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Computer Networking : Principles,

Protocols and Practice

Release 0.25

Olivier Bonaventure

October 30, 2011

Saylor URL: http://www.saylor.org/courses/cs402/

The Saylor Foundation

Computer Networking: Principles, Protocols, and Practice was written by Dr. Olivier Bonaventure of the
Universit catholique de Louvain for teaching Local Area Networks. After The Saylor Foundation accepted
his submission to Wave I of the Open Textbook Challenge, this textbook was relicensed as CC-BY 3.0.
Information on The Saylor Foundations Open Textbook Challenge can be found at www.saylor.org/otc/.
Computer Networking: Principles, Protocols and Practices October 31, 2011 by Olivier Bonaventure, is
licensed under a Creative Commons Attribution (CC BY) license made possible by funding from The Saylor
Foundation's Open Textbook Challenge in order to be incorporated into Saylor.org's collection of open courses
available at: http://www.saylor.org. Full license terms may be viewed at: http://creativecommons.org/licenses/
by/3.0/legalcode

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The Saylor Foundation

Contents

Preface

Introduction
2.1 Services and protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.2 The reference models . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3 Organisation of the book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

5
11
20
25

The application Layer

3.1 Principles . . . . . . . . . . . . . . . .
3.2 Application-level protocols . . . . . .
3.3 Writing simple networked applications
3.4 Summary . . . . . . . . . . . . . . . .
3.5 Exercises . . . . . . . . . . . . . . . .

27
27
32
55
61
61

.
.
.
.
.

The transport layer

4.1 Principles of a reliable transport protocol
4.2 The User Datagram Protocol . . . . . . .
4.3 The Transmission Control Protocol . . .
4.4 Summary . . . . . . . . . . . . . . . . .
4.5 Exercises . . . . . . . . . . . . . . . . .

.
.
.
.
.

67
. 67
. 87
. 89
. 113
. 114

The network layer

5.1 Principles . . . . . . .
5.2 Internet Protocol . . .
5.3 Routing in IP networks
5.4 Summary . . . . . . .
5.5 Exercises . . . . . . .

.
.
.
.
.

127
127
140
170
195
195

The datalink layer and the Local Area Networks

6.1 Principles . . . . . . . . . . . . . . . . . . .
6.2 Medium Access Control . . . . . . . . . . .
6.3 Datalink layer technologies . . . . . . . . .
6.4 Summary . . . . . . . . . . . . . . . . . . .
6.5 Exercises . . . . . . . . . . . . . . . . . . .

.
.
.
.
.

211
211
214
228
246
246

.
.
.
.
.

Glossary

249

Bibliography

255

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Indices and tables

257

Bibliography

259

Index

273

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Contents

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Contents

The Saylor Foundation

CHAPTER 1

Preface

This textbook came from a frustration of its main author. Many authors chose to write a textbook because there
are no textbooks in their field or because they are not satisfied with the existing textbooks. This frustration
has produced several excellent textbooks in the networking community. At a time when networking textbooks
were mainly theoretical, Douglas Comer chose to write a textbook entirely focused on the TCP/IP protocol suite
[Comer1988], a difficult choice at that time. He later extended his textbook by describing a complete TCP/IP
implementation, adding practical considerations to the theoretical descriptions in [Comer1988]. Richard Stevens
approached the Internet like an explorer and explained the operation of protocols by looking at all the packets
that were exchanged on the wire [Stevens1994]. Jim Kurose and Keith Ross reinvented the networking textbooks
by starting from the applications that the students use and later explained the Internet protocols by removing one
layer after the other [KuroseRoss09].
The frustrations that motivated this book are different. When I started to teach networking in the late 1990s,
students were already Internet users, but their usage was limited. Students were still using reference textbooks and
spent time in the library. Todays students are completely different. They are avid and experimented web users
who find lots of information on the web. This is a positive attitude since they are probably more curious than
their predecessors. Thanks to the information that is available on the Internet, they can check or obtain additional
information about the topics explained by their teachers. This abundant information creates several challenges for
a teacher. Until the end of the nineteenth century, a teacher was by definition more knowledgeable than his students
and it was very difficult for the students to verify the lessons given by their teachers. Today, given the amount
of information available at the fingertips of each student through the Internet, verifying a lesson or getting more
information about a given topic is sometimes only a few clicks away. Websites such as wikipedia provide lots of
information on various topics and students often consult them. Unfortunately, the organisation of the information
on these websites is not well suited to allow students to learn from them. Furthermore, there are huge differences
in the quality and depth of the information that is available for different topics.
The second reason is that the computer networking community is a strong participant in the open-source movement. Today, there are high-quality and widely used open-source implementations for most networking protocols.
This includes the TCP/IP implementations that are part of linux, freebsd or the uIP stack running on 8bits controllers, but also servers such as bind, unbound, apache or sendmail and implementations of routing protocols such
as xorp or quagga . Furthermore, the documents that define almost all of the Internet protocols have been developed within the Internet Engineering Task Force (IETF) using an open process. The IETF publishes its protocol
specifications in the publicly available RFC and new proposals are described in Internet drafts.
This open textbook aims to fill the gap between the open-source implementations and the open-source network
specifications by providing a detailed but pedagogical description of the key principles that guide the operation of
the Internet. The book is released under a creative commons licence. Such an open-source license is motivated
by two reasons. The first is that we hope that this will allow many students to use the book to learn computer
networks. The second is that I hope that other teachers will reuse, adapt and improve it. Time will tell if it is
possible to build a community of contributors to improve and develop the book further. As a starting point, the
first release contains all the material for a one-semester first upper undergraduate or a graduate networking course.
As of this writing, most of the text has been written by Olivier Bonaventure. Laurent Vanbever, Virginie Van den

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Schriek, Damien Saucez and Mickael Hoerdt have contributed to exercises. Pierre Reinbold designed the icons
used to represent switches and Nipaul Long has redrawn many figures in the SVG format. Stephane Bortzmeyer
sent many suggestions and corrections to the text. Additional information about the textbook is available at
http://inl.info.ucl.ac.be/CNP3

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Chapter 1. Preface

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CHAPTER 2

Introduction

When the first computers were built during the second world war, they were expensive and isolated. However,
after about twenty years, as their prices gradually decreased, the first experiments began to connect computers
together. In the early 1960s, researchers including Paul Baran, Donald Davies or Joseph Licklider independently
published the first papers describing the idea of building computer networks [Baran] [Licklider1963] . Given
the cost of computers, sharing them over a long distance was an interesting idea. In the US, the ARPANET
started in 1969 and continued until the mid 1980s [LCCD09]. In France, Louis Pouzin developed the Cyclades
network [Pouzin1975]. Many other research networks were built during the 1970s [Moore]. At the same time,
the telecommunication and computer industries became interested in computer networks. The telecommunication
industry bet on the X25. The computer industry took a completely different approach by designing Local Area
Networks (LAN). Many LAN technologies such as Ethernet or Token Ring were designed at that time. During
the 1980s, the need to interconnect more and more computers led most computer vendors to develop their own
suite of networking protocols. Xerox developed [XNS] , DEC chose DECNet [Malamud1991] , IBM developed
SNA [McFadyen1976] , Microsoft introduced NetBIOS [Winston2003] , Apple bet on Appletalk [SAO1990] . In
the research community, ARPANET was decommissioned and replaced by TCP/IP [LCCD09] and the reference
implementation was developed inside BSD Unix [McKusick1999]. Universities who were already running Unix
could thus adopt TCP/IP easily and vendors of Unix workstations such as Sun or Silicon Graphics included TCP/IP
in their variant of Unix. In parallel, the ISO, with support from the governments, worked on developing an open
1
Suite of networking protocols. In the end, TCP/IP became the de facto standard that is not only used within the
research community. During the 1990s and the early 2000s, the growth of the usage of TCP/IP continued, and
today proprietary protocols are seldom used. As shown by the figure below, that provides the estimation of the
number of hosts attached to the Internet, the Internet has sustained large growth throughout the last 20+ years.

Figure 2.1: Estimation of the number of hosts on the Internet

1 Open in ISO terms was in contrast with the proprietary protocol suites whose specification was not always publicly available. The US
government even mandated the usage of the OSI protocols (see RFC 1169), but this was not sufficient to encourage all users to switch to the
OSI protocol suite that was considered by many as too complex compared to other protocol suites.

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Recent estimations of the number of hosts attached to the Internet show a continuing growth since 20+ years.
However, although the number of hosts attached to the Internet is high, it should be compared to the number
of mobile phones that are in use today. More and more of these mobile phones will be connected to the Internet. Furthermore, thanks to the availability of TCP/IP implementations requiring limited resources such as uIP
[Dunkels2003], we can expect to see a growth of TCP/IP enabled embedded devices.

Figure 2.2: Estimation of the number of mobile phones

Before looking at the services provided by computer networks, it is useful to agree on some terminology that
is widely used in networking literature. First of all, computer networks are often classified in function of the
geographical area that they cover
LAN : a local area network typically interconnects hosts that are up to a few or maybe a few tens of kilometers apart.
MAN : a metropolitan area network typically interconnects devices that are up to a few hundred kilometers
apart
WAN : a wide area network interconnect hosts that can be located anywhere on Earth 2
Another classification of computer networks is based on their physical topology. In the following figures, physical
links are represented as lines while boxes show computers or other types of networking equipment.
Computer networks are used to allow several hosts to exchange information between themselves. To allow any
host to send messages to any other host in the network, the easiest solution is to organise them as a full-mesh, with
a direct and dedicated link between each pair of hosts. Such a physical topology is sometimes used, especially
when high performance and high redundancy is required for a small number of hosts. However, it has two major
drawbacks :
for a network containing n hosts, each host must have n-1 physical interfaces. In practice, the number of
physical interfaces on a node will limit the size of a full-mesh network that can be built
for a network containing n hosts, n(n1)
links are required. This is possible when there are a few nodes
2
in the same room, but rarely when they are located several kilometers apart
The second possible physical organisation, which is also used inside computers to connect different extension
cards, is the bus. In a bus network, all hosts are attached to a shared medium, usually a cable through a single
interface. When one host sends an electrical signal on the bus, the signal is received by all hosts attached to the bus.
A drawback of bus-based networks is that if the bus is physically cut, then the network is split into two isolated
networks. For this reason, bus-based networks are sometimes considered to be difficult to operate and maintain,
especially when the cable is long and there are many places where it can break. Such a bus-based topology was
used in early Ethernet networks.
A third organisation of a computer network is a star topology. In such topologies, hosts have a single physical
interface and there is one physical link between each host and the center of the star. The node at the center of
the star can be either a piece of equipment that amplifies an electrical signal, or an active device, such as a piece
2 In this book, we focus on networks that are used on Earth. These networks sometimes include satellite links. Besides the network
technologies that are used on Earth, researchers develop networking techniques that could be used between nodes located on different planets.
Such an Inter Planetary Internet requires different techniques than the ones discussed in this book. See RFC 4838 and the references therein
for information about these techniques.

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Figure 2.3: A Full mesh network

Figure 2.4: A network organised as a Bus

of equipment that understands the format of the messages exchanged through the network. Of course, the failure
of the central node implies the failure of the network. However, if one physical link fails (e.g. because the cable
has been cut), then only one node is disconnected from the network. In practice, star-shaped networks are easier
to operate and maintain than bus-shaped networks. Many network administrators also appreciate the fact that
they can control the network from a central point. Administered from a Web interface, or through a console-like
connection, the center of the star is a useful point of control (enabling or disabling devices) and an excellent
observation point (usage statistics).

Figure 2.5: A network organised as a Star

A fourth physical organisation of a network is the Ring topology. Like the bus organisation, each host has a single
physical interface connecting it to the ring. Any signal sent by a host on the ring will be received by all hosts
attached to the ring. From a redundancy point of view, a single ring is not the best solution, as the signal only
travels in one direction on the ring; thus if one of the links composing the ring is cut, the entire network fails. In
practice, such rings have been used in local area networks, but are now often replaced by star-shaped networks.
In metropolitan networks, rings are often used to interconnect multiple locations. In this case, two parallel links,
composed of different cables, are often used for redundancy. With such a dual ring, when one ring fails all the
traffic can be quickly switched to the other ring.
A fifth physical organisation of a network is the tree. Such networks are typically used when a large number of
customers must be connected in a very cost-effective manner. Cable TV networks are often organised as trees.
In practice, most real networks combine part of these topologies. For example, a campus network can be organised
as a ring between the key buildings, while smaller buildings are attached as a tree or a star to important buildings.

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Figure 2.6: A network organised as a Ring

Figure 2.7: A network organised as a Tree

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Chapter 2. Introduction

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Or an ISP network may have a full mesh of devices in the core of its network, and trees to connect remote users.
Throughout this book, our objective will be to understand the protocols and mechanisms that are necessary for a
network such as the one shown below.

ISP2

beta.be

alpha.com
R

ISP2
R
R

ISP2

societe.fr
PSTN
R

ISP1

R
tux@linux#

ADSL

Figure 2.8: A simple internetwork

The figure above illustrates an internetwork, i.e. a network that interconnects other networks. Each network is
illustrated as an ellipse containing a few devices. We will explain throughout the book the different types of
devices and their respective roles enabling all hosts to exchange information. As well as this, we will discuss how
networks are interconnected, and the rules that guide these interconnections. We will also analyse how the bus,
ring and mesh topologies are used to build real networks.
The last point of terminology we need to discuss is the transmission modes. When exchanging information through
a network, we often distinguish between three transmission modes. In TV and radio transmission, broadcast is
often used to indicate a technology that sends a video or radio signal to all receivers in a given geographical area.
Broadcast is sometimes used in computer networks, but only in local area networks where the number of recipients
is limited.
The first and most widespread transmission mode is called unicast . In the unicast transmission mode, information
is sent by one sender to one receiver. Most of todays Internet applications rely on the unicast transmission mode.
The example below shows a network with two types of devices : hosts (drawn as computers) and intermediate
nodes (drawn as cubes). Hosts exchange information via the intermediate nodes. In the example below, when
host S uses unicast to send information, it sends it via three intermediate nodes. Each of these nodes receives the
information from its upstream node or host, then processes and forwards it to its downstream node or host. This
is called store and forward and we will see later that this concept is key in computer networks.
A second transmission mode is multicast transmission mode. This mode is used when the same information must
be sent to a set of recipients. It was first used in LANs but later became supported in wide area networks. When
a sender uses multicast to send information to N receivers, the sender sends a single copy of the information and
the network nodes duplicate this information whenever necessary, so that it can reach all recipients belonging to
the destination group.
To understand the importance of multicast transmission, consider source S that sends the same information to
destinations A, C and E. With unicast, the same information passes three times on intermediate nodes 1 and 2 and
twice on node 4. This is a waste of resources on the intermediate nodes and on the links between them. With
multicast transmission, host S sends the information to node 1 that forwards it downstream to node 2. This node
creates a copy of the received information and sends one copy directly to host E and the other downstream to node
4. Upon reception of the information, node 4 produces a copy and forwards one to node A and another to node
C. Thanks to multicast, the same information can reach a large number of receivers while being sent only once on
each link.

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Figure 2.9: Unicast transmission

Figure 2.10: Multicast transmission

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The last transmission mode is the anycast transmission mode. It was initially defined in RFC 1542. In this
transmission mode, a set of receivers is identified. When a source sends information towards this set of receivers,
the network ensures that the information is delivered to one receiver that belongs to this set. Usually, the receiver
closest to the source is the one that receives the information sent by this particular source. The anycast transmission
mode is useful to ensure redundancy, as when one of the receivers fails, the network will ensure that information
will be delivered to another receiver belonging to the same group. However, in practice supporting the anycast
transmission mode can be difficult.

Figure 2.11: Anycast transmission

In the example above, the three hosts marked with * are part of the same anycast group. When host S sends
information to this anycast group, the network ensures that it will reach one of the members of the anycast group.
The dashed lines show a possible delivery via nodes 1, 2 and 4. A subsequent anycast transmission from host
S to the same anycast group could reach the host attached to intermediate node 3 as shown by the plain line.
An anycast transmission reaches a member of the anycast group that is chosen by the network in function of the
current network conditions.

2.1 Services and protocols

An important aspect to understand before studying computer networks is the difference between a service and a
protocol.
In order to understand the difference between the two, it is useful to start with real world examples. The traditional
Post provides a service where a postman delivers letters to recipients. The Post defines precisely which types of
letters (size, weight, etc) can be delivered by using the Standard Mail service. Furthermore, the format of the
envelope is specified (position of the sender and recipient addresses, position of the stamp). Someone who wants
to send a letter must either place the letter at a Post Office or inside one of the dedicated mailboxes. The letter
will then be collected and delivered to its final recipient. Note that for the regular service the Post usually does
not guarantee the delivery of each particular letter, some letters may be lost, and some letters are delivered to the
wrong mailbox. If a letter is important, then the sender can use the registered service to ensure that the letter will
be delivered to its recipient. Some Post services also provide an acknowledged service or an express mail service
that is faster than the regular service.
In computer networks, the notion of service is more formally defined in [X200] . It can be better understood by
considering a computer network, whatever its size or complexity, as a black box that provides a service to users ,
as shown in the figure below. These users could be human users or processes running on a computer system.
Many users can be attached to the same service provider. Through this provider, each user must be able to
exchange messages with any other user. To be able to deliver these messages, the service provider must be able
to unambiguously identify each user. In computer networks, each user is identified by a unique address, we will
discuss later how these addresses are built and used. At this point, and when considering unicast transmission, the
main characteristic of these addresses is that they are unique. Two different users attached to the network cannot
use the same address.
2.1. Services and protocols

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User B

User A
Service Access Point

Primitives
Service provider ("the network")

Figure 2.12: Users and service provider

Throughout this book, we will define a service as a set of capabilities provided by a system (and its underlying
elements) to its user. A user interacts with a service through a service access point. Note that as shown in the figure
above, users interact with one service provider. In practice, the service provider is distributed over several hosts,
but these are implementation details that are not important at this stage. These interactions between a user and a
service provider are expressed in [X200] by using primitives, as show in the figure below. These primitives are
an abstract representation of the interactions between a user and a service provider. In practice, these interactions
could be implemented as system calls for example.
User B

User A

X.request

X.confirm

X.response

X.indication

Service provider ("the network")

Figure 2.13: The four types of primitives

Four types of primitives are defined :
X.request. This type of primitive corresponds to a request issued by a user to a service provider
X.indication. This type of primitive is generated by the network provider and delivered to a user (often
related to an earlier and remote X.request primitive)
X.response. This type of primitive is generated by a user to answer to an earlier X.indication primitive
X.confirm. This type of primitive is delivered by the service provide to confirm to a user that a previous
X.request primitive has been successfully processed.
Primitives can be combined to model different types of services. The simplest service in computer networks is
called the connectionless service 3 . This service can be modelled by using two primitives :
Data.request(source,destination,SDU). This primitive is issued by a user that specifies, as parameters, its
(source) address, the address of the recipient of the message and the message itself. We will use Service
Data Unit (SDU) to name the message that is exchanged transparently between two users of a service.
Data.indication(source,destination,SDU). This primitive is delivered by a service provider to a user. It
contains as parameters a Service Data Unit as well as the addresses of the sender and the destination users.
When discussing the service provided in a computer network, it is often useful to be able to describe the interactions between the users and the provider graphically. A frequently used representation is the time-sequence
diagram. In this chapter and later throughout the book, we will often use diagrams such as the figure below. A
time-sequence diagram describes the interactions between two users and a service provider. By convention, the
users are represented in the left and right parts of the diagram while the service provider occupies the middle of the
diagram. In such a time-sequence diagram, time flows from the top, to the bottom of the diagram. Each primitive
3 This service is called the connectionless service because there is no need to create a connection before transmitting any data in contrast
with the connection-oriented service.

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is represented by a plain horizontal arrow, to which the name of the primitive is attached. The dashed lines are
used to represent the possible relationship between two (or more) primitives. Such a diagram provides information
about the ordering of the different primitives, but the distance between two primitives does not represent a precise
amount of time.
The figure below provides a representation of the connectionless service as a time-sequence diagram. The user on
the left, having address S, issues a Data.request primitive containing SDU M that must be delivered by the service
provider to destination D. The dashed line between the two primitives indicates that the Data.indication primitive
that is delivered to the user on the right corresponds to the Data.request primitive sent by the user on the left.
Source

Provider

Destination

DATA.request(S, D, "M")

DATA.indication(S, D, "M")

Time

Figure 2.14: A simple connectionless service

There are several possible implementations of the connectionless service, which we will discuss later in this book.
Before studying these realisations, it is useful to discuss the possible characteristics of the connectionless service.
A reliable connectionless service is a service where the service provider guarantees that all SDUs submitted in
Data.requests by a user will eventually be delivered to their destination. Such a service would be very useful for
users, but guaranteeing perfect delivery is difficult in practice. For this reason, computer networks usually support
an unreliable connectionless service.
An unreliable connectionless service may suffer from various types of problems compared to a reliable connectionless service. First of all, an unreliable connectionless service does not guarantee the delivery of all SDUs.
This can be expressed graphically by using the time-sequence diagram below.
In practice, an unreliable connectionless service will usually deliver a large fraction of the SDUs. However, since
the delivery of SDUs is not guaranteed, the user must be able to recover from the loss of any SDU.
A second imperfection that may affect an unreliable connectionless service is that it may duplicate SDUs. Some
unreliable connectionless service providers may deliver an SDU sent by a user twice or even more. This is
illustrated by the time-sequence diagram below.
Finally, some unreliable connectionless service providers may deliver to a destination a different SDU than the
one that was supplied in the Data.request. This is illustrated in the figure below.
When a user interacts with a service provider, it must precisely know the limitations of the underlying service to
be able to overcome any problem that may arise. This requires a precise definition of the characteristics of the
underlying service.
Another important characteristic of the connectionless service is whether it preserves the ordering of the SDUs
sent by one user. From the users viewpoint, this is often a desirable characteristic. This is illustrated in the figure
below.
However, many connectionless services, and in particular the unreliable services, do not guarantee that they will
always preserve the ordering of the SDUs sent by each user. This is illustrated in the figure below.
2.1. Services and protocols

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Source

Provider

Destination

DATA.request(S, D, "Msg")

Time

Figure 2.15: An unreliable connectionless service may loose SDUs

Source

Provider

Destination

DATA.request(S, D, "Msg")

DATA.indication(S, D, "Msg")

Time

Figure 2.16: An unreliable connectionless service may duplicate SDUs

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Source

Provider

Destination

DATA.request(S, D, "Msg")

DATA.indication(S, D, "XYZ")

Time

Figure 2.17: An unreliable connectionless service may deliver erroneous SDUs

Figure 2.18: A connectionless service that preserves the ordering of SDUs sent by a given user
Source

Provider

Destination

DATA.request(S, D, "A")

DATA.request(S, D, "B")
DATA.indication(S, D, "B")

DATA.indication(S, D, "A")

Time

Figure 2.19: A connectionless service that does not preserve the ordering of SDUs sent by a given user
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The connectionless service is widely used in computer networks as we will see later in this book. Several variations
to this basic service have been proposed. One of these is the confirmed connectionless service. This service uses
a Data.confirm primitive in addition to the classical Data.request and Data.indication primitives. This primitive
is issued by the service provider to confirm to a user the delivery of a previously sent SDU to its recipient. Note
that, like the registered service of the post office, the Data.confirm only indicates that the SDU has been delivered
to the destination user. The Data.confirm primitive does not indicate whether the SDU has been processed by the
destination user. This confirmed connectionless service is illustrated in the figure below.
Source

Provider

Destination

DATA.request(S, D, "M")

DATA.indication(S, D, "M")

DATA.confirm

Time

Figure 2.20: A confirmed connectionless service

The connectionless service we have described earlier is frequently used by users who need to exchange small
SDUs. Users needing to either send or receive several different and potentially large SDUs, or who need structured
exchanges often prefer the connection-oriented service.
An invocation of the connection-oriented service is divided into three phases. The first phase is the establishment
of a connection. A connection is a temporary association between two users through a service provider. Several
connections may exist at the same time between any pair of users. Once established, the connection is used to
transfer SDUs. Connections usually provide one bidirectional stream supporting the exchange of SDUs between
the two users that are associated through the connection. This stream is used to transfer data during the second
phase of the connection called the data transfer phase. The third phase is the termination of the connection. Once
the users have finished exchanging SDUs, they request to the service provider to terminate the connection. As we
will see later, there are also some cases where the service provider may need to terminate a connection itself.
The establishment of a connection can be modelled by using four primitives : Connect.request, Connect.indication,
Connect.response and Connect.confirm. The Connect.request primitive is used to request the establishment of a
connection. The main parameter of this primitive is the address of the destination user. The service provider
delivers a Connect.indication primitive to inform the destination user of the connection attempt. If it accepts to
establish a connection, it responds with a Connect.response primitive. At this point, the connection is considered
to be open and the destination user can start sending SDUs over the connection. The service provider processes
the Connect.response and will deliver a Connect.confirm to the user who initiated the connection. The delivery
of this primitive terminates the connection establishment phase. At this point, the connection is considered to be
open and both users can send SDUs. A successful connection establishment is illustrated below.
The example above shows a successful connection establishment. However, in practice not all connections are
successfully established. One reason is that the destination user may not agree, for policy or performance reasons,
to establish a connection with the initiating user at this time. In this case, the destination user responds to the
Connect.indication primitive by a Disconnect.request primitive that contains a parameter to indicate why the
connection has been refused. The service provider will then deliver a Disconnect.indication primitive to inform
the initiating user. A second reason is when the service provider is unable to reach the destination user. This
might happen because the destination user is not currently attached to the network or due to congestion. In these

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Source

Provider

Destination

CONNECT.request
CONNECT.indication

CONNECT.response

Destination considers
connection open

CONNECT.confirm

Source considers
connection open
Time

Figure 2.21: Connection establishment

cases, the service provider responds to the Connect.request with a Disconnect.indication primitive whose reason
parameter contains additional information about the failure of the connection.
Source

Provider

Destination

CONNECT.request
CONNECT.indication

DISCONNECT.request
DISCONNECT.indication

Connection rejected by destination

CONNECT.request
DISCONNECT.indication

Connection rejected
by provider
Time

Figure 2.22: Two types of rejection for a connection establishment attempt

Once the connection has been established, the service provider supplies two data streams to the communicating
users. The first data stream can be used by the initiating user to send SDUs. The second data stream allows
the responding user to send SDUs to the initiating user. The data streams can be organised in different ways. A
first organisation is the message-mode transfer. With the message-mode transfer, the service provider guarantees
that one and only one Data.indication will be delivered to the endpoint of the data stream for each Data.request
primitive issued by the other endpoint. The message-mode transfer is illustrated in the figure below. The main
advantage of the message-transfer mode is that the recipient receives exactly the SDUs that were sent by the other
user. If each SDU contains a command, the receiving user can process each command as soon as it receives a
SDU.
Unfortunately, the message-mode transfer is not widely used on the Internet. On the Internet, the most popular
connection-oriented service transfers SDUs in stream-mode. With the stream-mode, the service provider supplies a
byte stream that links the two communicating users. The sending user sends bytes by using Data.request primitives
that contain sequences of bytes as SDUs. The service provider delivers SDUs containing consecutive bytes to the
receiving user by using Data.indication primitives. The service provider ensures that all the bytes sent at one end
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Source

Provider

Destination

CONNECT.request
CONNECT.indication

CONNECT.response
CONNECT.confirm

DATA.request("A")
DATA.indication("A")

DATA.request("BCD")

DATA.indication("BCD")

DATA.request("EF")

DATA.indication("EF")
Time

Figure 2.23: Message-mode transfer in a connection oriented service

of the stream are delivered correctly in the same order at the other endpoint. However, the service provider does
not attempt to preserve the boundaries of the SDUs. There is no relation enforced by the service provider between
the number of Data.request and the number of Data.indication primitives. The stream-mode is illustrated in the
figure below. In practice, a consequence of the utilisation of the stream-mode is that if the users want to exchange
structured SDUs, they will need to provide the mechanisms that allow the receiving user to separate successive
SDUs in the byte stream that it receives. As we will see in the next chapter, application layer protocols often use
specific delimiters such as the end of line character to delineate SDUs in a bytestream.
Source

Provider

Destination

CONNECT.request
CONNECT.indication

CONNECT.response
CONNECT.confirm
DATA.request("AB")
DATA.indication("A")

DATA.request("CD")

DATA.indication("B")

DATA.request("EF")

DATA.indication("C")

Time

DATA.indication("DEF")

Figure 2.24: Stream-mode transfer in a connection oriented service

The third phase of a connection is when it needs to be released. As a connection involves three parties (two users
and one service provider), any of them can request the termination of the connection. Usually, connections are
terminated upon request of one user once the data transfer is finished. However, sometimes the service provider
may be forced to terminate a connection. This can be due to lack of resources inside the service provider or
because one of the users is not reachable anymore through the network. In this case, the service provider will issue
Disconnect.indication primitives to both users. These primitives will contain, as parameter, some information
about the reason for the termination of the connection. Unfortunately, as illustrated in the figure below, when a
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service provider is forced to terminate a connection it cannot guarantee that all SDUs sent by each user have been
delivered to the other user. This connection release is said to be abrupt as it can cause losses of data.
Source

Provider

Connection opened

Destination
Connection opened

DATA.request("A")
DATA.request("B")
DATA.indication("A")
DATA.indication("C")
DISCONNECT.indication

DISCONNECT.indication

Time

Figure 2.25: Abrupt connection release initiated by the service provider

An abrupt connection release can also be triggered by one of the users. If a user needs, for any reason, to terminate
a connection quickly, it can issue a Disconnect.request primitive and to request an abrupt release. The service
provider will process the request, stop the two data streams and deliver the Disconnect.indication primitive to the
remote user as soon as possible. As illustrated in the figure below, this abrupt connection release may cause losses
of SDUs.
Source

Provider

Connection opened

Destination
Connection opened

DATA.request("A")
DATA.request("B")
DATA.indication("A")
DISCONNECT.req(abrupt)

DATA.request("C")

DISCONNECT.indication

Time

Figure 2.26: Abrupt connection release initiated by a user

To ensure a reliable delivery of the SDUs sent by each user over a connection, we need to consider the two streams
that compose a connection as independent. A user should be able to release the stream that it uses to send SDUs
once it has sent all the SDUs that it planned to send over this connection, but still continue to receive SDUs over
the opposite stream. This graceful connection release is usually performed as shown in the figure below. One user
issues a Disconnect.request primitive to its provider once it has issued all its Data.request primitives. The service
provider will wait until all Data.indication primitives have been delivered to the receiving user before issuing the
Disconnnect.indication primitive. This primitive informs the receiving user that it will no longer receive SDUs
over this connection, but it is still able to issue Data.request primitives on the stream in the opposite direction.
Once the user has issued all of its Data.request primitives, it issues a Disconnnect.request primitive to request the
termination of the remaining stream. The service provider will process the request and deliver the corresponding
Disconnect.indication to the other user once it has delivered all the pending Data.indication primitives. At this

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point, all data has been delivered and the two streams have been released successfully and the connection is
completely closed.
Source

Provider

Connection opened

Destination
Connection opened

DATA.request("A")
DATA.request("C")

DATA.request("B")

DATA.indication("A")

DISCONNECT.req(graceful)

Source -> Destination

connection closed

DATA.indication("B")
DISCONNECT.ind(graceful)

DATA.indication("C")

DATA.request("D")
DATA.indication("D")

DISCONNECT.req(graceful)

DISCONNECT.ind(graceful)

Connection closed

Time

Figure 2.27: Graceful connection release

Note: Reliability of the connection-oriented service

An important point to note about the connection-oriented service is its reliability. A connection-oriented service
can only guarantee the correct delivery of all SDUs provided that the connection has been released gracefully. This
implies that while the connection is active, there is no guarantee for the actual delivery of the SDUs exchanged as
the connection may need to be released abruptly at any time.

2.2 The reference models

Given the growing complexity of computer networks, during the 1970s network researchers proposed various
reference models to facilitate the description of network protocols and services. Of these, the Open Systems
Interconnection (OSI) model [Zimmermann80] was probably the most influential. It served as the basis for the
standardisation work performed within the ISO to develop global computer network standards. The reference
model that we use in this book can be considered as a simplified version of the OSI reference model 4 .

2.2.1 The five layers reference model

Our reference model is divided into five layers, as shown in the figure below.
Starting from the bottom, the first layer is the Physical layer. Two communicating devices are linked through a
physical medium. This physical medium is used to transfer an electrical or optical signal between two directly
connected devices. Several types of physical mediums are used in practice :
electrical cable. Information can be transmitted over different types of electrical cables. The most common
ones are the twisted pairs that are used in the telephone network, but also in enterprise networks and coaxial
cables. Coaxial cables are still used in cable TV networks, but are no longer used in enterprise networks.
Some networking technologies operate over the classical electrical cable.
optical fiber. Optical fibers are frequently used in public and enterprise networks when the distance between the communication devices is larger than one kilometer. There are two main types of optical fibers
: multimode and monomode. Multimode is much cheaper than monomode fiber because a LED can be
4

An interesting historical discussion of the OSI-TCP/IP debate may be found in [Russel06]

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Application
Transport
Network
Datalink
Physical
Physical transmission medium

Figure 2.28: The five layers of the reference model

used to send a signal over a multimode fiber while a monomode fiber must be driven by a laser. Due to the
different modes of propagation of light, monomode fibers are limited to distances of a few kilometers while
multimode fibers can be used over distances greater than several tens of kilometers. In both cases, repeaters
can be used to regenerate the optical signal at one endpoint of a fiber to send it over another fiber.
wireless. In this case, a radio signal is used to encode the information exchanged between the communicating devices. Many types of modulation techniques are used to send information over a wireless channel
and there is lot of innovation in this field with new techniques appearing every year. While most wireless
networks rely on radio signals, some use a laser that sends light pulses to a remote detector. These optical
techniques allow to create point-to-point links while radio-based techniques, depending on the directionality
of the antennas, can be used to build networks containing devices spread over a small geographical area.
An important point to note about the Physical layer is the service that it provides. This service is usually an
unreliable connection-oriented service that allows the users of the Physical layer to exchange bits. The unit of
information transfer in the Physical layer is the bit. The Physical layer service is unreliable because :
the Physical layer may change, e.g. due to electromagnetic interferences, the value of a bit being transmitted
the Physical layer may deliver more bits to the receiver than the bits sent by the sender
the Physical layer may deliver fewer bits to the receiver than the bits sent by the sender
The last two points may seem strange at first glance. When two devices are attached through a cable, how is it
possible for bits to be created or lost on such a cable ?
This is mainly due to the fact that the communicating devices use their own clock to transmit bits at a given bit
rate. Consider a sender having a clock that ticks one million times per second and sends one bit every tick. Every
microsecond, the sender sends an electrical or optical signal that encodes one bit. The senders bit rate is thus 1
Mbps. If the receiver clock ticks exactly 5 every microsecond, it will also deliver 1 Mbps to its user. However, if
the receivers clock is slightly faster (resp. slower), than it will deliver slightly more (resp. less) than one million
bits every second. This explains why the physical layer may lose or create bits.
Note: Bit rate
In computer networks, the bit rate of the physical layer is always expressed in bits per second. One Mbps is one
million bits per second and one Gbps is one billion bits per second. This is in contrast with memory specifications that are usually expressed in bytes (8 bits), KiloBytes ( 1024 bytes) or MegaBytes (1048576 bytes). Thus
transferring one MByte through a 1 Mbps link lasts 8.39 seconds.
5 Having perfectly synchronised clocks running at a high frequency is very difficult in practice. However, some physical layers introduce a
feedback loop that allows the receivers clock to synchronise itself automatically to the senders clock. However, not all physical layers include
this kind of synchronisation.

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Bit rate
1 Kbps
1 Mbps
1 Gbps
1 Tbps

Bits per second

103
106
109
1012

Bits

Physical layer 01010010100010101001010

Physical layer

Physical transmission medium

Figure 2.29: The Physical layer

The physical layer allows thus two or more entities that are directly attached to the same transmission medium to
exchange bits. Being able to exchange bits is important as virtually any information can be encoded as a sequence
of bits. Electrical engineers are used to processing streams of bits, but computer scientists usually prefer to deal
with higher level concepts. A similar issue arises with file storage. Storage devices such as hard-disks also store
streams of bits. There are hardware devices that process the bit stream produced by a hard-disk, but computer
scientists have designed filesystems to allow applications to easily access such storage devices. These filesystems
are typically divided into several layers as well. Hard-disks store sectors of 512 bytes or more. Unix filesystems
group sectors in larger blocks that can contain data or inodes representing the structure of the filesystem. Finally, applications manipulate files and directories that are translated in blocks, sectors and eventually bits by the
operating system.
Computer networks use a similar approach. Each layer provides a service that is built above the underlying layer
and is closer to the needs of the applications.
The Datalink layer builds on the service provided by the underlying physical layer. The Datalink layer allows
two hosts that are directly connected through the physical layer to exchange information. The unit of information
exchanged between two entities in the Datalink layer is a frame. A frame is a finite sequence of bits. Some
Datalink layers use variable-length frames while others only use fixed-length frames. Some Datalink layers
provide a connection-oriented service while others provide a connectionless service. Some Datalink layers provide
reliable delivery while others do not guarantee the correct delivery of the information.
An important point to note about the Datalink layer is that although the figure below indicates that two entities
of the Datalink layer exchange frames directly, in reality this is slightly different. When the Datalink layer entity
on the left needs to transmit a frame, it issues as many Data.request primitives to the underlying physical layer
as there are bits in the frame. The physical layer will then convert the sequence of bits in an electromagnetic
or optical signal that will be sent over the physical medium. The physical layer on the right hand side of the
figure will decode the received signal, recover the bits and issue the corresponding Data.indication primitives to
its Datalink layer entity. If there are no transmission errors, this entity will receive the frame sent earlier.
Datalink

Frames

Physical

Datalink
Physical

Figure 2.30: The Datalink layer

The Datalink layer allows directly connected hosts to exchange information, but it is often necessary to exchange
information between hosts that are not attached to the same physical medium. This is the task of the network
layer. The network layer is built above the datalink layer. Network layer entities exchange packets. A packet is
a finite sequence of bytes that is transported by the datalink layer inside one or more frames. A packet usually

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Network
Datalink
Physical layer

Figure 2.33: The Application layer

2.2.2 The TCP/IP reference model

In contrast with OSI, the TCP/IP community did not spend a lot of effort defining a detailed reference model; in
fact, the goals of the Internet architecture were only documented after TCP/IP had been deployed [Clark88]. RFC
1122 , which defines the requirements for Internet hosts, mentions four different layers. Starting from the top,
these are :
an Application layer
a Transport layer
an Internet layer which is equivalent to the network layer of our reference model
a Link layer which combines the functionalities of the physical and datalink layers of our five-layer reference
model
Besides this difference in the lower layers, the TCP/IP reference model is very close to the five layers that we use
throughout this document.
2.2. The reference models

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2.2.3 The OSI reference model

Compared to the five layers reference model explained above, the OSI reference model defined in [X200] is
divided in seven layers. The four lower layers are similar to the four lower layers described above. The OSI
reference model refined the application layer by dividing it in three layers :
the Session layer. The Session layer contains the protocols and mechanisms that are necessary to organize
and to synchronize the dialogue and to manage the data exchange of presentation layer entities. While one
of the main functions of the transport layer is to cope with the unreliability of the network layer, the sessions
layer objective is to hide the possible failures of transport-level connections to the upper layer higher. For
this, the Session Layer provides services that allow to establish a session-connection, to support orderly data
exchange (including mechanisms that allow to recover from the abrupt release of an underlying transport
connection), and to release the connection in an orderly manner.
the Presentation layer was designed to cope with the different ways of representing information on computers. There are many differences in the way computer store information. Some computers store integers as
32 bits field, others use 64 bits field and the same problem arises with floating point number. For textual
information, this is even more complex with the many different character codes that have been used 6 . The
situation is even more complex when considering the exchange of structured information such as database
records. To solve this problem, the Presentation layer contains provides for a common representation of the
data transferred. The ASN.1 notation was designed for the Presentation layer and is still used today by some
protocols.
the Application layer that contains the mechanisms that do not fit in neither the Presentation nor the Session
layer. The OSI Application layer was itself further divided in several generic service elements.
Note: Where are the missing layers in TCP/IP reference model ?
The TCP/IP reference places the Presentation and the Session layers implicitly in the Application layer. The
main motivations for simplifying the upper layers in the TCP/IP reference model were pragmatic. Most Internet
applications started as prototypes that evolved and were later standardised. Many of these applications assumed
that they would be used to exchange information written in American English and for which the 7 bits US-ASCII
character code was sufficient. This was the case for email, but as well see in the next chapter, email was able to
evolve to support different character encodings. Some applications considered the different data representations
explicitly. For example, ftp contained mechanisms to convert a file from one format to another and the HTML
language was defined to represent web pages. On the other hand, many ISO specifications were developed by
committees composed of people who did not all participate in actual implementations. ISO spent a lot of effort
analysing the requirements and defining a solution that meets all of these requirements. Unfortunately, some of the
specifications were so complex that it was difficult to implement them completely and the standardisation bodies
defined recommended profiles that contained the implemented sets of options...

Figure 2.34: The seven layers of the OSI reference model

6 There is now a rough consensus for the greater use of the Unicode character format. Unicode can represent more than 100,000 different
characters from the known written languages on Earth. Maybe one day, all computers will only use Unicode to represent all their stored
characters and Unicode could become the standard format to exchange characters, but we are not yet at this stage today.

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2.3 Organisation of the book

This document is organised according to the TCP/IP reference model and follows a top-down approach. Most
of the classical networking textbooks chose a bottom-up approach, i.e. they first explained all the electrical and
optical details of the physical layer then moved to the datalink layer. This approach worked well during the infancy
of computer networks and until the late 1990s. At that time, most students were not users of computer networks
and it was useful to explain computer networks by building the corresponding protocols from the simplest, in the
physical layer, up to the application layer. Today, all students are active users of Internet applications, and starting
to learn computer networking by looking at bits is not very motivating. Starting from [KuroseRoss09], many
textbooks and teachers have chosen a top-down approach. This approach starts from applications such as email
and web that students already know and explores the different layers, starting from the application layer. This
approach works quite well with todays students. The traditional bottom-up approach could in fact be considered
as an engineering approach as it starts from the simple network that allows the exchange of bits, and explains how
to combine different protocols and mechanisms to build the most complex applications. The top-down approach
could on the other hand be considered as a scientific approach. Like biologists, it starts from an existing (manbuilt) system and explores it layer by layer.
Besides the top-down versus bottom-up organisation, computer networking books can either aim at having an
in-depth coverage of a small number of topics, or at having a limited coverage of a wide range of topics. Covering
a wide range of topics is interesting for introductory courses or for students who do not need a detailed knowledge
of computer networks. It allows the students to learn a little about everything and then start from this basic
knowledge later if they need to understand computer networking in more detail. This books chose to cover, in
detail, a smaller number of topics than other textbooks. This is motivated by the fact that computer networks often
need to be pushed to their limits. Understanding the details of the main networking protocols is important to be
able to fully grasp how a network behaves or extend it to provide innovative services 7 .
The book is organised as follows: We first describe the application layer in chapter The application Layer. Given
the large number of Internet-based applications, it is of course impossible to cover them all in detail. Instead we
focus on three types of Internet-based applications. We first study the Domain Name System (DNS) and then
explain some of the protocols involved in the exchange of electronic mail. The discussion of the application layer
ends with a description of the key protocols of the world wide web.
All these applications rely on the transport layer that is explained in chapter The transport layer. This is a key
layer in todays networks as it contains all the mechanisms necessary to provide a reliable delivery of data over an
unreliable network. We cover the transport layer by first developing a simple reliable transport layer protocol and
then explain the details of the TCP and UDP protocols used in TCP/IP networks.
After the transport layer, we analyse the network layer in chapter The network layer. This is also a very important
layer as it is responsible for the delivery of packets from any source to any destination through intermediate routers.
In the network layer, we describe the two possible organisations of the network layer and the routing protocols
based on link-state and distance vectors. Then we explain in detail the IPv4, IPv6, RIP, OSPF and BGP protocols
that are actually used in todays Internet.
The last chapter of the book is devoted to the datalink layer. In chapter The datalink layer and the Local Area
Networks, we begin by explaining the principles of the datalink layers on point-to-point links. Then, we focus on
the Local Area Networks. We first describe the Medium Access Control algorithms that allow multiple hosts to
share one transmission medium. We consider both opportunistic and deterministic techniques. We then explain in
detail two types of LANs that are important from a deployment viewpoint today : Ethernet and WiFi.

7 A popular quote says, the devil is in the details. This quote reflects very well the operation of many network protocols, where the change
of a single bit may have huge consequences. In computer networks, understanding all the details is sometimes necessary.

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CHAPTER 3

The application Layer

The Application Layer is the most important and most visible layer in computer networks. Applications reside in
this layer and human users interact via those applications through the network.
In this chapter, we first briefly describe the main principles of the application layer and focus on the two most
important application models : the client-server and the peer-to-peer models. Then, we review in detail two
families of protocols that have proved to be very useful in the Internet : electronic mail and the protocols that
allow access to information on the world wide web. We also describe the Domain Name System that allows
humans to use user-friendly names while the hosts use 32 bits or 128 bits long IP addresses.

3.1 Principles
The are two important models used to organise a networked application. The first and oldest model is the clientserver model. In this model, a server provides services to clients that exchange information with it. This model is
highly asymmetrical : clients send requests and servers perform actions and return responses. It is illustrated in
the figure below.

Figure 3.1: The client-server model

The client-server model was the first model to be used to develop networked applications. This model comes
naturally from the mainframes and minicomputers that were the only networked computers used until the 1980s.
A minicomputer is a multi-user system that is used by tens or more users at the same time. Each user interacts
with the minicomputer by using a terminal. Those terminals, were mainly a screen, a keyboard and a cable directly
connected to the minicomputer.
There are various types of servers as well as various types of clients. A web server provides information in
response to the query sent by its clients. A print server prints documents sent as queries by the client. An
email server will forward towards their recipient the email messages sent as queries while a music server will
deliver the music requested by the client. From the viewpoint of the application developer, the client and the
server applications directly exchange messages (the horizontal arrows labelled Queries and Responses in the
above figure), but in practice these messages are exchanged thanks to the underlying layers (the vertical arrows in
the above figure). In this chapter, we focus on these horizontal exchanges of messages.
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Networked applications do not exchange random messages. In order to ensure that the server is able to understand
the queries sent by a client, and also that the client is able to understand the responses sent by the server, they must
both agree on a set of syntactical and semantic rules. These rules define the format of the messages exchanged as
well as their ordering. This set of rules is called an application-level protocol.
An application-level protocol is similar to a structured conversation between humans. Assume that Alice wants
to know the current time but does not have a watch. If Bob passes close by, the following conversation could take
place :
Alice : Hello
Bob : Hello
Alice : What time is it ?
Bob : 11:55
Alice : Thank you
Bob : Youre welcome
Such a conversation succeeds if both Alice and Bob speak the same language. If Alice meets Tchang who only
speaks Chinese, she wont be able to ask him the current time. A conversation between humans can be more
complex. For example, assume that Bob is a security guard whose duty is to only allow trusted secret agents to
enter a meeting room. If all agents know a secret password, the conversation between Bob and Trudy could be as
follows :
Bob : What is the secret password ?
Trudy : 1234
Bob : This is the correct password, youre welcome
If Alice wants to enter the meeting room but does not know the password, her conversation could be as follows :
Bob : What is the secret password ?
Alice : 3.1415
Bob : This is not the correct password.
Human conversations can be very formal, e.g. when soldiers communicate with their hierarchy, or informal such
as when friends discuss. Computers that communicate are more akin to soldiers and require well-defined rules to
ensure an successful exchange of information. There are two types of rules that define how information can be
exchanged between computers :
syntactical rules that precisely define the format of the messages that are exchanged. As computers only
process bits, the syntactical rules specify how information is encoded as bit strings
organisation of the information flow. For many applications, the flow of information must be structured and
there are precedence relationships between the different types of information. In the time example above,
Alice must greet Bob before asking for the current time. Alice would not ask for the current time first and
greet Bob afterwards. Such precedence relationships exist in networked applications as well. For example,
a server must receive a username and a valid password before accepting more complex commands from its
clients.
Let us first discuss the syntactical rules. We will later explain how the information flow can be organised by
analysing real networked applications.
Application-layer protocols exchange two types of messages. Some protocols such as those used to support
electronic mail exchange messages expressed as strings or lines of characters. As the transport layer allows hosts
to exchange bytes, they need to agree on a common representation of the characters. The first and simplest method
to encode characters is to use the ASCII table. RFC 20 provides the ASCII table that is used by many protocols
on the Internet. For example, the table defines the following binary representations :
A : 1000011b
0 : 0110000b
z : 1111010b
28

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@ : 1000000b
space : 0100000b
In addition, the ASCII table also defines several non-printable or control characters. These characters were designed to allow an application to control a printer or a terminal. These control characters include CR and LF, that
are used to terminate a line, and the Bell character which causes the terminal to emit a sound.
carriage return (CR) : 0001101b
line feed (LF) : 0001010b
Bell: 0000111b
The ASCII characters are encoded as a seven bits field, but transmitted as an eight-bits byte whose high order bit
is usually set to 0. Bytes are always transmitted starting from the high order or most significant bit.
Most applications exchange strings that are composed of fixed or variable numbers of characters. A common
solution to define the character strings that are acceptable is to define them as a grammar using a Backus-Naur
Form (BNF) such as the Augmented BNF defined in RFC 5234. A BNF is a set of production rules that generate
all valid character strings. For example, consider a networked application that uses two commands, where the
user can supply a username and a password. The BNF for this application could be defined as shown in the figure
below.

Figure 3.2: A simple BNF specification

The example above defines several terminals and two commands : usercommand and passwordcommand. The
ALPHA terminal contains all letters in upper and lower case. In the ALPHA rule, %x41 corresponds to ASCII
character code 41 in hexadecimal, i.e. capital A. The CR and LF terminals correspond to the carriage return and
linefeed control characters. The CRLF rule concatenates these two terminals to match the standard end of line
termination. The DIGIT terminal contains all digits. The SP terminal corresponds to the white space characters.
The usercommand is composed of two strings separated by white space. In the ABNF rules that define the
messages used by Internet applications, the commands are case-insensitive. The rule user corresponds to all
possible cases of the letters that compose the word between brackets, e.g. user, uSeR, USER, usER, ... A username
contains at least one letter and up to 8 letters. User names are case-sensitive as they are not defined as a string
between brackets. The password rule indicates that a password starts with a letter and can contain any number of
letters or digits. The white space and the control characters cannot appear in a password defined by the above rule.
Besides character strings, some applications also need to exchange 16 bits and 32 bits fields such as integers. A
naive solution would have been to send the 16- or 32-bits field as it is encoded in the hosts memory. Unfortunately,
there are different methods to store 16- or 32-bits fields in memory. Some CPUs store the most significant byte
of a 16-bits field in the first address of the field while others store the least significant byte at this location. When
networked applications running on different CPUs exchange 16 bits fields, there are two possibilities to transfer
them over the transport service :
send the most significant byte followed by the least significant byte
send the least significant byte followed by the most significant byte
The first possibility was named big-endian in a note written by Cohen [Cohen1980] while the second was named
little-endian. Vendors of CPUs that used big-endian in memory insisted on using big-endian encoding in networked applications while vendors of CPUs that used little-endian recommended the opposite. Several studies
were written on the relative merits of each type of encoding, but the discussion became almost a religious issue
[Cohen1980]. Eventually, the Internet chose the big-endian encoding, i.e. multi-byte fields are always transmitted by sending the most significant byte first, RFC 791 refers to this encoding as the network-byte order. Most
3.1. Principles

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libraries 1 used to write networked applications contain functions to convert multi-byte fields from memory to the
network byte order and vice versa.
Besides 16 and 32 bit words, some applications need to exchange data structures containing bit fields of various
lengths. For example, a message may be composed of a 16 bits field followed by eight, one bit flags, a 24 bits
field and two 8 bits bytes. Internet protocol specifications will define such a message by using a representation
such as the one below. In this representation, each line corresponds to 32 bits and the vertical lines are used to
delineate fields. The numbers above the lines indicate the bit positions in the 32-bits word, with the high order bit
at position 0.

Figure 3.3: Message format

The message mentioned above will be transmitted starting from the upper 32-bits word in network byte order. The
first field is encoded in 16 bits. It is followed by eight one bit flags (A-H), a 24 bits field whose high order byte is
shown in the first line and the two low order bytes appear in the second line followed by two one byte fields. This
ASCII representation is frequently used when defining binary protocols. We will use it for all the binary protocols
that are discussed in this book.
We will discuss several examples of application-level protocols in this chapter.

3.1.1 The peer-to-peer model

The peer-to-peer model emerged during the last ten years as another possible architecture for networked applications. In the traditional client-server model, hosts act either as servers or as clients and a server serves a large
number of clients. In the peer-to-peer model, all hosts act as both servers and clients and they play both roles.
The peer-to-peer model has been used to develop various networked applications, ranging from Internet telephony
to file sharing or Internet-wide filesystems. A detailed description of peer-to-peer applications may be found in
[BYL2008]. Surveys of peer-to-peer protocols and applications may be found in [AS2004] and [LCP2005].

3.1.2 The transport services

Networked applications are built on top of the transport service. As explained in the previous chapter, there are
two main types of transport services :
the connectionless or datagram service
the connection-oriented or byte-stream service
The connectionless service allows applications to easily exchange messages or Service Data Units. On the Internet,
this service is provided by the UDP protocol that will be explained in the next chapter. The connectionless transport
service on the Internet is unreliable, but is able to detect transmission errors. This implies that an application will
not receive an SDU that has been corrupted due to transmission errors.
The connectionless transport service allows networked application to exchange messages. Several networked
applications may be running at the same time on a single host. Each of these applications must be able to exchange
SDUs with remote applications. To enable these exchanges of SDUs, each networked application running on a
host is identified by the following information :
the host on which the application is running
the port number on which the application listens for SDUs
1

For example, the htonl(3) (resp. ntohl(3)) function the standard C library converts a 32-bits unsigned integer from the byte order
used by the CPU to the network byte order (resp. from the network byte order to the CPU byte order). Similar functions exist in other
programming languages.

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Figure 3.7: BNF of the fully qualified host names

This grammar specifies that a host name is an ordered list of labels separated by the dot (.) character. Each label
can contain letters, numbers and the hyphen character (-) 4 . Fully qualified domain names are read from left to
right. The first label is a hostname or a domain name followed by the hierarchy of domains and ending with the
root implicitly at the right. The top-level domain name must be one of the registered TLDs 5 . For example, in the
above figure, www.whitehouse.gov corresponds to a host named www inside the whitehouse domain that belongs
to the gov top-level domain. info.ucl.ac.be corresponds to the info domain inside the ucl domain that is included
in the ac sub-domain of the be top-level domain.
This hierarchical naming scheme is a key component of the Domain Name System (DNS). The DNS is a distributed database that contains mappings between fully qualified domain names and IP addresses. The DNS uses
the client-server model. The clients are hosts that need to retrieve the mapping for a given name. Each nameserver
stores part of the distributed database and answers the queries sent by clients. There is at least one nameserver that
is responsible for each domain. In the figure below, domains are represented by circles and there are three hosts
inside domain dom (h1, h2 and h3) and three hosts inside domain a.sdom1.dom. As shown in the figure below, a
sub-domain may contain both host names and sub-domains.

Figure 3.8: A simple tree of domain names

An important point to note regarding the Domain Name System is its extensibility. Thanks to the Type and
RDLength fields, the format of the Resource Records can easily be extended. Furthermore, a DNS implementation
that receives a new Resource Record that it does not understand can ignore the record while still being able to
process the other parts of the message. This allows, for example, a DNS server that only supports IPv4 to ignore
the IPv6 addresses listed in the DNS reply for www.ietf.org while still being able to correctly parse the Resource
Records that it understands. This extensibility allowed the Domain Name System to evolve over the years while
still preserving the backward compatibility with already deployed DNS implementations.

3.2.2 Electronic mail

Electronic mail, or email, is a very popular application in computer networks such as the Internet. Email appeared
in the early 1970s and allows users to exchange text based messages. Initially, it was mainly used to exchange
short messages, but over the years its usage has grown. It is now not only used to exchange small, but also long
messages that can be composed of several parts as we will see later.
Before looking at the details of Internet email, let us consider a simple scenario illustrated in the figure below,
where Alice sends an email to Bob. Alice prepares her email by using an email clients and sends it to her email
server. Alices email server extracts Bobs address from the email and delivers the message to Bobs server. Bob
retrieves Alices message on his server and reads it by using his favourite email client or through his webmail
interface.

Figure 3.12: Simplified architecture of the Internet email

The email system that we consider in this book is composed of four components :
a message format, that defines how valid email messages are encoded
protocols, that allow hosts and servers to exchange email messages

3.2. Application-level protocols

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client software, that allows users to easily create and read email messages
software, that allows servers to efficiently exchange email messages
We will first discuss the format of email messages followed by the protocols that are used on todays Internet to
exchange and retrieve emails. Other email systems have been developed in the past [Bush1993] [Genilloud1990]
[GC2000], but today most email solutions have migrated to the Internet email. Information about the software
that is used to compose and deliver emails may be found on wikipedia among others, for both email clients and
email servers. More detailed information about the full Internet Mail Architecture may be found in RFC 5598.
Email messages, like postal mail, are composed of two parts :
a header that plays the same role as the letterhead in regular mail. It contains metadata about the message.
the body that contains the message itself.
Email messages are entirely composed of lines of ASCII characters. Each line can contain up to 998 characters
and is terminated by the CR and LF control characters RFC 5322. The lines that compose the header appear
before the message body. An empty line, containing only the CR and LF characters, marks the end of the header.
This is illustrated in the figure below.

Figure 3.13: The structure of email messages

The email header contains several lines that all begin with a keyword followed by a colon and additional information. The format of email messages and the different types of header lines are defined in RFC 5322. Two of these
header lines are mandatory and must appear in all email messages :
The sender address. This header line starts with From:. This contains the (optional) name of the sender
followed by its email address between < and >. Email addresses are always composed of a username
followed by the @ sign and a domain name.
The date. This header line starts with Date:. RFC 5322 precisely defines the format used to encode a date.
Other header lines appear in most email messages. The Subject: header line allows the sender to indicate the topic
discussed in the email. Three types of header lines can be used to specify the recipients of a message :
the To: header line contains the email addresses of the primary recipients of the message
addresses can be separated by using commas.

. Several

the cc: header line is used by the sender to provide a list of email addresses that must receive a carbon copy
of the message. Several addresses can be listed in this header line, separated by commas. All recipients of
the email message receive the To: and cc: header lines.
the bcc: header line is used by the sender to provide a list of comma separated email addresses that must
receive a blind carbon copy of the message. The bcc: header line is not delivered to the recipients of the
email message.
A simple email message containing the From:, To:, Subject: and Date: header lines and two lines of body is shown
below.
11

It could be surprising that the To: is not mandatory inside an email message. While most email messages will contain this header line an
email that does not contain a To: header line and that relies on the bcc: to specify the recipient is valid as well.

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From: Bob Smith <[email protected]>

To: Alice Doe <[email protected]>, Alice Smith <[email protected]>
Subject: Hello
Date: Mon, 8 Mar 2010 19:55:06 -0600
This is the "Hello world" of email messages.
This is the second line of the body

Value
0
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16

Encoding
A
B
C
D
E
F
G
H
I
J
K
L
M
N
O
P
Q

Value
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33

Encoding
R
S
T
U
V
W
X
Y
Z
a
b
c
d
e
f
g
h

Value
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50

Encoding
i
j
k
l
m
n
o
p
q
r
s
t
u
v
w
x
y

Value
51
52
53
54
55
56
57
58
59
60
61
62
63

Encoding
z
0
1
2
3
4
5
6
7
8
9
+
/

The example below, from RFC 4648, illustrates the Base64 encoding.
Input data
8-bit
6-bit
Decimal
Encoding

0x14fb9c03d97e
00010100 11111011 10011100 00000011 11011001 01111110
000101 001111 101110 011100 000000 111101 100101 111110
5 15 46 28 0 61 37 62
FPucA9l+

The last point to be discussed about base64 is what happens when the length of the sequence of bytes to be
encoded is not a multiple of three. In this case, the last group of bytes may contain one or two bytes instead of
three. Base64 reserves the = character as a padding character. This character is used twice when the last group
contains two bytes and once when it contains one byte as illustrated by the two examples below.
Input data
8-bit
6-bit
Decimal
Encoding

0x14
00010100
000101 000000
50
FA==

Input data
8-bit
6-bit
Decimal
Encoding

0x14b9
00010100 11111011
000101 001111 101100
5 15 44
FPs=

Now that we have explained the format of the email messages, we can discuss how these messages can be exchanged through the Internet. The figure below illustrates the protocols that are used when Alice sends an email
message to Bob. Alice prepares her email with an email client or on a webmail interface. To send her email to
Bob, Alices client will use the Simple Mail Transfer Protocol (SMTP) to deliver her message to her SMTP server.
Alices email client is configured with the name of the default SMTP server for her domain. There is usually at
least one SMTP server per domain. To deliver the message, Alices SMTP server must find the SMTP server that
contains Bobs mailbox. This can be done by using the Mail eXchange (MX) records of the DNS. A set of MX
records can be associated to each domain. Each MX record contains a numerical preference and the fully qualified
domain name of a SMTP server that is able to deliver email messages destined to all valid email addresses of this
domain. The DNS can return several MX records for a given domain. In this case, the server with the lowest
preference is used first. If this server is not reachable, the second most preferred server is used etc. Bobs SMTP
server will store the message sent by Alice until Bob retrieves it using a webmail interface or protocols such as the
Post Office Protocol (POP) or the Internet Message Access Protocol (IMAP).

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Figure 3.14: Email delivery protocols

The Simple Mail Transfer Protocol
The Simple Mail Transfer Protocol (SMTP) defined in RFC 5321 is a client-server protocol. The SMTP specification distinguishes between five types of processes involved in the delivery of email messages. Email messages
are composed on a Mail User Agent (MUA). The MUA is usually either an email client or a webmail. The MUA
sends the email message to a Mail Submission Agent (MSA). The MSA processes the received email and forwards
it to the Mail Transmission Agent (MTA). The MTA is responsible for the transmission of the email, directly or
via intermediate MTAs to the MTA of the destination domain. This destination MTA will then forward the message to the Mail Delivery Agent (MDA) where it will be accessed by the recipients MUA. SMTP is used for the
interactions between MUA and MSA 13 , MSA-MTA and MTA-MTA.
SMTP is a text-based protocol like many other application-layer protocols on the Internet. It relies on the bytestream service. Servers listen on port 25. Clients send commands that are each composed of one line of ASCII
text terminated by CR+LF. Servers reply by sending ASCII lines that contain a three digit numerical error/success
code and optional comments.
The SMTP protocol, like most text-based protocols, is specified as a BNF. The full BNF is defined in RFC 5321.
The main SMTP commands are defined by the BNF rules shown in the figure below.

Figure 3.15: BNF specification of the SMTP commands

In this BNF, atext corresponds to printable ASCII characters. This BNF rule is defined in RFC 5322. The five
main commands are EHLO, MAIL FROM:, RCPT TO:, DATA and QUIT 14 . Postmaster is the alias of the system
administrator who is responsible for a given domain or SMTP server. All domains must have a Postmaster alias.
The SMTP responses are defined by the BNF shown in the figure below.

Figure 3.16: BNF specification of the SMTP responses

SMTP servers use structured reply codes containing three digits and an optional comment. The first digit of
13

During the last years, many Internet Service Providers, campus and enterprise networks have deployed SMTP extensions RFC 4954 on
their MSAs. These extensions force the MUAs to be authenticated before the MSA accepts an email message from the MUA.
14 The first versions of SMTP used HELO as the first command sent by a client to a SMTP server. When SMTP was extended to support
newer features such as 8 bits characters, it was necessary to allow a server to recognise whether it was interacting with a client that supported
the extensions or not. EHLO became mandatory with the publication of RFC 2821.

3.2. Application-level protocols

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the reply code indicates whether the command was successful or not. A reply code of 2xy indicates that the
command has been accepted. A reply code of 3xy indicates that the command has been accepted, but additional
information from the client is expected. A reply code of 4xy indicates a transient negative reply. This means that
for some reason, which is indicated by either the other digits or the comment, the command cannot be processed
immediately, but there is some hope that the problem will only be transient. This is basically telling the client to
try the same command again later. In contrast, a reply code of 5xy indicates a permanent failure or error. In this
case, it is useless for the client to retry the same command later. Other application layer protocols such as FTP
RFC 959 or HTTP RFC 2616 use a similar structure for their reply codes. Additional details about the other reply
codes may be found in RFC 5321.
Examples of SMTP reply codes include the following :
500
501
502
503
220
221
421
250
450
452
550
354

Syntax error, command unrecognized

Syntax error in parameters or arguments
Command not implemented
Bad sequence of commands
<domain> Service ready
<domain> Service closing transmission channel
<domain> Service not available, closing transmission channel
Requested mail action okay, completed
Requested mail action not taken: mailbox unavailable
Requested action not taken: insufficient system storage
Requested action not taken: mailbox unavailable
Start mail input; end with <CRLF>.<CRLF>

The first four reply codes correspond to errors in the commands sent by the client. The fourth reply code would
be sent by the server when the client sends commands in an incorrect order (e.g. the client tries to send an email
before providing the destination address of the message). Reply code 220 is used by the server as the first message
when it agrees to interact with the client. Reply code 221 is sent by the server before closing the underlying
transport connection. Reply code 421 is returned when there is a problem (e.g. lack of memory/disk resources)
that prevents the server from accepting the transport connection. Reply code 250 is the standard positive reply that
indicates the success of the previous command. Reply codes 450 and 452 indicate that the destination mailbox
is temporarily unavailable, for various reasons, while reply code 550 indicates that the mailbox does not exist or
cannot be used for policy reasons. Reply code 354 indicates that the client can start transmitting its email message.
The transfer of an email message is performed in three phases. During the first phase, the client opens a transport
connection with the server. Once the connection has been established, the client and the server exchange greetings
messages (EHLO command). Most servers insist on receiving valid greeting messages and some of them drop the
underlying transport connection if they do not receive a valid greeting. Once the greetings have been exchanged,
the email transfer phase can start. During this phase, the client transfers one or more email messages by indicating
the email address of the sender (MAIL FROM: command), the email address of the recipient (RCPT TO: command)
followed by the headers and the body of the email message (DATA command). Once the client has finished sending
all its queued email messages to the SMTP server, it terminates the SMTP association (QUIT command).
A successful transfer of an email message is shown below
S:
C:
S:
C:
S:
C:
S:
C:
S:
C:
C:
C:
C:
C:
C:
C:
C:

220 smtp.example.com ESMTP MTA information

EHLO mta.example.org
250 Hello mta.example.org, glad to meet you
MAIL FROM:<[email protected]>
250 Ok
RCPT TO:<[email protected]>
250 Ok
DATA
354 End data with <CR><LF>.<CR><LF>
From: "Alice Doe" <[email protected]>
To: Bob Smith <[email protected]>
Date: Mon, 9 Mar 2010 18:22:32 +0100
Subject: Hello
Hello Bob
This is a small message containing 4 lines of text.
Best regards,

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C:
C:
S:
C:
S:

Alice
.
250 Ok: queued as 12345
QUIT
221 Bye

In the example above, the MTA running on mta.example.org opens a TCP connection to the SMTP server on host
smtp.example.com. The lines prefixed with S: (resp. C:) are the responses sent by the server (resp. the commands
sent by the client). The server sends its greetings as soon as the TCP connection has been established. The client
then sends the EHLO command with its fully qualified domain name. The server replies with reply-code 250 and
sends its greetings. The SMTP association can now be used to exchange an email.
To send an email, the client must first provide the address of the recipient with RCPT TO:. Then it uses the MAIL
FROM: with the address of the sender. Both the recipient and the sender are accepted by the server. The client
can now issue the DATA command to start the transfer of the email message. After having received the 354 reply
code, the client sends the headers and the body of its email message. The client indicates the end of the message
by sending a line containing only the . (dot) character 15 . The server confirms that the email message has been
queued for delivery or transmission with a reply code of 250. The client issues the QUIT command to close the
session and the server confirms with reply-code 221, before closing the TCP connection.
Note: Open SMTP relays and spam
Since its creation in 1971, email has been a very useful tool that is used by many users to exchange lots of
information. In the early days, all SMTP servers were open and anyone could use them to forward emails towards
their final destination. Unfortunately, over the years, some unscrupulous users have found ways to use email for
marketing purposes or to send malware. The first documented abuse of email for marketing purposes occurred in
1978 when a marketer who worked for a computer vendor sent a marketing email to many ARPANET users. At
that time, the ARPANET could only be used for research purposes and this was an abuse of the acceptable use
policy. Unfortunately, given the extremely low cost of sending emails, the problem of unsolicited emails has not
stopped. Unsolicited emails are now called spam and a study carried out by ENISA in 2009 reveals that 95% of
email was spam and this number seems to continue to grow. This places a burden on the email infrastructure of
Internet Service Providers and large companies that need to process many useless messages.
Given the amount of spam messages, SMTP servers are no longer open RFC 5068. Several extensions to SMTP
have been developed in recent years to deal with this problem. For example, the SMTP authentication scheme
defined in RFC 4954 can be used by an SMTP server to authenticate a client. Several techniques have also been
proposed to allow SMTP servers to authenticate the messages sent by their users RFC 4870 RFC 4871 .

The Post Office Protocol

When the first versions of SMTP were designed, the Internet was composed of minicomputers that were used by
an entire university department or research lab. These minicomputers were used by many users at the same time.
Email was mainly used to send messages from a user on a given host to another user on a remote host. At that
time, SMTP was the only protocol involved in the delivery of the emails as all hosts attached to the network were
running an SMTP server. On such hosts, an email destined to local users was delivered by placing the email in a
special directory or file owned by the user. However, the introduction of personal computers in the 1980s, changed
this environment. Initially, users of these personal computers used applications such as telnet to open a remote
session on the local minicomputer to read their email. This was not user-friendly. A better solution appeared
with the development of user friendly email client applications on personal computers. Several protocols were
designed to allow these client applications to retrieve the email messages destined to a user from his/her server.
Two of these protocols became popular and are still used today. The Post Office Protocol (POP), defined in RFC
1939, is the simplest one. It allows a client to download all the messages destined to a given user from his/her
email server. We describe POP briefly in this section. The second protocol is the Internet Message Access Protocol
(IMAP), defined in RFC 3501. IMAP is more powerful, but also more complex than POP. IMAP was designed to
allow client applications to efficiently access in real-time to messages stored in various folders on servers. IMAP
15 This implies that a valid email message cannot contain a line with one dot followed by CR and LF. If a user types such a line in an email,
his email client will automatically add a space character before or after the dot when sending the message over SMTP.

3.2. Application-level protocols

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assumes that all the messages of a given user are stored on a server and provides the functions that are necessary
to search, download, delete or filter messages.
POP is another example of a simple line-based protocol. POP runs above the bytestream service. A POP server
usually listens to port 110. A POP session is composed of three parts : an authorisation phase during which
the server verifies the clients credential, a transaction phase during which the client downloads messages and an
update phase that concludes the session. The client sends commands and the server replies are prefixed by +OK
to indicate a successful command or by -ERR to indicate errors.
When a client opens a transport connection with the POP server, the latter sends as banner an ASCII-line starting
with +OK. The POP session is at that time in the authorisation phase. In this phase, the client can send its
username (resp. password) with the USER (resp. PASS) command. The server replies with +OK if the username
(resp. password) is valid and -ERR otherwise.
Once the username and password have been validated, the POP session enters in the transaction phase. In this
phase, the client can issue several commands. The STAT command is used to retrieve the status of the server.
Upon reception of this command, the server replies with a line that contains +OK followed by the number of
messages in the mailbox and the total size of the mailbox in bytes. The RETR command, followed by a space and
an integer, is used to retrieve the nth message of the mailbox. The DELE command is used to mark for deletion
the nth message of the mailbox.
Once the client has retrieved and possibly deleted the emails contained in the mailbox, it must issue the QUIT
command. This command terminates the POP session and allows the server to delete all the messages that have
been marked for deletion by using the DELE command.
The figure below provides a simple POP session. All lines prefixed with C: (resp. S:) are sent by the client (resp.
server).
S:
C:
S:
C
S:
C:
S:
C:
S:
S:
S:
S:
C:
S:
S:
S:
C:
S:
C:
S:

+OK POP3 server ready

USER alice
+OK
PASS 12345pass
+OK alices maildrop has 2 messages (620 octets)
STAT
+OK 2 620
LIST
+OK 2 messages (620 octets)
1 120
2 500
.
RETR 1
+OK 120 octets
<the POP3 server sends message 1>
.
DELE 1
+OK message 1 deleted
QUIT
+OK POP3 server signing off (1 message left)

In this example, a POP client contacts a POP server on behalf of the user named alice. Note that in this example,
Alices password is sent in clear by the client. This implies that if someone is able to capture the packets sent by
Alice, he will know Alices password 16 . Then Alices client issues the STAT command to know the number of
messages that are stored in her mailbox. It then retrieves and deletes the first message of the mailbox.

3.2.3 The HyperText Transfer Protocol

In the early days of the Internet was mainly used for remote terminal access with telnet, email and file transfer.
The default file transfer protocol, FTP, defined in RFC 959 was widely used and FTP clients and servers are still
included in most operating systems.
16

RFC 1939 defines the APOP authentication scheme that is not vulnerable to such attacks.

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Many FTP clients offer a user interface similar to a Unix shell and allow the client to browse the file system on
the server and to send and retrieve files. FTP servers can be configured in two modes :
authenticated : in this mode, the ftp server only accepts users with a valid user name and password. Once
authenticated, they can access the files and directories according to their permissions
anonymous : in this mode, clients supply the anonymous userid and their email address as password. These
clients are granted access to a special zone of the file system that only contains public files.
ftp was very popular in the 1990s and early 2000s, but today it has mostly been superseded by more recent
protocols. Authenticated access to files is mainly done by using the Secure Shell (ssh) protocol defined in RFC
4251 and supported by clients such as scp or sftp. Nowadays, anonymous access is mainly provided by web
protocols.
In the late 1980s, high energy physicists working at CERN had to efficiently exchange documents about their
ongoing and planned experiments. Tim Berners-Lee evaluated several of the documents sharing techniques that
were available at that time [B1989]. As none of the existing solutions met CERNs requirements, they choose to
develop a completely new document sharing system. This system was initially called the mesh, but was quickly
renamed the world wide web. The starting point for the world wide web are hypertext documents. An hypertext
document is a document that contains references (hyperlinks) to other documents that the reader can immediately
access. Hypertext was not invented for the world wide web. The idea of hypertext documents was proposed in
1945 [Bush1945] and the first experiments were done during the 1960s [Nelson1965] [Myers1998] . Compared to
the hypertext documents that were used in the late 1980s, the main innovation introduced by the world wide web
was to allow hyperlinks to reference documents stored on remote machines.

Figure 3.17: World-wide web clients and servers

A document sharing system such as the world wide web is composed of three important parts.
1. A standardised addressing scheme that allows unambiguous identification of documents
2. A standard document format : the HyperText Markup Language
3. A standardised protocol that facilitates efficient retrieval of documents stored on a server
Note: Open standards and open implementations
Open standards have, and are still playing a key role in the success of the world wide web as we know it today. Without open standards, the world wide web would never have reached its current size. In addition to open
standards, another important factor for the success of the web was the availability of open and efficient implementations of these standards. When CERN started to work on the web, their objective was to build a running
system that could be used by physicists. They developed open-source implementations of the first web servers and
web clients. These open-source implementations were powerful and could be used as is, by institutions willing to

3.2. Application-level protocols

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share information on the web. They were also extended by other developers who contributed to new features. For
example, NCSA added support for images in their Mosaic browser that was eventually used to create Netscape
Communications.
The first components of the world wide web are the Uniform Resource Identifiers (URI), defined in RFC 3986. A
URI is a character string that unambiguously identifies a resource on the world wide web. Here is a subset of the
BNF for URIs
URI
scheme
authority
query
fragment
pchar
query
fragment
pct-encoded
unreserved
reserved
gen-delims
sub-delims

=
=
=
=
=

scheme ":" "//" authority path [ "?" query ] [ "#" fragment ]

By parsing this line, a server cannot determine which index.html file is requested.
Thanks to the
Host: header line, the server knows whether the request is for http://web.example.com/index.html or
http://www.dummy.net/index.html. Without the Host: header, this is impossible. The Host: header line allowed
web hosting companies to develop their business by supporting a large number of independent web servers on the
same physical server.
The status line of the HTTP response begins with the version of HTTP used by the server (usually HTTP/1.0
defined in RFC 1945 or HTTP/1.1 defined in RFC 2616) followed by a three digit status code and additional
information in English. HTTP status codes have a similar structure as the reply codes used by SMTP.
All status codes starting with digit 2 indicate a valid response. 200 Ok indicates that the HTTP request was
successfully processed by the server and that the response is valid.
All status codes starting with digit 3 indicate that the requested document is no longer available on the
server. 301 Moved Permanently indicates that the requested document is no longer available on this server.
A Location: header containing the new URI of the requested document is inserted in the HTTP response.
304 Not Modified is used in response to an HTTP request containing the If-Modified-Since: header. This
status line is used by the server if the document stored on the server is not more recent than the date indicated
in the If-Modified-Since: header.
All status codes starting with digit 4 indicate that the server has detected an error in the HTTP request sent
by the client. 400 Bad Request indicates a syntax error in the HTTP request. 404 Not Found indicates that
the requested document does not exist on the server.

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All status codes starting with digit 5 indicate an error on the server. 500 Internal Server Error indicates that
the server could not process the request due to an error on the server itself.
In both the HTTP request and the HTTP response, the MIME document refers to a representation of the document
with the MIME headers indicating the type of document and its size.
As an illustration of HTTP/1.0, the transcript below shows a HTTP request for http://www.ietf.org and the corresponding HTTP response. The HTTP request was sent using the curl command line tool. The User-Agent: header
line contains more information about this client software. There is no MIME document attached to this HTTP
request, and it ends with a blank line.
GET / HTTP/1.0
User-Agent: curl/7.19.4 (universal-apple-darwin10.0) libcurl/7.19.4 OpenSSL/0.9.8l zlib/1.2.3
Host: www.ietf.org

The HTTP response indicates the version of the server software used with the modules included. The LastModified: header indicates that the requested document was modified about one week before the request. A
HTML document (not shown) is attached to the response. Note the blank line between the header of the HTTP
response and the attached MIME document. The Server: header line has been truncated in this output.
HTTP/1.1 200 OK
Date: Mon, 15 Mar 2010 13:40:38 GMT
Server: Apache/2.2.4 (Linux/SUSE) mod_ssl/2.2.4 OpenSSL/0.9.8e (truncated)
Last-Modified: Tue, 09 Mar 2010 21:26:53 GMT
Content-Length: 17019
Content-Type: text/html
<!DOCTYPE HTML PUBLIC .../HTML>

HTTP was initially designed to share self-contained text documents. For this reason, and to ease the implementation of clients and servers, the designers of HTTP chose to open a TCP connection for each HTTP request.
This implies that a client must open one TCP connection for each URI that it wants to retrieve from a server as
illustrated on the figure below. For a web page containing only text documents this was a reasonable design choice
as the client usually remains idle while the (human) user is reading the retrieved document.

Figure 3.20: HTTP 1.0 and the underlying TCP connection

However, as the web evolved to support richer documents containing images, opening a TCP connection for each
URI became a performance problem [Mogul1995]. Indeed, besides its HTML part, a web page may include
dozens of images or more. Forcing the client to open a TCP connection for each component of a web page
has two important drawbacks. First, the client and the server must exchange packets to open and close a TCP
connection as we will see later. This increases the network overhead and the total delay of completely retrieving
all the components of a web page. Second, a large number of established TCP connections may be a performance
bottleneck on servers.
This problem was solved by extending HTTP to support persistent TCP connections RFC 2616. A persistent
connection is a TCP connection over which a client may send several HTTP requests. This is illustrated in the
figure below.

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Figure 3.21: HTTP 1.1 persistent connections

To allow the clients and servers to control the utilisation of these persistent TCP connections, HTTP 1.1 RFC
2616 defines several new HTTP headers :
The Connection: header is used with the Keep-Alive argument by the client to indicate that it expects the
underlying TCP connection to be persistent. When this header is used with the Close argument, it indicates
that the entity that sent it will close the underlying TCP connection at the end of the HTTP response.
The Keep-Alive: header is used by the server to inform the client about how it agrees to use the persistent
connection. A typical Keep-Alive: contains two parameters : the maximum number of requests that the
server agrees to serve on the underlying TCP connection and the timeout (in seconds) after which the server
will close an idle connection
The example below shows the operation of HTTP/1.1 over a persistent TCP connection to retrieve three URIs
stored on the same server. Once the connection has been established, the client sends its first request with the
Connection: keep-alive header to request a persistent connection.
GET / HTTP/1.1
Host: www.kame.net
User-Agent: Mozilla/5.0 (Macintosh; U; Intel Mac OS X 10_6_2; en-us)
Connection: keep-alive

The server replies with the Connection: Keep-Alive header and indicates that it accepts a maximum of 100 HTTP
requests over this connection and that it will close the connection if it remains idle for 15 seconds.
HTTP/1.1 200 OK
Date: Fri, 19 Mar 2010 09:23:37 GMT
Server: Apache/2.0.63 (FreeBSD) PHP/5.2.12 with Suhosin-Patch
Keep-Alive: timeout=15, max=100
Connection: Keep-Alive
Content-Length: 3462
Content-Type: text/html
<html>...

</html>

The client sends a second request for the style sheet of the retrieved web page.
GET /style.css HTTP/1.1
Host: www.kame.net
Referer: http://www.kame.net/
User-Agent: Mozilla/5.0 (Macintosh; U; Intel Mac OS X 10_6_2; en-us)
Connection: keep-alive

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The server replies with the requested style sheet and maintains the persistent connection. Note that the server only
accepts 99 remaining HTTP requests over this persistent connection.
HTTP/1.1 200 OK
Date: Fri, 19 Mar 2010 09:23:37 GMT
Server: Apache/2.0.63 (FreeBSD) PHP/5.2.12 with Suhosin-Patch
Last-Modified: Mon, 10 Apr 2006 05:06:39 GMT
Content-Length: 2235
Keep-Alive: timeout=15, max=99
Connection: Keep-Alive
Content-Type: text/css
...

Then the client automatically requests the web servers icon 18 , that could be displayed by the browser. This server
does not contain such URI and thus replies with a 404 HTTP status. However, the underlying TCP connection is
not closed immediately.
GET /favicon.ico HTTP/1.1
Host: www.kame.net
Referer: http://www.kame.net/
User-Agent: Mozilla/5.0 (Macintosh; U; Intel Mac OS X 10_6_2; en-us)
Connection: keep-alive
HTTP/1.1 404 Not Found
Date: Fri, 19 Mar 2010 09:23:40 GMT
Server: Apache/2.0.63 (FreeBSD) PHP/5.2.12 with Suhosin-Patch
Content-Length: 318
Keep-Alive: timeout=15, max=98
Connection: Keep-Alive
Content-Type: text/html; charset=iso-8859-1
<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML 2.0//EN"> ...

As illustrated above, a client can send several HTTP requests over the same persistent TCP connection. However,
it is important to note that all of these HTTP requests are considered to be independent by the server. Each HTTP
request must be self-contained. This implies that each request must include all the header lines that are required
by the server to understand the request. The independence of these requests is one of the important design choices
of HTTP. As a consequence of this design choice, when a server processes a HTTP request, it doesnt use any
other information than what is contained in the request itself. This explains why the client adds its User-Agent:
header in all of the HTTP requests it sends over the persistent TCP connection.
However, in practice, some servers want to provide content tuned for each user. For example, some servers
can provide information in several languages or other servers want to provide advertisements that are targeted to
different types of users. To do this, servers need to maintain some information about the preferences of each user
and use this information to produce content matching the users preferences. HTTP contains several mechanisms
that enable to solve this problem. We discuss three of them below.
A first solution is to force the users to be authenticated. This was the solution used by FTP to control the files that
each user could access. Initially, user names and passwords could be included inside URIs RFC 1738. However,
placing passwords in the clear in a potentially publicly visible URI is completely insecure and this usage has now
been deprecated RFC 3986. HTTP supports several extension headers RFC 2617 that can be used by a server
to request the authentication of the client by providing his/her credentials. However, user names and passwords
have not been popular on web servers as they force human users to remember one user name and one password
per server. Remembering a password is acceptable when a user needs to access protected content, but users will
not accept the need for a user name and password only to receive targeted advertisements from the web sites that
they visit.
A second solution to allow servers to tune that content to the needs and capabilities of the user is to rely on
the different types of Accept-* HTTP headers. For example, the Accept-Language: can be used by the client to
18 Favorite icons are small icons that are used to represent web servers in the toolbar of Internet browsers. Microsoft added this feature
in their browsers without taking into account the W3C standards. See http://www.w3.org/2005/10/howto-favicon for a discussion on how to
cleanly support such favorite icons.

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indicate its preferred languages. Unfortunately, in practice this header is usually set based on the default language
of the browser and it is not possible for a user to indicate the language it prefers to use by selecting options on
each visited web server.
The third, and widely adopted, solution are HTTP cookies. HTTP cookies were initially developed as a private
extension by Netscape. They are now part of the standard RFC 6265. In a nutshell, a cookie is a short string that
is chosen by a server to represent a given client. Two HTTP headers are used : Cookie: and Set-Cookie:. When a
server receives an HTTP request from a new client (i.e. an HTTP request that does not contain the Cookie: header),
it generates a cookie for the client and includes it in the Set-Cookie: header of the returned HTTP response. The
Set-Cookie: header contains several additional parameters including the domain names for which the cookie is
valid. The client stores all received cookies on disk and every time it sends a HTTP request, it verifies whether
it already knows a cookie for this domain. If so, it attaches the Cookie: header to the HTTP request. This is
illustrated in the figure below with HTTP 1.1, but cookies also work with HTTP 1.0.

Figure 3.22: HTTP cookies

Note: Privacy issues with HTTP cookies

The HTTP cookies introduced by Netscape are key for large e-commerce websites. However, they have also
raised many discussions concerning their potential misuses. Consider ad.com, a company that delivers lots of
advertisements on web sites. A web site that wishes to include ad.coms advertisements next to its content will
add links to ad.com inside its HTML pages. If ad.com is used by many web sites, ad.com could be able to track the
interests of all the users that visit its client websites and use this information to provide targeted advertisements.
Privacy advocates have even sued online advertisement companies to force them to comply with the privacy
regulations. More recent related technologies also raise privacy concerns

3.3 Writing simple networked applications

Networked applications were usually implemented by using the socket API. This API was designed when TCP/IP
was first implemented in the Unix BSD operating system [Sechrest] [LFJLMT], and has served as the model for
many APIs between applications and the networking stack in an operating system. Although the socket API is
very popular, other APIs have also been developed. For example, the STREAMS API has been added to several
Unix System V variants [Rago1993]. The socket API is supported by most programming languages and several
textbooks have been devoted to it. Users of the C language can consult [DC2009], [Stevens1998], [SFR2004] or
[Kerrisk2010]. The Java implementation of the socket API is described in [CD2008] and in the Java tutorial. In
this section, we will use the python implementation of the socket API to illustrate the key concepts. Additional
information about this API may be found in the socket section of the python documentation .

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The socket API is quite low-level and should be used only when you need a complete control of the network
access. If your application simply needs, for instance, to retrieve data with HTTP, there are much simpler and
higher-level APIs.
A detailed discussion of the socket API is outside the scope of this section and the references cited above provide
a detailed discussion of all the details of the socket API. As a starting point, it is interesting to compare the
socket API with the service primitives that we have discussed in the previous chapter. Let us first consider the
connectionless service that consists of the following two primitives :
DATA.request(destination,message) is used to send a message to a specified destination. In this socket API,
this corresponds to the send method.
DATA.indication(message) is issued by the transport service to deliver a message to the application. In the
socket API, this corresponds to the return of the recv method that is called by the application.
The DATA primitives are exchanged through a service access point. In the socket API, the equivalent to the service
access point is the socket. A socket is a data structure which is maintained by the networking stack and is used by
the application every time it needs to send or receive data through the networking stack. The socket method in the
python API takes two main arguments :
an address family that specifies the type of address family and thus the underlying networking
stack that will be used with the socket. This parameter can be either socket.AF_INET or
socket.AF_INET6. socket.AF_INET, which corresponds to the TCP/IPv4 protocol stack is the
default. socket.AF_INET6 corresponds to the TCP/IPv6 protocol stack.
a type indicates the type of service which is expected from the networking stack. socket.STREAM (the
default) corresponds to the reliable bytestream connection-oriented service. socket.DGRAM corresponds
to the connectionless service.
A simple client that sends a request to a server is often written as follows in descriptions of the socket API.
# A simple client of the connectionless service
import socket
import sys
HOSTIP=sys.argv[1]
PORT=int(sys.argv[2])
MSG="Hello, World!"
s = socket.socket( socket.AF_INET, socket.SOCK_DGRAM )
s.sendto( MSG, (HOSTIP, PORT) )

A typical usage of this application would be

python client.py 127.0.0.1 12345

where 127.0.0.1 is the IPv4 address of the host (in this case the localhost) where the server is running and
12345 the port of the server.
The first operation is the creation of the socket. Two parameters must be specified while creating a socket.
The first parameter indicates the address family and the second the socket type. The second operation is the
transmission of the message by using sendto to the server. It should be noted that sendto takes as arguments
the message to be transmitted and a tuple that contains the IPv4 address of the server and its port number.
The code shown above supports only the TCP/IPv4 protocol stack. To use the TCP/IPv6 protocol stack the
socket must be created by using the socket.AF_INET6 address family. Forcing the application developer to select TCP/IPv4 or TCP/IPv6 when creating a socket is a major hurdle for the deployment and usage
of TCP/IPv6 in the global Internet [Cheshire2010]. While most operating systems support both TCP/IPv4 and
TCP/IPv6, many applications still only use TCP/IPv4 by default. In the long term, the socket API should be
able to handle TCP/IPv4 and TCP/IPv6 transparently and should not force the application developer to always
specify whether it uses TCP/IPv4 or TCP/IPv6.
Another important issue with the socket API as supported by python is that it forces the application to deal with
IP addresses instead of dealing directly with domain names. This limitation dates from the early days of the
socket API in Unix 4.2BSD. At that time, the DNS was not widely available and only IP addresses could be
used. Most applications rely on DNS names to interact with servers and this utilisation of the DNS plays a very
important role to scale web servers and content distribution networks. To use domain names, the application needs
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to perform the DNS resolution by using the getaddrinfo method. This method queries the DNS and builds
the sockaddr data structure which is used by other methods of the socket API. In python, getaddrinfo takes
several arguments :
a name that is the domain name for which the DNS will be queried
an optional port number which is the port number of the remote server
an optional address family which indicates the address family used for the DNS request.
socket.AF_INET (resp. socket.AF_INET6) indicates that an IPv4 (IPv6) address is expected. Furthermore, the python socket API allows an application to use socket.AF_UNSPEC to indicate that it is
able to use either IPv4 or IPv6 addresses.
an optional socket type which can be either socket.SOCK_DGRAM or socket.SOCK_STREAM
In todays Internet hosts that are capable of supporting both IPv4 and IPv6, all applications should be
able to handle both IPv4 and IPv6 addresses. When used with the socket.AF_UNSPEC parameter, the
socket.getaddrinfo method returns a list of tuples containing all the information to create a socket.
import socket
socket.getaddrinfo(www.example.net,80,socket.AF_UNSPEC,socket.SOCK_STREAM)
[ (30, 1, 6, , (2001:db8:3080:3::2, 80, 0, 0)),
(2, 1, 6, , (203.0.113.225, 80))]

In the example above, socket.getaddrinfo returns two tuples. The first one corresponds to the sockaddr
containing the IPv6 address of the remote server and the second corresponds to the IPv4 information. Due to some
peculiarities of IPv6 and IPv4, the format of the two tuples is not exactly the same, but the key information in
both cases are the network layer address (2001:db8:3080:3::2 and 203.0.113.225) and the port number
(80). The other parameters are seldom used.
socket.getaddrinfo can be used to build a simple client that queries the DNS and contact the server by
using either IPv4 or IPv6 depending on the addresses returned by the socket.getaddrinfo method. The
client below iterates over the list of addresses returned by the DNS and sends its request to the first destination
address for which it can create a socket. Other strategies are of course possible. For example, a host running in
an IPv6 network might prefer to always use IPv6 when IPv6 is available 19 . Another example is the happy eyeballs
approach which is being discussed within the IETF [WY2011]. For example, [WY2011] mentions that some web
browsers try to use the first address returned by socket.getaddrinfo. If there is no answer within some
small delay (e.g. 300 milliseconds), the second address is tried.

import socket
import sys
HOSTNAME=sys.argv[1]
PORT=int(sys.argv[2])
MSG="Hello, World!"
for a in socket.getaddrinfo(HOSTNAME, PORT, socket.AF_UNSPEC,socket.SOCK_DGRAM,0, socket.AI_PASSIV
address_family,sock_type,protocol,canonicalname, sockaddr=a
try:
s = socket.socket(address_family, sock_type)
except socket.error:
s = None
print "Could not create socket"
continue
if s is not None:
s.sendto(MSG, sockaddr)
break

Now that we have described the utilisation of the socket API to write a simple client using the connectionless
transport service, let us have a closer look at the reliable byte stream transport service. As explained above, this
service is invoked by creating a socket of type socket.SOCK_STREAM. Once a socket has been created, a
client will typically connect to the remote server, send some data, wait for an answer and eventually close the
connection. These operations are performed by calling the following methods :
19 Most operating systems today by default prefer to use IPv6 when the DNS returns both an IPv4 and an IPv6 address for a name. See
http://ipv6int.net/systems/ for more detailed information.

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socket.connect : this method takes a sockaddr data structure, typically returned by

socket.getaddrinfo, as argument. It may fail and raise an exception if the remote server cannot
be reached.
socket.send : this method takes a string as argument and returns the number of bytes that were actually
sent. The string will be transmitted as a sequence of consecutive bytes to the remote server. Applications
are expected to check the value returned by this method and should resend the bytes that were not send.
socket.recv : this method takes an integer as argument that indicates the size of the buffer that has been
allocated to receive the data. An important point to note about the utilisation of the socket.recv method
is that as it runs above a bytestream service, it may return any amount of bytes (up to the size of the buffer
provided by the application). The application needs to collect all the received data and there is no guarantee
that some data sent by the remote host by using a single call to the socket.send method will be received
by the destination with a single call to the socket.recv method.
socket.shutdown : this method is used to release the underlying connection. On some platforms, it is
possible to specify the direction of transfer to be released (e.g. socket.SHUT_WR to release the outgoing
direction or socket.SHUT_RDWR to release both directions).
socket.close: this method is used to close the socket. It calls socket.shutdown if the underlying
connection is still open.
With these methods, it is now possible to write a simple HTTP client. This client operates over both IPv6 and IPv4
and writes the homepage of the remote server on the standard output. It also reports the number of socket.recv
calls that were used to retrieve the homepage 20 .
#!/usr/bin/python
# A simple http client that retrieves the first page of a web site
import socket, sys
if len(sys.argv)!=3 and len(sys.argv)!=2:
print "Usage : ",sys.argv[0]," hostname [port]"
hostname = sys.argv[1]
if len(sys.argv)==3 :
port=int(sys.argv[2])
else:
port = 80
READBUF=16384
s=None

# size of data read from web server

for res in socket.getaddrinfo(hostname, port, socket.AF_UNSPEC, socket.SOCK_STREAM):

af, socktype, proto, canonname, sa = res
# create socket
try:
s = socket.socket(af, socktype, proto)
except socket.error:
s = None
continue
# connect to remote host
try:
print "Trying "+sa[0]
s.connect(sa)
except socket.error, msg:
# socket failed
s.close()
s = None
continue
if s :
20 Experiments with the client indicate that the number of socket.recv calls can vary at each run. There are various factors that influence
the number of such calls that are required to retrieve some information from a server. Well discuss some of them after having explained the
operation of the underlying transport protocol.

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This server is far from a production-quality web server. A real web server would use multiple threads and/or
non-blocking IO to process a large number of concurrent requests 21 . Furthermore, it would also need to handle
all the errors that could happen while receiving data over a transport connection. These are outside the scope
of this section and additional information on more complex networked applications may be found elsewhere.
For example, [RG2010] provides an in-depth discussion of the utilisation of the socket API with python while
[SFR2004] remains an excellent source of information on the socket API in C.

3.4 Summary
In this chapter, we began by describing the client-server and peer-to-peer models. We then described, in detail,
three important families of protocols in the application layer. The Internet identifies hosts by using 32 bits IPv4
or 128 bits IPv6. However, using these addresses directly inside applications would be difficult for the humans
that use them. We have explained how the Domain Name System allows the mapping of names to corresponding
addresses. We have described both the DNS protocol that runs above UDP and the naming hierarchy. We have
then discussed one of the oldest applications on the Internet : electronic mail. We have described the format of
email messages and described the SMTP protocol that is used to send email messages as well as the POP protocol
that is used by email recipients to retrieve their email messages from their server. Finally, we have explained the
protocols that are used in the world wide web and the HyperText Transfer Protocol in particular.

3.5 Exercises
This section contains several exercises and small challenges about the application layer protocols.

3.5.1 The Domain Name System

The Domain Name System (DNS) plays a key role in the Internet today as it allows applications to use fully
qualified domain names (FQDN) instead of IPv4 or IPv6 addresses. Many tools allow to perform queries through
DNS servers. For this exercise, we will use dig which is installed on most Unix systems.
A typical usage of dig is as follows
dig @server -t type fqdn

where
server is the IP address or the name of a DNS server or resolver
type is the type of DNS record that is requested by the query such as NS for a nameserver, A for an IPv4
address, AAAA for an IPv6 address, MX for a mail relay, ...
fqdn is the fully qualified domain name being queried
1. What are the IP addresses of the resolvers that the dig implementation you are using relies on 22 ?
21 There are many production quality web servers software available. apache is a very complex but widely used one. thttpd and lighttpd are
less complex and their source code is probably easier to understand.
22 On a Linux machine, the Description section of the dig manpage tells you where dig finds the list of nameservers to query.

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The minimum command sent to a HTTP server is GET / HTTP/1.0 followed by CRLF and a blank line
Use dig to find the IP address used by totem.info.ucl.ac.be
33 The list of available language tags can be found at http://www.loc.gov/standards/iso639-2/php/code_list.php Additional information about
the support of multiple languages in Internet protocols may be found in RFC 5646
32

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Chapter 3. The application Layer

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CHAPTER 4

The transport layer

As the transport layer is built on top of the network layer, it is important to know the key features of the network
layer service. There are two types of network layer services : connectionless and connection-oriented. The
connectionless network layer service is the most widespread. Its main characteristics are :
the connectionless network layer service can only transfer SDUs of limited size 1
the connectionless network layer service may discard SDUs
the connectionless network layer service may corrupt SDUs
the connectionless network layer service may delay, reorder or even duplicate SDUs

Figure 4.1: The transport layer in the reference model

These imperfections of the connectionless network layer service will become much clearer once we have explained
the network layer in the next chapter. At this point, let us simply assume that these imperfections occur without
trying to understand why they occur.
Some transport protocols can be used on top of a connection-oriented network service, such as class 0 of the ISO
Transport Protocol (TP0) defined in [X224] , but they have not been widely used. We do not discuss in further
detail such utilisation of a connection-oriented network service in this book.
This chapter is organised as follows. We will first explain how it is possible to provide a reliable transport service
on top of an unreliable connectionless network service. For this, we explain the main mechanisms found in such
protocols. Then, we will study in detail the two transport protocols that are used in the Internet. We begin with the
User Datagram Protocol (UDP) which provides a simple connectionless transport service. Then, we will describe
in detail the Transmission Control Protocol (TCP), including its congestion control mechanism.

4.1 Principles of a reliable transport protocol

In this section, we depict a reliable transport protocol running above a connectionless network layer service. For
this, we first assume that the network layer provides a perfect service, i.e. :
the connectionless network layer service never corrupts SDUs
1

Many network layer services are unable to carry SDUs that are larger than 64 KBytes.

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the connectionless network layer service never discards SDUs

the connectionless network layer service never delays, reorders nor duplicate SDUs
the connectionless network layer service can support SDUs of any size
We will then remove each of these assumptions one after the other in order to better understand the mechanisms
used to solve each imperfection.

4.1.1 Reliable data transfer on top of a perfect network service

The transport layer entity interacts with both a user in the application layer and an entity in the network layer.
According to the reference model, these interactions will be performed using DATA.req and DATA.ind primitives.
However, to simplify the presentation and to avoid confusion between a DATA.req primitive issued by the user
of the transport layer entity, and a DATA.req issued by the transport layer entity itself, we will use the following
terminology :
the interactions between the user and the transport layer entity are represented by using the classical
DATA.req, DATA.ind primitives
the interactions between the transport layer entity and the network layer service are represented by using
send instead of DATA.req and recvd instead of DATA.ind
This is illustrated in the figure below.

Data.req

Data.ind

Transport
entity

Send

Recvd

Figure 4.2: Interactions between the transport layer, its user, and its network layer provider
When running on top of a perfect connectionless network service, a transport level entity can simply issue a
send(SDU) upon arrival of a DATA.req(SDU). Similarly, the receiver issues a DATA.ind(SDU) upon receipt of a
recvd(SDU). Such a simple protocol is sufficient when a single SDU is sent.
Unfortunately, this is not always sufficient to ensure a reliable delivery of the SDUs. Consider the case where a
client sends tens of SDUs to a server. If the server is faster that the client, it will be able to receive and process
all the segments sent by the client and deliver their content to its user. However, if the server is slower than the
client, problems may arise. The transport layer entity contains buffers to store SDUs that have been received
as a Data.request from the application but have not yet been sent via the network service. If the application is
faster than the network layer, the buffer becomes full and the operating system suspends the application to let the
transport entity empty its transmission queue. The transport entity also uses a buffer to store the segments received
from the network layer that have not yet been processed by the application. If the application is slow to process the
data, this buffer becomes full and the transport entity is not able to accept anymore the segments from the network
layer. The buffers of the transport entity have a limited size 2 and if they overflow, the transport entity is forced to
2 In the application layer, most servers are implemented as processes. The network and transport layer on the other hand are usually
implemented inside the operating system and the amount of memory that they can use is limited by the amount of memory allocated to the
entire kernel.

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Data.request(a)

Segment(a)
Data.ind(a)

Figure 4.3: The simplest transport protocol

discard received segments.
To solve this problem, our transport protocol must include a feedback mechanism that allows the receiver to inform
the sender that it has processed a segment and that another one can be sent. This feedback is required even though
the network layer provides a perfect service. To include such a feedback, our transport protocol must process two
types of segments :
data segments carrying a SDU
control segments carrying an acknowledgment indicating that the previous segment was processed correctly
These two types of segments can be distinguished using a segment composed of two parts :
the header that contains one bit set to 0 in data segments and set to 1 in control segments
the payload that contains the SDU supplied by the user application
The transport entity can then be modelled as a finite state machine, containing two states for the receiver and two
states for the sender. The figure below provides a graphical representation of this state machine with the sender
above and the receiver below.

Figure 4.4: Finite state machine of the simplest transport protocol

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The above FSM shows that the sender has to wait for an acknowledgement from the receiver before being able to
transmit the next SDU. The figure below illustrates the exchange of a few segments between two hosts.
A

Data.req(a)

D(a)

Data.req(b)

C(OK)

Data.ind(a)

D(b)
Data.ind(b)
C(OK)

Figure 4.5: Time sequence diagram illustrating the operation of the simplest transport protocol

4.1.2 Reliable data transfer on top of an imperfect network service

The transport layer must deal with the imperfections of the network layer service. There are three types of imperfections that must be considered by the transport layer :
1. Segments can be corrupted by transmission errors
2. Segments can be lost
3. Segments can be reordered or duplicated
To deal with these types of imperfections, transport protocols rely on different types of mechanisms. The first
problem is transmission errors. The segments sent by a transport entity is processed by the network and datalink
layers and finally transmitted by the physical layer. All of these layers are imperfect. For example, the physical
layer may be affected by different types of errors :
random isolated errors where the value of a single bit has been modified due to a transmission error
random burst errors where the values of n consecutive bits have been changed due to transmission errors
random bit creations and random bit removals where bits have been added or removed due to transmission
errors
The only solution to protect against transmission errors is to add redundancy to the segments that are sent. Information Theory defines two mechanisms that can be used to transmit information over a transmission channel
affected by random errors. These two mechanisms add redundancy to the information sent, to allow the receiver
to detect or sometimes even correct transmission errors. A detailed discussion of these mechanisms is outside the
scope of this chapter, but it is useful to consider a simple mechanism to understand its operation and its limitations.
Information theory defines coding schemes. There are different types of coding schemes, but let us focus on coding
schemes that operate on binary strings. A coding scheme is a function that maps information encoded as a string
of m bits into a string of n bits. The simplest coding scheme is the even parity coding. This coding scheme takes
an m bits source string and produces an m+1 bits coded string where the first m bits of the coded string are the bits
of the source string and the last bit of the coded string is chosen such that the coded string will always contain an
even number of bits set to 1. For example :
1001 is encoded as 10010
1101 is encoded as 11011

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This parity scheme has been used in some RAMs as well as to encode characters sent over a serial line. It is easy
to show that this coding scheme allows the receiver to detect a single transmission error, but it cannot correct it.
However, if two or more bits are in error, the receiver may not always be able to detect the error.
Some coding schemes allow the receiver to correct some transmission errors. For example, consider the coding
scheme that encodes each source bit as follows :
1 is encoded as 111
0 is encoded as 000
For example, consider a sender that sends 111. If there is one bit in error, the receiver could receive 011 or 101 or
110. In these three cases, the receiver will decode the received bit pattern as a 1 since it contains a majority of bits
set to 1. If there are two bits in error, the receiver will not be able anymore to recover from the transmission error.
This simple coding scheme forces the sender to transmit three bits for each source bit. However, it allows the
receiver to correct single bit errors. More advanced coding systems that allow to recover from errors are used in
several types of physical layers.
Transport protocols use error detection schemes, but none of the widely used transport protocols rely on error
correction schemes. To detect errors, a segment is usually divided into two parts :
a header that contains the fields used by the transport protocol to ensure reliable delivery. The header contains a checksum or Cyclical Redundancy Check (CRC) [Williams1993] that is used to detect transmission
errors
a payload that contains the user data passed by the application layer.
Some segment headers also include a length , which indicates the total length of the segment or the length of the
payload.
The simplest error detection scheme is the checksum. A checksum is basically an arithmetic sum of all the bytes
that a segment is composed of. There are different types of checksums. For example, an eight bit checksum can
be computed as the arithmetic sum of all the bytes of (both the header and trailer of) the segment. The checksum
is computed by the sender before sending the segment and the receiver verifies the checksum upon reception
of each segment. The receiver discards segments received with an invalid checksum. Checksums can be easily
implemented in software, but their error detection capabilities are limited. Cyclical Redundancy Checks (CRC)
have better error detection capabilities [SGP98], but require more CPU when implemented in software.
Note: Checksums, CRCs, ...
Most of the protocols in the TCP/IP protocol suite rely on the simple Internet checksum in order to verify that the
received segment has not been affected by transmission errors. Despite its popularity and ease of implementation,
the Internet checksum is not the only available checksum mechanism. Cyclical Redundancy Checks (CRC) are
very powerful error detection schemes that are used notably on disks, by many datalink layer protocols and file
formats such as zip or png. They can easily be implemented efficiently in hardware and have better error-detection
capabilities than the Internet checksum [SGP98] . However, when the first transport protocols were designed,
CRCs were considered to be too CPU-intensive for software implementations and other checksum mechanisms
were used instead. The TCP/IP community chose the Internet checksum, the OSI community chose the Fletcher
checksum [Sklower89] . Now, there are efficient techniques to quickly compute CRCs in software [Feldmeier95]
, the SCTP protocol initially chose the Adler-32 checksum but replaced it recently with a CRC (see RFC 3309).
The second imperfection of the network layer is that segments may be lost. As we will see later, the main cause
of packet losses in the network layer is the lack of buffers in intermediate routers. Since the receiver sends an
acknowledgement segment after having received each data segment, the simplest solution to deal with losses is to
use a retransmission timer. When the sender sends a segment, it starts a retransmission timer. The value of this
retransmission timer should be larger than the round-trip-time, i.e. the delay between the transmission of a data
segment and the reception of the corresponding acknowledgement. When the retransmission timer expires, the
sender assumes that the data segment has been lost and retransmits it. This is illustrated in the figure below.
Unfortunately, retransmission timers alone are not sufficient to recover from segment losses. Let us consider, as
an example, the situation depicted below where an acknowledgement is lost. In this case, the sender retransmits

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A
Data.req(a)

Data.req(b)

start timer

cancel timer
start timer

timer expires

D(a)
C(OK)
D(b)

Data.ind(a)
Lost segment

D(b)
Data.ind(b)
C(OK)

Figure 4.6: Using retransmission timers to recover from segment losses

the data segment that has not been acknowledged. Unfortunately, as illustrated in the figure below, the receiver
considers the retransmission as a new segment whose payload must be delivered to its user.

A
Data.req(a)

B
start timer

D(a)
C(OK)

Data.req(b)

Data.ind(a)

cancel timer
start timer

D(b)
C(OK)

timer expires
Lost segment

Data.ind(b)

D(b)
C(OK)

Data.ind(b)

!!!!!!

Figure 4.7: Limitations of retransmission timers

To solve this problem, transport protocols associate a sequence number to each data segment. This sequence
number is one of the fields found in the header of data segments. We use the notation D(S,...) to indicate a
data segment whose sequence number field is set to S. The acknowledgements also contain a sequence number
indicating the data segments that it is acknowledging. We use OKS to indicate an acknowledgement segment that
confirms the reception of D(S,...). The sequence number is encoded as a bit string of fixed length. The simplest
transport protocol is the Alternating Bit Protocol (ABP).
The Alternating Bit Protocol uses a single bit to encode the sequence number. It can be implemented easily. The
sender and the receivers only require a four states Finite State Machine.
72

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Wait
for
D(0,...)

Data.req(SDU)
send(D(0,SDU,CRC))
start_timer()

Wait
for
OK0/NAK

Recvd(C(NAK?)) OR
Recvd(C(OK1)) or timer expires
send(D(0,SDU,CRC))
restart_timer()

Recvd(C(OK0))
cancel_timer()

Recvd(C(OK1))
cancel_timer()

Wait
for
OK1/NAK

Data.req(SDU)
send(D(1,SDU,CRC))
start_timer()

Wait
for
D(1,...)

All corrupted
segments are
discarded in all states

Recvd(C(NAK?)) or recvd(C(OK0)) or timer expires

send(D(1,SDU,CRC))
restart-timer()

Figure 4.8: Alternating bit protocol : Sender FSM

The initial state of the sender is Wait for D(0,...). In this state, the sender waits for a Data.request. The first data
segment that it sends uses sequence number 0. After having sent this segment, the sender waits for an OK0 acknowledgement. A segment is retransmitted upon expiration of the retransmission timer or if an acknowledgement
with an incorrect sequence number has been received.
The receiver first waits for D(0,...). If the segment contains a correct CRC, it passes the SDU to its user and sends
OK0. If the segment contains an invalid CRC, it is immediately discarded. Then, the receiver waits for D(1,...).
In this state, it may receive a duplicate D(0,...) or a data segment with an invalid CRC. In both cases, it returns an
OK0 segment to allow the sender to recover from the possible loss of the previous OK0 segment.
Note: Dealing with corrupted segments
The receiver FSM of the Alternating bit protocol discards all segments that contain an invalid CRC. This is the
safest approach since the received segment can be completely different from the segment sent by the remote host.
A receiver should not attempt at extracting information from a corrupted segment because it cannot know which
portion of the segment has been affected by the error.
The figure below illustrates the operation of the alternating bit protocol.
The Alternating Bit Protocol can recover from transmission errors and segment losses. However, it has one
important drawback. Consider two hosts that are directly connected by a 50 Kbits/sec satellite link that has a 250
milliseconds propagation delay. If these hosts send 1000 bits segments, then the maximum throughput that can be
achieved by the alternating bit protocol is one segment every 20 + 250 + 250 = 520 milliseconds if we ignore the
transmission time of the acknowledgement. This is less than 2 Kbits/sec !
Go-back-n and selective repeat
To overcome the performance limitations of the alternating bit protocol, transport protocols rely on pipelining.
This technique allows a sender to transmit several consecutive segments without being forced to wait for an
acknowledgement after each segment. Each data segment contains a sequence number encoded in an n bits field.
Pipelining allows the sender to transmit segments at a higher rate, but we need to ensure that the receiver does not

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Recvd(D(1,SDU,CRC))
AND IsOK(CRC,SDU)
Send(C(OK1))
Wait
for
D(0,...)

Recvd(D(0,SDU,CRC))
AND IsOK(CRC,SDU)
Send(C(OK0))

Recvd(D(0,SDU,CRC))
AND IsOK(CRC,SDU)
Data.ind(SDU)

Process
SDU0
OK

Send(C(OK0))
Wait
for
D(1,...)

Recvd(D(1,SDU,CRC))
AND IsOK(CRC,SDU)
Data.ind(SDU)

Send(C(OK1))

All corrupted
segments are
discarded in all states

Process
SDU1
OK

Figure 4.9: Alternating bit protocol : Receiver FSM

Data.req(a)

D(0,a)
C(OK0)

Data.req(b)

Data.ind(a)

D(1, b)

Retransmission timer

D(1,b)
C(OK1)

Data.ind(b)
Duplicate detected

C(OK1)
Data.req(c)
D(0, c)
Retransmission timer
D(0, c)
Data.ind(c)

Figure 4.10: Operation of the alternating bit protocol

Figure 4.11: Pipelining to improve the performance of transport protocols

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become overloaded. Otherwise, the segments sent by the sender are not correctly received by the destination. The
transport protocols that rely on pipelining allow the sender to transmit W unacknowledged segments before being
forced to wait for an acknowledgement from the receiving entity.
This is implemented by using a sliding window. The sliding window is the set of consecutive sequence numbers
that the sender can use when transmitting segments without being forced to wait for an acknowledgement. The
figure below shows a sliding window containing five segments (6,7,8,9 and 10). Two of these sequence numbers
(6 and 7) have been used to send segments and only three sequence numbers (8, 9 and 10) remain in the sliding
window. The sliding window is said to be closed once all sequence numbers contained in the sliding window have
been used.

Figure 4.12: The sliding window

The figure below illustrates the operation of the sliding window. The sliding window shown contains three segments. The sender can thus transmit three segments before being forced to wait for an acknowledgement. The
sliding window moves to the higher sequence numbers upon reception of acknowledgements. When the first
acknowledgement (OK0) is received, it allows the sender to move its sliding window to the right and sequence
number 3 becomes available. This sequence number is used later to transmit SDU d.

Figure 4.13: Sliding window example

In practice, as the segment header encodes the sequence number in an n bits string, only the sequence numbers
between 0 and 2n 1 can be used. This implies that the same sequence number is used for different segments and
that the sliding window will wrap. This is illustrated in the figure below assuming that 2 bits are used to encode
the sequence number in the segment header. Note that upon reception of OK1, the sender slides its window and
can use sequence number 0 again.
Unfortunately, segment losses do not disappear because a transport protocol is using a sliding window. To recover
from segment losses, a sliding window protocol must define :
a heuristic to detect segment losses
a retransmission strategy to retransmit the lost segments.
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Figure 4.14: Utilisation of the sliding window with modulo arithmetic

The simplest sliding window protocol uses go-back-n recovery. Intuitively, go-back-n operates as follows. A
go-back-n receiver is as simple as possible. It only accepts the segments that arrive in-sequence. A go-backn receiver discards any out-of-sequence segment that it receives. When go-back-n receives a data segment, it
always returns an acknowledgement containing the sequence number of the last in-sequence segment that it has
received. This acknowledgement is said to be cumulative. When a go-back-n receiver sends an acknowledgement
for sequence number x, it implicitly acknowledges the reception of all segments whose sequence number is earlier
than x. A key advantage of these cumulative acknowledgements is that it is easy to recover from the loss of an
acknowledgement. Consider for example a go-back-n receiver that received segments 1, 2 and 3. It sent OK1,
OK2 and OK3. Unfortunately, OK1 and OK2 were lost. Thanks to the cumulative acknowledgements, when the
receiver receives OK3, it knows that all three segments have been correctly received.
The figure below shows the FSM of a simple go-back-n receiver. This receiver uses two variables : lastack and
next. next is the next expected sequence number and lastack the sequence number of the last data segment that has
been acknowledged. The receiver only accepts the segments that are received in sequence. maxseq is the number
of different sequence numbers (2n ).

All corrupted
segments are
discarded in all states

Recvd(D(next,SDU,CRC))
AND IsOK(CRC,SDU)
Data.ind(SDU)
Process
SDU
OK

Wait

Recvd(D(t<>next,SDU,CRC))
AND IsOK(CRC,SDU)
discard(SDU);
send(C(OK,lastack ,CRC));

send(C(OK,next,CRC));
lastack = next
next=(next+1) % maxseq;

Figure 4.15: Go-back-n : receiver FSM

A go-back-n sender is also very simple. It uses a sending buffer that can store an entire sliding window of
segments 3 . The segments are sent with increasing sequence number (modulo maxseq). The sender must wait for
3 The size of the sliding window can be either fixed for a given protocol or negotiated during the connection establishment phase. Well
see later that it is also possible to change the size of the sliding window during the connections lifetime.

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an acknowledgement once its sending buffer is full. When a go-back-n sender receives an acknowledgement, it
removes from the sending buffer all the acknowledged segments and uses a retransmission timer to detect segment
losses. A simple go-back-n sender maintains one retransmission timer per connection. This timer is started when
the first segment is sent. When the go-back-n sender receives an acknowledgement, it restarts the retransmission
timer only if there are still unacknowledged segments in its sending buffer. When the retransmission timer expires,
the go-back-n sender assumes that all the unacknowledged segments currently stored in its sending buffer have
been lost. It thus retransmits all the unacknowledged segments in the buffer and restarts its retransmission timer.
All corrupted
segments are
discarded in all states

Recvd(C(OK,t,CRC))
and CRCOK(C(OK,t,CRC))
remove acked seqs from buffer;
unpack = (t+1) % maxseq;
if (unpack==seq) {
cancel_timer(); } else
{ restart_timer(); }

Data.req(SDU)
size(buffer) < w
if (seq==unack) { start_timer ; }
insert_in_buffer(seq, SDU);
send(D(seq,SDU,CRC));
seq=(seq+1) % maxseq;
timer expires
for all (i, SDU) in buffer
{ send(D(i,SDU,CRC)); }
restart_timer();

Figure 4.16: Go-back-n : sender FSM

The operation of go-back-n is illustrated in the figure below. In this figure, note that upon reception of the outof-sequence segment D(2,c), the receiver returns a cumulative acknowledgement C(OK,0) that acknowledges all
the segments that have been received in sequence. The lost segment is retransmitted upon the expiration of the
retransmission timer.

Figure 4.17: Go-back-n : example

The main advantage of go-back-n is that it can be easily implemented, and it can also provide good performance
when only a few segments are lost. However, when there are many losses, the performance of go-back-n quickly
drops for two reasons :
the go-back-n receiver does not accept out-of-sequence segments
the go-back-n sender retransmits all unacknowledged segments once its has detected a loss
Selective repeat is a better strategy to recover from segment losses. Intuitively, selective repeat allows the receiver
to accept out-of-sequence segments. Furthermore, when a selective repeat sender detects losses, it only retransmits
the segments that have been lost and not the segments that have already been correctly received.
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A selective repeat receiver maintains a sliding window of W segments and stores in a buffer the out-of-sequence
segments that it receives. The figure below shows a five segment receive window on a receiver that has already
received segments 7 and 9.

Figure 4.18: The receiving window with selective repeat

A selective repeat receiver discards all segments having an invalid CRC, and maintains the variable lastack as
the sequence number of the last in-sequence segment that it has received. The receiver always includes the value
of lastack in the acknowledgements that it sends. Some protocols also allow the selective repeat receiver to
acknowledge the out-of-sequence segments that it has received. This can be done for example by placing the
list of the sequence numbers of the correctly received, but out-of-sequence segments in the acknowledgements
together with the lastack value.
When a selective repeat receiver receives a data segment, it first verifies whether the segment is inside its receiving
window. If yes, the segment is placed in the receive buffer. If not, the received segment is discarded and an
acknowledgement containing lastack is sent to the sender. The receiver then removes all consecutive segments
starting at lastack (if any) from the receive buffer. The payloads of these segments are delivered to the user,
lastack and the receiving window are updated, and an acknowledgement acknowledging the last segment received
in sequence is sent.
The selective repeat sender maintains a sending buffer that can store up to W unacknowledged segments. These
segments are sent as long as the sending buffer is not full. Several implementations of a selective repeat sender
are possible. A simple implementation is to associate a retransmission timer to each segment. The timer is started
when the segment is sent and cancelled upon reception of an acknowledgement that covers this segment. When a
retransmission timer expires, the corresponding segment is retransmitted and this retransmission timer is restarted.
When an acknowledgement is received, all the segments that are covered by this acknowledgement are removed
from the sending buffer and the sliding window is updated.
The figure below illustrates the operation of selective repeat when segments are lost. In this figure, C(OK,x) is
used to indicate that all segments, up to and including sequence number x have been received correctly.

Figure 4.19: Selective repeat : example

Pure cumulative acknowledgements work well with the go-back-n strategy. However, with only cumulative acknowledgements a selective repeat sender cannot easily determine which data segments have been correctly received after a data segment has been lost. For example, in the figure above, the second C(OK,0) does not inform

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explicitly the sender of the reception of D(2,c) and the sender could retransmit this segment although it has already been received. A possible solution to improve the performance of selective repeat is to provide additional
information about the received segments in the acknowledgements that are returned by the receiver. For example,
the receiver could add in the returned acknowledgement the list of the sequence numbers of all segments that
have already been received. Such acknowledgements are sometimes called selective acknowledgements. This is
illustrated in the figure below.
In the figure above, when the sender receives C(OK,0,[2]), it knows that all segments up to and including D(0,...)
have been correctly received. It also knows that segment D(2,...) has been received and can cancel the retransmission timer associated to this segment. However, this segment should not be removed from the sending buffer
before the reception of a cumulative acknowledgement (C(OK,2) in the figure above) that covers this segment.
Note: Maximum window size with go-back-n and selective repeat
A transport protocol that uses n bits to encode its sequence number can send up to 2n different segments. However,
to ensure a reliable delivery of the segments, go-back-n and selective repeat cannot use a sending window of 2n
segments. Consider first go-back-n and assume that a sender sends 2n segments. These segments are received
in-sequence by the destination, but all the returned acknowledgements are lost. The sender will retransmit all
segments and they will all be accepted by the receiver and delivered a second time to the user. It is easy to see that
this problem can be avoided if the maximum size of the sending window is 2n 1 segments. A similar problem
occurs with selective repeat. However, as the receiver accepts out-of-sequence segments, a sending window of
2n 1 segments is not sufficient to ensure a reliable delivery of all segments. It can be easily shown that to avoid
n
this problem, a selective repeat sender cannot use a window that is larger than 22 segments.
Go-back-n or selective repeat are used by transport protocols to provide a reliable data transfer above an unreliable
network layer service. Until now, we have assumed that the size of the sliding window was fixed for the entire
lifetime of the connection. In practice a transport layer entity is usually implemented in the operating system and
shares memory with other parts of the system. Furthermore, a transport layer entity must support several (possibly
hundreds or thousands) of transport connections at the same time. This implies that the memory which can be
used to support the sending or the receiving buffer of a transport connection may change during the lifetime of the
connection 4 . Thus, a transport protocol must allow the sender and the receiver to adjust their window sizes.
To deal with this issue, transport protocols allow the receiver to advertise the current size of its receiving window
in all the acknowledgements that it sends. The receiving window advertised by the receiver bounds the size of
the sending buffer used by the sender. In practice, the sender maintains two state variables : swin, the size of its
sending window (that may be adjusted by the system) and rwin, the size of the receiving window advertised by
the receiver. At any time, the number of unacknowledged segments cannot be larger than min(swin,rwin) 5 . The
utilisation of dynamic windows is illustrated in the figure below.
The receiver may adjust its advertised receive window based on its current memory consumption, but also to limit
the bandwidth used by the sender. In practice, the receive buffer can also shrink as the application may not able to
process the received data quickly enough. In this case, the receive buffer may be completely full and the advertised
receive window may shrink to 0. When the sender receives an acknowledgement with a receive window set to 0,
it is blocked until it receives an acknowledgement with a positive receive window. Unfortunately, as shown in the
figure below, the loss of this acknowledgement could cause a deadlock as the sender waits for an acknowledgement
while the receiver is waiting for a data segment.
To solve this problem, transport protocols rely on a special timer : the persistence timer. This timer is started
by the sender whenever it receives an acknowledgement advertising a receive window set to 0. When the timer
expires, the sender retransmits an old segment in order to force the receiver to send a new acknowledgement, and
hence send the current receive window size.
To conclude our description of the basic mechanisms found in transport protocols, we still need to discuss the
impact of segments arriving in the wrong order. If two consecutive segments are reordered, the receiver relies on
their sequence numbers to reorder them in its receive buffer. Unfortunately, as transport protocols reuse the same
sequence number for different segments, if a segment is delayed for a prolonged period of time, it might still be
accepted by the receiver. This is illustrated in the figure below where segment D(1,b) is delayed.
4

For a discussion on how the sending buffer can change, see e.g. [SMM1998]
Note that if the receive window shrinks, it might happen that the sender has already sent a segment that is not anymore inside its window.
This segment will be discarded by the receiver and the sender will retransmit it later.
5

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B
Rwin=1

Swin=3, rwin=1
0123

Data.req(a)
D(0,a)

0123
Data.ind(a)

0123

Swin=3, rwin=1
0123

Data.req(b)

C(OK,0, w=1)
2 new buffers become
available
0123

C(OK,0,w=3)
Data.req(c)
Swin=3, rwin=3
0123

Data.req(d)

D(1,b)
D(2,c)

Data.ind(b)

C(OK,2,w=3)

0123

D(3,d)

Figure 4.20: Dynamic receiving window

Figure 4.21: Risk of deadlock with dynamic windows

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Figure 4.22: Ambiguities caused by excessive delays

To deal with this problem, transport protocols combine two solutions. First, they use 32 bits or more to encode
the sequence number in the segment header. This increases the overhead, but also increases the delay between
the transmission of two different segments having the same sequence number. Second, transport protocols require
the network layer to enforce a Maximum Segment Lifetime (MSL). The network layer must ensure that no packet
remains in the network for more than MSL seconds. In the Internet the MSL is assumed 6 to be 2 minutes RFC
793. Note that this limits the maximum bandwidth of a transport protocol. If it uses n bits to encode its sequence
numbers, then it cannot send more than 2n segments every MSL seconds.
Transport protocols often need to send data in both directions. To reduce the overhead caused by the acknowledgements, most transport protocols use piggybacking. Thanks to this technique, a transport entity can place inside the
header of the data segments that it sends, the acknowledgements and the receive window that it advertises for the
opposite direction of the data flow. The main advantage of piggybacking is that it reduces the overhead as it is not
necessary to send a complete segment to carry an acknowledgement. This is illustrated in the figure below where
the acknowledgement number is underlined in the data segments. Piggybacking is only used when data flows
in both directions. A receiver will generate a pure acknowledgement when it does not send data in the opposite
direction as shown in the bottom of the figure.

Figure 4.23: Piggybacking

6 As we will see in the next chapter, the Internet does not strictly enforce this MSL. However, it is reasonable to expect that most packets
on the Internet will not remain in the network during more than 2 minutes. There are a few exceptions to this rule, such as RFC 1149 whose
implementation is described in http://www.blug.linux.no/rfc1149/ but there are few real links supporting RFC 1149 in the Internet.

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The last point to be discussed about the data transfer mechanisms used by transport protocols is the provision of a
byte stream service. As indicated in the first chapter, the byte stream service is widely used in the transport layer.
The transport protocols that provide a byte stream service associate a sequence number to all the bytes that are
sent and place the sequence number of the first byte of the segment in the segments header. This is illustrated in
the figure below. In this example, the sender chooses to put two bytes in each of the first three segments. This is
due to graphical reasons, a real transport protocol would use larger segments in practice. However, the division of
the byte stream into segments combined with the losses and retransmissions explain why the byte stream service
does not preserve the SDU boundaries.

Figure 4.24: Provision of the byte stream service

Connection establishment and release

The last points to be discussed about the transport protocol are the mechanisms used to establish and release a
transport connection.
We explained in the first chapters the service primitives used to establish a connection. The simplest approach to
establish a transport connection would be to define two special control segments : CR and CA. The CR segment is
sent by the transport entity that wishes to initiate a connection. If the remote entity wishes to accept the connection,
it replies by sending a CA segment. The transport connection is considered to be established once the CA segment
has been received and data segments can be sent in both directions.

Figure 4.25: Naive transport connection establishment

Unfortunately, this scheme is not sufficient for several reasons. First, a transport entity usually needs to maintain
several transport connections with remote entities. Sometimes, different users (i.e. processes) running above a
given transport entity request the establishment of several transport connections to different users attached to the
same remote transport entity. These different transport connections must be clearly separated to ensure that data
from one connection is not passed to the other connections. This can be achieved by using a connection identifier,
chosen by the transport entities and placed inside each segment to allow the entity which receives a segment to
easily associate it to one established connection.
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Second, as the network layer is imperfect, the CR or CA segment can be lost, delayed, or suffer from transmission
errors. To deal with these problems, the control segments must be protected by using a CRC or checksum to detect
transmission errors. Furthermore, since the CA segment acknowledges the reception of the CR segment, the CR
segment can be protected by using a retransmission timer.
Unfortunately, this scheme is not sufficient to ensure the reliability of the transport service. Consider for example
a short-lived transport connection where a single, but important transfer (e.g. money transfer from a bank account)
is sent. Such a short-lived connection starts with a CR segment acknowledged by a CA segment, then the data
segment is sent, acknowledged and the connection terminates. Unfortunately, as the network layer service is
unreliable, delays combined to retransmissions may lead to the situation depicted in the figure below, where a
delayed CR and data segments from a former connection are accepted by the receiving entity as valid segments,
and the corresponding data is delivered to the user. Duplicating SDUs is not acceptable, and the transport protocol
must solve this problem.

Figure 4.26: Duplicate transport connections ?

To avoid these duplicates, transport protocols require the network layer to bound the Maximum Segment Lifetime
(MSL). The organisation of the network must guarantee that no segment remains in the network for longer than
MSL seconds. On todays Internet, MSL is expected to be 2 minutes. To avoid duplicate transport connections,
transport protocol entities must be able to safely distinguish between a duplicate CR segment and a new CR
segment, without forcing each transport entity to remember all the transport connections that it has established in
the past.
A classical solution to avoid remembering the previous transport connections to detect duplicates is to use a clock
inside each transport entity. This transport clock has the following characteristics :
the transport clock is implemented as a k bits counter and its clock cycle is such that 2k cycle >>
M SL. Furthermore, the transport clock counter is incremented every clock cycle and after each connection
establishment. This clock is illustrated in the figure below.
the transport clock must continue to be incremented even if the transport entity stops or reboots

Figure 4.27: Transport clock

It should be noted that transport clocks do not need and usually are not synchronised to the real-time clock.
Precisely synchronising real-time clocks is an interesting problem, but it is outside the scope of this document.
See [Mills2006] for a detailed discussion on synchronising the real-time clock.

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The transport clock is combined with an exchange of three segments, called the three way handshake, to detect
duplicates. This three way handshake occurs as follows :
1. The initiating transport entity sends a CR segment. This segment requests the establishment
of a transport connection. It contains a connection identifier (not shown in the figure) and a
sequence number (seq=x in the figure below) whose value is extracted from the transport clock
. The transmission of the CR segment is protected by a retransmission timer.
2. The remote transport entity processes the CR segment and creates state for the connection attempt. At this stage, the remote entity does not yet know whether this is a new connection
attempt or a duplicate segment. It returns a CA segment that contains an acknowledgement
number to confirm the reception of the CR segment (ack=x in the figure below) and a sequence
number (seq=y in the figure below) whose value is extracted from its transport clock. At this
stage, the connection is not yet established.
3. The initiating entity receives the CA segment. The acknowledgement number of this segment
confirms that the remote entity has correctly received the CA segment. The transport connection
is considered to be established by the initiating entity and the numbering of the data segments
starts at sequence number x. Before sending data segments, the initiating entity must acknowledge the received CA segments by sending another CA segment.
4. The remote entity considers the transport connection to be established after having received the
segment that acknowledges its CA segment. The numbering of the data segments sent by the
remote entity starts at sequence number y.
The three way handshake is illustrated in the figure below.

Figure 4.28: Three-way handshake

Thanks to the three way handshake, transport entities avoid duplicate transport connections. This is illustrated by
the three scenarios below.
The first scenario is when the remote entity receives an old CR segment. It considers this CR segment as a
connection establishment attempt and replies by sending a CA segment. However, the initiating host cannot match
the received CA segment with a previous connection attempt. It sends a control segment (REJECT in the figure
below) to cancel the spurious connection attempt. The remote entity cancels the connection attempt upon reception
of this control segment.
A second scenario is when the initiating entity sends a CR segment that does not reach the remote entity and
receives a duplicate CA segment from a previous connection attempt. This duplicate CA segment cannot contain
a valid acknowledgement for the CR segment as the sequence number of the CR segment was extracted from the
transport clock of the initiating entity. The CA segment is thus rejected and the CR segment is retransmitted upon
expiration of the retransmission timer.
The last scenario is less likely, but it it important to consider it as well. The remote entity receives an old CR
segment. It notes the connection attempt and acknowledges it by sending a CA segment. The initiating entity

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Figure 4.29: Three-way handshake : recovery from a duplicate CR

Figure 4.30: Three-way handshake : recovery from a duplicate CA

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does not have a matching connection attempt and replies by sending a REJECT. Unfortunately, this segment never
reaches the remote entity. Instead, the remote entity receives a retransmission of an older CA segment that contains
the same sequence number as the first CR segment. This CA segment cannot be accepted by the remote entity as
a confirmation of the transport connection as its acknowledgement number cannot have the same value as the
sequence number of the first CA segment.

Figure 4.31: Three-way handshake : recovery from duplicates CR and CA

When we discussed the connection-oriented service, we mentioned that there are two types of connection releases
: abrupt release and graceful release.
The first solution to release a transport connection is to define a new control segment (e.g. the DR segment) and
consider the connection to be released once this segment has been sent or received. This is illustrated in the figure
below.

Figure 4.32: Abrupt connection release

As the entity that sends the DR segment cannot know whether the other entity has already sent all its data on the
connection, SDUs can be lost during such an abrupt connection release.
The second method to release a transport connection is to release independently the two directions of data transfer.
Once a user of the transport service has sent all its SDUs, it performs a DISCONNECT.req for its direction of data
transfer. The transport entity sends a control segment to request the release of the connection after the delivery of
all previous SDUs to the remote user. This is usually done by placing in the DR the next sequence number and by
delivering the DISCONNECT.ind only after all previous DATA.ind. The remote entity confirms the reception of
the DR segment and the release of the corresponding direction of data transfer by returning an acknowledgement.
This is illustrated in the figure below.

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Figure 4.33: Graceful connection release

4.2 The User Datagram Protocol

The User Datagram Protocol (UDP) is defined in RFC 768. It provides an unreliable connectionless transport
service on top of the unreliable network layer connectionless service. The main characteristics of the UDP service
are :
the UDP service cannot deliver SDUs that are larger than 65507 bytes 7
the UDP service does not guarantee the delivery of SDUs (losses and desquencing can occur)
the UDP service will not deliver a corrupted SDU to the destination
Compared to the connectionless network layer service, the main advantage of the UDP service is that it allows
several applications running on a host to exchange SDUs with several other applications running on remote hosts.
Let us consider two hosts, e.g. a client and a server. The network layer service allows the client to send information
to the server, but if an application running on the client wants to contact a particular application running on the
server, then an additional addressing mechanism is required other than the IP address that identifies a host, in order
to differentiate the application running on a host. This additional addressing is provided by port numbers. When
a server using UDP is enabled on a host, this server registers a port number. This port number will be used by the
clients to contact the server process via UDP.
The figure below shows a typical usage of the UDP port numbers. The client process uses port number 1234 while
the server process uses port number 5678. When the client sends a request, it is identified as originating from port
number 1234 on the client host and destined to port number 5678 on the server host. When the server process
replies to this request, the servers UDP implementation will send the reply as originating from port 5678 on the
server host and destined to port 1234 on the client host.
UDP uses a single segment format shown in the figure below.
The UDP header contains four fields :
a 16 bits source port
a 16 bits destination port
a 16 bits length field
a 16 bits checksum
As the port numbers are encoded as a 16 bits field, there can be up to only 65535 different server processes that are
bound to a different UDP port at the same time on a given server. In practice, this limit is never reached. However,
it is worth noticing that most implementations divide the range of allowed UDP port numbers into three different
ranges :
7 This limitation is due to the fact that the network layer (IPv4 and IPv6) cannot transport packets that are larger than 64 KBytes. As UDP
does not include any segmentation/reassembly mechanism, it cannot split a SDU before sending it.

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Figure 4.34: Usage of the UDP port numbers

Figure 4.35: UDP Header Format

the privileged port numbers (1 < port < 1024 )
the ephemeral port numbers ( officially 8 49152 <= port <= 65535 )
the registered port numbers (officially 1024 <= port < 49152)
In most Unix variants, only processes having system administrator privileges can be bound to port numbers smaller
than 1024. Well-known servers such as DNS, NTP or RPC use privileged port numbers. When a client needs to
use UDP, it usually does not require a specific port number. In this case, the UDP implementation will allocate
the first available port number in the ephemeral range. The range of registered port numbers should be used by
servers. In theory, developers of network servers should register their port number officially through IANA, but
few developers do this.
Note: Computation of the UDP checksum
The checksum of the UDP segment is computed over :
a pseudo header containing the source IP address, the destination IP address and a 32 bits bit field containing
the most significant byte set to 0, the second set to 17 and the length of the UDP segment in the lower two
bytes
the entire UDP segment, including its header
This pseudo-header allows the receiver to detect errors affecting the IP source or destination addresses placed in
the IP layer below. This is a violation of the layering principle that dates from the time when UDP and IP were
elements of a single protocol. It should be noted that if the checksum algorithm computes value 0x0000, then
value 0xffff is transmitted. A UDP segment whose checksum is set to 0x0000 is a segment for which the
transmitter did not compute a checksum upon transmission. Some NFS servers chose to disable UDP checksums
for performance reasons, but this caused problems that were difficult to diagnose. In practice, there are rarely
good reasons to disable UDP checksums. A detailed discussion of the implementation of the Internet checksum
may be found in RFC 1071
Several types of applications rely on UDP. As a rule of thumb, UDP is used for applications where delay must
be minimised or losses can be recovered by the application itself. A first class of the UDP-based applications are
applications where the client sends a short request and expects a quick and short answer. The DNS is an example of
8 A discussion of the ephemeral port ranges used
http://www.ncftp.com/ncftpd/doc/misc/ephemeral_ports.html

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different

TCP/UDP

implementations

may

found

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a UDP application that is often used in the wide area. However, in local area networks, many distributed systems
rely on Remote Procedure Call (RPC) that is often used on top of UDP. In Unix environments, the Network File
System (NFS) is built on top of RPC and runs frequently on top of UDP. A second class of UDP-based applications
are the interactive computer games that need to frequently exchange small messages, such as the players location
or their recent actions. Many of these games use UDP to minimise the delay and can recover from losses. A
third class of applications are multimedia applications such as interactive Voice over IP or interactive Video over
IP. These interactive applications expect a delay shorter than about 200 milliseconds between the sender and the
receiver and can recover from losses directly inside the application.

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Figure 4.38: Establishment of a TCP connection

and allows the system administrator to login from this host without giving a password 12 . Consider now an attacker
who knows this particular configuration and is able to send IP packets having the clients address as source. He
can send fake TCP segments to the server, but does not receive the servers answers. If he can predict the ISN that
is chosen by the server, he can send a fake SYN segment and shortly after the fake ACK segment confirming the
reception of the SYN+ACK segment sent by the server. Once the TCP connection is open, he can use it to send
any command to the server. To counter this attack, current TCP implementations add randomness to the ISN. One
of the solutions, proposed in RFC 1948 is to compute the ISN as
ISN = M + H(localhost, localport, remotehost, remoteport, secret).

where M is the current value of the TCP clock and His a cryptographic hash function. localhost and remotehost
(resp. localport and remoteport ) are the IP addresses (port numbers) of the local and remote host and secret is a
random number only known by the server. This method allows the server to use different ISNs for different clients
at the same time. Measurements performed with the first implementations of this technique showed that it was
difficult to implement it correctly, but todays TCP implementation now generate good ISNs.
A server could, of course, refuse to open a TCP connection upon reception of a SYN segment. This refusal may be
due to various reasons. There may be no server process that is listening on the destination port of the SYN segment.
The server could always refuse connection establishments from this particular client (e.g. due to security reasons)
or the server may not have enough resources to accept a new TCP connection at that time. In this case, the server
would reply with a TCP segment having its RST flag set and containing the sequence number of the received SYN
segment as its acknowledgment number. This is illustrated in the figure below. We discuss the other utilizations
of the TCP RST flag later (see TCP connection release).

Figure 4.39: TCP connection establishment rejected by peer

TCP connection establishment can be described as the four state Finite State Machine shown below. In this FSM,
!X (resp. ?Y) indicates the transmission of segment X (resp. reception of segment Y) during the corresponding
transition. Init is the initial state.
A client host starts in the Init state. It then sends a SYN segment and enters the SYN Sent state where it waits
for a SYN+ACK segment. Then, it replies with an ACK segment and enters the Established state where data can
12 On many departmental networks containing Unix workstations, it was common to allow users on one of the hosts to use rlogin RFC 1258
to run commands on any of the workstations of the network without giving any password. In this case, the remote workstation authenticated
the client host based on its IP address. This was a bad practice from a security viewpoint.

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Figure 4.40: TCP FSM for connection establishment

be exchanged. On the other hand, a server host starts in the Init state. When a server process starts to listen to
a destination port, the underlying TCP entity creates a TCP control block and a queue to process incoming SYN
segments. Upon reception of a SYN segment, the servers TCP entity replies with a SYN+ACK and enters the SYN
RCVD state. It remains in this state until it receives an ACK segment that acknowledges its SYN+ACK segment,
with this it then enters the Established state.
Apart from these two paths in the TCP connection establishment FSM, there is a third path that corresponds to the
case when both the client and the server send a SYN segment to open a TCP connection 13 . In this case, the client
and the server send a SYN segment and enter the SYN Sent state. Upon reception of the SYN segment sent by the
other host, they reply by sending a SYN+ACK segment and enter the SYN RCVD state. The SYN+ACK that arrives
from the other host allows it to transition to the Established state. The figure below illustrates such a simultaneous
establishment of a TCP connection.

Figure 4.41: Simultaneous establishment of a TCP connection

13 Of course, such a simultaneous TCP establishment can only occur if the source port chosen by the client is equal to the destination
port chosen by the server. This may happen when a host can serve both as a client as a server or in peer-to-peer applications when the
communicating hosts do not use ephemeral port numbers.

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Denial of Service attacks

When a TCP entity opens a TCP connection, it creates a Transmission Control Block (TCB). The TCB
contains the entire state that is maintained by the TCP entity for each TCP connection. During connection
establishment, the TCB contains the local IP address, the remote IP address, the local port number, the
remote port number, the current local sequence number, the last sequence number received from the remote
entity. Until the mid 1990s, TCP implementations had a limit on the number of TCP connections that could
be in the SYN RCVD state at a given time. Many implementations set this limit to about 100 TCBs. This limit
was considered sufficient even for heavily load http servers given the small delay between the reception of a
SYN segment and the reception of the ACK segment that terminates the establishment of the TCP connection.
When the limit of 100 TCBs in the SYN Rcvd state is reached, the TCP entity discards all received TCP SYN
segments that do not correspond to an existing TCB.
This limit of 100 TCBs in the SYN Rcvd state was chosen to protect the TCP entity from the risk of overloading its memory with too many TCBs in the SYN Rcvd state. However, it was also the reason for a new type of
Denial of Service (DoS) attack RFC 4987. A DoS attack is defined as an attack where an attacker can render
a resource unavailable in the network. For example, an attacker may cause a DoS attack on a 2 Mbps link
used by a company by sending more than 2 Mbps of packets through this link. In this case, the DoS attack
was more subtle. As a TCP entity discards all received SYN segments as soon as it has 100 TCBs in the SYN
Rcvd state, an attacker simply had to send a few 100 SYN segments every second to a server and never reply
to the received SYN+ACK segments. To avoid being caught, attackers were of course sending these SYN
segments with a different address than their own IP address a . On most TCP implementations, once a TCB
entered the SYN Rcvd state, it remained in this state for several seconds, waiting for a retransmission of the
initial SYN segment. This attack was later called a SYN flood attack and the servers of the ISP named panix
were among the first to be affected by this attack.
To avoid the SYN flood attacks, recent TCP implementations no longer enter the SYN Rcvd state upon reception of a SYN segment. Instead, they reply directly with a SYN+ACK segment and wait until the reception
of a valid ACK. This implementation trick is only possible if the TCP implementation is able to verify that
the received ACK segment acknowledges the SYN+ACK segment sent earlier without storing the initial sequence number of this SYN+ACK segment in a TCB. The solution to solve this problem, which is known as
SYN cookies is to compute the 32 bits of the ISN as follows :
the high order bits contain the low order bits of a counter that is incremented slowly
the low order bits contain a hash value computed over the local and remote IP addresses and ports and
a random secret only known to the server
The advantage of the SYN cookies is that by using them, the server does not need to create a TCB upon
reception of the SYN segment and can still check the returned ACK segment by recomputing the SYN cookie.
a

Sending a packet with a different source IP address than the address allocated to the host is called sending a spoofed packet.

Retransmitting the first SYN segment

As IP provides an unreliable connectionless service, the SYN and SYN+ACK segments sent to open a TCP
connection could be lost. Current TCP implementations start a retransmission timer when they send the first
SYN segment. This timer is often set to three seconds for the first retransmission and then doubles after each
retransmission RFC 2988. TCP implementations also enforce a maximum number of retransmissions for
the initial SYN segment.
As explained earlier, TCP segments may contain an optional header extension. In the SYN and SYN+ACK segments, these options are used to negotiate some parameters and the utilisation of extensions to the basic TCP
specification.
The first parameter which is negotiated during the establishment of a TCP connection is the Maximum Segment
Size (MSS). The MSS is the size of the largest segment that a TCP entity is able to process. According to RFC
879, all TCP implementations must be able to receive TCP segments containing 536 bytes of payload. However,
most TCP implementations are able to process larger segments. Such TCP implementations use the TCP MSS
Option in the SYN/SYN+ACK segment to indicate the largest segment they are able to process. The MSS value
indicates the maximum size of the payload of the TCP segments. The client (resp. server) stores in its TCB the
MSS value announced by the server (resp. the client).

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Another utilisation of TCP options during connection establishment is to enable TCP extensions. For example,
consider RFC 1323 (which is discussed in TCP reliable data transfer). RFC 1323 defines TCP extensions to
support timestamps and larger windows. If the client supports RFC 1323, it adds a RFC 1323 option to its SYN
segment. If the server understands this RFC 1323 option and wishes to use it, it replies with an RFC 1323
option in the SYN+ACK segment and the extension defined in RFC 1323 is used throughout the TCP connection.
Otherwise, if the servers SYN+ACK does not contain the RFC 1323 option, the client is not allowed to use this
extension and the corresponding TCP header options throughout the TCP connection. TCPs option mechanism
is flexible and it allows the extension of TCP while maintaining compatibility with older implementations.
The TCP options are encoded by using a Type Length Value format where :
the first byte indicates the type of the option.
the second byte indicates the total length of the option (including the first two bytes) in bytes
the last bytes are specific for each type of option
RFC 793 defines the Maximum Segment Size (MSS) TCP option that must be understood by all TCP implementations. This option (type 2) has a length of 4 bytes and contains a 16 bits word that indicates the MSS supported
by the sender of the SYN segment. The MSS option can only be used in TCP segments having the SYN flag set.
RFC 793 also defines two special options that must be supported by all TCP implementations. The first option
is End of option. It is encoded as a single byte having value 0x00 and can be used to ensure that the TCP header
extension ends on a 32 bits boundary. The No-Operation option, encoded as a single byte having value 0x01, can
be used when the TCP header extension contains several TCP options that should be aligned on 32 bit boundaries.
All other options 14 are encoded by using the TLV format.
Note: The robustness principle
The handling of the TCP options by TCP implementations is one of the many applications of the robustness
principle which is usually attributed to Jon Postel and is often quoted as Be liberal in what you accept, and
conservative in what you send RFC 1122
Concerning the TCP options, the robustness principle implies that a TCP implementation should be able to accept
TCP options that it does not understand, in particular in received SYN segments, and that it should be able to parse
any received segment without crashing, even if the segment contains an unknown TCP option. Furthermore, a
server should not send in the SYN+ACK segment or later, options that have not been proposed by the client in the
SYN segment.

4.3.2 TCP connection release

TCP, like most connection-oriented transport protocols, supports two types of connection release :
graceful connection release, where each TCP user can release its own direction of data transfer
abrupt connection release, where either one user closes both directions of data transfer or one TCP entity
is forced to close the connection (e.g. because the remote host does not reply anymore or due to lack of
resources)
The abrupt connection release mechanism is very simple and relies on a single segment having the RST bit set. A
TCP segment containing the RST bit can be sent for the following reasons :
a non-SYN segment was received for a non-existing TCP connection RFC 793
by extension, some implementations respond with an RST segment to a segment that is received on an
existing connection but with an invalid header RFC 3360. This causes the corresponding connection to be
closed and has caused security attacks RFC 4953
by extension, some implementations send an RST segment when they need to close an existing TCP connection (e.g. because there are not enough resources to support this connection or because the remote host
is considered to be unreachable). Measurements have shown that this usage of TCP RST was widespread
[AW05]
14

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When an RST segment is sent by a TCP entity, it should contain the current value of the sequence number for the
connection (or 0 if it does not belong to any existing connection) and the acknowledgement number should be set
to the next expected in-sequence sequence number on this connection.
Note: TCP RST wars
TCP implementers should ensure that two TCP entities never enter a TCP RST war where host A is sending a RST
segment in response to a previous RST segment that was sent by host B in response to a TCP RST segment sent by
host A ... To avoid such an infinite exchange of RST segments that do not carry data, a TCP entity is never allowed
to send a RST segment in response to another RST segment.
The normal way of terminating a TCP connection is by using the graceful TCP connection release. This mechanism uses the FIN flag of the TCP header and allows each host to release its own direction of data transfer. As for
the SYN flag, the utilisation of the FIN flag in the TCP header consumes one sequence number. The figure FSM
for TCP connection release shows the part of the TCP FSM used when a TCP connection is released.

Figure 4.42: FSM for TCP connection release

Starting from the Established state, there are two main paths through this FSM.
The first path is when the host receives a segment with sequence number x and the FIN flag set. The utilisation of
the FIN flag indicates that the byte before sequence number x was the last byte of the byte stream sent by the remote
host. Once all of the data has been delivered to the user, the TCP entity sends an ACK segment whose ack field is
set to (x + 1) mod 232 to acknowledge the FIN segment. The FIN segment is subject to the same retransmission
mechanisms as a normal TCP segment. In particular, its transmission is protected by the retransmission timer.
At this point, the TCP connection enters the CLOSE_WAIT state. In this state, the host can still send data to the
remote host. Once all its data have been sent, it sends a FIN segment and enter the LAST_ACK state. In this state,
the TCP entity waits for the acknowledgement of its FIN segment. It may still retransmit unacknowledged data
segments e.g. if the retransmission timer expires. Upon reception of the acknowledgement for the FIN segment,
the TCP connection is completely closed and its TCB can be discarded.
The second path is when the host decides first to send a FIN segment. In this case, it enters the FIN_WAIT1
state. It this state, it can retransmit unacknowledged segments but cannot send new data segments. It waits for an
acknowledgement of its FIN segment, but may receive a FIN segment sent by the remote host. In the first case,
the TCP connection enters the FIN_WAIT2 state. In this state, new data segments from the remote host are still
accepted until the reception of the FIN segment. The acknowledgement for this FIN segment is sent once all data
received before the FIN segment have been delivered to the user and the connection enters the TIME_WAIT state.
In the second case, a FIN segment is received and the connection enters the Closing state once all data received
from the remote host have been delivered to the user. In this state, no new data segments can be sent and the host
waits for an acknowledgement of its FIN segment before entering the TIME_WAIT state.
The TIME_WAIT state is different from the other states of the TCP FSM. A TCP entity enters this state after
having sent the last ACK segment on a TCP connection. This segment indicates to the remote host that all the
data that it has sent have been correctly received and that it can safely release the TCP connection and discard
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the corresponding TCB. After having sent the last ACK segment, a TCP connection enters the TIME_WAIT and
remains in this state for 2 M SL seconds. During this period, the TCB of the connection is maintained. This
ensures that the TCP entity that sent the last ACK maintains enough state to be able to retransmit this segment
if this ACK segment is lost and the remote host retransmits its last FIN segment or another one. The delay of
2 M SL seconds ensures that any duplicate segments on the connection would be handled correctly without
causing the transmission of an RST segment. Without the TIME_WAIT state and the 2 M SL seconds delay, the
connection release would not be graceful when the last ACK segment is lost.
Note: TIME_WAIT on busy TCP servers
The 2 M SL seconds delay in the TIME_WAIT state is an important operational problem on servers having
thousands of simultaneously opened TCP connections [FTY99]. Consider for example a busy web server that
processes 10.000 TCP connections every second. If each of these connections remain in the TIME_WAIT state
for 4 minutes, this implies that the server would have to maintain more than 2 million TCBs at any time. For this
reason, some TCP implementations prefer to perform an abrupt connection release by sending a RST segment to
close the connection [AW05] and immediately discard the corresponding TCB. However, if the RST segment is
lost, the remote host continues to maintain a TCB for a connection no longer exists. This optimisation reduces the
number of TCBs maintained by the host sending the RST segment but at the potential cost of increased processing
on the remote host when the RST segment is lost.

4.3.3 TCP reliable data transfer

The original TCP data transfer mechanisms were defined in RFC 793. Based on the experience of using TCP
on the growing global Internet, this part of the TCP specification has been updated and improved several times,
always while preserving the backward compatibility with older TCP implementations. In this section, we review
the main data transfer mechanisms used by TCP.
TCP is a window-based transport protocol that provides a bi-directional byte stream service. This has several
implications on the fields of the TCP header and the mechanisms used by TCP. The three fields of the TCP header
are :
sequence number. TCP uses a 32 bits sequence number. The sequence number placed in the header of a
TCP segment containing data is the sequence number of the first byte of the payload of the TCP segment.
acknowledgement number. TCP uses cumulative positive acknowledgements. Each TCP segment contains
the sequence number of the next byte that the sender of the acknowledgement expects to receive from the
remote host. In theory, the acknowledgement number is only valid if the ACK flag of the TCP header is set.
In practice almost all 15 TCP segments have their ACK flag set.
window. a TCP receiver uses this 16 bits field to indicate the current size of its receive window expressed
in bytes.
Note: The Transmission Control Block
For each established TCP connection, a TCP implementation must maintain a Transmission Control Block (TCB).
A TCB contains all the information required to send and receive segments on this connection RFC 793. This
includes 16 :
the local IP address
the remote IP address
the local TCP port number
the remote TCP port number
the current state of the TCP FSM
the maximum segment size (MSS)
15

In practice, only the SYN segment do not have their ACK flag set.
A complete TCP implementation contains additional information in its TCB, notably to support the urgent pointer. However, this part of
TCP is not discussed in this book. Refer to RFC 793 and RFC 2140 for more details about the TCB.
16

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snd.nxt : the sequence number of the next byte in the byte stream (the first byte of a new data segment that
you send uses this sequence number)
snd.una : the earliest sequence number that has been sent but has not yet been acknowledged
snd.wnd : the current size of the sending window (in bytes)
rcv.nxt : the sequence number of the next byte that is expected to be received from the remote host
rcv.wnd : the current size of the receive window advertised by the remote host
sending buffer : a buffer used to store all unacknowledged data
receiving buffer : a buffer to store all data received from the remote host that has not yet been delivered
to the user. Data may be stored in the receiving buffer because either it was not received in sequence or
because the user is too slow to process it
The original TCP specification can be categorised as a transport protocol that provides a byte stream service and
uses go-back-n.
To send new data on an established connection, a TCP entity performs the following operations on the corresponding TCB. It first checks that the sending buffer does not contain more data than the receive window advertised by
the remote host (rcv.wnd). If the window is not full, up to MSS bytes of data are placed in the payload of a TCP
segment. The sequence number of this segment is the sequence number of the first byte of the payload. It is set to
the first available sequence number : snd.nxt and snd.nxt is incremented by the length of the payload of the TCP
segment. The acknowledgement number of this segment is set to the current value of rcv.nxt and the window field
of the TCP segment is computed based on the current occupancy of the receiving buffer. The data is kept in the
sending buffer in case it needs to be retransmitted later.
When a TCP segment with the ACK flag set is received, the following operations are performed. rcv.wnd is set to
the value of the window field of the received segment. The acknowledgement number is compared to snd.una. The
newly acknowledged data is remove from the sending buffer and snd.una is updated. If the TCP segment contained
data, the sequence number is compared to rcv.nxt. If they are equal, the segment was received in sequence and
the data can be delivered to the user and rcv.nxt is updated. The contents of the receiving buffer is checked to see
whether other data already present in this buffer can be delivered in sequence to the user. If so, rcv.nxt is updated
again. Otherwise, the segments payload is placed in the receiving buffer.
Segment transmission strategies
In a transport protocol such as TCP that offers a bytestream, a practical issue that was left as an implementation
choice in RFC 793 is to decide when a new TCP segment containing data must be sent. There are two simple and
extreme implementation choices. The first implementation choice is to send a TCP segment as soon as the user
has requested the transmission of some data. This allows TCP to provide a low delay service. However, if the
user is sending data one byte at a time, TCP would place each user byte in a segment containing 20 bytes of TCP
header 17 . This is a huge overhead that is not acceptable in wide area networks. A second simple solution would
be to only transmit a new TCP segment once the user has produced MSS bytes of data. This solution reduces the
overhead, but at the cost of a potentially very high delay.
An elegant solution to this problem was proposed by John Nagle in RFC 896. John Nagle observed that the
overhead caused by the TCP header was a problem in wide area connections, but less in local area connections
where the available bandwidth is usually higher. He proposed the following rules to decide to send a new data
segment when a new data has been produced by the user or a new ack segment has been received
if rcv.wnd>= MSS and len(data) >= MSS :
send one MSS-sized segment
else
if there are unacknowledged data:
place data in buffer until acknowledgement has been received
else
send one TCP segment containing all buffered data
17 This TCP segment is then placed in an IP header. We describe IPv4 and IPv6 in the next chapter. The minimum size of the IPv4 (resp.
IPv6) header is 20 bytes (resp. 40 bytes).

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The first rule ensures that a TCP connection used for bulk data transfer always sends full TCP segments. The
second rule sends one partially filled TCP segment every round-trip-time.
This algorithm, called the Nagle algorithm, takes a few lines of code in all TCP implementations. These lines of
code have a huge impact on the packets that are exchanged in TCP/IP networks. Researchers have analysed the
distribution of the packet sizes by capturing and analysing all the packets passing through a given link. These
studies have shown several important results :
in TCP/IPv4 networks, a large fraction of the packets are TCP segments that contain only an acknowledgement. These packets usually account for 40-50% of the packets passing through the studied link
in TCP/IPv4 networks, most of the bytes are exchanged in long packets, usually packets containing up to
1460 bytes of payload which is the default MSS for hosts attached to an Ethernet network, the most popular
type of LAN
The figure below provides a distribution of the packet sizes measured on a link. It shows a three-modal distribution
of the packet size. 50% of the packets contain pure TCP acknowledgements and occupy 40 bytes. About 20%
of the packets contain about 500 bytes 18 of user data and 12% of the packets contain 1460 bytes of user data.
However, most of the user data is transported in large packets. This packet size distribution has implications on
the design of routers as we discuss in the next chapter.

Figure 4.43: Packet size distribution in the Internet

Recent measurements indicate that these packet size distributions are still valid in todays Internet, although the
packet distribution tends to become bimodal with small packets corresponding to TCP pure acks (40-64 bytes
depending on the utilisation of TCP options) and large 1460-bytes packets carrying most of the user data.
TCP windows
From a performance point of view, one of the main limitations of the original TCP specification is the 16 bits
window field in the TCP header. As this field indicates the current size of the receive window in bytes, it limits the
TCP receive window at 65535 bytes. This limitation was not a severe problem when TCP was designed since at
that time high-speed wide area networks offered a maximum bandwidth of 56 kbps. However, in todays network,
this limitation is not acceptable anymore. The table below provides the rough 19 maximum throughput that can be
achieved by a TCP connection with a 64 KBytes window in function of the connections round-trip-time
RTT
1 msec
10 msec
100 msec
500 msec

Maximum Throughput
524 Mbps
52.4 Mbps
5.24 Mbps
1.05 Mbps

To solve this problem, a backward compatible extension that allows TCP to use larger receive windows was
proposed in RFC 1323. Today, most TCP implementations support this option. The basic idea is that instead of
18 When these measurements were taken, some hosts had a default MSS of 552 bytes (e.g. BSD Unix derivatives) or 536 bytes (the default
MSS specified in RFC 793). Today, most TCP implementation derive the MSS from the maximum packet size of the LAN interface they use
(Ethernet in most cases).
19 A precise estimation of the maximum bandwidth that can be achieved by a TCP connection should take into account the overhead of the
TCP and IP headers as well.

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storing snd.wnd and rcv.wnd as 16 bits integers in the TCB, they should be stored as 32 bits integers. As the TCP
segment header only contains 16 bits to place the window field, it is impossible to copy the value of snd.wnd in
each sent TCP segment. Instead the header contains snd.wnd >> S where S is the scaling factor ( 0 S 14)
negotiated during connection establishment. The client adds its proposed scaling factor as a TCP option in the
SYN segment. If the server supports RFC 1323, it places in the SYN+ACK segment the scaling factor that it uses
when advertising its own receive window. The local and remote scaling factors are included in the TCB. If the
server does not support RFC 1323, it ignores the received option and no scaling is applied.
By using the window scaling extensions defined in RFC 1323, TCP implementations can use a receive buffer
of up to 1 GByte. With such a receive buffer, the maximum throughput that can be achieved by a single TCP
connection becomes :
RTT
1 msec
10 msec
100 msec
500 msec

Maximum Throughput
8590 Gbps
859 Gbps
86 Gbps
17 Gbps

These throughputs are acceptable in todays networks. However, there are already servers having 10 Gbps interfaces... Early TCP implementations had fixed receiving and sending buffers 20 . Todays high performance
implementations are able to automatically adjust the size of the sending and receiving buffer to better support high
bandwidth flows [SMM1998]
TCPs retransmission timeout
In a go-back-n transport protocol such as TCP, the retransmission timeout must be correctly set in order to achieve
good performance. If the retransmission timeout expires too early, then bandwidth is wasted by retransmitting
segments that have already been correctly received; whereas if the retransmission timeout expires too late, then
bandwidth is wasted because the sender is idle waiting for the expiration of its retransmission timeout.
A good setting of the retransmission timeout clearly depends on an accurate estimation of the round-trip-time of
each TCP connection. The round-trip-time differs between TCP connections, but may also change during the
lifetime of a single connection. For example, the figure below shows the evolution of the round-trip-time between
two hosts during a period of 45 seconds.

Figure 4.44: Evolution of the round-trip-time between two hosts

The easiest solution to measure the round-trip-time on a TCP connection is to measure the delay between the
transmission of a data segment and the reception of a corresponding acknowledgement 21 . As illustrated in the
20

See http://fasterdata.es.net/tuning.html for more information on how to tune a TCP implementation

In theory, a TCP implementation could store the timestamp of each data segment transmitted and compute a new estimate for the roundtrip-time upon reception of the corresponding acknowledgement. However, using such frequent measurements introduces a lot of noise in
practice and many implementations still measure the round-trip-time once per round-trip-time by recording the transmission time of one
segment at a time RFC 2988
21

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figure below, this measurement works well when there are no segment losses.

Figure 4.45: How to measure the round-trip-time ?

However, when a data segment is lost, as illustrated in the bottom part of the figure, the measurement is ambiguous
as the sender cannot determine whether the received acknowledgement was triggered by the first transmission of
segment 123 or its retransmission. Using incorrect round-trip-time estimations could lead to incorrect values of
the retransmission timeout. For this reason, Phil Karn and Craig Partridge proposed, in [KP91], to ignore the
round-trip-time measurements performed during retransmissions.
To avoid this ambiguity in the estimation of the round-trip-time when segments are retransmitted, recent TCP
implementations rely on the timestamp option defined in RFC 1323. This option allows a TCP sender to place
two 32 bit timestamps in each TCP segment that it sends. The first timestamp, TS Value (TSval) is chosen by the
sender of the segment. It could for example be the current value of its real-time clock 22 . The second value, TS
Echo Reply (TSecr), is the last TSval that was received from the remote host and stored in the TCB. The figure
below shows how the utilization of this timestamp option allows for the disambiguation of the round-trip-time
measurement when there are retransmissions.

Figure 4.46: Disambiguating round-trip-time measurements with the RFC 1323 timestamp option
Once the round-trip-time measurements have been collected for a given TCP connection, the TCP entity must
compute the retransmission timeout. As the round-trip-time measurements may change during the lifetime of a
connection, the retransmission timeout may also change. At the beginning of a connection 23 , the TCP entity that
sends a SYN segment does not know the round-trip-time to reach the remote host and the initial retransmission
timeout is usually set to 3 seconds RFC 2988.
The original TCP specification proposed in RFC 793 to include two additional variables in the TCB :
srtt : the smoothed round-trip-time computed as srrt = ( srtt) + ((1 ) rtt) where rtt is the
round-trip-time measured according to the above procedure and a smoothing factor (e.g. 0.8 or 0.9)
22

Some security experts have raised concerns that using the real-time clock to set the TSval in the timestamp option can leak information
such as the systems up-time. Solutions proposed to solve this problem may be found in [CNPI09]
23 As a TCP client often establishes several parallel or successive connections with the same server, RFC 2140 has proposed to reuse for
a new connection some information that was collected in the TCB of a previous connection, such as the measured rtt. However, this solution
has not been widely implemented.

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rto : the retransmission timeout is computed as rto = min(60, max(1, srtt)) where is used to take
into account the delay variance (value : 1.3 to 2.0). The 60 and 1 constants are used to ensure that the rto is
not larger than one minute nor smaller than 1 second.
However, in practice, this computation for the retransmission timeout did not work well. The main problem was
that the computed rto did not correctly take into account the variations in the measured round-trip-time. Van Jacobson proposed in his seminal paper [Jacobson1988] an improved algorithm to compute the rto and implemented
it in the BSD Unix distribution. This algorithm is now part of the TCP standard RFC 2988.
Jacobsons algorithm uses two state variables, srtt the smoothed rtt and rttvar the estimation of the variance of
the rtt and two parameters : and . When a TCP connection starts, the first rto is set to 3 seconds. When a first
estimation of the rtt is available, the srtt, rttvar and rto are computed as
srtt=rtt
rttvar=rtt/2
rto=srtt+4*rttvar

Then, when other rtt measurements are collected, srtt and rttvar are updated as follows :
rttvar = (1 ) rttvar + |srtt rtt|
srtt = (1 ) srtt + rtt
rto = srtt + 4 rttvar
The proposed values for the parameters are = 81 and = 41 . This allows a TCP implementation, implemented
in the kernel, to perform the rtt computation by using shift operations instead of the more costly floating point
operations [Jacobson1988]. The figure below illustrates the computation of the rto upon rtt changes.

Figure 4.47: Example computation of the rto

Advanced retransmission strategies

The default go-back-n retransmission strategy was defined in RFC 793. When the retransmission timer expires,
TCP retransmits the first unacknowledged segment (i.e. the one having sequence number snd.una). After each
expiration of the retransmission timeout, RFC 2988 recommends to double the value of the retransmission timeout. This is called an exponential backoff. This doubling of the retransmission timeout after a retransmission
was included in TCP to deal with issues such as network/receiver overload and incorrect initial estimations of the
retransmission timeout. If the same segment is retransmitted several times, the retransmission timeout is doubled
after every retransmission until it reaches a configured maximum. RFC 2988 suggests a maximum retransmission
timeout of at least 60 seconds. Once the retransmission timeout reaches this configured maximum, the remote
host is considered to be unreachable and the TCP connection is closed.
This retransmission strategy has been refined based on the experience of using TCP on the Internet. The first
refinement was a clarification of the strategy used to send acknowledgements. As TCP uses piggybacking, the
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easiest and less costly method to send acknowledgements is to place them in the data segments sent in the other
direction. However, few application layer protocols exchange data in both directions at the same time and thus this
method rarely works. For an application that is sending data segments in one direction only, the remote TCP entity
returns empty TCP segments whose only useful information is their acknowledgement number. This may cause
a large overhead in wide area network if a pure ACK segment is sent in response to each received data segment.
Most TCP implementations use a delayed acknowledgement strategy. This strategy ensures that piggybacking is
used whenever possible, otherwise pure ACK segments are sent for every second received data segments when
there are no losses. When there are losses or reordering, ACK segments are more important for the sender and
they are sent immediately RFC 813 RFC 1122. This strategy relies on a new timer with a short delay (e.g. 50
milliseconds) and one additional flag in the TCB. It can be implemented as follows
reception of a data segment:
if pkt.seq==rcv.nxt:
# segment received in sequence
if delayedack :
send pure ack segment
cancel acktimer
delayedack=False
else:
delayedack=True
start acktimer
else:
# out of sequence segment
send pure ack segment
if delayedack:
delayedack=False
cancel acktimer
transmission of a data segment:
if delayedack:
delayedack=False
cancel acktimer

# piggyback ack

acktimer expiration:
send pure ack segment
delayedack=False

Due to this delayed acknowledgement strategy, during a bulk transfer, a TCP implementation usually acknowledges every second TCP segment received.
The default go-back-n retransmission strategy used by TCP has the advantage of being simple to implement, in
particular on the receiver side, but when there are losses, a go-back-n strategy provides a lower performance than
a selective repeat strategy. The TCP developers have designed several extensions to TCP to allow it to use a
selective repeat strategy while maintaining backward compatibility with older TCP implementations. These TCP
extensions assume that the receiver is able to buffer the segments that it receives out-of-sequence.
The first extension that was proposed is the fast retransmit heuristic. This extension can be implemented on TCP
senders and thus does not require any change to the protocol. It only assumes that the TCP receiver is able to
buffer out-of-sequence segments.
From a performance point of view, one issue with TCPs retransmission timeout is that when there are isolated
segment losses, the TCP sender often remains idle waiting for the expiration of its retransmission timeouts. Such
isolated losses are frequent in the global Internet [Paxson99]. A heuristic to deal with isolated losses without
waiting for the expiration of the retransmission timeout has been included in many TCP implementations since
the early 1990s. To understand this heuristic, let us consider the figure below that shows the segments exchanged
over a TCP connection when an isolated segment is lost.
As shown above, when an isolated segment is lost the sender receives several duplicate acknowledgements since
the TCP receiver immediately sends a pure acknowledgement when it receives an out-of-sequence segment. A
duplicate acknowledgement is an acknowledgement that contains the same acknowledgement number as a previous
segment. A single duplicate acknowledgement does not necessarily imply that a segment was lost, as a simple
reordering of the segments may cause duplicate acknowledgements as well. Measurements [Paxson99] have
shown that segment reordering is frequent in the Internet. Based on these observations, the fast retransmit heuristic
has been included in most TCP implementations. It can be implemented as follows

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Figure 4.48: Detecting isolated segment losses

ack arrival:
if tcp.ack==snd.una:
# duplicate acknowledgement
dupacks++
if dupacks==3:
retransmit segment(snd.una)
else:
dupacks=0
# process acknowledgement

This heuristic requires an additional variable in the TCB (dupacks). Most implementations set the default number
of duplicate acknowledgements that trigger a retransmission to 3. It is now part of the standard TCP specification
RFC 2581. The fast retransmit heuristic improves the TCP performance provided that isolated segments are lost
and the current window is large enough to allow the sender to send three duplicate acknowledgements.
The figure below illustrates the operation of the fast retransmit heuristic.

Figure 4.49: TCP fast retransmit heuristics

When losses are not isolated or when the windows are small, the performance of the fast retransmit heuristic
decreases. In such environments, it is necessary to allow a TCP sender to use a selective repeat strategy instead
of the default go-back-n strategy. Implementing selective-repeat requires a change to the TCP protocol as the
receiver needs to be able to inform the sender of the out-of-order segments that it has already received. This can
be done by using the Selective Acknowledgements (SACK) option defined in RFC 2018. This TCP option is
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negotiated during the establishment of a TCP connection. If both TCP hosts support the option, SACK blocks can
be attached by the receiver to the segments that it sends. SACK blocks allow a TCP receiver to indicate the blocks
of data that it has received correctly but out of sequence. The figure below illustrates the utilisation of the SACK
blocks.

Figure 4.50: TCP selective acknowledgements

An SACK option contains one or more blocks. A block corresponds to all the sequence numbers between the left
edge and the right edge of the block. The two edges of the block are encoded as 32 bit numbers (the same size as
the TCP sequence number) in an SACK option. As the SACK option contains one byte to encode its type and one
byte for its length, a SACK option containing b blocks is encoded as a sequence of 2 + 8 b bytes. In practice,
the size of the SACK option can be problematic as the optional TCP header extension cannot be longer than 44
bytes. As the SACK option is usually combined with the RFC 1323 timestamp extension, this implies that a TCP
segment cannot usually contain more than three SACK blocks. This limitation implies that a TCP receiver cannot
always place in the SACK option that it sends, information about all the received blocks.
To deal with the limited size of the SACK option, a TCP receiver currently having more than 3 blocks inside its
receiving buffer must select the blocks to place in the SACK option. A good heuristic is to put in the SACK option
the blocks that have most recently changed, as the sender is likely to be already aware of the older blocks.
When a sender receives an SACK option indicating a new block and thus a new possible segment loss, it usually
does not retransmit the missing segment(s immediately. To deal with reordering, a TCP sender can use a heuristic
similar to fast retransmit by retransmitting a gap only once it has received three SACK options indicating this gap.
It should be noted that the SACK option does not supersede the acknowledgement number of the TCP header. A
TCP sender can only remove data from its sending buffer once they have been acknowledged by TCPs cumulative
acknowledgements. This design was chosen for two reasons. First, it allows the receiver to discard parts of its
receiving buffer when it is running out of memory without loosing data. Second, as the SACK option is not
transmitted reliably, the cumulative acknowledgements are still required to deal with losses of ACK segments
carrying only SACK information. Thus, the SACK option only serves as a hint to allow the sender to optimise its
retransmissions.
TCP congestion control
In the previous sections, we have explained the mechanisms that TCP uses to deal with transmission errors and
segment losses. In a heterogeneous network such as the Internet or enterprise IP networks, endsystems have very
different levels of performance. Some endsystems are high-end servers attached to 10 Gbps links while others are
mobile devices attached to a very low bandwidth wireless link. Despite these huge differences in performance, a
mobile device should be able to efficiently exchange segments with a high-end server.
To understand this problem better, let us consider the scenario shown in the figure below, where a server (A)

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attached to a 10 Mbps link is sending TCP segments to another computer (C) through a path that contains a 2
Mbps link.

Figure 4.51: TCP over heterogeneous links

In this network, the TCP segments sent by the server reach router R1. R1 forwards the segments towards router R2.
Router R2 can potentially receive segments at 10 Mbps, but it can only forward them at 2 Mbps to router R2 and
then to host C. Router R2 contains buffers that allow it to store the packets that cannot immediately be forwarded
to their destination. To understand the operation of TCP in this environment, let us consider a simplified model of
this network where host A is attached to a 10 Mbps link to a queue that represents the buffers of router R2. This
queue is emptied at a rate of 2 Mbps.

Figure 4.52: TCP self clocking

Let us consider that host A uses a window of three segments. It thus sends three back-to-back segments at 10
Mbps and then waits for an acknowledgement. Host A stops sending segments when its window is full. These
segments reach the buffers of router R2. The first segment stored in this buffer is sent by router R2 at a rate of 2
Mbps to the destination host. Upon reception of this segment, the destination sends an acknowledgement. This
acknowledgement allows host A to transmit a new segment. This segment is stored in the buffers of router R2
while it is transmitting the second segment that was sent by host A... Thus, after the transmission of the first
window of segments, TCP sends one data segment after the reception of each acknowledgement returned by the
destination 24 . In practice, the acknowledgements sent by the destination serve as a kind of clock that allows
the sending host to adapt its transmission rate to the rate at which segments are received by the destination. This
TCP self-clocking is the first mechanism that allows TCP to adapt to heterogeneous networks [Jacobson1988]. It
depends on the availability of buffers to store the segments that have been sent by the sender but have not yet been
transmitted to the destination.
However, TCP is not always used in this environment. In the global Internet, TCP is used in networks where a
large number of hosts send segments to a large number of receivers. For example, let us consider the network
24

If the destination is using delayed acknowledgements, the sending host sends two data segments after each acknowledgement.

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depicted below which is similar to the one discussed in [Jacobson1988] and RFC 896. In this network, we assume
that the buffers of the router are infinite to ensure that no packet is lost.

Figure 4.53: The congestion collapse problem

If many TCP senders are attached to the left part of the network above, they all send a window full of segments.
These segments are stored in the buffers of the router before being transmitted towards their destination. If there
are many senders on the left part of the network, the occupancy of the buffers quickly grows. A consequence of
the buffer occupancy is that the round-trip-time, measured by TCP, between the sender and the receiver increases.
Consider a network where 10,000 bits segments are sent. When the buffer is empty, such a segment requires 1
millisecond to be transmitted on the 10 Mbps link and 5 milliseconds to be the transmitted on the 2 Mbps link.
Thus, the round-trip-time measured by TCP is roughly 6 milliseconds if we ignore the propagation delay on the
links. Most routers manage their buffers as a FIFO queue 25 . If the buffer contains 100 segments, the round-triptime becomes 1 + 100 5 + 5 milliseconds as new segments are only transmitted on the 2 Mbps link once all
previous segments have been transmitted. Unfortunately, TCP uses a retransmission timer and performs go-back-n
to recover from transmission errors. If the buffer occupancy is high, TCP assumes that some segments have been
lost and retransmits a full window of segments. This increases the occupancy of the buffer and the delay through
the buffer... Furthermore, the buffer may store and send on the low bandwidth links several retransmissions of
the same segment. This problem is called congestion collapse. It occurred several times in the late 1980s. For
example, [Jacobson1988] notes that in 1986, the usable bandwidth of a 32 Kbits link dropped to 40 bits per second
due to congestion collapse 26 !
The congestion collapse is a problem that all heterogeneous networks face. Different mechanisms have been
proposed in the scientific literature to avoid or control network congestion. Some of them have been implemented
and deployed in real networks. To understand this problem in more detail, let us first consider a simple network
with two hosts attached to a high bandwidth link that are sending segments to destination C attached to a low
bandwidth link as depicted below.

Figure 4.54: The congestion problem

To avoid congestion collapse, the hosts must regulate their transmission rate 27 by using a congestion control
mechanism. Such a mechanism can be implemented in the transport layer or in the network layer. In TCP/IP
networks, it is implemented in the transport layer, but other technologies such as Asynchronous Transfer Mode
(ATM) or Frame Relay include congestion control mechanisms in lower layers.
Let us first consider the simple problem of a set of i hosts that share a single bottleneck link as shown in the
example above. In this network, the congestion control scheme must achieve the following objectives [CJ1989] :
25

We discuss in another chapter other possible organisations of the routers buffers.

At this time, TCP implementations were mainly following RFC 791. The round-trip-time estimations and the retransmission mechanisms
were very simple. TCP was improved after the publication of [Jacobson1988]
27 In this section, we focus on congestion control mechanisms that regulate the transmission rate of the hosts. Other types of mechanisms have been proposed in the literature. For example, credit-based flow-control has been proposed to avoid congestion in ATM networks
[KR1995]. With a credit-based mechanism, hosts can only send packets once they have received credits from the routers and the credits depend
on the occupancy of the routers buffers.
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1. The congestion control scheme must avoid congestion. In practice, this means that the bottleneck link cannot be overloaded. If ri (t) is the transmission rate allocated to host i at time t and
R the bandwidth
P of the bottleneck link, then the congestion control scheme should ensure that,
on average, t ri (t) R.
2. The congestion control scheme must be efficient. The bottleneck link is usually both a shared
and an expensive resource. Usually, bottleneck links are wide area links that are much more
expensive to upgrade than the local area networks. The congestion control scheme should ensure
thatPsuch links are efficiently used. Mathematically, the control scheme should ensure that
t ri (t) R.
3. The congestion control scheme should be fair. Most congestion schemes aim at achieving maxmin fairness. An allocation of transmission rates to sources is said to be max-min fair if :
no link in the network is congested
the rate allocated to source j cannot be increased without decreasing the rate allocated to a
source i whose allocation is smaller than the rate allocated to source j [Leboudec2008] .
Depending on the network, a max-min fair allocation may not always exist. In practice, max-min fairness is an
ideal objective that cannot necessarily be achieved. When there is a single bottleneck link as in the example above,
max-min fairness implies that each source should be allocated the same transmission rate.
To visualise the different rate allocations, it is useful to consider the graph shown below. In this graph, we plot
on the x-axis (resp. y-axis) the rate allocated to host B (resp. A). A point in the graph (rB , rA ) corresponds to a
possible allocation of the transmission rates. Since there is a 2 Mbps bottleneck link in this network, the graph
can be divided into two regions. The lower left part of the graph contains all allocations (rB , rA ) such that the
bottleneck link is not congested (rA + rB < 2). The right border of this region is the efficiency line, i.e. the set
of allocations that completely utilise the bottleneck link (rA + rB = 2). Finally, the fairness line is the set of fair
allocations.

Figure 4.55: Possible allocated transmission rates

As shown in the graph above, a rate allocation may be fair but not efficient (e.g. rA = 0.7, rB = 0.7), fair and
efficient ( e.g. rA = 1, rB = 1) or efficient but not fair (e.g. rA = 1.5, rB = 0.5). Ideally, the allocation should
be both fair and efficient. Unfortunately, maintaining such an allocation with fluctuations in the number of flows
that use the network is a challenging problem. Furthermore, there might be several thousands of TCP connections
or more that pass through the same link 28 .
To deal with these fluctuations in demand, which result in fluctuations in the available bandwidth, computer
networks use a congestion control scheme. This congestion control scheme should achieve the three objectives
listed above. Some congestion control schemes rely on a close cooperation between the endhosts and the routers,
while others are mainly implemented on the endhosts with limited support from the routers.
28 For example, the measurements performed in the Sprint network in 2004 reported more than 10k active TCP connections on a link, see
https://research.sprintlabs.com/packstat/packetoverview.php. More recent information about backbone links may be obtained from caida s
realtime measurements, see e.g. http://www.caida.org/data/realtime/passive/

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A congestion control scheme can be modelled as an algorithm that adapts the transmission rate (ri (t)) of host i
based on the feedback received from the network. Different types of feedbacks are possible. The simplest scheme
is a binary feedback [CJ1989] [Jacobson1988] where the hosts simply learn whether the network is congested or
not. Some congestion control schemes allow the network to regularly send an allocated transmission rate in Mbps
to each host [BF1995].
Let us focus on the binary feedback scheme which is the most widely used today. Intuitively, the congestion
control scheme should decrease the transmission rate of a host when congestion has been detected in the network,
in order to avoid congestion collapse. Furthermore, the hosts should increase their transmission rate when the
network is not congested. Otherwise, the hosts would not be able to efficiently utilise the network. The rate
allocated to each host fluctuates with time, depending on the feedback received from the network. The figure
below illustrates the evolution of the transmission rates allocated to two hosts in our simple network. Initially, two
hosts have a low allocation, but this is not efficient. The allocations increase until the network becomes congested.
At this point, the hosts decrease their transmission rate to avoid congestion collapse. If the congestion control
scheme works well, after some time the allocations should become both fair and efficient.

Figure 4.56: Evolution of the transmission rates

Various types of rate adaption algorithms are possible. Dah Ming Chiu and Raj Jain have analysed, in [CJ1989],
different types of algorithms that can be used by a source to adapt its transmission rate to the feedback received
from the network. Intuitively, such a rate adaptation algorithm increases the transmission rate when the network
is not congested (ensure that the network is efficiently used) and decrease the transmission rate when the network
is congested (to avoid congestion collapse).
The simplest form of feedback that the network can send to a source is a binary feedback (the network is congested
or not congested). In this case, a linear rate adaptation algorithm can be expressed as :
rate(t + 1) = C + C rate(t) when the network is congested
rate(t + 1) = N + N rate(t) when the network is not congested
With a linear adaption algorithm, C , N , C and N are constants. The analysis of [CJ1989] shows that to
be fair and efficient, such a binary rate adaption mechanism must rely on Additive Increase and Multiplicative
Decrease. When the network is not congested, the hosts should slowly increase their transmission rate (N =
1 and N > 0). When the network is congested, the hosts must multiplicatively decrease their transmission rate
(C < 1 and C = 0). Such an AIMD rate adaptation algorithm can be implemented by the pseudo-code below
# Additive Increase Multiplicative Decrease
if congestion :
rate=rate*betaC
# multiplicative decrease, betaC<1
else
rate=rate+alphaN
# additive increase, v0>0

Note: Which binary feedback ?

Two types of binary feedback are possible in computer networks. A first solution is to rely on implicit feedback.
This is the solution chosen for TCP. TCPs congestion control scheme [Jacobson1988] does not require any cooperation from the router. It only assumes that they use buffers and that they discard packets when there is congestion.
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TCP uses the segment losses as an indication of congestion. When there are no losses, the network is assumed to
be not congested. This implies that congestion is the main cause of packet losses. This is true in wired networks,
but unfortunately not always true in wireless networks. Another solution is to rely on explicit feedback. This
is the solution proposed in the DECBit congestion control scheme [RJ1995] and used in Frame Relay and ATM
networks. This explicit feedback can be implemented in two ways. A first solution would be to define a special
message that could be sent by routers to hosts when they are congested. Unfortunately, generating such messages
may increase the amount of congestion in the network. Such a congestion indication packet is thus discouraged
RFC 1812. A better approach is to allow the intermediate routers to indicate, in the packets that they forward,
their current congestion status. Binary feedback can be encoded by using one bit in the packet header. With such a
scheme, congested routers set a special bit in the packets that they forward while non-congested routers leave this
bit unmodified. The destination host returns the congestion status of the network in the acknowledgements that it
sends. Details about such a solution in IP networks may be found in RFC 3168. Unfortunately, as of this writing,
this solution is still not deployed despite its potential benefits.
The TCP congestion control scheme was initially proposed by Van Jacobson in [Jacobson1988]. The current
specification may be found in RFC 5681. TCP relies on Additive Increase and Multiplicative Decrease (AIMD).
To implement AIMD, a TCP host must be able to control its transmission rate. A first approach would be to use
timers and adjust their expiration times in function of the rate imposed by AIMD. Unfortunately, maintaining such
timers for a large number of TCP connections can be difficult. Instead, Van Jacobson noted that the rate of TCP
congestion can be artificially controlled by constraining its sending window. A TCP connection cannot send data
where window is the maximum between the hosts sending window and the window advertised
faster than window
rtt
by the receiver.
TCPs congestion control scheme is based on a congestion window. The current value of the congestion window
(cwnd) is stored in the TCB of each TCP connection and the window that can be used by the sender is constrained
by min(cwnd, rwin, swin) where swin is the current sending window and rwin the last received receive window. The Additive Increase part of the TCP congestion control increments the congestion window by MSS bytes
every round-trip-time. In the TCP literature, this phase is often called the congestion avoidance phase. The Multiplicative Decrease part of the TCP congestion control divides the current value of the congestion window once
congestion has been detected.
When a TCP connection begins, the sending host does not know whether the part of the network that it uses
to reach the destination is congested or not. To avoid causing too much congestion, it must start with a small
congestion window. [Jacobson1988] recommends an initial window of MSS bytes. As the additive increase part
of the TCP congestion control scheme increments the congestion window by MSS bytes every round-trip-time,
the TCP connection may have to wait many round-trip-times before being able to efficiently use the available
bandwidth. This is especially important in environments where the bandwidth rtt product is high. To avoid
waiting too many round-trip-times before reaching a congestion window that is large enough to efficiently utilise
the network, the TCP congestion control scheme includes the slow-start algorithm. The objective of the TCP
slow-start is to quickly reach an acceptable value for the cwnd. During slow-start, the congestion window is
doubled every round-trip-time. The slow-start algorithm uses an additional variable in the TCB : sshtresh (slowstart threshold). The ssthresh is an estimation of the last value of the cwnd that did not cause congestion. It is
initialised at the sending window and is updated after each congestion event.
In practice, a TCP implementation considers the network to be congested once its needs to retransmit a segment.
The TCP congestion control scheme distinguishes between two types of congestion :
mild congestion. TCP considers that the network is lightly congested if it receives three duplicate acknowledgements and performs a fast retransmit. If the fast retransmit is successful, this implies that only one
segment has been lost. In this case, TCP performs multiplicative decrease and the congestion window is
divided by 2. The slow-start threshold is set to the new value of the congestion window.
severe congestion. TCP considers that the network is severely congested when its retransmission timer
expires. In this case, TCP retransmits the first segment, sets the slow-start threshold to 50% of the congestion
window. The congestion window is reset to its initial value and TCP performs a slow-start.
The figure below illustrates the evolution of the congestion window when there is severe congestion. At the
beginning of the connection, the sender performs slow-start until the first segments are lost and the retransmission
timer expires. At this time, the ssthresh is set to half of the current congestion window and the congestion window
is reset at one segment. The lost segments are retransmitted as the sender again performs slow-start until the

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congestion window reaches the sshtresh. It then switches to congestion avoidance and the congestion window
increases linearly until segments are lost and the retransmission timer expires ...

Figure 4.57: Evaluation of the TCP congestion window with severe congestion
The figure below illustrates the evolution of the congestion window when the network is lightly congested and
all lost segments can be retransmitted using fast retransmit. The sender begins with a slow-start. A segment is
lost but successfully retransmitted by a fast retransmit. The congestion window is divided by 2 and the sender
immediately enters congestion avoidance as this was a mild congestion.

Figure 4.58: Evaluation of the TCP congestion window when the network is lightly congested
Most TCP implementations update the congestion window when they receive an acknowledgement. If we assume
that the receiver acknowledges each received segment and the sender only sends MSS sized segments, the TCP
congestion control scheme can be implemented using the simplified pseudo-code 29 below
# Initialisation
cwnd = MSS;
ssthresh= swin;
# Ack arrival
if tcp.ack > snd.una : # new ack, no congestion
if cwnd < ssthresh :
# slow-start : increase quickly cwnd
# double cwnd every rtt
cwnd = cwnd + MSS
else:
# congestion avoidance : increase slowly cwnd
# increase cwnd by one mss every rtt
cwnd = cwnd+ mss*(mss/cwnd)
else: # duplicate or old ack
if tcp.ack==snd.una:
# duplicate acknowledgement
dupacks++
if dupacks==3:
29 In this pseudo-code, we assume that TCP uses unlimited sequence and acknowledgement numbers. Furthermore, we do not detail how
the cwnd is adjusted after the retransmission of the lost segment by fast retransmit. Additional details may be found in RFC 5681.

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retransmitsegment(snd.una)
ssthresh=max(cwnd/2,2*MSS)
cwnd=ssthresh
else:
dupacks=0
# ack for old segment, ignored
Expiration of the retransmission timer:
send(snd.una)
# retransmit first lost segment
sshtresh=max(cwnd/2,2*MSS)
cwnd=MSS

Furthermore when a TCP connection has been idle for more than its current retransmission timer, it should reset its
congestion window to the congestion window size that it uses when the connection begins, as it no longer knows
the current congestion state of the network.
Note: Initial congestion window
The original TCP congestion control mechanism proposed in [Jacobson1988] recommended that each TCP connection should begin by setting cwnd = M SS. However, in todays higher bandwidth networks, using such a
small initial congestion window severely affects the performance for short TCP connections, such as those used
by web servers. Since the publication of RFC 3390, TCP hosts are allowed to use an initial congestion window
of about 4 KBytes, which corresponds to 3 segments in many environments.
Thanks to its congestion control scheme, TCP adapts its transmission rate to the losses that occur in the network. Intuitively, the TCP transmission rate decreases when the percentage of losses increases. Researchers have
proposed detailed models that allow the prediction of the throughput of a TCP connection when losses occur
[MSMO1997] . To have some intuition about the factors that affect the performance of TCP, let us consider a
very simple model. Its assumptions are not completely realistic, but it gives us good intuition without requiring
complex mathematics.
This model considers a hypothetical TCP connection that suffers from equally spaced segment losses. If p is the
segment loss ratio, then the TCP connection successfully transfers p1 1 segments and the next segment is lost.
If we ignore the slow-start at the beginning of the connection, TCP in this environment is always in congestion
avoidance as there are only isolated losses that can be recovered by using fast retransmit. The evolution of the
congestion window is thus as shown in the figure below. Note the that x-axis of this figure represents time measured
in units of one round-trip-time, which is supposed to be constant in the model, and the y-axis represents the size
of the congestion window measured in MSS-sized segments.

Figure 4.59: Evolution of the congestion window with regular losses

As the losses are equally spaced, the congestion window always starts at some value ( W
2 ), and is incremented by
one MSS every round-trip-time until it reaches twice this value (W). At this point, a segment is retransmitted and
the cycle starts again. If the congestion window is measured in MSS-sized segments, a cycle lasts W
2 round-triptimes. The bandwidth of the TCP connection is the number of bytes that have been transmitted during a given
period of time. During a cycle, the number of segments that are sent on the TCP connection is equal to the area of
the yellow trapeze in the figure. Its area is thus :
2
area = ( W
2 ) +

1
2

2
(W
2 ) =

3W 2
8

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However, given the regular losses that we consider, the q

number of segments that are sent between two losses (i.e.
8
1
= kp . The throughput (in bytes per second) of the
during a cycle) is by definition equal to p . Thus, W = 3p
TCP connection is equal to the number of segments transmitted divided by the duration of the cycle :
q
3W 2
3
SS
M SS
8

T hroughput = areaM
or,
after
having
eliminated
W,
T
hroughput
=
=
W
time
2 rtt p
rtt
2

More detailed models and the analysis of simulations have shown that a first order model of the TCP throughput
SS

when losses occur was T hroughput kM

rtt p . This is an important result which shows that :
TCP connections with a small round-trip-time can achieve a higher throughput than TCP connections having
a longer round-trip-time when losses occur. This implies that the TCP congestion control scheme is not
completely fair since it favors the connections that have the shorter round-trip-time
TCP connections that use a large MSS can achieve a higher throughput that the TCP connections that use
a shorter MSS. This creates another source of unfairness between TCP connections. However, it should be
noted that today most hosts are using almost the same MSS, roughly 1460 bytes.
In general, the maximum throughput that can be achieved by a TCP connection depends on its maximum window
size and the round-trip-time if there are no losses. If there are losses, it depends on the MSS, the round-trip-time
and the loss ratio.
kM
SS
T hroughput < min( window
rtt , rtt p )

Note: The TCP congestion control zoo

The first TCP congestion control scheme was proposed by Van Jacobson in [Jacobson1988]. In addition to writing
the scientific paper, Van Jacobson also implemented the slow-start and congestion avoidance schemes in release
4.3 Tahoe of the BSD Unix distributed by the University of Berkeley. Later, he improved the congestion control
by adding the fast retransmit and the fast recovery mechanisms in the Reno release of 4.3 BSD Unix. Since
then, many researchers have proposed, simulated and implemented modifications to the TCP congestion control
scheme. Some of these modifications are still used today, e.g. :
NewReno ( RFC 3782), which was proposed as an improvement of the fast recovery mechanism in the Reno
implementation
TCP Vegas, which uses changes in the round-trip-time to estimate congestion in order to avoid it [BOP1994]
CUBIC, which was designed for high bandwidth links and is the default congestion control scheme in the
Linux 2.6.19 kernel [HRX2008]
Compound TCP, which was designed for high bandwidth links is the default congestion control scheme in
several Microsoft operating systems [STBT2009]
A search of the scientific literature will probably reveal more than 100 different variants of the TCP congestion
control scheme. Most of them have only been evaluated by simulations. However, the TCP implementation in
the recent Linux kernels supports several congestion control schemes and new ones can be easily added. We can
expect that new TCP congestion control schemes will always continue to appear.

4.4 Summary
In this chapter, we have studied the transport layer. This layer provides two types of services to the application
layer. The unreliable connectionless service is the simplest service offered to applications. On the Internet, this is
the service offered by UDP. However, most applications prefer to use a reliable and connection-oriented transport
service. We have shown that providing this service was much more complex than providing an unreliable service
as the transport layer needs to recover from the errors that occur in the network layer. For this, transport layer
protocols rely on several mechanisms. First, they use a handshake mechanism, such as the three-way handshake
mechanism, to correctly establish a transport connection. Once the connection has been established, transport
entities exchange segments. Each segment contains a sequence number, and the transport layer uses acknowledgements to confirm the segments that have been correctly received. In addition, timers are used to recover from
segment losses and sliding windows are used to avoid overflowing the buffers of the transport entities. Finally,

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we explained how a transport connection can be safely released. We then discussed the mechanisms that are used
in TCP, the reliable transport protocol, used by most applications on the Internet. Most notably, we described the
congestion control mechanism that has been included in TCP since the late 1980s and explained how the reliability
mechanisms used by TCP have been tuned over the years.

4.5 Exercises
This section is divided in two parts. The first part contains exercises on the principles of transport protocols,
including TCP. The second part contains programming challenges packet analysis tools to observe the behaviour
of transport protocols.

30. Same question as above, but assume now that both the sender and the receiver implement selective repeat.
Note the the answer will be different from the above question.
31. Consider a transport that supports window of one hundred 1250 Bytes segments. What is the maximum
bandwidth that this protocol can achieve if the round-trip-time is set to one second ? What happens if,
instead of advertising a window of one hundred segments, the receiver decides to advertise a window of 10
segments ?
32. Explain under which circumstances a transport entity could advertise a window of 0 segments ?
33. To understand the operation of the TCP congestion control mechanism, it is useful to draw some time
sequence diagrams. Let us consider a simple scenario of a web client connected to the Internet that wishes
to retrieve a simple web page from a remote web server. For simplicity, we will assume that the delay
between the client and the server is 0.5 seconds and that the packet transmission times on the client and
the servers are negligible (e.g. they are both connected to a 1 Gbps network). We will also assume that the
client and the server use 1 KBytes segments.
1. Compute the time required to open a TCP connection, send an HTTP request and retrieve a 16 KBytes web
page. This page size is typical of the results returned by search engines like google_ or bing. An important
factor in this delay is the initial size of the TCP congestion window on the server. Assume first that the initial
window is set to 1 segment as defined in RFC 2001, 4 KBytes (i.e. 4 segments in this case) as proposed in
RFC 3390 or 16 KBytes as proposed in a recent paper.
2. Perform the same analysis with an initial window of one segment is the third segment sent by the server is
lost and the retransmission timeout is fixed and set to 2 seconds.
3. Same question as above but assume now that the 6th segment is lost.
4. Same question as above, but consider now the loss of the second and seventh acknowledgements sent by the
client.
5. Does the analysis above changes if the initial window is set to 16 KBytes instead of one segment ?
34. Several MBytes have been sent on a TCP connection and it becomes idle for several minutes. Discuss which
values should be used for the congestion window, slow start threshold and retransmission timers.
35. To operate reliably, a transport protocol that uses Go-back-n (resp. selective repeat) cannot use a window
that is larger than 2n 1 (resp. 2n1 ) segments. Does this limitation affects TCP ? Explain your answer.
36. Consider the simple network shown in the figure below. In this network, the router between the client and
the server can only store on each outgoing interface one packet in addition to the packet that it is currently
transmitting. It discards all the packets that arrive while its buffer is full. Assuming that you can neglect
the transmission time of acknowledgements and that the server uses an initial window of one segment and
has a retransmission timer set to 500 milliseconds, what is the time required to transmit 10 segments from
the client to the server. Does the performance increases if the server uses an initial window of 16 segments
instead ?

Figure 4.60: Simple network

37. The figure below describes the evolution of the congestion window of a TCP connection. Can you find the
reasons for the three events that are marked in the figure ?

38. The figure below describes the evolution of the congestion window of a TCP connection. Can you find the
reasons for the three events that are marked in the figure ?

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130.104.78.8.22 > 192.168.1.101.54150:
192.168.1.101.54150 > 130.104.78.8.22:
130.104.78.8.22 > 192.168.1.101.54150:

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S
S
.
P

1385328972:1385328972(0) win 65535 <ms

3627767479:3627767479(0) ack 138532897
ack 1 win 65535 <nop,nop,timestamp 274
1:21(20) ack 1 win 49248 <nop,nop,time

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21:05:56.279241
21:05:56.279534
21:05:56.303527
21:05:56.303623

IP
IP
IP
IP

192.168.1.101.54150 > 130.104.78.8.22:

192.168.1.101.54150 > 130.104.78.8.22:
130.104.78.8.22 > 192.168.1.101.54150:
192.168.1.101.54150 > 130.104.78.8.22:

.
P
.
P

ack 21 win 65535 <nop,nop,timestamp 27

1:22(21) ack 21 win 65535 <nop,nop,ti
ack 22 win 49248 <nop,nop,timestamp 12
22:814(792) ack 21 win 65535 <nop,nop,

You can easily recognise in the output above the SYN segment containing the MSS, window scale, timestamp and
sackOK options, the SYN+ACK segment whose wscale option indicates that the server uses window scaling for
this connection and then the first few segments exchanged on the connection.
wireshark is more recent than tcpdump. It evolved from the ethereal packet trace analysis software. It can be used
as a text tool like tcpdump. For a TCP connection, wireshark would provide almost the same output as tcpdump.
The main advantage of wireshark is that it also includes a graphical user interface that allows to perform various
types of analysis on a packet trace.

Figure 4.65: Wireshark : default window

The wireshark window is divided in three parts. The top part of the window is a summary of the first packets from
the trace. By clicking on one of the lines, you can show the detailed content of this packet in the middle part of
the window. The middle of the window allows you to inspect all the fields of the captured packet. The bottom part
of the window is the hexadecimal representation of the packet, with the field selected in the middle window being
highlighted.
wireshark is very good at displaying packets, but it also contains several analysis tools that can be very useful.
The first tool is Follow TCP stream. It is part of the Analyze menu and allows you to reassemble and display all
the payload exchanged during a TCP connection. This tool can be useful if you need to analyse for example the
commands exchanged during a SMTP session.
The second tool is the flow graph that is part of the Statistics menu. It provides a time sequence diagram of the
packets exchanged with some comments about the packet contents. See blow for an example.

Figure 4.66: Wireshark : flow graph

The third set of tools are the TCP stream graph tools that are part of the Statistics menu. These tools allow you
to plot various types of information extracted from the segments exchanged during a TCP connection. A first
interesting graph is the sequence number graph that shows the evolution of the sequence number field of the
captured segments with time. This graph can be used to detect graphically retransmissions.
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Figure 4.67: Wireshark : sequence number graph

A second interesting graph is the round-trip-time graph that shows the evolution of the round-trip-time in function
of time. This graph can be used to check whether the round-trip-time remains stable or not. Note that from a
packet trace, wireshark can plot two round-trip-time graphs, One for the flow from the client to the server and
the other one. wireshark will plot the round-trip-time graph that corresponds to the selected packet in the top
wireshark window.

Figure 4.68: Wireshark : round-trip-time graph

Emulating a network with netkit

Netkit is network emulator based on User Mode Linux. It allows to easily set up an emulated network of Linux
machines, that can act as end-host or routers.
Note: Where can I find Netkit?
Netkit
is
available
at
http://www.netkit.org.
Files
can
be
downloaded
from
http://wiki.netkit.org/index.php/Download_Official, and instructions for the installations are available here :
http://wiki.netkit.org/download/netkit/INSTALL .
There are two ways to use Netkit : The manual way, and by using pre-configured labs. In the first case, you
boot and control each machine individually, using the commands starting with a v (for virtual machine). In the
second case, you can start a whole network in a single operation. The commands for controlling the lab start with
a l. The man pages of those commands is available from http://wiki.netkit.org/man/man7/netkit.7.html
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You must be careful not to forgot to stop your virtual machines and labs, using either vhalt or lhalt.
A netkit lab is simply a directory containing at least a configuration file called lab.conf, and one directory for each
virtual machine. In the case the lab available on iCampus, the network is composed of two pcs, pc1 and pc2, both
of them being connected to a router r1. The lab.conf file contains the following lines :
pc1[0]=A
pc2[0]=B
r1[0]=A
r1[1]=B

This means that pc1 and r1 are connected to a virtual LAN named A via their interface eth0, while pc2 and r1
are connected to the virtual LAN B via respectively their interfaces eth0 and eth1.
The directory of each device is initially empty, but will be used by Netkit to store their filesystem.
The lab directory can contain optional files. In the lab provided to you, the pc1.startup file contains the shell
instructions to be executed on startup of the virtual machine. In this specific case, the script configures the interface
eth0 to allow traffic exchanges between pc1 and r1, as well as the routing table entry to join pc2.
Starting a lab consists thus simply in unpacking the provided archive, going into the lab directory and typing lstart
to start the network.
Note: File sharing between virtual machines and host
Virtual machines can access to the directory of the lab they belong to. This repertory is mounted in their filesystem
at the path /hostlab.
In the netkit lab (exercises/netkit/netkit_lab_2hosts_1rtr_ipv4.tar.tar.gz, you can find
a simple python client/server application that establishes TCP connections. Feel free to re-use this code to perform
your analysis.
Note: netkit tools
As the virtual machines run Linux, standard networking tools such as hping, tcpdump, netstat etc. are available as
usual.
Note that capturing network traces can be facilitated by using the uml_dump extension available at
http://kartoch.msi.unilim.fr/blog/?p=19 . This extension is already installed in the Netkit installation on the student lab. In order to capture the traffic exchanged on a given virtual LAN, you simply need to issue the command
vdump <LAN name> on the host. If you want to pipe the trace to wireshark, you can use vdump A | wireshark -i -k
1. A TCP/IP stack receives a SYN segment with the sequence number set to 1234. What will be the value of
the acknowledgement number in the returned SYN+ACK segment ?
2. Is it possible for a TCP/IP stack to return a SYN+ACK segment with the acknowledgement number set to 0
? If no, explain why. If yes, what was the content of the received SYN segment.
3. Open the tcpdump packet trace exercises/traces/trace.5connections_opening_closing.pcap
and identify the number of different TCP connections that are established and closed. For each connection,
explain by which mechanism they are closed. Analyse the initial sequence numbers that are used in the
SYN and SYN+ACK segments. How do these initial sequence numbers evolve ? Are they increased every
4 microseconds ?
4. The tcpdump packet trace exercises/traces/trace.5connections.pcap contains several
connection attempts. Can you explain what is happening with these connection attempts ?
5. The tcpdump packet trace exercises/traces/trace.ipv6.google.com.pcap was collected
from a popular website that is accessible by using IPv6. Explain the TCP options that are supported by the
client and the server.

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6. The tcpdump packet trace exercises/traces/trace.sirius.info.ucl.ac.be.pcap Was

collected on the departmental server. What are the TCP options supported by this server ?
7. A TCP implementation maintains a Transmission Control Block (TCB) for each TCP connection. This TCB
is a data structure that contains the complete state of each TCP connection. The TCB is described in RFC
793. It contains first the identification of the TCP connection :
localip : the IP address of the local host
remoteip : the IP address of the remote host
remoteport : the TCP port used for this connection on the remote host
localport : the TCP port used for this connection on the local host. Note that when a client
opens a TCP connection, the local port will often be chosen in the ephemeral port range ( 49152
<= localport <= 65535 ).
sndnxt : the sequence number of the next byte in the byte stream (the first byte of a new data
segment that you send will use this sequence number)
snduna : the earliest sequence number that has been sent but has not yet been acknowledged
rcvnxt : the sequence number of the next byte that your implementation expects to receive
from the remote host. For this exercise, you do not need to maintain a receive buffer and your
implementation can discard the out-of-sequence segments that it receives
sndwnd : the current sending window
rcvwnd : the current window advertised by the receiver
Using the exercises/traces/trace.sirius.info.ucl.ac.be.pcap packet trace,
what is the TCB of the connection on host 130.104.78.8 when it sends the third segment of the trace ?
8. The tcpdump packet trace exercises/traces/trace.maps.google.com was collected by containing a popular web site that provides mapping information. How many TCP connections were used to
retrieve the information from this server ?
9. Some network monitoring tools such as ntop collect all the TCP segments sent and received by a host or
a group of hosts and provide interesting statistics such as the number of TCP connections, the number of
bytes exchanged over each TCP connection, ... Assuming that you can capture all the TCP segments sent
by a host, propose the pseudo-code of an application that would list all the TCP connections established
and accepted by this host and the number of bytes exchanged over each connection. Do you need to count
the number of bytes contained inside each segment to report the number of bytes exchanged over each TCP
connection ?
10. There are two types of firewalls 30 : special devices that are placed at the border of campus or enterprise
networks and software that runs on endhosts. Software firewalls typically analyse all the packets that are
received by a host and decide based on the packets header and contents whether it can be processed by
the hosts network stack or must be discarded. System administrators often configure firewalls on laptop
or student machines to prevent students from installing servers on their machines. How would you design a simple firewall that blocks all incoming TCP connections but still allows the host to establish TCP
connections to any remote server ?
11. Using the netkit lab explained above, perform some tests by using hping3(8). hping3(8) is a command line tool that allows anyone (having system administrator privileges) to send special IP packets and
TCP segments. hping3(8) can be used to verify the configuration of firewalls 33 or diagnose problems.
We will use it to test the operation of the Linux TCP stack running inside netkit.
1. On the server host, launch tcpdump(1) with -vv as parameter to collect all packets received from the
client and display them. Using hping3(8) on the client host, send a valid SYN segment to one unused
port on the server host (e.g. 12345). What are the contents of the segment returned by the server ?
30 A firewall is a software or hardware device that analyses TCP/IP packets and decides, based on a set of rules, to accept or discard the
packets received or sent. The rules used by a firewall usually depend on the value of some fields of the packets (e.g. type of transport protocols,
ports, ...). We will discuss in more details the operation of firewalls in the network layer chapter.

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2. Perform the same experiment, but now send a SYN segment towards port 7. This port is the default port for
the discard service (see services(5)) launched by xinetd(8)). What segment does the server sends
in reply ? What happens upon reception of this segment ? Explain your answer.
12. The Linux TCP/IP stack can be easily configured by using sysctl(8) to change kernel configuration
variables. See http://fasterdata.es.net/TCP-tuning/ip-sysctl-2.6.txt for a recent list of the sysctl variables on
the Linux TCP/IP stack. Try to disable the selective acknowledgements and the RFC1323 timestamp and
large window options and open a TCP connection on port 7 on the server by using :manpage:telnet(1).
Check by using tcpdump(1) the effect of these kernel variables on the segments sent by the Linux stack
in netkit.
13. Network administrators sometimes need to verify which networking daemons are active on a server. When
logged on the server, several tools can be used to verify this. A first solution is to use the netstat(8)
command. This command allows you to extract various statistics from the networking stack on the Linux
kernel. For TCP, netstat can list all the active TCP connections with the state of their FSM. netstat supports
the following options that could be useful during this exercises :
-t requests information about the TCP connections
-n requests numeric output (by default, netstat sends DNS queries to resolve IP addresses in
hosts and uses /etc/services to convert port number in service names, -n is recommended on
netkit machines)
-e provides more information about the state of the TCP connections
-o provides information about the timers
-a provides information about all TCP connections, not only those in the Established state
On the netkit lab, launch a daemon and start a TCP connection by using telnet(1) and use
netstat(8) to verify the state of these connections.
A second solution to determine which network daemons are running on a server is to use a tool like
nmap(1). nmap(1) can be run remotely and thus can provide information about a host on which
the system administrator cannot login. Use tcpdump(1) to collect the segments sent by nmap(1)
running on the client and explain how nmap(1) operates.
14. Long lived TCP connections are susceptible to the so-called RST attacks. Try to find additional information
about this attack and explain how a TCP stack could mitigate such attacks.
15. For the exercises below, we have performed measurements in an emulated
shown below.

network similar to the one

Figure 4.69: Emulated network

The emulated network is composed of three UML machines 32 : a client, a server and a router. The
client and the server are connected via the router. The client sends data to the server. The link between
the router and the client is controlled by using the netem Linux kernel module. This module allows
us to insert additional delays, reduce the link bandwidth and insert random packet losses.
31

With an emulated network, it is more difficult to obtain quantitative results than with a real network since all the emulated machines
need to share the same CPU and memory. This creates interactions between the different emulated machines that do not happen in the real
world. However, since the objective of this exercise is only to allow the students to understand the behaviour of the TCP congestion control
mechanism, this is not a severe problem.
32
For more information about the TCP congestion control schemes implemented in the Linux kernel, see
http://linuxgazette.net/135/pfeiffer.html and http://www.cs.helsinki.fi/research/iwtcp/papers/linuxtcp.pdf or the source code of a recent
Linux. A description of some of the sysctl variables that allow to tune the TCP implementation in the Linux kernel may be found in
http://fasterdata.es.net/TCP-tuning/linux.html. For this exercise, we have configured the Linux kernel to use the NewReno scheme RFC 3782
that is very close to the official standard defined in RFC 5681

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We used netem to collect several traces :

exercises/traces/trace0.pcap
exercises/traces/trace1.pcap
exercises/traces/trace2.pcap
exercises/traces/trace3.pcap
Using wireshark or tcpdump, carry out the following analyses :
1. Identify the TCP options that have been used on the TCP connection
2. Try to find explanations for the evolution of the round-trip-time on each of these TCP
connections. For this, you can use the round-trip-time graph of wireshark, but be
careful with their estimation as some versions of wireshark are buggy
3. Verify whether the TCP implementation used implemented delayed acknowledgements
4. Inside each packet trace, find :
1. one segment that has been retransmitted by using fast retransmit. Explain this retransmission in details.
2. one segment that has been retransmitted thanks to the expiration of TCPs retransmission timeout. Explain why this segment could not have been retransmitted by using
fast retransmit.
5. wireshark contains several two useful graphs : the round-trip-time graph and the time
sequence graph. Explain how you would compute the same graph from such a trace .
6. When displaying TCP segments, recent versions of wireshark contain expert analysis
heuristics that indicate whether the segment has been retransmitted, whether it is a
duplicate ack or whether the retransmission timeout has expired. Explain how you
would implement the same heuristics as wireshark.
7. Can you find which file has been exchanged during the transfer ?
16. You have been hired as an networking expert by a company. In this company, users of a networked application complain that the network is very slow. The developers of the application argue that any delays are
caused by packet losses and a buggy network. The network administrator argues that the network works
perfectly and that the delays perceived by the users are caused by the applications or the servers where the
application is running. To resolve the case and determine whether the problem is due to the network or the
server on which the application is running. The network administrator has collected a representative packet
trace that you can download from exercises/traces/trace9.pcap. By looking at the trace, can
you resolve this case and indicate whether the network or the application is the culprit ?

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CHAPTER 5

The network layer

The transport layer enables the applications to efficiently and reliably exchange data. Transport layer entities
expect to be able to send segment to any destination without having to understand anything about the underlying
subnetwork technologies. Many subnetwork technologies exist. Most of them differ in subtle details (frame size,
addressing, ...). The network layer is the glue between these subnetworks and the transport layer. It hides to the
transport layer all the complexity of the underlying subnetworks and ensures that information can be exchanged
between hosts connected to different types of subnetworks.
In this chapter, we first explain the principles of the network layer. These principles include the datagram and
virtual circuit modes, the separation between the data plane and the control plane and the algorithms used by
routing protocols. Then, we explain, in more detail, the network layer in the Internet, starting with IPv4 and IPv6
and then moving to the routing protocols (RIP, OSPF and BGP).

5.1 Principles
The main objective of the network layer is to allow endsystems, connected to different networks, to exchange
information through intermediate systems called router. The unit of information in the network layer is called a
packet.

Figure 5.1: The network layer in the reference model

Before explaining the network layer in detail, it is useful to begin by analysing the service provided by the datalink
layer. There are many variants of the datalink layer. Some provide a connection-oriented service while others
provide a connectionless service. In this section, we focus on connectionless datalink layer services as they are
the most widely used. Using a connection-oriented datalink layer causes some problems that are beyond the scope
of this chapter. See RFC 3819 for a discussion on this topic.
There are three main types of datalink layers. The simplest datalink layer is when there are only two communicating systems that are directly connected through the physical layer. Such a datalink layer is used when there is a
point-to-point link between the two communicating systems. The two systems can be endsystems or routers. PPP
(Point-to-Point Protocol), defined in RFC 1661, is an example of such a point-to-point datalink layer. Datalink
layers exchange frames and a datalink frame sent by a datalink layer entity on the left is transmitted through the
physical layer, so that it can reach the datalink layer entity on the right. Point-to-point datalink layers can either

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Figure 5.2: The point-to-point datalink layer

provide an unreliable service (frames can be corrupted or lost) or a reliable service (in this case, the datalink layer
includes retransmission mechanisms similar to the ones used in the transport layer). The unreliable service is
frequently used above physical layers (e.g. optical fiber, twisted pairs) having a low bit error ratio while reliability
mechanisms are often used in wireless networks to recover locally from transmission errors.
The second type of datalink layer is the one used in Local Area Networks (LAN). Conceptually, a LAN is a set of
communicating devices such that any two devices can directly exchange frames through the datalink layer. Both
endsystems and routers can be connected to a LAN. Some LANs only connect a few devices, but there are LANs
that can connect hundreds or even thousands of devices.

Figure 5.3: A local area network

In the next chapter, we describe the organisation and the operation of Local Area Networks. An important difference between the point-to-point datalink layers and the datalink layers used in LANs is that in a LAN, each
communicating device is identified by a unique datalink layer address. This address is usually embedded in the
hardware of the device and different types of LANs use different types of datalink layer addresses. A communicating device attached to a LAN can send a datalink frame to any other communicating device that is attached to the
same LAN. Most LANs also support special broadcast and multicast datalink layer addresses. A frame sent to the
broadcast address of the LAN is delivered to all communicating devices that are attached to the LAN. The multicast addresses are used to identify groups of communicating devices. When a frame is sent towards a multicast
datalink layer address, it is delivered by the LAN to all communicating devices that belong to the corresponding
group.
The third type of datalink layers are used in Non-Broadcast Multi-Access (NBMA) networks. These networks are
used to interconnect devices like a LAN. All devices attached to an NBMA network are identified by a unique
datalink layer address. However, and this is the main difference between an NBMA network and a traditional
LAN, the NBMA service only supports unicast. The datalink layer service provided by an NBMA network
supports neither broadcast nor multicast.
Unfortunately no datalink layer is able to send frames of unlimited side. Each datalink layer is characterised by a
maximum frame size. There are more than a dozen different datalink layers and unfortunately most of them use a
different maximum frame size. The network layer must cope with the heterogeneity of the datalink layer.
The network layer itself relies on the following principles :
1. Each network layer entity is identified by a network layer address. This address is independent of the
datalink layer addresses that it may use.
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2. The service provided by the network layer does not depend on the service or the internal organisation of the
underlying datalink layers.
3. The network layer is conceptually divided into two planes : the data plane and the control plane. The data
plane contains the protocols and mechanisms that allow hosts and routers to exchange packets carrying user
data. The control plane contains the protocols and mechanisms that enable routers to efficiently learn how
to forward packets towards their final destination.
The independence of the network layer from the underlying datalink layer is a key principle of the network layer. It
ensures that the network layer can be used to allow hosts attached to different types of datalink layers to exchange
packets through intermediate routers. Furthermore, this allows the datalink layers and the network layer to evolve
independently from each other. This enables the network layer to be easily adapted to a new datalink layer every
time a new datalink layer is invented.
There are two types of service that can be provided by the network layer :
an unreliable connectionless service
a connection-oriented, reliable or unreliable, service
Connection-oriented services have been popular with technologies such as X.25 and ATM or frame-relay, but
nowadays most networks use an unreliable connectionless service. This is our main focus in this chapter.

5.1.1 Organisation of the network layer

There are two possible internal organisations of the network layer :
datagram
virtual circuits
The internal organisation of the network is orthogonal to the service that it provides, but most of the time a
datagram organisation is used to provide a connectionless service while a virtual circuit organisation is used in
networks that provide a connection-oriented service.
Datagram organisation
The first and most popular organisation of the network layer is the datagram organisation. This organisation is
inspired by the organisation of the postal service. Each host is identified by a network layer address. To send
information to a remote host, a host creates a packet that contains :
the network layer address of the destination host
its own network layer address
the information to be sent
The network layer limits the maximum packet size. Thus, the information must have been divided in packets by
the transport layer before being passed to the network layer.
To understand the datagram organisation, let us consider the figure below. A network layer address, represented
by a letter, has been assigned to each host and router. To send some information to host J, host A creates a packet
containing its own address, the destination address and the information to be exchanged.
With the datagram organisation, routers use hop-by-hop forwarding. This means that when a router receives
a packet that is not destined to itself, it looks up the destination address of the packet in its routing table. A
routing table is a data structure that maps each destination address (or set of destination addresses) to the outgoing
interface over which a packet destined to this address must be forwarded to reach its final destination.
The main constraint imposed on the routing tables is that they must allow any host in the network to reach any
other host. This implies that each router must know a route towards each destination, but also that the paths
composed from the information stored in the routing tables must not contain loops. Otherwise, some destinations
would be unreachable.

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When a router boots, it does not know any destination in the network and its routing table only contains itself. It
thus sends to all its neighbours a distance vector that contains only its address at a distance of 0. When a router
receives a distance vector on link l, it processes it as follows.
# V : received Vector
# l : link over which vector is received
def received(V,l):
# received vector from link l
for d in V[]

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if not (d in R[]) :
# new route
R[d].cost=V[d].cost+l.cost
R[d].link=l
R[d].time=now
else :
# existing route, is the new better ?
if ( ((V[d].cost+l.cost) < R[d].cost) or ( R[d].link == l) )
# Better route or change to current route
R[d].cost=V[d].cost+l.cost
R[d].link=l
R[d].time=now

The router iterates over all addresses included in the distance vector. If the distance vector contains an address
that the router does not know, it inserts the destination inside its routing table via link l and at a distance which is
the sum between the distance indicated in the distance vector and the cost associated to link l. If the destination
was already known by the router, it only updates the corresponding entry in its routing table if either :
the cost of the new route is smaller than the cost of the already known route ( (V[d].cost+l.cost) < R[d].cost)
the new route was learned over the same link as the current best route towards this destination ( R[d].link
== l)
The first condition ensures that the router discovers the shortest path towards each destination. The second condition is used to take into account the changes of routes that may occur after a link failure or a change of the metric
associated to a link.
To understand the operation of a distance vector protocol, let us consider the network of five routers shown below.

Figure 5.9: Operation of distance vector routing in a simple network

Assume that A is the first to send its distance vector [A=0].
B and D process the received distance vector and update their routing table with a route towards A.
D sends its distance vector [D=0,A=1] to A and E. E can now reach A and D.
C sends its distance vector [C=0] to B and E
E sends its distance vector [E=0,D=1,A=2,C=2] to D, B and C. B can now reach A, C, D and E
B sends its distance vector [B=0,A=1,C=1,D=2,E=1] to A, C and E. A, B, C and E can now reach all
destinations.
A sends its distance vector [A=0,B=1,C=2,D=1,E=2] to B and D.
At this point, all routers can reach all other routers in the network thanks to the routing tables shown in the figure
below.
To deal with link and router failures, routers use the timestamp stored in their routing table. As all routers send
their distance vector every N seconds, the timestamp of each route should be regularly refreshed. Thus no route
should have a timestamp older than N seconds, unless the route is not reachable anymore. In practice, to cope
with the possible loss of a distance vector due to transmission errors, routers check the timestamp of the routes
stored in their routing table every N seconds and remove the routes that are older than 3 N seconds. When a
router notices that a route towards a destination has expired, it must first associate an cost to this route and send
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Figure 5.10: Routing tables computed by distance vector in a simple network

its distance vector to its neighbours to inform them. The route can then be removed from the routing table after
some time (e.g. 3 N seconds), to ensure that the neighbouring routers have received the bad news, even if some
distance vectors do not reach them due to transmission errors.
Consider the example above and assume that the link between routers A and B fails. Before the failure, A used B
to reach destinations B, C and E while B only used the A-B link to reach A. The affected entries timeout on routers
A and B and they both send their distance vector.
A sends its distance vector [A = 0, D = , C = , D = 1, E = ]. D knows that it cannot reach B
anymore via A
D sends its distance vector [D = 0, B = , A = 1, C = 2, E = 1] to A and E. A recovers routes towards
C and E via D.
B sends its distance vector [B = 0, A = , C = 1, D = 2, E = 1] to E and C. D learns that there is no
route anymore to reach A via B.
E sends its distance vector [E = 0, A = 2, C = 1, D = 1, B = 1] to D, B and C. D learns a route towards
B. C and B learn a route towards A.
At this point, all routers have a routing table allowing them to reach all another routers, except router A, which
cannot yet reach router B. A recovers the route towards B once router D sends its updated distance vector [A =
1, B = 2, C = 2, D = 1, E = 1]. This last step is illustrated in figure Routing tables computed by distance vector
after a failure, which shows the routing tables on all routers.

Figure 5.11: Routing tables computed by distance vector after a failure

Consider now that the link between D and E fails. The network is now partitioned into two disjoint parts : (A , D)
and (B, E, C). The routes towards B, C and E expire first on router D. At this time, router D updates its routing
table.
If D sends [D = 0, A = 1, B = , C = , E = ], A learns that B, C and E are unreachable and updates its
routing table.
Unfortunately, if the distance vector sent to A is lost or if A sends its own distance vector ( [A = 0, D = 1, B =
3, C = 3, E = 2] ) at the same time as D sends its distance vector, D updates its routing table to use the
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shorter routes advertised by A towards B, C and E. After some time D sends a new distance vector : [D =
0, A = 1, E = 3, C = 4, B = 4]. A updates its routing table and after some time sends its own distance vector
[A = 0, D = 1, B = 5, C = 5, E = 4], etc. This problem is known as the count to infinity problem in networking
literature. Routers A and D exchange distance vectors with increasing costs until these costs reach . This
problem may occur in other scenarios than the one depicted in the above figure. In fact, distance vector routing
may suffer from count to infinity problems as soon as there is a cycle in the network. Cycles are necessary to
have enough redundancy to deal with link and router failures. To mitigate the impact of counting to infinity, some
distance vector protocols consider that 16 = . Unfortunately, this limits the metrics that network operators can
use and the diameter of the networks using distance vectors.
This count to infinity problem occurs because router A advertises to router D a route that it has learned via router
D. A possible solution to avoid this problem could be to change how a router creates its distance vector. Instead
of computing one distance vector and sending it to all its neighbors, a router could create a distance vector that is
specific to each neighbour and only contains the routes that have not been learned via this neighbour. This could
be implemented by the following pseudocode.
Every N seconds:
# one vector for each interface
for l in interfaces:
v=Vector()
for d in R[]:
if (R[d].link != i) :
v=v+Pair(d,R[d.cost])
send(v)
# end for d in R[]
#end for l in interfaces

To deal with the memory corruption problem, link state packets contain a checksum. This checksum is computed
by the router that generates the LSP. Each router must verify the checksum when it receives or floods an LSP.
Furthermore, each router must periodically verify the checksums of the LSPs stored in its LSDB.
Flooding is illustrated in the figure below. By exchanging HELLO messages, each router learns its direct neighbours. For example, router E learns that it is directly connected to routers D, B and C. Its first LSP has sequence
number 0 and contains the directed links E->D, E->B and E->C. Router E sends its LSP on all its links and routers
D, B and C insert the LSP in their LSDB and forward it over their other links.

Figure 5.14: Flooding : example

Flooding allows LSPs to be distributed to all routers inside the network without relying on routing tables. In the
example above, the LSP sent by router E is likely to be sent twice on some links in the network. For example,
routers B and C receive Es LSP at almost the same time and forward it over the B-C link. To avoid sending the
same LSP twice on each link, a possible solution is to slightly change the pseudo-code above so that a router waits
for some random time before forwarding a LSP on each link. The drawback of this solution is that the delay to
flood an LSP to all routers in the network increases. In practice, routers immediately flood the LSPs that contain
new information (e.g. addition or removal of a link) and delay the flooding of refresh LSPs (i.e. LSPs that contain
exactly the same information as the previous LSP originating from this router) [FFEB2005].
To ensure that all routers receive all LSPs, even when there are transmissions errors, link state routing protocols
use reliable flooding. With reliable flooding, routers use acknowledgements and if necessary retransmissions
to ensure that all link state packets are successfully transferred to all neighbouring routers. Thanks to reliable
flooding, all routers store in their LSDB the most recent LSP sent by each router in the network. By combining
the received LSPs with its own LSP, each router can compute the entire network topology.

Figure 5.15: Link state databases received by all routers

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Note: Static or dynamic link metrics ?

As link state packets are flooded regularly, routers are able to measure the quality (e.g. delay or load) of their
links and adjust the metric of each link according to its current quality. Such dynamic adjustments were included
in the ARPANET routing protocol [MRR1979] . However, experience showed that it was difficult to tune the
dynamic adjustments and ensure that no forwarding loops occur in the network [KZ1989]. Todays link state
routing protocols use metrics that are manually configured on the routers and are only changed by the network
operators or network management tools [FRT2002].
When a link fails, the two routers attached to the link detect the failure by the lack of HELLO messages received
in the last k N seconds. Once a router has detected a local link failure, it generates and floods a new LSP that
no longer contains the failed link and the new LSP replaces the previous LSP in the network. As the two routers
attached to a link do not detect this failure exactly at the same time, some links may be announced in only one
direction. This is illustrated in the figure below. Router E has detected the failures of link E-B and flooded a new
LSP, but router B has not yet detected the failure.

Figure 5.16: The two-way connectivity check

When a link is reported in the LSP of only one of the attached routers, routers consider the link as having failed and
they remove it from the directed graph that they compute from their LSDB. This is called the two-way connectivity
check. This check allows link failures to be flooded quickly as a single LSP is sufficient to announce such bad
news. However, when a link comes up, it can only be used once the two attached routers have sent their LSPs. The
two-way connectivity check also allows for dealing with router failures. When a router fails, all its links fail by
definition. Unfortunately, it does not, of course, send a new LSP to announce its failure. The two-way connectivity
check ensures that the failed router is removed from the graph.
When a router has failed, its LSP must be removed from the LSDB of all routers 1 . This can be done by using the
age field that is included in each LSP. The age field is used to bound the maximum lifetime of a link state packet
in the network. When a router generates a LSP, it sets its lifetime (usually measured in seconds) in the age field.
All routers regularly decrement the age of the LSPs in their LSDB and a LSP is discarded once its age reaches 0.
Thanks to the age field, the LSP from a failed router does not remain in the LSDBs forever.
To compute its routing table, each router computes the spanning tree rooted at itself by using Dijkstras shortest
path algorithm [Dijkstra1959]. The routing table can be derived automatically from the spanning as shown in the
figure below.

5.2 Internet Protocol

The Internet Protocol (IP) is the network layer protocol of the TCP/IP protocol suite. IP allows the applications
running above the transport layer (UDP/TCP) to use a wide range of heterogeneous datalink layers. IP was
1 It should be noted that link state routing assumes that all routers in the network have enough memory to store the entire LSDB. The
routers that do not have enough memory to store the entire LSDB cannot participate in link state routing. Some link state routing protocols
allow routers to report that they do not have enough memory and must be removed from the graph by the other routers in the network.

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Figure 5.17: Computation of the routing table

designed when most point-to-point links were telephone lines with modems. Since then, IP has been able to use
Local Area Networks (Ethernet, Token Ring, FDDI, ...), new wide area data link layer technologies (X.25, ATM,
Frame Relay, ...) and more recently wireless networks (802.11, 802.15, UMTS, GPRS, ...). The flexibility of IP
and its ability to use various types of underlying data link layer technologies is one of its key advantages.

Figure 5.18: IP and the reference model

The current version of IP is version 4 specified in RFC 791. We first describe this version and later explain IP
version 6, which is expected to replace IP version 4 in the not so distant future.

5.2.1 IP version 4
IP version 4 is the data plane protocol of the network layer in the TCP/IP protocol suite. The design of IP version
4 was based on the following assumptions :
IP should provide an unreliable connectionless service (TCP provides reliability when required by the application)
IP operates with the datagram transmission mode
IP addresses have a fixed size of 32 bits
IP must be usable above different types of datalink layers
IP hosts exchange variable length packets
The addresses are an important part of any network layer protocol. In the late 1970s, the developers of IPv4
designed IPv4 for a research network that would interconnect some research labs and universities. For this utilisation, 32 bits wide addresses were much larger than the expected number of hosts on the network. Furthermore, 32
bits was a nice address size for software-based routers. None of the developers of IPv4 were expecting that IPv4
would become as widely used as it is today.
IPv4 addresses are encoded as a 32 bits field. IPv4 addresses are often represented in dotted-decimal format as a
sequence of four integers separated by a dot. The first integer is the decimal representation of the most significant
byte of the 32 bits IPv4 address, ... For example,
1.2.3.4 corresponds to 00000001000000100000001100000100

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127.0.0.1 corresponds to 01111111000000000000000000000001

255.255.255.255 corresponds to 11111111111111111111111111111111
An IPv4 address is used to identify an interface on a router or a host. A router has thus as many IPv4 addresses as
the number of interfaces that it has in the datalink layer. Most hosts have a single datalink layer interface and thus
have a single IPv4 address. However, with the growth of wireless, more and more hosts have several datalink layer
interfaces (e.g. an Ethernet interface and a WiFi interface). These hosts are said to be multihomed. A multihomed
host with two interfaces has thus two IPv4 addresses.
An important point to be defined in a network layer protocol is the allocation of the network layer addresses. A
naive allocation scheme would be to provide an IPv4 address to each host when the host is attached to the Internet
on a first come first served basis. With this solution, a host in Belgium could have address 2.3.4.5 while another
host located in Africa would use address 2.3.4.6. Unfortunately, this would force all routers to maintain a specific
route towards each host. The figure below shows a simple enterprise network with two routers and three hosts and
the associated routing tables if such isolated addresses were used.

Figure 5.19: Scalability issues when using isolated IP addresses

To preserve the scalability of the routing system, it is important to minimize the number of routes that are stored on
each router. A router cannot store and maintain one route for each of the almost 1 billion hosts that are connected
to todays Internet. Routers should only maintain routes towards blocks of addresses and not towards individual
hosts. For this, hosts are grouped in subnets based on their location in the network. A typical subnet groups
all the hosts that are part of the same enterprise. An enterprise network is usually composed of several LANs
interconnected by routers. A small block of addresses from the Enterprises block is usually assigned to each
LAN. An IPv4 address is composed of two parts : a subnetwork identifier and a host identifier. The subnetwork
identifier is composed of the high order bits of the address and the host identifier is encoded in the low order bits
of the address. This is illustrated in the figure below.

Figure 5.20: The subnetwork and host identifiers inside an IPv4 address
When a router needs to forward a packet, it must know the subnet of the destination address to be able to consult
its forwarding table to forward the packet. RFC 791 proposed to use the high-order bits of the address to encode
the length of the subnet identifier. This led to the definition of three classes of unicast addresses 2
2 In addition to the A, B and C classes, RFC 791 also defined the D and E classes of IPv4 addresses. Class D (resp. E) addresses are those
whose high order bits are set to 1110 (resp. 1111). Class D addresses are used by IP multicast and will be explained later. Class E addresses
are currently unused, but there are some discussions on possible future usages [WMH2008] [FLM2008]

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Class
Class A
Class B
Class C

High-order bits
0
10
110

Length of subnet id
8 bits
16 bits
24 bits

Number of networks
128
16,384
2,097,152

Addresses per network

16,777,216 (224 )
65,536 (216 )
256 (28 )

However, these three classes of addresses were not flexible enough. A class A subnet was too large for most
organisations and a class C subnet was too small. Flexibility was added by the introduction of variable-length
subnets in RFC 1519. With variable-length subnets, the subnet identifier can be any size, from 1 to 31 bits.
Variable-length subnets allow the network operators to use a subnet that better matches the number of hosts that
are placed inside the subnet. A subnet identifier or IPv4 prefix is usually 3 represented as A.B.C.D/p where A.B.C.D
is the network address obtained by concatenating the subnet identifier with a host identifier containing only 0 and
p is the length of the subnet identifier in bits. The table below provides examples of IP subnets.
Subnet
10.0.0.0/8
192.168.0.0/16
198.18.0.0/15
192.0.2.0/24
10.0.0.0/30
10.0.0.0/31

Number of addresses
16,777,216
65,536
131,072
256
4
2

Smallest address
10.0.0.0
192.168.0.0
198.18.0.0
192.0.2.0
10.0.0.0
10.0.0.0

Highest address
10.255.255.255
192.168.255.255
198.19.255.255
192.0.2.255
10.0.0.3
10.0.0.1

The figure below provides a simple example of the utilisation of IPv4 subnets in an enterprise network. The length
of the subnet identifier assigned to a LAN usually depends on the expected number of hosts attached to the LAN.
For point-to-point links, many deployments have used /30 prefixes, but recent routers are now using /31 subnets
on point-to-point links RFC 3021 or do not even use IPv4 addresses on such links 4 .

Figure 5.21: IP subnets in a simple enterprise network

A second issue concerning the addresses of the network layer is the allocation scheme that is used to allocate
blocks of addresses to organisations. The first allocation scheme was based on the different classes of addresses.
The pool of IPv4 addresses was managed by a secretariat who allocated address blocks on a first-come first served
basis. Large organisations such as IBM, BBN, as well as Stanford or the MIT were able to obtain a class A address
block. Most organisations requested a class B address block containing 65536 addresses, which was suitable for
most enterprises and universities. The table below provides examples of some IPv4 address blocks in the class B
space.
3 Another way of representing IP subnets is to use netmasks. A netmask is a 32 bits field whose p high order bits are set to 1 and the low
order bits are set to 0. The number of high order bits set 1 indicates the length of the subnet identifier. Netmasks are usually represented in
the same dotted decimal format as IPv4 addresses. For example 10.0.0.0/8 would be represented as 10.0.0.0 255.0.0.0 while 192.168.1.0/24
would be represented as 192.168.1.0 255.255.255.0. In some cases, the netmask can be represented in hexadecimal.
4 A point-to-point link to which no IPv4 address has been allocated is called an unnumbered link. See RFC 1812 section 2.2.7 for a
discussion of such unnumbered links.

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Subnet
130.100.0.0/16
130.101.0.0/16
130.102.0.0/16
130.103.0.0/16
130.104.0.0/16
130.105.0.0/16

Organisation
Ericsson, Sweden
University of Akron, USA
The University of Queensland, Australia
Lotus Development, USA
Universite catholique de Louvain, Belgium
Open Software Foundation, USA

However, the Internet was a victim of its own success and in the late 1980s, many organisations were requesting
blocks of IPv4 addresses and started connecting to the Internet. Most of these organisations requested class B
address blocks, as class A address blocks were too large and in limited supply while class C address blocks were
considered to be too small. Unfortunately, there were only 16,384 different class B address blocks and this address
space was being consumed quickly. As a consequence, the routing tables maintained by the routers were growing
quickly and some routers had difficulties maintaining all these routes in their limited memory 5 .

Figure 5.22: Evolution of the size of the routing tables on the Internet (Jul 1988- Dec 1992 - source : RFC 1518)
Faced with these two problems, the Internet Engineering Task Force decided to develop the Classless Interdomain
Routing (CIDR) architecture RFC 1518. This architecture aims at allowing IP routing to scale better than the
class-based architecture. CIDR contains three important modifications compared to RFC 791.
1. IP address classes are deprecated. All IP equipment must use and support variable-length subnets.
2. IP address blocks are no longer allocated on a first-come-first-served basis. Instead, CIDR introduces a
hierarchical address allocation scheme.
3. IP routers must use longest-prefix match when they lookup a destination address in their forwarding table
The last two modifications were introduced to improve the scalability of the IP routing system. The main drawback of the first-come-first-served address block allocation scheme was that neighbouring address blocks were
allocated to very different organisations and conversely, very different address blocks were allocated to similar organisations. With CIDR, address blocks are allocated by Regional IP Registries (RIR) in an aggregatable manner.
A RIR is responsible for a large block of addresses and a region. For example, RIPE is the RIR that is responsible
for Europe. A RIR allocates smaller address blocks from its large block to Internet Service Providers RFC 2050.
Internet Service Providers then allocate smaller address blocks to their customers. When an organisation requests
an address block, it must prove that it already has or expects to have in the near future, a number of hosts or
customers that is equivalent to the size of the requested address block.
The main advantage of this hierarchical address block allocation scheme is that it allows the routers to maintain
fewer routes. For example, consider the address blocks that were allocated to some of the Belgian universities as
shown in the table below.
5

Example routers from this period include

http://www.ciscopress.com/articles/article.asp?p=25296

the

Cisco

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Address block
130.104.0.0/16
134.58.0.0/16
138.48.0.0/16
139.165.0.0/16
164.15.0.0/16

Organisation
Universite catholique de Louvain
Katholiek Universiteit Leuven
Facultes universitaires Notre-Dame de la Paix
Universite de Liege
Universite Libre de Bruxelles

These universities are all connected to the Internet exclusively via Belnet. As each university has been allocated
a different address block, the routers of Belnet must announce one route for each university and all routers on
the Internet must maintain a route towards each university. In contrast, consider all the high schools and the
government institutions that are connected to the Internet via Belnet. An address block was assigned to these
institutions after the introduction of CIDR in the 193.190.0.0/15 address block owned by Belnet. With CIDR,
Belnet can announce a single route towards 193.190.0.0/15 that covers all of these high schools.
However, there is one difficulty with the aggregatable variable length subnets used by CIDR. Consider for example
FEDICT, a government institution that uses the 193.191.244.0/23 address block. Assume that in addition to being
connected to the Internet via Belnet , FEDICT also wants to be connected to another Internet Service Provider.
The FEDICT network is then said to be multihomed. This is shown in the figure below.

Figure 5.23: Multihoming and CIDR

With such a multihomed network, routers R1 and R2 would have two routes towards IPv4 address 193.191.245.88
: one route via Belnet (193.190.0.0/15) and one direct route (193.191.244.0/23). Both routes match IPv4 address
193.192.145.88. Since RFC 1519 when a router knows several routes towards the same destination address, it
must forward packets along the route having the longest prefix length. In the case of 193.191.245.88, this is the
route 193.191.244.0/23 that is used to forward the packet. This forwarding rule is called the longest prefix match
or the more specific match. All IPv4 routers implement this forwarding rule.
To understand the longest prefix match forwarding, consider the figure below. With this rule, the route 0.0.0.0/0
plays a particular role. As this route has a prefix length of 0 bits, it matches all destination addresses. This route
is often called the default route.
a packet with destination 192.168.1.1 received by router R is destined to the router itself. It is delivered to
the appropriate transport protocol.
a packet with destination 11.2.3.4 matches two routes : 11.0.0.0/8 and 0.0.0.0/0. The packet is forwarded
on the West interface.
a packet with destination 130.4.3.4 matches one route : 0.0.0.0/0. The packet is forwarded on the North
interface.
a packet with destination 4.4.5.6 matches two routes : 4.0.0.0/8 and 0.0.0.0/0. The packet is forwarded on
the West interface.

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a packet with destination 4.10.11.254 matches three routes : 4.0.0.0/8, 4.10.11.0/24 and 0.0.0.0/0. The
packet is forwarded on the South interface.

Figure 5.24: Longest prefix match example

The longest prefix match can be implemented by using different data structures. One possibility is to use a trie.
The figure below shows a trie that encodes six routes having different outgoing interfaces.

Figure 5.25: A trie representing a routing table

Note: Special IPv4 addresses

Most unicast IPv4 addresses can appear as source and destination addresses in packets on the global Internet.
However, it is worth noting that some blocks of IPv4 addresses have a special usage, as described in RFC 5735.
These include :

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0.0.0.0/8, which is reserved for self-identification. A common address in this block is 0.0.0.0, which is
sometimes used when a host boots and does not yet know its IPv4 address.
127.0.0.0/8, which is reserved for loopback addresses. Each host implementing IPv4 must have a loopback
interface (that is not attached to a datalink layer). By convention, IPv4 address 127.0.0.1 is assigned to this
interface. This allows processes running on a host to use TCP/IP to contact other processes running on the
same host. This can be very useful for testing purposes.
10.0.0.0/8, 172.16.0.0/12 and 192.168.0.0/16 are reserved for private networks that are not directly attached
to the Internet. These addresses are often called private addresses or RFC 1918 addresses.
169.254.0.0/16 is used for link-local addresses RFC 3927. Some hosts use an address in this block when
they are connected to a network that does not allocate addresses as expected.

IPv4 packets
Now that we have clarified the allocation of IPv4 addresses and the utilisation of the longest prefix match to
forward IPv4 packets, we can have a more detailed look at IPv4 by starting with the format of the IPv4 packets.
The IPv4 packet format was defined in RFC 791. Apart from a few clarifications and some backward compatible
changes, the IPv4 packet format did not change significantly since the publication of RFC 791. All IPv4 packets
use the 20 bytes header shown below. Some IPv4 packets contain an optional header extension that is described
later.

Figure 5.26: The IP version 4 header

The main fields of the IPv4 header are :
a 4 bits version that indicates the version of IP used to build the header. Using a version field in the header
allows the network layer protocol to evolve.
a 4 bits IP Header Length (IHL) that indicates the length of the IP header in 32 bits words. This field allows
IPv4 to use options if required, but as it is encoded as a 4 bits field, the IPv4 header cannot be longer than
64 bytes.
an 8 bits DS field that is used for Quality of Service and whose usage is described later.
an 8 bits Protocol field that indicates the transport layer protocol that must process the packets payload at
the destination. Common values for this field 6 are 6 for TCP and 17 for UDP
a 16 bits length field that indicates the total length of the entire IPv4 packet (header and payload) in bytes.
This implies that an IPv4 packet cannot be longer than 65535 bytes.
a 32 bits source address field that contains the IPv4 address of the source host
a 32 bits destination address field that contains the IPv4 address of the destination host
a 16 bits checksum that protects only the IPv4 header against transmission errors
The other fields of the IPv4 header are used for specific purposes. The first is the 8 bits Time To Live (TTL) field.
This field is used by IPv4 to avoid the risk of having an IPv4 packet caught in an infinite loop due to a transient
6

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or permanent error in routing tables 7 . Consider for example the situation depicted in the figure below where
destination D uses address 11.0.0.56. If S sends a packet towards this destination, the packet is forwarded to router
B which forwards it to router C that forwards it back to router A, etc.

Figure 5.27: Forwarding loops in an IP network

Unfortunately, such loops can occur for two reasons in IP networks. First, if the network uses static routing, the
loop can be caused by a simple configuration error. Second, if the network uses dynamic routing, such a loop can
occur transiently, for example during the convergence of the routing protocol after a link or router failure. The
TTL field of the IPv4 header ensures that even if there are forwarding loops in the network, packets will not loop
forever. Hosts send their IPv4 packets with a positive TTL (usually 64 or more 8 ). When a router receives an IPv4
packet, it first decrements the TTL by one. If the TTL becomes 0, the packet is discarded and a message is sent
back to the packets source (see section ICMP). Otherwise, the router performs a lookup in its forwarding table to
forward the packet.
A second problem for IPv4 is the heterogeneity of the datalink layer. IPv4 is used above many very different
datalink layers. Each datalink layer has its own characteristics and as indicated earlier, each datalink layer is
characterised by a maximum frame size. From IPs point of view, a datalink layer interface is characterised by its
Maximum Transmission Unit (MTU). The MTU of an interface is the largest IPv4 packet (including header) that
it can send. The table below provides some common MTU sizes 9 .
Datalink layer
Ethernet
WiFi
ATM (AAL5)
802.15.4
Token Ring
FDDI

MTU
1500 bytes
2272 bytes
9180 bytes
102 or 81 bytes
4464 bytes
4352 bytes

Although IPv4 can send 64 KBytes long packets, few datalink layer technologies that are used today are able to
send a 64 KBytes IPv4 packet inside a frame. Furthermore, as illustrated in the figure below, another problem
is that a host may send a packet that would be too large for one of the datalink layers used by the intermediate
routers.

Figure 5.28: The need for fragmentation and reassembly

7 The initial IP specification in RFC 791 suggested that routers would decrement the TTL at least once every second. This would ensure
that a packet would never remain for more than TTL seconds in the network. However, in practice most router implementations simply chose
to decrement the TTL by one.
8 The initial TTL value used to send IP packets vary from one implementation to another. Most current IP implementations use an initial
TTL of 64 or more. See http://members.cox.net/~ndav1/self_published/TTL_values.html for additional information.
9 Supporting IP over the 802.15.4 datalink layer technology requires special mechanisms. See RFC 4944 for a discussion of the special
problems posed by 802.15.4

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To solve these problems, IPv4 includes a packet fragmentation and reassembly mechanism. Both hosts and intermediate routers may fragment an IPv4 packet if the packet is too long to be sent via the datalink layer. In IPv4,
fragmentation is completely performed in the IP layer and a large IPv4 is fragmented into two or more IPv4 packets (called fragments). The IPv4 fragments of a large packet are normal IPv4 packets that are forwarded towards
the destination of the large packet by intermediate routers.
The IPv4 fragmentation mechanism relies on four fields of the IPv4 header : Length, Identification, the flags and
the Fragment Offset. The IPv4 header contains two flags : More fragments and Dont Fragment (DF). When the
DF flag is set, this indicates that the packet cannot be fragmented.
The basic operation of the IPv4 fragmentation is as follows. A large packet is fragmented into two or more
fragments. The size of all fragments, except the last one, is equal to the Maximum Transmission Unit of the link
used to forward the packet. Each IPv4 packet contains a 16 bits Identification field. When a packet is fragmented,
the Identification of the large packet is copied in all fragments to allow the destination to reassemble the received
fragments together. In each fragment, the Fragment Offset indicates, in units of 8 bytes, the position of the payload
of the fragment in the payload of the original packet. The Length field in each fragment indicates the length of
the payload of the fragment as in a normal IPv4 packet. Finally, the More fragments flag is set only in the last
fragment of a large packet.
The following pseudo-code details the IPv4 fragmentation, assuming that the packet does not contain options.
#mtu : maximum size of the packet (including header) of outgoing link
if p.len < mtu :
send(p)
# packet is too large
maxpayload=8*int((mtu-20)/8) # must be n times 8 bytes
if p.flags==DF :
discard(p)
# packet must be fragmented
payload=p[IP].payload
pos=0
while len(payload) > 0 :
if len(payload) > maxpayload :
toSend=IP(dest=p.dest,src=p.src,
ttl=p.ttl, id=p.id,
frag=p.frag+(pos/8),
len=mtu, proto=p.proto)/payload[0:maxpayload]
pos=pos+maxpayload
payload=payload[maxpayload+1:]
else
toSend=IP(dest=p.dest,src=p.src,
ttl=p.ttl, id=p.id,
frag=p.frag+(pos/8),
flags=p.flags,
len=len(payload), proto=p.proto)/payload
forward(toSend)

The fragments of an IPv4 packet may arrive at the destination in any order, as each fragment is forwarded independently in the network and may follow different paths. Furthermore, some fragments may be lost and never
reach the destination.
The reassembly algorithm used by the destination host is roughly as follows. First, the destination can verify
whether a received IPv4 packet is a fragment or not by checking the value of the More fragments flag and the
Fragment Offset. If the Fragment Offset is set to 0 and the More fragments flag is reset, the received packet has not
been fragmented. Otherwise, the packet has been fragmented and must be reassembled. The reassembly algorithm
relies on the Identification field of the received fragments to associate a fragment with the corresponding packet
being reassembled. Furthermore, the Fragment Offset field indicates the position of the fragment payload in the
original unfragmented packet. Finally, the packet with the More fragments flag reset allows the destination to
determine the total length of the original unfragmented packet.
Note that the reassembly algorithm must deal with the unreliability of the IP network. This implies that a fragment
may be duplicated or a fragment may never reach the destination. The destination can easily detect fragment
duplication thanks to the Fragment Offset. To deal with fragment losses, the reassembly algorithm must bound the

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time during which the fragments of a packet are stored in its buffer while the packet is being reassembled. This
can be implemented by starting a timer when the first fragment of a packet is received. If the packet has not been
reassembled upon expiration of the timer, all fragments are discarded and the packet is considered to be lost.
The original IP specification, in RFC 791, defined several types of options that can be added to the IP header.
Each option is encoded using a type length value format. They are not widely used today and are thus only briefly
described. Additional details may be found in RFC 791.
The most interesting options in IPv4 are the three options that are related to routing. The Record route option was
defined to allow network managers to determine the path followed by a packet. When the Record route option was
present, routers on the packets path had to insert their IP address in the option. This option was implemented,
but as the optional part of the IPv4 header can only contain 44 bytes, it is impossible to discover an entire path
on the global Internet. traceroute(8), despite its limitations, is a better solution to record the path towards a
destination.
The other routing options are the Strict source route and the Loose source route option. The main idea behind
these options is that a host may want, for any reason, to specify the path to be followed by the packets that it sends.
The Strict source route option allows a host to indicate inside each packet the exact path to be followed. The Strict
source route option contains a list of IPv4 address and a pointer to indicate the next address in the list. When a
router receives a packet containing this option, it does not lookup the destination address in its routing table but
forwards the packet directly to the next router in the list and advances the pointer. This is illustrated in the figure
below where S forces its packets to follow the RA-RB-RD path.

Figure 5.29: Usage of the Strict source route option

The maximum length of the optional part of the IPv4 header is a severe limitation for the Strict source route
option as for the Record Route option. The Loose source route option does not suffer from this limitation. This
option allows the sending host to indicate inside its packet some of the routers that must be traversed to reach the
destination. This is shown in the figure below. S sends a packet containing a list of addresses and a pointer to the
next router in the list. Initially, this pointer points to RB. When RA receives the packet sent by S, it looks up in its
forwarding table the address pointed in the Loose source route option and not the destination address. The packet
is then forwarded to router RB that recognises its address in the option and advances the pointer. As there is no
address listed in the Loose source route option anymore, RB and other downstream routers forward the packet by
performing a lookup for the destination address.

Figure 5.30: Usage of the Loose source route option

These two options are usually ignored by routers because they cause security problems RFC 6274.

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5.2.2 ICMP version 4

It is sometimes necessary for intermediate routers or the destination host to inform the sender of the packet of a
problem that occurred while processing a packet. In the TCP/IP protocol suite, this reporting is done by the Internet
Control Message Protocol (ICMP). ICMP is defined in RFC 792. ICMP messages are carried as the payload of
IP packets (the protocol value reserved for ICMP is 1). An ICMP message is composed of an 8 byte header and
a variable length payload that usually contains the first bytes of the packet that triggered the transmission of the
ICMP message.

Figure 5.31: ICMP version 4 ( RFC 792)

In the ICMP header, the Type and Code fields indicate the type of problem that was detected by the sender of the
ICMP message. The Checksum protects the entire ICMP message against transmission errors and the Data field
contains additional information for some ICMP messages.
The main types of ICMP messages are :
Destination unreachable : a Destination unreachable ICMP message is sent when a packet
cannot be delivered to its destination due to routing problems. Different types of unreachability
are distinguished :
Network unreachable : this ICMP message is sent by a router that does not have a route for
the subnet containing the destination address of the packet
Host unreachable : this ICMP message is sent by a router that is attached to the subnet
that contains the destination address of the packet, but this destination address cannot be
reached at this time
Protocol unreachable : this ICMP message is sent by a destination host that has received a
packet, but does not support the transport protocol indicated in the packets Protocol field
Port unreachable : this ICMP message is sent by a destination host that has received a
packet destined to a port number, but no server process is bound to this port
Fragmentation needed : this ICMP message is sent by a router that receives a packet with the
Dont Fragment flag set that is larger than the MTU of the outgoing interface
Redirect : this ICMP message can be sent when there are two routers on the same LAN. Consider
a LAN with one host and two routers : R1 and R2. Assume that R1 is also connected to subnet
130.104.0.0/16 while R2 is connected to subnet 138.48.0.0/16. If a host on the LAN sends a
packet towards 130.104.1.1 to R2, R2 needs to forward the packet again on the LAN to reach
R1. This is not optimal as the packet is sent twice on the same LAN. In this case, R2 could send
an ICMP Redirect message to the host to inform it that it should have sent the packet directly to
R1. This allows the host to send the other packets to 130.104.1.1 directly via R1.
Parameter problem : this ICMP message is sent when a router or a host receives an IP packet
containing an error (e.g. an invalid option)

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Figure 5.32: ICMP redirect

Source quench : a router was supposed to send this message when it had to discard packets due
to congestion. However, sending ICMP messages in case of congestion was not the best way
to reduce congestion and since the inclusion of a congestion control scheme in TCP, this ICMP
message has been deprecated.
Time Exceeded : there are two types of Time Exceeded ICMP messages
TTL exceeded : a TTL exceeded message is sent by a router when it discards an IPv4 packet
because its TTL reached 0.
Reassembly time exceeded : this ICMP message is sent when a destination has been unable
to reassemble all the fragments of a packet before the expiration of its reassembly timer.
Echo request and Echo reply : these ICMP messages are used by the ping(8) network debugging software.
Note: Redirection attacks
ICMP redirect messages are useful when several routers are attached to the same LAN as hosts. However, they
should be used with care as they also create an important security risk. One of the most annoying attacks in an
IP network is called the man in the middle attack. Such an attack occurs if an attacker is able to receive, process,
possibly modify and forward all the packets exchanged between a source and a destination. As the attacker
receives all the packets it can easily collect passwords or credit card numbers or even inject fake information in an
established TCP connection. ICMP redirects unfortunately enable an attacker to easily perform such an attack. In
the figure above, consider host H that is attached to the same LAN as A and R1. If H sends to A an ICMP redirect
for prefix 138.48.0.0/16, A forwards to H all the packets that it wants to send to this prefix. H can then forward
them to R2. To avoid these attacks, hosts should ignore the ICMP redirect messages that they receive.
ping(8) is often used by network operators to verify that a given IP address is reachable. Each host is supposed
10
to reply with an ICMP Echo reply message when its receives an ICMP Echo request message. A sample usage
of ping(8) is shown below.
ping 130.104.1.1
PING 130.104.1.1 (130.104.1.1): 56 data bytes
64 bytes from 130.104.1.1: icmp_seq=0 ttl=243 time=19.961 ms
64 bytes from 130.104.1.1: icmp_seq=1 ttl=243 time=22.072 ms
64 bytes from 130.104.1.1: icmp_seq=2 ttl=243 time=23.064 ms
64 bytes from 130.104.1.1: icmp_seq=3 ttl=243 time=20.026 ms
64 bytes from 130.104.1.1: icmp_seq=4 ttl=243 time=25.099 ms
--- 130.104.1.1 ping statistics --5 packets transmitted, 5 packets received, 0% packet loss
round-trip min/avg/max/stddev = 19.961/22.044/25.099/1.938 ms
10 Until a few years ago, all hosts replied to Echo request ICMP messages. However, due to the security problems that have affected TCP/IP
implementations, many of these implementations can now be configured to disable answering Echo request ICMP messages.

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Another very useful debugging tool is traceroute(8). The traceroute man page describes this tool as print
the route packets take to network host. traceroute uses the TTL exceeded ICMP messages to discover the intermediate routers on the path towards a destination. The principle behind traceroute is very simple. When a router
receives an IP packet whose TTL is set to 1 it decrements the TTL and is forced to return to the sending host a
TTL exceeded ICMP message containing the header and the first bytes of the discarded IP packet. To discover all
routers on a network path, a simple solution is to first send a packet whose TTL is set to 1, then a packet whose
TTL is set to 2, etc. A sample traceroute output is shown below.
traceroute www.ietf.org
traceroute to www.ietf.org (64.170.98.32), 64 hops max, 40 byte packets
1 CsHalles3.sri.ucl.ac.be (192.168.251.230) 5.376 ms 1.217 ms 1.137 ms
2 CtHalles.sri.ucl.ac.be (192.168.251.229) 1.444 ms 1.669 ms 1.301 ms
3 CtPythagore.sri.ucl.ac.be (130.104.254.230) 1.950 ms 4.688 ms 1.319 ms
4 fe.m20.access.lln.belnet.net (193.191.11.9) 1.578 ms 1.272 ms 1.259 ms
5 10ge.cr2.brueve.belnet.net (193.191.16.22) 5.461 ms 4.241 ms 4.162 ms
6 212.3.237.13 (212.3.237.13) 5.347 ms 4.544 ms 4.285 ms
7 ae-11-11.car1.Brussels1.Level3.net (4.69.136.249) 5.195 ms 4.304 ms 4.329 ms
8 ae-6-6.ebr1.London1.Level3.net (4.69.136.246) 8.892 ms 8.980 ms 8.830 ms
9 ae-100-100.ebr2.London1.Level3.net (4.69.141.166) 8.925 ms 8.950 ms 9.006 ms
10 ae-41-41.ebr1.NewYork1.Level3.net (4.69.137.66) 79.590 ms
ae-43-43.ebr1.NewYork1.Level3.net (4.69.137.74) 78.140 ms
ae-42-42.ebr1.NewYork1.Level3.net (4.69.137.70) 77.663 ms
11 ae-2-2.ebr1.Newark1.Level3.net (4.69.132.98) 78.290 ms 83.765 ms 90.006 ms
12 ae-14-51.car4.Newark1.Level3.net (4.68.99.8) 78.309 ms 78.257 ms 79.709 ms
13 ex1-tg2-0.eqnwnj.sbcglobal.net (151.164.89.249) 78.460 ms 78.452 ms 78.292 ms
14 151.164.95.190 (151.164.95.190) 157.198 ms 160.767 ms 159.898 ms
15 ded-p10-0.pltn13.sbcglobal.net (151.164.191.243) 161.872 ms 156.996 ms 159.425 ms
16 AMS-1152322.cust-rtr.swbell.net (75.61.192.10) 158.735 ms 158.485 ms 158.588 ms
17 mail.ietf.org (64.170.98.32) 158.427 ms 158.502 ms 158.567 ms

The above traceroute(8) output shows a 17 hops path between a host at UCLouvain and one of the main
IETF servers. For each hop, traceroute provides the IPv4 address of the router that sent the ICMP message and the
measured round-trip-time between the source and this router. traceroute sends three probes with each TTL value.
In some cases, such as at the 10th hop above, the ICMP messages may be received from different addresses. This
is usually because different packets from the same source have followed different paths 11 in the network.
Another important utilisation of ICMP messages is to discover the maximum MTU that can be used to reach a
destination without fragmentation. As explained earlier, when an IPv4 router receives a packet that is larger than
the MTU of the outgoing link, it must fragment the packet. Unfortunately, fragmentation is a complex operation
and routers cannot perform it at line rate [KM1995]. Furthermore, when a TCP segment is transported in an IP
packet that is fragmented in the network, the loss of a single fragment forces TCP to retransmit the entire segment
(and thus all the fragments). If TCP was able to send only packets that do not require fragmentation in the
network, it could retransmit only the information that was lost in the network. In addition, IP reassembly causes
several challenges at high speed as discussed in RFC 4963. Using IP fragmentation to allow UDP applications to
exchange large messages raises several security issues [KPS2003].
ICMP, combined with the Dont fragment (DF) IPv4 flag, is used by TCP implementations to discover the largest
MTU size that is allowed to reach a destination host without causing network fragmentation. This is the Path MTU
discovery mechanism defined in RFC 1191. A TCP implementation that includes Path MTU discovery (most do)
requests the IPv4 layer to send all segments inside IPv4 packets having the DF flag set. This prohibits intermediate
routers from fragmenting these packets. If a router needs to forward an unfragmentable packet over a link with a
smaller MTU, it returns a Fragmentation needed ICMP message to the source, indicating the MTU of its outgoing
link. This ICMP message contains in the MTU of the routers outgoing link in its Data field. Upon reception of
this ICMP message, the source TCP implementation adjusts its Maximum Segment Size (MSS) so that the packets
containing the segments that it sends can be forwarded by this router without requiring fragmentation.
11

A detailed analysis of traceroute output is outside the scope of this document. Additional information may be found in [ACO+2006] and
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Interactions between IPv4 and the datalink layer

As mentioned in the first section of this chapter, there are three main types of datalink layers : point-to-point links,
LANs supporting broadcast and multicast and NBMA networks. There are two important issues to be addressed
when using IPv4 in these types of networks. The first issue is how an IPv4 device obtains its IPv4 address. The
second issue is how IPv4 packets are exchanged over the datalink layer service.
On a point-to-point link, the IPv4 addresses of the communicating devices can be configured manually or by
using a simple protocol. IPv4 addresses are often configured manually on point-to-point links between routers.
When point-to-point links are used to attach hosts to the network, automatic configuration is often preferred in
order to avoid problems with incorrect IPv4 addresses. For example, the PPP, specified in RFC 1661, includes
an IP network control protocol that can be used by the router in the figure below to send the IPv4 address that the
attached host must configure for its interface. The transmission of IPv4 packets on a point-to-point link will be
discussed in chapter chap:lan.

Figure 5.33: IPv4 on point-to-point links

Using IPv4 in a LAN introduces an additional problem. On a LAN, each device is identified by its unique datalink
layer address. The datalink layer service can be used by any host attached to the LAN to send a frame to any other
host attached to the same LAN. For this, the sending host must know the datalink layer address of the destination
host. For example, the figure below shows four hosts attached to the same LAN configured with IPv4 addresses
in the 10.0.1.0/24 subnet and datalink layer addresses represented as a single character 12 . In this network, if host
10.0.1.22/24 wants to send an IPv4 packet to the host having address 10.0.1.8, it must know that the datalink layer
address of this host is C.

Figure 5.34: A simple LAN

In a simple network such as the one shown above, it could be possible to manually configure the mapping between
the IPv4 addresses of the hosts and the corresponding datalink layer addresses. However, in a larger LAN this is
impossible. To ease the utilisation of LANs, IPv4 hosts must be able to automatically obtain the datalink layer
address corresponding to any IPv4 address on the same LAN. This is the objective of the Address Resolution
Protocol (ARP) defined in RFC 826. ARP is a datalink layer protocol that is used by IPv4. It relies on the ability
of the datalink layer service to easily deliver a broadcast frame to all devices attached to the same LAN.
12 In practice, most local area networks use addresses encoded as a 48 bits field [802]_ . Some recent local area network technologies use
64 bits addresses.

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The easiest way to understand the operation of ARP is to consider the simple network shown above and assume
that host 10.0.1.22/24 needs to send an IPv4 packet to host 10.0.1.8. As this IP address belongs to the same
subnet, the packet must be sent directly to its destination via the datalink layer service. To use this service, the
sending host must find the datalink layer address that is attached to host 10.0.1.8. Each IPv4 host maintains an
ARP cache containing the list of all mappings between IPv4 addresses and datalink layer addresses that it knows.
When an IPv4 hosts boots, its ARP cache is empty. 10.0.1.22 thus first consults its ARP cache. As the cache
does not contain the requested mapping, host 10.0.1.22 sends a broadcast ARP query frame on the LAN. The
frame contains the datalink layer address of the sending host (A) and the requested IPv4 address (10.0.1.8). This
broadcast frame is received by all devices on the LAN and only the host that owns the requested IPv4 address
replies by returning a unicast ARP reply frame with the requested mapping. Upon reception of this reply, the
sending host updates its ARP cache and sends the IPv4 packet by using the datalink layer service. To deal with
devices that move or whose addresses are reconfigured, most ARP implementations remove the cache entries that
have not been used for a few minutes. Some implementations re-validate ARP cache entries from time to time by
sending ARP queries 13 .
Note: Security issues with the Address Resolution Protocol
ARP is an old and widely used protocol that was unfortunately designed when security issues were not a concern.
ARP is almost insecure by design. Hosts using ARP can be subject to several types of attack. First, a malicious
host could create a denial of service attack on a LAN by sending random replies to the received ARP queries.
This would pollute the ARP cache of the other hosts on the same LAN. On a fixed network, such attacks can be
detected by the system administrator who can physically remove the malicious hosts from the LAN. On a wireless
network, removing a malicious host is much more difficult.
A second type of attack are the man-in-the-middle attacks. This name is used for network attacks where the
attacker is able to read and possibly modify all the messages sent by the attacked devices. Such an attack is
possible in a LAN. Assume, in the figure above, that host 10.0.1.9 is malicious and would like to receive and
modify all the packets sent by host 10.0.1.22 to host 10.0.1.8. This can be achieved easily if host 10.0.1.9 manages,
by sending fake ARP replies, to convince host 10.0.1.22 (resp. 10.0.1.8) that its own datalink layer address must
be used to reach 10.0.1.8 (resp. 10.0.1.22).
ARP is used by all devices that are connected to a LAN and implement IPv4. Both routers and endhosts implement
ARP. When a host needs to send an IPv4 packet to a destination outside of its local subnet, it must first send the
packet to one of the routers that reside on this subnet. Consider for example the network shown in the figure
below. Each host is configured with an IPv4 address in the 10.0.1.0/24 subnet and uses 10.0.1.1 as its default
router. To send a packet to address 1.2.3.4, host 10.0.1.8 will first need to know the datalink layer of the default
router. It will thus send an ARP request for 10.0.1.1. Upon reception of the ARP reply, host 10.0.1.8 updates its
ARP table and sends its packet in a frame to its default router. The router will then forward the packet towards its
final destination.

Figure 5.35: A simple LAN with a router

In the early days of the Internet, IP addresses were manually configured on both hosts and routers and almost never
changed. However, this manual configuration can be complex 14 and often causes errors that are sometimes difficult to debug. Recent TCP/IP implementations are able to detect some of these misconfigurations. For example,
if two hosts are attached to the same subnet with the same IPv4 address they will be unable to communicate. To
detect this problem hosts send an ARP request for their configured address each time their addressed is changed
RFC 5227. If they receive an answer to this ARP request, they trigger an alarm or inform the system administrator.
13
14

See chapter 28 of [Benvenuti2005] for a description of the implementation of ARP in the Linux kernel.
For example, consider all the options that can be specified for the ifconfig utility<http://en.wikipedia.org/wiki/Ifconfig> on Unix hosts.

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To ease the attachment of hosts to subnets, most networks now support the Dynamic Host Configuration Protocol
(DHCP) RFC 2131. DHCP allows a host to automatically retrieve its assigned IPv4 address. A DHCP server
is associated to each subnet 15 . Each DHCP server manages a pool of IPv4 addresses assigned to the subnet.
When a host is first attached to the subnet, it sends a DHCP request message in a UDP segment (the DHCP server
listens on port 67). As the host knows neither its IPv4 address nor the IPv4 address of the DHCP server, this
UDP segment is sent inside an IPv4 packet whose source and destination addresses are respectively 0.0.0.0 and
255.255.255.255. The DHCP request may contain various options such as the name of the host, its datalink layer
address, etc. The server captures the DHCP request and selects an unassigned address in its address pool. It then
sends the assigned IPv4 address in a DHCP reply message which contains the datalink layer address of the host
and additional information such as the subnet mask of the IPv4 address, the address of the default router or the
address of the DNS resolver. This DHCP reply message is sent in an IPv4 packet whose source and destination
addresses are respectively the IPv4 address of the DHCP server and the 255.255.255.255 broadcast address. The
DHCP reply also specifies the lifetime of the address allocation. This forces the host to renew its address allocation
once it expires. Thanks to the limited lease time, IP addresses are automatically returned to the pool of addresses
hosts are powered off. This reduces the waste of IPv4 addresses.
In an NBMA network, the interactions between IPv4 and the datalink layer are more complex as the ARP protocol
cannot be used as in a LAN. Such NBMA networks use special servers that store the mappings between IP addresses and the corresponding datalink layer address. Asynchronous Transfer Mode (ATM) networks for example
can use either the ATMARP protocol defined in RFC 2225 or the NextHop Resolution Protocol (NHRP) defined
in RFC 2332. ATM networks are less frequently used today and we will not describe the detailed operation of
these servers.
Operation of IPv4 devices
At this point of the description of IPv4, it is useful to have a detailed look at how an IPv4 implementation sends,
receives and forwards IPv4 packets. The simplest case is when a host needs to send a segment in an IPv4 packet.
The host performs two operations. First, it must decide on which interface the packet will be sent. Second it must
create the corresponding IP packet(s).
To simplify the discussion in this section, we ignore the utilisation of IPv4 options. This is not a severe limitation
as today IPv4 packets rarely contain options. Details about the processing of the IPv4 options may be found in the
relevant RFCs, such as RFC 791. Furthermore, we also assume that only point-to-point links are used. We defer
the explanation of the operation of IPv4 over Local Area Networks until the next chapter.
An IPv4 host having n datalink layer interfaces manages n + 1 IPv4 addresses :
the 127.0.0.1/32 IPv4 address assigned by convention to its loopback address
one A.B.C.D/p IPv4 address assigned to each of its n datalink layer interfaces
Such a host maintains a routing table containing one entry for its loopback address and one entry for each subnet
identifier assigned to its interfaces. Furthermore, the host usually uses one of its interfaces as the default interface
when sending packets that are not addressed to a directly connected destination. This is represented by the default
route : 0.0.0.0/0 that is associated to one interface.
When a transport protocol running on the host requests the transmission of a segment, it usually provides the IPv4
destination address to the IPv4 layer in addition to the segment 16 . The IPv4 implementation first performs a
longest prefix match with the destination address in its routing table. The lookup returns the identification of the
interface that must be used to send the packet. The host can then create the IPv4 packet containing the segment.
The source IPv4 address of the packet is the IPv4 address of the host on the interface returned by the longest prefix
match. The Protocol field of the packet is set to the identification of the local transport protocol which created
the segment. The TTL field of the packet is set to the default TTL used by the host. The host must now choose
the packets Identification. This Identification is important if the packet becomes fragmented in the network, as it
ensures that the destination is able to reassemble the received fragments. Ideally, a sending host should never send
a packet twice with the same Identification to the same destination host, in order to ensure that all fragments are
correctly reassembled by the destination. Unfortunately, with a 16 bits Identification field and an expected MSL of
15 In practice, there is usually one DHCP server per group of subnets and the routers capture on each subnet the DHCP messages and
forward them to the DHCP server.
16 A transport protocol implementation can also specify whether the packet must be sent with the DF set or set. A TCP implementation
using Path MTU Discovery would always request the transmission of IPv4 packets with the DF flag set.

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2 minutes, this implies that the maximum bandwidth to a given destination is limited to roughly 286 Mbps. With
a more realistic 1500 bytes MTU, that bandwidth drops to 6.4 Mbps RFC 4963 if fragmentation must be possible
17
. This is very low and is another reason why hosts are highly encouraged to avoid fragmentation. If; despite all
of this, the MTU of the outgoing interface is smaller than the packets length, the packet is fragmented. Finally,
the packets checksum is computed before transmission.
When a host receives an IPv4 packet destined to itself, there are several operations that it must perform. First,
it must check the packets checksum. If the checksum is incorrect, the packet is discarded. Then, it must check
whether the packet has been fragmented. If yes, the packet is passed to the reassembly algorithm described earlier.
Otherwise, the packet must be passed to the upper layer. This is done by looking at the Protocol field (6 for TCP,
17 for UDP). If the host does not implement the transport layer protocol corresponding to the received Protocol
field, it sends a Protocol unreachable ICMP message to the sending host. If the received packet contains an ICMP
message (Protocol field set to 1), the processing is more complex. An Echo-request ICMP message triggers the
transmission of an ICMP Echo-reply message. The other types of ICMP messages indicate an error that was
caused by a previously transmitted packet. These ICMP messages are usually forwarded to the transport protocol
that sent the erroneous packet. This can be done by inspecting the contents of the ICMP message that includes
the header and the first 64 bits of the erroneous packet. If the IP packet did not contain options, which is the case
for most IPv4 packets, the transport protocol can find in the first 32 bits of the transport header the source and
destination ports to determine the affected transport flow. This is important for Path MTU discovery for example.
When a router receives an IPv4 packet, it must first check the packets checksum. If the checksum is invalid, it is
discarded. Otherwise, the router must check whether the destination address is one of the IPv4 addresses assigned
to the router. If so, the router must behave as a host and process the packet as described above. Although routers
mainly forward IPv4 packets, they sometimes need to be accessed as hosts by network operators or network
management software.
If the packet is not addressed to the router, it must be forwarded on an outgoing interface according to the routers
routing table. The router first decrements the packets TTL. If the TTL reaches 0, a TTL Exceeded ICMP message is
sent back to the source. As the packet header has been modified, the checksum must be recomputed. Fortunately,
as IPv4 uses an arithmetic checksum, a router can incrementally update the packets checksum as described in
RFC 1624. Then, the router performs a longest prefix match for the packets destination address in its forwarding
table. If no match is found, the router must return a Destination unreachable ICMP message to the source.
Otherwise, the lookup returns the interface over which the packet must be forwarded. Before forwarding the
packet over this interface, the router must first compare the length of the packet with the MTU of the outgoing
interface. If the packet is smaller than the MTU, it is forwarded. Otherwise, a Fragmentation needed ICMP
message is sent if the DF flag was sent or the packet is fragmented if the DF was not set.
Note: Longest prefix match in IP routers
Performing the longest prefix match at line rate on routers requires highly tuned data structures and algorithms.
Consider for example an implementation of the longest match based on a Radix tree on a router with a 10 Gbps
link. On such a link, a router can receive 31,250,000 40 bytes IPv4 packets every second. To forward the packets
at line rate, the router must process one IPv4 packet every 32 nanoseconds. This cannot be achieved by a software
implementation. For a hardware implementation, the main difficulty lies in the number of memory accesses that
are necessary to perform the longest prefix match. 32 nanoseconds is very small compared to the memory accesses
that are required by a naive longest prefix match implement. Additional information about faster longest prefix
match algorithms may be found in [Varghese2005].

5.2.3 IP version 6
In the late 1980s and early 1990s the growth of the Internet was causing several operational problems on routers.
Many of these routers had a single CPU and up to 1 MByte of RAM to store their operating system, packet buffers
and routing tables. Given the rate of allocation of IPv4 prefixes to companies and universities willing to join the
Internet, the routing tables where growing very quickly and some feared that all IPv4 prefixes would quickly be
allocated. In 1987, a study cited in RFC 1752, estimated that there would be 100,000 networks in the near future.
In August 1990, estimates indicated that the class B space would be exhausted by March 1994. Two types of
17 It should be noted that only the packets that can be fragmented (i.e. whose DF flag is reset) must have different Identification fields. The
Identification field is not used in the packets having the DF flag set.

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solution were developed to solve this problem. The first short term solution was the introduction of Classless Inter
Domain Routing (CIDR). A second short term solution was the Network Address Translation (NAT) mechanism,
defined in RFC 1631. NAT allowed multiple hosts to share a single public IP address, it is explained in section
Middleboxes.
However, in parallel with these short-term solutions, which have allowed the IPv4 Internet to continue to be usable
until now, the Internet Engineering Task Force started to work on developing a replacement for IPv4. This work
started with an open call for proposals, outlined in RFC 1550. Several groups responded to this call with proposals
for a next generation Internet Protocol (IPng) :
TUBA proposed in RFC 1347 and RFC 1561
PIP proposed in RFC 1621
SIPP proposed in RFC 1710
The IETF decided to pursue the development of IPng based on the SIPP proposal. As IP version 5 was already
used by the experimental ST-2 protocol defined in RFC 1819, the successor of IP version 4 is IP version 6. The
initial IP version 6 defined in RFC 1752 was designed based on the following assumptions :
IPv6 addresses are encoded as a 128 bits field
The IPv6 header has a simple format that can easily be parsed by hardware devices
A host should be able to configure its IPv6 address automatically
Security must be part of IPv6
Note: The IPng address size
When the work on IPng started, it was clear that 32 bits was too small to encode an IPng address and all proposals
used longer addresses. However, there were many discussions about the most suitable address length. A first
approach, proposed by SIP in RFC 1710, was to use 64 bit addresses. A 64 bits address space was 4 billion times
larger than the IPv4 address space and, furthermore, from an implementation perspective, 64 bit CPUs were being
considered and 64 bit addresses would naturally fit inside their registers. Another approach was to use an existing
address format. This was the TUBA proposal ( RFC 1347) that reuses the ISO CLNP 20 bytes addresses. The
20 bytes addresses provided room for growth, but using ISO CLNP was not favored by the IETF partially due to
political reasons, despite the fact that mature CLNP implementations were already available. 128 bits appeared to
be a reasonable compromise at that time.

IPv6 addressing architecture

The experience of IPv4 revealed that the scalability of a network layer protocol heavily depends
on its addressing architecture.
The designers of IPv6 spent a lot of effort defining its addressing architecture RFC 3513.
All IPv6 addresses are 128 bits wide.
This implies that there are
340, 282, 366, 920, 938, 463, 463, 374, 607, 431, 768, 211, 456(3.4 1038 ) different IPv6 addresses. As the surface of the Earth is about 510,072,000 km2 , this implies that there are about 6.67 1023 IPv6 addresses per
square meter on Earth. Compared to IPv4, which offers only 8 addresses per square kilometer, this is a significant
improvement on paper.
IPv6 supports unicast, multicast and anycast addresses. As with IPv4, an IPv6 unicast address is used to identify
one datalink-layer interface on a host. If a host has several datalink layer interfaces (e.g. an Ethernet interface and
a WiFi interface), then it needs several IPv6 addresses. In general, an IPv6 unicast address is structured as shown
in the figure below.
An IPv6 unicast address is composed of three parts :
1. A global routing prefix that is assigned to the Internet Service Provider that owns this block of addresses
2. A subnet identifier that identifies a customer of the ISP
3. An interface identifier that identifies a particular interface on an endsystem

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Figure 5.36: Structure of IPv6 unicast addresses

In todays deployments, interface identifiers are always 64 bits wide. This implies that while there are 2128
different IPv6 addresses, they must be grouped in 264 subnets. This could appear as a waste of resources, however
using 64 bits for the host identifier allows IPv6 addresses to be auto-configured and also provides some benefits
from a security point of view, as explained in section ICMPv6
Note: Textual representation of IPv6 addresses
It is sometimes necessary to write IPv6 addresses in text format, e.g. when manually configuring addresses or for
documentation purposes. The preferred format for writing IPv6 addresses is x:x:x:x:x:x:x:x, where the x s are
hexadecimal digits representing the eight 16-bit parts of the address. Here are a few examples of IPv6 addresses :
ABCD:EF01:2345:6789:ABCD:EF01:2345:6789
2001:DB8:0:0:8:800:200C:417A
FE80:0:0:0:219:E3FF:FED7:1204
IPv6 addresses often contain a long sequence of bits set to 0. In this case, a compact notation has been defined.
With this notation, :: is used to indicate one or more groups of 16 bits blocks containing only bits set to 0. For
example,
2001:DB8:0:0:8:800:200C:417A is represented as 2001:DB8::8:800:200C:417A
FF01:0:0:0:0:0:0:101 is represented as FF01::101
0:0:0:0:0:0:0:1 is represented as ::1
0:0:0:0:0:0:0:0 is represented as ::
An IPv6 prefix can be represented as address/length, where length is the length of the prefix in bits. For example,
the three notations below correspond to the same IPv6 prefix :
2001:0DB8:0000:CD30:0000:0000:0000:0000/60
2001:0DB8::CD30:0:0:0:0/60
2001:0DB8:0:CD30::/60
In practice, there are several types of IPv6 unicast address. Most of the IPv6 unicast addresses are allocated in
blocks under the responsibility of IANA. The current IPv6 allocations are part of the 2000::/3 address block.
Regional Internet Registries (RIR) such as RIPE in Europe, ARIN in North-America or AfriNIC in Africa have
each received a block of IPv6 addresses that they sub-allocate to Internet Service Providers in their region. The
ISPs then sub-allocate addresses to their customers.
When considering the allocation of IPv6 addresses, two types of address allocations are often distinguished. The
RIRs allocate provider-independent (PI) addresses. PI addresses are usually allocated to Internet Service Providers
and large companies that are connected to at least two different ISPs [CSP2009]. Once a PI address block has
been allocated to a company, this company can use its address block with the provider of its choice and change
its provider at will. Internet Service Providers allocate provider-aggregatable (PA) address blocks from their own
PI address block to their customers. A company that is connected to only one ISP should only use PA addresses.

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The drawback of PA addresses is that when a company using a PA address block changes its provider, it needs to
change all the addresses that it uses. This can be a nightmare from an operational perspective and many companies
are lobbying to obtain PI address blocks even if they are small and connected to a single provider. The typical size
of the IPv6 address blocks are :
/32 for an Internet Service Provider
/48 for a single company
/64 for a single user (e.g. a home user connected via ADSL)
/128 in the rare case when it is known that no more than one endhost will be attached
For the companies that want to use IPv6 without being connected to the IPv6 Internet, RFC 4193 defines the
Unique Local Unicast (ULA) addresses (FC00::/7). These ULA addresses play a similar role as the private IPv4
addresses defined in RFC 1918. However, the size of the FC00::/7 address block allows ULA to be much more
flexible than private IPv4 addresses.
Furthermore, the IETF has reserved some IPv6 addresses for a special usage. The two most important ones are :
0:0:0:0:0:0:0:1 (::1 in compact form) is the IPv6 loopback address. This is the address of a logical interface
that is always up and running on IPv6 enabled hosts. This is the equivalent of 127.0.0.1 in IPv4.
0:0:0:0:0:0:0:0 (:: in compact form) is the unspecified IPv6 address. This is the IPv6 address that a host
can use as source address when trying to acquire an official address.
The last type of unicast IPv6 addresses are the Link Local Unicast addresses. These addresses are part of the
FE80::/10 address block and are defined in RFC 4291. Each host can compute its own link local address by
concatenating the FE80::/64 prefix with the 64 bits identifier of its interface. Link local addresses can be used
when hosts that are attached to the same link (or local area network) need to exchange packets. They are used
notably for address discovery and auto-configuration purposes. Their usage is restricted to each link and a router
cannot forward a packet whose source or destination address is a link local address. Link local addresses have also
been defined for IPv4 RFC 3927. However, the IPv4 link local addresses are only used when a host cannot obtain
a regular IPv4 address, e.g. on an isolated LAN.

Figure 5.37: IPv6 link local address structure

An important consequence of the IPv6 unicast addressing architecture and the utilisation of link-local addresses is
that an IPv6 host has several IPv6 addresses. This implies that an IPv6 stack must be able to handle multiple IPv6
addresses. This was not always the case with IPv4.
RFC 4291 defines a special type of IPv6 anycast address. On a subnetwork having prefix p/n, the IPv6 address
whose 128-n low-order bits are set to 0 is the anycast address that corresponds to all routers inside this subnetwork. This anycast address can be used by hosts to quickly send a packet to any of the routers inside their own
subnetwork.
Finally, RFC 4291 defines the structure of the IPv6 multicast addresses 18 . This structure is depicted in the figure
below
The low order 112 bits of an IPv6 multicast address are the groups identifier. The high order bits are used as a
marker to distinguish multicast addresses from unicast addresses. Notably, the 4 bits flag field indicates whether
the address is temporary or permanent. Finally, the scope field indicates the boundaries of the forwarding of
packets destined to a particular address. A link-local scope indicates that a router should not forward a packet
destined to such a multicast address. An organisation local-scope indicates that a packet sent to such a multicast
destination address should not leave the organisation. Finally the global scope is intended for multicast groups
spanning the global Internet.
18

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Figure 5.38: IPv6 multicast address structure

Among these addresses, some are well known. For example, all endsystem automatically belong to the FF02::1
multicast group while all routers automatically belong to the FF02::2 multicast group. We discuss IPv6 multicast
later.
IPv6 packet format
The IPv6 packet format was heavily inspired by the packet format proposed for the SIPP protocol in RFC 1710.
The standard IPv6 header defined in RFC 2460 occupies 40 bytes and contains 8 different fields, as shown in the
figure below.

Figure 5.39: The IP version 6 header ( RFC 2460)

Apart from the source and destination addresses, the IPv6 header contains the following fields :
version : a 4 bits field set to 6 and intended to allow IP to evolve in the future if needed
Traffic class : this 8 bits field plays a similar role as the DS byte in the IPv4 header
Flow label : this field was initially intended to be used to tag packets belonging to the same flow. However,
as of this writing, there is no clear guideline on how this field should be used by hosts and routers
Payload length : this is the size of the packet payload in bytes. As the length is encoded as a 16 bits field,
an IPv6 packet can contain up to 65535 bytes of payload.
Next Header : this 8 bits field indicates the type 19 of header that follows the IPv6 header. It can be a
transport layer header (e.g. 6 for TCP or 17 for UDP) or an IPv6 option. Handling options as a next header
allows simplifying the processing of IPv6 packets compared to IPv4.
19 The IANA maintains the list of all allocated Next Header types at http://www.iana.org/assignments/protocol-numbers/ The same registry
is used for the IPv4 protocol field and for the IPv6 Next Header.

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Hop Limit : this 8 bits field indicates the number of routers that can forward the packet. It is decremented
by one by each router and has the same purpose as the TTL field of the IPv4 header.
In comparison with IPv4, the IPv6 packets are much simpler and easier to process by routers. A first important
difference is that there is no checksum inside the IPv6 header. This is mainly because all datalink layers and
transport protocols include a checksum or a CRC to protect their frames/segments against transmission errors.
Adding a checksum in the IPv6 header would have forced each router to recompute the checksum of all packets,
with limited benefit in detecting errors. In practice, an IP checksum allows for catching errors that occur inside
routers (e.g. due to memory corruption) before the packet reaches its destination. However, this benefit was found
to be too small given the reliability of current memories and the cost of computing the checksum on each router.
A second difference with IPv4 is that the IPv6 header does not support fragmentation and reassembly. Experience
with IPv4 has shown that fragmenting packets in routers was costly [KM1995] and the developers of IPv6 have
decided that routers would not fragment packets anymore. If a router receives a packet that is too long to be
forwarded, the packet is dropped and the router returns an ICMPv6 messages to inform the sender of the problem.
The sender can then either fragment the packet or perform Path MTU discovery. In IPv6, packet fragmentation is
performed only by the source by using IPv6 options.
The third difference are the IPv6 options, which are simpler and easier to process than the IPv4 options.
Note: Header compression on low bandwidth links
Given the size of the IPv6 header, it can cause huge overhead on low bandwidth links, especially when small
packets are exchanged such as for Voice over IP applications. In such environments, several techniques can be
used to reduce the overhead. A first solution is to use data compression in the datalink layer to compress all the
information exchanged [Thomborson1992]. These techniques are similar to the data compression algorithms used
in tools such as compress(1) or gzip(1) RFC 1951. They compress streams of bits without taking advantage
of the fact that these streams contain IP packets with a known structure. A second solution is to compress the IP
and TCP header. These header compression techniques, such as the one defined in RFC 2507 take advantage of
the redundancy found in successive packets from the same flow to significantly reduce the size of the protocol
headers. Another solution is to define a compressed encoding of the IPv6 header that matches the capabilities of
the underlying datalink layer RFC 4944.

IPv6 options
In IPv6, each option is considered as one header containing a multiple of 8 bytes to ensure that IPv6 options in a
packet are aligned on 64 bit boundaries. IPv6 defines several type of options :
the hop-by-hop options are options that must be processed by the routers on the packets path
the type 0 routing header, which is similar to the IPv4 loose source routing option
the fragmentation option, which is used when fragmenting an IPv6 packet
the destination options
the security options that allow IPv6 hosts to exchange packets with cryptographic authentication (AH
header) or encryption and authentication (ESP header)
RFC 2460 provides lots of detail on the encodings of the different types of options. In this section, we only discus
some of them. The reader may consult RFC 2460 for more information about the other options. The first point to
note is that each option contains a Next Header field, which indicates the type of header that follows the option.
A second point to note is that in order to allow routers to efficiently parse IPv6 packets, the options that must be
processed by routers (hop-by-hop options and type 0 routing header) must appear first in the packet. This allows
the router to process a packet without being forced to analyse all the packets options. A third point to note is
that hop-by-hop and destination options are encoded using a type length value format. Furthermore, the type field
contains bits that indicate whether a router that does not understand this option should ignore the option or discard
the packet. This allows the introduction of new options into the network without forcing all devices to be upgraded
to support them at the same time.
Two hop-by-hop options have been defined. RFC 2675 specifies the jumbogram that enables IPv6 to support
packets containing a payload larger than 65535 bytes. These jumbo packets have their payload length set to 0 and
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the jumbogram option contains the packet length as a 32 bits field. Such packets can only be sent from a source
to a destination if all the routers on the path support this option. However, as of this writing it does not seem that
the jumbogram option has been implemented. The router alert option defined in RFC 2711 is the second example
of a hop-by-hop option. The packets that contain this option should be processed in a special way by intermediate
routers. This option is used for IP packets that carry Resource Reservation Protocol (RSVP) messages. Its usage
is explained later.
The type 0 routing header defined in RFC 2460 is an example of an IPv6 option that must be processed by some
routers. This option is encoded as shown below.

Figure 5.40: The Type 0 routing header ( RFC 2460)

The type 0 routing option was intended to allow a host to indicate a loose source route that should be followed by
a packet by specifying the addresses of some of the routers that must forward this packet. Unfortunately, further
work with this routing header, including an entertaining demonstration with scapy [BE2007] , revealed some
severe security problems with this routing header. For this reason, loose source routing with the type 0 routing
header has been removed from the IPv6 specification RFC 5095.
In IPv6, fragmentation is performed exclusively by the source host and relies on the fragmentation header. This
64 bits header is composed of six fields :
a Next Header field that indicates the type of the header that follows the fragmentation header
a reserved field set to 0.
the Fragment Offset is a 13-bit unsigned integer that contains the offset, in 8 bytes units, of the data following
this header, relative to the start of the original packet.
the More flag, which is set to 0 in the last fragment of a packet and to 1 in all other fragments.
the 32 bits Identification field indicates to which original packet a fragment belongs. When a host sends
fragmented packets, it should ensure that it does not reuse the same identification field for packets sent to
the same destination during a period of MSL seconds. This is easier with the 32 bits identification used in
the IPv6 fragmentation header, than with the 16 bits identification field of the IPv4 header.
Some IPv6 implementations send the fragments of a packet in increasing fragment offset order, starting from the
first fragment. Others send the fragments in reverse order, starting from the last fragment. The latter solution can
be advantageous for the host that needs to reassemble the fragments, as it can easily allocate the buffer required to

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reassemble all fragments of the packet upon reception of the last fragment. When a host receives the first fragment
of an IPv6 packet, it cannot know a priori the length of the entire IPv6 packet.
The figure below provides an example of a fragmented IPv6 packet containing a UDP segment. The Next Header
type reserved for the IPv6 fragmentation option is 44.

Figure 5.41: IPv6 fragmentation example

Finally, the last type of IPv6 options is the Encaspulating Security Payload (ESP) defined in RFC 4303 and the
Authentication Header (AH) defined in RFC 4302. These two headers are used by IPSec RFC 4301. They are
discussed in another chapter.

5.2.4 ICMP version 6

ICMPv6 defined in RFC 4443 is the companion protocol for IPv6 as ICMPv4 is the companion protocol for IPv4.
ICMPv6 is used by routers and hosts to report problems when processing IPv6 packets. However, as we will see in
chapter The datalink layer and the Local Area Networks, ICMPv6 is also used when auto-configuring addresses.
The traditional utilisation of ICMPv6 is similar to ICMPv4. ICMPv6 messages are carried inside IPv6 packets
(the Next Header field for ICMPv6 is 58). Each ICMP message contains an 8 bits header with a type field, a code
field and a 16 bits checksum computed over the entire ICMPv6 message. The message body contains a copy of
the IPv6 packet in error.

Figure 5.42: ICMP version 6 packet format

ICMPv6 specifies two classes of messages : error messages that indicate a problem in handling a packet and
informational messages. Four types of error messages are defined in RFC 4443 :
1 [Destination Unreachable. Such an ICMPv6 message is sent when the destination address of a packet
is unreachable. The code field of the ICMP header contains additional information about the type of
unreachability. The following codes are specified in RFC 4443]
0 : No route to destination. This indicates that the router that sent the ICMPv6 message did not
have a route towards the packets destination

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1 : Communication with destination administratively prohibited. This indicates that a firewall has
refused to forward the packet towards its destination.
2 : Beyond scope of source address. This message can be sent if the source is using link-local
addresses to reach a global unicast address outside its subnet.
3 : Address unreachable. This message indicates that the packet reached the subnet of the destination, but the host that owns this destination address cannot be reached.
4 : Port unreachable. This message indicates that the IPv6 packet was received by the destination,
but there was no application listening to the specified port.
2 : Packet Too Big. The router that was to send the ICMPv6 message received an IPv6 packet that is larger
than the MTU of the outgoing link. The ICMPv6 message contains the MTU of this link in bytes. This
allows the sending host to implement Path MTU discovery RFC 1981
3 : Time Exceeded. This error message can be sent either by a router or by a host. A router would set code
to 0 to report the reception of a packet whose Hop Limit reached 0. A host would set code to 1 to report that
it was unable to reassemble received IPv6 fragments.
4 : Parameter Problem. This ICMPv6 message is used to report either the reception of an IPv6 packet with
an erroneous header field (type 0) or an unknown Next Header or IP option (types 1 and 2). In this case, the
message body contains the erroneous IPv6 packet and the first 32 bits of the message body contain a pointer
to the error.
Two types of informational ICMPv6 messages are defined in RFC 4443 : echo request and echo reply, which are
used to test the reachability of a destination by using ping6(8).
ICMPv6 also allows the discovery of the path between a source and a destination by using traceroute6(8).
The output below shows a traceroute between a host at UCLouvain and one of the main IETF servers. Note
that this IPv6 path is different than the IPv4 path that was described earlier although the two traceroutes were
performed at the same time.

traceroute6 www.ietf.org
traceroute6 to www.ietf.org (2001:1890:1112:1::20) from 2001:6a8:3080:2:217:f2ff:fed6:65c0, 30 hop
1 2001:6a8:3080:2::1 13.821 ms 0.301 ms 0.324 ms
2 2001:6a8:3000:8000::1 0.651 ms 0.51 ms 0.495 ms
3 10ge.cr2.bruvil.belnet.net 3.402 ms 3.34 ms 3.33 ms
4 10ge.cr2.brueve.belnet.net 3.668 ms 10ge.cr2.brueve.belnet.net 3.988 ms 10ge.cr2.brueve.beln
5 belnet.rt1.ams.nl.geant2.net 10.598 ms 7.214 ms 10.082 ms
6 so-7-0-0.rt2.cop.dk.geant2.net 20.19 ms 20.002 ms 20.064 ms
7 kbn-ipv6-b1.ipv6.telia.net 21.078 ms 20.868 ms 20.864 ms
8 s-ipv6-b1-link.ipv6.telia.net 31.312 ms 31.113 ms 31.411 ms
9 s-ipv6-b1-link.ipv6.telia.net 61.986 ms 61.988 ms 61.994 ms
10 2001:1890:61:8909::1 121.716 ms 121.779 ms 121.177 ms
11 2001:1890:61:9117::2 203.709 ms 203.305 ms 203.07 ms
12 mail.ietf.org 204.172 ms 203.755 ms 203.748 ms

Note: Rate limitation of ICMP messages

High-end hardware based routers use special purpose chips on their interfaces to forward IPv6 packets at line
rate. These chips are optimised to process correct IP packets. They are not able to create ICMP messages at line
rate. When such a chip receives an IP packet that triggers an ICMP message, it interrupts the main CPU of the
router and the software running on this CPU processes the packet. This CPU is much slower than the hardware
acceleration found on the interfaces [Gill2004]. It would be overloaded if it had to process IP packets at line rate
and generate one ICMP message for each received packet. To protect this CPU, high-end routers limit the rate at
which the hardware can interrupt the main CPU and thus the rate at which ICMP messages can be generated. This
implies that not all erroneous IP packets cause the transmission of an ICMP message. The risk of overloading the
main CPU of the router is also the reason why using hop-by-hop IPv6 options, including the router alter option is
discouraged 20 .
20 For a discussion of the issues with the router alert IP option, see http://tools.ietf.org/html/draft-rahman-rtg-router-alert-dangerous-00 or
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Interactions between IPv6 and the datalink layer

There are several differences between IPv6 and IPv4 when considering their interactions with the datalink layer.
In IPv6, the interactions between the network and the datalink layer is performed using ICMPv6.
First ICMPv6 is used to resolve the datalink layer address that corresponds to a given IPv6 address. This part
of ICMPv6 is the Neighbour Discovery Protocol (NDP) defined in RFC 4861. NDP is similar to ARP, but there
are two important differences. First, NDP messages are exchanged in ICMPv6 messages while ARP messages
are sent as datalink layer frames. Second, an ARP request is sent as a broadcast frame while an NDP solicitation
message is sent as a multicast ICMPv6 packet that is transported inside a multicast frame. The operation of the
NDP protocol is similar to ARP. To obtain an address mapping, a host sends a Neighbour Solicitation message.
This message is sent inside an ICMPv6 message that is placed in an IPv6 packet whose source address is the
IPv6 address of the requesting host and the destination address is the all-hosts IPv6 multicast address (FF02::1)
to which all IPv6 hosts listen. The Neighbour Solicitation contains the requested IPv6 address. The owner of the
requested address replies by sending a unicast Neighbour Advertisement message to the requesting host. NDP
suffers from similar security issues as the ARP protocol. However, it is possible to secure NDP by using the
Cryptographically Generated IPv6 Addresses (CGA) defined in RFC 3972. The Secure Neighbour Discovery
Protocol is defined in RFC 3971, but a detailed description of this protocol is outside the scope of this chapter.
IPv6 networks also support the Dynamic Host Configuration Protocol. The IPv6 extensions to DHCP are defined
in RFC 3315. The operation of DHCPv6 is similar to DHCP that was described earlier. In addition to DHCPv6,
IPv6 networks support another mechanism to assign IPv6 addresses to hosts. This is the Stateless Address Configuration (SLAC) defined in RFC 4862. When a host boots, it derives its identifier from its datalink layer address 21
and concatenates this 64 bits identifier to the FE80::/64 prefix to obtain its link-local IPv6 address. It then sends
a Neighbour Solicitation with its link-local address as a target to verify whether another host is using the same
link-local address on this subnet. If it receives a Neighbour Advertisement indicating that the link-local address
is used by another host, it generates another 64 bits identifier and sends again a Neighbour Solicitation. If there is
no answer, the host considers its link-local address to be valid. This address will be used as the source address for
all NDP messages sent on the subnet. To automatically configure its global IPv6 address, the host must know the
globally routable IPv6 prefix that is used on the local subnet. IPv6 routers regularly send ICMPv6 Router Advertisement messages that indicate the IPv6 prefix assigned to each subnet. Upon reception of this message, the host
can derive its global IPv6 address by concatenating its 64 bits identifier with the received prefix. It concludes the
SLAC by sending a Neighbour Solicitation message targeted at its global IPv6 address to ensure that another host
is not using the same IPv6 address.

5.2.5 Middleboxes
When the TCP/IP architecture and the IP protocol were defined, two type of devices were considered in the
network layer : endhosts and routers. Endhosts are the sources and destinations of IP packets while routers
forward packets. When a router forwards an IP packet, it consults its forwarding table, updates the packets TTL,
recomputes its checksum and forwards it to the next hop. A router does not need to read or change the contents of
the packets payload.
However, in todays Internet, there exist devices that are not strictly routers but which process, sometimes modify,
and forward IP packets. These devices are often called middleboxes RFC 3234. Some middleboxes only operate
in the network layer, but most middleboxes are able to analyse the payload of the received packets and extract the
transport header, and in some cases the application layer protocols.
In this section, we briefly describe two type of middleboxes : firewalls and network address translation (NAT)
devices. A discussion of the different types of middleboxes with references may be found in RFC 3234.
Firewalls
When the Internet was only a research network interconnecting research labs, security was not a concern, and
most hosts agreed to exchange packets over TCP connections with most other hosts. However, as more and more
21 Using a datalink layer address to derive a 64 bits identifier for each host raises privacy concerns as the host will always use the same
identifier. Attackers could use this to track hosts on the Internet. An extension to the Stateless Address Configuration mechanism that does not
raise privacy concerns is defined in RFC 4941. These privacy extensions allow a host to generate its 64 bits identifier randomly every time it
attaches to a subnet. It then becomes impossible for an attacker to use the 64-bits identifier to track a host.

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Figure 5.43: IP middleboxes and the reference model

users and companies became connected to the Internet, allowing unlimited access to hosts that they managed
started to concern companies. Furthermore, at the end of the 1980s, several security issues affected the Internet,
such as the first Internet worm [RE1989] and some widely publicised security breaches [Stoll1988] [CB2003]
[Cheswick1990] .
These security problems convinced the industry that IP networks are a key part of a companys infrastructure, that
should be protected by special devices like security guards and fences are used to protect buildings. These special
devices were quickly called firewalls. A typical firewall has two interfaces :
an external interface connected to the global Internet
an internal interface connected to a trusted network
The first firewalls included configurable packet filters. A packet filter is a set of rules defining the security policy
of a network. In practice, these rules are based on the values of fields in the IP or transport layer headers. Any
field of the IP or transport header can be used in a firewall rule, but the most common ones are:
filter on the source address. For example, a company may decide to discard all packets received from one
of its competitors. In this case, all packets whose source address belong to the competitors address block
would be rejected
filter on destination address. For example, the hosts of the research lab of a company may receive packets
from the global Internet, but not the hosts of the financial department
filter on the Protocol number found in the IP header. For example, a company may only allow its hosts to
use TCP or UDP, but not other, more experimental, transport protocols
filter on the TCP or UDP port numbers. For example, only the DNS server of a company should received
UDP segments whose destination port is set to 53 or only the official SMTP servers of the company can
send TCP segments whose source ports are set to 25
filter on the TCP flags. For example, a simple solution to prohibit external hosts from opening TCP connections with hosts inside the company is to discard all TCP segments received from the external interface
with only the SYN flag set.
Such firewalls are often called stateless firewalls because they do not maintain any state about the TCP connections
that pass through them.
Another type of firewalls are stateful firewalls. A stateful firewall tracks the state of each TCP connection passing
through it and maintains a TCB for each of these TCP connection. This TCB allows it to reassemble the received
segments in order to extract their payload and perform verifications in the application layer. Some firewalls are
able to inspect the URLs accessed using HTTP and log all URLs visited or block TCP connections where a
dangerous URL is exchanged. Some firewalls can verify that SMTP commands are used when a TCP connection
is established on port 25 or that a TCP connection on port 80 carries HTTP commands and responses.
Note: Beyond firewalls
Apart from firewalls, different types of security devices have been installed at the periphery of corporate networks. Intrusion Detection Systems (IDS), such as the popular snort , are stateful devices that are capable of
matching reassembled segments against regular expressions corresponding to signatures of viruses, worms or
other types of attacks. Deep Packet Inspection (DPI) is another type of middlebox that analyses the packets payload and is able to reassemble TCP segments in order to detect inappropriate usages. While IDS are mainly used
in corporate networks, DPI is mainly used in Internet Service Providers. Some ISPs use DPI to detect and limit

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the bandwidth consumed by peer-to-peer applications. Some countries such as China or Iran use DPI to detect
inappropriate Internet usage.

NAT
Network Address Translation (NAT) was proposed in [TE1993] and RFC 3022 as a short term solution to deal
with the expected shortage of IPv4 addresses in the late 1980s - early 1990s. Combined with CIDR, NAT helped
to significantly slow down the consumption of IPv4 addresses. A NAT is a middlebox that interconnects two
networks that are using IPv4 addresses from different addressing spaces. Usually, one of these addressing spaces
is the public Internet while the other is using the private IPv4 addresses defined in RFC 1918.
A very common deployment of NAT is in broadband access routers as shown in the figure below. The broadband
access router interconnects a home network, either WiFi or Ethernet based, and the global Internet via one ISP
over ADSL or CATV. A single IPv4 address is allocated to the broadband access router and network address
translation allows all of the hosts attached to the home network to share a single public IPv4 address.

Figure 5.44: A simple NAT with one public IPv4 address

A second type of deployment is in enterprise networks as shown in the figure below. In this case, the NAT
functionality is installed on a border router of the enterprise. A private IPv4 address is assigned to each enterprise
host while the border router manages a pool containing several public IPv4 addresses.

Figure 5.45: An enterprise NAT with several public IPv4 addresses

As the name implies, a NAT is a device that translates IP addresses. A NAT maintains a mapping table between
the private IP addresses used in the internal network and the public IPv4 addresses. NAT allows a large number of
hosts to share a pool of IP addresses, as these hosts do not all access the global Internet at the same time.
The simplest NAT is a middlebox that uses a one-to-one mapping between a private IP address and a public IP
address. To understand its operation, let us assume that a NAT, such as the one shown above, has just booted.
When the NAT receives the first packet from source S in the internal network which is destined to the public
Internet, it creates a mapping between internal address S and the first address of its pool of public addresses (P1).
Then, it translates the received packet so that it can be sent to the public Internet. This translation is performed as
followed :

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the source address of the packet (S) is replaced by the mapped public address (P1)
the checksum of the IP header is incrementally updated as its content has changed
if the packet carried a TCP or UDP segment, the transport layer checksum found in the included segment
must also be updated as it is computed over the segment and a pseudo-header that includes the source and
destination addresses
When a packet destined to P1 is received from the public Internet, the NAT consults its mapping table to find S.
The received packet is translated and forwarded in the internal network.
This works as long as the pool of public IP addresses of the NAT does not become empty. In this case, a mapping
must be removed from the mapping table to allow a packet from a new host to be translated. This garbage
collection can be implemented by adding to each entry in the mapping table a timestamp that contains the last
utilisation time of a mapping entry. This timestamp is updated each time the corresponding entry is used. Then,
the garbage collection algorithm can remove the oldest mapping entry in the table.
A drawback of such a simple enterprise NAT is the size of the pool of public IPv4 addresses which is often too
small to allow a large number of hosts share such a NAT. In this case, a better solution is to allow the NAT to
translate both IP addresses and port numbers.
Such a NAT maintains a mapping table that maps an internal IP address and TCP port number with an external
IP address and TCP port number. When such a NAT receives a packet from the internal network, it performs a
lookup in the mapping table with the packets source IP address and source TCP port number. If a mapping is
found, the source IP address and the source TCP port number of the packet are translated with the values found
in the mapping table, the checksums are updated and the packet is sent to the global Internet. If no mapping is
found, a new mapping is created with the first available couple (IP address, TCP port number) and the packet is
translated. The entries of the mapping table are either removed at the end of the corresponding TCP connection
as the NAT tracks TCP connection state like a stateful firewall or after some idle time.
When such a NAT receives a packet from the global Internet, it looks up its mapping table for the packets destination IP address and destination TCP port number. If a mapping is found, the packet is translated and forwarded
into the internal network. Otherwise, the packet is discarded as the NAT cannot determine to which particular
internal host the packet should be forwarded. For this reason,
With 216 different port numbers, a NAT may support a large number of hosts with a single public IPv4 address.
However, it should be noted that some applications open a large number of TCP connections [Miyakawa2008].
Each of these TCP connections consumes one mapping entry in the NATs mapping table.
NAT allows many hosts to share one or a few public IPv4 addresses. However, using NAT has two important
drawbacks. First, it is difficult for external hosts to open TCP connections with hosts that are behind a NAT. Some
consider this to be a benefit from a security perspective. However, a NAT should not be confused with a firewall
as there are some techniques to traverse NATs. Second, NAT breaks the end-to-end transparency of the network
and transport layers. The main problem is when an application layer protocol uses IP addresses in some of the
ADUs that it sends. A popular example is ftp defined in RFC 959. In this case, there is a mismatch between the
packet header translated by the NAT and the packet payload. The only solution to solve this problem is to place
an Application Level Gateway (ALG) on the NAT that understands the application layer protocol and can thus
translate the IP addresses and port numbers found in the ADUs. However, defining an ALG for each application
is costly and application developers should avoid using IP addresses in the messages exchanged in the application
layer RFC 3235.
Note: IPv6 and NAT
NAT has been very successful with IPv4. Given the size of the IPv6 addressing space, the IPv6 designers expected
that NAT would never be useful with IPv6. The end-to-end transparency of IPv6 has been one of its key selling
points compared to IPv4. However, the expected shortage of IPv4 addresses lead enterprise network administrators
to consider IPv6 more seriously. One of the results of this analysis is that the IETF defined NAT devices [WB2008]
that are IPv6 specific. Another usage of NAT with IPv6 is to allow IPv6 hosts to access IPv4 destinations and
conversely. The early IPv6 specifications included the Network Address Translation - Protocol Translation (NATPT) mechanism defined in RFC 2766. This mechanism was later deprecated in RFC 4966 but has been recently
restarted under the name NAT64 [BMvB2009]. A NAT64 is a middlebox that performs the IPv6<->IPv4 packet
translation to allow IPv6 hosts to contact IPv4 servers RFC 6144.

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5.3 Routing in IP networks

In a large IP network such as the global Internet, routers need to exchange routing information. The Internet is
an interconnection of networks, often called domains, that are under different responsibilities. As of this writing,
the Internet is composed on more than 30,000 different domains and this number is still growing. A domain can
be a small enterprise that manages a few routers in a single building, a larger enterprise with a hundred routers
at multiple locations, or a large Internet Service Provider managing thousands of routers. Two classes of routing
protocols are used to allow these domains to efficiently exchange routing information.

Figure 5.46: Organisation of a small Internet

The first class of routing protocols are the intradomain routing protocols (sometimes also called the interior gateway protocols or IGP). An intradomain routing protocol is used by all routers inside a domain to exchange routing
information about the destinations that are reachable inside the domain. There are several intradomain routing
protocols. Some domains use RIP, which is a distance vector protocol. Other domains use link-state routing protocols such as OSPF or IS-IS. Finally, some domains use static routing or proprietary protocols such as IGRP or
EIGRP.
These intradomain routing protocols usually have two objectives. First, they distribute routing information that
corresponds to the shortest path between two routers in the domain. Second, they should allow the routers to
quickly recover from link and router failures.
The second class of routing protocols are the interdomain routing protocols (sometimes also called the exterior
gateway protocols or EGP). The objective of an interdomain routing protocol is to distribute routing information
between domains. For scalability reasons, an interdomain routing protocol must distribute aggregated routing
information and considers each domain as a black box.
A very important difference between intradomain and interdomain routing are the routing policies that are used
by each domain. Inside a single domain, all routers are considered equal, and when several routes are available
to reach a given destination prefix, the best route is selected based on technical criteria such as the route with the
shortest delay, the route with the minimum number of hops or the route with the highest bandwidth.
When we consider the interconnection of domains that are managed by different organisations, this is no longer
true. Each domain implements its own routing policy. A routing policy is composed of three elements : an import
filter that specifies which routes can be accepted by a domain, an export filter that specifies which routes can be
advertised by a domain and a ranking algorithm that selects the best route when a domain knows several routes
towards the same destination prefix. As we will see later, another important difference is that the objective of the
interdomain routing protocol is to find the cheapest route towards each destination. There is only one interdomain
routing protocol : BGP.

5.3.1 Intradomain routing

In this section, we briefly describe the key features of the two main intradomain unicast routing protocols : RIP
and OSPF.

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RIP
The Routing Information Protocol (RIP) is the simplest routing protocol that was standardised for the TCP/IP
protocol suite. RIP is defined in RFC 2453. Additional information about RIP may be found in [Malkin1999]
RIP routers periodically exchange RIP messages. The format of these messages is shown below. A RIP message
is sent inside a UDP segment whose destination port is set to 521. A RIP message contains several fields. The
Cmd field indicates whether the RIP message is a request or a response. Routers send one of more RIP response
messages every 30 seconds. These messages contain the distance vectors that summarize the routers routing table.
The RIP request messages can be used by routers or hosts to query other routers about the content of their routing
table. A typical usage is when a router boots and quickly wants to receive the RIP responses from its neighbours
to compute its own routing table. The current version of RIP is version 2 defined in RFC 2453 for IPv4 and RFC
2080 for IPv6.

Figure 5.47: RIP message format

The RIP header contains an authentication field. This authentication can be used by network administrators to
ensure that only the RIP messages sent by the routers that they manage are used to build the routing tables. RFC
2453 only supports a basic authentication scheme where all routers are configured with the same password and
include this password in all RIP messages. This is not very secure since an attacker can know the password
by capturing a single RIP message. However, this password can protect against configuration errors. Stronger
authentication schemes are described in RFC 2082 and RFC 4822, but the details of these mechanisms are outside
the scope of this section.
Each RIP message contains a set of route entries. Each route entry is encoded as a 20 bytes field whose format
is shown below. RIP was initially designed to be suitable for different network layer protocols. Some implementations of RIP were used in XNS or IPX networks. The first field of the RIP route entry is the Address Family
Identifier (AFI). This identifier indicates the type of address found in the route entry 22 . IPv4 uses AFI=1. The
other important fields of the route entry are the IPv4 prefix, the netmask that indicates the length of the subnet
identifier and is encoded as a 32 bits netmask and the metric. Although the metric is encoded as a 32 bits field, the
maximum RIP metric is 15 (for RIP, 16 = )

Figure 5.48: Format of the RIP IPv4 route entries ( RFC 2453)
With a 20 bytes route entry, it was difficult to use the same format as above to support IPv6. Instead of defining a
variable length route entry format, the designers of RFC 2080 defined a new format that does not include an AFI
field. The format of the route entries used by RFC 2080 is shown below. Plen is the length of the subnet identifier
in bits and the metric is encoded as one byte. The maximum metric is still 15.
22

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Figure 5.49: Format of the RIP IPv6 route entries

Note: A note on timers
The first RIP implementations sent their distance vector exactly every 30 seconds. This worked well in most
networks, but some researchers noticed that routers were sometimes overloaded because they were processing
too many distance vectors at the same time [FJ1994]. They collected packet traces in these networks and found
that after some time the routers timers became synchronised, i.e. almost all routers were sending their distance
vectors at almost the same time. This synchronisation of the transmission times of the distance vectors caused
an overload on the routers CPU but also increased the convergence time of the protocol in some cases. This
was mainly due to the fact that all routers set their timers to the same expiration time after having processed the
received distance vectors. Sally Floyd and Van Jacobson proposed in [FJ1994] a simple solution to solve this
synchronisation problem. Instead of advertising their distance vector exactly after 30 seconds, a router should
send its next distance vector after a delay chosen randomly in the [15,45] interval RFC 2080. This randomisation
of the delays prevents the synchronisation that occurs with a fixed delay and is now a recommended practice for
protocol designers.

OSPF
Link-state routing protocols are used in IP networks. Open Shortest Path First (OSPF), defined in RFC 2328, is the
link state routing protocol that has been standardised by the IETF. The last version of OSPF, which supports IPv6,
is defined in RFC 5340. OSPF is frequently used in enterprise networks and in some ISP networks. However,
ISP networks often use the IS-IS link-state routing protocol [ISO10589] , which was developed for the ISO CLNP
protocol but was adapted to be used in IP RFC 1195 networks before the finalisation of the standardisation
of OSPF. A detailed analysis of ISIS and OSPF may be found in [BMO2006] and [Perlman2000]. Additional
information about OSPF may be found in [Moy1998].
Compared to the basics of link-state routing protocols that we discussed in section Link state routing, there are
some particularities of OSPF that are worth discussing. First, in a large network, flooding the information about
all routers and links to thousands of routers or more may be costly as each router needs to store all the information
about the entire network. A better approach would be to introduce hierarchical routing. Hierarchical routing
divides the network into regions. All the routers inside a region have detailed information about the topology of
the region but only learn aggregated information about the topology of the other regions and their interconnections.
OSPF supports a restricted variant of hierarchical routing. In OSPFs terminology, a region is called an area.
OSPF imposes restrictions on how a network can be divided into areas. An area is a set of routers and links that
are grouped together. Usually, the topology of an area is chosen so that a packet sent by one router inside the area
can reach any other router in the area without leaving the area 23 . An OSPF area contains two types of routers
RFC 2328:
Internal router : A router whose directly connected networks belong to the area
Area border routers : A router that is attached to several areas.
For example, the network shown in the figure below has been divided into three areas : area 1, containing routers
R1, R3, R4, R5 and RA, area 2 containing R7, R8, R9, R10, RB and RC. OSPF areas are identified by a 32 bit
integer, which is sometimes represented as an IP address. Among the OSPF areas, area 0, also called the backbone
area has a special role. The backbone area groups all the area border routers (routers RA, RB and RC in the figure
below) and the routers that are directly connected to the backbone routers but do not belong to another area (router
23 OSPF can support virtual links to connect routers together that belong to the same area but are not directly connected. However, this goes
beyond this introduction to OSPF.

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RD in the figure below). An important restriction imposed by OSPF is that the path between two routers that
belong to two different areas (e.g. R1 and R8 in the figure below) must pass through the backbone area.

Figure 5.50: OSPF areas

Inside each non-backbone area, routers distribute the topology of the area by exchanging link state packets with
the other routers in the area. The internal routers do not know the topology of other areas, but each router knows
how to reach the backbone area. Inside an area, the routers only exchange link-state packets for all destinations
that are reachable inside the area. In OSPF, the inter-area routing is done by exchanging distance vectors. This is
illustrated by the network topology shown below.
Let us first consider OSPF routing inside area 2. All routers in the area learn a route towards 192.168.1.0/24 and
192.168.10.0/24. The two area border routers, RB and RC, create network summary advertisements. Assuming
that all links have a unit link metric, these would be:
RB advertises 192.168.1.0/24 at a distance of 2 and 192.168.10.0/24 at a distance of 3
RC advertises 192.168.1.0/24 at a distance of 3 and 192.168.10.0/24 at a distance of 2
These summary advertisements are flooded through the backbone area attached to routers RB and RC. In its routing
table, router RA selects the summary advertised by RB to reach 192.168.1.0/24 and the summary advertised by
RC to reach 192.168.10.0/24. Inside area 1, router RA advertises a summary indicating that 192.168.1.0/24 and
192.168.10.0/24 are both at a distance of 3 from itself.
On the other hand, consider the prefixes 10.0.0.0/24 and 10.0.1.0/24 that are inside area 1. Router RA is the only
area border router that is attached to this area. This router can create two different network summary advertisements :
10.0.0.0/24 at a distance of 1 and 10.0.1.0/24 at a distance of 2 from RA
10.0.0.0/23 at a distance of 2 from RA
The first summary advertisement provides precise information about the distance used to reach each prefix. However, all routers in the network have to maintain a route towards 10.0.0.0/24 and a route towards 10.0.1.0/24 that
are both via router RA. The second advertisement would improve the scalability of OSPF by reducing the number
of routes that are advertised across area boundaries. However, in practice this requires manual configuration on
the border routers.
The second OSPF particularity that is worth discussing is the support of Local Area Networks (LAN). As shown
in the example below, several routers may be attached to the same LAN.
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Figure 5.51: Hierarchical routing with OSPF

Figure 5.52: An OSPF LAN containing several routers

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A first solution to support such a LAN with a link-state routing protocol would be to consider that a LAN is
equivalent to a full-mesh of point-to-point links as if each router can directly reach any other router on the LAN.
However, this approach has two important drawbacks :
1. Each router must exchange HELLOs and link state packets with all the other routers on the LAN. This
increases the number of OSPF packets that are sent and processed by each router.
2. Remote routers, when looking at the topology distributed by OSPF, consider that there is a full-mesh of
links between all the LAN routers. Such a full-mesh implies a lot of redundancy in case of failure, while in
practice the entire LAN may completely fail. In case of a failure of the entire LAN, all routers need to detect
the failures and flood link state packets before the LAN is completely removed from the OSPF topology by
remote routers.
To better represent LANs and reduce the number of OSPF packets that are exchanged, OSPF handles LAN differently. When OSPF routers boot on a LAN, they elect 24 one of them as the Designated Router (DR) RFC 2328.
The DR router represents the local area network, and advertises the LANs subnet (138.48.4.0/24 in the example
above). Furthermore, LAN routers only exchange HELLO packets with the DR. Thanks to the utilisation of a DR,
the topology of the LAN appears as a set of point-to-point links connected to the DR as shown in the figure below.

Figure 5.53: OSPF representation of a LAN

Note: How to quickly detect a link failure ?

Network operators expect an OSPF network to be able to quickly recover from link or router failures [VPD2004].
In an OSPF network, the recovery after a failure is performed in three steps [FFEB2005] :
the routers that are adjacent to the failure detect it quickly. The default solution is to rely on the regular
exchange of HELLO packets. However, the interval between successive HELLOs is often set to 10 seconds... Setting the HELLO timer down to a few milliseconds is difficult as HELLO packets are created and
processed by the main CPU of the routers and these routers cannot easily generate and process a HELLO
packet every millisecond on each of their interfaces. A better solution is to use a dedicated failure detection
protocol such as the Bidirectional Forwarding Detection (BFD) protocol defined in [KW2009] that can be
implemented directly on the router interfaces. Another solution to be able to detect the failure is to instrument the physical and the datalink layer so that they can interrupt the router when a link fails. Unfortunately,
such a solution cannot be used on all types of physical and datalink layers.
the routers that have detected the failure flood their updated link state packets in the network
all routers update their routing table

5.3.2 Interdomain routing

As explained earlier, the Internet is composed of more than 30,000 different networks 25 called domains. Each
domain is composed of a group of routers and hosts that are managed by the same organisation. Example domains
include belnet, sprint, level3, geant, abilene, cisco or google ...
24 The OSPF Designated Router election procedure is defined in RFC 2328. Each router can be configured with a router priority that
influences the election process since the router with the highest priority is preferred when an election is run.
25 An analysis of the evolution of the number of domains on the global Internet during the last ten years may be found in
http://www.potaroo.net/tools/asn32/

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Each domain contains a set of routers. From a routing point of view, these domains can be divided into two
classes : the transit and the stub domains. A stub domain sends and receives packets whose source or destination
are one of its own hosts. A transit domain is a domain that provides a transit service for other domains, i.e. the
routers in this domain forward packets whose source and destination do not belong to the transit domain. As of
this writing, about 85% of the domains in the Internet are stub domains 26 . A stub domain that is connected to a
single transit domain is called a single-homed stub. A multihomed stub is a stub domain connected to two or more
transit providers.

Figure 5.54: Transit and stub domains

The stub domains can be further classified by considering whether they mainly send or receive packets. An accessrich stub domain is a domain that contains hosts that mainly receive packets. Typical examples include small
ADSL- or cable modem-based Internet Service Providers or enterprise networks. On the other hand, a contentrich stub domain is a domain that mainly produces packets. Examples of content-rich stub domains include google,
yahoo, microsoft, facebook or content distribution networks such as akamai or limelight For the last few years, we
have seen a rapid growth of these content-rich stub domains. Recent measurements [ATLAS2009] indicate that a
growing fraction of all the packets exchanged on the Internet are produced in the data centers managed by these
content providers.
Domains need to be interconnected to allow a host inside a domain to exchange IP packets with hosts located
in other domains. From a physical perspective, domains can be interconnected in two different ways. The first
solution is to directly connect a router belonging to the first domain with a router inside the second domain. Such
links between domains are called private interdomain links or private peering links. In practice, for redundancy or
performance reasons, distinct physical links are usually established between different routers in the two domains
that are interconnected.

Figure 5.55: Interconnection of two domains via a private peering link

Such private peering links are useful when, for example, an enterprise or university network needs to be connected
to its Internet Service Provider. However, some domains are connected to hundreds of other domains 27 . For some
of these domains, using only private peering links would be too costly. A better solution to allow many domains
to interconnect cheaply are the Internet eXchange Points (IXP). An IXP is usually some space in a data center that
hosts routers belonging to different domains. A domain willing to exchange packets with other domains present
at the IXP installs one of its routers on the IXP and connects it to other routers inside its own network. The IXP
contains a Local Area Network to which all the participating routers are connected. When two domains that are
present at the IXP wish 28 to exchange packets, they simply use the Local Area Network. IXPs are very popular
26 Several web sites collect and analyse data about the evolution of BGP in the global Internet. http://bgp.potaroo.net provides lots of
statistics and analyses that are updated daily.
27 See http://as-rank.caida.org/ for an analysis of the interconnections between domains based on measurements collected in the global
Internet
28 Two routers that are attached to the same IXP only exchange packets when the owners of their domains have an economical incentive to
exchange packets on this IXP. Usually, a router on an IXP is only able to exchange packets with a small fraction of the routers that are present
on the same IXP.

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in Europe and many Internet Service Providers and Content providers are present in these IXPs.

Figure 5.56: Interconnection of two domains at an Internet eXchange Point

In the early days of the Internet, domains would simply exchange all the routes they know to allow a host inside
one domain to reach any host in the global Internet. However, in todays highly commercial Internet, this is no
longer true as interdomain routing mainly needs to take into account the economical relationships between the
domains. Furthermore, while intradomain routing usually prefers some routes over others based on their technical
merits (e.g. prefer route with the minimum number of hops, prefer route with the minimum delay, prefer high
bandwidth routes over low bandwidth ones, etc) interdomain routing mainly deals with economical issues. For
interdomain routing, the cost of using a route is often more important than the quality of the route measured by its
delay or bandwidth.
There are different types of economical relationships that can exist between domains. Interdomain routing converts
these relationships into peering relationships between domains that are connected via peering links.
The first category of peering relationship is the customer->provider relationship. Such a relationship is used when
a customer domain pays an Internet Service Provider to be able to exchange packets with the global Internet over
an interdomain link. A similar relationship is used when a small Internet Service Provider pays a larger Internet
Service Provider to exchange packets with the global Internet.

Figure 5.57: A simple Internet with peering relationships

To understand the customer->provider relationship, let us consider the simple internetwork shown in the figure
above. In this internetwork, AS7 is a stub domain that is connected to one provider : AS4. The contract between
AS4 and AS7 allows a host inside AS7 to exchange packets with any host in the internetwork. To enable this
exchange of packets, AS7 must know a route towards any domain and all the domains of the internetwork must
know a route via AS4 that allows them to reach hosts inside AS7. From a routing perspective, the commercial
contract between AS7 and AS4 leads to the following routes being exchanged :
over a customer->provider relationship, the customer domain advertises to its provider all its routes and all
the routes that it has learned from its own customers.
over a provider->customer relationship, the provider advertises all the routes that it knows to its customer.
The second rule ensures that the customer domain receives a route towards all destinations that are reachable via
its provider. The first rule allows the routes of the customer domain to be distributed throughout the Internet.

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Coming back to the figure above, AS4 advertises to its two providers AS1 and AS2 its own routes and the routes
learned from its customer, AS7. On the other hand, AS4 advertises to AS7 all the routes that it knows.
The second type of peering relationship is the shared-cost peering relationship. Such a relationship usually does
not involve a payment from one domain to the other in contrast with the customer->provider relationship. A
shared-cost peering relationship is usually established between domains having a similar size and geographic
coverage. For example, consider the figure above. If AS3 and AS4 exchange many packets via AS1, they both need
to pay AS1. A cheaper alternative for AS3 and AS4 would be to establish a shared-cost peering. Such a peering
can be established at IXPs where both AS3 and AS4 are present or by using private peering links. This shared-cost
peering should be used to exchange packets between hosts inside AS3 and hosts inside AS4. However, AS3 does
not want to receive on the AS3-AS4 shared-cost peering links packets whose destination belongs to AS1 as AS3
would have to pay to send these packets to AS1.
From a routing perspective, over a shared-cost peering relationship a domain only advertises its internal routes
and the routes that it has learned from its customers. This restriction ensures that only packets destined to the
local domain or one of its customers is received over the shared-cost peering relationship. This implies that the
routes that have been learned from a provider or from another shared-cost peer is not advertised over a shared-cost
peering relationship. This is motivated by economical reasons. If a domain were to advertise the routes that it
learned from a provider over a shared-cost peering relationship that does not bring revenue, it would have allowed
its shared-cost peer to use the link with its provider without any payment. If a domain were to advertise the routes
it learned over a shared cost peering over another shared-cost peering relationship, it would have allowed these
shared-cost peers to use its own network (which may span one or more continents) freely to exchange packets.
Finally, the last type of peering relationship is the sibling. Such a relationship is used when two domains exchange
all their routes in both directions. In practice, such a relationship is only used between domains that belong to the
same company.
These different types of relationships are implemented in the interdomain routing policies defined by each domain.
The interdomain routing policy of a domain is composed of three main parts :
the import filter that specifies, for each peering relationship, the routes that can be accepted from the neighbouring domain (the non-acceptable routes are ignored and the domain never uses them to forward packets)
the export filter that specifies, for each peering relationship, the routes that can be advertised to the neighbouring domain
the ranking algorithm that is used to select the best route among all the routes that the domain has received
towards the same destination prefix
A domains import and export filters can be defined by using the Route Policy Specification Language (RPSL)
specified in RFC 2622 [GAVE1999] . Some Internet Service Providers, notably in Europe, use RPSL to document
29
their import and export policies. Several tools help to easily convert a RPSL policy into router commands.
The figure below provides a simple example of import and export filters for two domains in a simple internetwork.
In RPSL, the keyword ANY is used to replace any route from any domain. It is typically used by a provider to
indicate that it announces all its routes to a customer over a provider->customer relationship. This is the case
for AS4s export policy. The example below clearly shows the difference between a provider->customer and a
shared-cost peering relationship. AS4s export filter indicates that it announces only its internal routes (AS4) and
the routes learned from its clients (AS7) over its shared-cost peering with AS3, while it advertises all the routes
that it uses (including the routes learned from AS3) to AS7.
The Border Gateway Protocol
The Internet uses a single interdomain routing protocol : the Border Gateway Protocol (BGP). The current version of BGP is defined in RFC 4271. BGP differs from the intradomain routing protocols that we have already
discussed in several ways. First, BGP is a path-vector protocol. When a BGP router advertises a route towards a
prefix, it announces the IP prefix and the interdomain path used to reach this prefix. From BGPs point of view,
each domain is identified by a unique Autonomous System (AS) number 30 and the interdomain path contains the
AS numbers of the transit domains that are used to reach the associated prefix. This interdomain path is called the
29

See ftp://ftp.ripe.net/ripe/dbase for the RIPE database that contains the import and export policies of many European ISPs
In this text, we consider Autonomous System and domain as synonyms. In practice, a domain may be divided into several Autonomous
Systems, but we ignore this detail.
30

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Figure 5.58: Import and export policies

AS Path. Thanks to these AS-Paths, BGP does not suffer from the count-to-infinity problems that affect distance
vector routing protocols. Furthermore, the AS-Path can be used to implement some routing policies. Another difference between BGP and the intradomain routing protocols is that a BGP router does not send the entire contents
of its routing table to its neighbours regularly. Given the size of the global Internet, routers would be overloaded
by the number of BGP messages that they would need to process. BGP uses incremental updates, i.e. it only
announces the routes that have changed to its neighbours.
The figure below shows a simple example of the BGP routes that are exchanged between domains. In this example,
prefix 1.0.0.0/8 is announced by AS1. AS1 advertises a BGP route towards this prefix to AS2. The AS-Path of this
route indicates that AS1 is the originator of the prefix. When AS4 receives the BGP route from AS1, it re-announces
it to AS2 and adds its AS number to the AS-Path. AS2 has learned two routes towards prefix 1.0.0.0/8. It compares
the two routes and prefers the route learned from AS4 based on its own ranking algorithm. AS2 advertises to AS5
a route towards 1.0.0.0/8 with its AS-Path set to AS2:AS4:AS1. Thanks to the AS-Path, AS5 knows that if it sends
a packet towards 1.0.0.0/8 the packet first passes through AS2, then through AS4 before reaching its destination
inside AS1.

Figure 5.59: Simple exchange of BGP routes

BGP routers exchange routes over BGP sessions. A BGP session is established between two routers belonging to
two different domains that are directly connected. As explained earlier, the physical connection between the two
routers can be implemented as a private peering link or over an Internet eXchange Point. A BGP session between
two adjacent routers runs above a TCP connection (the default BGP port is 179). In contrast with intradomain
routing protocols that exchange IP packets or UDP segments, BGP runs above TCP because TCP ensures a reliable
delivery of the BGP messages sent by each router without forcing the routers to implement acknowledgements,
checksums, etc. Furthermore, the two routers consider the peering link to be up as long as the BGP session and
the underlying TCP connection remain up 31 . The two endpoints of a BGP session are called BGP peers.
31 The BGP sessions and the underlying TCP connection are typically established by the routers when they boot based on information found
in their configuration. The BGP sessions are rarely released, except if the corresponding peering link fails or one of the endpoints crashes or

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AS3
R1

BGP
session
R2

AS4

Figure 5.60: A BGP peering session between two directly connected routers
In practice, to establish a BGP session between routers R1 and R2 in the figure above, the network administrator
of AS3 must first configure on R1 the IP address of R2 on the R1-R2 link and the AS number of R2. Router R1 then
regularly tries to establish the BGP session with R2. R2 only agrees to establish the BGP session with R1 once it
has been configured with the IP address of R1 and its AS number. For security reasons, a router never establishes
a BGP session that has not been manually configured on the router.
The BGP protocol RFC 4271 defines several types of messages that can be exchanged over a BGP session :
OPEN : this message is sent as soon as the TCP connection between the two routers has been established.
It initialises the BGP session and allows the negotiation of some options. Details about this message may
be found in RFC 4271
NOTIFICATION : this message is used to terminate a BGP session, usually because an error has been detected by the BGP peer. A router that sends or receives a NOTIFICATION message immediately shutdowns
the corresponding BGP session.
UPDATE: this message is used to advertise new or modified routes or to withdraw previously advertised
routes.
KEEPALIVE : this message is used to ensure a regular exchange of messages on the BGP session, even
when no route changes. When a BGP router has not sent an UPDATE message during the last 30 seconds,
it shall send a KEEPALIVE message to confirm to the other peer that it is still up. If a peer does not receive
any BGP message during a period of 90 seconds 32 , the BGP session is considered to be down and all the
routes learned over this session are withdrawn.
As explained earlier, BGP relies on incremental updates. This implies that when a BGP session starts, each router
first sends BGP UPDATE messages to advertise to the other peer all the exportable routes that it knows. Once
all these routes have been advertised, the BGP router only sends BGP UPDATE messages about a prefix if the
route is new, one of its attributes has changed or the route became unreachable and must be withdrawn. The BGP
UPDATE message allows BGP routers to efficiently exchange such information while minimising the number of
bytes exchanged. Each UPDATE message contains :
a list of IP prefixes that are withdrawn
a list of IP prefixes that are (re-)advertised
the set of attributes (e.g. AS-Path) associated to the advertised prefixes
In the remainder of this chapter, and although all routing information is exchanged using BGP UPDATE messages,
we assume for simplicity that a BGP message contains only information about one prefix and we use the words :
Withdraw message to indicate a BGP UPDATE message containing one route that is withdrawn
Update message to indicate a BGP UPDATE containing a new or updated route towards one destination
prefix with its attributes
needs to be rebooted.
32 90 seconds is the default delay recommended by RFC 4271. However, two BGP peers can negotiate a different timer during the
establishment of their BGP session. Using a too small interval to detect BGP session failures is not recommended. BFD [KW2009] can be
used to replace BGPs KEEPALIVE mechanism if fast detection of interdomain link failures is required.

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From a conceptual point of view, a BGP router connected to N BGP peers, can be described as being composed
of four parts as shown in the figure below.

Figure 5.61: Organisation of a BGP router

In this figure, the router receives BGP messages on the left part of the figure, processes these messages and possibly
sends BGP messages on the right part of the figure. A BGP router contains three important data structures :
the Adj-RIB-In contains the BGP routes that have been received from each BGP peer. The routes in the
Adj-RIB-In are filtered by the import filter before being placed in the BGP-Loc-RIB. There is one import
filter per BGP peer.
the Local Routing Information Base (Loc-RIB) contains all the routes that are considered as acceptable by
the router. The Loc-RIB may contain several routes, learned from different BGP peers, towards the same
destination prefix.
the Forwarding Information Base (FIB) is used by the dataplane to forward packets towards their destination.
The FIB contains, for each destination, the best route that has been selected by the BGP decision process.
This decision process is an algorithm that selects, for each destination prefix, the best route according to the
routers ranking algorithm that is part of its policy.
the Adj-RIB-Out contains the BGP routes that have been advertised to each BGP peer. The Adj-RIB-Out for
a given peer is built by applying the peers export filter on the routes that have been installed in the FIB.
There is one export filter per BGP peer. For this reason, the Adj-RIB-Out of a peer may contain different
routes than the Adj-RIB-Out of another peer.
When a BGP session starts, the routers first exchange OPEN messages to negotiate the options that apply throughout the entire session. Then, each router extracts from its FIB the routes to be advertised to the peer. It is important
to note that, for each known destination prefix, a BGP router can only advertise to a peer the route that it has itself
installed inside its FIB. The routes that are advertised to a peer must pass the peers export filter. The export filter
is a set of rules that define which routes can be advertised over the corresponding session, possibly after having
modified some of its attributes. One export filter is associated to each BGP session. For example, on a shared-cost
peering, the export filter only selects the internal routes and the routes that have been learned from a customer.
The pseudo-code below shows the initialisation of a BGP session.
def initiliaze_BGP_session( RemoteAS, RemoteIP):
# Initialize and start BGP session
# Send BGP OPEN Message to RemoteIP on port 179
# Follow BGP state machine
# advertise local routes and routes learned from peers*/
for d in BGPLocRIB :
B=build_BGP_Update(d)
S=Apply_Export_Filter(RemoteAS,B)
if (S != None) :
send_Update(S,RemoteAS,RemoteIP)
# entire RIB has been sent
# new Updates will be sent to reflect local or distant
# changes in routers

In the above pseudo-code, the build_BGP_UPDATE(d) procedure extracts from the BGP Loc-RIB the best path
towards destination d (i.e. the route installed in the FIB) and prepares the corresponding BGP UPDATE message.

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This message is then passed to the export filter that returns NULL if the route cannot be advertised to the peer or
the (possibly modified) BGP UPDATE message to be advertised. BGP routers allow network administrators to
specify very complex export filters, see e.g. [WMS2004]. A simple export filter that implements the equivalent of
split horizon is shown below.
def apply_export_filter(RemoteAS, BGPMsg) :
# check if RemoteAS already received route
if RemoteAS is BGPMsg.ASPath :
BGPMsg=None
# Many additional export policies can be configured :
# Accept or refuse the BGPMsg
# Modify selected attributes inside BGPMsg
return BGPMsg

At this point, the remote router has received all the exportable BGP routes. After this initial exchange, the router
only sends BGP UPDATE messages when there is a change (addition of a route, removal of a route or change in
the attributes of a route) in one of these exportable routes. Such a change can happen when the router receives a
BGP message. The pseudo-code below summarizes the processing of these BGP messages.
def Recvd_BGPMsg(Msg, RemoteAS) :
B=apply_import_filer(Msg,RemoteAS)
if (B== None): # Msg not acceptable
return
if IsUPDATE(Msg):
Old_Route=BestRoute(Msg.prefix)
Insert_in_RIB(Msg)
Run_Decision_Process(RIB)
if (BestRoute(Msg.prefix) != Old_Route) :
# best route changed
B=build_BGP_Message(Msg.prefix);
S=apply_export_filter(RemoteAS,B);
if (S!=None) : # announce best route
send_UPDATE(S,RemoteAS,RemoteIP);
else if (Old_Route != None) :
send_WITHDRAW(Msg.prefix,RemoteAS, RemoteIP)
else : # Msg is WITHDRAW
Old_Route=BestRoute(Msg.prefix)
Remove_from_RIB(Msg)
Run_Decision_Process(RIB)
if (Best_Route(Msg.prefix) !=Old_Route):
# best route changed
B=build_BGP_Message(Msg.prefix)
S=apply_export_filter(RemoteAS,B)
if (S != None) : # still one best route towards Msg.prefix
send_UPDATE(S,RemoteAS, RemoteIP);
else if(Old_Route != None) : # No best route anymore
send_WITHDRAW(Msg.prefix,RemoteAS,RemoteIP);

When a BGP message is received, the router first applies the peers import filter to verify whether the message is
acceptable or not. If the message is not acceptable, the processing stops. The pseudo-code below shows a simple
import filter. This import filter accepts all routes, except those that already contain the local AS in their AS-Path.
If such a route was used, it would cause a routing loop. Another example of an import filter would be a filter used
by an Internet Service Provider on a session with a customer to only accept routes towards the IP prefixes assigned
to the customer by the provider. On real routers, import filters can be much more complex and some import filters
modify the attributes of the received BGP UPDATE [WMS2004] .
def apply_import_filter(RemoteAS, BGPMsg):
if MysAS in BGPMsg.ASPath :
BGPMsg=None
# Many additional import policies can be configured :
# Accept or refuse the BGPMsg
# Modify selected attributes inside BGPMsg
return BGPMsg

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Note: The bogon filters

Another example of frequently used import filters are the filters that Internet Service Providers use to ignore
bogon routes. In the ISP community, a bogon route is a route that should not be advertised on the global Internet.
Typical examples include the private IPv4 prefixes defined in RFC 1918, the loopback prefixes (127.0.0.1/8 and
::1/128) or the IP prefixes that have not yet been allocated by IANA. A well managed BGP router should ensure
that it never advertises bogons on the global Internet. Detailed information about these bogons may be found at
http://www.team-cymru.org/Services/Bogons/
If the import filter accepts the BGP message, the pseudo-code distinguishes two cases. If this is an Update
message for prefix p, this can be a new route for this prefix or a modification of the routes attributes. The router
first retrieves from its RIB the best route towards prefix p. Then, the new route is inserted in the RIB and the BGP
decision process is run to find whether the best route towards destination p changes. A BGP message only needs to
be sent to the routers peers if the best route has changed. For each peer, the router applies the export filter to verify
whether the route can be advertised. If yes, the filtered BGP message is sent. Otherwise, a Withdraw message
is sent. When the router receives a Withdraw message, it also verifies whether the removal of the route from its
RIB caused its best route towards this prefix to change. It should be noted that, depending on the content of the
RIB and the export filters, a BGP router may need to send a Withdraw message to a peer after having received an
Update message from another peer and conversely.
Let us now discuss in more detail the operation of BGP in an IPv4 network. For this, let us consider the simple
network composed of three routers located in three different ASes and shown in the figure below.
AS10

AS30

AS20
195.100.0.4/30

195.100.0.0/30

195.100.0.1

194.100.0.0/24

BGP

195.100.0.2

195.100.0.5

195.100.0.6

BGP

194.100.1.0/24

194.100.2.0/23

Figure 5.62: Utilisation of the BGP nexthop attribute

This network contains three routers : R1, R2 and R3. Each router is attached to a local IPv4 subnet that it advertises
using BGP. There are two BGP sessions, one between R1 and R2 and the second between R2 and R3. A /30 subnet
is used on each interdomain link (195.100.0.0/30 on R1-R2 and 195.100.0.4/30 on R2-R3). The BGP sessions
run above TCP connections established between the neighbouring routers (e.g. 195.100.0.1 - 195.100.0.2 for the
R1-R2 session).
Let us assume that the R1-R2 BGP session is the first to be established. A BGP Update message sent on such a
session contains three fields :
the advertised prefix
the BGP nexthop
the attributes including the AS-Path
We use the notation U(prefix, nexthop, attributes) to represent such a BGP Update message in this section. Similarly, W(prefix) represents a BGP withdraw for the specified prefix. Once the R1-R2 session has been established,
R1 sends U(194.100.0.0/24,195.100.0.1,AS10) to R2 and R2 sends U(194.100.2.0/23,195.100.0.2,AS20). At this
point, R1 can reach 194.100.2.0/23 via 195.100.0.2 and R2 can reach 194.100.0.0/24 via 195.100.0.1.
Once the R2-R3 has been established, R3 sends U(194.100.1.0/24,195.100.0.6,AS30). R2 announces on the R2R3 session all the routes inside its RIB. It thus sends to R3 : U(194.100.0.0/24,195.100.0.5,AS20:AS10) and
U(194.100.2.0/23,195.100.0.5,AS20). Note that when R2 advertises the route that it learned from R1, it updates the
BGP nexthop and adds its AS number to the AS-Path. R2 also sends U(194.100.1.0/24,195.100.0.2,AS20:AS30)
to R1 on the R1-R3 session. At this point, all BGP routes have been exchanged and all routers can reach
194.100.0.0/24, 194.100.2.0/23 and 194.100.1.0/24.

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If the link between R2 and R3 fails, R3 detects the failure as it did not receive KEEPALIVE messages recently from
R2. At this time, R3 removes from its RIB all the routes learned over the R2-R3 BGP session. R2 also removes
from its RIB the routes learned from R3. R2 also sends W(194.100.1.0/24) to R1 over the R1-R3 BGP session
since it does not have a route anymore towards this prefix.
Note: Origin of the routes advertised by a BGP router
A frequent practical question about the operation of BGP is how a BGP router decides to originate or advertise a
route for the first time. In practice, this occurs in two situations :
the router has been manually configured by the network operator to always advertise one or several routes
on a BGP session. For example, on the BGP session between UCLouvain and its provider, belnet , UCLouvains router always advertises the 130.104.0.0/16 IPv4 prefix assigned to the campus network
the router has been configured by the network operator to advertise over its BGP session some of the routes
that it learns with its intradomain routing protocol. For example, an enterprise router may advertise over a
BGP session with its provider the routes to remote sites when these routes are reachable and advertised by
the intradomain routing protocol
The first solution is the most frequent. Advertising routes learned from an intradomain routing protocol is not
recommended, this is because if the route flaps 33 , this would cause a large number of BGP messages being
exchanged in the global Internet.
Most networks that use BGP contain more than one router. For example, consider the network shown in the figure
below where AS20 contains two routers attached to interdomain links : R2 and R4. In this network, two routing
protocols are used by R2 and R4. They use an intradomain routing protocol such as OSPF to distribute the routes
towards the internal prefixes : 195.100.0.8/30, 195.100.0.0/30, ... R2 and R4 also use BGP. R2 receives the routes
advertised by AS10 while R4 receives the routes advertised by AS30. These two routers need to exchange the
routes that they have respectively received over their BGP sessions.

194.100.2.0/23

AS30

AS10

195.100.0.2
R2

195.100.0.0/30
R1

194.100.0.0/23

195.100.0.1

195.100.0.10

BGP
195.100.0.8/30

AS20

195.100.0.6

BGP
195.100.0.4/30

195.100.0.9

194.100.4.0/23

195.100.0.5

Figure 5.63: A larger network using BGP

A first solution to allow R2 and R3 to exchange the interdomain routes that they have learned over their respective
BGP sessions would be to configure the intradomain routing protocol to distribute inside AS20 the routes learned
over the BGP sessions. Although current routers support this feature, this is a bad solution for two reasons :
1. Intradomain routing protocols cannot distribute the attributes that are attached to a BGP route. If R4 received
via the intradomain routing protocol a route towards 194.100.0.0/23 that R2 learned via BGP, it would not
know that the route was originated by AS10 and the only advertisement that it could send to R3 would
contain an incorrect AS-Path
2. Intradomain routing protocols have not been designed to support the hundreds of thousands of routes that a
BGP router can receive on todays global Internet.
33 A link is said to be flapping if it switches several between an operational state and a disabled state within a short period of time. A router
attached to such a link would need to frequently send routing messages.

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The best solution to allow BGP routers to distribute, inside an AS, all the routes learned over BGP sessions is to
establish BGP sessions among all the BGP routers inside the AS. In practice, there are two types of BGP sessions
:
eBGP session or external BGP session. Such a BGP session is established between two routers that are
directly connected and belong to two different domains.
iBGP session or internal BGP session. Such a BGP session is established between two routers belonging to
the same domain. These two routers do not need to be directly connected.
In practice, each BGP router inside a domain maintains an iBGP session with every other BGP router in the
domain 34 . This creates a full-mesh of iBGP sessions among all BGP routers of the domain. iBGP sessions, like
eBGP sessions run over TCP connections. Note that in contrast with eBGP sessions that are established between
directly connected routers, iBGP sessions are often established between routers that are not directly connected.
An important point to note about iBGP sessions is that a BGP router only advertises a route over an iBGP session
provided that :
the router uses this route to forward packets, and
the route was learned over one of the routers eBGP sessions
A BGP router does not advertise a route that it has learned over an iBGP session over another iBGP session. Note
that a router can, of course, advertise over an eBGP session a route that it has learned over an iBGP session.
This difference between the behaviour of a BGP router over iBGP and eBGP session is due to the utilisation of a
full-mesh of iBGP sessions. Consider a network containing three BGP routers : A, B and C interconnected via a
full-mesh of iBGP sessions. If router A learns a route towards prefix p from router B, router A does not need to
advertise the received route to router C since router C also learns the same route over the C-B iBGP session.
To understand the utilisation of an iBGP session, let us consider what happens when router R1 sends
U(194.100.0.0/23,195.100.0.1,AS10) in the network shown below. This BGP message is processed by R2 which
advertises it over its iBGP session with R4. The BGP Update sent by R2 contains the same nexthop and the same
AS-Path as in the BGP Update received by R2. R4 then sends U(194.100.0.0/23,195.100.0.5,AS20:AS10) to R3.
Note that the BGP nexthop and the AS-Path are only updated 35 when a BGP route is advertised over an eBGP
session.

194.100.2.0/23

AS10
R1

194.100.0.0/23

195.100.0.2
195.100.0.0/30
195.100.0.1 eBGP

AS20

AS30
R2

195.100.0.10

195.100.0.8/30

iBGP

eBGP

195.100.0.4/30

195.100.0.9
194.100.4.0/23

195.100.0.6

195.100.0.5

Figure 5.64: iBGP and eBGP sessions

Note: Loopback interfaces and iBGP sessions

In addition to their physical interfaces, routers can also be configured with a special loopback interface. A loopback interface is a software interface that is always up. When a loopback interface is configured on a router,
the address associated to this interface is advertised by the intradomain routing protocol. Consider for example
34

Using a full-mesh of iBGP sessions is suitable in small networks. However, this solution does not scale in large networks containing
n(n1)
hundreds or more routers since
iBGP sessions must be established in a domain containing n BGP routers. Large domains use either
2
Route Reflection RFC 4456 or confederations RFC 5065 to scale their iBGP, but this goes beyond this introduction.
35 Some routers, when they receive a BGP Update over an eBGP session, set the nexthop of the received route to one of their own addresses.
This is called nexthop-self. See e.g. [WMS2004] for additional details.

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a router with two point-to-point interfaces and one loopback interface. When a point-to-point interface fails, it
becomes unreachable and the router cannot receive anymore packets via this IP address. This is not the case for
the loopback interface. It remains reachable as long as at least one of the routers interfaces remains up. iBGP
sessions are usually established using the routers loopback addresses as endpoints. This allows the iBGP session
and its underlying TCP connection to remain up even if physical interfaces fail on the routers.
Now that routers can learn interdomain routes over iBGP and eBGP sessions, let us examine what happens when
router R3 sends a packet destined to 194.100.1.234. R3 forwards this packet to R4. R4 uses an intradomain routing
protocol and BGP. Its BGP routing table contains the following longest prefix match :
194.100.0.0/23 via 195.100.0.1
This routes indicates that to forward a packet towards 194.100.0.0/23, R4 needs to forward the packet along the
route towards 195.100.0.1. However, R4 is not directly connected to 195.100.0.1. R4 learned a route that matches
this address thanks to its intradomain routing protocol that distributed the following routes :
195.100.0.0/30 via 195.100.0.10
195.100.0.4/30 East
195.100.0.8/30 North
194.100.2.0/23 via 195.100.0.10
194.100.0.4/23 West
To build its forwarding table, R4 must combine the routes learned from the intradomain routing protocol with the
routes learned from BGP. Thanks to its intradomain routing table, for each interdomain route R4 replaces the BGP
nexthop with its shortest path computed by the intradomain routing protocol. In the figure above, R4 forwards
packets to 194.100.0.0/23 via 195.100.0.10 to which it is directly connected via its North interface. R4 s resulting
forwarding table, which associates an outgoing interface for a directly connected prefix or a directly connected
nexthop and an outgoing interface for prefixes learned via BGP, is shown below :
194.100.0.0/23 via 195.100.0.10 (North)
195.100.0.0/30 via 195.100.0.10 (North)
195.100.0.4/30 East
195.100.0.8/30 North
194.100.2.0/23 via 195.100.0.10 (North)
194.100.4.0/23 West
There is thus a coupling between the interdomain and the intradomain routing tables. If the intradomain routes
change, e.g. due to link failures or changes in link metrics, then the forwarding table must be updated on each
router as the shortest path towards a BGP nexthop may have changed.
The last point to be discussed before looking at the BGP decision process is that a network may contain routers
that do not maintain any eBGP session. These routers can be stub routers attached to a single router in the network
or core routers that reside on the path between two border routers that are using BGP as illustrated in the figure
below.
In the scenario above, router R2 needs to be able to forward a packet towards any destination in the 12.0.0.0/8
prefix inside AS30. Such a packet would need to be forwarded by router R5 since this router resides on the path
between R2 and its BGP nexthop attached to R4. Two solutions can be used to ensure that R2 is able to forward
such interdomain packets :
enable BGP on router R5 and include this router in the iBGP full-mesh. Two iBGP sessions would be added
in the figure above : R2-R5 and R4-R5. This solution works and is used by many ASes. However, it forces
all routers to have enough resources (CPU and memory) to run BGP and maintain a large forwarding table
encapsulate the interdomain packets sent through the AS so that router R5 never needs to forward a packet
whose destination is outside the local AS. Different encapsulation mechanisms exist. MultiProtocol Label
Switching (MPLS) RFC 3031 and the Layer 2 Tunneling Protocol (L2TP) RFC 3931 are frequently used
in large domains, but a detailed explanation of these techniques is outside the scope of this section. The

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194.100.2.0/23

AS10

195.100.0.2
195.100.0.0/30
195.100.0.1 eBGP

194.100.0.0/23

AS20

AS30
R2

195.100.0.6

iBGP

195.100.0.4/30

195.100.0.9
194.100.4.0/23

eBGP

195.100.0.5

Figure 5.65: How to deal with non-BGP routers ?

simplest encapsulation scheme to understand is in IP in IP defined in RFC 2003. This encapsulation scheme
places an IP packet (called the inner packet), including its payload, as the payload of a larger IP packet
(called the outer packet). It can be used by border routers to forward packets via routers that do not maintain
a BGP routing table. For example, in the figure above, if router R2 needs to forward a packet towards
destination 12.0.0.1, it can add at the front of this packet an IPv4 header whose source address is set to one
of its IPv4 addresses and whose destination address is one of the IPv4 addresses of R4. The Protocol field
of the IP header is set to 4 to indicate that it contains an IPv4 packet. The packet is forwarded by R5 to
R4 based on the forwarding table that it built thanks to its intradomain routing table. Upon reception of the
packet, R4 removes the outer header and consults its (BGP) forwarding table to forward the packet towards
R3.
The BGP decision process

Besides the import and export filters, a key difference between BGP and the intradomain routing protocols is that
each domain can define is own ranking algorithm to determine which route is chosen to forward packets when
several routes have been learned towards the same prefix. This ranking depends on several BGP attributes that can
be attached to a BGP route.
The first BGP attribute that is used to rank BGP routes is the local-preference (local-pref) attribute. This attribute
is an unsigned integer that is attached to each BGP route received over an eBGP session by the associated import
filter.
When comparing routes towards the same destination prefix, a BGP router always prefers the routes with the
highest local-pref. If the BGP router knows several routes with the same local-pref, it prefers among the routes
having this local-pref the ones with the shortest AS-Path.
The local-pref attribute is often used to prefer some routes over others. This attribute is always present inside
BGP Updates exchanged over iBGP sessions, but never present in the messages exchanged over eBGP sessions.
A common utilisation of local-pref is to support backup links. Consider the situation depicted in the figure below.
AS1 would always like to use the high bandwidth link to send and receive packets via AS2 and only use the backup
link upon failure of the primary one.
As BGP routers always prefer the routes with the highest local-pref attribute, this policy can be implemented
using the following import filter on R1
import: from AS2 RA at R1 set localpref=100;
from AS2 RB at R1 set localpref=200;
accept ANY

With this import filter, all the BGP routes learned from RB over the high bandwidth links are preferred over the
routes learned over the backup link. If the primary link fails, the corresponding routes are removed from R1s RIB
and R1 uses the route learned from RA. R1 reuses the routes via RB as soon as they are advertised by RB once the
R1-RB link comes back.

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AS2

Backup: 2Mbps

Primary: 34Mbps

Figure 5.66: How to create a backup link with BGP ?

The import filter above modifies the selection of the BGP routes inside AS1. Thus, it influences the route followed
by the packets forwarded by AS1. In addition to using the primary link to send packets, AS1 would like to receive
its packets via the high bandwidth link. For this, AS2 also needs to set the local-pref attribute in its import filter.
import: from AS1 R1 at RA set localpref=100;
from AS1 R1 at RB set localpref=200;
accept AS1

Sometimes, the local-pref attribute is used to prefer a cheap link compared to a more expensive one. For example,
in the network below, AS1 could wish to send and receive packets mainly via its interdomain link with AS4.
AS3

RB
RA
R3

AS2
Expensive

AS1

Cheap
R1

AS5

AS4

Figure 5.67: How to prefer a cheap link over an more expensive one ?
AS1 can install the following import filter on R1 to ensure that it always sends packets via R2 when it has learned
a route via AS2 and another via AS4.
import: from AS2 RA at R1 set localpref=100;
from AS4 R2 at R1 set localpref=200;
accept ANY

However, this import filter does not influence how AS3 , for example, prefers some routes over others. If the link
between AS3 and AS2 is less expensive than the link between AS3 and AS4, AS3 could send all its packets via AS2
and AS1 would receive packets over its expensive link. An important point to remember about local-pref is that
it can be used to prefer some routes over others to send packets, but it has no influence on the routes followed by
received packets.
Another important utilisation of the local-pref attribute is to support the customer->provider and shared-cost peering relationships. From an economic point of view, there is an important difference between these three types of
peering relationships. A domain usually earns money when it sends packets over a provider->customer relationship. On the other hand, it must pay its provider when it sends packets over a customer->provider relationship.

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Using a shared-cost peering to send packets is usually neutral from an economic perspective. To take into account
these economic issues, domains usually configure the import filters on their routers as follows :
insert a high local-pref attribute in the routes learned from a customer
insert a medium local-pref attribute in the routes learned over a shared-cost peering
insert a low local-pref attribute in the routes learned from a provider
With such an import filter, the routers of a domain always prefer to reach destinations via their customers whenever
such a route exists. Otherwise, they prefer to use shared-cost peering relationships and they only send packets
via their providers when they do not know any alternate route. A consequence of setting the local-pref attribute
like this is that Internet paths are often asymmetrical. Consider for example the internetwork shown in the figure
below.

AS2

AS3

AS1

$
$

AS4

AS8

AS6
AS5

$
Shared-cost
$

Customer-provider

AS7

Figure 5.68: Asymmetry of Internet paths

Consider in this internetwork the routes available inside AS1 to reach AS5. AS1 learns the AS4:AS6:AS7:AS5 path
from AS4, the AS3:AS8:AS5 path from AS3 and the AS2:AS5 path from AS2. The first path is chosen since it was
from learned from a customer. AS5 on the other hand receives three paths towards AS1 via its providers. It may
select any of these paths to reach AS1 , depending on how it prefers one provider over the others.
Coming back to the organisation of a BGP router shown in figure Organisation of a BGP router, the last part to
be discussed is the BGP decision process. The BGP Decision Process is the algorithm used by routers to select
the route to be installed in the FIB when there are multiple routes towards the same prefix. The BGP decision
process receives a set of candidate routes towards the same prefix and uses seven steps. At each step, some routes
are removed from the candidate set and the process stops when the set only contains one route 36 :
1. Ignore routes having an unreachable BGP nexthop
2. Prefer routes having the highest local-pref
3. Prefer routes having the shortest AS-Path
4. Prefer routes having the smallest MED
5. Prefer routes learned via eBGP sessions over routes learned via iBGP sessions
6. Prefer routes having the closest next-hop
36

Some BGP implementations can be configured to install several routes towards a single prefix in their FIB for load-balancing purposes.
However, this goes beyond this introduction to BGP.

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7. Tie breaking rules : prefer routes learned from the router with lowest router id
The first step of the BGP decision process ensures that a BGP router does not install in its FIB a route whose
nexthop is considered to be unreachable by the intradomain routing protocol. This could happen, for example,
when a router has crashed. The intradomain routing protocol usually advertises the failure of this router before
the failure of the BGP sessions that it terminates. This rule implies that the BGP decision process must be re-run
each time the intradomain routing protocol reports a change in the reachability of a prefix containing one of more
BGP nexthops.
The second rule allows each domain to define its routing preferences. The local-pref attribute is set by the import
filter of the router that learned a route over an eBGP session.
In contrast with intradomain routing protocols, BGP does not contain an explicit metric. This is because in the
global Internet it is impossible for all domains to agree on a common metric that meets the requirements of all
domains. Despite this, BGP routers prefer routes having a short AS-Path attribute over routes with a long ASPath. This step of the BGP decision process is motivated by the fact that operators expect that a route with a long
AS-Path is lower quality than a route with a shorter AS-Path. However, studies have shown that there was not
always a strong correlation between the quality of a route and the length of its AS-Path [HFPMC2002].
Before explaining the fourth step of the BGP decision process, let us first describe the fifth and the sixth steps of
the BGP decision process. These two steps are used to implement hot potato routing. Intuitively, when a domain
implements hot potato routing, it tries to forward packets that are destined to addresses outside of its domain, to
other domains as quickly as possible.
To understand hot potato routing, let us consider the two domains shown in the figure below. AS2 advertises prefix
1.0.0.0/8 over the R2-R6 and R3-R7 peering links. The routers inside AS1 learn two routes towards 1.0.0.0/8: one
via R6-R2 and the second via R7-R3.

AS1
R8

C=50

C=1
R7

C=98

C=1
R2

1.0.0.0/8

AS2

Figure 5.69: Hot and cold potato routing

With the fifth step of the BGP decision process, a router always prefers to use a route learned over an eBGP session
compared to a route learned over an iBGP session. Thus, router R6 (resp. R7) prefers to use the route via router
R2 (resp. R3) to reach prefix 1.0.0.0/8.
The sixth step of the BGP decision process takes into account the distance, measured as the length of the shortest
intradomain path, between a BGP router and the BGP nexthop for routes learned over iBGP sessions. This rule is
used on router R8 in the example above. This router has received two routes towards 1.0.0.0/8:
1.0.0.0/8 via R7 that is at a distance of 1 from R8
1.0.0.0/8 via R6 that is at a distance of 50 from R8
The first route, via R7 is the one that router R8 prefers, as this is the route that minimises the cost of forwarding
packets inside AS1 before sending them to AS2.

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Hot potato routing allows AS1 to minimise the cost of forwarding packets towards AS2. However, there are
situations where this is not desirable. For example, assume that AS1 and AS2 are domains with routers on both
the East and the West coast of the US. In these two domains, the high metric associated to links R6-R8 and R0-R2
correspond to the cost of forwarding a packet across the USA. If AS2 is a customer that pays AS1, it would prefer
to receive the packets destined to 1.0.0.0/8 via the R2-R6 link instead of the R7-R3 link. This is the objective of
cold potato routing.
Cold potato routing is implemented using the Multi-Exit Discriminator (MED) attribute. This attribute is an
optional BGP attribute that may be set 37 by border routers when advertising a BGP route over an eBGP session.
The MED attribute is usually used to indicate over an eBGP session the cost to reach the BGP nexthop for the
advertised route. The MED attribute is set by the router that advertises a route over an eBGP session. In the
example above, router R2 sends U(1.0.0.0/8,R2,AS2,MED=1) while R3 sends U(1.0.0.0/8,R3,AS2,MED=98).
Assume that the BGP session R7-3 is the first to be established. R7 sends U(1.0.0.0/8,R3,AS2,MED=98) to both
R8 and R6. At this point, all routers inside AS1 send the packets towards 1.0.0.0/8 via R7-R3. Then, the R6R2 BGP session is established and router R6 receives U(1.0.0.0/8,R2,AS2,MED=1). Router R6 runs its decision
process for destination 1.0.0.0/8 and selects the route via R2 as its chosen route to reach this prefix since this is the
only route that it knows. R6 sends U(1.0.0.0/8,R2,AS2,MED=1) to routers R8 and R7. They both run their decision
process and prefer the route advertised by R6, as it contains the smallest MED. Now, all routers inside AS1 forward
the packets to 1.0.0.0/8 via link R6-R2 as expected by AS2. As router R7 no longer uses the BGP route learned
via R3, it must stop advertising it over iBGP sessions and sends W(1.0.0.0/8) over its iBGP sessions with R6 and
R8. However, router R7 still keeps the route learned from R3 inside its Adj-RIB-In. If the R6-R2 link fails, R6
sends W(1.0.0.0/8) over its iBGP sessions and router R7 responds by sending U(1.0.0.0/8,R3,AS2,MED=98) over
its iBGP sessions.
In practice, the fifth step of the BGP decision process is slightly more complex, as the routes towards a given prefix
can be learned from different ASes. For example, assume that in figure Hot and cold potato routing, 1.0.0.0/8 is
also advertised by AS3 (not shown in the figure) that has peering links with routers R6 and R8. If AS3 advertises a
route whose MED attribute is set to 2 and another with a MED set to 3, how should AS1s router compare the four
BGP routes towards 1.0.0.0/8 ? Is a MED value of 1 from AS2 better than a MED value of 2 from AS3 ? The fifth
step of the BGP decision process solves this problem by only comparing the MED attribute of the routes learned
from the same neighbour AS. Additional details about the MED attribute may be found in RFC 4451. It should
be noted that using the MED attribute may cause some problems in BGP networks as explained in [GW2002]. In
practice, the MED attribute is not used on eBGP sessions unless the two domains agree to enable it.
The last step of the BGP decision allows the selection of a single route when a BGP router has received several
routes that are considered as equal by the first six steps of the decision process. This can happen for example
in a dual-homed stub attached to two different providers. As shown in the figure below, router R1 receives two
equally good BGP routes towards 1.0.0.0/8. To break the ties, each router is identified by a unique router-id which
in practice is one of the IP addresses assigned to the router. On some routers, the lowest router id step in the
BGP decision process is replaced by the selection of the oldest route RFC 5004. Preferring the oldest route when
breaking ties is used to prefer stable paths over unstable paths. However, a drawback of this approach is that the
selection of the BGP routes depends on the arrival times of the corresponding messages. This makes the BGP
selection process non-deterministic and can lead to problems that are difficult to debug.
R0

AS1

1.0.0.0/8

AS2
R2

AS3

Figure 5.70: A stub connected to two providers

37 The MED attribute can be used on customer->provider peering relationships upon request of the customer. On shared-cost peering
relationship, the MED attribute is only enabled when there is a explicit agreement between the two peers.

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BGP convergence

In the previous sections, we have explained the operation of BGP routers. Compared to intradomain routing
protocols, a key feature of BGP is its ability to support interdomain routing policies that are defined by each
domain as its import and export filters and ranking process. A domain can define its own routing policies and
router vendors have implemented many configuration tweaks to support complex routing policies. However, the
routing policy chosen by a domain may interfere with the routing policy chosen by another domain. To understand
this issue, let us first consider the simple internetwork shown below.
1.0.0.0/8

AS1
Preferred paths for AS3
1. AS4:AS1
2. AS1

AS3

Preferred paths for AS4

1. AS3:AS1
2. AS1

AS4

Figure 5.71: The disagree internetwork

In this internetwork, we focus on the route towards 1.0.0.0/8 which is advertised by AS1. Let us also assume that
AS3 (resp. AS4) prefers, e.g. for economic reasons, a route learned from AS4 (AS3) over a route learned from AS1.
When AS1 sends U(1.0.0.0/8,AS1) to AS3 and AS4, three sequences of exchanges of BGP messages are possible :
1. AS3 sends first U(1.0.0.0/8,AS3:AS1) to AS4. AS4 has learned two routes towards 1.0.0.0/8. It runs its BGP
decision process and selects the route via AS3 and does not advertise a route to AS3
2. AS4 first sends U(1.0.0.0/8,AS3:AS1) to AS3. AS3 has learned two routes towards 1.0.0.0/8. It runs its BGP
decision process and selects the route via AS4 and does not advertise a route to AS4
3. AS3 sends U(1.0.0.0/8,AS3:AS1) to AS4 and, at the same time, AS4 sends U(1.0.0.0/8,AS4:AS1). AS3 prefers
the route via AS4 and thus sends W(1.0.0.0/8) to AS4. In the mean time, AS4 prefers the route via AS3 and
thus sends W(1.0.0.0/8) to AS3. Upon reception of the BGP Withdraws, AS3 and AS4 only know the direct
route towards 1.0.0.0/8. AS3 (resp. AS4) sends U(1.0.0.0/8,AS3:AS1) (resp. U(1.0.0.0/8,AS4:AS1)) to AS4
(resp. AS3). AS3 and AS4 could in theory continue to exchange BGP messages for ever. In practice, one of
them sends one message faster than the other and BGP converges.
The example above has shown that the routes selected by BGP routers may sometimes depend on the ordering of
the BGP messages that are exchanged. Other similar scenarios may be found in RFC 4264.
From an operational perspective, the above configuration is annoying since the network operators cannot easily
predict which paths are chosen. Unfortunately, there are even more annoying BGP configurations. For example,
let us consider the configuration below which is often named Bad Gadget [GW1999]
In this internetwork, there are four ASes. AS0 advertises one route towards one prefix and we only analyse the
routes towards this prefix. The routing preferences of AS1, AS3 and AS4 are the following :
AS1 prefers the path AS3:AS0 over all other paths
AS3 prefers the path AS4:AS0 over all other paths
AS4 prefers the path AS1:AS0 over all other paths
AS0 sends U(p,AS0) to AS1, AS3 and AS4. As this is the only route known by AS1, AS3 and AS4 towards p, they
all select the direct path. Let us now consider one possible exchange of BGP messages :
1. AS1 sends U(p, AS1:AS0) to AS3 and AS4. AS4 selects the path via AS1 since this is its preferred path. AS3
still uses the direct path.
2. AS4 advertises U(p,AS4:AS1:AS0) to AS3.

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AS1

Preferred paths for AS3

1. AS4:AS0
2. other paths

AS0

AS3

Preferred paths for AS1

1. AS3:AS0
2. other paths

Preferred paths for AS4

1. AS1:AS0
2. other paths

AS4

Figure 5.72: The bad gadget internetwork

3. AS3 sends U(p, AS3:AS0) to AS1 and AS4. AS1 selects the path via AS3 since this is its preferred path. AS4
still uses the path via AS1.
4. As AS1 has changed its path, it sends U(p,AS1:AS3:AS0) to AS4 and W(p) to AS3 since its new path is via
AS3. AS4 switches back to the direct path.
5. AS4 sends U(p,AS4:AS0) to AS1 and AS3. AS3 prefers the path via AS4.
6. AS3 sends U(p,AS3:AS4:AS0) to AS1 and W(p) to AS4. AS1 switches back to the direct path and we are
back at the first step.
This example shows that the convergence of BGP is unfortunately not always guaranteed as some interdomain
routing policies may interfere with each other in complex ways. [GW1999] have shown that checking for global
convergence is either NP-complete or NP-hard. See [GSW2002] for a more detailed discussion.
Fortunately, there are some operational guidelines [GR2001] [GGR2001] that can guarantee BGP convergence in
the global Internet. To ensure that BGP will converge, these guidelines consider that there are two types of peering
relationships : customer->provider and shared-cost. In this case, BGP convergence is guaranteed provided that
the following conditions are fulfilled :
1. The topology composed of all the directed customer->provider peering links is an acyclic graph
2. An AS always prefers a route received from a customer over a route received from a shared-cost peer or a
provider.
The first guideline implies that the provider of the provider of ASx cannot be a customer of ASx. Such a relationship
would not make sense from an economic perspective as it would imply circular payments. Furthermore, providers
are usually larger than customers.
The second guideline also corresponds to economic preferences. Since a provider earns money when sending
packets to one of its customers, it makes sense to prefer such customer learned routes over routes learned from
providers. [GR2001] also shows that BGP convergence is guaranteed even if an AS associates the same preference
to routes learned from a shared-cost peer and routes learned from a customer.
From a theoretical perspective, these guidelines should be verified automatically to ensure that BGP will always
converge in the global Internet. However, such a verification cannot be performed in practice because this would
force all domains to disclose their routing policies (and few are willing to do so) and furthermore the problem is
known to be NP-hard [GW1999].
In practice, researchers and operators expect that these guidelines are verified 38 in most domains. Thanks to the
large amount of BGP data that has been collected by operators and researchers 39 , several studies have analysed
38 Some researchers such as [MUF+2007] have shown that modelling the Internet topology at the AS-level requires more than the sharedcost and customer->provider peering relationships. However, there is no publicly available model that goes beyond these classical peering
relationships.
39 BGP data is often collected by establishing BGP sessions between Unix hosts running a BGP daemon and BGP routers in different ASes. The Unix hosts stores all BGP messages received and regular dumps of its BGP routing table. See http://www.routeviews.org,
http://www.ripe.net/ris, http://bgp.potaroo.net or http://irl.cs.ucla.edu/topology/

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the AS-level topology of the Internet. [SARK2002] is one of the first analysis. More recent studies include
[COZ2008] and [DKF+2007]
Based on these studies and [ATLAS2009], the AS-level Internet topology can be summarised as shown in the
figure below.

Figure 5.73: The layered structure of the global Internet

The domains on the Internet can be divided in about four categories according to their role and their position in
the AS-level topology.
the core of the Internet is composed of a dozen-twenty Tier-1 ISPs. A Tier-1 is a domain that has no provider.
Such an ISP has shared-cost peering relationships with all other Tier-1 ISPs and provider->customer relationships with smaller ISPs. Examples of Tier-1 ISPs include sprint, level3 or opentransit
the Tier-2 ISPs are national or continental ISPs that are customers of Tier-1 ISPs. These Tier-2 ISPs have
smaller customers and shared-cost peering relationships with other Tier-2 ISPs. Example of Tier-2 ISPs
include France Telecom, Belgacom, British Telecom, ...
the Tier-3 networks are either stub domains such as enterprise or campus networks networks and smaller
ISPs. They are customers of Tier-1 and Tier-2 ISPs and have sometimes shared-cost peering relationships
the large content providers that are managing large datacenters. These content providers are producing a
growing fraction of the packets exchanged on the global Internet [ATLAS2009]. Some of these content
providers are customers of Tier-1 or Tier-2 ISPs, but they often try to establish shared-cost peering relationships, e.g. at IXPs, with many Tier-1 and Tier-2 ISPs.
Due to this organisation of the Internet and due to the BGP decision process, most AS-level paths on the Internet
have a length of 3-5 AS hops.

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5.4 Summary
5.5 Exercises
5.5.1 Principles
1. Routing protocols used in data networks only use positive link weights. What would happen with a distance
vector routing protocol in the network below that contains a negative link weight ?

Figure 5.74: Simple network

2. When a network specialist designs a network, one of the problems that he needs to solve is to set the metrics
the links in his network. In the USA, the Abilene network interconnects most of the research labs and
universities. The figure below shows the topology 40 of this network in 2009.

Figure 5.75: The Abilene network

In this network, assume that all the link weights are set to 1. What is the paths followed by a packet
sent by the router located in Los Angeles to reach :
the router located in New York
the router located in Washington ?
Is it possible to configure the link metrics so that the packets sent by the router located in Los Angeles
to the routers located in respectively New York and Washington do not follow the same path ?
Is it possible to configure the link weights so that the packets sent by the router located in Los Angeles
to router located in New York follow one path while the packets sent by the router located in New York
to the router located in Los Angeles follow a completely different path ?
40 This figure was downloaded from the Abilene observatory http://www.internet2.edu/observatory/archive/data-views.html. This observatory contains a detailed description of the Abilene network including detailed network statistics and all the configuration of the equipment
used in the network.

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Assume that the routers located in Denver and Kansas City need to exchange lots of packets. Can you
configure the link metrics such that the link between these two routers does not carry any packet sent
by another router in the network ?
3. In the five nodes network shown below, can you configure the link metrics so that the packets sent by router
E to router A use link B->A while the packets sent by router B use links B->D and D->A?

Figure 5.76: Simple five nodes network

4. In the five nodes network shown above, can you configure the link weights so that the packets sent by router
E (resp. F) follow the E->B->A path (resp. F->D->B->A) ?
5. In the above questions, you have worked on the stable state of the routing tables computed by routing
protocols. Let us now consider the transient problems that main happen when the network topology changes
41
. For this, consider the network topology shown in the figure below and assume that all routers use a
distance vector protocol that uses split horizon.

Figure 5.77: Simple network with redundant links

If you compute the routing tables of all routers in this network, you would obtain a table such as the
table below :
41

The main events that can affect the topology of a network are : - the failure of a link. Measurements performed in IP networks
have shown that such failures happen frequently and usually for relatively short periods of time - the addition of one link in the network. This may be because a new link has been provisioned or more frequently because the link failed some time ago and is now
back - the failure/crash of a router followed by its reboot. - a change in the metric of a link by reconfiguring the routers attached to
the link See http://totem.info.ucl.ac.be/lisis_tool/lisis-example/ for an analysis of the failures inside the Abilene network in June 2005 or
http://citeseer.ist.psu.edu/old/markopoulou04characterization.html for an analysis of the failures affecting a larger ISP network

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Destination
A
B
C
D
E

Routes on A
0
1 via B
2 via B
3 via B
4 via B

Routes on B
1 via A
0
1 via C
2 via C
3 via C

Routes on C
2 via B
1 via B
0
1 via D
2 via D

Routes on D
3 via C
2 via C
1 via C
0
1 via E

Routes on E
4 via D
3 via D
2 via D
1 via D
0

Distance vector protocols can operate in two different modes : periodic updates and triggered updates.
Periodic updates is the default mode for a distance vector protocol. For example, each router could
advertise its distance vector every thirty seconds. With the triggered updates a router sends its distance
vector when its routing table changes (and periodically when there are no changes).
Consider a distance vector protocol using split horizon and periodic updates. Assume that the
link B-C fails. B and C update their local routing table but they will only advertise it at the end
of their period. Select one ordering for the periodic updates and every time a router sends its
distance vector, indicate the vector sent to each neighbor and update the table above. How many
periods are required to allow the network to converge to a stable state ?
Consider the same distance vector protocol, but now with triggered updates. When link B-C
fails, assume that B updates its routing table immediately and sends its distance vector to A and
D. Assume that both A and D process the received distance vector and that A sends its own
distance vector, ... Indicate all the distance vectors that are exchanged and update the table
above each time a distance vector is sent by a router (and received by other routers) until all
routers have learned a new route to each destination. How many distance vector messages must
be exchanged until the network converges to a stable state ?
6. Consider the network shown below. In this network, the metric of each link is set to 1 except link A-B whose
metric is set to 4 in both directions. In this network, there are two paths with the same cost between D and
C. Old routers would randomly select one of these equal cost paths and install it in their forwarding table.
Recent routers are able to use up to N equal cost paths towards the same destination.

Figure 5.78: A simple network running OSPF

On recent routers, a lookup in the forwarding table for a destination address returns a set of outgoing
interfaces. How would you design an algorithm that selects the outgoing interface used for each
packet, knowing that to avoid reordering, all segments of a given TCP connection should follow the
same path ?
7. Consider again the network shown above. After some time, OSPF converges and all routers compute the
following routing tables :
Destination
A
B
C
D
E

Routes on A
0
2 via C
1 via C
3 via C
2 via C

Routes on B
2 via C
0
1 via C
1 via D
2 via C,D

Routes on C
1 via A
1 via B
0
2 via B,E
1 via E

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Routes on D
3 via B,E
1 via B
2 via B,E
0
1 via E

Routes on E
2 via C
2 via D,C
1 via C
1 via D
0

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An important difference between OSPF and RIP is that OSPF routers flood link state packets that
allow the other routers to recompute their own routing tables while RIP routers exchange distance
vectors. Consider that link B-C fails and that router B is the first to detect the failure. At this point,
B cannot reach anymore A, C and 50% of its paths towards E have failed. C cannot reach B anymore
and half of its paths towards D have failed.
Router B will flood its updated link state packet through the entire network and all routers will recompute their forwarding table. Upon reception of a link state packet, routers usually first flood the
received link-state packet and then recompute their forwarding table. Assume that B is the first to
recompute its forwarding table, followed by D, A, C and finally E
8. After each update of a forwarding table, verify which pairs of routers are able to exchange packets. Provide
your answer using a table similar to the one shown above.
9. Can you find an ordering of the updates of the forwarding tables that avoids all transient problems ?
10. Consider the network shown in the figure below and explain the path that will be followed by the packets to
reach 194.100.10.0/23

Figure 5.79: A stub connected to one provider

11. Consider, now, as shown in the figure below that the stub AS is now also connected to provider AS789. Via
which provider will the packets destined to 194.100.10.0/23 will be received by AS4567 ? Should AS123
change its configuration ?

12. Consider that stub shown in the figure below decides to advertise two /24 prefixes instead of its allocated
/23 prefix.
1. Via which provider does AS4567 receive the packets destined to 194.100.11.99 and 194.100.10.1
?
2. How is the reachability of these addresses affected when link R1-R3 fails ?
3. Propose a configuration on R1 that achieves the same objective as the one shown in the figure but
also preserves the reachability of all IP addresses inside AS4567 if one of AS4567s interdomain

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Figure 5.80: A stub connected to two providers

links fails ?
13. Consider the network shown in the figure below. In this network, each AS contains a single BGP router.
Assume that R1 advertises a single prefix. R1 receives a lot of packets from R9. Without any help from R2,
R9 or R4, how could R1 configure its BGP advertisement such that it receives the packets from R9 via R3 ?
What happens when a link fails ?

14. Consider the network show in the figure below.

1. Show which BGP messages are exchanged when router R1 advertises prefix 10.0.0.0/8.
2. How many and which routes are known by router R5 ? Which route does it advertise to R6?
3. Assume now that the link between R1 and R2 fails. Show the messages exchanged due to this
event. Which BGP messages are sent to R6 ?
15. Consider the network shown in the figure below where R1 advertises a single prefix. In this network, the
link between R1 and R2 is considered as a backup link. It should only be used only when the primary link
(R1-R4) fails. This can be implemented on R2 by setting a low local-pref to the routes received on link
R2-R1
1. In this network, what are the paths used by all routers to reach R1 ?
2. Assume now that the link R1-R4 fails. Which BGP messages are exchanged and what are now
the paths used to reach R1 ?
3. Link R1-R4 comes back. Which BGP messages are exchanged and what do the paths used to
reach R1 become ?
16. On February 22, 2008, the Pakistan Telecom Authority issued an order to Pakistan ISPs to block access to
three IP addresses belonging to youtube: 208.65.153.238, 208.65.153.253, 208.65.153.251. One operator
noted that these addresses were belonging to the same /24 prefix. Read http://www.ripe.net/news/studyyoutube-hijacking.html to understand what happened really.
1. What should have done youtube to avoid this problem ?

3. Researchers and network operators collect and expose lots of BGP data. For this, they establish eBGP
sessions between data collection routers and production routers located in operational networks. Several
data collection routers are available, the most popular ones are :
http://www.routeviews.org
http://www.ripe.net/ris
For this exercise, you will use one of the routeviews BGP routers. You can access one of these routers
by using telnet. Once logged on the router, you can use the routers command line interface to analyse
its BGP routing table.
telnet route-views.routeviews.org
Trying 128.223.51.103...
Connected to route-views.routeviews.org.
Escape character is ^].
C
**********************************************************************
Oregon Exchange BGP Route Viewer
route-views.oregon-ix.net / route-views.routeviews.org
route views data is archived on http://archive.routeviews.org
This hardware is part of a grant from Cisco Systems.
Please contact [email protected] if you have questions or
comments about this service, its use, or if you might be able to
contribute your view.
This router has views of the full routing tables from several ASes.
The list of ASes is documented under "Current Participants" on
http://www.routeviews.org/.
**************
route-views.routeviews.org is now using AAA for logins.
username "rviews". See http://routeviews.org/aaa.html

**********************************************************************
User Access Verification
Username: rviews
route-views.oregon-ix.net>

This router has eBGP sessions with routers from several ISPs.
See
http://www.routeviews.org/peers/route-views.oregon-ix.net.txt for an up-to-date list of all eBGP
sessions maintained by this router.
Among all the commands supported by this router, the show ip bgp command is very useful. This
command takes an IPv4 prefix as parameter and allows you to retrieve all the routes that this routers
has received in its Adj-RIB-In for the specified prefix.
1. Use show ip bgp 130.104.0.0/16 to find the best path used by this router to reach UCLouvain
2. Knowing that 130.104.0.0/16 is announced by belnet (AS2611), what are, according to this BGP
routing tables, the ASes that peer with belnet
3. Do the same analysis for one of the IPv4 prefixes assigned to Skynet (AS5432) : 62.4.128.0/17.
The output of the show ip bgp 62.4.128.0/17 reveals something strange as it seems that one of
the paths towards this prefix passes twice via AS5432. Can you explain this ?
2905 702 1239 5432 5432
196.7.106.245 from 196.7.106.245 (196.7.106.245)
Origin IGP, metric 0, localpref 100, valid, external

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machines running User Model Linux. netkit allows to setup a small network in a lab and configure it as if
you had access to several PCs interconnected by using cables and network equipments.
A netkit lab is defined as a few configuration files and scripts :
lab.conf is a textfile that defines the virtual machines and the network topology. A simple lab.conf
file is shown below.
LAB_DESCRIPTION="a string describing the lab"
LAB_VERSION=1.0
LAB_AUTHOR="the author of the lab"
LAB_EMAIL="email address of the author"
h1[0]="lan"
h2[0]="lan"

This configuration file requests the creation of two virtual machines, named h1 and h2. Each of these hosts has one net
A host.startup file for each host (h1.startup and h2.startup in the example above). This file is
a shell script that is executed at the end of the boot of the virtual host. This is typically in this
script that the network interfaces are configured and the daemons are launched. A directory for
each host (h1 and h2 in the example above). This directory is used to store configuration files
that must be copied on the virtual machines filesystems when they are first created.
netkit contains several scripts that can be used to run a lab. lstart allows to launch a lab and lhalt
allows to halt the machines at the end of a lab. If you need to exchange files between the virtual
machines and the Linux host on which netkit runs, note that the virtual hosts mount the directory that
contains the running lab in /hostlab and your home directory in /hosthome.
For this exercise, you will use a netkit lab containing 4 hosts and two routers. The configuration files
are available exercises/labs/lab-2routers.tar.gz. The network topology of this lab is
shown in the figure below.

Figure 5.86: The two routers lab

The lab.conf file for this lab is shown below.
h1[0]="lan1"
h2[0]="lan1"
h3[0]="lan2"
router1[0]="lan1"
router1[1]="lan2"
router2[0]="lan2"
router2[1]="lan3"
h4[0]="lan3"

In this network, we will use subnet 172.12.1.0/24 for lan1, 172.12.2.0/24 for lan2 and 172.12.3.0/24
for lan3.
On Linux, the IP addresses assigned on an interface can be configured by using ifconfig(8).
When ifconfig(8) is used without parameters, it lists all the existing interfaces of the host with
their configuration. A sample ifconfig(8) output is shown below.
host:~# ifconfig
eth0
Link encap:Ethernet HWaddr FE:3A:59:CD:59:AD
Inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0

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inet6 addr: fe80::fc3a:59ff:fecd:59ad/64 Scope:Link

UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:3 errors:0 dropped:0 overruns:0 frame:0
TX packets:3 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:216 (216.0 b) TX bytes:258 (258.0 b)
Interrupt:5
lo

Link encap:Local Loopback

inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:0 errors:0 dropped:0 overruns:0 frame:0
TX packets:0 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:0 (0.0 b) TX bytes:0 (0.0 b)

This host has two interfaces : the loopback interface (lo with IPv4 address 127.0.0.1 and IPv6
address ::1) and the eth0 interface. The 192.168.1.1/24 address and a link local IPv6 address
(fe80::fc3a:59ff:fecd:59ad/64) have been assigned to interface eth0. The broadcast address is used
in some particular cases, this is outside the scope of this exercise. ifconfig(8) also provides
statistics such as the number of packets sent and received over this interface. Another important information that is provided by ifconfig(8) is the hardware address (HWaddr) used by the datalink
layer of the interface. On the example above, the eth0 interface uses the 48 bits FE:3A:59:CD:59:AD
hardware address.
You can configure the IPv4 address assigned to an interface by specifying the address and the netmask.
ifconfig eth0 192.168.1.2 netmask 255.255.255.128 up

You can also specify the prefix length

.. code-block:: text

ifconfig eth0 192.168.1.2/25 up

In both cases, ifconfig eth0 allows you to verify that the interface has been correctly configured.
eth0

Link encap:Ethernet HWaddr FE:3A:59:CD:59:AD

inet addr:192.168.1.2 Bcast:192.168.1.127 Mask:255.255.255.128
inet6 addr: fe80::fc3a:59ff:fecd:59ad/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:3 errors:0 dropped:0 overruns:0 frame:0
TX packets:3 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:216 (216.0 b) TX bytes:258 (258.0 b)
Interrupt:5

Another important command on Linux is route(8) that allows to look at the contents of the routing
table stored in the Linux kernel and change it. For example, route -n returns the contents of the IPv4
routing table. See route(8) for a detailed description on how you can configure routes by using
this tool.
1. Use ifconfig(8) to configure the following IPv4 addresses :
172.16.1.11/24 on interface eth0 on h1
172.16.1.12/24 on interface eth0 on h2
2. Use route -n to look at the contents of the routing table on the two hosts.
3. Verify by using ping(8) that h1 can reach 172.16.1.12
4. Use ifconfig(8) to configure IPv4 address 172.16.1.1/24 on the eth0 interface of router1
and 172.16.2.1/24 on the eth1 interface on this router.

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5. Since hosts h1 and h2 are attached to a local area network that contains a single router, this
router can act as a default router. Add a default route on h1 and h2 so that they can use router1
as their default router to reach any remote IPv4 address. Verify by using ping(8) that h1 can
reach address 172.16.2.1.
6. What do you need to configure on router2, h3 and h4 so that all hosts and routers can reach all
hosts and routers in the emulated network ? Add the ifconfig and route commands in the .startup
files of all the hosts so that the network is correctly configured when it is started by using lstart.
5. Use the network configured above to test how IP packets are fragmented. The ifconfig command allows you
to specify the Maximum Transmission Unit (MTU), i.e. the largest size of the frames that are allowed on a
given interface. The default MTU on the eth? interfaces is 1500 bytes.
1. Force an MTU of 500 bytes on the three interfaces attached to lan2.
2. Use ping -s 1000 to send a 1000 bytes ping packet from h3 to one of the routers attached to lan2 and capture
the packets on the other router by using tcpdump(8). In which order does the emulated host sends the IP
fragments ?
3. Use ping -s 2000 to send a 2000 bytes ping packet from h1 to h4 and capture the packets on lan2 and lan3
by using tcpdump(8). In which order does the emulated host sends the IP fragments ?
4. From your measurements, how does an emulated host generate the identifiers of the IP packets that it sends
?
5. Reset the MTU on the eth1 interface of router r1 at 1500 bytes, but leave the MTU on the eth0 interface
of router r2 at 500 bytes. Check whether host h1 can ping host h4. Use tcpdump(8) to analyse what is
happening.
6. The Routing Information Protocol (RIP) is a distance vector protocol that is often used in small IP networks.
There are various implementations of RIP. For this exercise, you will use quagga, an open-source implementation of several IP routing protocols that runs on Linux and other Unix compatible operating systems.
quagga(8) is in fact a set of daemons that interact together and with the Linux kernel. For this exercise,
you will use two of these daemons : zebra(8) and ripd(8). zebra(8) is the master daemon that
handles the interactions between the Linux kernel routing table and the routing protocols. ripd(8) is
the implementation of the RIP protocol. It interacts with the Linux routing tables through the zebra(8)
daemon.
To use a Linux real or virtual machine as a router, you need to first configure the IP addresses of the interfaces of the machine. Once this configuration has been verified, you can configure the zebra(8)
and ripd(8) daemons. The configuration files for these daemons reside in /etc/zebra. The first
configuration file is /etc/zebra/daemons. It lists the daemons that are launched when zebra is started
by /etc/init.d/zebra. To enable ripd(8) and zebra(8), this file will be configured as follows.
# This file tells the zebra package
# which daemons to start.
# Entries are in the format: <daemon>=(yes|no|priority)
# where yes is equivalent to infinitely low priority, and
# lower numbers mean higher priority. Read
# /usr/doc/zebra/README.Debian for details.
# Daemons are: bgpd zebra ospfd ospf6d ripd ripngd
zebra=yes
bgpd=no
ospfd=yes
ospf6d=no
ripd=no
ripngd=no

The second configuration file is the /etc/zebra/zebra.conf file. It defines the global configuration rules
that apply to zebra(8). For this exercise, we use the default configuration file shown below.
! -*- zebra -*!
! zebra configuration file
!

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hostname zebra
password zebra
enable password zebra
!
! Static default route sample.
!
!ip route 0.0.0.0/0 203.181.89.241
!
log file /var/log/zebra/zebra.log

In the zebra configuration file, lines beginning with ! are comments. This configuration defines the
hostname as zebra and two passwords. The default password (password zebra) is the one that must
be given when connecting to the zebra(8) management console over a TCP connection. This
management console can be use like a shell on a Unix host to specify commands to the zebra(8)
daemons. The second one (enable password zebra) specifies the password to be provided before
giving commands that change the configuration of the daemon. It is also possible to specify static
routes in this configuration file, but we do not use this facility in this exercise. The last parameter that
is specified is the log file where zebra(8) writes debugging information. Additional information
about quagga are available from http://www.quagga.net/docs/docs-info.php
The most interesting configuration file for this exercise is the /etc/zebra/ripd.conf file. It contains all
the parameters that are specific to the operation of the RIP protocol. A sample ripd(8) configuration file is shown below.
!
hostname ripd
password zebra
enable password zebra
!
router rip
network 100.1.0.0/16
redistribute connected
!
log file /var/log/zebra/ripd.log

This configuration files shown the two different ways to configure ripd(8). The statement router
rip indicates the beginning of the configuration for the RIP routing protocol. The indented lines that
follow are part of the configuration of this protocol. The first line, network 100.1.0.0/16 is used to
enable RIP on the interface whose IP subnet matches 100.1.0.0/16. The second line, redistribute
connected indicates that all the subnetworks that are directly connected on the router should be advertised. When this configuration line is used, ripd(8) interacts with the Linux kernel routing table
and advertises all the subnetworks that are directly connected on the router. If a new interface is enabled and configured on the router, its subnetwork prefix will be automatically advertised. Similarly,
the subnetwork prefix will be automatically removed if the subnetwork interface is shutdown.
To experiment with RIP, you will use the emulated routers shown in the figure below. You can
download the entire lab from exercises/labs/lab-5routers-rip.tar.gz

Figure 5.87: The five routers lab

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The lab.conf describing the topology and the interfaces used on all hosts is shown below/
r1[0]="A"
r1[1]="B"
r1[2]="F"
r1[3]="V"
r2[0]="A"
r2[1]="C"
r2[2]="W"
r3[0]="B"
r3[1]="C"
r3[2]="D"
r3[3]="X"
r4[0]="D"
r4[1]="E"
r4[2]="Y"
r5[0]="E"
r5[1]="F"
r5[2]="Z"

There are two types of subnetworks in this topology. The subnetworks from the 172.16.0.0/16 prefix
are used on the links between routers while the subnetworks from the 192.168.0.0/16 prefix are used
on the local area networks that are attached to a single router.
A router can be configured in two different ways : by specifying configuration files and by typing the
commands directly on the router by using telnet(1). The first four routers have been configured
in the provided configuration files. Look at r1.startup and the configurations files in r1/tmp/zebra in
the labs directory for router r1. The r?.startup files contain the ifconfig(8) commands that are
used to configure the interfaces of each virtual router. The configuration files located in r?/tmp/zebra
are also copied automatically on the virtual router when it boots.
1. Launch the lab by using lstart and verify that router r1 can reach 192.168.1.1, 192.168.2.2,
192.168.3.3 and 192.168.4.4. You can also traceroute(8) to determine what is the route
followed by your packets.
2. The ripd(8) daemon can also be configured by typing commands over a TCP connection.
ripd(8) listens on port 2602. On router r1, use telnet 127.0.0.1 2602 to connect to the
ripd(8) daemon. The default password is zebra. Once logged on the ripd(8) daemon,
you reach the > prompt where you can query the status of the router. By typing ? at the prompt,
you will find the list of supported commands. The show command is particularly useful, type
show ? to obtain the list of its sub options. For example, show ip rip will return the routing table
that is maintained by the ripd(8) daemon.
3. Disable interface eth3 on router r1 by typing ifconfig eth3 down on this router. Verify the
impact of this command on the routing tables of the other routers in the network. Re-enable
this interface by typing ifconfig eth3 up.
4. Do the same with the eth1 interface on router r3.
5. Edit the /etc/zebra/ripd.conf configuration file on router r5 so that this router becomes part of
the network. Verify that 192.168.5.5 is reachable by all routers inside the network.
7. The Open Shortest Path First (OSPF) protocol is a link-state protocol that is often used in enterprise IP networks. OSPF is implemented in the ospfd(8) daemon that is part of quagga.
We use the same topology as in the previous exercise. The netkit lab may be downloaded from
exercises/labs/lab-5routers-ospf.tar.gz.
The ospfd(8) daemon supports a more complex configuration that the ripd(8) daemon. A
sample configuration is shown below.
!
hostname ospfd
password zebra
enable password zebra
!

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interface eth0
ip ospf cost 1
interface eth1
ip ospf cost 1
interface eth2
ip ospf cost 1
interface eth3
ip ospf cost 1
!
router ospf
router-id 192.168.1.1
network 172.16.1.0/24 area 0.0.0.0
network 172.16.2.0/24 area 0.0.0.0
network 172.16.3.0/24 area 0.0.0.0
network 192.168.1.0/24 area 0.0.0.0
passive-interface eth3
!
log file /var/log/zebra/ospfd.log

In this configuration file, the ip ospf cost 1 specify a metric of 1 for each interface. The ospfd(8)
configuration is composed of three parts. First, each router must have one identifier that is unique
inside the network. Usually, this identifier is one of the IP addresses assigned to the router. Second,
each subnetwork on the router is associated with an area. In this example, we only use the backbone
area (i.e. 0.0.0.0). The last command specifies that the OSPF Hello messages should not be sent over
interface eth3 although its subnetwork will be advertised by the router. Such a command is often used
on interfaces that are attached to endhosts to ensure that no problem will occur if a student configures
a software OSPF router on his laptop attached to this interface.
The netkit lab contains already the configuration for routers r1 - r4.
The ospfd(8) daemon listens on TCP port 2604. You can follow the evolution of the OSPF protocol
by using the show ip ospf ? commands.
1. Launch the lab by using lstart and verify that the 192.168.1.1, 192.168.2.2, 192.168.3.3 and
192.168.4.4 addresses are reachable from any router inside the network.
2. Configure router r5 by changing the /etc/zebra/ospfd.conf file and restart the daemon. Verify
that the 192.168.5.5 address is reachable from any router inside the network.
3. How can you update the network configuration so that the packets sent by router r1 to router r5
use the direct link between the two routers while the packets sent by r5 are forwarded via r4 ?
4. Disable interface eth3 on router r1 and see how quickly the network converges ? You can follow
the evolution of the routing table on a router by typing netstat -rnc. Re-enable interface eth3 on
router r1.
5. Change the MTU of eth0 on router r1 but leave it unchanged on interface eth0 of router r2.
What is the impact of this change ? Can you explain why ?
6. Disable interface eth1 on router r3 and see how quickly the network converges ? Re-enable this
interface.
7. Halt router r2 by using vcrash r2. How quickly does the network react to this failure ?

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CHAPTER 6

The datalink layer and the Local Area

Networks

The datalink layer is the lowest layer of the reference model that we discuss in detail. As mentioned previously,
there are two types of datalink layers. The first datalink layers that appeared are the ones that are used on pointto-point links between devices that are directly connected by a physical link. We will briefly discuss one of these
datalink layers in this chapter. The second type of datalink layers are the ones used in Local Area Networks
(LANs). The main difference between the point-to-point and the LAN datalink layers is that the latter need to
regulate the access to the Local Area Network which is usually a shared medium.
This chapter is organised as follows. We first discuss the principles of the datalink layer as well as the services
that it uses from the physical layer. We then describe in more detail several Medium Access Control algorithms
that are used in Local Area Networks to regulate the access to the shared medium. Finally we discuss in detail
important datalink layer technologies with an emphasis on Ethernet and WiFi networks.

6.1 Principles
The datalink layer uses the service provided by the physical layer. Although there are many different implementations of the physical layer from a technological perspective, they all provide a service that enables the datalink
layer to send and receive bits between directly connected devices. The datalink layer receives packets from the
network layer. Two datalink layer entities exchange frames. As explained in the previous chapter, most datalink
layer technologies impose limitations on the size of the frames. Some technologies only impose a maximum
frame size, others enforce both minimum and maximum frames sizes and finally some technologies only support
a single frame size. In the latter case, the datalink layer will usually include an adaptation sublayer to allow the
network layer to send and receive variable-length packets. This adaptation layer may include fragmentation and
reassembly mechanisms.

Figure 6.1: The datalink layer and the reference model

The physical layer service facilitates the sending and receiving of bits. Furthermore, it is usually far from perfect
as explained in the introduction :
the Physical layer may change, e.g. due to electromagnetic interferences, the value of a bit being transmitted
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the Physical layer may deliver more bits to the receiver than the bits sent by the sender
the Physical layer may deliver fewer bits to the receiver than the bits sent by the sender
The datalink layer must allow endsystems to exchange frames containing packets despite all of these limitations.
On point-to-point links and Local Area Networks, the first problem to be solved is how to encode a frame as a
sequence of bits, so that the receiver can easily recover the received frame despite the limitations of the physical
layer.
If the physical layer were perfect, the problem would be very simple. The datalink layer would simply need to
define how to encode each frame as a sequence of consecutive bits. The receiver would then easily be able to
extract the frames from the received bits. Unfortunately, the imperfections of the physical layer make this framing
problem slightly more complex. Several solutions have been proposed and are used in practice in different datalink
layer technologies.

6.1.1 Framing
This is the framing problem. It can be defined as : How does a sender encode frames so that the receiver can
efficiently extract them from the stream of bits that it receives from the physical layer.
A first solution to solve the framing problem is to require the physical layer to remain idle for some time after the
transmission of each frame. These idle periods can be detected by the receiver and serve as a marker to delineate
frame boundaries. Unfortunately, this solution is not sufficient for two reasons. First, some physical layers cannot
remain idle and always need to transmit bits. Second, inserting an idle period between frames decreases the
maximum bandwidth that can be achieved by the datalink layer.
Some physical layers provide an alternative to this idle period. All physical layers are able to send and receive
physical symbols that represent values 0 and 1. However, for various reasons that are outside the scope of this
chapter, several physical layers are able to exchange other physical symbols as well. For example, the Manchester
encoding used in several physical layers can send four different symbols. The Manchester encoding is a differential
encoding scheme in which time is divided into fixed-length periods. Each period is divided in two halves and two
different voltage levels can be applied. To send a symbol, the sender must set one of these two voltage levels
during each half period. To send a 1 (resp. 0), the sender must set a high (resp. low) voltage during the first half
of the period and a low (resp. high) voltage during the second half. This encoding ensures that there will be a
transition at the middle of each period and allows the receiver to synchronise its clock to the senders clock. Apart
from the encodings for 0 and 1, the Manchester encoding also supports two additional symbols : InvH and InvB
where the same voltage level is used for the two half periods. By definition, these two symbols cannot appear
inside a frame which is only composed of 0 and 1. Some technologies use these special symbols as markers for
the beginning or end of frames.

Figure 6.2: Manchester encoding

Unfortunately, multi-symbol encodings cannot be used by all physical layers and a generic solution which can be
used with any physical layer that is able to transmit and receive only 0 and 1 is required. This generic solution is
called stuffing and two variants exist : bit stuffing and character stuffing. To enable a receiver to easily delineate
the frame boundaries, these two techniques reserve special bit strings as frame boundary markers and encode the
frames so that these special bit strings do not appear inside the frames.
Bit stuffing reserves the 01111110 bit string as the frame boundary marker and ensures that there will never be
six consecutive 1 symbols transmitted by the physical layer inside a frame. With bit stuffing, a frame is sent as
follows. First, the sender transmits the marker, i.e. 01111110. Then, it sends all the bits of the frame and inserts

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an additional bit set to 0 after each sequence of five consecutive 1 bits. This ensures that the sent frame never
contains a sequence of six consecutive bits set to 1. As a consequence, the marker pattern cannot appear inside the
frame sent. The marker is also sent to mark the end of the frame. The receiver performs the opposite to decode a
received frame. It first detects the beginning of the frame thanks to the 01111110 marker. Then, it processes the
received bits and counts the number of consecutive bits set to 1. If a 0 follows five consecutive bits set to 1, this bit
is removed since it was inserted by the sender. If a 1 follows five consecutive bits sets to 1, it indicates a marker if
it is followed by a bit set to 0. The table below illustrates the application of bit stuffing to some frames.
Original frame
0001001001001001001000011
0110111111111111111110010
01111110

Transmitted frame
01111110000100100100100100100001101111110
01111110011011111011111011111011001001111110
0111111001111101001111110

For example, consider the transmission of 0110111111111111111110010. The sender will first send the 01111110
marker followed by 011011111. After these five consecutive bits set to 1, it inserts a bit set to 0 followed by 11111.
A new 0 is inserted, followed by 11111. A new 0 is inserted followed by the end of the frame 110010 and the
01111110 marker.
Bit stuffing increases the number of bits required to transmit each frame. The worst case for bit stuffing is of course
a long sequence of bits set to 1 inside the frame. If transmission errors occur, stuffed bits or markers can be in
error. In these cases, the frame affected by the error and possibly the next frame will not be correctly decoded by
the receiver, but it will be able to resynchronise itself at the next valid marker.
Bit stuffing can be easily implemented in hardware. However, implementing it in software is difficult given the
higher overhead of bit manipulations in software. Software implementations prefer to process characters than
bits, software-based datalink layers usually use character stuffing. This technique operates on frames that contain
an integer number of 8-bit characters. Some characters are used as markers to delineate the frame boundaries.
Many character stuffing techniques use the DLE, STX and ETX characters of the ASCII character set. DLE STX
(resp. DLE ETX) is used to mark the beginning (end) of a frame. When transmitting a frame, the sender adds a
DLE character after each transmitted DLE character. This ensures that none of the markers can appear inside the
transmitted frame. The receiver detects the frame boundaries and removes the second DLE when it receives two
consecutive DLE characters. For example, to transmit frame 1 2 3 DLE STX 4, a sender will first send DLE STX
as a marker, followed by 1 2 3 DLE. Then, the sender transmits an additional DLE character followed by STX 4
and the DLE ETX marker.
Original frame
1234
1 2 3 DLE STX 4
DLE STX DLE ETX

Transmitted frame
DLE STX 1 2 3 4 DLE ETX
DLE STX 1 2 3 DLE DLE STX 4 DLE ETX
DLE STX DLE DLE STX DLE DLE ETX DLE ETX

Character stuffing , like bit stuffing, increases the length of the transmitted frames. For character stuffing, the worst
frame is a frame containing many DLE characters. When transmission errors occur, the receiver may incorrectly
decode one or two frames (e.g. if the errors occur in the markers). However, it will be able to resynchronise itself
with the next correctly received markers.

6.1.2 Error detection

Besides framing, datalink layers also include mechanisms to detect and sometimes even recover from transmission
error. To allow a receiver to detect transmission errors, a sender must add some redundant information as an error
detection code to the frame sent. This error detection code is computed by the sender on the frame that it transmits.
When the receiver receives a frame with an error detection code, it recomputes it and verifies whether the received
error detection code matches the computer error detection code. If they match, the frame is considered to be valid.
Many error detection schemes exist and entire books have been written on the subject. A detailed discussion of
these techniques is outside the scope of this book, and we will only discuss some examples to illustrate the key
principles.
To understand error detection codes, let us consider two devices that exchange bit strings containing N bits. To
allow the receiver to detect a transmission error, the sender converts each string of N bits into a string of N+r
bits. Usually, the r redundant bits are added at the beginning or the end of the transmitted bit string, but some
techniques interleave redundant bits with the original bits. An error detection code can be defined as a function
that computes the r redundant bits corresponding to each string of N bits. The simplest error detection code is the
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parity bit. There are two types of parity schemes : even and odd parity. With the even (resp. odd) parity scheme,
the redundant bit is chosen so that an even (resp. odd) number of bits are set to 1 in the transmitted bit string of
N+r bits. The receiver can easily recompute the parity of each received bit string and discard the strings with an
invalid parity. The parity scheme is often used when 7-bit characters are exchanged. In this case, the eighth bit is
often a parity bit. The table below shows the parity bits that are computed for bit strings containing three bits.
3 bits string
000
001
010
100
111
110
101
011

Odd parity
1
0
0
0
0
1
1
1

Even parity
0
1
1
1
1
0
0
0

The parity bit allows a receiver to detect transmission errors that have affected a single bit among the transmitted
N+r bits. If there are two or more bits in error, the receiver may not necessarily be able to detect the transmission
error. More powerful error detection schemes have been defined. The Cyclical Redundancy Checks (CRC) are
widely used in datalink layer protocols. An N-bits CRC can detect all transmission errors affecting a burst of
less than N bits in the transmitted frame and all transmission errors that affect an odd number of bits. Additional
details about CRCs may be found in [Williams1993].
It is also possible to design a code that allows the receiver to correct transmission errors. The simplest error
correction code is the triple modular redundancy (TMR). To transmit a bit set to 1 (resp. 0), the sender transmits
111 (resp. 000). When there are no transmission errors, the receiver can decode 111 as 1. If transmission errors
have affected a single bit, the receiver performs majority voting as shown in the table below. This scheme allows
the receiver to correct all transmission errors that affect a single bit.
Received bits
000
001
010
100
111
110
101
011

Decoded bit
0
0
0
0
1
1
1
1

Other more powerful error correction codes have been proposed and are used in some applications. The Hamming
Code is a clever combination of parity bits that provides error detection and correction capabilities.
In practice, datalink layer protocols combine bit stuffing or character stuffing with a length indication in the frame
header and a checksum or CRC. The checksum/CRC is computed by the sender and placed in the frame before
applying bit/character stuffing.

6.2 Medium Access Control

Point-to-point datalink layers need to select one of the framing techniques described above and optionally add
retransmission algorithms such as those explained for the transport layer to provide a reliable service. Datalink
layers for Local Area Networks face two additional problems. A LAN is composed of several hosts that are
attached to the same shared physical medium. From a physical layer perspective, a LAN can be organised in four
different ways :
a bus-shaped network where all hosts are attached to the same physical cable
a ring-shaped where all hosts are attached to an upstream and a downstream node so that the entire network
forms a ring
a star-shaped network where all hosts are attached to the same device

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a wireless network where all hosts can send and receive frames using radio signals
These four basic physical organisations of Local Area Networks are shown graphically in the figure below. We
will first focus on one physical organisation at a time.

Figure 6.3: Bus, ring and star-shaped Local Area Network

The common problem among all of these network organisations is how to efficiently share the access to the Local
Area Network. If two devices send a frame at the same time, the two electrical, optical or radio signals that
correspond to these frames will appear at the same time on the transmission medium and a receiver will not be
able to decode either frame. Such simultaneous transmissions are called collisions. A collision may involve frames
transmitted by two or more devices attached to the Local Area Network. Collisions are the main cause of errors
in wired Local Area Networks.
All Local Area Network technologies rely on a Medium Access Control algorithm to regulate the transmissions to
either minimise or avoid collisions. There are two broad families of Medium Access Control algorithms :
1. Deterministic or pessimistic MAC algorithms. These algorithms assume that collisions are a very severe
problem and that they must be completely avoided. These algorithms ensure that at any time, at most one
device is allowed to send a frame on the LAN. This is usually achieved by using a distributed protocol which
elects one device that is allowed to transmit at each time. A deterministic MAC algorithm ensures that no
collision will happen, but there is some overhead in regulating the transmission of all the devices attached
to the LAN.
2. Stochastic or optimistic MAC algorithms. These algorithms assume that collisions are part of the normal
operation of a Local Area Network. They aim to minimise the number of collisions, but they do not try to
avoid all collisions. Stochastic algorithms are usually easier to implement than deterministic ones.
We first discuss a simple deterministic MAC algorithm and then we describe several important optimistic algorithms, before coming back to a distributed and deterministic MAC algorithm.

6.2.1 Static allocation methods

A first solution to share the available resources among all the devices attached to one Local Area Network is to
define, a priori, the distribution of the transmission resources among the different devices. If N devices need to
share the transmission capacities of a LAN operating at b Mbps, each device could be allocated a bandwidth of Nb
Mbps.
Limited resources need to be shared in other environments than Local Area Networks. Since the first radio transmissions by Marconi more than one century ago, many applications that exchange information through radio
signals have been developed. Each radio signal is an electromagnetic wave whose power is centered around a
given frequency. The radio spectrum corresponds to frequencies ranging between roughly 3 KHz and 300 GHz.
Frequency allocation plans negotiated among governments reserve most frequency ranges for specific applications
such as broadcast radio, broadcast television, mobile communications, aeronautical radio navigation, amateur radio, satellite, etc. Each frequency range is then subdivided into channels and each channel can be reserved for a
given application, e.g. a radio broadcaster in a given region.
Frequency Division Multiplexing (FDM) is a static allocation scheme in which a frequency is allocated to each
device attached to the shared medium. As each device uses a different transmission frequency, collisions cannot

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occur. In optical networks, a variant of FDM called Wavelength Division Multiplexing (WDM) can be used. An
optical fiber can transport light at different wavelengths without interference. With WDM, a different wavelength
is allocated to each of the devices that share the same optical fiber.
Time Division Multiplexing (TDM) is a static bandwidth allocation method that was initially defined for the telephone network. In the fixed telephone network, a voice conversation is usually transmitted as a 64 Kbps signal.
Thus, a telephone conservation generates 8 KBytes per second or one byte every 125 microsecond. Telephone
conversations often need to be multiplexed together on a single line. For example, in Europe, thirty 64 Kbps voice
signals are multiplexed over a single 2 Mbps (E1) line. This is done by using Time Division Multiplexing (TDM).
TDM divides the transmission opportunities into slots. In the telephone network, a slot corresponds to 125 microseconds. A position inside each slot is reserved for each voice signal. The figure below illustrates TDM on a
link that is used to carry four voice conversations. The vertical lines represent the slot boundaries and the letters
the different voice conversations. One byte from each voice conversation is sent during each 125 microsecond
slot. The byte corresponding to a given conversation is always sent at the same position in each slot.

Figure 6.4: Time-division multiplexing

TDM as shown above can be completely static, i.e. the same conversations always share the link, or dynamic. In
the latter case, the two endpoints of the link must exchange messages specifying which conversation uses which
byte inside each slot. Thanks to these signalling messages, it is possible to dynamically add and remove voice
conversations from a given link.
TDM and FDM are widely used in telephone networks to support fixed bandwidth conversations. Using them
in Local Area Networks that support computers would probably be inefficient. Computers usually do not send
information at a fixed rate. Instead, they often have an on-off behaviour. During the on period, the computer tries
to send at the highest possible rate, e.g. to transfer a file. During the off period, which is often much longer than
the on period, the computer does not transmit any packet. Using a static allocation scheme for computers attached
to a LAN would lead to huge inefficiencies, as they would only be able to transmit at N1 of the total bandwidth
during their on period, despite the fact that the other computers are in their off period and thus do not need to
transmit any information. The dynamic MAC algorithms discussed in the remainder of this chapter aim solve this
problem.

6.2.2 ALOHA
In the 1960s, computers were mainly mainframes with a few dozen terminals attached to them. These terminals
were usually in the same building as the mainframe and were directly connected to it. In some cases, the terminals
were installed in remote locations and connected through a modem attached to a dial-up line. The university
of Hawaii chose a different organisation. Instead of using telephone lines to connect the distant terminals, they
developed the first packet radio technology [Abramson1970]. Until then, computer networks were built on top of
either the telephone network or physical cables. ALOHANet showed that it was possible to use radio signals to
interconnect computers.
The first version of ALOHANet, described in [Abramson1970], operated as follows: First, the terminals and the
mainframe exchanged fixed-length frames composed of 704 bits. Each frame contained 80 8-bit characters, some
control bits and parity information to detect transmission errors. Two channels in the 400 MHz range were reserved
for the operation of ALOHANet. The first channel was used by the mainframe to send frames to all terminals.
The second channel was shared among all terminals to send frames to the mainframe. As all terminals share the
same transmission channel, there is a risk of collision. To deal with this problem as well as transmission errors,
the mainframe verified the parity bits of the received frame and sent an acknowledgement on its channel for each
correctly received frame. The terminals on the other hand had to retransmit the unacknowledged frames. As for
TCP, retransmitting these frames immediately upon expiration of a fixed timeout is not a good approach as several
terminals may retransmit their frames at the same time leading to a network collapse. A better approach, but still
far from perfect, is for each terminal to wait a random amount of time after the expiration of its retransmission
timeout. This avoids synchronisation among multiple retransmitting terminals.
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The pseudo-code below shows the operation of an ALOHANet terminal. We use this python syntax for all Medium
Access Control algorithms described in this chapter. The algorithm is applied to each new frame that needs to be
transmitted. It attempts to transmit a frame at most max times (while loop). Each transmission attempt is performed
as follows: First, the frame is sent. Each frame is protected by a timeout. Then, the terminal waits for either a
valid acknowledgement frame or the expiration of its timeout. If the terminal receives an acknowledgement, the
frame has been delivered correctly and the algorithm terminates. Otherwise, the terminal waits for a random time
and attempts to retransmit the frame.
# ALOHA
N=1
while N<= max :
send(frame)
wait(ack_on_return_channel or timeout)
if (ack_on_return_channel):
break # transmission was successful
else:
# timeout
wait(random_time)
N=N+1
else:
# Too many transmission attempts

[Abramson1970] analysed the performance of ALOHANet under particular assumptions and found that ALOHANet worked well when the channel was lightly loaded. In this case, the frames are rarely retransmitted and the
channel traffic, i.e. the total number of (correct and retransmitted) frames transmitted per unit of time is close to
the channel utilization, i.e. the number of correctly transmitted frames per unit of time. Unfortunately, the analysis
1
= 0.186 times the channel bandwidth. At
also reveals that the channel utilization reaches its maximum at 2e
higher utilization, ALOHANet becomes unstable and the network collapses due to collided retransmissions.
Note: Amateur packet radio
Packet radio technologies have evolved in various directions since the first experiments performed at the University
of Hawaii. The Amateur packet radio service developed by amateur radio operators is one of the descendants
ALOHANet. Many amateur radio operators are very interested in new technologies and they often spend countless
hours developing new antennas or transceivers. When the first personal computers appeared, several amateur radio
operators designed radio modems and their own datalink layer protocols [KPD1985] [BNT1997]. This network
grew and it was possible to connect to servers in several European countries by only using packet radio relays.
Some amateur radio operators also developed TCP/IP protocol stacks that were used over the packet radio service.
Some parts of the amateur packet radio network are connected to the global Internet and use the 44.0.0.0/8 prefix.
Many improvements to ALOHANet have been proposed since the publication of [Abramson1970], and this technique, or some of its variants, are still found in wireless networks today. The slotted technique proposed in
[Roberts1975] is important because it shows that a simple modification can significantly improve channel utilization. Instead of allowing all terminals to transmit at any time, [Roberts1975] proposed to divide time into slots
and allow terminals to transmit only at the beginning of each slot. Each slot corresponds to the time required to
transmit one fixed size frame. In practice, these slots can be imposed by a single clock that is received by all
terminals. In ALOHANet, it could have been located on the central mainframe. The analysis in [Roberts1975]
reveals that this simple modification improves the channel utilization by a factor of two.

6.2.3 Carrier Sense Multiple Access

ALOHA and slotted ALOHA can easily be implemented, but unfortunately, they can only be used in networks that
are very lightly loaded. Designing a network for a very low utilisation is possible, but it clearly increases the cost
of the network. To overcome the problems of ALOHA, many Medium Access Control mechanisms have been
proposed which improve channel utilization. Carrier Sense Multiple Access (CSMA) is a significant improvement
compared to ALOHA. CSMA requires all nodes to listen to the transmission channel to verify that it is free before
transmitting a frame [KT1975]. When a node senses the channel to be busy, it defers its transmission until the
channel becomes free again. The pseudo-code below provides a more detailed description of the operation of
CSMA.
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# persistent CSMA
N=1
while N<= max :
wait(channel_becomes_free)
send(frame)
wait(ack or timeout)
if ack :
break # transmission was successful
else :
# timeout
N=N+1
# end of while loop
# Too many transmission attempts

The above pseudo-code is often called persistent CSMA [KT1975] as the terminal will continuously listen to the
channel and transmit its frame as soon as the channel becomes free. Another important variant of CSMA is the
non-persistent CSMA [KT1975]. The main difference between persistent and non-persistent CSMA described
in the pseudo-code below is that a non-persistent CSMA node does not continuously listen to the channel to
determine when it becomes free. When a non-persistent CSMA terminal senses the transmission channel to be
busy, it waits for a random time before sensing the channel again. This improves channel utilization compared to
persistent CSMA. With persistent CSMA, when two terminals sense the channel to be busy, they will both transmit
(and thus cause a collision) as soon as the channel becomes free. With non-persistent CSMA, this synchronisation
does not occur, as the terminals wait a random time after having sensed the transmission channel. However, the
higher channel utilization achieved by non-persistent CSMA comes at the expense of a slightly higher waiting
time in the terminals when the network is lightly loaded.
# Non persistent CSMA
N=1
while N<= max :
listen(channel)
if free(channel):
send(frame)
wait(ack or timeout)
if received(ack) :
break # transmission was successful
else :
# timeout
N=N+1
else:
wait(random_time)
# end of while loop
# Too many transmission attempts

[KT1975] analyzes in detail the performance of several CSMA variants. Under some assumptions about the transmission channel and the traffic, the analysis compares ALOHA, slotted ALOHA, persistent and non-persistent
CSMA. Under these assumptions, ALOHA achieves a channel utilization of only 18.4% of the channel capacity.
Slotted ALOHA is able to use 36.6% of this capacity. Persistent CSMA improves the utilization by reaching
52.9% of the capacity while non-persistent CSMA achieves 81.5% of the channel capacity.

6.2.4 Carrier Sense Multiple Access with Collision Detection

CSMA improves channel utilization compared to ALOHA. However, the performance can still be improved,
especially in wired networks. Consider the situation of two terminals that are connected to the same cable. This
cable could, for example, be a coaxial cable as in the early days of Ethernet [Metcalfe1976]. It could also be built
with twisted pairs. Before extending CSMA, it is useful to understand more intuitively, how frames are transmitted
in such a network and how collisions can occur. The figure below illustrates the physical transmission of a frame
on such a cable. To transmit its frame, host A must send an electrical signal on the shared medium. The first step
is thus to begin the transmission of the electrical signal. This is point (1) in the figure below. This electrical signal
will travel along the cable. Although electrical signals travel fast, we know that information cannot travel faster
than the speed of light (i.e. 300.000 kilometers/second). On a coaxial cable, an electrical signal is slightly slower
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than the speed of light and 200.000 kilometers per second is a reasonable estimation. This implies that if the cable
has a length of one kilometer, the electrical signal will need 5 microseconds to travel from one end of the cable to
the other. The ends of coaxial cables are equipped with termination points that ensure that the electrical signal is
not reflected back to its source. This is illustrated at point (3) in the figure, where the electrical signal has reached
the left endpoint and host B. At this point, B starts to receive the frame being transmitted by A. Notice that there is
a delay between the transmission of a bit on host A and its reception by host B. If there were other hosts attached
to the cable, they would receive the first bit of the frame at slightly different times. As we will see later, this timing
difference is a key problem for MAC algorithms. At point (4), the electrical signal has reached both ends of the
cable and occupies it completely. Host A continues to transmit the electrical signal until the end of the frame. As
shown at point (5), when the sending host stops its transmission, the electrical signal corresponding to the end of
the frame leaves the coaxial cable. The channel becomes empty again once the entire electrical signal has been
removed from the cable.

Figure 6.5: Frame transmission on a shared bus

Now that we have looked at how a frame is actually transmitted as an electrical signal on a shared bus, it is
interesting to look in more detail at what happens when two hosts transmit a frame at almost the same time. This
is illustrated in the figure below, where hosts A and B start their transmission at the same time (point (1)). At this
time, if host C senses the channel, it will consider it to be free. This will not last a long time and at point (2) the
electrical signals from both host A and host B reach host C. The combined electrical signal (shown graphically as
the superposition of the two curves in the figure) cannot be decoded by host C. Host C detects a collision, as it
receives a signal that it cannot decode. Since host C cannot decode the frames, it cannot determine which hosts
are sending the colliding frames. Note that host A (and host B) will detect the collision after host C (point (3) in
the figure below).
As shown above, hosts detect collisions when they receive an electrical signal that they cannot decode. In a wired
network, a host is able to detect such a collision both while it is listening (e.g. like host C in the figure above)
and also while it is sending its own frame. When a host transmits a frame, it can compare the electrical signal
that it transmits with the electrical signal that it senses on the wire. At points (1) and (2) in the figure above,
host A senses only its own signal. At point (3), it senses an electrical signal that differs from its own signal and
can thus detects the collision. At this point, its frame is corrupted and it can stop its transmission. The ability
to detect collisions while transmitting is the starting point for the Carrier Sense Multiple Access with Collision
Detection (CSMA/CD) Medium Access Control algorithm, which is used in Ethernet networks [Metcalfe1976]
[802.3] . When an Ethernet host detects a collision while it is transmitting, it immediately stops its transmission.
Compared with pure CSMA, CSMA/CD is an important improvement since when collisions occur, they only last
until colliding hosts have detected it and stopped their transmission. In practice, when a host detects a collision, it
sends a special jamming signal on the cable to ensure that all hosts have detected the collision.
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Figure 6.6: Frame collision on a shared bus

To better understand these collisions, it is useful to analyse what would be the worst collision on a shared bus
network. Let us consider a wire with two hosts attached at both ends, as shown in the figure below. Host A
starts to transmit its frame and its electrical signal is propagated on the cable. Its propagation time depends on the
physical length of the cable and the speed of the electrical signal. Let us use to represent this propagation delay
in seconds. Slightly less than seconds after the beginning of the transmission of As frame, B decides to start
transmitting its own frame. After seconds, B senses As frame, detects the collision and stops transmitting. The
beginning of Bs frame travels on the cable until it reaches host A. Host A can thus detect the collision at time
+ 2 . An important point to note is that a collision can only occur during the first 2 seconds of
its transmission. If a collision did not occur during this period, it cannot occur afterwards since the transmission
channel is busy after seconds and CSMA/CD hosts sense the transmission channel before transmitting their
frame.

Figure 6.7: The worst collision on a shared bus

Furthermore, on the wired networks where CSMA/CD is used, collisions are almost the only cause of transmission
errors that affect frames. Transmission errors that only affect a few bits inside a frame seldom occur in these wired
networks. For this reason, the designers of CSMA/CD chose to completely remove the acknowledgement frames
in the datalink layer. When a host transmits a frame, it verifies whether its transmission has been affected by a
collision. If not, given the negligible Bit Error Ratio of the underlying network, it assumes that the frame was
received correctly by its destination. Otherwise the frame is retransmitted after some delay.
Removing acknowledgements is an interesting optimisation as it reduces the number of frames that are exchanged
on the network and the number of frames that need to be processed by the hosts. However, to use this optimisation,
we must ensure that all hosts will be able to detect all the collisions that affect their frames. The problem is
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important for short frames. Let us consider two hosts, A and B, that are sending a small frame to host C as
illustrated in the figure below. If the frames sent by A and B are very short, the situation illustrated below may
occur. Hosts A and B send their frame and stop transmitting (point (1)). When the two short frames arrive at the
location of host C, they collide and host C cannot decode them (point (2)). The two frames are absorbed by the
ends of the wire. Neither host A nor host B have detected the collision. They both consider their frame to have
been received correctly by its destination.

Figure 6.8: The short-frame collision problem

To solve this problem, networks using CSMA/CD require hosts to transmit for at least 2 seconds. Since
the network transmission speed is fixed for a given network technology, this implies that a technology that uses
CSMA/CD enforces a minimum frame size. In the most popular CSMA/CD technology, Ethernet, 2 is called
the slot time 1 .
The last innovation introduced by CSMA/CD is the computation of the retransmission timeout. As for ALOHA,
this timeout cannot be fixed, otherwise hosts could become synchronised and always retransmit at the same time.
Setting such a timeout is always a compromise between the network access delay and the amount of collisions. A
short timeout would lead to a low network access delay but with a higher risk of collisions. On the other hand,
a long timeout would cause a long network access delay but a lower risk of collisions. The binary exponential
back-off algorithm was introduced in CSMA/CD networks to solve this problem.
To understand binary exponential back-off, let us consider a collision caused by exactly two hosts. Once it has
detected the collision, a host can either retransmit its frame immediately or defer its transmission for some time.
If each colliding host flips a coin to decide whether to retransmit immediately or to defer its retransmission, four
cases are possible :
1. Both hosts retransmit immediately and a new collision occurs
2. The first host retransmits immediately and the second defers its retransmission
3. The second host retransmits immediately and the first defers its retransmission
4. Both hosts defer their retransmission and a new collision occurs
In the second and third cases, both hosts have flipped different coins. The delay chosen by the host that defers
its retransmission should be long enough to ensure that its retransmission will not collide with the immediate
retransmission of the other host. However the delay should not be longer than the time necessary to avoid the
collision, because if both hosts decide to defer their transmission, the network will be idle during this delay. The
1 This name should not be confused with the duration of a transmission slot in slotted ALOHA. In CSMA/CD networks, the slot time is
the time during which a collision can occur at the beginning of the transmission of a frame. In slotted ALOHA, the duration of a slot is the
transmission time of an entire fixed-size frame.

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slot time is the optimal delay since it is the shortest delay that ensures that the first host will be able to retransmit
its frame completely without any collision.
If two hosts are competing, the algorithm above will avoid a second collision 50% of the time. However, if the
network is heavily loaded, several hosts may be competing at the same time. In this case, the hosts should be able
to automatically adapt their retransmission delay. The binary exponential back-off performs this adaptation based
on the number of collisions that have affected a frame. After the first collision, the host flips a coin and waits
0 or 1 slot time. After the second collision, it generates a random number and waits 0, 1, 2 or 3 slot times, etc.
The duration of the waiting time is doubled after each collision. The complete pseudo-code for the CSMA/CD
algorithm is shown in the figure below.
# CSMA/CD pseudo-code
N=1
while N<= max :
wait(channel_becomes_free)
send(frame)
wait_until (end_of_frame) or (collision)
if collision detected:
stop transmitting
send(jamming)
k = min (10, N)
r = random(0, 2k - 1) * slotTime
wait(r*slotTime)
N=N+1
else :
wait(inter-frame_delay)
break
# end of while loop
# Too many transmission attempts

The inter-frame delay used in this pseudo-code is a short delay corresponding to the time required by a network
adapter to switch from transmit to receive mode. It is also used to prevent a host from sending a continuous
stream of frames without leaving any transmission opportunities for other hosts on the network. This contributes
to the fairness of CSMA/CD. Despite this delay, there are still conditions where CSMA/CD is not completely fair
[RY1994]. Consider for example a network with two hosts : a server sending long frames and a client sending
acknowledgments. Measurements reported in [RY1994] have shown that there are situations where the client
could suffer from repeated collisions that lead it to wait for long periods of time due to the exponential back-off
algorithm.

6.2.5 Carrier Sense Multiple Access with Collision Avoidance

The Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) Medium Access Control algorithm was
designed for the popular WiFi wireless network technology [802.11]. CSMA/CA also senses the transmission
channel before transmitting a frame. Furthermore, CSMA/CA tries to avoid collisions by carefully tuning the
timers used by CSMA/CA devices.
CSMA/CA uses acknowledgements like CSMA. Each frame contains a sequence number and a CRC. The CRC
is used to detect transmission errors while the sequence number is used to avoid frame duplication. When a
device receives a correct frame, it returns a special acknowledgement frame to the sender. CSMA/CA introduces
a small delay, named Short Inter Frame Spacing (SIFS), between the reception of a frame and the transmission of
the acknowledgement frame. This delay corresponds to the time that is required to switch the radio of a device
between the reception and transmission modes.
Compared to CSMA, CSMA/CA defines more precisely when a device is allowed to send a frame. First,
CSMA/CA defines two delays : DIFS and EIFS. To send a frame, a device must first wait until the channel
has been idle for at least the Distributed Coordination Function Inter Frame Space (DIFS) if the previous frame
was received correctly. However, if the previously received frame was corrupted, this indicates that there are
collisions and the device must sense the channel idle for at least the Extended Inter Frame Space (EIFS), with
SIF S < DIF S < EIF S. The exact values for SIFS, DIFS and EIFS depend on the underlying physical layer
[802.11].

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The figure below shows the basic operation of CSMA/CA devices. Before transmitting, host A verifies that the
channel is empty for a long enough period. Then, its sends its data frame. After checking the validity of the
received frame, the recipient sends an acknowledgement frame after a short SIFS delay. Host C, which does not
participate in the frame exchange, senses the channel to be busy at the beginning of the data frame. Host C can
use this information to determine how long the channel will be busy for. Note that as SIF S < DIF S < EIF S,
even a device that would start to sense the channel immediately after the last bit of the data frame could not decide
to transmit its own frame during the transmission of the acknowledgement frame.

Figure 6.9: Operation of a CSMA/CA device

The main difficulty with CSMA/CA is when two or more devices transmit at the same time and cause collisions.
This is illustrated in the figure below, assuming a fixed timeout after the transmission of a data frame. With
CSMA/CA, the timeout after the transmission of a data frame is very small, since it corresponds to the SIFS plus
the time required to transmit the acknowledgement frame.

Figure 6.10: Collisions with CSMA/CA

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To deal with this problem, CSMA/CA relies on a backoff timer. This backoff timer is a random delay that is
chosen by each device in a range that depends on the number of retransmissions for the current frame. The
range grows exponentially with the retransmissions as in CSMA/CD. The minimum range for the backoff timer
is [0, 7 slotT ime] where the slotTime is a parameter that depends on the underlying physical layer. Compared
to CSMA/CDs exponential backoff, there are two important differences to notice. First, the initial range for
the backoff timer is seven times larger. This is because it is impossible in CSMA/CA to detect collisions as
they happen. With CSMA/CA, a collision may affect the entire frame while with CSMA/CD it can only affect
the beginning of the frame. Second, a CSMA/CA device must regularly sense the transmission channel during
its back off timer. If the channel becomes busy (i.e. because another device is transmitting), then the back off
timer must be frozen until the channel becomes free again. Once the channel becomes free, the back off timer
is restarted. This is in contrast with CSMA/CD where the back off is recomputed after each collision. This is
illustrated in the figure below. Host A chooses a smaller backoff than host C. When C senses the channel to be
busy, it freezes its backoff timer and only restarts it once the channel is free again.

Figure 6.11: Detailed example with CSMA/CA

The pseudo-code below summarises the operation of a CSMA/CA device. The values of the SIFS, DIFS, EIFS
and slotTime depend on the underlying physical layer technology [802.11]
# CSMA/CA simplified pseudo-code
N=1
while N<= max :
waitUntil(free(channel))
if correct(last_frame) :
wait(channel_free_during_t >=DIFS)
else:
wait(channel_free_during_t >=EIFS)
back-off_time = int(random[0,min(255,7*(2^(N-1)))])*slotTime
wait(channel free during backoff_time)
# backoff timer is frozen while channel is sensed to be busy
send(frame)
wait(ack or timeout)
if received(ack)
# frame received correctly
break
else:
# retransmission required
N=N+1

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Another problem faced by wireless networks is often called the hidden station problem. In a wireless network,
radio signals are not always propagated same way in all directions. For example, two devices separated by a wall
may not be able to receive each others signal while they could both be receiving the signal produced by a third
host. This is illustrated in the figure below, but it can happen in other environments. For example, two devices that
are on different sides of a hill may not be able to receive each others signal while they are both able to receive the
signal sent by a station at the top of the hill. Furthermore, the radio propagation conditions may change with time.
For example, a truck may temporarily block the communication between two nearby devices.

Figure 6.12: The hidden station problem

To avoid collisions in these situations, CSMA/CA allows devices to reserve the transmission channel for some
time. This is done by using two control frames : Request To Send (RTS) and Clear To Send (CTS). Both are very
short frames to minimize the risk of collisions. To reserve the transmission channel, a device sends a RTS frame
to the intended recipient of the data frame. The RTS frame contains the duration of the requested reservation. The
recipient replies, after a SIFS delay, with a CTS frame which also contains the duration of the reservation. As the
duration of the reservation has been sent in both RTS and CTS, all hosts that could collide with either the sender
or the reception of the data frame are informed of the reservation. They can compute the total duration of the
transmission and defer their access to the transmission channel until then. This is illustrated in the figure below
where host A reserves the transmission channel to send a data frame to host B. Host C notices the reservation and
defers its transmission.

Figure 6.13: Reservations with CSMA/CA

The utilization of the reservations with CSMA/CA is an optimisation that is useful when collisions are frequent.
If there are few collisions, the time required to transmit the RTS and CTS frames can become significant and in
particular when short frames are exchanged. Some devices only turn on RTS/CTS after transmission errors.

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6.2.6 Deterministic Medium Access Control algorithms

During the 1970s and 1980s, there were huge debates in the networking community about the best suited Medium
Access Control algorithms for Local Area Networks. The optimistic algorithms that we have described until
now were relatively easy to implement when they were designed. From a performance perspective, mathematical
models and simulations showed the ability of these optimistic techniques to sustain load. However, none of the
optimistic techniques are able to guarantee that a frame will be delivered within a given delay bound and some
applications require predictable transmission delays. The deterministic MAC algorithms were considered by a
fraction of the networking community as the best solution to fulfill the needs of Local Area Networks.
Both the proponents of the deterministic and the opportunistic techniques lobbied to develop standards for Local
Area networks that would incorporate their solution. Instead of trying to find an impossible compromise between
these diverging views, the IEEE 802 committee that was chartered to develop Local Area Network standards
chose to work in parallel on three different LAN technologies and created three working groups. The IEEE
802.3 working group became responsible for CSMA/CD. The proponents of deterministic MAC algorithms agreed
on the basic principle of exchanging special frames called tokens between devices to regulate the access to the
transmission medium. However, they did not agree on the most suitable physical layout for the network. IBM
argued in favor of Ring-shaped networks while the manufacturing industry, led by General Motors, argued in
favor of a bus-shaped network. This led to the creation of the IEEE 802.4 working group to standardise Token Bus
networks and the IEEE 802.5 working group to standardise Token Ring networks. Although these techniques are
not widely used anymore today, the principles behind a token-based protocol are still important.
The IEEE 802.5 Token Ring technology is defined in [802.5]. We use Token Ring as an example to explain the
principles of the token-based MAC algorithms in ring-shaped networks. Other ring-shaped networks include the
almost defunct FDDI [Ross1989] or the more recent Resilient Pack Ring [DYGU2004] . A good survey of the
token ring networks may be found in [Bux1989] .
A Token Ring network is composed of a set of stations that are attached to a unidirectional ring. The basic principle
of the Token Ring MAC algorithm is that two types of frames travel on the ring : tokens and data frames. When the
Token Ring starts, one of the stations sends the token. The token is a small frame that represents the authorization
to transmit data frames on the ring. To transmit a data frame on the ring, a station must first capture the token by
removing it from the ring. As only one station can capture the token at a time, the station that owns the token can
safely transmit a data frame on the ring without risking collisions. After having transmitted its frame, the station
must remove it from the ring and resend the token so that other stations can transmit their own frames.

Figure 6.14: A Token Ring network

While the basic principles of the Token Ring are simple, there are several subtle implementation details that add
complexity to Token Ring networks. To understand these details let us analyse the operation of a Token Ring
interface on a station. A Token Ring interface serves three different purposes. Like other LAN interfaces, it must
be able to send and receive frames. In addition, a Token Ring interface is part of the ring, and as such, it must be
able to forward the electrical signal that passes on the ring even when its station is powered off.
When powered-on, Token Ring interfaces operate in two different modes : listen and transmit. When operating
in listen mode, a Token Ring interface receives an electrical signal from its upstream neighbour on the ring,
introduces a delay equal to the transmission time of one bit on the ring and regenerates the signal before sending
it to its downstream neighbour on the ring.
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The first problem faced by a Token Ring network is that as the token represents the authorization to transmit, it
must continuously travel on the ring when no data frame is being transmitted. Let us assume that a token has been
produced and sent on the ring by one station. In Token Ring networks, the token is a 24 bits frame whose structure
is shown below.

Figure 6.15: 802.5 token format

The token is composed of three fields. First, the Starting Delimiter is the marker that indicates the beginning of a
frame. The first Token Ring networks used Manchester coding and the Starting Delimiter contained both symbols
representing 0 and symbols that do not represent bits. The last field is the Ending Delimiter which marks the end
of the token. The Access Control field is present in all frames, and contains several flags. The most important is
the Token bit that is set in token frames and reset in other frames.
Let us consider the five station network depicted in figure A Token Ring network above and assume that station S1
sends a token. If we neglect the propagation delay on the inter-station links, as each station introduces a one bit
delay, the first bit of the frame would return to S1 while it sends the fifth bit of the token. If station S1 is powered
off at that time, only the first five bits of the token will travel on the ring. To avoid this problem, there is a special
station called the Monitor on each Token Ring. To ensure that the token can travel forever on the ring, this Monitor
inserts a delay that is equal to at least 24 bit transmission times. If station S3 was the Monitor in figure A Token
Ring network, S1 would have been able to transmit the entire token before receiving the first bit of the token from
its upstream neighbour.
Now that we have explained how the token can be forwarded on the ring, let us analyse how a station can capture
a token to transmit a data frame. For this, we need some information about the format of the data frames. An
802.5 data frame begins with the Starting Delimiter followed by the Access Control field whose Token bit is reset,
a Frame Control field that allows for the definition of several types of frames, destination and source address, a
payload, a CRC, the Ending Delimiter and a Frame Status field. The format of the Token Ring data frames is
illustrated below.

Figure 6.16: 802.5 data frame format

To capture a token, a station must operate in Listen mode. In this mode, the station receives bits from its upstream
neighbour. If the bits correspond to a data frame, they must be forwarded to the downstream neighbour. If they
correspond to a token, the station can capture it and transmit its data frame. Both the data frame and the token
are encoded as a bit string beginning with the Starting Delimiter followed by the Access Control field. When the
station receives the first bit of a Starting Delimiter, it cannot know whether this is a data frame or a token and
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must forward the entire delimiter to its downstream neighbour. It is only when it receives the fourth bit of the
Access Control field (i.e. the Token bit) that the station knows whether the frame is a data frame or a token. If
the Token bit is reset, it indicates a data frame and the remaining bits of the data frame must be forwarded to the
downstream station. Otherwise (Token bit is set), this is a token and the station can capture it by resetting the
bit that is currently in its buffer. Thanks to this modification, the beginning of the token is now the beginning of
a data frame and the station can switch to Transmit mode and send its data frame starting at the fifth bit of the
Access Control field. Thus, the one-bit delay introduced by each Token Ring station plays a key role in enabling
the stations to efficiently capture the token.
After having transmitted its data frame, the station must remain in Transmit mode until it has received the last bit
of its own data frame. This ensures that the bits sent by a station do not remain in the network forever. A data
frame sent by a station in a Token Ring network passes in front of all stations attached to the network. Each station
can detect the data frame and analyse the destination address to possibly capture the frame.
The Frame Status field that appears after the Ending Delimiter is used to provide acknowledgements without
requiring special frames. The Frame Status contains two flags : A and C. Both flags are reset when a station sends
a data frame. These flags can be modified by their recipients. When a station senses its address as the destination
address of a frame, it can capture the frame, check its CRC and place it in its own buffers. The destination of
a frame must set the A bit (resp. C bit) of the Frame Status field once it has seen (resp. copied) a data frame.
By inspecting the Frame Status of the returning frame, the sender can verify whether its frame has been received
correctly by its destination.
The text above describes the basic operation of a Token Ring network when all stations work correctly. Unfortunately, a real Token Ring network must be able to handle various types of anomalies and this increases the
complexity of Token Ring stations. We briefly list the problems and outline their solutions below. A detailed
description of the operation of Token Ring stations may be found in [802.5]. The first problem is when all the
stations attached to the network start. One of them must bootstrap the network by sending the first token. For this,
all stations implement a distributed election mechanism that is used to select the Monitor. Any station can become
a Monitor. The Monitor manages the Token Ring network and ensures that it operates correctly. Its first role is to
introduce a delay of 24 bit transmission times to ensure that the token can travel smoothly on the ring. Second,
the Monitor sends the first token on the ring. It must also verify that the token passes regularly. According to
the Token Ring standard [802.5], a station cannot retain the token to transmit data frames for a duration longer
than the Token Holding Time (THT) (slightly less than 10 milliseconds). On a network containing N stations, the
Monitor must receive the token at least every N T HT seconds. If the Monitor does not receive a token during
such a period, it cuts the ring for some time and then reinitialises the ring and sends a token.
Several other anomalies may occur in a Token Ring network. For example, a station could capture a token and
be powered off before having resent the token. Another station could have captured the token, sent its data frame
and be powered off before receiving all of its data frame. In this case, the bit string corresponding to the end of a
frame would remain in the ring without being removed by its sender. Several techniques are defined in [802.5] to
allow the Monitor to handle all these problems. If unfortunately, the Monitor fails, another station will be elected
to become the new Monitor.

6.3 Datalink layer technologies

In this section, we review the key characteristics of several datalink layer technologies. We discuss in more detail
the technologies that are widely used today. A detailed survey of all datalink layer technologies would be outside
the scope of this book.

6.3.1 The Point-to-Point Protocol

Many point-to-point datalink layers 2 have been developed, starting in the 1960s [McFadyen1976]. In this section,
we focus on the protocols that are often used to transport IP packets between hosts or routers that are directly
connected by a point-to-point link. This link can be a dedicated physical cable, a leased line through the telephone
network or a dial-up connection with modems on the two communicating hosts.
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LAPB and HDLC were widely used datalink layer protocols.

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The first solution to transport IP packets over a serial line was proposed in RFC 1055 and is known as Serial
Line IP (SLIP). SLIP is a simple character stuffing technique applied to IP packets. SLIP defines two special
characters : END (decimal 192) and ESC (decimal 219). END appears at the beginning and at the end of each
transmitted IP packet and the sender adds ESC before each END character inside each transmitted IP packet.
SLIP only supports the transmission of IP packets and it assumes that the two communicating hosts/routers have
been manually configured with each others IP address. SLIP was mainly used over links offering bandwidth of
often less than 20 Kbps. On such a low bandwidth link, sending 20 bytes of IP header followed by 20 bytes of
TCP header for each TCP segment takes a lot of time. This initiated the development of a family of compression
techniques to efficiently compress the TCP/IP headers. The first header compression technique proposed in RFC
1144 was designed to exploit the redundancy between several consecutive segments that belong to the same TCP
connection. In all these segments, the IP addresses and port numbers are always the same. Furthermore, fields
such as the sequence and acknowledgement numbers do not change in a random way. RFC 1144 defined simple
techniques to reduce the redundancy found in successive segments. The development of header compression
techniques continued and there are still improvements being developed now RFC 5795.
While SLIP was implemented and used in some environments, it had several limitations discussed in RFC 1055.
The Point-to-Point Protocol (PPP) was designed shortly after and is specified in RFC 1548. PPP aims to support
IP and other network layer protocols over various types of serial lines. PPP is in fact a family of three protocols
that are used together :
1. The Point-to-Point Protocol defines the framing technique to transport network layer packets.
2. The Link Control Protocol that is used to negotiate options and authenticate the session by using username
and password or other types of credentials
3. The Network Control Protocol that is specific for each network layer protocol. It is used to negotiate options
that are specific for each protocol. For example, IPv4s NCP RFC 1548 can negotiate the IPv4 address to
be used, the IPv4 address of the DNS resolver. IPv6s NCP is defined in RFC 5072.
The PPP framing RFC 1662 was inspired by the datalink layer protocols standardised by ITU-T and ISO. A typical
PPP frame is composed of the fields shown in the figure below. A PPP frame starts with a one byte flag containing
01111110. PPP can use bit stuffing or character stuffing depending on the environment where the protocol is used.
The address and control fields are present for backward compatibility reasons. The 16 bit Protocol field contains
the identifier 3 of the network layer protocol that is carried in the PPP frame. 0x002d is used for an IPv4 packet
compressed with RFC 1144 while 0x002f is used for an uncompressed IPv4 packet. 0xc021 is used by the Link
Control Protocol, 0xc023 is used by the Password Authentication Protocol (PAP). 0x0057 is used for IPv6 packets.
PPP supports variable length packets, but LCP can negotiate a maximum packet length. The PPP frame ends with
a Frame Check Sequence. The default is a 16 bits CRC, but some implementations can negotiate a 32 bits CRC.
The frame ends with the 01111110 flag.

Figure 6.17: PPP frame format

PPP played a key role in allowing Internet Service Providers to provide dial-up access over modems in the late
1990s and early 2000s. ISPs operated modem banks connected to the telephone network. For these ISPs, a key
issue was to authenticate each user connected through the telephone network. This authentication was performed
by using the Extensible Authentication Protocol (EAP) defined in RFC 3748. EAP is a simple, but extensible
protocol that was initially used by access routers to authenticate the users connected through dialup lines. Several
authentication methods, starting from the simple username/password pairs to more complex schemes have been
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defined and implemented. When ISPs started to upgrade their physical infrastructure to provide Internet access
over Asymmetric Digital Subscriber Lines (ADSL), they tried to reuse their existing authentication (and billing)
systems. To meet these requirements, the IETF developed specifications to allow PPP frames to be transported over
other networks than the point-to-point links for which PPP was designed. Nowadays, most ADSL deployments
use PPP over either ATM RFC 2364 or Ethernet RFC 2516.

6.3.2 Ethernet
Ethernet was designed in the 1970s at the Palo Alto Research Center [Metcalfe1976]. The first prototype 4 used
a coaxial cable as the shared medium and 3 Mbps of bandwidth. Ethernet was improved during the late 1970s
and in the 1980s, Digital Equipment, Intel and Xerox published the first official Ethernet specification [DIX]. This
specification defines several important parameters for Ethernet networks. The first decision was to standardise
the commercial Ethernet at 10 Mbps. The second decision was the duration of the slot time. In Ethernet, a long
slot time enables networks to span a long distance but forces the host to use a larger minimum frame size. The
compromise was a slot time of 51.2 microseconds, which corresponds to a minimum frame size of 64 bytes.
The third decision was the frame format. The experimental 3 Mbps Ethernet network built at Xerox used short
frames containing 8 bit source and destination addresses fields, a 16 bit type indication, up to 554 bytes of payload
and a 16 bit CRC. Using 8 bit addresses was suitable for an experimental network, but it was clearly too small
for commercial deployments. Although the initial Ethernet specification [DIX] only allowed up to 1024 hosts on
an Ethernet network, it also recommended three important changes compared to the networking technologies that
were available at that time. The first change was to require each host attached to an Ethernet network to have a
globally unique datalink layer address. Until then, datalink layer addresses were manually configured on each host.
[DP1981] went against that state of the art and noted Suitable installation-specific administrative procedures are
also needed for assigning numbers to hosts on a network. If a host is moved from one network to another it may
be necessary to change its host number if its former number is in use on the new network. This is easier said than
done, as each network must have an administrator who must record the continuously changing state of the system
(often on a piece of paper tacked to the wall !). It is anticipated that in future office environments, hosts locations
will change as often as telephones are changed in present-day offices. The second change introduced by Ethernet
was to encode each address as a 48 bits field [DP1981]. 48 bit addresses were huge compared to the networking
technologies available in the 1980s, but the huge address space had several advantages [DP1981] including the
ability to allocate large blocks of addresses to manufacturers. Eventually, other LAN technologies opted for 48 bits
addresses as well [802]_ . The third change introduced by Ethernet was the definition of broadcast and multicast
addresses. The need for multicast Ethernet was foreseen in [DP1981] and thanks to the size of the addressing
space it was possible to reserve a large block of multicast addresses for each manufacturer.
The datalink layer addresses used in Ethernet networks are often called MAC addresses. They are structured as
shown in the figure below. The first bit of the address indicates whether the address identifies a network adapter
or a multicast group. The upper 24 bits are used to encode an Organisation Unique Identifier (OUI). This OUI
identifies a block of addresses that has been allocated by the secretariat 5 who is responsible for the uniqueness
of Ethernet addresses to a manufacturer. Once a manufacturer has received an OUI, it can build and sell products
with one of the 16 million addresses in this block.

Figure 6.18: 48 bits Ethernet address format

The original 10 Mbps Ethernet specification [DIX] defined a simple frame format where each frame is composed
of five fields. The Ethernet frame starts with a preamble (not shown in the figure below) that is used by the physical
layer of the receiver to synchronise its clock with the senders clock. The first field of the frame is the destination
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Additional information about the history of the Ethernet technology may be found at http://ethernethistory.typepad.com/
Initially, the OUIs were allocated by Xerox [DP1981]. However, once Ethernet became an IEEE and later an ISO standard, the allocation
of the OUIs moved to IEEE. The list of all OUI allocations may be found at http://standards.ieee.org/regauth/oui/index.shtml
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address. As this address is placed at the beginning of the frame, an Ethernet interface can quickly verify whether
it is the frame recipient and if not, cancel the processing of the arriving frame. The second field is the source
address. While the destination address can be either a unicast or a multicast/broadcast address, the source address
must always be a unicast address. The third field is a 16 bits integer that indicates which type of network layer
packet is carried inside the frame. This field is often called the EtherType. Frequently used EtherType values 6
include 0x0800 for IPv4, 0x86DD for IPv6 7 and 0x806 for the Address Resolution Protocol (ARP).
The fourth part of the Ethernet frame is the payload. The minimum length of the payload is 46 bytes to ensure a
minimum frame size, including the header of 512 bits. The Ethernet payload cannot be longer than 1500 bytes.
This size was found reasonable when the first Ethernet specification was written. At that time, Xerox had been
using its experimental 3 Mbps Ethernet that offered 554 bytes of payload and RFC 1122 required a minimum
MTU of 572 bytes for IPv4. 1500 bytes was large enough to support these needs without forcing the network
adapters to contain overly large memories. Furthermore, simulations and measurement studies performed in
Ethernet networks revealed that CSMA/CD was able to achieve a very high utilization. This is illustrated in the
figure below based on [SH1980], which shows the channel utilization achieved in Ethernet networks containing
different numbers of hosts that are sending frames of different sizes.

Figure 6.19: Impact of the frame length on the maximum channel utilisation [SH1980]
The last field of the Ethernet frame is a 32 bit Cyclical Redundancy Check (CRC). This CRC is able to catch a
much larger number of transmission errors than the Internet checksum used by IP, UDP and TCP [SGP98]. The
format of the Ethernet frame is shown below.
Note: Where should the CRC be located in a frame ?
The transport and datalink layers usually chose different strategies to place their CRCs or checksums. Transport
layer protocols usually place their CRCs or checksums in the segment header. Datalink layer protocols sometimes
place their CRC in the frame header, but often in a trailer at the end of the frame. This choice reflects implementation assumptions, but also influences performance RFC 893. When the CRC is placed in the trailer, as in Ethernet,
the datalink layer can compute it while transmitting the frame and insert it at the end of the transmission. All Ethernet interfaces use this optimisation today. When the checksum is placed in the header, as in a TCP segment,
it is impossible for the network interface to compute it while transmitting the segment. Some network interfaces
provide hardware assistance to compute the TCP checksum, but this is more complex than if the TCP checksum
were placed in the trailer 8 .
6

The official list of all assigned Ethernet type values is available from http://standards.ieee.org/regauth/ethertype/eth.txt
The attentive reader may question the need for different EtherTypes for IPv4 and IPv6 while the IP header already contains a version
field that can be used to distinguish between IPv4 and IPv6 packets. Theoretically, IPv4 and IPv6 could have used the same EtherType.
Unfortunately, developers of the early IPv6 implementations found that some devices did not check the version field of the IPv4 packets that
they received and parsed frames whose EtherType was set to 0x0800 as IPv4 packets. Sending IPv6 packets to such devices would have caused
disruptions. To avoid this problem, the IETF decided to apply for a distinct EtherType value for IPv6.
8 These network interfaces compute the TCP checksum while a segment is transferred from the host memory to the network interface
7

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Figure 6.20: Ethernet DIX frame format

The Ethernet frame format shown above is specified in [DIX]. This is the format used to send both IPv4 RFC 894
and IPv6 packets RFC 2464. After the publication of [DIX], the Institute of Electrical and Electronic Engineers
(IEEE) began to standardise several Local Area Network technologies. IEEE worked on several LAN technologies, starting with Ethernet, Token Ring and Token Bus. These three technologies were completely different, but
they all agreed to use the 48 bits MAC addresses specified initially for Ethernet [802]_ . While developing its
Ethernet standard [802.3], the IEEE 802.3 working group was confronted with a problem. Ethernet mandated a
minimum payload size of 46 bytes, while some companies were looking for a LAN technology that could transparently transport short frames containing only a few bytes of payload. Such a frame can be sent by an Ethernet
host by padding it to ensure that the payload is at least 46 bytes long. However since the Ethernet header [DIX]
does not contain a length field, it is impossible for the receiver to determine how many useful bytes were placed
inside the payload field. To solve this problem, the IEEE decided to replace the Type field of the Ethernet [DIX]
header with a length field 9 . This Length field contains the number of useful bytes in the frame payload. The payload must still contain at least 46 bytes, but padding bytes are added by the sender and removed by the receiver.
In order to add the Length field without significantly changing the frame format, IEEE had to remove the Type
field. Without this field, it is impossible for a receiving host to identify the type of network layer packet inside a
received frame. To solve this new problem, IEEE developed a completely new sublayer called the Logical Link
Control [802.2]. Several protocols were defined in this sublayer. One of them provided a slightly different version
of the Type field of the original Ethernet frame format. Another contained acknowledgements and retransmissions
to provide a reliable service... In practice, [802.2] is never used to support IP in Ethernet networks. The figure
below shows the official [802.3] frame format.

Figure 6.21: Ethernet 802.3 frame format

Note: What is the Ethernet service ?

An Ethernet network provides an unreliable connectionless service. It supports three different transmission modes
[SH2004].
9 Fortunately, IEEE was able to define the [802.3] frame format while maintaining backward compatibility with the Ethernet [DIX] frame
format. The trick was to only assign values above 1500 as EtherType values. When a host receives a frame, it can determine whether the
frames format by checking its EtherType/Length field. A value lower smaller than 1501 is clearly a length indicator and thus an [802.3] frame.
A value larger than 1501 can only be type and thus a [DIX] frame.

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[unicast, multicast and broadcast. While the Ethernet service is unreliable in theory, a good Ethernet
network should, in practice, provide a service that :]
delivers frames to their destination with a very high probability of successful delivery
does not reorder the transmitted frames
The first property is a consequence of the utilisation of CSMA/CD. The second property is a consequence of
the physical organisation of the Ethernet network as a shared bus. These two properties are important and all
evolutions of the Ethernet technology have preserved them.
Several physical layers have been defined for Ethernet networks. The first physical layer, usually called 10Base5,
provided 10 Mbps over a thick coaxial cable. The characteristics of the cable and the transceivers that were used
then enabled the utilisation of 500 meter long segments. A 10Base5 network can also include repeaters between
segments.
The second physical layer was 10Base2. This physical layer used a thin coaxial cable that was easier to install
than the 10Base5 cable, but could not be longer than 185 meters. A 10BaseF physical layer was also defined
to transport Ethernet over point-to-point optical links. The major change to the physical layer was the support
of twisted pairs in the 10BaseT specification. Twisted pair cables are traditionally used to support the telephone
service in office buildings. Most office buildings today are equipped with structured cabling. Several twisted pair
cables are installed between any room and a central telecom closet per building or per floor in large buildings.
These telecom closets act as concentration points for the telephone service but also for LANs.
The introduction of the twisted pairs led to two major changes to Ethernet. The first change concerns the physical
topology of the network. 10Base2 and 10Base5 networks are shared buses, the coaxial cable typically passes
through each room that contains a connected computer. A 10BaseT network is a star-shaped network. All the
devices connected to the network are attached to a twisted pair cable that ends in the telecom closet. From
a maintenance perspective, this is a major improvement. The cable is a weak point in 10Base2 and 10Base5
networks. Any physical damage on the cable broke the entire network and when such a failure occurred, the
network administrator had to manually check the entire cable to detect where it was damaged. With 10BaseT,
when one twisted pair is damaged, only the device connected to this twisted pair is affected and this does not
affect the other devices. The second major change introduced by 10BaseT was that is was impossible to build a
10BaseT network by simply connecting all the twisted pairs together. All the twisted pairs must be connected to
a relay that operates in the physical layer. This relay is called an Ethernet hub. A hub is thus a physical layer
relay that receives an electrical signal on one of its interfaces, regenerates the signal and transmits it over all its
other interfaces. Some hubs are also able to convert the electrical signal from one physical layer to another (e.g.
10BaseT to 10Base2 conversion).

Figure 6.22: Ethernet hubs in the reference model

Computers can directly be attached to Ethernet hubs. Ethernet hubs themselves can be attached to other Ethernet
hubs to build a larger network. However, some important guidelines must be followed when building a complex
network with hubs. First, the network topology must be a tree. As hubs are relays in the physical layer, adding
a link between Hub2 and Hub3 in the network below would create an electrical shortcut that would completely
disrupt the network. This implies that there cannot be any redundancy in a hub-based network. A failure of a
hub or of a link between two hubs would partition the network into two isolated networks. Second, as hubs are
relays in the physical layer, collisions can happen and must be handled by CSMA/CD as in a 10Base5 network.
This implies that the maximum delay between any pair of devices in the network cannot be longer than the 51.2

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microseconds slot time. If the delay is longer, collisions between short frames may not be correctly detected. This
constraint limits the geographical spread of 10BaseT networks containing hubs.

Figure 6.23: A hierarchical Ethernet network composed of hubs

In the late 1980s, 10 Mbps became too slow for some applications and network manufacturers developed several
LAN technologies that offered higher bandwidth, such as the 100 Mbps FDDI LAN that used optical fibers. As the
development of 10Base5, 10Base2 and 10BaseT had shown that Ethernet could be adapted to different physical
layers, several manufacturers started to work on 100 Mbps Ethernet and convinced IEEE to standardise this new
technology that was initially called Fast Ethernet. Fast Ethernet was designed under two constraints. First,
Fast Ethernet had to support twisted pairs. Although it was easier from a physical layer perspective to support
higher bandwidth on coaxial cables than on twisted pairs, coaxial cables were a nightmare from deployment
and maintenance perspectives. Second, Fast Ethernet had to be perfectly compatible with the existing 10 Mbps
Ethernets to allow Fast Ethernet technology to be used initially as a backbone technology to interconnect 10
Mbps Ethernet networks. This forced Fast Ethernet to use exactly the same frame format as 10 Mbps Ethernet.
This implied that the minimum Fast Ethernet frame size remained at 512 bits. To preserve CSMA/CD with this
minimum frame size and 100 Mbps instead of 10 Mbps, the duration of the slot time was decreased to 5.12
microseconds.
The evolution of Ethernet did not stop. In 1998, the IEEE published a first standard to provide Gigabit Ethernet
over optical fibers. Several other types of physical layers were added afterwards. The 10 Gigabit Ethernet standard
appeared in 2002. Work is ongoing to develop standards for 40 Gigabit and 100 Gigabit Ethernet and some are
thinking about Terabit Ethernet. The table below lists the main Ethernet standards. A more detailed list may be
found at http://en.wikipedia.org/wiki/Ethernet_physical_layer
Standard
10Base5
10Base2
10BaseT
10Base-F
100Base-Tx
100Base-FX
1000Base-CX
1000Base-SX
10 Gbps
40-100 Gbps

Comments
Thick coaxial cable, 500m
Thin coaxial cable, 185m
Two pairs of category 3+ UTP
10 Mb/s over optical fiber
Category 5 UTP or STP, 100 m maximum
Two multimode optical fiber, 2 km maximum
Two pairs shielded twisted pair, 25m maximum
Two multimode or single mode optical fibers with lasers
Optical fiber but also Category 6 UTP
Being developed, standard expected in 2010

Ethernet Switches
Increasing the physical layer bandwidth as in Fast Ethernet was only one of the solutions to improve the performance of Ethernet LANs. A second solution was to replace the hubs with more intelligent devices. As Ethernet
hubs operate in the physical layer, they can only regenerate the electrical signal to extend the geographical reach

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of the network. From a performance perspective, it would be more interesting to have devices that operate in the
datalink layer and can analyse the destination address of each frame and forward the frames selectively on the link
that leads to the destination. Such devices are usually called Ethernet switches 10 . An Ethernet switch is a relay
that operates in the datalink layer as is illustrated in the figure below.

Figure 6.24: Ethernet switches and the reference model

An Ethernet switch understands the format of the Ethernet frames and can selectively forward frames over each
interface. For this, each Ethernet switch maintains a MAC address table. This table contains, for each MAC
address known by the switch, the identifier of the switchs port over which a frame sent towards this address must
be forwarded to reach its destination. This is illustrated below with the MAC address table of the bottom switch.
When the switch receives a frame destined to address B, it forwards the frame on its South port. If it receives a
frame destined to address D, it forwards it only on its North port.

Figure 6.25: Operation of Ethernet switches

One of the selling points of Ethernet networks is that, thanks to the utilisation of 48 bits MAC addresses, an
Ethernet LAN is plug and play at the datalink layer. When two hosts are attached to the same Ethernet segment
or hub, they can immediately exchange Ethernet frames without requiring any configuration. It is important to
retain this plug and play capability for Ethernet switches as well. This implies that Ethernet switches must be
able to build their MAC address table automatically without requiring any manual configuration. This automatic
configuration is performed by the the MAC address learning algorithm that runs on each Ethernet switch. This
algorithm extracts the source address of the received frames and remembers the port over which a frame from
10 The first Ethernet relays that operated in the datalink layers were called bridges. In practice, the main difference between switches and
bridges is that bridges were usually implemented in software while switches are hardware-based devices. Throughout this text, we always use
switch when referring to a relay in the datalink layer, but you might still see the word bridge.

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each source Ethernet address has been received. This information is inserted into the MAC address table that the
switch uses to forward frames. This allows the switch to automatically learn the ports that it can use to reach each
destination address, provided that this host has previously sent at least one frame. This is not a problem since
most upper layer protocols use acknowledgements at some layer and thus even an Ethernet printer sends Ethernet
frames as well.
The pseudo-code below details how an Ethernet switch forwards Ethernet frames. It first updates its MAC address
table with the source address of the frame. The MAC address table used by some switches also contains a
timestamp that is updated each time a frame is received from each known source address. This timestamp is
used to remove from the MAC address table entries that have not been active during the last n minutes. This limits
the growth of the MAC address table, but also allows hosts to move from one port to another. The switch uses its
MAC address table to forward the received unicast frame. If there is an entry for the frames destination address
in the MAC address table, the frame is forwarded selectively on the port listed in this entry. Otherwise, the switch
does not know how to reach the destination address and it must forward the frame on all its ports except the port
from which the frame has been received. This ensures that the frame will reach its destination, at the expense of
some unnecessary transmissions. These unnecessary transmissions will only last until the destination has sent its
first frame. Multicast and Broadcast frames are also forwarded in a similar way.
# Arrival of frame F on port P
# Table : MAC address table dictionary : addr->port
# Ports : list of all ports on the switch
src=F.SourceAddress
dst=F.DestinationAddress
Table[src]=P #src heard on port P
if isUnicast(dst) :
if dst in Table:
ForwardFrame(F,Table[dst])
else:
for o in Ports :
if o!= P : ForwardFrame(F,o)
else:
# multicast or broadcast destination
for o in Ports :
if o!= P : ForwardFrame(F,o)

Note: Security issues with Ethernet hubs and switches

From a security perspective, Ethernet hubs have the same drawbacks as the older coaxial cable. A host attached to
a hub will be able to capture all the frames exchanged between any pair of hosts attached to the same hub. Ethernet
switches are much better from this perspective thanks to the selective forwarding, a host will usually only receive
the frames destined to itself as well as the multicast, broadcast and unknown frames. However, this does not imply
that switches are completely secure. There are, unfortunately, attacks against Ethernet switches. From a security
perspective, the MAC address table is one of the fragile elements of an Ethernet switch. This table has a fixed
size. Some low-end switches can store a few tens or a few hundreds of addresses while higher-end switches can
store tens of thousands of addresses or more. From a security point of view, a limited resource can be the target
of Denial of Service attacks. Unfortunately, such attacks are also possible on Ethernet switches. A malicious
host could overflow the MAC address table of the switch by generating thousands of frames with random source
addresses. Once the MAC address table is full, the switch needs to broadcast all the frames that it receives. At
this point, an attacker will receive unicast frames that are not destined to its address. The ARP attack discussed in
the previous chapter could also occur with Ethernet switches [Vyncke2007]. Recent switches implement several
types of defences against these attacks, but they need to be carefully configured by the network administrator. See
[Vyncke2007] for a detailed discussion on security issues with Ethernet switches.
The MAC address learning algorithm combined with the forwarding algorithm work well in a tree-shaped network
such as the one shown above. However, to deal with link and switch failures, network administrators often add
redundant links to ensure that their network remains connected even after a failure. Let us consider what happens
in the Ethernet network shown in the figure below.
When all switches boot, their MAC address table is empty. Assume that host A sends a frame towards host C.
Upon reception of this frame, switch1 updates its MAC address table to remember that address A is reachable

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Figure 6.26: Ethernet switches in a loop

via its West port. As there is no entry for address C in switch1s MAC address table, the frame is forwarded to
both switch2 and switch3. When switch2 receives the frame, its updates its MAC address table for address A
and forwards the frame to host C as well as to switch3. switch3 has thus received two copies of the same frame.
As switch3 does not know how to reach the destination address, it forwards the frame received from switch1 to
switch2 and the frame received from switch2 to switch1... The single frame sent by host A will be continuously
duplicated by the switches until their MAC address table contains an entry for address C. Quickly, all the available
link bandwidth will be used to forward all the copies of this frame. As Ethernet does not contain any TTL or
HopLimit, this loop will never stop.
The MAC address learning algorithm allows switches to be plug-and-play. Unfortunately, the loops that arise
when the network topology is not a tree are a severe problem. Forcing the switches to only be used in tree-shaped
networks as hubs would be a severe limitation. To solve this problem, the inventors of Ethernet switches have
developed the Spanning Tree Protocol. This protocol allows switches to automatically disable ports on Ethernet
switches to ensure that the network does not contain any cycle that could cause frames to loop forever.
The Spanning Tree Protocol (802.1d)
The Spanning Tree Protocol (STP), proposed in [Perlman1985], is a distributed protocol that is used by switches
to reduce the network topology to a spanning tree, so that there are no cycles in the topology. For example,
consider the network shown in the figure below. In this figure, each bold line corresponds to an Ethernet to which
two Ethernet switches are attached. This network contains several cycles that must be broken to allow Ethernet
switches that are using the MAC address learning algorithm to exchange frames.
In this network, the STP will compute the following spanning tree. Switch1 will be the root of the tree. All the
interfaces of Switch1, Switch2 and Switch7 are part of the spanning tree. Only the interface connected to LANB
will be active on Switch9. LANH will only be served by Switch7 and the port of Switch44 on LANG will be
disabled. A frame originating on LANB and destined for LANA will be forwarded by Switch7 on LANC, then by
Switch1 on LANE, then by Switch44 on LANF and eventually by Switch2 on LANA.
Switches running the Spanning Tree Protocol exchange BPDUs. These BPDUs are always sent as frames with
destination MAC address as the ALL_BRIDGES reserved multicast MAC address. Each switch has a unique 64
bit identifier. To ensure uniqueness, the lower 48 bits of the identifier are set to the unique MAC address allocated
to the switch by its manufacturer. The high order 16 bits of the switch identifier can be configured by the network
administrator to influence the topology of the spanning tree. The default value for these high order bits is 32768.

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Figure 6.27: Spanning tree computed in a switched Ethernet network

The switches exchange BPDUs to build the spanning tree. Intuitively, the spanning tree is built by first selecting
the switch with the smallest identifier as the root of the tree. The branches of the spanning tree are then composed
of the shortest paths that allow all of the switches that compose the network to be reached. The BPDUs exchanged
by the switches contain the following information :
the identifier of the root switch (R)
the cost of the shortest path between the switch that sent the BPDU and the root switch (c)
the identifier of the switch that sent the BPDU (T)
the number of the switch port over which the BPDU was sent (p)
We will use the notation <R,c,T,p> to represent a BPDU whose root identifier is R, cost is c and that was sent
on the port p of switch T. The construction of the spanning tree depends on an ordering relationship among the
BPDUs. This ordering relationship could be implemented by the python function below.
# returns True if bpdu b1 is better than bpdu b2
def better( b1, b2) :
return ( (b1.R < b2.R) or
( (b1.R==b2.R) and (b1.c<b2.c) ) or
( (b1.R==b2.R) and (b1.c==b2.c) and (b1.T<b2.T) ) or
( (b1.R==b2.R) and (b1.c==b2.c) and (b1.T==b2.T) and (b1.p<b2.p) ) )

In addition to the identifier discussed above, the network administrator can also configure a cost to be associated
to each switch port. Usually, the cost of a port depends on its bandwidth and the [802.1d] standard recommends
the values below. Of course, the network administrator may choose other values. We will use the notation cost[p]
to indicate the cost associated to port p in this section.
Bandwidth
10 Mbps
100 Mbps
1 Gbps
10 Gbps
100 Gbps

Cost
2000000
200000
20000
2000
200

The Spanning Tree Protocol uses its own terminology that we illustrate in the figure above. A switch port can be
in three different states : Root, Designated and Blocked. All the ports of the root switch are in the Designated
state. The state of the ports on the other switches is determined based on the BPDU received on each port.
The Spanning Tree Protocol uses the ordering relationship to build the spanning tree. Each switch listens to
BPDUs on its ports. When BPDU=<R,c,T,p> is received on port q, the switch computes the ports priority
vector: V[q]=<R,c+cost[q],T,p,q> , where cost[q] is the cost associated to the port over which the BPDU was
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received. The switch stores in a table the last priority vector received on each port. The switch then compares its
own identifier with the smallest root identifier stored in this table. If its own identifier is smaller, then the switch
is the root of the spanning tree and is, by definition, at a distance 0 of the root. The BPDU of the switch is then
<R,0,R,p>, where R is the switch identifier and p will be set to the port number over which the BPDU is sent.
Otherwise, the switch chooses the best priority vector from its table, bv=<R,c,T,p>. The port over which this best
priority vector was learned is the switch port that is closest to the root switch. This port becomes the Root port of
the switch. There is only one Root port per switch. The switch can then compute its BPDU as BPDU=<R,c,S,p>
, where R is the root identifier, c the cost of the best priority vector, S the identifier of the switch and p will be
replaced by the number of the port over which the BPDU will be sent. The switch can then determine the state
of all its ports by comparing its own BPDU with the priority vector received on each port. If the switchs BPDU
is better than the priority vector of this port, the port becomes a Designated port. Otherwise, the port becomes a
Blocked port.
The state of each port is important when considering the transmission of BPDUs. The root switch regularly sends
its own BPDU over all of its (Designated) ports. This BPDU is received on the Root port of all the switches that
are directly connected to the root switch. Each of these switches computes its own BPDU and sends this BPDU
over all its Designated ports. These BPDUs are then received on the Root port of downstream switches, which
then compute their own BPDU, etc. When the network topology is stable, switches send their own BPDU on
all their Designated ports, once they receive a BPDU on their Root port. No BPDU is sent on a Blocked port.
Switches listen for BPDUs on their Blocked and Designated ports, but no BPDU should be received over these
ports when the topology is stable. The utilisation of the ports for both BPDUs and data frames is summarised in
the table below.
Port state
Blocked
Root
Designated

Receives BPDUs
yes
yes
yes

Sends BPDU
no
no
yes

Handles data frames

no
yes
yes

To illustrate the operation of the Spanning Tree Protocol, let us consider the simple network topology in the figure
below.

Figure 6.28: A simple Spanning tree computed in a switched Ethernet network

Assume that Switch4 is the first to boot. It sends its own BPDU=<4,0,4,?> on its two ports. When Switch1
boots, it sends BPDU=<1,0,1,1>. This BPDU is received by Switch4, which updates its table and computes a
new BPDU=<1,3,4,?>. Port 1 of Switch4 becomes the Root port while its second port is still in the Designated
state.
Assume now that Switch9 boots and immediately receives Switch1 s BPDU on port 1. Switch9 computes its own
BPDU=<1,1,9,?> and port 1 becomes the Root port of this switch. This BPDU is sent on port 2 of Switch9 and

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reaches Switch4. Switch4 compares the priority vector built from this BPDU (i.e. <1,2,9,2>) and notices that it is
better than Switch4 s BPDU=<1,3,4,2>. Thus, port 2 becomes a Blocked port on Switch4.
During the computation of the spanning tree, switches discard all received data frames, as at that time the network
topology is not guaranteed to be loop-free. Once that topology has been stable for some time, the switches again
start to use the MAC learning algorithm to forward data frames. Only the Root and Designated ports are used
to forward data frames. Switches discard all the data frames received on their Blocked ports and never forward
frames on these ports.
Switches, ports and links can fail in a switched Ethernet network. When a failure occurs, the switches must be
able to recompute the spanning tree to recover from the failure. The Spanning Tree Protocol relies on regular
transmissions of the BPDUs to detect these failures. A BPDU contains two additional fields : the Age of the
BPDU and the Maximum Age. The Age contains the amount of time that has passed since the root switch initially
originated the BPDU. The root switch sends its BPDU with an Age of zero and each switch that computes its own
BPDU increments its Age by one. The Age of the BPDUs stored on a switchs table is also incremented every
second. A BPDU expires when its Age reaches the Maximum Age. When the network is stable, this does not
happen as BPDU s are regularly sent by the root switch and downstream switches. However, if the root fails or
the network becomes partitioned, BPDU will expire and switches will recompute their own BPDU and restart the
Spanning Tree Protocol. Once a topology change has been detected, the forwarding of the data frames stops as
the topology is not guaranteed to be loop-free. Additional details about the reaction to failures may be found in
[802.1d]
Virtual LANs
Another important advantage of Ethernet switches is the ability to create Virtual Local Area Networks (VLANs).
A virtual LAN can be defined as a set of ports attached to one or more Ethernet switches. A switch can support
several VLANs and it runs one MAC learning algorithm for each Virtual LAN. When a switch receives a frame
with an unknown or a multicast destination, it forwards it over all the ports that belong to the same Virtual LAN
but not over the ports that belong to other Virtual LANs. Similarly, when a switch learns a source address on a
port, it associates it to the Virtual LAN of this port and uses this information only when forwarding frames on this
Virtual LAN.
The figure below illustrates a switched Ethernet network with three Virtual LANs. VLAN2 and VLAN3 only require
a local configuration of switch S1. Host C can exchange frames with host D, but not with hosts that are outside of
its VLAN. VLAN1 is more complex as there are ports of this VLAN on several switches. To support such VLANs,
local configuration is not sufficient anymore. When a switch receives a frame from another switch, it must be able
to determine the VLAN in which the frame originated to use the correct MAC table to forward the frame. This
is done by assigning an identifier to each Virtual LAN and placing this identifier inside the headers of the frames
that are exchanged between switches.

Figure 6.29: Virtual Local Area Networks in a switched Ethernet network

IEEE defined in the [802.1q] standard a special header to encode the VLAN identifiers. This 32 bit header
includes a 20 bit VLAN field that contains the VLAN identifier of each frame. The format of the [802.1q] header
is described below.
The [802.1q] header is inserted immediately after the source MAC address in the Ethernet frame (i.e. before the
EtherType field). The maximum frame size is increased by 4 bytes. It is encoded in 32 bits and contains four

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Figure 6.30: Format of the 802.1q header

fields. The Tag Protocol Identifier is set to 0x8100 to allow the receiver to detect the presence of this additional
header. The Priority Code Point (PCP) is a three bit field that is used to support different transmission priorities
for the frame. Value 0 is the lowest priority and value 7 the highest. Frames with a higher priority can expect to
be forwarded earlier than frames having a lower priority. The C bit is used for compatibility between Ethernet and
Token Ring networks. The last 12 bits of the 802.1q header contain the VLAN identifier. Value 0 indicates that
the frame does not belong to any VLAN while value 0xFFF is reserved. This implies that 4094 different VLAN
identifiers can be used in an Ethernet network.

6.3.3 802.11 wireless networks

The radio spectrum is a limited resource that must be shared by everyone. During most of the twentieth century,
governments and international organisations have regulated most of the radio spectrum. This regulation controls
the utilisation of the radio spectrum, in order to ensure that there are no interferences between different users.
A company that wants to use a frequency range in a given region must apply for a license from the regulator.
Most regulators charge a fee for the utilisation of the radio spectrum and some governments have encouraged
competition among companies bidding for the same frequency to increase the license fees.
In the 1970s, after the first experiments with ALOHANet, interest in wireless networks grew. Many experiments
were done on and outside the ARPANet. One of these experiments was the first mobile phone , which was developed and tested in 1973. This experimental mobile phone was the starting point for the first generation analog
mobile phones. Given the growing demand for mobile phones, it was clear that the analog mobile phone technology was not sufficient to support a large number of users. To support more users and new services, researchers in
several countries worked on the development of digital mobile telephones. In 1987, several European countries
decided to develop the standards for a common cellular telephone system across Europe : the Global System for
Mobile Communications (GSM). Since then, the standards have evolved and more than three billion users are
connected to GSM networks today.
While most of the frequency ranges of the radio spectrum are reserved for specific applications and require a
special licence, there are a few exceptions. These exceptions are known as the Industrial, Scientific and Medical
(ISM) radio bands. These bands can be used for industrial, scientific and medical applications without requiring
a licence from the regulator. For example, some radio-controlled models use the 27 MHz ISM band and some
cordless telephones operate in the 915 MHz ISM. In 1985, the 2.400-2.500 GHz band was added to the list of
ISM bands. This frequency range corresponds to the frequencies that are emitted by microwave ovens. Sharing
this band with licensed applications would have likely caused interferences, given the large number of microwave
ovens that are used. Despite the risk of interferences with microwave ovens, the opening of the 2.400-2.500 GHz
allowed the networking industry to develop several wireless network techniques to allow computers to exchange
data without using cables. In this section, we discuss in more detail the most popular one, i.e. the WiFi [802.11]
family of wireless networks. Other wireless networking techniques such as BlueTooth or HiperLAN use the same
frequency range.
Today, WiFi is a very popular wireless networking technology. There are more than several hundreds of millions
of WiFi devices. The development of this technology started in the late 1980s with the WaveLAN proprietary
wireless network. WaveLAN operated at 2 Mbps and used different frequency bands in different regions of the
world. In the early 1990s, the IEEE created the 802.11 working group to standardise a family of wireless network
technologies. This working group was very prolific and produced several wireless networking standards that use
different frequency ranges and different physical layers. The table below provides a summary of the main 802.11
standards.

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Standard
802.11
802.11a
802.11b
802.11g
802.11n

Frequency
2.4 GHz
5 GHz
2.4 GHz
2.4 GHz
2.4/5 GHz

Typical throughput
0.9 Mbps
23 Mbps
4.3 Mbps
19 Mbps
74 Mbps

Max bandwidth
2 Mbps
54 Mbps
11 Mbps
54 Mbps
150 Mbps

Range (m) indoor/outdoor

20/100
35/120
38/140
38/140
70/250

When developing its family of standards, the IEEE 802.11 working group took a similar approach as the IEEE
802.3 working group that developed various types of physical layers for Ethernet networks. 802.11 networks use
the CSMA/CA Medium Access Control technique described earlier and they all assume the same architecture and
use the same frame format.
The architecture of WiFi networks is slightly different from the Local Area Networks that we have discussed until
now. There are, in practice, two main types of WiFi networks : independent or adhoc networks and infrastructure
networks 11 . An independent or adhoc network is composed of a set of devices that communicate with each other.
These devices play the same role and the adhoc network is usually not connected to the global Internet. Adhoc
networks are used when for example a few laptops need to exchange information or to connect a computer with a
WiFi printer.

Figure 6.31: An 802.11 independent or adhoc network

Most WiFi networks are infrastructure networks. An infrastructure network contains one or more access points
that are attached to a fixed Local Area Network (usually an Ethernet network) that is connected to other networks
such as the Internet. The figure below shows such a network with two access points and four WiFi devices. Each
WiFi device is associated to one access point and uses this access point as a relay to exchange frames with the
devices that are associated to another access point or reachable through the LAN.
An 802.11 access point is a relay that operates in the datalink layer like switches. The figure below represents the
layers of the reference model that are involved when a WiFi host communicates with a host attached to an Ethernet
network through an access point.
802.11 devices exchange variable length frames, which have a slightly different structure than the simple frame
format used in Ethernet LANs. We review the key parts of the 802.11 frames. Additional details may be found
in [802.11] and [Gast2002] . An 802.11 frame contains a fixed length header, a variable length payload that
may contain up 2324 bytes of user data and a 32 bits CRC. Although the payload can contain up to 2324 bytes,
11 The 802.11 working group defined the basic service set (BSS) as a group of devices that communicate with each other. We continue to
use network when referring to a set of devices that communicate.

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Figure 6.32: An 802.11 infrastructure network

Figure 6.33: An 802.11 access point

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most 802.11 deployments use a maximum payload size of 1500 bytes as they are used in infrastructure networks
attached to Ethernet LANs. An 802.11 data frame is shown below.

Figure 6.34: 802.11 data frame format

The first part of the 802.11 header is the 16 bit Frame Control field. This field contains flags that indicate the
type of frame (data frame, RTS/CTS, acknowledgement, management frames, etc), whether the frame is sent to
or from a fixed LAN, etc [802.11]. The Duration is a 16 bit field that is used to reserve the transmission channel.
In data frames, the Duration field is usually set to the time required to transmit one acknowledgement frame after
a SIFS delay. Note that the Duration field must be set to zero in multicast and broadcast frames. As these frames
are not acknowledged, there is no need to reserve the transmission channel after their transmission. The Sequence
control field contains a 12 bits sequence number that is incremented for each data frame.
The astute reader may have noticed that the 802.11 data frames contain three 48-bits address fields 12 . This is
surprising compared to other protocols in the network and datalink layers whose headers only contain a source and
a destination address. The need for a third address in the 802.11 header comes from the infrastructure networks. In
such a network, frames are usually exchanged between routers and servers attached to the LAN and WiFi devices
attached to one of the access points. The role of the three address fields is specified by bit flags in the Frame
Control field.
When a frame is sent from a WiFi device to a server attached to the same LAN as the access point, the first address
of the frame is set to the MAC address of the access point, the second address is set to the MAC address of the
source WiFi device and the third address is the address of the final destination on the LAN. When the server
replies, it sends an Ethernet frame whose source address is its MAC address and the destination address is the
MAC address of the WiFi device. This frame is captured by the access point that converts the Ethernet header into
an 802.11 frame header. The 802.11 frame sent by the access point contains three addresses : the first address is
the MAC address of the destination WiFi device, the second address is the MAC address of the access point and
the third address the MAC address of the server that sent the frame.
802.11 control frames are simpler than data frames. They contain a Frame Control, a Duration field and one or
two addresses. The acknowledgement frames are very small. They only contain the address of the destination of
the acknowledgement. There is no source address and no Sequence Control field in the acknowledgement frames.
This is because the acknowledgement frame can easily be associated to the previous frame that it acknowledges.
Indeed, each unicast data frame contains a Duration field that is used to reserve the transmission channel to ensure
that no collision will affect the acknowledgement frame. The Sequence Control field is mainly used by the receiver
to remove duplicate frames. Duplicate frames are detected as follows. Each data frame contains a 12 bits Sequence
Control field and the Frame Control field contains the Retry bit flag that is set when a frame is transmitted. Each
802.11 receiver stores the most recent sequence number received from each source address in frames whose Retry
bit is reset. Upon reception of a frame with the Retry bit set, the receiver verifies its sequence number to determine
whether it is a duplicated frame or not.
802.11 RTS/CTS frames are used to reserve the transmission channel, in order to transmit one data frame and its
acknowledgement. The RTS frames contain a Duration and the transmitter and receiver addresses. The Duration
12 In fact, the [802.11] frame format contains a fourth optional address field. This fourth address is only used when an 802.11 wireless
network is used to interconnect bridges attached to two classical LAN networks.

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Figure 6.35: IEEE 802.11 ACK and CTS frames

field of the RTS frame indicates the duration of the entire reservation (i.e. the time required to transmit the CTS,
the data frame, the acknowledgements and the required SIFS delays). The CTS frame has the same format as the
acknowledgement frame.

Figure 6.36: IEEE 802.11 RTS frame format

Note: The 802.11 service

Despite the utilization of acknowledgements, the 802.11 layer only provides an unreliable connectionless service
like Ethernet networks that do not use acknowledgements. The 802.11 acknowledgements are used to minimize
the probability of frame duplication. They do not guarantee that all frames will be correctly received by their
recipients. Like Ethernet, 802.11 networks provide a high probability of successful delivery of the frames, not a
guarantee. Furthermore, it should be noted that 802.11 networks do not use acknowledgements for multicast and
broadcast frames. This implies that in practice such frames are more likely to suffer from transmission errors than
unicast frames.
In addition to the data and control frames that we have briefly described above, 802.11 networks use several types
of management frames. These management frames are used for various purposes. We briefly describe some of
these frames below. A detailed discussion may be found in [802.11] and [Gast2002].
A first type of management frames are the beacon frames. These frames are broadcasted regularly by access
points. Each beacon frame contains information about the capabilities of the access point (e.g. the supported
802.11 transmission rates) and a Service Set Identity (SSID). The SSID is a null-terminated ASCII string that can
contain up to 32 characters. An access point may support several SSIDs and announce them in beacon frames. An
access point may also choose to remain silent and not advertise beacon frames. In this case, WiFi stations may
send Probe request frames to force the available access points to return a Probe response frame.
Note: IP over 802.11
Two types of encapsulation schemes were defined to support IP in Ethernet networks : the original encapsulation
scheme, built above the Ethernet DIX format is defined in RFC 894 and a second encapsulation RFC 1042
scheme, built above the LLC/SNAP protocol [802.2]. In 802.11 networks, the situation is simpler and only the
RFC 1042 encapsulation is used. In practice, this encapsulation adds 6 bytes to the 802.11 header. The first four
bytes correspond to the LLC/SNAP header. They are followed by the two bytes Ethernet Type field (0x800 for IP
and 0x806 for ARP). The figure below shows an IP packet encapsulated in an 802.11 frame.
The second important utilisation of the management frames is to allow a WiFi station to be associated with an
access point. When a WiFi station starts, it listens to beacon frames to find the available SSIDs. To be allowed to
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Figure 6.37: IP over IEEE 802.11

send and receive frames via an access point, a WiFi station must be associated to this access point. If the access
point does not use any security mechanism to secure the wireless transmission, the WiFi station simply sends an
Association request frame to its preferred access point (usually the access point that it receives with the strongest
radio signal). This frame contains some parameters chosen by the WiFi station and the SSID that it requests to
join. The access point replies with an Association response frame if it accepts the WiFI station.

6.4 Summary
In this chapter, we first explained the principles of the datalink layer. There are two types of datalink layers : those
used over point-to-point links and those used over Local Area Networks. On point-to-point links, the datalink layer
must at least provide a framing technique, but some datalink layer protocols also include reliability mechanisms
such as those used in the transport layer. We have described the Point-to-Point Protocol that is often used over
point-to-point links in the Internet.
Local Area Networks pose a different problem since several devices share the same transmission channel. In this
case, a Medium Access Control algorithm is necessary to regulate the access to the transmission channel because
whenever two devices transmit at the same time a collision occurs and none of these frames can be decoded
by their recipients. There are two families of MAC algorithms. The statistical or optimistic MAC algorithms
reduce the probability of collisions but do not completely prevent them. With such algorithms, when a collision
occurs, the collided frames must be retransmitted. We have described the operation of the ALOHA, CSMA,
CSMA/CD and CSMA/CA MAC algorithms. Deterministic or pessimistic MAC algorithms avoid all collisions.
We have described the Token Ring MAC where stations exchange a token to regulate the access to the transmission
channel.
Finally, we have described in more detail two successful Local Area Network technologies : Ethernet and WiFi.
Ethernet is now the de facto LAN technology. We have analysed the evolution of Ethernet including the operation
of hubs and switches. We have also described the Spanning Tree Protocol that must be used when switches
are interconnected. For the last few years, WiFi became the de facto wireless technology at home and inside
enterprises. We have explained the operation of WiFi networks and described the main 802.11 frames.

6.5 Exercises
1. Consider the switched network shown in the figure below. What is the spanning tree that will be computed
by 802.1d in this network assuming that all links have a unit cost ? Indicate the state of each port.
2. Consider the switched network shown in the figure above. In this network, assume that the LAN between
switches 3 and 12 fails. How should the switches update their port/address tables after the link failure ?
3. Many enterprise networks are organized with a set of backbone devices interconnected by using a full mesh
of links as shown in the figure below. In this network, what are the benefits and drawbacks of using Ethernet

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Figure 6.38: A small network composed of Ethernet switches

switches and IP routers running OSPF ?

Figure 6.39: A typical enterprise backbone network

4. Most commercial Ethernet switches are able to run the Spanning tree protocol independently on each
VLAN. What are the benefits of using per-VLAN spanning trees ?

6.5. Exercises

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CHAPTER 7

Glossary

AIMD Additive Increase, Multiplicative Decrease. A rate adaption algorithm used notably by TCP where a host
additively increases its transmission rate when the network is not congested and multiplicatively decreases
when congested is detected.
anycast a transmission mode where an information is sent from one source to one receiver that belongs to a
specified group
API Application Programming Interface
ARP The Address Resolution Protocol is a protocol used by IPv4 devices to obtain the datalink layer address
that corresponds to an IPv4 address on the local area network. ARP is defined in RFC 826
ARPANET The Advanced Research Project Agency (ARPA) Network is a network that was built by network
scientists in USA with funding from the ARPA of the US Ministry of Defense. ARPANET is considered as
the grandfather of todays Internet.
ascii The American Standard Code for Information Interchange (ASCII) is a character-encoding scheme that
defines a binary representation for characters. The ASCII table contains both printable characters and
control characters. ASCII characters were encoded in 7 bits and only contained the characters required to
write text in English. Other character sets such as Unicode have been developed later to support all written
languages.
ASN.1 The Abstract Syntax Notation One (ASN.1) was designed by ISO and ITU-T. It is a standard and flexible
notation that can be used to describe data structures for representing, encoding, transmitting, and decoding
data between applications. It was designed to be used in the Presentation layer of the OSI reference model
but is now used in other protocols such as SNMP.
ATM Asynchronous Transfer Mode
BGP The Border Gateway Protocol is the interdomain routing protocol used in the global Internet.
BNF A Backus-Naur Form (BNF) is a formal way to describe a language by using syntactic and lexical rules.
BNFs are frequently used to define programming languages, but also to define the messages exchanged
between networked applications. RFC 5234 explains how a BNF must be written to specify an Internet
protocol.
broadcast a transmission mode where is same information is sent to all nodes in the network
CIDR Classless Inter Domain Routing is the current address allocation architecture for IPv4. It was defined in
RFC 1518 and RFC 4632.
dial-up line A synonym for a regular telephone line, i.e. a line that can be used to dial any telephone number.
DNS The Domain Name System is a distributed database that allows to map names on IP addresses.
DNS The Domain Name System is defined in RFC 1035

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Plea

for

Peace,

IEN

137,

April

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Bibliography

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Index

Symbols

beacon frame (802.11), 245

BGP, 178, 249
::, 160
BGP Adj-RIB-In, 181
::1, 160
BGP Adj-RIB-Out, 181
0.0.0.0, 146, 155
BGP decision process, 187
100BaseTX, 234
BGP KEEPALIVE, 180
10Base2, 233
BGP local-preference, 187
10Base5, 233
BGP nexthop, 183
10BaseT, 233
BGP NOTIFICATION, 180
127.0.0.1, 146
BGP OPEN, 180
255.255.255.255, 155
BGP peer, 179
802.11 frame format, 242
BGP RIB, 181
802.5 data frame, 227
BGP UPDATE, 180
802.5 token frame, 227
binary exponential back-off (CSMA/CD), 221
bit stuffing, 212
A
BNF, 249
abrupt connection release, 18, 86
Additive Increase Multiplicative Decrease (AIMD), 109 Border Gateway Protocol, 178
bridge, 234
address, 11
broadcast, 249
address learning, 235
BSS, 242
Address Resolution Protocol, 154
adhoc network, 242
AF_INET, 56
AF_INET6, 56
AF_UNSPEC, 56
AIMD, 249
ALG, 169
ALOHA, 216
Alternating Bit Protocol, 72
anycast, 249
API, 249
Application layer, 23
Application Level Gateway, 169
ARP, 154, 249
ARP cache, 154
ARPANET, 249
ascii, 249
ASN.1, 249
ATM, 249

B
Base64 encoding, 41
Basic Service Set (BSS), 242

C
Carrier Sense Multiple Access, 217
Carrier Sense Multiple Access with Collision Avoidance, 222
Carrier Sense Multiple Access with Collision Detection, 218
character stuffing, 213
Checksum computation, 88
CIDR, 249
Class A IPv4 address, 142
Class B IPv4 address, 142
Class C IPv4 address, 142
Classless Interdomain Routing, 144
Clear To Send, 225
Cold potato routing, 191
collision, 215
collision detection, 219
collision domain, 233
confirmed connectionless service, 13
congestion collapse, 107
connection establishment, 16
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connection-oriented service, 16
Connectionless service, 12
connectionless service, 13
count to infinity, 135
CSMA, 217
CSMA (non-persistent), 218
CSMA (persistent), 217
CSMA/CA, 222
CSMA/CD, 218
CTS, 225
CTS frame (802.11), 244
cumulative acknowledgements, 76
customer-provider peering relationship, 177

framing, 212
Frequency Division Multiplexing, 215
FTP, 250
ftp, 250

G
getaddrinfo, 56
go-back-n, 75
graceful connection release, 19, 86

H
Hello message, 137
hidden station problem, 225
hop-by-hop forwarding, 129
hosts.txt, 32, 250
Hot potato routing, 190
HTML, 250
HTTP, 250
hub, 250

datagram, 129
Datalink layer, 22
delayed acknowledgements, 102
Denial of Service, 93
DHCP, 155
DHCPv6, 166
I
dial-up line, 249
IANA, 250
DIFS, 222
iBGP, 185, 250
Distance vector, 133
ICANN, 250
Distributed Coordination Function Inter Frame Space, ICMP, 150
222
IETF, 250
DNS, 249
IGP, 250
DNS message format, 34
IGRP, 250
Dynamic Host Configuration Protocol, 155
IMAP, 250
import policy, 178
E
independent network, 242
EAP, 229
infrastructure network, 242
eBGP, 185, 250
interdomain routing policy, 178
EGP, 250
Internet, 250
EIFS, 222
internet, 250
EIGRP, 250
Internet Control Message Protocol, 150
electrical cable, 20
inverse query, 250
email message format, 38
IP, 250
Ending Delimiter (Token Ring), 227
IP options, 150
Ethernet bridge, 234
IP prefix, 142
Ethernet DIX frame format, 231
IP subnet, 142
Ethernet hub, 233
IPv4, 250
Ethernet switch, 234
IPv4 fragmentation and reassembly, 149
Ethernet Type field, 230
IPv6, 251
EtherType, 230
IPv6 fragmentation, 163
exponential backoff, 102
IS-IS, 251
export policy, 178
ISN, 251
Extended Inter Frame Space, 222
ISO, 251
Extensible Authentication Protocol, 229
ISO-3166, 251
ISP, 251
F
ITU, 251
Fairness, 107
IXP, 251
Fast Ethernet, 234
J
FDM, 215
firewall, 166
jamming, 219
Five layers reference model, 20
jumbogram, 162
frame, 22, 250
Frame-Relay, 250
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Index

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L
label, 131
LAN, 251
large window, 99
leased line, 251
Link Local address, 160
link local IPv4 addresses, 146
link-state routing, 137
LLC, 232
Logical Link Control (LLC), 232
loopback interface, 185

M
MAC address learning, 235
MAC address table (Ethernet switch), 235
MAN, 251
man-in-the-middle attack, 155
Manchester encoding, 212
max-min fairness, 107
Maximum Segment Lifetime (MSL), 83
maximum segment lifetime (MSL), 79
Maximum Segment Size, 94
Maximum Transmission Unit, 148
Maximum Transmission Unit (MTU), 149
message-mode data transfer, 17
Middlebox, 166
MIME, 251
MIME document, 251
minicomputer, 251
modem, 251
Monitor station, 228
monomode optical fiber, 20
MSS, 94, 251
MTU, 148
Multi-Exit Discriminator (MED), 191
multicast, 251
multihomed host, 142
multihomed network, 145
multimode optical fiber, 20

TCP selective acknowledgements, 104
TCP self clocking, 105
TCP SYN, 90
TCP SYN+ACK, 90
TCP/IP, 252
TCP/IP reference model, 23
telnet, 253
Tier-1 ISP, 194
Time Division Multiplexing, 216
Time To Live (IP), 147
time-sequence diagram, 12
TLD, 253
TLS, 253
Token Holding Time, 228
Token Ring data frame, 227
Token Ring Monitor, 227
Token Ring token frame, 227
traceroute, 153
traceroute6, 165
transit domain, 175
Transmission Control Block, 97
transport clock, 83
Transport layer, 23
two-way connectivity, 140

U
UDP, 86, 253
UDP Checksum, 88
UDP segment, 87
unicast, 253
Unique Local Unicast IPv6, 160
unreliable connectionless service, 13

V
virtual circuit, 129
Virtual LAN, 240
VLAN, 240
vnc, 253

W
W3C, 253
WAN, 253
Wavelength Division Multiplexing, 215
WDM, 215
WiFi, 241

X
X.25, 253
X11, 253
XML, 253

278