COURSE

GUIDE

CIT 423
COMPUTER NETWORKS AND COMMUNICATION

Course Team

NATIONAL OPEN UNIVERSITY OF NIGERIA

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© 2022 by NOUN Press
National Open University of Nigeria
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e-mail: [email protected]
URL: www.nou.edu.ng

All rights reserved. No part of this book may be reproduced, in any

form or by any means, without permission in writing from the publisher.

Printed 2022

ISBN:

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CONTENT PAGE

Introduction……………………………………………….………… iv
What you will Learn in this Course................................................ iv
Working through the Course........................................................... iv
Assessment ..................................................................................... iv
How to Get the Most from this Course ............................................ iv
Summary ......................................................................................... v

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Introduction

The aim and objective of computer networks as a course is computer

literacy. Information Technology is the frontier hero of the new century,
driven by ambition and full of courage, replicating itself like a virus and
sweeping all before it. So in order not to be marginalized one needs to get
involved as Information Technology influences virtually all the areas of
human endeavour.

What You will Learn in this Course

This course will give you in brief what you need to know in Computer
Science and Technology. At the end of the course you will be an expert of
some sort in Computer Science and Technology.

Aim of the Course

Each module, each unit, in the course contains notes as well as set exercises.
The set exercises are listed in form of a table. The table has two volumes.
The last column contains the heading “what you do” while the right column
is headed “comments/prompts”. The left column contains the steps that you
must follow. The right column serves as additional information.

Computers will be provided at study centres and will be equipped with the
required programs. If you have the means, buy your PC and the stipulated
software.

Each unit contains a Tutor-Marked Assignment, which must be done as

stipulated and handed to the tutor on schedule.

Working through this Course

To complete this course you are required to read each study unit, read the textbooks and read
other materials which may be provided by the National Open University of Nigeria.

Each unit contains self-assessment exercises and at certain points in the course you will be
required to submit assignments for assessment purposes. At the end of the course there is final
examination. Below you will find listed all the components of the course, what you have to do
and how you should allocate your time to each unit in order to complete the course on time
and successfully.

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This course demands that you spend a lot of time to study. My advice is that you optimise the
opportunity provided by the tutorial sessions where you have the opportunity of comparing
your knowledge with that of your colleagues.

The Course Materials

The main components of the course are:

1. The Course Guide

2. Study Units
3. References/Further Readings
4. Assignments
5. Presentation Schedule

Study Unit

The study units in this course are as follows:

Module 1: Introduction to Computer Networks

Unit 1: Network Classification and Reference Models
Unit 2: Network Structures

Module 2: Network Devices and Technology

Unit 1 Network Technology
Unit 2 Network Devices–I
Unit 3 Network Devices–II
Unit 4 Integrated Service Digital Network (ISDN)
Unit5 Asynchronous Transfer Mode (ATM)
Unit 6 Data Transmission and Multiplexing
Unit 7 Medium Access Control and Data Link Layer

Module 3: Network Administration

Unit1 Network Administration: Scope, Goals, Philosophy and Standards

Unit2 Network Protocols
Unit3 Network, Transport and Application Layers

Note: each unit consists of one or two weeks work and includes introduction, objectives,
reading materials, exercises, conclusion and summary, Tutor-Marked Assignment (TMAs),
references and other resources. The unit directs you to work on these exercises related
to required reading. In general, these exercises test you on the materials thereby assisting you
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to evaluate your progress and to reinforce your comprehension of the material. Together with
the TMAs these exercises will help you in achieving the stated learning objectives of the
individual units and of the course as a whole.

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Presentation Schedule

Your course materials have important dates for early and timely completion and submission
of your TMAs and attending tutorials. You should remember that you are required to
submit all your assignments by the stipulated time and date. You should guide against
falling behind in your work.

Assessment

There are three aspects to the assessment of the course. First is made up of self-assessment
exercises, second consists of the TMA and third is the written examination/end of course
examination.

You are advised to do the exercises. In tackling the assignments, you are expected to apply
information, knowledge and techniques you gathered during the course. The assignments
must be submitted to your facilitator for formal assessment in accordance with the
deadlines stated in the presentation schedule and the assignment file. The work you submit
to your tutor for assessment accounts for 30% of your total course work. At the end of the
course you will need to sit for a final examination or end of course examination of about
three hour duration. This examination will count for 70% of the total course mark.

Tutor-Marked Assignment

This is the continuous assessment component of your course. It accounts for 30% of the total
score. You will be given four TMAs (4) to answer. Three of these must be answered before
you are allowed to sit for the end of the course examination. The assignment questions for
the units in the course are contained in the assignment file. You will be able to complete
them through your reading the information contained in the reading materials,
references and the study units. You are advised to research deeper into topics so as
have a broader view of the discussions.

Endeavour to get the assignments to the facilitator on or before the deadline. If for any
reason you cannot complete the work on time, contact your facilitator before the
assignment is due to discuss the possibility of an extension. Extension will not be granted
after the due date has passed unless on exceptional circumstances.

Final Examination and Grading

The end of course examination for Network Administration will be for about 3 hours and it
has a value of 70% of the total course work. The examination will reflect the type of self-
testing, practice exercise and tutor-marked assignment problems you have previously
encountered. All these areas of the course will be assessed.

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Use the time between finishing the last unit and sitting for the examination to revise the
whole course. It might be useful to review your self tests, TMAs and the comments on
them before the course examination. The end of course examination covers information
from all parts of the course.

Course Marking Scheme

Facilitators/Tutors and
Tutorial
There are 21 hours of tutorials provided in support of this course. You will be notified
of the dates, times and venues of these tutorials as well as the name and phone
numbers of the facilitator, as soon as you are allocated to a tutorial group.

Your facilitator will mark and comment on your assignments, keep a close watch on
your progress and any difficulties you might face and provide assistance to you during
the course. You are expected to mail TMA to your facilitator at least two working days
before the schedule date. The TMAs will be marked by your tutor returned back to you as
soon as possible.

Do not delay to contact your facilitator by telephone or email if you need

assistance. The following might lead to your needing your facilitator’s

assistance:

You do not understand any part of the study or assigned reading

You have difficulty with the self test
You have a question or a problem with an assignment or with the grading
of an assignment

Endeavour to attend tutorials. It affords you the opportunity of face to face contact with
the facilitator and to ask questions which are answered instantly. You also raise problems
encountered in the course of study.

SUMMARY

Computers in Society intend to make you computer literate. At the end of the
course you will achieve the objective if you follow the instructions and do
what you are expected to do.

We wish you a huge success.

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CONTENTS PAGE

Module 1: Introduction to Computer Networks

Unit 1: Network Classification and Reference Models
Unit 2: Network Structures

Module 2: Network Devices and Technology

Unit 1 Network Technology
Unit 2 Network Devices–I
Unit 3 Network Devices–II
Unit 4 Integrated Service Digital Network (ISDN)
Unit5 Asynchronous Transfer Mode (ATM)
Unit 6 Data Transmission and Multiplexing
Unit 7 Medium Access Control and Data Link Layer

Module 3: Network Administration

Unit1 Network Administration: Scope, Goals, Philosophy and Standards

Unit2 Network Protocols
Unit3 Network, Transport and Application Layers

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MODULE 1 INTRODUCTION TO COMPUTER NETWORKS
UNIT 1: NETWORK CLASSIFICATION AND REFERENCE MODELS
UNIT 2: NETWORK STRUCTURE

UNIT 1: NETWORK CLASSIFICATION AND REFERENCE

MODELS

1.0 INTRODUCTION

Earlier computers used to be stand alone. Different computers were used

for information gathering, processing or distribution. Due to rapid
technological progress, the areas of information gathering, processing
and distribution are rapidly converging and differences between them
are quickly disappearing. In this unit, we will learn about the different
types of networks, their applications, networking models and topologies.
We will also examine references, the various layers and functions of
each layer.

2.0 OBJECTIVES

By the end of this unit, you will be able to:

 define and classify network

 distinguish between different types of networks
 understand what OSI model is, and TCP reference model and
functions of each layer.

3.0 MAIN CONTENT

3.1 What Is A Network

In the simplest form, data transfer can take place between two devices
which are directly connected by some form of communication medium.
But it is not practical for two devices to be directly Point–to–Point
connected. This is due to the following reasons:

(i) The devices are very far apart.

(ii) There is a set of devices, each of which may require to connect to
others at various times.

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 Solution to this problem is to connect each device to a communication
network. Computer network means interconnected set of autonomous
systems that permit distributed processing of information.

In order to meet the needs of various applications, networks are

available with different interconnection layouts and pLANs, method of
access, protocols and media. Networks can be classified on the basis of
geographical coverage.

3.1.1 Classification of Networks

 Local Area Network (LAN)

 Metropolitan Area Network (MAN)
 Wide Area Network (WAN)

3.1.2 Local Area Network (LAN)

A local area network is a relatively smaller and privately owned network with
maximum span of 10km to provide local connectivity within a building or small
geographical area. The LANs are distinguished from other kinds of networks by three
characteristics:

(i) Size
(ii) Transmission technology, and
(iii) Topology

Accordingly, there are many LAN standards known as IEEE area standards 802
x.

3.1.3 Metropolitan Area Network (MAN)

Metropolitan Area Network is defined as less than 50km and provides regional
connectivity typically within a campus or small geographical area. It is designed to
extend over an entire city. It may be a single network, such as cable television
network, or it may be a means of connecting a number of LANs into a large
network, so that resources may be shared LAN–to–LAN as well as device to device.
For example, a company can use a MAN to connect to the LANs in all of its offices
throughout a city.

3.1.4 Wide Area Network (WAN)

Wide Area Network provides no limit of distance. In most WAN, the subnet consists
of two distinct components. Transmission lines, also called circuits or channels, and
routers. Transmission lines are used for moving bits between machines, whereas
routers are used to connect two or more transmission lines

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A WAN provides long distance transmission of data, voice, image and video
information over large geographical areas that may comprise a country, a continent or
even the whole world. In contrast to LANs (which depend on their own hardware for
transmission), WANs may utilize public, leased or private
communication devices usually in combination, and span own unlimited number of
miles.

A WAN that is wholly owned by a single company is often referred to

as an enterprise network.

3.2 Computer Network Goals/Motivation

The main goal of a computer network is to enable its users to share

resources and to access these resources (i.e hard disks, high quality
expensive laser printer, modems, peripheral devices, licensed software.
etc.), regardless of their physical locations. Physical locations may be a
few feet or even thousands of miles apart, but users exchange data and
programs in the same way. In other words, distance is removed as a
barrier for the above application. The computer network thus creates a
global environment for its users and computers. Another goal is to
provide communication services (such as E–mail) and in general, to
provide robust transport network. i.e., (highway) over which application
can be built.

3.3 Applications of Networks

The following is the list of some applications of computer network.

Generic application

 Resource sharing (CPU, peripherals, information and software)

 Personal communication (text+graphics+audio+video)
 Network wide information discovery and retrieval.

We are now moving from personalized computing to network

computing. Therefore, its applications are increasing everyday.

3.4 Types of Network

There are basically two types of network based on whether the network
contains switching elements or not. These are Point–to–Point network
and Broadcast network.

3.5.1 Point–to–Point Network or Switch Network

Point–to–Point networks consist of many connections between

individual pairs of machines. To go from the to the source destination, a
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packet on this type of network may have to first visit one or more
intermediate machine routers. When a packet is sent from one router to
another intermediate router, the entire packet is stored at each
intermediate router, till the output line is free and then forwarded. A
subnet using this principle is called Point–to–Point or Packet switched
network.

Some possible topologies for a Point–to–Point subnet are:

Star

In a star topology, each device has a dedicated Point–to–Point link only to a central
controller, usually called a hub. These devices are not linked to each other. If one
device wants to send data to another, it sends to the hub which then relays the data
to the other connected devices. In a star, each device needs only one link and one
I/O Port to connect it to any number of other devices. This factor makes it
easy to install and configure. Far less cabling need to be housed and additions,
moves and deletions involve only one connection between that device and the hub.

Tree

A tree topology is a variation of a star. As in a star modes in a tree are linked to a

central hub that controls the traffic to the network. However, not every device plugs
directly into the central hub. The majority of devices connect to a secondary hub
that in turn is connected to the central hub.

The advantages and disadvantages of a tree topology are generally the same as
those of stars. The addition of secondary hubs, however, brings two further
advantages. First, it allows more devices to be attached to a single central hub and
can, therefore, increase the distance a signal can travel between devices. Second, it
isolates the network and prioritizes communication from different computers.

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Ring

In a ring topology, each device has a dedicated Point–to–Point line

configuration only, with the two devices on either side of it. A signal is
passed along the ring in one direction from device to device, until it
reaches its destination. Each device in the ring incorporates a repeater.
When a device receives a signal intended for another device, its repeater
regenerates the bits and passes them along.

A ring is relatively easy to install and reconfigure. Each device is linked

to its immediate neighbours. However, unidirectional traffic can be a
disadvantage. In a simple ring, a break in ring can disable the entire
network. This weakness can be solved by using a dual ring or switch
capable of closing off the break.

Bus

Bus, unlike other topologies, is a multi–point configuration. One long

cable acts as a backbone to link all the devices in the network.
Advantages of a bus topology include use of installation. A disadvantage
includes difficult reconfiguration and fault isolation.

3.4.2 Broadcast Networks

Broadcast networks have a single communication channel that is shared

by all the machines on the network. Short messages, called packets, sent
by any machine are received by all the others. An address field within
the packet specifies for when it is intended. Upon receiving a packet, a
machine checks the address field. If the packet is intended for itself, it
processes the packet; if the packet is intended for some other machine, it
is just ignored.

Broadcast systems generally also allow the possibility of addressing a

packet to all destinations by using a special code in the address field.
When a packet with this code is transmitted, it is received and processed
by very machine on the network, and this mode of operation is called
broadcasting. Some broadcast systems also support transmission to a
subset of the machines, something known as multicasting. One possible
scheme is to reserve one bit multicasting. The remaining (n–1) address
bits can hold a group number. Each machine can “subscribe” to any or
all of the groups. When a packet is sent to a certain group, it is delivered
to all machines subscribing to that group.

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3.5 Reference Model

In this section, we will discuss two important network architectures: the

OSI reference model and the TCP/IP reference model.

3.5.1 OSI (Open System Interconnection) Reference Model

The OSI model is based on a proposal developed by the International Standards

Organisation as a first step towards international standardization of the protocols used
in the various layers. The model is called the ISO – OSI (International Standard
Organisation–Open Systems Interconnection) Reference Model because it deals with
connecting open systems – that is, systems that are open for communication with
other systems.

Its main objectives were to:

(i) Allow the manufacture of different systems to

interconnect equipment through standard interfaces.
(ii) Allow software and hardware to integrate well and be portable on
different systems.

The OSI model has seven layers shown in figure 2. The principles that were applied
to arrive at the seven layers are as follows:

1. Each layer should perform a well–defined function.

2. The function of each layer should be chosen with an eye
toward defining internationally standardized protocols.
3. The layer boundaries should be chosen minimize the information
flow across the interfaces.

The seven layers of ISO OSI Reference Model are: (a)

Physical Layer
(b) Data Link Layer
(c) Network Layer
(d) Transport Layer
(e) Session Layer
(f) Presentation Layer
(g) Application Layer.

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 Layer
Application Protocol
Application Application
7
Interface
Presentation Protocol
6 Presentation Presentation

Session Protocol
5 Session Session

4 Transport Transport
Communication Subnet Boundary

3 Internet subnet protocol

Network Network
Network Network

Data Link Data Link Data Link Data Link

2

1 Physical Physical Physical Physical

Transmission medium

Figure 2: OSI Reference Model

3.5.1.1 The Physical Layer

Physical Layer defines electrical and mechanical specifications of

cables, connectors and signaling options that physically link two nodes
on a network.

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3.5.1.2 The Data Link Layer

The main task of the Data Link Layer is to provide error free transmission. It
accomplishes this task by having the sender break the input data up into data frames,
transmit the frames sequentially, and process the acknowledgement frames sent back
to the receiver.

The Data Link Layer creates and recognises frame boundaries. This can be
accomplished by attaching special bit patterns to the beginning and end of the frame.
If these bit patterns can accidentally occur in the data, special care must be taken to
make sure these patterns are not incorrectly interpreted as frame delimiters

3.5.1.3 The Network Layer

Whereas the Data Link Layer is responsible for end to end delivery, the network
layer ensures that each packet travels from its source to destination successfully and
efficiently. A key design issue is determining how packets are routed from source to
destination. Routes can be based on static tables that are “wired into” the network and
rarely changed.

They can also be determined at the start of each conversation, for example, a
terminal session. Finally, they can be highly dynamic, being determined anew for
each packet, to reflect the current network load.

When a packet has to travel from one network to another to get to its destination,
many problems can arise. The addressing used by the second network may be
different from the first one. The second one may not accept the packet at all because it
is too large. The protocols may differ, and so on. It is up to the network layer to
overcome all these problems to allow heterogeneous networks to be interconnected.

3.5.1.4 The Transport Layer

The basic function of the transport layer is to accept data from the session layer,
split it up into smaller units if need be, pass these to the Network Layer, and ensure
the pieces all arrive correctly at the other end. Furthermore, all this must be done
efficiently, and in a way that isolates the upper layers from the inevitable changes in
the hardware technology.

The transport layer provides location and media independent data transfer
service to session and upper layers.

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3.5.1.5 The Session Layer

The main tasks of the session layer are to provide:

 Session establishment
 Session Release– Orderly or Abort
 Data Exchange
 Expedited Data Exchange.

The session layer allows users on different machines to establish

sessions between them. A session allows ordinary data transport, as does
the transport layer, but it also provides enhanced services useful in some
applications. A session might be used to allow a user to log into a
remote time sharing system or to transfer a file between two machines.

One of the services of the session layer is to manage dialogue control.

Sessions can allow traffic to go in both directions at the same time, or in
only one direction at a time. If traffic can only go one way at a time
(analogous to a single railroad track), the session layer can help keep
track of whose turn it is.

A related session service is token management. For some protocols, it is

essential that both sides do not attempt the same operation at the same
time. To manage these activities, the session layer provides tokens that
can be exchanged. Only the side holding the token may perform the
critical operation.

Another session service is synchronization. Consider the problem that

might occur when trying to do a two–hour file transfer between two
machines with a one hour mean time between crashes. After each
transfer is aborted, the whole transfer would have to start over again and
would probably fail again the next time as well. To eliminate this
problem, the session layer provides a way to insert after the last
checkpoint has to be repeated.

3.5.1.6 The Presentation Layer

Unlike all the lower layers which are just interested in moving bits
reliably from here to there, the presentation layer is concerned with the
syntax and semantics of the information transmitted.

A typical example of a presentation service is encoding data in a

standard agreed upon way. Most user programs do not exchange random
binary bit strings, they exchange things such as people’s names, dates,
amounts of money and invoices. These items are represented as
character strings, integers, floating–point number, and data structures

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composed of several simpler items. Different computers have different codes for
representing character strings, (e.g., ASCII and Unicode), integers (e.g., one’s
complement and two’s complement), and so on. In order to make it possible for
computers with different representations to communicate, the data structure to be
exchanged can be defined in an abstract way, along with a standard encoding to be
used “on the wire”. The presentation layer manages these abstract data structure
and converts from the representation used inside the computer to the network
standard representation and back.

3.5.1.7 Application Layer

Application layer supports functions that control and supervise OSI application
processes such as start/maintain/stop application, allocate/de–allocate OSI resources,
accounting, check point and recovering. It also supports remote job execution, file
transfer protocol, message transfer and virtual terminal.

3.5.2 TCP Reference Model

The TCP/IP network architecture is a set of protocols that allow communication across
multiple device networks. The architecture evolved out of research that had the
original objective of transferring packets across three different packet networks: the
ARPANET packet– switching networks, a packet radio network, and a packet satellite
network. The military orientation of the research placed a premium on robustness with
regards to failures in the network and on flexibility in operating over diverse
networks. The environment led to a set of protocols that are highly effective in
enabling communication among the many different types of computer systems and
networks. Today, the internet has become the primary fabric for interconnecting the
world’s computers. In this section, we introduce the TCP/IP network architecture and
TCP/IP is the main protocol for carrying information.

Figure 3 shows the TCP/IP network architecture, which consists of four layers. The
Application Layer provides services that can be used by other applications. For
example, protocols have been developed for remote login, for e–mail, for file
transfer, and for network management.

The Application Layer programs are intended to run directly over the transport layer.
Two basic types of services are offered in the transport layer. The first service
consists of reliable connection–oriented transfer of a byte stream, which is
provided by the Transmission Control Protocol (TCP). The second service
consists of best–effort connectionless transfer of individual messages, which is
provided by the User Datagram Protocol (UDP). This service provides no
mechanisms

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for error recovery or flow control. UDP is used for applications that
require quick but necessary or flow control. UDP is used for application
that require but necessarily reliable delivery layer.

Application
Layer
Transport
Layer
Internet
Layer
Network
Interface Layer

Figure 3: TCP/IP Network Architecture

The TCP/IP model does not require strict layering. In other words, the
application layer has the option or bypassing intermediate layers. For
example, an application layer may run directly over the internet.

The Internet Layer handles the transfer of information across multiple

networks through the use of gateways of routers, as shown in figure 4.
The Internet Layer corresponds to the part of the OSI network layer that
is concerned with the transfer of packets between machines that are
connected to different networks. It must, therefore, deal with the routing
of packets across these networks as well as with the control of
congestion. A key aspect of the internet layer is the definition of
globally unique addresses for machines that are attached to the Internet.
The internet layer provides a single service, namely: best–effort
connectionless packet transfer. IP packets are exchanged between
routers without a connection set up; the packets are routed
independently, and so they may traverse different paths. For this reason,
IP packets also called datagrams. The connectionless approach makes
the system robust; that is, if failures occur in the network, the packets
are routed around the points of failure; there is no need to set up the
connections. The gateways that interconnect the intermediate networks
may discard packets when congestion occurs. The responsibility for
recovery from these losses is passed on to the transport layer.

Finally, the Network Interface layer is concerned with the network–

specific aspects of the transfer of packets. As such, it must deal with the
part of the OSI network layer and data link layer. Various interfaces are
available for connecting end computer systems to specific networks such
as X.25, ATM, frame relay, Ethernet, and token ring.

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The network interface layer is particularly concerned with the protocols that access
the intermediate networks. At each gateway, the network access protocol
encapsulates the IP packet into a packet or frame of the underlying network or
link. The IP packet is recovered at the exit gateway of the given network. This
gateway must then encapsulate the IP packet into a packet or frame of the type of
the next network or link.

This approach provides a clear separation of the internet layer from the technology
dependent network interface layer. This approach also allows the internet layer
to provide a data transfer service that is transparent sense of not depending on
the details of the underlying networks. The next section provides a detailed
example of how IP operates over the underlying networks.

Figure 5 shows some of the protocols of the TCP/IP protocol suite. The figure
shows two of the many protocols that operate over TCP, namely, HTTP and SMTP.
The figure also shows DNS and Real time Protocol (RTP), which operate over
UDP. The transport layer protocols TCP and UDP, on the other hand, operate over
IP. Many network interfaces are defined to support IP. The salient part of figure 5 is
that all higher–layer protocols access the network interfaces through IP. This feature
provides the capability to operate over multiple networks. The IP protocol is
complemented by additional protocols (ICMP, IGMP, ARP, and RARP) that are
required to operate an internet.

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 HTTP SMTP DNS RTP

TCP UDP

IP

Network Network Network

Interface 1 Interface 2 Interface 3

Figure 5: TCP/IP Protocol Graph

The hourglass shape of the TCP/P protocol graph underscores the

features that make TCP/IP so powerful. The operation of the single IP
protocol over various networks provides independence from the
underlying network technologies. The communication services of TCP
and UDP provide a network independent platform on which applications
can be developed. By allowing multiple network technologies to coexist,
the internet is able to provide ubiquitous connectivity and to achieve
enormous economies of scale.

3.5.3 Difference between OSI Reference Model & TCP

Reference Model

OSI Reference Model TCP Reference Model

1. Seven layers 1. 4 layers
2. It distinguishes between service, 2. Does not clearly distinguish
interface and protocol. between service, interface
3. First comes description of model and and protocol
protocol comes next 3. protocol comes first and
4. Both have Network description of model later.
5. supports connectionless and 4. Transport and Application
connection oriented communication layer.
in network layer and only 5. TCP/IP has only one mode in
connection–oriented communication Network layer (connectionless)
in transport layer (Co2 T. service is but supports both modes in
Transport layer.
visual to the User)
6. Protocols in TCP/IP are not
6. Protocol in OSI model are better hidden and thus, cannot be
hidden and can be replaced relatively easily replaced.
easily (No Transparency) (Transparency)

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3.6 IEEE Standards for LAN

Although there are many standards, we will configure here to just three of them:

 IEEE Standard 802.3 and Ethernet

 IEEE Standard 804 Token Bus
 IEEE Standard 802.5 Token Ring

IEEE Standard 802.3 and Ethernet

1. 802.3 is a simple protocol, Station can be installed on fly without taking network
down. A passive cable is used and modems are not required. Delay at low load is
practically zero. A station does not have to wait for a token, they just transmit
immediately. Each station has to be able to detect the signal of the weakest station
even when it is transmitting itself and all of the collision detect circuiting in the
transceiver is analog. Minimum valid frame is 64 bytes.

2. 802.4 Bus – It uses highly reliable cable envision equipment which is

available from numerous vendors. It is more deterministic than 802.3
although repeated losses of the token at critical moments can introduce
more uncertainty than its supporters like to admit–Token Bus also supports
priorities.

3 Token Ring–Point–to–Point connection means that the

engineering is easy and can be fully digital. Ring can be built virtually in a
transmission medium from carrier pigeon to fibre optics. The standard twisted pair is
cheap and simple to install like the Token bus in token ring priorities are possible.

4.0 CONCLUSION

This unit has introduced you to Computer Networks. We have classified the different
types of networks, goal and motivation of Computer Networks. This unit has
introduced you to the two types of network models as well as the difference between
these two. The unit has also done a good job of defining various standards of LANs.

5.0 SUMMARY

A communication system that supports many udders is called a network. In a network,

many computers are connected to each other by various topologies like star, ring,
complete, interconnected or irregular.

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Depending on the area of coverage, a network can be classified as LAN, MAN,
or WAN. A network is required for better utilisation of expensive resources,
sharing information, collaboration among different groups, multimedia
communication and video conferencing.

Two different types of networking models OSI and TCP/IP exist. The
difference between these models was discussed in detail.

6.0 TUTOR–MARKED ASSIGNMENT

i. What are the various types of networks?

ii. What is the difference between broadcasting and multicasting?

7.0 REFERENCES/FURTHER READING

7
 UNIT 2: NETWORK STRUCTURE

1.0 INTRODUCTION

This unit provides a survey of the basic network structures or topologies.

Topology can be considered as a virtual shape or structure of a network.
2.0 OBJECTIVES
By the end of this unit, you will be able to:

 name the four basic network topologies

 cite advantages and disadvantages of each type
 state the criteria necessary for an effective and efficient network.

3.0 MAIN CONTENT

3.1 PHYSICAL TOPOLOGY
The term physical topology refers to the way in which a network is laid out
physically. Two or more devices connect to a link; two or more links form a
topology. The topology of a network is the geometric representation of the
relationship of all the links and linking devices (usually called nodes) to one
another. There are four basic topologies possible: mesh, star, bus and ring (see
figure 1).

Topolog
y

Mesh Ring
Star Bus

Figure 1 Categories of topology

8
3.1.1 Mesh

In a mesh topology, every device has a dedicated point to point link to every
other device. The term dedicated means that the link carries traffic only
between the two devices it connects. (see figure 2)

Statio
n

Station Station

Station Station

Figure 2 A fully connected mesh topology (five devices)

A mesh offers several advantages over other network topologies. First, the
use of dedicated links guarantees that each connection can carry its own data
load, thus eliminating the traffic problems that can occur when links must be
shared by multiple devices. Second, a mesh topology is robust. If one link
becomes unavailable it does not incapacitate the entire system. Third, there is
the advantage of privacy or security. Whenever message travels along a
dedicated line, only the intended recipient sees it. Finally, point to point links
make fault identification and fault isolation easy. Traffic can be routed to avoid
links with suspected problems. This enables the network manger to discover
the precise location of the fault and aids in finding its cause and solution.

The main disadvantages of a mesh are related to the amount of cabling and that
of I/O ports required. First, because every device must be connected to every
other device, installation and reconnection are difficult. Second, the sheer bulk
9
of the wiring can be greater than the available space (in walls, ceilings or floors)
can accommodate. Finally, the hardware required to connect each link (I/O
ports and cables) can be prohibitively expensive. For these reasons, a mesh
topology is usually implemented in a limited fashion, for example, as a
backbone connecting the main computers of a hybrid network that can include
several other topologies.

One practical example of a mesh topology is the connection of a telephone

regional office in which each regional office needs to be connected to every
other regional office.

3.1.2 Star Topology

In a star topology, each device has a dedicated point-to-point link only to a

central controller, usually called a hub. These devices are not directly
linked to one another. Unlike a mesh topology, a star topology does not
allow direct traffic between devices. The controller acts as an exchange. If one
device wants to send data to another it sends the data to the controller, which
then relay the data to the other connected devices (see figure 3)

Hub

Statio Statio Statio Station

n n n

Figure 3 A star topology connecting four station

A Star topology is less expensive than a mesh topology. In a star, each device
needs only one link and one I/O port to connect to any number of others.
9
This factor also makes it easy to install and reconfigure. Far less calling needs
to be housed, and additions, moves and deletions involve only one connection
between that device and the hub.

Other advantages include robustness. If one link fails, only that link is affected.
All other remain active. This factor also lends itself to easy fault identification
and fault isolation. As long as the hub is working, it can be used to monitor link
problems and bypass detective links.

One big disadvantage of a star topology is the dependency of the whole

topology on one single point, the hub. If the hub goes down, the whole system is
dead.
Although a star requires far less cable than a mesh, each node must be linked to
a central hub. For this reason, often more cabling is registered in a star then in
some other topologies (such as ring or bus)

3.1.3 Bus Topology

A bus topology is multipoint. One long cable acts as a backbone to link all the
devices in the network (see figure 4)

Station Station Station

Drop line Drop line Drop line

Cable Cable end
end

Figure 4 A bus topology connecting three stations

Nodes are connected to the bus cable by drop lines and taps. A drop line is
a connection running between the device and the main cable. A tap is a
connector that either splices into the main cable or punctures the sheathing of a
9
cable to create a contact with the metallic core. As a signal travels along the
backbone, some of its energy is transformed into heat. Therefore, it becomes
weaker and weaker as it travels farther and farther. For this reason, there is a
limit on the number of taps a bus can support and on the distance between those
taps.

Advantages of a bus topology include ease of installation. Backbone cable can

be laid along the most efficient path and then connected to the nodes by drop
lines of various lengths. In this way, a bus uses less cabling than mesh or star
topologies.

Disadvantages include difficult reconnection and fault isolation. A bus is usually

designed to be optimally efficient at installation. It can therefore be difficult to
add new devices. Signal reflection at the taps can cause degradation of quality.

In addition, a fault or break in the bus cable stops all transmission, even
between devices on the same side of the problem. The damaged area reflects
signals back in the direction of origin, creating noise in both directions.

Bus topology was the one of the first topologies used in the design of early
local area networks

3.4 Ring Topology

In a ring topology, each device has a dedicated point to point connection with
only the two devices on either side of it. A signal is passed along the ring in
one direction from device to device, until it reaches its destination. Each device
in the ring incorporates a repeater. When a device receives a signal intended
for another device, its repeater regenerates the bits and
passes them along (see
figure 5).
 Station Station

Repeater Repeater Repeater

Statio Repeater Station
n
Repeater Repeater

Station Station

Figure 5 A ring topology connecting six stations

A ring is relatively easy to install and reconfigure. Each device is linked to only
its immediate neighbors (either physically or logically). To add or delete a
device requires changing only two connections. The only constraints are media
and traffic considerations (maximum ring length and number of devices). In
addition, fault isolation is simplified. Generally in a ring, a signal is
circulating at all times. If one device does not receive a signal within a specified
period, it can issue an alarm. The alarm alerts the network operator to the problem
and its location.
However, unidirectional traffic can be a disadvantage. In a simple ring, a break
in the ring (such as a disabled station) can disable the entire network. This
weakness can be solved by using a dual ring or a switch capable of closing off
the break.

Today, the need for higher speed LANS has made this
topology less popular

3.5 Hybrid Topology

A network can be hybrid. For example, we can have a main star topology with
each branch connecting several stations in a bus topology as shown in Figure 6.

Station Station Station

Station Station Station

Hub

Station Station Station

Figure 6 A hybrid topology: a star backbone with three bus networks

SELF-ASSESSMENT EXERCISES

a) What are the three criteria necessary for an effective and efficient network?
b) What is network topology?
4.0 CONCLUSION

In the context of a communication network, the term topology refers to the

very in which the end points, or stations attached to the network are
interconnected. Topologies are the important part of the network design
theory. A better network can be built if you have the knowledge of these
topologies and if you know the difference between each topology.

5.0 SUMMARY

Topology refers to the physical or logical arrangement of a network. Devices

may be arranged in a mesh, star, bus or ring topology.

A mesh offers several advantages over other network topologies. The main
disadvantages of a mesh are the number of I/O ports required.

Star topology is less expensive than a mesh topology. One big disadvantage of a
star topology is the dependency of the whole topology on one single point, the
hub.

A bus topology is multipoint unlike mesh and star topologies that are
point to point connections. An advantage of a bus topology is ease of
installation. Disadvantages include difficult reconnection and fault isolation.

Ring topology is relatively easy to install and reconfigure. However,

unidirectional traffic can be a disadvantage.

Hybrid topology is complex which can be built of two or more above

networked topologies.
6.0 TUTOR-MARKED ASSIGNMENT

1. For each of the following four networks, discuss the consequences if a

connection fails.

a) Five devices arranged in a bus topology

b) Five devices arranged in a ring topology
2. For n devices in a network, what is the number of cable links required for a
mesh, ring, bus and star topology?
3. Name the four basic networking topologies and cite on advantage of each type.

7.0 REFERENCES/FURTHER READING

1. Burgess, M. (2004). Principles of Network and System Administration. (2nd

Ed.). Chichester, West Sussex , England: Wiley.

2. Forouzan, B.A, & Fegan, S.C. (2007). Data communications and Networking
(4th Ed). Mc
Graw Hill.
3. Limoncelli, T. A.,Hogan, C. J. & Chalup, S. R (2007}. The Practice of System
and Network
Administration. (2nd Ed.). Upper Saddle River, NJ: Addison-Wesley

4. Stallings, W. (2009). Data and computer communications ( 8th ed.). Upper

saddle River, NJ.: Pearson Education Inc.
5. Subramanian, M. (2000). Network Management: Principles and Practice,
Addison-Wesley
Module 2: Network Devices & Technology
Unit 1 Network Technology
Unit 2 Network Devices–I
Unit 3 Network Devices–II
Unit 4 Integrated Service Digital Network (ISDN)
Unit5 Asynchronous Transfer Mode (ATM)
Unit 6 Data Transmission and Multiplexing
Unit 7 Medium Access Control and Data Link Layer
UNIT 1: NETWORK TECHNOLOGY

1.0 INTRODUCTION
This unit looks at what constitutes a local area network (LAN), then a wide area
network (WAN) and then discusses the differences between the two. We then
discuss the technologies for implementing WAN.

2.0 OBJECTIVES

o explain the categories of networks

o state the distinctions between LAN and WAN
o explain the technologies used in implementing WAN.

3.0 MAIN CONTENT

3.1 Categories of Networks/Network Technologies

Today when we speak of networks, we are generally referring to two primary

categories: local area networks (LANs) and wide area networks (WANs).

The category into which a network falls is determined by its size. A LAN
normally covers an area less than 2 miles; a WAN can be worldwide.
Networks of a size in between are normally referred to as metropolitan area
networks (MANs) and span tens of miles.

3.1.1 LOCAL AREA NETWORK

(LAN)
One type of network that becomes ubiquitous is the local area network. Indeed,
LAN is to be found in virtually all medium and large size office buildings.
Depending on the needs of an organization and the type of technology used, a
LAN can be as simple as two PCs and a printer in someone’s home office; or it
t
can extend throughout a company and include audio and video peripherals.
Currently LAN size is limited to a few kilometers. LANs are
designed to allow resources to be shared between personal computers or
workstations. The resources to be shared can include hardware (e.g. a printer),
software (e.g. an application program) or data. In addition to size, LANs are
distinguished from other types of networks by transmission media

and topology. In general, a given LAN will use only one type of transmission
medium. The most common LAN topologies are bus, ringed star.

Early LANS had data rates in the 4 to 16 megabits per seconds (mbps) ranges. LANs
come in a parallel of different configurations. The most common is switched LANs
and wireless LANs. The most switched LAN is a switched Ethernet LAN, which
may consist of a single switch with a parallel of attached devices, or parallel of
interconnected switches. Today, however, speeds are normally 100 or 1000 mbps.
Wireless LANs are the newest evolution in LAN technology.

3.1.2 WIDE AREA NETWORK (WAN)

A wide area network (WAN) provides long distance transmission of data, image,
audio, video information over large geographic area that may comprise a
country, a continent or even the whole world. WAN can be as complex as the
backbones that connect the Internet or as simple as a dial-up line that a home
computer to the Internet. We normally refer to the first as a switched WAN
and to the second as a point to point WAN. The switched WAN connects the
end systems which usually comprise a router (internet – working
connecting devices) that connects together LAN or WAN. The point to
point WAN is normally a line leased from a telephone or cable T.V provider
that connects a home computer or a small LAN to an Internet service provider
(ISP). This type of WAN is often used to provide Internet access. Wireless
WANs are become more and more popular. Traditionally, WANs have been
implemented using one of two technologies: Circuit switching and packet
switching. More recently, frame relay and asynchronous transfer mode (ATM)
networks have assumed major roles.

CIRCUIT SWITCHING
In a circuit- switching network, a dedicated communications path is established
between two stations through the nodes of the network. That path is a connected
sequence of physical links between nodes. On each link, a logical channel is
dedicated to the connection. Data generated by the source station are transmitted
along the dedicated path as rapidly as possible. At each mode, incoming data
are routed or switched to the appropriate outgoing channel without delay.
The most common example of circuit switching is the telephone network.

PACKET SWITCHING
A quite different approach is used in a packet switching network. In this case, it
is not necessary to dedicate transmission capacity along a path through the
network. Rather, data are sent out in a sequence of small chunks, called packets.
Each packet is passed through the network from node to node along some path
leading from source to destination. At each node, the entire packet is
received, stored briefly, and then transmitted to the next node. Packet
switching networks are commonly used for terminal to computer
communications.

FRAME RELAY

Packet switching was developed at a time when digital long distance

transmission facilities exhibited a relatively high error rate compared to
today’s facilities. As a result, there is a considerable amount of overhead
built into packet switching schemes to compensate for errors. The overhead
includes additional bits added to each packet to introduce redundancy and
additional processing at the end stations and the intermediate switching nodes
to detect and recover from errors.

With modern high-speed communication systems, this overhead is unnecessary

and counterproductive. It is unnecessary because the rate of errors has been
dramatically lowered and any remaining errors can easily be caught in the end
systems by logic that operates above the level of the packet-switching logic. It
is counterproductive because the overhead involved soaks up a significant
fraction of the high capacity provided by the network.
Frame relay was developed to take advantage of these high data rates and low
error rates whereas the original packet-switching networks were designed with a
data rate to the end user of about 64 kbps. Frame relay networks are designed
to operate efficiently at user data rate of up to 2mbps. The key to achieving
these high data rates is to strip out most of the overhead involved with errors
control.

ASYNCHRONOUS TRANSFER MODE (ATM)

Sometimes referred to as cell relay is a culmination of developments in

circuit switching and packet switching. ATM can be viewed as an evolution
from frame relay. The most obvious difference between frame relay and ATM
is that frame relay uses variable length packets called frames and ATM uses
fixed length packets, called cells. As with frame relay, ATM provides little
overhead for error control depending on the inherent reliability of the
transmission system and on higher layers of logic in the end of systems to
catch and correct errors. By wiring a fixed packet length, the processing
overhead is reduced even further for ATM compared to frame relay. The
result is that ATM is designed to work in the range of 10s and 100s of mbps
and in the Gbps range. ATM can also be viewed as an evolution from circuit
switching; only fixed-data- rate circuits are available to the end system.
ATM allows the definition of multiple virtual channels with date rate that
are dynamically defined at the time the virtual channel is created. By using
small, fixed-size cells, ATM is so efficient that it can offer a contant-data rate
channel even though it is using a packet-switching technique. Thus ATM
extends circuit switching to allow multiple channels with the data rate on each
channel dynamically set on demand.

3.1.3 Distinctions between LANs and WANs

There are several key distinctions between LANs and WANs.
Among which are:

1. The scope of the LAN is small, typically a single building or a cluster

of buildings. This difference in geographic scope leads to different
technical solution.

2. It is usually the case that the LAN is owned by the same

organization that owns the attached devices. For WANs, this is less often
the case, or at least a significant fraction of the network assets is not
owned. This has two implications. First, care must be taken in the choice of
LAN, because there may be a substantial capital investment (compared to
dial-up or leased charges of WANs) for both purchase and maintenance.
Second, the network management responsibility for a LAN falls solely on
the user.
3. The internal data rates of LANs are typically much greater than
those of WANs.
3.1.4 Metropolitan Area Network (MAN)

A MAN is a network with a size between a LAN and WAN. It normally covers
the area inside a town or a city. It is designed for customers who need a high-
speed connectivity, normally to the Internet and have end points spread over a
city or part of a city.
19

3.1.5 Interconnection of Networks: Internetwork

Today, it is very rare to see a LAN, a MAN or a WAN in isolation; they are
connected to one another. When two or more networks are connected, they
become an internetwork or internet.

SELF ASSESSMENT EXERCISES

What is an internet?

What is the Internet?

4.0 CONCLUSION

Whereas wide area networks may be public or private, LANs usually are owned
by the organization that is using the network to interconnect equipment.
LANs have much greater capacity than WANS to carry what is generally a
greater internal communication load.

5.0 SUMMARY

In general terms, communications networks can be categorized as local area

networks (LANs) and wide area networks (WANs).
A LAN consists of a shared transmission medium and a set of hardware
and software for interfacing devices to the medium and regulating the orderly
access of the medium. LAN size is limited. In addition to size, LANs are
distinguished from other types of networks by their transmission media and
topology. A WAN provides long-distance transmission over large geographic
areas. WAN is often used to provide Internet access. Traditionally, WANs have
been implemented using one of two technologies: circuit switching and packet
switching. Wireless WANs are becoming more and more popular.

20
6.0 TUTOR-MARKED ASSIGNMENTS

1. What are the advantages of packed switching compared to circuit switching?

2. What are some of the factors that determine whether a communication
system is LAN or WAN?
3. Outline the distinctions between LAN and WAN.
4. Discuss circuit-switching network.

7.0 REFERENCES/FURTHER READING

1. Burgess, M. (2004). Principles of Network and System Administration. (2nd

Ed.). Chichester, West Sussex, England: Wiley.

2. Forouzan, B.A, & Fegan, S.C. (2007). Data communications and Networking
(4th Ed). McGraw Hill.
3. Limoncelli, T. A.,Hogan, C. J. & Chalup, S. R (2007}. The Practice of System
and Network Administration. (2nd Ed.). Upper Saddle River, NJ: Addison-Wesley

4. Stallings, W. (2009). Data and computer communications (8th ed.). Upper

saddle River, NJ.: Pearson Education Inc.
UNIT 2: NETWORK DEVICES–I

1.0 INTRODUCTION

As corporations grow, network designers need to extend the area of a

network, the number of users on a particular network, and the bandwidth
available to the network users. To solve these problems, network
designers break a network into smaller portions and connect them with
networking devices such as bridges, switches and gateways etc.
Depending on the complexities of each of the networks being connected,
a choice is made between these different network devices.

In this unit, and the next unit, we will examine features of several
network devices.

2.0 OBJECTIVES

By the end of this unit, you will be able to

explain:

 repeaters
 bridges
 witches
 hubs.

3.0 MAIN CONTENT

3.1 Network Devices

Most common features of network devices are to interconnect networks,

boost signals etc. The basic difference between them is that they operate
at different layers. Now let us examine each device separately.

3.1.1 Repeaters

When a signal is sent over a long network cable, signal gets weakened
due to attenuation. This results in some data getting lost in the way. In
order to boost the data signal, Repeaters are needed to amplify the
weakened signal. They are known as signal boosters or amplifiers. They
are physical layer devices. They are like small boxes that connect two
segments of networks, refine and regenerate the digital signals on the
cable and send them on their way.
Repeaters help in increasing the geographical coverage of networks i.e.
LAN for example, IEEE802.3 Standard allows for up to four repeaters
connecting five cable segments to a maximum of 3000 metres distance.

Repeaters use different physical media as:

This Ethernet cable and fibre optic cable: Token ring networks
translate between electrical signals on shielded or unshielded twisted
pair wiring and light pulse on fibre–optic cabling.

In modern installations, repeaters are housed in the central wiring hubs

of 10 Base–T and fibre optic cable systems.

Repeaters send every bit of data appearing on either cable segment

through to the other side, even if the data consist of malformed packets
from a malfunctioning Ethernet adapter or packets not destined for use
of the local LAN segment.

Figure 1 : Repeater Action

Bridges

Segmenting a large network with a device has numerous benefits.

Among these are reduced collisions (in an Ethernet network), contained
bandwidth utilization, and the ability to filter out unwanted packets.
However, if the addition of the interconnect device required extensive
reconfiguration of stations, the benefits of the device would be
outweighed by the administrative overhead required to keep the network
running. Bridges were created to allow network administrators to
segment their networks transparently. This means that individual
stations need not know whether there is a bridge separating them or not.
It is up to the bridge to make sure that packets get properly forwarded to
their destinations. This is the fundamental principle underlying all of the
bridging behaviours.

Bridges work at the Data Link Layer of the OSI model. Since bridges
work in the Data Link Layer they do not examine the network layer
addresses. They just look at the MAC addresses for Ethernet and Token
Ring, Token Bus and determine whether or not to forward or ignore a
packet.

Purpose of a Bridge

The purposes of a Bridge are as followings:

1. Isolates networks by MAC addresses

2. Manages network traffic by filtering packets
3. Translates from one protocol to another.

Now let us examine each functionality of a bridge in detail.

1. Isolates networks by MAC addresses

A bridge divides a network into separate collision domains (Fig. 2). This
reduces congestion as only frames that need to be forwarded are sent
across interfaces. All transmissions between nodes connected to same
segment are not forwarded and therefore, do not load the rest of the
network.

Collision Collision
Domain A Bridge Domain B

Thus, bridges effectively improve the bandwidth of the network by

reducing the unnecessary traffic in the network.

For example, if you have one segment called 100: it has 50 users (in
several departments) using this network segment. The Engineering
Department is CAD (Computer–Aided Design)–oriented, while the
Accounting Department is into heavy number crunching (year end
reports, month end statements, etc.). On this network, any traffic
between clients of Accounting Department and the Accounting File
Server (in the Accounting Department) will be heard across the Segment
100. Likewise, any traffic between the Engineering Dept clients (to the
CAD File Server) will be heard throughout the Network Segment. The
result is that “Other” Departments accesses to the Generic File Server
are incredibly slow: this is because of the unnecessary traffic that’s
being generated from other departments (Engineering and Accounting).

The solution is to use one bridge to isolate the Accounting Department

and another bridge to isolate the Engineering Department. The Bridges
will only allow packets to pass through that are not on the local segment.
The bridge will first check its “routing” table to see if the packet is on
the local segment. If it is, it will ignore the packet, and not forward it to
the remote segment. If a client of Accounting Department sends a packet
to the Accounting File Server, then Bridge #1 will check its routing table
(to see if the Accounting File Server is on the local port). If it is on the
local port, then Bridge #1 will not forward the packet to the other
segments. If a client of Accounting Department sends a packet to the
Generic File Server, Bridge #1 will again check its routing table to see if
the Generic File Server is on the local port. If it is not, then Bridge #1
will forward the packet to the remote port.
2. Manages network traffic by filtering packets

Bridges listen to the network traffic, and build an image of the network
on each side of the bridge. This image of the network indicates the
location of each node (and the bridge’s port that accesses it). With this
information, a bridge can make a decision whether to forward the packet
across the bridge – if the destination address is not on the same port – or,
it can decide not to forward the packet (if the destination is on the same
port).

3. Translates from one protocol to another

The MAC layer also contains the Bus Arbitration method used by the
network. This can be CSMA/CD, as used in Ethernet, or Token Passing,
as used in Token Ring. Bridges are aware of bus arbitration and special
translation bridges can be used to translate between Ethernet and Token
Ring.

Bridges physically separate a network segment by managing the traffic

(that’s based on the MAC address). Bridges are store and forward
devices. They receive a packet on the local segment, store it, and wait
for the remote segments to be clear before forwarding the packet. The
two physical types of bridges are Local and Remote bridges.

4. Local Bridges

Local Bridges are used (as in the previous examples) where the network
is being locally (talking about physical location now) segmented. The 2
segments are physically close together: same building, same floor, etc.
Only one bridge is required.

5. Remote Bridges

Remote Bridges are used in pairs, and also used where the network is
remotely segmented (again, talking of physical locations). The two
segments are physically far apart: different buildings, different floors,
etc. 2 x half–bridges are required; one at each segment. The remote
bridges are half of a normal bridge, and may use several different
communications media in between.

6. Bridging Methodologies

Transparent Bridges examine the MAC address of the frames to

determine whether the packet is on the local segment or on the distant
segment. Early bridges required the system administrator to manually
build the routing table to tell a bridge which addresses were on which
side of the bridge. Manually building a routing table is called fixed or
static routing. Modern bridges are self–learning: they listen to the
network in promiscuous mode, meaning that they accept all packets,
regardless of the packets’ addressing. The bridge then looks up each
packet’s destination DLC Address in its internal tables to find out which
port the destination NIC is attracted to. Finally, it forwards the packet
onto only the necessary port. In the case of a broadcasting message, the
bridge forwards the packet onto every port except the port that the
packet came from. Promiscuous listening is the key to the bridge’s
transparent operation. Since the bridge effectively “hears” all packets
that are transmitted, it can decide whether forwarding is necessary
without any special behaviour from the individual stations.

Consider a situation where there are two bridges, Bridge A and B. As

frames flow on Bridge A’s local port, Bridge A examines the source
address of each frame. Any frames with a destination address (other than
the nodes on the local port) are forwarded to the remote port. As far as
Bridge A is concerned, nodes on Bridge B’s local port appear as if they
were on Bridge A’s remote port and therefore are mapped in the table
accordingly. Similarly, Bridge B also develops its routing table for
various nodes.

The algorithm used by transparent bridges is backward learning. As

mentioned above, the bridges operate in promiscuous mode and track
the source addresses of different frames. Because it knows what ports
different addresses come from, it also knows onto what port to send
packets going to those addresses. The backward learning algorithm can
be written in Pseudo Code as follows:

if the address is in the tables then

forward the packet onto the necessary port.
if the address is not in the tables, then
forward the packet onto every port except for the port that
the packet was received on, just to make sure the destination
gets the message. add an entry in your internal tables
linking the source address of the packet to whatever port the
packet was received from.

Take, for example, a simple network consisting of a four–port

transparent bridge with five stations attached to it. The ports on the
bridge shall be numbered one through four, with Station A and Station B
on port 1, no station on port 2, Station C on port 3, and Station D and
Station E on port 4. The bridge has just been brought on–line, and its
tables are empty.
Station B transmits a packet destined for station C. Since the bridge doesn’t know
what port station B is on yet, it puts the packet out onto every port except Port 1 (the
packet came from Port 1, so the bridge knows that the packet has already been seen by
stations on port 1). This behaviour is known as flooding. The bridge also examines the
source address in the packet and determines that Station B is attached to Port 1. It
updates its stables to reflect this.

Now that the bridge knows where Station B is, it will forward packets destined for
Station B only onto Port 1. As stations transmit packets, the bridge will learn the
location of more and more stations until, finally, it knows the location of every station
that is attached to its ports. The beauty of the system is that even if the bridge doesn’t
know the location of a station, packets still get sent to their destination, just with a
tiny bit of wasted bandwidth.

Finally, the bridge ages each entry in its internal tables and deletes the entry if, after a
period of time known as the aging time, the bridge has not received any traffic from
that station. This is just an extra safeguard to keep the bridge’s tables up–to–date.

7. Advantages of Transparent Bridges

 Self learning: Requires no manual configuration, considered plug

and work.
 Independent of higher level protocols (TCP/IP, IPX/SPX,
Netbeui, etc.).
 No hardware changes required, no software changes required.

8. Disadvantages of Transparent Bridges

Can only work with one path between segments: loops are not allowed: A loop would
confuse the bridges as to which side of the bridge a node was really on (i.e., local
or remote)? Transparent Bridges are not suitable for use on MANs on WANs,
because many paths can be taken to reach a destination. In a LAN, it is simple to
determine that a loop occurs, but in a large corporate network (with several hundred
bridges), it may be next to impossible to determine. As such, bridges are most
commonly used in LAN–to–LAN connectivity (and not in MANs or WANs).

9. Spanning Tree Bridges

The Spanning Tree Protocol was developed to address the problem of loops in
Transparent Bridging. The IEEE 802.ID (Institute of Electrical
and Electronic Engineers) committee formed the Spanning Tree
Protocol.

The Spanning Tree Protocol (STP) converts a loop into a tree topology
by disabling a bridge link. This action ensures that there is a unique path
from any node to every other node (in a MAN or WAN). Disabled
bridges are kept in a stand–by–mode of operation until a network failure
occurs. At a time, the Spanning Tree Protocol will attempt to construct a
new tree, using any of the previously disabled links.

The Spanning Tree Protocol is a Bridge–to–Bridge communication

where all bridges cooperate to form the overall bridge topology. The
Spanning Tree algorithm is dynamic, and periodically checks every one
to four seconds to see if the bridge topology has changed.

Each bridge is assigned an arbitrary number that assigns priority to the

bridge in the Internetwork. The number is concatenated with the bridge
MAC address. If 2 bridges have the same priority, the MAC address is
used as a tie breaker mechanism. The lower the assigned number, the
higher the bridge priority.

During initial power–up, a Bridge Protocol Data Unit (BPDU) is

flooded out each network port of the bridge. The BPDU contains the
following: the current spanning tree root, the distance to the root
(measured in hops through other bridges), the bridge address
information, and the age of the information in the BPDU. Bridges
priorities are usually controlled manually so as to configure the traffic
flow – over the Internetwork – on a preferred path.

Problems can arise where, for example, the Spanning Tree Algorithm
may select a path from Los Angeles to New York City – and back to San
Francisco rather than the preferred route of Los Angeles to San
Francisco.

10. Source Routing Bridges

Source–Routing is mostly used to interconnect token ring LANs. In

Source–Routing, the source station must determine, in advance, the
route to the LAN of the destination station, and include this route in the
header of each frame. To determine the routing information, the source
station first issues a search frame which is generally an LLC Test
command, on its ring. If a response is received from the desired
destination station, it indicates that both source and destination stations
are on the same ring and that no routing information is required.
However, if no response is received, the source station issues a route discovery frame,
which fans–out on every ring in the LAN segment. As the frame is forwarded from
one ring to another, each bridge updates the routing information in the search frame.
When the search frame reaches the destination, it contains the route between the
source and destination stations. The destination station then sends a response
frame to the source station, with the routing information. Both stations then use the
routing information in each subsequent frame sent to each other.

Source–Routing uses two key parameters to identify a route between a source station
and a destination station. These parameters are ring numbers and bridge numbers. Each
ring is assigned a unique number.

These numbers generally range between 1 and FFF (hex). Each bridge is assigned a
bridge number, ranging between 0 and F (hex). The only restriction when assigning
bridge numbers is that parallel bridges connecting identical rings, must have
different bridge numbers. The route between the source and the destination stations
consists of LAN numbers and bridge numbers. The route is obtained by thus: each
bridge which receives the route discovery frame adds to the existing route, its number
and the ring number that it forwards this frame to.

The Pseudo Code for Source Routing Bridges can be written as:

 The host uses its known path to the destination if it has one that is not
old.
 Else, the host sends a probe message.
 The probe will be forwarded by every bridge that sees it, on every
LAN to which the bridge is attached (except the one the probe
came in on).
 If the bridge sees its own ID already in the path the probe is
accumulating, it will drop the probe without forwarding it (preventing
a loop).
 The probe will eventually get to the destination by every possible
path, including the shortest.
 The destination will return the probe to the sender, using the
discovered route as its source routing path.
 The source will then send its “real” message using the newly
discovered route.

3.1.3 Switches

A switch is a device that incorporates bridge functions as well as point– to–point

‘dedicated connections’. They connect devices or networks, filter, forward and flood
frames based on the MAC destination address of each frame. Switch operates at
Data link layer of the OSI model.
They are technically called bridges. They move data without contention.
Ethernet switches provide a combinations of shaed/dedicated
10/100/1000 Mbps connection. Some E–net switches support cut–
through switching: frame forwarded immediately to destination without
waiting for assembling of the entire frame in the switch buffer. They
significantly increase throughput. It provides express lane for traffic.

3.1.4 Hubs

If multiple incoming connections need to be connected with multiple

out–going connections, then a hub (Figure 4) is required. In data
communications, a hub is a place of convergence where data arrive from
one or more directions and are forwarded out in one or more other
directions. Hubs are multi–port repeaters and as such, they obey the
same rule as repeaters. They operate at the OSI Model Physical Layer.

Hubs are used to provide a Physical Star Topology. At the centre of the
star is the Hub, with the network nodes located on the tips of the star.

Star Topology

The hub is installed in a central wiring closet, with all the cables
extending out to the network nodes. The advantage of having a central
wiring location is that it’s easier to maintain and troubleshoot large
networks. All of the network cables come to the central hub. This way, it
is especially easy to detect and fix cable problems. You can easily move
a workstation in a star topology by changing the connection to the hub at
the central wiring closet.

The disadvantages of a star topology are shown below:

 Failure of the Hub can disable a major section of the network.

 The Star Topology requires more cabling than does the Ring or
the Bus topology because all stations must be connected to the
hub, not to the next station.

Hub’s Segment–to–Segment Characteristics

To understand the Ethernet segment–to–segment characteristics of a

hub, let us first determine how the Ethernet Hubs operate. Logically,
they appear as Topology, and physically, as a Star Topology. Looking
inside an Ethernet, we can see that it consists of an electronic printed
circuit board. Understating that inside the Hub is only more repeaters,
we can draw the conclusion that all connections attached to a Hub are on
the same segment (and have the same segment number). A single
repeater is said to exist from any port to any port, even though it is
indicated as a path of 2 repeaters.

Cascaded Hub Network

Connecting hubs together through ports creates Cascading Hubs. One

Master Hub (Level 1) is connected to many Level 2 (slave) Hubs, which
are masters to Level 3 (slave) Hubs in a hierarchical tree (or clustered
star). The maximum number of stations in a Cascaded Hub Network is
limited to 128.

Backbone Networks

In a Backbone Network, there is no master Hub. The level 1 Hubs are

connected through their AUI port to a Coax backbone. For thin coax, up
to 30 hubs can be connected together. For thick coax, up to 100 hubs can
be connected to the backbone. The backbone is considered to be a
populated segment.

Level 2 Hubs are allowed to be connected to Level 1 Hubs’ 10 Base T

ports. This connection between the two hubs is considered an
unpopulated segment, or link segment. Up to 1,024 stations (or nodes)
can be attached to the Level 2 Hubs’ 10 BaseT ports.

All stations and segments would appear as 1 Logical segment, with 1

Network Number. In the real world, 1024 stations are never attached to
1 segment; as the resulting traffic would slow the network to a crawl.
Hub’s Addressing

Because a Hub is just many repeaters in the same box, any network
traffic between nodes is heard over the complete network. As far as the
stations are concerned, they are connected on 1 long logical bus (wire).

Half–Duplex and Full–Duplex Ethernet Hubs

Normal Ethernet operation is Half–Duplex: only 1 station or node is

talking at a time. The stations take turns talking on the bus (CSMA/CD–
bus arbitration). Full–Duplex Ethernet Hubs are hubs which allow two–
way communication, thus doubling the available bandwidth from 10
Mbps to 20 Mbps. Full–duplex hubs are proprietary products, and
normally only work within their own manufacturer’s line.

For example, if A wanted to talk to C, a direct 10 Mbps line would be

connected through the two switching hubs. Simultaneously, if D wanted
to talk to B, another direct 10 Mbps line (in the opposite direction)
would be connected through the two switching hubs (doubling the
available bandwidth to 20 Mbps).

There are no official standards for Full–Duplex Ethernet although

proprietary standards do exist.

Switching Hubs

Switching hubs are hubs that will directly switch ports to each other.
They are similar to full duplex hubs, except that they allow dedicated 10
Mbps channels between ports.

If A wanted to communicate with B, a dedicated 10 Mbps connection

would be established between the two. If C wanted to communicate with
D, another dedicated 10 Mbps connection would be established.
3.1.5 Comparison Of Switches And Hubs

HUBS SWITCHES
1. Collision Domain Broadcast Domain
2. All of the parts on a hub are Each part on a switch may be
part of the same Ethernet regarded as a separate Ethernet
(but all are part of the same local
area network).
3. All parts on a hub share the Each part on a switch has its own
same 10Mb (100 Mb) 10Mb (100 Mb) bandwidth
bandwidth)
4. Any frame appearing on one A directed frame appearing on one
port of a hub is repeated to part of a switch is forwarded only to
all other ports on the hub the destination port.
5. A sniffer on any hub port
can see all of the traffic on
the network
6. A hub will repeat defective Switched networks are difficult to
frames sniff.

4.0 CONCLUSION

In this unit, we have examined the features of several network devices such as
repeaters, bridges, switches, hubs, etc. and their various purposes in networks.

This unit has exposed you to when and how to use any of these devices. But you
should note that all the network devices discussed in this unit are used at physical
layer and Data link layer.

5.0 SUMMARY

In this unit we have studied about features of different network devices namely:
repeaters, bridges, hubs and switches.

Repeaters are used in long distance network cable to enhance the signals that get
weakened due to attenuation.
Bridges are used to interconnect multiple LANs two devices at the data link layers
of the OSI model.
Switches are used for performing the functions of bridges as well as point–to–
point dedicated connections.
Hubs are used to interconnect various incoming connections with
different outgoing connections at the Physical layer of the OSI
Model.
6.0 TUTOR–MARKED ASSIGNMENT

1. Which of the following network devices is used at the physical

layer? (a) Routers (b) Bridges (c) Repeaters (d) Switches
2. List the major functionality of a bridge
3. Compare Switches and Hubs.
4. What are Switching Hubs?

In the next unit, we will examine another set of network devices.

7.0 REFERENCES/FURTHER READING

 UNIT 3 NETWORK DEVICES–II

1.0 INTRODUCTION

In the previous unit, we studied some of the network devices which are used at
physical layer and data link layer. In this unit, we continue our discussion about
devices/operating at lower layers, and also look at higher layer devices. Routers
and Gateways work at network layers and above, whereas modem work at a lower
layer. We will also examine the differences between bridges and routers.

2.0 OBJECTIVES

By the end of this unit, you will be able to:

 routers
 gateways
 modems.

3.0 MAIN CONTENT

3.1 Network Devices

3.1.1 Routers

In an environment consisting of several network segments with different protocols

and architecture, a bridge may not be adequate for ensuring fast communication
among all of the segments. A complex network needs a device which not only
knows the address of each segment, but
also can determine the best path for sending data and filtering broadcast
traffic to the local segment. Such a device is called a Router.

Routers are both hardware and software devices. They can be cards that
plug into a collapsed backbone, stand–alone devices or software that
would run on a file server.

Purpose of Routers

The purpose of a router is to connect nodes across an Internetwork,

regardless of the Physical Layer and Data Link Layer protocol that is
used. Routers are hardware and topology–independent. Routers are not a
ware of the type of medium or frame that is being used (Ethernet, Token
Ring, FDDI, X.25, etc.). Routers are a ware of the Network Layer
protocol that is used (e.g., Novell’s IPX, UNIX’s IP, XNS, Apple’s
DDP, and so on).

Router OSI Operating Layer

Routers operate on the OSI Model’s Network Layer. The Internetwork

must use the same Network Layer protocol. Routers allow the
transportation of the Network Layer PDU through the Internetwork,
even though the Physical and Data Link Frame size and addressing
scheme may change.

Routers that only know Novell IPX (Internetwork Packet Exchange) will
not forward Unix’s IP (Internetwork Packet) PDUs, and vice versa.
Routers only see the Network Layer protocols that they have been
configured for. This means that a network can have multiple protocols
running on it (e.g., SPX/IP, TCPIIP AppleTalk, XNS, etc.).

For example, a Novell SPX/IPX router; only sees the Network Layer
protocol, IPX. This means that any TCP/IP PDUs will not pass through:
the router does not recognise the PDUs, and doesn’t know what to do
with them. Therefore, routers allow network traffic to be isolated – or
segmented – based on the Network Layer Protocol. This provides a
functional segmentation of the network.

Routers that can only see one protocol are called Protocol–Dependent
Routers. Routers that can see many different protocols (two or more) are
called Multi–protocol Routers.

Routing Protocols
Routing Protocols are a “sub–protocol” of the Network Layer Protocol.
They deal specifically with the routing of packets from the source, to the
destination (across an Internetwork). Examples of Routing Protocols are: RIP, IGRP
and OSPF. Let us look at each of these protocols in some more detail.

RIP–Routing Information Protocol

RIP was one of the first routing protocols to gain widespread acceptance. It is
described in RFC1058, which is an internet standard. Commercial NOS, such as
Novell, Apple, Banyan Vines, and 3Com, use RIP as the base routing algorithm for
their respective protocol suites.

RIP is a distance vector algorithm. Routers maintain a detailed view of locally–

attached network segments, and a partial view of the remainder of the routing
table. The routers contain information on the umber of Hop counts of each
segment. A hop is considered to be one transverse through a router. Pass through a
router and the hop count increases by 1.

The routers are updated every 30 seconds, when each router sends out a RIP
broadcast. This advertisement process is what enables RIP routing to be dynamic.
Dynamic routers can change routing tables on the fly (as the network
configuration changes). By using the Hop Count information from their
routing tables, routers can select the shortest path (the least number of hops) to the
destination.

Apple uses RTMP (Routing Table Maintenance Protocol):

This adds a good, bad or suspect route status indicator, depending on the age of the
route information.

Novell adds Ticks to the RIP Algorithm:

Ticks are dynamically assigned values that represent the delay associated
with a given route. Each tick is considered 1/18 of a second. LAN segments are
typically assigned a value of 1 tick. A T1 link may have a value of 5 to 6 ticks and
a 56 Kbps line may have a value of 20 ticks. A larger number of ticks indicate a
slower routing path.

Three commonest problems that can occur with RIP are shown below:

1. Routing loops
The router indicates that the shortest path is going back the way the
packet came from
2. Slow Route Convergence

Routers have delay timers that start counting after the RIP advertising
packet is broadcast. This gives the routers time to receive and formulate
a proper routing table from the other routers. If the delay timer is too
short, the routing table can be implemented with incomplete data
causing routing loops.

3. Hop Count Exceeded

The maximum number of hop counts is 15 for RIP. A hop count of 15 is

classified as unreachable which makes RIP unsuitable for large networks
where hop counts of 15 and above are normal.

EGRP–Exterior Gateway Routing Protocol

EGRP was created to solve many of the problems with RIP, and has
become the default routing protocol across the internet. EGRP is an
enhanced distance vectoring protocol; it uses up to5 metrics (conditions)
to determine the best route as shown below:

1. Bandwidth
2. Hop Count (Delay)–maximum of 255
3. Maximum Packet size
4. Reliability
5. Traffic (Load).

These routing metrics are much more realistic indicators (of the best
routes) than simple hop counts.

OSPF–Open Shortest Path First

OSPF is a link state premises:

It has several states of routers that are linked together in a hierarchical

routing model. This means that each router maintains link status
information and this is exchanged between routers wishing to build
routing tables. Unlike RIP, OSPF uses IP directly, OSPF packets being
identified by a special value in the IP datagram protocol field.

The top of the root is the Autonomous Router that connects to the
autonomous systems (the Internet). The next is the Backbone Routers,
the highest area in the OSPF system. Border routers are attached to
multiple areas and they run multiple copies of the routing algorithm.
Last are internal routers that run a single routing database for one area.
Basically, by dividing the network into a routing hierarchy, both substantial
reduction of routing update traffic and faster route convergence – result on a local
basis. Each level has a smaller routing table and less to update.

3.1.2 Comparison of Bridges and Routers

 Both are stored–and forward devices, but Routers are Network Layer
devices (examine network layer headers) and Bridges are Link Layer devices.

 Routers maintain routing tables (hierarchical, aggregatable addresses)

and implement routing algorithms, bridges maintain filtering tables (flat
addresses) and implement filtering, learning and spanning tree algorithms.

3.1.3 Gateways

This device (Figure 1) is used to connect totally dissimilar networks. They function
at a high end of OSI model. They perform protocol conversion for all seven
layers of the OSI model. They are commonly used to connect a LAN and a main
frame computer. Gateways handle conversions of messages, addresses and protocol,
to deliver a message from one network to another. They offer greatest flexibility in
internetworking communications. Gateway’s decision – making is more complex
than Routers. They are very costly and their implementation, maintenance and
operations, are also very complex. They are slower than other devices. They can
recover e–mail messages in one format and convert them into another format.

Gateways provide an interface between IPX–based LANs and the IP protocols of

the internet. This provides a centralised and secure way to connect IPX–based
LANs to IP networks. Because of this, a single IP address can be used for an entire
network. Therefore, this eliminates configuration and maintenance problems.

Dual–homed Gateway is also present in the network. It is a system that has two or
more network interfaces. It acts to block or filter some or all of the traffic trying to
pass between the networks in firewall configuration.
 A B C

GATEWAYS

WAN

GATEWAY GATEWAY

PCs Mainframe

Figure 1: Gateway

Gateway has its main memory and processor to perform protocol

conversion.

Typical corporate gateways connect the PC world of token Ring,

Ethernet and AppleTalk LANs to IBM’s main frame SNA environment
with x.25 packet switched networks or DECnet networks.

At the lowest level, gateway provides terminal emulation so all LAN

workstations can emulate varies considerably depending on the gateway.

Second level of gateway functionality includes file sharing between

LAN & host. Novell has developed a platform – independent version of
netware that will run on several different platforms, including several
traditional mini–computer platforms.

At the higher level of functionality, a gateway would provide peer–to–

peer communications between micro computer programs running on the
LAN, and mainframe programs running on the host. These types of
client/server relationships will become more and more important in the
near future as programs are written to distribute databases among LAN’s
mini–computers and mainframes, with the machine users
communicating with the programs, using the same type of user interface.
How Do Gateways Link Hosts and LANs?

Using gateway’s micro–mainframe connection is much more cost effective than

other types of connections like using coaxial cable via PC
3270 emulation card etc. The gateway board emulates a cluster controller so
each network workstation is seen by the mainframe as a
terminal linked to the cluster controller. The gateway’s multiple mainframe sessions
are split among the network’s workstations, so the
channel rarely sits idle. Only the gateway needs to have a circuit card
and the software necessary for protocol conversion and terminal emulation.

Remote LAN Gateways

These gateways (Figure 2) are becoming very common because of the evolution of
enterprise networks and WAN. A PC on the remote site’s LAN functions as a
gateway and runs gateway software. This gateway PC functions as a cluster
controller and communications with a front- end processor using IBM’s
Synchronous Data Link Control (SDLC) protocol via synchronous modems located
at both sites.

The limitation of these gateways has speed. A synchronous modem can dial up a
front–end processor at speeds up to 64Kbps. Companies with heavy micro–
mainframe traffic might require multiple remote gateways to solve this congestion
problem.

X.25 Gateways

Remote LAN can also communicate with IBM mainframe viz., x.25 gateway. A
gateway PC with an adapter card functions as a cluster controller and runs special
gateway software that Contains the QLLC protocol, an IBM defined protocol that
runs over the X.25 suite. The other LAN workstations emulate IBM 3270 terminals.
The IBM host simply assumes it’s communicating with the remote cluster controller.
 HOST
SDLC TRANSMISSION
FEP SYNCHRONOUS NODE
MODEM
SDLC Gateway
SYNCHRONOUS
MODEM
NODE LAN

Figure 2: Remote LAN Gateway

Netware Workstation running 3270

Terminal Emulation Software

3.14 Modem

This is a device which is used to convert digital signals generated by the

computer into an analog signal to be carried by a public access
telephone line. It is also the device that converts the analog signal
received over a phone line into digital signal usable by the computer. A
modem derives it meaning from a modulation, and demodulation is a
composite word that refers to two functional units that make up a device.
A signal modulator and a signal demodulator. A modulator converts
digital signal into an analog signal. A demodulator converts analog
signal into digital signal.

Modem can be classified into many categories to include the mode of

transmission and their techniques, as well as by the application features
they contain and the type of lines they are built to service.

Figure 3: Signal conversion by modems i.e Modulation and

Demodulation
Speed

Modem speed ranges from 300 bps to 56kbps. It normally transmits about 10
bits/character (each character has 8 bits); maximum rate of characters for a high
speed modem is 2,880 characters/sec. For example, a compressed image of 20KB
(equivalent to 20,000 characters) will take nearly 6 seconds to load on the
fastest modem. The tasks which a modem can perform are:

1. Automatically dials another modem using either touch–tone or pulse

dialing.
2. Auto answer i.e., automatically answers another modem for making
connection.
3. Disconnects a telephone connection when data transfer has been completed
or if an error occurs.
4. Automatic speed negotiation between two modems
5. Converts bits into the form suitable for the line (Modulator)
6. Transfer data reliably with the correct type of handshaking
7. Convert received signals back into bits (demodulator)

Modem standards

The CCIT (now known as ITU) has defined standards for modem communication.
Each uses v number to define their type.

v.22 bis – It operates at 1200 or 2400bps v.32

– Operates at 9600 bps
v.32 bis – Operates at 19,200 bps v. 33
– Operates at 14,400 bps v. 34 –
Operates at 28,800 bps

Modem Commands

They are provided by Hayes Company that pioneered Modems and defined the
standard method of programming the mode of modem, which is the AT
command language. A computer gets the attention of the modem by sending “AT”
command. For example, ‘ATDT’ is the touch–tone dial command. Initially, a
modem is in the command mode and accepts commands from the computer. These
commands are sent at either 300 bps or 1200bps.

Most commands are sent with AT prefix. Each command is followed by carriage
return character; a command without this is ignored. More than one command can
be placed in a single line and spaces can be entered to improve readability, either
character case can be used.
Modem can enter two states; the normal state and command state. In the
normal state, the modem transmits or receives characters from the
computer an in the command state, characters sent to the modem are
interpreted as commands. Once a command is interpreted, the modem
goes into the normal state. Any character sent to the modem is then sent
along with line. To interpret the modem or to end a connection so that it
goes back into command mode, three consecutive ‘+’ characters are sent
i.e. ‘+++’.

Example:

When a computer wants to make a connection using telephone no.

17325, it sends the command. ‘ATCH 17325’ using tone dialing. The
modem then replies with an OK response i.e., ‘O’ value and it tries to
make connection with remote modem. If it is not able to make
connection, it sends a message in form of a code as (3) for no carrier, (7)
for busy (6) for no dial tone etc. If it gets connected then it returns a
connect code as it sends ‘+++’ and then waits for a command from host
computer. In this case, command is “hang–up the connection” (ATH).
The modem will then return an OK response when it has successfully
cleared the connection.

Figure 4: Connection establish & release

The modem contains various status registers called s–register which

store modem settings.
Modem Set Up

The following figure shows a sample window from the MS Windows terminal
program (in both MS Windows 3.x and Windows 95/98). It shows the modem
command window. It can be seen that when the modem dials a number, the prefix
to the number dialed is ‘ATDT’. The hang–up command sequence is ‘+++’ ATH.

MODEM COMMANDS X
COMMAND
DIAL PREFIX SUFFX OK

HANG UP: ATDT CANCEL

BINARY IX: +++ ATH
MODEM
BINARY RX: DEFAULT
О HAYES
О MULTITECH
О TRAILBLAZER
ORIGINATE: ATQOV
О NONE
IEISO=0
Figure 5: Modem commands window

Modem Indicator

These are used to inform the user about current status of a connection. Typically
the indicator lights are:

 AA – ON when receiving call. OFF when not receiving calls, flash when
call is incoming.
 CD – ON when modem detects the remote modem’s carrier,
else it is off.
 OH – ON when modem is on the hook else off.
 RD – Flashes when modem is getting data or a command from the
computer.
 SD – Flashes when Modem is sending data.
 TR – Shows that DTR line is active i.e., computer is ready to send or
receive data.
 MR – Shows that modem is powered up.

The following table illustrates widely used modems with bit rates &
modulation techniques
Typical Modems:

ITU Recommendations Bit rate (bps) Modulation

V.21 300 FSK
V.22 1200 PSK
V.22 bis 2400 ASK/PSK
V.27 ter 4800 PSK
V.29 9600 ASK/PSK
V.32 9600 ASK/PSK
V.32 bis 14400 ASK/PSK
V.34 28800 ASK/PSK

Most modems operate with V .22 bis (2400bps), V.32 (9600bps), V.32
bis (14400bps) The V.32 and V.32 bis modems can be enhanced with
echo cancellation. They also typically have built–in compression using
either the V.42 bis standards or MNPC (Microcom Networking
Protocol) level 5.

4.0 CONCLUSION

In this unit you have been taken through network devices such as routers
and gateways that work at network layers and above, and modems that
work at a lower layer.

This unit has also exposed you to the differences between bridges and
routers.

5.0 SUMMARY

In this unit, we have studied about some networking devices which are
used at higher layers of OSI model. The devices which were covered are
the following:

1. Router

Used to connect two devices at the network layer of the OSI Model

2. Gateway

Used to connect totally dissimilar networks because they can perform

protocol conversion for all seven layers of the OSI Model.
3. Modem

Used to connect the computer with the telephone lines. A Modem can convert
digital signal of a computer to analog signals, so that it can be transferred through
the telephone lines.

6.0 TUTOR–MARKED ASSIGNMENT

1. Which layer does the Router operates?

(a) Physical Layer (b) MAC Layer (d) Network Layer
(c) Session Layer
2. List few standards of modems.
3. List the names of routing protocols.

7.0 REFERENCES/FURTHER READING

Unit 5 Asynchronous Transfer Mode (ATM)

1.0 INTRODUCTION

Asynchronous Transfer Mode (ATM) is a form of data transmission that

allows voice, video and data to be sent along the same network. In
contrast to ATM, in the past, voice, video and data were transferred
using separate networks. For example, voice was transmitted over the
phone, video over cable networks, and data over an internetwork. ATM
is the ultimate culmination of all the developments in switching and
transmission in the last twenty years and has the best of circuit switching
and packet switching (discussed, in the previous block).

Asynchronous Transfer Mode (ATM) is a technology that has its history

in the development of broadband ISDN in the 1970s and 1980s. In this
unit, first we will have a re–look at different types of switching
techniques (technologies) and then we will examine how ATM is
compatible with the existing technologies and then compare the
architectural difference between ATM and the OSI model and finally,
spend some time on how ATM protocol works.
2.0 OBJECTIVES

By the end of this unit, you will be able to:

 explain the term ATM

 discuss the compatibility of ATM as technology
 compare ATM-layered architecture with OSI Model
 describe how ATM protocol works
 describe the structure of ATM cell
 identify the various ATM classes of services
 define the various ATM classes of service
 discuss the approach and tools used for ATM traffic control
 discuss the benefits of ATM technology
 explain the various applications of ATM technology

3.0 MAIN CONTENT

3.1 Switching Techniques

In this section, we will discuss different types of switching techniques.

Circuit Switching

This was the first type of data transfer mechanism used. Circuit
switching is used in the telephone networks to transmit voice and data
signals. In a synchronous transmission, which involves transmission of
voice, a synchronized connection must be made between the sender and
receiver because there must be a constant time interval between each
successive bit, character, or event. To enable synchronized transmission,
circuit switching establishes a dedicated connection between the sender
and the receiver involved in the data transfer over the network. As a
result, the connection consumes network capacity whether or not there is
an active transmission taking place; for example, the network capacity is
used even when a caller is put on hold. For different applications,
utilisation of the line can vary enormously. However, there is little delay
and effective transparency for the user. It is very efficient for Constant
Bit Rate (CBR).

Packet Switching

In contrast to circuit switching, packet switching ensures that the

network is utilised at all times. It does this by sending signals even in
the small unused segments of the transmission – for example, between
the words of a conversation or when a caller is put on hold. However, in
packet switching, there can be variations in the timing when the digital
bits are received. For normal voice and data communications, this is not
a problem, but for broadband signals, such as television, it is a huge
problem that causes the picture to jerk and the audio to be out of
synchronization with the picture. Data to be sent is broken down into
chunks or packets. Each packet contains data and header information for
control e.g., routing. At each node the packet is received, stored briefly
and passed on. At each node, the packets may be put on a queue for
further movement into the network.

There are two approaches to transport–

1. Datagram, where each packet can take any path through the
network as long as they all reach the destination.

2. Virtual Circuit, where all the packets are routed through the
same path without having the path dedicated.

Datagram allows for dynamic handling of congestion and no call setup

is necessary. Virtual channels allow for sequencing, error and flow
control.

Though, Packet switching is much more efficient than circuit switching,

Packet–switched networks have been slow. The public data networks
that use the x.25 standard for public switching allow users to operate
typically at speeds of 9.6 kbps. The standard leased lines that large
companies use for their high-speed data communications, operate at 56
kbps. ATM can transmit bits through the network at speeds up to 622
Mbps.

Multirate Circuit Switching

This is an enhancement of the synchronous Time–Division

Multiplexing (TDM) approach used initially in circuit switching. In
circuit switching, a station must operate at which must be used
regardless of application. In multirate switching, multiplexing is
introduced. A station attaches to the network by means of a single
physical link which carries multiple fixed data–rate channels (B–channel
@ 64kbps). Traffic on each channel can be switched independently
through the network to various destinations. This is used for simple
ISDN. So the user has a number of data rate choice but they are fixed so
Variable Bit Rate (VBR) is difficult to accommodate efficiently.

Frame Relay

Frame relay is essentially identical to packet switching. Frame relay saw

its development as a result of high data rates and low error rates in
modern high–speed communications systems. In old packet switching,
there was considerable overhead involved in error recovery, redundancy
enhancement and routing information. With Frame relay, the packets are
now of variable length and not fixed length, meaning that they were
designed to operate at up to 2Mbps. This was very good for VBR.

Cell Relay

This is an evolution from Frame relay and multirate circuit switching.

Cell relay uses fixed sized packets called calls. Multirate circuit
switching also has fixed channels. Cells relay allows for the definition of
virtual channels with data rates dynamically defined. Using a small cell
size allows almost constant data rate even though it uses packets. From
frame relay, cell relay takes improved error control into account, and
allows more errors to be handled at a higher logical level. The fixed–size
cells reduce overhead even more and thus allow rates of tens to
hundreds of Mbps.

So, in the evolution of switching technology there has been a change

from two areas – circuit switching for CBR, and packet switching for
VBR

3.2 How Compatible is ATM as Technology?

ATM is emerging as a viable technology. Some of its application are as

follows:

 ATM is used in many networks today including both private and

public environments. ATM is used extensively by most public
service providers today to integrate different types of traffic into
one network.
 ATM can be used in existing twisted pair, fibre–optic, coaxial,
and hybrid fibre/coax (HFC) networks for local area network
(LAN) and wide area network (WAN) communications. Because
ATM was developed to have such a wide range of compatibility
with existing networks, its implementation does not require
replacement or over–building of telephone, data, or cable
networks.
 ATM is also compatible with wireless and satellite
communications.
3.3 ATM Layered Architecture in Comparison with OSI
Model

ATM has a layered structure that is similar to the 7–layered OSI model.
However, ATM only addresses the functionality of the two lowest layers
of the OSI, i.e;

 The physical layer, and

 The data link layer.

Apart from these two layers, all other layers of the OSI model are
irrelevant in ATM, as these layers are only part of the encapsulated
information portion of the cell which is not used by the ATM network.

In ATM, the functionality of the two lower OSI layers is handled by

three layers.

Application Layer
User Layers
ATM Adaptation Layer (AAL):

Convergence sublayer
Segmentation and Reassembly sublayer
ATM Layer
Physical Layer
Transmission Convergence Sub layer

ATM Protocol Model

i) Physical Layer

The Physical layer defines the specification of a transmission medium

(copper, fibre optic, coaxial, HFC, wireless) and a signal–encoding
scheme and electrical to optical transformation. It provides Convergence
with physical transport protocols such as SONET, as well as the
mechanism for transforming the flow of cells into a flow of bits.

The ATM form has left most of the specification for this level to the
implementer.

ii) The ATM layer deals with cells and cell transport. It defines the
layout of a cell and tells what the header fields mean. The size of
a cell is 53 bytes (5 bytes of header and 48 bytes of payload).
Because each cell is the same size and all are relatively small,
 delay and other problems with multiplexing different sized
packets are avoided.

It also deals with establishment and release of virtual circuits.

Congestive control is also located here. It resembles the network layer of
the OSI model as it has got the characteristics of the network layer
protocol of OSI model like;

 Routing
 Switching
 End-to-end virtual circuit set up
 Traffic management

Switches in ATM provides both switching and multiplexing cell format

of ATM Layer are distinguished as

 UNI (User Network Interface)

 UNI (Network–Network Interface

In both cases, the cell consists of a 5–byte header followed by a 48–

byte pay–load but the two headers are slightly different.

iii) ATM Adaptation Layer

The ATM Adaptation Layer (AAL) maps the higher-level data into
ATM cells to be transported over the ATM network, i.e., this layer
segments the data and adds appropriate error control information as
necessary. It is dependent on the type of services (voice, data, etc.) being
transported by the higher layer.

This is the adaptation layer that divides all types of user data into 48–
byte cells. The ATM layer that adds the five–byte header information
to direct the user data to its destination.

Depending on the type of data, several AAL protocols have been

defined. However, no AAL is restricted to a specific data class or type;
all types of data could conceivably be handled by any of the AALs. The
various AAL protocols define are:

1. AAL 1
2. AAL 2
3. AAL ¾
4. AAL 5
It is divided into two sublayers

 SAR (Segmentation & Reassembly)

 CS (Convergence Sublayer)

Segmentation & Reassemble

This is the lower part of the AAL. The SAR sublayer breaks packets up
into cells on the transmission side and puts them back together again at
the destination. It can add headers and trailers to the data units given to
it by the CS to form payloads. It is basically concerned with cells.

Convergence Sublayer

The CS sublayer makes it possible to have ATM system offer different

kinds of services to different applications. The CS is responsible for
accepting bit streams or arbitrary length messages from the application
and breaking them into units of 44 or 48 bytes for transmission.

How ATM Protocol Works

When a user sends data over the ATM network, the higher–level data
unit is passed down to the Convergence Sublayer of the AAL Layer,
which prepares data for the ATM Layer according to the designated
AAL protocol. The data is then passed down to the Segmentation and
Reassembly sublayer of the AAL Layer, which divides the data unit into
appropriately sized segments.

These segments are then passed down to the ATM Layer, which defines
an appropriate cell header for each segment and encapsulates the header
and payload segment into a 53–byte ATM cell. The cells are then passed
down to the Physical Layer; which streams the cells at an appropriate
pace for the transmission medium being used, adding empty cells as
needed.

ATM circuits are of two types:

1. Virtual Paths and,

2. Virtual Channels.

A virtual channel is a unidirectional pipe made up from the

concatenation of a sequence of connection elements.
A virtual path consists of a set of these channels.

Each virtual channel and virtual path has an identifier associated with it.
Virtual path is identified by Virtual Path Identifiers (VPI) and a virtual
channel is identified by a Virtual Channel Identifier (VCI). All channels
within a single path must have distinct channel identifiers but may have
the same channel identifier as channels in different virtual paths.

An individual channel can, therefore, be uniquely identified by its

virtual channel and virtual path number. Cell sequence is maintained
through a virtual channel connection.

ATM connections can be categorised into two types:

i) Point–to–point connections: – These are the connections which

connect two ATM end–systems. Such connections can be
unidirectional or bidirectional.

ii) Point–to–multipoint connection: These are the connections

which connect a single source end–system known as the root
node, to multiple destination end–systems (known as leaves).

The basic operation of an ATM switch is very simple to understand.

1. The ATM switch receives a cell across a link on a known VCI or

VPI value.

2. The ATM switch looks up to the connection value in a local

translation table to determine the outgoing port (or ports) of the
connection and the new VPI/VCI value of the connection on that
link.

3. The ATM switch then retransmits the cell on that outgoing link
with the appropriate connection identifiers.

The manner in which the local translation tables are set up determines
the two fundamental types of ATM connections:

 Permanent Virtual Connections (PVC): A PVC is a connection

set up by some external mechanism, typically network
management, in which a set of switches between an ATM source
and destination ATM system are programmed with the
appropriate VPI/VCI values.

 Switched Virtual Connections (SVC): An SVC is a connection

that is set up automatically through a signal protocol. SVCs do
 not require the manual to set up PVCs and, as such, are likely to
be much more widely used.

The ATM Network

An ATM networks consist of a set of ATM switches interconnected by

point–to–point ATM links or interfaces. ATM switches support three
kinds of interfaces:

 User–Network Interfaces (UNI)

 Network–Node Interfaces (NNI)
 Inter–Carrier Interface (ICI)

ATM ATM
User 1 SWITCH SWITCH

ATM Network 1

ATM ATM
Private ATM SWITCH SWITCH
Switch

ATM Network 2

User 2

Figure 1: ATM Network

 The UNI exists between a single end-user and a public ATM

network, between a single end-user and a private ATM switch, or
between a private ATM switch and the public ATM network.
 The NNI exists between switches in a single public ATM
network. NNIs may also exist between two private ATM
switches.
 The ICI is located between two public ATM networks.

The major differences between these two types of interfaces are

administrative and signaling related. The only type of signaling
exchanged across the UNI is that required to set up a Virtual Channel
for the transmission.

Communication across the NNI and the ICI will require signaling for
virtual–path and virtual–channel establishment, together with various
exchange mechanisms for the exchange of information such as routing
tables, etc.

Let us take an example to understand how the ATM network

works

 Let there be a user 1 in Delhi who wishes to transfer a data file-to

user 2 in Bangalore. A virtual channel is created and a virtual
path is established from switch to switch within the public ATM
network in Delhi (ATM Network 1) which, in turn, establishes
contact with the public ATM network in Bangalore (ATM
Network 2).
 ATM Network 2 also establishes a virtual path from switch to
switch within the network and with the private ATM Switch at
the destination. The private ATM network completes the virtual
path by establishing a virtual channel with User 2 in Bangalore.
 At each interface in this network, a unique virtual path identifier
(VPI) and the virtual channel identifier (VCI) is established for
this transmission. These identifiers are significant only for a
specific switch and two nodes adjacent to it in the virtual path.
Each node within the virtual path (including both the end-users
and the switches) maintain a pool of inactive identifiers to be
used as needed.
 User 1 or User 2 terminates the cell and the virtual path is
discontinued. The VCI and VPI values are returned to the pool of
available values for each switch.

Notice that only the user at either end of the transmission deal with the
48–byte information load within the cell. At each stage of the
transmission, the switch is only concerned with accepting the cell from
one port, changing the VPI/VCI according to its tables, and routing the
cell out the appropriate switch port.

The ATM Cell

ATM transmits all the information in small, fixed–size packets called

cells. Each individual ATM cell consists of a 5–byte cell header and 48
bytes of data. The ATM network uses the header to support the virtual
path and the virtual channel routing, and to perform a quick error check
for corrupted cells.
Bytes
5 48
Header User data

Figure 2: An ATM Cell

The Header Format

The structure of the header is different in UNI and NNI. In the network–
network interface, the virtual path identifier field is expanded from 8 to
12 bits.

8 7 6 5 4 3 2 1

Generic Flow Control* Virtual Path Identifier

Virtual Path Identifier Virtual Channel Identifier

Virtual Channel Identifier

Virtual Channel Identifier Payload Type ID CLP

Header Error Control

INFORMATION PAYLOAD (48 Bytes)

Figure 3: User–Network Interface

8 7 6 5 4 3 2 1

Virtual Path Identifier

Virtual Path Identifier Virtual Channel Identifier

Virtual Channel Identifier

Virtual Channel Identifier Payload Type ID
CLP
Header Error Control

INFORMATION PAYLOAD (48 Bytes)

Figure 4: Network–Network Interface
Let’s now look at the characteristics of each of the fields of the header
format of an ATM cell.

Generic Flow Control (GFC)

The GFC field of the header is only defined across the UNI and does not
appear in the NNI.

Function

 It controls the traffic flow across the UNI.

Virtual Path Identifier (VPI)

The VPI is an 8–bit field for the UNI and a 12–bit field for the NNI

Function

 It constitutes a routing field for the network and is used to

identify virtual paths. In an idle cell, the VPI is set to all 0’s.

 Together with the Virtual Channel Identifier, the VPI provides a

unique local identification for the transmission.

Virtual Channel Identifier (VCI)

It is a 16–bit field used to identify a virtual channel. For idle cells, the
VCI is set to all 0’s.

Function

 It functions as a service access point it and is used for routing to

and from the end-user.
 Together with the Virtual Path Identifier, the VCI provides a
unique local identification for the transmission.

Payload Type Identifier (PTI)

The PTI field indicates the type of information in the information field.
The value in each of the three bits of PTI indicates different conditions.

Bit 1 is set to 1 to identify operation, administration, or maintenance

cells (i.e., anything other than data cells).
Bits 2 is set to 1 to indicate that congestion was experienced by a data
cell in transmission and is only valid when bit 4 is set to 0.

Bit 3 is used to convey information between end-users.

Cell Loss Priority (CLP)

The 1–bit CLP field is used for indication of the priority of the cell. It is
used to provide guidance to the network in the event of congestion.
When set to value 1, it indicates that the cell is subject to discard within
the network. When the CLP value is set to 0, it indicates that the cell is
of relatively high priority and should be discarded only in situations
when no alternative is available.

Header Error Control (HEC)

Each ATM cell includes an 8–bit HEC that is calculated based on the
remaining 32 bits of the header.

Function:

 It detects all single–bit errors and some multiple–bit errors. As an

ATM cell is received at a switch, the HEC of the cell is compared
and all cells with HEC discrepancies (errors) are discarded. Cells
with single–bit errors may be subject to error correction if
supported or discarded. When a cell is passed through the switch
and the VPI/VCI values are altered, the HEC is recalculated for
the cell prior to being passed out the port.

Advantages of small, fixed-sized cells

Here is a list of some advantages of a cell.

1. Reduced queuing delay for a high priority cell;

2. Easy to implement the switching mechanism in hardware;
3. The fixed cell size ensures that time–critical information such as
voice or video, is not adversely affected by long data frames or
packets;
4. The header is organised for efficient switching in high–speed
hardware implementations and carries pay–load–type
information, virtual – circuit identifiers, and header error check.

ATM Classes Of Service

ATM is connection oriented and allows the user to specify the resources
required on a per–connection basis (per SVC) dynamically. There are
five classes of service defined for ATM (as per ATM Forum UNI 4.0
specification).

Service class Quality of Service Parameter

Constant bit rate CBR class is used for emulating circuit switching.
(CBR) The cell rate is constant with time. CBR applications
are sensitive to cell–delay variation. Examples of
applications that can use CBR are telephone traffic
(i.e., nx64 kbps), video conferencing, and television.
Variable bit rate VBR–NRT class allows users to send traffic at a rate
–real time (VBR that varies with time depending on the availability of
– RT) user information. Statistical multiplexing is provided
to make optimum use of network resources.
Multimedia e–mail is an example of VBR–NRT.
Variable bit This class is similar to VBR–NRT but is designed for
rate–non real applications that are sensitive to cell–delay variation.
time (VBR– Examples of real–time VBR are voice with speech
NRT) activity detection (SAD) and interactive compressed
video.
Available bit ABR class provides rate–based flow control and is
rate (ABR) aimed at data traffic such as file transfer and e–mail.
Although the standard does not require the cell
transfer delay and cell–loss ratio to be guaranteed or
minimised, it is desirable for switches to minimise
delay and loss as much as possible. Depending upon
the state of congestion in the network, the source is
required to control its rate. The users are allowed to
declare a minimum cell rate, which is guaranteed to
the connection by the network.
Unspecified bit UBR class is the catch–all other class and is widely
Rate (UBR) used today for TCP/IP.

The ATM Forum has identified certain technical parameters to be

associated with a connection.
ATM Technical Parameters

Technical Definition
Parameters
Cell loss CLR is the percentage of cells not delivered at their
ratio (CLR) destination because they were lost in the network due
to congestion and buffer overflow.
Cell transfer The delay experienced by a cell between network
delay entry and exit points is called the CTD. It includes
(CTD) propagation delays, queuing delays at various
intermediate switches, and service times at queuing
points.
Cell delay CVD is a measure of the variance of the cell transfer
variation delay. High variation implies larger buffering for
(CVD) delay–sensitive traffic such as voice and video.
Peak cell The maximum cell rate at which the user will
rate (PCR) transmit. PCR is the inverse of the minimum cell
inter–arrival time.
Sustained This is the average rate, as measured over a long
cell rate interval, in the order of the connection lifetime.
(SCR)
Burst This parameter determines the maximum burst that
tolerance can be sent at the peak rate. This is the bucket–size
(BT) parameter for the enforcement algorithm that is used
to control the traffic entering the network.

ATM Technical Parameters

Finally, there are a number of ATM classes of service. These classes

are:

ATM Classes of Service

Classes of Service CBR VBR –NRT VBR – RT ABR UBR

CLR Yes Yes Yes yes No
CTD Yes No Yes no No
CDV Yes Yes Yes No No
PRC Yes Yes Yes No Yes
SCR No Yes Yes No No
BT @ PCR No Yes Yes No No
Flow control No No No Yes No

Its extensive class–of–service capabilities make ATM the technology of

choice for multimedia communications.
3.8 ATM Traffic Control

An ATM network needs efficient traffic control mechanisms to allocate

network resources in such a way as to separate traffic flows according to
the various service classes and to cope with potential errors within the
network at anytime. The network should have the following traffic
control mechanisms:

 Network Resource Management

 Connection Admission Control
 Usage Parameter Control and Network Parameter Control
 Priority Control
 Congestion Control.

Network Resource Management

Network Resource management deals with allocation of network

resources in such a way that traffic is separated on the basis of the
service characteristics. A tool of network resource management which
can be used for traffic control is the virtual path technique. A Virtual
Path Connection (VPC) groups several Virtual Channel Connections
(VCCs) together such that only the collective traffic of an entire virtual
path has to be handled. In this type of set up, priority can be supported
by re–aggregating traffic types requiring different qualities of service
through virtual paths. Messages for the operation of traffic control can
be more easily distributed, a single message referring to all the virtual
channels within a virtual path will do.

Connection Admission Control

Connection Admission Control is the set of actions taken by the network

in protecting itself from excessive loads. When a user requests a new
virtual path connection or virtual channel connection, the user needs to
specify the traffic characteristics in both directions for that connection.
The network establishes such a connection only if sufficient network
resources are available to establish the end–to–end connection with the
required quality of service. The agreed quality of service for any of the
existing channels must not be affected by the new connection.

Usage Parameter Control and Network Parameter Control

After a connection is accepted by the Connection Admission Control

function, the UPC function of network monitors the connection to check
whether the traffic conforms to the traffic contract.
The main purpose of UPC/NPC is to protect the network resources from
an overload on one connection that would affect the quality of service of
other already established connections.

Usage Parameter Control (UPC) and Network Parameter Control (NPC)

do the same job at different interfaces. The UPC function is performed
at the user network interface, while the NPC function is performed at the
network node interface.

Functions performed by the Usage Parameter Control include:

 Checking the validity of VPI/VCI values

 Monitoring the traffic volume entering the network from all
active VP and VC connections to ensure that the agreed
parameters are not violated.
 Monitoring the total volume of the accepted traffic on the access
link.
 Detecting violations of assigned parameters and taking
appropriate actions.

Priority Control

Priority control is an important function as its main objective is to

discard lower priority cells in order to protect the performance of
higher–priority cells.

Congestion Control

Congestion is a state of network wherein the network resources are

overloaded. This situation indicates that the network is not able to
guarantee the negotiated quality of service to established connections
and to the new connection requests. ATM Congestion Control refers to
the measures taken by the network to minimise the intensity, spread and
duration of network congestion.

3.9 Benefits of ATM

1. As a high–bandwidth medium with low delay and the capability

to be switched or routed to a specific destination, ATM provides
a uniformity that meets the needs of the telephone, cable
television, video, and data industries. This universal compatibility
makes it possible to interconnect the networks – something that is
not currently possible because of the various transmission
standards used by each industry.
2. One of the key advantages of ATM is its ability to transmit video
 without creating a jittery picture of losing the synchronization of the sound and
picture.
3. ATM is also extremely fast and provides dynamic bandwidth for bursty traffic.
4. AT&T has developed ATM switches capable of transmitting 20
gigabits of data per second (Gbps) and a shared switch that can transmit up to 662
Gbps.
5. Telephone networks connect every telephone to every other telephone using
a dedicated path, but carry narrow bandwidth
signals. Cable networks carry broadband signals, but only
connect subscribers to centralised locations. To build a network that would provide
a dedicated connection between sender and receiver for broadband
communications would be prohibitively expensive. For this reason, ATM seems to
be the best hope since it can use existing networks to deliver simple voice and data
as well as complex and time–sensitive television signals. ATM can also handle bi–
directional communications easily.
6. Unlike packet switching, ATM is designed for high–performance multimedia
networking.

3.10 ATM Applications

ATM technologies, standards, and services are being applied in a wide range of
network environments.

ATM Services

Service providers globally are introducing or already offering ATM

services to their business users.

ATM Work Group and Campus Networks

Enterprise users are deploying ATM campus networks based on the ATM
LANE standards. Workgroup ATM is more of a niche market with the wide
acceptance of switched–Ethernet desktop technologies.

ATM Enterprise Network Consolidation

A new class of products has evolved as an ATM multimedia network–

consolidation vehicle. It is called an ATM Enterprise Network switch. A full–
featured ATM ENS offers a broad range of in–building (e.g., voice, video. LAN,
and ATM) and wide-area interfaces (e.g leased line, circuit switched, frame relay
and ATM at narrowband and broadband speeds) and supports ATM switching,
voice networking, frame–relay SVCs, and integrated multi-protocol routing.
Multimedia Virtual Private Networks and Managed Services

Service providers are building on their ATM networks to offer a broad

range of services. Examples include managed ATM, LAN, voice and
video services.

Frame–Relay Backbones

Frame–relay service providers are deploying ATM backbones to meet

the rapid growth of their frame–relay services to use as a networking
infrastructure for a range of data services, and to enable frame relay to
ATM service interworking services.

Internet Backbones

Internet service providers are likewise deploying ATM backbones to

meet the rapid growth of their frame–relay services, to use as a
networking infrastructure for a range of data services, and to enable
Internet class–of–service offerings and virtual private intranet services.

Residential Broadband Networks

ATM is the networking infrastructure of choice for carriers establishing

residential broadband services, driven by the need for highly scalable
solutions.

Carrier Infrastructure for the Telephone and Private–Line

Networks

Some carriers have identified opportunities to make more effective use

of their SONET/SDH fibre infrastructure by building an ATM
infrastructure to carry their telephony and private–line traffic.

4.0 CONCLUSION

This unit has discussed the different types of switching techniques and
how the Asynchronous Transfer Mode (ATM) – layered architecture
compares with the OSI model.

Also, you have been taken through how the ATM protocol works,
together with a detailed discussion on the ATM network, the ATM cell,
the ATM traffic control and classes of services, benefits of ATM and its
applications.
5.0 SUMMARY

 Asynchronous Transfer Mode (ATM) is a high–performance,

cell–oriented switch and multiplexing technology that utilises
fixed–length packets to carry different types of traffic.

 ATM is a technology defined by protocol standards created by

the ITU–T, ANSI, ETSI and the ATM Forum

 ATM is asynchronous because cells are not transferred

periodically. Cells are given time slots on demand.

 ATM is a technology that will enable carriers to capitalise on a

number of revenue opportunities through multiple ATM classes
of service; high speed Local–Area Network (LAN)
interconnection; voice, video, and future multimedia applications
in business markets in the short term; and in community and
residential markets in the longer term.

 ATM reduces infrastructure costs through efficient bandwidth

management, operational simplicity, and the consolidation of
overlay networks.

6.0 TUTOR–MARKED ASSIGNMENT

i. Fill in the blanks;

1. ATM cells are ………………….with ……………header

formats.
2. Asynchronous Transfer Mode (ATM) is also known as
…………..
3. ATM is fundamentally a ………….. switching technology
4. Multirate circuit switching is an enhancement of
….............
5. The two main approaches to packet switching
are………….and ………….

ii. What is ATM?

iii. Differentiate between Datagram and Virtual circuit.
iv. List the types of ATM connections
v. ATM switches support three kinds of interfaces. List and explain
each of them.
vi. Draw and explain the structure of an ATM cell.

7.0 REFERENCES/FURTHER READING

 UNIT 6 DATA TRANSMISSION AND MULTIPLEXING
1.0 INTRODUCTION

In the previous unit, the basics of computer network were discussed. This unit
covers topics related to the physical layer, which will comprise the difference
between data rate and bandwidth, analog and digital and finally, characteristics of
different transmission media.

2.0 OBJECTIVES

By the end of this unit, you will be able to:

 types of transmission
 domain concepts – time and frequency
 difference between analog & digital signal
 type of transmission impairments
 transmission media.
3.0 MAIN CONTENT

3.1 Transmission Terminology

Data transmission occurs between transmitters and receivers over some

transmission medium.

Transmission media may be classified as:

 Guided
 Unguided
 In both cases, communication is in the form of electromagnetic
waves.

With guided media, the waves are guided along a physical path.
Examples of guided media are twisted pair, coaxial cable, and optical
fibre. Unguided media provide a means for transmitting electromagnetic
waves but do not guide them; examples are propagation through air,
vacuum and seawater. In this unit, we will discuss about guided media
only.

A transmission may be

 Simplex
 Half–duplex
 Full duplex

In simplex transmissions, signals are transmitted in only one direction;

one station is a transmitter and the other is the receiver. In the half–
diplex operation, both stations may transmit but only one at a time. In
full–duplex, operation, both stations may transmit simultaneously. In the
latter case, the medium is carrying signals in both directions at the same
time.

3.1.1 Time–Domain Concept

As a function of time, an electromagnetic signal can be either

continuous or discrete. A continuous signal is one in which the signal
intensity varies in smooth fashion over time. There are no breaks or
discontinuities in the signal. A discrete signal is one in which the signal
intensity maintains a constant level for some period of time and then
changes to another constant level.
3.1.2 Frequency Domain Concepts
In practice, an electromagnetic signal will be made up of many frequencies.
It can be shown, using a discipline known as Fourier analysis, that any signal is
made up of components at various frequencies, in which each
component is sinusoidal.

So, we can say that for each signal, there is a time–domain function (t) that
specifies the amplitude of the signal at each instance of time. Similarly, there
is a frequency–domain function S(t) that specifies the constituent frequency of the
signal. The spectrum of the signal is the range of frequencies that it contains.

3.1.3 Relationship between Data Rate and Bandwidth

The concept of effective bandwidth is somewhat a fuzzy one. It is the band within
which most of the energy is confined. The term “most” in this context is somewhat
arbitrary. The important issue here is that, although a given waveform may contain
frequencies over a very broad range, as a practical matter, any transmission
medium that is used will be able to accommodate only a limited band of
frequencies. This, in turn, limits the data rate that can be carried on the
transmission.

3.2 Analog and Digital Data Transmission

The terms ‘analog’ and ‘digital’ correspond, roughly, to continuous and discrete,
respectively. These two terms are used frequently in data communications at least in
three contexts:

 Data
 Signaling
 Transmission

3.2.1 Data

Analog signal takes on continuous values on some interval. For example,

voice and video are continuously varying patterns of intensity. Most data collected
by sensors, such as temperature and pressure, take on continuous values. Digital
data take on discrete values; examples are text and integers.

3.2.2 Signals

In a communication system, data are propagated from one point to another by

means of electrical signals. An analog signal is a continuously varying
electromagnetic wave that may be propagated over a variety of media, depending on
spectrum.
A digital signal is a sequence of voltage pulses that may be transmitted
over a wire medium; for example, a constant positive voltage level may
represent binary 1, and a constant negative voltage level may represent
binary 0.

3.2.3 Transmissions

Both analog and digital signals may be transmitted on suitable

transmission media. Analog transmission is a means of transmitting
analog signal without regard to their context.

Analog Analog signal Digital signal

data  Signal occupies  Analog data are encoded
the same using a
spectrum as the codec to produce a digital bit
analog data. stream
 Analog data are
encoded to
occupy different
portions
of spectrum.
Digital  Digital data are  Signal consists of two voltage
Data encoded levels to represent the two
using a modem binary values
to produce  Digital data are encoded to
Analog signal produce a digital signal with
desired properties.

3.3 Transmission Media

The purpose of the physical layer is to transport a raw bit stream from
one machine to another. Various physical media can be used for the
actual transmission. Each one has its own niche in terms of bandwidth,
delay, cost, and ease of installation and maintenance. Media are roughly
grouped into guided media, such as copper wire and fibre optics, and
unguided media, such as radio and lasers through the air. We will look at
these in this section and next one.

3.3.1 Twisted Pair

Although the bandwidth characteristic of magnetic tape is excellent, the

delay characteristics are poor. Transmission time is measured in minutes
or hours, not milliseconds. For many applications, an on–line connection
is needed. The oldest and still most common transmission medium is
twisted pair. A twisted pair consists of two insulated copper wires, typically
about 1mm thick. The wires are twisted together in a helical form, just like a DNA
molecule. The purpose of twisting the wires is to reduce electrical interference from
similar pairs close by. The common application of the twisted pair is the telephone
systems.

Twisted pairs can be used for either analog or digital transmission. The bandwidth
depends on the thickness of the wire and the distance travelled, but several
megabytes/sec can be achieved for a few kilometres in many cases. Due to their
adequate performance and low cost, twisted pairs are widely used and are likely to
remain so for years to come.

Twisted pair cabling comes in several varieties, two of which are important for
computer networks Category 3 twisted pairs consist of two insulated wires gently
twisted together. Four such pairs are typically grouped together in a plastic sheath
for protection and to keep the eight wires together.

Starting around 1988, the more advanced category 5 twisted pairs were introduced.
They are similar to Category 3 pairs, but with more twists per centimetres and
insulation, which result in less cross talk and a better quality signal over longer
distances, making them more suitable for high–speed computer communication.
Both of these wiring types are often referred to as UTP (Unshielded Twisted Pair, to
contrast them with the bulky, expensive, shielded twisted pair cables IBM
introduced in the early 1980s, but which have not proven popular outside of IBM
installations.

3.3.2 Baseband Coaxial Cable

Another communication transmission medium is the coaxial cable. It has better

shielding than twisted pairs, so it can span longer distances at higher speeds. Two
kinds of coaxial cable are widely used. One kind,
50–ohm cable is commonly used for digital transmission and is the subject of this
section. The other kind, 75–ohm cable, is commonly used for analog transmission
and will be described in the next section. This distinction is based on historical,
rather than technical factor, (e.g., early dipole antennas had an impedance of 300
ohms, and it was easy to build
4:1 impedance matching transformers).

A coaxial cable consists of a stiff copper wire as the core, surrounded by an

insulating material. The construction and shielding of the coaxial cable give it a
good combination of high bandwidth and excellent noise immunity. The bandwidth
possible depends on the cable length. For 1 km cables, a data rate of 1 or 2 Gbps
is feasible. Longer cables can also
be used, to be widely used within the telephone systems but have now
largely been replaced by fibre optics on long–haul routes. In the United
States alone, 1000 km of fibre is installed every day (counting a 100 km
bundle with 10 strands of fibre as 1000 km). Sprint is already 100 per
cent fibre, and the other major carriers are rapidly approaching that.
Coax is still widely used for cable television and some local area
networks.

3.3.3 Broadband Coaxial Cable

The other kind of coaxial cable system uses analog transmission on

standard cable television cabling. It is cabled broadband. Although the
term “broadband” comes from the telephone world, where it refers to
anything wider than 4kHz, in the computer networking world,
“broadband cable” means any cable network using analog transmission.

Since broadband networks use standard cable television technology, the

cables can be used up to 300 MHz (and up to 450 MHz) and can run for
nearly 100 km due to the analog signaling, which is much less critical
than digital signaling. To transmit digital signals on an analog network,
each interface must contain electronics to convert the outgoing bit
stream to an analog signal, and the incoming analog signal to a bit
stream. Depending on the type of these electronics, 1 bps may occupy
roughly 1 Hz of bandwidth. At higher frequencies, many bits per Hz are
possible using advanced modulation techniques.

Broadband systems are divided up into multiple channels frequently, the

6MHz channels used for television broadcasting. Each channel can be
used for analog television, CD–quality audio or a digital bit stream at,
say, 3 Mbps, independent of the others. Television and data can be
mixed on one cable.

3.4 Multiplexing
In communication, multiplexing is a technique that transmits signals
from several sources over a single communication channel. So in order
to minimize the cost of communication bearer, various means of sharing
a communication channel between several users, have been devised;
these are known as multiplexing. In this section, we will discuss about
two multiplexing techniques: FDM & TDM.

Frequency Division Multiplexing (FDM)

In FDM, the frequency spectrum is divided among the logical channels

with each user having exclusive possession of some frequency band.
Time Division Multiplexing (TDM)

In TDM, the users take turns (in a round robin), each one is periodically getting the
entire bandwidth for a little burst of time. Television broadcasting provides an
example of both kinds of multiplexing. Each TV channel operates in a different
frequency range, which is a portion of the allocated spectrum, with the inter–
channel separation great enough to prevent interference. This system is an example
of FDM. During the transmission of any program (Serial/film), there is an
advertisement as well. These two alternate in time on the same frequency. This is an
example of TDM.

4.0 CONCLUSION

This unit covered topics relating to the physical layer, which comprises the
difference between data rate and bandwidth, analog and digital transmission
and characteristics of transmission media except wireless transmission.

Also, the unit has taken you through the types of transmission impairments and how
to minimize them.

5.0 SUMMARY

Transmissions can take place through the media of guided and unguided type and it
can be simplex, half–duplex and full–duplex. In simplex, the data/signals are
transmitted in one direction by a station i.e., by the sender; in half–duplex, the
transmission can be done in one direction at a time whereas in full–duplex, the
transmission can take place in directions. The concept of time domain and
frequency domain deals with the electromagnetic signals and components at
various frequencies spectrum. The concept of analog and digital transmission deals
with data signaling and transmission which can be analog data i.e., signal occupies
same spectrum and digital data are encoded using a modem to produce analog
signal. The other type of signal is digital, which uses a bit stream.

Media used in transmission are of the magnetic type and it is one of the most
common ways to store data physically on tapes, floppy disks and hard disks.
Twisted pairs are used both for analog as well as digital transmission. Twisted pair
can be Cat 3, or Cat 5. Both of them are UTP cables. Baseband cable is used for
longer distances at high–speed 50 ohm and 75 ohm are normally used. Broadband
Coaxial cable refers to anything wider than 4KHz. Broadband is divided into
multiple channels and each channel can be used for analog signal also. It is used for
CD– quality audio or a bit stream.
6.0 TUTOR MARKED ASSIGNMENT

i. What is the difference between data rate and bandwidth?

………………………………………………………………………
………………………………………………………………………
……………………………………………………………………….
……………………………………………………………………….

ii. List the characteristics of broadband coaxial cable

…………………………………………………………………………
…………………………………………………………………………..
…………………………………………………………………………..
……………………………………………………………………………

7.0 REFERENCES/FURTHER READING

UNIT 7 MEDIUM ACCESS CONTROL AND DATA LINK LAYER

1.0 INTRODUCTION

This unit introduces the design of Data Link Layer and its Medium
Access Control Sublayer. This includes various protocols for achieving
reliable, efficient communication. It also covers the study of nature of
errors, causes and how they can be detected and corrected.

The MAC sublayer contains protocols which determine who goes next
on a multi access channel.

2.0 OBJECTIVES

By the end of this unit, you will be able to:

 understand various error–handling methods.

 explain the various flow control methods.
 Identify MAC sublayer protocols like CSMA/CD, Token Passing

3.0 MAIN CONTENT

3.1 Data Link Layer

To exchange digital information between devices A and B, we require

an interconnecting transmission medium to carry the electrical signals; a
standard interface and the physical layer to convert bits into electrical
signals and vice–versa.
This has certain limitations:

 If the electrical signal gets impaired due to the encountered

interference with other signals or electromagnetic waves from
external sources, error may be introduced in the data bits.

 Errors can also be introduced if the receiving device is not ready

for the incoming signal, hence resulting in the loss of some
information.

The Data Link Layer constitutes the second layer of the hierarchical OSI
Model. The Data Link Layer together with Physical Layer and the
interconnectivity medium provide a data link connection for reliable
transfer of data bits over an imperfect physical connection.

It accomplishes the task by having the sender break the input data up
into data frames, transmit the frames sequentially and process the
acknowledgement frames sent back by the receiver. It is up to the Data
Link Layer to create and recognise frame boundaries.

Another issue that arises in Data Link Layer is how to keep a fast
transmitter from overflowing a slow receiver in data.

The Data Link Layer incorporates certain processes, which carry out
error control, flow control and the associated link management
functions. The data block along with the control bits is called a frame.

Data Link Layer is divided into two sublayers:

Logical Link Control (LLC) concerned with providing a reliable

communication path between two devices. It is also involved with flow
control and sequencing of class. The LLC is non–architecture–specific
which is the same for all IEEE–defined LANs.

Medium Access Control focuses on methods of sharing a single

transmission medium.
3.1.1 Services Provided By Data Link Layer (Logical Link
Control)

 Framing: Some control bits are added to the data packets from
network layer to mark the start and end of a frame. This is done
using character count, character or bit stuffing.

 Flow Control: Flow Control deals with how to keep the fast
sender from overflowing a slow receiver by buffer at the receiver
sides and acknowledgement.

 Error Detection and Correction Codes: Various methods used

for error detection and corrections are: parity bit, cyclic
redundancy check, checksum, hamming code, etc.

3.1.2 Retransmission Strategies.

In this section, we will discuss several retransmission strategies, which

are also considered as a flow control and error control mechanism.

Stop and Wait

The sender allows one message to be transmitted, checked for errors and
an appropriate ACK (Positive Acknowledgement) or NAK (Negative
Acknowledgement) returned to the sending station. No other data
messages can be transmitted until the receiving station sends back a
reply, thus the same STOP & WAIT is derived from the originating
station sending a message, stopping further transmission and waiting for
a reply.

Its major drawback is the idle line time that results when the stations are
in the waiting period. If the ACK is lost, then the master station
retransmits the same message to the receiver side. The redundant
transmission could possibly create a duplicate frame. A typical approach
to solving this problem is the provision for a sequence number in the
header of the message. The receiver can then check for the sequence
number to determine if the message is a duplicate. The Stop and Wait
mechanism requires a very small sequence number, since only one
message is outstanding at any time. The sending and the receiving
station only use a one bit alternating sequence of 0 and 1 to maintain the
relationship of the transmitted message and its ACK/NAK status.

Sliding Window
The data control signals flow from sender to receiver in a more
continuous manner and several messages can be outstanding at any one
time.
The transmitting station maintains a sending window that maintains the
number of frames it is permitted to send to the receiving station and the
receiving station also maintains a receiving window that performs
complementary functions. The two sides use the window to coordinate
the flow of frames between each other. The window wrap around is used
to reuse the same set of numbers for different frames. There are sliding
window techniques:

(1) Go Back N
(2) Selective Repeat

The following two diagrams (Figure 1 and Figure 2) explain the function
of Go Back N and Selective Repeat respectively.

GO Back N

| Time Period |

Figure 1: Go Back N

This is a sliding window technique. It allows data and control messages

to be transmitted continuously without waiting for its acknowledgement
from the receiver. In the event, if an error is detected at the receiving
side, the erroneous message is retransmitted, as well as all other frames
that were transmitted after the erroneous message.

Selective Repeat
 Figure 2: Selective Repeat

This method provides for a more refined approach. In contrast to the Go

Back N, the only messages retransmitted are those for which negative
acknowledgement is received.

Studies reveal that the selective repeat mechanism obtains greater

throughout than the Go Back N. Selective Repeat mechanism requires
additional logic to maintain the sequence of the recent message and
merge it into the proper place as they queue at the proper site.

3.2 Medium Access Control Sublayer

In any broadcast network, the key issue is how to determine who gets to
use the channel when there is competition for it. The protocols used to
determine who goes next on a multi–access channel belong to a sub–
layer of a Data Link Layer called MAC sublayer.

3.2.1 Contention Based Media Access Protocols

Contention is what happens at a staff meeting when several people start

to talk at the same time. In contention protocol, no policeman controls
usage of the communication channel.

All workstations on a contention Network share a common transmission

channel. Messages are broadcast on that channel and may be overheard
by all attached workstations. A workstation responds only to a message
with its address: Message intended for different modes are ignored.

Messages to be transmitted are converted to packets and are sent when

ready, without verifying the availability of the channel. When
transmission of a channel overlap with that of another, collision occurs.
Colliding packets with their messages are destroyed.
3.2.2 Polling–Based MAC Protocols

Polling involves the channel control of all workstations in a network.

The primary workstation acts like a teacher going down the rows of the
classroom asking each student for homework. When one student has
answered, the next is given a chance to respond.

A polling network contains two classes of workstations, the primary

workstation and the multiple secondary workstations connected to it. A
buffer that can temporarily store messages is associated with each
secondary workstation. When a workstation has information to transmit,
the data is passed to the buffer. The frames are held until the central
controller polls the workstation.

These are the two possibilities for the path of a message from source to
destination workstation:

 All messages may require passing to the central workstation,

which route them to their destination.

 Messages may be sent directly.

Polling technique can be said to maintain a tight control over the

network than do contention–based protocols.

Token Passing

The network continuously circulates a special bit pattern known as a

token, among all the modes in the network.

Each token contains network information, comprising a header, a data

field and a trailer. Any mode willing to send a frame has to grass a token
first. Now, let us talk about some standards.

3.2.3 IEEE Standard 802.3 and Ethernet

It is for CSMA/CD LAN. When a station wants to transmit, it listens to

the cable. If the cable is busy, the station waits until it goes idle,
otherwise, it transmits immediately. If two or more stations
simultaneously begin transmitting on an idle cable, they will collide. All
colliding stations then terminate their transmissions, wait a random time
and repeat the whole process all over again.
3.2.4 IEEE Standard 802.4 Token Bus

Token Bus combines features of Ethernet and token ring (discussed in

the next section). It combines the physical configuration of Ethernet (bus
topology) and collision–free (predictable delay) feature of token ring.
Token bus is a physical bus that operates as logical ring using tokens.

It is a linear cable onto which the stations are attached. When the logical
ring is initialized, the highest numbered station may send the first frame
after it is done, it passes permission to its immediate neighbour by
sending the neighbour a special control frame called a token.

3.2.5 IEEE Standard 802.5 Token Ring

In a token ring, the token circulates around the ring whenever all stations
are idle. When a station wants to transmit a frame, it is required to seize
the token and remove it from the ring before transmitting. This action is
done by inverting a single bit in the 3–byte token which instantly
changes it into the first 3 bytes of a normal data frame. Because there is
only one token, only one station can transmit at a given instant, thus
solving the channel access problem.

4.0 CONCLUSION

In this unit, you have been introduced to a number of protocols such as

IEEE Standard 802.3 and Ethernet, IEEE Standard 802.4 Token Bus,
IEEE Standard 802.5 Token Ring; etc. for MAC sub–layer, which is a
part of data link layer.

Also, the unit has discussed issues relating to error handling methods.

5.0 SUMMARY

In this unit, an introduction to issues of DLL and various methods for

allocation of a common channel to the competing users were discussed.
LANS are dominated by four types of architecture: Ethernet, Token Bus,
Token Ring and Fibre Distributed Data Interface (FDDI). Each of them
has its own advantages and disadvantages. Depending upon the
requirements, the choice is made.
6.0 TUTOR MARKED ASSIGNMENT

i. How is Selective Repeat better than Go Back N?

……………………………………………………………………
……………………………………………………………………
……………………………………………………………………
……………………………………………………………………

ii. What are the four LAN architectures?

……………………………………………………………………
……………………………………………………………………
……………………………………………………………………
……………………………………………………………………
……………………………………………………………………
…………………………………………………………………..

7.0 REFERENCES/FURTHER READING

Module 3: Network Administration
Unit1 Network Administration: Scope, Goals, Philosophy and
Standards
Unit2 Network Protocols
Unit3 Network, Transport and Application Layers

Unit1: Network Administration: Scope, Goals, Philosophy and

Standards

1.0
INTRODUCTION
Network and distribution processing systems are of critical and growing
importance in business, government and other organizations. Therefore,
networks must be managed for effectiveness and efficiency. This unit discusses
fundamental aspects of network administration.

2.0
OBJECTIVES

At the end of this unit, you should be

also to: Define network

administration
Know the scope of network administration
State the goals of system administration
Understand the challenges of system administration
State the Meta principles of system administration

3
3.0 MAIN CONTENT

3.1 What is network and system administration

Network and system administration is a branch of engineering that concerns the

operational management of human–computer systems. It is about putting
together a network of computers (workstations, PCs and supercomputers),
getting them running and then keeping them running in spite of the activities of
users who tend to cause the systems to fail.
A system administrator works for users, so that they can use the system to
produce work. However, a system administrator should not just cater for one
or two selfish needs, but also work for the benefit of a whole community.
Today, that community is a global community of machines and organizations,
which spans every niche of human society and culture, thanks to the Internet.
It is often a difficult balancing act to determine the best policy, which accounts
for the different needs of everyone with a stake in a system. Once a computer is
attached to the Internet, we have to consider the consequences of being directly
connected to all the other computers in the world.
In the future, improvements in technology might render system
administration a somewhat easier task – one of pure resource administration –
but, today, system administration is not just an administrative job, it is an
extremely demanding engineer’s job. It’s about hardware, software, user
support, diagnosis, repair and prevention. System administrators need to know a
bit of everything: the skills are technical, administrative and socio-
psychological.
The terms network administration and system administration exist separately
and are used both variously and inconsistently by industry and by academics.
System administration is the term used traditionally by mainframe and
Unix engineers to describe the management of computers whether they are
coupled by a network or not. To this community, network administration means
the management of network infrastructure devices (routers and switches). The
world of personal computers (PCs) has no tradition of managing individual
computers and their subsystems, and thus does not speak of system
administration, per se. To this community, network administration is the
management of PCs in a network. In this material, we shall take the first view,
since this is more precise.

4
Network and system administration are increasingly challenging. The
complexity of computer systems is increasing all the time. Even a single PC
today, running Windows NT, and attached to a network, approaches the level
of complexity that mainframe computers had ten years ago. We are now
forced to think systems not just computers.

3.2 Scope of Network administration

The management of a network, usually called network administration, consists
of procedures and services that keep the network running properly. An
important part of network management entails making sure that the network
is available (or up and running as IT professionals say) when employees and
managers need it. Other admin activities are:
- Monitoring the network capacity to ensure that all transmission
requirements can be met.
- Adding capacity to the network by increasing band width interconnecting
additional modes, or creating and interconnecting additional networks.
- Training people to use the network
effectively
- Assisting IT professionals in organizational applications that will make
good use of the network’s capabilities.

Backing up the network software and data regularly to protect against

the failure of network or any of its components
- Putting security procedures in place to make certain that only authorized
users have access to the network and ensuring that all security procedures are
followed
- Making sure the network personnel can respond quickly and effectively in
the event of a
network operational or security failure.
- Diagnosing and troubleshooting problems on the network and determining
the best course of action to take to solve them.

3.3 The goal of Network administration

The goal is to keep the network running properly and configuring and
managing services that are provided over the network.
There are many services that we use regularly. There are some which work in
the background enabling other services to run smoothly.

5
3.4 The challenges of system administration

System administration is not just about installing operating systems. It is about

planning and designing an efficient community of computers so that real users
will be able to get their jobs done. That means:
• Designing a network which is logical and efficient.
• Deploying large numbers of machines which can be easily upgraded later.
• Deciding what services are needed.
• Planning and implementing adequate security.
• Providing a comfortable environment for users.
• Developing ways of fixing errors and problems which occur.
• Keeping track of and understanding how to use the enormous amount
of knowledge which increases every year.
Some system administrators are responsible for both the hardware of the
network and the computers which it connects, i.e. the cables as well as the
computers. Some are only responsible for the computers. Either way, an
understanding of how data flow from machine to machine is essential as well as
an understanding of how each machine affects every other.

3.5 The Meta principles of system administration

Many of the principles in this course material derive from a single overriding
issue: they address the predictability of a system. The term system clearly
implies an operation that is systematic, or predictable – but, unlike simple
mechanical systems, like say a clock, computers interact with humans in a
complex cycle of feedback, where uncertainty can enter at many levels.
That makes human–computer systems difficult to predict, unless we somehow
fix the boundaries of what is allowed, as a matter of policy.

Principle (Policy is the foundation). System administration begins with a

policy – a decision about what we want and what should be, in relation to what
we can afford.
Policy speaks of what we wish to accomplish with the system, and what
we are willing to tolerate of behavior within it. It must refer to both the
component parts and to the environment with which the system interacts.
6
If we cannot secure predictability, then we cannot expect long-term
conformance with a policy.

Principle (Predictability). The highest level aim in system administration is to

work towards a predictable system. Predictability has limits. It is the basis
of reliability, hence trust and therefore security.
Policy and predictability are intertwined. What makes system administration
difficult is that it involves a kind of ‘search’ problem. It is the hunt for a
stable region in the landscape of all policies, i.e. those policies that can lead to
stable and predictable behavior. In choosing policy, one might easily promote
a regime of cascading failure, of increasing unpredictability that degenerates
into chaos. Avoiding these regimes is what makes system administration
difficult.
As networks of computers and people grow, their interactions become
increasingly complex and they become non-deterministic, i.e. not predictable in
terms of any manageable number of variables. We therefore face another
challenge that is posed by inevitable growth:

Principle (Scalability). Scalable systems are those that grow in accordance

with policy; i.e. they continue to function predictably, even as they increase in
size.
These meta-themes will recur throughout this material. The important point
to understand about predictability is that it has limits. Human–computer
systems are too complex and have too many interactions and dependencies to
be deterministic. When we speak of predictability, it must always be within a
margin of error. If this were not the case, system administration would not be
difficult.

3.6 Advice to the students

To study this subject, we need to cultivate a way of thinking which embodies a

basic scientific humility and some core principles:
• Independence or self-sufficiency in learning. We cannot always ask
someone for the right answer to every question.
7
 • Systematic and tidy work practices.
• An altruistic view of the system. Users come first: collectively and only then

individually.
• Balancing a fatalistic view (the inevitability of errors) with a
determination to gain firmer control of the system.
Some counter-productive practices could be
avoided:
• The belief that there exists a right answer to every problem.
• Getting fraught and upset when things do not work the way we expect.
• Expecting that every problem has a beginning, middle and an end
(some problems are chronic and cannot be solved without impractical
restructuring).
We can begin with a
checklist:
• Look for answers in manuals and newsgroups.
• Use controlled trial and error to locate problems.
• Consider all the information; listen to people who tell you that there is
a problem. It might be true, even if you can’t see it yourself.
• Write down experiences in an A–Z so that you learn how to solve the
same problem again in the future.
• Take responsibility for your actions. Be prepared for accidents.
They are going to happen and they will be your fault. You will have to
fix them.
• Remember tedious jobs like vacuum cleaning the hardware once a year.
• After learning about something new, always pose the question: how
does this apply to me?

SELF-ASSESSMENT EXERCISES

1. Is system administration management or engineering?

2. Why does the physical environment play a role in system
administration?
8
4.0 CONCLUSION

Network administration is concerned with establishing and administering

overall goals, policies and procedures of network management. This requires a
well rounded skills set and not just technical skills.

5.0 SUMMARY
Network and system administration is a branch of engineering that concerns
the operational management of human–computer systems. System
administration is not just about installing operating systems. It is about
planning and designing an efficient community of computers so that real users
will be able to get their jobs done. System administration begins with a policy –
a decision about what we want and what should be, in relation to what we
can afford. Policy speaks of what we wish to accomplish with the system, and
what we are willing to tolerate of behavior within it. To study this subject, we
need to cultivate a way of thinking which embodies a basic scientific humility
and some core principles:

6.0 TUTOR-MARKED ASSIGNMENTS

1. State the top-most principles that guide network and system
administrators
2. What kinds of issues does system
administration cover?
3. State the meta principles of system
administration
4. What are the challenges of system
administration?

7.0 REFERENCES/FURTHER READING

1. Burgess, M. (2004). Principles of Network and System Administration. (2nd
Ed.). Chichester, West Sussex , England: Wiley.

2. Forouzan, B.A, & Fegan, S.C. (2007). Data communications and

Networking (4th Ed). Mc Graw Hill.
9
3. Limoncelli, T. A.,Hogan, C. J. & Chalup, S. R (2007}. The Practice of
System and Network Administration. (2nd Ed.). Upper Saddle River, NJ:
Addison-Wesley.

4. Stallings, W. (2009). Data and computer communications ( 8th ed.). Upper

saddle River, NJ.: Pearson Education Inc.

10
UNIT 2: NETWORK PROTOCOL

1.0 INTRODUCTION

This unit discusses packets and protocols which are the fundamental building
blocks of data transmission over the network.

2.0 OBJECTIVES

By the end of this unit, you will be able to:

 state why are protocols and standards needed?

 state the function of a packet
 state the principal function of protocols in a network
 explain the layered approach to computer communications
 discuss some of the different protocols and their uses
 state how protocols work
 state the advantages and disadvantages of standards.

3.0 MAIN CONTENT

3.1 PROTOCOLS, PACKETS AND STANDARDS
All data that is transmitted across the network is put into packets containing
information about the source and destination of the data. These packets are
created using standards or protocols. Since there are many different network
configurations, there are many different protocols. By having a variety of
protocols, you can choose the one that best fulfills the needs of your
network.

3.1.1 Function of packets

The function of a packet is to carry data from one point to another.
Protocols require that packet contain some basic information about their source
and their destination and in many cases, protocols require that the packet
include a checksum. A checksum is a number that can be used to verify that 11
the packet has been transferred across the network without being
corrupted.
3.1.2 Packet Structure
The structure of the packet is extremely important. Useless a packet is
structured exactly as it is supposed to be; it is ignored by the receiving party
and assumed to be corrupted. Basic packet structure requires that the packet
include a header section, a data section, and in most cases, a cyclic
redundancy check (CRC) section (also called a trailer). Not every protocol
requires that a CRC be attached.

3.1.2.1 Header
The header section of a packet contains the routing information. This
information includes the source and destination of the packet. The header also
contains the number of the packet, which is generated when the packet is
created. In addition, the header can contain a protocol version number, the length
of the header, the type of service, the length of the entire packet, the flags, the
time to live, and other information.

3.1.2.2 Data
The data is the actual information that is being transmitted over the
network from one application to another. Each protocol has a predefined
maximum data size. If the data is larger than this maximum data size, the data is
broken into smaller pieces and transmitted in multiple packets.

3.1.2.3 CRC
A CRC (Cyclic redundancy check) is calculated prior to the data being sent and
attached to the bottom of a packet. At the destination, a new CRC is computed
and compared to the original to verify that the packet was not corrupted. A
CRC is usually attached to the bottom of a packet, but some protocols include
CRC within the header.

3.1.3 Creating packets

Before data is transmitted across the network, it is broken into smaller,
more manageable pieces called packets. All packets are numbered so they can
be put back together when they reach their destination. The header, which 12

contains the source address, destination address, and packet number, along
with other information, is attached to the beginning of the packet. A
CRC is then calculated and added to the end of
the packet.
3.1.4 Encapsulation
Encapsulation is the process of encoding data for transmitting it across the
network. Once a packet is created as described previously, in order for the
packet to be transmitted to its final destination, it may need to use a protocol in
addition to the one that it is currently using. A header and CRC are then added
to the newly created packet. This packet is an encapsulated packet. Figure 1
illustrates an encapsulated packet.

13
 Header CRC

Data
Figure 1 Encapsulated packet

3.2 PROTOCOLS
In computer networking, communication occurs between entities in different
systems. An entity is anything capable of sending or receiving information.
However, two entities cannot simply send bit streams to each other and expect to
be understood. For communication to occur, the entities must agree on a protocol.
A protocol is a set of nodes that govern data communications. A protocol defines
what is communicated, how it is communicated, and when it is communicated. The
key elements of protocol are syntax, semantics and timing.
Syntax: The term syntax refers to the structure or format of the data, meaning
the order in which they are presented. For example, a simple protocol might expect
the first 8 bits of data to be the address of the sender, the second 8 bits to be the
address of the receiver, and the rest of the stream to be message itself.
SEMANTICS:
The word semantics refers to the meaning of each section of bits. How is a
particular pattern to be interpreted, and what action is to be taken based on that
interpretation? For example, does the address identify the route to be taken or the
final destination of the message?
TIMING:
The term timing refers to two characteristics: When data should be sent and how
fast they can be sent. For example, if a sender produces data of 100mbps and the
receiver can process data
at only 1 mbps, the transmission will overload the receiver and some data will be lost.

oo
3.2.1 How Protocols Work at

When an application first attempts to transfer data across the network, the data is
usually t 24 large to be sent in a single transmission. To meet the need of
networking, the protocol th governs the transmission of the data first breaks the
data into packets. The protocol numbers
each of the packets so can later be put back together when they arrive at their
destination and transmits each of the packets across the network. In addition to this
numbering, information on the source, destination, and the protocol used is added
to the header of the packet.
Protocols are the first software layer to receive data that has been transmitted
across the network. After all packets have been received, they are put back
together using the numbers that were placed in the header at the origin of the
packet. Once the data has all been put back together, it can be used by the
application that the data was sent across the network to.

3.2.2 Functions of Protocols

The principal functions of protocols in a network are line access and collision
avoidance. Line access concerns how the sending device gains access the network
to send a message. Collision avoidance refers to managing message transmission
so that the messages do not collide with each other on the network. Other
functions of protocols are to identify each device in the communication path,
to secure the attention of the other device, to verify correct receipt of the
transmitted message, to verify that a message requires transmission because
it cannot be correctly interpreted and to perform recovery when errors occur.
3.3 The layered approach to computer communications
In order to enable two or more computers to communicate in a meaningful
manner, we must define with great care all aspects of the communication
process (i.e. we must define a ‘communications protocol’). By way of a
useful analogy, let us consider the situation in which the director of a company
in the UK wishes to communicate with a person in another company located in
China. The director may ask a secretary to put a call through and will
provide sufficient information for the secretary to identify the person who is to be
contacted. Here, the director will not give the actual phone number- it may be left
to the secretary to obtain this information. From this point, the director has
no further involvement until the phone connection is in place. The secretary
will locate and dial the number and this will initiate various electronic/software
activities. Neither the director nor the secretary has any interest in knowing
how the electronic and software systems will route the call. It may be
carried by electronic cables, fiber optic cables, or be routed via a satellite.
Additionally, it may use communications systems that route the call across the
Atlantic through the US and then across the Pacific Ocean, or it may be routed
in an easterly direction. These low-level issues are of little interest to the
secretary – a number is dialed and processes occur that result in a phone ringing
in an office somewhere in China. Hopefully, the intended recipient is available
and the secretary notifies the director. Both parties must now adopt/agree on a
common language and must exercise a degree of hand-shaking (in this sense we
mean that only one person should talk
at any one time). Finally, at the end of the conversation, an acceptable convention
is used to bring the call to a conclusion. All these issues form part of the
‘communications protocol’ that is needed to enable a useful dialogue and it is
important to note that the elements that underpin
the communication do not need to have any knowledge of the overall purpose
that they will serve. For example:

The establishment of a communications protocol that enables computers (and

other digital systems) to communicate is, in many ways, similar to the protocols
used to support the sort of phone conversation referred to in the above analogy
(although computer communications are perhaps more complex). To handle
the design implementation and maintenance of such systems, a ‘layered’
approach is adopted. In figure 2, we indicate two computers that need to
communicate. Perhaps, for example, an applications program running on
Node A wishes to send a data file to a similar program running on Node B (just
as in the same way the company director mentioned above wishes to talk to a
person in a remote location). In order to transmit the data a number of tasks must
be performed, and these are carried out by layers of software located on both
nodes.
Each layer carries out a number of specific tasks and directly
communicates with the immediately adjacent software layers. However, from a
logical point of view each layer communicates with a corresponding layer on
the remote computer – i.e. corresponding software layers located on the two
nodes have similar/equivalent functionality. The lowest
layer on either node is responsible for interfacing with the physical
interconnect.

Figure 2: An applications program on Node A wishes to send data to an

equivalent program on Node B
In order for Node A to transmit a data file to Node B, various events

must take place. For example:

Node A must inform the communications network of the identity of the

destination system
(Node B)
Node A must ensure that Node B is prepared to
receive the data.
The file transfer applications program on Node A must ensure that the
file management program on the destination system is ready to accept and
store the file.
If the file formats used on the two systems are incompatible, one or other
system must perform a format translation function.
File transfer must be carried out in an orderly manner and in such a
way that the two communicating machines do not block other network
traffic. This will involve splitting the data file into packets (chunks) and
appending various information to each packet.
 Node B provides acknowledgement of receipt
Node B reassembles the packet in other to reconstruct the
original data file
Node B must attempt to detect any errors in data it has received. In some
cases Node B may be able to correct errors.
In the case that secure transmission is required, the data may be encrypted
by Node A prior to transmission. Node B must then perform the reverse
process.

To achieve this high degree of cooperation between computers, the tasks

are broken into subtasks that are implemented individually using a layered
approach. These layers form the data communication protocol architecture.
Example of such layer architectures are: the Open System Interconnection (OSI)
model, and the Transmission Control Protocol/ Internet Protocol
(TCP/IP). Key advantages of a layered
structure include:
The complex communication protocol is divided into subtasks and these
are implemented within a layered structure. Each layer has limited
functionality and this ‘divide and conquer’ approach facilitates the design and
the implementation of the system.
Higher-level layers need have no knowledge of tasks performed by the lower
layers. Thus,
for example, a higher-level layer needs no knowledge of the type of
interconnect that is in use. Again, this facilitates the design process.
When changes are made to the communications protocol, only certain relevant
layers need
to be modified/replaced. This makes it easier to upgrade software and
undertake software testing.
Structuring software using a layered approach tends to result in larger
programs which run more slowly than if a non-layered approach were to
be adopted. However, these two weaknesses are outweighed by the benefits
that are associated with the layered approach - especially in terms of providing
a structured framework within which the complex issues associated with
computer communications may be resolved.

3.4 Standards
Are essential in creating and maintaining an open and competitive market for
equipment manufacturers and in guaranteeing national and international
interoperability of data and telecommunications technology and processes.
Standards provide guidelines to manufacturers, vendors, government
agencies and other service providers to ensure the kind of
interconnectivity necessary in today’s market place and in international
communication.

Standards play an important role in our everyday lives and facilitate the
operation of products produced by different manufacturers. For example:
Countries adopt a standard type of mains plug and socket. Without such
a standard, we would find that we had to continually rewire mains plugs or
employ some form of adaptor. This provides an example of national
standard.
Car manufacturers adopt a standard for the relative placement of the
clutch, brake and accelerator pedals. This provides an example of global
standard.
Computers are equipped with standard interface sockets (e.g. serial,
parallel and USB) via
which they are able to connect to peripheral devices. This provides an
example of global standard.
Standards may come into being in various ways.
For example:
A standard may be established (imposed) by the company that plays the
most dominant role in any particular area. For example, the serial and
parallel ports employed by today’s PC were implemented on the earliest
PCs introduced by IBM. They soon became standard for desktop computing
A standard may gradually evolve
A standard may be developed/defined by a committee of experts.
Although standardization can facilitate our use of technologies and products,
standards seldom reflect an optimal solution. For example, the VHS videotape
format became a standard, while 28 other superior and equally cost-effective
formats fell by the wayside. Furthermore, in the case
of standards developed by committees, these often reflect many technological
compromises and take long periods to develop. Such standards are often out of
date even before they are released.

From a computer user’s perspective, standards are extremely important because

they allow a combination of products from different manufacturers to be used
together. Standards ensure greater compatibility and interoperability
between various types of equipment and technologies.

In data communications, standards provide guidelines to manufacturers and

service providers to ensure compatibility, connectivity, and interoperability of
technologies – an essential requirement in today’s global market. Key
advantages of standards are:

To ensure a large market for hardware or software products – thus

encouraging mass production
To allow products from different vendors to communicate, thus giving
customers more flexibility in the selection and use of equipment.

On the other hand, standards do have

limitations:

They tend to slow down technological change. This is due to the fact that, in
some cases, by the time a standard is developed, subjected to scrutiny,
reviewed, compromised and endorsed by all concerned parties – and then
disseminated, more efficient technologies could have developed.
Many standards may exist for the same thing. It is often difficult to decide
which standard will provide better compatibility and remain in place for the
greatest amount of time.

Many official computer-related standards are defined by the following

organizations:

ANSI (America National Standards Institute)

ITU (International Telecommunication Union)
IEEE (Institute of Electrical and Electronic Engineers)
ISO (International Organization for Standardization)
VESA (Video Electronics Standards Association).
Car drivers generally use agreed signals when turning left or right. Aero
plane pilots follow specific standardized rules for communicating
throughout the world. Similarly, for any
computer-based systems to communicate successfully, they need to use ‘the same
language’.
This means that what is communicated, how it is communicated, and when it is
communicated must conform to some mutually acceptable conventions agreed
between the parties involved. These conventions are known as a ‘protocol’,
which can be defined as a set of rules governing the exchange of data between
two or more devices.
Typical tasks performed by protocols are as
follows:
 To make sure that the source device activates the data
communication line
To inform the transmission system about the
destination system.
To make sure that the source device communicates with the destination
device before sending data
To make sure the destination device is ready to
accept the data
To make sure that the destination file management system is ready to store
incoming files
To ensure compatibility between source and destination, and to perform
format translation.

In the 1980s, many companies entered the desktop computing market and this
led to a rich diversity of products. Unfortunately, these systems would often
not operate together, nor could software developed for use on one particular
type of machine necessarily be used on another. In short, although the lack
of standards enabled product diversity, it hampered computer usage.
Quickly, standards were developed (and/or evolved) and these impacted on
many areas of computing. For example:
Compatibility improved. By conformance to standards, hardware and
software systems developed by different manufacturers could be used
together (although there were often unforeseen problems)
The diversity of available products
decreased
Backwards compatibility became an important issue. For example, a
new model of computer, or a new release of an operating system should
support the function of older products. This has greatly increased hardware
and software complexity and retarded the development of radically new
computer products.

3.5 The OSI model

The Open System Interconnection (OSI) reference model was developed by the
International Standards Organization (ISO) and provides a framework for
protocol development. By implementing a communication protocol that adheres
to the OSI model, systems developed by different manufacturers are able to
communicate. The tasks that must be performed to enable machines to
communicate in an effective and efficient manner are incorporated within a
seven-layer hierarchy, as indicated in figure 3. Although the protocols detailed
within this reference 0 model are seldom used, the model provides us with an
excellent conceptual framework for understanding the tasks performed by the
various software layers. Below we briefly summarize
aspects of the functionality of the various layers.

3.5.1 Application
layer
This should not be confused with the applications programs that may be running on
a computer. The application layer provides network access to the user and to applications
programs. This layer passes data to (and receives data from) the presentation layer,
and logically communicates directly to the application layer on the remote computer.
This is indicated in figure 3 where the horizontal lines indicate the logical communication of
each layer with its remote counterpart. The application layer needs know nothing of the
tasks carried out by the lower layers – it needs only interface with the user (and applications
programs) and with the presentation layer.

3.5.2 Presentation
layers

Different computers may employ different character set formats. A user is not interested in
31 such differences and one of the tasks undertaken by the presentation layer is to
translate between different formats that may be used to represent numbers, characters
and other
symbols. Additionally, the presentation layer is also involved in ensuring secure
data transmission (consequently, when data is being transmitted the presentation layer
undertakes encryption, and when data is being received it performs decryption).
3.5.3 Session
layer

A user applications program may need to open a ‘session’ with a remote machine. For
example, a user may wish to log on to a remote computer and carry out various tasks and
this will involve the transmission and reception of data over a period of time. This
necessitates synchronization whereby each node knows when it can transmit and when it is
to receive data (i.e. when it must
‘listen’). The session layer deals with this synchronization and additionally is involved in error
recovery. Consider the case that a file is being transmitted between two nodes, and during
this process the network fails. Without the support of the session layer it would be
necessary to start the transmission process again from the beginning. However, the session
layer inserts checkpoints into the transmitted data stream and these are used to
efficiently recover from such failures. Following a failure, transmission can be
recommenced from the point at which the last checkpoint was successfully delivered to the
destination node. The session layer carries out various other activities, such as bracketing a
set of related and non-independent activities. For example, there may be a need to carry
out a task on a remote machine, which involves the execution of series of commands.
Perhaps if only some of these commands are executed (i.e. they are not carried out in their
entirety) problems will ensue. If the individual commands are executed as each arrives at the
remote machine then, in the case that the network connection fails, there is the likelihood of
incomplete execution. One task performed by the session layer relates to the buffering of
such commands – as each arrives it is temporarily stored and not passed to higher layers
until all commands (and any associated data) have been received. The series of commands
may then execute in full.

3.5.4 Transport
layer

This acts as the intermediary between the lower layers (whose implementation is dependent
on the underlying network architecture) and the three upper layers which provide user
services and whose architecture is (at least in principle) independent of the
detailed network
characteristics
.

The type of transport service that is provided to the session layer is determined by
the transport layer. Suppose a node wished to send an extremely large file to a remote
machine via a shared network (or set of interconnected networks). Without judicious
design (in relation to
the type of transport service used), there is the possibility that such a transmission could block
32
the network(s) in such a way that whilst the transmission is in progress no other
machines
could communicate. The approach commonly used to prevent such a situation is to split
the data into chunks (‘packets’) which are individually encapsulated within a frame
containing all the necessary data needed to enable a packet delivery to the intended
destination. The splitting of the data into smaller units is carried out by the transport layer.
These packets may traverse a
set of networks by different routes and so arrive at their destination out of order. The
transport layer reorders packets and so enables them to be correctly reassembled.

3.5.5 Network
Layer

This layer decides on routing issues, determining the path that should be followed by
packets when they traverse networks. In fact, in such a situation the path taken is not
defined solely by the source node but by all the nodes (network devices) through which
packets pass on their way to the destination. Consider the situation illustrated in figure 4

Node A Node C

Node E

Node D Node B

Figure 4: A simple network in which a packet may be sent from Node A to B via
different routes. The circles represent nodes, and the lines network interconnects.

Suppose that a packet is to be sent from Node A to Node B. The packet will have to
pass through at least one intermediate node (network device). These nodes may simply
forward the packet, or may decide on the direction of the next step in its voyage. Thus, for
example, Node D simply performs a forwarding function, whereas Nodes C and E are able to
make routing decisions. The transport layer plays a critical role in determining the time it
will take for packets to reach their destination and in this sense the actions of the transport
layer impact on transmission latency.

3.5.6 The data link

layer

This layer is responsible for various low-level network specific tasks and plays a crucial part in
33
the detection and correction of errors that may occur during the transmission
process.

Correction may be achieved by means of additional information inserted into messages prior
to their transmission that can be used to modify bits corrupted during the transmission
process.

Alternatively, correction may involve requesting re-transmission. Additionally, the data

link layer plays a pivotal role in managing network access and ensuring that network
‘collisions’ (which occur when two or more nodes attempt to transmit onto the same
LAN at the same
time) are handled correctly. Devices connected together via networks do not necessarily
demonstrate the same transmission/reception characteristics.

Thus a device able to transmit at high speed (i.e. that has high bit-rate) could readily swamp
a slower recipient. Buffering techniques are used to circumvent this problem and
this necessitates a protocol that ensures that the capacity of the buffer is not exceeded.
This is referred to as flow control.

3.5.7 The physical

layer

This layer deals with the transmission of the bit stream through the transmission medium,
and the connection strategy used to enable the transfer of bits from one node to another.
Thus the physical layer defines the signal levels, the type of transmission medium employed
(e.g. twisted pair cable, coaxial cable, fiber optic cable), and also the techniques that will
be used to permit the passage of data, such as circuit switching (in which a dedicated path is
set up between two communicating nodes), packet switching, etc.

3.6 The TCP/IP

protocol

In the late 1960s, the US Department of Defence’s Advance Research Project Agency (ARPA)
initiated a project that centered upon the interconnection of geographically
dispersed computing systems. Gradually a large-scale network of university and government
computing facilities evolved (this network was named ARPANET), which used packet
switching techniques and initially employed leased phone lines. Early networking protocols
were slow and unreliable and in 1974 a new set of protocols were proposed. These formed
the basis for TCP/IP (Transmission Control Protocol/Internet Protocol (TCP/IP) which today
underpins the operation of the Internet.

A protocol such as TCP/IP must support a number of essential requirements such

as: ey
Reliability: in terms of both data integrity and timely delivery
Fault tolerance: the failure of a network segment should not seriously disrupt
overall network operation; it must be possible to route packets along different paths so
that th 34 can still reach their destination
Transparent communications: different computer systems and LANs should be able
to communicate transparently.

It is convenient to employ a layered model in order to most readily conceptualize TCP/IP.

We can therefore consider TCP/IP within a four-layer framework (a five–layer model is
sometimes
preferred). In figure 5 these layers are depicted, and are placed alongside the layers
that comprise the OSI model. Below we briefly summarize aspects of their role

3.6.1 Application
layer

This layer provides communication services to the user and to applications programs. It can
be viewed as corresponding to the application, presentation and session layers found in the
OSI model. The application layer contains all the high-level protocols (such as those
that we commonly encounter when accessing the Internet – such as DNS (Domain Name
System) and HTTP).

3.6.2 Transport
layer

Two different protocols are defined in this layer (TCP and UDP (User Datagram Protocol)).
These differ in a number of important respects. For example:

Reliability: in the case of UDP, error correction is not implemented – the onus for
this activity is placed on the applications program. This contrast with TCP in which
error detection and correction form an integral part. Free from error correction
overheads, UDP can (under some circumstances) demonstrate high performance
Flow control: in the case of TCP, flow control is implemented and this prevents a
faster machine from swamping a recipient that operates more slowly.
Figure 5: A conceptual model of TCP/IP set alongside the layers that
comprise the OSI model
A stream of data that is to be transmitted is fragmented into chunks and the
transport layer appends various information, before passing these to the
internet layer. At the receiving node, the transport layer reassembles these
data chunks. In the case of TCP, the transport layer encapsulates the data
chunks into a TCP segment (in the case of UDP, the encapsulated data is
usually referred to as a packet. There are differences between the information
contained in the UDP and TCP headers.) Here the data is provided with a
‘header’ containing various important information; see Figure 6. It is
instructive to consider the purpose of several pieces of information
contained in the header:
Source and Destination ports: many well known (widely used) application
protocols are designated by unique identification numbers provided by the
‘Internet Assigned Numbers Authority’. For example, the File Transfer
Protocol (FTP) is identified as “port21’, and the Simple Mail Transfer
Protocol (SMTP) as ‘port 25’. TCP inserts this information into the header
and thereby provides information on the source and destination applications
protocol associated with the data to be transferred. The source port and
destination port fields are each two bytes
long, and values below 256 are used to reference ‘well-
known’ ports.
 0 16 31 bits

Source Destination port

Sequence number

TCP header
Acknowledge number
Offset Reserved U A P R S F Window
Checksum Urgent pointer
Options Padding

Data
Data
…

Figure 6: Information contained within a TCP segment

Sequence number: TCP numbers each byte of data that is transmitted
between two nodes during the transfer process. The sequence number
references the first byte of data encapsulated within frame. This is most
readily understood by means of an example. Suppose that a set of frames
are transmitted between node A and node B, and that each contains 256 bytes
of data. Then the sequence numbers contained in the first four frames
transmitted by Node A could be 1,257,513, 769 (the process is slightly more
complex since the sequence number of the first frame need not be 1). Node
B these sequence numbers to
37
 reconstruct the data chunks and correct for frames being received out of
their transmitted order
Header length: this enables the receiving node to determine the point at
which the header ends and the data starts. It is necessary to specify this
length as not of fixed size.
Checksum: this enables the transport layer to perform
error detection
Options: various options can be included. For example, one option enables
the recipient to inform the source node about the maximum segment size that
it is able to accept. This is indicated during the establishment of a
communication and ensures that the recipient’s buffer will not be swamped
by a high-speed transfer.

Internet
layer
At the sending node, the Internet layer takes packets or segments generated by
the transport layer, and further encapsulate these to produce datagrams. The
additional information appended by the Internet layer (the ‘IP header’) is
intended to enable the datagrams to be injected onto any network and travel (via
intermediate networks) to the intended destination. During their transit,
intermediate network devices will use this information to determine the direction
they should take. Since the routing of packets is fundamental to the Internet
layer, it may be considered to be equivalent to the network layer used in the OSI
model.

Network interface layer

In terms of its functionality, this layer is equivalent to the lowest two layers
used in the OSI model. It further encapsulates a datagram received from the
Internet layer producing a ‘frame’. This layer makes the connection to the
transmission medium and employs the appropriate protocol for launching and
receiving frames.
The process of encapsulation referred to above is summarized in figure 7 and in
Table 1 an overview of the functionality of the layers that have been
conceptualized in connection with TCP/IP is presented.
 Data to be transmitted

TCP or UDP header Data to be transmitted

IP header TCP or UDP header Data to be transmitted

Figure 7F:rame
Theheader
process IP encapsul ation TCP
ofheader used headIePr .This isData
by TCP/
or UDP deptiocted for the sou rce
be transmitted
(sending) node. At the receiving node, the process operates in the reverse: bottom up.

38
Application (4)

Similar to OSI application layer

Serves as communication interface by providing specific application services
Examples include email, virtual terminal, file transfer, WWW

Transport (3)

Defined by two protocols:

User Datagram protocol (UDP)

- a connectionless protocol
- provides unreliable datagram service (no end -to-end error detection or correction)
- does not retransmit any unreceived data
- requires little overhead
- application protocols include Trivial File Transfer Protocol (TFTP), Network File System (NFS),
Simple Network Management Protocol (SNMP), Bootstrap Protocol (BOOTP), and Domain
Name Service (DNS)

Transmission Control Protocol (TCP)

- (the TCP of TCP/IP)

- connection–oriented protocol
- provides reliable data transmission via end-to-end detection and correction
- guarantees data is transferred across a network accurately and in correct order
- retransmits any data not received by destination node
- guarantees against data duplication between sending and receiving nodes
- application protocols include Telnet, FTP, SMTP and POP

Internet (2)

(The IP of TCP/IP)
Transfers user messages from source host to destination host
Connectionless datagram service
Route selection is based on a metric
Uses Internet or IUP addresses to locate a host within the Internet
Relies on routers or switches
Integral part is Internet Control Message Protocol (ICMP); this uses an IP datagram to carry
messages about state of communications environment

Network Interface (1)

Connects host to the local network hardware

Makes a connection to the physical medium
Uses a specific protocol for accessing the medium
Places data into frames
Effectively performs all functions of the first two layers of the OSI model

Table 1: A summary of some aspects of the functionally of the conceptualized four-layer

TCP/IP model
39
4.0 CONCLUSION
You have been introduced to the basic concepts of packets, protocols and
standards. You are now in a position to relate these concepts to how
communication occurs between entities in different systems in your
environment.

5.0 SUMMARY

Packets and protocols are the fundamental building blocks of data

transmission over the network. A packet is a segment of data that has a
header with destination and addressing information attached to it.

A protocol is a set of rules that govern data communication; the key elements of
a protocol are syntax, semantics and timing.

Standards are necessary to ensure that products from different

manufacturers can work together as expected. The ISO, ITU-T, ANSI, IEEE
and EIA are some of the organizations involved in standards creation.
To enable two or more computers to communicate in a meaningful manner, a
communication protocol must be defined.

We briefly summarize aspects of the functionality of the various layers of the

OSI model and layers of the TCP/IP protocol.

TCP/IP today underpins the operation of the

Internet.

6.0 TUTOR-MARKED ASSIGNMENTS

1 List the major disadvantages with the layered approach to protocols.

2a. Why are protocols needed?

b Why are standards needed?

3. How does the protocol travel through the OSI model?
4. What does OSI stand for and what do we use it for?
7.0 REFERENCES/FURTHER READING

1. Burgess, M. (2004). Principles of Network and System Administration. (2nd Ed.).

Chichester, West Sussex , England: Wiley.

2. Forouzan, B.A, & Fegan, S.C. (2007). Data communications and Networking
(4th Ed). Mc
Graw Hill.

3. Limoncelli, T. A.,Hogan, C. J. & Chalup, S. R (2007}. The Practice of System

and
Network Administration. (2nd Ed.). Upper Saddle River, NJ: Addison-Wesley

4. Stallings, W. (2009). Data and computer communications ( 8th ed.). Upper

saddle River, NJ.: Pearson Education Inc.
Unit 3 Network, Transport and Application Layers

1.0 INTRODUCTION

This unit covers three important layers in networking, namely: the

network, transport and the application layers. Important algorithms and
mechanisms related to each layer are introduced. We will also discuss
certain types of network devices.

2.0 OBJECTIVES

By the end of this unit, you will be able to:

 to explain the difference between the network, transport and

application layers
 to make use of the routing and congestion algorithms and explain
transport control mechanisms.

3.0 MAIN CONTENT

3.1 Network Layer

The Network Layer provides services to the Transport Layer. It can be

based on either virtual circuits or datagrams. In both cases, its main job
is routing packets from the source to the destination. In virtual circuit
subnets, a routing decision is made when the virtual circuit is set up. In
datagram subnets, it is made on every packet.

The network layer services have been designed with the following goals:

 The services should be independent of the subnet technology.

 The transport layer should be shielded from the number, type,

and topology of the subnets present.

 The network addresses made available to the transport layer

should use a uniform numbering plan, even across LANs and
WANs.

The discussion centres on the question of whether the network layer

should provide connection – oriented service or connectionless service.

3.1.1 Routing Algorithms

Many routing algorithms are used in computer networks. Static

algorithms include shortest path routing, flooding, and flow–based
routing. Dynamic algorithms include distance vector routing and link
state routing. Most actual networks use one of these. Other important routing topics
are hierarchical routing, routing for mobile hosts, broadcast routing, and
multicast routing.

The function of the network layer is routing packets from the source machine to
the destination machine. In most subnets, packets will require multiple hops to
make the journey. The only notable exception is for broadcast networks, but even
here, routing is an issue if the source and destination are not on the same network.
The algorithms that choose the routes and the data structures that they use are a
major area of network layer design.

The routing algorithm is that part of the network layer software responsible for
deciding which output line an incoming packet should be transmitted on.

Regardless of whether routes are chosen independently for each packet or only
when new connection is established, there are certain properties that are desirable in
a routing algorithm: correctness, simplicity, robustness, stability, fairness and
optimality.

Stability is also an important goal for the routing algorithm. Routing algorithms can
be grouped into two major classes: non–adaptive and adaptive. Non–adaptive
algorithms do not base their routing decisions on measurements or estimates of
the current traffic and topology. Instead, the choice of the route to use to get from
I to J is computed in advance, off–line, and downloaded to the routers when the
network is booted. This procedure is something called static routing.

Adaptive algorithms, in contrast, change their routing decisions to reflect

changes in the topology, and usually the traffic as well. Adaptive algorithms differ
in where they get their information (e.g., every sec, when the load changes, or when
the topology changes), and what metric is used for optimization (e.g., distance,
number of hops, or estimated transit time). In the following sections, we will
introduce a variety of routing algorithms, both static and dynamic.

3.1.1.1 Shortest Path Routing

Let us begin our study of routing algorithms with a technique that is widely used in
many forms because it is simple and easy to understand. The idea is to build a
graph of the subnet, with each node of the graph representing a router and each arc
of the graph representing a communication line. To choose a route between a given
pair of routers, the algorithm just finds the shortest path between them on the
graph. One way of measuring path length is the number of hops.
3.1.1.2 Flooding

Another static algorithm is flooding, in which every incoming packet is

sent out on every outgoing line except the one it arrived on. Flooding
obviously generates vast numbers of duplicate packets, in fact an infinite
number unless some measures are taken, but it is one of the simplest
routing algorithms.

Routers need to communicate with other routers so they can exchange

routing information. Most network operating systems have associated
routing protocols which support the transfer of routing information.
Typical routing protocols and their associated network operating
systems are:

 BGP (Border Gateway Protocol)–TCP/IP.

 EGP (Exterior Gateway Protocol)–TCP/IP.
 IS–IS (Immediate System to Intermediate System) – DECent,
OSI.
 NLSP (NetWare Link State Protocol) – Net Ware 4.1
 OSPF (Open Shortest Path First) – TCP/IP.
 RIP (Routing Information Protocols) – XNS, Net Ware, TCP/IP.
 RTMP (Routing Table Maintenance Protocol) – Apple Talk.

3.1.2 Congestion Control Algorithms

The situation in which when too many packets are present in the subnet,
performance degrades. This situation is called congestion.

Subnet can become congested, increasing the delay and lowering the
throughput for packets. Network designers attempt to avoid congestion
by proper design. Techniques include traffic shaping, flow
specifications, and bandwidth reservation. If congestion does occur, it
must be dealt with. Choke packets can be sent back, load can be shed,
and other methods applied.

Congestion can be brought about by several factors.

 If all of a sudden, streams of packets begin arriving on three or

four input lines and all need the same output line, a queue will
build up. If there is insufficient memory to hold all of them,
packets will be lost. If the router has an infinite amount of
memory, congestion gets worse, not better, because by the time
packets get to the front of the queue, they have already timed out
and duplicates have been sent. All these packets will be dutifully
forwarded to the next router, increasing the load all the way to the
destination.
 Slow processors can also cause congestion. If the routers; CPUs are slow at
performing the book–keeping tastes required of them, queues can build up,
even though there is excess line capacity. Similarly, low–bandwidth lines can
also cause congestion.

 Congestion control has to do with making sure the subnet is able to carry the
offered traffic. It is a global issue, involving the behavior of all the hosts, all
the routers, the store and forwarding process within the routers, and all the
other factors that tend to diminish the carrying capacity of the subnet.

3.1.3 Comparison of Virtual Circuit and Datagram Subnets.

Inside the subnet, several trade–offs exist between virtual circuits and datagrams.

 One trade–off is between router memory space and bandwidth.

Virtual circuits allow packets to contain circuit numbers instead of full
destination addresses. If the packets tend to be fairly short,
a full destination address in every packet may represent a
significant amount of overhead, and hence, wasted bandwidth. The price
paid for using virtual circuits internally is the table space within the
routers. Depending upon the relative cost of communication circuits versus
router memory, one or the other may be cheaper.

 Another trade–off is setup time versus address passing time.

Using virtual circuit requires a setup phase, which takes time and consumes
resources. However, figuring out what to do with a data packet in a
virtual circuit subnet is easy: the router just uses the circuit number to index
into a table to find out where the packet goes. In datagram subnet, a more
complicated procedure is required to determine where the packet goes.

 Virtual circuits have some advantages in avoiding congestion within the

subnet because resources can be reserved in advance, when the connection
is established. Once the packets start arriving, the necessary bandwidth
and router capacity will be there. With a datagram subnet, congestion
avoidance is more difficult.

 For transaction processing systems, the overhead is required to set up and

clear a virtual circuit. If the majority of the traffic is expected to be of this
kind, the use of switched virtual circuits inside the subnet makes little
sense. On the other hand,
 permanent virtual circuits, which are set up manually and last for
months or years, may be useful here.

 Virtual circuits also have vulnerability problem. If a router

crashes and loses its memory, even if it comes back up a second
later, all the virtual circuits passing through it will have to be
aborted. In contrast, if a datagram router goes down, only those
users whose packets were queued up in the router at the time, will
suffer, and may be not even all those, depending upon whether
they have already been acknowledged or not. The loss of
communication line is fatal to virtual circuits using it but can be
easily compensated for if datagrams are used.

 Datagrams also allow the routers to balance the traffic throughout

the subnet, since routes can be changed halfway through a
connection.

 It is worth explicitly pointing out that the service offered is a

separate issue from the subnet structure. In theory, all four
combinations are possible. Obviously, a virtual circuit
implementation of a connection–oriented service and datagram
implementation of a connectionless service are reasonable.
Implementing connections using datagrams also makes sense
when the subnet is trying to provide a highly robust service.

3.1.4 Internetworking

In internetworking modes are connected together through different

network equipment. Networks connect to other networks through
repeaters, bridges or routers. A repeater corresponds to the physical
layer and always routes signals from one network segment to another.
Bridges route using Data Link Layer and routers route using the network
layer.

Networks differ in various ways, so when multiple networks are

connected together, problems can occur. Sometimes the problems can be
finessed by tunneling a packet through a hostile network, but if the
source and destination networks are different, this approach fails. When
different networks have different maximum packet sizes, fragmentation
may be called for.

3.1.4.1 Repeaters

All types of network connections suffer attenuation and pulse distortion.

For a given cable specification and bit rate, each has a maximum length
of cable. Repeaters can be used to increase the maximum
interconnection length and will do the following:

 Clean signal pulses.

 Passes all signals between attached segments
 Boost signal power
 Possibly translate between two different media types (e.g., fibre –
optic to twisted–pair cable).

3.1.4.2 Bridges

Bridges filter input and output traffic so that only packets intended for a network are
actually routed into the network and only packets intended for the outside are
allowed out of the network.

3.1.4.3 Routers

Routers examine the network address field and determine the best route for the
packet. They have the great advantage that they normally support several different
types of network layer protocols.

Routers, which only read one type of protocol, will normally have high filtering and
forwarding rates. If they support multiple protocols, then there is normally an
overhead in that the router must detect the protocol and look into the correct place
for the destination address.

3.2 Transport Layer

The Transport Layer provides reliable cost effective data transport from the source
machine to destination machine.

3.2.1 Transport Service And Mechanism

The Transport Layer provides various services, the most important being an end–to–
end, reliable, connection–oriented byte stream from sender to receiver. It is accessed
through service primitives that permit the establishment, use and release of
connections.

Transport protocols must be able to do connection management over unreliable

networks. Connection establishment is complicated by the existence of delayed
duplicate packets that can reappear at opportune moments. To deal with them,
three–way handshakes are needed to establish connections. Releasing a
connection is easier than establishing one but is still far from trivial due to the two–
army problem.
3.2.2 Types of Service/Quality of Service (QoS)

The need to define quality of service arises from the realisation that
users require different quality presentations at different times. The
different quality presentations map onto different parameter values.
When a multimedia presentation is transmitted via a network, it
translates into different requirements of network performance. To be
able to specify QoS aspects concisely and to request them of a network,
QoS must be specified as a set of parameters that can be assigned
numerical values. In a multimedia presentation, the ultimate user of the
system is a human being. Thus, the quality of the presentation is a matter
of the user’s perception, which is limited by the response of the human
vision and auditory senses. This perceptual nature of QoS makes it
subjective and difficult to quantify precisely. Thus, it is easier to specify
a range of values rather than a single value.

3.2.3 Transport Control Mechanism

The transport control service is implemented by a transport protocol

used between the two transport entities. It is similar to the Data Link
Protocol, but with some differences:

 Environments in which they operate (at the data link layer, two
routers communicate directly via a physical channel, whereas at the
transport layer, this physical channel is replaced by the entire subnet.

 In Data Link Protocol, it is not necessary for a router to specify

which router to talk to–each outgoing line uniquely specifies a
particular route.

3.2.3.1 Addressing

When an application process wishes to set up a connection to a remote

application process, it must specify which one to connect to. In Internet,
these end–points are (IP address + Local Port) pairs. The end–point in
this context:

 TSAP (Transport Service Access Point)

 NSAP (Network Service Access Point)

A transport entity supports multiple TSAPs.

3.2.3.2 Flow Control And Buffering

Flow control of transport layer is similar to that of Data Link Layer, but in
Transport layers, the number of connections open is numerous as compared to Data
Link Layer.

If the subnet provides datagram service, the sending transport entity must also
be buffered, for re–transmitting in the case of loss. If the receiver knows that the
sender buffer all TPDUs (Transport Protocol Data Units) until they are
acknowledged, the receiver may or may not dedicate specific buffers to specific
connections.

In summary, if the network service is unreliable, the sender must buffer all TPDUs.
However, with reliable network services, other trade–off becomes possible.

The optimum trade–off between source buffering and destination buffering depends
on the type of traffic carried by the connection. For low–bandwidth bursty traffic,
such as that produced by an interactive terminal, it is better not to dedicate any
buffer, but rather to acquire them dynamically at both ends.

3.2.3.3 Multiplexing

Multiplexing is multiple things on to one i.e., multiplexing several

conversations onto connections Virtual circuits and physical links play a role in
several layers of the network architecture.

Need for Multiplexing

 A number of virtual circuits are open by the users or one user opening
more than one, which requires a lot of buffer in the router; this gives a
solid reason for packet switched network.

 To bill the users based on the amount of data sent, not the connection
time.

Upward Multiplexing: Multiplexing of different transport connections onto the

same network connection is attractive.

Downward Multiplexing: The transport layer opens multiple network connections

and distributes the traffic among them on a round–robin basis.
3.2.3.4 Connection Establishment and Management

Connection Establishment is not as easy as it sounds, but it is in fact, a

complicated task, we have to take care of the losses that occur during
transmission. At first glance, it would seem sufficient for one transport
entity to just send a CONNECTION REQUEST TPDU to the
destination and wait for a CONNECTON ACCEPTANCE reply. The
problem occurs when the network is not reliable.

The solution could be to give each connection a connection identifier,

chosen by the initiating party, and put it in each TPDU, including the
one requesting the connection. After each connection is released, each
transport entity could update a table listing absolute connection as (peer
transport entity, connection identifier) pair.

Unfortunately, this scheme has a basic flaw: it requires each transport

entity to maintain a certain amount of history information indefinitely. If
a machine crashes and loses its memory, it will no longer know which
connection identifiers have already been used.

3.2.3.5 Crash Recovery

If host and routers are subject to crashes, recoveries from these crashes
become an issue. If the transport entity is entirely within the hosts,
recovery from network and router crashes is straightforward. If the
network later provides datagram services, the transport entity expects
lost TPDUs all the time and knows how to cope with them. If the
network layer provides connection–oriented service, then loss of virtual
circuits is handled by establishing a new one and then probing the
remote transport entity to ask it which TPDUs it has received and which
one it has not received.

3.2.4 TCP/UDP

In this section we will discuss two important transport layer protocols:

TCP and UDP.

3.2.4.1 Transmission Control Protocol (TCP)

TCP provides a highly reliable, connection–oriented, end–to–end

transport service between processes in end systems connected to the
subnet. TCP only assumes that the layer below offers an unreliable
datagram service. TCP provides the types of facility associated with the
ISO Class 4 transport service, including error recovery, sequencing of
packets, flow control by the windowing method, and the support of
multiplexed connections from the layer above.
3.2.4.2 Format of TCP Header

The sender’s TCP layer communicates with the receiver’s TCP layer using the
TCP protocol data unit. It defines parameters such as the source port,
destination port, sequence number and so on. Its descriptions are given
below:

 Source and destination port number – which are 16 bit values to identify the
local port number.

 Sequence number – which identifies the current sequence number of the data
segment. This allows the receiver to keep track of the data segments
received. Any segment that is missing can be easily identified.

 Data offset – which is a 32–bit value and identifies the start of the data.

 Flags – the flag field is defined as UAPRSE, where U is the urgent flag,
a the acknowledgement flag, P the push function, R the reset flag, S the
sequence synchronize flag and E the end of transmission flag.

 Windows – is a 16 bit values and gives the number of data blocks that the
receiving host can accept at a time.

 Checksum – is a 16 bit checksum for the data and header.

 UrgPtr – is the urgent pointer and is used to identify an important area of

data.

3.2.4.3 User Data Protocol (UDP)

The internet protocol suite also supports a connectionless transport protocol, UDP
(User Data Protocol). UDP provides a way for applications to send encapsulated
raw IP datagrams and send them without having to establish a connection.
Many client–server applications that have one request and one response use UDP
rather than go through the trouble of establishing and later releasing a connection.

A UDP segment consists of an 8–byte header followed by the data. The two ports
serve the same function as they do in TCP: to identify the end–points within the
source and destination machines. The UDP length field includes the 8–byte
header and the data. The UDP checksum includes the same format pseudo–
header, the UDP header, and the UDP
data, padded out to an even number of bytes if need be. It is optional and
stored as 0 if not computed.

3.3 Application Layer

The Application Layer contains a variety of protocols that are

commonly needed. For example, there are hundreds of incompatible
terminal types in the world. To solve this problem there is need to define
an abstract network virtual terminal that editors and other programs can
be written to deal with.

Another application layer function is file transfer. Different file systems

have different file naming conventions.

3.3.1 The Domain Name System (DNS)

DNS is a scheme for assigning meaningful high–level name to a large

set of machines, and discusses a mechanism that maps between high–
level machine names and IP addresses. It considers both the translation
from–high level name to IP addresses and the translation from IP
addresses to high–level machines names. It has been used to assign
machine names throughout the global Internet. It uses a geographically
distributed set of servers to map names to addresses, the implementation
of the name mapping mechanism provides a large scale example of the
client sever paradigm.

In a TCP/IP internet, hierarchical machine names are assigned according

to the structure of organisations that obtain authority for parts of the
namespace, not necessarily according to the structure of the physical
network interconnections.

3.3.2 TCP/IP Internet Domain Name

The mechanism that implements a machine name hierarchy for TCP/IP

Internets is called the Domain Name Systems (DNS). DNS has two
conceptually independent aspects. The first is abstract; it specifies the
name syntax and rules for delegating authority over names. The second
is concrete; it specifies the implementation of a distributed computing
system that efficiently maps names to addresses. This section considers
the name syntax, and later sections examine the implementation.

The domain name system uses a hierarchical naming scheme as domain

names. As in our earlier examples, a domain name consists of a
sequence of sub names separated by a delimiter character, the period. In
our examples we said that individual sections of the name might
represent sites or groups, but the domain system simply calls each
section, a label. Thus, the domain name cs.ignou.org contains three labels: CS,
IGNOU, and ORG. Any suffix of a label in a domain name is called a domain. In
the above example, the lowest level domain is CS. ignou.org (the domain name for
ignou) and the top level domain is org. As the example shows, domain names are
written with the local label first and the top domain last. As we will see, writing
them in this order makes it possible to compress messages that contain multiple
domain names.

Domain Name Meaning

Com Commercial Organisations

Edu Educational Institutes Gov
Government Institutions Mil Military
Groups
Net Major network support centres
Org Organisations other than those above
Arpa Temporary ARPANET domain (obsolete) INT
International Organisations
Country code Each country (geographic scheme)

3.3.3 Electronic Mail

This is the most widely used service facilitating users to send and receive
messages electronically in a store and forward manner. Different E–mail standards,
viz., SMTP, UUCP and X400 Message Handling system, are supported on
ERNET.

Electronic Mail is a system whereby a computer user can exchange messages with
other computer users or group of users via a communications network.

The backbone of an electronic mail system is a communication network that

connects remote terminals to a central system or a local area network that
interconnects personal computers. Users can send mails to a single recipient or
they can broadcast it to any number of selected users on the systems. When
multi–tasking personal computer and workstation are used, mail can be delivered to
users while they are working on something else. Otherwise, users have to
interrogate their mail boxes in a central system, or file server.

Many users first encounter computer networks when they send or receive
electronic mail to or from a remote site. E–mail is the most widely used application
service. Indeed, many computer users access networks only through electronic mail.
E–mail is popular because it offers a fast, convenient method of
transferring information. E–mail can accommodate small notes or large
voluminous memos with a single mechanism. It should not surprise you
to learn that more users send files with electronic mail than with file
transfer protocols.

Characteristics

 Store and forward

 Delivery time ranging from few seconds to hours
 Largely textual
 Binary files may be appended or “uuencoded”
 Multimedia (“mime” standard)
 Distribution lists with “cc:”, “bcc:”, “fcc:”
 Mail forwarding
 Auto–processing
 Statistics collection
 Secure email
 Several mailers: smtp,uucp (smtp requires IP connectivity; uucp
works with dial–up).

3.3.4 WWW (World Wide Web)

The World Wide Web is a system for linking up hypertext documents.

Each document is a page written in HTML, possible with hyperlinks to
other documents. A browser can display a document by establishing a
TCP connection to its server, asking for the document, and then closing
the connection. When a hyperlink is selected by the user, that document
can also be fetched in the same way. In this manner, documents all over
the world are linked together in a giant web. Some facts about WWW:

 Fastest growing discovery and retrieval system

 Presently 10,000 servers, growing at an astounding rate
 Retrieve “hypermedia” documents, with text, graphics, audio, video,
and links to other hypermedia documents.
 A navigational system based on “hyperlink”
 State–less interaction between client and server, conforming to
“http” protocol.

3.3.5 Mail–Based Applications

Plain–old mail
Notices
Auto–save and processing
News dissemination through LISTSRV
Archival search and retrieval
Access to network–wide news (bulletin boards)

3.4 Remote Procedure Call (RPC)

The designers chose to build three independent pieces; the NFS protocol itself, a
general purpose Remote Procedure Call (RPC) mechanism, and a general purpose
External Data Representation (XDR). Their intent was to separate the three to make
it possible to use RPC and XDR in other software, including application programs
as well as other protocols. For example, a programmer can divide a program into a
client side and a server side that use RPC as the chief communication mechanism
can one of the client sides, the programmer designates some procedures as remote,
forcing the compiler to incorporate RPC code into those procedures. On the server
side, the programmer implements the desired procedures and uses other RPC
facilities to declare them to be part of a server. When the executing client
program calls one of the remote procedures, RPC automatically collects values
for argument, from a message, sends the message to the remote server, awaits a
response, and stores returned values in the designated arguments. In essence,
communication with the remote server occurs automatically as a side– effect of a
remote call. The RPC mechanism hides all the details of protocols, making it
possible for programmers who know little about the underlying communication
protocols to write distributed programs.

3.5 File Transfer Protocol (FTP)

FTP (File Transfer Protocol) is the primary method of transferring files over the
Internet. “FTP” transfers files to and from a remote network site. Some sites
maintain Anonymous accounts on the system for retrieval of public domain
softwares stored on the system.

The ftp protocol is used to access files by FTP, the Internet’s file transfer
protocol. FTP has been around more than two decades and is well entrenched.
Numerous FTP servers all over the world allow people anywhere on the internet to
log in and download whatever files have been placed on the FTP server. The web
does not change this; it just makes obtaining files by FTP easier, as FTP has a
somewhat arcane interface.

3.6 Telnet

Telnet is a program that allows you to establish a virtual terminal connection

between two machines using TCP/IP. For this, you must have its internet
address or host name of computer.
4.0 CONCLUSION

In this unit, you have been taken through Network Layer, Transport
Layer and Application layer. Also the various algorithms, mechanisms,
and protocols relating to each of these layers have been discussed.

It has also discussed other concepts like multiplexing, crash recovery,

electronic mail, WWW, etc.

5.0 SUMMARY

In this unit, you have been introduced to the network, transport and
application layers, their features, services offered by them and the
algorithms used by them. Other concepts covered include
internetworking, repeaters, routes, bridges, multiplexing, addressing and
transport control mechanisms. Standards and definitions of commonly
used terms in Application Layer are covered briefly to familiarise you
with current trends.

6.0 TUTOR-MARKED ASSIGNMENT

i. What is the difference between virtual circuit and Datagram

subnets?…………………………………………………………...
……………………………………………………………………
……………………………………………………………………
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ii. Distinguish between non–adaptive and adaptive algorithms.

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7.0 REFERENCES/FURTHER READING

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