite aligned; the same happens at
T=45 and T=100.
We also have several multiple-loss-response clusters, at T=40, T=130, T=190 and T=230.
The queue graph gives the appearance of much more solid yellow; this is due to a large number of transient
queue spikes. Under greater magnification it becomes clear these spikes are still relatively sparse, however.
There are transient queue overflows at T=46 and T=48, following a normal overflow at T=44, at T=101
following a normal overflow at T=99, at T=129 and T=131 following a normal overflow at T=126, and at
T=191 following a normal overflow at T=188.

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16.4.1.5 delayB = 120

Here we have (without the queue data) a good example of highly unsynchronized teeth: the green graph has
12 teeth, after the start, while the red graph has five. But note that the red-graph loss events are a subset of
the green loss events.

16.4.2 SACK TCP and Avoiding Loss Anomalies

Neither the coarse timeouts nor the clustered loss responses in the graphs above were anticipated in our
original fairness model in 14.3 TCP Fairness with Synchronized Losses. It is time to see if we can avoid
these anomalies and thus obtain behavior closer to the classic sawtooth. It turns out that SACK TCP fills the
bill quite nicely.
In the following subsection (16.4.2.1 Counting teeth in Python) is a script for counting loss responses
(teeth). If we run it on the tracefiles from our TCP Reno simulations with bottleneckBW = 8.0 and
time = 300, we find that, for each flow, about 30-35% of all loss responses are coarse timeouts. There is
little if any dependence on the value for delayB.
If we change the two tcp connections in the simulation to Agent/TCP/Newreno, the coarse-timeout
fraction falls by over half, to under 15%. This is presumably because TCP NewReno is better able to handle
multiple packet drops.
However, when we change the TCP connections to use SACK TCP, the situation improves dramatically.
We get essentially no coarse timeouts. Runs for 3000 seconds, typically involving 100-200 cwnd-halving
adjustments per flow, almost never have more than one coarse timeout and the majority of the time have
none.
Clusters of multiple loss responses also vanish almost entirely. In these 3000-second runs, there are usually
1-2 cases where two loss responses occur within 1.0 seconds (over 4 RTTs for the AD connection) of one
another.
SACK TCPs smaller number of packet losses results in a marked improvement in goodput. Total link
utilization ranges from 80.4% when delayB = 0 down to 68.3% when delayB = 400. For TCP Reno, the
corresponding utilization range is from 77.5% down to 51.5%.
Note that switching to SACK TCP is unlikely to have a significant impact on the distribution of packet
losses; SACK TCP differs from TCP Reno only in the way it responds to losses.
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The SACK TCP sender is specified in ns-2 via Agent/TCP/Sack1. The receiver must also be SACKaware; this is done by creating the receiver with Agent/TCPSink/Sack1.
When we switch to SACK TCP, the underlying fairness situation does not change much. Here is a graph
similar to that above in 16.3.10 Raising the Bandwidth. The run time is 3000 seconds, bottleneckBW
is 8 Mbps, and delayB runs from 0 to 400 in increments of 5. (We did not use an increment of 1, as in the
similar graph in 16.3.10 Raising the Bandwidth, because we are by now confident that phase effects have
been taken care of.)

Ratio1 is shown in blue and ratio2 in green. We again try to fit curves as in 16.3.10.1 Possible models above.
For the exponential model, goodput_ratio = K(RTT_ratio) , the value for the exponent comes out this
time to about 1.31, noticeably below TCP Renos value of 1.57. There remains, however, considerable
noise despite the less-frequent sampling interval, and it is not clear the difference is significant. The
second model, goodput_ratio 0.5 (RTT_ratio)2 , still also appears to remain a good fit.
16.4.2.1 Counting teeth in Python
Using our earlier Python nstrace.py module, we can easily count the times cwnd_ is reduced; these events
correspond to loss responses. Anticipating more complex scenarios, we define a Python class flowstats
to hold the per-flow information, though in this particular case we know there are exactly two flows.
We count every time cwnd_ gets smaller; we also keep a count of coarse timeouts. As a practical matter,
two closely spaced reductions in cwnd_ are always due to a fast-recovery event followed about a second
later by a coarse timeout. If we want a count of each separate TCP response, we count them both.
We do not count anything until after STARTPOINT. This is to avoid counting losses related to slow start.
#!/usr/bin/python3
import nstrace
import sys
# counts all points where cwnd_ drops.
STARTPOINT = 3.0

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class flowstats:
def __init__(self):
self.toothcount = 0
self.prevcwnd = 0
self.CTOcount = 0

# python object to hold per-flow data

# Coarse TimeOut count

def countpeaks(filename):
global STARTPOINT
nstrace.nsopen(filename)
flow0 = flowstats()
flow1 = flowstats()
while not nstrace.isEOF():
if nstrace.isVar():
# counting cwnd_ trace lines
(time, snode, dummy, dummy, dummy, varname, cwnd) = nstrace.getVar()
if (time < STARTPOINT):
continue
if varname != "cwnd_":
continue
if snode == 0: flow=flow0
else: flow=flow1
if cwnd < flow.prevcwnd:
flow.toothcount += 1
# count this as a tooth
if cwnd == 1.0: flow.CTOcount += 1
# coarse timeout
flow.prevcwnd=cwnd
else:
nstrace.skipline()

print ("flow 0 teeth:", flow0.toothcount, "flow 1 teeth:", flow1.toothcount)

print ("flow 0 coarse timeouts:", flow0.CTOcount, "flow 1 coarse timeouts:", flow1.CTOco
countpeaks(sys.argv[1])

A more elaborate version of this script, set up to count clusters of teeth and clusters of drops, is available in
teeth.py. If a new cluster starts at time T, all teeth/drops by time T + granularity are part of the same cluster;
the next tooth/drop after T + granularity starts the next new cluster.
16.4.2.2 Relative loss rates
At this point we accept that the AD/BD throughput ratio is generally smaller than the value predicted by
the synchronized-loss hypothesis, and so the AD flow must have additional losses to account for this. Our
next experiment is to count the loss events in each flow, and to identify the loss events common to both
flows.
The following graph demonstrates the rise in AD losses as a percentage of the total, using SACK TCP. We
use the tooth-counting script from the previous section, with a granularity of 1.0 sec. With SACK TCP it is
rare for two drops in the same flow to occur close to one another; the granularity here is primarily to decide
when two teeth in the two different flows are to be counted as shared.

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The blue region at the bottom represents the percentage of all loss-response events (teeth) that are shared
between both flows. The green region in the middle represents the percentage of all loss-response events
that apply only to the faster AD connection (flow 0); these rise to 30% very quickly and remain at about
50% as delayB ranges from just over 100 up to 400. The uppermost yellow region represents the loss
events that affected only the slower BD connection (flow 1); these are usually under 10%.
As a rough approximation, if we assume a 50/50 division between shared (blue) and flow-0-only (green)
loss events, we have losscount0 /losscount1 = = 2. Applying the formula of 14.5.2 Unsynchronized TCP
Losses, we get bandwidth0 /bandwidth1 = (RTT_ratio)2 /2, or ratio2 = 1/2. While it is not at all clear this
will continue to hold as delayB continues to increase beyond 400, it does suggest a possible underlying
explanation for the second formula of 16.3.10.1 Possible models.

16.4.3 Loss rate versus cwnd: part 2

In 14.5 TCP Reno loss rate versus cwnd we argued that the average value of cwnd was about K/ p, where
the constant K was somewhere between 1.225 and 1.309. In 16.2.6.5 Loss rate versus cwnd: part 1 above
we tested this for a single connection with very regular teeth; in this section we will test this hypothesis in
the two-connections simulation. In order to avoid coarse timeouts, which were not included in the original
model, we will use SACK TCP.
Over the lifetime of a connection, the average cwnd is the number of packets sent divided by the number
of RTTs. This simple relationship is slightly complicated by the fact that the RTT varies with the fullness
of the bottleneck queue, but, as before, this effect can be minimized if we choose models where RTTnoLoad
is large compared with the queuing delay. We will as before set bottleneckBW = 8.0; at this bandwidth
queuing delays add less than 10% to RTTnoLoad . For the longer-path flow1, RTTnoLoad 220 + 2delayB
ms. We will for both flows use the appropriate RTTnoLoad as an approximation for RTT, and will use the
number of packets acknowledged as an approximation for the number transmitted. These calculations give
the true average cwnd values in the table below. The estimated average cwnd values are from the formula

K/ p, where the loss ratio p is the total number of teeth, as counted earlier, divided by the number of that
flows packets.
With the relatively high value for bottleneckBW it takes a long simulation to get a reasonable number
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of losses. The simulations used here ran 3000 seconds, long enough that each connection ended up with
100-200 losses.
Here is the data, for each flow, using K=1.225. The bold columns represent the extent by which the ratio of
the estimated average cwnd to the true average cwnd differs from unity; these error values are reasonably
close to zero.
est avg
cwnd0
93.5
95.9
99.2
108.9
121.6
126.9
129.8

error0

0
10
20
40
70
100
150

true avg
cwnd0
89.2
92.5
95.6
104.9
117.0
121.9
125.2

4.8%
3.6%
3.7%
3.8%
3.9%
4.1%
3.7%

true avg
cwnd1
87.7
95.6
98.9
103.3
101.5
104.1
112.7

est avg
cwnd1
92.2
99.2
101.9
105.6
102.8
104.1
109.7

200

133.0

137.5

3.4%

95.9

93.3

250

133.4

138.2

3.6%

81.0

78.6

300

134.6

138.8

3.1%

74.2

70.9

350

135.2

139.1

2.9%

69.8

68.4

400

137.2

140.6

2.5%

60.4

58.5

delayB

error1
5.1%
3.8%
3.0%
2.2%
1.3%
0%
2.6%
2.7%
3.0%
4.4%
2.0%
3.2%

The table also clearly shows the rise in flow0s cwnd, cwnd0, and also the fall in cwnd1.
A related observation is that the absolute number of losses for either flow slowly declines as delayB
increases, at least in the delayB 400 range we have been considering. For flow 1, with the longer RTT,
this is because the number of packets transmitted drops precipitously (almost sevenfold) while cwnd and
therefore the loss-event probability p stays relatively constant. For flow 0, the total number of packets
sent rises as flow 1 drops to insignificance, but only by about 50%, and the average cwnd0 (above) rises
sufficiently fast (due to flow 1s decline) that
total_losses = total_sent p = total_sent K/cwnd2
generally does not increase.

16.4.4 Clusters of Packet Drops

During a multiple-packet-loss event involving two flows, just how many packets are lost from each flow? Are
these losses usually divided evenly between the flows? The synchronized-loss hypothesis simply requires
that each flow have at least one loss; it says nothing about how many.
As discussed in 16.4.2.1 Counting teeth in Python, we can use the tracefile to count clusters of packet
drops, given a suitable granularity value. We can also easily count the number of packet drops, for each

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flow, in each cluster. Each packet loss is indicated by an ns-2 event-trace line beginning with d. In our
simple arrangement, all losses are at the router R and all losses are of data (rather than ACK) packets.
At each drop cluster (loss event), each flow loses zero or more packets. For each pair of integers (X,Y)
we will count the number of drop clusters for which flow 0 experienced X losses and flow 1 experienced
Y losses. In the simplest model of synchronized losses, we might expect the majority of drop clusters to
correspond to the pair (1,1).
Whenever we identify a new drop cluster, we mark its start time (dcstart below); one cluster then consists
of all drop events from that time until one granularity interval of length DC_GRANULARITY has elapsed,
and the next loss after the end of that interval marks the start of the next drop cluster. The granularity interval
must be larger than the coarse-grained timeout interval and smaller than the tooth-to-tooth interval. Because
the timeout interval is often 1.0 sec, it is convenient to set DC_GRANULARITY to 1.5 or 2.0; in that range
the clusters we get are not very sensitive to the exact value.
We will represent a drop cluster as a Python tuple (pair) of integers (d0,d1) where d0 represents the
number of packet losses for flow 0 and d1 the losses for flow 1. As we start each new cluster, we add the
old one to a Python dictionary clusterdict, mapping each cluster to the count of how many times it has
occurred.
The code looks something like this, where dropcluster is initialized to the pair (0,0) and
addtocluster(c,f) adds to cluster c a drop for flow f. (The portion of the script here fails to include the final cluster in clusterdict.)

if nstrace.istrace():
# counting regular trace lines for DROP CLUSTERS
(event, time, sendnode, dnode, proto, dummy, dummy, flow, dummy, dummy, seqno, pktid
if (event==d):
# ignore all others
if (time > dcstart + DC_GRANULARITY):
# save old cluster, start ne
dcstart = time
if (dropcluster != (0,0)):
# dropcluster==(0,0) means i
inc_cluster(dropcluster, clusterdict)
# add dropcluster to
dropcluster = addtocluster((0,0), flow)
# start new cluster
else:
# add to dropcluster
dropcluster = addtocluster(dropcluster, flow)

At the end of the tracefile, we can print out clusterdict; if (f0,f1) had an occurrence count of N then
there were N drop events where flow 0 had f0 losses and flow 1 had f1. We can also count the number of
loss events that involved only flow 0 by adding up the occurrence counts of all drop clusters of the form
(x,0), that is, with 0 as the number of flow-1 losses in the cluster; similarly for flow 1.
A full Python script for this is at teeth.py.
If we do this for our earlier 16.3 Two TCP Senders Competing scenario, with bottleneckBW = 0.8 and
with a runtime of 3000 seconds, we find out that for equal paths, the two connections have completely
synchronized loss events (that is, no flow-0-only or flow-1-only losses) for overhead values of 0, 0.0012,
0.003, 0.0055 and 0.008 sec (recall the bottleneck-link packet send time is 0.01 sec). For all these overhead
values but 0.008, each loss event does indeed consist of exactly one loss for flow 0 and one loss for flow 1.
For overhead = 0.008, the final clusterdict is {(1,2):1, (1,1):674, (2,1):1}, meaning there were 674 loss
events with exactly one loss from each flow, one loss event with one loss from flow 0 and two from flow 1,
and one loss event with two losses from flow 0 and one from flow 1.
As overhead increases further, however, the final clusterdicts become more varied; below is a diagram
representing the dictionary for overhead = 0.02.
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The numbers are the occurrence counts for each dropcluster pair; for example, the entry of 91 in row 1
column 2 means that there were 91 times flow 0 had two losses and flow 1 had a single loss.

16.5 TCP Reno versus TCP Vegas

In 15.4 TCP Vegas we described an alternative to TCP Reno known as TCP Vegas; in 15.4.1 TCP Vegas
versus TCP Reno we argued that when the queue capacity was significant relative to the transit capacity,
TCP Vegas would be at a disadvantage, but competition might be fairer when the queue capacity was small.
Here we will test that theory.
In our standard model above with bottleneckBW = 8.0 Mbps, the transit capacity for the AD path is
220 packets. We will simulate a competition between TCP Reno and TCP Vegas for queue sizes from 10 to
220; as earlier, we will use overhead = 0.002.
In our first attempt, with the default TCP Vegas parameters of =1 and =3, things start out rather evenly:
when the queuesize is 10 the Reno/Vegas goodput ratio is 1.02. However, by the time the queue size is 220,
the ratio has climbed almost to 22; TCP Vegas is swamped. Raising and helps a little for larger queues
(perhaps at the expense of performance at small queue sizes); here is the graph for =3 and =6:

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The vertical axis plots the Reno-to-Vegas goodput ratio; the horizontal axis represents the queue capacity.
The performance of TCP Vegas falls off quite quickly as the queue size increases. One reason the larger
may help is that this slightly increases the range in which TCP Vegas behaves like TCP Reno.
To create an ns-2 TCP Vegas connection and set and one uses
set tcp1 [new Agent/TCP/Vegas]
$tcp1 set v_alpha_ 3
$tcp1 set v_beta_ 6

In prior simulations we have also made the following setting, in order to make the total TCP packetsize
including headers be 1000:
Agent/TCP set packetSize_ 960

It turns out that for TCP Vegas objects in ns-2, the packetSize_ includes the headers, as can be verified
by looking at the tracefile, and so we need
Agent/TCP/Vegas set packetSize_ 1000

Here is a cwnd-v-time graph comparing TCP Reno and TCP Vegas; the queuesize is 20, bottleneckBW
is 8 Mbps, overhead is 0.002, and =3 and =6. The first 300 seconds are shown. During this period the
bandwidth ratio is about 1.1; it rises to close to 1.3 (all in TCP Renos favor) when T=1000.

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The red plot represents TCP Reno and the green represents TCP Vegas. The green plot shows some spikes
that probably represent implementation artifacts.
Five to ten seconds before each sharp TCP Reno peak, TCP Vegas has its own softer peak. The RTT has
begun to rise, and TCP Vegas recognizes this and begins decrementing cwnd by 1 each RTT. At the point of
packet loss, TCP Vegas begins incrementing cwnd again. During the cwnd-decrement phase, TCP Vegas
falls behind relative to TCP Reno, but it may catch up during the subsequent increment phase because TCP
Vegas often avoids the cwnd = cwnd/2 multiplicative decrease and so often continues after a loss event
with a larger cwnd than TCP Reno.
We conclude that, for smaller bottleneck-queue sizes, TCP Vegas does indeed hold its own. Unfortunately,
in the scenario here the bottleneck-queue size has to be quite small for this to work; TCP Vegas suffers in
competition with TCP Reno even for moderate queue sizes. That said, queue capacities out there in the real
Internet tend to increase much more slowly than bandwidth, and there may be real-world situations were
TCP Vegas performs quite well when compared to TCP Reno.

16.6 Wireless Simulation

When simulating wireless networks, there are no links; all the configuration work goes into setting up the
nodes, the traffic and the wireless behavior itself. Wireless nodes have multiple wireless-specific attributes,
such as the antenna type and radio-propagation model. Nodes are also now in charge of packet queuing;
before this was the responsibility of the links. Finally, nodes have coordinates for position and, if mobility
is introduced, velocities.
For wired links the user must set the bandwidth and delay. For wireless, these are both generally provided
by the wireless model. Propagation delay is simply the distance divided by the speed of light. Bandwidth
is usually built in to the particular wireless model chosen; for the Mac/802_11 model, it is available in
attribute dataRate_ (which can be set). To find the current value, one can print [Mac/802_11 set
dataRate_]; in ns-2 version 2.35 it is 1mb.
Ad hoc wireless networks must also be configured with a routing protocol, so that paths may be found from
one node to another. We looked briefly at DSDV in 9.4.1 DSDV; there are many others.
The maximum range of a node is determined by its power level; this can be set with node-config
below (using the txPower attribute), but the default is often used. In the ns-2 source code, in file
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wireless-phy.cc, the variable Pt_ for transmitter power is declared; the default value of
0.28183815 translates to a physical range of 250 meters using the appropriate radio-attenuation model.
We create a simulation here in which one node (mover) moves horizontally above a sequence of fixedposition nodes (stored in the Tcl array rownodes). The leftmost fixed-position node transmits continuously
to the mover node; as the mover node progresses, packets must be routed through other fixed-position
nodes. The fixed-position nodes here are 200 m apart, and the mover node is 150 m above their line; this
means that the mover reaches the edge of the range of the ith rownode when it is directly above the i+1th
rownode.
We use Ad hoc On-demand Distance Vector (AODV) as the routing protocol. When the mover moves out
of range of one fixed-position node, AODV finds a new route (which will be via the next fixed-position
node) quite quickly; we return to this below. DSDV (9.4.1 DSDV) is much slower, which leads to many
packet losses until the new route is discovered. Of course, whether a given routing mechanism is fast enough
depends very much on the speed of the mover; the simulation here does not perform nearly as well if the
time is set to 10 seconds rather than 100 as the mover moves too fast even for AODV to keep up.
Because there are so many configuration parameters, to keep them together we adopt the common convention of making them all attributes of a single Tcl object, named opt.
We list the simulation file itself in pieces, with annotation; the complete file is at wireless.tcl. We begin with
the options.
# ======================================================================
# Define options
# ======================================================================
set opt(chan)
Channel/WirelessChannel ;# channel type
set opt(prop)
Propagation/TwoRayGround ;# radio-propagation model
set opt(netif)
Phy/WirelessPhy
;# network interface type
set opt(mac)
Mac/802_11
;# MAC type
set opt(ifq)
Queue/DropTail/PriQueue ;# interface queue type
set opt(ll)
LL
;# link layer type
set opt(ant)
Antenna/OmniAntenna
;# antenna model
set opt(ifqlen)
50
;# max packet in ifq
set
set
set
set
set
set

opt(bottomrow)
opt(spacing)
opt(mheight)
opt(brheight)
opt(x)
opt(y)

set opt(adhocRouting)
set opt(finish)

5
;# number of bottom-row nodes
200
;# spacing between bottom-row nodes
150
;# height of moving node above bottom-row
50
;# height of bottom-row nodes from bottom
[expr ($opt(bottomrow)-1)*$opt(spacing)+1]
;# x coordinate of to
300
;# y coordinate of topology
AODV
100

;# routing protocol
;# time to stop simulation

# the next value is the speed in meters/sec to move across the field
set opt(speed)
[expr 1.0*$opt(x)/$opt(finish)]

The Channel/WirelessChannel class represents the physical terrestrial wireless medium; there is
also a Channel/Sat class for satellite radio. The Propagation/TwoRayGround is a particular radiopropagation model. The TwoRayGround model takes into account ground reflection; for larger inter-node
distances d, the received power level is proportional to 1/d4 . Other models are the free-space model (in
which received power at distance d is proportional to 1/d2 ) and the shadowing model, which takes into
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account other types of interference. Further details can be found in the Radio Propagation Models chapter
of the ns-2 manual.
The Phy/WirelessPhy class specifies the standard wireless-node interface to the network; alternatives include Phy/WirelessPhyExt with additional options and a satellite-specific Phy/Sat. The
Mac/802_11 class specifies IEEE 802.11 (that is, Wi-Fi) behavior; other options cover things like generic
CSMA/CA, Aloha, and satellite. The Queue/DropTail/PriQueue class specifies the queuing behavior of each node; the opt(ifqlen) value determines the maximum queue length and so corresponds to
the queue-limit value for wired links. The LL class, for Link Layer, defines things like the behavior of
ARP on the network.
The Antenna/OmniAntenna class defines a standard omnidirectional antenna. There are many kinds
of directional antennas in the real world eg parabolic dishes and waveguide cantennas and a few have
been implemented as ns-2 add-ons.
The next values are specific to our particular layout. The opt(bottomrow) value determines the
number of fixed-position nodes in the simulation. The spacing between adjacent bottom-row nodes is
opt(spacing) meters. The moving node mover moves at height 150 meters above this fixed row.

When mover is directly above a fixed node, it is thus at distance (2002 + 1502 ) = 250 from the previous
fixed node, at which point the previous node is out of range. The fixed row itself is 50 meters above the
bottom of the topology. The opt(x) and opt(y) values are the dimensions of the simulation, in meters;
the number of bottom-row nodes and their spacing determine opt(x).
As mentioned earlier, we use the AODV routing mechanism. When the mover node moves out of range
of the bottom-row node that it is currently in contact with, AODV receives notice of the failed transmission
from the Wi-Fi link layer (ultimately this news originates from the absence of the Wi-Fi link-layer ACK).
This triggers an immediate search for a new route, which typically takes less than 50 ms to complete. The
earlier DSDV (9.4.1 DSDV) mechanism does not use Wi-Fi link-layer feedback and so does not look for
a new route until the next regularly scheduled round of distance-vector announcements, which might be
several seconds away. Other routing mechanisms include TORA, PUMA, and OLSR.
The finishing time opt(finish) also represents the time the moving node takes to move across all the
bottom-row nodes; the necessary speed is calculated in opt(speed). If the finishing time is reduced, the
mover speed increases, and so the routing mechanism has less time to find updated routes.
The next section of Tcl code sets up general bookkeeping:
# create the simulator object
set ns [new Simulator]
# set up tracing
$ns use-newtrace
set tracefd [open wireless.tr w]
set namtrace [open wireless.nam w]
$ns trace-all $tracefd
$ns namtrace-all-wireless $namtrace $opt(x) $opt(y)
# create and define the topography object and layout
set topo [new Topography]
$topo load_flatgrid $opt(x) $opt(y)
# create an instance of General Operations Director, which keeps track of nodes and

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# node-to-node reachability. The parameter is the total number of nodes in the simulation.
create-god [expr $opt(bottomrow) + 1]

The use-newtrace option enables a different tracing mechanism, in which each attribute except the first
is prefixed by an identifying tag, so that parsing is no longer position-dependent. We look at an example
below.
Note the special option namtrace-all-wireless for tracing for nam, and the dimension parameters
opt(x) and opt(y). The next step is to create a Topography object to hold the layout (still to be
determined). Finally, we create a General Operations Director, which holds information about the layout
not necessarily available to any node.
The next step is to call node-config, which passes many of the opt() parameters to ns and which
influences future node creation:
# general node configuration
set chan1 [new $opt(chan)]
$ns node-config -adhocRouting $opt(adhocRouting) \
-llType $opt(ll) \
-macType $opt(mac) \
-ifqType $opt(ifq) \
-ifqLen $opt(ifqlen) \
-antType $opt(ant) \
-propType $opt(prop) \
-phyType $opt(netif) \
-channel $chan1 \
-topoInstance $topo \
-wiredRouting OFF \
-agentTrace ON \
-routerTrace ON \
-macTrace OFF

Finally we create our nodes. The bottom-row nodes are created within a Tcl for-loop, and are stored in a
Tcl array rownode(). For each node we set its coordinates (X_, Y_ and Z_); it is at this point that the
rownode() nodes are given positions along the horizontal line y=50 and spaced opt(spacing) apart.
# create the bottom-row nodes as a node array $rownode(), and the moving node as $mover
for {set i 0} {$i < $opt(bottomrow)} {incr i} {
set rownode($i) [$ns node]
$rownode($i) set X_ [expr $i * $opt(spacing)]
$rownode($i) set Y_ $opt(brheight)
$rownode($i) set Z_ 0
}

set mover [$ns node]

$mover set X_ 0
$mover set Y_ [expr $opt(mheight) + $opt(brheight)]
$mover set Z_ 0

We now make the mover node move, using setdest. If the node reaches the destination supplied in
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setdest, it stops, but it is also possible to change its direction at later times using additional setdest
calls, if a zig-zag path is desired. Various external utilities are available to create a file of Tcl commands
to create a large number of nodes each with a designated motion; such a file can then be imported into the
main Tcl file.
set moverdestX [expr $opt(x) - 1]
$ns at 0 "$mover setdest $moverdestX [$mover set Y_] $opt(speed)"

Next we create a UDP agent and a CBR (Constant Bit Rate) application, and set up a connection from
rownode(0) to mover. CBR traffic does not use sliding windows.
# setup UDP connection, using CBR traffic
set udp [new Agent/UDP]
set null [new Agent/Null]
$ns attach-agent $rownode(0) $udp
$ns attach-agent $mover $null
$ns connect $udp $null
set cbr1 [new Application/Traffic/CBR]
$cbr1 set packetSize_ 512
$cbr1 set rate_ 200Kb
$cbr1 attach-agent $udp
$ns at 0 "$cbr1 start"
$ns at $opt(finish) "$cbr1 stop"

The remainder of the Tcl file includes additional bookkeeping for nam, a finish{} procedure, and the
startup of the simulation.
# tell nam the initial node position (taken from node attributes)
# and size (supplied as a parameter)
for {set i 0} {$i < $opt(bottomrow)} {incr i} {
$ns initial_node_pos $rownode($i) 10
}
$ns initial_node_pos $mover 20
# set the color of the mover node in nam
$mover color blue
$ns at 0.0 "$mover color blue"
$ns at $opt(finish) "finish"
proc finish {} {
global ns tracefd namtrace
$ns flush-trace
close $tracefd
close $namtrace
exit 0
}
# begin simulation

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$ns run

The simulation can be viewed from the nam file, available at wireless.nam. In the simulation, the mover
node moves across the topography, over the bottom-row nodes. The CBR traffic reaches mover from
rownode(0) first directly, then via rownode(1), then via rownode(1) and rownode(2), etc. The
motion of the mover node is best seen by speeding up the animation frame rate using the nam control for
this, though doing this means that aliasing effects often make the CBR traffic appear to be moving in the
opposite direction.

Above is one frame from the animation, with the mover node is almost (but not quite) directly over
rownode(3), and so is close to losing contact with rownode(2). Two CBR packets can be seen en
route; one has almost reached rownode(2) and one is about a third of the way from rownode(2) up to
the blue mover node. The packets are not shown to scale; see exercise 17.
The tracefile is specific to wireless networking, and even without the use of use-newtrace has a rather
different format from the link-based simulations earlier. The newtrace format begins with a letter for
send/receive/drop/forward; after that, each logged attribute is identified with a prefixed tag rather than by
position. Full details can be found in the ns-2 manual. Here is an edited record of the first packet drop (the
initial d indicates a drop-event record):
d -t 22.586212333 -Hs 0 -Hd 5 ... -Nl RTR -Nw CBK ... -Ii 1100 ... -Pn cbr -Pi 1100 ...

The -t tag indicates the time. The -Hs and -Hd tags indicate the source and destination, respectively.
The -Nl tag indicates the level (RouTeR) at which the loss was logged, and the -Nw tag indicates the
cause: CBK, for CallBacK, means that the packet loss was detected at the link layer but the information
was passed up to the routing layer. The -Ii tag is the packets unique serial number, and the P tags supply
information about the constant-bit-rate agent.
We can use the tracefile to find clusters of drops beginning at times 22.586, 47.575, 72.707 and 97.540,
corresponding roughly to the times when the route used to reach the $mover node shifts to passing through
one more bottom-row node. Between t=72.707 and t=97.540 there are several other somewhat more mysterious clusters of drops; some of these clusters may be related to ordinary queue overflow but others may
reflect decreasing reliability of the forwarding mechanism as the path grows longer.

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16.7 Epilog
Simulations using ns (either ns-2 or ns-3) are a central part of networks research. Most scientific papers
addressing comparisons between TCP flavors refer to ns simulations to at least some degree; ns is also
widely used for non-TCP research (especially wireless).
But simulations are seldom a matter of a small number of runs. New protocols must be tested in a wide range
of conditions, with varying bandwidths, delays and levels of background traffic. Head-to-head comparisons
in isolation, such as our first runs in 16.3.3 Unequal Delays, can be very misleading. Good simulation
design, in other words, is not easy.
Our simulations here involved extremely simple networks. A great deal of effort has been expended by the
ns community in the past decade to create simulations involving much larger sets of nodes; the ultimate goal
is to create realistic simulations of the Internet itself. We refer the interested reader to [FP01].

16.8 Exercises
1. In the graph in 16.2.1 Graph of cwnd v time, examine the trace file to see what accounts for the dot at
the start of each tooth (at times approximately 4.1, 6.1 and 8.0). Note that the solid parts of the graph are
as solid as they are because fractional cwnd values are used; cwnd is incremented by 1/cwnd on receipt of
each ACK.
2. A problem with the single-sender link-utilization experiment at 16.2.6 Single-sender Throughput Experiments was that the smallest practical value for queue-limit was 3, which is 10% of the path transit
capacity. Repeat the experiment but arrange for the path transit capacity to be at least 100 packets, making a
queue-limit of 3 much smaller proportionally. Be sure the simulation runs long enough that it includes
multiple teeth. What link utilization do you get? Also try queue-limit values of 4 and 5.
3. Create a single-sender simulation in which the path transit capacity is 90 packets, and the bottleneck
queue-limit is 30. This should mean cwnd varies between 60 and 120. Be sure the simulation runs
long enough that it includes many teeth.
1. What link utilization do you observe?
2. What queue utilization do you observe?
4. Use the basic2 model with equal propagation delays (delayB = 0), but delay the starting time for the
first connection. Let this delay time be startdelay0; at the end of the basic2.tcl file, you will have
$ns at $startdelay0 "$ftp0 start"

Try this for startdelay0 ranging from 0 to 40 ms, in increments of 1.0 ms. Graph the ratio1 or ratio2
values as a function of startdelay0. Do you get a graph like the one in 16.3.4.2 Two-sender phase
effects?
5. If the bottleneck link forwards at 10 ms/packet (bottleneckBW = 0.8), then in 300 seconds we can
send 30,000 packets. What percentage of this, total, are sent by two competing senders as in 16.3 Two TCP
Senders Competing, for delayB = 0, 50, 100, 200 and 400.
6. Repeat the previous exercise for bottleneckBW = 8.0, that is, a bottleneck rate of 1 ms/packet.

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7. Pick a case above where the total is less than 100%. Write a script that keeps track of cwnd0 + cwnd1 ,
and measure how much time this quantity is less than the transit capacity. What is the average of cwnd0 +
cwnd1 over those periods when it is less than the transit capacity?
8. In the model of 16.3 Two TCP Senders Competing, it is often the case that for small delayB, eg delayB
< 5, the longer-path BD connection has greater throughput. Demonstrate this. Use an appropriate value of
overhead (eg 0.02 or 0.002).
9. In generating the first graph in 16.3.7 Phase Effects and telnet traffic, we used a packetSize_ of
210 bytes, for an actualSize of 250 bytes. Generate a similar graph using in the simulations a much
smaller value of packetSize_, eg 10 or 20 bytes. Note that for a given value of tndensity, as the
actualSize shrinks so should the tninterval.
10. In 16.3.7 Phase Effects and telnet traffic the telnet connections ran from A and B to D, so the telnet
traffic competed with the bulk ftp traffic on the bottleneck link. Change the simulation so the telnet traffic
runs only as far as R. The variable tndensity should now represent the fraction of the AR and BR
bandwidth that is used for telnet. Try values for tndensity of from 5% to 50% (note these densities
are quite high). Generate a graph like the first graph in 16.3.7 Phase Effects and telnet traffic, illustrating
whether this form of telnet traffic is effective at reducing phase effects.
11. Again using the telnet simulation of the previous exercise, in which the telnet traffic runs only as far as
R, generate a graph comparing ratio1 for the bulk ftp traffic when randomization comes from:
overhead values in the range discussed in the text (eg 0.01 or 0.02 for bottleneckBW = 0.8
Mbps)
AR and BR telnet traffic with tndensity in the range 5% to 50%.
The goal should be a graph comparable to that of 16.3.8 overhead versus telnet. Do the two randomization
mechanisms overhead and telnet still yield comparable values for ratio1?
12. Repeat the same experiment as in exercise 8, but using telnet instead of overhead. Try it with AD/B
D telnet traffic and tndensity around 1%, and also AR/BR traffic as in exercise 10 with a tndensity
around 10%. The effect may be even more marked.
13. Analyze the packet drops for each flow for the Reno-versus-Vegas competition shown in the second
graph (red v green) of 16.5 TCP Reno versus TCP Vegas. In that simulation, bottleneckBW = 8.0
Mbps, delayB = 0, queuesize = 20, =3 and =6; the full simulation ran for 1000 seconds.

(a). How many drop clusters are for the Reno flow only?
(b). How many drop clusters are for the Vegas flow only?
(c). How many shared drop clusters are there?

Use a drop-cluster granularity of 2.0 seconds.

14. Generate a cwnd-versus-time graph for TCP Reno versus TCP Vegas, like the second graph in
16.5 TCP Reno versus TCP Vegas, except using the default =1. Does the TCP Vegas connection perform better or worse?
15. Compare two TCP Vegas connections as in 16.3 Two TCP Senders Competing, for delayB varying
from 0 to 400 (perhaps in steps of about 20). Use bottleneckBW = 8 Mbps, and run the simulations for
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at least 1000 seconds. Use the default and (usually =1 and =3). Is ratio1 1 or ratio2 1 a better
fit? How does the overall fairness compare with what ratio1 1 or ratio2 1 would predict? Does either
ratio appear roughly constant? If so, what is its value?
16. Repeat the previous exercise for a much larger value of , say =10. Set =+2, and be sure
queuesize is larger than 2 (recall that, in ns, is v_alpha_ and is v_beta_). If both connections manage to keep the same number of packets in the bottleneck queue at R, then both should get about
the same goodput, by the queue-competition rule of 14.2.2 Example 2: router competition. Do they? Is the
fairness situation better or worse than with the default and ?
17. In nam animations involving point-to-point links, packet lengths are displayed proportionally: if a link
has a propagation delay of 10 ms and a bandwidth of 1 packet/ms, then each packet will be displayed with
length 1/10 the link length. Is this true of wireless as well? Consider the animation (and single displayed
frame) of 16.6 Wireless Simulation. Assume the signal propagation speed is c 300 m/sec, that the nodes
are 300 m apart, and that the bandwidth is 1 Mbps.

(a). How long is a single bit? (That is, how far does the signal travel in the time needed to send a single
bit?)
(b). If a sender transmits continuously, how many bits will it send before its first bit reaches its destination?
(c). In the nam animation of 16.6 Wireless Simulation, is it plausible that what is rendered for the CBR
packets represents just the first bit of the packet? Is the scale about right for a single bit?
(d). What might be a more accurate animated representation of wireless packets?

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17 THE NS-3 NETWORK SIMULATOR

In this chapter we take a somewhat cursory look at the ns-3 simulator, intended as a replacement for ns-2.
The project is managed by the NS-3 Consortium, and all materials are available at nsnam.org.
Ns-3 represents a rather sharp break from ns-2. Gone is the Tcl programming interface; instead, ns-3 simulation programs are written in the C++ language, with extensive calls to the ns-3 library, although they are often
still referred to as simulation scripts. As the simulator core itself is also written in C++, this in some cases
allows improved interaction between configuration and execution. However, configuration and execution
are still in most cases quite separate: at the end of the simulation script comes a call Simulator::Run()
akin to ns-2s $ns run at which point the user-written C++ has done its job and the library takes over.
To configure a simple simulation, an ns-2 Tcl script had to create nodes and links, create network-connection
agents attached to nodes, and create traffic-generating applications attached to agents. Much the same
applies to ns-3, but in addition each node must be configured with its network interfaces, and each network
interface must be assigned an IP address.

17.1 Installing and Running ns-3

We here outline the steps for installing ns-3 under linux from the allinone tar file, assuming that all
prerequisite packages (such as gcc) are already in place. Much more general installation instructions can be
found at www.nsnam.org. In particular, serious users are likely to want to download the current Mercurial
repository directly. Information is also available for Windows and Macintosh installation, although perhaps
the simplest option for Windows users is to run ns-3 in a linux virtual machine.
The first step is to unzip the tar file; this should leave a directory named ns-allinone-3.nn, where nn reflects
the version number (20 in the authors installation as of this 2014 writing). This directory is the root of the
ns-3 system; it contains a build.py (python) script and the primary ns-3 directory ns-3.nn. All that is
necessary is to run the build.py script:
./build.py
Considerable configuration and then compiler output should ensue, hopefully terminating with a list of
Modules built and Modules not built.
From this point on, most ns-3 work will take place in the subdirectory ns-3.nn, that is, in ns-allinone-3.nn/ns3.nn. This development directory contains the source directory src, the script directory scratch, and the
execution script waf.
The development directory also contains a directory examples containing a rich set of example scripts.
The scripts in examples/tutorial are described in depth in the ns-3 tutorial in doc/tutorial.

17.1.1 Running a Script

Let us now run a script, for example, the file first.cc included in the examples/tutorial directory.
We first copy this file into the directory scratch, and then, in the parent development directory, enter the
command
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./waf --run first

The program is compiled and, if compilation is successful, is run.
In fact, every uncompiled program in the scratch directory is compiled, meaning that projects in progress
that are not yet compilable must be kept elsewhere. One convenient strategy is to maintain multiple project
directories, and link them symbolically to scratch as needed.
The ns-3 system includes support for command-line options; the following example illustrates the passing
by command line of the value 3 for the variable nCsma:
./waf --run "second --nCsma=3"

17.1.2 Compilation Errors

By default, ns-3 enables the -Werror option to the compiler, meaning that all warnings are treated as errors.
This is good practice for contributed or published scripts, but can be rather exasperating for beginners. To
disable this, edit the file waf-tools/cflags.py (relative to the development directory). Change the line
self.warnings_flags = [[-Wall], [-Werror], [-Wextra]]

to
self.warnings_flags = [[-Wall], [-Wextra]]

Then, in the development directory, run

./waf configure
./waf build

17.2 A Single TCP Sender

We begin by translating the single-TCP-sender script of 16.2 A Single TCP Sender. The full program is in
basic1.cc; we now review most of it line-by-line; some standard things such as #include directives are
omitted.
/*
Network topology:
A----R----B
A--R: 10 Mbps / 10 ms delay
R--B: 800Kbps / 50 ms delay
queue at R: size 7
/
*
using namespace ns3;
std::string fileNameRoot = "basic1";

// base name for trace files, etc

void CwndChange (Ptr<OutputStreamWrapper> stream, uint32_t oldCwnd, uint32_t newCwnd)

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{
*stream->GetStream () << Simulator::Now ().GetSeconds () << " " <<

newCwnd << std::endl;

static void
TraceCwnd ()
// Trace changes to the congestion window
{
AsciiTraceHelper ascii;
Ptr<OutputStreamWrapper> stream = ascii.CreateFileStream (fileNameRoot + ".cwnd");
Config::ConnectWithoutContext ("/NodeList/0/$ns3::TcpL4Protocol/SocketList/0/CongestionWi
}

The function TraceCwnd() arranges for tracing of cwnd; the function CwndChange is a callback, invoked by the ns-3 system whenever cwnd changes. Such callbacks are common in ns-3.
The parameter string beginning /NodeList/0/... is an example of the configuration namespace. Each
ns-3 attribute can be accessed this way. See 17.2.2 The Ascii Tracefile below.
int main (int argc, char *argv[])
{
int tcpSegmentSize = 1000;
Config::SetDefault ("ns3::TcpSocket::SegmentSize", UintegerValue (tcpSegmentSize));
Config::SetDefault ("ns3::TcpSocket::DelAckCount", UintegerValue (0));
Config::SetDefault ("ns3::TcpL4Protocol::SocketType", StringValue ("ns3::TcpReno"));
Config::SetDefault ("ns3::RttEstimator::MinRTO", TimeValue (MilliSeconds (500)));

The use of Config::SetDefault() allows us to configure objects that will not exist until some later point, perhaps not until the ns-3 simulator is running.
The first parameter
is an attribute string, of the form ns3::class::attribute.
A partial list of attributes is at
https://www.nsnam.org/docs/release/3.19/doxygen/group___attribute_list.html. Attributes of a class can
also be determined by a command such as the following:
./waf --run "basic1 --PrintAttributes=ns3::TcpSocket
The advantage of the Config::SetDefault mechanism is that often objects are created indirectly,
perhaps by helper classes, and so direct setting of class properties can be problematic.
It is perfectly acceptable to issue some Config::SetDefault calls, then create some objects (perhaps
implicitly), and then change the defaults (again with Config::SetDefault) for creation of additional
objects.
We pick the TCP congesion-control algorithm by setting ns3::TcpL4Protocol::SocketType.
Options are TcpRfc793 (no congestion control), TcpTahoe, TcpReno, TcpNewReno and
TcpWestwood. TCP Cubic and SACK TCP are not supported natively (though they are available if the
Network Simulation Cradle is installed).
Setting the DelAckCount attribute to 0 disables delayed ACKs. Setting the MinRTO value to 500 ms
avoids some unexpected hard timeouts. We will return to both of these below in 17.2.3 Unexpected
Timeouts and Other Phenomena.
Next comes our local variables and command-line-option processing. In ns-3 the latter is handled via
the CommandLine object, which also recognized the --PrintAttributes option above. Using the
--PrintHelp option gives a list of variables that can be set via command-line arguments.

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unsigned int runtime = 20;

int delayAR = 10;
int delayRB = 50;
double bottleneckBW= 0.8;
double fastBW = 10;
uint32_t queuesize = 7;
uint32_t maxBytes = 0;

//
//
//
//
//

seconds
ms
ms
Mbps
Mbps

// 0 means "unlimited"

CommandLine cmd;
// Here, we define command line options overriding some of the above.
cmd.AddValue ("runtime", "How long the applications should send data", runtime);
cmd.AddValue ("delayRB", "Delay on the R--B link, in ms", delayRB);
cmd.AddValue ("queuesize", "queue size at R", queuesize);
cmd.AddValue ("tcpSegmentSize", "TCP segment size", tcpSegmentSize);
cmd.Parse (argc, argv);
std::cout << "queuesize=" << queuesize << ", delayRB=" << delayRB << std::endl;

Next we create three nodes, illustrating the use of smart pointers and CreateObject().
Ptr<Node> A = CreateObject<Node> ();
Ptr<Node> R = CreateObject<Node> ();
Ptr<Node> B = CreateObject<Node> ();

Class Ptr is a smart pointer that manages memory through reference counting. The template function
CreateObject acts as the ns-3 preferred alternative to operator new. Parameters for objects created
this way can be supplied via Config::SetDefault, or by some later method call applied to the Ptr
object. For Node objects, for example, we might call A -> AddDevice(...).
A convenient alternative to creating nodes individually is to create a container of nodes all at once:
NodeContainer allNodes;
allNodes.Create(3);
Ptr<Node> A = allNodes.Get(0);
...

After the nodes are in place we create our point-to-point links, using the PointToPointHelper
class. We also create NetDeviceContainer objects; we dont use these here (we could simply call
AR.Install(A,R)), but will need them below when assigning IPv4 addresses.
// use PointToPointChannel and PointToPointNetDevice
NetDeviceContainer devAR, devRB;
PointToPointHelper AR, RB;
// create point-to-point link from A to R
AR.SetDeviceAttribute ("DataRate", DataRateValue (DataRate (fastBW * 1000 * 1000)));
AR.SetChannelAttribute ("Delay", TimeValue (MilliSeconds (delayAR)));
devAR = AR.Install(A, R);
// create point-to-point link from R to B
RB.SetDeviceAttribute ("DataRate", DataRateValue (DataRate (bottleneckBW * 1000 * 1000)));
RB.SetChannelAttribute ("Delay", TimeValue (MilliSeconds (delayRB)));

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RB.SetQueue("ns3::DropTailQueue", "MaxPackets", UintegerValue(queuesize));

devRB = RB.Install(R,B);

Next we hand out IPv4 addresses. The Ipv4AddressHelper class can help us with individual LANs (eg
AR and RB), but it is up to us to make sure our two LANs are on different subnets. If we attempt to put
A and B on the same subnet, routing will simply fail, just as it would if we were to do this with real network
nodes.
InternetStackHelper internet;
internet.Install (A);
internet.Install (R);
internet.Install (B);
// Assign IP addresses
Ipv4AddressHelper ipv4;
ipv4.SetBase ("10.0.0.0", "255.255.255.0");
Ipv4InterfaceContainer ipv4Interfaces;
ipv4Interfaces.Add (ipv4.Assign (devAR));
ipv4.SetBase ("10.0.1.0", "255.255.255.0");
ipv4Interfaces.Add (ipv4.Assign(devRB));
Ipv4GlobalRoutingHelper::PopulateRoutingTables ();

Next we print out the addresses assigned. This gives us a peek at the GetObject template and the ns-3
object-aggregation model. The original Node objects we created earlier were quite generic; they gained their
Ipv4 component in the code above. Now we retrieve that component with the GetObject<Ipv4>()
calls below.
Ptr<Ipv4> A4 = A->GetObject<Ipv4>(); // gets node As IPv4 subsystem
Ptr<Ipv4> B4 = B->GetObject<Ipv4>();
Ptr<Ipv4> R4 = R->GetObject<Ipv4>();
Ipv4Address Aaddr = A4->GetAddress(1,0).GetLocal();
Ipv4Address Baddr = B4->GetAddress(1,0).GetLocal();
Ipv4Address Raddr = R4->GetAddress(1,0).GetLocal();
std::cout
std::cout
std::cout
std::cout

<<
<<
<<
<<

"As
"Bs
"Rs
"Rs

address: " <<

address: " <<
#1 address: "
#2 address: "

Aaddr << std::endl;

Baddr << std::endl;
<< Raddr << std::endl;
<< R4->GetAddress(2,0).GetLocal() << std::endl;

In general, A->GetObject<T> returns the component of type T that has been aggregated to
Ptr<Object> A; often this aggregation is invisible to the script programmer but an understanding of
how it works is sometimes useful. The aggregation is handled by the ns-3 Object class, which contains
an internal list m_aggregates of aggregated companion objects. At most one object of a given type
can be aggregated to another, making GetObject<T> unambiguous. Given a Ptr<Object> A, we
can obtain an iterator over the aggregated companions via A->GetAggregateIterator(), of type
Object::AggregateIterator. From each Ptr<const Object> B returned by this iterator, we
can call B->GetInstanceTypeId().GetName() to get the class name of B.
The GetAddress() calls take two parameters; the first specfies the interface (a value of 0
gives the loopback interface) and the second distinguishes between multiple addresses assigned to

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the same interface (which is not happening here). The call A4->GetAddress(1,0) returns an
Ipv4InterfaceAddress object containing, among other things, an IP address, a broadcast address
and a netmask; GetLocal() returns the first of these.
Next we create the receiver on B, using a PacketSinkHelper. A receiver is, in essense, a read-only
form of an application server.
// create a sink on B
uint16_t Bport = 80;
Address sinkAaddr(InetSocketAddress (Ipv4Address::GetAny (), Bport));
PacketSinkHelper sinkA ("ns3::TcpSocketFactory", sinkAaddr);
ApplicationContainer sinkAppA = sinkA.Install (B);
sinkAppA.Start (Seconds (0.01));
// the following means the receiver will run 1 min longer than the sender app.
sinkAppA.Stop (Seconds (runtime + 60.0));
Address sinkAddr(InetSocketAddress(Baddr, Bport));

Now comes the sending application, on A. We must configure and create a BulkSendApplication,
attach it to A, and arrange for a connection to be created to B. The BulkSendHelper class simplifies this.
BulkSendHelper sourceAhelper ("ns3::TcpSocketFactory", sinkAddr);
sourceAhelper.SetAttribute ("MaxBytes", UintegerValue (maxBytes));
sourceAhelper.SetAttribute ("SendSize", UintegerValue (tcpSegmentSize));
ApplicationContainer sourceAppsA = sourceAhelper.Install (A);
sourceAppsA.Start (Seconds (0.0));
sourceAppsA.Stop (Seconds (runtime));

If we did not want to use the helper class here, the easiest way to create the BulkSendApplication
is with an ObjectFactory. We configure the factory with the type we want to create and the
relevant configuration parameters, and then call factory.Create(). (We could have used the
Config::SetDefault() mechanism and CreateObject() as well.)
ObjectFactory factory;
factory.SetTypeId ("ns3::BulkSendApplication");
factory.Set ("Protocol", StringValue ("ns3::TcpSocketFactory"));
factory.Set ("MaxBytes", UintegerValue (maxBytes));
factory.Set ("SendSize", UintegerValue (tcpSegmentSize));
factory.Set ("Remote", AddressValue (sinkAddr));
Ptr<Object> bulkSendAppObj = factory.Create();
Ptr<Application> bulkSendApp = bulkSendAppObj -> GetObject<Application>();
bulkSendApp->SetStartTime(Seconds(0.0));
bulkSendApp->SetStopTime(Seconds(runtime));
A->AddApplication(bulkSendApp);

The above gives us no direct access to the actual TCP connection. Yet another alternative is to start by
creating the TCP socket and connecting it:
Ptr<Socket> tcpsock = Socket::CreateSocket (A, TcpSocketFactory::GetTypeId ());
tcpsock->Bind();
tcpsock->Connect(sinkAddr);

However, there is then no mechanism for creating a BulkSendApplication that uses a pre-existing
socket. (For a workaround, see the tutorial example fifth.cc.)
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Before beginning execution, we set up tracing; we will look at the tracefile format later. We use the AR
PointToPointHelper class here, but both ascii and pcap tracing apply to the entire ARB network.
// Set up tracing
AsciiTraceHelper ascii;
std::string tfname = fileNameRoot + ".tr";
AR.EnableAsciiAll (ascii.CreateFileStream (tfname));
// Setup tracing for cwnd
Simulator::Schedule(Seconds(0.01),&TraceCwnd);
// this Time cannot be 0.0
// This tells ns-3 to generate pcap traces, including "-node#-dev#-" in filename
AR.EnablePcapAll (fileNameRoot);
// ".pcap" suffix is added automatically

This last creates four .pcap files, eg

basic1-0-0.pcap
basic1-1-0.pcap
basic1-1-1.pcap
basic1-2-0.pcap
The first number refers to the node (A=0, R=1, B=2) and the second to the interface. A packet arriving at
R but dropped there will appear in the second .pcap file but not the third. These files can be viewed with
WireShark.
Finally we are ready to start the simulator! The BulkSendApplication will stop at time runtime,
but traffic may be in progress. We allow it an additional 60 seconds to clear. We also, after the simulation
has run, print out the number of bytes received by B.
Simulator::Stop (Seconds (runtime+60));
Simulator::Run ();
Ptr<PacketSink> sink1 = DynamicCast<PacketSink> (sinkAppA.Get (0));
std::cout << "Total Bytes Received from A: " << sink1->GetTotalRx () << std::endl;
return 0;
}

17.2.1 Running the Script

When we run the script and plot the cwnd trace data (here for about 12 seconds), we get the following:

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Compare this graph to that in 16.2.1 Graph of cwnd v time produced by ns-2. The slow-start phase earlier
ended at around 2.0 and now ends closer to 3.0. There are several modest differences, including the halving
of cwnd just before T=1 and the peak around T=2.6; these were not apparent in the ns-2 graph.
After slow-start is over, the graphs are quite similar; cwnd ranges from 10 to 21. The period before was
1.946 seconds; here it is 2.0548; the difference is likely due to a more accurate implementation of the
recovery algorithm.
One striking difference is the presence of the near-vertical line of dots just after each peak. What is happening here is that ns-3 implements the cwnd inflation/deflation algorithm outlined at the tail end of 13.4 TCP
Reno and Fast Recovery. When three dupACKs are received, cwnd is set to cwnd/2 + 3, and is then allowed
to increase to 1.5cwnd. See the end of 13.4 TCP Reno and Fast Recovery.

17.2.2 The Ascii Tracefile

Below are four lines from the tracefile, starting with the record showing packet 271 (Seq=271001) being
dropped by R.
d
r
+
-

4.9823
4.98312
4.98312
4.98312

/NodeList/1/DeviceList/1/$ns3::PointToPointNetDevice/TxQueue/Drop ns3::PppHeader
/NodeList/2/DeviceList/0/$ns3::PointToPointNetDevice/MacRx ns3::Ipv4Header (tos 0
/NodeList/2/DeviceList/0/$ns3::PointToPointNetDevice/TxQueue/Enqueue ns3::PppHead
/NodeList/2/DeviceList/0/$ns3::PointToPointNetDevice/TxQueue/Dequeue ns3::PppHead

As with ns-2, the first letter indicates the action: r for received, d for dropped, + for enqueued, - for
dequeued. For Wi-Fi tracefiles, t is for transmitted. The second field represents the time.
The third field represents the name of the event in the configuration namespace, sometimes called the configuration path name. The NodeList value represents the node (A=0, etc), the DeviceList represents
the interface, and the final part of the name repeats the action: Drop, MacRx, Enqueue, Dequeue.
After that come a series of class names (eg ns3::Ipv4Header, ns3::TcpHeader), from the ns-3
attribute system, followed in each case by a parenthesized list of class-specific trace information.

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In the output above, the final three records all refer to node B (/NodeList/2/). Packet 258 has just
arrived (Seq=258001), and ACK 259001 is then enqueued and sent.

17.2.3 Unexpected Timeouts and Other Phenomena

In the discussion of the script above at 17.2
A Single TCP Sender we mentioned
that we set ns3::TcpSocket::DelAckCount to 0, to disable delayed ACKs, and
ns3::RttEstimator::MinRTO to 500 ms, to avoid unexpected timeouts.
If we comment out the line disabling delayed ACKs, little changes in our graph, except that the spacing
between consecutive TCP teeth now almost doubles to 3.776. This is because with delayed ACKs the
receiver sends only half as many ACKs, and the sender does not take this into account when incrementing
cwnd (that is, the sender does not implement the suggestion of RFC 3465 mentioned in 13.2.1 Per-ACK
Responses).
If we leave out the MinRTO adjustment, and set tcpSegmentSize to 960, we get a more serious problem:
the graph now looks something like this:

We can enable ns-3s internal logging in the TcpReno class by entering the commands below, before
running the script. (In some cases, as with WifiHelper::EnableLogComponents(), logging output
can be enabled from within the script.) Once enabled, logging output is written to stderr.
NS_LOG=TcpReno=level_info
export NS_LOG

The log output shows the initial dupACK at 8.54:

8.54069 [node 0] Triple dupack. Reset cwnd to 12960, ssthresh to 10080

But then, despite Fast Recovery proceding normally, we get a hard timeout:

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8.71463 [node 0] RTO. Reset cwnd to 960, ssthresh to 14400, restart from seqnum 510721

What is happening here is that the RTO interval was just a little too short, probably due to the use of the
awkward segment size of 960.
After the timeout, there is another triple-dupACK!
8.90344 [node 0] Triple dupack. Reset cwnd to 6240, ssthresh to 3360

Shortly thereafter, at T=8.98, cwnd is reset to 3360, in accordance with the Fast Recovery rules.
The overall effect is that cwnd is reset, not to 10, but to about 3.4 (in packets). This significantly slows
down throughput.
In recovering from the hard timeout, the sequence number is reset to Seq=510721 (packet 532), as this was
the last packet acknowledged. Unfortunately, several later packets had in fact made it through to B. By
looking at the tracefile, we can see that at T=8.7818, B received Seq=538561, or packet 561. Thus, when
A begins retransmitting packets 533, 534, etc after the timeout, Bs response is to send the ACK the highest
packet it has received, packet 561 (Ack=539521).
This scenario is not what the designers of Fast Recovery had in mind; it is likely triggered by a tooconservative timeout estimate. Still, exactly how to fix it is an interesting question; one approach might
be to ignore, in Fast Recovery, triple dupACKs of packets now beyond what the sender is currently sending.

17.3 Wireless
We next present the wireless simulation of 16.6 Wireless Simulation. The full script is at wireless.cc; the
animation output for the netanim player is at wireless.xml. As before, we have one mover node moving
horizontally 150 meters above a row of five fixed nodes spaced 200 meters apart. The limit of transmission is
set to be 250 meters, meaning that a fixed node goes out of range of the mover node just as the latter passes
over directly above the next fixed node. As before, we use Ad hoc On-demand Distance Vector (AODV) as
the routing protocol. When the mover passes over fixed node N, it goes out of range of fixed node N-1, at
which point AODV finds a new route to mover through fixed node N.

As in ns-2, wireless simulations tend to require considerably more configuration than point-to-point simulations. We now review the source code line-by-line. We start with two callback functions and the global
variables they will need to access.
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using namespace ns3;

Ptr<ConstantVelocityMobilityModel> cvmm;
double position_interval = 1.0;
std::string tracebase = "scratch/wireless";
// two callbacks
void printPosition()
{
Vector thePos = cvmm->GetPosition();
Simulator::Schedule(Seconds(position_interval), &printPosition);
std::cout << "position: " << thePos << std::endl;
}
void stopMover()
{
cvmm -> SetVelocity(Vector(0,0,0));
}

Next comes the data rate:

int main (int argc, char *argv[])
{
std::string phyMode = "DsssRate1Mbps";

The phyMode string represents the Wi-Fi data rate (and modulation technique). DSSS rates are
DsssRate1Mbps, DsssRate2Mbps, DsssRate5_5Mbps and DsssRate11Mbps. Also available
are ErpOfdmRate constants to 54 Mbps and OfdmRate constants to 150 Mbps with a 40 MHz bandwidth (GetOfdmRate150MbpsBW40MHz). All these are defined in src/wifi/model/wifi-phy.cc.
Next are the variables that determine the layout and network behavior. The factor variable allows slowing
down the speed of the mover node but correspondingly extending the runtime (though the new-routediscovery time is not scaled):
int
int
int
int

bottomrow = 5;
spacing = 200;
mheight = 150;
brheight = 50;

//
//
//
//

number of bottom-row nodes

between bottom-row nodes
height of mover above bottom row
height of bottom row

int X = (bottomrow-1)*spacing+1;
// X is the horizontal dimension of the field
int packetsize = 500;
double factor = 1.0; // allows slowing down rate and extending runtime; same total # of pa
int endtime = (int)100*factor;
double speed = (X-1.0)/endtime;
double bitrate = 80*1000.0/factor; // *average* transmission rate, in bits/sec
uint32_t interval = 1000*packetsize*8/bitrate*1000;
// in microsec
uint32_t packetcount = 1000000*endtime/ interval;
std::cout << "interval = " << interval <<", rate=" << bitrate << ", packetcount=" << packet

There are some niceties in calculating the packet transmission interval above; if we do it instead as
1000000*packetsize*8/bitrate then we sometimes run into 32-bit overflow problems or integer-divisionroundoff problems.
Now we configure some Wi-Fi settings.
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// disable fragmentation for frames below 2200 bytes

Config::SetDefault ("ns3::WifiRemoteStationManager::FragmentationThreshold", StringValue ("
// turn off RTS/CTS for frames below 2200 bytes
Config::SetDefault ("ns3::WifiRemoteStationManager::RtsCtsThreshold", StringValue ("2200"))
// Set non-unicast data rate to be the same as that of unicast
Config::SetDefault ("ns3::WifiRemoteStationManager::NonUnicastMode", StringValue (phyMode))

Here we create the mover node with CreateObject<Node>(), but the fixed nodes are created via a
NodeContainer, as is more typical with larger simulations
// Create nodes
NodeContainer fixedpos;
fixedpos.Create(bottomrow);
Ptr<Node> lowerleft = fixedpos.Get(0);
Ptr<Node> mover = CreateObject<Node>();

Now we put together a set of helper objects for more Wi-Fi configuration. We must configure both the
PHY (physical) and MAC layers.
// The below set of helpers will help us to put together the desired Wi-Fi behavior
WifiHelper wifi;
wifi.SetStandard (WIFI_PHY_STANDARD_80211b);
wifi.SetRemoteStationManager ("ns3::AarfWifiManager"); // Use AARF rate control

The AARF rate changes can be viewed by enabling the appropriate logging with, at the shell level before
./waf, NS_LOG=AarfWifiManager=level_debug. We are not otherwise interested in rate scaling
(3.3.3 Dynamic Rate Scaling) here, though.
The PHY layer helper is YansWifiPhyHelper. The YANS project (Yet Another Network Simulator)
was an influential precursor to ns-3; see [LH06]. Note the AddPropagationLoss configuration, where
we set the Wi-Fi range to 250 meters. The MAC layer helper is NqosWifiMacHelper; the nqos means
no quality-of-service, ie no use of Wi-Fi PCF (3.3.7 Wi-Fi Polling Mode).
// The PHY layer here is "yans"
YansWifiPhyHelper wifiPhyHelper = YansWifiPhyHelper::Default ();
// for .pcap tracing
// wifiPhyHelper.SetPcapDataLinkType (YansWifiPhyHelper::DLT_IEEE802_11_RADIO);

YansWifiChannelHelper wifiChannelHelper;
// *not* ::Default() !
wifiChannelHelper.SetPropagationDelay ("ns3::ConstantSpeedPropagationDelayModel"); // pld:
// the following has an absolute cutoff at distance > 250
wifiChannelHelper.AddPropagationLoss ("ns3::RangePropagationLossModel", "MaxRange", DoubleV
Ptr<YansWifiChannel> pchan = wifiChannelHelper.Create ();
wifiPhyHelper.SetChannel (pchan);
// Add a non-QoS upper-MAC layer "AdhocWifiMac", and set rate control
NqosWifiMacHelper wifiMacHelper = NqosWifiMacHelper::Default ();
wifiMacHelper.SetType ("ns3::AdhocWifiMac");
NetDeviceContainer devices = wifi.Install (wifiPhyHelper, wifiMacHelper, fixedpos);
devices.Add (wifi.Install (wifiPhyHelper, wifiMacHelper, mover));

At this point the basic Wi-Fi configuration is done! The next step is to work on the positions and motion.
First we establish the positions of the fixed nodes.

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MobilityHelper sessile;
// for fixed nodes
Ptr<ListPositionAllocator> positionAlloc = CreateObject<ListPositionAllocator> ();
int Xpos = 0;
for (int i=0; i<bottomrow; i++) {
positionAlloc->Add(Vector(Xpos, brheight, 0.0));
Xpos += spacing;
}
sessile.SetPositionAllocator (positionAlloc);
sessile.SetMobilityModel ("ns3::ConstantPositionMobilityModel");
sessile.Install (fixedpos);

Next we set up the mover node.

ConstantVelocityMobilityModel is a subclass of
MobilityModel. At the end we print out a couple things just for confirmation.

Vector pos (0, mheight+brheight, 0);

Vector vel (speed, 0, 0);
MobilityHelper mobile;
mobile.SetMobilityModel("ns3::ConstantVelocityMobilityModel");
// no Attributes
mobile.Install(mover);
cvmm = mover->GetObject<ConstantVelocityMobilityModel> ();
cvmm->SetPosition(pos);
cvmm->SetVelocity(vel);
std::cout << "position: " << cvmm->GetPosition() << " velocity: " << cvmm->GetVelocity() <<
std::cout << "mover mobility model: " << mobile.GetMobilityModelType() << std::endl;

Now we configure Ad hoc On-demand Distance Vector routing.

AodvHelper aodv;
OlsrHelper olsr;
Ipv4ListRoutingHelper listrouting;
//listrouting.Add(olsr, 10);
listrouting.Add(aodv, 10);

// generates less traffic

// fastest to find new routes

Uncommenting the olsr line (and commenting out the last line) is all that is necessary to change to OLSR
routing. OLSR is slower to find new routes, but sends less traffic.
Now we set up the IP addresses. This is straightforward as all the nodes are on a single subnet.
InternetStackHelper internet;
internet.SetRoutingHelper(listrouting);
internet.Install (fixedpos);
internet.Install (mover);
Ipv4AddressHelper ipv4;
NS_LOG_INFO ("Assign IP Addresses.");
ipv4.SetBase ("10.1.1.0", "255.255.255.0");
Ipv4InterfaceContainer i = ipv4.Assign (devices);

// there is only one subnet

Now we create a receiving application UdpServer on node mover, and a sending application
UdpClient on the lower-left node. These applications generate their own sequence numbers, which show
up in the ns-3 tracefiles marked with ns3::SeqTsHeader. As in 17.2 A Single TCP Sender, we use
Config::SetDefault() and CreateObject<>() to construct the applications.

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uint16_t port = 80;

// create a receiving application (UdpServer) on node mover
Address sinkaddr(InetSocketAddress (Ipv4Address::GetAny (), port));
Config::SetDefault("ns3::UdpServer::Port", UintegerValue(port));
Ptr<UdpServer> UdpRecvApp = CreateObject<UdpServer>();
UdpRecvApp->SetStartTime(Seconds(0.0));
UdpRecvApp->SetStopTime(Seconds(endtime+60));
mover->AddApplication(UdpRecvApp);
Ptr<Ipv4> m4 = mover->GetObject<Ipv4>();
Ipv4Address Maddr = m4->GetAddress(1,0).GetLocal();
std::cout << "IPv4 address of mover: " << Maddr << std::endl;
Address moverAddress (InetSocketAddress (Maddr, port));

Here is the UdpClient sending application:

Config::SetDefault("ns3::UdpClient::MaxPackets", UintegerValue(packetcount));
Config::SetDefault("ns3::UdpClient::PacketSize", UintegerValue(packetsize));
Config::SetDefault("ns3::UdpClient::Interval",
TimeValue (MicroSeconds (interval)));
Ptr<UdpClient> UdpSendApp = CreateObject<UdpClient>();
UdpSendApp -> SetRemote(Maddr, port);
UdpSendApp -> SetStartTime(Seconds(0.0));
UdpSendApp -> SetStopTime(Seconds(endtime));
lowerleft->AddApplication(UdpSendApp);

We now set up tracing. The first, commented-out line enables pcap-format tracing, which we do not need
here. The YansWifiPhyHelper object supports tracing only of receive (r) and transmit (t) records;
the PointtoPointHelper of 17.2 A Single TCP Sender also traced enqueue and drop records.
//wifiPhyHelper.EnablePcap (tracebase, devices);
AsciiTraceHelper ascii;
wifiPhyHelper.EnableAsciiAll (ascii.CreateFileStream (tracebase + ".tr"));
// create animation file, to be run with netanim
AnimationInterface anim (tracebase + ".xml");
anim.SetMobilityPollInterval(Seconds(0.1));

If we view the animation with netanim, the moving nodes motion is clear. The mover node, however,
sometimes appears to transmit back to both the fixed-row node below left and the fixed-row node below
right. These transmissions represent the Wi-Fi link-layer ACKs; they appear to be sent to two fixed-row
nodes because what netanim is actually displaying with its blue links is transmission every other node in
range.
We can also view the motion in text format by uncommenting the first line below.
//Simulator::Schedule(Seconds(position_interval), &printPosition);
Simulator::Schedule(Seconds(endtime), &stopMover);

Finally it is time to run the simulator, and print some final output.

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Simulator::Stop(Seconds (endtime+60));
Simulator::Run ();
Simulator::Destroy ();
int pktsRecd
std::cout <<
std::cout <<
std::cout <<

= UdpRecvApp->GetReceived();
"packets received: " << pktsRecd << std::endl;
"packets recorded as lost: " << (UdpRecvApp->GetLost()) << std::endl;
"packets actually lost: " << (packetcount - pktsRecd) << std::endl;

return 0;
}

17.3.1 Tracefile Analysis

The tracefile provides no enqueue records, and Wi-Fi doesnt have fixed links; how can we verify that
packets are being forwarded correctly? One thing we can do with the tracefile is to look at each value of the
UdpServer application sequence number, and record
when it was received by node mover
when it was transmitted by any fixed-row node
If we do this, we get output like the following:
packet
packet
...
packet
packet
packet
packet
...
packet
packet
packet
...
packet
packet
packet
packet
packet
...

0 received at 0.0248642, forwarded by 0 at 0.0201597

1 received at 0.0547045, forwarded by 0 at 0.05
499
500
501
502

received at 24.9506, forwarded by 0 at 24.95

NOT recd, forwarded by 0 at 25, forwarded by 0 at 25.0019, forwarded by 0 at 25.
received at 25.0864, forwarded by 0 at 25.0767, forwarded by 1 at 25.0817
received at 25.1098, forwarded by 0 at 25.1, forwarded by 1 at 25.1051

1000 NOT recd, forwarded by 0 at 50, forwarded by 1 at 50.001, forwarded by 1 at 50.

1001 received at 50.082, forwarded by 0 at 50.0683, forwarded by 1 at 50.0722, forwa
1002 received at 50.1107, forwarded by 0 at 50.1, forwarded by 1 at 50.101, forwarde
1499
1500
1501
1502
1503

received at 74.9525, forwarded by 0 at 74.95, forwarded by 1 at 74.951, forward

NOT recd, forwarded by 0 at 75, forwarded by 1 at 75.001, forwarded by 2 at 75.
NOT recd, forwarded by 0 at 75.05
received at 75.1484, forwarded by 0 at 75.1287, forwarded by 1 at 75.1299, forw
received at 75.1621, forwarded by 0 at 75.15, forwarded by 1 at 75.151, forward

That is, packets 0-499 were transmitted only by node 0. Packet 500 was never received by mover, but
there were seven transmission attempts; these seven attempts follow the rules described in 3.3.2 Wi-Fi and
Collisions. Packets starting at 501 were transmitted by node 0 and then later by node 1. Similarly, packet
1000 was lost, and after that each packet arriving at mover was first transmitted by nodes 0, 1 and 2, in that
order. In other words, packets are indeed being forwarded rightward along the line of fixed-row nodes until
a node is reached that is in range of mover.

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17.3.2 AODV Performance

If we change the line
listrouting.Add(aodv, 10);

to
listrouting.Add(dsdv, 10);

we find that the loss count goes from 4 packets out of 2000 to 398 out of 2000; for OLSR routing the loss
count is 426. As we discussed in 16.6 Wireless Simulation, the loss of one data packet triggers the AODV
implementation to look for a new route. The DSDV and OLSR implementations, on the other hand, only
look for new routes at regularly spaced intervals.

17.4 Exercises
In preparation.

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Is giving all control of congestion to the TCP layer really the only option? As the Internet has evolved,
so have situations in which we may not want routers handling all traffic on a first-come, first-served basis.
Traffic with delay bounds so-called real-time traffic, often involving either voice or video is likely to
perform much better with preferential service, for example; we will turn to this in 19 Quality of Service.
But even without real-time traffic, we might be interested in guaranteeing that each of several customers gets
an agreed-upon fraction of bandwidth, regardless of what the other customers are receiving. If we trust only
to TCP Renos bandwidth-allocation mechanisms, and if one customer has one connection and another has
ten, then the bandwidth received may also be in the ratio of 1:10. This may make the first customer quite
unhappy.
The fundamental mechanism for achieving these kinds of traffic-management goals in a shared network is
through queuing; that is, in deciding how the routers prioritize traffic. In this chapter we will take a look at
what router-based strategies are available; in the following chapter we will study how some of these ideas
have been applied to develop distributed quality-of-service options.
Previously, in 14.1 A First Look At Queuing, we looked at FIFO queuing both tail-drop and random-drop
variants and (briefly) at priority queuing. These are examples of queuing disciplines, a catchall term
for anything that supports a way to accept and release packets. The RED gateway strategy (14.8.3 RED
gateways) could qualify as a separate queuing discipline, too, although one closely tied to FIFO.
Queuing disciplines provide tools for meeting administratively imposed constraints on traffic. Two senders,
for example, might be required to share an outbound link equally, or in the proportion 60%-40%, even if
one participant would prefer to use 100% of the bandwidth. Alternatively, a sender might be required not to
send in bursts of more than 10 packets at a time.
Closely allied to the idea of queuing is scheduling: deciding what packets get sent when. Scheduling may
take the form of sending someone elses packets right now, or it may take the form of delaying packets that
are arriving too fast.
While priority queuing is one practical alternative to FIFO queuing, we will also look at so-called fair
queuing, in both flat and hierarchical forms. Fair queuing provides a straightforward strategy for dividing
bandwidth among multiple senders according to preset percentages.
Also introduced here is the token-bucket mechanism, which can be used for traffic scheduling but also for
traffic description.
Some of the material here in particular that involving fair queuing and the Parekh-Gallager theorem may
give this chapter a more mathematical feel than earlier chapters. Mostly, however, this is confined to the
proofs; the claims themselves are more straightforward.

18.1 Queuing and Real-Time Traffic

One application for advanced queuing mechanisms is to support real-time transport that is, traffic with
delay constraints on delivery.

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In its original conception, the Internet was arguably intended for non-time-critical transport. If you wanted
to place a digital phone call where every (or almost every) byte was guaranteed to arrive within 50 ms, your
best bet might be to use the (separate) telephone network instead.
And, indeed, having an entirely separate network for real-time transport is definitely a workable solution.
It is, however, expensive; there are many economies of scale to having just a single network. There is,
therefore, a great deal of interest in figuring out how to get the Internet to support real-time traffic directly.
The central strategy for mixing real-time and bulk traffic is to use queuing disciplines to give the real-time
traffic the service it requires. Priority queuing is the simplest mechanism, though the fair-queuing approach
below offers perhaps greater flexibility.
We round out the chapter with the Parekh-Gallager theorem, which provides a precise delay bound for realtime traffic that shares a network with bulk traffic. All that is needed is that the real-time traffic satisfies a
token-bucket specification and is assigned bandwidth guarantees through fair queuing; the volume of bulk
traffic does not matter. This is exactly what is needed for real-time support.
While this chapter contains some rather elegant theory, it is not at all clear how much it is put into practice
today, at least for real-time traffic at the ISP level. We will return to this issue in the following chapter, but
for now we acknowledge that real-time traffic management using the queuing mechanisms described here
has seen limited acceptance in the Internet marketplace.

18.2 Traffic Management

Even if none of your traffic has real-time constraints, you still may wish to allocate bandwidth according
to administratively determined percentages. For example, you may wish to give each of three departments
an equal share of download (or upload) capacity, or you may wish to guarantee them shares of 55%, 35%
and 10%. If you want any unused capacity to be divided among the non-idle users, fair queuing is the
tool of choice, though in some contexts it may require cooperation from your ISP. If the users are more
like customers receiving only the bandwidth they pay for, you might want to enforce flat caps even if some
bandwidth thus goes unused; token-bucket filtering would then be the way to go. If bandwidth allocations
are not only by department (or customer) but also by workgroup (or customer-specific subcategory), then
hierarchical queuing offers the necessary control.
In general, network management divides into managing the hardware and managing the traffic; the tools in
this chapter address this latter component. These tools can be used internally by ISPs and at the customer/ISP
interconnection, but traffic management often makes good economic sense even when entirely contained
within a single organization.

18.3 Priority Queuing

To get started, let us fill in the details for priority queuing, which we looked at briefly in 14.1.1 Priority
Queuing. Here a given outbound interface can be thought of as having two (or more) physical queues representing different priority levels. Packets are placed into the appropriate subqueue based on some packet
attribute, which might be an explicit priority tag, or which might be the packets destination socket. Whenever the outbound link becomes free and the router is able to send the next packet, it always looks first to

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the higher-priority queue; if it is nonempty then a packet is dequeued from there. Only if the higher-priority
queue is empty is the lower-priority queue served.
Note that priority queuing is nonpreemptive: if a high-priority packet arrives while a low-priority packet is
being sent, the latter is not interrupted. Only when the low-priority packet has finished transmission does
the router again check its high-priority subqueue(s).
Priority queuing can lead to complete starvation of low-priority traffic, but only if the high-priority traffic
consumes 100% of the outbound bandwidth. Often we are able to guarantee (for example, through admission
control) that the high-priority traffic is limited to a designated fraction of the total outbound bandwidth.

18.4 Queuing Disciplines

As an abstract data type, a queuing discipline is simply a data structure that supports the following operations:
enqueue()
dequeue()
is_empty()
Note that the enqueue() operation includes within it a way to handle dropping a packet in the event that
the queue is full. For FIFO queuing, the enqueue() operation needs only to know the correct outbound
interface; for priority queuing enqueue() also needs to be told or be able to infer the packets priority
classification.
We may also in some cases find it convenient to add a peek() operation to return the next packet that
would be dequeued if we were actually to do that, or at least to return some important statistic (eg size or
arrival time) about that packet.
As with FIFO and priority queuing, any queuing discipline is always tied to a specific outbound interface.
In that sense, any queuing discipline has a single output.
On the input side, the situation may be more complex. The FIFO queuing discipline has a single input
stream, though it may be fed by multiple physical input interfaces: the enqueue() operation puts all
packets in the same physical queue. A queuing discipline may, however, have multiple input streams; we
will call these classes, or subqueues, and will refer to the queuing discipline itself as classful. Priority
queues, for example, have an input class for each priority level.
When we want to enqueue a packet for a classful queuing discipline, we must first invoke a classifier
possibly external to the queuing discipline itself to determine the input class. (In the linux documentation,
what we have called classifiers are often called filters.) For example, if we wish to use a priority queue to give
priority to VoIP packets, the classifiers job is to determine which arriving packets are in fact VoIP packets
(perhaps taking into account things like size or port number or source host), so as to be able to provide this
information to the enqueue() operation. The classifier might also take into account pre-existing traffic
reservations, so that packets that belong to flows with reservations get preferred service, or else packet tags
that have been applied by some upstream router; we return to both of these in 19 Quality of Service.
The number and configuration of classes is often fixed at the time of queuing-discipline creation; this is
typically the case for priority queues. Abstractly, however, the classes can also be dynamic; an example of

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this might be fair queuing (below), which often supports a configuration in which a separate input class is
created on the fly for each separate TCP connection.
FIFO and priority queuing are both work-conserving, meaning that the associated outbound interface is not
idle unless the queue is empty. A non-work-conserving queuing discipline might, for example, artificially
delay some packets in order to enforce an administratively imposed bandwidth cap. Non-work-conserving
queuing disciplines are often called traffic shapers or policers; see 18.9 Token Bucket Filters below for an
example.

18.5 Fair Queuing

An important alternative to FIFO and priority is fair queuing. Where FIFO and its variants have a single
input class and put all the incoming traffic into a single physical queue, fair queuing maintains a separate
logical FIFO subqueue for each input class; we will refer to these as the per-class subqueues. Division into
classes can be fine-grained eg a separate class for each TCP connection or coarse-grained eg a separate
class for each arrival interface, or a separate class for each designated internal subnet.
Suppose for a moment that all packets are the same size; this makes fair queuing much easier to visualize.
In this (special) case sometimes called Nagle fair queuing, and proposed in RFC 970 the router simply
services the per-class subqueues in round-robin fashion, sending one packet from each in turn. If a per-class
subqueue is empty, it is simply skipped over. If all per-class subqueues are always nonempty this resembles
time-division multiplexing (4.2 Time-Division Multiplexing). However, unlike time-division multiplexing
if one of the per-class subqueues does become empty then it no longer consumes any outbound bandwidth.
Recalling that all packets are the same size, the total bandwidth is then divided equally among the nonempty
per-class subqueues; if there are K such queues, each will get 1/K of the output.
Fair queuing was extended to streams of variable-sized packets in [DKS89] and [LZ89]. Since then there
has been considerable work in trying to figure out how to implement fair queuing efficiently and to support
appropriate variants.

18.5.1 Weighted Fair Queuing

A straightforward extension of fair queuing is weighted fair queuing (WFQ), where instead of giving each
class an equal share, we assign each class a different percentage. For example, we might assign bandwidth
percentages of 10%, 30% and 60% to three different departments. If all three subqueues are active, each gets
the listed percentage. If the 60% subqueue is idle, then the others get 25% and 75% respectively, preserving
the 1:3 ratio of their allocations. If the 10% subqueue is idle, then the other two subqueues get 33.3% and
66.7%.
Weighted fair queuing is, conceptually, a straightforward generalization of fair queuing, although the actual
implementation details are sometimes nontrivial as the round-robin implementation above naturally yields
equal shares. If all packets are still the same size, and we have two per-class subqueues that are to receive
allocations of 40% and 60% (that is, in the ratio 2:3), then we could implement WFQ by having one perclass subqueue send two packets and the other three. Or we might intermingle the two: class 1 sends its first
packet, class 2 sends its first packet, class 1 sends its second, class 2 sends its second and its third. If the

allocation is to be in the ratio 1: 2, the first sender might always send 1 packet while the second might send

in a pattern an irregular one that averages 2: 1, 2, 1, 2, 1, 1, 2, ....

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The fluid-based GPS model approach to fair queuing, 18.5.4 The GPS Model, does provide an algorithm
that has direct, natural support for weighted fair queuing.

18.5.2 Different Packet Sizes

If not all the packets are the same size, fair queuing and weighted fair queuing are still possible but we have
a little more work to do. This is an important practical case, as fair queuing is often used when one input
class consists of small-packet real-time traffic.
We present two mechanisms for handling different-sized packets; the two are ultimately equivalent. The first
18.5.3 Bit-by-bit Round Robin is a straightforward extension of the round-robin idea, and the second
18.5.4 The GPS Model uses a fluid model of simultaneous packet transmission. Both mechanisms
share the idea of a virtual clock that runs at a rate proportional to the number of active subqueues; as we
shall see, the point of varying the clock rate in this way is so that the virtual-clock time at which a given
packet would theoretically finish transmission does not depend on activity in any of the other subqueues.
Finally, we present the quantum algorithm 18.5.5 The Quantum Algorithm which is a more-efficient
approximation to either of the exact algorithms, but which being an approximation no longer satisfies a
sometimes-important time constraint.
For a straightforward generalization of the round-robin idea to different packet sizes, we start with a simplification: let us assume that each per-class subqueue is always active, where a subqueue is active if it is
nonempty whenever the router looks at it.
If each subqueue is always active for the equal-sized-packets case, then packets are transmitted in order
of increasing (or at least nondecreasing) cumulative data sent by each subqueue. In other words, every
subqueue gets to send its first packet, and only then do we go on to begin transmitting second packets, and
so on.
Still assuming each subqueue is always active, we can handle different-sized packets by the same idea. For
packet P, let CP be the cumulative number of bytes that will have been sent by Ps subqueue as of the end of
P. Then we simply need to send packets in nondecreasing order of CP .

In the diagram above, transmission in nondecreasing order of CP means transmission in left-to-right order
of the vertical lines marking packet divisions, eg Q1, P1, R1, Q2, Q3, R2, .... This ensures that, in the long
run, each subqueue gets an equal share of bandwidth.
A completely equivalent strategy, better suited for generalization to the case where not all subqueues are
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always active, is to send each packet in nondecreasing order of virtual finishing times, calculated for each
packet with the assumption that only that packets subqueue is active. The virtual finishing time FP of packet
P is equal to CP divided by the output bandwidth. We use finishing times rather than starting times because
if one packet is very large, shorter packets in other subqueues that would finish sooner should be sent first.
18.5.2.1 A first virtual-finish example
As an example, suppose there are two subqueues, P and Q. Suppose further that a stream of 1001-byte
packets P1 , P2 , P3 , ... arrives for P, and a stream of 400-byte packets Q1 , Q2 , Q3 , ... arrives for Q; each
stream is steady enough that each subqueue is always active. Finally, assume the output bandwidth is 1 byte
per unit time, and let T=0 be the starting point.
For the P subqueue, the virtual finishing times calculated as above would be P1 at 1001, P2 at 2002, P3
at 3003, etc; for Q the finishing times would be Q1 at 400, Q2 at 800, Q3 at 1200, etc. So the order of
transmission of all the packets together, in increasing order of virtual finish, will be as follows:
Packet
Q1
Q2
P1
Q3
Q4
Q5
P2

virtual finish
400
800
1001
1200
1600
2000
2002

actual finish
400
800
1801
2201
2601
3001
4002

For each packet we have calculated in the table above its virtual finishing time, and then its actual wallclock
finishing time assuming packets are transmitted in nondecreasing order of virtual finishing time (as shown).
Because both subqueues are always active, and because the virtual finishing times assumed each subqueue
received 100% of the output bandwidth, in the long run the actual finishing times will be about double the
virtual times. This, however, is irrelevant; all that matters is the relative virtual finishing times.
18.5.2.2 A second virtual-finish example
For the next example, however, we allow a subqueue to be idle for a while and then become active; in this
situation virtual finishing times do not work quite so well. We return to our initial simplification that all
packets are the same size, which we take to be 1 unit; this allows us to apply the round-robin mechanism to
determine the transmission order and compare this to the virtual-finish order. Assume there are three queues
P, Q and R, and P is empty until wallclock time 20. Q is constantly busy; its Kth packet QK has virtual
finishing time FK = K.
For the first case, assume R is completely idle. When Ps first packet P1 arrives at time 20, its virtual
finishing time will be 21. At time 20 the head packet in Q will be Q21 ; the two packets therefore have
identical virtual finishing times. And, encouragingly, under round-robin queue service P1 and Q21 will be
sent in the same round.
For the second case, however, suppose R is also constantly busy. Up until time 20, Q and R have each sent
10 packets; their next packets are Q11 and R11 , with virtual finishing times of T=11. When Ps first packet
arrives at T=20, again with virtual finishing time 21, under round-robin service it should be sent in the same

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round as Q11 and R11 . Yet their virtual finishing times are off by a factor of about two; queue Ps stretch of
inactivity has left it far behind. Virtual finishing times, as we have been calculating them so far, simply do
not work.
The trick, as it turns out, is to measure elapsed time not in terms of packet-transmission times (ie wallclock
time), but rather in terms of rounds of round-robin transmission. This amounts to scaling the clock used
for measuring arrival times; counting in rounds rather than packets means that we run this clock at rate 1/N
when N subqueues are active. If we do this in case 1, with N=1, then the finishing times are unchanged.
However, in case 2, with N=2, packet P1 arrives after 20 time units but only 10 rounds; the clock runs at half
rate. Its calculated finishing time is thus 11, exactly matching the finishing times of the two long-queued
packets Q11 and R11 with which P1 shares a round-robin transmission round.
We formalize this in the next section, extending the idea to include both variable-sized packets and
sometimes-idle subqueues. Note that only the clock that measures arrival times is scaled; we do not scale
the calculated transmission times.

18.5.3 Bit-by-bit Round Robin

Imagine sending a single bit at a time from each active input subqueue, in round-robin fashion. While not
directly implementable, this certainly meets the goal of giving each active subqueue equal service, even if
packets are of different sizes. We will use bit-by-bit round robin, or BBRR, as a way of modeling packetfinishing times, and then, as in the previous example, send the packets the usual way one full packet at a
time in order of increasing BBRR-calculated virtual finishing times.
It will sometimes happen that a larger packet is being transmitted at the point a new, shorter packet arrives
for which a smaller finishing time is computed. The current transmission is not interrupted, though; the
algorithm is non-preemptive.
The trick to making the BBRR approach workable is to find an invariant formulation of finishing time that
does not change as later packets arrive, or as other subqueues become active or inactive. To this end, taking
the lead from the example of the previous section, we define the rounds counter R(t), where t is the time
measured in units of the transmission time for one bit. When there are any active (nonempty or currently
transmitting) input subqueues, R(t) counts the number of round-robin 1-bit cycles that have occurred since
the last time all the subqueues were empty. If there are K active input subqueues, then R(t) increments by 1
as t increments by K; that is, R(t) grows at rate 1/K.
An important attribute of R(t) is that, if a packet of size S bits starts transmission via BBRR at R0 = R(t0 ),
then it will finish when R(t) = R0 +S, regardless of whether any other input subqueues become active or
become empty. For any packet actively being sent via BBRR, R(t) increments by 1 for each bit of that
packet sent. If for a given round-robin cycle there are K subqueues active, then K bits will be sent in all, and
R(t) will increment by 1.
To calculate the virtual BBRR finishing time of a packet P, we first record RP = R(tP ) at the moment of
arrival. We now compute the BBRR-finishing R-value FP as follows; we can think of this as a time
measured via the rounds counter R(t). That is, R(t) represents a virtual clock that happens sometimes to
run slow. Let S be the size of the packet P in bits. If P arrived on a previously empty input subqueue, then its
BBRR transmission can begin immediately, and so its finishing R-value FP is simply RP +S. If the packets
subqueue was nonempty, we look up the (future) finishing R-value of the packet immediately ahead of P in
its subqueue, say Fprev ; the finishing R-value of P is then FP = Fprev + S. This is sometimes described as:

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Start = max(R(now), Fprev )

FP = Start + S
(S = packet size, measured in bits)
As each new packet P arrives, we calculate its BBRR-finishing R-value FP , and then send packets the
conventional one-packet-at-a-time way in increasing order of FP . As stated above, FP will not change if
other subqueues empty or become active, thus changing the rate of the rounds-counter R(t).
The router maintaining R(t) does not have to increment it on every bit; it suffices to update it whenever a
packet arrives or a subqueue becomes empty. If the previous value of R(t) was Rprev , and from then to now
exactly K subqueues were nonempty, and M bit-times have elapsed according to the wall clock, then the
current value of R(t) is Rprev + M/K.
18.5.3.1 BBRR example
As an example, suppose the fair queuing router has three input subqueues P, Q and R, initially empty. The
following packets arrive at the wall-clock times shown.
Packet
P1
P2
Q1
Q2
Q3
R1
R2

Queue
P
P
Q
Q
Q
R
R

Size
1000
1000
600
400
400
200
200

Arrival time, t
0
0
800
800
800
1200
2100

At t=0, we have R(t)=0 and we assign finishing R-values F(P1 )=1000 to P1 and F(P2 ) = F(P1 )+1000 = 2000
to P2 . Transmission of P1 begins.
When the three Q packets arrive at t=800, we have R(t)=800 as well, as only one subqueue has been active.
We assign finishing R-values for the newly arriving Q1 , Q2 and Q3 of F(Q1 ) = 800+600 = 1400, F(Q2 ) =
1400+400 = 1800, and F(Q3 ) = 1800+400 = 2200. At this point, BBRR begins serving two subqueues, so
the R(t) rate is cut in half.
At t=1000, transmission of packet P1 is completed; R(t) is 800 + 200/2 = 900. The smallest finishing R-value
on the books is F(Q1 ), at 1400, so Q1 is the second packet transmitted. Q1 s real finishing time will be t =
1000+600 = 1600.
At t=1200, R1 arrives; transmission of Q1 is still in progress. R(t) is 800 + 400/2 = 1000; we calculate F(R1 )
= 1000 + 200 = 1200. Note this is less than the finishing R-value for Q1 , which is currently transmitting, but
Q1 is not preempted. At this point (t=1200, R(t)=1000), the R(t) rate falls to 1/3.
At t=1600, Q1 has finished transmission. We have R(t) = 1000 + 400/3 = 1133. The next smallest finishing
R-value is F(R1 ) = 1200 so transmission of R1 commences.
At t=1800, R1 finishes. We have R(1800) = R(1200) + 600/3 = 1000 + 200 = 1200 (3 subqueues have been
busy since t=1200). Queue R is now empty, so the R(t) rate rises from 1/3 to 1/2. The next smallest finishing
R-value is F(Q2 )=1800, so transmission of Q2 begins. It will finish at t=2200.
At t=2100, we have R(t) = R(1800) + 300/2 = 1200 + 150 = 1350. R2 arrives and is assigned a finishing time
of F(R2 ) = 1350 + 200 = 1550. Again, transmission of Q2 is not preempted even though F(R2 ) < F(Q2 ). The
R(t) rate again falls to 1/3.
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At t=2200, Q2 finishes. R(t) = 1350 + 100/3 = 1383. The next smallest finishing R-value is F(R2 )=1550, so
transmission of R2 begins.
At t=2400, transmission of R2 ends. R(t) is now 1350 + 300/3 = 1450. The next smallest finishing R-value
is F(P2 ) = 2000, so transmission of P2 begins. The R(t) rate rises to 1/2, as queue R is again empty.
At t=3400, transmission of P2 ends. R(t) is 1450 + 1000/2 = 1950. The only remaining unsent packet is Q3 ,
with F(Q3 )=2200. We send it.
At t=3800, transmission of Q3 ends. R(t) is 1950 + 400/1 = 2350.
To summarize:
Packet
P1
Q1
R1
Q2
R2
P2
Q3

send-time, wall clock

t
0
1000
1600
1800
2200
2400
3400

calculated finish
R-value
1000
1400
1200*
1800
1550*
2000
2200

R-value when
sent
0
900
1133
1200
1383
1450
1950

R-value at
finish
900
1133
1200
1383
1450
1950
2350

Packets arrive, begin transmission and finish in real time. However, the number of queues active in real
time affects the rate of the rounds-counter R(t); this value is then attached to each packet as it arrives as its
virtual finishing time, and determines the order of packet transmission.
The change in R-value from start to finish exactly matches the packet size when the packet is virtually sent
via BBRR. When the packet is sent as an indivisible unit, as in the table above, the change in R-value is
usually much smaller, as the R-clock runs slower whenever at least two subqueues are in use.
The calculated-finish R-values are not in fact increasing, as can be seen at the starred (*) values. This is
because, for example, R1 was not yet available when it was time to send Q1 .
Computationally, maintaining the R-value counter is inconsequential. The primary performance issue with
BBRR simulation is the need to find the smallest R-value whenever a new packet is to be sent. If n is the
number of packets waiting to be sent, then we can do this in time O(log(n)) by keeping the R-values sorted
in an appropriate data structure.
The BBRR approach assumes equal weights for each subqueue; this does not generalize completely straightforwardly to weighted fair queuing as the number of subqueues cannot be fractional. If there are two queues,
one which is to have weight 40% and the other 60%, we could use BBRR with five subqueues, two of which
(2/5) are assigned to the 40%-subqueue and the other three (3/5) to the 60% subqueue. But this becomes
increasingly awkward as the fractions become less simple; the GPS model, next, is a better option.

18.5.4 The GPS Model

An almost-equivalent model to BBRR is the generalized processor sharing model, or GPS; it was first
developed as an application to CPU scheduling. In this approach we imagine the packets as liquid, and the
outbound interface as a pipe that has a certain total capacity. The head packets from each subqueue are all
squeezed into the pipe simultaneously, each at its designated fractional rate. The GPS model is essentially

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an infinitesimal variant of BBRR. The GPS model has an advantage of generalizing straightforwardly to
weighted fair queuing.
Other fluid models have also been used in the analysis of networks, eg for the study of TCP, though we do
not consider these here. See [MW00] for one example.
For the GPS model, assume there are N input subqueues, and the ith subqueue, 0i<N, is to receive fraction
i > 0, where 0 +1 + ... + N1 =1. If at some point a set A of input subqueues is active , say A = {0,2,4},
then subqueue 0 will receive fraction 0 /(0 +2 +4 ), and subqueues 2 and 4 similarly. The router forwards
packets from each active subqueue simultaneously, each at its designated rate.
The GPS model (and the BBRR model) provides an ideal degree of isolation between input flows: each flow
is insulated from any delay due to packets on competing flows. The ith flow receives bandwidth of at least
i and packets wait only for other packets belonging to the same flow. When a packet arrives for an inactive
subqueue, forwarding begins immediately, interleaved with any other work the router is doing. Traffic on
other flows can reduce the real rate of a flow, but not its virtual rate.
While GPS is convenient as a model, it is even less implementable, literally, than BBRR. As with BBRR,
though, we can use the GPS model to determine the order of one-packet-at-a-time transmission. As each
real packet arrives, we calculate the time it would finish, if we were using GPS. Packets are then transmitted
under WFQ one at a time, in order of increasing GPS finishing time.
In lieu of the BBRR rounds counter R(t), a virtual clock VC(t) is used that runs at an increased rate 1/
1 where is the sum of the i for the active subqueues. That is, if subqueues 0, 2 and 4 are active,
then the VC(t) clock runs at a rate of 1/(0 +2 +4 ). If all the i are equal, each to 1/N, then VC(t) always
runs N times faster than R(t), and so VC(t) = NR(t); the VC clock runs at wallclock speed when all input
subqueues are active and speeds up as subqueues become idle.
For any one active subqueue i, the GPS rate of transmission relative to the virtual clock (that is, in units of
bits per virtual-second) is always equal to fraction i of the full output-interface rate. That is, if the output
rate is 10 Mbps and an active flow has fraction = 0.4, then it will always transmit at 4 bits per virtual
microsecond. When all the subqueues are active, and the VC clock runs at wallclock speed, the flows actual
rate will be 4 bits/sec. When the subqueue is active alone, its speed measured by a real clock will be 10
bit/sec but the virtual clock will run 2.5 times faster so 10 bits/sec is 10 bits per 2.5 virtual microseconds,
or 4 bits per virtual microsecond.
To make this claim more precise, let A be the set of active queues, and let again be the sum of the j for j
in A. Then VC(t) runs at rate 1/ and active subqueue is data is sent at rate i / relative to wallclock time.
Subqueue is transmission rate relative to virtual time is thus (i /)/(1/) = i .
As other subqueues become inactive or become active, the VC(t) rate and the actual transmission rate move
in lockstep. Therefore, as with BBRR, a packet P of size S on subqueue i that starts transmission at virtual
time T will finish at T + S/(ri ) by the VC clock, where r is the actual output rate of the router, regardless
of what is happening in the other subqueues. In other words, VC-calculated finishing times are invariant.

To round out the calculation of finishing times, suppose packet P of size S arrives on an active GPS subqueue i. The p
FP = max(VC(now), Fprev ) + S/(ri )
In 18.8.1.1 WFQ with non-FIFO subqueues below, we will consider WFQ routers that, as part of a hierarchy, are in effect only allowed to transmit intermittently. In such a case, the virtual clock should be

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suspended whenever output is blocked. This is perhaps easiest to see for the BBRR scheduler: the roundscounter RR(t) is to increment by 1 for each bit sent by each active subqueue. When no bits may be sent, the
clock should not increase.
As an example of what happens if this is not done, suppose R has two subqueues A and B; the first is empty
and the second has a long backlog. R normally processes one packet per second. At T=0/VC=0, Rs output
is suspended. Packets in the second subqueue b1 , b2 , b3 , ... have virtual finishing times 1, 2, 3, .... At T=10,
R resumes transmission, and packet a1 arrives on the A subqueue. If Rs virtual clock had been suspended
for the interval 0T10, a1 would be assigned finishing time T=1 and would have priority comparable to
b1 . If Rs virtual clock had continued to run, a1 would be assigned finishing time T=11 and would not be
sent until b11 reached the head of the B queue.
18.5.4.1 The WFQ scheduler
To schedule actual packet transmission under weighted fair queuing, we calculate upon arrival each packets
virtual-clock finishing time assuming it were to be sent using GPS. Whenever the sender is ready to start
transmission of a new packet, it selects from the available packets the one with the smallest GPS-finishingtime value. By the argument above, a packets GPS finishing time does not depend on any later arrivals or
idle periods on other subqueues. As with BBRR, small but later-arriving packets might have smaller virtual
finishing times, but a packet currently being transmitted will not be interrupted.
18.5.4.2 Finishing Order under GPS and WFQ
We now look at the order in which packets finish transmission under GPS versus WFQ. The goal is to
provide in 18.5.4.7 Finishing-Order Bound a tight bound on how long packets may have to wait under
WFQ compared to GPS. We emphasize again:
GPS finishing time: the theoretical finishing time based on parallel multi-packet transmissions under
the GPS model
WFQ finishing time: the real finishing time assuming packets are sent sequentially in increasing order
of calculated GPS finishing time
One way to view this is as a quantification of the informal idea that WFQ provides a natural priority for
smaller packets, at least smaller packets sent on previously idle subqueues. This is quite separate from
the bandwidth guarantee that a given small-packet input class might receive; it means that small packets
are likely to leapfrog larger packets waiting in other subqueues. The quantum algorithm, below, provides
long-term WFQ bandwidth guarantees but does not provide the same delay assurances.
First, if all subqueues are always active (or if a fixed subset of subqueues is always active), then packets finish
under WFQ in the same order as they do under GPS. This is because under WFQ packets are transmitted
in the order of GPS finishing times according the virtual clock, and if all subqueues are always active the
virtual clock runs at a rate identical to wallclock time (or, if a fixed subset of subqueues is always active, at
a rate proportional to wallclock time).
If all subqueues are always active, we can assume that all packets were in their subqueues as of time T=0;
the finishing order is the same as long as each packet arrived before its subqueue went inactive.
Finally, if all subqueues are always active then each packet finishes at least as early under WFQ as under
GPS. To see this, let Pj be the jth packet to finish, under either GPS or WFQ. At the time when Pj finishes
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under WFQ, the router R will have devoted 100% of its output bandwidth exclusively to P1 through Pj .
When Pj finishes under GPS, R will also have transmitted P1 through Pj , and may have transmitted fractions
of later packets as well. Therefore, the Pj finishing time under GPS cannot be earlier.
The finishing order and the relative GPS/WFQ finishing times may change, however, if as will usually
be the case some subqueues are sometimes idle; that is, if packets sometimes arrive late for some
subqueues.
18.5.4.3 GPS Example 1
As a first example we return to the scenario of 18.5.2.1 A first virtual-finish example. The routers two
subqueues are always active; each has an allocation of =50%. Packets P1 , P2 , P3 , ..., all of size 1001, wait
in the first queue; packets Q1 , Q2 , Q3 , ..., all of size 400, wait in the second queue. Output bandwidth is 1
byte per unit time, and T=0 is the starting point.
The routers virtual clock runs at wallclock speed, as both subqueues are always active. If Fi represents the
virtual finishing time of Ri , then we now calculate Fi as Fi-1 + 400/ = Fi-1 + 800. The virtual finishing times
of P1 , P2 , etc are similarly at multiples of 2002.
Packet
Q1
Q2
P1
Q3
Q4
Q5
P2

virtual finish
800
1600
2002
2400
3200
4000
4004

actual finish time

400
800
1801
2201
2601
3001
4002

In the table above, the virtual finish column is simply double that of the BBRR version, reflecting the fact
that the virtual finishing times are now scaled by a factor of 1/ = 2. The actual finish times are identical to
what we calculated before.
Note that, in every case, the actual WFQ finish time is always less than or equal to the virtual GPS finish
time.
18.5.4.4 GPS Example 2
If the router has only a single active subqueue, with share and packets P1 , P2 , P3 , ..., then the calculated
virtual-clock packet finishing times will be equal to the time on the virtual clock at the point of actual finish,
at least if this has been the case since the virtual clock last restarted at T=VC=0. Let r be the output rate of
the router, let S1 , S2 , S3 be the sizes of the packets and let F1 , F2 , F3 be their virtual finishing times with
F0 =0. Then
Fi = Fi-1 + Si /(r) = S1 /(r) + ... + Si /(r)
The ith packets actual finishing time Ai is (S1 + ... + Si )/r, which is Fi . But the virtual clock runs fast by
a factor of 1/, so the actual finishing time on the virtual clock is Ai / = Fi .

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18.5.4.5 GPS Example 3

The next example illustrates a smaller but later-arriving packet, in this case Q2 , that finishes ahead of P2
under GPS but not under WFQ. P2 can be said to leapfrog Q2 and R1 under WFQ.
Suppose packets P1 , Q1 , P2 , Q2 and R1 arrive at a router at the following times T, and with the following
lengths L. The output bandwidth is 1 length unit per time unit; that is, r=1. The total number of length units
is 24. Each subqueue is allocated an equal share of the bandwidth; eg =1/3.
subqueue 1
P1 : T=0, L=1
P2 : T=2, L=10

subqueue 2
Q1 : T=0, L=2
Q2 : T=4, L=6

subqueue 3
R1 : T=10, L=5

Under WFQ, we send P1 and then Q1 ; Q1 is second because its finishing time is later. When Q1 finishes the
wallclock time is T=3. At this point, P2 is the only packet available to send; it finishes at T=13.
Up until T=10, we have two packets in progress under GPS (because Q1 finishes under GPS at T=4 and
Q2 arrives at T=4), and so the GPS clock runs at rate 3/2 of wallclock time and the BBRR clock runs at
rate 1/2 of wallclock time. At T=4, when Q2 arrives, the BBRR clock is at 2 and the VC clock is at 6 and
we calculate the BBRR finishing time as 2+6=8 and the GPS finishing time as 6+6/(1/3) = 24. At T=10,
the BBRR clock is at 5 and the GPS clock is 15. R1 arrives then; we calculate its BBRR finishing time as
5+5=10 and its GPS finishing time as 15+5/ = 30.
Because Q2 has the earlier virtual-clock finishing time, WFQ sends it next after P2 , followed by R1 .
Here is a diagram of transmission under GPS. The chart itself is scaled to wallclock times. The BBRR clock
is on the scale below; the VC clock always runs three times faster.

The circled numbers represent the size of the portion of the packet sent in the intervals separated by the
dotted vertical lines; for each packet, these add up to the packets total size.
Note that, while the transmission order under WFQ is P1 , Q1 , P2 , Q2 , R1 , the finishing order under GPS is
P1 , Q1 , Q2 , R1 , P2 . That is, P2 managed to leapfrog Q2 and R1 under WFQ by the simple expedient of being
the only packet available for transmission at T=3.
18.5.4.6 GPS Example 4
As a second example of leapfrogging, suppose we have the following arrivals; in this scenario, the smaller
but later-arriving R1 finishes ahead of P1 and Q2 under GPS, but not under WFQ.

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subqueue 1
P1 : T=0, L=1000

subqueue 2
Q1 : T=0, L=200
Q2 : T=0, L=300

subqueue3
R1 : T=600, L=100

The following diagram shows how the packets shared the link under GPS over time. As can be seen, the
GPS finishing order is Q1 , R1 , Q2 , P1 .

Under WFQ, the transmission order is Q1 , Q2 , P1 , R1 , because when Q2 finishes at T=500, R1 has not yet
arrived.
18.5.4.7 Finishing-Order Bound
These examples bring us to the following delay-bound claim, due to Parekh and Gallager [PG93]; we will
make use of it below in 18.13.3 Parekh-Gallager Theorem. It is arguably the deepest part of the ParekhGallager theorem.
Claim: For any packet P, the wallclock finishing time of P at a router R under WFQ cannot be later than the
wallclock finishing time of P at R under GPS by more than the time R needs to transmit the maximum-sized
packet that can appear.
Expressed symbolically, if FWFQ and FGPS are the finishing times for P under WFQ and GPS, Rs outbound
transmission rate is r, and Lmax is the maximum packet size that can appear at R, then
FWFQ FGPS + Lmax /r
This is the best possible bound; Lmax /r is the time packet P must wait if it has arrived an instant too late and
another packet of size Lmax has started instead.
Note that, if a packets subqueue is inactive, the packet starts transmitting immediately upon arrival under
GPS; however, GPS may send the packet relatively slowly.
To prove this claim, let us number the packets P1 through Pk in order of WFQ transmission, starting from
the most recent point when at least one subqueue of the router became active. (Note that these packets may
be spread over multiple input subqueues.) For each i, let Fi be the finishing time of Pi under WFQ, let Gi be
the finishing time of Pi under GPS, and let Li be the length of Pi ; note that, for each i, Fi+1 = Li+1 /r + Fi .
If Pk finishes after P1 through Pk-1 under GPS, then the argument above (18.5.4.2 Finishing Order under
GPS and WFQ) for the all-subqueues-active case still applies to show Pk cannot finish earlier under GPS
than it does under WFQ; that is, we have Fk Gk .
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Otherwise, some packet Pi with i<k must finish after Pk under GPS; Pi has leapfrogged Pk under WFQ,
presumably because Pk was late in arriving. Let Pm be the most recent (largest m<k) such leapfrogger
packet, so that Pm finishes after Pk under GPS but Pm+1 through Pk-1 finish earlier (or are tied); this was
illustrated above in 18.5.4.5 GPS Example 3 for k=5 and m=3.
We next claim that none of the packets Pm+1 through Pk could have yet arrived at R at the time Tstart when
Pm began transmission under WFQ. If some Pi with i>m were present at time Tstart , then the fact that it is
transmitted after Pm under WFQ would imply that the calculated GPS finishing time came after that of Pm .
But, as we argued earlier, calculated virtual-clock GPS finishing times are always the actual virtual-clock
GPS finishing times, and we cannot have Pi finishing both ahead of Pm and behind it.
Recalling that Fm is the finishing time of Pm under WFQ, the time Tstart above is simply Fm - Lm /r. Between
Tstart and Gk , all the packets Pm+1 through Pk must arrive and then, under GPS, depart. The absolute minimum time to send these packets under GPS is the WFQ time for end-to-end transmission, which is (Lm+1 +
.. + Lk )/r = Fk - Fm . Therefore we have
Gk - Tstart Fk - Fm
Gk - (Fm - Lm /r) Fk - Fm
Fk Gk + Lm /r Gk + Lmax /r
The last line represents the desired conclusion.

18.5.5 The Quantum Algorithm

The BBRR approach has cost O(log n), where n is the number of packets waiting, as outlined earlier. One
simple though approximate alternative that has O(1) performance, introduced in [SV96], is to give each
input subqueue a quantum. At each round, a subqueue will be able to send a quantums worth of packets
(plus, as we shall see, a carryover from the previous round). The quantum must be a number of bytes at least
as large as the network MTU (so each time a subqueue has a chance to send, it will be able to send at least
one packet). Then, we again service the subqueues in round-robin fashion, but each time we take as many
packets from the subqueue as we can such that the total size does not exceed the quantum.
One cost of the quantum approach is that the elegant delay bound of 18.5.4.7 Finishing-Order Bound no
longer holds. In fact, a sender may be forced to wait for all other senders each to send a full quantum, even
if the sender has only a small packet.
If we just allow subqueues to send up to a quantums worth of bytes, the strategy will be biased against
classes that, for example, only include packets with size just over half the quantum. Fairness can be improved
significantly by keeping track of the difference between the quantum and what was actually sent; we will
call this the deficit. If the quantum is 1000 bytes and we have two 600-byte packets, the first of the packets
is sent and the deficit is recorded as 400; if we have two 400-byte packets then the deficit is 0 as sending the
two 400-byte packets empties the subqueue.
The next time that subqueue is serviced, we add the previous deficit to the quantum, and send packets up to
a total cumulative size of deficit+quantum. With this variation, if the quantum is 1000 bytes and a steady
stream of 600-byte packets arrives on one subqueue, they are sent as follows
round
1
2
3

quantum + prev deficit

1000
1400
1200

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1
2,3
4,5

new deficit
400
200
0
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In three rounds the subqueue has been allowed to send 5600 = 3000 bytes, which is exactly 3 quantums. We
will refer to this strategy, with provision for deficits as above, as the quantum algorithm for fair queuing.
If a subqueue is ever empty, its deficit should immediately expire.
We can implement weighted fair queuing using the quantum algorithm by adjusting the quantum sizes in
proportion to the weights; that is, if the weights are 40% and 60% then the respective quanta might be 1000
bytes and 1500 bytes. The quantum should be at least as large as the largest packet (eg the MTU), so that
if one input class is to have 10% weight and the other 90%, then the second class will have a quantum of 9
times the MTU. Of course, if the smaller-weighted class happens to be a VoIP stream with smaller packets
as well, this is less of an issue.

18.5.6 Stochastic Fair Queuing

Stochastic fair queuing [McK90] is a different kind of approximation to fair queuing. The ideal is to give
each individual TCP connection through a router its own fair queuing class, so that no connection has an
incentive to keep more than the minimum number of packets in the queue. However, managing so many
separate (and dynamically changing) input classes is computationally intensive. Instead, a hash function is
used to put each TCP connection (as identified by its socketpair) into one of several buckets (the current
default bucket count is 1024). The buckets are then used as the fair queuing input queues.
If two connections hash to the same bucket, each will get only half the bandwidth to which it is entitled.
Therefore the hash function has some additional parameters that allow it to be perturbed at regular intervals;
after perturbation, socketpairs that formerly hashed to the same bucket likely now will not.
Stochastic fair queuing is available in the linux kernel.

18.6 Applications of Fair Queuing

As we saw in the Queue-Competition Rule in 14.2.2 Example 2: router competition, if a bottleneck router
uses FIFO queuing then it pays a sender, in terms of the bandwidth it receives, to keep as many packets as
possible in the queue. Fair queuing, however, can behave quite differently. If fair queuing is used with a
separate class for each connection, this greediness benefit evaporates; a sender need only keep its per-class
queue nonempty in order to be guaranteed its assigned share. Senders can limit their queue utilization to
one or two packets without danger of being crowded out by competing traffic.
As another application of fair queuing, suppose we have three web servers, A, B and C, to which we want to
give equal shares at sending along a 6 Mbps link; all three reach the link through router R. One way would
be to have R restrict them each to 2 Mbps, but that is not work-conserving: if one server is idle then its share
goes unused. Fair queuing offers a better approach: if all three are busy, each gets 2 Mbps; if two are busy
then each gets 3 Mbps, and if only one is busy it gets 6 Mbps. ISPs can also use this strategy internally
whenever they have several flows competing for one outbound link.

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Unfortunately, this strategy does not work quite as well with three receivers. Suppose now A, B and C are
three pools of users (eg three departments), and we want to give them each equal shares of an inbound 6
Mbps link:

Inbound fair queuing could be used at the ISP (upstream) end of the connection, where the three flows
would be identified by their destinations. But the ISP router is likely not ours to play with. Once the three
flows arrive at R, it is too late to allocate shares; there is nothing to do but deliver the traffic to A, B and C
respectively. We will return to this scenario below in 18.10 Applications of Token Bucket.
Note that for these kinds of applications we need to be able to designate administratively how packets get
classified into different input classes. The automatic classification of stochastic fair queuing, above, will not
help here.

18.7 Hierarchical Queuing

Any queuing discipline with multiple input classes can participate in a hierarchy: the root queuing discipline
is at the top of the tree, and each of its input classes is fed by a separate lower-level queuing discipline.
Hierarchical Queuing Today
Hierarchical queuing is admittedly somewhat out of fashion. One of its original goals was to support
complex sharing of a single link by multiple data streams with different bandwidth and delay requirements; in todays Internet providers often instead address this with the brute-force approach of excess
bandwidth capacity. That said, hierarchical queuing offers an elegant and economical alternative, and
despite todays trends it is not clear if providing excess capacity will always be an option in the future.
The usual understanding of hierarchical queuing is that the non-leaf queuing-discipline nodes are virtual:
they do not store data, but only make decisions as to which subqueue to serve. The only physical output
interface is at the root, and all physical queues are attached to the leaf nodes. Each non-leaf queuingdiscipline node is allowed to peek at what packet, if any, its subqueues would dequeue if asked, but any
node will only actually dequeue a packet unless that packet will immediately be forwarded up the tree to the
root. This might be referred to as a leaf-storage hierarchy.
An immediate corollary is that leaf nodes, which now hold all the physical packets, will do all the packet
dropping.
While it is possible to chain real queue objects together with real links, to construct composite queuing
structures in which packets are not stored only at leaf nodes, the resultant internal-storage hierarchy will
often not function as might be expected. For example, a priority-queuing node that immediately forwards
each arriving packet on to its parent has forfeited any control over the priority order of resultant packet
transmissions. Similarly, if we try to avoid the complexity of hierarchical fair queuing, below, by connecting
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separate WFQ nodes into a tree with real links, then either packets simply pile up at the root, or else the
interior links become the bottleneck. For a case where the internal-storage construction does work, see the
end of 18.12 Hierarchical Token Bucket.

18.7.1 Generic Hierarchical Queuing

One simple way to enable lazy dequeuing is to require the following properties of the hierarchys component
queuing disciplines:
each node supports a peek() operation to return the packet it would dequeue if asked, and
any non-leaf node can determine what packet it would itself dequeue simply by calling peek() on
each of its child nodes
With these in place, a dequeue operation at the root node will trigger a depth-first traversal of the entire tree
through peek() calls; eventually one leaf node will dequeue its packet and pass it up towards the root. We
will call this generic hierarchical queuing.
Note that if we have a peek() operation at the leaf nodes, and the second property above for the interior
nodes, we can recursively define peek() at every node.
Many homogeneous queuing-discipline hierarchies support this generic mechanism. Priority queuing, as in
the example below, works, because when a parent node needs to dequeue a packet it finds the highest-priority
nonempty child subqueue, dequeues a packet from it, and sends it up the tree. In this case we do not even
need peek(); all we need is is_empty(). Homogeneous FIFO queuing also works, assuming packets
are each tagged with their arrival time, because for a node to determine which packet it would dequeue it
needs only to peek() at its child nodes and find the earliest-arriving packet.
As an example of where generic hierarchical queuing does not work, imagine a hierarchy consisting of a
queuing discipline node that sends the smallest packet first, fed by two FIFO child nodes. The parent would
have to dequeue all packets from the child nodes to determine the smallest.
Fair queuing is not quite well-behaved with respect to generic hierarchical queuing. There is no problem if
all packets are the same size; in this case the nodes can all implement simple round-robin selection in which
they dequeue a packet from the next nonempty child node and send it immediately up the tree. Suppose,
however, that to accommodate packets of different sizes all nodes use BBRR (or GPS). The problem is that
without timely notification of when packets arrive at the leaf nodes, the interior nodes may not have enough
information to determine the correct rates for their virtual clocks. An appropriate fair-queuing-specific
modification to the generic hierarchical-queuing algorithm is below in 18.8.1 A Hierarchical Weighted
Fair Queuing Algorithm.

18.7.2 Hierarchical Examples

Our first example is of hierarchical priority queuing, though as we shall see shortly it turns out in this case
that the hierarchy collapses. It may still serve to illustrate some principles, however. The basic queuingdiscipline unit will be the two-input-class priority queue, with priorities 0 (high) and 1 (low). We put one
of these at the root, named QR, and then put a pair of such queues (QA and QB in the diagram) at the next
level down, each feeding into one of the input classes of the root queue.

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We now need a classifier, as with any multi-subqueue queuing discipline, to place input packets into one of
the four leaf nodes labeled A, B, C and D above; these likely represent FIFO queues. Once everything is set
up, the dequeuing rules are as follows:
at the root node, we first see if packets are available in the left (high-priority) subqueue, QA. If so, we
dequeue the packet from there; if not, we go on to the right subqueue QB.
at either QA or QB, we first see if packets are available in the left input, labeled 0; if so, we dequeue
from there. Otherwise we see if packets are available from the right input, labeled 1. If neither, then
the subqueue is empty.
The catch here is that hierarchical priority queuing collapses to a single layer, with four priority levels 00,
01, 10, and 11 (from high to low):

For many other queuing disciplines, however, the hierarchy does not collapse. One example of this might
be a root queuing discipline PQR using priority queuing, with two leaf nodes using fair queuing, FQA and
FQB. (Fair queuing does not behave well as an interior node without modification, as mentioned above, but
it can participate in generic hierarchical queuing just fine as a leaf node.)

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Senders A1 and A2 receive all the bandwidth, if either is active; when this is the case, B1 and B2 get
nothing. If both A1 and A2 are active, they get equal shares. Only when neither is active do B1 and B2
receive anything, in which case they divide the bandwidth fairly between themselves. The root node will
check on each dequeue() operation whether FQA is nonempty; if it is, it dequeues a packet from FQA
and otherwise dequeues from FQB. Because the root does not dequeue a packet from FQA or FQB unless it
is about to return it via its own dequeue() operation, those two child nodes do not have to decide which
of their internal per-class queues to service until a packet is actually needed.

18.8 Hierarchical Weighted Fair Queuing

Hierarchical weighted fair queuing is an elegant mechanism for addressing naturally hierarchical bandwidthallocation problems, even if it cannot be properly implemented by generic methods of creating queuingdiscipline hierarchies. There are, fortunately, algorithms specific to hierarchical fair queuing.
The fluid model provides a convenient reference point for determining the goal of hierarchical weighted fair
queuing. Each interior node divides its bandwidth according to the usual one-level WFQ mechanism: if N is
an interior node, and if the ith child of N is guaranteed fraction i of Ns bandwidth, and A is the set of Ns
currently active children, then WFQ on N will allocate to active child j the fraction j equal to j divided by
the sum of the i for i in A.
Now we can determine the bandwidth allocated to each leaf node L. Suppose L is at level r (with the root at
level 0), and let k be the fraction currently assigned to the node on the root-to-L path at level 0<kr. Then
L should receive service in proportion to 1 ... r , the product of the fractions along the root-to-L path.
For example, in the following hierarchical model, leaf node L1 should receive fraction 70%40%50% =
14% of the total bandwidth.

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If, however, node FQD becomes inactive, then FQA will assign 100% to FQC, in which case L1 will receive
70%100%50% = 35% of the bandwidth.
Alternatively, we can stick to real packets, but simplify by assuming all are the same size and that child
allocations are all equal; in this situation we can implement hierarchical fair queuing via generic hierarchical
queuing. Each parent node simply dequeues from its child nodes in round-robin fashion, skipping over any
empty children.
Hierarchical fair queuing unlike priority queuing does not collapse; the hierarchical queuing discipline
is not equivalent to any one-level queuing discipline. Here is an example illustrating this. We form a tree of
three fair queuing nodes, labeled FQR, FQA and FQB in the diagram. Each grants 50% of the bandwidth to
each of its two input classes.

If all four input classes A,B,C,D are active in the hierarchical structure, then each gets 25% of the total
bandwidth, just as for a flat four-input-class fair queuing structure. However, consider what happens when
only A, B and C are active, and D is idle. In the flat model, A, B and C each get 33%. In the hierarchical
model, however, as long as FQA and FQB are both active then each gets 50%, meaning that A and B split

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FQAs allocation and receive 25% each, while C gets 50%.

As an application, suppose two teams, Left and Right, are splitting a shared outbound interface; each team
has contracted to receive at least 50% of the bandwidth. Each team then further subdivides its allocation
among its own members, again using fair queuing. When A, B and C are the active senders, then Team
Right having active sender C still expects to receive its contractual 50%. Team Right may in fact have
decided to silence D specifically so C would receive the full allocation.
Hierarchical fair queuing can not be implemented by computing a finishing time for each packet at the point
it arrives. By way of demonstration, consider the following example from [BZ97], with root node R and
interior child node C.

If A is idle, but B and D are active, then B should receive four times the bandwidth of D. Assume for
a moment that a large number of packets have arrived for each of B and D. Then B should receive the
entire 80% of Cs share. If transmission order can be determined by packet arrival, then the order will look
something like
d1 , b1 , b2 , b3 , b4 ,
d2 , b5 , b6 , b7 , b8 , ...
Now suppose A wakes up and becomes active. At that point, B goes from receiving 80% of the total
bandwidth to 5% = 80% 1/16, or from four times Ds bandwidth to a quarter of it. The new b/d relative
rate, not showing As packets, must be
b1 , d1 , d2 , d3 , d4 ,
b2 , d5 , d6 , d7 , d8 , ...
The relative frequencies of B and D packets have been reversed by the arrival of later A packets. The
packet order for B and D is thus dependent on later arrivals on another queue entirely, and thus cannot be
determined at the point the packets arrive.
After presenting the actual hierarchical-WFQ virtual-clock algorithm, we will return to this example in
18.8.1.2 A Hierarchical-WFQ Example.

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18.8.1 A Hierarchical Weighted Fair Queuing Algorithm

The GPS-based WFQ scheduling algorithm is almost suitable for use in the generic-hierarchical-queuing
framework; two adjustments must be made. The first adjustment is that each non-leaf node must be notified
whenever any of its formerly empty subqueues becomes active; the second adjustment is a modification to
how and more importantly when a packets virtual finishing time is calculated.
As for the active-subqueue notification, each non-leaf node can, of course, check after dequeuing each
packet which of its subqueues is active, using the is_empty() operation. This, however, may be significantly after the subqueue-activating packet has arrived. A WFQ node needs to know the exact time when
a subqueue becomes active both to record the GPS virtual-clock start time for the subqueue, and to know
when to change the rate of its virtual clock.
Addition of this subqueue-activation notification to hierarchical queuing is straightforward. When a previously empty leaf node receives a packet, it must send a notification to its parent. Each interior node of the
hierarchy must in turn forward any received subqueue-activation notification to its parent, provided that none
of its other child subqueues were already active. If the interior node already had other active subqueues, then
that node is itself active and no new notification needs to be sent. In this way, when a leaf node becomes
active, the news will be propagated towards the root of the tree until either the root or an already-active
interior node is reached.
To complete the hierarchical WFQ algorithm, we next describe how to modify the algorithm of 18.5.4 The
GPS Model to support subqueues of any type (eg FIFO, priority, or in our case hierarchical subtrees), provided inactive subqueues notify the WFQ parent when they become active.
18.8.1.1 WFQ with non-FIFO subqueues
Suppose we want to implement WFQ where the per-class subqueues, instead of being FIFO, can be arbitrary
queuing disciplines; again, the case in which we are interested is when the subqueue represents a subhierarchy. The order of dequeuing from each subqueue might be changed by later arrivals (eg as in priority
queuing), and packets in the subqueues might even disappear (as with random-drop queuing). (We will
assume that nonempty subqueues can only become empty through a dequeuing operation; this holds in all
the cases we will consider here.) The original WFQ algorithm envisioned labeling each packet P with its
finishing time as follows:
FP = max(VC(now), Fprev ) + S/i
Clearly, this labeling of each packet upon its arrival is incompatible with subqueues that might place laterarriving packets ahead of earlier-arriving ones. The original algorithm must be modified.
It turns out, though, that a WFQ node need only calculate finishing times FP for the packets P that are at the
heads of each of its subqueues, and even that needs only to be done at the time the dequeue() operation is
invoked. No waiting packet needs to be labeled. In fact, a packet P1 might be at the head of one subqueue,
and be passed over in a dequeue() operation for too large an FP1 , only to be replaced by a different packet
P2 during the next dequeue() call.
It is sufficient for the WFQ node to maintain, for each of its subqueues, a variable NextStart, representing
the virtual-clock time (according to the WFQ nodes own virtual clock) to be used as the start time for that
subqueues next transmitted packet. A subqueues NextStart value serves as the max(VC(now), Fprev )
of the single-layer WFQ formula above. When the parent WFQ node is called upon to dequeue a packet, it

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calls the peek() operation on each of its subqueues and then calculates the finishing time FP for the packet
P currently at the head of each subqueue as NextStart + size(P)/, where is the bandwidth fraction
assigned to the subqueue.
If a formerly inactive subqueue becomes active, it by hypothesis notifies the parent WFQ node. The parent
node records the time on its virtual clock as the NextStart value for that subqueue. Whenever a subqueue
is called upon to dequeue packet P, its NextStart value is updated to NextStart + size(P)/, the
virtual-clock finishing time for P.
The active-subqueue notification is also exactly what is necessary for the WFQ node to maintain its virtual
clock. If A is the set of active subqueues, and the ith subqueue has bandwidth share i , then the clock is to
run at rate equal to the reciprocal of the sum of the i for i in A. This rate needs to be updated whenever
a subqueue becomes active or inactive. In the first case, the parent node is notified by hypothesis, and the
second case happens only after a dequeue() operation.
There is one more modification needed for non-root WFQ nodes: we must suspend their virtual clocks when
they are not transmitting, following the argument at the end of 18.5.4 The GPS Model. Non-root nodes
do not have real interfaces and do not physically transmit. However, we can say that an interior node N is
logically transmitting when the root node R is currently transmitting a packet from leaf node L, and N lies
on the path from R to L. Note that all interior nodes on the path from R to L will be logically transmitting
simultaneously. For a specific non-root node N, whenever it is called upon at time T to dequeue a packet P,
its virtual clock should run during the wallclock interval from T to T + size(P)/r, where r is the root nodes
physical bandwidth. The virtual finishing time of P need not have any direct correspondence to this actual
finishing time T + size(P)/r. The rate of Ns virtual clock in the interval from T to T + size(P)/r will depend,
of course, on the number of Ns active child nodes.
We remark that the dequeue() operation described above is relatively inefficient; each dequeue()
operation by the root results in recursive traversal of the entire tree. There have been several attempts to
improve the algorithm performance. Other algorithms have also been used; the mechanism here has been
taken from [BZ97].
18.8.1.2 A Hierarchical-WFQ Example
Let us consider again the example at the end of 18.8 Hierarchical Weighted Fair Queuing:

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Assume that all packets are of size 1 and R transmits at rate 1 packet per second. Initially, suppose FIFO
leaf nodes B and D have long backlogs (eg b1 , b2 , b3 , ...) but A is idle. Both of Rs subqueues are active, so
Rs virtual clock runs at the wall-clock rate. Cs virtual clock is running 16 fast, though. Rs NextStart
values for both its subqueues are 0.
The finishing time assigned by R to di will be 5i. Whenever packet di reaches the head of the D queue,
Rs NextStart for D will be 5(i-1). (Although we claimed in the previous section that hierarchical WFQ
nodes shouldnt need to assign finishing times beyond that for the current head packet, for FIFO subqueues
this is safe.)
At least during the initial A-idle period, whenever R checks Cs subqueue, if bi is the head packet then Rs
NextStart for C will be 1.25(i-1) and the calculated virtual finishing time will be 1.25i. If ties are decided
in Bs favor then in the first ten seconds R will send
b1 , b2 , b3 , b4 , d1 , b5 , b6 , b7 , b8 , d2
During the ten seconds needed to send the ten packets above, all of the packets dequeued by C come from B.
Having only one active subqueue puts C is in the situation of 18.5.4.4 GPS Example 2, and so its packets
calculated finishing times will exactly match Cs virtual-clock value at the point of actual finish. C dequeues
eight packets, so its virtual clock runs for only those 8 of the 10 seconds during which one of the bi is being
transmitted. As a result, packet bi finishes at time 16i by Cs virtual clock. At T=10, Cs virtual clock is
816 = 128.
Now, at T=10, as the last of the ten packets above completes transmission, let subqueue A become backlogged with a1 , a2 , a3 , etc. C will assign a finishing time of 128 + 1.0667i to ai (1.0667 = 16/15); C has
already assigned a virtual finishing time of 916=144 to b9 . None of the virtual finishing times assigned by
C to the remaining bi will change.
At this point the virtual finishing times for Cs packets are as follows:
packet
a1
a2
a3
a4
...
a15
b9

C finishing time
128 + 1.0667
128 + 21.0667
128 + 31.0667
128 + 41.0667

R finishing time
10 + 1.25
10 + 2.50
10 + 3.75
10 + 5

128 + 151.0667 = 144

144

10 + 151.25
10 + 161.25 = 30

During the time the 16 packets in the table above are sent from C, R will also send four of Ds packets, for
a total of 20.
The virtual finishing times assigned by C to b9 and b10 have not changed, but note that the virtual finishing
times assigned to these packets by R are now very different from what they would have been had A remained
idle. With A idle, these finishing times would have been F9 = 11.25 and F10 = 12.50, etc. Now, with A active,
it is a1 and a2 that finish at 11.25 and 12.50; b9 will now be assigned by R a finishing time of 30 and b10
will be assigned a finishing time of 50. R is still assigning successive finishing times at increments of 1.25
to packets dequeued by C, but Bs contributions to this stream of packets have been bumped far back.
Rs assignments of virtual finishing times to the di are immutable, as are Cs assignments of virtual finishing
times, but R can not assign a final virtual finishing time to any of Cs packets (that is, As or Bs) until the
packet has reached the head of Cs queue. R assigns finishing times to Cs packets in the order they are

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dequeued, and until a packet is dequeued by C it is subject to potential preemption.

18.9 Token Bucket Filters

Token-bucket filters provide an alternative to fair queuing for providing a traffic allocation to each of several
groups. The main practical difference between fair queuing and token bucket is that if one sender is idle, fair
queuing distributes that senders bandwidth among the other senders. Token bucket does not: the bandwidth
a sender is allocated is a bandwidth cap.
Suppose the outbound bandwidth is 4 packets/ms and we want to allocate to one particular sender, A, a
bandwidth of 1 packet/ms. We could use fair queuing and give sender A a bandwidth fraction of 25%, but
suppose we do not want A ever to get more bandwidth than 1 packet/ms. We might do this, for example,
because A is paying a reduced rate, and any excess available bandwidth is to be divided among the other
senders.
The catch is that we want the flexibility to allow As packets to arrive at irregular intervals. We could simply
wait 1 ms after each of As packets begins transmission, before the next can begin, but this may be too strict.
Suppose A has been dutifully submitting packets at 1ms intervals and then the packet that was supposed to
arrive at T=6ms instead arrives at T=6.5. If the following packet then arrives on time at T=7, does this mean
it should now be held until T=7.5, etc? Or do we allow A to send one late packet at T=6.5 and the next at
T=7, on the theory that the average rate is still 1 packet/ms?
The latter option is generally what we want, and the solution is to define As quota in terms of a token-bucket
specification, which allows for specification of both an average rate and also a burst capacity.
If a packet does not meet the token-bucket specification, it is non-compliant; we can do any of the following
things:
delay the packet until the bucket is ready
drop the packet
mark the packet as non-compliant
The first option here is often called shaping; the second, more authoritarian option is sometimes known as
policing.
Another use for token-bucket specifications is as a theoretical traffic description, rather than a rule to be
enforced; in this context compliance is a non-issue.
A token-bucket filter can be thought of as a queuing discipline, with an underlying FIFO queue. If noncompliant packets are delayed, it is non-work-conserving. Dropping non-compliant packets can be viewed
as an alternative to tail-drop. The queuing-discipline definition above in 18.4 Queuing Disciplines does not
provide for marking packets, but this is a straightforward extension.

18.9.1 Token Bucket Definition

The idea behind a token bucket is that there is a notional bucket somewhere, being filled at a steady rate
with tokens (or, if more divisibility is needed, with fluid); any overflow from the bucket is discarded. To
send a packet, we need to be able to take one token from the bucket; if the bucket is empty then the packet

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is non-compliant and must suffer special treatment as above. If the bucket is full, however, then the sender
may send a burst of packets corresponding to the bucket capacity (at which point the bucket will be empty).
A common variation is requiring one token per byte rather than per packet, with the fill rate correspondingly
scaled; this allows packet size to be taken into account.
More precisely, a token-bucket specification TB(r,Bmax ) includes a token fill rate of r tokens/sec, representing the rate at which the bucket fills with tokens, and also a bucket capacity (or depth) Bmax >0. The bucket
fills at the rate specified, subject to a maximum of Bmax ; we will denote the current capacity by B, or by
B(t) if we need to specify the time. In order for a packet of size S (possibly S=1 for counting size in units of
whole packets) to be within the specification, the bucket must have at least S tokens; that is, BS. Otherwise
the packet is non-compliant. When the packet is sent, S tokens are removed from the bucket, that is, B=BS.
It is possible for the packets of a given flow all to be compliant with a given token-bucket specification at
one point (eg one router) in the network but not at another point; this can happen, for example, if more than
Bmax packets pile up at a downstream router due to momentary congestion.
The following graph is a visual representation of a token-bucket constraint. The black and purple curves
plotted are of cumulative bits sent as a function of time, that is, bits(t). When bits(t) is horizontal, the sender
is idle.

The blue line represents a sender sending linearly at the rate r, with no burstiness. At vertical distance Bmax
above the blue line is the red line. Graphs for compliant senders cannot cross this, because that would entail
a burst of more than Bmax above the blue line; we give a more formal argument below. As a senders graph
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approaches the red line, the senders current bucket contents decreases; the instantaneous bucket contents
for the black sender is shown at one point as B(t).
The purple sender has fallen below the blue line at one point; as a result, it can never catch up. In fact, after
passing through the vertex at point A the purple graph can never cross the dashed red line. A proof is in
18.11 Token Bucket Queue Utilization, following some numeric token-bucket examples that illustrate how
a token-bucket filter works.
Satellite Token Bucket
When I first got satellite Internet, my service was limited by a token-bucket filter with rate 56 Kbps
and bucket 300 megabytes. When the bucket emptied, it took 12 hours to refill. The idea was that
someone could use the Internet intensely but relatively briefly; satellite access is expensive. Within a
year, the provider switched to a flat 300 MB cap per day; the token-bucket rule was apparently not well
understood by customers.
If a packet arrives when there are not enough tokens in the bucket to send it, then as indicated above there are
three options. The sender can engage in shaping, making the packet wait until sufficient tokens accumulate.
The sender can engage in policing, dropping the packet. Or the sender can send the packet immediately but
mark it as noncompliant.
One common strategy is to send noncompliant packets as marked in the third option above with lower
priority. Alternatively, marked packets may face a greater chance of being dropped by some downstream
router. In ATM networks (3.8 Asynchronous Transfer Mode: ATM) the cell-loss priority bit is often used
to mark noncompliant packets.
Token-bucket specifications supply a framework for making decisions about admission control: a router
can decide whether to accept a new connection (or whether to accept the connections quality-of-service
request) based on the requested rate and bucket (queue) requirements.
Token-bucket specifications are the mirror-image equivalent to leaky-bucket specifications, in which the
fluid leaks out of the leaky bucket at rate r and to send a packet we must add S units without overflowing.
The two forms are completely equivalent.
So far we have been using token-bucket specifications to describe traffic; eg traffic arriving at a router. It is
also possible to use token buckets to describe the router itself; in this setting, the leaky-bucket formulation
may be clearer. The routers queue represents the bucket, and the routers packet transmissions represent
tokens leaking out of the bucket. Arriving packets are added to the bucket; a bucket overflow represents lost
packets. We will not pursue this interpretation further.

18.9.2 Token-Bucket Examples

Suppose the token-bucket specification is TB(1/3 packet/ms, 4 packets), and packets arrive at the following
times, with the bucket initially full:
0, 0, 0, 2, 3, 6, 9, 12
After all the T=0 packets are processed, the bucket holds 1 token. By the time the fourth packet arrives at
T=2, the bucket volume has risen to 1 2/3; it immediately drops to 2/3 when packet 4 is sent. By T=3, the

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bucket volume has reached 1 and the fifth packet can be sent. The bucket is now empty, but fortunately the
remaining packets arrive at 3-ms intervals and can all be sent.
In the next set of packet arrival times, again with TB(1/3,4), we have three bursts of four packets each.
0, 0, 0, 0, 12, 12, 12, 12, 24, 24, 24, 24
Each burst empties the bucket, which then takes 12 ms to refill. All packets are compliant.
In the following set of packet arrival times, still with TB(1/3,4), the burst of four packets at T=0 drains the
bucket. At T=3 the bucket size has increased back to 1, allowing the packet that arrives then to be sent but
also draining the bucket again.
0, 0, 0, 0, 3, 6, 12, 12
At T=6 the same thing happens. From T=6 to T=12 the bucket contents rise from 0 to 2, allowing the two
packets arriving at T=12 to be sent.
Finally, suppose packets arrive at the following times at our TB(1/3,4) filter.
0, 1, 2, 3, 4, 5

Just after T=0 the bucket size is 3; just before T=1 it is 3 1/3.
Just after T=1 the bucket size is 2 1/3; just before T=2 it is 2 2/3
Just after T=2 the bucket size is 1 2/3; just before T=3 it is 2
Just after T=3 the bucket size is 1; just before T=4 it is 1 1/3
Just after T=4 the bucket size is 1/3; just before T=5 it is 2/3
At T=5 the bucket size is 2/3 and the arriving packet is noncompliant.

We can also represent this in tabular form as follows; note that for the noncompliant packet the bucket is not
decremented.
packet arrival
bucket just before
bucket just after

0
4
3

1
3 1/3
2 1/3

2
2 2/3
1 2/3

3
2
1

4
1 1/3
1/3

5
2/3
2/3

18.9.3 Multiple Token Buckets

It often makes sense to require that a sender comply with two (or more) separate token-bucket specifications.
We can think of these being applied to the traffic sequentially. Often one filter will specify a peak rate, with
a small bucket size, and the other will specify an average rate, with a larger bucket size. Consider, for
example, the following pair:
1. TB(1 packet/ms, 1.5 packets)
2. TB(1/5 packet/ms, 6 packets)
The first specification, meant to apply to the peak rate, mandates 1 ms on average between packets, but
packets can be only 0.5 ms early without being noncompliant. The second specification, meant to apply

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over the longer term, states that on average there will be 5 ms between packets, subject to a burst of 6. The
following is compliant, assuming both buckets are initially full.
0, 1, 2.5, 3, 4, 5, 6, 10, 15, 20
The first seven packets arrive at 1 ms intervals (the rate of the first filter) except for the packet that arrived at
T=2.5 instead of T=2. The sender was allowed to send again at T=3 instead of waiting until T=3.5 because
the bucket size in the first filter was 1.5 instead of 1.0. Here are the packet arrivals with the current size
of each bucket at the time of packet arrival, just before the bucket is decremented. At T=2.0, the filter2
bucket would be 4.4.
arrival:
Filter 1:
Filter 2:

T=0
1.5
6

T=1
1.5
5.2

T=2.5
1.5
4.5

T=3
1.0
3.6

T=4
1.0
2.8

T=5
1.0
2

T=6
1.0
1.2

T=10
1.5
1.0

T=15
1.5
1.0

T=20
1.5
1.0

If the packet arriving at T=2.5 had instead arrived at T=2, we would have the fastest compliant sequence for
this pair of filters. This is the sequence generated by a token-bucket shaper when there is a steady backlog
of packets and each is sent as soon as the bucket capacity (or capacities, when applicable) is full enough to
allow sending.

18.9.4 GCRA
Another formulation of the token-bucket specifications is the Generalized Cell Rate Algorithm, or GCRA;
this formulation is frequently used in classification of ATM traffic. A GCRA specification takes two parameters, a mean packet spacing time T, and an early-arrival allowance . For each packet we compute a
theoretical arrival time, tat, initially zero. A packet may arrive earlier by amount at most . Specifically, if
t is the time of actual arrival, we have two cases:
1. t tat : the packet is compliant, and we update tat to max(t,tat) + T
2. t < tat : the packet is too early and is noncompliant; tat is unchanged.
A flow satisfying GCRA(T, ) is equivalent to a token-bucket specification with rate 1/T packets/unit time,
and bucket size ( +1)/T; tat represents the time the bucket would next be full. The time to fill an empty
bucket is +1; if the bucket becomes full at time tat then, working backwards, it would contain enough to
send one packet at time tat .
For traffic flows with a more-or-less constant rate, represents the time by which one packet can be late
without permanently falling behind its regular 1/T rate. The GCRA formulation is sometimes more convenient than the token-bucket formulation, particularly when <T.

18.10 Applications of Token Bucket

Unlike fair queuing, token-bucket filtering can be implemented at the downstream end of a link, though possibly with results not quite in agreement with expectations. Let us return to the final scenario of 18.6 Applications of Fair Queuing:

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While fair queuing cannot be applied at R to enforce equal shares to A, B and C, we can implement a
token-bucket filter at R that limits each of A, B and C to 2 Mbps.
There are two drawbacks. First, the filter is not work-conserving: if A is idle, B and C will still only receive
2 Mbps. Second, in the absence of feedback there is no guarantee that limiting the traffic at R will eventually
result in reduced utilization of the ISPR link. While this is true for TCP traffic, due to the self-clocking
property, it is conceivable that a sender D somewhere is trying to send 8 Mbps of real-time UDP traffic to
A, via ISP and R. Three-quarters of the traffic would then fail to be compliant, and might be dropped by R,
but unless D gets feedback from A that not much of the traffic is getting through, and that it should reduce
its sending rate, the token-bucket filter at R will not achieve what we want. Most protocols do provide this
kind of feedback, but not all.

18.10.1 Guaranteeing VoIP Bandwidth

As a particular instance of the previous situation, suppose we have an Internet connection from our ISP and
want to begin using VoIP for telephony. We would like to reserve something like 64 Kbps of bandwidth for
VoIP (plus room for headers), so that large downloads do not degrade voice quality. We can easily do this
for the upstream direction, either with fair or priority queuing; priority queuing is an option here as the total
VoIP traffic is limited by the number of lines.
However, the downstream direction may be more of a problem, if we are unable to enlist the ISP to apply
fair queuing at the upstream end. As we argued in 18.6 Applications of Fair Queuing, fair queuing at the
downstream end has no effect.
Token-bucket at the downstream end might be an option. If we knew that the total link bandwidth was 500
bits/ms we might reserve 100 bits/ms, say, for VoIP traffic by limiting further delivery of non-VoIP traffic
to 400 bits/ms. This is indeed sometimes done. Unfortunately, we encounter three problems. The first is
that if no VoIP traffic is flowing then we probably do not want the 400 bits/ms cap on other traffic; we
might arrange this by applying the cap only when the phone is in use, or by setting aside a small enough
bandwidth fraction that it does not have a material affect on overall bulk bandwidth. The second problem
is the (remote) possibility discussed in the previous example that the sender might keep sending anyway, at
500 bits/ms; our downstream router can throw away as many bits as it wants but the link itself will still be
saturated. Finally, it is often quite difficult to determine exactly what the bandwidth of a particular Internet
connection is. Especially if, as is often the case, it is shared, or configured to change with time, or subject
to time-varying caps by the ISP.
If the competing downstream traffic is TCP, we theoretically could reduce the rate of upstream ACKs until
the downstream VoIP traffic no longer encounters excessive losses, but that is largely hypothetical and likely
to respond slowly.
Fortunately, typical VoIP bandwidth needs are low enough that one can often muddle through without providing any quality-of-service guarantees at all. This remains, however, a good example of the difficulties
often faced by real-time traffic.
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18.11 Token Bucket Queue Utilization

Suppose traffic meeting token-bucket specification TB(r,Bmax ) arrives at a router R, with no competition
from other traffic. The bucket fill rate r corresponds to the minimum outbound link bandwidth needed by R
to guarantee that the traffic does not build up; we do not want traffic on average arriving faster than it can
depart.
Intuitively, the bucket size Bmax corresponds to the amount of queue space at R that the flow can consume.
To make this more precise, we will argue that, if the output rate from R is at least r, then the number of
untransmitted bits stored at R is never more than Bmax .
To show this more formally, we start by proving the red line lemma implicit in the discussion of the graph
in 18.9.1 Token Bucket Definition above, that the sender can never cross the red line. Specifically, assume
the flow satisfies TB(r,Bmax ) and has a full bucket at time t=0. Let bits(t) be the cumulative number of bits
sent (packetized or not) by time t. The blue line is the graph bits = rt and the red line is the graph bits = rt +
Bmax ; we show the following:
bits(t) <= rt + Bmax
We first prove this so long as the graph is above the blue line; that is, bits(t) >= rt. We claim that the righthand side minus the left-hand side above, rt + Bmax - bits(t), represents the volume B(t) of fluid (or tokens)
in the bucket. Equating and rearranging slightly, we need to show B(t) + bits(t) - rt is always equal to Bmax .
This is true at t=0 when bits(t) = rt = 0 and the bucket is full. We next establish that its rate of change is also
0, and so it is constant.
While the bucket is not full, B(t) is always being filled at rate r. Correspondingly, rt is increasing at rate r, so
B(t) - rt is not affected by the fill rate. Similarly, B(t) is being reduced at exactly the rate bits(t) is increasing.
If we use the packet formulation, then when a packet arrives B(t) is reduced by the packet size and bits(t)
increases by exactly the same amount.
The Calculus Version
For readers familiar with calculus it may help to note dB(t)/dt = r - bits(t), at least if we assume, say, a
fluid bit-arrival model where bits(t) is differentiable. That is, the bucket volume B(t) increases at rate r
and also decreases at a rate equal to that of arriving data. Therefore, the rate of change of B(t) + bits(t)
- rt is just r - bits(t) + bits(t) - r = 0.
This does not quite apply when bits(t) falls below the blue line. However, we have nothing to prove then.
If bits(t) has a later interval above the blue line, starting at time t1 , we can reapply the argument above
re-starting the clock and the bits counter at t=t1 .
In fact, we can argue that whenever the bits(t) graph passes through a point below the blue line, such as
point A in the diagram above, then bits(t) cannot in the future climb above the new red line (the dashed red
line in the diagram) Bmax units above point A.

18.11.1 Token Bucket Through One Router

We now return to the claim about accumulation at a router R with outbound flow at least r; as before, let
bits(t) represent the cumulative amount of data received. As long as bits(t) is above the blue line, the router

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can continuously transmit at rate r and the net number of bits held within the router is bits(r) - rt. By the
argument above, this is bounded by Bmax . If bits(t) falls below the blue line, the routers queue is empty and
the router can transmit incoming data at least as fast as it is arriving.
While R can never be holding more than Bmax bytes, at the instant just before a packet finishes transmission
it can have Bmax bytes in the queue, plus the currently transmitting packet still taking up an entire buffer. As
a practical matter, then, we may need space equal to Bmax plus one packet.
While a token-bucket specification does not include a delay bound specifically, we can compute an upper
bound to the queuing delay at a router R as Bmax /r; this is the time it takes for one full buckets worth of
packets to be transmitted.
If we have N flows each individually satisfying TB(r,B), then the collective traffic will satisfy TB(Nr, NB)
(see exercise 12). However, a bucket size of NB will be needed only when all N individual flows have
their bursts gang up at a particular instant. Often it is possible to take advantage of theoretical or empirical statistical information and conclude that the collective traffic most of the time meets a token-bucket
specification TB(Nr, BN ) for BN significantly less than NB.

18.11.2 Token Bucket Through Multiple Routers

If we have a single TB(r,Bmax ) flow through N routers, however, the queuing delay is not larger than for a
single router, again assuming no competition. More specifically, assume that the traffic flow arrives at router
R1 satisfying TB(r,B), and passes in turn through R1 to RN . Each router Ri has an outbound bandwidth at
least as large as r. Then the total queuing delay through all N routers remains Bmax /r. If the packets pile up
to the maximum size Bmax , they only do so once.
To prove this we compare the TB sequence of packets with the same sequence of packets sent at a steady
rate r through the same series of routers. If the last bit of packet k is the Nth bit since we began, then for the
steady stream we send packet k at time N/r. We assume the link rates are all reduced to r.
Let t=0 represent the time we start counting bits. For every n, we established above that the nth bit of the
TB packet flow can be transmitted at most Bmax /r seconds ahead of the nth steady-stream bit, which is sent
at time n/r. The steady-stream packets do not encounter queuing delays at all, as each router has always
finished the previous one. The TB packets can each arrive no later than the steady-stream packets, as they
were sent earlier and they cannot cross. Therefore, the maximum delay faced by any TB packet is Bmax /r,
exactly as for traffic through a single router.

18.11.3 Delay Constraints

If a traffic flow arriving at a router R is compliant for token-bucket specification TB(r,B), then as we showed
above the amount of Rs queue space used by the flow will be bounded by B so long as R can devote at least
rate r to the flows traffic.
Now let us add a real-time delay constraint: suppose that R is not to be allowed to delay any of the flows
packets by more than time D. For the time being, assume that there is no other traffic at R. We now need to
make sure that R has sufficient bandwidth to forward a bucketful of size B within the time interval D. To
send a burst of size B in time D, bandwidth B/D is needed. Therefore, to satisfy the real-time constraint, R
needs outbound bandwidth
s = max(r, B/D)
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Example 1: suppose the traffic specification is TB(1/3, 10), where the rate is in (equal-sized) packets/sec,
and D is 40 sec. Then B/D is 1/4 packets/sec, and the necessary outbound bandwidth s is simply r=1/3.
Example 2: now suppose in the previous example that the delay limit D is 20 sec. In this case, we need s
= B/D = 1/2 packets/sec.
If there is other traffic, the delay constraint still holds, provided s represents the bandwidth allocated by R
to the flow, and the flows packets receive priority service at R, and we first subtract the largest-packet delay
as in 5.3.2 Packet Size and Real-Time Traffic.
Calculations of this sort often play a role in a routers decision on whether to accept a reservation for an
additional TB(r,B) flow with associated delay constraint.

18.12 Hierarchical Token Bucket

Token-bucket filters can also be used to form a hierarchy, as in 18.7.1 Generic Hierarchical Queuing. In
this section we will assume that token bucket is used only for shaping; that is, delaying packets until the
bucket has sufficiently filled. As usual, packets will remain in the leaf FIFO queues until they are ready to
be transmitted.
Central to the hierarchy is the conceptual time each internal token-bucket node releases its next packet;
that is, becomes able to inform its parent node (when asked) that it has a packet ready to send, even if the
packet physically remains waiting in one of the leaf queues. If a node N is informed by a child node that
a packet has been released and Ns bucket has sufficient capacity, then N releases the packet in turn to its
parent immediately; otherwise N waits until its bucket fills sufficiently to make the packet compliant. When
a packet arrives at a leaf node, it will be progressively released by each node along the path to the root; when
it is released by the root node it can be sent.
To make token-bucket filters classful, we will assume that each node may have multiple input subqueues,
but treats these as if they were consolidated into a single FIFO subqueue. That is, the node releases packets
to its parent in the order they were released to the node by its children.
Leaf nodes can mark each packet with its release time at the moment of arrival. Interior nodes may only be
able to determine their release times for packets that have been released by their child nodes.
It is now straightforward to define the peek() operation of 18.7.1 Generic Hierarchical Queuing: a node
looks at the set of packets it has released and returns the one with the earliest release time.
In a token-bucket hierarchy it makes a sense to say that two child flows have bucket sizes of 200 and 300,
respectively, while the combined flow is to be limited to a bucket size of 400.
The following diagram illustrates an example of a token-bucket hierarchy. The three token-bucket filters
TB1, TB2 and TB3 have rates in packets/ms and bucket sizes in packets.

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If TB1s rate is, as here, less than the sum of its child rates, then as long as its children always have packets
ready to send, the children will receive bandwidth in proportion to their token bucket rates. In the example
above, TB1s rate is 4 packets/ms and yet the sum of the rates of its children is 5 packets/ms. Each child
will therefore receive 4/5 its promised rate: TB2 will send at a rate of 2(4/5) packets/ms while TB3 will
send at rate of 3(4/5) packets/ms.
To see this, assume FIFO2 and FIFO3 remain nonempty for a period long enough for their buckets to
empty. TB2 and TB3 will then each release packets to TB1 at their respective rates of 2 packets/ms and 3
packets/ms. In the following sequence of release times to TB1, we assume TB3 starts at T=0 and TB2 at
T=0.01, to avoid ties. Packets from A released by TB2 are shown in italic:
0, 0.01, .33, 0.51, .67, 1.0, 1.01, 1.33, 1.51, 1.67, 2.0, 2.01
They will be dequeued by TB1 at 4 packets/ms, once TB1s bucket is empty. In the long run, TB3 has
released three packets into this sequence for every two of TB2s, so sender B will receive 3/5 of the dequeuings, and thus 3/5 of the 4 packet/ms root bandwidth.
We can also have each token-bucket node physically forward released packets to FIFO queues attached
to each parent node; we called this an internal-storage hierarchy in 18.7 Hierarchical Queuing. In this
particular case, the leaf-storage and internal-storage mechanisms function identically, provided the internal
links are infinitely fast and the internal queues infinitely large. See exercise 18.
There is no point in having a node with a bucket larger than the sum of its child buckets and also a rate larger
than the sum of its child rates. In the example above, in which the sum of the child rates exceeds the parent
rates, A would be able to send at a sustained rate of 2 packets/ms provided B sends at only 2 packets/ms as
well; reducing the child rates to 2(4/5) and 3(4/5) packets/ms respectively is not equivalent. If a nodes
rate is larger than the sum of the child rates, then it will be able to handle the child traffic without delay once
the child buckets have emptied. Before that, though, the parent bucket may be the limiting factor.

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18.13 Fair Queuing / Token Bucket combinations

At first glance, combining fair queuing with token bucket might seem improbable: the goal of fair queuing is
to be work-conserving, allowing the bandwidth assigned to an idle input class to be divided among the active
input classes, and the goal of token bucket is generally to limit a class to its token-bucket-defined maximum
transmission rate. The usual approach to a hierarchy-based synthesis is to allow the administrator to decide,
at each node of the hierarchy, whether or not the node can borrow (without repayment) bandwidth from
inactive siblings. If it can, the set of siblings with mutual borrowing privileges resembles a fair-queuing
scheduler; if not, the node is more like a token-bucket scheduler.

18.13.1 CBQ
CBQ was introduced in [CJ91] and analyzed in [FJ95]. It did not actually use the token-bucket mechanism,
but instead implemented shaping by keeping track of the average idle time (more precisely, non-transmitting
time) for a given input class. Input classes that tried to send too much were restricted, unless the node was
permitted to borrow bandwidth from a sibling. When an input class sent less than it was allowed, its
average utilization would fall; if a burst arrived then it would take some time for the average to catch up
and thus the node could briefly send faster than its assigned rate. However, the size of the bucket could be
controlled only indirectly.

18.13.2 Linux htb

The linux htb queuing discipline allows the same general functionality of CBQ, but replaces the averageidle calculations with token-bucket filters. This permits more direct control of burst sizes, and also avoided
some technical timing issues that CBQ users had to watch out for. For the sake of efficiency, htb uses the
quantum algorithm for fair queuing; as noted in 18.5.5 The Quantum Algorithm, this means less precise
control over packet delay.
It is not uncommon to arrange for htb to apply token-bucket shaping only at the leaf nodes, and to configure
the interior nodes do fair queuing only.
Each node in the tree has the following attributes:
its guaranteed rate, r, corresponding to the token-bucket rate
its burst allowance B, corresponding to the bucket size
its ceiling rate rceil ; nodes never send faster than this
In many cases rceil may simply be the output rate of the root node. For interior nodes, if the requested ceiling
rate is less than the sum of the child rates, the actual ceiling rate is adjusted upwards to match the sum of
the child rates; this means that interior nodes cannot do rate-limiting.
The most important attribute of each node is its guaranteed rate. The requested rate at each node should be at
least as large as the sum of the child rates. In the following diagram, all rates are in Kbps; burst allowances
are not shown. We will assume the root guaranteed rate, 600 Kbps, is also its ceiling rate.

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Packets are marked green, yellow or red depending on their situation. Red packets are those that must wait;
eventually they will turn yellow and then green.
Packets are considered green if they are now compliant (perhaps after waiting earlier) for one of the leaf
token-bucket nodes; green packets are sent as soon as possible.
After L1, L2 and L3 have each emptied their buckets, they will not exhaust node Ns rate. Similarly, after
N and M have emptied their buckets they will use only half of Rs rate. Nodes are allowed to borrow
bandwidth without payback from their parents rates; packets benefiting from such borrowed bandwidth
are marked yellow, and may also be sent immediately if no green packets are waiting. Borrowing is always
in proportion to a nodes guaranteed rate, in the manner of fair queuing. That is, the guaranteed rates of the
child nodes are treated as unnormalized fair-queuing weights; normalized weight fractions are obtained by
dividing by their total. N above would have normalized weight fraction 200/(200+100) = 2/3.
If L1, L2 and L3 engage in borrowing from N, and each has traffic to send, then each gets a total bandwidth
of 50, 50 and 100 Kbps, respectively. If L3 is idle, then L1 and L2 each would get 100 Kbps. If N and M
borrow in turn from R, they each can send at 400 and 200 Kbps respectively, in which case L1, L2 and L3
(again assuming all are active) get 100, 100 and 200 Kbps. If M elects not do do any borrowing, because it
has nothing to send, then N will get 600 Kbps and L1, L2 and L3 will get 150, 150 and 300.
If fair-queuing behavior is not desired, we can set rceil = r so that a node can never send faster than its
guaranteed rate. This allows htb to model the token-bucket-only hierarchy of 18.12 Hierarchical Token
Bucket.

18.13.3 Parekh-Gallager Theorem

As a final example relating token-bucket specifications and fair queuing, we present the Parekh-Gallager
Theorem, which provides a precise queuing-delay bound on traffic that enters a network meeting a tokenbucket specification TB(r,B) and which has a guaranteed weighted-fair-queuing fraction through each router
along the path.
Specifically, let us assume that the traffic travels from sender A to destination B through N routers R1 ... RN .
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The output rate of the ith router Ri is ri , of which our flow is guaranteed rate fi ri . Let f = min {fi |i<N}.
Suppose the maximum packet size for packets in our flow is S, and the maximum packet size including
competing traffic is Smax . Then the total delay encountered by the flows packets is bounded by the sum of
the following:
1. propagation delay (total single-bit delay along all N+1 links)
2. B/f
3. The sum from 1 to N of S/fi
4. The sum from 1 to N of Smax /ri
The second term B/f represents the queuing delay introduced by a single burst of size B; we showed in
18.11.2 Token Bucket Through Multiple Routers that this delay bound applied regardless of the number of
routers.
The third term represents the total store-and-forward delay at each router for packets belonging to our flow
under GPS; the delay at Ri is S/fi .
The final term represents the degree to which fair-queuing may delay a packet beyond the theoretical GPS
time expressed in the third term. If the routers were to use GPS, then the first three terms above would
bound the packet delay; we established in 18.5.4.7 Finishing-Order Bound that router Ri may introduce an
additional delay above and beyond the GPS delay of at most Smax /ri .

18.14 Epilog
If we want to use 100% the outbound bandwidth, but divide it among several senders according to a predetermined ratio, fair queuing is the tool to use. If we want to impose an absolute rather than a relative cap on
traffic, token bucket is appropriate.
Fair queuing has applications to the routing of ordinary packets; for example, if routers implement fair
queuing on a per-connection basis, then TCP senders will have no incentive to maximize queue utilization
and TCP Reno will lose its competitive advantage.
It is for real-time traffic, however, that queuing disciplines such as fair queuing, token bucket and even
priority queuing come into their own as fundamental building blocks. These tools allow us to guarantee a
bandwidth fraction to VoIP traffic, or to allow such traffic to be sent with minimal delay. In the next chapter
19 Quality of Service we will encounter fair queuing and token-bucket specifications repeatedly.

18.15 Exercises
1. Suppose a router uses fair queuing with three input classes, and uses the quantum algorithm of 18.5.2 Different Packet Sizes. The first class sends packets of size 900 bytes, the second sends packets of 400 bytes,
and the third sends packets of 200 bytes. List what would be sent by each flow in each of the first five
rounds.
2. Suppose we attempt to simulate BBRR as follows with the following strategy we will call SBBRR. Each
subqueue has a bit-position marker that advances by one bit for each bit we have available to send from that

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queue. If three queues are active, then in the time it takes us to send three bits, each marker would advance
by one bit. When all the bits of a packet have been ticked through, the packet is sent.
(a). Explain why this is not the same as BBRR fair queuing (even with equal-sized packets).
(b). Is it the same as BBRR if all input queues are active?
3. Suppose we modify the SBBRR strategy of the previous exercise so that, if the output link is ever idle,
and no packet has yet had all its bits ticked through by the bit-position marker, then we immediately send
the packet with the fewest bits remaining to be ticked through by the bit-position marker.
Suppose packets P1 of size 1000 and P2 of size 100 arrive on subqueue 1. Just after P1 begins transmission,
packet Q1 of size 400 arrives on subqueue 2. Fair queuing should send the packets in the order P1, Q1, P2;
show that the mechanism described here does not do that.
4. Suppose we attempt to implement fair queuing by calculating the finishing time Fj for Pj , the jth packet
in subqueue i, as follows.
Startj+1 = max(Fj , now) (now by wallclock time)
Fj = Startj + NLj
where N is the total number of subqueues, active or not.

(a). Suppose a router has three subqueues; ie N=3. The outbound bandwidth is 1 size unit / 1 time unit. At
T=0, packets P1, P2, P3, P4 and P5 arrive for subqueue 1, each of size 1 unit. At T=2 (by which point P1
and P2 will have finished transmission), packets Q1 and Q2 arrive on subqueue 2, also of size 1. What
finishing times will all the packets be assigned? In what order will they be transmitted?
(b). Is this strategy approximately equivalent to fair queuing if we are given that all subqueues of the router
are always active?

5. Suppose we modify the strategy of the previous exercise by letting N be the number of active subqueues
at the time of arrival of packet Pj . What happens if we have three input subqueues, and at T=0 five packets
arrive for subqueue 1, and at T=1 five packets arrive for subqueue 2. Assume all ten packets are of size 1,
and the output bandwidth is again 1 size unit per time unit.
6. The following packets all arrive at time T=0 at a router with an output rate of one size unit per time unit.
Subqueue 1: P1 of size 100, P2 of size 500, P3 of size 400
Subqueue 2: Q1 of size 300, Q2 of size 200, Q3 of size 600
Subqueue 3: R1 of size 400, R2 of size

(a). Find the BBRR virtual finishing time of each packet

(b). Give the actual wallclock finishing time of each packet, if the packets were sent via BBRR

6.5. Is byte-by-byte round-robin the same as bit-by-bit round-robin, if someone found a way to implement
these literally? Does byte-by-byte round-robin lead to the same transmission order as BBRR?

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7. Calculate GPS finishing times for the following packets, all present at T=0. There are two subqueues,

and their bandwidth fractions are and where = ( 5-1)/2 0.618 and = 2 = 1-. The packet sizes
for the two subqueues are as follows (they follow the Fibonacci sequence, except 2 appears twice):
: 2, 3, 8, 21, 55, 144, 377, 987
: 1, 2, 5, 13, 34, 89, 233, 610
Hint: you will have to evaluate to more decimal places than is shown here.
8. Suppose a WFQ router has two subqueues, each with a bandwidth fraction of =50%. The router
transmits 1 byte per ms. Initially, the subqueues are empty and T=0 and the GPS virtual clock is 0. At that
moment a packet P1 of size 1000 bytes arrives at the first subqueue. At T=500, a similarly sized packet P2
arrives at the second subqueue. Give, for each of P1 and P2,

(a). Its finishing time under the GPS virtual clock

(b). Its wallclock finishing time
(c). The value of the GPS virtual clock at the moment of WFQ finishing.

9. Suppose a router has three subqueues, and an outbound bandwidth of 1 packet per unit time. Twelve
packets arrive at or after T=0, timed so that the router remains busy until finishing the packets at T=12.

(a). What packet arrival schedule leads to the minimum final BBRR clock value?
(b). What schedule leads to the maximum final BBRR clock value?

Hint: the rate of the BBRR clock depends only on the number of active subqueues.
10. Suppose packets from three subqueues are sent using the quantum algorithm of 18.5.5 The Quantum
Algorithm. The packets are listed below in order of arrival for each subqueue, along with their lengths L;
the packets are all available at time T=0. The quantum is 1000 bytes. Give the order of transmission.
Subqueue 1
P1, L=700
P2, L=700
P3, L=700
P4, L=700

Subqueue 2
Q1, L=400
Q2, L=500
Q3, L=1000
Q4, L=200

Subqueue 3
R1, L=500
R2, L=600
R3, L=200
R4, L=900

11. At Disneyland, patrons often wait in a queue that winds slowly through one large waiting room, only to
feed into another queue in another room. Is this an example of hierarchical queuing, eg of one FIFO queue
feeding another, without classes?
12. If two traffic streams meet token-bucket specifications of TB(r1,b1) and TB(r2,b2) respectively, show
their commingled traffic must meet TB(r1+r2,b1+b2). Hint: imagine a common bucket of size b1+b2, filled
at rate r1 with red tokens and at rate r2 with blue tokens.
13. For each sequence of arrival times, indicate which packets are compliant for the given token-bucket
specification. If a packet is noncompliant, go on to the next arrival without decrementing the bucket.

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(a). TB(1/4,5): 0, 0, 0, 2, 3, 4, 5, 7, 9, 11, 15, 18

(b). TB(1/3,6): 0, 0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9
(c). TB(1/3,6): 0, 2, 4, 6, 8, 10, 12, 14, 16, 18

14. Find the fastest sequence (see the end of 18.9.3 Multiple Token Buckets) for the following flows. Both
start at T=0, and all buckets are initially full.

(a). TB(1/4, 4); packets can depart at a minimum of 1 time unit apart. Continue the sequence to at least
T=10
(b). TB(1/2, 4) and TB(1/8, 8); multiple packets can depart at the same instant. Continue to at least T=25.

15. Give the fastest sequence of packets compliant for all three of the following token-bucket specifications.
Continue the sequence at least until T=60.
TB(1/2, 1)
TB(1/6, 4)
TB(1/12, 8)
Hint: the first specification means arrival times must always be separated by at least 2. The middle specification should kick in by T=12.
16. Show that if a GPS traffic flow satisfies a token-bucket specification TB(r,B), then in any interval of time
t1 tt2 the amount of traffic is at most B + r(t2 t1 ). Hint: during the interval t1 tt2 the amount of fluid
added to the bucket is exactly r(t2 t1 ).
17. Show that a generic hierarchy of FIFO queuing disciplines, described in 18.7.1 Generic Hierarchical
Queuing, collapses to a single FIFO queue.
18. Show that the token-bucket leaf-storage hierarchy of 18.12 Hierarchical Token Bucket produces the
same result as an internal-storage hierarchy in which each intermediate token-bucket node contained a
real, infinite-capacity FIFO queue, and each node instantaneously transmitted each packet to the parents
FIFO queue as soon as it was released. Show that packets are transmitted by each hierarchy at the same
times. Hint: show that each node in the leaf-storage hierarchy releases a packet at the same time the
corresponding internal-storage hierarchy forwards the packet upwards.
19. The following linux htb hierarchies are labeled with their guaranteed rates. Is there any difference in
terms of the bandwidth allocations that would be received by senders A and B?
(a)
R
100
/
\
/
\
L1
L2
60
40
|
|
A
B

18.15 Exercises

(b)
R
100
/
\
/
\
L1
L2
30
20
|
|
A
B

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20. Suppose we know that the real-time traffic through a given router R uses at most 1 Mbps of the total 10
Mbps bandwidth. Consider the following two ways of giving the real-time traffic special treatment:
i. Using priority queuing, and giving the real-time traffic higher priority.
ii. Using weighted fair queuing, and giving the real-time traffic a 10% share

(a). Show that, if the real-time traffic meets a token-bucket specification with rate 1 Mbps and negligible
bucket size, then the two mechanisms are equivalent, in the sense that if the real-time and non-real-time
traffic flows are sending at fractions and , respectively, of the 10 Mbps outbound rate, with +=1 (and
with 10%), then the two methods above will actually send at the same rates.
(b). What differences can be expected if the bucket size is not negligible? Which approach will favor the
real-time fraction?

21. In the previous exercise, now suppose we have two separate real-time flows, each guaranteed by a
token-bucket specification not to exceed 1 Mbps. Is there a material difference between any pair of the
following?
i. Sending the two real-time flows at priority 1, and the remaining traffic at priority 2.
ii. Sending the first real-time flow at priority 1, the second at priority 2, and the remaining
traffic at priority 3.
iii. Giving each real-time flow a WFQ share of 10%, and the rest a WFQ share of 80%
22. Suppose a router uses priority queuing. There is one low-priority and one high-priority input. The
outbound bandwidth is r.

(a). If the high-priority queue is currently empty, what is the maximum time that an arriving high-priority
packet must wait?

(b). If the high-priority traffic follows a token-bucket description TB(rp,B), with rp < r, what is the
maximum time an arriving high-priority packet must wait? Hint: use 18.11 Token Bucket Queue
Utilization.

Your answer may include symbolic representations of any necessary additional parameters.

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19 QUALITY OF SERVICE

So far, the Internet has been presented as a place where all traffic is sent on a best-effort basis and routers
handle all traffic on an equal footing; indeed, this is often seen as a fundamental aspect of the IP layer. Delays
and losses due to congestion are nearly universal. For bulk file-transfers this is usually quite sufficient; one
way to look at TCP congestive losses, after all, is as part of a mechanism to ensure optimum utilization of
the available bandwidth.
Sometimes, however, at least some senders may wish to arrange in advance for a certain minimum level of
network services. Such arrangements are known as quality of service (QoS) assurances, and may involve
bandwidth, delay, loss rates, or any combination of these. Even bulk senders, for example, might sometimes
wish to negotiate ahead of time for a specified amount of bandwidth.
While any sender might be interested in quality-of-service levels, they are an especially common concern
for those sending and receiving real-time traffic such as voice-over-IP or videoconferencing. Real-time
senders are likely to have not only bandwidth constraints, but constraints on delay and on loss rates as well.
Furthermore, real-time applications may simply fail at least temporarily if these bandwidth, delay and
loss constraints are not met.
In any network, large or small, in which bulk traffic may sometimes create queue backlogs large enough to
cause unacceptable delay, quality-of-service assurances must involve the cooperation of the routers. These
routers will then use the queuing and scheduling mechanisms of 18 Queuing and Scheduling to set aside
bandwidth for designated traffic. This is a major departure from the classic Internet model of stateless
routers that have no information about specific connections or flows, though it is a natural feature of virtualcircuit routing.
In this chapter, we introduce some quality-of-service mechanisms for the Internet. We introduce the theory,
anyway; some of these mechanisms have not exactly been adopted with warm arms by the ISP industry.
Sometimes this is simply the chicken-and-egg problem: ISPs do not like to implement features nobody is
using, but often nobody is using them because their ISPs dont support them. However, often an ISPs
problem with a QoS feature comes down costs: routers will have more state to manage and more work to
do, and this will require upgrades. There are two specific cost issues:
it is not clear how to charge endpoints for their QoS requests (as with RSVP), particularly when the
endpoints are not direct customers
it is not clear how to compare special traffic like multicast, for the purpose of pricing, with standard
unicast traffic
Nonetheless, VoIP at least is here to stay on a medium scale. Streaming video is here to stay on a large scale.
Quality-of-service issues are no longer quite being ignored, or, at least, not blithely.
Note that a fundamental part of quality-of-service requests on the Internet is sharing among multiple traffic classes; that is, the transmission of best-effort and various grades of premium traffic on the same
network. One can, after all, lease a SONET line and construct a private, premium-only network; such an
approach is, however, expensive. Support for multiple traffic classes on the same network is sometimes
referred to generically as integrated services, not to be confused with the specific IETF protocol suite of
that name (19.4 Integrated Services / RSVP). There are two separate issues in an integrated network:
how to make sure premium traffic gets the service it requires
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how to integrate different traffic classes on the same network

The first issue is addressed largely through the techniques presented in 18 Queuing and Scheduling, and
applies as well to networks with only a single service level. The present chapter addresses mostly the second
issue.
Quality-of-service requests may be made for both TCP and UDP traffic. For example, an interactive TCP
connection might request a minimum-delay path, while a bulk connection might request a maximumbandwidth path and a streaming-prerecorded-video connection might request a specific guaranteed bandwidth. However, service quality is a particular concern of real-time traffic such as voice and interactive
video, where delay can be a major difficulty. Such traffic is more likely to use UDP than TCP, because of
the head-of-line blocking problem with the latter. We return to this in 19.3.3 UDP and Real-Time Traffic.

19.1 Net Neutrality

There is a school of thought that says carriers must carry all traffic on an equal footing, and should be
forbidden to charge extra for premium service. This is a strong formulation of the net neutrality principle,
and it potentially complicates the implementation of some of the services described here. In principle, it
may not matter whether the premium service involves interaction with routers, as is described in some of
the mechanisms below, or simply involves improved access to best-effort carriage.
There are, however, many weaker formulations of net neutrality; for example, one is that traffic carriage
must be non-discriminatory. That is, a carrier could charge more for premium service so long as it did not
single out individual providers for rate throttling.
Without taking sides on the net-neutrality debate, there are good reasons for arguing that additional charges
for specific services from backbone routers are more appropriate than additional charges to avoid rate throttling.

19.2 Where the Wild Queues Are

We stated above that to ensure quality-of-service standards, it is necessary to have the participation of routers
that have significant queue backlogs. It is not always clear, however, which routers these are. In the late
1990s, it was often claimed there was significant congestion at many (or at least some) backbone routers,
where the word is in quotes here as a reminder that the term does not have a precise technical meaning. It is
worth noting that this was also the era of 56 Kbps dialup access for most residential users.
In 2003, just a few years later, an analysis of the Internet in [CM03] concluded, the Internet cloud does
not contribute heavily to congestion. At the time of this writing (2013), it is often still claimed that the
Internet backbone has excess capacity, and therefore is seldom congested; delays, therefore, are more likely
to be encountered more locally.
More concretely, suppose traffic from A to B goes through routers R1, R2 and R3:

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If queuing delays (and losses) occur only at R1 and at R3, then there is no need to involve R2 in any
bandwidth-reservation scheme; the same is true of R1 and R3 if the delays occur only at R2. Unfortunately,
it is not always easy to determine the location of Internet congestion, and an abundance of bandwidth today
may become a dearth tomorrow: bandwidth usage ineluctably grows to consume supply. That said, the early
models for quality-of-service requests assumed that all (or at least most) routers would need to participate;
it is quite possible that this is practically speaking no longer true. Another possible consequence is that
adequate QoS levels might might be attainable with only the participation of ones immediate ISP.

19.3 Real-time Traffic

Real-time traffic is traffic with some sort of hard or soft delay bound, presumably larger than the one-way
no-load propagation delay. Such traffic can be said to be delay-intolerant. For voice or video traffic, a
packet arriving after the time at which it is to be played back might as well have been lost.
Fortunately, voice and video are also loss-tolerant, at least to a degree: a lost voice packet simply results in
a momentary voice dropout; a lost video packet might result in replay of the previous video frame. Handling
traffic that is both loss- and delay-intolerant is very difficult; we will not consider that case further.
Much (but not all) real-time traffic is also rate-adaptive. For example, the online-video service hulu.com
can send at resolutions of 288p, 360p, 480p and 720p, approximately corresponding to bandwidths of 480
Kbps, 700 Kbps, 1 Mbps, and 2.5 Mbps [2012 data]. Hulus software can dynamically choose the appropriate rate. (Hulu transmissions, where the consequence of delay is a pause in the video replay rather than a
loss, are not necessarily real-time; see 19.3.2 Streaming Video below.)
As another example of rate-adaptiveness, the voice-grade audio-compression codec Opus (a successor to
an earlier codec known as Speex) might normally be used at a 64 Kbps rate, but supports a more-or-less
continuous range of rates down to 8 Kbps. While the lower rates have lower voice quality, they can be used
as a fallback in the event that congestion prevents successful use of the 64 Kbps rate.
Generally speaking, rate-adaptivity notwithstanding, real-time traffic needs sufficient management that congestion becomes minimal. Real-time traffic should not be allowed to arrive at any router, for example, faster
than it can depart. The most definitive way to achieve this is via some sort of reservation or admissioncontrol mechanism, where new connections will not be accepted unless resources are available.

19.3.1 Playback Buffer

Real-time applications cannot avoid delay completely, of course, so the received stream will be delivered to
the receiving application (for playback, if it is a voice or video stream) slightly behind the time when it was
sent. Applications can intentionally increase this time by creating a playback buffer or jitter buffer; this
allows the smoothing over of variations in delay (known as jitter).
For example, suppose a VoIP application sends a packet every 20 ms (a typical rate). If the delay is exactly
50 ms, then packets sent at times T=0, 20, 40, etc will arrive at times T=50, 70, 90, etc. Now suppose the
delay is not so uniform, so packets sent at T=0, 20, 40, 60, 80 arrive at times 50, 65, 120, 125, 130.

19.3 Real-time Traffic

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packet
1
2
3
4
5

sent
0
20
40
60
80

expected
50
70
90
110
130

recd
50
55
120
125
130

(recd expected)
0
-5
30
15
0

The first and the last packet have arrived on time, the second is early, and the third and fourth are late by 30
and 15 ms respectively. Setting the playback buffer to 25 ms means that the third packet is not received in
time to be played back, and so must be discarded; setting the buffer to a value at least 30 ms means that all
the packets are received.
For non-interactive voice and video, there is essentially no penalty to making the playback buffer quite
long. But for telephony, if one speaker stops and the other starts, they will perceive a gap of length equal to
the RTT including playback-buffer delay. For voice, this becomes increasingly annoying if the RTT delay
reaches about 200-400 ms.

19.3.2 Streaming Video

Streaming video is something of a gray area. On the one hand, it is not really real-time; viewers do not
necessarily care how long the playback-buffer delay is as long as it is consistent. Playback-buffer delays
of ten seconds to up to a minute are not uncommon. As long as the sender can stay ahead of the playback
application, all is well. The real issue is bandwidth; a playback-buffer delay in the tens of seconds is two
orders of magnitude larger than the delay bounds that a genuine real-time application might request.
For very large videos, the sender probably coordinates with the playback application to limit how far ahead
it gets; this amounts to using a fixed-size playback buffer. That playback buffer can accommodate some
fluctuation in the delivery rate, but in the long run the delivery bandwidth must be at least as large as the
playback rate. For smaller videos, eg traditional ten-minute YouTube clips, the sender sends as fast as it can
without stopping. If the delivery bandwidth is less than the playback rate then the viewer can hit pause
and wait until the entire video has been downloaded.
On the other hand, viewers do not particularly like pauses, especially during long videos. If a viewer starts a
two-hour movie, average network congestion levels may change materially many times during that interval.
The viewer would prefer a more-or-less consistent bandwidth, even if competing traffic ramps up; rateadaptive playback is fine until one has signed up for HD-quality viewing. What the viewer here wants is an
average-bandwidth guarantee. This can be supplied by overbuilding the network, by implementing some of
the mechanisms of this chapter, or by various intermediate approaches.
For example, a large-enough network-content provider N might negotiate with a residential Internet carrier
C so that there is sufficient long-haul bandwidth from N to the end-user viewers. If the problem is backbone
congestion, then N might arrange to use BGPs MED option (10.6.5.3 MULTI_EXIT_DISC) to carry the
traffic as far as possible in its own network; this may also entail the creation of a large number of peering
points with Cs network (10.4.1 Internet Exchange Points). Alternatively, C might be persuaded to set aside
one very large traffic category in its own backbone network (perhaps using 19.7 Differentiated Services)
that is reserved for Ns traffic.

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19.3.3 UDP and Real-Time Traffic

The main difficulty with using TCP for real-time traffic is head-of-line blocking: when a loss does occur,
the TCP layer will hold any later data until the lost segment times out and is retransmitted. Even if the
receiving application is able simply to ignore the lost packet, it is not granted that option.
In theory, the TCP timeout interval may be only slightly more than the RTT, which is often well under 100
ms. In practice, however, it is often much larger, due in part to the use of a coarse-grained clock to keep
track of timeouts. TCP implementations are actually discouraged from using smaller timeout intervals; the
following is from RFC 6298 (2011); RTO stands for Retransmission TimeOut:
Whenever RTO is computed, if it is less than 1 second, then the RTO SHOULD be rounded up
to 1 second.
Such a policy makes TCP unsuitable, except in environments with vanishingly small loss rates, whenever
any genuine interactive response is needed; this includes telephony, video telephony, and most forms of
video conferencing that involve real-time feedback between participants. (TCP does work well with video
streaming.) While it is true that the popular video-telephony package Skype does in fact use TCP, this is
not because Skype has figured a way around this limitation; Skype sessions are not infrequently plagued by
congestion-related difficulties. The use of UDP allows an application the option of deciding that lost or late
data should simply be skipped over.

19.4 Integrated Services / RSVP

Integrated Services, or IntServ, was developed by the IETF in the late 1990s as a first attempt at providing
quality-of-service guarantees for selected Internet traffic. The IntServ model assumed all routers would
participate, or at least all routers along the connection path, though this is not strictly necessary. Connections
were to make reservations using the Resource ReSerVation Protocol RSVP.
Note that this is a major retreat from the datagram-routing stateless-router model. Were virtual circuits
the better routing model all along? For better or for worse, the marketplace appears to have answered this
question with an unambiguous no; IntServ has seen very limited adoption in the core Internet.
Under IntServ, routers maintain soft state about a connection, meaning that reservations must be refreshed
by the sender at regular intervals (eg 30 seconds); if not, the reservation can be discarded. This also means
that reservations can be recovered if the router crashes (though with some small probability of failure).
Traditional virtual-circuit switches maintain hard state, so that if a sender stops sending but fails to hang
up properly then the connection is still maintained (and perhaps charged for), and a router crash means the
connection is lost.
IntServ has mostly not been supported by the backbone ISP industry. Partly this is because of practical
difficulties in figuring out how to charge for reservations; if the charge is zero then everyone might make
reservations for every connection. Another issue is that a busy router might have to maintain thousands of
reservations; IntServ thus adds a genuine expense to the ISPs infrastructure.
At the time IntServ was developed, the dominant application envisioned was teleconferencing, in which
one speakers audio/video stream would be sent to a large number of receivers. Because of this, IntServ
was based on IP multicast, to which we therefore turn next. This decision made IntServ somewhat more
complex than would be necessary if point-to-point VoIP were all that was required, and future Internet
reservation mechanisms may jettison multicast support (see 19.9 NSIS). However, multicast and IntServ
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do share something fundamental in common: both require the participation of intermediate routers; neither
can be effectively implemented solely in end systems.

19.5 Global IP Multicast

The idea behind IP multicast, following up on 7.3.1 Multicast addresses, is that sender A transmits a stream
of packets (real-time or not) to a set of receivers {B1, B2, ..., Bn}, in such a way that no one packet of the
stream is transmitted more than once on any one link. This means that it is up to routers on the way to
duplicate packets that need to be forwarded on multiple outbound links. For example, suppose A below
wishes to send to B1, B2 and B3 in the following diagram:

Then R1 will receive packet 1 from A and will forward it to both B1 and to R2. R2 will receive packet 1
from R1 and forward it to B2 and R3. R3 will forward only to B3; R4 does not see the traffic at all. The
set of paths used by the multicast traffic, from the sender to all destinations, is known as the multicast tree;
these are shown in bold.
We should acknowledge upfront that, while IP multicast is potentially a very useful technology, as with
IntServ there is not much support for it within the mainstream ISP industry. The central issues are the need
for routers to maintain complex new information and the difficulty in figuring out how to charge for this
kind of traffic. At larger scales ISPs normally charge by total traffic carried; now suppose an ISPs portion
of a multicast tree is a single path all across the continent, but the tree branches into ten different paths near
the point of egress. Should the ISP consider this to be like one unicast connection, or more like ten?
Once Upon A Time, an ideal candidate for multicast might have been the large-scale delivery of broadcast
television. Ironically, the expansion of the Internet backbone has meant that large-scale video delivery is
now achieved with an individual unicast connection for every viewer. This is the model of YouTube.com,
Netflix.com and Hulu.com, and almost every other site delivering video. Online education also once upon a
time might have been a natural candidate for multicast, and here again separate unicast connections are now
entirely affordable. The bottom line for the future of multicast, then, is whether there is an application out
there that really needs it.
Note that IP multicast is potentially straightforward to implement within a single large (or small) organization. In this setting, though, the organization is free to set its own budget rules for multicast use.
Multicast traffic will consist of UDP packets; there is no provision in the TCP specification for multicast
connections. For large groups, acknowledgment by every receiver of the multicast UDP packets is impractical; the returning ACKs could consume more bandwidth than the outbound data. Fortunately, complete
acknowledgments are often unnecessary; the archetypal example of multicast traffic is loss-tolerant voice
and video traffic. The RTP protocol (19.11 Real-time Transport Protocol (RTP)) includes a response mechanism from receivers to senders; the RTP response packets are intended to at least give the sender some idea
of the loss rate. Some effort is expended in the RTP protocol (more precisely, in the companion protocol

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RTCP) to make sure that these response packets, from multiple recipients, do not end up amounting to more
traffic than the data traffic.
In the original Ethernet model for LAN-level multicast, nodes agree on a physical multicast address, and
then receivers subscribe to this address, by instructing their network-interface cards to forward on up to the
host system all packets with this address as destination. Switches, in turn, were expected to treat packets
addressed to multicast addresses the same as broadcast, forwarding on all interfaces other than the arrival
interface.
Global broadcast, however, is not an option for the Internet as a whole. Routers must receive specific
instructions about forwarding packets. Even on large switched Ethernets, newer switches generally try to
avoid broadcasting every multicast packet, preferring instead to attempt to figure out where the subscribers
to the multicast group are actually located.
In principle, IP multicast routing can be thought of as an extension of IP unicast routing. Under this
model, IP forwarding tables would have the usual udest,next_hop entries for each unicast destination,
and mdest,set_of_next_hops entries for each multicast destination. In the diagram above, if G represents
the multicast group of receivers {B1,B2,B3}, then R1 would have an entry G,{B1,R2}. All that is needed
to get multicast routing to work are extensions to distance-vector, link-state and BGP router-update algorithms to accommodate multicast destinations. (We are using G here to denote both the actual multicast
group and also the multicast address for the group; we are also for the time being ignoring exactly how a
group would be assigned an actual multicast address.)
These routing-protocol extensions can be done (and in fact it is quite straightforward in the link-state case,
as each node can use its local network map to figure out the optimal multicast tree), but there are some
problems. First off, if any Internet host might potentially join any multicast group, then each router must
maintain a separate entry for each multicast group; there are no opportunities for consolidation or for hierarchical routing. For that matter, there is no way even to support for multicast the basic unicast-IP separation
of addresses into network and host portions that was a crucial part of the continued scalability of IP routing. The Class-D multicast address block contains 228 270 million entries, far to many to support a
routing-table entry for each.
The second problem is that multicast groups, unlike unicast destinations, may be ephemeral; this would
place an additional burden on routers trying to keep track of routing to such groups. An example of an
ephemeral group would be one used only for a specific video-conference speaker.
Finally, multicast groups also tend to be of interest only to their members, in that hosts far and wide on the
Internet generally do not send to multicast groups to which they do not have a close relationship. In the
diagram above, the sender A might not actually be a member of the group {B1,B2,B3}, but there is a strong
tie. There may be no reason for R4 to know anything about the multicast group.
So we need another way to think about multicast routing. Perhaps the most successful approach has been the
subscription model, where senders and receivers join and leave a multicast group dynamically, and there is
no route to the group except for those subscribed to it. Routers update the multicast tree on each join/leave
event. The optimal multicast tree is determined not only by the receiving group, {B1,B2,B3}, but also by
the sender, A; if a different sender wanted to send to the group, a different tree might be constructed. In
practice, sender-specific trees may be constructed only for senders transmitting large volumes of data; less
important senders put up with a modicum of inefficiency.
The multicast protocol most suited for these purposes is known as PIM-SM, defined in RFC 2362. PIM
stands for Protocol-Independent Multicast, where protocol-independent means that it is not tied to a
specific routing protocol such as distance-vector or link-state. SM here stands for Sparse Mode, meaning
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that the set of members may be widely scattered on the Internet. We acknowledge again that, while PIM-SM
is a reasonable starting point for a realistic multicast implementation, it may be difficult to find an ISP that
implements it.
The first step for PIM-SM, given a multicast group G as destination, is for the designation of a router to
serve as the rendezvous point, RP, for G. If the multicast group is being set up by a particular sender, RP
might be a router near that sender. The RP will serve as the default root of the multicast tree, up until such
time as sender-specific multicast trees are created. Senders will send packets to the RP, which will forward
them out the multicast tree to all group members.
At the bottom level, the Internet Group Management Protocol, IGMP, is used for hosts to inform one of
their local routers (their designated router, or DR) of the groups they wish to join. When such a designated
router learns that a host B wishes to join group G, it forwards a join request to the RP.
The join request travels from the DR to the RP via the usual IP-unicast path from DR to RP. Suppose that
path for the diagram below is DR,R1,R2,R3,RP. Then every router in this chain will create (or update)
an entry for the group G; each router will record in this entry that traffic to G will need to be sent to the
previous router in the list (starting from DR), and that traffic from G must come from the next router in the
list (ultimately from the RP).

In the above diagram, suppose B1 is first to join the group. B1s designated router is R5, and the join packet
is sent R5R2R3RP. R5, R2 and R3 now have entries for G; R2s entry, for example, specifies that
packets addressed to G are to be sent to {R5} and must come from R3. These entries are all tagged with
*,G, to use RFC 2362s notation, where the * means any sender; we will return to this below when
we get to sender-specific trees.
Now B2 wishes to join, and its designated router DR sends its join request along the path
DRR1R2R3RP. R2 updates its entry for G to reflect that packets addressed to G are to be forwarded to the set {R5,R1}. In fact, R2 does not need to forward the join packet to R3 at all.
At this point, a sender S can send to the group G by creating a multicast-addressed IP packet and encapsulating it in a unicast IP packet addressed to RP. RP opens the encapsulation and forwards the packet down
the tree, represented by the bold links above.
Note that the data packets sent by RP to DR will follow the path RPR3R2R1, as set up above, even if

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the normal unicast path from R3 to R1 were R3R4R1. The multicast path was based on R1s preferred
next_hop to RP, which was assumed to be R2. Traffic here from sender to a specific receiver takes the
exact reverse of the path that a unicast packet would take from that receiver to the sender; as we saw in
10.4.3 Provider-Based Hierarchical Routing, it is common for unicast IP traffic to take a different path
each direction.
The next step is to introduce source-specific trees for high-volume senders. Suppose that sender S above is
sending a considerable amount of traffic to G, and that there is also an R6R2 link (in blue below) that can
serve as a shortcut from S to {B1,B2}:

We will still suppose that traffic from R2 reaches RP via R3 and not R6. However, we would like to allow
S to send to G via the more-direct path R6R2. RP would initiate this through special join messages sent
to R6; a message would then be sent from RP to the group G announcing the option of creating a sourcespecific tree for S (or, more properly, for Ss designated router R6). For R1 and R5, there is no change; these
routers reach RP and R6 through the same next_hop router R2.
However, R2 receives this message and notes that it can reach R6 directly (or, in general, at least via a
different path than it uses to reach RP), and so R2 will send a join message to R6. R2 and R6 will now each
have general entries for *,G but also a source-specific entry S,G, meaning that R6 will forward traffic
addressed to G and coming from S to R2, and R2 will accept it. R6 may still also forward these packets
to RP (as RP does belong to group G), but RP might also by then have an S,G entry that says (unless the
diagram above is extended) not to forward any further.
The tags *,G and S,G thus mark two different trees, one rooted at RP and the other rooted at R6. Routers
each use an implicit closest-match strategy, using a source-specific entry if one is available and the wildcard
*,G entry otherwise.
As mentioned repeatedly above, the necessary ISP cooperation with all this has not been forthcoming. As
a stopgap measure, the multicast backbone or Mbone was created as a modest subset of multicast-aware
routers. Most of the routers were actually within Internet leaf-customer domains rather than ISPs, let alone
backbone ISPs. To join a multicast group on the Mbone, one first identified the nearest Mbone router and
then connected to it using tunneling. The Mbone gradually faded away after the year 2000.
We have not discussed at all how a multicast address would be allocated to a specific set of hosts wishing to
form a multicast group. There are several large blocks of class-D addresses assigned by the IANA. Some of
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these are assigned to specific protocols; for example, the multicast address for the Network Time Protocol is
224.0.1.1 (though you can use NTP quite happily without using multicast). The 232.0.0.0/8 block is reserved
for source-specific multicast, and the 233.0.0.0/8 block is allocated by the GLOP standard; if a site has a
16-bit Autonomous System number with bytes x and y, then that site automatically gets the multicast block
233.x.y.0/24. A fuller allocation scheme waits for the adoption and development of a broader IP-multicast
strategy.

19.6 RSVP
We next turn to the RSVP (ReSerVation) protocol, which forms the core of IntServ.
The original model for RSVP was to support multicast, so as to support teleconferencing. For this reason,
reservations are requested not by senders but by receivers, as a multicast sender may not even know who all
the receivers are. Reservations are also for one direction; bidirectional traffic needs to make two reservations.
Like multicast, RSVP generally requires participation of intermediate routers.
Reservations include both a flowspec describing the traffic flow (eg a unicast or multicast destination) and
also a filterspec describing how to identify the packets of the flow. We will assume that filterspecs simply
define unidirectional unicast flows, eg by specifying source and destination sockets, but more general filterspecs are possible. A component of the flowspec is the Tspec, or traffic spec; this is where the token-bucket
specification for the flow appears. Tspecs do not in fact include a bound on total delay; however, the degree
of queuing delay at each router can be computed from the TB(r,Bmax ) token-bucket parameters as Bmax /r.
The two main messages used by RSVP are PATH packets, which move from sender to receiver, and the
subsequent RESV packets, which move from receiver to sender.
Initially, the sender (or senders) sends a PATH message to the receiver (or receivers), either via a single
unicast connection or to a multicast group. The PATH message contains the senders Tspec, which the
receivers need to know to make their reservations. But the PATH messages are not just for the ultimate
recipients: every router on the path examines these packets and learns the identity of the next_hop RSVP
router in the upstream direction. The PATH messages inform each router along the path of the path state
for the sender.
As an example, imagine that sender A sends a PATH message to receiver B, using normal unicast delivery.
Suppose the route taken is AR1R2R3B. Suppose also that if B simply sends a unicast message to
A, however, then the route is the rather different BR3R4R1A.

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As As PATH message heads to B, R2 must record that R1 is the next hop back to A along this particular
PATH, and R3 must record that R2 is the next reverse-path hop back to A, and even B needs to note R3 is
the next hop back to A (R1 presumably already knows this, as it is directly connected to A). To convey this
reverse-path information, each router inserts its own IP address at a specific location in the PATH packet,
so that the next router to receive the PATH packet will know the reverse-path next hop. All this path state
stored at each router includes not only the address of the previous router, but also the senders Tspec. All
these path-state records are for this particular PATH only.
The PATH packet, in other words, tells the receiver what the Tspec is, and prepares the routers along the
way for future reservations.
Each receiver now responds with its RESV message, requesting its reservation. The RESV packets are
passed back to the sender not by the default unicast route, but along the reverse path created by the PATH
message. In the example above, the RESV packet would travel BR3R2R1A. Each router (and also
B) must look at the RESV message and look up the corresponding PATH record in order to figure out how
to pass the reservation message back up the chain. If the RESV message were sent using normal unicast,
via BR3R4R1A, then R2 would not see it.
Each router seeing the RESV path must also make a decision as to whether it is able to grant the reservation.
This is the admission control decision. RSVP-compliant routers will typically set aside some fraction of
their total bandwidth for their reservations, and will likely use priority queuing to give preferred service to
this fraction. However, as long as this fraction is well under 100%, bulk unreserved traffic will not be shut
out. Fair queuing can also be used.
Reservations must be resent every so often (eg every ~30 seconds) or they will time out and go away; this
means that a receiver that is shut down without canceling its reservation will not continue to tie up resources.
If the RESV messages are moving up a multicast tree, rather than backwards along a unicast path, then they
are likely to reach a router that already has granted a reservation of equal or greater capacity. In the diagram
below, router R has granted a reservation for traffic from A to receiver B1, reached via Rs interface 1, and
now has a similar reservation from receiver B2 reached via Rs interface 2.

Assuming R is able to grant B2s reservation, it does not have to send the RESV packet upstream any further
(at least not as requests for a new reservation); B2s reservation can be merged with B1s. R simply will
receive packets from A and now forward them out both of its interfaces 1 and 2, to the two receivers B1 and
B2 respectively.
It is not necessary that every router along the path be RSVP-aware. Suppose A sends its PATH messages to
B via AR1R2aR2bR3B, where every router is listed but R2a and R2b are part of a non-RSVP
cloud. Then R2a and R2b will not store any path state, but also will not mark the PATH packets with
their IP addresses. So when a PATH packet arrives at R3, it still bears R1s IP address, and R3 records
R1 as the reverse-path next hop. It is now possible that when R3 sends RESV packets back to R1, they
will take a different path R3R2cR1, but this does not matter as R2a, R2b and R2c are not accepting

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reservations anyway. Best-effort delivery will be used instead for these routers, but at least part of the path
will be covered by reservations. As we outlined in 19.2 Where the Wild Queues Are, it is quite possible
that we do not need the participation of the various R2s to get the quality of service wanted; perhaps only
R1 and R3 contribute to delays.
In the multicast multiple-sender/multiple-receiver model, not every receiver must make a reservation for
all senders; some receivers may have no interest in some senders. Also, if rate-adaptive data transmission
protocols are used, some receivers may make a reservation for a sender at a lower rate than that at which
the sender is sending. For this to work, some node between sender and receiver must be willing to decode
and re-encode the data at a lower rate; the RTP protocol provides some support for this in the form of RTP
mixers (19.11.1 RTP Mixers). This allows different members of the multicast receiver group to receive the
same audio/video stream but at different resolutions.
From an ISPs perspective, the problems with RSVP are that there are likely to be a lot of reservations,
and the ISP must figure out how to decide who gets to reserve what. One model is simply to charge for
reservations, but this is complicated when the ISP doing the charging is not the ISP providing service to the
receivers involved. Another model is to allow anyone to ask for a reservation, but to maintain a cap on the
number of reservations from any one site.
These sorts of problems have largely prevented RSVP from being implemented in the Internet backbone.
That said, RSVP is apparently quite successful within some corporate intranets, where it can be used to
support voice traffic on the same LANs as data.

19.7 Differentiated Services

Differentiated Services, or DiffServ, was created as a low-overhead alternative to IntServ. The idea behind
DiffServ is to replace IntServs many reservations with just two service classes: regular (for everyones
bulk traffic) and premium (for what in IntServ would have been everyones reserved traffic). For this
reason, DiffServ is sometimes describe as providing coarse-grained resource allocation versus RSVPs
fine-grained per-flow allocations. Like IntServ, DiffServ is easiest to implement within a single ISP or
Autonomous System; however, DiffServ may be reasonably practical to implement across ISP boundaries
as well. A set of routers agreeing on a common DiffServ policy is called a DS domain; a DS domain might
consist of a single ISP but might also comprise a larger backbone ISP and some regional customer ISPs
that have negotiated a single DiffServ policy.
Packets are marked at (and only at) the border routers of each DS domain, as the traffic in question enters
that domain. The DS domains interior routers do no marking and no traffic policing. At these interior
routers, priority queuing is generally used to give premium service to the marked premium packets.
Packets are marked using the six bits of the DS field of the IPv4 header (7.1 The IPv4 Header); these were
part of what was originally the IP Type-of-Service bits. DiffServ traffic is divided into several classes called
Per Hop Behaviors, or PHBs; PHBs are perhaps best thought of as service classes. The simplest PHB is the
default PHB, meaning the packet gets no special processing.
The Class Selector PHB is for backwards compatibility with the three-bit precedence subfield of the old
IPv4 Type-of-Service field.
The Expedited Forwarding (EF) PHB (19.7.1 Expedited Forwarding) is arguably the best service. It is
meant for traffic that has delay constraints in addition to rate constraints. The newer Voice Admit PHB is
closely related and has been recommended as the preferred PHB for voice telephony.
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Assured Forwarding, or AF (19.7.2 Assured Forwarding), is really four separate PHBs, corresponding
to four classes 1 through 4. It is meant for providing (soft) service guarantees that are contingent on the
senders staying within a certain rate specification. Each AF class has its own rate specification; these rate
specifications are entirely at the discretion of the implementing DS domain. AF uses the first three bits to
denote the AF class, and the second three bits to denote the drop precedence.
Here are the six-bit patterns for the above PHBs; the AF drop-precedence bits are denoted ddd.
000 000: default PHB (best-effort delivery)
001 ddd: AF class 1 (the lowest priority)
010 ddd: AF class 2
011 ddd: AF class 3
100 ddd: AF class 4 (the best)
101 110: Expedited Forwarding
101 100: Voice Admit
11x 000: Network control traffic (RFC 2597)
xxx 000: Class Selector (traditional IPv4 Type-of-Service)
The goal behind premium PHBs such as EF and AF is for the DS domain to set some rules on admitting
premium packets, and hope that their total numbers to any given destination are small enough that high-level
service targets can be met. This is not exactly the same as a guarantee, and to a significant degree depends on
statistics. The actual specifications are written as per-hop behaviors (hence the PHB name); with appropriate
admission control these per-hop behaviors will translate into the desired larger-scale behavior.
One possible Internet arrangement is that a leaf domain or ISP would support RSVP, but hands traffic off
to some larger-scale carrier that runs DiffServ instead. Traffic with RSVP reservations would be marked on
entry to the second carrier with the appropriate DS class.

19.7.1 Expedited Forwarding

The goal of the EF PHB is to provide low queuing delay to the marked packets; the canonical example
is VoIP traffic, even though the latter now has its own PHB. EF is generally considered to be the best DS
service, though this depends on how much EF traffic is accepted by the DS domain.
Each router in a DS domain supporting EF is configured with a committed rate, R, for EF traffic. Different
routers can have different committed rates. At any one router, RFC 3246 spells out the rule this way (note
that this rule does indeed express a per-hop behavior):
Intuitively, the definition of EF is simple: the rate at which EF traffic is served at a given output
interface should be at least the configured rate R, over a suitably defined interval, independent
of the offered load of non-EF traffic to that interface.
To the EF traffic, in other words, each output interface should appear to offer bandwidth R, with no competing non-EF traffic. In general this means that the network should appear to be lightly loaded, though that
appearance depends very much on strict control of entering EF traffic. Normally R will be well below the
physical bandwidths of the routers interfaces.

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RFC 3246 goes on to specify how this apparent service should work. Roughly, if EF packets have length
L then they should be sent at intervals L/R. If an EF packet arrives when no other EF traffic is waiting, it
can be held in a queue, but it should be sent soon enough so that, when physical transmission has ended,
no more than L/R time has elapsed in total. That is, if R and L are such that L/R is 10s, but the physical
bandwidth delay in sending is only 2s, then the packet can be held up to 8s for other traffic.
Note that this does not mean that EF traffic is given strict priority over all other traffic (though implementation of EF-traffic processing via priority queuing is a reasonable strategy); however, the sending interface
must provide service to the EF queue at intervals of no more than L/R; the EF rate R must be in effect at
per-packet time scales. Queuing ten EF packets and then sending the lot of them after time 10L/R is not
allowed. Fair queuing can be used instead of priority queuing, but if quantum fair queuing is used then the
quantum must be small.
An EF routers committed rate R means simply that the router has promised to reserve bandwidth R for EF
traffic; if EF traffic arrives at a router faster than rate R, then a queue of EF packets may build up (though
the router may be in a position to use some of its additional bandwidth to avoid this, at least to a degree).
Queuing delays for EF traffic may mean that someones application somewhere fails rather badly, but the
router cannot be held to account. As long as the total EF traffic arriving at a given router is limited to that
routers EF rate R, then at least that router will be able to offer good service. If the arriving EF traffic meets
a token-bucket specification TB(R,B), then the maximum number of EF packets in the queue will be B and
the maximum time an EF packet should be held will be B/R.
So far we have been looking at individual routers. A DS domain controls EF traffic only at its border; how
does it arrange things so none of its routers receives EF traffic at more than its committed rate?
One very conservative approach is to limit the total EF traffic entering the DS domain to the common
committed rate R. This will likely mean that individual routers will not see EF traffic loads anywhere close
to R.
As a less-conservative, more statistical, approach, suppose a DS domain has four border routers R1, R2, R3
and R4 through which all traffic must enter and exit, as in the diagram below:

Suppose in addition the domain knows from experience that exiting EF traffic generally divides equally
between R1-R4, and also that these border routers are the bottlenecks. Then it might allow an EF-traffic
entry rate of R at each router R1-R4, meaning a total entering EF traffic volume of 4R. Of course, if on
some occasion all the EF traffic entering through R1, R2 and R3 happened to be addressed so as to exit via
R4, then R4 would see an EF rate of 3R, but hopefully this would not happen often.
If an individual ISP wanted to provide end-user DiffServ-based VoIP service, it might mark VoIP packets
for EF service as they entered (or might let the customer mark them, subject to the ISPs policing). The
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rate of marked packets would be subject to some ceiling, which might be negotiated with the customer as
a certain number of voice lines. These marked VoIP packets would receive EF service as they were routed
within the ISP.
For calls also terminating within that ISP or switching over to the traditional telephone network at an
interface within that ISP this would be all that was necessary, but some calls will likely be to customers of
other ISPs. To address this, the original ISP might negotiate with its ISP neighbors for continued preferential
service; such service might be at some other DS service class(eg AF). Packets would likely need to be remarked as they left the original ISP and entered another.
The original ISP may have one larger ISP in particular with which it has a customer-provider relationship.
The larger ISP might feel that with its high-volume internal network it has no need to support preferential
service, but might still agree to carry along the original ISPs EF marking for use by a third ISP down the
road.

19.7.2 Assured Forwarding

AF, documented in RFC 2597, is simpler than EF, but with little by way of a delay guarantee. The macroscopic goal is to grant specific rate assurances to certain traffic classes, eg the traffic from a certain set of
customers. Distinct traffic classes are represented by distinct AF classes, which are limited to four. If a
contributor to a traffic class sends more traffic than that particular contributor is permitted, the excess traffic
is simply marked with a higher drop precedence. Further on in the network, the traffic may or may not be
dropped.
The drop-precedence values for the second three bits for the AF classes are as follows:
010: do not drop
100: medium
110 high
Different priority-queuing levels may used for the different AF classes; this would ensure that all the traffic
needs of AH class 4 are met before AH class 3, and down the line to AH class 1 at the lowest AH priority, just
above bulk (default) traffic. Provided careful limits are placed on how much traffic is admitted to each class,
strict priority queuing need not lead to the starvation of bulk traffic. Use of priority queuing is not mandatory;
another option is fair queuing where the higher-precedence AH classes get a greater fair-queuing-guaranteed
fraction of the total bandwidth, at least relative to their traffic volumes.
Traffic with different drop-precedence values within a single AF class, however, is not assigned to different
subqueues (Priority, Fair or otherwise). Doing that would almost certainly lead to significant reordering
of the overall traffic, and reordering tends to be bad for TCP traffic. If different priority-queuing levels
were used here, for example, higher-drop-precedence packets would be delayed until lower-drop-precedence
queues were emptied; almost any form of queuing using multiple queues for a different output interface
would likely lead to at least some reordering. So long as one TCP connection remains in a single AF class
(eg because all traffic to or from a given site remains in a single AF class), the packets of that connection
should not be reordered.
A classic application of AF is for an ISP to be able to grant different performance levels to different customers: gold, silver and bronze (again from the Appendix to RFC 2597). Customers would pay an ISP more
to carry their traffic in a higher category. Along with their AF level, each customer would negotiate with the

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ISP their average rate and also their committed and excess burst (bucket) capacities. As the customers
traffic entered the network, it would encounter two token-bucket filters, with rate equal to the agreed-upon
rate and with the two different bucket sizes: TB(r,Bcommitted ) and TB(r,Bexcess ). Traffic compliant with the
first token-bucket specification would be marked do not drop; traffic noncompliant for the first but compliant for the second would be marked medium and traffic noncompliant for either specification would be
marked with a drop precedence of high. (The use of Bexcess here corresponds to the sum of committed
burst size and excess burst size in the Appendix to RFC 2597.)
Customers would thus be free to send faster than their agreed-upon rate, subject to the excess traffic being
marked with a lower drop precedence.
This process means that an ISP can have many different Gold customers, each with widely varying rate
agreements, all lumped together in the same AF-4 class. The individual customer rates, and even the sum of
the rates, may have only a tenuous relationship to the actual internal capacity of the ISP, although it is not in
the ISPs long-term interest to oversubscribe any AF level.
If the ISP keeps the AF class 4 traffic sparse enough, it might outperform EF traffic in terms of delay. The
EF PHB rules, however, explicitly address delay issues while the AF rules do not.

19.8 RED with In and Out

Differentiated Services Assured Forwarding (AF) fits nicely with RIO routers: RED with In and Out (or In,
Middle, and Out); see 14.8.3 RED gateways. In RIO routers, each drop-precedence value (the In, Middle and Out here) is subject to a different drop threshold. Packets with higher drop-precedence values
would experience RED signaling losses at correspondingly lower degrees of RED queue utilization. This
means that TCP connections with a significant fraction of higher-drop-precedence values would encounter
RED-induced losses sooner and would thus reduce their congestion window sooner than connections
that stayed within their committed rate. Because each AF class is sent at a single priority, TCP connections
within a single AF class should not experience problems(eg reordering) other than these RED signaling
losses. All this has the effect of gently encouraging customers to stay within their committed traffic limits.
RIO is likely to have a meaningful effect only on TCP connections; real-time UDP traffic may be affected
slightly or not at all by RED signaling-type losses.

19.9 NSIS
The practical problem with RSVP is the need for routers to participate. One approach to gaining ISP cooperation might be a lighter-weight version of RSVP, though that is speculative; Differentiated Services was
supposed to be just that and it too has not been widely adopted within the commercial Internet backbone.
That said, work has been underway for quite some time now on a replacement protocol suite. One candidate
is Next Steps In Signaling, or NSIS, documented in RFC 4080 and RFC 5974.
NSIS breaks the RSVP design into two separate layers: the signal transport layer, charged with figuring out
how to reach the intermediate routers, and the higher signaling application layer, charged with requesting
actual reservations. One advantage of this two-layer approach is that NSIS can be adapted for other kinds
of signaling, although most NSIS signaling can be expected to be related to a specific network flow. For

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example, NSIS can be used to tell NAT routers to open up access to a given inside port, in order to allow a
VoIP (or other) connection to proceed.
Generally speaking, NSIS also abandons the multicast-centric approach of RSVP. Signaling traffic travels
hop-by-hop from one NSIS Element, or NE, to the next NE on the path. In cases when the signaling traffic
follows the same path as the data (the path-coupled case), the signaling packet would likely be addressed
to the ultimate destination, but recognized by each NE router on the path. NE routers would then add
something to the packet, and perhaps update their own internal state. Nonparticipating (non-NE) routers
would simply forward the signaling packet, like any other IP packet, further towards its ultimate destination.

19.10 Comcast Congestion-Management System

The large ISP Comcast introduced a DiffServ-like scheme to reduce internal congestion; this was eventually
documented in RFC 6057. The effect of the scheme is to reduce bandwidth for those subscribers who
are using the largest amount of bandwidth; this reduction is done only when the ISPs local network is
experiencing significant congestion. The goal is to avoid saturation of the ISP-level backbone, thus perhaps
improving fairness for other users.
By throttling bulk traffic throughout its network before maximum capacities are reached, this strategy has the
potential to improve the performance of real-time protocols without IntServ- or DiffServ-type mechanisms.
During non-congested operation, all user traffic is tagged within Comcasts network as Priority Best Effort (PBE). This is akin to a medium drop precedence for a particular DiffServ Assured Forwarding class
(19.7.2 Assured Forwarding), though RFC 6057 does not claim that DiffServ is actually used. When congestion occurs, traffic of some high-volume users may be tagged instead as Best Effort (BE), meaning that
it has a lower priority and a higher drop precedence.
The mechanism first defines a Near-Congestion State for ports on routers at a certain level of Comcasts
network. These routers are known as Cable Modem Termination Systems, or CMTSs. Each CMTS node
serves about 5,000 subscribers. Near-Congestion State is declared when the port utilization exceeding a
specific percentage of its maximum for a specific amount of time; typically the utilization threshold is 7080% and the time interval is 15 minutes. One CMTS port typically serves 200-300 customers. Previous
experience had established CMTS ports as the likely locus for congestion.
When a particular CMTS port enters the Near-Congestion State, the ports traffic is monitored on a persubscriber basis to see if any subscribers are in an Extended High Consumption State, or EHCS, again
triggered when a subscriber exceeds a given percentage of their subscriber rate for a given number of minutes. The threshold for entering EHCS state is typically 70% of the subscriber rate either the upstream rate
or the downstream rate for 15 minutes.
When the switch port is in Near-Congestion State and the subscribers traffic is in EHCS, then all that subscribers traffic is marked Best Effort (BE) rather than PBE, corresponding to a higher AF drop precedence.
Downstream traffic is marked (at the CMTS or even higher up in the network) if it is the users downstream utilization that is high; upstream traffic is marked at the users connection to the network if upstream
utilization is high.
Other routers may then drop BE-marked traffic preferentially, versus PBE-marked traffic. This will occur
only at routers that are actively experiencing congestion, perhaps using a mechanism like RIO (19.8 RED
with In and Out), though RIO was originally intended only for TCP traffic.

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A subscriber leaves the EHCS state when the subscribers bandwidth drops below 50% of the subscription
rate for 15 minutes, where presumably the 50% rate is measured over some very short timescale. Note that
this means that a user with a TCP sawtooth ranging from 30% to 60% of the maximum might remain in the
EHCS state indefinitely.
Also note that all the subscribers traffic will be marked as BE traffic, not just the overage. The intent is to
provide a mild disincentive for sustained utilization within 70% of the subscriber maximum rate.
Token bucket specifications are not used, except possibly over very small timescales to define utilizations
of 50% and 70%. The (larger-scale) token-bucket alternative might be to create a token-bucket specification
for each customer TB(r,B) where r is some fraction of the subscription rate and B is a bucket of modest
size. All compliant traffic would then be marked PBE and noncompliant traffic would be marked BE. Such
a mechanism might behave quite differently, as only traffic actually over the ceiling would be marked.

19.11 Real-time Transport Protocol (RTP)

RTP is a convenient framework for carrying real-time traffic. It runs on top of UDP, largely to avoid headof-line blocking (11.1 User Datagram Protocol UDP), and offers several advantages over raw UDP. It
does not, however, involve interactions with the intervening routers, and therefore cannot offer any service
guarantees.
RTP headers include a sequence number, an application-provided timestamp representing when the attached
data was recorded, and basic mechanisms for handling traffic that is a merger of several original sources
(contributing sources) of related content. RTP also includes support for multicast transmission, eg for the
voice and video streams that make up a teleconference. Indeed, teleconferencing is arguably the application
for which RTP was developed, although it is also widely used for point-to-point applications such as VoIP.
Perhaps the most striking feature of RTP is the absence of frequent acknowledgments. There is indeed a
provision for receiver responses, but they are often sent only at several-second intervals, and are frequently
omitted entirely. These responses use the companion RTCP protocol, and the receiver report format.
RTCP receiver-report packets are often thought of not as acknowledgments but as a source of statistics about
how the RTP flow is being delivered; in particular, the RTCP packets contain a ratio of how many sender
packets arrived since the previous RTCP report, the highest sequence number received, and a measure of the
degree of jitter.
For multicast, the infrequent acknowledgments make sense. If every receiver of a multicast group sent
acknowledgments totaling 3% of the received-content bandwidth (about the TCP ratio), and if there were
100 receivers in the group (not large for a teleconference), then the total acknowledgment traffic arriving
at the sender would be triple the content traffic. There are, in fact, mechanisms in place to limit the total
RTCP bandwidth to no more than 5% of the outbound content stream; if a multicast stream has thousands
of receivers then those receivers will each respond relatively infrequently (eg at intervals of many seconds,
if not many minutes).
Even in one-to-one transmission settings, though, RTP may not send many acknowledgments. Many VoIP
systems are configured not to send RTCP responses at all by default; when these are enabled, the rate has a
typical minimum of one RTCP response per second. In one second, a VoIP sender may transmit 50 packets.
One explanation for this is that when a voice call is encountering significant congestion, the participants are
expected to hang up, rather than keep the line open.

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For two-way voice calls, symmetric RTP is often used (RFC 4961). This means that each party uses the
same port for sending and receiving. This is done only for convenience and to handle NAT routers; the two
separate directions are completely independent and do not serve as acknowledgments for one another, as
would be the case for bidirectional TCP traffic. Indeed, one can block one direction of a VoIP symmetricRTP stream and the call continues on indefinitely, transmitting voice only in the other direction. When the
block is removed, the blocked voice flow simply resumes.

19.11.1 RTP Mixers

Mixers are network nodes that take one or more RTP source streams and perform any combination of the
following operations
consolidation of multiple streams into one
translation to a different audio or video format
re-encoding to a lower-bandwidth format
As an example of consolidation, video of several panelists might be consolidated into a single video frame,
in which the current speaker has a larger window. At the audio level, a mixer might combine the streams
from several individual microphones. If all parties are at the same location this kind of consolidation can
be performed by the primary sender, but mixers may be useful if conference participants are geographically
dispersed. As for translation, some participants may prefer to receive in a different format.
Perhaps the most important task for mixers, however, is rate-adaptation. With a unicast connection, if
congestion is experienced then the sender itself can adapt to it by switching the encoding. This is not
appropriate for multicast, however; if one receiver is experiencing congestion it is not unlikely that other
receivers may be doing just fine. Therefore, if a multicast receiver is having trouble receiving at a high data
rate, it is up to it to find a mixer that offers a lower-rate encoding, and switch its subscription/reservation
to that mixer instead. Mixers offering a range of rates might be set up near the sender (or at least logically
near); they might also be geographically distributed to be nearer to clusters of likely receivers. One host
might provide several mixers offering a range of bandwidth options.
One thing mixers do not do is mix audio and video together; separate audio and video streams should be
carried by separate RTP sessions. RTP packets carry timestamps to allow a receiver to synchronize audio
and video playback, so mixing is not necessary. But more importantly, mixing of audio and video makes it
difficult to re-encode to a different format, difficult for a receiver to subscribe only to the audio, and difficult
for the RTCP reports to indicate which original stream was encountering more losses.
An individual sending point in RTP, complete with timing information, is called a synchronization source
or SSRC. At the point of origin, each camera and microphone might be an SSRC. SSRCs are identified by
a 32-bit identifier. Actual SSRC identifiers are to be chosen pseudo-randomly (and there is a provision in
the protocol for making sure two different sources do not choose the same SSRC identifier), but for many
practical purposes the SSRC can be thought of as a host identifier, like an IP address.
When a mixer combines multiple SSRCs into one stream (or even when a mixer takes as input only a single
SSRC), the mixer becomes the new synchronization source and creates a new SSRC identifier for all its
output packets; the mixer also generates a new series of synchronization timestamps. However, the SSRC
identifiers for the streams that the mixer used as input are attached to all RTP packets from the mixer; each
of these is now known as a contributing source or CSRC.

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19.11.2 RTP Packet Format

The basic RTP header has the following format, from RFC 3550:

The Ver field holds the version, currently 2. The P bit indicates that the RTP packet includes at least one
padding byte at the end; the final padding byte contains the exact count.
The X bit is set if an extension header follows the basic header; we do not consider this further.
The CC field contains the number of contributing source (CSRC) identifiers (below). The total header size,
in 32-bit words, is 3+CC.
The M bit is set to allow the marking, for example, of the first packet of each new video frame. Of course,
the actual video encoding will also contain this information.
The Payload Type field allows (or, more precisely, originally allowed) for specification of the audio/visual
encoding mechanism (eg -law/G.711), as described in RFC 3551. Of course, there are more than 27 possible encodings now, and so these are typically specified via some other mechanism, eg using an extension
header or the separate Session Description Protocol (RFC 4566) or as part of the RTP streams announcement. RFC 3551 put it this way:
During the early stages of RTP development, it was necessary to use statically assigned
payload types because no other mechanism had been specified to bind encodings to payload
types.... Now, mechanisms for defining dynamic payload type bindings have been specified in
the Session Description Protocol (SDP) and in other protocols....
The sequence-number field allows for detection of lost packets and for correct reassembly of reordered
packets. Typically, if packets are lost then the receiver is expected to manage without them; there is no
timeout/retransmission mechanism in RTP.
The timestamp is for synchronizing playback. The timestamp should be sufficiently fine-grained to support
not only smooth playback but also the measurement of jitter, that is, the degree of variation in packet arrival.
Each encoding mechanism chooses its own timestamp granularity. For most telephone-grade voice encodings, for example, the timestamp clock increments at the canonical sampling rate of 8,000 times a second,
corresponding to one DS0 channel (4.2 Time-Division Multiplexing). RFC 3551 suggests a timestamp
clock rate of 90,000 times a second for most video encodings.
Many VoIP applications that use RTP send 20 ms of voice per packet, meaning that the timestamp is incremented by 160 for each packet. The actual amount of data needed to send 20 ms of voice can vary from 160
bytes down to 20 bytes, depending on the encoding used, but the timestamp clock always increments at the
8,000/sec, 160/packet rate.

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The SSRC identifier identifies the primary data source (the synchronization source) of the stream. In
most cases this is either the identifier for the originating camera/microphone, or the identifier for the mixer
that repackaged the stream.
If the stream was processed by a mixer, the SSRC field identifies the mixer, and the SSRC identifiers of the
original sources are now listed in the CSRC (contributing sources) section of the header. If there was no
mixer involved, the CSRC section is empty.

19.11.3 RTP Control Protocol

The RTP Control Protocol, or RTCP, provides a mechanism for exchange of a variety of extra information
between RTP participants. RTCP packets may be Sender Reports (SR packets) or Receiver Reports (RR
packets); we are mostly interested in the latter.
RTCP receiver reports are the only form of acknowledgments for RTP transmission. These are, however,
unlike any other form of acknowledgment we have considered; for one thing, there is generally no question
of retransmitting any lost data. RTCP RR packets should perhaps be thought of as statistical summaries
regarding delivery rather than acknowledgments per se; in some cases they may in effect simply relay to
senders the information multicast is not working today.
RTCP packets contain a list of several SSRCs; for each SSRC, the message includes
the fraction of RTP packets that were lost in the most recent interval
the cumulative number of packets lost since the session began
the highest sequence number received
a measure of the interarrival jitter (below)
for RR packets, information about the most recent RTCP SR packet
Jitter is measured as the mean deviation in actual arrival times, versus theoretical arrival times, and is measured in units of the RTP timestamp clock (above). The actual formula is as follows. For each packet Pi ,
we can record the actual time of arrival in RTP timestamp units as Ri and we can save the RTP timestamp
included in the packet as Si . We first calculate the ith deviation as follows:
Devi = (Ri - Ri-1 ) - (Si - Si-1 )
For evenly spaced packets, the deviation will be zero. For VoIP packets, Si - Si-1 is likely exactly 160; Ri Ri-1 is the actual arrival interval in eighths of milliseconds.
We then calculate the cumulative jitter as follows, where = 15/16:
Ji = Ji-1 + (1-)Devi
In a multicast setting, if an RTP receiver is experiencing excessive losses, its most practical option is probably to find a mixer offering a lower-bandwidth encoding, or that is otherwise more likely to provide lowcongestion service. RTCP packets are not needed for this.
In a unicast setting, an RTP sender can at least theoretically use the information provided by RTCP RRs
to adapt its transmission rate downwards, to better make use of the available bandwidth. A sender might
also use TFRC (14.6.1 TFRC) to calculate a TCP-friendly sending rate; there are Internet drafts for this

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([GP11]), but as yet no RFC. TFRC also allows for a gradual adjustment in rate to accommodate the new
conditions.
Of course, TFRC can only be used to adjust the sending rate if the sending rate is in fact adaptive, at the
application level. This, of course, requires a rate-adaptive encoding.
RFC 3550 specifies a mechanism to make sure that RTCP packets do not consume more than 5% of the total
RTP bandwidth, and, of that 5%, RTCP RR packets do not account for more than 75%. This is achieved by
each receiver learning from RTCP SR packets how many other receivers there are, and what the total RTP
bandwidth is. From this, and from some related information, each RTP receiver can calculate an acceptable
RTCP reporting interval. This interval can easily be in the hundreds of seconds.
Mixers are allowed to consolidate RTCP information from their direct subscribers, and forward it on to the
originating sources.

19.11.4 RTP and VoIP

Voice-over-IP data is often carried by RTP, although VoIP calls are initially set up using the Session Initiation
Protocol, SIP (RFC 3261), or some other setup protocol such as H.323. As part of the call set-up process,
the SIP nodes organizing the call exchange IP addresses and port numbers for the actual endpoints. Each
endpoint then is informed of the address and port for the other endpoint, and the endpoints more-or-less
immediately begin sending one another RTP packets. If an endpoint is behind a NAT firewall, its associated
SIP server may have to continue forwarding packets.
VoIP calls typically send voice data in packets containing 20ms of audio input. If the standard DS0 G.711
encoding is used, 8 bytes are needed for each millisecond of voice and so each packet has 160ms of data.
The G.711 encoding includes both -law and A-law encoders; these are both variations on the idea of having
the 8-bit samples be proportional to the logarithm of the actual sound intensity.
Other voice encoders provide a greater degree of compression. For example, when G.729 voice encoding is
used for VoIP then each packet will carry 20 ms of voice in 20 bytes of data. Each such packet will also,
typically, have 54 bytes of header (14 bytes of Ethernet header, 20 bytes of IP header, 8 bytes of UDP header
and 12 bytes of RTP header). Despite a payload efficiency of only 20/74 27%, this is generally considered
acceptable, if not ideal.
VoIP connections often do not make much use of the data in the RTCP packets; it is not uncommon, in
fact, for RTCP packets not even to be sent. The general presumption is that if the voice quality is not
adequate, the users involved should simply hang up. The most common voice encoders (eg G.711 and, for
that matter, G.729) are not rate-adaptive, and so if a call is encountering congestion there is nothing that
can be done. That said, RTCP packets may provide useful information regarding jitter experienced by the
call. Furthermore, as we mentioned above in 19.3 Real-time Traffic, some voice codecs such as Opus and
Speex do support rate-adaptiveness.

19.12 Multi-Protocol Label Switching (MPLS)

Currently, MPLS is a way of tagging certain classes of traffic, and perhaps routing them altogether differently
from other classes of traffic. While IPv4 has always allowed routing based on tagged QoS levels, this feature
was seldom used, and a common assumption for both Integrated and Differentiated Services was that the

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priority traffic essentially took the same route as everything else. MPLS allows priority traffic to take an
entirely different route.
MPLS started out as a way of routing IP and non-IP traffic together, and later became a way to avoid the
then-inefficient lookup of an IP address at every router. It has, however, now evolved into a way to support
the following:
creation of explicit routes (virtual circuits) for IP traffic, either in bulk or by designated class.
large-scale virtual private networks (VPNs)
We are mostly interested in MPLS for the first of these; as such, it provides an alternative way for ISPs to
handle real-time traffic.
In effect, virtual-circuit tags are added to each packet, on entrance to the routing domain, allowing packets
to be routed along predefined virtual circuits. The virtual circuits need not follow the same route that normal
IP routing would use, though note that link-state routing protocols such as OSPF already allow different
routes for different classes of service.
We note also that the MPLS-labeled traffic might very well use the same internal routers as bulk traffic;
priority or fair queuing would then still be needed at those routers to make sure the MPLS traffic gets the
desired level of service. However, the use of MPLS at least makes the classification problem easier: internal
routers need only look at the tag to determine the priority or fair-queuing class, and deep-packet inspection
can be avoided. MPLS would also allow the option that a high-priority flow would travel on a special path
through its own set of routers that do not also service low-priority traffic.
Generally MPLS is used only within one routing domain or administrative system; that is, within the scope
of one ISP. Traffic enters and leaves looking like ordinary IP traffic, and the use of MPLS internally is
completely invisible. This local scope of MPLS, however, has meant that it has seen relatively widespread
adoption, at least compared to RSVP and IP multicast: no coordination with other ISPs is necessary.
To implement MPLS, we start with a set of participating routers, called label-switching routers or LSRs.
(The LSRs can comprise an entire ISP, or just a subset.) Edge routers partition (or classify) traffic into
large flow classes; one distinct flow (which might, for example, correspond to all VoIP traffic) is called a
forwarding equivalence class or FEC. Different FECs can have different quality-of-service targets. Bulk
traffic not receiving special MPLS treatment is not considered to be part of any FEC.
A one-way virtual circuit is then created for each FEC. An MPLS header is prepended to each IP packet,
using for the VCI a label value related to the FEC. The MPLS label is a 32-bit field, but only the first 20
bits are part of the VCI itself. The last 12 bits may carry supplemental connection information, for example
ATM virtual-channel identifiers and virtual-path identifiers (3.8 Asynchronous Transfer Mode: ATM).
It is likely that some traffic (perhaps even a majority) does not get put into any FEC; such traffic is then
delivered via the normal IP-routing mechanism.
MPLS-aware routers then add to their forwarding tables an MPLS table that consists of labelin , interfacein ,
labelout , interfaceout quadruples, just as in any virtual-circuit routing. A packet arriving on interface
interfacein with label labelin is forwarded on interface interfaceout after the label is altered to labelout .
Routers can also build their MPLS tables incrementally, although if this is done then the MPLS-routed traffic
will follow the same path as the IP-routed traffic. For example, downstream router R1 might connect to two
customers 200.0.1/24 and 200.0.2/24. R1 might assign these customers labels 37 and 38 respectively.

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R1 might then tell its upstream neighbors (eg R2 above) that any arriving traffic for either of these customers
should be labeled with the corresponding label. R2 now becomes the ingress router for the MPLS domain
consisting of R1 and R2.
R2 can push this further upstream (eg to R3) by selecting its own labels, eg 17 and 18, and asking R3 to
label 200.0.1/24 traffic with label 17 and 200.0.2/24 with 18. R2 would then rewrite VCIs 17 and 18 with 37
and 38, respectively, before forwarding on to R1, as usual in virtual-circuit routing. R2 might not be able to
continue with labels 37 and 38 because it might already be using those for inbound traffic from somewhere
else. At this point R2 would have an MPLS virtual-circuit forwarding table like the following:
interfacein
R3
R3

labelin
17
18

interfaceout
R1
R1

labelout
37
38

One advantage here of MPLS is that labels live in a flat address space and thus are easy and simple to look
up, eg with a big array of 65,000 entries for 16-bit labels.
MPLS can be adapted to multicast, in which case there might be two or more labelout , interfaceout combinations for a single input.
Sometimes, packets that already have one MPLS label might have a second (or more) label pushed on the
front, as the packet enters some designated subdomain of the original routing domain.
When MPLS is used throughout a domain, ingress routers attach the initial label; egress routers strip it
off. A label information base, or LIB, is maintained at each node to hold any necessary packet-handling
information (eg queue priority). The LIB is indexed by the labels, and thus involves a simpler lookup than
examination of the IP header itself.
MPLS has a natural fit with Differentiated Services (19.7 Differentiated Services): the ingress routers could
assign the DS class and then attach an MPLS label; the interior routers would then need to examine only the
MPLS label. Priority traffic could be routed along different paths from bulk traffic.
MPLS also allows an ISP to create multiple, mutually isolated VPNs; all that is needed to ensure isolation is
that there are no virtual circuits crossing over from one VPN to another. If the ISP has multi-site customers
A and B, then virtual circuits are created connecting each pair of As sites and each pair of Bs sites. A and
B each probably have at least one gateway to the whole Internet, but A and B can communicate with each
other only through those gateways.

19.13 Epilog
Quality-of-service guarantees for real-time and other classes of traffic have been an area of active research
on the Internet for over 20 years, but have not yet come into the mainstream. The tools, however, are there.
Protocols requiring global ISP coordination such as RSVP and IP Multicast may come slowly if at all,

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but other protocols such as Differentiated Services and MPLS can be effective when rolled out within a
single ISP.
Still, after twenty years it is not unreasonable to ask whether integrated networks are in fact the correct
approach. One school of thought says that real-time applications (such as VoIP) are only just beginning to
come into the mainstream, and integrated networks are sure to follow, or else that video streaming will take
over the niche once intended for real-time traffic. Perhaps IntServ was just ahead of its time. But the other
perspective is that the marketplace has had plenty of opportunity to make up its mind and has answered with
a resounding no, and it is time to move on. Perhaps having separate networks for bulk traffic and for voice
is not unreasonable or inefficient after all. Or perhaps the Internet will evolve into a network where all traffic
is handled by real-time mechanisms. Time, as usual, may tell, but not, perhaps, quickly.

19.14 Exercises
1. Suppose someone proposes TCP over multicast, in which each router collects the ACKs returning from
the group members reached through it, and consolidates them into a single ACK. This now means that, like
the multicast traffic itself, no ACK is duplicated on any single link.
What problems do you foresee with this proposal? (Hint: who will send retransmissions? How long will
packets need to be buffered for potential retransmission?)
2. In the following network, suppose traffic from RP to R3-R5 is always routed first right and then down,
while traffic from R3R5 to RP is always routed first left and then up. What is the multicast tree for the
group G = {B1,B2,B3}?
RP

R1

R2

R3

R4

R5

B1

B2

B3

3. What should an RSVP router do if it sees a PATH packet but there is no subsequent RESV packet?
4. In 19.7.1 Expedited Forwarding there is an example of an EF router with committed rate R for packets
with length L. If R and L are such that L/R is 10s, but the physical bandwidth delay in sending is only 2s,
then the packet can be held up to 8s for other traffic.
How large a bulk-traffic packet can this router send, in between EF packets? Your answer will involve L.
5. Suppose, in the diagram in 19.7.1 Expedited Forwarding, EF was used for voice telephony and at some
point calls entering through R1, R2 and R3 were indeed all directed to R4.

(a). How might the problem be perceived by users?

(b). How might the ISP respond to reduce the problem?

Note that the ISP has no control over who calls whom.

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20 NETWORK MANAGEMENT AND SNMP

Network management, broadly construed, consists of all the administrative actions taken to keep a network
running efficiently. This may include a number of non-technical considerations, eg staffing the help desk
and negotiating contracts with vendors, but we will restrict attention exclusively to the technical aspects of
network management.
The ISO and the International Telecommunications Union have defined a formal model for telecommunications and network management. The original model defined five areas of concern, and was sometimes
known as FCAPS after the first letter of each area:
fault
configuration
accounting
performance
security
Most non-ISP organizations have little interest in network accounting (the A in FCAPS is often replaced
with administration for that reason, but that is a rather vague category). Network security is arguably its
own subject entirely. As for the others, we can identify some important subcategories:
fault:
device management: monitoring of all switches, routers, servers and other network hardware to make
sure they are running properly.
server management: monitoring of the networks application layer, that is, all network-based software services; these include login authentication, email, web servers, business applications and file
servers.
link management: monitoring of long-haul links to ensure they are working.
configuration:
network architecture: the overall design, including topology, switching vs routing and subnet layout.
configuration management: arranging for the consistent configuration of large numbers of network
devices.
change management: how does a site roll out new changes, from new IP addresses to software
updates and patches?
performance:
traffic management: using the techniques of 18 Queuing and Scheduling to allocate bandwidth
shares (and perhaps bucket sizes) among varying internal or external clients or customers.
service-level management: making sure that agreed-upon service targets eg bandwidth are met
(depending on the focus, this could also be placed in the fault category).

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While all these aspects are important, technical network management narrowly construed often devolves to
an emphasis on fault management and its companion, reliability management: making sure nothing goes
wrong, and, when it does, fixing it promptly. It is through fault management that some network providers
achieve the elusive availability goal of 99.999% uptime.
SNMP versus Management
While SNMP is a very important tool for network management, it is just a tool. Network management
is the process of making decisions to achieve the goals outlined above, subject to resource constraints.
SNMP simply provides some input for those decisions.
By far the most common device-monitoring protocol, and the primary focus for this chapter, is the Simple
Network Management Protocol or SNMP (20.2 SNMP Basics). This protocol allows a device to report
information about its current operational state; for example, a switch or router may report the configuration
of each interface and the total numbers of bytes and packets sent via each interface.
Implicit in any device-monitoring strategy is initial device discovery: the process by which the monitor
learns of new devices. The ping protocol (7.9 Internet Control Message Protocol) is common here, though
there are other options as well; for example, it is possible to probe the UDP port on a node used for SNMP
usually 161. As was the case with router configuration (9 Routing-Update Algorithms), manual entry is
simply not a realistic alternative.
SNMP and the Application Layer
SNMP can be studied entirely from a network-management perspective, but it also makes an excellent
self-contained case study of the application layer. Like essentially all applications, SNMP defines
rules for client and server roles and for the format of requests and responses. SNMP also contains its
own authentication mechanisms (20.11 SNMPv1 communities and security and 20.15 SNMPv3),
generally unrelated to any operating-system-based login authentication.
It is a practical necessity, for networks of even modest size, to automate the job of checking whether everything is working properly. Waiting for complaints is not an option. Such a monitoring system is known as a
network management system or NMS; there are a wide range of both proprietary and open-source NMSs
available. At its most basic, an NMS consists of a library of scripts to discover new network devices and
then to poll each device (possibly but not necessarily using SNMP) at regular intervals. Generally the data
received is recorded and analyzed, and alarms are sounded if a failure is detected.
When SNMP was first established, there was a common belief that it would soon be replaced by the OSIs
Common Management Interface Protocol. CMIP is defined in the International Telecommunication Unions
X.711 protocol and companion protocols. CMIP uses the same ASN.1 syntax as SNMP, but has a richer
operations set. It remains the network management protocol of choice for OSI networks, and was once upon
a time believed to be destined for adoption in the TCP/IP world as well.
But it was not to be. TCP/IP-based network-equipment vendors never supported CMIP widely, and so any
network manager had to support SNMP as well. With SNMP support essentially universal, there was never
a need for a site to bother with CMIP.
CMIP, unlike SNMP, is supported over TCP, using the CMIP Over Tcp, or CMOT, protocol. One advantage
of using TCP is the elimination of uncertainty as to whether a request or a reply was received.
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20.1 Network Architecture

Before turning to SNMP in depth, we offer a few references to other parts of this book relating to network architecture. At the LAN and Internetwork layers local to a site, perhaps the main issues are redundancy, bandwidth and cost. Cabling between buildings, in particular, needs to provide redundancy. See
2.5 Spanning Tree Algorithm and 2.6 Virtual LAN (VLAN) for some considerations at the Ethernet level,
and 7.6.3 Subnets versus Switching.
Next, a site must determine what sort of connection to the Internet it will have. ISP contracts vary greatly
in terms of bandwidth, burst bandwidth, and agreed-upon responses in the event of an outage. Some aspects
of service-level specification appear in 18.9 Token Bucket Filters and 19.7.2 Assured Forwarding.
Organizations with geographically dispersed internal networks ISPs and larger corporations must decide
how their internal sites should be connected. Should they communicate over the public Internet? Should a
VPN be used (3.1 Virtual Private Network)? Or should private lines (such as SONET, 4.2.2 SONET, or
carrier Ethernet, 3.2 Carrier Ethernet) be leased between sites? If private lines are used, link monitoring
becomes essential.
One increasingly important architectural decision at the application layer is the extent to which network
software services are outsourced to the cloud, and run on remote servers managed by third parties.

20.2 SNMP Basics

SNMP is far and away the most popular protocol for supporting network device monitoring. At its most
basic level, SNMP allows polling of individual designated device attributes, such as the system name or the
number of packets received via interface eth0. Attributes may, however, be organized into records, sets
and tables. Tables may be indexed contiguously, like an array eg interface[1], interface[2], interface[3],
etc or sparsely eg interface[1], interface[32767].
While the simplest routers and switches may have quite limited provisions for SNMP, virtually all serious
networking hardware provides extensive support, for both standard sets of basic device attributes and for
proprietary attributes as well. Devices with significant SNMP support are sometimes referred to as managed
devices.
An SNMP node that replies to requests for information is known as an SNMP agent. The network node
doing the SNMP querying is known as the manager; it may be part of an NMS or more simply be a
standalone tool known as an SNMP browser or MIB browser (where MIB stands for Management Information Base, below). While most MIB browsers are understood to have a graphical user interface, there
are also command-line tools to make SNMP queries, such as the snmpget and snmpwalk commands of
the Net-SNMP project at net-snmp.org. These can be invoked by scripting languages to build a simple if
rudimentary NMS.
SNMP runs exclusively over UDP. The choice of UDP was made to avoid the connection overhead of what
was envisioned to be a simple request-reply protocol; if a manager polls 1,000 devices once a minute, that
is 2,000 packets in all over UDP but might easily be 8,000 packets with TCP and the necessary SYN/FIN
packets. This may be especially significant when the network is congested to near the point of failure.
The use of UDP does raise two problems: lost packets and having more data than will fit in one packet. For
the simplest case of manager-initiated data requests, a manager can handle packet loss by polling a device

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until a response is received. If a response (or even a request) is too big, the usual strategy is to use IP-layer
fragmentation (7.4 Fragmentation and 8.4.4 Fragment Header). This is not ideal, but as most SNMP data
stays within local and organizational networks it is at least workable.
Another consequence of the use of UDP is that every SNMP device needs an IP address. For devices acting
at the IP layer and above, this is not an issue, but an Ethernet switch or hub has no need of an IP address for
its basic function. Devices like switches and hubs that are to use SNMP must be provided with a suitable IP
interface and address. Sometimes, for security, these IP addresses for SNMP management are placed on a
private, more-or-less hidden subnet.
SNMP also supports the writing of attributes to devices, thus implementing a remote-configuration mechanism. As writing to devices has rather more security implications than reading from them, this mechanism
must be used with care. Even for read-only SNMP, however, security is an important issue; see 20.11 SNMPv1 communities and security and 20.15 SNMPv3.
Writing to agents may be done either to configure the network behavior of the device eg to bring an
interface up or down, or to set an IP address or specifically to configure the SNMP behavior of the agent.
Finally, SNMP supports asynchronous notification through its traps mechanism: a device can be configured
to report an error immediately rather than wait until asked. While traps are quite important at many sites,
we will largely ignore them here.
SNMP is sometimes used for server monitoring as well as device monitoring; alternatively, simple scripts
can initiate connections to services and discern at least something about their readiness. It is one thing,
however, to verify that an email server is listening on TCP port 25 and responds with the correct RFC 5321
(originally RFC 821) EHLO message; it is another to verify that messages are accepted and then actually
delivered.

20.2.1 SNMP versions

SNMP has three official versions, SNMPv1, SNMPv2 and SNMPv3. SNMPv1 made its first appearance
in 1988 in a collection of RFCs starting with RFC 1065 (updated in RFC 1155); the current definition for
core attribute reporting was released as RFC 1213 in 1991. We will return to this below in 20.10 MIB-2.
SNMPv2 was introduced in 1993 with RFC 1441. Loosely speaking, SNMPv2 expanded on the basic
information, starting with RFC 1442 (currently RFC 2578), and also introduced improved techniques for
managing tables.
SNMPv2 also included a proposed security mechanism, but it was largely rejected by the marketplace.
Ultimately, a version of SNMPv2 that used the SNMPv1 community security mechanism (20.11 SNMPv1
communities and security) was introduced; see RFC 1901. This community-security version became
known as SNMPv2c.
SNMPv3 then finally delivered a model for reasonably effective security. The User-based Security Model
or USM was first proposed in 1998 in RFC 2264; the final 2002 version is in RFC 3414.

20.3 SNMP Naming and OIDs

A central part of the SNMP protocol is how to name each device attribute in a consistent manner. The
naming scheme chosen was the Object Identifier, or OID, hierarchy.
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Starting in 1985, the International Standards Organization (ISO) and the standardization sector of the International Telecommunications Union (ITU-T, then known as CCITT) developed the Object Identifier scheme
for naming anything in the world that needed naming. Names, or OIDs, consist of strings of non-negative integers. In textual representation these component integers are often written separated by periods, eg 1.3.6.1;
the notation { 1 3 6 1 } is also used. The scheme is standardized in ITU-T X.660.
All OIDs used by SNMP begin with the prefix 1.3.6.1. For example, the sysName attribute corresponds to
OID 1.3.6.1.2.1.1.5. The prefix 1.3.6.1.2.1 is known as mib-2, and the one-step-longer prefix 1.3.6.1.2.1.1
is system. Occasionally we will abuse notation and act as if names referred to single additional levels
rather than full prefixes, eg the latter two names in mib-2.system.sysName.
The basic SNMP read operation is Get() (sent via the GetRequest protocol message), which takes an
OID as parameter. The agent receiving the Get request will, if authentication checks out and if the OID
corresponds to a valid attribute, return a pair consisting of the OID and the attribute value. We will return to
this in 20.8 SNMP Operations, and see how multiple attribute values can be requested in a single operation.
OIDs form a tree or hierarchy; the immediate child nodes of, say, 1.3.6.1 of length 4 are all nodes 1.3.6.1.N
of length 5. The root node, with this understanding, is anonymous; OIDs are sometimes rendered with a
leading . to emphasize this: .1.3.6.1. Alternatively, the numbers can be thought of as labels on the arcs of
the tree rather than the nodes.
There is no intrinsic way to distinguish internal OID nodes prefixes from leaf OID nodes that correspond
to actual named objects. Context is essential here.
It is common to give the numeric labels at any specific level human-readable string equivalents. The three
nodes immediately below the root are, with their standard string equivalents
itu-t(0)
iso(1)
joint-iso-itu-t(2)
The string equivalents can be thought of as external data; when OIDs are encoded only the numeric values
are included.
OID naming has been adopted in several non-SNMP contexts as well. ISO and ITU-T both use OIDs
to identify official standards documents. The X.509 standard uses OID naming in encryption certificates;
see RFC 5280. The X.500 directory standard also uses OIDs; the related RFC 4524 defines several OID
classes prefixed by 0.9.2342.19200300.100.1. As a non-computing example, Health Level Seven names US
healthcare information beginning with the OID prefix 2.16.840.1.113883.
As mentioned above, SNMP uses OIDs beginning with 1.3.6.1; in the OID tree these levels correspond to
iso(1)
org(3): organizations
dod(6): the US Department of Defense, the original sponsor of the Internet
internet(1)
This is spelled out in RFC 1155 via the syntax
internet

OBJECT IDENTIFIER ::= { iso org(3) dod(6) 1 }

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which means that the string internet has the type OBJECT IDENTIFIER and its actual value is the list
to the right of ::=: 1.3.6.1. The use of iso here represents the OID prefix .1, conceptually different from
just the level iso(1). The formal syntax here is ASN.1, as defined by the ITU-T standard X.680. We will
expand on this below in 20.6 ASN.1 Syntax and SNMP, though the presentation will mostly be informal;
further details can be found at http://www.itu.int/en/ITU-T/asn1/Pages/introduction.aspx.
After the above, RFC 1155 then defines the OID prefixes for management information, mgmt, and for
private-vendor use, private, as
mgmt
private

OBJECT IDENTIFIER ::= { internet 2 }

OBJECT IDENTIFIER ::= { internet 4 }

Again, internet represents the newly defined prefix 1.3.6.1. Most general-purpose SNMP OIDs begin
with the mgmt prefix. The private prefix is for OIDs representing vendor-specific information; for
example, the prefix 1.3.6.1.4.1.9 has been delegated to Cisco Systems, Inc. RFC 1155 states that both the
mgmt and private prefixes are delegated by the IAB to the IANA.
Responsibility for assigning names that begin with a given OID prefix can easily be delegated. The Internet
Activities Board, for example, is in charge of 1.3.6.1. (According to RFC 1155, this delegation by the
Department of Defense to the IAB was never made officially; the IAB just began using it.)

20.4 MIBs
A MIB, or Management Information Base, is a set of definitions that associate OIDs with specific attributes, and, along the way, associates each numeric level of the OIDs with a symbolic name. For example,
below is the set of so-called System definitions in the core RFC 1213 MIB known as MIB-2 (20.10 MIB2). The mib-2 and system prefixes are first defined as
mib-2
system

OBJECT IDENTIFIER ::= { mgmt 1 }

OBJECT IDENTIFIER ::= { mib-2 1 }

that is, 1.3.6.1.2.1 and 1.3.6.1.2.1.1 respectively. The system definitions, which represent actual attributes
that can be retrieved, are as follows
sysDescr
sysObjectID
sysUpTime
sysContact
sysName
sysLocation
sysServices

{
{
{
{
{
{
{

system
system
system
system
system
system
system

1
2
3
4
5
6
7

}
}
}
}
}
}
}

Most of these attributes represent string values that need to be administratively defined; we will return to
these in 20.10 MIB-2.
The MIB is not the actual attributes themselves; the set of all OID,value pairs stored by an SNMP manager
perhaps the result of a single set of queries, perhaps the result of a sequence of queries over time is
sometimes known as the management database, or MDB, though that term is not as universal.
Colloquially, a MIB can be either the abstract set of OID definitions or a particular ASCII file that defines the
set. The latter must be run through a MIB compiler to be used by other software; users of a MIB browser
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generally find the information much more useful if the appropriate MIB file is loaded before making SNMP
queries. Some RFCs can be read directly by the MIB compiler, which knows to edit out the non-MIB
discussion.
The Net-SNMP package contains SNMP agents for linux and Macintosh computers (Microsoft has their
own SNMP-agent software), and command-line tools for making SNMP queries; these latter tools are also
available for Windows. The sysName value, for example, can be retrieved with the snmpget command
snmpget -v1 -c public localhost 1.3.6.1.2.1.1.5.0

The OID here corresponds to { system 5 0 }; the final 0 represents a SNMP convention that indicates this is
a scalar value and is not part of a table. See 20.9 MIB Browsing for further examples.
MIBs serve as both documentation and data. As documentation, they tell the implementer of a given SNMP
agent what information is to be returned for each OID request. As data, they serve to translate numeric OIDs
into human-readable data. For example, if the above snmpget command understands the appropriate MIB
(eg because the MIB file is in the right place), we can type
snmpget -v1 -c public localhost sysName.0

It is not uncommon for an installation to involve several hundred different (possibly overlapping) MIBs, each
defining a different portion of the OID tree. This is particularly true of the private subtree 1.3.6.1.4.1,
for which one might have a separate MIB file for each manufacturer and device.

20.5 SNMPv1 Data Types

SNMP uses OIDs as the indexes for retrieval of attributes. The attributes themselves can have the following
types, as of RFC 1155:
Datatype
INTEGER
OCTET STRING
Counter
Gauge
TimeTicks
OBJECT IDENTIFIER
IpAddress
NetworkAddress
Opaque

Description
A 32-bit integer
Either a text string or a network address
A non-negative INTEGER that increases from 0, and can wrap around.
An INTEGER that can rise and fall but that never wraps around
An INTEGER used to measure time in 1/100ths of a second
An OID appearing as data
An IPv4 address, as an OCTET STRING of length 4
An IpAddress
Any other data, ultimately an OCTET STRING

The type of the sysName attribute, for example, is OCTET STRING.

When SNMP data values are returned by an agent, they are tagged with their type. Thus, the receiver can
See 20.12 SNMP and ASN.1 Encoding for further details.
When the INTEGER type is used to represent an enumerated type, the value of zero must not be used; an
example is the status type up(1), down(2), testing(3).

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20.6 ASN.1 Syntax and SNMP

We have already seen the ASN.1 definition of some OBJECT IDENTIFIERs in 20.3 SNMP Naming and
OIDs.
The attribute corresponding to any single OID is always a scalar type, eg one of the above. However, SNMP
also uses some composite type definitions:
SEQUENCE { fieldname1 type1, fieldname2 type2, ... }: This defines a record type.

SEQUENCE OF type: This defines a homogeneous list an array or table of objects of type
type. In most cases, type represents a SEQUENCE type, and so the list is a list of records, or
table in the database sense.

Note the importance of the keyword OF here to distinguish records from lists.
The OBJECT-TYPE macro is heavily used in MIB definitions. It has the format
objname OBJECT-TYPE
SYNTAX type
ACCESS read-write or read-only or write-only or not-accessible
STATUS mandatory or optional or obsolete
DESCRIPTION descriptive string
INDEX OID used for table indexing
::= OID assigned to objname
The objname, sometimes called the OBJECT DESCRIPTOR, is intended to be globally unique. The value
for SYNTAX must be a valid ASN.1 syntax specification; see 20.10 MIB-2 for examples. The values for ACCESS and STATUS are, in each case, one of the literal strings shown above. The value for
DESCRIPTION is optional.
Many of the objects so defined will represent tables or other higher-level structures; in this case the Object
ID of the last line will represent an internal node of the OID tree rather than a leaf node, and the ACCESS
will be not-accessible. It is not that the table is actually inaccessible, but rather that the attributes
must be retrieved one by one rather than collectively.
Here is a concrete simple example; other examples appear in the following section. The OID value of
ifEntry here is 1.3.6.1.2.1.2.2.1.
ifInOctets OBJECT-TYPE
SYNTAX Counter
ACCESS read-only
STATUS mandatory
DESCRIPTION
"The total number of octets received on the
interface, including framing characters."
::= { ifEntry 10 }

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The data type definitions and the above ASN.1 syntax used for are collectively known as the structure of
management information, or SMI. The SMI in effect defines all the types and structures needed by the
various MIBs. For SNMPv1, the SMI is defined in RFC 1155.

20.7 SNMP Tables

We have seen in 20.4 MIBs how SNMP organizes the OID names for the system group, which consists
of scalar values. Each agent attribute is assigned a permanent OID name; eg { system 5 0 } is the full OID
for the sysName attribute. OIDs for scalar (non-table) attributes always include a final .0.
Most agent data, however, comes in the form of tables, that is, lists of records. In SNMP, they are sometimes
referred to as conceptual tables, as the SNMP agent offering the table is often not responsible for its storage
or physical organization. In this sense, SNMP tables resemble database views.
These tables can be accessed by linear search or by primary key lookup; the key can be a single attribute
or a set of attributes and is known as the index. SNMP does not support secondary keys on tables, though
it would in principle be possible for an agent to present the same data in two tables, each with a different
index (a related example of this trick appears in 20.14.1.5 Matrix).
SNMP tables are almost always small enough that the agent keeps all the records in main memory; databasestyle concerns about disk-storage organization and indexing are not an issue.
As in databases, an SNMP table index acts as a constraint, disallowing two distinct rows with identical index
values. When an SNMP table mirrors a real table (eg the IP forwarding table), problems sometimes occur
when the SNMP index is too small has too few attributes and so the SNMP table may not be able to
accurately represent the original table. See 20.13.11.2 IP-Forward MIB for an example.
In the world of databases, a key with too many attributes may fail to capture an important constraint; it may
also be less efficient. Neither is a concern with SNMP, with the exception of tables that support row creation,
which we may take to be a special case. SNMP table indexes may thus sometimes include attributes that
from a database perspective would likely be omitted from the key; again, see 20.13.11.2 IP-Forward MIB
for examples.
We will look here at three examples of SNMPv1 tables, each part of MIB-2. The goal right now is to present
how the tabular structure is mapped onto the underlying OID-tree structure; we will return to the syntactic
definition of tables in 20.10.2 Table definitions and the interfaces Group and to the semantics of each table
in 20.10 MIB-2. SNMPv2 has made some modest improvements to how tables are defined.
The first example will be the SNMP interface table, ifTable. For each network interface (eg loopback,
Ethernet 0, Ethernet 1, Wireless 0, Point-to-Point 0, etc) a collection of attributes including the device name,
MTU, bitrate, physical address, and the number of bytes and packets received and sent. The index here (as
we shall see later) is an abstractly assigned interface number, known as ifIndex. Here is some sample
data, not showing all columns that the actual MIB-2 table includes, and not (yet) showing any OIDs. The
header of the index column, ifIndex, is in italic; the value is a single integer. The eth0 interface has a
higher number for packets than bytes (octets) because the 32-bit inOctets value has wrapped around.

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ifIndex
1
2
3
4

name
lo
eth0
eth1
ppp0

MTU
16436
1500
1500
1420

bitrate
10,000,000
100,000,000
100,000,000
0

inOctets
171
37155014
0
2906687015

inPackets
3
1833455677
0
2821825

Routes
Although the data here is mostly a mash-up of records from different sources, the 10.38.0.0 route below
was a VPN route using the ppp0 tunnel above. Otherwise, workstation forwarding tables usually have
just two entries, for the local subnet and the default route.
The SNMP routing table, ipRouteTable, based on the IP forwarding table of 1.10 IP - Internet
Protocol, has a column representing the destination network, a variety of status and information columns
(eg for RouteAge), and a NextHop column; the index is the destination-network attribute. Again, here is
some sample data with the index column again in italic. The index here an IP address is a compound
object: a list of four integers (yes, we can represent an IPv4 address as a single 32-bit integer, but usually do
not).
dest
0.0.0.0
10.0.0.0
10.38.0.0

mask
0.0.0.0
255.255.255.0
255.255.0.0

metric
1
0
1

next_hop
192.168.1.1
0.0.0.0
192.168.4.1

type
indirect(4)
direct(3)
indirect(4)

(The type column indicates whether the destination is local or nonlocal; the host is on subnet 10.0.0.0/24
and so the middle entry involves local delivery. The use of indirect(4) and direct(3) is an example
of an SNMP enumerated type.)
The indexing of the interfaces table is (usually) dense: the values for ifIndex are typically consecutive
numbers. The routing table, on the other hand, has a sparse index; the index values are likely nowhere near
one another.
The TCP Connections table tcpConnTable lists every TCP connection together with its connection
state as in 12.6 TCP state diagram; one can obtain this on most linux, windows or macintosh systems
with the command netstat -a. The index in this case consists of the four attributes of the connectiondefining socketpair: the local address, the local port, the remote address and the remote port. In this table,
the only attribute not part of the index is an integer representing the connection state (again represented by
an SNMP enumerated type).
localAddr
10.0.0.3
10.0.0.3
10.0.0.3

localPort
31895
40113
20459

remoteAddr
147.126.1.209
74.125.225.98
10.38.2.42

remotePort
993
80
22

state
established(5)
timeWait(11)
established(5)

SNMP has adopted conventions for how tabular data such as the above is to be encoded in the OID tree.
The basic strategy is first to define, statically, an OID prefix for the subtree representing the table; in this
section we will denote this by T (later, in 20.10.2 Table definitions and the interfaces Group, we will see
that T often represents two OID levels of the form Table.Entry, or T.E). Each attribute of the table (that is,
each column) is then assigned a non-leaf OID by appending successive integers to the table prefix, starting
at 1. For example, the root prefix for the interfaces table is T = 1.3.6.1.2.1.2.2.1, known as ifEntry. The
columns shown in the fraction of the interfaces table above have the following OIDs:
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OID
{ ifEntry
{ ifEntry
{ ifEntry
{ ifEntry
{ ifEntry
{ ifEntry

1 } or T.1
2 } or T.2
4 } or T.4
5 } or T.5
10 } or T.10
11 } or T.11

table column
The interface number, ifIndex
The interface name or description
The interface MTU
The interface bitrate
The number of inbound octets (bytes)
The number of inbound unicast packets

The second step is to establish a convention for converting the index attributes to a list of integers that can
be used as an OID suffix. This index OID suffix is then appended to the attribute OID prefix to obtain the
full OID (with no trailing .0 this time) that represents the name of the table data value. The prefix specifies
the table and the column, and the suffix specifies the row.
For the case of the interfaces table, the interface number is used as a single-level suffix. The eth1 value
for inOctets thus has OID name T.10.3: 10 is the number assigned to the inOctets column and 3 is
the ifIndex value for the eth1 row. Written in full, this is 1.3.6.1.2.1.2.2.1.10.3. Note this numbering
has the form T.col.row, the transposition of the more common programming-language row-first convention
T[row][col].
For the routing table, a destination IP address, eg 10.38.0.0, is converted to a four-level OID suffix, eg
10.38.0.0. (Note that 10.38.0.0 has exactly the same textual representation as an IPv4 address and as a fourlevel OID suffix, but logically they are entirely different things). An IP address might have been encoded
as a single OID level using the address as a single 32-bit integer, but this option was not chosen. In the
table above, the nextHop column is assigned the number 7, so the nextHop for 10.38.0.0 thus has the OID
T.7.10.38.0.0.
For the TCP connections table, the two IP addresses involved each translate to four-level OID chunks, as in
the routing table, and the two port numbers each translate to one-level chunks; the resultant OID suffix for
the state of the first connection in the table above would be
10.0.0.3.31895.147.126.1.209.993
where, for clarity, the port-number levels are shown in italic. The state column is assigned the identifier 1,
so this would all be appended to T.1 (it is common, but, as we see here, not universal, for column-number
assignment to begin with the index columns).
Here is a column-oriented diagram of the interfaces-table fragment from above. As we have been doing, the
table prefix is denoted T and nodes are labeled T.col.row. The topmost T and the first row (with the T.col
OIDs) are internal nodes; these are drawn with the heavier, blue boxes. The lower four rows, with black
boxes, are leaf nodes.

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The diagram above emphasizes the arrangement into columns. The actual OID tree structure, however, is as
in the diagram below; all the leaf nodes in one column above are actually sibling nodes.

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In all three tables here, the index columns are part of the table data. This is unnecessary; all the index
columns are of necessity encoded in the full OID of any table entry. In SNMPv2 it is required (though not
necessarily enforced) to in effect omit the index columns from the table as presented by the agent (though
they are still declared); see 20.13.4 SNMPv2 Indexes.
We also note that, in all three tables here (and in most SNMP tables that serve purely as sources of information), new rows cannot be added via SNMP. Some SNMP tables, however, do support row creation; see
20.14 Table Row Creation.

20.8 SNMP Operations

As mentioned earlier, the fundamental read operation is Get(); the protocol message itself is known as
GetRequest, and also contains the authentication information (that is, the community string for SNMPv1
and SNMPv2c). A request-id is included so multiple outstanding GetRequests arent mixed up.
The GetRequest message can contain a list of OIDs (actually encoded as a list of OID,null pairs called
a VarBind list). A single-OID get simply entails a VarBind list of length 1.
The agent receiving the GetRequest validates the authentication and then, if successful, looks up each
OID in the received list to find the corresponding value. The filled-in VarBind list of all OID,value pairs
is returned.
If any value cannot be found, eg because the OID represented an interior node of the OID tree instead of an
actual value, or because the OID lay outside the portion of the OID tree that the request was authorized to
access, then in SNMPv1 no values are returned. Only if every OID in the list corresponds to a valid value is
a list of all the OID,value pairs returned. SNMPv2 later relaxed this rule to allow the return of whatever
individual requests were successful.
If the exact OID is not known which is often the case for tabular data the GetNext() operation can
be used. It works like Get(), except that for each OID oid in the request list the agent finds the next leaf
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OID strictly following oid in lexicographical order, which we will call oid . The agent then includes the pair
oid ,value in its response, where value is the value corresponding to oid . Note that in the GetNext()
case oid is not previously known to the manager. Note also that SNMP always returns, with each value, the
corresponding OID, which for tabular data encodes the full index for the value.
The GetNext() operation allows a manager to walk through any subtree of the OID tree. If the root
of the subtree is prefix T, then the first call is to GetNext(T). This returns oid1 ,value1 . The next call
to GetNext has parameter oid1 ; the agent returns oid2 ,value2 . The manager continues with the series of
GetNext calls until, finally, the subtree is exhausted and the agent returns either an error or else (more likely)
an oidN ,valueN for which oidN is no longer an extension of the original prefix p.
As an example, let us start with the prefix 1.3.6.1.2.1.1, the start of the system group. The first call to
GetNext() will return the pair
1.3.6.1.2.1.1.1.0, sysDescr_value
The OID 1.3.6.1.2.1.1.1.0 is the first leaf node below the interior node 1.3.6.1.2.1.1.
The second call to GetNext will take OID 1.3.6.1.2.1.1.1.0 as parameter and GetNext() will return
1.3.6.1.2.1.1.2.0, sysObjectID_value
Here, the OID returned is the next leaf node strictly following 1.3.6.1.2.1.1.1.0. The process will continue on through {system 3 0}, {system 4 0}, etc, until an OID is found that is not part of the system
group. As we have seen, this will likely be the first entry of the interfaces group, ifNumber.0, with OID
1.3.6.1.2.1.2.1.0.
The action of GetNext() is particularly useful when retrieving a table, such as the interfaces table presented in the preceding section. In that example the index values are consecutive integers, and so an SNMP
manager could likely guess them. However, guessing the index values for the other two tables the IP
forwarding table and the TCP connections table would be well nigh impossible. And without the index
values, we do not have a complete OID.
Consider again the partial-column version of the interfaces table as diagrammed in this format:

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Let us initially call GetNext(T). The next leaf node is the upper left black box, with OID T.1.1. The call to
GetNext(T) returns the pair T.1.1,1. We now continue with a call to GetNext(T.1.1); this returns T.1.2,2
and represents the black box immediately below the first one. The next two calls to GetNext() return,
successively, T.1.3,3 and T.1.4,4.
Now we call GetNext(T.1.4). The next leaf node following is the first leaf node in the second column, T.2.1;
the value returned is T.2.1,lo. The next three calls to GetNext, each time supplied with parameter the
value returned by the previous GetNext(), return the next three values making up the second, ifDesc
column of the table.
At the bottom of the second column, the call to GetNext(T.2.4) returns the OID,value pair that is the first
leaf node of the third column, T.3.1,16436. At the bottom of the third column, GetNext() jumps to
the top of the fourth column, and so on. In this manner GetNext() iterates through the entire table, one
column at a time.
When we finally get to the last leaf node of the table, shown here as the lower-right T.11.4 (though the actual
ifTable has additional columns). The call to GetNext(T.11.4) returns something outside the table. It will
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either return an error in the event that there is no next OID that the manager is authorized to receive or
the next leaf node up and to the right of T. (In the normal MIB-2 collection, this is would be the first entry
of the atTable table.)
Either way, the manager receiving the data can tell that its request for GetNext(T.11.4) has gone past the
end of table T, and so T has been completely traversed.
Here is a diagram of the above sequence of GetNext() operations involved in traversing our partial
interfaces table; it shows the GetNext(T.11.4) at the tables lower right returning the beyond-the-table
OID,value pair A,avalue:

Some manager utilities performing this kind of table retrieval often called a walk will present the data
in the order retrieved, one column at a time, and some will (perhaps as an option) format the data visually
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to look more like a table. See Net-SNMPs snmpwalk and snmptable in 20.9 MIB Browsing.

20.8.1 Multi-attribute Get()

A single Get()/GetNext() request can in fact include a list of attributes to be retrieved. This provides
an efficient way to request entire rows of a table.
A manager in general does not know all the index values of a table, but likely does know all the column
OIDs (as in the blue full-length row above). The manager can speed up the table-retrieval process by asking
for entire rows at a time. The first such GetNext() might be
GetNext(T.1, T.2, T.3, T.4, T.10, T.11)
These are the OIDs of the blue full-length row; these are all non-leaf OIDs. The result is the list of pairs
T.1.1,1, T.2.1,lo, T.3.1,16436, T.4.1,10000000, T.10.1,171, T.11.1,3
That is, the tables entire first row is returned (where we are assuming again that the only columns are 1, 2,
3, 4, 10 and 11). The next call to GetNext() would use the OIDs returned above:
GetNext(T.1.1, T.2.1, T.3.1, T.4.1, T.10.1, T.11.1)
This would then return the tables entire second row, and so on. When we got to the very last row, the call
GetNext(T.1.4, T.2.4, T.3.4, T.4.4, T.10.4, T.11.4)
would return (assuming now that the columns shown are the only columns that exist)
T.2.1,lo, T.3.1,16436, T.4.1,10000000, T.10.1,171, T.11.1,3, A,avalue
where A is the next leaf OID above and to the right of T (assuming no error). At this point the manager
knows, as in the single-attribute-get case, that the entire table has been retrieved.
It does not matter whether the manager uses this multi-attribute form of GetNext() to return all the
columns of the relevant table, or only the subset of columns in which the manager is interested.
We will see in 20.13.3 SNMPv2 GetBulk() that SNMPv2 introduced an operation GetBulk() that can
return multiple rows of multiple columns in a single operation.
Tables are not necessarily static. New dynamic interfaces may be added, and new routes and TCP connections are added all the time. A manager reading a table using the single-attribute form of GetNext() may
find that only the latter part of a new row is retrieved, or, alternatively, that the first part of a deleted row is
retrieved. The multi-attribute GetNext() offers a little more protection from this.

20.8.2 Set()
SNMP also allows a Set() operation. Not all attributes are writable, of course, and a manager must have
write authority for those that are writable. And there is no SetNext(); the exact OID of each value must
be supplied.
In the system group, the sysName attribute is writable. It has OID 1.3.6.1.2.1.1.5.0; to update this we
would invoke
Set(1.3.6.1.2.1.1.5.0, newsysname)

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Like Get(), Set() can be invoked on multiple attributes. Suppose table T has three columns T.1, T.2
and T.3, and two rows:
T.1
10
11

T.2
eth0
eth1

T.3
3.14159
2.71828

We can then change the second eth1 row to 11,ppp1,0.577216 with

Set( (T.2.11,ppp1), (T.3.11,0.577216) )
We cannot change the indexing this way, but see 20.14 Table Row Creation.
The semantics for multi-attribute Set() explicitly require and still require that either all the assignments
succeed, or all fail; that is, multi-attribute Set() operations are atomic. (The usual reason for a failure is
that the manager issuing the Set() lacked write permission for some item.) SNMPv1 had the same all-ornothing rule for multi-attribute Get(), but that requirement was relaxed in SNMPv2. The all-or-nothing
Set() semantics mean that certain updates cannot end up completed only half-way.
The multi-attribute Set() semantics do not mean, however, that near-simultaneous assignments by different managers are serialized. Suppose, for example, two separate managers both attempt to update attributes
specified by OIDs X and Y, one to 10 and 100 and the other to 20 and 200. If the updates are scheduled at
about the same time,
Set((X,10), (Y,100))
Set((X,20), (Y,200))
then it is possible for the Set()s to be performed in the order (X,20), (X,10), (Y,100), (Y,200), leaving X =
10 and Y = 200. See 20.13.5 TestAndIncr for a workaround.

20.9 MIB Browsing

Tools for individual SNMP reading are widely available, and are very helpful for viewing what is going on.
We have already mentioned the Net-SNMP package. The command snmpget issues a single (perhaps
multi-attribute) Get() request; the very similar snmpgetnext issues a GetNext request. The command
snmpwalk takes an OID representing the root of a subtree, and returns everything in that subtree. Here is
an example that assumes the snmp agent on localhost is configured to use community string tengwar (see
20.11 SNMPv1 communities and security):
snmpwalk -v 1 -c tengwar localhost 1.3.6.1.2.1.2.2

The OID here is that of ifTable. Multiple OIDs are permitted for snmpget and snmpgetnext but not
for snmpwalk.
If the appropriate MIB files are loaded, the above command can also be entered as
snmpwalk -v 1 -c tengwar localhost ifTable

This entails putting the RFC1213-MIB file into a special directory (eg $HOME/.snmp/mibs), and then either
adding a line like

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mibs +RFC1213-MIB

to the snmp.conf file, or by including the MIB name on the command line:
snmpwalk -v 1 -c tengwar -m RFC1213-MIB localhost ifTable

Note that RFC1213-MIB is the identifier assigned to the MIB file in its first line, as below, and not necessarily the name of the file containing the MIB.
RFC1213-MIB DEFINITIONS ::= BEGIN

If the appropriate MIB file is loaded, the OIDs may be displayed symbolically rather than numerically, but
the data presentation will not change:
iso.3.6.1.2.1.1.5.0 = STRING: "valhal"
RFC1213-MIB::sysName.0 = STRING: "valhal"

;; without MIB
;; with MIB

(Actually, if the data is of type Object ID, then without the MIB the OID will be displayed numerically, and
with the appropriate MIB all or part of it may be displayed symbolically.)
Finally, the Net-SNMP manager package includes snmptable, which is like snmpwalk except that the
data is displayed as a table rather than one column at a time. For the snmptable command, the appropriate
MIB file must be installed. Net-SNMP comes with a mib browser with a graphical interface, tkmib.
The Java SNMP open-source package at http://jsevy.com/snmp/ includes both a Java library to encode and
decode SNMP packets and the SNMP Inquisitor browser.
The personal edition of the iReasoning MIB browser is not open-source, but it is free, and works well on
windows, macs and linux; SNMPv3 is not supported. The license does prohibit the publication of benchmark
tests without consent.

20.10 MIB-2
We can now turn to the MIB that represents the core of SNMPv1 data, known as MIB-2, and defined in RFC
1213. The 2 here often represented with a Roman numeral: MIB-II represents the second iteration of
the definition, but it is still part of SNMPv1. The predecessor MIB-1 was first documented in RFC 1066,
1988.
As we saw in 20.4 MIBs, the MIB-2 OID prefix is 1.3.6.1.2.1.
MIB-2 has since been extended. We look at a few extensions below in 20.13.7 SNMPv2 MIB Changes,
20.13.9 IF-MIB and ifXTable, 20.13.12 TCP-MIB and 20.13.11.2 IP-Forward MIB. In general, serious
network management should make use of these newer versions. However, MIB-2 is still an excellent place
to get started, even if partly obsolete.
The original MIB-2 is divided into ten groups, not all of which are in current use:
system(1): above
interfaces(2): above, in brief
at(3): the ARP cache, 7.7 Address Resolution Protocol: ARP

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ip(4): including the IP forwarding table used as an example above

icmp(5): information about ICMP, 7.9 Internet Control Message Protocol
tcp(6): including the TCP connections table used as an example above
udp(7): information about UDP activity
egp(8): information about the now-obsolete EGP protocol, replaced by BGP
transmission(10): never used
snmp(11): about the SNMP agent itself
Originally group 9 was for CMOT, above, but this entry is commented out in RFC 1213 (though it does not
appear at all in MIB-1).
The at group is officially deprecated; the same ARP-cache information is available within the ip group.
The EGP protocol has been entirely replaced by BGP, and management of BGP routers is highly specialized;
RFC 4273 contains a BGP MIB but private MIBs are almost universally used here as well.
The transmission group was included anticipating the day when definitions for managed transmission media
are defined. This day has not come to pass.
The Net-SNMP agent implementation returns nothing for the egp and transmission groups.

20.10.1 The system Group

We have already looked at this in 20.4 MIBs and will not say much more, except to note that several
of the entries in this group must be manually configured. This can be done either through editing of the
configuration file, eg snmpd.conf, or else (in the case of read-write attributes) by using SNMP itself.
The sysObjectID value represents a vendor-specific OID for this particular systems SNMP agent. For
example, the Net-SNMP package installed on my system returns 1.3.6.1.4.1.8072.3.2.10. The 1.3.6.1.4.1 is
the root of the private, vendor-specific, OID tree, and 8072 has been assigned to the Net-SNMP project.
This often represents, in practice, the way an SNMP manager can figure out the vendor of an agent, and
thus determine what vendor-specific information to query. One can use the 8072 discovered here to ask
for the subtree 1.3.6.1.4.1.8072, and find all sorts of Net-SNMP-specific information. Similarly, on one
particular Windows XP installation with Microsoft Network Monitoring installed, the sysObjectID value is
1.3.6.1.4.1.311.1.1.3.1.1; 311 is the OID level assigned to Microsoft in the private subtree 1.3.6.1.4.1.
The system group has been expanded in SNMPv2 with an Object Resource table, sysORTable; see
20.13.7 SNMPv2 MIB Changes.

20.10.2 Table definitions and the interfaces Group

The interfaces group consists of a single INTEGER value ifNumber representing the number of current
network interfaces (including down interfaces), and the interfaces table partially introduced earlier. We
take the opportunity here to outline exactly how MIB table definitions and enumerated types are structured using ASN.1.

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Some of the interfaces-group definitions have later been updated. See, for example, RFC 2863, which also
redefines the group to use the additional features of the SNMPv2 SMI. We will return to an extension of the
interfaces group in 20.13.9 IF-MIB and ifXTable.
MIB table definitions almost always involve a two-level process: an OID is defined for the table, and then
a second OID is defined for a table entry, that is, for one row of the table. This second OID is usually
generated from the first by appending .1, and it is this second OID that represents the table prefix in the
sense of 20.7 SNMP Tables, denoted there by T.
The actual ASN.1, slightly annotated, is as follows:
ifTable OBJECT-TYPE
SYNTAX SEQUENCE OF IfEntry
-- note UPPER-CASE-I IfEntry
ACCESS not-accessible
STATUS mandatory
DESCRIPTION
"A list of interface entries. The number of
entries is given by the value of ifNumber."
::= { interfaces 2 }
-- that is, 1.3.6.1.2.1.2.2
ifEntry OBJECT-TYPE
-- note lower-case-i ifEntry
SYNTAX IfEntry
-- note UPPER-CASE-I IfEntry
ACCESS not-accessible
STATUS mandatory
DESCRIPTION
"An interface entry containing objects at the
subnetwork layer and below for a particular interface."
INDEX
{ ifIndex }
::= { ifTable 1 }
-- that is, 1.3.6.1.2.1.2.2.1

Both these entries are not-accessible as they do not represent leaf nodes. The second declaration
above is for the lower-case-i ifEntry; the next definition in RFC 1213 is for the UPPER-CASE-I version.
The latter represents the complete list of all columns of an ifEntry/IfEntry object, together with their types
from (in most cases) 20.5 SNMPv1 Data Types. The PhysAddress type is defined in RFC 1213 as a
synonym for OCTET STRING. Definitions for the OID associated with each column comes later.
It is ifEntry that represents an actual row, and thus includes an INDEX entry to specify the attribute or
attributes that make up the primary key for that row.
We now define IfEntry:
IfEntry ::=
SEQUENCE {
ifIndex
ifDescr
ifType
ifMtu
ifSpeed
ifPhysAddress
ifAdminStatus
ifOperStatus
ifLastChange
ifInOctets
ifInUcastPkts

20.10 MIB-2

INTEGER,
DisplayString,
INTEGER,
INTEGER,
Gauge,
PhysAddress,
INTEGER,
INTEGER,
TimeTicks,
Counter,
Counter,

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ifInNUcastPkts
ifInDiscards
ifInErrors
ifInUnknownProtos
ifOutOctets
ifOutUcastPkts
ifOutNUcastPkts
ifOutDiscards
ifOutErrors
ifOutQLen
ifSpecific

Counter,
Counter,
Counter,
Counter,
Counter,
Counter,
Counter,
Counter,
Counter,
Gauge,
OBJECT IDENTIFIER

Attributes
There can be a certain eye-glazing tedium in some of SNMPs lengthy attribute lists, such as the one
above. This can be replaced by panic in a hurry, though, when a problem has arisen and the proper
attribute to diagnose it doesnt seem to be included anywhere.
It would have been possible to plug in the above definition of IfEntry into the SYNTAX specification of the
previous ifEntry, but that is cumbersome. The IfEntry definition is not a stand-alone OBJECT-TYPE and
does not have its own OID.
If all we wanted to do was to implement the interfaces table using the shortest possible OIDs, we would
not have created separate ifTable and ifEntry OIDs. This would mean, however, that we could not use
the OBJECT-TYPE macro to define ifTable, which would have been less consistent (especially as the
ASN.1 syntax also determines the packet encoding, as in 20.12 SNMP and ASN.1 Encoding). Essentially
every table prefix in SNMP is defined using two additional OID levels, as here, rather than one.
We now turn to the 22 specific interface attributes. Here is the definition for the first, ifIndex; it defines
column 1 of the interfaces table to be, in effect, ifEntry.1. As we are now talking about a leaf node, once the
OID suffix is appended to represent the index, the ACCESS is no longer not-accessible.
ifIndex OBJECT-TYPE
SYNTAX INTEGER
ACCESS read-only
STATUS mandatory
DESCRIPTION
"A unique value for each interface. Its value ranges
between 1 and the value of ifNumber. The value for each
interface must remain constant at least from one
reinitialization of the entitys network management system
to the next reinitialization."
::= { ifEntry 1 }

The DESCRIPTION clearly indicates that the values for ifIndex, used to specify interfaces, are to be
consecutive integers. Compliance with this rule has been an early casualty, for various reasons, and is
formally withdrawn in RFC 2863. Some vendors simply number physical interfaces non-consecutively. In
other cases, there is some underlying issue with consecutive numbering. For example, one of the authors
systems running Net-SNMP returns ifNumber = 3, and then the following table values:

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ifIndex
1
2
549

ifName
lo
eth0
ppp0

It turns out that ppp0 is a virtual interface corresponding to a VPN tunnel, 3.1 Virtual Private Network,
and the underlying tunnel regularly fails and is then automatically re-instantiated. Each time it does so, the
ifIndex is incremented by 1.
The rule that interfaces be numbered consecutively was formally deprecated in RFC 2863, an SNMPv2
update of the interface group. This, in turn, makes the ifNumber value rather less useful than it might be;
most SNMP tables are not associated with a count attribute and seem to do just fine.
Most SNMP data values correspond straightforwardly with attributes defined by the hardware and the underlying operating system. The ifIndex value does not, at least not necessarily. Generally the agent must
maintain at least some state to keep the ifIndex value consistent. In some cases, the ifIndex value
may be taken from the relative position of the interface in some internal operating-system table. This is not,
however, universally the case, as with the ifIndex value of 549 for ppp0 in the table above.
The ifIndex value is widely used throughout SNMP, and is often referenced in other tables. SNMPv2 even
defines a special type (a TEXTUAL CONVENTION) for it, named InterfaceIndex (20.13.1 SNMPv2
SMI and Data Types).
As we mentioned earlier, there is no reason to include ifIndex as an actual column in the table; the value
of ifIndex can always be calculated from the OID of any component of the row. The SNMPv2 approach
here basically to define ifIndex as not-accessible is described below in 20.13.4 SNMPv2
Indexes.
The ifDescr is a textual description of the interface; it is usually the device name associated with the
interface. RFC 1213 states this string should include the name of the manufacturer, the product name and
the version of the hardware interface, but this is inconsistent with linux device-naming conventions. The
current rule is that it is merely unique.
The ifType attribute is our first example of how ASN.1 handles enumerated types. The value is a small
integer and the hardware type associated with each integer is spelled out as follows. The majority of the
networking technologies in this 1991 list have pretty much vanished from the face of the earth.
ifType OBJECT-TYPE
SYNTAX INTEGER {
other(1),
-- none of the following
regular1822(2),
hdh1822(3),
ddn-x25(4),
rfc877-x25(5),
ethernet-csmacd(6),
iso88023-csmacd(7),
iso88024-tokenBus(8),
iso88025-tokenRing(9),
iso88026-man(10),
starLan(11),
proteon-10Mbit(12),
proteon-80Mbit(13),
hyperchannel(14),

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fddi(15),
lapb(16),
sdlc(17),
ds1(18),
-- T-1
e1(19),
-- European equiv. of T-1
basicISDN(20),
primaryISDN(21),
-- proprietary serial
propPointToPointSerial(22),
ppp(23),
softwareLoopback(24),
eon(25),
-- CLNP over IP [11]
ethernet-3Mbit(26),
nsip(27),
-- XNS over IP
slip(28),
-- generic SLIP
ultra(29),
-- ULTRA technologies
ds3(30),
-- T-3
sip(31),
-- SMDS
frame-relay(32)
}
ACCESS read-only
STATUS mandatory
DESCRIPTION
"The type of interface, distinguished according to
the physical/link protocol(s) immediately below
the network layer in the protocol stack."
::= { ifEntry 3 }

A more serious problem is that over two hundred new technologies are unlisted here. To address this, the
values for ifType have been placed under control of the IANA, as defined in IANAifType; see RFC
2863 and https://www.iana.org/assignments/ianaiftype-mib/ianaiftype-mib. The IANA can then add new
types without formally updating any RFC.
For the meaning of ifMtu, the interface MTU, see 7.4 Fragmentation. The 32-bit ifSpeed value will
be unusable once speeds exceed 2 Gbps; RFC 2863 defines an ifHighSpeed object with speed measured
in units of Mbps. This entry is part of the ifXTable table, 20.13.9 IF-MIB and ifXTable. RFC 2863
also clarifies that, for virtual interfaces that do not really have a bandwidth, the value to be reported is zero
(though in the example earlier the loopback interface lo was reported to have a bandwidth of 10 Mbps).
The ifPhysAddress is, on Ethernets, the Ethernet address of the interface.
The ifAdminStatus and ifOperStatus attributes are enumerated types: up(1), down(2), testing(3).
If the ifAdminStatus is up(1), and the ifOperStatus disagrees, then there is a likely hardware
malfunction. The ifLastChange attribute reflects the last time, in TimeTicks, there was a change in
ifOperStatus.
The next eleven entries count bytes, packets and errors. The ifInOctets and ifOutOctets count
bytes received and sent (including framing bytes, 4.1.5 Framing, if applicable). The problem with these
is that they may wrap around too fast: in 34 seconds at 1 Gbps. The MIB-2 values are still used, but are
generally supplemented with 64-bit counters defined in ifXTable, 20.13.9 IF-MIB and ifXTable.
The packet counters are for unicast packets, non-unicast packets, discarded packets, errors, and, for inbound
packets only, packets with unknown protocols. The count of non-unicast packets is separated in ifXTable
into separate counts of broadcast and multicast packets.

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The interface queue length is available in ifOutQLen. It takes a considerable amount of traffic to make
this anything other than 0. RFC 1213 says nothing about the timescale for averaging the queue length.
The last member of the classic MIB-2 interfaces group is ifSpecific, which has type ObjectID. It was
to return an OID that may be queried for additional ifType-specific information about the interface. It was
formally deprecated in RFC 2233.

20.10.3 The ip Group

The ip group contains several scalar attributes and three tables.
The first two attributes are writable:
ipForwarding: a Boolean attribute to enable or disable forwarding
ipDefaultTTL: the default TTL in outgoing packets (7.1 The IPv4 Header)
The next scalar attributes are as follows. Here and later, we summarize these informally.
ipInReceives: the number of received IP packets
ipInHdrErrors: the number of apparent IP packets received with header errors
ipInAddrErrors: the number of IP packets received with bad destination addresses
ipForwDatagrams: the number of forwarded IP packets
ipInUnknownProtos: the number of received IP packets with unknown higher-level protocol
ipInDiscards: the number of received IP packets that had no errors, but which were still discarded
due to resource limits
ipInDelivers: the number of received IP packets delivered locally
ipOutRequests: the number of IP packets originated by this node for delivery elsewhere
ipOutDiscards: the number of outbound IP packets discarded due to resource limits
ipOutNoRoutes: the number of outbound IP packets for which there was no route in the IP forwarding table
ipReasmTimeout: the value, in seconds, of the IP fragment-reassembly timer
ipReasmReqds: the number of IP proper fragments received that needed reassembly at this node
(forwarded fragments are reassembled at their destination)
ipReasmOKs: the number of fragmented IP packets successfully reassembled
ipReasmFails: the number of fragmented IP packets not successfully reassembled
ipFragFails: primarily, packets that needed fragmentation but their Dont Fragment bit (7.4 Fragmentation) was set
ipFragCreates: the number of IP fragments created by this node
ipRoutingDiscards: the number of packets that should have been forwarded, but were discarded
due to resource limits

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The first table is the ipAddrTable, a list of all IP addresses assigned to this node together with interfaces
and netmasks.
The second table is the ipRouteTable, that is, the forwarding table. We looked briefly at this in
20.7 SNMP Tables. Note that the index consists solely of the destination IP address ipRouteDest,
which is not sufficient for CIDR-based forwarding in which the forwarding-table key is the dest,netmask
pair. Traffic to 10.38.0.0/16 might be routed differently than traffic to 10.38.0.0/24!
This table contains four attributes ipRouteMetric1 through ipRouteMetric4. RFC 1213 states
that unused metrics should be set to -1, but many agents simply omit such columns entirely. Agents may
similarly omit ipRouteAge.
The third table is ipNetToMediaTable, which is the ARP-cache table and which replaces a similar nowdeprecated table in the at group. The ipNetToMedia table adds one column, ipNetToMediaType, not
present in the at-group table; this column indicates whether an physical-to-IP-address mapping is invalid(2),
dynamic(3) or static(4). Most ARP entries are dynamic.
For an updated version of the ip group, see 20.13.11.1 IP-MIB and 20.13.11.2 IP-Forward MIB.

20.10.4 The icmp Group

The icmp group consists of 26 counters for various ICMP events. ICMP Echo Request (ping request) packets
have their own counter.

20.10.5 The tcp Group

The tcp group includes the following scalars:
tcpRtoAlgorithm: the method for calculating the retransmission timeout, normally vanj(4) for
the Jacobson-Karels algorithm in 12.17 TCP Timeout and Retransmission.
tcpRtoMin: the minimum TCP retransmission timeout, in milliseconds
tcpRtoMax: the maximum TCP retransmission timeout, again in ms
tcpMaxConn: if the system imposes a static cap on the number of allowed TCP connections, that is
returned. Most systems have no static cap, though, and return -1.
tcpActiveOpens: the number of TCP connections opened from this node; specifically, the number
of TCP state transitions CLOSED SYN_SENT
tcpPassiveOpens: literally, the number of TCP state transitions LISTEN SYN_RECD
tcpAttemptFails: the number of TCP connections that went to CLOSED without ever being
ESTABLISHED
tcpEstabResets: the number of TCP connections that went from ESTABLISHED or
CLOSE_WAIT directly to CLOSED, via RST packets
tcpCurrEstab: the number of TCP connections currently in either state ESTABLISHED or state
CLOSE_WAIT

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tcpInSegs: the count of received TCP packets, including errors and duplicates (though duplicate
reception is relatively rare)
tcpOutSegs: the count of sent TCP packets, not including retransmissions
tcpRetransSegs: the count of TCP packets with at least one retransmitted byte; the packet may
also contain new data
tcpInErrs: the number of TCP packets received with errors, including checksum errors
tcpOutRsts: the number of RST packets sent
Perhaps surprisingly, there is no counter provided for total number of TCP bytes sent or received. There is
also no entry for the congestion-management strategy, eg Reno v NewReno v TCP Cubic (15 Newer TCP
Implementations).
The tcp group also includes the tcpConnTable table, which lists, for each connection (identified by
localAddress,localPort,remoteAddress,remotePort) the connection state. We looked at this earlier in
20.7 SNMP Tables. The table is noteworthy in that four of its five columns are part of the INDEX.
A consequence is that to extract all the information from the table, a manager need only retrieve the
tcpConnState column: the other four attributes will all be encoded in the OID index that is returned
with each tcpConnState value.
We will look at the newer SNMPv2 replacement in 20.13.12 TCP-MIB.

20.10.6 The udp Group

The udp group contains the following scalars:
udpInDatagrams: a count of the number of UDP packets received and deliverable to a socket
udpNoPorts: a count of UDP packets received but undeliverable because the port was not open
udpInErrors: a count of the number of UDP packets undeliverable due to any other errors
udpOutDatagrams: a count of the number of UDP packets sent
The udp group also contains the table udpTable. This table lists those UDP ports that the node has open.
The table rows have the form localAddr,localPort, with both columns included in the index. If a UDP
socket accepts connections from anywhere, the localAddr will appear as 0.0.0.0.

20.10.7 The snmp Group

The SNMP group consists of 28 SNMP status and error attributes, numbered from 1 to 30 with 7 and 23 not
used.

20.11 SNMPv1 communities and security

An SNMPv1 manager authenticates itself to an agent by providing a string known as a community string
that is a combination of both userid and password. That is, the single string identifies the manager (or
manager group) and at the same time authenticates the manager.
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The community string identifies a manager as a member of a designated community, in the conventional
use of the word, of managers. Of course, at many sites there will be only one manager that interacts with
any given agent.
Community strings are sent unencrypted, and so are vulnerable to sniffing. They should be obscure, and
changed frequently. In actual practice, far and away the most popular value the default for many agents
is the string public.
If a manager sends an incorrect community string, then in SNMPv1 and SNMPv2c there is no reply. SNMPv3 relaxed this rule somewhat, 20.15.3 SNMPv3 Engines.
A single agent can support multiple community strings. Each community has an associated subset of the
MIB (a view) that it allows. For example, an agent can be configured so that the community string system
allows the manager access to the system group, the community string tengwar allows access to the entire
MIB-2 group, and the community string galadriel allows access to the preceding plus the private
group(s). Using Net-SNMP (the most common agent on linux and Macintosh machines) this would be
achieved by the following entries in the snmpd.conf file:
rocommunity
rocommunity
rocommunity
view
view

system
default
tengwar
default
galadriel default

mib2+private included
mib2+private included

1.3.6.1.2.1.1
1.3.6.1.2.1
-V mib2+private
1.3.6.1.2.1
1.3.6.1.4.1

In order that the galadriel community could contain two disconnected OID subtrees, it was necessary to
make use of the View-based Access Control Model (VACM). Community galadriel has access to the OIDtree view named mib2+private; this view is in turn defined in the last two lines above.
For our purposes here, VACM can be seen as an implementation mechanism for specifying what portion of
the OID tree is accessible to a given community. VACM allows read and write permissions to be granted
to specific OID trees, and also to be excluded from specific subtrees (eg table columns). Using the mask
mechanism, access can even be granted to specific rows of a table. We return to VACM for SNMPv3 very
briefly in 20.15.9 VACM for SNMPv3.
On Microsoft operating systems, an SNMP agent is generally included but must be enabled, eg from Windows components or Programs and Features. After that, the agent must still be configured. This is done
by accessing Services (eg through Control Panel Administrative Tools or by launching services.msc),
selecting SNMP service, and clicking on Properties. This applet only allows setting the community
name and read vs write permissions; specifying collections of visible OID subtrees (views) is not supported
here (though it may be via SNMP itself).
The community mechanism can offer a reasonable degree of security, if community names are changed
frequently and if eavesdropping is not a concern. Perhaps the real problem with community-based security
is that just how much information can leak out if an attacker knows the community string is not always well
understood. SNMP access can reveal a sites complete network and host structure, including VPNs, subnets,
TCP connections, host-to-host trust relationships, and much more. Most sites block the SNMP port 161 at
their firewall; some even go so far as to run SNMP only on a hidden network largely invisible even within
the organization.
The view model for OID-tree access is formalized in RFC 3415 as part of SNMPv3; it is called the Viewbased Access Control Model or VACM. It allows the creation of named views. The vacmAccessTable

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spells out the viewing rights assigned to a given VACM group, which, in a rough sense, corresponds to an
SNMPv1/SNMPv2c community. VACM supports, in addition to views consisting of disjoint unions of OID
subtrees, table views that limit access to a specific set of columns.

20.12 SNMP and ASN.1 Encoding

When SNMP data is entered into a packet, it is encoded according to the ASN.1 Basic Encoding Rules, or
BER. This is a hierarchical syntax-driven binary encoding strategy in which each atomic value (INTEGER,
OCTET STRING, OBJECT IDENTIFIER, etc) is tagged with its type, and each compound structure is
also tagged. This means that the receiver can understand a complex packet format without prior knowledge
of its structure. The BER rules are part of the ITU standard X.690. (ASN.1 also supports several other
encoding-rule formats, including the human-readable XML Encoding Rules, but SNMP uses BER.)
The general representation of an object is a type-length-value triple of the following form; the type tag
makes the data self-identifying.

The first two bits of the type field, the class bits, identify the context. Universal types such as INTEGER
and OBJECT IDENTIFIER have class bits 00; application-specific types such as Counter32 and TimeTicks
have class bits 01.
The third bit is 0 for primitive types and 1 for constructed types such as STRUCTURE.
The rest of the first byte is the type tag. If a second byte is needed, the tag bits of the first byte are 11111.

20.12.1 Primitive Types

The standard ASN.1 universal-type primitive tag values used by SNMP are as follows:
00000
00010
00100
00110

NULL
INTEGER
OCTET STRING
OBJECT IDENTIFIER

SNMP also defines several application-specific tags (the following are from RFC 2578):
00000
00001
00010
00011
00100
00110

IpAddress
Counter32
Gauge32
TimeTicks
Opaque
Counter64

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The second field of the type-length-value structure is the length of the value portion, in bytes. Most lengths
of primitive types will fit into a single byte. If a data item is longer than 127 bytes (true for many composite
types), the multi-byte integer encoding below is used.
The actual data is then encoded into the value field. For nonnegative integers, the integer is converted
to a twos-complement bitstring and then encoded in as few bytes as possible, provided there is at least
one leading 0-bit to represent the sign. Similarly, negative numbers must have at least one leading 1-bit to
represent the sign.
For example, 127 can be encoded as a length=1 INTEGER with value byte 0111 1111; 128 must be
encoded as a length=2 integer with value bytes 0000 0000 and 1000 0000; the INTEGER represented by a
length of 1 and aa value byte of 1000 0000 is 128. Similarly, to encode decimal 10,000,000 (0x989680),
four value bytes are needed as otherwise the leading bit would be 1 and the number would be interpreted
as negative.
For OCTET STRINGs, the string bytes are placed in the value portion and the number of bytes is placed
in the length portion.
Object IDs are generally encoded using one byte per level; there are two exceptions. First, the initial two
levels x.y of the OID are encoded using a single level with value 40x + y; all SNMP OIDs begin with 1.3
and so the first byte is 43 (0x2b). Second, if a level is greater than 127, it is encoded as multiple bytes. The
first bit of the last byte is 0; the first bit of each of the preceding bytes is 1. The seven remaining bits of each
byte contain the bits of the OID level. For example, 1.3.6.1.2.1.128.9 would have a value encoding of the
following bytes in hexadecimal:
2b 06 01 02 01 81 00 09
where 128 is represented as the two seven-bit blocks 0000001 0000000 and the first block is prefixed by 1
and the second block by 0.

20.12.2 Composite Types

Now that we have the encodings of the primitive types we can build structures. For composite types the P/C
bit will be set to 1. The only universal composite type we will consider is
10000

SEQUENCE and SEQUENCE OF

There are also several application-specific composite types; the first three bits of the tag field for these will
be 101.
00000
00001
00010
00011
00100

Get-Request
Get-Next-Request
Get-Response
Set-Request
Trap

A VarBind pair is encoded in SNMPv1 as

SEQUENCE { name OBJECT IDENTIFIER, value ObjectSyntax }

ObjectSyntax is specified as a CHOICE that can contain any of the tagged SNMP primitive types
above; the CHOICE syntax adds no bytes so the value is simply encoded as above. The VarBind pair

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1.3.6.1.2.1.1.3.0, 8650000 the OID is sysUpTime and the value is 24 hours would then be represented
in hexadecimal bytes as below; the hexadecimal representation of 8650000 is 0x83d600.

The first byte of 0x30 marks this as a SEQUENCE; recall that the type byte for a SEQUENCE has a P/C
bit of 1 and five low-order bits of 10000, for a numeric value of 48 decimal or 0x30.
The VarBindList is defined to be a SEQUENCE OF VarBind. If we have only the single VarBind
above, the resultant enclosing VarBindList would be as follows; the length field is 0x12 = 18.
30 12 30 10 06 08 2b 06 01 02 01 01 03 00 02 04 00 83 d6 00
The BER encoding rules do not stop with the VarBindList. A slightly simplified ASN.1 specification
for an entire SNMPv1 Get-Request packet portion (or protocol data unit) is
SEQUENCE {
request-id
INTEGER,
error-status INTEGER,
error-index INTEGER,
variable-bindings VarBindList
}

The encoding of the whole packet would also be by the above BER rules.
The BER encoding mechanism represents a very different approach from the fixed-field layout of, say, the
IP and TCP headers (7.1 The IPv4 Header and 12.2 TCP Header). The latter approach is generally quite
a bit more compact, as only four bytes are needed for a larger integer versus six under BER, and no bytes
are used for SEQUENCE specifications. The biggest advantage of the SNMP BER approach, however, is
that all objects, from entire packets down to individual values, are self-describing. Given the variety of
types used by SNMP, the fact that many are of variable length, and the fact that value readers such as MIB
browsers often operate without having all the type-specifying MIBs loaded, this self-describing feature is
quite useful.

20.13 SNMPv2
SNMPv2 introduced multiple evolutionary changes: to the SMI, to various MIBs, and to the basic protocol
operation. Many new MIBs were added. SNMPv2 also contained a proposal for improved security, but this
was not widely adopted. Eventually most of the SNMP community settled on SNMPv2c, the version of
SNMPv2 that stayed with the community-based security model.
Most of the specification of SNMPv2c is in RFC 1901 through RFC 1909.
Generally SNMPv1 and SNMPv2c agents and managers can interoperate quite easily. Essentially all SNMPv2 agents also support SNMPv1 queries, and answer according to the version of the request received.
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There is slightly greater confusion between SNMPv1 and SNMPv2 MIB files, but almost all browsers and
managers support both.

20.13.1 SNMPv2 SMI and Data Types

SNMPv2 introduced the OBJECT-IDENTITY macro, which acts like the OBJECT-TYPE macro except
that it leaves out the SYNTAX clause; it serves as a way to define OIDs separately from syntax. In
the OBJECT-TYPE macro, the ACCESS attribute is renamed MAX-ACCESS, and a new access option
read-create (for reading plus row creation) is added.
SNMPv2 also provided new 64-bit versions of some of the basic types, and added some other types. These
definitions are in RFC 2578, originally RFC 1442. A very practical problem with SNMPv1 is that, for
example, the 32-bit inOctets counter can wrap around in 34 seconds at 1 Gbps.
The INTEGER type remains, now declared synonymous with Integer32. The SNMPv1 Counter and Gauge
types have been replaced with Counter32 and Gauge32, and 64-bit versions of both Counter64 and
Gauge64 were added (Gauge64 was added somewhat later, in RFC 2856).
A BITS type, for representing individual bits in a word, was also added.
The OBJECT IDENTIFIER type was formally limited to 128 levels.
SNMPv2 also introduced the TEXTUAL-CONVENTION macro, which is an alternative name for a type (or
a primary name for an enumerated type) together with a description of what the type is actually to represent.
Examples include OwnerString, which is an OCTET STRING intended to describe a management station
and perhaps its human operator, EntryStatus which is an enumerated type meant to be used for row
additions (20.14.1 RMON), and InterfaceIndex, which is meant to be the number of an interface
appearing in the MIB-2 ifTable, 20.10.2 Table definitions and the interfaces Group, but used in another
table. For casual MIB reading, textual conventions can simply be thought of as types.

20.13.2 SNMPv2 Get Semantics

When an SNMPv1 agent receives a list of OIDs as part of a Get() request, and one or more of them is unavailable, then an error is returned. Under SNMPv2, the agent returns a list containing the appropriate value
for each valid OID. For request OIDs for which the agent is not able to return an actual value, the special
value noSuchInstance (for missing table entries with legal column specifications) or noSuchObject
(for other missing values) is substituted.
The Set() semantics remain unchanged: an SNMPv2 agent does not update any of the OID values in the
request unless it is able to update all of them.

20.13.3 SNMPv2 GetBulk()

SNMPv2 introduced the GetBulk operation as an extension of GetNext. A manager includes an integer N
in its request and the agent then iterates the action of GetNext() N times. All N results (which can each
represent an entire row) can then be returned in a single operation.
For example, suppose a table T has five rows indexed by 11-15, and three columns with OIDs T.1 through
T.3. The OIDs for the table are as follows:
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T.1.11
T.1.12
T.1.13
T.1.14
T.1.15

T.2.11
T.2.12
T.2.13
T.2.14
T.2.15

T.3.11
T.3.12
T.3.13
T.3.14
T.3.15

Then a GetBulk request for (T.1,T.2,T.3) with a repetition count of 3 will return the first three rows, with
the following OIDs:
T.1.11 T.2.11 T.3.11
T.1.12 T.2.12 T.3.12
T.1.13 T.2.13 T.3.13
To continue, the next such request would include OIDs (T.1.13, T.2.13, T.3.13) and the result (again assuming a count of 3) would be of values with these OIDs:
T.1.14 T.2.14 T.3.14
T.1.15 T.2.15 T.3.15
T.2.11 T.3.11 A
where A is the next leaf OID above and to the right of T.
Note the third row of the second request: the first leaf OID following T.1.15 the last row of column 1 is
T.2.11, that is, the first row of column 2. Similarly, T.3.11 follows T.2.15. As T.3.15 is the last leaf OID in
the table, its leaf-OID successor (A) is outside the table.
The GetBulk request format actually partitions the list of requested OIDs into two parts: those OIDs that are
to be requested only once and those for which the request is repeated. Two additional parameters besides
the OID list are included: non-repeaters indicating the number of OIDs to be requested only once and
max-repetitions indicating the number of times the remaining OIDs are retrieved.
As with GetNext, it is possible that a request for rows of the table will return OID,value pairs outside the
table, that is, following the table in the OID-tree order.
If the total number of OIDs in the request is N non-repeaters, then the return packet will contain a
list of OID,value variable bindings of up to length
non-repeaters + (N non-repeaters)max-repetitions
GetBulk has considerable potential to return more data than there is room for, so an agent may return fewer
repetitions as it sees fit.

20.13.4 SNMPv2 Indexes

As we saw in 20.7 SNMP Tables, in SNMPv1 the index columns were included in the table. Because every
attribute returned by an agent comes paired with the attributes OID, and because the OID of a table item
encodes the row and thus the index value, index columns are unnecessary except in rare cases (one such case
is when all table columns are part of the index, as in udpTable).
SNMPv2 deals with this by requiring the usual INDEX clause in the ASN.1 tableEntry definition, but
then when the tableIndex attribute is defined with OBJECT-TYPE it is assigned a MAX-ACCESS of

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not-accessible. Here is an example from the sysORTable of 20.13.7

is the second not-accessible that represents the change.

SNMPv2 MIB Changes; it

sysOREntry OBJECT-TYPE
SYNTAX
SysOREntry
MAX-ACCESS not-accessible
STATUS
current
DESCRIPTION
"An entry (conceptual row) in the sysORTable."
INDEX
{ sysORIndex }
::= { sysORTable 1 }
...
sysORIndex OBJECT-TYPE
SYNTAX
INTEGER (1..2147483647)
MAX-ACCESS not-accessible
STATUS
current
DESCRIPTION ...
::= { sysOREntry 1 }

The INDEX attribute appears in the declaration of sysOREntry in the usual way. But it is now classed as
an auxiliary object, and access to sysORIndex is not-accessible; in SNMPv1 it would have been
read-only and thus an ordinary column.
When direct access to index attributes is suppressed this way, the data is still available in the accompanying
OID, but it is no longer tagged by type as in 20.12 SNMP and ASN.1 Encoding. The sysORIndex value
above, for example, would appear as a single OID level, but the manager would have to use the MIB to
determine that it was meant as an INTEGER and not a TimeTicks or Counter. An IPv4 address used as an
index would appear in the OID as four levels, readily recognizable as an IPv4 address provided the manager
knew where to look. When STRING values appear in the index, the lack of an index column can be a
particular nuisance; for example, the only indication of the username alice in the usmUserTable of
20.15.9.1 The usmUserTable might be the OID fragment 97.108.105.99.101, representing the ASCII codes
for the letters a.l.i.c.e.
Generally, an SNMPv2 agent will send back the noSuchObject special value (see 20.13.2 SNMPv2 Get
Semantics) when asked for a not-accessible auxiliary object.

20.13.5 TestAndIncr
SNMPv2 introduced the TestAndIncr textual convention, which introduces something of an aberration
to the usual semantics of Set(). The underlying type is INTEGER, in the range 0..231 1, and Get()
works the usual way. However, if TI is the OID name of a TestAndIncr object, then Set(TI,val) never
sets the value of TI to val. Instead, if the value of the object is already equal to val, then the Set() succeeds
and the value of the object is incremented, that is, set to val + 1. If the value of the object is not equal to val,
an error occurs and no change is made.
A TestAndIncr object acts like a kind of semaphore, though not exactly as there is no way to decrement
the object (though the value does wrap around from 231 -1 back to 0). The goal here is to provide a voluntary mechanism to enforce serialization when more than one manager may be writing to the same set of
attributes.
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As we saw at the example at the end of 20.8.2 Set(), such serialization is not automatic. But now let us revisit
that example using TestAndIncr. Recall that two managers are updating attributes with OIDs X and Y;
this time, however, they agree also to include a TestAndIncr object with OID TI. Then serialization is
assured as long as each manager executes each multi-attribute Set() in the following form, where val :=
Get(TI) means that the manager uses Get() to retrieve the value of TI and stores it in its own local variable
val.
val := Get(TI)
Set((TI,val), (X,xval), (Y,yval))
To see this, suppose manager Bs Get(TI) occurs after manager As Set(TI,val) has incremented val. Then
manager Bs Set() operations occur even later, after As Set() has completed. The alternative is that
both managers Get() the same value val. Let A be the manager who first succeeds with Set(TI,val). Now
the other manager B will have its Set(TI,val) fail, as the value stored at TI is now val+1. Thus all of Bs
Set() operations will fail.
A consequence here is that manager B will have to try again, probably immediately. However, the likelihood
of conflict is low, and B can expect to succeed soon.
Usually only one TestAndIncr object needs to be provided for an entire table, not one per row. For an
actual example, see 20.15.9.1 The usmUserTable.

20.13.6 Table Augmentation

SNMPv2 introduced a mechanism for extending an existing table by adding, in effect, more columns, known
as augmentation. In the new table-entry definition, instead of an INDEX clause there is an AUGMENTS
clause; the argument to which is a table-entry name for the existing table to be extended. The new tableentry row will be indexed by whatever attributes indexed the table-entry in the AUGMENTS clause.
For example, the ifXTable augments the MIB-2 ifTable, meaning in essence that the ifXTable is
automatically indexed by ifTables index, ifIndex. Here is the ifXEntry definition that establishes
that:
ifXEntry
OBJECT-TYPE
SYNTAX
IfXEntry
MAX-ACCESS not-accessible
STATUS
current
DESCRIPTION
"An entry containing additional management information
applicable to a particular interface."
AUGMENTS
{ ifEntry }
::= { ifXTable 1 }

The intent here is that every ifTable row is now extended to include the nineteen additional values defined
for IfXEntry; that is, there is a one-to-one correspondence between rows if ifTable and ifXTable.
If, on the other hand, the new table extends only a few rows of the original table, ie is a sparse extension,
then the new table entry should have an INDEX clause that repeats that of the original table. An example is the EtherLike-MIB of 20.13.10 ETHERLIKE-MIB, in which the dot3StatsTable extends the
MIB-2 ifTable by providing additional information for those interfaces that behave like Ethernets. The
dot3StatsEntry definition is

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dot3StatsEntry OBJECT-TYPE
SYNTAX
Dot3StatsEntry
MAX-ACCESS not-accessible
STATUS
current
DESCRIPTION "Statistics for a particular interface to an ethernet-like medium."
INDEX
{ dot3StatsIndex }
::= { dot3StatsTable 1 }

The INDEX is dots3StatsIndex, which is then defined as follows; note the statement in the DESCRIPTION (and the REFERENCE) that the dot3StatsIndex is to correspond to ifIndex.
dot3StatsIndex OBJECT-TYPE
SYNTAX
InterfaceIndex
MAX-ACCESS read-only
STATUS
current
DESCRIPTION "An index value that uniquely identifies an interface
to an ethernet-like medium. The interface identified
by a particular value of this index is the same interface
as identified by the same value of ifIndex."
REFERENCE
"RFC 2863, ifIndex"
::= { dot3StatsEntry 1 }

Finally, it is possible that a new table has a many-to-one, or dense, correspondence to the rows (or a subset
of the rows) of an existing table. In this case, the new table will have an INDEX clause that will include the
index attributes of the original table, and one or more additional attributes. An example is the EtherLikeMIB dot3CollTable, which keeps, for each interface, a set of histogram buckets giving, for each N, a
count of the number of packets that experienced exactly N collisions before successful transmission. The
dot3CollEntry definition is as follows:
dot3CollEntry OBJECT-TYPE
SYNTAX
Dot3CollEntry
MAX-ACCESS not-accessible
STATUS
current
DESCRIPTION ...
INDEX
{ ifIndex, dot3CollCount }
::= { dot3CollTable 1 }

The ifIndex entry in the INDEX represents the original table, as before except that here there is also a
second, new INDEX attribute, dot3CollCount.

20.13.7 SNMPv2 MIB Changes

The core MIB for SNMPv2 essentially the SNMPv2 version of MIB-2 is RFC 3418, originally RFC
1907.
Some changes appear under the MIB-2 OID prefix: 1.3.6.1.2.1. SNMPv2 also defines a prefix 1.3.6.1.6
specifically for SNMPv2 information.
This MIB adds the Object Resource table, sysORTable, to the system group. It also slightly modifies the
mib-2.snmp group and provides a new snmp group under the SNMPv2 1.3.6.1.6 prefix.

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20.13.8 sysORTable
The original system group contained the attribute sysObjectID that identifies the agent and at the same
time suggests a private OID tree that could provide additional information about the agent (20.10.1 The
system Group).
The sysORTable, 1.3.6.1.2.1.1.9 or mib-2.system.9, is an attempt to extend this. It consists of a list of
OIDs that can be queried for further agent information; each OID also has an associated description string
and a sysORUpTime value indicating the time that OID was added.
For example, my system lists the following (where snmpModules = 1.3.6.1.6.3 and mib-2 = 1.3.6.1.2.1):
snmpModules.11.3.1.1
snmpModules.15.2.1.1
snmpModules.10.3.1.1
snmpModules.1
mib-2.49
mib-2.4
mib-2.50
snmpModules.16.2.2.1
snmpModules.13.3.1.3
mib-2.92

Message Processing MIB

User-based security MIB
SNMP management architecture
SNMPv2 information
TCP
IP
UDP
VACM
SNMP notification
SNMP notification logging

Each of the above OID prefixes can theoretically then be accessed for further information. Unfortunately,
on my system several of them are not configured, and a query returns nothing, but sysORTable does not
know that.

20.13.9 IF-MIB and ifXTable

RFC 2863 has addressed the problems with ifSpeed and the byte counters by introducing a new table,
ifXTable (formerly ifExtnsTable with OID mib-2.31.1.2. As such, it is still under the aegis of the
MIB-2 OID, but outside the traditional interfaces group. It does use the SNMPv2 SMI.
The ifXTable table includes definitions for high capacity 64-bit counters for in and out octets, in and
out unicast packets, in and out multicast packets and in and out broadcast packets. It also includes the
ifHighSpeed attribute for interface speed, still a 32-bit quantity but now measured in units of 1 Mbps.
The full definition of an ifXTable row is as follows:
IfXEntry ::=
SEQUENCE {
ifName
ifInMulticastPkts
ifInBroadcastPkts
ifOutMulticastPkts
ifOutBroadcastPkts
ifHCInOctets
ifHCInUcastPkts
ifHCInMulticastPkts
ifHCInBroadcastPkts

20.13 SNMPv2

DisplayString,
Counter32,
Counter32,
Counter32,
Counter32,
Counter64,
Counter64,
Counter64,
Counter64,

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ifHCOutOctets
Counter64,
ifHCOutUcastPkts
Counter64,
ifHCOutMulticastPkts
Counter64,
ifHCOutBroadcastPkts
Counter64,
ifLinkUpDownTrapEnable INTEGER,
ifHighSpeed
Gauge32,
ifPromiscuousMode
TruthValue,
ifConnectorPresent
TruthValue,
ifAlias
DisplayString,
ifCounterDiscontinuityTime TimeStamp
}

The original MIB-2 interfaces group counted multicast and broadcast packets together, eg in
ifInNUcastPkts.

20.13.10 ETHERLIKE-MIB
RFC 3635 (originally RFC 1650) defines a MIB for Ethernet-like interfaces. The primary goal is to
enable the collection of statistics on collisions and other Ethernet-specific behaviors. Several new tables are
defined.
The table dot3StatsTable contains additional per-interface attributes; the name refers to the IEEE
designation for Ethernet of 802.3. The table represents a sparse extension of the original ifTable, in the
sense of 20.13.6 Table Augmentation (where this table was used as the example).
The rows of the table mostly consist of counters for various errors and other noteworthy conditions:
Alignment errors: the number of bits in the frame is not divisible by 8
CRC checksum failure
Frames that experienced exactly one collision
Frames that experienced more than one collision
Signal quality errors. SQE is specific to 10 Mbps Ethernet
Deferred transmissions; when the station tried to send, the line was not idle
Late collisions: the only way a collision can occur after the slot time is passed is if the physical
Ethernet is too big or if collision-detection is failing. See 2.1.3 The Slot Time and Collisions
Excessive collisions: the frame experienced 16 collisions and the sender gave up
Other
hardware
errors
(dot3StatsInternalMacTransmitErrors
dot3StatsInternalMacReceiveErrors)

and

Carrier sense errors (carrier sense refers to the collision-detection mechanism; there is no actual
carrier)
Frames longer than 1500 octets
For Ethernets that encode data as symbols (eg 100 Mbps Ethernets 4B/5B), frames arriving with a
corrupted symbol

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There is also an attribute to describe the Ethernet chipset.

The dot3HCStatsTable is like the dot3StatsTable except its counters are 64 bits instead of 32
bits.
The table dot3CollTable lists, for each N, how many packets experienced exactly N collisions before
successful transmission or discard. Ethernet generally allows a maximum N of 16. The table is indexed by
the pair ifIndex,N. The dot3StatsTable contains this information for N=0,1,16.
The table dot3ControlTable contains information on those Ethernet-like interfaces that also support
the so-called MAC Control sublayer; the dot3PauseTable is similar.

20.13.11 IP-MIB and IP-Forward MIB

The MIB-2 ip group is really about two separate things: strictly local information about IP packets and
delivery, and the forwarding table, which affects a much larger collection of nodes. All nodes have local
IP information, but only routers do significant forwarding. It did not take long for these two subgroups to
separate.
The IP-Forward MIB began in RFC 1354, just over a year after the original MIB-2 in RFC 1312; this
created a new home for forwarding-table information. By the time the first SNMPv2 MIB for the ip group
came out in RFC 2011, the ipRouteTable once part of that group was declared obsolete.
20.13.11.1 IP-MIB
The current version of the IP-MIB is RFC 4293. It allows enabling or disabling of forwarding
(ipForwarding, setting the default TTL (ipDefaultTTL), and, for IPv4 only (as it is not a configurable parameter for IPv6) the fragment-reassembly timeout (ipReasmTimeout).
When IP addresses appear in these tables, the defining entry is almost always prefaced by an object of type
InetAddressType, which can take values ipv4(1) and ipv6(2) (and a few other values for special cases).
Two tables are devoted to IP statistics: the ipSystemStatsTable for system-wide IP information and
the ipIfStatsTable for interface-specific information. Each table includes the IP address type (that is
the IP version) in its index, meaning that separate statistics are kept for IPv4 and IPv6 traffic. The system
table is indexed by the IP address type alone; the interface table is indexed by that and the interface number
(the MIB-2 ifIndex).
The ipIfStatsTable contains, for each interface, counts of packets (and octets) in and out for broadcast,
multicast and unicast, counts of forwarded, reassembled and fragmented packets, counts of packets with any
of several kinds of errors, and related additional counts. An entry for the counter refresh rate is also provided.
When appropriate, 64-bit counters are provided; usually the equivalent 32-bit counter is also provided. The
ipSystemStatsTable provides all-interface summaries of these same counts.
The tables ipv4InterfaceTable and ipv6InterfaceTable represent information about what IP
addresses are assigned to each interface. These are indexed by the interface index alone, meaning that the
tables cannot accurately represent an interface with more than one IP address of the same type.
Additional IP-address information is contained in the ipAddressPrefixTable primarily for IPv6 and
the ipAddressTable for both IPv4 and IPv6. These tables contain information about what IP addresses

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are assigned to what interfaces and where these addresses came from (eg DHCP, 7.8 Dynamic Host Configuration Protocol (DHCP), or Prefix Discovery, 8.6.2 Prefix Discovery). Indexed by the IP address
itself (and also the ipAddressAddrType), these tables thus support the possibility that one interface has
multiple IP addresses (this is particularly common for IPv6).
The ipNetToPhysicalTable represents the map from local IP addresses to physical LAN addresses, as
created by either ARP for IPv4 or Neighbor Discovery for IPv6. In addition to the interface, the IP address
and the physical address, tle table also contains a timestamp indicating when a given entry was last updated
or refreshed, an indication of whether the address mapping is dynamic, static or invalid, and, finally, an
attribute ipNetToPhysicalState. The values for this last are reachable(1), stale(2) for expired reachability, delay(3) and probe(4) relating to active updates of the reachability, invalid(5),
unknown(6) and incomplete(7) indicating that ARP is in progress. See 7.7.1 ARP Finer Points.
There is also a simple version of the forwarding table known as the Default Router Table. This contains
a list of default, or, more accurately, initially configured routes. While this does represent a genuine
forwarding table, it is intended for nodes that do not act as routers and do not engage in routing-update
protocols. The table represents a list of default routes by IP address and interface, and also contains
route-lifetime and route-preference values.
The ipv6RouterAdvertTable is used for specifying timers and other attributes of IPv6 router advertisements, 8.6.1 Router Advertisement.
Finally, the IP-MIB contains two tables for ICMP statistics icmpStatsTable and icmpMsgTable.
The latter keeps track, for example, of how many pings (ICMP Echo) and other ICMP messages were sent
out; see 7.9 Internet Control Message Protocol.
20.13.11.2 IP-Forward MIB
Information specific to a hosts IP-forwarding capability was first split out from the MIB-2 ip group in RFC
1354; it was updated to SNMPv2 in RFC 2096 and the current version is RFC 4292. The original MIB-2
ip group left off at OID mib-2.ip.23; the new IP-Forward MIB begins at mib-2.ip.24.
There have been three iterations of an SNMP-viewable IP forwarding table since the original RFC 1213 ip
groups ipRouteTable at mib-2.ip.21. Here are all four:
ipRouteTable
ipForwardTable
ipCidrRouteTable
inetCidrRouteTable
Each new version has formally deprecated its predecessors.
The first replacement was ipForwardTable, described in RFC 1354. It defines the OID ipForward
to be mib-2.ip.24; the new table is at ipForward.2. This table added several routing attributes, but
perhaps more importantly changed the indexing. The index for ipRouteTable was the IP destination network ipRouteDest, alone. The new tables index also includes a quality-of-service attribute
ipForwardPolicy, usually representing the IPv4 Type of Service field (now usually known as the DS
field, 7.1 The IPv4 Header). This inclusion allows the ipForwardTable to accurately represent routing
based on dest,QoS, as discussed in 9 Routing-Update Algorithms. Such routing is sometimes called

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multipath routing, because it allows multiple paths to a given destination based on different QoS values. However, the mask length is not included in the index, making ipForwardTable inadequate for
representing CIDR routing.
The index also includes the next_hop, which for the actual forwarding table does not make sense the
next_hop is what one is looking up, given the destination but which works fine for SNMP. See the comments about SNMP indexes with more attributes than expected in 20.7 SNMP Tables. The index even includes an attribute ipForwardProto that represents the routing-update protocol that is the source of the
table entry: icmp(4), rip(8) (a common distance-vector implementation), is-is(9) and ospf(13)
(two link-state implementations) and bgp(14).
In addition to the next_hop, this table also includes attributes for ipForwardType (eg local vs remote),
ipForwardAge (the time since the last update), ipForwardInfo, ipForwardNextHopAS, and several routing metrics. The purpose of ipForwardInfo is to provide an OID that can be accessed to provide additional information specific to the routing-update algorithm in use. The ipForwardNextHopAS
allows the specification of the next_hop Autonomous System number, usually relevant only when BGP
(10.6 Border Gateway Protocol, BGP) is involved. (If the AS number is not relevant, it is set to zero.)
The second iteration of the SNMP-viewable IP forwarding table is ipCidrRouteTable, appearing in
RFC 2096 and located at ipForward.4 (and returning to the practice of calling it a route rather than a
forward table). This table adds the address mask, ipCidrRouteMask, to the index, finally allowing
distinct routes to 10.38.0.0/16 and 10.38.0.0/24. The quality-of-service field ipCidrRouteTos remains
in the index (as does the destination), and is now firmly identified with the IPv4 Type of Service (DS) field.
The routing-update algorithm was dropped from the index.
This table also adds an attribute ipCidrRouteStatus of type RowStatus and used for the creation
and deletion of entire rows (that is, forwarding table entries) under the control of SNMP. We will return to
this process in 20.14 Table Row Creation.
The third (and still current) version of the IP forwarding table is inetCidrRouteTable, introduced in
RFC 4292 and located at ipForward.7. The main change introduced by this table is the extension to
support IPv6: the IP-address columns (eg for destination and next_hop) have companion columns for the
address type: ipv4(1) and ipv6(2).
The next_hop attribute (now two columns, with the addition of the address type) is still part of the index.
The address mask used in ipCidrRouteTable is now updated to be a prefix length,
inetCidrRoutePfxLen. The quality-of-service field inetCidrRoutePolicy is an Object ID, declared to be an opaque object without any defined semantics; that is, it is at the implementers discretion.
The IPv4 ToS/DS field evolved in IPv6 to the Traffic Class field, 8.1 The IPv6 Header.
Finally, routes are no longer required to list a single associated interface. The table makes use of the
InterfaceIndexOrZero textual convention of RFC 2863, covering just this situation.

20.13.12 TCP-MIB
RFC 4022 contains some extensions to MIB-2s tcp group. The SNMPv2-based MIB embedded in RFC
4022 repeats the MIB-2 tcp group and then adds new features within the mib-2.tcp (1.3.6.1.2.1.6) tree. RFC
1213 stopped at mib-2.tcp.15; RFC 4022 defines new objects starting at mib-2.tcp.17.

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The new tcpConnectionTable is defined at mib-2.tcp.19, versus the original tcpConnTable at mib2.tcp.13. The newer table supports IPv6; like the inetCidrRouteTable above, each IP-address column
now also has a companion address-type column. For IPv4 users used to the earlier tcpConnTable, this
means that there are extra 1s prefixing the IP addresses in the index, as in this (fictitious) example of host
10.0.0.5 connecting from port 54321 to web server 147.126.1.230:
tcpConnectionState.1.10.0.0.5.54321.1.147.126.1.230.80
The new table also includes a column representing the process ID of the process that has open the local end
of the connection.
This table adheres to the SNMPv2 convention that index columns are not included in the data the attributes
are marked not-accessible. There are only two accessible columns, tcpConnectionState and
tcpConnectionProcess.
TCP-MIB does not make available per-interface TCP statistics, eg the number of TCP bytes sent by eth0.
Nor does it make available per-connection statistics such as packet-loss and retransmission counts or total
bytes transmitted each way.

20.14 Table Row Creation

SNMPv2 also refined the mechanisms by which a manager can add a row to an agents table. In principle,
adding a row to a table is done with the Set() operation. Imagine a table T with three columns indexT,
fruitT and primeT corresponding to T.1, T.2 and T.3. Imagine also that, right now, the table has three
rows with indexT values 10, 11 and 12:
indexT
T.1
10
11
12

fruitT
T.2
apple
blueberry
cantaloupe

primeT
T.3
37
59
67

Then a new row 13,durian,101 might be added with the following multi-attribute Set() operation. The
entries of the new row will have OIDs T.1.13, T.2.13 and T.3.13, and all we have to do to create the row is
assign to these. Note that we are assigning to OIDs that, in the agents current database, do not yet exist.
Set((T.1.13,13), (T.2.13,durian), (T.3.13,101))
Of course, this raises some questions. Will the agent actually allow this? What happens if these Set()
operations are performed individually rather than as a group? Is there any way to delete this newly added
row? And, more seriously, what happens if some other manager tries to insert at the same time the row
13,feijoa,103?
We now turn to the specific row-creation mechanism of RMON.

20.14.1 RMON
RMON, for Remote MONitoring, was an early attempt (first appearing in RFC 1271 eight months after
MIB-2s RFC 1213) at having an SNMP agent take on some monitoring responsibilities of its own. The

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current version is in RFC 2819. The original RMON, now often called RMON1, only implemented LANlayer monitoring, but this was later extended to the IP and Transport layers in RMON2, RFC 4502. We will
here consider only RMON1.
RMON implements only passive monitoring; there is no capability for the remote agent to send out its own
SNMP queries, or even pings (though see 20.14.3 PING-MIB). Monitoring is implemented by putting
the designated interface into promiscuous mode (2.1 10-Mbps classic Ethernet) and capturing all traffic.
In modern fully-switched Ethernets, hosts simply do not see traffic not actually addressed to them, and so
RMON would need to be implemented on a switch or router to be of much practical use.
An agents RMON activity is controlled by an SNMP manager through the insertion of new rows in various
control tables. The mechanism for doing this is our primary concern here.
RMON statistics are divided into ten groups, of which we will consider only the following:
statistics: counts of errors and counts of packets in size ranges 0-64, 65-127, 128-255, 255-511,
512-1023 and 1024-1518 octets.
history: The statistics group data, taken at regular intervals
hosts: The Ethernet senders and receivers seen by an interface
host top N: The top-N senders or receivers
matrix: Information on traffic by sender,receiver pair
20.14.1.1 Statistics
The etherStatsTable is, by default, empty, and is indexed by etherStatsIndex which is an
opaque INTEGER. The column etherStatsDataSource represents the OID of a specific interface
number as defined by ifTable; for example, the interface with ifNumber = 6 would be represented by
mib-2.2.2.1.1.6. One column, etherStatsStatus, has type EntryStatus as follows:
valid(1),
createRequest(2),
underCreation(3),
invalid(4)

The attributes etherStatsDataSource and etherStatsStatus were initially read-write,

which was later changed to status read-create as they can only be written to as part of the creation
of a new row. There is one more read-create attribute, etherStatsOwner, which is a managersupplied string identifying that particular manager (perhaps by IP address and hostname and, if a human is
involved, appropriate additional identification).
To enable statistics collection, the SNMP manager creates a row in the agents etherStatsTable by
setting the three attributes Status, DataSource and Owner (where we have left off the attribute-name
prefix etherStats for readability). Once this is done and the Status is set to valid, the agent begins
collecting data about the desired interface. The manager can read the data as it desires, by accessing that
particular row. Data collection ends when the manager sets the etherStatsStatus for the row to
invalid, though it is up to the agent whether the row is actually deleted at that point.
The etherStatsIndex column is not actually writable, but the manager must still select a value for the
index. One approach is to read the entire table and identify the first unused index value. Other managers,
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however, may be creating new rows in the agent at the same moment, and so a row index that was available
moments before may now be unavailable. If that happens, though, the Set() operation will fail (recall that
in a multi-attribute Set() all the assignments will succeed or all will fail), and the manager can choose a
different index and try again.
One common strategy for reducing the chance of such row-creation collisions is to choose a value for the
index at random.
It is often possible for a manager to create the row with a single Set() operation, though RFC 2819
prohibits setting Status initially to valid. If T is the tree OID etherStatsEntry and 157 is the
managers chosen index, and the manager wants to monitor ifIndex = 6, it could send the following as a
single operation.
Set((T.Status.137,createRequest), (T.DataSource.137,mib-2.2.2.1.1.6), (T.Owner.137,owner))
at which point the agent will create the row with status underCreation. The value createRequest is
used only in manager requests and never appears in actual rows on the agent. The manager will then follow
with
Set(T.Status.137,valid)
If the manager wants or needs to create the entry piecemeal, it can do so as follows:
Set(T.Status.137, createRequest)
Set(T.DataSource.137, mib-2.2.2.1.1.6)
Set(T.Owner.137, owner)
...
Set(T.Status.137, valid)
Immediately following the first Set(T.Status.137, createRequest) the agent will again create the row and
mark it as underCreation, but this time the new row will be missing several columns.
The Status attribute for a row being created should be specified in the very first Set() operation for the
row.
Existing rows may not have their Status set to createRequest. The primary legal state transitions are
as follows:

In some cases, a row can be edited by the manager by changing the status from valid back to

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underConstruction. However, many specific row-creation implementations require that no changes

can be made after a row is marked valid. To change an existing row in such a case, the row should be
marked invalid and a new row created.
Any authorized SNMP manager can read all the etherStatsTable records; there is no requirement
that only the creator of a row can read that row. One manager can also mark another managers record as
invalid, thus effectively deleting it.
It is quite possible for multiple rows to be created all referring to the same interface, eg by multiple managers.
Although it is not forbidden, there is no reason for one manager to create two rows for the same interface.
The four-state EntryStatus type appeared in the original SNMPv1 RFC 1271. Updates to RMON to
SNMPv2 have left EntryStatus alone, but SNMPv2 has also introduced the six-state RowStatus type
which is more likely to be used by new MIBs. We will turn to this in 20.14.2 SNMPv2 RowStatus.
..index:: single: history, RMON
20.14.1.2 History
The history group allows for collection of a series of samples of data from the Statistics group, above, at
regular intervals. The group consists of two tables, historyControlTable where the manager creates
rows to manage the process, and etherHistoryTable containing the actual data.
The historyControlTable contains entries for (again omitting the historyControl prefix from
attribute names) DataSource, representing the interface to be examined, and the Status and Owner
attributes as above. As with the etherStatsTable, when the manager creates a new row it specifies a
value for the index, with full name historyControlIndex, again perhaps chosen at random.
The manager must also specify two additional attributes: Interval representing the time in seconds
between data-collection events, and BucketsRequested representing the requested maximum number
of interface records to be kept by the agent. When the agent reaches the maximum, each new record replaces
the oldest previous record.
The control table contains one agent-created attribute: BucketsGranted indicating the actual maximum
number of interface records to be kept.
Once the control table row is marked valid, the agent starts accumulating one statistics record for the
specified interface every Interval seconds. These records are numbered consecutively; the number
of a given record is its SampleIndex. The agent keeps only BucketsRequested records, so once
SampleIndex reaches that value then, whenever record N is added, record N BucketsRequested
is deleted.
All this data collected by the agent is stored in etherHistoryTable, which contains columns for all
the statistics in etherStatsTable except for the counts of packets in the various size ranges. The table
is indexed by the pair of values Index corresponding to etherControlIndex and the record number
SampleIndex. As such, this table is the disjoint union of multiple independent series of consecutively
numbered records, one series for each Index value in the control table, that is, one series for each manager
request.
As an example, suppose the manager asked for history statistics to be kept for a given interface at a rate
of once a minute, the BucketsGranted is 70, and the control-index value for this request is 491. After
one hour, the table contains records with sample indexes 1-60; the full row-index values are 491,1 through
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491,60. After one hour and eleven minutes, record 491,71 replaces record 491,1. After two hours,
records with sample indexes of 51-120 are available. The manager might return once an hour and retrieve
the most recent 60 records.
A manager might also create two control-table records, one holding 25 records taken at 1-hour intervals and
another holding 60 records taken at 1-minute intervals. If all is well, that manager might download the first
table once a day, and entirely ignore the second table. The manager always has available these 1-minute
records for the last hour, though, and can access them as needed if a problem arises (perhaps signaled by
something else entirely).
A manager can easily retrieve only its own rows from the etherHistoryTable. Let T be the root of the
etherHistoryTable, which has columns 1-15. Suppose again a manager has created its controlTable
row with a value for historyControlIndex of 491; the manager can then retrieve the first of its data
rows with the following; note that each OID contains the column number and part of the row index.
GetNext(T.1.491, T.2.491, ..., T.15.491)
If the history-table rows associated with 491 have sample-index values ranging from 37 to 66, the above
GetNext() will return the row indexed by 491,37; that is, the values paired with the following OIDs:
T.1.491.37, T.2.491.37, ..., T.15.491.37
Subsequent GetNext()s will return the subsequent rows associated with control entry 491: row 491,38,
row 491,39, etc. If the etherHistoryTable had been indexed in the reverse order, with the sample
index first and the historyControlIndex second, a substantial linear search would be necessary to
locate the first row with a given value for the control index.
20.14.1.3 Hosts
The host group allows an agent to keep track of what other hosts in RMON1 identified by their Ethernet
address are currently active.
Like the historyControlTable, the hostControlTable allows a manager to specify
DataSource, Status and Owner; the manager also specifies TableSize.
Once the control-table entry is valid, the agent starts recording hosts in the hostTable, which is indexed by the control-table index and the host Ethernet address. The agent also records each new hosts
CreationOrder value, an integer record number starting at 1.
The hostTable also maintains counters for the following per-host attributes; these are updated whenever
the agent sees another packet to or from that host. We will revisit these in the hostTopNTable, following.
InPackets
OutPackets
InOctets
OutOctets
OutErrors
OutBroadcastPkts
OutMulticastPkts
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When the number of host entries for a particular control-table index value exceeds TableSize that is,
the new hosts CreationOrder would exceed TableSize, old entries are removed and all hosts in the
table are given updated CreationOrder values. That is, if the table of size three contains entries for
hosts A, B and C with creationOrder values of 1, 2 and 3, and host D comes along, then A will be
deleted and B, C and D will be given creationOrder values of 1, 2 and 3 respectively. This is quite
different from the record-number assignments in the etherHistoryTable, where the SampleIndex
record numbers are immutable.
A consequence of this is that the CreationOrder values are always contiguous integers starting at 1.
Entries are deleted based on order of creation, not order of last update. The hostTable does not even
have an attribute representing the time a given hosts entry (or entries) was last updated.
As a convenience, the data in hostTable is also made available in hostTimeTable, but there indexed
by the control-table index and the CreationOrder time. This makes for potentially faster lookup; for
example, the manager always knows that the record with CreationOrder = 1 is the oldest record. More
importantly, this alternative index allows the manager to download the most recent entries in a single step.
Whenever a host is deleted because the table is full, the CreationOrder values assigned to other hosts
all change, and so the indexing to hostTimeTable changes. Thus, a manager downloading rows from
hostTimeTable one at a time must be prepared for the possibility that what had earlier been row 3 is
now row 2, and that a host might be duplicated or skipped. To help managers deal with this, the control table
has an entry LastDeleteTime representing the time in TimeTicks since startup of the last deletion
and thus CreationOrder renumbering. If a manager sees that this value changes, it can, for example,
start the data request over from the beginning.
20.14.1.4 Host Top N
The hostTopN table is a report prepared by the agent about the top N entries (where N is manager-supplied)
in the hostTable, over an interval of time. The manager specifies in the control table the Status and
Owner attributes, the HostIndex control-table index value from hostControlTable, and also the
Duration (in seconds) and the RateBase indicating which of the following hostTable statistics is to
be used in the ranking:
InPackets
OutPackets
InOctets
OutOctets
OutErrors
OutBroadcastPkts
OutMulticastPkts
The value of N is in the attribute RequestedSize; the agent may reduce this and communicates any
change (or lack of it) through the attribute GrantedSize.
Once the control-table row becomes valid, the agent then starts maintaining counters for all the entries
in the part of hostTable indexed by HostIndex, and at the end of the Duration sorts the data and

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places its top-N results in the hostTopN table. If, during the interval, some hosts were removed from the
hostTable because the table was full, the results may be inaccurate.
All result statistics are inaccessible until the Duration has elapsed and the particular hostTopN
report has run its course, at which point the results become read-only.
20.14.1.5 Matrix
The matrix group allows an agent to collect information on traffic flow indexed by the source and destination
addresses. The manager begins the process by supplying attributes for Status, Owner, the usual Index,
the interface DataSource, and the maximum table size.
Once the row is valid, the agent begins collecting, for every source,destination pair, counts of the number
of packets, octets and errors. The record for A,B counts these things sent by A; the corresponding record
for B,A counts the reverse direction. The actual matrixSDTable is indexed by the manager-supplied
Index value and the source and destination Ethernet addresses in that order.
A companion table (or view) is also maintained called matrixDSTable, that lists the same information
but indexed by destination first and then source. This view is not present to supply information about the
reverse direction; that would be obtained by reversing source and destination in the matrixSDTable.
Rather, the matrixDSTable allows a manager to extract all information about a single destination D in a
single SNMP tree-walk operation of the prefix matrixDSTable.Index.D. This is similar to the indexing
discussion at the end of 20.14.1.2 History. See exercise 9.

20.14.2 SNMPv2 RowStatus

RMON made do with four values for its EntryStatus attribute: createRequest, underCreation,
valid and invalid. SNMPv2 still supports EntryStatus, but the preferred form is the six-value
RowStatus, RFC 2579, with options
active: like valid
notReady: the agent knows that the row is not complete
notInService: the row is complete, but has not yet been activated
createAndGo: like createRequest followed by valid
createAndWait: like createRequest
destroy: like invalid
The createAndGo option allows a manager to make a single new-row request to the agent, and, if successful, the agent will immediately set the row status to active. Using EntryStatus requires two steps,
as in 20.14.1.1 Statistics
The manager can ask for any status except notReady. Every row created will, however, be marked at the
agent with one of notReady, notInService or active.
As with EntryStatus, the manager must still choose an index value. Pseudo-random selection
remains appropriate in many cases, but the agent may also supply, if the MIB file supports it, a
recommendedNextIndex attribute.

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Here is a simplified RowStatus state-transition diagram. Not all links to destroy, or from a node back
to itself, are shown. See RFC 2579, p 9, for more details.

After the manager issues a createAndWait, the agent fills in the attributes provided by the manager, and
any other default attributes it has available. The manager, if desired, can now Get() the entire row, and
find out what values are still missing. These values will be reported as noSuchInstance. Alternatively,
the manager may simply know that it has more attributes to Set().
If the agent knows the row is missing attributes necessary for activation, it will set the RowStatus to
notReady, otherwise to notInService. A notInService row can, of course, still have undefined
read-only attributes that the agent will later set after activation. A notInService row can also still
have attributes the manager intends to set before activation, but the agent has given default values in the
interim.
Once the manager has set all the attributes required for activation, it sets the RowStatus attribute to
active, and agent activity begins.

20.14.3 PING-MIB
The idea behind the ping MIB is to allow a manager to ask an agent to repeatedly ping a target, 7.9 Internet Control Message Protocol, and then to report the success rate. Cisco Systems has a ping MIB
at ftp://ftp.cisco.com/pub/mibs/v2/CISCO-PING-MIB.my, but we will use the IETF alternative in RFC
2925/RFC 4560. The latter is officially titled DISMAN-PING-MIB (DISMAN is short for DIStributed
MANagement). Both ping MIBs make use of the RowStatus convention of the previous section.
The manager can ping the target itself, if there are no firewalls in the way, but the result may likely be
different from that obtained by the agent.
The actual MIB supports multiple types of ping besides the usual ICMP Echo Request, but we will consider
only the latter.
The DISMAN version has control table pingCtlTable and the results appear in pingResultsTable;
in the Cisco version the same table is used for both control and results. The results include the minimum,
maximum and average RTT, the number of pings sent and the number of responses. The results table also
has an attribute OperStatus to indicate whether the test has stopped.
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The pingCtlTable contains a wide range of options for the actual ping request, including options to
specify the outgoing IP address for multi-homed agents. The more familiar options, however, include
pingCtlTargetAddress: and address type, as IPv6 is supported
pingCtlDataSize: how big each ping packet is
pingCtlTimeout: how long before the agent gives up on any one ping
pingCtlProbeCount: the number of pings to be sent
pingCtlFrequency: the interval between pings
The table also includes pingCtlRowStatus of type RowStatus, above, and pingCtlOwnerIndex
of type SnmpAdminString, which is a fancier way of identifying the manager than the RMON Owner
attributes, and which can include some SNMPv3 credentials as desired.
The new row is created, RowStatus is set to active (perhaps through createAndGo), and the test
begins.
By setting the control tables RowStatus to destroy, a manager can attempt to halt a ping series in
progress.

20.15 SNMPv3
SNMP version 3 added authentication and encryption to SNMPv2c, but made relatively few other changes
except to nomenclature. The original definitions are in RFC 3410 through RFC 3415; RFC 3410 first
appeared as RFC 2570. SNMPv3 introduced the User Security Model, or USM, in which agents allow
manager access only if the manager has presented an appropriate key, derived ultimately from a passphrase.
The agents response can be either digitally signed or encrypted, as desired.
SNMPv3 did make several terminology changes. Any SNMP node either manager or agent is now
known as an SNMP entity. An entity is divided into two parts; the first is the SNMP engine consisting of
the message dispatcher, the message processor, and the security and access-control subsystems. The second
part consists of the various SNMP applications, consisting, for agents, primarily of the command responder
(responding to Get() and GetNext(), etc). One goal of this architectural division is to separate the
applications from the authentication mechanisms. Another goal, however, is to provide a framework in
which future new applications can easily be supported.
It is the SNMP engine that must implement all the new security provisions.

20.15.1 What Could Possibly Go Wrong?

RFC 3414 analyzes the the risks of inadequate SNMP security, and identifies two primary and two secondary
threats. The primary threats are as follows:
Unauthorized modification of information in transit: that is, man-in-the-middle attacks in which As
message to B is intercepted and modified by an intermediate node M.
Masquerade: unauthorized operations by a manager (or agent) masquerading as an authorized entity.
Use of guessed or cracked passwords would fall into this category.

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The secondary threats are these:

Message Stream Modification: This includes re-ordering messages, and, perhaps most importantly,
replay attacks. For example, even if no keys are ever discovered, an attacker might learn that a
certain message causes a certain server to reboot. Retransmitting that message could have serious
consequences.
Disclosure: reading by an eavesdropper of the management information in transit.
All but the last, disclosure, can be addressed by appropriate digital signatures and timestamps; encryption
is only needed when disclosure is an active concern. Because disclosure is not always a significant concern,
and perhaps because when RFC 2574 was written in 1999 the unlicensed export of encryption technology
from the United States was illegal, SNMPv3 defines two separate new levels of security:
authentication-only, addressing modification, masquerade and replay
encryption, addressing the above and disclosure
These correspond to three values for the snmpSecurityLevel textual convention (in which priv abbreviates privacy):
noAuthNoPriv: no authentication
authNoPriv: authentication, but no encryption
authPriv: encryption of all messages
Encryption (privacy) implies authentication; encrypted messages also include the authentication signatures
described below.

20.15.2 Cryptographic Fundamentals

SNMPv3 authentication is based on cryptographic hash functions: functions which take a data string of
arbitrary length and return a fixed-length hash, in such a way that
Knowing the hash value sheds no practical light on the original data
Given a hash value, there is no feasible way to find a message yielding that hash
Further details are at 21.5 Secure Hashes.
The Internet checksum of 5.4 Error Detection fails as a cryptographic hash, for example, because given a
message m and a hash value h , one can calculate hash(m) = h and then append to m a two-byte string based
on h h, yielding a message m for which hash(m ) = h .
The two cryptographic hash functions originally supported by SNMPv3 in RFC 3414 are MD5, which
produces a 128-bit hash, and SHA-1, which produces a 160-bit hash. Since the publication of RFC 3414
in 2002, vulnerabilities in each of these hash functions have been discovered. The SNMP framework in
principle allows easy substitution of new hash functions, but to date new hashes have not been standardized.
If two parties share a secret key k, the basic hash-based way to sign a message m is to append to it the value
hash(m k), where m k is the result of appending k to m. An eavesdropper will see m and hash(m k) but
this will provide no information about k, and, similarly, an attacker will, given a message m, not be able
to generate the value hash(m k) without knowing k. Because some hash functions are vulnerable to the
so-called length-extension attack when used this way, the actual signature is often slightly more involved
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(20.15.5 Passwords and Keys and 21.5.1 Secure Hashes and Authentication), but hash(m k) is a good
example of the basic concept.
The SNMP encryption mechanism is also based on shared secret keys (21.6 Shared-Key Encryption);
that is, public-key encryption is not used. RFC 3414 describes the use of the Data Encryption Standard
cipher, or DES, which uses 56-bit keys. Later, RFC 3826 introduced the use of the Advanced Encryption
Standard cipher, or AES, which in SNMP users a 128-bit key. (Both DES and AES are discussed briefly in
21.6.3 Block Ciphers.) DES is, if anything, even more vulnerable than MD5 and SHA-1, due to the limited
key length; AES is a much stronger choice.
Shared-secret encryption is based, abstractly, on an encrypting function E(p,k) that takes a plaintext message
p and a key k and returns the encrypted ciphertext. Similarly, there is a decrypting function D(c,k) that takes
an encrypted ciphertext message c and the key k and returns the original plaintext.

20.15.3 SNMPv3 Engines

The SNMPv3 engine is the component of an SNMP entity charged with ensuring security. Each SNMPv3
engine manager or agent has an snmpEngineID. This is a string that, by default, incorporates the nodes
IP or Ethernet address and additional standard information. While it is
In any SNMPv3 exchange, one SNMPv3 engine is designated authoritative. For a Get() or Set()
request, or any other request that requires a response, it is always the agent that is authoritative. It is the job
of the authoritative engine to validate the message received from the other engine.
For either authentication-only or encrypting security levels, the nonauthoritative engine must present a pair
consisting of the following:
userName: a human-readable string identifying a person or manager-computer
authParameters: a string holding data specific to the authentication/encryption mechanism in
use that serves to authenticate the user. This might be thought of as a passphrase, except one goal is
never to send passphrases in the clear.
The authoritative engine must keep a table of all the userName values it recognizes, and, for each, the
appropriate key with which to validate the user.
Any SNMPv3 engine also keeps track of two 32-bit attributes
snmpEngineTime: the number of seconds since the engine last rebooted
snmpEngineBoots: the number of times the engine has rebooted
The pair of these, which we will abbreviate as Time,Boots, uniquely identifies a point in time for an entity
(to the nearest second). They are used to prevent replay attacks; two messages sent more than one second
apart will never have the same Time,Boots timestamp.
Note that keeping track of snmpEngineBoots implies that the entity have some form of persistent storage.
When a nonauthoritative engine (for our purposes, the manager) wants to send a request to an authoritative engine (the agent), the first step is to find out the agents snmpEngineID and then its current
Time,Boots value. This is done through an initial two-step discovery process. The first request contains an empty varBindList, an empty engineID, a username of the empty string and a security level of

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noAuthNoPriv. The authoritative side sends a response containing its engineID. The second step is to
send a request, again with an empty varBindList but now containing a valid username,key pair. The
value for Time,Boots is 0,0. The authoritative engine now responds with a message including its actual
Time,Boots.

20.15.4 Message Authentication

After discovering the authoritative engines Time,Boots, the nonauthoritative engine stores the authoritative engines snmpEngineBoots and also the difference between the authoritative engines
snmpEngineTime and its own clock time. It can now use this difference to approximate the authoritative engines snmpEngineTime at any point in the future. Every message from the nonauthoritative
engine will include its estimate of the authoritative engines current Time,Boots value. Relative drift between the two engines clocks will eventually mean this estimate fails, but, as we shall see, it can be expected
to be close enough for quite a while.
The authoritative engine accepts a non-empty request only if all three of the following hold, where
Time and Boots are the values submitted by the nonauthoritative engine and snmpEngineTime and
snmpEngineBoots are the authoritative engines own values:
Boots = snmpEngineBoots
snmpEngineBoots < 231 1
Time and snmpEngineTime differ by less than 150 seconds
The maximum allowable clock drift, in other words, is 150 seconds. If the two clocks drift by more than
that, the nonauthoritative side must again go through the synchronization process outlined at the end of the
previous section.
The actual message signing is based on a shared secret key, authKey, negotiated previously between user
and agent. Because the key is usually specific to the agent, it is sometimes called a local key, kl.

20.15.5 Passwords and Keys

Users are identified by userName values, and also (usually) have human-readable passwords. These passwords must be converted first into keys, in such a way that each user,agent pair has its own unique key.
We will start with a mechanism commonly used for authentication-only security; it is the same for both the
MD5 and SHA-1 hashes. We will denote the hash function (either MD5 or SHA-1) by hash().
The first step is to create a digest based on the password. It would suffice to set the digest to hash(password).
In order to make the passwordkey conversion process relatively slow, however, so as to hamper brute-force
attacks, Appendix A of RFC 3414 recommends repeating the password (logically, at least) to fill a buffer
220 bytes in length. The hash() function is then applied to this megabyte string. (For another perspective
on intentionally slow password operations, see 21.5.2 Password Hashes.)
The next step is to take the engineID of the agent for which the user is generating this particular key and
take the hash of the concatenation of the digest, engineID and digest again:
kl = hash(digest engineID digest)

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This is now the user,agent shared key, or local key. It must be entered (or computed) on the agent, and
stored there. For MD5 it is 16 bytes long; for SHA-1 it is 20 bytes long.
This mechanism is, strictly speaking, optional; an agent does not know how manager keys were generated,
and thus cannot enforce any particular mechanism when a managers key is later changed as below. Any
secure way to generate a unique key kl for each user and each agent would be sufficient. The mechanism
here, though, has the advantage that any manager node can compute a users local key directly from the
password and the agent engineID; no keys need be stored by the manager. This allows a manager to use one
password for multiple agents; compromise of any one agent and its attendant local key kl should not affect
the security of other agents or of the original password.

20.15.6 Message Signing

An SNMP message mesg is now signed by a local key kl using the Hash Message Authentication Code,
or HMAC (RFC 2104), as follows:
kl is extended to a 64-byte ek by padding with zeroes.
A constant string ipad is formed by repeating the octet 0x36 64 times.
A constant string opad is formed by repeating 0x5c 64 times.
Set k1 = ek XOR ipad.
Set k2 = ek XOR opad.
The authParameter field the actual digital signature is now set to be the first 12 bytes of
hash(k2 hash(k1 mesg))
This is slightly more complicated than the hash(mesg kl) mechanism suggested in 20.15.2 Cryptographic
Fundamentals, in order to make it more resistant to potential vulnerabilities in the hash() function; see
21.5.1 Secure Hashes and Authentication.
The receiver can replicate this authParameter calculation and verify that it matches the value transmitted
in the packet.

20.15.7 Key Changes

Suppose a user wants to change his or her password, and thus the key k. The manager will need to communicate the new key to the agent in such a way that it is not exposed to eavesdroppers. Here is the mechanism,
where, again, hash() is either MD5 or SHA-1 as appropriate; we will use N to denote the length in bytes
of the result of hash() (either 16 or 20).
The original key is here denoted oldkey and the new key is newkey.
The manager first chooses a string random of N bytes chosen as randomly as possible. A second N-byte
string delta is then calculated as follows:
temp = hash(oldkey random)
delta = temp XOR newkey

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At this point, random and delta are sent in the clear from the manager to the agent. The agent can
use random and oldkey to compute temp, and thus newkey = delta XOR temp. An eavesdropper
cannot use random to find out anything about temp without knowing oldkey, and cannot get anything
useful out of delta unless either newkey or temp is known.
Note that if an eavesdropper saves random and delta, and later discovers the users oldkey, then
newkey can be calculated easily. Using the language of 21.8.2 Forward Secrecy, forward secrecy fails
badly. However, if an eavesdropper later discovers newkey, there is no obvious way to find oldkey; see
exercise 10.
The actual process is to combine random and delta into a single 2N-byte keyChange string.

20.15.8 Creating Additional Users

Suppose we want to add user alice to an agent, together with Alices local key kl. The problem is that
there is no pre-existing shared key, and so no way to transmit kl to the agent via SNMP without risk of
eavesdropping.
One solution is to require out-of-band account creation, that is, for Alice to log on to the agent via, perhaps,
an ssh or direct serial-line connection, and enter there her name and password. This works, but is awkward
on a large network.
The usual way is to create an account for Alice by cloning some other account on the agent. We will assume
that an initial account has been created as in 20.15.5 Passwords and Keys for user master. Cloning the
account just requires specification of the names master and alice; no keys need be sent. The second step
is to update the password for the new alice account, which is done via the key-change mechanism of the
previous section. We look at the details of the cloning procedure in the next section.

20.15.9 VACM for SNMPv3

For SNMPv1 and SNMPv2c, VACM created associations between community strings and security groups,
an then between security groups and permitted views of the OID tree. For SNMPv3, user names replace
community strings.
There are three primary VACM tables, defined in RFC 3415.
The first is
vacmSecurityToGroupTable, indexed by the security model and the security name. For
SNMPv3 the model is 3 and the security name is the username; the SNMPv3 rows of the table are thus
effectively indexed by the username. It is not unreasonable to have a separate security group for each user,
eg grppld for user pld.
The second table is vacmAccessTable, indexed by security group, security model (community or SNMPv3) and security level (eg authNoPriv). The table can be used to look up a named view for each
of read and write privileges (also trap privileges, if desired). (This table is also indexed by contextName, but often there is only one contextName, the default.)
Finally, the named views themselves are stored in vacmViewTreeFamilyTable, indexed by view name
and an OID prefix representing an OID subtree. A given view v consists of all the OID prefixes op for which
v,op appears in the table. Some prefixes may have an associated mask, typically to allow access to a
designated table row, and some prefixes may represent exclusion of that subtree from the view. Typical

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view names are _all_, _none_ and systemonly; we created a view mib2+private in 20.11 SNMPv1
communities and security.
20.15.9.1 The usmUserTable
Now it is time to look at the actual table in which agents store user names and their associated authentication
credentials, usmUserTable. It has the attributes below, which should all be prefixed by usmUser; the
index attributes are the first two, EngineID and Name. We will consider only those attributes related to
authentication-only security.
EngineID
Name
SecurityName
CloneFrom
AuthProtocol
AuthKeyChange
OwnAuthKeyChange
PrivProtocol
PrivKeyChange
OwnPrivKeyChange
Public
StorageType
Status

usually the agents own engineID

eg alice
usually the same as Name
an indication of the row this entry is cloned from
MD5 or SHA-1
the random, delta keyChange object of 20.15.7 Key Changes
the same but with different permissions; see below
used for encryption
used for encryption
used for encryption
used for confirming key changes; see below
hopefully permanent, meaning values persist between reboots
of type RowStatus, for row insertion

One can think of the table as also having an implicit authKey column, representing the local key corresponding to Name, that is never directly readable or writable. RFC 3414 states flatly the authKey is not
accessible via SNMP. However, the agent must still keep the authKey somewhere, tied to the Name, so it
can validate a given user based on the Name and authParameter supplied in a request.
Because EngineID is usually the agents own EngineID, the table is de facto indexed just by Name.
Recall that, as was discussed in 20.13.4 SNMPv2 Indexes, the index attributes, EngineID and Name, will
not be directly accessible, but will be encoded in the OID associated with every other retrieved attribute.
The username alice will thus be encoded, using the ASCII string encoding, as 97.108.105.99.101, not
necessarily easily readable by human managers.
We are now in a position explain the cloning process and the key-change process as they play out with this
table.
A specific semantic rule for this table is that the use of Set() to assign a random, delta keyChange object
to AuthKeyChange or OwnAuthKeyChange causes that users hidden authKey to be updated via the
process of 20.15.7 Key Changes. The user cannot confirm directly that this change succeeded, as a read
of these keyChange attributes returns the empty string, so the usual recommended strategy is also to write a
random value to the Public attribute; because Set() operations are atomic, a change to the latter means
the former change succeeded as well.
Because it is possible for multiple managers to be updating the table simultaneously, a single
TestAndIncr object named usmUserSpinLock may be used to enforce serialization, as in 20.13.5 TestAndIncr. So the full recommended sequence for updating a key is as follows, where keyChange is the
random, delta keyChange object and index encodes the EngineID and the Name:

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rand = random value chosen by the manager

val := Get(usmUserSpinLock)
Set((usmUserSpinLock, val), (AuthKeyChange.index, keyChange), (Public.index,
rand))
This is repeated as necessary until Get(Public) returns rand, indicating that the Set() operations all
succeeded.
The above applies for a key change being done by a privileged manager, ie one with write privileges to the
AuthKeyChange column of usmUserTable, on behalf of someone else. Ordinary users should not
have write access to this column; this can be enforced using VACM to construct a view omitting this column.
KeyChange Privileges
While it is good practice to ensure that ordinary SNMPv3 users cannot change one anothers keys, it
is not unheard of for all users to be fully trusted, and all granted write permission to the entire view
.1.3.6.1.
If an ordinary user wants to change his or her own key, the column OwnAuthKeyChange may be used.
Any user may write to this column, but the write will only succeed if the userName by which the request
is authenticated is equal to the Name representing the index for this particular keyChange. In other words,
anyone with write access to the OwnAuthKeyChange column can change his or her own key, and only his
or her own key.
To clone a new user from an existing one, the first step is to choose the row to be cloned, represented by the
OID of an index value. Call this cloneRow, and let index again encode the EngineID and Name:
val := Get(usmUserSpinLock)
Set((usmUserSpinLock, val), (CloneFrom.index, cloneRow), (Status.index,
createAndWait))
Typically the key change is then executed before changing the Status to active, though this is not
required.
20.15.9.2 Creating Accounts in Net-SNMP
To create an initial account, the configuration files are used. On the authors installation, the administratively written configuration file is /etc/snmp/snmpd.conf and the system-written configuration file is
/var/lib/snmp/snmpd.conf; the latter contains snmpEngineBoots and other persistent SNMP data. The
following line goes neither place, but rather in /var/net-snmp/snmpd.conf; it creates an SNMPv3 user named
master with MD5-authentication password saskatchewan.
createUser master MD5 saskatchewan
The next time SNMP is restarted or is otherwise induced to reload its configuration files, the line above
causes the following entry to be created (on a single line) in /var/lib/snmp/snmpd.conf:
usmUser 1 3 0x80001f8880889cb038b1aca650 "master" "master" NULL
\

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.1.3.6.1.6.3.10.1.1.2 0x7293f49a82fc950f5c344efd94dbb7db
.1.3.6.1.6.3.10.1.2.1 0x 0x
The new localized key is 0x7293f49a82fc950f5c344efd94dbb7db; the first hex string beginning 0x80001 is
the engineID.
For this new account to be able to do anything, we must also add the following permissions entry to
/etc/snmp/snmpd.conf:
rwuser master
The effect of this entry is to create an entry in vacmSecurityToGroupTable associating user
master with its own security group (Net-SNMP names it grpmaster), and then an entry in the
vacmAccessTable granting this new group SNMPv3 read and write access to the entire OID tree. (In
general we could also have used rouser, except that we will need master to be able to create new users.)
After permissions (at least read permissions) are enabled, the following snmpget should work
snmpget -v 3 -u master -l authNoPriv -a MD5 -A saskatchewan
localhost 1.3.6.1.2.1.1.4.0
The hostname is localhost if this is being run on the same machine that the Net-SNMP agent is running
on; it can also of course be run remotely.
We can make this a little shorter by editing the Net-SNMP per-user manager configuration file
$HOME/.snmp/snmp.conf to add the following lines:
defSecurityName master
defAuthType MD5
defSecurityLevel authNoPriv
defAuthPassphrase saskatchewan
Now the snmpget command can be shortened to
snmpget -v 3 localhost 1.3.6.1.2.1.1.4.0
If we take the password, saskatchewan, and repeat it to 220 bytes, the MD5 checksum is
0x3da9dbfc3a78acb675e436746e1f4f8a; this is the digest of 20.15.5 Passwords and Keys. From
the /var/lib/snmp/snmpd.conf file (or from the created usmUser entry above) we find the engineID is
0x80001f8880889cb038b1aca650. If we convert these two strings to binary data, concatenate them as
digest engineID digest, and take the MD5 checksum, we indeed get
7293f49a82fc950f5c344efd94dbb7db
which is exactly the key entry in the /var/lib/snmp/snmpd.conf file.
20.15.9.2.1 Cloning in Net-SNMP

We can now clone this account while SNMP is running, using the snmpusm command-line utility. The
following line creates an account pld cloned from master, using the master account (and assuming that
master is an rwuser and not an rouser); we have abbreviated by auth the full credentials -u master -l
authNoPriv -a MD5 -A saskatchewan. We have switched from localhost to HOST to emphasize that this can be run remotely.

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snmpusm -v3 auth HOST create pld master

We then change the password to ramblers, still using the authority of user master; the -Ca option tells
snmpusm to change only the authentication key.
snmpusm -v3 auth -Ca HOST passwd saskatchewan ramblers pld
But before new user pld can do anything (including changing plds password with plds own credentials), we
must grant access. We can again edit /etc/snmp/snmpd.conf on the machine running the Net-SNMP agent,
adding the directive rwuser pld and then restarting SNMP. We can also, however, use the Net-SNMP
snmpvacm command here, which works remotely:
snmpvacm -v3 auth HOST createSec2Group 3 pld grppld
snmpvacm -v3 auth HOST createAccess grppld 3 2 0 _all_ _none_
_none_
These two commands create entries in the vacmSecurityToGroupTable and the
vacmAccessTable respectively. The parameters following grppld in the second command include the
USM security model (3), the authNoPriv security level (2), a placeholder (0) for the contextmatch flag
which we are not using, and the views _all_ and _none_ predefined by Net-SNMP. The command here gives
grppld read-only access to everything. If user pld is to be able to change plds password, write-access
must be granted at least to the usmUserOwnAuthKeyChange column of usmUserTable.
After all this, user pld can then issue an snmpget using plds own credentials with
snmpget -v 3 -u pld -l authNoPriv -a MD5 -A ramblers HOST
1.3.6.1.2.1.1.4.0
The commands snmpusm and snmpvacm do not support making general modifications; if a USM or
VACM entry here is made incorrectly, it must be deleted and re-created.

20.16 Exercises
1. Consider the table below. The first column is the index.
index
1
3
57
92

count
401
523
607
727

veggie
kale
kohlrabi
mche
okra

Give the OID for each data value, assuming this table were encoded in SNMP. Assume the columns are
assigned OID levels 1, 2, and 3 in order (including the index column), and the root of the subtree (the
tableEntry OID) is represented by T. Note that some data values are missing.
2. Recall that GetBulk() acts like a repeated GetNext(). Why is there no GetBulk() equivalent of
Get()?
3. What happens if we have a three-row, three-column table and ask, using GetBulk(), for the first two
columns with a repetition count of four? What is retrieved? Assume entries are of the form T.col.row, for
col and row each ranging from 1 to 3.
4. Consider the following multi-attribute GetNext() as presented in 20.8.1 Multi-attribute Get():
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GetNext(T.1.4, T.2.4, T.3.4, T.4.4, T.10.4, T.11.4)

The answer given in the text explicitly assumed that the only columns existing were those shown: 1, 2, 3, 4,
10 and 11. What would be the result if all columns of ifTable were present? Assume, as before, that row
4 is the final row.
5. Suppose you want an SNMP table identTable to hold idNum,userName pairs, where idnum is to
be an INTEGER and username an OCTET STRING. The INDEX is idnum. Give ASN.1 definitions for the
following:
identTable
identEntry
IdentEntry
idNum
userName
Follow the SNMPv2 convention of not including the index column in the actual table data, 20.13.4 SNMPv2
Indexes.
6. (a) What can an SNMP agent do to detect that a manager has created a row in state underCreation
and then crashed, leaving the row abandoned?
(b). What can an SNMP agent do to detect that a manager has created a valid row, and has later crashed?
7. In the RMON hostTopN table (20.14.1.4 Host Top N), the agent does all the report-building work.
Would it be possible for the manager to do this? If not, why not? If so, why do you think RMON provides
this table?
8. In the RMON matrixSDTable (20.14.1.5 Matrix), the table constructed collects data from only a
single interface on the agent (that interface specified by DataSource). Could we get additional Matrix
information by considering other interfaces? Why or why not?
9. The RMON matrix group (20.14.1.5
matrixSDTable and matrixDSTable.

Matrix) provides two tables with the same information,

(a). Suppose we want all the table data about a given destination D, and have only the matrixSDTable.
Explain why every row of matrixSDTable would need to be examined.
(b). Now suppose we repeat the investigation of (a), but this time the manager has previously downloaded
the complete list of all hosts on the LAN by using the RMON hosts tables. If N is the number of hosts on
the LAN, explain how to find all hosts that communicated with D using only N retrieval requests.
(c). Explain how to find all hosts S that sent packets to D even more quickly using the matrixDSTable.
If MN is the number of such S, your answer should involve N+1 retrieval requests.
10. In the keychange operation of 20.15.7 Key Changes, suppose the manager simply transmitted delta2
= oldkey XOR newkey to the agent.
(a). Suppose an eavesdropper discovers delta2 and also knows a few bits of oldkey. What can the
eavesdropper learn about newkey? Would the same vulnerability apply to the mechanism of 20.15.7 Key
Changes?

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(b). Suppose an eavesdropper later discovers newkey. Explain how to recover oldkey, and why this does
not work when the mechanism of 20.15.7 Key Changes is used.
11. List the OID prefixes for which a manager would need to be granted write permission if the manager
were to be able to modify all settings in .1.3.6.1.2.1 and .1.3.6.1.4.1, and change their own local key, but
not have access to columns in usmUserTable that would allow modification of other manager accounts.
(The VACM table has an exception option to make this easier).
12. Suppose a manager has write permission only for those columns in usmUserTable necessary for
password changes, but has full write access to the VACM tables. Explain how the manager can modify the
local keys of other managers.
13. Use Wireshark to monitor localhost traffic while you use the Net-SNMP snmpusm command to
change a managers own local key. Does the command use the usmUserAuthKeyChange column or
the usmUserOwnAuthKeyChange column?

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21 SECURITY

[This chapter is undergoing active editing pld, 10/31/2014]

How do we keep intruders out of our computers? How do we keep them from impersonating us, or from
listening in to our conversations or downloading our data? Computer security problems are in the news on
almost a daily basis. In this chapter we take a look at just a few of the issues involved in building secure
networks.
For our limited overview here, we will divide attacks into three categories:
1. Attacks that execute the intruders code on the target computer
2. Attacks that extract data from the target, without code injection
3. Eavesdropping on or interfering with computer-to-computer communications
The first category is arguably the most serious; this usually amounts to a complete takeover, though occasionally the attackers code is limited by operating-system privileges. A computer taken over this way is
sometimes said to have been owned. We discuss these attacks below in 21.1 Code-Execution Intrusion.
Perhaps the simplest form of such an attack is through stolen or guessed passwords to a system that offers
remote login to command-shell accounts. More technical forms of attack may involve a virus, a buffer overflow (21.2 Stack Buffer Overflow and 21.3 Heap Buffer Overflow), a protocol flaw (21.1.2 Christmas
Day Attack), or some other software flaw (21.1.1 The Morris Worm).
In the second category are intrusions that do not involve remote code execution; a server application may be
manipulated to give up data in ways that its designers did not foresee.
For example, in 2008 David Kernell gained access to the Yahoo email account of then-vice-presidential
candidate Sarah Palin, by guessing or looking up the answers to the forgotten-password security questions
for the account. One question was Palins birthdate. Another was where did you meet your spouse?,
which, after some research and trial-and-error, Kernell was able to guess was Wasilla High; Palin grew up
in and was at one point mayor of Wasilla, Alaska. Much has been made since of the idea that the answers to
many security questions can be found on social-networking sites.
As a second example in this category, in 2010 Andrew weev Auernheimer and Daniel Spitler were charged
in the AT&T iPad hack. IPad owners who signed up for network service with AT&T had their iPads ICCID recorded along with their other information. If one of these owners later revisited the AT&T website,
the site would automatically request the iPads ICC-ID and then populate the web form with the users
information. If a randomly selected ICC-ID were presented to the AT&T site that happened to match a real
account, that users name, phone number and email address would be returned. ICC-ID strings contain 20
decimal digits, but the individual-device portion of the identifier is much smaller and this brute-force attack
yielded 114,000 accounts.
This attack is somewhat like a password intrusion, except that there was no support for running commands
via the compromised accounts.
Auernheimer was convicted for this intrusion in November 2012, but his sentence was set aside on appeal
in April 2014. Auernheimers conviction remains controversial as the AT&T site never requested a password
in the usual sense, though the site certainly released information not intended by its designers.

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Finally, the third category here includes any form of eavesdropping. If the password for a login-shell account
is obtained this way, a first-category attack may follow. The usual approach to prevention is the use of
encryption. Encryption is closely related to secure authentication; encryption and authentication are
addressed in this chapter in 21.5 Secure Hashes through 21.9 SSH and TLS.
Encryption does not always work as desired. In 2006 intruders made off with 40 million credit-card records
from TJX Corp by breaking the WEP Wi-Fi encryption (21.6.8 Wi-Fi WEP Encryption Failure) used by
the company, and thus gaining access to account credentials and to file servers. Albert Gonzalez pleaded
guilty to this intrusion in 2009. This was the largest retail credit-card breach until the Target hack of late
2013.

21.1 Code-Execution Intrusion

The most serious intrusions are usually those in which a vulnerability allows the attacker to run executable
code on the target system.
The classic computer virus is broadly of this form, though usually without a network vulnerability: the user
is tricked often involving some form of social engineering into running the attackers program on the target computer; the program then makes itself at home more or less permanently. In one form of this attack, the
user receives a file interesting_picture.jpg.exe or IRS_deficiency_notice.pdf.exe.
The attack is made slightly easier by the default setting in Windows of not displaying the final file extension
.exe.
Early viruses had to be completely self-contained, but, for networked targets, once an attacker is able to run
some small initial executable then that program can in turn download additional malware. The target can
also be further controlled via the network.
The reach of an executable-code intrusion may be limited by privileges on the target operating system; if
I am operating a browser on my computer as user pld and an intruder takes advantage of a flaw in that
browser, then the intruders code will also run as pld and not as root or Administrator. This may
prevent the intruder from rewriting my kernel, though that is small comfort to me if my files are encrypted
and held for ransom.
On servers, it is standard practice to run network services with the minimum privileges practical, though see
21.2.3 Defenses Against Buffer Overflows.
Exactly what is executable code is surprisingly hard to state. Scripting languages usually qualify. In 2000, the ILOVEYOU virus began spreading on Windows systems; users received a file
LOVE-LETTER.TXT.vbs (often with an enticing Subject: line such as love letter for you). The .vbs
extension, again not displayed by default, meant that when the file was opened it was automatically run as a
visual basic script. The ILOVEYOU virus was later attributed to Reonel Ramones and Onel de Guzman of
the Philippines, though they were never prosecuted. The year before, the Melissa virus spread as an emailed
Microsoft Word attachment; the executable component was a Word macro.
Under Windows, a number of configuration-file formats are effectively executable; among these are the
program-information-file format .PIF and the screen-saver format .SCR.

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21.1.1 The Morris Worm

The classic Morris Worm was launched on the infant Internet in 1988 by Robert Tappan Morris. Once one
machine was infected, it would launch attacks against other machines, either on the same LAN or far away.
The worm used a number of techniques, including stack buffer overflows (21.2 Stack Buffer Overflow). Two
of the worms techniques, however, had nothing to do with code injection. One worm module contained a
dictionary of popular passwords that were used to try against various likely system accounts. Another
module relied on a (broken) diagnostic feature of the sendmail email server. Someone could connect
to the sendmail TCP port 25 and send the command WIZ <password>; that person would then get
a shell and be able to execute arbitrary commands. It was the intent to require a legitimate sendmailspecific password, but an error in sendmails frozen-configuration-file processing meant that an empty
password often worked.

21.1.2 Christmas Day Attack

The 1994 Christmas day attack (12.9.1 ISNs and spoofing) used a TCP protocol weakness combined with
a common computer-trust arrangement to gain privileged login access to several computers at UCSD. The
rlogin trust arrangement meant that computer A might be configured to trust requests for remote-command
execution from computer B, often on the same subnet. But the ISN-spoofing attack meant that an attacker M
could send a command request to A that would appear to come from the trusted peer B, at least until it was
too late. The command might be as simple as open up a shell connection to M. At some point the spoofed
connection would fail, but by then the harmful command would have been executed. (Ironically, the ISN
spoofing attack was discovered by Morris but was not used in the Morris worm above; see [RTM85].)
Note that, as with the sendmail WIZ attack of the Morris worm, this attack did not involve network delivery
of an executable fragment (a shellcode).

21.2 Stack Buffer Overflow

The stack buffer overflow is perhaps the classic way for an attacker to execute a short piece of machine code
on a remote machine, thus compromising it. In this attack, a server application reads attacker-supplied data
into a buffer, buf, of length buflen. Due to a software flaw, however, the server reads more than buflen
bytes of data, and the additional data is written into memory past the end of buf, corrupting memory. In
the C language, there is no bounds checking with native arrays, and so such an overflow is not detected at
the time it occurs.
In most memory layouts, the stack grows downwards; that is, a function call creates a new stack frame with
a numerically lower address. Array indexing, however, grows upwards: buf[i+1] is at a higher address
than buf[i]. As a consequence, overwriting the buffer allows rewriting the most recent return address on
the stack. A common goal for the attacker is to supply an overflowing buffer that does two things:
1. it includes a shellcode - a small snippet of machine code that, when executed, does something bad
(traditionally but not necessarily by starting a shell with which the attacker can invoke arbitrary commands).
2. it overwrites the stack return address so that, when the current function exits, control is returned not
to the caller but to the supplied shellcode.

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In the diagram below, the left side shows the normal layout: the current stack frame has a buffer into which
the attackers message is read. When the current function exits, control is returned to the address stored in
return_address.

The right side shows the result after the attacker has written shellcode to the buffer, and, by overflow, has
also overwritten the return_address field on the stack so that it now points to the shellcode. When the
function exits, control will be passed to the code at return_address, that is, to the shellcode rather than
to the original caller. The shellcode is here shown below return_address in memory, but it can also be
above it.

21.2.1 Return to libc

A variant attack is for the attacker to skip the shellcode, but instead to overwrite the stack return address
with the address of a known library procedure. The libc library is popular here, making this known as
the return-to-libc approach. The goal of the attacker is to identify a library procedure that, in the current
context of the server process being attacked, will do something that the attacker finds useful.
One version of this attack is to overwrite the stack address with the address of the system() call, and to
place on the stack just above this return address a pointer to the string /bin/sh (often present in the environment strings of the attacked process). When the current function exits, control branches to system(),
which now thinks it has been called with parameter /bin/sh. A shell is launched.
Return-to-libc attacks often involve no shellcode at all. The attack of 21.3.2 A JPEG heap vulnerability
uses some return-to-libc elements but also does involve injected shellcode.
A practical problem with any form of the stack-overflow attack is knowing enough about the memory layout
of the target machine so that the attacker can determine exactly where the shellcode is loaded. This is
made simpler by standardized versions of Windows in which many library routines are at fixed, well-known
addresses.

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21.2.2 An Actual Stack-Overflow Example

Why the code here?

In many accounts of computer vulnerabilities, there is an understandable reluctance to explain the
actual mechanics, lest some attacker learn how to exploit the flaw. This secrecy, however, sometimes
has the unfortunate side-effect of making vulnerabilities seem almost magical. The goal in presenting
this twenty-year-old example in detail is simply to strip away the aura of mystery surrounding many
exploits.
To put together an actual example, we modify a version of TCP simplex-talk (12.5 TCP simplex-talk) written in the C language. On the server side, all we have to do is read the input with the infamous gets(char
* buf), which reads in data up to a newline character into the array buf, with no size restrictions. To do
this, we must arrange for the standard-input stream of the reading process to be the network connection; the
version of gets() that reads from an arbitrary input stream is fgets(char * buf, int bufsize,
FILE *stream); this is much safer as it will not read more than bufsize characters into buf, though
it is still up to the programmer to supply the correct value of bufsize.
Alternatively, we can also have the server its the data with the call
recv(int socket, char * buf, int bufsize, int flags)

but supply an incorrect (and much too large) value for the parameter bufsize. This approach has the
practical advantage of allowing the attacker to supply a buffer with NUL characters (zero bytes) and with
additional embedded newline characters.
On the client side the attackers side we need to come up with a suitable shellcode and create a too-large
buffer containing, in the correct place, the address of the start of the shellcode. Guessing this address used
to be easy, in the days when every process always started with the same virtual-memory address. It is now
much harder precisely to make this kind of buffer-overflow attack more difficult; we will cheat and have our
server print out an address on startup that we can then supply to the client.
An attack like this depends on knowing the target operating system, the target cpu architecture (so as to
provide an executable shellcode), the target memory layout, and something about the target server implementation (so the attacker knows what overflow to present, and when). Alas, all but the last of these are
readily guessable. Once upon a time vulnerabilities in server implementations were discovered by reading
source code, but it has long been possible to search for overflow opportunities making use only of the binary
executable code.
21.2.2.1 The server
The overflow-vulnerable server, oserver.c, is more-or-less a C implementation of the tcp simplex-talk server
of 12.5 TCP simplex-talk; note the explicit call to bind() which was handled by the ServerSocket()
constructor in the Java version. For each new connection, a new process to handle it is created with fork();
that new process then calls process_connection().
The process_connection() function then reads a line at a time from the client into a buffer pcbuf
of 80 bytes. Unfortunately for the server, it may read well more than 80 bytes into pcbuf.

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For the stack overflow to work, it is essential that the function that read in the oversized buffer thus corrupting the stack must return. Therefore the protocol has been modified so that process_connection()
returns if the arriving string begins with quit.
We must also be careful that the stack overflow does not so badly corrupt any local variables
that the code fails afterwards to continue running, even before returning. All local variables in
process_connection() are overwritten by the overflow, so we save the socket itself in the global
variable gsock.
We also call setstdinout(gsock) so that the standard input and standard output within
process_connection() is the network connection. This allows the use of gets(), which reads
from the standard input (alternatively, recv() with an incorrect value for bufsize may be used). It also
means that the shell launched by the shellcode will have its standard input and output correctly set up; we
could, of course, make the appropriate dup()/fcntl() calls from the shellcode, but that would increase
its complexity and size.
Because the standard output is now the TCP connection, the server prints incoming strings to the terminal
via the standard-error stream.
On startup, the server prints an address (that of mbuf[]) within its stack frame; we will refer to this as
mbuf_addr. The attacking client must know this value. No real server is so accommodating as to print its
internal addresses, but in the days before address randomization, 21.2.3.2 ASLR, the servers stack address
was typically easy to guess.
Whenever a connection is made, the server here also prints out the distance, in bytes, between the
start of mbuf[] in main() and the start of pcbuf the buffer that receives the overflow in
process_connection(). This latter number, which we will call addr_diff, is constant, and must
be compiled into the exploit program (it does change if new variables are added to the servers main() or
process_connection()). The actual address of pcbuf[] is thus mbuf_addr addr_diff. This will
be the address where the shellcode is located, and so is the address with which we want to overwrite the
stack. We return to this below in 21.2.2.3 The exploit, where we introduce a NOPslide so that the attacker
does not have to get the address of pcbuf[] exactly right.
Linux provides some protection against overflow attacks (21.2.3 Defenses Against Buffer Overflows), so the
server must disable these. As mentioned above, one protection, address randomization, we defeat by having
the server print a stack address. The server must also be compiled with the -fno-stack-protector
option to disable the stack canary of 21.2.3.1 Stack canary, and the -z execstack option to disable
making the stack non-executable, 21.2.3.3 Making the stack non-executable.
gcc -fno-stack-protector -z execstack -o oserver oserver.c

Even then we are dutifully warned by both the compiler and the linker:
warning: gets is deprecated ....
warning: the gets function is dangerous and should not be used.

In other words, getting this vulnerability still to work in 2014 takes a bit of effort.
The server here does work with the simplex-talk client of 12.5 TCP simplex-talk, but with the #USE_GETS
option enabled it does not handle a client exit gracefully.

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21.2.2.2 The shellcode

The shellcode must be matched to the operating system and to the cpu architecture; we will assume linux
running on a 32-bit Intel x86. Our goal is to create a shellcode that launches /bin/sh. The approach here
is taken from [SH04].
The shellcode must be a string of machine instructions that is completely self-contained; data references
cannot depend on the linker for address resolution. Not only must the code be position-independent, but the
code together with its data must be position-independent.
So-called Intel syntax is used here, in which the destination operand comes first, eg mov eax,37. The
nasm assembler is one of many that supports this format.
System calls in linux are easily made using interrupts, using the int 0x80 opcode and parameter. The x86
architecture has four general-purpose registers eax, ebx, ecx and edx; the first of these holds a code for
the particular system routine to be invoked and the others hold up to three parameters. (Below, in 21.3.2 A
JPEG heap vulnerability, we will also make use of register edi.) The system call we want to make is
execve(char * filename, char *argv[], char *envp[])

We need eax to contain the numeric value 11, the 32-bit-linux syscall value corresponding to execve
(perhaps defined in /usr/include/i386-linux-gnu/asm/unistd_32.h). (64-bit linux uses 59
as the syscall value for execve.) We load this with mov al 11; al is a shorthand for the low-order byte
of register eax. We first zero eax by subtracting it from itself. We can also, of course, use mov eax 11,
but then the 11 expands into four bytes 0x0b000000, and we want to avoid including NUL bytes in the code.
We also need ebx to point to the NUL-terminated string /bin/sh. The register ecx should point to an array of pointers [/bin/sh, 0] in memory (the null-terminated argv[]), and edx should point to a null word
in memory (the null-terminated envp[]). We include in the shellcode the string /bin/shNAAAABBBB,
and have the shellcode insert a NUL byte to replace the N and a zero word to replace the BBBB, as the
shellcode must contain no NUL bytes. The shellcode will also replace the AAAA with the address of
/bin/sh. We then load ecx with the address of AAAA (now containing the address of /bin/sh followed
by a zero word) and edx with the address of BBBB (now just a zero word).
Loading the address of a string is tricky in the x86 architecture. We want to calculate this address relative to
the current instruction pointer IP, but no direct access is provided to IP. The workaround in the code below
is to jump to shellstring near the end, but then invoke call start, where start: is the label for
our main code up at the top. The action of call start is to push the address of the byte following call
start onto the stack; this happens to be the address of shellstring:. Back up at start:, the pop
ebx pops this address off the stack and leaves it in ebx, where we want it.
Our complete shellcode is as follows (the actual code is in shellcode.s):
jmp short shellstring
start:
pop ebx
sub eax, eax
mov [ebx+7 ], al
mov [ebx+8 ], ebx
mov [ebx+12], eax
mov al, 11
lea ecx, [ebx+8]
lea edx, [ebx+12]

;get the address of the string in ebx

;zero eax by subtracting it from itself
;put a NUL byte where the N is in the string
;put the address of the string where the AAAA is
;put 4 null bytes into where the BBBB is
;execve is syscall 11
;load the address of where the AAAA was
;load the address of the NULLS;

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sub ecx,ecx
sub edx,edx
int 0x80
shellstring:
call start
db /bin/shNAAAABBBB

;call the kernel, WE HAVE A SHELL!

;pushes address of string below and jumps to start:
;the string. AAAA and BBBB get filled in as above

We run this through the commands

nasm -f elf shellcode.s
ld -o shellcode shellcode.o
objdump -d shellcode

and come up with the following string:

char shellcode[] =
"\xeb\x1a\x5b\x29\xc0\x88\x43\x07\x89\x5b\x08\x89\x43\x0c"
"\xb0\x0b\x8d\x4b\x08\x8d\x53\x0c\x29\xc9\x29\xd2\xcd\x80"
"\xe8\xe1\xff\xff\xff/bin/sh/NAAAABBBB";

We can test this with a simple C program defining the above and including
int main(int argc, char **argv) {
int (*func)();
func = (int (*)()) shellcode;
(int)(*func)();
}

We can verify that the resulting shell has not inherited the parent environment with the env and set
commands.
Additional shellcode examples can be found in [AHLR07].
21.2.2.3 The exploit
Now we can assemble the actual attack. We start with a C implementation of the simplex-talk client, and
add a feature so that when the input string is doit, the client
sends the oversized buffer containing the shellcode, terminated with a newline to make gets()
happy
sends quit, terminated with a newline, to force the servers process_connection() to return,
and thus to transfer control to the code pointed to by the now-overwritten return_address field
of the stack
begin a loop copylines() to copy the local terminals standard input to the network connection (hopefully now with a shell at the other end), and to copy the network connection to the local
terminals standard output
On startup, the client accepts a command-line parameter representing the address (in hex) of a variable close
to the start of the servers main() stack frame. When the server starts, it prints this address out; we simply
copy that when starting the client.
The full client code is in netsploit.c.
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All that remains is to describe the layout of the malicious oversized buffer, created by buildbadbuf().
We first compute our guess for the address of the start of the vulnerable buffer pcbuf in the servers
process_connection() function: we take the address passed in on the command line, which is actually the address of mbuf in the servers main(), and add to it the known constant (pcbuf - mbuf). This
latter value, 147 in the version tested by the author, is stored in netsploit.cs BUF_OFFSET.
This calculation of the address of the servers pcbuf should be spot-on; if we now store our shellcode at the
start of pcbuf and arrange for the server to jump to our calculated address, the shellcode should run. Real
life, however, is not always so accommodating, so we introduce a NOP slide: we precede our shellcode
with a run of NOP instructions. A jump anywhere into the NOPslide should lead right into the shellcode. In
our example here, we make the NOPslide 20 bytes long, and add a fudge factor of between 0 and 20 to our
calculated address (FUDGE is 10 in the actual code).
We need the attack buffer to be large enough that it overwrites the stack return-address field, but small
enough that the server does not incur a segmentation fault when overfilling its buffer. We settle on a
BADBUFSIZE of 161 (160 plus 1 byte for a final newline character); it should be comparable to but perhaps
slightly larger than the BUF_OFFSET value of, in our case, 147).
The attack buffer is now
20 bytes of NOPs
50 bytes of shellcode
90 bytes worth of the repeated calculated address (baseaddr-BUF_OFFSET+FUDGE in the code
1 byte newline, as the servers gets() expects a newline-terminated string
Here is a diagram like the one above, but labeled for this particular attack. Not all memory regions are drawn
to scale, and there are more addresses between the two stack frames than just return address.

We double-check the bad buffer to make sure it contains no NUL bytes and no other newline characters.

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If we wanted a longer NOPslide, we would have to hope there was room above the stacks return-address
field. In that case, the attack buffer would consist of a bunch of repeated address guesses, then the NOPslide,
and finally the shellcode.
After the command doit, the netsploit client prompt changes to 1>. We can then type ls, and, if the
shellcode has successfully started, we get back a list of files on the server. As earlier, we can also type env
and set to verify that the shell did not inherit its environment from any normal shell.

21.2.3 Defenses Against Buffer Overflows

How to prevent this sort of attack? The most basic approach is to make sure that array bounds are never
violated. Many languages enforce this automatically; in C it can be achieved through careful programming.
Another basic approach, applicable to almost all remote-code-execution attacks, is to make sure that the
server runs with the minimum permissions possible. The server may not have write permission to anything
of substance, and may in fact be run in a so-called chroot environment in which any access to the bulk of
the servers filesystem is disabled.
One issue with this approach is that a server process on a unix-derived system that wants to listen on a port
less than 1024 needs special privileges to open that port. Traditionally, the process would start with root
privileges and, once the socket was opened, would downgrade its privileges with calls to setgid() and
setuid(). This is safe in principle, but requires careful attention to the man pages; use of seteuid(),
for example, allows the shellcode to recover the original privileges. Linux systems now support assigning to
an unprivileged process any of several capabilities (see man capabilities). The one most relevant
here is CAP_NET_BIND_SERVICE, which, if set, allows a process to open privileged ports. Still, to assign
these capabilities still requires some privileged intervention when the process is started.
21.2.3.1 Stack canary
The underlying system can also provide some defenses. One of these, typically implemented within the
compiler, is the stack canary: an additional word on the stack near the return address that is set to a pseudorandom value. A copy of this word is saved elsewhere. When the function calls return (either explicitly
or implicitly), this word is checked to make sure the stack copy still agrees with the saved-elsewhere copy;
a discrepancy indicates that the stack was overwritten.
In the gcc compiler, a stack canary can be enabled with the compiler option -fstack-protector. To
compile the stack exploit in 21.2.2.3 The exploit, we needed to add -fno-stack-protector.
21.2.3.2 ASLR
Another operating-system-based defense is Address-Space Layout Randomization, or ASLR. In its simplest form, this means that each time the server process is restarted, the stack will have a different starting
address. For example, restarting the oserver program above five times yields addresses (in hex) of bf8d22b7,
bf84ed87, bf977d87, bfcc5cb7 and bfa302d7. There are at least four hex digits (16 bits) of entropy here; if
the server did not print its stack address it might take 216 guesses for the attacker to succeed.
Still, 216 guesses might take an attacker well under an hour. Worse, the attacker might simply create a
stack-buffer-overflow attack with a very long NOPslide; the longer the NOPslide the more room for error
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in guessing the shellcode address. With a NOPslide of length 210 = 1024 bits, guessing the correct stack
address will take only 26 = 64 tries. (Some implementations squeeze out 19 bits of address-space entropy,
meaning that guessing the correct address increases to 29 = 512 tries.)
For 64-bit systems, however, ASLR is much more effective. Brute-force guessing of the stack address takes
a prohibitively long time.
ALSR also changes the heap layout and the location of libraries each time the server process is restarted.
This is to prevent return-to-libc attacks, 21.2.1 Return to libc. For a concrete example of an attackers use
of non-randomized library addresses, see 21.3.2 A JPEG heap vulnerability.
On linux systems, ASLR can be disabled by writing a 0 to /proc/sys/kernel/randomize_va_space; values 1
and 2 correspond to partial and full randomization.
Windows systems since Vista (2007) have had ASLR support, though earlier versions of the linker required
the developer to request ASLR with the /DYNAMICBASE switch.
21.2.3.3 Making the stack non-executable
A more sophisticated idea, if the virtual-memory hardware supports it, is to mark those pages of memory
allocated to the stack as non-executable, meaning that if the processors instruction register is loaded with
an address on those pages (due to branching to stack-based shellcode), a hardware-level exception will
immediately occur. This immediately prevents attacks that place a shellcode on the stack, though return-tolibc attacks (21.2.1 Return to libc) are still possible.
In the x86 world, AMD introduced the per-page NX bit, for No eXecute, in their x86-64 architecture;
CPUs with this architecture began appearing in 2003. Intel followed with its XD, for eXecute Disabled,
bit. Essentially all x86-64 CPUs now provide hardware NX/XD support; support on 32-bit CPUs generally
requires so-called Physical Address Extension mode.
The NX idea applies to all memory pages, not just stack pages. This sometimes causes problems with
applications such as just-in-time compilation, where the code page is written and then immediately executed.
As a result, it is common to support NX-bit disabling in software. On linux systems, the compiler option
-z execstack disables NX-bit protection; this was used above in 21.2.2.1 The server. Windows has a
similar /NXCOMPAT option for requesting NX-bit protection.
While a non-executable stack prevents the stack-overflow attack described above, injecting shellcode onto
the heap is still potentially possible. The OpenBSD operating system introduced write or execute in 2003;
this is abbreviated W^X after the use of ^ as the XOR operator in C. A memory page may be writeable or
executable, but not both. This is strong protection against virtually all shellcode-injection attacks, but may
still be vulnerable to some return-to-libc attacks (21.2.1 Return to libc).
See [AHLR07], chapter 14, for some potential attack strategies against systems with non-executable pages.

21.3 Heap Buffer Overflow

As with stack overflows, heap overflows all rely on some software flaw that allows data to be written beyond
the confines of the designated buffer. A buffer on the heap is subject to the same software-failure overflow
prospects as a buffer on the stack. An important difference, however, is that buffers on the heap are not in

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clear proximity to an obvious return address. Despite that difference, heap overflows can also be used to
enable remote-code-execution attacks.
Perhaps the simplest heap overflow is to take advantage of the fact that some heap pages contain executable
code, representing application-loaded library pages. If the page with the overflowable buffer is pointed to
by p, and the following page in memory pointed to by q contains code, then all an attacker has to do is to
have the overflow fill the q page with a long NOPslide and a shellcode. When at some point a call is made
to the code pointed to by q, the shellcode is executed instead. A drawback to this attack is that the layout of
heap pages like this is seldom known. On the other hand, the fact that heap pages do sometimes legitimately
contain executable code means that uniformly marking the heap as non-executable, as is commonly done
with the stack, may not be an option.

21.3.1 A linux heap vulnerability

We now describe an actual linux heap vulnerability from 2003, following [AB03], based on version 2.2.4 of
the glibc library. The vulnerable server code is simply the following:
char *p = malloc(1024);
char *q = malloc(1024);
gets(p);
// read the attackers input
free(q);
// block q is last allocated and first freed
free(p);

As with the stack-overflow example, the gets(p) results in an overflow from block p into block q, overwriting not just the data in block q but also the block headers maintained by malloc(). While there is
no guarantee in general that block q will immediately follow block p in memory, in practice this usually
happens unless there has been a great deal of previous allocate/free activity.
The vulnerability here relies on some features of the 2003-era (glibc-2.2.4) malloc(). All blocks are
either allocated to the user or are free; free blocks are kept on a doubly linked list. We will assume the data
portion of each block is always a power of 2; there is a separate free-block list for each block size. When
two adjacent blocks are both free, malloc() coalesces them and moves the joined block to the free-block
list of the next size up.
All blocks are prefixed by a four-byte field containing the size, which we will assume here is the actual size
1032 including the header and alignment rather than the user-available size of 1024. As the three low-order
bits of the size are always zero, one of these bits is used as a flag to indicate whether the block is allocated
or free. Crucially, another bit is used to indicate whether the previous block is allocated or free.
In addition to the size-plus-flags field, the first two 32-bit words of a free block are the forward and backward
pointers linking the block into the doubly linked free-block list; the size-plus-flag field is also replicated as
the last word of the block:

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The strategy of the attacker, in brief, is to overwrite the p block in such a way that, when block q is freed,
malloc() thinks block p is also free, and so attempts to coalesce them, in the process unlinking block
p. But p was not in fact free, and the forward and backward pointers manipulated by malloc() as
part of the unlinking process are in fact provided by the attacker. As we shall see, this allows writing two
attacker-designated 32-bit words to two attacker-designated memory locations; if one of the locations holds
a return address and is updated so as to point to attacker-supplied shellcode also in block p, the system has
been compromised.
The normal operation of free(q), for an arbitrary block q, includes the following:
Get the block size (size) and flags at address q-4
Check the following block at address p+size to see if it is free; if so, merge (we are not interested
in this case)
Check the flags to see if the preceding block is free; if so, load its size prev_size from address
q-8, the address of the copy of size field in the diagram above; from that calculate a pointer to
the previous block as p = q - prev_size; then unlink block p (as the coalesced block will go
on a different free-block list).
For our specific block q, however, the attacker can overwrite the final size field of block p, prev_size
above, and can also overwrite the flag at address q-4 indicating whether or not block p is free. The attacker
can not overwrite the header of block p that would properly indicate that block p was still in use, but the
free() code did not double-check that.
We will suppose that the attacker has overwritten block p to include the following:
setting the previous-block-is-free flag in the header of block q to true
setting the final size field of block p to a desired value, badsize
placing value ADDR_F at address q-badsize; this is where free() will believe the previous
blocks forward pointer is located
placing the value ADDR_B at address q-badsize+4; this is where free() will believe the previous
blocks backward pointer is located
When the free(q) operation is now executed, the system will calculate the previous block as at address
p1 = q-badsize and attempt to unlink the false block p1. The normal unlink is
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(p->backward) -> forward = p->forward;

(p->forward) -> backward = p->backward;

Alas, when unlinking p1 the result is

= ADDR_F
*ADDR_B
*(ADDR_F + 4) = ADDR_B

where, for the pointer increment in the second line, we take the type of ADDR_F to be char * or void
*.
At this point the jig is pretty much up. If we take ADDR_B to be the location of a return address on the
stack, and ADDR_F to be the location of our shellcode, then the shellcode will be executed when the current
stack frame returns. Extending this to a working example still requires a fair bit of attention to details; see
[AB03].
One important detail is that the data we use to overwrite block p generally cannot contain NUL bytes,
and yet a small positive number for badsize will have several NUL bytes. One workaround is to have
badsize be a small negative number, meaning that the false previous-block pointer p1 will actually come
after q in memory.

21.3.2 A JPEG heap vulnerability

In 2004 Microsoft released vulnerability notice MS04-028 (and patch), relating to a heap buffer overflow
in Windows XP SP1. Microsoft had earlier provided a standard library for displaying JPEG images, known
as GDI for Graphics Device Interface. If a specially formed JPEG image were opened via the GDI library,
an overflow on the heap would occur that would launch a shellcode. While most browsers had their own,
unaffected, JPEG-rendering subroutines, it was often not difficult to convince users to open a JPEG file
supplied as an email attachment or via an html download link.
The problem occurred in the processing of the JPEG comment section, normally invisible to the user. The
section begins with the flag 0xFFFE, followed by a two-byte length field (in network byte order) that was to
include its own two bytes in the total length. During the image-file loading, 2 was subtracted from the length
to get the calculated length of the actual data. If the length field had contained 1, for example, the calculated
length would be -1, which would be interpreted as the unsigned value 232 - 1. The library, thinking it had
that amount of space, would then blithely attempt to write that many bytes of comment into the next heap
page, overflowing it. These bytes in fact would come from the image portion of the JPEG file; the attacker
would place here a NOPslide, some shellcode, and, as we shall see, a few other carefully constructed values.
As with the linux heap described above in 21.3.1 A linux heap vulnerability, blocks on the WinXP heap
formed a doubly-linked list, with forward and backward pointers known as flink and blink. As before,
the allocator will attempt to unlink a block via
blink -> forward = flink;
flink -> backward = blink;

The first of these simplifies to *blink = flink, as the offset to field forward is 0; this action allows
the attacker to write any word of memory (at address blink) with any desired value.
The JPEG-comment-overflow operation eventually runs out of heap and results in a segmentation fault, at
which point the heap manager attempts to allocate more blocks. However, the free list has already been
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overwritten, and so, as above, this block-allocation attempt instead executes *blink = flink.
The attackers conceptual goal is to have flink hold an instruction that branches to the shellcode, and
blink hold the address of an instruction that will soon be executed, or, equivalently, to have flink hold
the address of the shellcode and blink represent a location soon to be loaded and used as the target of a
branch. The catch is that the attacker doesnt exactly know where the heap is, so a variant of the return-tolibc approach described in 21.2.1 Return to libc is necessary. The strategy described in the remainder of
this section, described in [JA05], is one of several possible approaches.
In Windows XP SP1, location 0x77ed73b4 holds a pointer to the entry point of the Undefined
Exception Filter handler; if an otherwise-unhandled program exception occurs, Windows creates an
EXCEPTION_POINTERS structure and branches to the address stored here. It is this address, which we
will refer to as UEF, the attack will overwrite, by setting blink = UEF. A call to the Undefined Exception
Filter will be triggered by a suitable subsequent program crash.
When the exception occurs (typically by the second operation above, flink -> backward = blink),
before the branch to the address loaded from UEF, the EXCEPTION_POINTERS structure is created on the
heap, overwriting part of the JPEG comment buffer. The address of this structure is stored in register edi.
It turns out that, scattered among some standard libraries, there are half a dozen instructions at known
addresses of the form call DWORD [edi+0x74], that is, call the subroutine at 32-bit address edi +
0x74 ([AHLR07], p 186). All these call instructions are intended for contexts in which register edi
has been set up by immediately preceding instructions to point to something appropriate. In our attackers
context, however, edi points to the EXCEPTION_POINTERS structure; 0x74 bytes past edi is part of the
attackers overflowed JPEG buffer that is safely past this structure and will not have been overwritten by it.
One such call instruction happens to be at address 0x77d92a34, in user32.dll. This address is the
value the attacker will put in flink.
So now, when the exception comes, control branches to the address stored in UEF. This address now points
to the above call DWORD [edi+0x74], so control branches again, this time into the attacker-controlled
buffer. At this point, the processor lands on the NOPslide and ends up executing the shellcode (sometimes
one more short jump instruction is needed, depending on layout).
This attack depends on the fact that a specific instruction, call DWORD [edi+0x74], can be found at a
specific, fixed address, 0x77d92a34. Address-space layout randomization (21.2.3.2 ASLR) would have
prevented this; it was introduced by Microsoft in Windows Vista in 2007.

21.3.3 Cross-Site Scripting (XSS)

In its simplest form, cross-site scripting involves the attacker posting malicious javascript on a third-party
website that allows user-posted content; this javascript is then executed by the target computer when the
victim visits that website. The attacker might leave a comment on the website of the following form:
I agree with the previous poster completely
<script> do_something_bad() </script>

Unless the website in question does careful html filtering of what users upload, any other site visitor who so
much as views this comment will have the do_something_bad() script executed by his or her browser.
The script might email information about the target user to the attacker, or might attempt to exploit a browser
vulnerability on the target system in order to take it over completely. The script and its enclosing tags will
not appear in what the victim actually sees on the screen.
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The do_something_bad() code block will usually include javascript function definitions as well as
function calls.
In general, the attacker can achieve the same effect if the victim visits the attackers website. However, the
popularity (and apparent safety) of the third-party website is usually important in practice; it is also common
for the attack to involve obtaining private information from the victims account on that website.

21.3.4 SQL Injection

SQL is the close-to-universal query language for databases; in a SQL-injection attack the attacker places a
carefully chosen SQL fragment into a website form, in such a way that it gets executed. Websites typically
construct SQL queries based on form data; the attackers goal is to have his or her data treated as additional
SQL. This is supposed to be prevented by careful quoting, but quoting often ends up not quite careful
enough. A successful attacker has managed to run SQL code on the server that gives him or her unintended
privileges or information.
As an example, suppose that the form has two fields, username and password. The system then runs
the following sub-query that returns an empty set of records if the username,password pair is not found;
otherwise the user is considered to be authenticated:
select * from PASSWORDS p
where p.user = username and p.password = password;
The strings username and password are taken from the web form and spliced in; note that each is enclosed
in single quotation marks supplied by the server. The attackers goal is to supply username/password values
so that a nonempty set of records will be returned, thus authenticating the attacker. The following values are
successful here, where the quotation marks in username are supplied by the attacker:
username: OR 1=1 OR x=y
password: foo
The spliced-together query built by the server is now
select * from PASSWORDS p
where p.user = OR 1=1 OR x=y and p.password = foo;
Note that of the eight single-quote marks in the where-clause, four (the first, sixth, seventh and eighth) came
from the server, and four (the second through fifth) came from the attacker.
The where-clause here appears to SQL to be the disjunction of three OR clauses (the last of which is
x=y and p.password = foo). The middle OR clause is 1=1 which is always true. Therefore, the login succeeds.
For this attack to work, the attacker must have a pretty good idea what query the server builds from the
user input. There are two things working in the attackers favor here: first, these queries are often relatively
standard, and second, the attacker can often discover a great deal from the error messages returned for
malformed entries. In many cases, these error messages even reveal the table names used.
See also xkcd.com/327.

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21.4 Network Intrusion Detection

The idea behind network intrusion detection is to monitor ones network for signs of attack. Many newer
network intrusion-detection systems (NIDS) also attempt to halt the attack, but the importance of simple
monitoring and reporting should not be underestimated. Many attacks (such as password guessing, or buffer
overflows in the presence of ASLR) take multiple (thousands or millions) of tries to succeed, and a NIDS
can give fair warning.
There are also host-based intrusion-detection systems (HIDS) that run on and monitor a specific host; we
will not consider these further.
Most NIDS detect intrusions based either on traffic anomalies or on pattern-based signatures. As an
example of the former, a few pings (7.9 Internet Control Message Protocol) might be acceptable but a large
number, or a modest number addressed to nonexistent hosts, might be cause for concern.
As for signatures, the attack in 21.3.2 A JPEG heap vulnerability can be identified by the hex strings
0xFFFE0000 or 0xFFFE0001. What about the attack in 21.2.2.3 The exploit? Using the shellcode itself as
signature tends to be ineffective as shellcode is easy to vary. The NOPslide, too, can be replaced with a wide
variety of other instructions that just happen to do nothing in the present context, such as sub eax,eax.
One of the most common signatures used by NIDSs for overflow attacks is simply the presence of overly
long strings; the false-positive rate is relatively low. In general, however, coming up with sufficiently specific
signatures can be nontrivial. An attack that keeps changing in relatively trivial ways to avoid signature-based
detection is sometimes said to be polymorphic.

21.4.1 Evasion
The NIDS will have to reassemble TCP streams (and sometimes sequences of UDP packets) in order to
match signatures. This raises the possibility of evasion: the attacker might arrange the packets so that the
NIDS reassembles them one way and the target another way. The possibilities for evasion are explored in
great detail in [PN98]; see also [SP03].
One simple way to do this is with overlapping TCP segments. What happens if one packet contains bytes
1-6 of a TCP connection as help and a second packet contains bytes 2-7 as armful?
h

e
a

l
r

p
m

These can be reassembled as either helpful or harmful; the TCP specification does not say which is preferred and different operating systems routinely reassemble these in different ways. If the NIDS reassembles
the packets one way, and the target the other, the attack goes undetected. If the attack is spread over multiple
packets, there may be many more than two ways that reassembly can occur, increasing the probability that
the NIDS and the target will differ.
Another possibility is that one of the overlapping segments has a header irregularity (in either the IP or TCP
header) that causes it to be rejected by the target but not by the NIDS, or vice-versa. If the packets are
h
h

e
a

l
r

p
m

and both systems normally prefer data from the first segment received, then both would reassemble as
helpful. But if the first packet is rejected by the target due to a header flaw, then the target receives
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harmful. If the flaw is not recognized by the NIDS, the NIDS does not detect the problem.
A very similar problem occurs with IPv4 fragment reassembly, although IPv4 fragments are at this point
intrinsically suspicious.
One approach to preventing evasion is to configure the NIDS with information about how the actual target
does its reassembly, so the NIDS can match it. An even safer approach is to have the NIDS reassemble any
overlapping packets and then forward the result on to the potential target.

21.5 Secure Hashes

How can we tell if a file has been changed? One way is to create a record of each files checksum, using,
for the sake of argument, the Internet checksum of 5.4 Error Detection. If the potential file corruption is
random, this will fail to detect a modified file with probability only 1 in 216 .
Alas, if the modification is intentional, it is trivial to modify the file so that it has the same checksum as
the original. If the original file has checksum c1 , and after the initial modification the checksum is now c2 ,
then all we have to do to create a file with checksum c1 is to append the 16-bit quantity c2 c1 (where the
subtraction is done using ones-complement; we need to modify this slightly if the number of bytes in the
first-draft modified file is odd). The CRC family of checksums is almost as easily tricked.
The goal of a cryptographically secure hash is to provide a hash function we will not call it a checksum
as hashes based on simple sums all share the insecurity above for which this trick is well-nigh impossible.
Specifically, we want a hash function such that:
Knowing the hash value should shed no practical light on the message
Given a hash value, there should be no feasible way to find a message yielding that hash
A slightly simpler problem than the second one above is to find two messages that have the same hash; this
is sometimes called the collision problem. When the collision problem for a hash function has been solved,
it is (high) time to abandon it as no longer secure.
If a single bit of the input is changed, the secure-hash-function output is usually entirely different.
Hashes popular in the 1990s were the 128-bit MD5 (RFC 1321, based on MD4, [RR91]) and the 160bit SHA-1 (NSA); SNMPv3 originally supported both of these (20.15.2 Cryptographic Fundamentals).
MD5 collisions (two messages hashing to the same value) were reported in 2004, and collisions where both
messages were meaningful were reported in 2005; such collision attacks mean its use for security is no
longer appropriate.
Hash functions currently (2014) believed secure include the SHA-2 family, which includes variants ranging
from 224 bits to 512 bits.
A common framework for constructing n-bit secure-hash functions is the Merkle-Dmgard construction
([RM88], [ID89]); it is used by MD5, SHA-1 and SHA-2. The initial n-bit state is specified. To this is
then applied a set of transformations Hi (x,y) that each take an n-bit block x and some additional bits y and
return an updated n-bit block. These transformations are usually similar to the rounds functions of a block
cipher; see 21.6.3 Block Ciphers for a typical example. In encryption, the parameter y would be used to
hold the key, or a substring of the key; in secure-hash functions, the parameter y holds a substring of the
input message. The set of transformations is applied repeatedly as the process iterates over the entire input
message; the result of the hash is the final n-bit state.
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In the MD5 hash function, the input is processed in blocks of 64 bytes. Each block is divided into sixteen
32-bit words. One such block of input results in 64 iterations from a set of sixteen rounds-functions Hi , each
applied four times in all. Each 32-bit input word is used as the key parameter to one of the Hi four times.
If the total input length is not a multiple of 512 bits, it is padded; the padding includes a length attribute so
two messages differing only by the amount of padding should not hash to the same value.
While this framework in general is believed secure, and is also used by the SHA-2 family, it does suffer
from what is sometimes called the length-extension vulnerability. If h = hash(m), then the value h is simply
the final state after the above mechanism has processed message m. An attacker knowing only h can then
initialize the above algorithm with h, and continue it to find the hash h = hash(m m ), for an arbitrary
message m concatenated to the end of m, without knowing m. If the original message m was padded to
message mp , then the attacker will find h = hash(mp m ), but that is often enough. This vulnerability must
be considered when using secure-hash functions for message authentication, next.
The SHA-3 family of hash functions does not use the Merkle-Dmgard construction and is believed not
vulnerable to length-extension attacks.

21.5.1 Secure Hashes and Authentication

Secure hash functions can also be used to implement message authentication. Suppose the sender and
receiver share a secret key, K, a random string of length comparable to the output of the hash. Then, in
principle, the sender can append to the message m the value h = hash(K m). The receiver, knowing K, can
recalculate this value and verify that the h appended to the message is correct. In theory, only someone who
knew K could calculate h.
Unfortunately for this particular strategy, it is undermined by the length-extension attack of the previous
section. If the sender transmits message m together with hash(K m), then an attacker can modify this to
message m m together with hash(K m m ), without knowing K.
The length-extension vulnerability is defeated by reversing the order and using hash(m K), but this now
introduces potential collision vulnerabilities: if two messages m1 and m2 hash to the same value, then so
will m1 K and m2 K.
Taking these vulnerabilities into account, RFC 2104 defines the Hash Message Authentication Code, or
HMAC, as follows; it can be used with any secure-hash function but is not necessary with SHA-3 and
other hash functions not sharing the length-extension vulnerability. The 64-byte length here comes from the
typical input-block length of many secure-hash functions, including the SHA-2 family except for the 512-bit
option; it may be increased as necessary.
The shared key K is extended to 64 bytes by padding with zeroes.
A constant string ipad is formed by repeating the octet 0x36 (0011 0110) 64 times.
A constant string opad is formed by repeating 0x5c (0101 1100) 64 times.
We set K1 = K XOR ipad.
We set K2 = K XOR opad.
Finally, the HMAC value is

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HMAC = hash(K2 hash(K1 mesg))

The values 0x36 (0011 0110) and 0x5c (0101 1100) are not critical, but the XOR of them has, by intent, a
reasonable number of both 0-bits and 1-bits; see [BCK96].
The HMAC algorithm is, somewhat surprisingly, believed to be secure even when the underlying hash
function is vulnerable to some kinds of collision attacks; see [MB06] and RFC 6151. That said, a hash
function vulnerable to collision attacks may have other vulnerabilities as well, and so HMAC-MD5 should
still be phased out.

21.5.2 Password Hashes

A site generally does not store user passwords directly; instead, a hash of the password, h(p), is stored.
The cryptographic hash functions above, eg MD5 and SHA-n, work well as long as the password file itself
is not compromised. However, these functions all execute very quickly by design and so an attacker
who succeeds in obtaining the password file can usually extract passwords simply by calculating the hash of
every possible password. There are about 214 six-letter English words, and so there are about 238 passwords
consisting of two such words and three digits. Such passwords are usually considered rather strong, but
brute-force calculation of h(p) for 238 possible values of p is quite feasible for the hashes considered so far.
Checking 108 MD5-hashed passwords per second is quite feasible; this means 238 = 256 230 256 109
passwords can be checked in under 45 minutes. Use of a GPU increases the speed at least 10-fold.
Special hashes have been developed to address this; two well-known ones are scrypt ([CP09] and this Internet Draft) and bcrypt. The goal here is to develop a hash that takes ideally the better part of a second
to calculate even once, and which cannot easily be sped up by large precomputed tables. Typically these
are something like a thousand to a million times slower than conventional hash functions like MD5 and the
SHA-2 family; a millionfold-slower hash function is the equivalent of making the passwords longer by 20
random bits (106 220 ). See also RFC 2898, section 4.2, though the proposal there is only a thousandfold
slower.
(Good password tables also incorporate a salt, or nonce: a random string s is chosen and appended to the
password before hashing; the password record is stored as the pair s,h(p s). This means that cracking the
password for one user will not expose a second user who has chosen exactly the same password, so long as
the second user has a different salt s. Salting, however, does not change the scenarios outlined above.)

21.6 Shared-Key Encryption

Secure hashes can provide authentication, but to prevent eavesdropping we need encryption. While publickey encryption (21.8 Public-Key Encryption) is perhaps more glamorous, the workhorse of the encryption
world is the shared-key cipher, or shared-secret or symmetric cipher, in which each party has possession
of a key K, used for both encryption and decryption.
Shared-key ciphers are often quite fast; public-key encryption is generally slower by several orders of magnitude. As a result, if two parties without a shared key wish to communicate, they will almost always used
public-key encryption only to exchange a key for a shared-key cipher, which will then be used for the actual
message encryption.

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Typical key lengths for shared-key ciphers believed secure range from 128 bits to 256 bits. Recent keylength recommendations for public-key RSA encryption are 2048 bits. The difference here is because for
most shared-key ciphers there are no known attacks that are much faster than brute force, and 2256 1077
is quite a large number. For most public-key encryption mechanisms, on the other hand, there are shortcuts;
for RSA an attacker can use a factoring shortcut.
Shared-key ciphers can be either block ciphers, encrypting data in units of blocks that might typically be 8
bytes long, or stream ciphers, which generate a pseudorandom keystream. Each byte (or even bit) of the
message is then encrypted by (typically) XORing it with the corresponding byte of the keystream.

21.6.1 Alice and Bob

Cryptographers usually use Alice and Bob, instead of A and B, as the names of the parties sending each
other encrypted messages. This tradition dates back at least to [RSA78]. Other parties sometimes include
Eve the eavesdropper and Mallory the active attacker (and launcher of man-in-the-middle attacks). (In this
article the names Aodh and Bea are introduced, though not specifically for cryptography; the Irish name
Aodh is pronounced, roughly, as Eh.)

21.6.2 Session Keys

The more messages a key is used to encrypt, the more information a potential eavesdropper may have with
which to launch a codebreaking attack. If Alice and Bob have a shared secret key K, is therefore quite
common for them to encrypt the bulk of their correspondence with temporary session keys, each one used
for a time period that may range from minutes to days. The secret key K might then be used only to exchange
new session keys and to forestall man-in-the-middle attacks (21.8.3 Trust and the Man in the Middle) by
signing important messages (21.5.1 Secure Hashes and Authentication). If Diffie-Hellman-Merkle key
exchange (ref:diffie-hellman) is used, K might be reserved only for message signing.
Sometimes session keys are entirely different keys, chosen randomly; if such a key is discovered, the attacker
learns nothing about the secret key K. Other times, the session key is a mash-up of K and some additional
information (such as an initialization vector, 21.6.4 Cipher Modes), that is transmitted in the clear. The
session key might then be the concatenation of K and the initialization vector, or the XOR of the two. Either
approach means that an eavesdropper who figures out the session key also has the secret key K, but the use
of such session keys still may make any codebreaking attempt much more difficult.
In 21.8.2 Forward Secrecy we consider the reverse problem of how Alice and Bob might keep a session
key private even if the secret key is compromised.

21.6.3 Block Ciphers

As mentioned, a block cipher encrypts data one block at a time, typically with a key length rather longer
than the block size. A typical block cipher proceeds by the iterated application of a sequence of round
functions, each updating the block and using some round-dependent substring of the key as auxiliary input.
If we start with a block of plaintext P and apply all the round functions in turn, the result is the ciphertext
block C = E(P,K) = EK (P). The process can be reversed so that P = D(C,K) = DK (C).
A common framework is that of the Feistel network. In such an arrangement, the block is divided into two
or more words sized for the machine architecture; an 8-byte block is typically divided into two 32-bit words
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which we can call L and H for Low and High. A typical round function is now of the following form, where
K is the key and F(x,K) takes a word x and the key K and returns a new word:
L,H H, L XOR F(H,K)
Visually, this is often diagrammed as

One round here scrambles only half the block, but the other half gets scrambled in the next round (sometimes the operation above is called a half-round for that reason). The total number of rounds is typically
somewhere between 10 and 50. Much of the art of designing high-quality block ciphers is to come up with
round functions that result in overall very fast execution, without introducing vulnerabilities. The more
rounds, the slower.
The internal function F, often different for each round, may look at only a designated subset of the bits of
the key K. Note that the operation above is invertible that is, can be decrypted regardless of F; given the
right-hand side the receiver can compute F(H,K) and thus, by XORing this with L XOR F(H,K), can recover
L. This remains true if, as is sometimes the case, the XOR operation is replaced with ordinary addition.
Crypto Law
The Salsa20 cipher mentioned here is a member of Daniel Bernsteins snuffle family of ciphers based
on secure-hash functions. In the previous century, the US government banned the export of ciphers but
not secure-hash functions. They also at one point banned the export (and thus publication) of one of
Bernsteins papers; he sued. In 1999, the US Court of Appeals for the Ninth Circuit found in his favor.
The decision, 176 F.3d 1132, is the only appellate decision of which the author is aware that contains
(in the footnotes) not only samples of C code, but also of Lisp.
A simple F might return the result of XORing H and a subset of the bits of K. This is usually a little too
simple, however. Ordinary modulo-32 addition of H and a subset of K often works well; the interaction of
addition and XORing introduces considerable bit-shuffling (or diffusion in the language of cryptography).
Other operations used in F(H,K) include Boolean And and Or operations. 32-bit multiplication introduces
considerable bit-shuffling, but is often computationally more expensive. The Salsa20 cipher of [DB08] uses
only XOR and addition, for speed.
It is not uncommon for the round function also to incorporate bit rotation of one or both of L and H; the

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result of bit-rotation of 1000 1110 two places to the left is 0011 1010.
If a larger blocksize is desired, say 128 bits, but we still want to avail ourselves of the efficiency of 32-bit
operations, the block can be divided into A,B,C,D. The round function might then become
A,B,C,D B,C,D, (A XOR F(B,C,D,K))
As mentioned above, many secure-hash functions use block-cipher round functions that then use chunks of
the message being hashed as the key. In the MD5 algorithm, block A above is transformed into the 32bit sum of input-message fragment M, a constant K, and G(B,C,D) which can be any of several Boolean
combinations of B, C and D.
An alternative framework for block ciphers is the so-called substitution-permutation network.
The first block cipher in widespread use was the federally sanctioned Data Encryption Standard, or DES
(commonly pronounced dez). It was developed at IBM by 1974 and then selected by the US National
Bureau of Standards (NBS) as a US standard after some alterations recommended by the National Security
Agency. One of the NSAs recommendations was that a key size of 56 bits was sufficient; this was in an
era when the US government was very concerned about the spread of strong encryption. For years, many
people assumed the NSA had intentionally introduced other weaknesses in DES to facilitate government
eavesdropping, but after forty years no such vulnerability has been discovered and this no longer seems so
likely. The suspicion that the NSA had in the 1970s the resources to launch brute-force attacks against
DES, however, has greater credence.
In 1997 an academic team launched a successful brute-force attack on DES. The following year the Electronic Frontier Foundation (EFF) built a hardware DES cracker for about US$250,000 that could break DES
in a few days.
In an attempt to improve the security of DES, triple-DES or 3DES was introduced. It did not become an
official standard until the late 1990s, but a two-key form was proposed in 1978. 3DES involves three
applications of DES with keys K1,K2,K3; encrypting a plaintext P to ciphertext C is done by C =
EK3 (DK2 (EK1 (P))). The middle deciphering option DK2 means the algorithm reduces to DES when K1 =
K2 = K3; it also reduces exposure to a particular vulnerability known as meet in the middle (no relation to man in the middle). In [MH81] it is estimated that 3DES with three distinct keys has a strength
roughly equivalent to 256 = 112 bits. That same paper also uses the meet-in-the-middle attack to show
that straightforward double-DES encryption C = EK2 (EK1 (P)) has an effective keystrength of only 56 bits
no better than single DES if sufficient quantities of plaintext and corresponding ciphertext are known.
As concerns about the security of DES continued to mount, the US National Institute of Standards and
Technology (NIST, the new name for the NBS) began a search for a replacement. The result was the
Advanced Encryption Standard, AES, officially approved in 2001. AES works with key lengths of 128, 192
and 256 bits. The algorithm is described in [DR02], and is based on the Rijndael family of ciphers; the name
Rijndael (rain-dahl) is a combination of the authors names.
Earlier non-DES ciphers include IDEA, the International Data Encryption Algorithm, described in [LM91],
and Blowfish, described in [BS93]. Both use 128-bit keys. The IDEA algorithm was patented; Blowfish
was intentionally not patented. A successor to Blowfish is Twofish.

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21.6.4 Cipher Modes

The simplest encryption mode for a block cipher is to encrypt each input block independently. That is, if
Pi is the ith plaintext block, then the ith ciphertext block Ci is E(Pi ,K). This is known as electronic codebook
or ECB mode.
ECB is vulnerable to known-plaintext attacks. Suppose the attacker knows that the message is of the following form, where the vertical lines are the block boundaries:
Bank Transfer

to MALLORY

Amount $1000

If the attacker also knows the ciphertext for Amount $100,000, and is in a position to rewrite
the message (or to inject a new message), then the attacker can combine the first two blocks above
with the third $100,000 block to create a rather different message, without knowing the key. At
http://en.wikipedia.org/wiki/Block_cipher_mode_of_operation there is a remarkable example of the failure
of ECB to fail to effectively conceal an encrypted image.
As a result, ECB is seldom used. A common alternative is cipher block chaining or CBC mode. In this
mode, each plaintext block is XORed with the previous ciphertext block before encrypting:
Ci = E(K,Ci-1 XOR Pi )
To decrypt, we use
Pi = D(K,Ci ) XOR Ci-1
If we stop here, this means that if two messages begin with several identical plaintext blocks, the encrypted
messages will also begin with identical blocks. To prevent this, the first ciphertext block C0 is a random
string, known as the initialization vector, or IV. The plaintext is then taken to start with block P1 . The IV
is sent in the clear, but contains no private information.
See exercise 5.

21.6.5 Stream Ciphers

Conceptually, a stream cipher encodes one byte at a time. The cipher generates a pseudo-random keystream.
If Ki is the ith byte of the keystream, then the ith plaintext byte Pi is encrypted as Ci = Pi XOR Ki . A stream
cipher can be viewed as a special form of pseudo-random number generator or PRNG, though PRNGs used
for numeric methods and simulations are seldom secure enough for cryptography.
Simple XORing of the plaintext with the keystream may not seem secure, but if the keystream is in fact
truly random then this cipher is unbreakable: to an attacker, all possible plaintexts are equally likely. A truly
random keystream means that the entire keystream must be shared ahead of time, and no portion may ever
be reused; this cipher is known as the one-time pad.
Stream ciphers do not provide authenticity; an attacker may have XORed something with the encrypted
message to change it. For example, if the message is known to be
Transfer $ 02000 to Mallory
^^

then the attacker can XOR the two bytes over the ^ with 0 XOR 2, changing the character 0 to a 2
and the 2 to a 0. The attacker does this to the encrypted stream, but the decrypted plaintext stream retains
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the change. Appending an authentication code such as HMAC, 21.5.1 Secure Hashes and Authentication,
prevents this.
Stream ciphers are, in theory, well suited to the encryption of data sent a single byte at a time, such as data
from a telnet session. The ssh protocol (21.9.1 SSH), however, generally uses block ciphers; when it has to
send a single byte it pads the block with random data.
21.6.5.1 RC4
The RC4 stream cipher is quite widely used. It was developed by Ron Rivest at RSA Security in 1987, but
never formally published. The code was leaked, however, and so the internal details are widely available.
Unofficial implementations are sometimes called ARC4 or ARCFOUR, the A for Alleged.
RC4 was designed for very fast implementation in software; to this end, all operations are on whole bytes.
RC4 generates a keystream of pseudorandom bytes, each of which is XORed with the corresponding plaintext byte. The keystream is completely independent of the plaintext.
The key length can range from 5 up to 256 bytes. The unofficial protocol contains no explicit mechanism
for incorporating an initialization vector, but an IV is well-nigh essential for stream ciphers; otherwise an
identical keystream is generated each time. One simple approach is to create a session key by concatenating
the IV and the secret RC4 key; alternatively (and perhaps more safely) the IV and the RC4 key can be
XORed together.
RC4 was the basis for the ill-fated WEP Wi-Fi encryption, 21.6.8 Wi-Fi WEP Encryption Failure, due in
part to WEPs requirement that the 3-byte IV precede the 5-byte RC4 key. The flaw there did not affect
other applications of RC4, but newer attacks have suggested that RC4 be phased out.
Internally, an RC4 implementation maintains the following state variables:
An array S[] representing a permutation iS[i] of all bytes
two 8-bit integer indexes to S[], i and j
S[] is guaranteed to represent a permutation (ie is guaranteed to be a 1-1 map) because it is initialized to the
identity and then all updates are transpositions (involving swapping S[i] and S[j]).
Initially, we set S[i] = i for all i, and then introduce 256 transpositions. This is known as the key-scheduling
algorithm. In the following, keylength is the length of the key key[] in bytes.
J=0;
for I=0 to 255:
J = J + S[I] + key[I mod keylength];
swap S[I] and S[J]

As we will see in 21.6.8 Wi-Fi WEP Encryption Failure, 256 transpositions is apparently not enough.
After initialization, I and J are reset to 0. Then, for each keystream byte needed, the following is executed,
where I and J retain their values between calls:
I = (I+1) mod 256
J = J + S[I] mod 256
swap S[I] and S[J]
return S[ S[I] + S[J] mod 256 ]

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For I = 1, the first byte returned is S[ S[1] + S[S[1]] ].

21.6.6 Block-cipher-based stream ciphers

It is possible to create a stream cipher out of a block cipher. The first technique we will consider is counter
mode, or CTR. The sender and receiver agree on an initial block, B, very similar to an initialization vector.
The first block of the keystream, K0 , is simply the encrypted block E(B,K). The ith keystream block, for
i>0, is Ki = E(B+i,K), where B+i is the result of treating B as a long number and performing ordinary
arithmetic.
Going from B+1 to B+(i+1) typically changes only one or two bits, but no significant vulnerability based
on this has ever been found, and this form of counter mode is now widely believed to be as secure as the
underlying block cipher.
A related technique is to generate the keystream by starting with a secret key K and taking a secure hash
of each of the successive concatenations K 1, K 2, etc. The drawback to this approach is that many
secure-hash functions have not been developed with this use in mind, and unforeseen vulnerabilities may
exist.
The keystream calculated by counter mode is completely independent of the plaintext. This is not always
the case. In cipher feedback mode, or CFB, we initialize C0 to the initialization vector, and then define the
keystream blocks Ki for i>0 as follows:
Ki = EK (Ci-1 )
As usual, Ci = Pi XOR Ki for i>0, so the right-hand side above is EK (Pi-1 XOR Ki-1 ). The evolution of the
keystream thus depends on earlier plaintext. Note that, by the time we need to use Ki , the earlier plaintext
Pi-1 has been entirely received.
Stream ciphers face an error-recovery problem, but we will ignore that here as we will assume the transport
layer provides sufficient assurances that the ciphertext is received accurately.

21.6.7 Encryption and Authentication

If the ciphertext is modified by an attacker in transit, most decryption algorithms will happily decrypt it
anyway, likely to gibberish. It is common practice to include HMAC authentication signatures (21.5.1 Secure Hashes and Authentication) with each encrypted message; the use of HMAC rather than a simple
checksum ensures that an attacker cannot modify messages and have them still appear to be valid.
Appending an HMAC signature does more than prevent garbled messages; there are attacks that, in the
absence of such signatures, allow full decryption of messages (particularly if the attacker can create multiple
ciphertexts and find out which ones the recipient was able to decrypt successfully). See [SV02] for an
example.
There are three options for combining encryption with HMAC; for consistency with the literature we replace
HMAC with the more general MAC (for Message Authentication Code):
MAC-then-encrypt: calculate the MAC signature on the plaintext, append it, and encrypt the whole
encrypt-and-MAC: calculate the MAC signature on the plaintext, encrypt the message, and append to
that the MAC

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encrypt-then-MAC: encrypt the plaintext and calculate the MAC of the ciphertext; append the MAC
to the ciphertext
These are analyzed in [BN00], in which it is proven that encrypt-then-MAC in general satisfies some stronger
cryptographic properties than the others, although these properties may hold for the others in special cases.
Encrypt-then-MAC means that no decryption is attempted of a forged or spoofed message.

21.6.8 Wi-Fi WEP Encryption Failure

The WEP (Wired-Equivalent Privacy) mechanism was the first Wi-Fi encryption option, introduced in 1999;
see 3.3.6 Wi-Fi Security. It used RC4 encryption with either a 5-byte or 13-byte secret key; this was always
appended to a 3-byte initialization vector that, of necessity, was sent in the clear. The RC4 session key was
thus either 8 or 16 bytes; the shorter key was more common.
There is nothing inherently wrong with that strategy, but the designers of WEP made some design choices
that were, in retrospect, infelicitous:
the secret key was appended to the IV (rather than, say, XORed with it)
the IV was only three bytes long
a new RC4 keystream and new IV was used for each packet
The third choice above was perhaps the more serious mistake. One justification for this choice, however,
was that otherwise lost packets (which are common in Wi-Fi) would represent gaps in the received stream,
and the receiver might have trouble figuring out the size of the gap and thus how to decrypt the later arrivals.
A contributing issue is that the first byte of the plaintext a header byte from the Wi-Fi packet is essentially
known. This first byte is generally from the Logical Link Control header, and contains the same value as
the Ethernet Type field (see 3.3.4 Wi-Fi Configurations). All IP packets share the same value. Thus, by
looking at the first byte of the encrypted packet, an attacker knows the first byte of the keystream.
A theoretical WEP vulnerability was published in [FMS01] that proved all-too-practical in the real world.
We will let K[] denote the 8-byte key, with K[0],K[1],K[2] representing the three-byte initialization vector
and K[i] for 3i<8 representing the secret key. The attacker monitors the airwaves for IVs of a particular
form; after about 60 of these are collected, the attacker can make an excellent guess at K[3]. Now a slightly
different group of IVs is collected, allowing a guess at K[4] and so on. The bytes of the secret key thus fail
sequentially. The attack requires about the same (modest) time for each key byte discovered, so a 16-byte
total key length does not add much more security versus the 8-byte key.
Because the IV is only three bytes, the time needed to collect the special values necessary is modest.
We outline the attack on the first secret byte, K[3]. The IVs we are looking for have the form
3,-1,X
where -1 = 255 for 8-bit bytes; these are sometimes called weak IVs. One IV in 216 is of this form, but a
more careful analysis (which we omit) can increase the usable fraction of IVs substantially.
To see why these IVs are useful, we need to go back to the RC4 key-scheduling algorithm, 21.6.5.1 RC4.
We start with an example in which the first 6 bytes of the key is
K[] = [3,-1,5,4,-14,3]

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The IV here is 3,-1,5.

Recall that the array S is initialized with S[i] = i; this represents S = S0 . At this point, I and J are also 0. We
now run the following loop, introducing one transposition to S per iteration:
for I=0 to 255:
J = J + S[I] + K[I mod keylength];
swap S[I] and S[J]

The first value of J, when I = 0, is K[0] which is 3. After the first transposition, S is as follows, where the
swapped values are in bold:

For the next iteration, I is 1 and J is 3+1+(-1) = 3 again. After the swap, S is

Next, I is 2 and J is 3+2+5 = 10. In general, if the IV were 3,-1,X, J would be 5+X.

At this point, we have processed the three-byte IV; the next iteration involves the first secret byte of K. I is
3 and J becomes 10+1+K[3] = 15:

Recall that the first byte returned in the keystream is S[ S[1] + S[S[1]] ]. At this point, that would
be S[0+3] = 15.
From this value, 15, and the IV 3,-1,5, the attacker can calculate, by repeating the process above, that
K[3] = 4.
If the IV were 3,-1,X, then, as noted above, in the I=2 step we would have J = 5+X, and then in the I=3
step we would have J = 6 + X + K[3]. If Y is the value of the first byte of the keystream, using S[] as of
this point, then K[3] = Y X 6 (assuming that X is not -5 or -4, so that S[0] and S[1] are not changed
in step 3).

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If none of S[0], S[1] and S[3]] are changed in the remaining 252 iterations, the value we predict here
after three iterations eg 15 would be the first byte of the actual keystream. Treating future selections of
the value J as random, the probability that one iteration does not select J in the set {0,1,3} and thus leaves
these values alone is 253/256. The probability that the remaining iterations do not change these values in
S[] is thus about (253/256)252 5%
A 5% success rate is not terribly impressive, on the face of it. But 5% of the time we identify K[3]
correctly, and the other 95% of the time the (incorrect) values we calculate for K[3] are uniformly, randomly
distributed in the range 0 to 255. With 60 guesses, we expect about three correct guesses. The other 57 are
spread about with, most likely, no more than two guesses for any one value. If we look for the value guessed
most often, it is likely to be the true value of K[3]; increasing the number of 3,-1,X IVs will increase the
certainty here.
For the sake of completeness, in the next two iterations, I is 4 and 5 respectively, and J is 15+4+(-14) = 5
and 5+4+3=12 respectively. The corresponding contents of S[] are

Now that we know K[3], the next step is to find K[4]. Here, we search the traffic for IVs of the form
4,-1,X, but the general strategy above still works.
The [FMS01] attack, as originally described, requires on average about 60 weak IVs for each secret key
byte, and a weak IV turns up about every 216 packets. Each key byte, however, requires a different IV, so
any one IV has one chance per key byte to be a weak IV. To guess a key thus requires 6065536 4 million
packets.
But we can do quite a bit better. In [TWP07], the number of packets needed is only about 40,000 on average.
At that point the attack is entirely practical.
This vulnerability was used in the attack on TJX Corp mentioned in the opening section.

21.7 Diffie-Hellman-Merkle Exchange

How do two parties agree on a secret key K? They can meet face-to-face, but that is expensive and not
always possible. We would like to encrypt, for example, online shopping transactions, which often occur at
the spur of the moment.

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Diffie, Hellman and Merkle

Diffie and Hellmans paper [DH76] appeared in 1976, but they reference Merkles later-published paper
[RM78] as submitted. In [MH04], Hellman suggested that the name of the key-exchange algorithm
here should bear the names of all three contributors. The algorithm was apparently discovered in 1974
by Malcolm Williamson at GCHQ, but Williamsons work was classified.
One approach is to use public-key encryption, below, but that came two years later. The Diffie-HellmanMerkle approach, from [DH76] and [RM78], allows private key exchange without public keys.
The goal here is for Alice and Bob to exchange information over a public network so that, at the end,
Alice and Bob have determined a common key, and no eavesdropper can determine it without extraordinary
computational effort.
As a warmup, we begin with a simple example. Alice and Bob each choose a number; say Alice chooses 7
and Bob chooses 9. Alice sends to Bob the value A = 37 = 2187, and Bob sends to Alice the value B = 39
= 19683. Alice then computes B7 = 397 , and Bob computes A7 = 379 . If they each use 32-bit arithmetic,
they have each arrived at 2111105451.
The catch here is that any intruder with a calculator can recover 7 from A and 9 from B. But if we make the
arithmetic a little more complicated, this becomes extremely difficult.
In the actual protocol, Alice and Bob agree, publicly, on a large prime p (typically several hundred digits)
and a small value g. Alice then chooses a<p, and sends to Bob the value A = ga mod p. Similarly, Bob
chooses b<p and sends to Alice the value B = gb mod p.
As before, Alice and Bob now compute Ba mod p and Ab mod p, which both equal gab mod p. This
becomes their shared key.
The question is whether an eavesdropper who knows A and B and g can compute a and b. This is the discrete logarithm problem, as in some mod-p sense a = logg (A). It is indeed believed to be computationally
intractable, provided that g is a so-called primitive root modulo p (meaning that the values g1 mod p, g2 mod
p, ..., gp-1 mod p are all distinct). A naive attempt to compute a from A is to try each gi for i<p until one
finds a match. There is some work involved in finding a suitable primitive root g given p, but that can be
public, and is done only once, and is usually not too time-consuming.
The Diffie-Hellman-Merkle exchange is vulnerable to a man-in-the-middle attack, discussed in greater
detail in 21.8.3 Trust and the Man in the Middle. If Mallory intercepts Alices ga and sends Bob his own
ga , and similarly intercepts Bobs gb and replaces it with his own gb , then the AliceMallory link will
negotiate key gab and the Mallory-Bob link will negotiate gab . Mallory then remains in the middle,
intercepting, decryping, re-encrypting and forwarding each message between Alice and Bob. The usual fix
is for Alice and Bob to sign their ga and gb ; see 21.8.2 Forward Secrecy.

21.7.1 Fast Arithmetic

It is important to note that the other arithmetic operations here eg calculating ga mod p can be done in
polynomial time with respect to the number of binary digits in p, sometimes called the bit-length of p. This
is not completely obvious, as the naive approach of multipying out gg...g, a times, takes O(p) steps
when ap, which is exponential with respect to the bit-length of p. The trick is to use repeated squaring: to
compute g41 , we note 41 = 101001 in binary, that is, 41 = 25 + 23 + 1, and so

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g41 = ((((g2 )2 )2 )2 )2 ((g2 )2 )2 g

A simple python3 implementation of this appears at 21.9.3 Creating an RSA Key.
It is perhaps also worth noting that even finding large primes is not obviously polynomial in the number of
digits. We can, however, use fast probabilistic primality testing, and note that the Prime Number Theorem
guarantees that the number of candidates we must test to find a prime near a given number n is O(log n).

21.8 Public-Key Encryption

In public-key encryption, each party has a public encryption key KE and a private decryption key KD . Alice
can publish her KE to the world. If Bob has Alices KE , he can encrypt a message with it and send it to her;
only someone in possession of Alices KD can read it.
Public-key systems can also be used for authentication, by reversing the process. Alice can write a message
to Bob and encrypt it with her own KD . Bob, and everyone else in the world, can decrypt the message, but
the fact that it was encrypted with Alices private key pretty much means that only Alice could have created
it. Alice thus has a way of applying a digital signature to her messages. If Alice doesnt want the signature
to be world-readable, she can encrypt it with Bobs public key after creating it. (Just as Alice probably
prefers not to sign undated documents with her handwritten signature, she will likely prefer to apply her
digital signature only to documents containing a timestamp.)

21.8.1 RSA
Public-key encryption was outlined in [DH76], but the best-known technique is arguably the RSA algorithm
of [RSA78]. (The RSA algorithm was discovered in 1973 by Clifford Cocks at GCHQ, but was classified.)
To construct RSA keys, one first finds two very large primes p and q, perhaps 1024 binary digits each. These
primes should be chosen at random, by choosing a random N and then testing N, N+1, ... for primality until
success. Let n = pq.
It now follows from Fermats little theorem that, for any integer m, m(p-1)(q-1) = 1 mod n (it suffices to show
m(p-1)(q-1) = 1 mod p and m(p-1)(q-1) = 1 mod q).
One then finds positive integers e and d so that ed = 1 mod (p-1)(q-1); given any e relatively prime to both
p-1 and q-1 it is possible to find d using the Extended Euclidean Algorithm (a simple Python implementation
appears below in 21.9.3 Creating an RSA Key). From the claim in the previous paragraph, we now know
med = (me )d = m mod p.
If we take m as a message (that is, as a bit-string of length less than the bit-length of n, rather than as an
integer), we encrypt it as c = me mod n. We can decrypt c and recover m by calculating cd mod n = med
mod n = m.
The public key is the pair n,e; the private key is d.

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Elliptic-curve cryptography
One of the concerns with RSA is that a faster factoring algorithm will someday make the encryption
useless. A newer alternative is the use of elliptic curves; for example, the set of solutions modulo a
large prime p of the equation y2 = x3 + ax + b. This set has a natural (but nonobvious) product operation
completely unrelated to modulo-p multiplication, and, as with modulo-p arithmetic, finding n given B
= An appears to be quite difficult. Several cryptographic protocols based on elliptic curves have been
proposed; see Wikipedia.
As with Diffie-Hellman-Merkle (21.7.1 Fast Arithmetic), the operations above can all be done in polynomial time with respect to the bit-lengths of p and q.
The theory behind the security of RSA is that, if one knows the public key, the only way to find d in practice
is to factor n. And factoring is a hard problem, with a long history. There are faster ways to factor than to

try every candidate less than n, but it is still believed to require, in general, exponential time with respect
to the bit-length of n.
RSA encryption is usually thousands of times slower than comparably secure shared-key ciphers. However,
RSA needs only to be used to encrypt a secret key for a shared-key cipher; it is this latter key that actually
protects the document.
For this reason, if a message is encrypted for multiple recipients, it is usually not much larger than if it were
encrypted for only one: the additional space needed for each additional recipient is just the encrypted key
for the shared-key cipher.
Similarly, for digital-signature purposes Alice doesnt have to encrypt the entire message with her private
key; it is sufficient (and much faster) to encrypt a secure hash of the message.
If Alice and Bob want to negotiate a session key using public-key encryption, it is sufficient for Alice to
know Bobs public key; Alice does not even need a public key. Alice can choose a session key, encrypt it
with Bobs public key, and send it to Bob.
We will look at an actual calculation of an RSA key and then an encrypted message in 21.9.3 Creating an
RSA Key.

21.8.2 Forward Secrecy

Suppose Alice and Bob exchange a shared-key-cipher session key KS using their RSA keys. Later, their
RSA keys are compromised. If the attacker has retained Alice and Bobs prior communications, the attacker
can go back and decrypt KS , and then use KS to decrypt the entire session protected by KS .
This is not true, however, if Alice and Bob had used Diffie-Hellman-Merkle key exchange. In that case,
there is no encryption used in the process of negotiating KS , so no later encryption compromise can reveal
KS .
This property is called forward secrecy, or, sometimes, perfect forward secrecy (other times, perfect forward secrecy adds the further requirement that the compromise of any other session key negotiated by Alice
and Bob does not reveal information about KS ).
The advantage of public-key encryption, however, is that Alice can sign the key KS she sends to Bob.
Assuming Bob is confident he has Alices real public key, a man-in-the-middle attack (21.8.3 Trust and the

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Man in the Middle) becomes impossible.

It is now common for public-key encryption to be used to sign all the transactions that are part of the DiffieHellman-Merkle exchange. When this is done, Alice and Bob gain both forward secrecy and prevention of
man-in-the-middle attacks.

21.8.3 Trust and the Man in the Middle

Suppose Alice wants to send Bob a message. Where does she find Bobs public key EB ?
If Alice goes to a public directory that is not completely secure and trustworthy, she may find a key that in
fact belongs to Mallory instead. Alice may now fall victim to a man-in-the-middle attack, like that at the
end of 21.7 Diffie-Hellman-Merkle Exchange:
Alice encrypts Bobs message using Mallorys public key EMal , thinking it is Bobs
Mallory intercepts and decrypts the message, using his own decryption key DMal
Mallory re-encrypts the message with Bobs real public key EBob
Bob decrypts the message with DBob
Despite Mallorys inability to break RSA directly, he has read (and may even modify) Alices message.
Alice can, of course, get Bobs key directly from Bob. Of course, if Alice met Bob, the two could also
exchange a key for a shared-key cipher.
Wi-Fi in the Middle
One of the easiest settings at which to launch a man-in-the-middle attack is a public Wi-Fi hotspot,
either as the hotspot owner or by setting up a rogue access point to which some customers mistakenly
connect.
We now come to the mysterious world of trust. Alice might trust Charlie for Bobs key, but not Dave. Alice
doesnt have to meet Charlie to get Bobs key; all she needs is
a trusted copy of Charlies public key
a copy of Bobs public key, together with Bobs name, signed by Charlie
At the same time, Alice might trust Dave but not Charlie for Evans key. And Bob might not trust Charlie or
Dave for Alices key. Mathematically, the trust relationship is neither symmetric nor transitive. To handle
the possibility that trust might erode with time, signed keys often have an expiration date.
At the small scale, key-signing parties are sometimes held in which participants exchange some keys directly and others indirectly through signing. At the large scale, certificate authorities are built into TLS
(21.9.2.1 Certificate Authorities); you are implicitly trusting these certificate authorities if your browser
vendor trusts them.

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21.8.4 End-to-End Encryption

Many communications applications are based on the model of encryption from each end-user to the central
server. For example, Alice and Bob might both use https (based on TLS, 21.9.2 TLS) to encrypt their interactions with their email provider. This means Alice and Bob are now trusting that provider, who decrypts
messages from Alice, stores them, and re-encrypts them when delivering them to Bob.
This model does protect Alice and Bob from Internet eavesdroppers who have not breached the security of
the email provider. However, it also allows government authorities to order the email provider to turn over
Alice and Bobs correspondence.
If Alice and Bob do not wish to trust an intermediary, or their (or someone elses) government, they need
to implement end-to-end encryption. That is, Alice and Bob must negotiate a key, use that key to encrypt
messages between them, and not divulge the key to anyone else. This is quite a bit more work for Alice
and Bob, and even more complicated if Alice wishes to use end-to-end encryption with a large number of
correspondents.
Of course, even with end-to-end encryption Alice may still be compelled by subpoena to turn over her
correspondence with Bob, but that is a different matter. Alices private key may also be seized under a
search warrant. It is common (though not universal) to protect private keys with a password; this is good
practice, but protecting a key having an effective length of 256 bits with a password having an effective
length of 32 bits leaves something to be desired. The mechanisms of 21.5.2 Password Hashes provide only
limited relief. The mechanism of 21.8.2 Forward Secrecy may be more useful here, assuming Alice can
communicate to Bob that her previous key is now compromised; see also 21.9.2.2 Certificate revocation.

21.9 SSH and TLS

We now look at two encryption mechanisms in popular use on the Internet. The first is the Secure Shell,
SSH, used to allow login to remote systems, remote command execution, file transfer and even some forms
of VPN tunneling. Public-key-encryption is optional; if it is used, the public keys are generally transported
manually.
POODLE
The 2014 POODLE attack was based in part on a long-known flaw in SSL 3.0. But SSL 3.0 is fifteen
years obsolete; the other flaw was a broken version-negotiation implementation in which the disruption
of a few packets by an attacker could force both client and server to settle on SSL 3.0 even when both
supported the latest TLS version. See 21.9.2.3 TLS Connection Setup.
The second example is the Transport Layer Security protocol, TLS, which is a successor of the Secure
Sockets Layer, or SSL. Many people still refer to TLS by the SSL name even though TLS replaced SSL in
1999, though, to be fair, TLS is effectively an update of SSL, and TLS v 1.0 could easily have been named
SSL v 4.0. TLS is used to encrypt web traffic for the HTTP Secure protocol, https; it is also used to encrypt
traffic for several other applications and can be used, with appropriate programming, for any application.
If Alice wants to contact a server S using either SSH or TLS, at some point she will have to trust a public
key claimed to be from S. A common approach with SSH is for Alice to accept the key on faith the first time
it is presented, but then to save it for all future verifications. Under TLS, the key from S that Alice receives
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will have been signed by a certificate authority (21.9.2.1 Certificate Authorities); Alice presumably trusts
this certificate authority. (SSH does now support certificate authorities too, but their use with SSH seems
not yet to be common.)
Both SSH and TLS eventually end up negotiating a shared-secret session key, which is then used for most
of the actual data encryption.

21.9.1 SSH
The SSH protocol establishes an encrypted communications channel between two hosts, after establishing
the identities of each endpoint. Its primary use is to enable secure remote-command execution with input
and output via the secure channel; this includes the remote execution of an interactive shell, which is in
effect a telnet-style terminal login with encryption. The companion program scp uses the SSH protocol to
implement secure file transfer. Finally, ssh supports secure port forwarding from one machine (and port)
to another; unrelated applications can then connect to one machine and find themselves securely talking to
another. The current version of the SSH protocol is 2.0, and is defined in RFC 4251. The authentication
and transport sub-protocols are defined in RFC 4252 and RFC 4253 respectively.
One of the first steps in an SSH connection is for the endpoints to negotiate which secret-key cipher (and
mode) to use. Support of the following ciphers is recommended and there is a much longer list of optional ciphers (which include RC4 and Blowfish); the table below includes those added by RFC 4344:
cipher
3DES
AES
AES
AES

modes
CBC, CTR
CBC, CTR
CTR
CTR

nominal keylength
168 bits
192 bits
128 bits
256 bits

SSH supports a special name format for including new ciphers for local use.
The SSH protocol also allows the endpoints to negotiate a public-key-encryption mechanism, a secure-hash
function, and even a key-exchange algorithm although only minor variants of Diffie-Hellman-Merkle key
exchange are implemented.
If Alice wishes to connect to a server S, the server clearly wants to verify Alices identity, either through a
password or some other means. But it is just as important for Alice to verify the identity of S, to prevent
man-in-the-middle attacks and the possibility that an attacker masquerading as S is simply collecting Alices
password for later use.
In the following subsections we focus on the common SSH configuration and ignore some advanced
options.
21.9.1.1 Server Authentication
To this end, one of the first steps in SSH connection negotiation is for the server to send the public half of
its host key to Alice. Alice verifies this key, which is typically in her known_hosts file. Alice also asks
S to sign something with its host key. If Alice can then decrypt this with the public host key of S, she is
confident that she is talking to the real S.
If this is Alices first attempt to connect to S, however, she should get a message like the one below:

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The authenticity of host S (10.2.5.1) cant be established.

RSA key fingerprint is da:2e:e3:94:84:6b:bf:6d:2f:e4:c3:76:68:72:a5:a0.
Are you sure you want to continue connecting (yes/no)?
If Alice is cautious, she may contact the administrator of S and verify the key fingerprint. Otherwise she
may simply choose yes, in which case the host key of S will be added to her own known_hosts file. If
she later re-connects to S after the host key of S has been changed, she will get a rather more dire message,
and, under the default configuration, she will not be allowed to continue until she manually removes the old,
now-incorrect, host key for S from her known_hosts file.
See also the scenario at the end of 7.7.2 ARP security.
SSH v2.0 now also supports a certificate-authority mechanism for verifying server host keys, replacing the
known_hosts file.
21.9.1.2 Key Exchange
The next step is for Alices computer and S to negotiate a session key. After the cipher and key-exchange
algorithms are negotiated, Alices computer and S use the chosen key-exchange algorithm to agree on a
session key for the chosen cipher.
The two directions of the connection generally get different session keys. They can even use different
encryption algorithms.
At this point the channel is encrypted.
21.9.1.3 Client Authentication
The server now asks Alice to authenticate herself. If password authentication is used, Alice types in her
password and it and her username are sent to the server. This does expose the server to brute-force password
attacks, and is not infrequently disabled.
Alice may also have set up RSA authentication (other types of public-key authentication are also possible).
For this, Alice must create a public/private key pair (often in files id_rsa.pub and id_rsa), and the
public key must have been previously installed on S. On Unix-based systems it is often installed on S in
the authorized_keys file in the .ssh subdirectory of Alices home directory. If Alice now sends a
message to S signed by her private key, S is in a position to verify the signature. If this succeeds, Alice can
log in without supplying a password to S.
It is common though not universal practice for Alices private-key file (on her own computer) to be protected
by a password. If this is the case, Alice will need to supply that password, but it is used only on Alices end of
the connection. (The default password hash is MD5, which may not be a good choice; see 21.5.2 Password
Hashes.)
Public-key authentication is tried before password authentication. If Alice has created a public key, it is
likely to be tried even if she has not copied it to S.
If S is set up to require public-key authentication, Alice may not have any direct way to install her public key
on S herself, and may need the cooperation of the administrators of S. It is possible that the latter will send
Alice a public/private keypair chosen by them specifically for S, rather than allowing Alice to choose her
own keypair. The standard SSH user configuration does support different private keys for different servers.
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In several command-line implementations of ssh, the various stages of authentication can be observed from
the client side by using the ssh or slogin command with the -v option.
21.9.1.4 The Session
Once an SSH connection has started, a new session key is periodically negotiated. RFC 4253 recommends
this after one hour or after 1 GB of data is exchanged.
Data is then sent in packets generally with size a multiple of the block size of the cipher. If not enough data
is available, eg because only a single keystroke (byte) is being sent, the packet is padded with random data
as needed. Every packet is required to be padded with at least four bytes of random data to thwart attacks
based on known plaintext/ciphertext pairs. Included in the encrypted part of the packet is a byte indicating
the length of the padding.

21.9.2 TLS
Transport Layer Security, or TLS, is an IETF extension of the Secure Socket Layer (SSL) protocol originally
developed by Netscape Communications. Its primary role is encrypting web connections and verifying for
the client the authenticity of the server; its current specification is RFC 5246.
Unlike SSH, client authentication, while possible, is not common; web servers often have no pre-existing
relationship with the client. Also unlike SSH, the public-key mechanisms are all based on established
certificate authorities, or CAs, whereas the most common way for an SSH servers host key to end up on
a client is for it to have been accepted by the user during the first connection. Browsers (and other TLS
applications as necessary) have embedded lists of certificate authorities trusted by the browser vendor. SSH
requires no such centralized trust.
If Bob wishes to use TLS on his web server SBob , he must first arrange for a certificate authority, say CA1 ,
to sign his certificate. A certificate contains the full DNS name of SBob , say bob.int, a public key KS used
by SBob , and also an expiration date. The use of the ITU-T X.509 certificate format is common.
Now imagine that Alice connect to Bobs server SBob using TLS. Early in the process SBob will send her its
signed certificate, claiming public key KS . Alices browser will note that the certificate is signed by CA1 ,
and will look up CA1 on its list of trusted certificate authorities. If found, the next step is for the browser to
use CA1 s public key, also on the list, to verify the signature on the certificate SBob sent.
If everything checks out, Alices browser now knows that CA1 certifies bob.int has public key KS . As SBob
has presented this key KS , and is able to verify that it possesses the matching private key, this is proof that
SBob is legitimately the server with domain name bob.int.
Assuming, of course, that CA1 is correct.
As with SSH, once Alice has accepted the public key KS of SBob , a secret-key cipher is negotiated and the
remainder of the exchange is encrypted.
21.9.2.1 Certificate Authorities
A certificate authority, or CA, is just an entity in the business of signing certificates. The purpose of the
certificate authority is to prevent man-in-the-middle attacks (21.8.3 Trust and the Man in the Middle); Alice
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wants to be sure she is not really connected to SBad instead of to SBob .

What the CA is actually signing is that the particular public key KS belongs to domain bob.int, just like
Charlie might sign Bobs public key on an individual basis. The difference here is that the CA is likely to be
a large organization, and the CAs public key is likely to be embedded in the network application software
somewhere.
A certificate for bob.int will not work for www.bob.int, even though both DNS names might direct to the
same server SBob . If SBob presents a certificate for bob.int to a browser that has arrived at SBob via the url
http://www.bob.int, the user should see a warning. All but the smallest sites generally work out the necessary
details here to support two certificates.
Firefox Certificates
As of this writing, certificate authorities for the popular Firefox browser are found in Preferences
Advanced View Certificates. It is an interesting list, if only because, for such highly trusted
organizations, few people have heard of more than a handful of them. The policy for becoming a
Firefox CA is here.
Popular browsers all contain embedded lists of CAs that the browser vendor has decided to trust; more
precisely, the browser embeds the names and public keys of these CAs. If Alice is also willing to trust
these CAs, she can feel comfortable using the key she receives to send private messages to Bob. That if,
however, is sometimes controversial. Just because Alice trusts her browser (eg not to contain malware), that
does not automatically imply that Alice should trust her browser vendors judgment when it comes to CAs.
On the face of it, CA1 is just signing that domain name bob.int has public key KS . This isnt quite the
same, however, as verifying that the certificate-signing request actually came from Bob. All depends on
how thorough CA1 is in checking the identity of its customer, and since those customers typically choose
the least expensive CA, there is sometimes an incentive to cut corners.
If the certificate authority CA1 were also the domain registrar with whom Bob registered the DNS name
bob.int, it would be especially well-positioned to verify that Bob is really the owner of the bob.int domain.
But this is not generally the case. Quite often, the CAs primary verification is that it sends an email to,
say, [email protected] (or perhaps to the DNS administrative contact listed in the WHOIS database for domain
bob.int). If someone responds to this email, it is assumed they must be legitimate.
That assumption is not always warranted, and a failure can enable a man-in-the-middle attack (21.8.3 Trust
and the Man in the Middle). Suppose, for example, that Mallory is able to obtain a signed certificate from
another certificate authority CA2 linking bob.int to key Kbad controlled by Mallory. If Mallory can now
position his server SBad in the middle between Alice and SBob ,
Alice

SBad

SBob

then Alice can be tricked into connecting to SBad instead of SBob . Alice will request a certificate, but from
SBad instead of SBob , and get Mallorys from CA2 instead of Bobs actual certificate from CA1 . Mallory
opens a second connection from SBad to SBob (this is easy, as Bob makes no attempt to verify Alices
identity), and forwards information from one connection to the other. As far as TLS is concerned everything
checks out. From Alices perspective, Mallorys false certificate vouches for the key Kbad of SBad , CA2 has
signed this certificate, and CA2 is trusted by Alices browser. There is no search at any point through the
other CAs to see if any of them have any contrary information about SBob . In fact, there is not necessarily
even contact with CA2 , though see 21.9.2.2 Certificate revocation below.
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If Alice is very careful, she may click on the lock icon in her browser and see that the certificate authority
signing her connection to SBad is CA2 rather than CA1 . But if Alice has a secure way of finding Bobs real
certificate authority, she might as well use it to find Bobs key KS . As she probably does not, this is of
limited utility in practice.
The second certificate authority CA2 might be a legitimate certificate authority that has been tricked, coerced
or bribed into signing Mallorys certificate. Alternatively, it might be Mallorys own creation, inserted by
Mallory into Alices browser through some other vulnerability.
Mallorys machinations here do require the man-in-the-middle attack as well as the bad certificate. If Alice
is able to establish a direct connection with SBob , then the latter will send its true key KS signed by CA1 .
As another attack, Mallory might obtain a certificate for b0b.int and hope Alice doesnt notice the spelling
difference between B0B and BOB. When this is done, Mallory often also sends Alice a disguised link to
b0b.int in the hope she will click on it. Unicode domain names make this even easier, as Unicode provides
many character pairs that are different but which look identical.
Extended-Validation certificates were introduced in 2007 as a way of providing greater assurances that the
certificate issued to bob.int was in fact generated by a request from Bob himself; Mallory should in theory
have a much harder time obtaining an EV certificate for bob.int from CA2 . Browsers that have secured a
TLS connection using an EV certificate typically add the name of the domain owner, highlighted in green
and/or with a green padlock icon, to the address bar. Financial institutions often use EV certificates. So does
mozila.org.
21.9.2.2 Certificate revocation
Suppose the key KS underlying the certificate for bob.int has been compromised, and Mallory has the private
key. Then, if Mallory can trick Alice into connecting to SBad instead of SBob , the original CA1 will validate
the connection. SBad can prove to Alice that it has the secret key corresponding to KS , and all the certificate
does is to attest that bob.int has key KS . Mallorys deception works even if Bob noticed the compromise and
updated SBob s key to K2 ; the point is that Mallory still has the original key, and CA1 s certificate attesting
to that original key is still valid.
Fortunately, there is a mechanism by which a certificate can be revoked. Revocation information, however,
must be kept at some central directory; a server can continue to serve up a revoked certificate and unless the
clients actively check, they will be none the wiser. This is one reason certificates have expiration dates.
At one time, for efficiency reasons browsers by default did not check for revoked certificates. Now mostly
they do.
The original revocation mechanism was the global certificate revocation list. A newer alternative is the
Online Certificate Status Protocol, OCSP, described in RFC 6960. If Alice receives a certificate signed by
CA1 , she can send the serial number of the certificate to a designated OCSP responder run by or on behalf
of CA1 ; this site will confirm the validity of the original certificate.
Of course, an eavesdropper watching Alices traffic arriving at the OCSP responder now knows that Alice is
visiting bob.int. An eavedropper closer to Alice, however, knows that anyway.

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21.9.2.3 TLS Connection Setup

The typical TLS client-side user is interested in viewing a web page as quickly as possible, placing a premium on rapid negotiation of the TLS connection. If sites were to load noticeably more slowly when
encryption was used, encryption might not be used routinely.
TLS connection negotiation typically takes two RTTs, as follows. The client first sends its ClientHello
message, which contains its preferred TLS/SSL-protocol version number, a session identifier, some random
bytes, and a list of cipher suites: tuples consisting of a key-exchange algorithm, a shared-key encryption
algorithm and a secure-hash algorithm. The list of cipher suites is ranked by the client according to preference.
Choosing your cipher
In theory, any TLS client can choose which cipher suites it will accept. In practice, for most software
applications this is non-trivial. In the Firefox browser, see about:config, under security.ssl3 (yes,
ssl3 is in theory not the same as tls).
The server then responds with its ServerHello. This contains the final protocol-version number, generally the maximum of the version proposed by the client and the highest version supported by the server.
The ServerHello also contains the servers choice of a cipher suite from the clients list, and some more
random bytes. The server also sends its certificate, if applicable, and its first key-exchange message (if this
was not implicit in the certificate). These last two components are considered to be separate messages at
the TLS protocol level, but everything is sent together.
The client then responds with its key-exchange response, and its own certificate if applicable. It immediately
follows that with its Finished message, the first message that is encrypted with the just-negotiated cipher
suite. The server then replies with its encrypted Finished message, and encrypted application-layer
communication can then begin.
Most browsers allow the user to click on the padlock icon for the TLS-secured connection to find out what
cipher suite was actually agreed upon.
Once upon a time, some broken TLS servers failed to respond properly if a client proposed a version number
greater than what the server supported; the server would close the connection instead of returning a lower
version number. As a result, many clients were programmed to try again with the next-lower version in the
event of any connection-setup failure. This meant that an attacker could force a client to propose an obsolete
version (eg SSL 3.0) simply by interrupting earlier connection-setup attempts, perhaps with a TCP RST
packet. A consequence was the POODLE vulnerability mentioned in the sidebar in 21.9 SSH and TLS.
An applications choice of TLS (versus unencrypted communication) is often signaled by the use of a special
port number; for example, the standard http port is 80 while the standard https port is 443. Alternatively,
there may be some initial negotiation; for example, in the SMTP email protocol (RFC 5321) it is common
for the client to connect to the server on the standard port 25 without encryption, but then to negotiate the
use of TLS using the STARTTLS extension to SMTP (RFC 3207). This adds multiple extra RTTs, but for
non-interactive protocols this is usually of only minor concern.

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21.9.3 Creating an RSA Key

In this section we create a short RSA key, using the openssl package, available for Windows, Macs and
linux.
We first create an RSA key with length 96 bits:
openssl genrsa -out key96.pem 96

The resultant file is as follows:

-----BEGIN RSA PRIVATE KEY----MFICAQACDQCo1hzP6/gTzbNAEHcCAwEAAQINAJzcEHi8aYSO0iizgQIHANkzC28P
jwIHAMb/VQH8mQIHALc+9ZqRyQIHAKkfQ43msQIGJxhmMMOs
-----END RSA PRIVATE KEY-----

This is in the so-called PEM format, which means that the two lines in the middle are the base64 encoding
of the ASN.1 encoding (20.12 SNMP and ASN.1 Encoding) of the actual data. Despite the PRIVATE KEY
label, the file in fact contains both private and public keys. SSH private keys, typically generated with the
ssh-keygen command, are also in PEM format.
We can extract the PEM-file data with the next command:
openssl rsa -in key96.pem -text

The output is the following:

Private-Key: (96 bit)
modulus:
00:a8:d6:1c:cf:eb:f8:13:cd:b3:40:10:77
publicExponent: 65537 (0x10001)
privateExponent:
00:9c:dc:10:78:bc:69:84:8e:d2:28:b3:81
prime1:
00:d9:33:0b:6f:0f:8f
prime2:
00:c6:ff:55:01:fc:99
exponent1:
00:b7:3e:f5:9a:91:c9
exponent2:
00:a9:1f:43:8d:e6:b1
coefficient:
27:18:66:30:c3:ac

The default OpenSSL encryption exponent, denoted e in 21.8.1 RSA, is 65537 = 216 +1.
We next convert all these hex numbers to decimal; the corresponding notation of 21.8.1 RSA is in parentheses.

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modulus (n)
privateExponent (d)
prime1 (p)
prime2 (q)
exponent1
exponent2
coefficient

52252327837124407964427358327
48545702997494592199601992577
238813258387343
218799945153689
201481036403145
185951742453425
42985747170220

We now verify some arithmetic, using any tool that supports large integers (eg python3, used here, or the
unix bc command). First we check n = pq:
>>> 238813258387343 * 218799945153689
52252327837124407964427358327

Next we check that ed = 1 mod (p-1)(q-1):

>>> e=65537
>>> d=48545702997494592199601992577
>>> p=238813258387343
>>> q=218799945153689
>>> (p-1)*(q-1)
52252327837123950351223817296
>>> e*d % 52252327837123950351223817296
1

To encrypt a message m, we must use efficient mod-n calculations; here is an implementation of the repeatedsquaring algorithm (mentioned above in 21.7.1 Fast Arithmetic) in python3. (This function is built into
python3 as pow(x,e,n).)
def power(x,e,n): # computes x^e mod n
pow = 1
while e>0:
if e%2 == 1: pow = pow*x % n
x = x* x % n
e = e//2
# // denotes integer division
return pow

Let m be the string Rivest. In hex this is 0x526976657374; in decimal, 90612911403892.

>>> m=0x526976657374
>>> c=power(m,e,n)
>>> c
38571433489059199500953769621
>>> power(c,d,n)
90612911403892

What about the last three numbers in the PEM file, exponent1, exponent2 and coefficient? These
are pre-computed values to speed up decryption. Their values are
exponent1 = d mod (p-1)
exponent2 = d mod (q-1)
coefficient is the solution of coefficient q = 1 mod p
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21.9.3.1 Breaking the key

Finally, let us break this 96-bit key and decrypt the message with ciphertext c above. The hard part is
factoring n; we use the Gnu/linux factor command:
> factor 52252327837124407964427358327
52252327837124407964427358327: 218799945153689 238813258387343

The factors are indeed the values of p and q, above. Factoring took 2.584 seconds on the authors laptop. Of
course, 96-bit RSA keys were never secure; recall that the current recommendation is to use 2048-bit keys.
The Gnu/linux factor command uses Pollards rho algorithm, and, while serviceable, is not especially
well suited to factoring the product of two large primes. The author was able to factor a 200-bit modulus in
just over 5 seconds using the msieve program, one of several large-number-factoring programs available on
the Internet.
We are almost done; we now need to find the decryption key d, knowing e, p-1 and q-1. For this we need an
implementation of the extended Euclidean algorithm; the following Python implementation is taken from
WikiBooks:
def egcd(a, b):
if a == 0:
return (b, 0, 1)
else:
g, y, x = egcd(b % a, a)
return (g, x - (b // a) * y, y)

A call to egcd(a,b) returns a triple (g,x,y) where g is the greatest common divisor of a and b, and
x and y are solutions to g = ax + by. From 21.8.1 RSA, we need d to be positive and to satisfy 1 =
de + (p-1)(q-1)y. The x value (the second value) returned by egcd(e, (p-1)*(q-1)) satisfies the
second part, but it may be negative in which case we need to add (p-1)(q-1) to get a positive value which is
congruent mod (p-1)(q-1). This x value is -3706624839629358151621824719; after adding (p-1)(q-1) we
get d=48545702997494592199601992577.
This is the value of d we started with. If c is the ciphertext, we now calculate m = pow(c,d,n) as before,
yielding m=90612911403892, or, in hex, 52:69:76:65:73:74, Rivest.

21.10 Exercises
1. Modify the netsploit.c program of 21.2.2.3 The exploit so that the NOPslide is about 1024 bytes long,
and the NOPslide and shellcode are now above the overwritten return address in memory.
2.
Disable ASLR on a linux system by writing the appropriate value to
/proc/sys/kernel/randomize_va_space. Now get the netsploit attack to work without having the attacked server print out a stack address. You are allowed beforehand to run a test copy of the server that
prints its address.
3. Below are a set of four possible TCP-segment reassembly rules. These rules (and more) appear in [SP03].
Recall that, once values for all positions ji are received, these values are acknowleged and released to the
application; there must be some earlier hole for segments to be subject to reassembly at all.

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1. Always accept the first value received for each position.

2. Always accept the last value received for each position, up until the value is released to the application.
3. [BSD] For each new segment, discard each position from that segment if that position was supplied
by an earlier packet with a left edge less than or equal to that of the new packet. Values in any
remaining positions replace any previously received values.
4. [Linux] Same as the previous rule, except that we only discard positions that were supplied by an
earlier packet with a left edge strictly smaller than that of the new packet.
For each of the following two sets of packet arrivals, give the reassembled packet. Values in the same column
have the same position. Reassembled packets will all begin with daa and ebb respectively.
(a).
a a a a a
b b b b b b b
c c c c
d

(b).
a a a a a
b b b b b
c c c c c c
d d d
e

4. Suppose a network intrusion-detection system is 10 hops from the attacker, but the target is 11 hops,
behind one more router R. Outline a strategy of tweaking the TTL field so that the NIDS receives
TCP stream helpful and the target receives harmful. Segments should either be disjoint or cover
exactly the same range of bytes; you may assume that the target accepts the first segment for any given
range of bytes.
5. Suppose Alice encrypts blocks P1, P2 and P3 using CBC mode (21.6.4 Cipher Modes). The initialization
vector is C0. The encrypt and decrypt operations are E(P) = C and D(C) = P. We have
C1 = E(C0 XOR P1)
C2 = E(C1 XOR P2)
C3 = E(C2 XOR P3)
Suppose Mallory intercepts C2 in transit, and replaces it with C2 = C2 XOR M; C1 and C3 are transmitted
normally; that is, Bob receives [C1,C2,C3] where C1 = C1 and C3 = C3. Bob now attempts to decrypt;
C1 decrypts normally to P1 while C2 decrypts to gibberish.
Show that C3 decrypts to P3 XOR M. (This is sometimes used in attacks to flip a specific bit or byte, in
cases where earlier gibberish does not matter.)
6. Suppose Alice uses a block-based stream cipher (21.6.6 Block-cipher-based stream ciphers); block i
of the keystream is Ki and Ci = Ki XOR Pi. Alice sends C1, C2 and C3 as in the previous exercise, and
Mallory again replaces C2 by C2 XOR M. What are the three plaintext blocks Bob deciphers?

588

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An Introduction to Computer Networks, Release 1.0

7. Show that if p and q are primes with p = 2q + 1, then g is a primitive root mod p if g=1, g2 =1, and gq =1.
(This exercise requires familiarity with modular arithmetic and primitive roots.)
8. Suppose we have a short message m, eg a bank PIN number. Alice wants to send message M to Bob that,
without revealing m, can be used later to verify that Alice knew m.
(a). Suppose Alice simply sends M = hash(m). Explain how Bob can quickly recover m.
(b). How can Alice construct M using a secure-hash function, avoiding the problem of (a)? Hint: as part of
the later verification, Alice can supply additional information to Bob.
9. In the example of 21.6.8 Wi-Fi WEP Encryption Failure, suppose the IV is 4,-1,5 and the first two
bytes of the key are 10,20. What is the first keystream byte S[ S[1] + S[S[1]] ]?
10. Suppose Alice uses encryption exponent e=3 to three friends, Bob, Charlie and Deborah, with respective encryption moduli nB , nC and nD , all of which are relatively prime. Alice sends message m to each,
encrypted as
CB = m3 mod nB
CC = m3 mod nC
CD = m3 mod nD
If Mallory intercepts all three encrypted messages, explain how he can efficiently decrypt m. Hint: the
Chinese Remainder Theorem implies that Mallory can find C < nB nC nD such that
C = CB mod nB
C = CC mod nC
C = CD mod nD
(One simple way to avoid this risk is for Alice to include a timestamp and the recipients name in each
message, ensuring that she never sends exactly the same message twice.)
11. Repeat the key-creation of 21.9.3 Creating an RSA Key using a 110-bit key. Extract the modulus from
the key file, convert it to decimal, and attempt to factor it. Can you do the factoring in under a minute?
12. Below are a series of public RSA keys and encrypted messages; the encrypted message is c and the
modulus is n=pq. In each case, find the original message, using the methods of 21.9.3.1 Breaking the
key; you will have to factor n and then find d. For some keys, the Gnu/Linux factor command will be
sufficient; for the larger keys consider msieve or some other fast factorer.
Each number below is in decimal. The encryption exponent e is always 65537; the encryption is c =
power(message,e,n). Each message is an ASCII string; that is, after the numeric message is converted
to a string, the byte values are each in the range 32-127. The following Python function may be useful in
converting numeric messages to strings:
def int2ascii(n):
if n==0: return ""
return int2ascii(n // 256) + chr(n % 256)

(a) [64 bits] c=13467824835325843134

n=15733922878520524621
21.10 Exercises

589

An Introduction to Computer Networks, Release 1.0

(b) [96 bits] c=8007751471156136764029275727

n=57644199986835279860947893727

(c) [104 bits] c=14709590754332048008398738106355

n=17058317327334907783258193953123

(d) [127 bits] c=95651949760509273124353927897611145475

n=122096824047754908887766043915630626757
Limit for Gnu/linux factor without the GMP library

(e) [185 bits] c=14898070767615435522751082309577192810119252877170446296

n=36881105206579952723396854897336450502002018818436622611

(f) [210 bits] c=1030865591241775131254813948981782525587720826169501849049177362

n=1089313781487492651628882855744766776820791642308868127824231021

(g) [280 bits]

c=961792929180423930042975913300802531765775361218349440433358416557620430721970697783
n=1265365011260907658483984328995361695181000868013851878620685943648084220336740539017

(h) [304 bits]

c=17860252858059565448950681133486824440185752167054796213786228492658586864179401029486173539

n=26294146550372428669569992924076340150116542153388301312743129088600749884420889043685502979

590

21 Security

BIBLIOGRAPHY

Note that RFCs are not included here.

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592

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INDICES AND TABLES

genindex
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An Introduction to Computer Networks, Release 1.0

594

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600

Bibliography

INDEX

Symbols
2-D parity, 104
2.4 GHz, 57
3DES, 567
4B/5B, 87
4G, 68
5 GHz, 57
802.11, 57
802.16, 68
802.1Q, 51
802.1X, IEEE, 64

A
accelerated open, TCP, 256
access point, Wi-Fi, 62
accurate costs, 176
ACD, IPv4, 141
ACK compression, 331
ACK, TCP, 239
acknowledgment, 22
acknowledgment number, TCP, 238
ACKs of unsent data, TCP, 251
ACK[N], 109
active close, TCP, 250
active queue management, 311
active subqueue, 419
ad hoc configuration, Wi-Fi, 62
ad hoc wireless network, 66
additive increase, multiplicative decrease, 266
address, 8
address randomization, 554
Address Resolution Protocol, 140
Administratively Prohibited, 146
admission control, RSVP, 467
advertised window size, 257
AES, 567
AF drop precedence, 471

AfriNIC, 21
agent configuration, SNMP, 509
agent, SNMP, 485
AIMD, 266, 309
algorithm, distance-vector, 169
algorithm, DSDV, 178
algorithm, EIGRP, 179
algorithm, exponential backoff, 38
algorithm, fair queuing bit-by-bit round-robin, 421
algorithm, fair queuing GPS, 423
algorithm, fair queuing, quantum, 429
algorithm, hierarchical weighted fair queuing, 436
algorithm, Karn/Partridge, 259
algorithm, link-state, 180
algorithm, loop-free distance vector, 177
algorithm, Nagle, 258
algorithm, Shortest-Path First, 181
algorithm, spanning-tree, 47
Alice, 565
all-nodes multicast address, 156
all-routers multicast address, 156
ALOHA, 41
AMI, 89
anycast address, 155
Aodh, 565
AODV, 178, 413
APNIC, 21
ARC4, 569
ARCFOUR, 569
architecture, network, 484
ARIN, 21
arithmetic, fast, 574
ARP, 140
ARP cache, 141
ARP failover, 143
ARP spoofing, 142
ARPANET, 27

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An Introduction to Computer Networks, Release 1.0

ARQ protocols, 109

AS-path, 198
AS-set, 199
ASLR, 554
ASN.1, 487, 489
ASN.1 enumerated type, 505
association, Wi-Fi, 62
Assured Forwarding, 471
Assured Forwarding PHB, 468
asymmetric routes, 196
Asynchronous Transfer Mode, 77
at-least-once semantics, 232
ATM, 8, 77
augmentation, SNMP, 517
authentication with secure-hash functions, 563
authenticator, WPA, 64
autoconfiguration, IPv4, 144
autonomous system, 169, 189, 197

B
B8ZS, 89
backbone, 20
backoff, Ethernet, 38
backup link, BGP, 205
backwards compatibility, TCP, 324
bad news, distance-vector, 170
band width, radio, 57
bandwidth, 7
bandwidth delay, 98, 114
bandwidth delay, 95
bandwidth guarantees, 445
base station, WiMAX, 68
basic encoding rules, 511
BBRR, 421
beacon, Wi-Fi, 63
BER, 511
Berkeley Unix, 28
best-effort, 17, 22
best-path selection, BGP, 200
BGP, 197
BGP relationships, 207
BGP speaker, 197
big-endian, 9
binary data, 222
bind(), 218
bit stuffing, 88
bit-by-bit round robin, 421
BLAM, 40
602

Blowfish, 567
Bob, 565
border gateway protocol, 197
border routers, 177
bottleneck link, 116, 117, 290
BPDU message, spanning tree, 47
bps, 7, 14
broadcast IP address, 131
broadcast, Ethernet, 15
BSD, 28
buffer overflow, 24
buffer overflow, heap, 556
byte stuffing, 88

C
CA, 581
Canopy, 72
capture effect, Ethernet, 39
care-of address, 149
carrier Ethernet, 56
CBC mode, 568
cell-loss priority bit, ATM, 78
certificate authorities, 577
certificate authority, 581
certificate revocation, 583
certificate revocation list, 583
CFB mode, 570
channel, Wi-Fi, 57
Christmas day attack, 255
CIDR, 189
cipher feedback mode, 570
cipher modes, 567
Cisco, 51, 179, 204, 281
class A/B/C addressing, 16
Class Selector PHB, 468
class, queuing discipline, 417
classful queuing discipline, 417
Classless Internet Domain Routing, 189
clear-to-send, Wi-Fi, 60
client, 23
cliff, 12, 265
CLNP, 27
clock recovery, 85
clock synchronization, 85
CMIP, 484
CMNS, 27
CMOT, 484
collision, 14, 33
Index

An Introduction to Computer Networks, Release 1.0

collision attack, MD5, 562

collision avoidance, 59
collision detection, 33, 41, 42
collision domain, 45
collision, Wi-Fi, 58
community attribute, BGP, 204
community, SNMP, 509
concave cwnd graph, 325
congestion, 12, 18, 24
congestion avoidance phase, 265
congestion bit, 311
congestion window, 264
connection, 8
connection table, virtual circuit, 75
connection-oriented, 237
connection-oriented networking, 17
connectionless networking, 17
conservative, 27
contention, 12
contention interval, 41
contributing source, RTP, 475
convergence to TCP fairness, 310, 324
convex cwnd graph, 325
counter mode, 570
CRC code, 103
cross-site scripting, 559
CSMA, 40
CSMA persistence, 40
CSMA/CA, 59
CSMA/CD, 33
CSRC, 475
CTR mode, 570
CTS, Wi-Fi, 60
Cubic, TCP, 335
cumulative ACK, 113
customer, BGP, 207
cut-through, 13
cwnd, 264
CWR, 239, 311
cyclical redundancy check, 103

D
DAD, IPv6, 161
DALLY, TFTP, 227
data rate, 7
data types, SNMPv1, 489
datagram forwarding, 8
Data[N], 109
Index

deadbeef, 166, 210

DECbit, 311
DECnet, 311
default forwarding entry, 134
default route, 10, 20
deflation, cwnd, 276
delay constraints, 447
delay, bandwidth, 95
delay, largest-packet, 101
delay, propagation, 95
delay, queuing, 95
delay, store-and-forward, 95
delayed ACKs, 257, 270
DES, 567
Destination Unreachable, 146
DHCP, 143
DHCP Relay, 145
DHCPv6, 160, 163
Differentiated Services, 128, 468
Diffie-Hellman-Merkle key exchange, 573
DiffServ, 468
DIFS, Wi-Fi, 59, 66
discrete logarithm problem, 574
distance vector, loop-free versions, 177
distance-vector, 20
distance-vector routing update, 169
distribution network, Wi-Fi, 64
DNS, 21, 144, 161, 162, 221, 222, 230, 247
domain name system, 21
Dont Fragment bit, 132
draft standard, 28
drop precedence, DiffServ AF, 471
DS, 128
DS domain, 468
DS field, IPv4 header, 468
DS1 line, 90
DS3 line, 90
DSDV, 178, 413
DSO channel, 90
dumbbell network topology, 297
duplicate address detection, IPv4, 141
duplicate connection request, 225
duplicate-address detection, IPv6, 161
durian, 524
dynamic rate scaling, 61

E
EAP, 64
603

An Introduction to Computer Networks, Release 1.0

ECB mode, 567

ECE, 239, 311
Echo Request/Reply, 146
ECN, 128, 311
ECT, 311
EFS, 274
EGP, 199
EIGRP, 179
elevator algorithm, 233
encoding, 222
encrypt-and-MAC, 570
encrypt-then-MAC, 570
encryption, 564
end-to-end encryption, 577
End-to-End principle, 109, 238
engines, SNMPv3, 534
enumerated type, SNMP, 505
error-correcting code, 104
error-detection code, 101
estimated flightsize, 274
Ethernet, 14
Ethernet address, 15
Ethernet hub, 34
Ethernet repeater, 34
Ethernet switch, 45
EUI-64 identifier, 155
evasion, intrusion detection, 561
Expedited Forwarding, 469
Expedited Forwarding PHB, 468
expiration date, certificate, 583
Explicit Congestion Notification, 128, 311
exponential backoff, Ethernet, 38
exponential growth, 268
export filtering, BGP, 200
extended interface table, SNMP, 519
Extensible Authentication Protocol, 64
extension headers, 156
exterior routing, 197
extreme TCP unfairness, 303

F
fair queuing, 418
fair queuing and AF, 471
fair queuing and EF, 470
fairness, 24
fairness, TCP, 290, 297, 304, 324
fast arithmetic, 574
Fast Ethernet, 43
604

Fast Open, TCP, 256

fast primality testing, 575
Fast Recovery, 273
fast retransmit, 272
fastest sequence, token-bucket, 444
FCAPS, 483
Feistel network, 565
Fibonacci sequence, 453
FIFO, 289
fill time, voice, 90
filtering, BGP, 200
filterspec, 466
FIN, 239
finishing order, WFQ, 425
firewall, 24
fixed wireless, 70
flights of packets, 265
flightsize, estimated, 274
flow control, 113
flow control, TCP, 258
Flow Label, 154
flow specification, 440
flow, IPv6, 154
flowspec, RSVP, 466
fluid model, fair queuing, 423
foreign agent, 149
forward secrecy, 576
forwarding delay, 13
forwarding table, 9
forwarding, IP, 18
foxes, 263
fragment header, IPv6, 157
fragment offset, 132
fragmentation, 17
Fragmentation Required, 146
fragmentation, IP, 131
fragmentation, Wi-Fi, 61
frame, 8
framing, 88
Friendliness, TCP, 307

G
generalized processor sharing, 423
generic hierarchical queuing, 432
geographical routing, 196
getAllByName(), java, 219
GetBulk, SNMPv2, 514
getByName(), java, 221
Index

An Introduction to Computer Networks, Release 1.0

gigabit Ethernet, 44
glibc-2.2.4, 556
global scope, IPv6 addresses, 164
goodput, 7, 349
GPS, fair queuing, 423
granularity, loss counting, 376
gratuitous ARP, 141
greediness in TCP, 301

H
H-TCP, 335
half-closed, TCP, 250
Hamilton TCP, 335
Hamming code, 104
Hash Message Authentication Code, 563
hash, password, 564
HDLC, 88
head-of-line blocking, 24, 216, 460
header, 8
header, Ethernet, 34
header, IPv4, 127
header, IPv6, 153
header, TCP, 238
header, UDP, 215
heap buffer overflow, 555
heap vulnerability, 556
Hello, spanning-tree, 47
henhouse, 263
hidden-node collisions, 60
hidden-node problem, 60
hierarchical routing, 135, 189, 191
hierarchical token bucket, 448
hierarchical token bucket, linux, 450
high-bandwidth TCP problem, 313
Highspeed TCP, 325
HMAC, 536, 563
hold down, 175
home address, 149
home agent, 149
host identifier, ipv6, 155
host key, ssh, 579
Host Top N, RMON, 529
Host Unreachable, 146
host-specific forwarding, 56
hot-potato routing, 195
htb, linux, 450
htonl, 224
htons, 224
Index

https, 578
hub, Ethernet, 34
hulu, 459
Hybla, TCP, 333

I
IAB, 26
IANA, 17, 21, 506
ICMP, 145
ICMPv6, 164
IDEA, 567
idempotent, 232
IDENT field, 132
IEEE, 33, 36
IEEE 802.11, 57
IEEE 802.1Q, 51
IEEE 802.1X, 64
IETF, 26
ifDescr, 505
ifIndex, 504
IFS, Wi-Fi, 59
ifType, SNMP, 505
ifXTable, SNMP, 519
Illinois, TCP, 333
implementations, at least two, 28
import filtering, BGP, 200
incarnation, connection, 224
inflation, cwnd, 276
infrastructure configuration, Wi-Fi, 62
initial sequence number, TCP, 239, 242, 253
initialization vector, 568
instability, BGP, 208
integrated services, 461
integrated services, generic, 457
interface, 129
interface table, extended, SNMP, 519
interior routing, 169
Internet Architecture Board, 26
Internet checksum, 101
Internet Engineering Task Force, 26
Internet exchange point, 194
Internet Society, 26
intrusion detection, 560
IntServ, 461
IP, 7, 16
IP forwarding, 18
IP fragmentation, 131
IP multicast, 462
605

An Introduction to Computer Networks, Release 1.0

IP-in-IP encapsulation, 149

iptables, 184
IPv4 header, 127
IPv6, 153, 155
IPv6 addresses, 154
IPv6 connections, link-local, 165
IPv6 extension headers, 156
IPv6 header, 153
ipv6 host identifier, 155
IPv6 link-local connections, 165
IPv6 multicast, 155
IPv6 Neighbor Discovery, 159
IPv6 programming, 219
irregular prime, 19
ISM band, 57
ISN, 242, 253
ISOC, 26
IV, 568
IXP, 194

J
jail, staying out of, 194
java getAllByName(), 219
java getByName(), 221
javascript, 559
jitter, 22, 459, 477
join, multicast, 464
JPEG heap vulnerability, 558
JSON, 224
jumbogram, IPv6, 157

K
Karn/Partridge algorithm, 259
KeepAlive, TCP, 259
key-scheduling algorithm, RC4, 569
key-signing parties, 577
keystream, 568
kings, 27
knee, 12, 265, 326
known_hosts, ssh, 579

L
LACNIC, 21
ladder diagram, 97
LAN, 7, 14
LAN layer, 36
LARTC, 183
layers, 7
606

leaky bucket, alternative formulation, 442

learning, Ethernet switch, 16, 45
legacy routing, 191
length-extension vulnerability, 563
liberal, 27
lightning, 93
link-layer ACK, 59
link-local address, 155
link-state packets, 180
link-state routing update, 180
linux, 63, 141, 143, 148, 165, 166, 251, 323, 334,
336, 417, 430
linux advanced routing, 183
linux htb, 450
linux IPv6 routing, 159
Lisp, 566
listen, 23
little-endian, 9
load-balancing, BGP, 205
local traffic, 200
local-area network, 14
logical link layer, 7
lollipop numbering, 181
longest-match rule, 190, 194
loopback address, 130
loopback interface, 129
loss recovery, sliding windows, 116
loss synchronization, 303
loss synchronization, TCP, 301
loss-tolerant, 458
lossy-link TCP problem, 314
lost final ACK, 225
lost final ACK, TFTP, 227
LSA, 180
LSP, 180
LTE, 68

M
MAC address, 15
MAC layer, 36
MAC-then-encrypt, 570
MAE, 194
MAE-East, 194
man-in-the-middle attack, 574, 577
managed device, SNMP, 485
manager, SNMP, 485
Manchester encoding, 86
MANET, 66
Index

An Introduction to Computer Networks, Release 1.0

market, IPv4 addresses, 21

Matrix, RMON, 530
max-min fairness, 304
Maximum Transfer Unit, 131
Mbone, 465
Mbps, 7, 14
MD5, 562
MED, 196, 201, 203
media-access control, 36
Merkle-Dmgard construction, 562
mesh network, 66
MIB browsers, 500
MIB-2, 501
Microsoft SNMP agent, 510
MIMO, 58
minimal network configuration, 144
minimizing route cost, 176
Mitnick, Kevin, 15, 255
mixer, RTP, 475
mobile IP, 149
mobile wireless network, 66
modified EUI-64 identifier, 155
MPLS, 478
MTU, 131
multi-exit discriminator, 201, 203
multi-homed, 193
multi-protocol label switching, 478
multicast, 155
multicast address allocation, 465
multicast IP address, 131
multicast subscription, 463
multicast tree, 462
multicast, Ethernet, 35
multicast, IP, 17, 462
multiple losses, 324
multiple token buckets, 443
MUST, 27

N
Nagle algorithm, 258
NAT, 25
NAT, IPv6-to-IPv4, 167
Neighbor Advertisement, IPv6, 160
Neighbor Discovery, IPv6, 159
Neighbor Solicitation, IPv6, 160
net neutrality, 458
Net-SNMP, 485, 489, 500, 502, 510
Net-SNMP and SNMPv3, 539
Index

network address, 17
network address translation, 25
network architecture, 484
network entry, WiMAX, 69
Network File Sharing, 232
network interface, Ethernet, 15
network management, 416
Network Management System, 484
network model
five layer, 7
four layer, 7
seven layer, 27
network number, 17
network prefix, 17
network prefix, IPv6, 158
Network Unreachable, 146
NewReno, TCP, 276
next_hop, 9
NEXT_HOP attribute, BGP, 202
NFS, 232
NMS, 484
no-transit, BGP, 205
no-valley theorem, BGP, 208
non-compliant, token-bucket, 440
non-congestive loss, 314
non-executable stack, 555
nonpersistence, 40
NOPslide, 553
noSuchObject, SNMP, 514
NRZ, 85
NRZI, 86
ns-2 trace file, 352
ns-2 tracefiles, reading with python, 355
NSFNet, 27
NSIS, 472
ntohl, 224
ntohs, 224
NX page bit, 555

O
Object ID, SNMP, 486
OBJECT-IDENTITY, SNMP, 514
OBJECT-TYPE, SNMP, 490
OC-3, 92
OCSP, 583
OID, SNMP, 486
old duplicate packets, 224
old duplicates, TCP, 251, 252
607

An Introduction to Computer Networks, Release 1.0

OLSR, 413
one-time pad, 568
ones-complement, 101
OpenBSD, 555
optimistic DAD, 161
opus, 459
OSI, 26
OSPF, 180
overhead, ns-2, 369

P
packet loss rate, 305
packet pairs, 296
packet size, 99
Parekh-Gallager claim, 428
Parekh-Gallager theorem, 451
parking-lot topology, 304
partial ACKs, 276
passive close, TCP, 250
password hash, 564
password sniffing, 15, 143
path attributes, BGP, 202
path bandwidth, 117
Path MTU Discovery, 133, 146
path MTU discovery, TCP, 256
PATH packet, RSVP, 466
PAWS, TCP, 255
PCF Wi-Fi, 65
peer, BGP, 207
per-hop behaviors, DiffServ, 468
perfect forward secrecy, 576
persist timer, TCP, 258
persistence, 40
phase effects, TCP, 366
PHBs, DiffServ, 468
physical address, Ethernet, 15
physical layer, 7
PIFS, Wi-Fi, 66
PIM-SM, 463
ping, 146
ping6, 165
pipe drain, 116
pipelining, SMTP, 105
playback buffer, 459
point-to-point protocol, 88
poison reverse, 175
policing, 440
policy routing, 197, 200
608

polling mode, Wi-Fi, 65

polling, TCP, 258
POODLE vulnerability, 578
port exhaustion, TCP, 253
port numbers, UDP, 215
Port Unreachable, 146
potatoes, hot, 195
PPP, 88
PPPoE, 88
prefix information, IPv6, 160
presentation layer, 27
presidents, 27
primality testing, fast, 575
Prime Number Theorem, 575
primitive root modulo p, 574
priority for small packets, WFQ, 425
priority queuing, 289, 416
priority queuing and AF, 471
privacy extensions, SLAAC, 162
private IPv4 address, 130
private IPv6 address, 158
promiscuous mode, 35
propagation delay, 95
proportional fairness, 305
protection against wrapped segments, TCP, 255
protocol graph, 28
protocol-independent multicast, 463
provider, BGP, 207
provider-based routing, 192
proxy ARP, 142
PSH, 239
public-key encryption, 575
pulse stuffing, TDM, 90
push, TCP, 239
python tracefile script, 358360, 382
python, reading ns-2 tracefiles, 355

Q
QoS, 457
QoS and routing, 169
quality of service, 10, 18, 169, 457
quantum algorithm, fair queuing, 429
queue capacity, typical, 281
queue overflow, 18
queue utilization, token bucket, 445
queue-competition rule, 291
queuing delay, 95
queuing discipline, 417
Index

An Introduction to Computer Networks, Release 1.0

queuing, priority, 416

QUIC, 216
quiet time on startup, TCP, 255

R
RADIUS, 64
radvd, 159
random drop, 289
Random Early Detection, 312
ranging intervals, WiMAX, 69
ranging, WiMAX, 69
rate control, Wi-Fi, 61
rate scaling, Wi-Fi, 61
rate-adaptive traffic, 459
RC4, 569
Real-Time Protocol, 308
real-time traffic, 457, 459
Real-time Transport Protocol, 24
reassembly, 17
reboots, 225
Record Route, 129
RED, 312
regional registry, 21
registry, regional, 21
reliable, 237
reliable flooding, 180
Remote Procedure Call, 230
rendezvous point, multicast, 464
Reno, 28, 265
repeater, Ethernet, 34
request for comment, 26
request-to-send, Wi-Fi, 60
request/reply, 230, 237
reservations, 18
reset, TCP, 239
RESV packet, RSVP, 466
retransmit-on-duplicate, 111
retransmit-on-timeout, 109
return-to-libc attack, 548
RFC, 26
RFC 1065, 486
RFC 1066, 501
RFC 1122, 10, 128, 142, 215, 253, 257, 260
RFC 1123, 227
RFC 1155, 486489, 491
RFC 1213, 486, 488, 501503, 505, 507, 508,
522524
RFC 1271, 524, 527
Index

RFC 1312, 521

RFC 1321, 562
RFC 1323, 255
RFC 1350, 225227
RFC 1354, 521, 522
RFC 1441, 486
RFC 1442, 486, 514
RFC 1518, 190
RFC 1519, 190
RFC 1550, 153
RFC 1644, 256
RFC 1650, 520
RFC 1661, 55
RFC 1700, 9, 223
RFC 1812, 148
RFC 1831, 232
RFC 1854, 105
RFC 1901, 486, 513
RFC 1907, 518
RFC 1909, 513
RFC 1948, 255
RFC 2001, 276
RFC 2003, 149
RFC 2011, 521
RFC 2026, 28
RFC 2096, 522, 523
RFC 2104, 536, 563
RFC 2119, 27
RFC 2131, 143
RFC 2136, 163
RFC 2233, 507
RFC 2264, 486
RFC 2362, 463
RFC 2386, 169
RFC 2453, 176
RFC 2460, 154, 157
RFC 2461, 159
RFC 2464, 156
RFC 2474, 128
RFC 2481, 128, 312
RFC 2570, 532
RFC 2574, 533
RFC 2578, 486, 511, 514
RFC 2579, 530, 531
RFC 2581, 257, 271
RFC 2582, 276
RFC 2597, 469, 471, 472
RFC 2675, 157
609

An Introduction to Computer Networks, Release 1.0

RFC 2766, 167

RFC 2819, 525, 526
RFC 2856, 514
RFC 2863, 503506, 519, 523
RFC 2865, 65
RFC 2898, 564
RFC 2925, 531
RFC 3168, 312
RFC 3207, 584
RFC 3246, 469
RFC 3261, 26, 478
RFC 3410, 532
RFC 3414, 486, 532535, 538
RFC 3415, 510, 532, 537
RFC 3418, 518
RFC 3448, 308
RFC 3465, 270, 407
RFC 3513, 155
RFC 3550, 476, 478
RFC 3551, 476
RFC 3561, 178
RFC 3635, 520
RFC 3649, 313, 325
RFC 3748, 65
RFC 3826, 534
RFC 3879, 158
RFC 4022, 523
RFC 4080, 472
RFC 4193, 158
RFC 4251, 579
RFC 4252, 579
RFC 4253, 579, 581
RFC 4271, 197, 198
RFC 4273, 502
RFC 4291, 155, 158
RFC 4292, 522, 523
RFC 4293, 521
RFC 4294, 159
RFC 4344, 579
RFC 4380, 129
RFC 4429, 161
RFC 4443, 164
RFC 4451, 204
RFC 4502, 525
RFC 4524, 487
RFC 4560, 531
RFC 4566, 476
RFC 4620, 166
610

RFC 4681, 289

RFC 4861, 159
RFC 4862, 161
RFC 4941, 162
RFC 4961, 475
RFC 5227, 141
RFC 5246, 581
RFC 5247, 65
RFC 5280, 487
RFC 5321, 486, 584
RFC 5348, 308
RFC 5944, 149
RFC 5974, 472
RFC 6057, 473
RFC 6106, 162
RFC 6151, 564
RFC 6298, 461
RFC 6472, 200
RFC 6633, 146, 312
RFC 6724, 162
RFC 6960, 583
RFC 783, 225, 227
RFC 791, 17, 128
RFC 793, 240, 249, 250
RFC 821, 486
RFC 896, 258
RFC 917, 135
RFC 950, 135
RFC 970, 418
RFC 988, 17
Rinjdael, 567
RIPE, 21
RMON, 524
roaming, Wi-Fi, 63
round-trip time, 97
router, 10
router advertisement, IPv6, 159
router discovery, IPv6, 159
router solicitation, IPv6, 159
routerless IPv6 examples, 165
routing and addressing, 127
routing domain, 169, 189
routing header, IPv6, 157
routing loop, 11, 174
routing loop, ephemeral, 181
routing policy database, 183
routing update algorithms, 168
routing, IP, 18
Index

An Introduction to Computer Networks, Release 1.0

RPC, 230
RSA, 575
RST, 239
RSVP, 461, 466
RTCP, 477
RTCP measurement of, 477
RTO, TCP, 259
RTP, 24, 308
RTP and VoIP, 476
RTP mixer, 475
RTS, Wi-Fi, 60
RTT, 97
RTT bias in TCP, 301
RTT inflation, 119
RTT-noLoad, 98

S
SACK TCP, 279
SACK TCP in ns-2, 381
Salsa20, 566
satellite Internet, 73
satellite-link TCP problem, 315
sawtooth, TCP, 266, 270, 314, 351, 377, 388
scalable routing, 127
scheduled transmission, WiMAX, 69
search warrant, 578
secure hash functions, 562
secure shell, 579
security, 543
segment, 8
segmentation, 17
segments, TCP, 239
Selective ACKs, TCP, 279
self-ARP, 141
self-clocking, 114
sequence number, TCP, 238
serialization, RPC, 233
server, 23
session key, 565
session layer, 27
SHA-1, 562
SHA-2, 562
shaping, 440
shared-key ciphers, 564
shellcode, 550
Shortest-Path First algorithm, 181
SHOULD, 27
sibling, BGP, 207
Index

SIFS, Wi-Fi, 59
signaling losses, 312
signatures, 561
Simple Network Management Protocol, 484
simplex talk, UDP, 216
simplex-talk, TCP, 244
simultaneous open, TCP, 250
single link-state, 20
singlebell network topology, 292
site-local IPv6 address, 158
size, packet, 99
SLAAC, 160, 161
SLAAC privacy extensions, 162
sliding windows, 23, 113
sliding windows, TCP, 257
slot time, Wi-Fi, 59
slow convergence, 174
small-packet priority, WFQ, 425
SMI, 491
SMTP, 105, 584
SNMP, 484
SNMP agent configuration, 509
SNMP agents and managers, 485
SNMP enumerated type, 505
SNMP versions, 486
SNMPv1 data types, 489
SNMPv3 engines, 534
socket, 23
socket address, 23
soft state, 461
SONET, 91
Sorcerers Apprentice bug, 112
Source Quench, 146, 312
source-specific multicast tree, 465
spanning-tree algorithm, 47
sparse-mode multicast, 463
spatial streams, Wi-Fi, 58
speex, 459
split horizon, 174
SQL injection, 560
SSH, 143
ssh, 578, 579
ssh host key, 579
SSID, Wi-Fi, 63
ssl, 578
SSRC, 475
stack canary, 554
STARTTLS, 584
611

An Introduction to Computer Networks, Release 1.0

state diagram, TCP, 248

stateless autoconfiguration, 160
stateless forwarding, 10
STM-1, 92
stochastic fair queuing, 430
stop-and-wait transport, 109
stop-and-wait, TFTP, 227
store-and-forward, 13
store-and-forward delay, 95
stream ciphers, 568
stream-oriented, 237
streaming video, 460
STS-1, 91
STS-3, 92
subnet mask, 136
subnets, 135
subnets, IPv6, 158
subpoena, 578
subqueue, 417
subscription, multicast, 463
Sun RPC, 232
supplicant, WPA, 64
switch, 10
switch fabrics, 46
symbol, data, 44, 87
symmetric ciphers, 564
SYN, 239
synchronization source, RTP, 475
synchronized loss hypothesis, TCP, 302
synchronized loss, TCP, 268, 297, 301

T
T/TCP, 256
T1 line, 90
T3 line, 90
tables, SNMP, 491
Tahoe, 28, 265
tail drop, 289
TCP, 7, 235
TCP accelerated open, 256
TCP checksum offloading, 244
TCP Cubic, 335
TCP fairness, 297, 304
TCP Fast Open, 256
TCP Friendliness, 307
TCP Hamilton, 335
TCP header, 238
TCP Hybla, 333
612

TCP Illinois, 333

TCP NewReno, 276
TCP NewReno in ns-2, 381
TCP old duplicates, 251
TCP Reno, 265, 273
TCP sawtooth, 266, 270, 314, 351, 377, 388
TCP state diagram, 248
TCP Tahoe, 265
TCP timeout interval, 460
TCP Vegas, 387
TCP Westwood, 330
TCP Westwood+, 331
TCP, Highspeed, 325
TCP, SACK, 279
TDM, 89
Teredo tunneling, 129
terrestrial broadband, 70
terrestrial wireless, 70
TestAndIncr, 516
TEXTUAL-CONVENTION, SNMP, 514
TFTP, 225
TFTP scenarios, 229
thermonuclear, 27
three-way handshake, 239
three-way handshake, TCP, 253
threshold slow start, 270
throughput, 7
Time to Live, 128
time-division multiplexing, 89
timeout and retransmission, 22
timeout interval, TCP, 259, 460
TIMEWAIT, TCP, 252
TJX attack, 546
tls, 578
token bucket, 440
token bucket queue utilization, 445
token bus Ethernet, 74
token ring, 73
token-bucket applications, 444
token-bucket, RSVP, 466
topology, 11
topology table, EIGRP, 179
ToS and routing, 169
TP4, 27
trace file, ns-2, 352
tracefiles, ns-2, reading with python, 355
traceroute, 147
trading, 95
Index

An Introduction to Computer Networks, Release 1.0

traffic anomalies, 561

traffic management, 416
tragedy of the commons, 263
Trango, 72
transient queue peak, 379
transit capacity, 114
transit traffic, 200, 205
transmission, Ethernet, 38
traps, SNMP, 486
tree, 11
triggered updates, 175
triple DES, 567
Trivial File Transport Protocol, 225
trust and public keys, 577
Tspec, 466
TTL, 128
tunneling, 26, 55
two implementations, 28
twos-complement, 101
Type of Service, 128

VoIP and RTP, 476

VoIP bandwidth guarantees, 445
voting, 27
VPN, 55

UDP, 24, 215

UDP, for real-time traffic, 460
unbounded slow start, 270
unicast, 15
unique-local IPv6 address, 158
unlicensed spectrum, 57
unnumbered IP interface, 148
upgrades, network, 13
uplink-map, WiMAX, 69
URG, 239
User Datagram Protocol, 24
usmUserTable, 538

W^X, 555
weighted fair queuing, 418
WEP encryption failure, 546
WEP, Wi-Fi, 64
Westwood, TCP, 330
Wi-Fi, 57
Wi-Fi fragmentation, 61
Wi-Fi polling mode, 65
WiMAX, 68
window, 114
window scale option, TCP, 257
window size, 23, 113
Windows, 129, 251
Windows XP SP1 vulnerability, 558
winsize, 113
wireless, 57
wireless, fixed, 70
wireless, satellite, 73
wireless, terrestrial, 70
WireShark, 405
WireShark, TCP example, 243
work-conserving queuing, 418
WPA authenticator, 64
WPA supplicant, 64
WPA, Wi-Fi, 64
WPA-Enterprise, 64
WPA-Personal, 64
write-or-execute, 555

VACM, 510
VarBind list, 495
VCI, 75
video, streaming, 460
virtual circuit, 8, 17, 74
Virtual LANs, 51
virtual link, 55
virtual private network, 55
virtual tributary, 92
VLANs, 51
voice over IP, 18
VoIP, 18

XD page bit, 555

xkcd, 197, 560
XML, 224
XSS, 559

Index

613