National Open University of Nigeria: School of Science and Technology

NATIONAL OPEN UNIVERSITY OF NIGERIA
SCHOOL OF SCIENCE AND TECHNOLOGY
COURSE CODE: CIT 311
COURSE TITLE: COMPUTR NETWORKS
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COURSE
GUIDE
CIT 311
COMPUTER NETWORKS
Course Team (Course Developer/Writer)

(Programme Leader)
(Course Co-ordinator)
NATIONAL OPEN UNIVERSITY OF NIGERIA
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National Open University of Nigeria
Headquarters
14/16 Ahmadu Bello Way
Victoria Island, Lagos
Abuja Office:
5 Dar–es–Salaam Street
Off Aminu Kano Crescent
Wuse II, Abuja
E–mail: [email protected]
URL: www.nou.edu.ng
Published by
National Open University of Nigeria
Printed 2009
Reprinted 2014
ISBN: 978-058-848-X
All Rights Reserved
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CONTENTS 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.
WORKING THROUGH THIS 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.
ASSESSMENT
The final grade in the course will have two components; continuous
assessment and final examination. Basically, the tutor-marked
assignment will form the continuous assessments while NOUN will
inform you about the final examination.
HOW TO GET THE MOST FROM THIS COURSE
In distance learning, the study units replace the University lecturer. This
is one of the great advantages of distance learning, you can read and
work through specially designed study materials at your own pace, and
at a time and place that suits you best. Think of it as reading the lecture
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instead of listening to a lecturer. Just as a lecturer might give you an in-
class exercise, your study units provide exercises for you to do at
appropriate points.
Working through the assignments and exercises will help you to achieve
the objectives of the unit and prepare you for the final examination. If
you run into trouble, arrange to meet your training tutor. Note the
following:
1. Read this course guide thoroughly

2. Organise a study schedule. Whatever method you choose to use,
you should decide on it and write in your own dates for working
on each.
3. Once you have created your own study schedule, do everything
you can to stick to it. The major reason that students fail is that
they get behind with their course work.
4. For each unit, read the introduction and the objectives.
5. Read through the materials. Work through the unit.
6. At the end of the unit, review the objective and see how many of
them you have achieved.
7. Do the tutor-marked assignment and submit as required.
8. Proceed unit by unit throughout the course.
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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MAIN
COURSE
CONTENTS PAGE
Module 1 An Introduction to Computer Networks…... 1
Unit 1 Network Classification and Reference

Models………………………………………….. 1
Unit 2 Data Transmission and Multiplexing……………. 17
Unit 3 Medium Access Control and Data Link Layer……..25
Unit 4 Network, Transport and Application Layers……… 33
Module 2 Network Devices & Technology………….. 49
Unit 1 Network Devices–I……………………………… 49

Unit 2 Network Devices–II……………………………… 62
Unit 3 Integrated Service Digital Network (ISDN)…… 75
Unit 4 Asynchronous Transfer Mode (ATM)…………. 85
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MODULE 1 AN INTRODUCTION TO COMPUTER
NETWORKS
Unit 1 Network classification and reference models

Unit 2 Data transmission and multiplexing
Unit 3 Medium access control and data link layer
Unit 4 Network, transport and application layers
UNIT 1 NETWORK CLASSIFICATION AND

REFERENCE MODELS
CONTENTS
1.0 Introduction
2.0 Objectives
3.0 Main Content
3.1 What is a network?
3.1.1 Classification of networks
3.1.2 Local area network
3.1.3 Metropolitan area network
3.1.4 Wide area network
3.2 Computer network goals / motivation
3.3 Applications of networks
3.4 Types of networks
3.4.1 Point–to–Point network
3.4.2 Broadcast network
3.5 Reference model
3.5.1 OSI reference model
3.5.1.1 The Physical Layer
3.5.1.2 The Data Link Layer
3.5.1.3 The Network Layer
3.5.1.4 The Transport Layer
3.5.1.5 The Session Layer
3.5.1.6 The Presentation Layer
3.5.1.7 The Application Layer
3.5.2 TCP Reference Model
3.5.3 Difference between OSI Reference Model and
TCP Reference Model
3.6 IEEE Standards for LANs
4.0 Conclusion
5.0 Summary
6.0 Tutor–Marked Assignments
7.0 References/Further Reading
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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
At the end of this unit, you should 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.
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.
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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
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 utilise public, leased or private
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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
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
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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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Machine A Machine B
Transport
layer
Internet Internet
Layer Internet layer Layer
Network
Network Network
Interface layer
Interface layer Interface layer
Network 1 Network 2
Figure 4: The Internet Layer and Network Interface Layers
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)
but supports both modes in
in transport layer (Co2 T. service is
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
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UNIT 2 DATA TRANSMISSION & MULTIPLEXING
CONTENTS
1.0 Introduction
2.0 Objectives
3.0 Main Content
3.1 Transmission
3.1.1 Time Domain Concepts
3.1.2 Frequency Domain Concepts
3.1.3 Relationship between Data rate and Bandwidth
3.2 Analog and digital Data Transmission
3.2.1 Data
3.2.2 Signals
3.2.3 Transmissions
3.3 Transmission Media
3.3.1 Twisted Pair
3.3.2 Baseband Coaxial Cable
3.3.3 Broadband Coaxial Cable
3.4 Multiplexing
4.0 Conclusion
5.0 Summary
6.0 Tutor–Marked Assignment
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
 types of transmission
 domain concepts – time and frequency
 difference between analog & digital signal
 type of transmission impairments
 transmission media.
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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
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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.
74
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
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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
76
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.
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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.
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6.0 TUTOR MARKED ASSIGNMENT
i. What is the difference between data rate and bandwidth?

………………………………………………………………………
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ii. List the characteristics of broadband coaxial cable

…………………………………………………………………………
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UNIT 3 MEDIUM ACCESS CONTROL AND DATA LINK LAYER
CONTENTS
1.0 Introduction
2.0 Objectives
3.0 Main Content
3.1 Data Link Layer
3.1.1 Services provided by Data Link Layer
3.1.2 Retransmission Strategies
3.2 Medium Access Control Sub Layer
3.2.1 Contention–Based Media Access Protocols
3.2.2 Polling Based Media Access Protocols
3.2.3 IEEE Standard 802.3 and Ethernet
3.2.4 IEEE Standard 802.4 Token bus
3.2.5 IEEE Standard 802.5 Token Ring
4.0 Conclusion
5.0 Summary
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
 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
80
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.
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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.
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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
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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.
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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.
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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.
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6.0 TUTOR MARKED ASSIGNMENT
i. How is Selective Repeat better than Go Back N?

……………………………………………………………………
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ii. What are the four LAN architectures?

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UNIT 4 NETWORK, TRANSPORT (TCP/IP) AND
APPLICATION LAYER
CONTENTS
1.0 Introduction
2.0 Objectives
3.0 Main Content
3.1 Network Layer
3.1.1 Routing Algorithms
3.1.1.1 Shortest path Routing
3.1.1.2 Flooding
3.1.2 Congestion Control Algorithms
3.1.3 Comparison of Virtual Circuit and Datagram subsets
3.1.4 Internetworking
3.1.4.1 Repeaters
3.1.4.2 Bridges
3.1.4.3 Routers
3.2 Transport Layer
3.2.1 Transport Service and Mechanism
3.2.2 Type of Service /Quality of Service
3.2.3 Transport Control Mechanism
3.2.3.1 Addressing
3.2.3.2 Flow Control and Buffering
3.2.3.3 Multiplexing
3.2.3.4 Connection Establishment and
Management
3.2.3.5 Crash Recovery.
3.2.4 TCP/UDP
3.2.4.1 Transmission Control Protocol (TCP)
3.2.4.2 Format of TCP Header
3.2.4.3 User Data Protocol (UDP)
3.3 Application Layer
3.3.1 The Domain Name System (DNS)
3.3.2 TCP/IP Internet Domain Name
3.3.3 Electronic Mail
3.3.4 World Wide Web (WWW)
3.3.5 Mail–Based Applications
3.4 Remote Procedure Call (RPC)
3.5 File Transfer Protocol (FTP)
3.6 Telnet
4.0 Conclusion
5.0 Summary
88
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
 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
89
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.
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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.
91
 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,
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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
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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.
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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.
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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.
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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.
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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
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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
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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.
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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
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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.
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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?…………………………………………………………...
……………………………………………………………………
……………………………………………………………………
……………………………………………………………………
ii. Distinguish between non–adaptive and adaptive algorithms.

………………………………………………………………….....
…………………………………………………………………….
…………………………………………………………………….
…………………………………………………………………….
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MODULE 2 NETWORK DEVICES & TECHNOLOGY
Unit 1 Network Devices–I

Unit 2 Network Devices–II
Unit 3 Integrated Services Digital Network (ISDN)
Unit 4 Asynchronous Transfer Mode (ATM)
UNIT 1 NETWORK DEVICES–I
CONTENTS
1.0 Introduction
2.0 Objectives
3.0 Main Content
3.1 Network Devices
3.1.1 Repeaters
3.1.2 Bridges
3.1.3 Switches
3.1.4 Hubs
3.1.5 Comparison of Switches and Hubs
4.0 Conclusion
5.0 Summary
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
 repeaters
 bridges
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 switches
 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.
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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.
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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.
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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
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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.
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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
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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.
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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.
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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
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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.
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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.
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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.
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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.
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UNIT 2 NETWORK DEVICES–II
CONTENTS
1.0 Introduction
2.0 Objectives
3.0 Main Content
3.1 Network Devices
3.1.1 Routers
3.1.2 Comparison of Bridges & Routers
3.1.3 Gateways
3.1.4 Modems
4.0 Conclusion
5.0 Summary
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
 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
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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
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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
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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.
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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.
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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.
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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.
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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
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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.
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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.
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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
ORIGINATE: ATQOV О TRAILBLAZER
IEISO=0 О NONE
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
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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.
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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.
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.
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UNIT 3 INTEGRATED SERVICE DIGITAL NETWORK
(ISDN)
CONTENTS
1.0 Introduction
2.0 Objectives
3.0 Main Content
3.1 Baseband and Broadband Communication
3.2 ISDN Services: BRI and PRI
3.3 Advantages of ISDN
3.4 ISDN Applications
4.0 Conclusion
5.0 Summary
1.0 INTRODUCTION
ISDN stands for Integrated Services Digital Network. It is an

international communications standard for sending voice, video and data
over digital telephone lines or normal telephone wires. This system
allows data to be transmitted simultaneously across the world using
end–to–end digital connectivity.
The early phone network consisted of a pure analog system that

connected telephone users directly by an interconnection of wires. This
system was very inefficient and was very prone to breakdown and noise,
and did not lend itself easily to long distance connections. Beginning in
the 1960s, the telephone system gradually began converting its internal
connections to a packet–based, digital switching system. Still, the final
connection from the local central office to the customer equipment was,
and still largely is, an analog Plain–old Telephone Service (POTS) line.
A standards movement was started by the International Telephone and

Telegraph Consultative Committee (CCIT), now known as International
Telecommunications Union (ITU). The ITU is a United Nations
organisation that coordinates and standardises international
telecommunications. Original recommendations of ISDN were in
CCITT Recommendation 1.120 (1984) which described some initial
guidelines for implementing ISDN.
In the term ISDN:
 “Integrated Services” refers to ISDN’s ability to deliver two

simultaneous connections, in any combination of data, voice, and
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fax, over a single line. Multiple devices can be attached to the
line and used as needed. That means an ISDN line can take care
of most people’s complete communications needs, without
forcing the purchase of multiple analog phone lines at a much
higher transmission rate.
 “Digital” refers to its purely digital transmission, as opposed to

the analog transmission of plain old telephone service. ISDN
transmits data digitally, resulting in a very clear transmission
quality.
 “Network” refers to the fact that ISDN is not simply a point–to–

point solution like a leased line. ISDN networks extend from the
local telephone exchange to the remote user and include all of the
telecommunications and switching equipment in–between. When
you have ISDN, you can make connections throughout the world
to other ISDN equipment. If your ISDN equipment include
analog capabilities, you can also connect to analog modems, fax
machines, and telephones, even though they may be connected to
plain old telephone service.
ISDN supports data transfer rates of 64Kbps. Most ISDN lines offered
by telephone companies give two lines at once, called Bearer channels
or B–Channels.
B–channels: You can use one line for voice and the other for data, or
you can use both lines for data to give you data rates of 128 Kbps, three
times the data rate provided by today’s fastest modems. Some switches
limit B channels to a capacity of 56 Kbps. A data channel (D–channel)
handles signalling at 16Kbps, depending on the service type.
2.0 OBJECTIVES
 explain the term ISDN

 differentiate between baseband and broadband communication
 explain the types of ISDN services
 discuss various advantages of ISDN
 explain the various applications of ISDN.
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3.0 MAIN CONTENT
3.1 Baseband and Broadband Communication
When the ISDN was originally designed, data rates of 64Kbps to 1.544
Mbps were sufficient to handle all existing transmission needs. As
application using the telecommunication network advanced, however,
these rates proved inadequate to provide for the needs of the next
generation of technology, called broadband. ISDN is under development
which uses broadband communication. The original version of ISDN
employs baseband transmission. Let us look at the difference between
the two.
Baseband Communication
A type of digital data transmission in which each medium (wire) carries

only one signal, or channel, at a time. In contrast, broadband
transmission enables a single wire to carry multiple signals
simultaneously.
Most communications involving computers use baseband transmission.

This includes communication from the computer to devices (printers,
monitors, and so on), communications via modems, and the majority of
networks. An exception is B–ISDN networks, which use broadband
transmission.
Broadband communication
This is a standard for transmitting voice, video and data at the same time
over fibre optic telephone lines. Broadband ISDN can support data rates
of 1.5 million bits per second (bps), but it has not been widely
implemented.
The original version of ISDN employs baseband transmission. Another

version called B–ISDN, uses broadband transmission and is able to
support transmission rates of 1.5Mbps. B-ISDN requires fibre optic
cables and is not widely available.
3.2 ISDN Services
There are two basic types of ISDN services:
 Basic Rate Interface (BRI)

 Primary Rate Interface (PRI)
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BRI consists of two 64kbps B – channels and one 16 kbps D – channel
for a total of 144 kbps. This basic service is intended to meet the needs
of most individual users. BRI service is by far the most common and is
typically found in homes and businesses alike.
H channels provide a way to aggregate B channels. They are
implemented as:
 H0 = 384 kbs (6 B–channels)

 H010 = 1472 kbps (23 B–channels)
 H011 = 1536 kbps (24 B–channels)
 H012 = 1920 kbps (30 B–channels)–International (E1) only.
To access BRI service, it is necessary to subscribe to an ISDN phone

line. The only limitation to access BRI service is that the customer must
be within 18,000 feet (about 3.4 miles or 5.5 km) of the telephone
company’s central office for BRI service; beyond that, expensive
repeater devices are required, or ISDN service may not be available at
all. Customers will also need special equipment to communicate with
the phone company switch and with the other ISDN devices. These
devices include ISDN Terminal Adapters and ISDN Routers.
In BRI service at an office or a home, the ISDN line consists of the same
twisted pair of wires traditionally used for analog telephones. Numerous
ISDN devices can connect to this single line.
The BRI line provides two types of ISDN communications channels
 Two “bearer service” B–channels, which carry data and services

at 64 kbps (or combine for 128 kbps), and
 A single 16 Kbps D–channel, which usually carries signaling and

administrative information used to set up and terminate calls.
Up to eight ISDN devices can be connected to a single BRI line, and can
all share the B–channel and D–channel. Individual devices are
distinguished through the use of multiple subscriber numbers, with a
different ISDN number assigned to each device. D–channel signals
automatically route communications to the appropriate ISDN device.
Although only two B–channels are available at any point in time,

numerous other calls may be put “on hold” via D–channel signaling, a
feature referred to as “multiple call appearances”.
PRI is intended for users with greater capacity requirements. Typically

the channel structure is 23 B–channels plus one 64kbps D–channel for a
total of 1536 kbps. (in Europe, PRI consists of 30 B channels plus one
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64 kbps D–channel for a total of 1984 kbps. It is also possible to support
multiple PRI lines with one 64kbps D–channel using Non–Facility
Associated Signaling (NFAS)
The higher capacity PRI service is a central–site solution for extending

applications to large numbers of remote users, who communicate
through their BRI connections. With its BRI and PRI services, ISDN has
the flexibility to meet the bandwidth needs of a home office, branch
office, or company headquarters. A small office can use ISDN BRI to
support all of its voice and data communications requirements. In a
larger office, multiple ISDN BRI lines can be divided among multiple
users and applications via a server of PBX. And users in a very large
office can benefit from ISDN PRI’s capacity to provide a large quantity
of multiple B–channels.
3.3 Advantages of ISDN
There are many advantages of ISDN. In this section we will look at

them.
 Speed
The modem was a big breakthrough in computer communications. It

allowed computers to communicate by converting their digital
information into analog signal to travel through the public phone
network. There is an upper limit to the amount of information that an
analog telephone line can hold. Currently, it is about 56kbps. Commonly
available modems have a maximum speed of 56kbps, but are limited by
the quality of the analog connection and actual speed goes to about 45
kbps. ISDN is a replacement for plain old telephone service, which was
never designed to meet the needs of the information age. ISDN allows
multiple digital channels to be operated simultaneously through the
same regular phone wiring used for analog lines. The major advantage
here is that the telephone company’s switches can support digital
connections. Therefore, the same physical wiring can be used, but a
digital signal, instead of an analog signal, is transmitted across the line.
You get ISDN service from the same companies who provide telephone
service, and you use it to connect telephones, computers, and fax
machines. The difference is that you get much faster, much more
dependable connections for voice, data, fax, and even video – all
through a single line. BRI ISDN, using a channel aggregation protocol
such as BONDING or Multilink–PPP, supports an uncompressed data
transfer speed of 128 kbps. In addition, the latency, or the amount of
time it takes for a communication to begin on an ISDN line is typically
about half that of an analog line. You can plug an ISDN adapter into a
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phone jack, like you would an analog modem, and get a much faster
connection with no “line noise”.
 Multiple Devices
Before the advent of ISDN, it was necessary to have a phone line for
each device that has to be used simultaneously. For example, one line
each was required for a telephone, fax, computer, bridge/router, and live
video conference system. Transferring a file to someone while talking
on the phone or seeing their live picture on a video screen would require
several potentially expensive phone lines. With ISDN, it has become
possible to combine many different digital data sources and have the
information routed to the proper destination. Since the line is digital, it is
easier to keep the noise and interference out while combining these
signals. ISDN technically refers to a specific set of digital services
provided through a single, standard interface. Without ISDN, distinct
interfaces are required instead.
 Signaling
Instead of the phone company sending a ring voltage signal to ring the
bell in your phone (“In–Band signal”), it sends a digital packet on a
separate channel (“Out–of–Band signal”). The Out–of–Band signal does
not disturb established connections, and call setup time is very fast. For
example, a V.34 modem typically takes 30–60 seconds to establish
connection, whereas an ISDN call usually takes less than 2 seconds. The
signaling also indicates who is calling, what type of call it is
(data/voice), and what number was dialled. Available ISDN phone
equipment is then capable of making intelligent decisions on how to
direct the call.
3.4 ISDN Applications
There are many applications of ISDN. In this section, we will briefly

describe some of the applications.
 Internet Access
Internet access is one of the most popular applications of ISDN.

Compared with even the fastest modem access, ISDN makes Web
graphics appear almost immediately, and can reduce download times by
over 75%. ISDN can even provide advantages over shared, higher–
bandwidth office connections.
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 Telephony
ISDN provides two exceptionally clear, all–digital telephone

connections per BRI line. Connections are established in 2 to 4 seconds,
versus the 10 to 30 seconds required for analog telephone connections.
The digital ISDN telephone sets include microprocessors that can drive
features unavailable in regular telephones, such as automated call–back
and direct lines to computer-based call centre applications. The savings
in connection establishment time alone can be extremely beneficial.
Though 10 to 30 seconds doesn’t sound like much, it can mount up
quickly in large–scale phone centres. For example, USA today uses
ISDN in its Maryland Customer Service Centre. The database links
provide the same benefits found in analog–fed CTI systems, but the
reduction in connect time has cut an average of 15 seconds each from
more than a million calls received each year – that’s a total savings of
over 4,000 hours, the equivalent of two full–time operators.
 Telecommuting
It is the latest buzzword in IT which is described as the modern way of

working from home. ISDN has made telecommuting a reality. With a
single ISDN line and an ISDN adapter, telephone, and fax machine,
remote professionals can enjoy the same communications capabilities as
office–bound workers. Because of the high transmission speed, ISDN
succeeds in making remote LAN access seem very much like being
locally attached to the LAN. And because a single ISDN line can carry
simultaneous data and analog communications, such as telephone calls
and faxes, one line can turn the home desktop into an efficient, cost–
effective communications centre.
 Video Conferencing
ISDN also supports videoconferencing. One channel is used for voice,

and the other channel is used for the display of moving video pictures.
Thus, the remote professionals can actually communicate face–to–face
with each other. Video conferencing is an emerging ISDN application
that’s growing fast in popularity, and ISDN is currently the only
practical way of making it happen. The ability to transmit quality video
and voice across distances used to require very expensive equipment and
costly leased lines that were justified only by the most rigorous needs in
the largest of companies. And because they depended on leased lines,
those video conferencing solutions were point–to–point; a headquarters
facility could be linked to satellite facilities, for example, but video
conferencing on the scale of teleconferencing was simply impractical.
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ISDN, along with new, lower–cost hardware, is changing that rapidly.
Both desktop video conferencing (where a participant joins from a PC
equipped with ISDN, a video camera, and a microphone), and
conference room video conferencing (where more sophisticated
equipment such as remote control cameras, allow group participation)
have become as easy to set up as voice conferencing. And because of
ISDN’s versatility, video conferences can include the sharing of
documents, images, and other files with all participants, through file
transfers or screen display. This capability is proving highly popular for
workgroup collaboration, telecommuting, security and surveillance, and
dozens of highly innovative applications.
For general business use, ISDN video conferencing can make meetings
much more practical and productive. There is far less travel time and
expense involved, as participants no longer need to physically
congregate. That also makes scheduling much easier. And since
participants are typically at their own offices, video conferencing means
everyone has access to their computers and file cabinets for information
during the meeting. Other people can also be called into the meeting as
needed.
 Education
“Distance learning” can be thought of as telecommuting for students.

And just as ISDN has made telecommuting a reality for thousands of
professional workers, it is making distance learning a practicality for
ground–breaking public and private grade schools, high schools, and
colleges throughout North America. Computer–based, interactive voice,
data, image, and video can go a long way toward helping students who
are unable to physically visit a classroom to participate in the learning
experience. For example, the College of Education at Ohio State
University links elementary schools in the Appalachian region to the
university via ISDN, bringing advanced courses even to the most remote
locations. At Michigan’s Upper Peninsula Intermediate school district,
students who formerly travelled up to 80 miles to school now attend 11
local facilities to participate in a full roster of classes, delivered by
ISDN. There are many additional examples of similar successful
programs throughout the U.S.
 Large–scale file transfers
As computer applications have become increasingly graphically

oriented, PC communications have the ability, more than ever, to
include the pictures, sounds, and even full–motion video that enable
people to show, rather than just tell. Before ISDN, It wasn’t practical to
communicate this way over distance – downloading a 1–megabyte file
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with even the fastest modem takes several minutes. With ISDN’s faster
speed and compression, it takes seconds.
If you’ve ever encountered a web site that features a video clip you’re
supposed to watch on your screen, you will understand the limits of
modems in multimedia communications. With ISDN, however,
multimedia is going to be practical. Many organisations are beginning to
turn to imaging systems to store documents, rather than keeping endless
files of papers. Once a document has been scanned, it is a graphical file
that can be viewed on a computer screen. To transmit it to someone
requires the bandwidth of ISDN, and ISDN is becoming very popular
with publishers, banks, photo agencies, hospitals, Police departments,
manufacturers, government agencies, and countless other industry
segments that are driven by digitally stored documents that need to be
sent to others.
4.0 CONCLUSION
In this unit, you have been exposed to the concepts of baseband and
broadband communication; ISDN, types of ISDN services, advantages
of ISDN and its applications.
You should be able to discuss these various concepts and differentiate

them where necessary.
5.0 SUMMARY
In this unit, we covered the following topics related to ISDN:
1. ISDN stands for Integrated Services Digital Network. It is an

international communications standard for sending voice, video,
and data over digital telephone lines or normal telephone wires.
2. ISDN supports data transfer rates of 64 kbps. Most ISDN lines
offered by telephone companies give two lines at once, called
Bearer channels or B–channels.
3. Baseband communication is a type of digital data transmission in
which each medium (wire) carries only one signal, or channel, at
a time.
4. Broadband communication is a standard for transmitting voice,
video and data at the same time over fibre optic telephone lines.
5. ISDN provides two types of services:
 Basic Rate Interface (BRI) and
 Primary Rate Interface (PRI)
6. Basic Rate Interface consists of two 64 kbps B–channels and one
16 kbps D–channel for a total of 144 kbps. BRI service is by far
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the most common and is typically found in homes and businesses
alike.
7. Primary Rate Interface is intended for users with greater capacity
requirements. Typically, the channel structure is 23 B–channels
plus one 64 kbps D–channel for a total of 1536 kbps. (In Europe,
PRI consists of 30 B–channels plus one 64 kbps D–channel for a
total of 1984 kbps. It is also possible to support multiple PRI
lines with one 64kpbs D–channel using Non–Facility Associated
Signaling (NFAS).
8. ISDN is a replacement for plain old telephone service, which was

never designed to meet the needs of the information age. ISDN
allows multiple digital channels to be operated simultaneously
through the same regular telephone wiring used for analog lines.
The major advantage here is that the telephone company’s
switches can support digital connections.
9. With ISDN, it has become possible to combine many digital data

sources and have the information routed to the proper destination.
1. Define ISDN
2. Differentiate between baseband and broadband communication.
3. List a few applications of ISDN.
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UNIT 4 ASYNCHRONOUS TRANSFER MODE (ATM)
CONTENTS
1.0 Introduction
2.0 Objectives
3.0 Main Content
3.1 Switching – Techniques
3.2 How compatible is ATM as technology
3.3 ATM–Layered Architecture in comparison with OSI
Model
3.4 How ATM Protocol Works
3.5 The ATM Network
3.6 The ATM Cell
3.7 ATM Classes of Service
3.8 ATM Traffic Control
3.9 Benefits of ATM
3.10 ATM Applications
4.0 Conclusion
5.0 Summary
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.
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2.0 OBJECTIVES
 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
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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
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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.
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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,
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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
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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.
147
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
148
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
149
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.
150
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
151
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).
152
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
153
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.
154
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.
155
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.
156
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
157
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.
158
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.
159
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.
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.
160

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NATIONAL OPEN UNIVERSITY OF NIGERIA

SCHOOL OF SCIENCE AND TECHNOLOGY

COURSE CODE: CIT 311

COURSE TITLE: COMPUTR NETWORKS

Course Team (Course Developer/Writer)

NATIONAL OPEN UNIVERSITY OF NIGERIA

All Rights Reserved

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

WHAT YOU WILL LEARN IN THIS COURSE

WORKING THROUGH THIS COURSE

Computers will be provided at study centres and will be equipped with

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

HOW TO GET THE MOST FROM THIS COURSE

1. Read this course guide thoroughly

Computers in Society intend to make you computer literate. At the end

We wish you a huge success.

Module 1 An Introduction to Computer Networks…... 1

Unit 1 Network Classification and Reference

Module 2 Network Devices & Technology………….. 49

Unit 1 Network Devices–I……………………………… 49

Unit 1 Network classification and reference models

UNIT 1 NETWORK CLASSIFICATION AND

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

At the end of this unit, you should be able to:

 define and classify network

3.0 MAIN CONTENT

3.1 What Is A Network

(i) The devices are very far apart.

Solution to this problem is to connect each device to a communication

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

 Local Area Network (LAN)

3.1.2 Local Area Network (LAN)

A local area network is a relatively smaller and privately owned network

Accordingly, there are many LAN standards known as IEEE area

3.1.3 Metropolitan Area Network (MAN)

Metropolitan Area Network is defined as less than 50km and provides

3.1.4 Wide Area Network (WAN)

Wide Area Network provides no limit of distance. In most WAN, the

A WAN provides long distance transmission of data, voice, image and

In contrast to LANs (which depend on their own hardware for

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

3.2 Computer Network Goals/Motivation

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

3.3 Applications of Networks

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

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

We are now moving from personalized computing to network

3.4 Types of Network

3.5.1 Point–to–Point Network or Switch Network

Point–to–Point networks consist of many connections between

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

In a star topology, each device has a dedicated Point–to–Point link only

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

The advantages and disadvantages of a tree topology are generally the

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

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

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

3.4.2 Broadcast Networks

Broadcast networks have a single communication channel that is shared