USIT303-Computer-Networks-munotes

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1Unit -I
1
INTRODUCTION
Unit Structure
1.0 Objectives
1.1 Introduction
1.2 Data communications
1.3 Networks
1.4 Types of Network
1.5 Internet history
1.6 Internet Standards
1.7 Internet Administration
1.8 Review questions
1.9 Summary
1.10 References
1.0 OBJECTIVES:
This chapter would make you understand the following concepts
What is data communication?
What is network? Types of network.
Brief history of internet
Different standards and administration of the internet.
1.1 INTRODUCTION
Communication is sharing information or providing entertainment
by speaking, writing or other methods. Probably the most important type
of communication is personal communication, which happens when
people make their thoughts and wishes k nown to each other. There are
many methods of communication. We have come a long way from the
prehistoric times. In those days methods like smoke signals, certain sounds
were used to communicate with each other. Then human speech developed
and people began to talk and share their thoughts with one another. Not to
mention the present days, the world of electronic communication. This
world will have to be reinvented if there was no communication. No
business would be done. No parent would understand what his child wants
from him. There would be no teaching in classes. We would be worst than
a rodent.munotes.in

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2The Concept of Data Communication evolved from sharing the
computation power of computer along with various resources available in
a computer environment such as printer, hard disk etc. With increasing
demand for exchange of information across the globe, the need of data
communication has increased in many folds. Data Communication, can be
used to transfer or exchange information with in one building, one city,
across cities, countries and continents. It is also possible to update and
share data at different locations.
By Data Communication we mean the transportation of
information from one point to another through a communication media.
The word data refers to f acts, concepts, and instructions presented in
whatever form is agreed upon by parties creating and using the data. In the
context of computer information systems data are represented by binary
information units (or bits) produced and consume in form of 0s and 1s.
Data communication is exchange of data (in the form of 0s and 1s)
between two devices via some form of transmission medium (such as wire
cables). Data communication is considered local if the communicating
devices are in the same building or a si milarly restricted graphical area,
and is considered remote if the devices are farther apart.
1.2 DATA COMMUNICATIONS
The main components of data communication are data sources,
data sinks and communication media. The source is the originator of
inform ation, while sink is the receiver of information. The media is the
path through which the information is transported to the sink from the
sources. This media could be a telephone wire, a microwave system on a
satellite circuit or a fiber optic line. Usuall y, the media is provided by one
or more common communication carriers. The computer equipment is
connected to the communication media through apiece of equipment
called MODEM. This piece of equipment converted the digital signal to
analog and passes it to the communication media through whom they are
propagated towards the si nk. The sink is similarly connected to the
communication media through a modem and receives the propagated
signals.
Media
Communication Protocol
All communications between devices require that the devices agree
on the format of data. The set of rules defining a format is called a
protocol. At the very least, a communications, protocol must define the
followingData Source Data Sinkmunotes.in

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3rate of trans mission (in baud or bps)
whether transmission is to be synchronous or asynchronous
whether data is to be transmitted in half -duplex or full -duplex
mode
In addition, protocols can include sophisticated techniques for
detecting and recovering from transm ission errors and for encoding and
decoding data.
Characteristics of Data Communication
The effectiveness of any data communications system depends
upon the following four fundamental characteristics:
1.Delivery : The data should be delivered to the correct destination and
correct user.
2.Accuracy : The communication system should deliver the data
accurately, without introducing any errors. The data may get corrupted
during transmission affecting the accuracy of t he delivered data.
3.Timeliness : Audio and Video data has to be delivered in a timely
manner without any delay; such a data delivery is called real time
transmission of data.
4.Jitter : It is the variation in the packet arrival time. Uneven Jitter may
affect the timeliness of data being transmitted
Data Communication Terminology
Data Communication is the process of transferring data from one
machine to another machine such that the sender and receiver both
interpret the data correctly.
1.Data Channel
In communications the term channel refers to a communications
path between two computers or devices. It may refer to the physical
medium, such as coaxial cable or to a specific carrier frequency (sub -
channel) within a larger channel or wireless medium.
2.Baud
Pronounced bawd, it is the number of signaling elements that occur
each second. The term is named after J.M.E. Baudot, the inventor of the
Baudot telegraph code.
At slow speeds, only one bit of information(signaling element) is
encoded in each elec trical change. The baud, therefore, indicates the
number of bits per second that are transmitted. For example, 300 baud
means that 300 bits are transmitted each second (abbreviated 300 bps).
Assuming asynchronous communication, which requires 10 bits per
character; this translates to 30 characters per second (cps). For slow ratesmunotes.in

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4you can divide the baud by 10 to see how many characters per second are
sent.
At higher speeds, it is possible to encode more than one bit in each
electrical change, 4,800 baud m ay allow 9,600 bits to be sent each second.
At high data transfer speeds, therefore, data transmission rates are usually
expressed in bits per second (bps) rather than baud. For example, a 9,600
bps modem may operate at only 2,400 baud.
3.Bandwidth
Band width is the amount of data that can be transmitted in a fixed
amount of time. For digital devices, the bandwidth is usually expressed in
bits per second (bps) or bytes per second. For analog devices, the
bandwidth is expressed in cycles per second, or Her tz(Hz).
The bandwidth is particularly important for I/O devices. For
example, a fast disk drive can be hampered by a bus with a low
bandwidth.
4.Data Transfer Rates
The amount of data transferred per second by a communication
channel is known as data transfer rate. It is measured in bits per second
(bps)
Components of Data Communication
A Data Communication system has five components as shown in
the diagram below:
Fig. Components of a Data Communication System
1.Message :
Message is the information to be communicated by the sender to the
receiver.
2.Sender
The sender is any device that is capable of sending the data (me ssage).munotes.in

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53.Receiver
The receiver is a device that the sender wants to communicate the data
(message).
4.Transmission Medium
It is the path by which the message travels from sender to receiver. It can
be wired or wireless and many subtypes in both.
5.Protocol
It is an agreed upon set or rules used by the sender and receiver to
communicate data. A protocol is a set of rules that governs data
communication. A Protocol is a necessity in data communications without
which the communicating entities are like two persons trying to talk to
each other in a different language without know the other language
Data Representation
Data is collection of raw facts which is processed to deduce
information. There may be different forms in which data may be
represented. Some of the forms of data used in communications are as
follows:
1.Text
Text includes combination of alphabets in small case as well as
upper case. It is stored as a pattern of bits. Prevalent encoding system :
ASCII, Unicode
2.Numbers
Numbers include combination of digits from 0 to 9. It is stored as a
pattern of bits. Prevalent encoding system : ASCII, Unicode
3.Images
An image is worth a thousand words ‖is a very famous saying. In
computers images are digitally stored. A Pixel is the smallest element of
an image. To put it in simple terms, a picture or image is a matrix of pixel
elements. The pixels are represented in the form of b its. Depending upon
the type of image (black n white or color) each pixel would require
different number of bits to represent the value of a pixel. The size of an
image depends upon the number of pixels (also called resolution) and the
bit pattern used to indicate the value of each pixel. Example: if an image is
purely black and white (two color) each pixel can be represented by a
value either 0 or 1, so an image made up of 10 x 10 pixel elements would
require only 100 bits in memory to be stored. On the ot her hand an image
that includes gray may require 2 bits to represent every pixel value (00 -
black, 01 –dark gray, 10 light gray, 11 –white). So the same 10 x 10 pixel
image would now require 200 bits of memory to be stored. Commonly
used Image formats : jpg, png, bmp, etcmunotes.in

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64.Audio
Data can also be in the form of sound which can be recorded and
broadcasted. Example: What we hear on the radio is a source of data or
information. Audio data is continuous, not discrete.
5.Video
Video refers to broadcast ing of data in form of picture or movie.
Data Flow
We devices communicate with each other by sending and receiving data.
The data can flow between the two devices in the following ways.
1.Simplex
2. Half Duplex
3. Full Duplex
1.Simplex
Figure: Simplex mode of communication
In Simplex, communication is unidirectional .Only one of the
devices sends the data and the other one only receives the data. Example:
in the above diagram: a CPU sends data while a mo nitor only receives
data.
2. Half Simplex
Figure: Half Duplex Mode of Communication
In half duplex both the stations can transmit as well as receive but
not at the same time. When one device is sending other can only receive
and vice -versa (as shown in figure above.) Example: A walkie -talkie.
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Figure: Full Duplex Mode of Communication
In Full duplex mode, both stati ons can transmit and receive at the same
time. Example: mobile phones
1.3 NETWORKS
A computer network is a group of computers that use a set of
common communication protocols over digital interconnections for the
purpose of sharing resources located on or provided by the network nodes.
The interconnections between nodes are formed from a broad spectrum of
telecommunication network technologies, based on physically wired,
optical, and wireless radio -frequency methods that may be arranged in a
variety of n etwork topologies.
The nodes of a computer network may include personal computers,
servers, networking hardware, or other specialized or general -purpose
hosts. They are identified by hostnames and network addresses.
Hostnames serve as memorable labels for the nodes, rarely changed after
initial assignment. Network addresses serve for locating and identifying
the nodes by communication protocols such as the Internet Protocol.
Computer networks may be classified by many criteria, including
the transmission medium used to carry signals, bandwidth, and
communications protocols to organize network traffic, the network size,
the topology, traffic control mechanism, and organizational intent.
Computer networks support many applications and services, such
as acce ss to the World Wide Web, digital video, digital audio, shared use
of application and storage servers, printers, and fax machines, and use of
email and instant messaging applications.
1.4 NETWORK TYPES
A computer network is a group of computers linked to each other
that enables the computer to communicate with another computer and
share their resources, data, and applications.
A computer network can be categorized by their size. A computer
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Figure : Types of network
1.LAN(Local Area Network)
Local Area Network is a group of computers connected to each
other in a small area such as building, office.
LAN is used for connecting two or more personal computers
through a communication medium such as twisted pair, coaxial
cable, etc.
It is less costly as it is built with inexpensive hardware such as
hubs, network adapters, and Ethernet cables.
The data is transferred at an extremely faster rate in Local Area
Network.
Local Area Network provides higher security.
Figure LAN (Local Area Network)
2.PAN (Personal Area Network)
Personal Area Network is a network arranged within an individual
person, typically within a range of 10 meters.
Personal Area Network is used for connecting the computer
devices of personal use is known as Personal Area Network.
Thomas Zimmerman was the first research scientist to bring the
idea of the Personal Area Network.
Personal Area Network covers an area of 30 feet .munotes.in

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9Personal computer devices that are used to develop the personal
area network are the laptop, mobile phones, media player and play
stations.
Figure PAN (Personal Are a Network)
There are two types of Personal Area Network:
Wired Personal Area Network
Wireless Personal Area Network: Wireless Personal Area Network is
developed by simply using wireless technologies such as WiFi, Bluetooth.
It is a low range network.
Wireless Personal Area Network
Wired Personal Area Network: Wired Personal Area Network is
created by using the USB.
Examples of Personal Area Network:
Body Area Network: Body Area Network is a network that moves
with a person. For example , a mobile network moves with a person.
Suppose a person establishes a network connection and then creates
a connection with another device to share the information.
Offline Network: An offline network can be created inside the
home, so it is also known as a home network . A home network is
designed to integrate the devices such as printers, computer,
television but they are not connected to the internet.
Small Home Office: It is used to connect a variety of devices to the
internet and to a corporate network using a VPN
3.MAN (Metropolitan Area Network)
A metropolitan area network is a network that covers a larger
geographic area by interconnecting a different LAN to form a
larger network.
Government agencies use MAN to connect to the citizens and
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10In M AN, various LANs are connected to each other through a
telephone exchange line.
The most widely used protocols in MAN are RS -232, Frame
Relay, ATM, ISDN, OC -3, ADSL, etc.
It has a higher range than Local Area Network(LAN).
Figure MAN (Metropolitan Area Network)
Uses of Metropolitan Area Network:
MAN is used in communication between the banks in a city.
It can be used in an Airline Reservation.
It can be used in a college within a city.
It can also be used for communication in the military.
4. WAN (Wide Area Network)
A Wide Area Network is a network that extends over a large
geographical area such as states or countries.
A Wide Area Network is quite bigger network than the LAN.
A Wide Ar ea Network is not limited to a single location, but it
spans over a large geographical area through a telephone line, fibre
optic cable or satellite links.
The internet is one of the biggest WAN in the world.
A Wide Area Network is widely used in the field of Business,
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Figure WAN( Wide Area Network)
Examples of Wide Area Network:
Mobile Broadband : A 4G network is widely used across a region or
country
Last mile : A telecom company is used to provide the internet services to
the customers in hundreds of cities by connecting their home with fiber.
Private network : A bank provides a private network that connects the 44
offices. This network is made by using the tel ephone leased line provided
by the telecom company.
1.5 INTERNET HISTORY
What is Internet?
The Internet (or internet) is the global system of interconnected
computer networks that uses the Internet protocol suite (TCP/IP) to
communicate between networks and devices. It is a network of networks
that consists of private, public, academic, business, and government
networks of local to global scope, linked by a broad array of electronic,
wireless, and optical networking technologies. The Internet carries a va st
range of information resources and services, such as the inter -linked
hypertext documents and applications of the World Wide Web (WWW),
electronic mail, telephony, and file sharing.
Brief History of Internet
A network is a group of connected communicat ing devices such as
computers and printers. An internet (note the lowercase letter i) is two or
more networks that can communicate with each other. The most notable
internet is called the Internet (uppercase letter I), a collaboration of more
than hundreds of thousands of interconnected networks. Private
individuals as well as various organizations such as government agencies,munotes.in

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12schools, research facilities, corporations, and libraries in more than 100
countries use the Internet. Millions of people are users. Yet this
extraordinary communication system only came into being in 1969.
In the mid -1960s, mainframe computers in research organizations
were standalone devices. Computers from different manufacturers were
unable to communicate with one another. The Ad vanced Research
Projects Agency (ARPA) in the Department of Defense (DoD) was
interested in finding a way to connect computers so that the researchers
they funded could share their findings, thereby reducing costs and
eliminating duplication of effort.
In 1967, at an Association for Computing Machinery (ACM)
meeting, ARPA presented its ideas for ARPANET, a small network of
connected computers. The idea was that each host computer (not
necessarily from the same manufacturer) would be attached to a
speciali zed computer, called an interface message processor (IMP). The
IMPs, in tum, would be connected to one another. Each IMP had to be
able to communicate with other IMPs as well as with its own attached
host.
By 1969, ARPANET was a reality. Four nodes, at t he University
of California at Los Angeles (UCLA), the University of California at
Santa Barbara (UCSB), Stanford Research Institute (SRI), and the
University of Utah, were connected via the IMPs to form a network.
Software called the Network Control Proto col (NCP) provided
communication between the hosts.
In 1972, Vint Cerf and Bob Kahn, both of whom were part of the
core ARPANET group, collaborated on what they called the Interknitting
Project. Cerf and Kahn's landmark 1973 paper outlined the protocols to
achieve end -to-end delivery of packets. This paper on Transmission
Control Protocol (TCP) included concepts such as encapsulation, the
datagram, and the functions of a gateway.
Shortly thereafter, authorities made a decision to split TCP into
two prot ocols: Transmission Control Protocol (TCP) and Internetworking
Protocol (lP). IP would handle datagram routing while TCP would be
responsible for higher -level functions such as segmentation, reassembly,
and error detection. The internetworking protocol bec ame known as
TCPIIP
1.6 INTERNET STANDARDS
An Internet standard is a thoroughly tested specification that is
useful to and adhered to by those who work with the Internet. It is a
formalized regulation that must be followed. There is a strict procedure by
which a specification attains Internet standard status. A specification
begins as an Internet draft. An Internet draft is a working document (amunotes.in

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13work in progress) with no official status and a six -month lifetime. Upon
recommendation from the Internet authorities, a draft may be published as
a Request for Comment (RFC). Each RFC is edited, assigned a number,
and made available to all interested parties. RFCs go through maturity
levels and are categorized according t o their requirement level.
Maturity Levels
An RFC, during its lifetime, falls into one of six maturity levels:
proposed standard, draft standard, Internet standard, historic,
experimental, and informational.
Proposed Standard
A proposed standard is a specification that is stable, well
understood, and of sufficient interest to the Internet community. At this
level, the specification is usually tested and implemented by several
different groups.
Draft Standard
A proposed standard is elevated to draft st andard status after at
least two successful independent and interoperable implementations.
Barring difficulties, a draft standard, with modifications if specific
problems are encountered, normally becomes an Internet standard.
Internet Standard
A draft s tandard reaches Internet standard status after
demonstrations of successful implementation
Historic
The historic RFCs are significant from a historical perspective.
They either have been superseded by later specifications or have never
passed the necessar y maturity levels to become an Internet standard.
Experimental
An RFC classified as experimental describes work related to an
experimental situation that does not affect the operation of the Internet.
Such an RFC should not be implemented in any functio nal Internet
service.
Informational
An RFC classified as informational contains general, historical, or
tutorial information related to the Internet. It is usually written by someone
in a non -Internet organization, such as a vendor.
Requirement Levels
RFCs are classified into five requirement levels: required,
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14Required
An RFC is labeled required if it must be implemented by all
Internets systems to achieve minimum conformance. For example, IF and
ICMP are required protocols.
Recommended
An RFC labeled recommended is not required for minimum
conformance; it is recommended because of its usefulness. For example,
FTP and TELNET are recommended protocols.
Elective
An RFC labeled elective is not required and not recommended.
However, a system can use it for its own benefit.
Limited Use
An RFC labeled limited use should be used only in limited
situations. Most of the experimental RFCs fall under this category.
Not Recommended
An RFC labeled no t recommended is inappropriate for general use.
Normally a historic (deprecated) RFC may fall under this category.
1.7 INTERNET ADMINISTRATION
The Internet, with its roots primarily in the research domain, has
evolved and gained a broader user base with significant commercial
activity. Various groups that coordinate Internet issues have guided this
growth and development. Appendix G gives the add resses, e -mail
addresses, and telephone numbers for some of these groups. Shows the
general organization of Internet administration. E -mail addresses and
telephone numbers for some of these groups.
Figure the general organization of Internet administration.
ISOCmunotes.in

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15The Internet Society (ISOC) is an international, nonprofit
organization formed in 1992 to provide support for the Internet standards
process. ISOC accomplishes this through maintaining and supporting
other Internet administrative bodies such as lAB, IETF,IRTF, and IANA
(see the following sections). ISOC also promotes research and other
scholarly activities relating to the Internet.
IAB
The Internet Architecture Board (IAB) is the technical advisor to
the ISOC. The main purposes of the lAB are to oversee the continuing
development of the TCP/IP Protocol Suite and to serve in a technical
advisory capacity to research members of the Internet community. IAB
accompli shes this through its two primary components, the Internet
Engineering Task Force (IETF) and the Internet Research Task Force
(IRTF). Another responsibility of the IAB is the editorial management of
the RFCs, described earlier. IAB is also the external lia ison between the
Internet and other standards organizations and forums.
JETF
The Internet Engineering Task Force (IETF) is a forum of working
groups managed by the Internet Engineering Steering Group (IESG). IETF
is responsible for identifying operationa l problems and proposing
solutions to these problems. IETF also develops and reviews specifications
intended as Internet standards. The working groups are collected into
areas, and each area concentrates on a specific topic. Currently nine areas
have been defined. The areas include applications, protocols, routing,
network management next generation (lPng), and security.
JRTF
The Internet Research Task Force (IRTF) is a forum of working
groups managed by the Internet Research Steering Group (IRSG). IRTF
focuses on long -term research topics related to Internet protocols,
applications, architecture, and technology .
1.8 REVIEW QUESTIONS
1. Explain the concept of Computer network.
2.How data communication is done ?Explain in brief.
3. What Computer Network ?Explain types of network.
4.What is Internet? Explain brief history of Internet .
5.Why do we require Internet standards? What are they?
1.9SUMMARY
Data communications are the transfer of data from one device to
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16A data communications system must transmit data to the correct
destination in an accurate and timely manner.
The five components that make up a data communications system are
the message, sender, receiver, medium, and protocol.
Text, n umbers, images, audio, and video are different forms of
information.
A network can be categorized as a local area network or a wide area
network.
A LAN is a data communication system within a building, plant, or
campus, or between nearby buildings.
AW A Ni s a data communication system spanning states, countries, or
the whole world.
An internet is a network of networks.
The Internet is a collection of many separate networks.
There are local, regional, national, and international Internet service
providers.
Aprotocol is a set of rules that govern data communication; the key
elements of a protocol are syntax, semantics, and timing .
Standards are necessary to ensure that products from different
manufacturers can work together as expected.
1.10 RE FERENCES
1.Data Communication & Networking –Behrouz Forouzan
2.TCP/IPProtocol Suite –Behrouz Forouzan
3.Computer Networks –Andrew Tanenbaum
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NETWORK MODELS
Unit Structure
2.0 Objectives
2.1 Protocol layering
2.2 TCP/IP protocol suite
2.3 The OSO model
2.4 Review questions
2.5 Summary
2.6 References
2.0 OBJECTIVES:
This chapter would make you understand the following concepts
What is Protocol Layering?
Principles of Protocol Layering
What is TCP/IP protocol in brief
Layers in the TCP/IP Protocol Suite
The OSI model.
Comparison of the OSI and TCP/IP Reference Models
Problems of the TCP/IP Reference Model
Problems of the OSI Model and Protocols
2.1 PROTOCOL LAYERING
We have discussed the term protocol in the previous chapter. In
data communication and networking, a protocol defines the rules that both
the sender and receiver and all intermediate devices need to follow to be
able to communicate effectively. When commun ication is simple, we may
need only one simple protocol; when the communication is complex, we
may need to divide the task between different layers, in which case we
need a protocol at each layer, or protocol layering.
To understand the protocol layering let us develop two simple scenarios.
In the first scenario, communication is so simple that it can occur
in only one layer. Assume Seeta and Kaveri are neighbors with a lot of
common ideas. Communication between Seeta and Kaveri takes place in
one layer, f ace to face, in the same language, as shown inmunotes.in

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Figure: A single -layer protocol
Even in this simple scenario, we can see that a set of rules needs to
be followed. First, Seeta and Kaveri know that they should greet each
other when they meet. Second, they know that they should confine their
vocabulary to the level of their friendship. Third, each party knows that
she should refrain from speaking when the other party is speaking. Fourth,
each party knows that the conv ersation should be a dialog, not a monolog:
both should have the opportunity to talk about the issue. Fifth, they should
exchange some nice words when they leave. We can see that the protocol
used by Seeta and Kaveri is different from the communication bet ween a
professor and the students in a lecture hall. The communication in the
second case is mostly monolog; the professor talks most of the time unless
a student has a question, a situation in which the protocol dictates that she
should raise her hand and wait for permission to speak. In this case, the
communication is normally very formal and limited to the subject being
taught.
Second Scenario
In the second scenario, we assume that Kaveri is offered a higher -
level position in her company, but needs to m ove to another branch
located in a city very far from Seeta. The two friends still want to continue
their communication and exchange ideas because they have come up with
an innovative project to start a new business when they both retire. They
decide to co ntinue their conversation using regular mail through the post
office. However, they do not want their ideas to be revealed by other
people if the letters are intercepted. They agree on an
encryption/decryption technique. The sender of the letter encrypts i tt o
make it unreadable by an intruder; the receiver of the letter decrypts it to
get the original letter. We discuss the encryption/decryption methods in
but for the moment we assume that Seeta and Kaveri use one technique
that makes it hard to decrypt th e letter if one does not have the key for
doing so. Now we can say that the communication between Seeta and
Kaveri takes place in three layers, as shown in Figure. We assume that
Seeta and Kaveri each have three machines (or robots) that can perform
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Figure: A three -layer protocol
Principles of Protocol Layering
Let us discuss two principles of protocol layering.
First Principle
The first principle dictates that if we want bidirectional
communication, we need to make each layer so that it is able to perform
two opposite tasks, one in each direction. For example, the third layer task
is to listen (in one direction) and talk (in the other direction). The second
layer needs to be able to encrypt and decrypt. The first layer needs to send
and receive mail.
Second Principle
The second principle that we need to follow in protocol layering is
that the two objects under each layer at bo th sites should be identical. For
example, the object under layer 3 at both sites should be a plaintext letter.
both sites should be a cipher text letter. The object under layer 1 at both
sites should be a piece of mail.
Logical Connections
After following the above two principles, we can think about
logical connection between each layer as shown in below figure. This
means that we have layer -to-layer communication. Seeta and Kaveri can
think that there is a logical (imaginary) connection at each layer through
which they can send the object created from that layer. We will see that
the concept of logical connection will help us better understand the task of
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Figure: Logical connection between peer layers
1.2 TCP/IP PROTOCOL SUITE
Now that we know about the concept of protocol layering and the
logical communication between layers in our second scenario, we can
introduce the TCP/IP (Transmissio n Control Protocol/Internet Protocol).
TCP/IP is a protocol suite (a set of protocols organized in different layers)
used in the Internet today. It is a hierarchical protocol made up of
interactive modules, each of which provides a specific functionality. The
term hierarchical means that each upper level protocol is supported by the
services provided by one or more lower level protocols. The original
TCP/IP protocol suite was defined as four software layers built upon the
hardware. Today, however, TCP/IP is thought of as a five -layer model.
Following figure shows both configurations.
Layered Architecture
To show how the layers in the TCP/IP protocol suite are involved
in communication between two hosts, we assume that we want to use the
suite in a small int ernet made up of three LANs (links), each with a link -
layer switch. We also assume that the links are connected by one router, as
shown in below Figure.munotes.in

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21
Figure: Layers in the TCP/IP protocol suite
Figure: Communication through an internet
Layers in the TCP/IP Protocol Suite
After the above introduction, we briefly discuss the functions and
duties of layers in the TCP/IP protocol suite. Each layer is discussed in
detail in the next five parts of the book. To better understand the duties of
each layer, we need to think about the logical connections between layers.
Below figure shows logical connections in our simple internet.
Using logical connections makes it easier for us to think about the
duty of each layer. As the figure shows, the duty of the application,
transport, and network layers is end -to-end. However, the duty of the data -
link and physical layers is hop -to-hop, in which a hop is a host or router.munotes.in

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22
Figure: Logical connections between layers in TCP/IP
In other words, the domain of duty of the top three layers is the
internet, and the domain of duty of the two lower layers is the link.
Another way of thinking of the logical connections is to think about the
data unit created from each layer. In the top t hree layers, the data unit
(packets) should not be changed by any router or link -layer switch. In the
bottom two layers, the packet created by the host is changed only by the
routers, not by the link -layer switches. Below figure shows the second
principle discussed previously for protocol layering. We show the
identical objects below each layer related to each device.
Figure: Identical objects in the TCP/IP protocol suite
Note that, although the logical connectio n at the network layer is
between the two hosts, we can only say that identical objects exist
between two hops in this case because a router may fragment the packet at
the network layer and send more packets than received. Note that the link
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23Description of Each Layer
After understanding the concept of logical communication, we are
ready to briefly discuss the duty of each layer
Physical Layer
We can say that the physical layer is responsible for carrying
individual bits in a frame across the link. Although the physical layer is
the lowest level in the TCPIIP protocol suite, the communication between
two devices at the physical layer is still a logical communication because
there is another, hidden layer, the transmission media, under the physical
layer. Two devices are connected by a transmission medium (cable or air).
We need to know that the transmission medium does not carry bits; it
carries electrical or optical signals. So the bits received in a frame fro m the
data-link layer are transformed and sent through the transmission media,
but we can think that the logical unit between two physical layers in two
devices is a bit. There are several protocols that transform a bit to a signal.
Data-link Layer
We ha ve seen that an internet is made up of several links (LANs
and WANs) connected by routers. There may be several overlapping sets
of links that a datagram can travel from the host to the destination. The
routers are responsible for choosing the best links. However, when the
next link to travel is determined by the router, the data -link layer is
responsible for taking the datagram and moving it across the link. The link
can be a wired LAN with a link -layer switch, a wireless LAN, a wired
WAN, or a wireless WA N. We can also have different protocols used with
any link type. In each case, the data -link layer is responsible for moving
the packet through the link. TCP/IP does not define any specific protocol
for the data -link layer. It supports all the standard and proprietary
protocols. Any protocol that can take the datagram and carry it through the
link suffices for the network layer. The data -link layer takes a datagram
and encapsulates it in a packet called «frame. Each link -layer protocol
may provide a differe nt service. Some link -layer protocols provide
complete error detection and correction, some provide only error
correction.
Network Layer
The network layer is responsible for creating a connection between
the source computer and the destination computer. T he communication at
the network layer is host -to-host. However, since there can be several
routers from the source to the destination, the routers in the path are
responsible for choosing the best route for each packet. We can say that
the network layer is responsible for host -to-host communication and
routing the packet through possible routes. Again, we may ask ourselves
why we need the network layer. We could have added the routing duty to
the transport layer and dropped this layer. One reason, as we sai d before,
is the separation of different tasks between different layers. The second
reason is that the routers do not need the application and transport layers.munotes.in

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24Transport Layer
The logical connection at the transport layer i s also end -to-end.
The transport layer at the source host gets the message from the
application layer, encapsulates it in a transport layer packet (called a
segment or a user datagram in different protocols) and sends it, through
the logical (imaginary) co nnection, to the transport layer at the destination
host. In other words, the transport layer is responsible for giving services
to the application layer: to get a message from an application program
running on the source host and deliver it to the corresp onding application
program on the destination host. We may ask why we need an end -to-end
transport layer when we already have an end -to-end application layer. The
reason is the separation of tasks and duties, which we discussed earlier.
The transport layer should be independent of the application layer. In
addition, we will see that we have more than one protocol in the transport
layer, which means that each application program can use the protocol that
best matches its requirement.
Application Layer
Thelogical connection between the two application layers is end
to-end. The two application layers exchange messages between each other
as though there were a bridge between the two layers. However, we should
know that the communication is done through all th e layers.
Communication at the application layer is between two processes (two
programs running at this layer). To communicate, a process sends a
request to the other process and receives a response. Process -to-process
communication is the duty of the appl ication layer. The application layer
in the Internet includes many predefined protocols, but a user can also
create a pair of processes to be run at the two hosts.
2.3THE OSI MODEL
Although, when speaking of the Internet, everyone talks about the
TCP/I Pprotocol suite, this suite is not the only suite of protocols defined.
Established in 1947, the International Organization for Standardization
(ISO) is a multinational body dedicated to worldwide agreement on
international standards. Almost three -fourths of the countries in the world
are represented in the ISO. An ISO standard that covers all aspects of
network communications is the Open Systems Interconnection (OSI)
model. It was first introduced in the late 1970s.
ISO is the organization; OSI is t he model
The OSI model is a layered framework for the design of network
systems that allows communication between all types of computer
systems. It consists of seven separate but related layers, each of which
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Figure: The OSI model
The TCP/IP Reference Model:
The TCP/IP reference model was developed prior to OSI model.
The major design goals of this model were,
1. To connect multiple networks together so that they appear as a single
network.
2. To survive after partial subnet hardware failures.
3. To provide a flexible architecture.
Unlike OSI reference model, TCP/IP reference model has only 4 layers.
They are,
1. Host -to-Network Layer
2. Internet Layer
3. Transport Layer
4. Application Layer
Application Layer
Transport Layer
Internet Layer
Host-to-Network Layer
Figure: TCP/IP Reference model
Host-to-Network Layer
The TCP/IP reference model does not really say much about what
happens here, except to point out that the host has to connect to themunotes.in

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26network using some protocol so it can send IP packets to it. This protocol
is not defined and varies from host to host and network to network.
Internet Layer
This layer, called the internet layer, is the linchpin that holds the
whole architecture together. Its job is to permit hosts to inject packets into
any network and have they travel independently to the destination
(pote ntially on a different network). They may even arrive in a different
order than they were sent, in which case it is the job of higher layers to
rearrange them, if in -order delivery is desired. Note that ''internet'' is used
here in a generic sense, even t hough this layer is present in the Internet.
The internet layer defines an official packet format and proto col
called IP (Internet Protocol). The job of the internet layer is to deliver IP
packets where they are supposed to go. Packet routing is clearly the major
issue here, as is avoiding congestion. For these reasons, it is reasonable to
say that the TCP/IP i nternet layer is similar in functionality to the OSI
network layer.
The Transport Layer
The layer above the internet layer in the TCP/IP model is now
usually called the transport lay er. It is designed to allow peer entities on
the source and destination hosts to carry on a conversation, just as in the
OSI transport layer. Two end -to-end transport protocols have been defined
here. The first one, TCP (Transmission Control Protocol), is a reliable
connection -oriented protocol that allows a byte stream originating on one
machine to be delivered without error on any other machine in the
internet. It fragments the incoming byte stream into discrete messages and
passes each one on to the inte rnet layer. At the destination, the receiving
TCP process reassembles the received messages into the output stream.
TCP also handles flow control to make sure a fast sender cannot swamp a
slow receiver with more messages than it can handle.
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27The second protocol in this layer, UDP (User Datagram Protocol),
is an unreliable, connectionless protocol for applications that do not want
TCP's sequencing or flow control and wish to pro vide their own. It is also
widely used for one -shot, client -server -type request reply queries and
applications in which prompt delivery is more important than accurate
delivery, such as transmitting speech or video. The relation of IP, TCP,
and UDP is show n
Figure: Protocol and networks in the TCP/IP model initially
Application layer:
The TCP/IP model does not have session or presentation layers.
On top of the transport layer is the application layer. It contains all the
higher -level protocols. The early ones included virtual terminal
(TELNET), file transfer (FTP), and electronic mail (SMTP), as shown in
the above figure. The virtual terminal protocol allows a user on one
machine to log onto a distant machine and w ork there. The file transfer
protocol provides a way to move data efficiently from one machine to
another. Electronic mail was originally just a kind of file transfer, but later
a specialized protocol (SMTP) was developed for it.
Many other protocols hav e been added to these over the years: the
Domain Name System (DNS) for mapping host names onto their network
addresses, NNTP, the protocol for moving USENET news articles around,
and HTTP, the protocol for fetching pages on the World Wide Web, and
many oth ers.
Comparison of the OSI and TCP/IP Reference Models
The OSI and TCP/IP reference models have much in common.
Both are based on the concept of a stack of independent protocols. Also,
the functionality of the layers is roughly similar. For example, in bo th
models the layers up through and including the transport layer are there to
provide an end -to-end, network -independent transport service to processes
wishing to communicate. These layers form the transport provider. Again
in both models, the layers abov e transport are application -oriented users of
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28Despite these fundamental similarities, the two models also have
many differences
Three concepts are central to the OSI model
1. Services.
2. Interfaces.
3. Protocols.
Probably the biggest contribution of the OSI model is to make the
distinction between these three concepts explicit. Each layer performs
some services for the layer above it. The service definition tells what the
layer does, not how entities above it acces s it or how the layer works. It
defines the layer's semantics.
A layer's interface tells the processes above it how to access it. It
specifies what the parameters are and what results to expect. It, too, says
nothing about how the layer works inside. Fin ally, the peer protocols used
in a layer are the layer's own business. It can use any protocols it wants to,
as long as it gets the job done (i.e., provides the offered services). It can
also change them at will without affecting software in higher layers.
The TCP/IP model did not originally clearly distinguish between
service, interface, and protocol, although people have tried to retrofit it
after the fact to make it more OSI -like. For example, the only real services
offered by the internet layer are SE ND IP PACKET and RECEIVE IP
PACKET.
As a consequence, the protocols in the OSI model are better hidden
than in the TCP/IP model and can be replaced relatively easily as the
technology changes. Being able to make such changes is one of the main
purposes o f having layered protocols in the first place.
The OSI reference model was devised before the corresponding
protocols were invented. This ordering means that the model was not
biased toward one particular set of protocols, a fact that made it quite
gener al. The downside of this ordering is that the designers did not have
much experience with the subject and did not have a good idea of which
functionality to put in which layer.
Another difference is in the area of connectionless versus
connection -oriented communication. The OSI model supports both
connectionless and connection -oriented communication in the network
layer, but only connection -oriented communication in the transport layer,
where it counts (because the transport service is visible to the users). The
TCP/IP model has only one mode in the network layer (connectionless)
but supports both modes in the transport layer, giving the users a choice.
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29Problems of the TCP/IP Reference Model
First, the model does not clearly distinguish the concepts of
service, interface, and protocol. Good software engineering practice
requires differentiating between the specification and the implementation,
something that OSI does very carefully, and TCP/IP does not.
Consequently, the TCP/IP model is not much of a guide for designing new
networks using new technologies.
Second, the TCP/IP model is not at all general and is poorly suited
to describing any protocol stack other than TCP/IP. Trying to use the
TCP/IP model to describe Bluetooth, for example, is completely
impossible.
Third, the host -to-network layer is not really a layer at all in the
normal sense of the te rm as used in the context of layered protocols. It is
an interface (between the network and data link layers). The distinction
between an interface and a layer is crucial, and one should not be sloppy
about it.
Fourth, the TCP/IP model does not distingui sh (or even mention)
the physical and data link layers. These are completely different. The
physical layer has to do with the transmission characteristics of copper
wire, fiber optics, and wireless communication. The data link layer's job is
to delimit the start and end of frames and get them from one side to the
other with the desired degree of reliability. A proper model should include
both as separate layers. The TCP/IP model does not do this. Finally,
although the IP and TCP protocols were carefully tho ught out and well
implemented, many of the other protocols were ad hoc, generally
produced by a couple of graduate students hacking away until they got
tired. The protocol implementations were then distributed free, which
resulted in their becoming widely used, deeply entrenched, and thus hard
to replace. Some of them are a bit of an embarrassment now. The virtual
terminal protocol, TELNET, for example, was designed for a ten -character
per second mechanical Teletype terminal. It knows nothing of graphical
user interfaces and mice. Nevertheless, 25 years later, it is still in
widespread use.
Problems of the OSI Model and Protocols:
Bad timing
Bad technology
Bad implementations
Bad politics
Bad Timing:
The time at which a standard is established is absolutely critical to
its success. David Clark of M.I.T. has a theory of standards that he calls
the apocalypse of the two elephants, which is illustrated in Fig. This figure
shows the amount of activity surr ounding a new subject. When the subject
is first discovered, there is a burst of research activity in the form ofmunotes.in

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30discussions, papers, and meetings. After a while this activity subsides,
corporations discover the subject, and the billion -dollar wave of
investment hits. It is essential that the standards be written in the trough in
between the two ''elephants.'' If the standards are written too early, before
the research is finished, the subject may still be poorly understood; the
result is bad standards. If they are written too late, so many companies
may have already made major investments in different ways of doing
things that the standards are effectively ignored. If the interval between the
two elephants is very short (because everyone is in a hurry to g et started),
the people developing the standards may get crushed.
Figure: The apocalypse of the two elephants
Bad Implementations:
Given the enormous complexity of the model and the protocols, it
will come as no surprise that the initial implementations were huge,
unwieldy, and slow. Everyone who tried them got burned. It did not take
long for people to associate ''OSI'' with ''poor quality.'' Although the
products improved in the course of time, the image stuck.
Bad Politics:
On account of the initial implementation, many people, especially
in academia, thought of TCP/IP as part of UNIX, and UNIX in the 1980s
in academia was not unlike parenthood (then incorrectly called
motherhood) and apple pie. OSI, on the o ther hand, was widely thought to
be the creature of the European telecommunication ministries, the
European Community, and later the U.S. Government. This belief was
only partly true, but the very idea of a bunch of government bureaucrats
trying to shove a technically inferior standard down the throats of the poor
researchers and programmers down in the trenches actually developing
computer networks did not help much. Some people viewed this
development in the same light as IBM announcing in the 1960s that PL/I
was the language of the future, or DoD correcting this later by announcing
that it was actually Ada.
Bad Technology:
The second reason that OSI never caught on is that both the model
and the protocols are flawed. The choice of seven layers was more
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31are nearly empty, whereas two other ones (data link and network) are
overfull.
The OSI model, along with the associated service definitions and
protocols, is extraordinarily compl ex. When piled up, the printed standards
occupy a significant fraction of a meter of paper. They are also difficult to
implement and inefficient in operation. In addition to being
incomprehensible, another problem with OSI is that some functions, such
as addressing, flow control, and error control, reappear again and again in
each layer.
2.4 REVIEW QUESTIONS
1. What is prototype layering?
2. What are the principles of protocol layering?
3. Explain the TCP/IP Protocol in brief.
4. Explain the OSI Model in brief.
5. Differentiate between the OSI and TCP/IP referential model
2.5SUMMARY
The International Standards Organization created a model called the
Open Systems Interconnection, which allows diverse systems to
communicate.
Theseven -layer OSI model provides guidelines for the development of
universally compatible networking protocols.
The physical, data link, and network layers are the network support
layers.
The session, presentation, and application layers are the user support
layers.
The transport layer links the network support layers and the user
support layers.
The physical layer coordinates the functions required to transmit a bit
stream overa physical medium.
The data link layer is responsible for delivering data units fr om one
station to the next without errors.
The network layer is responsible for the source -to-destination delivery
of a packet across multiple network links.
The transport layer is responsible for the process -to-process delivery
of the entire message.
The session layer establishes, maintains, and synchronizes the
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32The presentation layer ensures interoperability between
communicating devices through transformation of data into a mutually
agreed upon format.
Theapplication layer enables the users to access the network.
TCP/IP is a five -layer hierarchical protocol suite developed before the
OSI model.
The TCP/IP application layer is equivalent to the combined session,
presentation, and application layers of the OS I model.
Four levels of addresses are used in an internet following the TCP/IP
protocols: physical(link) addresses, logical (IP) addresses, port
addresses, and specific addresses.
The physical address, also known as the link address, is the address of
a node as defined by its LAN or WAN.
The IP address uniquely defines a host on the Internet.
The port address identifies a process on a host.
A specific address is a user -friendly address.
2.6 REFERENCES
1. Data Communication & Networking –Behrouz Forouzan
2. TCP/IP Protocol Suite –Behrouz Forouzan
3. Computer Networks –Andrew Tanenbaum1

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333
INTRODUCTION TO PHYSICAL LAYER
Unit Structure
3.0 Objectives
3.1Data and signals
3.2Periodic analog signals
3.3 Digital Signal
3.4Transmission Impairment
3.5 Data rate limits
3.6 Performance
3.7 Review questions
3.8 Summary
3.9 References
3.0OBJECTIVES:
This chapter would make you understand the following concepts
What is data and single?
What is analog and digital data?
Concept of periodic and non -periodic single
Concept of Sine wave, Wavelength, Bandwidth
Digital Signals, bit rates, bit length
Concep t of transmission impairment
Different causes of impairment
Data rate limits
Measuring the performance, Throughput, Latency and Queuing Time
3.1DATA AND SIGNALS
One of the major functions of the physical layer is to move data in
the form of electromagn etic signals across a transmission medium.
Whether you are collecting numerical statistics from another computer,
sending animated pictures from a design workstation, or causing a bell to
ring at a distant control center, you are working with the transmiss ion of
data across network connections .
Generally, the data usable to a person or application are not in a
form that can be transmitted over a network. For example, a photograph
must first be changed to a form that tra nsmission media can accept.
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34Analog and Digital Data
Data can be analog or digital. The term analog data refers to
information that is continuous; digital data refers to information that has
discrete states. For example, an analog clock that has hour, minute, and
second hands gives information in a continuous form; the movements of
the hands are continuous. On the other hand, a digital clock that re ports
the hours and the minutes will change suddenly from 9:02 to 9:03.
Analog data, such as the sounds made by a human voice, take on
continuous values. When someone speaks, an analog wave is created in
the air. This can be captured by a microphone and c onverted to an analog
signal or sampled and converted to a digital signal.
Digital data take on discrete values. For example, data are stored in
computer memory in the form of Os and 1s. They can be converted to a
digital signal or modulated into an analo g signal for transmission across a
medium.
Analog and Digital Signals
Like the data they represent, signals can be either analog or digital.
An analog signal has infinitely many levels of intensity over a period of
time. As the wave moves from value A to value B, it passes through and
includes an infinite number of values along its path. A digital signal, on
the other hand, can have only a limited number of defined values.
Although each value can be any number, it is often as simple as 1 and O.
The simple st way to show signals is by plotting them on a pair of
perpendicular axes. The vertical axis represents the value or strength of a
signal. The horizontal axis represents time. Figure 3.1 illustrates an analog
signal and a digital signal. The curve represe nting the analog signal passes
through an infinite number of points. The vertical lines of the digital
signal, however, demonstrate the sudden jump that the signal makes from
value to value.
Figure: Comparison of analog and digital signals
Periodic and Non -periodic Signals
Both analog and digital signals can take one of two forms: periodic
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35Greek means "non"). A periodic signal completes a pattern within a
measurable time frame, called a period, and repeats that pattern over
subsequent identical periods. The completion of one full pattern is called a
cycle. A non -periodic signal changes without exhibiting a pattern or cycle
thatrepeats over time. Both analog and digital signals can be periodic or
non-periodic. In data communications, we commonly use periodic analog
signals and non -periodic digital signals
3.2PERIODIC ANALOG SIGNALS
Periodic analog signals can be classified as simple or composite. A
simple periodic analog signal, a sine wave, cannot be decomposed into
simpler signals. A composite periodic analog signal is composed of
multiple sine waves.
Sine Wave
The sine wave is the most fundamental form of a periodic analog
signal. When we visualize it as a simple oscillating curve, its change over
the course of a cycle is smooth and consistent, a continuous, rolling flow.
Following Figure shows a sine wave. Each cycle consists of a single arc
above the time axis followed by a single arc below it
Figure: A sine wave
A sine wave can be represented by three parameters: the peak
amplitude, thefrequency, and the phase. These three parameters fully
describe a sine wave.
Peak Amplitude
The peak amplitude of a signal is the absolute value of its highest
intensity, proportional to the energy it carries. For electric signals, peak
amplitude is normally measured in volts.
Period and Frequency
Period refers to the amount of time, in seconds, a signal needs to
complete 1 cycle. Frequency refers to the number of periods in I s. Note
that period and frequency are just one characteristic defined in two ways.
Period is the inverse of frequency, and frequency is the inverse of period,
as the following formulas show.
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36Frequency and period are the inverse of each other.
Figure: Two signals with the same amplitude and phase, but different
frequencies
Period is formally expressed in seconds. Frequency is formally
expressed in Hertz (Hz), which is cycle per second. Units of period and
frequency are shown in Table
Table: Units of period and frequency
Phase
The term phase describes the position of the waveform relative to
time O. If we think of the wave as something that can be shifted backward
or forward along the time axis, phase describes the amount of that shift. It
indicates the status of the first cycle.
Phase is measured in degrees or radians [360° is 2n rad; 1° is
2n/360 rad, and 1 rad is 360/(2n)]. A phase shift of 360° corresponds to a
shift of a complete period; a phase shift of 180° c orresponds to a shift of
one-half of a period; and a phase shift of 90° corresponds to a shift of one -
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37
Figure: Three sine waves with the same amplitude and frequency, but
different phases
By the looking at above Figure, we can say that
A sine wave with a phase of 0° starts at time 0 with a zero amplitude.
The amplitude is increasing.
A sine wave with a phase of 90° starts at time 0 with a peak
amplitude. The amplitude is decreasing.
A sine wave with a phase of 180° starts at time 0 with a zero
amplitude. The amplitude is decreasing
Another way to look at the phase is in terms of shift or offset. We
can say that
A sine wave with a phase of 0° is not shifted.
A sine wave with a phase o f 90° is shifted to the left by 1/4 cycle.
However, note that the signal does not really exist before time O.
A sine wave with a phase of 180° is shifted to the left by 1/2 cycle.
However, note that the signal does not really exist before time O.
Waveleng th
Wavelength is another characteristic of a signal traveling through a
transmission medium. Wavelength binds the period or the frequency of a
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38
Figure: Wavel ength and period
While the frequency of a signal is independent of the medium, the
wavelength depends on both the frequency and the medium. Wavelength
is a property of any type of signal. In data communications, we often use
wavelength to describe the transmission of light in an optical fiber. The
wavelength is the distance a simple signal can travel in one period.
Wavelength can be calculated if one is given the propagation speed
(the speed of light) and the period of the signal. However, since period and
frequency are related to each other, if we represent wavelength by A,
propagation speed by ƛ(speed of light), and frequenc yb y 1,we get
The propagation speed of electromagnetic signals depends on the
medium and on the frequency of the signal. For example, in a vacuum,
light is propagated with a speed of 3 x 108mls. That speed is lo wer in air
and even lower in cable. The wavelength is normally measured in micro -
meters (microns) instead of meters.
Time and Frequency Domains
A sine wave is comprehensively defined by its amplitude,
frequency, and phase. We have been showing a sine wave by using what is
called a time -domain plot. The time -domain plot shows changes in signal
amplitude with respect to time (it is an amplitude -versus -time plot). Phase
is not explicitly shown on a time -domain plot.
To show the relationship between amplitude and frequency, we can
use what is called a frequency -domain plot. A frequency -domain plot is
concerned with only the peak value and the frequency. Changes of
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Figure: Th et i m e -domain and frequency -domain plots of a sine wave
It is obvious that the frequency domain is easy to plot and conveys
the information that one can find in a time domain plot. The advantage of
the frequency doma in is that we can immediately see the values of the
frequency and peak amplitude. A complete sine wave is represented by
one spike. The position of the spike shows the frequency; its height shows
the peak amplitude.
Composite Signals
So far, we have focus ed on simple sine waves. Simple sine waves
have many applications in daily life. We can send a single sine wave to
carry electric energy from one place to another. For example, the power
company sends a single sine wave with a frequency of 60 Hz to distrib ute
electric energy to houses and businesses. As another example, we can use
a single sine wave to send an alarm to a security center when a burglar
opens a door or window in the house. In the first case, the sine wave is
carrying energy; in the second, th e sine wave is a signal of danger.
Bandwidth
The range of frequencies contained in a composite signal is its
bandwidth. The bandwidth is normally a difference between two numbers.
For example, if a composite signal contains frequencie s between 1000 and
5000, its bandwidth is 5000 -1000, or 4000.
Following figure shows the concept of bandwidth. The figure
depicts two composite signals, one periodic and the other non -periodic.
The bandwidth of the periodic signal contains all integer f requencies
between 1000 and 5000 (1000, 100 I, 1002, ...). The bandwidth of the non -
periodic signals has the same range, but the frequencies are continuous.munotes.in

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Figure: The bandwidth of periodic and non -periodic comp osite
signals
3.3DIGITAL SIGNALS
In addition to being represented by an analog signal, information
can also be represented by a digital signal. For example, a 1 can be
encoded as a positive voltage and a 0 as zero voltage. A digital signal can
have more than two levels. In this case, we can send more than 1 bit for
each level. Following figure shows two signals, on e with two levels and
the other with four.munotes.in

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Figure: Two digital signals: one with two signal levels and the other
with four signal levels
We send 1 bit per level in part of the figure and 2 bits per level in
part b of the figure. In general, if a signal has Llevels, each level needs
log2L bits.
Bit Rate
Most digital signals are non -periodic, and thus period and
frequency are not appropriat e characteristics. Another term -bit rate
(instead of frequency) -isused to describe digital signals. The bit rate is the
number of bits sent in Is, expressed in bits per second (bps). Above figure
shows the bit rate for two signals.
Bit Length
We discusse d the concept of the wavelength for an analog signal:
the distance one cycle occupies on the transmission medium. We can
define something similar for a digital signal: the bit length. The bit length
is the distance one bit occupies on the transmission medi um.
Bit length = propagation speed x bit duration
Digital Signal as a Composite Analog Signal
Based on Fourier analysis, a digital signal is a composite analog
signal. The bandwidth is infinite, as you may have guessed. We can
intuitively corne up with t his concept when we consider a digital signal. A
digital signal, in the time domain, comprises connected vertical and
horizontal line segments. A vertical line in the time domain means amunotes.in

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42frequency of infinity (sudden change in time); a horizontal line in t he time
domain means a frequency of zero (no change in time). Going from a
frequency of zero to a frequency of infinity (and vice versa) implies all
frequencies in between are part of the domain.
Fourier analysis can be used to decompose a digital signal . If the
digital signal is periodic, which is rare in data communications, the
decomposed signal has a frequency domain representation with an infinite
bandwidth and discrete frequencies. If the digital signal is non -periodic,
the decomposed signal still h as an infinite bandwidth, but the frequencies
are continuous. Following figure shows a periodic and a non -periodic
digital signal and their bandwidths.
Figure: The time and frequency domains of periodic and non -periodic
digital signals
Note that both bandwidths are infinite, but the periodic signal has
discrete frequencies while the non -periodic signal has continuous
frequencies.
3.4 TRANSMISSION IMPAIRMENT
Signals travel through transmission media, which are not perfect.
The imperfection causes signal impairment. This means that the signal at
the beginning of the medium is not the same as the signal at the end of the
medium. What is sent is not what is recei ved. Three causes of impairment
are attenuation, distortion, and noise.
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43Figure: Causes of impairment
Attenuation
Attenuation means a loss of energy. When a signal, simple or
composite, travels through a medium, it loses some of its energy in
overcoming the resistance of the medium. That is why a wire carrying
electric signals gets warm, if not hot, after a while. Som e of the electrical
energy in the signal is converted to heat. To compensate for this loss,
amplifiers are used to amplify the signal. Figure 3.26 shows the effect of
attenuation and amplification.
Figure: Attenu ation
Decibel
To show that a signal has lost or gained strength, engineers use the
unit of the decibel. The decibel (dB) measures the relative strengths of two
signals or one signal at two different points. Note that the decibel is
negative if a signal is attenuated and positive if a signal is amplified.
Variables PIandP2are the powers of a signal at points 1 and 2,
respectively. Note that some engineering books define the decibel in terms
of voltage instea d of power. In this case, because power is proportional to
the square of the voltage, the formula is dB = 20 log 10 (V2IV1). In this
text, we express dB in terms of power.
Distortion
Distortion means that the signal changes its form or shape.
Distortion c an occur in a composite signal made of different frequencies.
Each signal component has its own propagation speed through a medium
and, therefore, its own delay in arriving at the final destination.
Differences in delay may create a difference in phase if the delay is not
exactly the same as the period duration. In other words, signal components
at the receiver have phases different from what they had at the sender. The
shape of the composite signal is therefore not the same. Following figure
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Figure: Distortion
Noise
Noise is another cause of impairment. Several types of noise, such
as thermal noise, induced noise, crosstalk, and impulse noise, may corrupt
the signal. Thermal noise is the random motion of electrons in a wire
which creates an extra signal not originally sent by the transmitter.
Induced noise comes from sources such as motors and appliances. These
devices act as a sending antenna, and the transmission medium acts as the
receiving antenna. Crosstalk is the effect of one wire on the other. One
wire acts as a sending antenna and the other as the receiving antenna.
Impulse noise is a spike (a signal with high energy in a very short time)
that comes from power lines, lightning, and so on. Following figure shows
the effect of noise on a signal.
Figure: Noise
Signal -to-Noise Ratio (SNR)
As we will see later, to find the theoretical bit rate limit, we need to
know the ratio of the signal power to the noise power. The signal -to-noise
ratio is defined as
SNR= averag e signal power/average noise power
We need to consider the average signal power and the average
noise power because these may change with time. Following figure shows
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Figure: Two cases of SNR: a h igh SNR and a low SNR
SNR is actually the ratio of what is wanted (signal) to what is not wanted
(noise). A high SNR means the signal is less corrupted by noise; a low
SNR means the signal is more corrupted by noise. Because SNR is the
ratio of two powers , it is often described in decibel units, SNRdB, defined
as
3.5 DATA RATE LIMITS
A very important consideration in data communications is how fast
we can send data, in bits per second over a channel. Data rate depends on
three factors:
1. The bandwidth available
2. The level of the signals we use
3. The quality of the channel (the level of noise)
Two theoretical formulas were developed to calculate the data rate:
one by Nyquist for a noiseless channels another by Shannon for a noisy
channel.
Noiseless Channel: Nyquist Bit Rate
For a noiseless channel, the Nyquist bit rate formula defines the theoretical
maximum bit rate
BitRate = 2 x bandwidth x 10g2 L
In this formula, bandwidth is the bandwidth of the channel, Lis the
number of signal levels used to represent data, and BitRate is the bit rate
in bits per second.
According to the formula, we might think that, given a specific
bandwidth, we can have any bit rate we want by increasing the number of
signa11eve1s. Although the idea is theoretically correct, practically there
is a limit. When we increase the number of sig nal levels, we impose amunotes.in

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46burden on the receiver. If the number of levels in a signal is just 2, the
receiver can easily distinguish between a 0 and a 1. If the level of a signal
is 64, the receiver must be very sophisticated to distinguish between 64
differ ent levels. In other words, increasing the levels of a signal reduces
the reliability of the system.
Noisy Channel: Shannon Capacity
In reality, we cannot have a noiseless channel; the channel is
always noisy. In 1944, Claude Shannon introduced a formula, called the
Shannon capacity, to determine the theoretical highest data rate for a noisy
channel:
Capacity =bandwidth X log2 (1 +SNR)
In this formula, bandwidth is the bandwidth of the channel, SNR is
the signal -to-noise ratio, and capacity is the capacit y of the channel in bits
per second. Note that in the Shannon formula there is no indication of the
signal level, which means that no matter how many levels we have, we
cannot achieve a data rate higher than the capacity of the channel. In other
words, the formula defines a characteristic of the channel, not the method
of transmission.
3.6 PERFORMANCE
Up to now, we have discussed the tools of transmitting data
(signals) over a network and how the data behave. One important issue in
networking is the performance of the network -how good is it? We discuss
quality of service, an overall measurement of network performance.
Bandwidth
One characteristic that measures network performance is
bandwidth. However, the term can be used in two different contexts w ith
two different measuring values: bandwidth in hertz and bandwidth in bits
per second.
Bandwidth in Hertz
We have discussed this concept. Bandwidth in hertz is the range of
frequencies contained in a composite signal or the range of frequencies a
channe l can pass. For example, we can say the bandwidth of a subscriber
telephone line is 4 kHz.
Bandwidth in Bits per Seconds
The term bandwidth can also refer to the number of bits per second
that a channel, a link, or even a network can transmit. For example, one
can say the bandwidth of a Fast Ethernet network (or the links in this
network) is a maximum of 100 Mbps. This means that this network can
send 100 Mbps
Relationshipmunotes.in

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47There is an explicit relationship between the bandwidth in hertz
and bandwid th in bits per seconds. Basically, an increase in bandwidth in
hertz means an increase in bandwidth in bits per second. The relationship
depends on whether we have base band transmission or transmission with
modulation.
In networking, we use the term band width in two contexts.
The first, bandwidth in hertz, refers to the range of frequencies in a
composite signal or the range of frequencies that a channel can pass.
The second, bandwidth in bits per second, refers to the speed of bit
transmission in a chan nel or link.
Throughput
The throughput is a measure of how fast we can actually send data
through a network. Although, at first glance, bandwidth in bits per second
and throughput seem the same, they are different. A link may have a
bandwidth of Bbps, but we can only send Tbps through this link with T
always less than B.In other words, the bandwidth is a potential
measurement of a link; the throughput is an actual measurement of how
fast we can send data. For example, we may have a link with a ba ndwidth
of 1 Mbps, but the devices connected to the end of the link may handle
only 200 kbps. This means that we cannot send more than 200 kbps
through this link.
Imagine a highway designed to transmit 1000 cars per minute from
one point to another. Howev er, if there is congestion on the road, this
figure may be reduced to 100 cars per minute. The bandwidth is 1000 cars
per minute; the throughput is 100 cars per minute.
Latency (Delay)
The latency or delay defines how long it takes for an entire
message t o completely arrive at the destination from the time the first bit
is sent out from the source. We can say that latency is made of four
components: propagation time, transmission time, queuing time and
processing delay.
Latency =propagation time +transmiss ion time +queuing time +
processing delay
Propagation Time
Propagation time measures the time required for a bit to travel
from the source to the destination. The propagation time is calculated by
dividing the distance by the propagation speed.
Propagatio n time= Distance/Propagation speed
The propagation speed of electromagnetic signals depends on the
medium and on the frequency of the signal. For example, in a vacuum,
light is propagated with a speed of 3 x 108mfs. It is lower in air; it is much
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48Transmission time
In data communications we don't send just 1 bit, we send a
message. The first bit may take a time equal to the propagation time to
reach its destination; the last bit also may take the same amount of time.
However, there is a tim e between the first bit leaving the sender and the
last bit arriving at the receiver. The first bit leaves earlier and arrives
earlier; the last bit leaves later and arrives later. The time required for
transmission of a message depends on the size of the message and the
bandwidth of the channel.
Transmission time= Message size/Bandwidth
Queuing Time
The third component in latency is the queuing time, the time
needed for each intermediate or end device to hold the message before it
can be processed. The queuing time is not a fixed factor; it changes with
the load imposed on the network. When there is heavy traffic on the
network, the queuing time increases. An intermediate device, such as a
router, queues the arrived messages and processes them one by one. If
there are many messages, each message will have to wait.
3.7REVIEW QUESTIONS
1. Differentiate between analog and digital signal
2. Explain periodic analog signals in brief
3. Explain the terms
a) Wavelength
b) Sine Wave
c)Bandwidth
4. Explain Digital Signals with suitable
5. What are the causes of impairment transmission?
6. Explain data rate limits with different factors
7. How performance is measured? Explain with suitable example
3.8 SUMMARY
Data must be transformed to electromagnetic signals to be transmitted.
Data can be analog or digital. Analog data are continuous and take
continuous values. Digital data have discrete states and take discrete
values.
Signals can be analog or digital. Analog signals can have an inf inite
number of values in a range; digital, signals can have only a limited
number of values.
In data communications, we commonly use periodic analog signals
and non -periodic digital signals.
Frequency and period are the inverse of each other.
Frequency is the rate of change with respect to time.munotes.in

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49Phase describes the position of the waveform relative to time O.
A complete sine wave in the time domain can be represented by one
single spike in the frequency domain.
A single -frequency sine wave is not useful in data communications;
we need to send a composite signal, a signal made of many simple sine
waves.
According to Fourier analysis, any composite signal is a combination
of simple sine waves with different frequencies, amplitudes, and
phases.
The bandwidth o f a composite signal is the difference between the
highest and the lowest frequencies contained in that signal.
A digital signal is a composite analog signal with an infinite
bandwidth.
Baseband transmission of a digital signal that preserves the shape of
the digital signal is possible only if we have a low -pass channel with
an infinite or very wide bandwidth.
If the available channel is a bandpass channel, we cannot send a digital
signal directly to the channel; we need to convert the digital signal to
ananalog signal before transmission.
For a noiseless channel, the Nyquist bit rate formula defines the
theoretical maximum bit rate. For a noisy channel, we need to use the
Shannon capacity to find the maximum bit rate.
Attenuation, distortion, and noise can impair a signal.
Attenuation is the loss of a signal's energy due to the resistance of the
medium.
Distortion is the alteration of a signal due to the differing propagation
speeds of each of the frequencies that make up a signal.
Noise is the external ene rgy that corrupts a signal.
The bandwidth -delay product defines the number of bits that can fill
the link.
3.9 REFERENCES
1. Data Communication & Networking –Behrouz Forouzan
2. TCP/IP Protocol Suite –Behrouz Forouzan
3. Computer Networks –Andrew Tanenbaum1
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DIGITAL AND ANALOG TRANSMISSION
Unit Structure
4.0 Objectives
4.1 Digital -to-digital conversion
4.2Analog -to-digital conversion
4.3Transmission modes
4.4Digital -to-analog conversion
4.5 An alog-to-analog conversion
4.6 Summary
4.7 References
4.0 OBJECTIVES:
This chapter would make you understand the following concepts
What is the need to conversion of data in different form?
How the data is converted from one form to another form
What ar e the different ways to convert it?
What are the different transmission modes
What are the ways of digital to analog conversion?
And conversion of analog to analog.
4.1 DIGITAL -TO-DIGITAL CONVERSION
As we have discussed in the previous topics data can be either
digital or analog. We also saw that signals that represent data can also be
digital or analog. In this section, we see how we can represent digital data
by using digital signals.
The conversi on involves three techniques: line coding ,block
coding , and scrambling . Line coding is always needed block coding and
scrambling mayor may not be needed.
4.1.1 Line Coding
Line coding is the process of converting digital data to digital
signals. We assume that data, in the form of text, numbers, graphical
images, audio, or video, are stored in computer memory as sequences of
bits Line coding converts a sequence of bits to a digital signal. At the
sender, digital data are encoded into a digital signal; at the receiver, the
digital data are recreated by decoding the digital signal. Following figure
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Figure: Line coding and decoding
Characteristics
Before discussing different line coding schemes, we address their
common characteristics Signal Element Versus Data Element Let us
distinguish between a data element and a signal element. In data
communications, our goal is to send data elements. A data element is the
smallest entity that can represent a piece of information: this is the bit. In
digital data communications, a signal element carries data elements. A
signal element is the shortest un it (time wise) of a digital signal. In other
words, data elements are what we need to send; signal elements are what
we can send. Data elements are being carried; signal elements are the
carriers.
We define a ratio rwhich is the number of data elements c arried
by each signal element. Following figure shows several situations with
different values of r.
Figure: Signal element versus data element
In part a of the figure, one data element is carried by one signal
element (r= 1). In part b of the figure, we need two signal elements (two
transitions) to carry each data element (r=½).We will se e later that the
extra signal element is needed to guarantee synchronization. In part c ofmunotes.in

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52the figure, a signal element carries two data elements (r=2). Finally, in
part d, a group of 4 bits is being carried by a group of three signal
elements (r=4/3).For every line coding scheme we discuss, we will give
the value of r.
An analogy may help here. Suppose each data element IS a person
who needs to be carried from one place to another. We can think of a
signal element as a vehicle that can carry people . When r=1, it means
each person is driving a vehicle. When r>1, it means more than one
person is travelling in a vehicle (a carpool, for example). We can also have
the case where one person is driving a car and a trailer (r=½ ) .
Data Rate Versus Sig nal Rate The data rate defines the number of
data elements (bits) sent in Is. The unit is bits per second (bps). The signal
rate is the number of signal elements sent in Is. The unit is the baud. There
are several common terminologies used in the literatur e. The data rate is
sometimes called the bit rate; the signal rate is sometimes called the pulse
rate, the modulation rate, or the baud rate.
One goal in data communications is to increase the data rate while
decreasing the signal rate. Increasing the da ta rate increases the speed of
transmission; decreasing the signal rate decreases the bandwidth
requirement. In our vehicle -people analogy, we need to carry more people
in fewer vehicles to prevent traffic jams. We have a limited bandwidth in
our transport ation system.
We now need to consider the relationship between data rate and
signal rate (bit rate and baud rate). This relationship, of course, depends on
the value of r.It also depends on the data pattern. If we have a data pattern
of all 1s or all Os, the signal rate may be different from a data pattern of
alternating Os and Is. To derive a formula for the relationship, we need to
define three cases: the worst, best, and average. The worst case is when we
need the maximum signal rate; the best case is when we need the
minimum. In data communications, we are usually interested in the
average case. We can formulate the relationship between data rate and
signal rate as
where Nis the data rate (bps); c is the case factor, which varies for each
case; S is the number of signal elements; and ris the previously defined
factor.
Bandwidth
The digital signal that carries information is non periodic. We also
showed that the bandwidth of a non periodic signa l is continuous with an
infinite range. However, most digital signals we encounter in real life have
a bandwidth with finite values. In other words, the bandwidth ismunotes.in

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53theoretically infinite, but many of the components have such small
amplitude that they can be ignored. The effective bandwidth is finite.
Baseline Wandering
In decoding a digital signal, the receiver calculates a running
average of the received signal power. This average is called the baseline.
The incoming signal power is evaluated against this baseline to determine
the value of the data element. A long string of Os or 1s can cause a drift in
the baseline (baseline wandering) and make it difficult for the receiver to
decode correctly. A good lin e coding scheme needs to prevent baseline
wandering.
DC Components
When the voltage level in a digital signal is constant for a while,
the spectrum creates very low frequencies (results of Fourier analysis).
These frequencies around zero, called DC (direc t-current) components,
present problems for a system that cannot pass low frequencies or a
system that uses electrical coupling (via a transformer). For example, a
telephone line cannot pass frequencies below 200 Hz. Also a long -distance
link may use one o r more transformers to isolate different parts of the line
electrically. For these systems, we need a scheme with no DC component.
Self-synchronization
To correctly interpret the signals received from the sender, the
receiver's bit intervals must corresp ond exactly to the sender's bit intervals.
If the receiver clock is faster or slower, the bit intervals are not matched
and the receiver might misinterpret the signals. Following figure shows a
situation in which the receiver has a shorter bit duration. Th e sender sends
10110001, while the receiver receives 110111000011.
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54Built -in Error Detection
It is desirable to have a built -in error -detecting capability in the
generat ed code to detect some of or all the errors that occurred during
transmission. Some encoding schemes that we will discuss have this
capability to some extent.
Immunity to Noise and Interference
Another desirable co de characteristic is a code that is immune to
noise and other interferences. Some encoding schemes that we will discuss
have this capability.
Complexity
A complex scheme is more costly to implement than a simple one.
For example, a scheme that uses four signal levels is more difficult to
interpret than one that uses only two levels.
4.1.2 Line Coding Schemes
We can roughly divide line coding schemes into five broad
categories, as shown in following figure.
Figur e:Line coding schemes
There are several schemes in each category as
In a unipolar scheme, all the signal levels are on one side of the
time axis, either above or below.
NRZ (Non -Return -to-Zero) traditionally, a unipolar scheme was
designed as a non -return-to-zero (NRZ) scheme in which the positive
voltage defines bit I and the zero voltage defines bit O. Itis called NRZ
because the signal does not return to zero at the middle of the bit.
Following figure show a unipolar NRZ scheme.
Figure : Unipolar NRZ schememunotes.in

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55Compared with its polar counterpart (see the next section), this
scheme is very costly. As we will see shortly, the normalized power
(power needed to send 1 bit per unit line resistance) is double that f or
polar NRZ. For this reason, this scheme is normally not used in data
communications today.
Polar Schemes
In polar schemes, the voltages are on the both sides of the time
axis. For example, the voltage level for 0 can be positive and the voltage
level for I can be negative.
Non-Return -to-Zero (NRZ) In polar NRZ encoding, we use two
levels of voltage amplitude . We can have two versions of polar NRZ:
NRZ -Land NRZ -I, as shown in following figure. The figure also shows
the value of r,the average baud rate, and the bandwidth. In the first
variation, NRZ -L( N R Z -Level), the level of the voltage determines the
value of the bit. In the second variation, NRZ -I( N R Z -Invert), the change
or lack of change in the level of the voltage determines the value of the bit.
Ifthere is no change, the bit is 0; if there is a change, the bit is 1.
Figure: Polar NRZ -La n dN R Z -I schemes
InNRZ -L the level of the voltage determines the value of the bit.
InNRZ -I the inversion or the lack of inversion determines the value of the
bit.
When we compare these two schemes based on the criteria
Although baseline wandering is a problem for both variations, it is
twice as severe in NRZ -L. If there is a long sequence of 0s or 1s in
NRZ -L, the average signal power becomes skewed. The receiver
might have difficulty discerning the bit value. In NRZ -It his problem
occurs only for a long sequence of as, If somehow we can eliminate
the long sequence of as, we can avoid baseline wandering.
Another problem with NRZ -L occurs when there is a sudden change of
polarity in the system. NRZ -I does not have this prob lem.
Note Both schemes have an average signal rate of N/2 Bd, NRZ -L and
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56Return to Zero (RZ)
The main problem with NRZ encoding occurs when the sender and
receiver clocks are not synchronized. The receiver does not know when
one bit has ended and the next bit is starting. One solution is the return -to-
zero (RZ) scheme, which uses three values: positive, negative, and zero. In
RZ, the signal changes not between bits but during the bit. In following
figure we see that the signal goes to 0 in the middle of each bit. Itremains
there until the beginning of the next bit. The main disadvantage of RZ
encoding is that it requires two signal changes to encode a bit and
therefore occupies greater bandwidth. The same problem we mentioned, a
sudden change of polarity resulting in all as interpreted as 1s and all 1s
interpreted as as, still exist here, but there is no DC component problem.
Another problem is the complexity: RZ uses three levels of voltage, which
is more complex to create and discern. As a result of all these deficiencies,
the scheme is not used today. Instead, it has been replaced by the better -
performing Manchester and differential Manchester schemes
Figure: Polar RZ
scheme
Biphase (Manchester and Differential Manchester)
The idea of RZ (transition at the middle of the bit) and the idea of
NRZ -L are combined into the Manchester scheme. In Manchester
encoding, the duration of the bit is divided into two halves. The voltage
remains at one level during the first half and moves to the other level in
the second half. The transition at the middle of the bit provides
synchronization. Differential Manchester, on the other hand, co mbines the
ideas of RZ and NRZ -I. There is always a transition at the middle of the
bit, but the bit values are determined at the beginning of the bit. Ifthe next
bit is 0, there is a transition; if the next bit is 1, there is none. Following
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Figure: Polar biphase: Manchester and differential Manchester schemes
In Manchester and differential Manchester encoding, the transition
at the middle of the bit is used for synchronization.
The Manchester scheme overcomes several problems associated
with NRZ -L, and differential Manchester overcomes several problems
associated with NRZ -I.First, there is no baseline wandering. There is no
DC component because each bit has a positive and negative voltage
contribution. The only drawback is the signal rate. The signal rate for
Manchester and differential Manchester is double that for NRZ. The
reason is that there is always one transition at the middle of the bit and
maybe one transition at the end of each bit. Above figure shows both
Manchester and differential Manchester encoding schemes. Note that
Manchester and differential Manchester schemes are also called biphase
schemes.
Bipolar Schemes
In bipolar encodin g (sometimes called multilevel binary), there are
three voltage levels: positive, negative, and zero. The voltage level for one
data element is at zero, while the voltage level for the other element
alternates between positive and negative.
AMI and Pseudo ternary
Following figure shows two variations of bipolar encoding: AMI
and pseudoternary. A common bipolar encoding scheme is called bipolar
alternate mark inversion (AMI). In the term alternate mark inversion, the
word mark comes from telegraphy and means 1. So AMI means alternate I
inversion. A neutral zero voltage represents binary O. Binary Is are
represented by alternating positive and negative voltages. A variation of
AMI encoding is called pseudoternary in which the 1bit is encoded as a
zero voltage and the 0 bit is encoded as alternating positive and negative
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Figure: Bipolar schemes: AMI and pseudoternary
The bipolar scheme was developed as an alternative to NRZ. The
bipolar scheme has the same si gnal rate as NRZ, but there is no DC
component. The NRZ scheme has most of its energy concentrated near
zero frequency, which makes it unsuitable for transmission over channels
with poor performance around this frequency. The concentration of the
energy in bipolar encoding is around frequency N12. Above figure shows
the typical energy concentration for a bipolar scheme.
One may ask why we do not have DC component in bipolar
encoding. We can answer this question by using the Fourier transform, but
we can al so think about it intuitively. Ifwe have a long sequence of 1s, the
voltage level alternates between positive and negative; it is not constant.
Therefore, there is no DC component. For a long sequence of Os, the
voltage remains constant, but its amplitude is zero, which is the same as
having no DC component. In other words, a sequence that creates a
constant zero voltage does not have a DC component.
AMI is commonly used for long -distance communication, but it
has a synchronization problem when a long sequence of Os is present in
the data.
Multilevel Schemes
Its goal is to increase the number of bits per baud by encoding a
pattern of mdata elements into a pattern of nsignal elements. Two types
of data elements (0s and 1s), which means that a group of mdata elements
can produce a combination of 2mdata patterns. We can have different
types of signal elements by allowing different signa l levels. If we have L
different levels, then we can produce Lncombinations of signal patterns.
If2m =Ln, then each data pattern is encoded into one signal pattern. If 2m
be care fully designed to prevent baseline wandering, to provide
synchronization, and to detect errors that occurred during data
transmission.
Data encoding is not possible if 2m>Lnbecause some of the data
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59Multiline Transmission
NRZ -I and differential Manchester are classified as differential
encoding but use two transition rules to encode binary data (no inversion,
inversion). If we have a signal with more than two levels, we can design a
differential encoding scheme with more th an two transition rules. MLT -3
is one of them. The multiline transmission, three level (MLT -3) scheme
uses three levels (+ v,0, and -v)and three transition rules to move between
the levels.
If the next bit is 0, there is no transition.
If the next bit is 1 and the current level is not 0, the next level is 0.
If the next bit is 1 and the cut Tent level is 0, the next level is the
opposite of the last non zero level.
Figure: Multi -transition: MLT -3 scheme
4.1.3 Block Coding
We need redundancy to ensure synchronization and to provide
some kind of inherent error detecting. Block coding can give us this
redundancy and improve the performance of line coding. In general, block
coding changes a block of mbits into a block of nbits, where nis larger
than m.Block coding is referred to as an mB/nB encoding technique.
The slash in block encoding (for example, 4B/5B) distinguishes
block encoding from multilevel encoding (for example, 8B6T), which is
written without a slash. Block coding normally involves three steps:
division, substitution, and combination. In the divisi on step, a sequence of
bits is divided into groups of mbits. For example, in 4B/5B encoding, the
original bit sequence is divided into 4 -bit groups. The heart of block
coding is the substitution step. In this step, we substitute an m -bit group
for an n -bitgroup. For example, in 4B/5B encoding we substitute a 4 -bit
code for a 5 -bit group. Finally, the n -bit groups are combined together to
form a stream. The new stream has more bits than the original bits.
Following figure shows the procedure.munotes.in

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60
Figure: Block coding concept
4.1.4 Scrambling
Biphase schemes that are suitable for dedicated links between
stations in a LAN are not suitable for long -distance communication
because of their wide bandwidth requirement. The combination of block
coding and NRZ line coding is not suitable for long -distance encoding
either, because of the DC component. Bipolar AMI encoding, on the other
hand, has a narrow bandwidth and does not create a DC component.
However, a long sequence of Os upsets the synchronization. Ifwe can find
a way to avoid a long sequence of Os in the original stream, we can use
bipolar AMI for long distances. We are looking for a technique that does
not increase the number of bits and does provide synchronization. We are
looking for a solution that substitutes long zero -level pulses with a
combination of other levels to provide synchronization. One solution is
called scrambling. We modify part of the AMI rule to include scrambling,
as shown in following figure. Note that scrambling, as opposed to block
coding, is done at the same time as enco ding. The system needs to insert
the required pulses based on the defined scrambling rules. Two common
scrambling techniques are B8ZS and HDB3.
Figure: AMI used with scrambling
R8ZS
Bipolar with S -zero substitution (BSZS) is commonly used in
North America. In this technique, eight consecutive zero -level voltages are
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61The V in the sequence denotes violation; this is a nonzero voltage
that breaks an AMI rule of encoding (opposite polarity fr om the previous).
The B in the sequence denotes bipolm; which means a nonzero level
voltage in accordance with the AMI rule. There are two cases, as shown in
following figure.
Figure: Two cases of B8ZS scrambling technique
HDB3
High -density bipolar 3 -zero (HDB3) is commonly used outside of
North America. In this technique, which is more conservative than B8ZS,
four consecutive zero -level voltages are replaced with a sequence of
OOOV or BOOV The reason for two different substitutions is to maintain
the even number of nonzero pulse s after each substitution. The two rules
can be stated as follows
If the number of nonzero pulses after the last substitution is odd,
the substitution pattern will be OOOV, which makes the total
number of nonzero pulses even.
Ifthe number of nonzero pulse s after the last substitution is even,
the substitution pattern will be BOOV, which makes the total
number of nonzero pulses even.
Following figure shows the example of HDB3
Figure: Different situations in HDB3 s crambling techniquemunotes.in

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624.2ANALOG -TO-DIGITAL CONVERSION
Sometimes, we have an analog signal such as one created by a
microphone or camera. We have seen in previous chapter that a digital
signal is superior to an analog signal. The tendency today is to change an
analog signal to digital data. For conversion two techniques are used,
pulse code modulation and delta modulation. After the digital data are
created (digitization), we can use one of the techniques described of line
coding to convert the digital data to a digital signal.
4.2.1 Pulse Code Modulation (PCM )
To convert analog wave into digital data we use Pulse Code
Modulation . Pulse Code Modulation is one of the most commonly used
method to convert analog data into digital form. It involves three steps:
Sampling, Quantization and Encoding.
Figure Components of PCM encoder
Steps involved are
1. The analog signal is sampled.
2. The sampled signal is quantized.
3. The quantized values are encoded as streams of bits.
Sampling
The sampling process is sometimes referred to as pulse amplitude
modulation (PAM). But, that result is still an analog signal with non
integral values.
The first step in PCM is sampling.
The analog signal is sampled every Tss, where Tsis the sample
interval or period. The inverse of the sampling interv al is called the
sampling rate or sampling frequency.
There are three sampling methods: Ideal, Natural, Flat -top
Inideal sampling , pulses from the analog signal are sampled. This
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63Innatural s ampling , a high -speed switch is turned on for only the
small period of time when the sampling occurs. The result is a sequence of
samples that retains the shape of the analog signal.
The most common sampling method, called sample and hold,
however, create sflat-topsamples by using a circuit.
Figure: Three different sampling methods for PCM
Delta Modulation (DM)
PCM is a very complex technique. Other techniques have been
developed to reduce the complexity of PCM. The simplest is delta
modulation. PCM finds the value of the signal amplitude for each sample;
DM finds the change from the previous sample. Following fi gure shows
the process. Note that there are no code words here; bits are sent one after
another.
Figure: The process of delta modulation
Modulator
The modulator is used at the sender site to create a stream of b its
from an analog signal. The process records the small positive or negative
changes, called delta . If the delta is positive, the process records a I; if it
is negative, the proc ess records a O. However, the process needs a base
against which the analog signal is compared. The modulator builds amunotes.in

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64second signal that resembles a staircase. Finding the change is then
reduced to comparing the input signal with the gradually made stairc ase
signal. Following figure shows a diagram of the process.
Figure: Delta modulation components
The modulator, at each sampling interval, compares the value of
the analog signal with the last value of the staircase sig nal. Ifthe
amplitude of the analog signal is larger, the next bit in the digital data is 1;
otherwise, it is O. The output of the comparator, however, also makes the
staircase itself. If the next bit is I, the staircase maker moves the last point
of the s taircase signal up; it the next bit is 0, it moves it down. Note
that we need a delay unit to hold the staircase function for a period
between two comparisons.
Demodulator
The demodulator tak es the digital data and, using the staircase
maker and the delay unit, creates the analog signal. The created analog
signal, however, needs to pass through a low -pass filter for smoothing.
Following figure shows the schematic diagram.
Figure: Delta demodulation components
Adaptive DM
A better performance can be achieved if the value of is not fixed.
In adaptive delta modulation , the value of changes according to the
amplitude of the analog signal.
Quantization Error
Itis obvious that DM is not perfect. Quantization error is always
introduced in the process. The quantization error of DM, however, is much
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654.3TRANSMIS SION MODES
The transmission of data from one device to another is the wiring,
and of primary concern when we are considering the wiring is the data
stream. The transmission of binary data (0 and 1) across a link can be
accomplished in either parallel orserial mode.
Inparallel mode, multiple bits are sent with each clock tick.
Inserial mode, 1 bit is sent with each clock tick. While there is
only one way to send parallel data, there are three sub classes of
serial transmission: asynchronous, synchronous , and
isochronous .
4.3.1 Parallel Transmission
Binary data, consisting of 1s and 0s, may be organized into groups
ofnbits each. Computers produce and consume data in groups of bits. By
grouping, we can send data nbits at a time instead of 1. This is called
parallel transmission.
Figure: Parallel transmission
The mechanism for parallel transmission is a conceptually simple one:
Usenwires to send nbits at one time.
That way each b it has its own wire, and all nbits of one group can be
transmitted with each clock tick from one device to another.
The advantage of parallel transmission is speed.
That is parallel transmission can increase the transfer speed by a factor
ofnover serial transmission.
But there is a significant disadvantage is cost: Parallel transmission
requires ncommunication lines (wires in the example) just to transmit the
data stream. Because this is expensive, parallel transmission is usually
limited to short distances.
4.3.2 Serial Transmission
In serial transmission one bit follows another, so we need only one
communication channel rather than nto transmit data between two
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66
In serial transmission one bit follows another, so we need only one
communication channel rather than nto transmit data between two
communicating devices.
Theadvantage ofserial over parallel transmission is that with only
one communication channel, ser ial transmission reduces the cost of
transmission over parallel by roughly a factor of n.Since communication
within devices is parallel, conversion devices are required at the interface
between the sender and the line (parallel -to-serial) and between the line
and the receiver (serial -to parallel).
Serial transmission occurs in one of three ways: asynchronous,
synchronous, and isochronous.
Asynchronous
Asynchronous transmission is so named because the timing of a
signal is unimportant. Instead, information is received and translated by
agreed upon patterns. Patterns are based on grouping the bit stream into
bytes. Each group, usually 8 bits, is sent alon g the link as a unit. The
sending system handles each group independently, relaying it to the link
whenever ready, without regard to a timer. Without synchronization, the
receiver cannot use timing to predict when the next group will arrive. To
alert the r eceiver to the arrival of a new group, therefore, an extra bit is
added to the beginning of each byte. This bit, usually a 0, is called the start
bit. To let the receiver know that the byte is finished, 1 or more additional
bits are appended to the end of the byte. These bits, usually 1s, are called
stop bits . The start and stop bits and the gap alert the receiver to the
beginning and end of each byte allow it to synchronize with the data
stream.
This mechanism is called asynchronous because, at the byte l evel,
the sender and receiver do not have to be synchronized. But within each
byte, the receiver must still be synchronized with the incoming bit stream.munotes.in

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67
Figure: Asynchronous transmission
Synchronous Transmission
In synchronous transmission, the bit stream is combined into longer
"frames," which may contain multiple bytes. Each byte is introduced onto
the transmission link without a gap between it and the next one. It is left to
the receiver to separate the bit str eam into bytes for decoding purposes.
Figure: Synchronous Transmission
Inother words , data are transmitted as an unbroken string of 1s and
0s, and the receiver separates that string into the bytes, or characters, it
needs to reconstruct the information.
Theadvantage of synchronous transmission is speed .
With no extra bits or gaps to introduce at the sending end and
remove at the receiving end, and, by extension, with fewer bits to
move across the link.
Synchronous transmission is faster than asynchronous
transmission.
For this reason, it is more useful for high -speed applications such
as the transmission of data from one computer to another.
Byte synchronization is accomplished in the data link layer.
Although there is no gap between characters in synchronous serial
transmission, there may be uneven gaps between frames.
Isochronous
In real -time audio and video, in which uneven delays between
frames are not acceptable synchronous transmission fails. For example,
TV images are broadcast at the rate of 30 images per second; they must bemunotes.in

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68viewed at the same rate. If each image is sent by using one or more
frames, there should be no delays between frames. For this type of
application, synchronization between characters is not enough; the entire
stream of bits must be synchronized. The isochronous transmission
guarantees that the data arrive at a fixed rate.
4.4DIGITAL -TO-ANALOG CONVERSION
Digital -to-analog conversion is the process of changing one of the
characteristics of an analog signal based on the information in digital data.
Following figure shows the relationship between the digita l information,
the digital -to-analog modulating process, and the resultant analog signal.
Figure: Digital -to-analog conversion
The above figure shows the relationship between the digital
information, the digital -to-analog modulating process, and the resultant
analog signal.
We have discussed that sine wave is defined by three
characteristics: amplitude, frequency, and phase. When we vary anyone of
these characteristics, we create a diffe rent version of that wave. So, by
changing one characteristic of a simple electric signal, digital data is
represented. Any of the three characteristics can be altered in least three
mechanisms for modulating digital data into an analog signal: amplitude
shift keying (ASK), frequency shift keying (FSK), and phase shift keying
(PSK). In addition, there is a fourth (and better) mechanism that combines
changing both the amplitude and phase, called quadrature amplitude
modulation (QAM). QAM is the most efficien tof these options and is the
mechanism commonly used today.
Figure: Types of digital -to-analog conversionmunotes.in

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694.4.1 Aspects of Digital -to-Analog Conversion
Before we discuss specific methods of digital -to-analog
modulation, two basic issues must be reviewed: bit and baud rates and the
carrier signal.
i) Data Element vs Signal Element: we have discussed the data element
as the smallest piece of information to be exchanged, the bit and the signal
element are also as the smalle st unit of a signal that is constant. These
terms are only little difference in digital to analog conversion.
ii) Data Rate vs Signal Rate (Bit rate vs Baud rate): Bit rate is the
number of bits transmitted during 1 sec. Baud rate refer to the number of
signal units per second that are required to represent those bits.
Relationship between these two are:
Baud Rate= Bit Rate/Number of Bits Per Signal Unit
In transportation, a baud is analogous to a vehicle, and a bit is
analogous to a passenger.
iii) Bandwidth: The required bandwidth for analog transmission of digital
data is proportional to the signal rate except for FSK, in which the
difference between the carrier signals needs to be added.
iv) Carrier Signal :In analog transmission, the sending device produces a
high frequency signal that acts as a base for the information signal. This
base signal is called the carrier signal or carrier frequency .T h e
receiving device is turned to the frequency of the carrier s ignal that it
expects from the sender. Digital information then changes the carrier
signal by modifying one or more of its characteristics (amplitude,
frequency, or phase). This kind of modification is called modulation
(shift keying).
4.4.2 Amplitude shi ft keying
In amplitude shift keying, the amplitude of the carrier signal is
varied to create signal elements. Both frequency and phase remain
constant while the amplitude changes.
ASK is normally implemented using only two levels.
This is referred to as bi nary amplitude shift keying or on-off
keying (OOK). The peak amplitude of one signal level is 0, the
other is the same as the amplitude of the carrier frequency.
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70Bandwidth for ASK: the bandwidth is proportional to the signa lr a t e
(baud rate). However, there is normally another factor involved, called d,
which depends on the modulation and filtering process. The value of dis
between 0 and 1.the relationship can be expressed as
B=( 1+d) xS(N baud)
Where, B is the bandwidth, S/N baudis the baud rate and d is the factor
related to the modulation process (with minimum value 0).
4.4.3 Frequency Shift Keying
In frequency shift keying, the frequency of the carrier signal is
varied to represent data. The frequency of the modulated signal is constant
for the duration of one signal element, but changes for the next signal
element if the data element changes. Both peak amplitude andphase
remain constant for all signal elements.
Binary FSK (BFSK)
Figure: Binary frequency shift keying
Here binary FSK (or BFSK) is to consider two carrier frequencies.
In the above figure we have select ed two carrier frequencies, f1 and f2.
It is assumed for first carrier the data element is 0 for second data
element is 1.
However, note that this is an unrealistic example used only for
demonstration purposes.
Normally the carrier frequencies are very hi gh, and the difference
between them is very small .
Here the middle of one bandwidth is f1 and the middle of the other
is f2 .Both f1 and f2 are Δf apart from the midpoint between the two
bands. The difference between the two frequencies is 2Δf.
Bandwidth for BFSK
Carrier signals are only simple sine waves, but the modulation
creates a non periodic composite signal with continuous frequencies. For
FSK it can think as two ASK signals, each with its own carrier frequency
f2. If the difference betwee n the two frequencies is 2Δf,then the required
bandwidth is
B=(l+d)xS+ 2Δf
Where, B is the bandwidth, S is the baud rate and d is the factor related to
the modulation process (with minimum value 0) and 2Δf is the frequency
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714.4.4 Phase Shift Keying
In phase shift keying, the phase of the carrier is varied to represent
two or more different signal elements 0 and 1.Both peak amplitude and
frequency remain constant as the phase changes. The phase of the signal
during each bit du ration is constant and its value depends on the bits (0
and 1). Today, PSK is more common than ASK or FSK. However QAM,
which combines ASK and PSK, is the dominant method of digital -to-
analog modulation.
Binary PSK (BPSK)
The simplest PSK is binary PSK, in which we have used only two
signal elements, one with a phase of 0°, and the other with a phase of
180°. Below figure shows a conceptual view of PSK and relationship of
phase to bit value.
Binary PSK is as simple as bin ary ASK with one big advantage -it
is less susceptible to noise .
Bandwidth
The bandwidth for BFSK is the same as that for binary ASK, but
less than that for BFSK. Here no bandwidth is wasted for separating two
carrier signa ls.
The implementation of BPSK is as simple as that for ASK. The
reason is that the signal element with phase 180° can be seen as the
complement of the signal element with phase 0°.This gives us a clue on
how to implement BPSK. Here it has been used same idea used for ASK
but with a polar NRZ signal instead of a uni polar NRZ signal. The polar
NRZ signal is multiplied by the carrier frequency, the 1 bit (positive
voltage) is represented by a phase starting at 0°, the a bit (negative
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724.4.5 Quadrature Amplitude Modulation
Quadrature Amplitude Modulation is the idea of using two carriers,
one in -phase and the other quadrature, with different amplitude levels for
each carrie r.
Figure: Constellation diagrams for some QAMs
The possible variations of QAM are numerous. Below shows some
of these schemes where 4 -QAM scheme (fo ur different signal element
types) is a simplest one using a unipolar NRZ signal to modulate each
carrier. This is the same mechanism used for ASK (OOK).
Similarly Part b shows another 4 -QAM using polar NRZ, but this
is exactly the same as QPSK. Part c sh ows another QAM -4 in which we
used a signal with two positive levels to modulate each of the two carriers.
Finally, figure shows a 16 -QAM constellation of a signal with
eight levels, four positive and four negative.
Bandwidth for QAM
The minimum bandwidt h required for QAM transmission is the
same as that required for ASK and PSK transmission. QAM has the same
advantages as PSK over ASK.
4.5 ANALOG -TO-ANALOG CONVERSION
Analog -to-analog conversion, or analog modulation, is the
representation of analog inf ormation by an analog signal. This modulation
is needed if the medium is band pass in nature or if only a band pass
channel is available to us. An example is radio . The government assigns a
narrow bandwidth to each radio station. The analog signal produced by
each station is a low-pass signal , all in the same range. To be able to listen
to different stations, the low -pass signals need to be shifted ,e a c ht oa
different range.
Analog -to-analog conversion can be accomplished in three ways:
amplitude modulati on (AM) ,frequency modulation (FM) , and phase
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73
4.5.1 Amplitude Modulation
InAM transmission, the carrier signal is modulated so that its
amplitude varies with the changing amplitudes of the modulating (audio)
signal. The frequency and phase of the carrier remain the same, only the
amplitude changes to follow variations in the information. Where the
modulating signal is act as the envelope of the carr ier.
AMis normally implemented by using a simple multiplier because
the amplitude of the carrier signal needs to be changed according to the
amplitude of the modulating signal (audio).
AM Bandwidth
The modulation creates a bandwidth that is twice the bandwidth of
the modulating signal and covers a range centered on the carrier
frequency. However, the signal components above and below the carrier
frequency carry exactly the same information. For this reason, some
implement ations discard one -half of the signals and cut the bandwidth in
half.
The total bandwidth required for AMcan be determined from the
bandwidth of the audio signal:
BAM=2Bm
Where, BAMbandwidth of AM signal and Bmis bandwidth of modulating
signal
4.5.2 Frequency Modulation
In FM transmission, the frequency of the carrier signal is
modulated to follow the changing voltage level (amplitude) of the
modulating signal. The peak amplitude andphase of the carrier signal
remain constant, but as the amplit udeof the information signal changes,
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74
Figure: Frequency Modulation
The actual bandwidth is difficult to determine exactly, but it can be
shown empirically that it is several times that of the analog signal or 2(1 +
β)Bwhere βis a factor depends on modulation technique with a common
value of 4. In some books it is given that:
FM Bandwidth
The total bandwidth required for FM can be determined from the
bandwidth of the audio signal as: BFM=10xB mWhere ,BFMis bandwidth
of FM signal and B mis the bandwidth of modulating signal.
4.5.3 Phase Modulation
InPMtransmission, the phase of the carrier signal is modulated to
follow the changing voltage level (amplitude) of the modulating signal.
Thepeak amplitude andfrequency of the carrier signal remain constant,
but as the amplitude of the information signal cha nges, the phase of the
carrier changes correspondingly.
It has been proved that PM is the same as FM with one difference.
In FM, the instantaneous change in the carrier frequency is proportional
to the amplitude of the modulating signal where as in PM the
instantaneous change in the carrier frequency is proportional to the
derivative of the amplitude of the modulating signal.
Figure: Phase modulation
As the above figure shows, PM is normally implemented by using
a voltage -controlled oscillator along with a derivative. The frequency ofmunotes.in

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75the oscillator changes according to the derivative of the input voltage
which is the amplitude of the modulating signal.
PM Bandwidth
The actual bandwidth is difficult to determine exactly, b ut it can be
shown empirically that it is several times that of the analog signal.
Although, the formula shows the same bandwidth for FM and PM, the
value of βis lower in the case of PM (around 1 for narrowband and 3 for
wideband).
The total bandwidth required for PM can be determined from the
bandwidth and maximum amplitude of the modulating signal:
Bpm = 2 (1 +β)B.
4.6 REVIEW QUESTIONS
1.What are the three techniques of digital -to-digital conversion?
2.Enlist different line coding schemes.
3.Explain block coding and give its purpose.
4.Define scrambling and give its purpose.
5.Explain analog transmission.
6.Explain digital -to-analog conversion.
7.Which of the four digital -to-analog conversion techniques (ASK, FSK,
PSK or QAM) is the most susceptible to noise? Why explain it?
8.Define constellation diagram and its role in analog transmission.
9.Explain analog -to-analog conversion in brief.
10.Which of the three analog -to-analog conversion techniques (AM, FM,
or PM) is the most susceptible to noise? Why explain it?
11.Distinguish between
Signal element and a data element.
Data rate and signal rate.
Parallel and serial transmission
4.7 SUMMARY
Digital -to-digital conversion involves three techniques: line coding,
block coding, and scrambling.
Line coding is the process of converting digital data to a digital signal.
We can roughly divide line coding schemes into five broad categories:
unipolar, polar, bipolar, multilevel, and multitransitio n.
Block coding provides redundancy to ensure synchronization and
inherent error detection. Block coding is normally referred to as
mB/nB coding; it replaces each m -bit group with an n -bit group.munotes.in

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76Scrambling provides synchronization without increasing the number of
bits. Two common scrambling techniques are B8ZS and HDB3.
The most common technique to change an analog signal to digital data
(digitization) is called pulse code modulation (PCM).
The first step in PCM is sampling. The analog signal is sampled e very
Tss, where Tsis the sample interval or period. The inverse of the
sampling interval is called the sampling rate orsampling frequency
and denoted by fs,where fs=lITs. There are three sampling methods -
ideal, natural, and flat -top.
According to the Nyquist theorem, to reproduce the original analog
signal, one necessary condition is that the sampling rate be at least
twice the highest frequency in the original signal.
The simplest is delta modulation. PCM finds the value of the signal
amplitude for e ach sample; DM finds the change from the previous
sample.
While there is only one way to send parallel data, there are three
subclasses of serial transmission: asynchronous, synchronous, and
isochronous.
In asynchronous transmission, we send 1 start bit ( 0) at the beginning
and 1 or more stop bits (1 s) at the end of each byte.
In synchronous transmission, we send bits one after another without
start or stop bits or gaps. It is the responsibility of the receiver to group
the bits.
The isochronous mode prov ides synchronized for the entire stream of
bits must. In other words, it guarantees that the data arrive at a fixed
rate.
Digital -to-analog conversion is the process of changing one of the
characteristics
of an analog signal based on the information in the digital data.
Digital -to-analog conversion can be accomplished in several ways:
amplitude shift keying (ASK), frequency shift keying (FSK), and
phase shift keying (PSK). Quadrature amplitude modulation (QAM)
combines ASK and PSK.
In amplitude shift keying , the amplitude of the carrier signal is varied
to create signal elements. Both frequency and phase remain constant
while the amplitude changes.
In frequency shift keying, the frequency of the carrier signal is varied
to represent data. The frequency of th e modulated signal is constant for
the duration of one signal element, but changes for the next signal
element if the data element changes. Both peak amplitude and phase
remain constant for all signal elements.munotes.in

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77In phase shift keying, the phase of the carri er is varied to represent two
or more different signal elements. Both peak amplitude and frequency
remain constant as the phase changes.
A constellation diagram shows us the amplitude and phase of a signal
element, particularly when we are using two carrie rs (one in -phase and
one quadrature).
Quadrature amplitude modulation (QAM) is a combination of ASK
and PSK. QAM uses two carriers, one in -phase and the other
quadrature, with different amplitude levels for each carrier.
Analog -to-analog conversion is the representation of analog
information by an analog signal. Conversion is needed if the medium
is bandpass in nature or if only a bandpass bandwidth is available to
us.
Analog -to-analog conversion can be accomplished in three ways:
amplitude modulation (AM), frequency modulation (FM), and phase
modulation (PM).
InAM transmission, the carrier signal is modulated so that its
amplitude varies with the changing amplitudes of the modulating
signal. The frequency and phase of the carrier remain the same; only
theamplitude changes to follow variations in the information.
In PM transmission, the frequency of the carrier signal is modulated to
follow the changing voltage level (amplitude) of the modulating
signal. The peak amplitude and phase of the carrier signal re main
constant, but as the amplitude of the information signal changes, the
frequency of the carrier changes correspondingly.
In PM transmission, the phase of the carrier signal is modulated to
follow the changing voltage level (amplitude) of the modulating
signal. The peak amplitude and frequency of the carrier signal remain
constant, but as the amplitude of the information signal changes, the
phase of the carrier changes correspondingly.
4.8 REFERENCES
1. Data Communication & Networking –Behrouz Forouzan
2. TCP/IP Protocol Suite –Behrouz Forouzan
3. Computer Networks –Andrew Tanenbaum1
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78Unit II
5
BANDWIDTH UTILIZATION:
MULTIPLEXING AND SPECTRUM
SPREADING
Unit Structure :
5.0 Objectives
5.1 Introduction
5.2 Multiplexing
5.2.1 Types of Multiplexing
5.2.2 Frequency -Division Multiplexing
5.2.3 Wavelength -Division Multiplexing
5.2.4 Time -Division Multiplexing
5.2.4.1 Synchronous Time -Division Multiplexing
5.2.4.2 Statistical Time -Division Multiplexing
5.3 Spread Spectrum
5.3.1 Types of Spread Spectrum
5.3.2 Frequency Hopping Spread Spectrum (FHSS)
5.3.3 Direct Sequence Spread Spectrum (DSSS)
5.4 Summary
5.5 Reference for further reading
5.6 Model Questions
5.0 OBJECTIVES:
This chapter would make you to understand the following
concepts:
Concept of Multiplexing.
Types of Multiplexing: Frequency –division multiplexing,
Wavelength –division multiplexing and Time –division multiplexing.
Concept of Spread Spectrum.
Types of S pread Spectrum: Frequency Hoping Spread Spectrum
(FHSS) and Direct Sequence Spread Spectrum (DSSS).munotes.in

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795.1 INTRODUCTION
In real life, we have links with limited bandwidths. The efficient
use of these bandwidths has been, and will be, one of the main chall enges
of electronic communications. However, the meaning of efficient may
depend on the application. Sometimes we need to combine several low -
bandwidth channels to make use of one channel with a larger bandwidth.
Sometimes we need to expand the bandwidth o f a channel to achieve goals
such as privacy and anti jamming. In this chapter, we will explore these
two broad categories of bandwidth utilization: Multiplexing and
Spreading. In multiplexing, our goal is efficiency; we combine several
channels into one. In spreading, our goals are privacy and anti-jamming;
we expand the bandwidth of a channel to insert redundancy, which is
necessary to achieve these goals.
5.2 MULTIPLEXING
Whenever the bandwidth of a medium linking two devices is
greater than the bandw idth needs of the devices, the link can be shared.
Multiplexing is the set of techniques that allows the simultaneous
transmission of multiple signals across a single data link. As data and
telecommunications use increases, so does traffic. We can accommod ate
this increase by continuing to add individual links each time a new
channel is needed; or we can install higher -bandwidth links and use each
to carry multiple signals. Today’s technology includes high -bandwidth
media such as optical fiber and terrestri al and satellite microwaves. Each
has a bandwidth far in excess of that needed for the average transmission
signal. If the bandwidth of a link is greater than the bandwidth needs of
the devices connected to it, the bandwidth is wasted. An efficient system
maximizes the utilization of all resources; bandwidth is one of the most
precious resources we have in data communications.
In a multiplexed system, ‘n’ lines share the bandwidth of one link.
Figure 5.1 shows the basic format of a multiplexed system. The lines on
the left direct their transmission streams to a multiplexer (MUX), which
combines them into a single stream (many -to-one). At the receiving end,
that stream is fed into a de -multiplexer (DEMUX), which separates the
stream back into its component transmissions (one -to-many) and directs
them to their corresponding lines. In the figure, the word link refers to the
physical path. The word channel refers t o the portion of a link that carries
a transmission between a given pair of lines. One link can have many (n)
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80
Figure 5.1: Dividing a link into channels
5.2.1 Types of Multiplexing
There are thr ee basic multiplexing techniques: Frequency -division
multiplexing, Wavelength -division multiplexing, and Time -division
multiplexing. The first two are techniques designed for analog signals, the
third, for digital signals (see Figure 5.2).
Figure 5.2: Types of Multiplexing
5.2.2 Frequency -Division Multiplexing
Frequency -division multiplexing (FDM) is an analog technique
that can be applied when the bandwidth of a link (in hertz) is greater than
the combined bandw idths of the signals to be transmitted. In FDM, signals
generated by each sending device modulate different carrier frequencies.
These modulated signals are then combined into a single composite signal
that can be transported by the link. Carrier frequenci es are separated by
sufficient bandwidth to accommodate the modulated signal. These
bandwidth ranges are the channels through which the various signals
travel. Channels can be separated by strips of unused bandwidth -guard
bands -to prevent signals from over lapping. In addition, carrier frequencies
must not interfere with the original data frequencies. Figure 5.3 gives a
conceptual view of FDM. In this figure, the transmission path is divided
into three parts, each representing a channel that carries one tran smission.
Figure 5.3: Frequency Division Multiplexing (FDM)munotes.in

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81We consider FDM to be an analog multiplexing technique;
however, this does not mean that FDM cannot be used to combine sources
sending digital signals. In such ca se a digital signal can be first converted
to an analog signal before FDM is used to multiplex them.
Multiplexing Process
Figure 5.4 is a conceptual illustration of the multiplexing process. Each
source generates a signal of a similar frequency range. Inside the
multiplexer, these similar signals modulate different carrier frequencies
(f1, f2 and f3). The resulting modulated signals are then combined into a
single composite signal that is sent out over a media link that has enough
bandw idth to accommodate it.
Figure 5.4: FDM –Multiplexing process
De-multiplexing Process
The de -multiplexer uses a series of filters to decompose the
multiplexed signal into its constituent component signals. The individua l
signals are then passed to a demodulator that separates them from their
carriers and passes them to the output lines. Figure 5.5 is a conceptual
illustration of de -multiplexing process.
Figure 5.5: FDM –De-multiplexing processmunotes.in

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82Example 5.1:
Assume that a voice channel occupies a bandwidth of 4 kHz. We
need to combine three voice channels into a link with a bandwidth of 12
kHz, from 20 to 32 kHz. Show the configuration, using the frequency
domain. Assume t here are no guard bands.
Solution:
We shift (modulate) each of the three voice channels to a different
bandwidth, as shown in Figure 5.6. We use the 20 -kHz to 24 -kHz
bandwidth for the first channel, the 24 -kHz to 28 -kHz bandwidth for the
second channel, a nd the 28 -kHz to 32 -kHz bandwidth for the third one.
Then we combine them as shown in Figure 5.6. At the receiver, each
channel receives the entire signal, using a filter to separate out its own
signal. The first channel uses a filter that passes frequenci es between 20
and 24 kHz and filters out (discards) any other frequencies. The second
channel uses a filter that passes frequencies between 24 and 28 kHz, and
the third channel uses a filter that passes frequencies between 28 and 32
kHz. Each channel then shifts the frequency to start from zero.
Figure 5.6: Example 5.1
Example 5.2:
Five channels, each with a 100 -kHz bandwidth, are to be
multiplexed together. What is the minimum bandwidth of the link if there
is a need for a guard band of 10 -kHz between the channels to prevent
interference?
Solution:
For five channels, we need at least four guard bands. This means
that the required bandwidth is at least 5 x 100 + 4 x 10 = 540 kHz .munotes.in

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83Applications of FDM
A very common application of FDM is AM and FM radio
broadcasting. Radio uses the air as the transmission medium. A special
band from 530 to 1700 kHz is assigned to AM radio. All radio stations
need to share this band. E ach AM station needs 10kHz of bandwidth. Each
station uses a different carrier frequency, which means it is shifting its
signal and multiplexing. The signal that goes to the air is a combination of
signals. A receiver receives all these signals, but filter s (by tuning) only
the one which is desired. Without multiplexing, only one AM station
could broadcast to the common link, the air. However, we need to know
that there is physical multiplexer or de -multiplexer here.
The situation is similar in FM broadca sting. However, FM has a
wider band of 88to 108 MHz because each station needs a bandwidth of
200 kHz.
Another common use of FDM is in television broadcasting. Each
TV channel has its own bandwidth of 6 MHz.
The first generation of cellular telephones (s till in operation) also
uses FDM. Eachuser is assigned two 30 -kHz channels, one for sending
voice and the other for receiving. The voice signal, which has a bandwidth
of 3 kHz (from 300 to 3300 Hz), is modulated by using FM. Remember
that an FM signal has a bandwidth 10 times that of the modulating signal,
which means each channel has 30 kHz (10 x 3) of bandwidth. Therefore,
each user is given, by the base station, a 60 -kHz bandwidth in a range
available at the time of the call.
FDM Implementation
FDM can be implemented very easily. In many cases, such as radio and
television broadcasting, there is no need for a physical multiplexer or de -
multiplexer. As long as the stations agree to send their broadcasts to the air
using different carrier frequencies, multiplexing is achieved. In other
cases, such as the cellular telephone system, a base station needs to assign
a carrier frequency to the telephone user. There is not enough bandwidth
in a cell to permanently assign a bandwidth range to every telephone user.
When a user hangs up, her or his bandwidth is assigned to another caller.
5.2.3 Wavelength -Division Multiplexing
Wavelength -division multiplexing (WDM) is designed to use the
high-data-rate capability of fiber -optic cable. The optical fiber data rate is
higher than the data rate of metallic transmission cable. Using a fiber -optic
cable for one single line wastes the available bandwidth. Multiplexing
allows us to combine several lines into one.
WDM is conceptually the same as FDM, except that the multi plexing and
de-multiplexing involve optical signals transmitted through fiber -optic
channels. The idea is the same: We are combining different signals of
different frequencies. The difference is that the frequencies are very high.munotes.in

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84Figure 5.7 gives a concep tual view of a WDM multiplexer and de -
multiplexer. Very narrow bands of light from different sources are
combined to make a wider band of light. At the receiver, the signals are
separated by the de -multiplexer.
Figure 5.7: Wavelength Division Multiplexing
Although WDM technology is very complex, the basic idea is very simple.
Wewant to combine multiple light sources into one single light at the
multiplexer and do the reverse at the de -multiplexer. The combining and
splitting of light sources are easily handled by a prism. Recall from basic
physics that a prism bends a beam of light based on the angle of incidence
and the frequency. Using this technique, a multiplexer can be made to
combine several input beams of light, each containing a narrow band of
frequencies, into one output beam of a wider band of frequencies. A de -
multiplexer can also be made to reverse the process. Figure 5.8 shows the
concept.
Figure 5.8: Prisms in Wave length Division Multiplexing and De -
multiplexing
Applications of WDM
One application of WDM is the SONET (Synchronous Optical
Network) in which multiple optical fiber lines are multiplexed and de -
multiplexed. A new method, called Dense WDM (DWDM), can mul tiplex
a very large number of channels by spacing channels very close to one
another. It achieves even greater efficiency.
5.2.4 Time -Division Multiplexing
Time -division multiplexing (TDM) i s a digital process that allows
several connections to share the high bandwidth of a link. Instead of
sharing a portion of the bandwidth as in FDM, time is shared. Each
connection occupies a portion of time in the link. Figure 5.9 gives amunotes.in

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85conceptual view o f TDM. Note that the same link is used as in FDM; here,
however, the link is shown sectioned by time rather than by frequency. In
the figure, portions of signals 1,2,3 and 4 occupy the link sequentially.
Figure 5.9: Tim e Division Multiplexing
Note that in Figure 5.9 we are concerned with only multiplexing,
not switching. This means that all the data in a message from source 1
always go to one specific destination, be it 1, 2, 3, or 4. The delivery is
fixed and unvarying , unlike switching.
We also need to remember that TDM is, in principle, a digital
multiplexing technique. Digital data from different sources are combined
into one timeshared link. However, this does not mean that the sources
cannot produce analog data; a nalog data can be sampled, changed to
digital data, and then multiplexed by using TDM.
We can divide TDM into two different schemes: synchronous and
statistical. We first discuss synchronous TDM and then show how
statistical TDM differs.
5.2.4.1 Synchro nous Time -Division Multiplexing
In synchronous TDM, each input connection has an allotment in
the output even if it is not sending data.
Time Slots and Frames
In synchronous TDM, the data flow of each input connection is
divided into units, where each input occupies one input time slot. A unit
can be 1 bit, one character, or one block of data. Each input unit becomes
one output unit and occupies one output time slot. However, the duration
of an output time slot is ntimes sho rter than the duration of an input time
slot. If an input time slot is Ts, the output time slot is Tins, where nis the
number of connections. In other words, a unit in the output connection has
a shorter duration; it travels faster. Figure 5.10 shows an e xample of
synchronous TDM where nis 3.munotes.in

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86
Figure 5.10: Synchronous TDM
In synchronous TDM, a round of data units from each input connection is
collected into a frame. If we have nconnections, a frame is divided into n
time slots and one slot is allocated for each unit, one for each input line. If
the duration of the input unit is T,the duration of each slot is Tinand the
duration of each frame is T.
The data rate of the output link must be ntimes the data rate of a
connection to guarantee the flow of data. In Figure 5.10, the data rate of
the link is 3 times the data rate of a connection; likewise, the duration of a
unit on a connection is 3 times that of the time slot (duration of a unit on
the link). In the figure we represent the data prior to multiplexing as 3
times the size of the data after multiplexing. This is just to convey the idea
that each unit is 3 times longer in duration before multiplexing than after.
Time slots are grouped into frames. A frame consis ts of one
complete cycle of time slots, with one slot dedicated to each sending
device. In a system with ninput lines, each frame has nslots, with each
slot allocated to carrying data from a specific input line.
Example 5.3:
In Figure 5.10, the data ra te for each input connection is 3 kbps. If
1 bit at a time is multiplexed (a unit is 1 bit), what is the duration of (a)
each input slot, (b) each output slot, and (c) each frame?
Solution:
We can answer the questions as follows:
a)The data rate of each input connection is 1 kbps. This means that
the bit duration is 1/1000 s or1m s. The duration of the input time
slot is 1m s (same as bit duration).
b)The duration of each output time slot is one -third of the input time
slot. This mea ns that the duration of the output time slot is 1/3 ms.
c)Each frame carries three output time slots. So the duration of a
frame is 3 x 1/3 ms , or1 ms. The duration of a frame is the same
as the duration of an input unit.munotes.in

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87Example 5.4:
Figure 5.11 shows syn chronous TOM with a data stream for each
input and one data stream for the output. The unit of data is 1 bit. Find (a)
the input bit duration, (b) the output bit duration,(c) the output bit rate, and
(d) the output frame rate.
Figure 5.11: Example 5.4
Solution:
We can answer the questions as follows:
a)The input bit duration is the inverse of the bit rate: 1/1 Mbps = 1
μs.
b)The output bit duration is one -fourth of the input bit duration, or
1/4μs.
c)The output bit rate is the inverse of the output bit duration or 1/(4
μs), or4M b p s . This can also be deduced from the fact that the
output rate is 4times as fast as any input rate; so the output rate = 4
x 1Mbps =4 Mbps.
d)The frame rate is always the same as any input rate. So the frame
rate is 1,000,000 frames per second. Because we are sending 4b i t s
in each frame, we can verify the result of the previous question by
multiplying the frame rate by the number of bits per frame.
Interleaving
TDM can be vis ualized as two fast -rotating switches, one on the
multiplexing side and the other on the de -multiplexing side. The switches
are synchronized and rotate at the same speed, but in opposite directions.
On the multiplexing side, as the switch opens in front of a connection, that
connection has the opportunity to send a unit onto the path. This process is
called interleaving. On the de -multiplexing side, as the switch opens in
front of a connection, that connection has the opportunity to receive a unit
from the path.
Figure 5.12 shows the interleaving process for the connection
shown in Figure 510.In this figure, we assume that no switching is
involved and that the data from the first connection at the multiplexer site
go to the first connection at the de -multip lexer.munotes.in

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88
Figure 5.12: Interleaving process
Example 5.5:
Four channels are multiplexed using TDM. If each channel sends 100
bytes/s and we multiplex
1 byte per channel, show the frame traveling on the link, the size of the
frame, the duration of a frame, the frame rate, and the bit rate for the link.
Solution:
The multiplexer is shown in Figure 5.13. Each frame carries 1 byte
from each channel; the size of each frame, therefore, is 4 bytes , or32 bits .
Because each channel is sending 100 bytes/s and a frame carries 1 byte
from each channel, the frame rate must be 100 frames per second. The
duration of a frame is therefore 1/100 s . The link is ca rrying 100 frames
per second, and since each frame contains 32 bits , the bit rate is 100 x 32 ,
or3200 bps . This is actually 4t i m e s the bit rate of each channel, which is
100 x 8 =800 bps .
Figure 5.13: Example 5.5
Empty S lots
Synchronous TDM is not as efficient as it could be. If a source
does not have data to send, the corresponding slot in the output frame is
empty. Figure 5.14 shows a case in which one of the input lines has no
data to send and one slot in another input line has discontinuous data.
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89The first output frame has three slots filled, the second frame has
two slots filled, and the third frame has three slots filled. No frame is full.
We learn in the next section that statistical TDM can improve the
efficiency by removing the empty slots from the frame.
Data Rate Management
One problem with TDM is how to handle a disparity in the input
data rates. In all our discussion so far, we assumed that the data rat es of all
input lines were the same. However, if data rates are not the same, three
strategies, or a combination of them, can be used. We call these three
strategies Multilevel multiplexing, Multiple -slot allocation, andPulse
stuffing.
Multilevel Multipl exing
Multilevel multiplexing is a technique used when the data rate of
an input line is a multiple of others. For example, in Figure 5.15, we have
two inputs of 20 kbps andthree inputs of 40 kbps. The first two input
lines can be multiplexed together to provide a data rate equal to the last
three. A second level of multiplexing can create an output of 160 kbps.
Figure 5.15: Multilevel Multiplexing
Multiple -Slot Allocation
Sometimes it is more efficient to allot more than one slot in a
frame to a single input line. For example, we might have an input line that
has a data rate that is a multiple of another input. In Figure 5.16, the input
line with a 50 -kbps data rate can be given two slots in the output. We
insert a serial -to-parallel converter in the line to make two inputs out of
one.
5.16: Multiple -slot Multiplexing
Pulse Stuffing
Sometimes the bit rates of sources are not multiple intege rs of each
other. Therefore, neither of the above two techniques can be applied. One
solution is to make the highest input data rate the dominant data rate and
then add dummy bits to the input lines with lower rates. This will increase
their rates. This te chnique is called Pulse stuffing , bit padding, or bitmunotes.in

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90stuffing. The idea is shown in Figure 5.17. The input with a data rate of 46
is pulse -stuffed to increase the rate to 50 kbps . Now multiplexing can take
place.
Figure 5 .17: Pulse Stuffing
Frame Synchronizing
The implementation of TDM is not as simple as that of FDM.
Synchronization between the multiplexer and de -multiplexer is a major
issue. If the multiplexer and the de -multiplexer are not synchronized, a bit
belongi ng to one channel may be received by the wrong channel. For this
reason, one or more synchronization bits are usually added to the
beginning of each frame. These bits, called framing bits , follow a pattern,
frame to frame, that allows the de -multiplexer to synchronize with the
incoming stream so that it can separate the time slots accurately. In most
cases, this synchronization information consists of 1 bit per frame,
alternating between 0 and 1, as shown in Figure 5.18.
Figure 5.18: Framing bits
Example 5.6:
We have four sources, each creating 250 characters per second. If
the interleaved unit is a character and 1 synchronizing bit is added to each
frame, find (a) the data rate of each source, (b) the duration of each
character in each source, (c) the frame rate, (d) the duration of each frame,
(e)the number of bits in each frame, and (f) the data rate of the link.
Solution:
We can answer the questions as follows:
a)The data rate of each source is 250 x 8 =2000 bps =2 kbps.
b)Each source sends 250 characters per second; therefore, the
duration of a c haracter is 1/250 s ,or4 ms.munotes.in

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91c)Each frame has one character from each source, which means the
link needs to send 250frames per second to keep the transmission
rate of each source.
d)The duration of each frame is 1/250s , or 4m s . Note that the
duration of each f rame is the same as the duration of each character
coming from each source.
e)Each frame carries 4characters and 1extra synchronizing bit. This
means that each frame is 4 x 8 + 1 =33 bits.
f)The link sends 250frames per second, and each frame contains 33
bits. This means that the data rate of the link is 250 x 33 , or8250
bps. Note that the bit rate of the link is greater than the combined
bit rates of the four channels. If we add the bit rates of four
channels, we get 8000 bps . Because 250frames are travelin gp e r
second and each contains 1extra bit for synchronizing, we need to
add250to the sum to get 8250 bps.
TDM Implementation
Telephone companies implement TDM through a hierarchy of
digital signals ,c a l l e d digital signal (DS) service ordigital hierarc hy.
Figure 5.19 shows the data rates supported by each level.
Figure 5.19: Digital Hierarchy
ADS-0service is a single digital channel of 64 kbps.
DS-1is a 1.544 -Mbps service; 1.544 Mbps is 24 times 64 kbps plus 8
kbps of overhead. It can be used as a single service for 1.544 -Mbps
transmissions, or it can be used to multiplex 24 DS -0 channels or to
carry any other combination desired by the user that can fit within its
1.544 -Mbps capacity.
DS-2is a 6.312 -Mbps service; 6.312 Mbps is 96 times 64 kbps plus
168 kbps of overhead. It can be used as a single service for 6.312 -
Mbps transmissions; or it can be used to multiplex 4 DS -l channels, 96
DS-0 channels, or a combination of thes e service types.
DS-3is a 44.376 -Mbps service; 44.376 Mbps is 672 times 64 kbps
plus 1.368 Mbps of overhead. It can be used as a single service formunotes.in

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9244.376 -Mbps transmissions; or it can be used to multiplex 7 DS -2
channels, 28 DS -l channels, 672 DS -0 chann els, or a combination of
these service types.
DS-4is a 274.176 -Mbps service; 274.176 is 4032 times 64 kbps plus
16.128 Mbps of overhead. It can be used to multiplex 6 DS -3
channels, 42 DS -2 channels, 168 DS -lchannels, 4032 DS -0 channels,
or a combination of these service types.
Applications of Synchronous TDM
Some second -generation cellular telephone companies use
synchronous TDM. For example, the digital version of cellular telephony
divides the available bandwidth into3D -kHzbands. For each band, TDM is
applied so that six users can share the band. This means that each 3D -kHz
band is now made of six time slots, and the digitized voice signals of the
users are inserted in the slots. Using TDM, the number of telephone users
ineach area is now 6 times greater.
5.2.4.2 Statistical Time -Division Multiplexing
As we saw in synchronous TDM, each input has a reserved slot in
the output frame. This can be inefficient if some input lines have no data
to send. In statistical time -division multiplexing, slots are dynamically
allocated to improve bandwidth efficiency. Only when an input line has a
slot’s worth of data to send is it given a slot in the output frame. In
statistical multiplexing, the number of slots in each frame is less than the
number of input lines. The multiplexer checks each input line in round
robin fashion; it allocates a slot for an input line if the line has data to
send; otherwise, it skips the line and checks the next line.
Figure 5.20 shows a synchronous and a statistical TDM example. In the
former, some slots are empty because the corresponding lin e does not have
data to send. In the latter, however, no slot is left empty as long as there
are data to be sent by any input line.
munotes.in

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93Figure 5.20 TDM slot comparison
Addressing
Figure 5.20 also shows a major difference between slots in
synchronous TDM and statistical TDM. An output slot in synchronous
TDM is totally occupied by data; in statistical TDM, a slot needs to carry
data as well as the address of the destination.
In synchronous TDM, there is no need for addressing;
synchronization and pre assigned relationships between the inputs and
outputs serve as an address. We know, for example, that input 1 always
goes to input 2. If the multiplexer and the de -multiplexer are
synchronized, this is guaran teed. In statistical multiplexing, there is no
fixed relationship between the inputs and outputs because there are no pre
assigned or reserved slots. We need to include the address of the receiver
inside each slot to show where it is to be delivered. The a ddressing in its
simplest form can be nbits to define Ndifferent output lines with n=
log2N. For example, for eight different output lines, we need a3 -bit
address.
Slot Size
Since a slot carries both data and an address in stat istical TDM, the
ratio of the data size to address size must be reasonable to make
transmission efficient. For example, it would be inefficient to send 1 bit
per slot as data when the address is 3 bits. This would mean an overhead
of 300 percent. In statis tical TDM, a block of data is usually many bytes
while the address is just a few bytes.
No Synchronization Bit
There is another difference between synchronous and statistical
TDM, but this time it is at the frame level. The frames in statistical TDM
need not be synchronized, so we do not need synchronization bits.
Bandwidth
In statistical TDM, the capacity of the link is normally less than the
sum of the capacities of each channel. The designers of statistical TDM
define the capacity of the link based on the statistics of the load for each
channel. If on average only xpercent of the input slots are filled, the
capacity of the link reflects this. Of course, during peak times, some slots
need to wait.
5.3 SPREAD SPECTRUM
Multiplexing combines signals from several sources to achieve
bandwidth efficiency; the available bandwidth of a link is divided between
the sources. In Spread Spectrum, we also combine signals from different
sources to fit into a larger bandwidth, but our goals are somewhat
different. Spr ead spectrum is designed to be used in wireless applications
(LANs and WANs). In these types of applications, we have some concerns
that outweigh bandwidth efficiency. In wireless applications, all stationsmunotes.in

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94use air (or a vacuum) as the medium for communica tion. Stations must be
able to share this medium without interception by an eavesdropper and
without being subject to jamming from a malicious intruder.
To achieve these goals, spread spectrum techniques add
redundancy; they spread the original spectrum n eeded for each station. If
the required bandwidth for each station is B,spread spectrum expands it to
BSSsuch that BSS>B . The expanded bandwidth allows the source to wrap
its message in a protective envelope for a more secure transmission.
An analogy i s the sending of a delicate, expensive gift. We can
insert the gift in a special box to prevent it from being damaged during
transportation, and we can use a superior delivery service to guarantee the
safety of the package. Figure 5.21 shows the idea of sp read spectrum.
Spread spectrum achieves its goals through two principles:
1.The bandwidth allocated to each station needs to be, by far, larger
than what is needed. This allows redundancy.
2.The expanding of the original bandwidth Bto the bandwidth BSS
must b e done by a process that is independent of the original
signal. In other words, the spreading process occurs after the signal
is created by the source.
Figure 5.21: Spread Spectrum
After the signal is created by the source , the spreading process uses
a spreading code and spreads the bandwidth. The figure shows the original
bandwidth Band the spreaded bandwidth BSS.The spreading code is a
series of numbers that look random, but are actually a pattern.
5.3.1 Types of Sprea dS p e c t r u m
There are two techniques to spread the bandwidth: frequency
hopping spread spectrum (FHSS) and direct sequence spread spectrum
(DSSS).
5.3.2 Frequency Hopping Spread Spectrum (FHSS)
The frequency hopping spread spectrum (FHSS) technique uses M
different carrier frequencies that are modulated by the source signal. At
one moment, the signal modulates one carrier frequency; at the next
moment, the signal modulates another carrier frequency. Although the
modulation is done using one carrier frequenc ya tat i m e , Mfrequenciesmunotes.in

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95are used in the long run. The bandwidth occupied by a source after
spreading is BpHSS>B.
Figure 5.22 shows the general layout for FHSS. A pseudorandom code
generator, called Pseudorandom Noise (PN), creates a k-bitpattern for
every hopping period Th. The frequency table uses the pattern to find the
frequency to be used for this hopping period and passes it to the frequency
synthesizer. The frequency synthesizer creates a carrier signal of that
frequency, and the source signal modulates the carrier signal.
Figure 5.22: Frequency Hopping Spread Spectrum (FHSS)
Suppose we have decided to have eight hopping frequencies. This is
extremely low for real applications and is just for illustration. In this case,
Mis8and kis3. The pseudorandom code generator will create eight
different 3 -bit patterns. These are mapped to eight different frequencies in
the frequency table (see Figure 5.23).
Figure 5.23: Frequency selection in FHSS
The pattern for this station is 101, 111, 001, 000, 010, all, 100.
Note that the pattern is pseudorandom it is repeated after eight hoppings.
This means that at hopping period 1, the pattern is 101. The frequency
selected is 700 kHz; the source signal modulates this carrier frequency.
The second k-bitpattern selected is 111, which selects the 900 -kHz
carrier; the eighth pattern is 100, the frequency is 600 kHz. After eight
hoppings, the pattern repeats, starting from 101 again. Figure 5.24 showsmunotes.in

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96how the signal hops around from carrier to carrier. We assume the
required bandwidth of the original signal is 100 kH z.
Figure 5.24: FHSS Cycles
It can be shown that this scheme can accomplish the previously
mentioned goals. If there are many k -bit patterns and the hopping period is
short, a sender and receiver can have privacy. If an in truder tries to
intercept the transmitted signal, she can only access a small piece of data
because she does not know the spreading sequence to quickly adapt
herself to the next hop. The scheme has also an anti jamming effect. A
malicious sender may be abl e to send noise to jam the signal for one
hopping period (randomly), but not for the whole period.
Bandwidth Sharing
If the number of hopping frequencies is M,we can multiplex M
channels into one by using the same BSSbandwidth. This is possible
because a station uses just one frequency in each hopping period; M-1
other frequencies can be used by other M-1stations. In other words, M
different stations can use the same BSSif an appropriate modulation
technique suc h as multiple FSK (MFSK) is used. FHSS is similar to FDM,
as shown in Figure 5.25.
Figure 5.25 shows an example of four channels using FDM and four
channels using FHSS. In FDM, each station uses 11M of the bandwidth,
but the allocation is fixed; in FHSS, each station uses 11M of the
bandwidth, but the allocation changes hop to hop.
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97Figure 5.25: Bandwidth sharing
5.3.3 Direct Sequence Spread Spectrum (DSSS)
The direct sequence spread spectrum (DSSS) technique also
expands the bandwidth of the original signal, but the process is different.
In DSSS, we replace each data bit with 11 bits using a spreading code. In
other words, each bit is assigned a code of 11 bits, called chips, where the
chip rate is 11 times that of the da ta bit. Figure 5.26 shows the concept of
DSSS.
Figure 5.26: DSSS
As an example, let us consider the sequence used in a wireless
LAN, the famous Barker sequence where nis 11. We assume that the
original signal and the chips in the chip generator use polar NRZ
encoding. Figure 5.27 shows the chips and the result of multiplying the
original data by the chips to get the spread signal.
In Figure 5.27, the spreading code is 11 chips having the pattern
10110111000 (in this case). If the original signal rate is N,the rate of the
spread signal is 11 N.This means that the required bandwidth for the
spread signal is 11 times larger than the bandwidth of the original signal.
The spread signal can provide pri vacy if the intruder does not know the
code. It can also provide immunity against interference if each station uses
a different code.
Figure 5.27: DSSS Example
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98Can we share a bandwidth in DSSS as we did in FHSS? The
answer is no and yes. If we use a spreading code that spreads signals (from
different stations) that cannot be combined and separated, we cannot share
a bandwidth. For example, some wireless LANs use DSSS and the spread
bandwidth cannot be shared. However, if we use a special type of
sequence code that allows the combining and separating of spread signals,
we can share the bandwidth. A special spreading code allows us to use
DSSS in cellular telephony and share a bandwidth between several users.
5.4 SUMMARY
Bandwidth utilization is the use of available bandwidth to achieve
specific goals. Efficiency can be achieved by using multiplexing; privacy
and ant jamming can be achieved by using spreading.
Multiplexing is the set of techniques that allows the simultaneous
transmission of multiple signals across a single data link. In a
multiplexed system, nlines share the bandwidth of one link. The word
link refers to the physical path. The word channel refers to the portion
of a link that carries a transmission. There are three basic multiplexing
techniques: frequency -division multiplexing, wavelength -division
multiplexing, and time -division multiplexing. The first two are
techniques designed for analog signal s, the third, for digital signals.
Frequency -division multiplexing (FDM) is an analog technique that
can be applied when the bandwidth of a link (in hertz) is greater than
the combined bandwidths of the signals to be transmitted.
Wavelength -division multi plexing (WDM) is designed to use the high
bandwidth capability of fiber -optic cable. WDM is an analog
multiplexing technique to combine optical signals.
Time -division multiplexing (TDM) is a digital process that allows
several connections to share the high bandwidth of a link. TDM is a
digital multiplexing technique for combining several low -rate channels
into one high -rate one.
We can divide TDM into two different schemes: synchronous or
statistical. In synchronous, each input connection has an allotment i n
the output even if it is not sending data. In statistical TDM, slots are
dynamically allocated to improve bandwidth efficiency.
In Spread Spectrum (SS), we combine signals from different sources
to fit into a larger bandwidth. Spread spectrum is designed to be used
in wireless applications in which stations must be able to share the
medium without interception by an eavesdropper and without being
subject to jamming from a malicious intruder.
The Frequency Hopping Spread Spectrum (FHSS) technique uses M
different carrier frequencies that are modulated by the source signal.
At one moment, the signal modulates one carrier frequency; at the next
moment, the signal modulates another carrier frequency.munotes.in

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99The Direct Sequence Spread Spectrum (DSSS) technique expands the
bandwidth of a signal by replacing each data bit with n bits using a
spreading code. In other words, each bit is assigned a code of n bits,
called chips.
5.5 REFERENCE FOR FURTHER READING
For more details about topics discussed in this chapter, we
recommend the following books.
1.Data Communication and Networking by Behrouz A. Forouzan ,
McGraw -Hill, 2007.
2.Basic Communication Theory byJ. E. Pearson, Prentice Hall, 1992.
3.Digital and Analog Communication Systems by L.W. Couch, Prentice
Hall, 2001.
4.Digital Baseband and Transmission and Recording by J. Bergman,
Kluwer Academic, 1996.
5.Data and Computer Communications by W. Stallings, Prentice Hall,
2004.
5.6 MODEL QUESTIONS
1.Describe the purpose of multiplexing.
2.What are the three main multiplexing techniques?
3.What is difference between a link and a channel in multiplexing?
4.Which of the three multiplexing techniques is common for fiber optic
links? Explain the reason.
5.Differentiate between multilevel TDM, multipl e slot TDM, and pulse -
stuffed TDM.
6.Differentiate between synchronous and statistical TDM.
7.Define spread spectrum and its goal. List the two spread spectrum.
8.Define FHSS and explain how it achieves bandwidth spreading.
9.Define DSSS and explain how it achi eves bandwidth spreading.
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1006
TRANSMISSION MEDIA AND
SWITCHING
Unit Structure
6.0 Objectives
6.1 Introduction
6.2 Guided Media –Wired
6.2.1 Twisted –Pair Cable
6.2.2 Coaxial Cable
6.2.3 Fiber –Optic Cable
6.3 Unguided Media –Wireless
6.3.1 Radio Waves
6.3.2 Microwaves
6.3.3 Infrared
6.4 Switching
6.4.1 Circuit –Switching (Circuit Switched Networks)
6.4.2 Packet –Switching (Packet Switched Networks)
6.4.2.1 Datagram switching
6.4.2.2 Virtual –Circuit Switching
6.4.3 Message –Switc hing (Message Switched Networks)
6.5 Structure of Switch
6.5.1Structure of Circuit Switch
6.5.2Structure of Packet Switch
6.6 Summary
6.7 Reference for further reading
6.8 Model Questions
6.0 OBJECTIVES:
This chapter would make you to understand the following concepts:
What is Transmission Medium?
Types of Transmission media –Guided and Unguided.
Types of Guided transmission media.
Types of Unguided transmission media.
Concept of Switching.
Types of switching –Circuit switching, Packet switching and Message
switching.
Structure of a Switch.munotes.in

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1016.1 INTRODUCTION
In this chapter, we are going to discuss transmission media.
Transmission media are actually located below the physical layer and are
directly controlled by the physical laye r. You could say that transmission
media belong to layer zero. Figure 6.1 shows the position of transmission
media in relation to the physical layer.
Figure 6.1: Transmission Medium and Physical layer
A transmiss ionmedium can be broadly defined as anything (either
wire or air) that can carry information from a source to a destination. For
example, the transmission medium for two people having a dinner
conversation is the air. For a written message, the transmissi on medium
might be a mail carrier, a truck, or an airplane. In data communications the
definition of the information and the transmission medium is more
specific. The transmission medium is usually free space, metallic cable, or
fiber-optic cable. The info rmation is usually a signal that is the result of a
conversion of data from another form.
In telecommunications, transmission media can be divided into
two broad categories: guided and unguided. Guided media include
twisted -pair cable, coaxial cable, and fiber -optic cable. Unguided medium
is free space. Figure 6.2 shows this classification.
Figure 6.2: Classification of Transmission media
6.2 GUIDED MEDIA –WIRED
Guided media, which are those that provide a conduit from one
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102cable. A signal traveling along any of these media is directed and
contained by the physical limits of the medium. Twisted -pair and coaxial
cable use metallic (copper) conductors that accept and transport signals in
the form of electric current. Optical fiber is a cable that accepts and
transports signals in the form of light.
6.2.1 Twisted –Pair Cable
A twisted pair consists of two conductors (normally copper), each
with its own plasticinsulation, twisted together, as shown in Figure 6.3.
Figure 6.3: Twisted –pair cable
One of the wires is used to carry signals to the receiver, and the
other is used only as a groun d reference. The receiver uses the difference
between the two. In addition to the signal sent by the sender on one of the
wires, interference (noise) and crosstalk may affect both wires and create
unwanted signals. If the two wires are parallel, the effect of these
unwanted signals is not the same in both wires because they are at
different locations relative to the noise or crosstalk sources (e.g., one is
closer and the other is farther). This results in a difference at the receiver.
By twisting the pairs, a balance is maintained. For example, suppose in
one twist, one wire is closer to the noise source and the other is farther; in
the next twist, the reverse is true. Twisting makes it probable that both
wires are equally affected by external influences (noise or crosstalk). This
means that the receiver, which calculates the difference between the two,
receives no unwanted signals. The unwanted signals are mostly canceled
out. Following Figure 6.4 shows how twisting of wires reduces the
crosstalk and outs ide interference. From the above discussion, it is clear
that the number of twists per unit of length (e.g., inch) has some effect on
the quality of the cable.
Figure 6.4: Twisting of wires reduces the cross talk a nd outside
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103Unshielded Vs Shielded Twisted -Pair Cable
The most common twisted -pair cable used in communications is
referred to as Unshielded Twisted -pair (UTP). IBM has also produced a
version of twisted -pair cable for its use called Shielded T wisted -pair
(STP). STP cable has a metal foil or braided mesh covering that encases
each pair of insulated conductors. Although metal casing improves the
quality of cable by preventing the penetration of noise or crosstalk, it is
bulkier and more expensive . Figure 6.5 shows the difference between UTP
and STP. Our discussion focuses primarily on UTP because STP is seldom
used outside of IBM.
Figure 6.5: UTP and STP cable
Categories
The Electronic Industries Association (EI A) has developed
standards to classify unshielded twisted -pair cable into seven categories.
Categories are determined by cable quality, with 1 as the lowest and 7 as
the highest. Each EIA category is suitable for specific uses. Table 6.1
shows these catego ries.
Category Bandwidth
(MHz)Maximum data
rateApplication
CAT1 <1 < 100 Kbps Telephone Lines
CAT2 4 4 Mbps IBM Token ring LANs
CAT3 1616 Mbps
(3-4 twists / foot)10 Base -TL A N s
Currently used in Telephone
Lines
CAT4 20 20 Mbps 16 Mbps Token ring LANs
CAT5 100100 Mbps
1000 Mbps (4 pairs)
(3-4 twists / inch)100 Base –T (Fast Ethernet)
155 Mbps ATM Network
&Gigabit Ethernet
CAT5E 100100 Mbps
1000 Mbps (4
pairs)100 Base –T (Fast Ethernet)
155 Mbps ATM Network
&Gigabit Ethernet
CAT6 200-250 1 Gbps Gigabit Ethernet
CAT7 600 1 GbpsGigabit Ethernet
(over long distance than CAT6)
Table 6.1: Categories of UTP cablemunotes.in

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104Connectors
The most common UTP connector is RJ45 (RJ stands for
registered jack), as shown in Figure 6.6. The RJ45 is a keyed connector,
meaning the connector can be inserted in only one way.
Figure 6.6: UTP –RJ-45 Connector
Applications
Twisted -pair cables are used in te lephone lines to provide voice
and data channels. Local -area networks, such as 10BaseT and l00Base -T,
also use twisted -pair cables.
6.2.2 Coaxial Cable
Consist of two conductors’ shares the same axis hence called as
coaxial. A solid copper wire runs down the center of the cable, surrounded
by insulator (PVC –Poly Vinyl Chloride), surrounded by second
conductor (shield), which is further surrounded by insulator and thick
plastic jacket forms the cover of the cable as shown in Figure 6.7.
Figure 6.7: Coaxial cable
Coaxial Cable Standards
Coaxial cables are classified by their radio government (RG) ratings and
cable’s resistance to DC (Direct current) and AC (Alternating current)
measured in Ω(ohms). Following Table 6.2 shows the categories of
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105Category Impedance Application
RG-8 and
RG-1150Ω10Base5 –Thick Ethernet
RG-58 50Ω10Base2 –Thin Ethernet
RG-59 75ΩCable TV
RG-62 93ΩARCnet (Attached Resource
Network)
Table 6.2: Categories of Coaxial cable
Connectors
To connect coaxial cable to devices, we need coaxial connectors. The
most common type of connector used today is the Bayone -Neill -
Concelman (BNe), connector. Figure 6.8 shows three popular types of
these co nnectors: the BNC connector, the BNC T connector, and the BNC
terminator. The BNC connector is used to connect the end of the cable to a
device, such as a TV set. The BNC T connector is used in Ethernet
networks to branch out to a connection to a computer or other device. The
BNC terminator is used at the end of the cable to prevent the reflection of
the signal.
Figure 6.8: BNC connectors
Applications
Coaxial cable was widely used in analog telephone networks
where a single coaxial network could carry 10,000 voice signals. Later it
was used in digital telephone networks where a single coaxial cable could
carry digital data up to 600 Mbps. However, coaxi al cable in telephone
networks has largely been replaced today with fiber -optic cable.
Cable TV networks also use coaxial cables. In the traditional cable
TV network, the entire network used coaxial cable. Later, however, cable
TV providers replaced most of the media with fiber -optic cable; hybrid
networks use coaxial cable only at the network boundaries, near the
consumer premises.
6.2.3 Fiber –Optic Cable
A fiber -optic cable is made of glass or plastic and transmits signals
in the form of light. To un derstand optical fiber, we first need to explore
several aspects of the nature of light. Light travels in a straight line as long
as it is moving through a single uniform substance. If a ray of light
traveling through one substance suddenly enters another substance (of amunotes.in

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106different density), the ray changes direction. Figure 6.9 shows how a ray
of light changes direction when going from a more dense to a less dense
substance.
Figure 6.9: Bending of Light rays
As the figure shows, if the angle of incidence Iis less than the critical
angle, the ray refracts and moves closer to the surface. If the angle of
incidence is equal to the critical angle, the light bends along the interface.
If the angle is greater than the critical angle, the ray reflects (makes a turn)
and travels again in the denser substance. Note that the critical angle is a
property of the substance, and its value differs from one substance to
another.
Optical fibers use reflection to guide light through a channel. A
glass or plastic core is surrounded by a cladding of less dense glass or
plastic. The difference in density of the two materials must be such that a
beam of light moving through the core is reflected off the c ladding instead
of being refracted into it as shown in the following Figure 6.10.
Figure 6.10: Optical –fiber
Propagation Modes
Current technology supports two modes (Multimode and Single
mode) for propagating light alo ng optical channels, each requiring fiber
with different physical characteristics. Multimode can be implemented in
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107
Figure 6.11: Propagation mode –classification
Multimode
It is called Multimode because multiple beams from a light source
move through the core in different paths. How these beams move within
the cable depends on the structure of the core, as shown in Figure 6.12.
In Multimode Step -index fiber, the den sity of the core remains
constant from the center to the edges. A beam of light moves through this
constant density in a straight -line until it reaches the interface of the core
and the cladding. At the interface, there is an abrupt change due to a lower
density; this alters the angle of the beam’s motion. The term step index
refers to the suddenness of this change, which contributes to the distortion
of the signal as it passes through the fiber.
A second type of fiber, called Multimode Graded -index fiber,
decreases this distortion of the signal through the cable. A Graded -index
fiber, therefore, is one with varying densities. Density is highest at the
center of the core and decreases gradually to its lowest at the edge. Figure
6.12 shows the impact of this variable density on the propagation of light
beams.
Single -Mode
Single -mode uses Step -index fiber and a highly focused source of
light that limits beams to a small range of angles, all close to the
horizontal. The single mode fibers itself is manufactur ed with a much
smaller diameter than that of multimode fiber, and with substantially
lower density. The decrease in density results in a critical angle that is
close enough to 90° to make the propagation of beams almost horizontal.
In this case, propagatio n of different beams is almost identical, and delays
are negligible. All the beams arrive at the destination “together” and can
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108
Figure 6.12: Propagation modes
Optical Fiber –Types
Optical fibers are defined by the ratio of the diameter of their core to the
diameter of their cladding, both expressed in micrometers. The common
sizes are shown in Table 6.3.
Table 6.3: Optical Fiber –types
Cable Composition
Figure 6.13 shows the composition of a typical fiber -optic cable. The outer
jacket is made of either PVC or Teflon. Inside the jacket are Kevlar
strands to strengthen the cable. Kevlar is a stro ng material used in the
fabrication of bulletproof vests. Below the Kevlar is another plastic
coating to cushion the fiber. The fiber is at the center of the cable, and it
consists of cladding and core.
Figure 6.13: Optica lF i b e r –Compositionmunotes.in

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109Optical Fiber Cable –Connectors
There are three types of connectors for fiber -optic cables, as shown
in Figure 6.14. The Subscriber Channel (SC) connector is used for cable
TV. It uses a push/pull locking system. The Straight -Tip(ST) connector
is used for connecting cable to networking devices. It uses a bayonet
locking system and is more reliable than SC. The Mechanical Transfer -
Registered Jack ( MT-RJ)is a connector that is the same size as RJ45 .
Figure 6.14: Optical Fiber Cable –Connectors
Applications
Fiber -optic cable is often found in backbone networks (The
SONET network) because its wider band width is cost -effective. Today,
with wavelength -division multiplexing (WDM), we can transfer data at a
rate of 1600 Gbps.
Some cable TV companies use a combination of optical fiber and
coaxial cable, thus creating a hybrid network. Optical fiber provides the
backbone structure while coaxial cable provides the connection to the user
premises.
LANs such as 100Base -FX network (Fast Ethernet) and 1000Base -X
(Gigabit Ethernet) also use fiber -optic cable.
Advantages of Optical Fiber
Fiber -optic cable has sever al advantages over metallic cable (twisted
pair or coaxial)
Higher bandwidth : Fiber -optic cable can support dramatically higher
bandwidths(data rates) than either twisted -pair or coaxial cable.
Currently, data rates and bandwidth utilization over fiber -optic cable
are limited not by the medium but by the signal generation and
reception technology available.
Less signal attenuation : Fiber -optic transmission distance is
significantly greater than that of other guided media. A signal can run
for 50 km without requiring regeneration. We need repeaters every 5
km for coaxial or twisted -pair cable.
Immunity to electromagnetic interference : Electromagnetic noise
cannot affect fiber -optic cables.
Resistance to corrosive materials : Glass is more resistant to
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110Light weight : Fiber -optic cables are much lighter than copper cables.
Greater immunity to tapping : Fiber -optic cables are more immune to
tapping than copper cables.
Disadvantages of Optical Fiber
There are some disadvantages in the use of optical fiber.
Installation and maintenance : Fiber -optic cable is a relatively new
technology. Its installation and maintenance require expertise that is
not yet available everywhere.
Unidirectional light propagation : Propagation of light is
unidirectional. If we need bidirectional communication, two fibers are
needed.
Cost: The cable and the interfaces are relatively more expensive than
those of other guided media
6.3 UNGUIDED MEDIA –WIRELESS
Unguided media transport electromagnetic waves without using a
physical conductor. This type of communication is often referred to as
wireless communication. Signals are normally broadcast through free
space and thus are available to anyone who has a device capable of
receiving them. Figure 6.15 shows the part of the electromagnetic
spectrum, ranging from 3 kHz to 900 THz, used for wireless
communication.
Figure 6.15: Electromagnetic Spectrum for Wireless commu nication
Propagation Methods
Unguided signals can travel from the source to destination in
several ways: ground propagation, sky propagation, and line -of-sight
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111
Figure 6.16: Propagation methods
In Ground propagation, radio waves travel through the lowest
portion of the atmosphere, hugging the earth. These low -frequency signals
emanate in all directions from the transmitting antenna and follow the
curvature of the planet. Distance depends on the amount of power in the
signal: The greater the power, the greater the distance.
In Sky propagation, higher -frequency radio waves radiate upward
into the ionosphere (the layer of atmosphere where particles exist as ions)
where they are reflected back to earth. This type of transmission allows for
greater distances with lower output power.
In Line -or-sight propagation, very high -frequency signals are
transmitted in straight -line directly from antenna to antenna. Antennas
must be directional, facing each other and either tall enough or close
enough together not to be affected by the curvature of the earth. Line -of-
sight propagation is tricky because radio transmissions cannot be
completely focused.
The section of the electro magnetic spectrum defined as radio
waves and microwaves is divided into eight ranges, called bands ,each
regulated by government authorities. These bands are rated from Very Low
Frequency (VLF) to Extremely High Frequency (EHF).Table 6.4 lists
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112
Table 6.4: Bands
We can divide wireless transmission into three broad groups:
Radio waves, Microwaves, and Infrared waves.
6.3.1 Radio Waves
Although t here is no clear -cut demarcation between radio waves
and microwaves, electromagnetic waves ranging in frequencies between 3
kHz and 1 GHz are normally called Radio waves; waves ranging in
frequencies between 1 and 300 GHz are called microwaves.
However, the behavior of the waves, rather than the frequencies, is
a better criterion for classification.
Radio waves, for the most part, are omni directional. When an
antenna transmits radio waves, they are propagated in all directions. This
means that the sending and receiving antennas do not have to be aligned.
The omni directional property has a disadvantage, too. The radio waves
transmitted by one antenna are susceptible to interference by another
antenna that may send signals using the same frequenc y or band.
Radio waves, particularly those waves that propagate in the sky
mode, can travel long distances. This makes radio waves a good candidate
for long -distance broadcasting such as AM radio.
Radio waves, particularly those of low and medium frequen cies,
can penetrate walls. This characteristic can be both an advantage and a
disadvantage. It is an advantage because, for example, an AM radio can
receive signals inside a building. It is a disadvantage because we cannot
isolate a communication to just i nside or outside a building. The radio
wave band is relatively narrow, just under 1 GHz, compared to the
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113Omni directional Antenna
Radio waves use omni directional antennas that send out signals in
all directions. Based on the wavelength, s trength, and the purpose of
transmission, we can have several types of antennas. Figure 6.17 shows an
omni directional antenna.
Figure 6.17: Omni directional antenna
Applications
The omni directional characteristics of ra dio waves make them
useful for multicasting, in which there is one sender but many receivers.
AM and FM radio, television, maritime radio, cordless phones, and paging
are examples of multicasting.
6.3.2 Microwaves
Electromagnetic waves having frequencies between 1 and 300 GHz
are called microwaves. Microwaves are unidirectional. When an antenna
transmits microwave waves, they can be narrowly focused. This means
that the sending and receiving antennas need to be alig ned. The
unidirectional property has an obvious advantage. A pair of antennas can
be aligned without interfering with another pair of aligned antennas. The
following describes some characteristics of microwave propagation:
Microwave propagation is line -of-sight. Since the towers with the
mounted antennas need to be in direct sight of each other, towers that
are far apart need to be very tall. Repeaters are often needed for long
distance communication.
Very high -frequency microwaves cannot penetrate walls. T his
characteristic can be a disadvantage if receivers are inside buildings.
The microwave band is relatively wide, almost 299 GHz. Therefore
wider sub -bands can be assigned, and a high data rate is possible.
Use of certain portions of the band requires pe rmission from
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114Unidirectional Antenna
Microwaves need unidirectional antennas that send out signals in one
direction. Two types of antennas are used for microwave communications:
the parabolic dish and the horn (see Figure 6.18).
Figure 6.18: Unidirectional antenna
A parabolic dish antenna is based on the geometry of a parabola:
Every line parallel to the line of symmetry (line of sight) reflects off the
curve at angles such that all the lines intersect in a common point called
the focus. The parabolic dish works as a funnel, catching a wide range of
waves and directing them to a common point. In this way, more of the
signal is recovered than would be possible with a single -point receiver.
Outgoing transmissi ons are broadcast through a horn aimed at the dish.
The microwaves hit the dish and are deflected outward in a reversal of the
receipt path.
A horn antenna loo ks like a gigantic scoop. Outgoing
transmissions are broadcast up a stem (resembling a handle) and deflected
outward in a series of narrow parallel beams by the curved head. Received
transmissions are collected by the scooped shape of the horn, in a manner
similar to the parabolic dish, and are deflected down into the stem.
We can categories the Microwave systems into two categories such as
terrestrial -microwave and satellite -microwave systems.
Terrestrial Microwaves
Terrestrial microwave uses parabolic d ish, signals are highly
focused and line of sight is maintained between sender and receiver (see
Figure 6.19). It is used in Long haul telecommunications.
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115Satellite Microw aves
In this, satellite acts as a relay station (repeater). Satellite receives
signals on one frequency (uplink) from sending earth station, amplifies or
repeats the signals and transmits on another frequency (downlink) to the
receiving earth station. Sate llite need to be positioned in the
geosynchronous orbit (height of 36,000 km from earth) with the help of
space shuttle. Geosynchronous means stationary with respect to the earth,
hence the position of particular Earth Station with respect to the satellite
remains constant at all times(see Figure 6.20). It is used in Television,
Long distance telephone and Private business networks.
Figure 6.20: Satellite Microwave system
Microwaves Applications
Microwaves, due to their unidirectional properties, are very useful
when unicast (one-to-one) communication is needed between the sender
and the receiver. They are used in cellular phones, satellite networks, and
wireless LANs.
6.3.3 Infrared
Infrared wav es, with frequencies from 300 GHz to 400 THz
(wavelengths from 1 mm to 770 nm), can be used for short -range
communication. Infrared waves, having high frequencies, cannot penetrate
walls. This advantageous characteristic prevents interference between one
system and another; a short -range communication system in one room
cannot be affected by another system in the next room. When we use our
infrared remote control, we do not interfere with the use of the remote by
our neighbors. However, this same characteri stic makes infrared signals
useless for long -range communication. In addition, we cannot use infrared
waves outside a building because the sun’s rays contain infrared waves
that can interfere with the communication.
Applications
The infrared band, almost 400 THz, has an excellent potential for
data transmission. Such a wide bandwidth can be used to transmit digital
data with a very high data rate. TheInfrared Data Association (IrDA), an
association for sponsoring the use of infra red waves, has established
standards for using these signals for communication between devices such
as keyboards, mice, PCs, and printers. For example, some manufacturers
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116to commun icate with a PC. The standard originally defined a data rate of
75 kbps for a distance up to 8 m. The recent standard defines a data rate of
4 Mbps.
Infrared signals defined by IrDA transmit through line of sight; the
IrDA port on the keyboard needs to po int to the PC for transmission to
occur.
6.4 SWITCHING
A network is a set of connected devices. Whenever we have
multiple devices, we have the problem of how to connect them to make
one-to-one communication possible. One solution is to make a point -to-
point connection between each pair of devices (a mesh topology) or
between a central device and every other device (a star topology). These
methods, however, are impractical and wasteful when applied to very
large networks.
A better solution is switching. A switched network consists of a
series of interlinked nodes, called switches. Switches are devices capable
of creating temporary connections between two or more devices linked to
the switch. In a switched network, some of these nodes are connected to
the e nd systems (computers or telephones). Others are used only for
routing. Figure 6.21 shows a switched network.
Figure 6.21: Switched network
Traditionally, three methods of switching have been important:
circuit switching, packet switching, and message switching. The first two
are commonly used today. The third has been phased out in general
communications but still has networking applications. We can then divide
today’s networks into three broad categories: circuit -switched networks,
packet -switched networks, and message -switched. Packet -switched
networks can further be divided into two subcategories -virtual -circuit
networks and datagram networ ksasshown in Figure 6.22.munotes.in

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Figure 6.22: Classification of Switched networks
6.4.1 Circuit –Switching (Circuit Switched Networks)
A circuit -switched network consists of a set of switches connected
by physical links. A connection between two stations is a dedicated path
made of one or more links. However, each connection uses only one
dedicated channel on each link. Each link is normally divided into n
channels by using FDM or TDM as discussed in Chapter 5.Figure 6.23
shows a trivial circuit -switched network with four switches and four links.
Each link is divided into n( n is 3 in the figure) channels by using FDM or
TDM.
Figure 6.23: Trivial Circuit -switched network
We have explicitly shown the multiplexing symbols to emphasize the
division of the link into channels even though multiplexing can be
implicitly included in the switch fabric. The end systems, such as
computers or telephones, are directly connected to a switch. We have
show n only two end systems for simplicity. When end system A needs to
communicate with end system M, system A needs to request a connection
to M that must be accepted by all switches as well as by M itself. This is
called the setup phase; a circuit (channel) i s reserved on each link, and the
combination of circuits or channels defines the dedicated path. After the
dedicated path made of connected circuits (channels)is established, data
transfer can take place. After all data have been transferred, the circuits are
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118We need to emphasize several points here:
Circuit switching takes place at the physical layer.
Before starting communication, the stations must make a
reservation for the resources to be used during the communication.
These resources, such as channels (bandwidth in FDM and time
slots in TDM), switch buffers, switch processing time, and switch
input/ output ports, must remain dedicated during the entire
duration of data transfer until the teardown phase.
Data transferred between the two stations are not packetized
(physical layer transfer of the signal). The data are a continuous
flow sent by the sour ce station and received by the destination
station, although there may be periods of silence.
There is no addressing involved during data transfer. The switches
route the data based on their occupied band (FDM) or time slot
(TDM). Of course, there is end -to-end addressing used during the
setup phase.
Example 6.1:
As a trivial example, let us use a circuit -switched network to
connect eight telephones in a small area. Communication is through 4 -kHz
voice channels. We assume that each link uses FDM to connec ta
maximum of two voice channels. The bandwidth of each link is then 8
kHz. Figure 6.24 shows the situation. Telephone 1 is connected to
telephone 7; 2 to 5 ;3 to 8; and 4 to 6. Of course the situation may change
when new connections are made. The switch controls the connections.
Figure 6.24: Example 6.1 (Circuit -switched network)
Example 6.2:
As another example, consider a circuit -switched network that connects
computers in two remote offices of a private company. The of fices are
connected using a T -l line leased from a service provider. There are two 4
X 8 (4 inputs and 8 outputs) switches in this network. For each switch,
four output ports are folded into the input ports to allow communication
between computers in the s ame office. Four other output ports allow
communication between the two offices. Figure 6.25 shows the situation.munotes.in

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119
Figure 6.25: Example 6.2 (Circuit -switched network)
Connection Phases
The actual communication in a circuit -switched network requires
three phases: connection setup, data transfer, and connection teardown.
Connection setup phase
Before the two parties can communicate, a dedicated circuit
(combination of channels in links) needs to be established. The end
systems are normally connected through dedicated lines to the switches, so
connection setup means creating dedicated channels be tween the switches.
For example, in Figure 6.23, when system A needs to connect to system
M, it sends a setup request that includes the address of system M, to
switch I. Switch I finds a channel between itself and switch IV that can be
dedicated for this p urpose. Switch I then sends the request to switch IV,
which finds a dedicated channel between itself and switch III. Switch III
informs system M of system A’s intention at this time.
In the next step to making a connection, an acknowledgment from
system M needs to be sent in the opposite direction to system A. Only
after system A receives this acknowledgment is the connection
established. Note that end -to-end addressing is required for creating a
connection between the two end systems. These can be, for ex ample, the
addresses of the computers assigned by the administrator in a TDM
network, or telephone numbers in an FDM network.
Data Transfer phase
After the establishment of the dedicated circuit (channels), the two
parties can transfer data.
Connection t eardown phase
When one of the parties needs to disconnect, a signal is sent to
each switch to release the resources.
Efficiency
It can be argued that circuit -switched networks are not as efficient
as the other two types of networks because resources are a llocated during
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120other connections. In a telephone network, people normally terminate the
communication when they have finished their conversation. However, in
computer networks, a co mputer can be connected to another computer
even if there is no activity for a long time. In this case, allowing resources
to be dedicated means that other connections are deprived.
Delay
Although a circuit -switched network normally has low efficiency,
the delay in this type of network is minimal. During data transfer the data
are not delayed at each switch; the resources are allocated for the duration
of the connection.
Circuit -switching in Telephone networks
Switching at the physical layer in the tradi tional telephone network
uses the circuit -switching approach.
6.4.2 Packet –Switching (Packet Switched Networks)
In data communications, we need to send messages from one end
system to another. If the message is going to pass through a packet -
switched network, it needs to be divided into packets of fixed or variable
size. The size of the packet is determined by the network and the
governing protocol. Each packet contains data and head er (priority, source
and destination address). In packet switching, there is no resource
allocation for a packet. This means that there is no reserved bandwidth on
the links, and there is no scheduled processing time for each packet.
Resources are allocate d on demand. The allocation is done on a first -
come, first -served basis. When a switch receives a packet, no matter what
is the source or destination, the packet must wait if there are other packets
being processed. Figure 6.26 shows the Packet -switched ne twork.
Figure 6.26: Packet -switched network
Packet -switched networks can further be divided into two
subcategories Virtual -circuit networks and Datagram networks.
6.4.2.1 Datagram switching (Datagram networks)
In a datagram network, each packet is treated independently from
all other packets and is referred as datagram packet. Datagram packets
belongs to same message may go by different paths to reach at the
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121layer reorders them at the destination. Datagram switching is normally
done at the network layer.
Figure 6.27 shows how the datagram approach is used to deliver four
packets from station A to station X.
Figure 6.27: Datagram network
The datagram networks are sometimes referred to as connectionless
networks. The term connectionless here means that the switch (packet
switch) does not keep information about the c onnection state. There are no
connection setup or teardown phases. Each packet is treated the same by a
switch regardless of its source or destination.
Efficiency
The efficiency of a datagram network is better than that of a
circuit -switched network; reso urces are allocated only when there are
packets to be transferred. If a source sends a packet and there is a delay of
a few minutes before another packet can be sent, the resources can be
reallocated during these minutes for other packets from other source s.
Delay
There may be greater delay in a datagram network than in a
virtual-circuit network. Although there are no setup and teardown phases,
each packet may experience a wait at a switch before it is forwarded. In
addition, since not all packets in a message necessarily travel through the
same switches, the delay is not unif orm for the packets of a message.
Datagram Networks in the Internet
The Internet has chosen the datagram approach to switching at the
network layer. It uses the universal addresses (IP addresses) defined in the
network layer to route packets from the sour ce to the destination.
6.4.2.2 Virtual –Circuit Switching ( Virtual -circuit networks)
A virtual -circuit network is a cross between a circuit -switched
network and a datagram network. It has some characteristics of both.munotes.in

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1221.As in a circuit -switched network, t here are setup and teardown phases
in addition to the data transfer phase.
2.Resources can be allocated during the setup phase, as in a circuit -
switched network, or on demand, as in a datagram network.
3.As in a datagram network, data are packetized and each p acket carries
an address in the header.
4.As in a circuit -switched network, all packets follow the same path
established during the connection.
5.A virtual -circuit network is normally implemented in the data link
layer, while acircuit -switched network is impl emented in the physical
layer and a datagram network in the network layer. But this may
change in the future.
Figure 6.28 is an example of a virtual -circuit network. The network
has switches that allow traffic from sources to destinations. A source or
destination can be a computer, packet switch, is a device that connects
other networks.
Figure 6.28: Virtual -circuit network
A Virtual -circuit switching further can be implemented in two
ways Switched Virtual -circuit (SVC) and Permanent Virtual -circuit
(PVC).
Switched Virtual -circuit (SVC)
It works like a dial up line in circuit switching. Circuit is
created whenever needed a nd exist only for the duration of specific
exchange. Figure 6.29 shows an example of switched virtual -circuit.munotes.in

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123
Figure 6.29: Switched virtual -circuit (SVC)
Permanent Virtual -circuit (PVC)
It works like a leased line in ci rcuit switching, where same virtual
circuit is provided between two users on a continuous basis. Virtual circuit
is dedicated to specific users and no one else can use it and can be used
without connection establishment and connection termination. Figure 6 .30
shows an example of permanent virtual -circuit.
Figure 6.30: Permanent Virtual -circuit (PVC)
Efficiency in Virtual -Circuit Networks
As we said before , resource reservation in a virtual -circuit network
can be made during the setup or can be on demand during the data transfer
phase. In the first case, the delay for each packet is the same; in the second
case, each packet may encounter different delays.
Delay in Virtual -Circuit Networks
In a virtual -circuit network, there is a one -time delay for setup and
a one -time delay for teardown. If resources are allocated during the setup
phase, there is no wait time for individual packets.
Circuit -Switched Techn ology in WANs
Virtual -circuit networks are used in switched WANs such as
Frame Relay and ATM networks. The data link layer of these technologies
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1246.4.3 Message –Switching (Message Switched Networks)
The t hird method of switching is a message switching; which has
been phased out in general communications but still has networking
applications. It was b est known by the term store and forward and most
common in the 1960s and 1970s.
In message -switching when as witching node receives a message,
stores it until the appropriate route is free, then it sends along and the
messages were stored and relayed from its secondary storage
(disk).Requirement of large capacity storage media at each switching node
was the ma jor disadvantage of this method. Figure 6.31 shows the
example of message switching.
Figure 6.31: Message switching
6.5 STRUCTURE OF SWITCH
We use switches in circuit -switched and packet -switched networks.
In this secti on, we discuss the structures of the switches used in each type
of network.
6.5.1 Structure of Circuit Switch
Circuit switching today can use either of two technologies: the
space -division switch or the time -division switch.
Space -Division Switch
In space -division switching, the paths in the circuit are separated
from one another spatially. This technology was originally designed for
use in analog networks but is used currently in both analog and digital
networks. It has evolved through a long hist oryof many designs.
Crossbar Switch
A crossbar switch connects ninputs to moutputs in a grid; using
electronic micro -switches (transistors) at each cross -point (see Figuremunotes.in

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1256.32). The major limitation of this design is the number of cross -points
required. To connect ninputs to moutputs using a crossbar switch requires
nxmcross -points. For examp le, to connect 100 inputs to 100 outputs
requires a switch with 10000 cross -points. A crossbar switch with this
number of cross -points is impractical. Such a switch is also inefficient
because statistics show that, in practice, fewer than 25 percent of the
cross -points are in use at any given time and the rest are idle.
Figure 6.32: Crossbar switch with 3 inputs and 4 outputs
Multistage Switch
The solution to the limitations of the crossbar switch is the
multistage switch, which combines crossbar switches in several (normally
three) stages, as shown in Figure 6.33. In a single crossbar switch, only
one row or column (one path) is active for any connection. So we need Nx
Ncross -points. If we can allow multiple paths inside the switch, we can
decrease the number of cross -points. Each cross -point in the middle stage
can be accessed by multiple cross -points in the first or third stage.
In a three -stage switch, the total number of cross -points is 2kN + k
(N/n)2which is much smaller than the number of cross -points in a single -
stage switch ( N2).
Figure 6.33: Multistage switch
Example 6.3:
Design a three -stage, 200 x 200 switch (N=200 )w i t h k=4andn=20.munotes.in

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126Solution:
In the first stage we have N/nor10crossbars switches, each of size
20 x 4 . In the second stage, we have 4crossbars switches, each of size 10 x
10. In the third stage, we have 10crossbars switches, each of size 4 x 20 .
The total number of cross -points is 2kN +k(N/n)2or2000 cross -
points. This is 5percent of the number of cross -points in a single -stage
switch ( 200 x 200 = 40,000 ).
Time -Division switch
Time -division switching uses time -division multiplexing (TDM)
inside a switch. The most popular technology is called the time -slot
interchange (TSI). Time -Slot Interchange Figure 6.34 shows a system
connecting four input lines to four output lines. Imagine that each input
line wants to send data to an output line according to the following pattern
for example, 1→3, 2→4, 3→1 and 4 →2.
Figure 6.34: Time -division switch
The figure combines a TDM multiplexer, a TDM de -multiplexer,
and a TSI consisting of random access memory (RAM) with several
memory locations. The size of each location is the same as the size of a
single time slot. The number of locations is the same as the number of
inputs (in most cases, the numbers of inputs and outputs are equal).The
RAM fills up with incoming data from ti me slots in the order received.
Slots are then sent out in an order based on the decisions of a control unit.
6.5.2 Structure of Packet Switch
A switch used in a packet -switched network has a different
structure from a switch used in a circuit -switched ne twork. We can say
that a packet switch has four components: input ports, output ports, the
routing processor, and the switching fabric, as shown in Figure 6.35.munotes.in

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Figure 6.35: Components of Packet -switch
Input Ports
An input port performs the physical and data link functions of the
packet switch. The bits are constructed from the received signal. The
packet is encapsulated from the frame. Errors are detected and corrected.
The packet is now ready to be routed by the n etwork layer. In addition to a
physical layer processor and a data link processor, the input port has
buffers (queues) to hold the packet before it is directed to the switching
fabric. Figure 6.36 shows a schematic diagram of an input port.
Figure 6.36: Input port
Output Port
The output port performs the same functions as the input port, but
in the reverse order. First the outgoing packets are queued, then the packet
is encapsulated in a frame, and finally the physical la yer functions are
applied to the frame to create the signal to be sent on the line. Figure 6.37
shows a schematic diagram of an output port.
Figure 6.37: Output port
Routing Processor
The routing processor performs the f unctions of the network layer.
The destination address is used to find the address of the next hop and, at
the same time, the output port number from which the packet is sent out.
This activity is sometimes referred to as table lookup because the routing
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128function of the routing processor is being moved to the input ports to
facilitate and expedite the process.
Switching Fabrics
The most difficult task in a packet switch is to move the packet
from the input queue to the output queue. The speed with which this is
done affects the size of the input/output queue and the overall delay in
packet delivery. In the past, when a packet switch was actually a dedicated
computer, the memory of the computer or a bus was used as the switching
fabric. The input port stored the packet in memory; the output port
retrieved the packet from memory. Today, packet switches are specialized
mechanisms that use a variety of switching fabrics.
6.6 SUMMARY
Transmission Media
Transmission media lie below the physical layer.
A guided medium provides a physical conduit from one device to
another. Twisted pair cable, coaxial cable, and optical fiber are the
most popular types of guided media.
Twisted -pair cable consists of two insulated copper wires twisted
together. Twisted pair cable is used for voice and data
communications.
Coaxial cable consists of a central conductor and a shield. Coaxial
cable can carry signals of higher frequency ranges than twisted -pair
cable. Coaxial cable is used in cable TV networks and traditional
Ethernet LANs.
Fiber -optic cables are composed of a glass or plastic inner core
surrounded by cladding, all encased in an outside jacket. Fiber -optic
cables carry data signals in the form of light. The signal is propagated
along the inner core by reflection. Fiber -optic cable is used in
backbone networks, cable TV networks, and Fast Ethernet networks.
Unguided media (free space) transport electromagnetic waves without
the use of a physical co nductor.
Wireless data are transmitted through ground propagation, sky
propagation, and line -of-sight propagation. Wireless waves can be
classified as radio waves, microwaves, or infrared waves. Radio waves
are omni directional; microwaves are unidirectional. Microwaves are
used for cellular phone, satellite, and wireless LAN communications.
Infrared waves are used for short -range communications such as those
between a PC and a peripheral device. It can also be used for indoor
LANs.
Switchingmunotes.in

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129A switched network consists of a series of interlinked nodes, called
switches. Traditionally three methods of switching have been important:
circuit switching, packet switching, and message switching.
We can divide today’ s networks into three broad categories: circuit -
switched networks, packet -switched networks, and message -switched.
Packet -switched networks can also be divided into two subcategories:
virtual -circuit networks and datagram networks.
A circuit -switched netwo rk is made of a set of switches connected by
physical links, in which each link is divided into nchannels. Circuit
switching takes place at the physical layer. In circuit switching, the
resources need to be reserved during the setup phase; the resources
remain dedicated for the entire duration of data transfer phase until the
teardown phase.
In packet switching, there is no resource allocation for a packet. This
means that there is no reserved bandwidth on the links, and there is no
scheduled processing ti me for each packet. Resources are allocated on
demand.
In a datagram network, each packet is treated independently of all
others. Packets in this approach are referred to as datagrams. There are
no setup or teardown phases.
A virtual -circuit network is a c ross between a circuit -switched network
and a datagram network. It has some characteristics of both.
Circuit switching uses either of two technologies: the space -division
switch or the time -division switch.
A switch in a packet -switched network has a diffe rent structure from a
switch used in a circuit -switched network. We can say that a packet
switch has four types of components: input ports, output ports, a
routing processor, and switching fabric.
6.7 REFERENCE FOR FURTHER READING
For more details about topic –Transmission media and Switching
discussed in this chapter, we recommend the following books.
1.Data Communication and Networking by Behrouz A. Forouzan,
McGraw -Hill, 2007.
2.Data and Computer Communications by W. Stallings, Prentice Hall,
2004.
3.Communication Networks by Alberto Leon -Gracia &Indra Widjaja,
McGraw -Hill, 2003.
4.Computer Networks by Andrew S. Tanenbaum, Prentice Hall, 2003.
5.Digital Telephony by J. Bellamy, Wiely, 2000.munotes.in

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1306.8 MODEL QUESTIONS
Transmission Media
1.What is th e position of the transmission media in the OSI or the
Internet model?
2.Name the two major categories of transmission media.
3.How do guided media differ from unguided media?
4.What are the three major classes of guided media?
5.What is the significance of the tw isting in twisted -pair cable?
6.What is refraction? What is reflection?
7.What is the purpose of cladding in an optical fiber?
8.How does sky propagation differ from line -of-sight propagation?
Switching
1.What is the need for switching and define a switch.
2.List the three traditional switching methods. What are the most
common today?
3.What are the two approaches to packet -switching?
4.Compare and contrast a circuit -switched network and a packet -
switched network.
5.What is the role of the address field in a packet traveling through a
datagram network?
6.What is the role of the address field in a packet traveling through a
virtual -circuit network?
7.What is TSI and its role in a time -division switching?
8.List four major components of a packet switch and their functions.
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1317
INTRODUCTION TO DATA LINK LAYER
Unit Structure
7.0 Objectives
7.1 Introduction
7.2 Data Link Layer addressing
7.3 Data Link Layer Design issues
7.4 Error Detection and Correction
7.5 Block Coding
7.6 Linear Block Coding
7.7 Cyclic Codes
7.8 Checksum
7.9 Summary
7.10 Reference for further reading
7.11 Model Questions
7.0 OBJECTIVES:
This chapter would make you to understand the following concep ts:
Concept of Link layer addressing and Design issues of Data Link
layer.
Concept of Error Detection and Correction.
Block Coding, Linear Block Coding and its different methods or
techniques.
Cyclic codes, CRC code and CRC Polynomial.
Checksum and Interne t Checksum.
7.1 INTRODUCTION
The data link layer transforms the physical layer, a raw
transmission facility, to a link responsible for node -to-node (hop -to-hop)
communication. Specific responsibilities of the data link layer include
framing, addressing, flow control, error control, and media access control.
The data link layer divides the strea m of bits received from the network
layer into manageable data units called frames. The data link layer adds a
header to the frame to define the addresses of the sender and receiver of
the frame. If the rate at which the data are absorbed by the receiver i s less
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132imposes a flow control mechanism to avoid overwhelming the receiver.
The data link layer also adds reliability to the physical layer by adding
mechanisms to detect and retra nsmit damaged, duplicate, or lost frames.
When two or more devices are connected to the same link, data
link layer protocols are necessary to determine which device has control
over the link at any given time.
7.2 DATA LINK LAYER ADDRESSING
As data lin k layer adds a header to the frame to define the
addresses of the sender and receiver of the frame. The address used by
data link layer to define sender as well as receiver’s address called as
Physical address or Media Access Control (MAC) address. Physica l
address is uniquely assigned to an individual device by its manufacturer
and it is 48 bit (6 bytes) written as 12 hexadecimal digits ;e v e r yb y t e (2
hexadecimal digits) is separated by a colon as shown in Figure 7.1.
Figure 7.1: Physical address or MAC address used by Data Link layer
7.3 DATA LINK LAYER DESIGN ISSUES
Following are the some design issues of the Data Link layer
1.Services provided tothenetwork layer: The data link layer act as a
service interface to the network layer. The principle service is
transferring data from network layer on sending machine to the
network layer on destination machine. This transfer also takes place
via DLL (Dynamic Link Library).
2.Frame synchronization: The source machine sends data in the form
of blocks (data units of manageable size) called frames to the
destination machine. The starting and ending of each frame should be
identified so that the frame can be recognized by the destination
machine.
3.Flow control: Flow control is done to prevent the flow of data frame
at the receiver end. The source machine must not send data frames at
a rate faster than the capacity of destination machine to accept them.
4.Error control: Error control is done to prevent duplic ation of
frames. The errors introduced during transmission from source to
destination machines must be detected and corrected at the
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1337.4 ERROR DETECTION AND CORRECTION
Networks must be able to transfer data from one device to ano ther
with acceptable accuracy. For most applications, a system must guarantee
that the data received are identical to the data transmitted. Any time data
are transmitted from one node to the next, they can become corrupted in
passage. Many factors can alte r one or more bits of a message. Some
applications require a mechanism for detecting and correcting errors.
Some applications can tolerate a small level of error. For example,
random errors in audio or video transmissions may be tolerable, but when
we tra nsfer text, we expect a very high level of accuracy.
Types of Errors
Whenever bits flow from one point to another, they are subject to
unpredictable changes because of interference. This interference can
change the shape of the signal. In a single -bit error, a 0 is changed to a 1
or a 1 to a 0. In a burst error, multiple bi ts are changed. For example, a
11100 s burst of impulse noise on a transmission with a data rate of 1200
bps might change all or some of the12 bits of information.
Single -Bit Error
The term single -bit error means that only 1 bit of a given data unit
(such as a byte, character, or packet) is changed from 1 to 0 or from 0 to
1.Figure 7.2 show the effect of a single -bit error on a data unit.
Figure 7.2: Single bit error
Burst Error
The t erm burst error means that 2 or more bits in the data unit have
changed from 1 to 0or from 0 to 1.Figure 7.3 shows the effect of a burst
error on a data unit.
Figure 7.3: Burst error of length 8
Redundancy
The central conc ept in detecting or correcting errors is redundancy.
To be able to detect or correct errors, we need to send some extra bits with
our data. These redundant bits are added by the sender and removed by the
receiver. Their presence allows the receiver to dete ct or correct corrupted
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134Detection versus Correction
The correction of errors is more difficult than the detection. In error
detection, we are looking only to see if any error has occurred. The answer
is a simple yes or no. We are not even interested in the number of errors.
A single -bit error is the same for us as a burst error.
In error correction, we need to know the exact number of bits that
are corrupted and more importantly, their location in the message. The
number of the errors and the size of the message are important factors. If
we need to correct one single error in an 8 -bit data unit, we need to
consider eight possible error locations; if we need to correct two errors in
a data unit of the same size, we need to consider 28 possibilities. Yo uc a n
imagine the Receiver’s difficulty in finding 10 errors in a data unit of 1000
bits.
Forward Error Correction versus Retransmission
There are two main methods of error correction.
Forward error correction is the process in which the receiver tries
to guess the message by using redundant bits. This is possible, as we see
later, if the number of errors is small.
Correction by retransmission is a technique in which the receiver
detects the occurrence of an error and asks the sender to resend the
messag e. Resending is repeated until a message arrives that the receiver
believes is error -free (usually, not all errors can be detected).
Coding
Redundancy is achieve d through various coding schemes. The
sender adds redundant bits through a process that creates a relationship
between the redundant bits and the actual data bits. The receiver checks
the relationships between the two sets of bits to detect or correct the errors.
The ratio of redundant bits to the data bits and the robustness of the
process are important factors in any coding scheme. Figure 7.4 shows the
general idea of coding.
Figure 7.4: Structure of Encoder and Decoder
Modular Arithmetic
In modular arithmetic, we use only a limited range of integers. We
define an upper limit, called a modulus N.We then use only the integers 0munotes.in

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135toN-1, inclusive. This is called as modulo -Narithmetic. For example, if
the modulus is 12, w e use only the integers 0 to11, inclusive.
Modulo -2A r i t h m e t i c
Our main interest is in modulo -2 arithmetic. In this arithmetic, the
modulus N is 2. We can use only 0 and 1. Operations (addition and
subtraction) in this arithmetic are very simple.
In thi s arithmetic we use the XOR (exclusive OR) operation for
both addition and subtraction. The result of an XOR operation is 0 if two
bits are the same; the result is 1 if two bits are different. Figure 7.5 shows
this operation.
Figure 7.5: XORing of 2 single bits or 2 words
7.5 BLOCK CODING
In block coding, we divide our message into blocks, each of kbits,
called data words. We add rredundant bits to each block to make the
length n=k+r.The resulting n-bitblocks are called code words. For the
moment, it is important to know that we have a set of data words, each of
size k,and a set of code words, each of size of n.With kbits,w ec a n
create a combination of 2kdatawords; with nbits,w ec a nc r e a t ea
combin ation of 2ncodewords.
Since n>k,the number of possible code words is larger than the number
of possible data words.
The block coding process is one -to-one; the same data word is
always encoded as the same codeword. This means that we have 2n-
2kcodewords that are not used. We call these code words invalid or illegal.
Figure 7.6 shows the situation.
Figure 7.6: Datawords and Codewords in Block Codingmunotes.in

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136Example 7.1:
The4B/5B block coding is a good example of this type of coding.
In this coding scheme, k=4 andn=5 . As we see, we have 2k= 16 data
words and 2n= 32 code words. We have 16 out of 32 code words are used
for message transfer and the rest are either used for other purposes or
unused.
Error Detection
How can errors be detected by using block coding? If the
following two conditions are met, the receiver can detect a change in the
original codeword.
1.The receiver has (or can find) a list of valid code words.
2.The original codeword has changed to an invalid one.
Figure 7.7 shows the role of block coding in error detection.
Figure 7.7: Process of Error detection in Block coding
Example 7.2:
Let us assume that k=2 andn=3 .Table 7.1 shows the list of data words
and code words.
Table 7.1: Code for Error detection
Assume the sender encodes the data word 01 as 011 and sends it to
the receiver. Consider the following cases:
1.The receiver receives 011. It is a valid codeword. The receiver
extracts the data word 01 from it.
2.The codeword is corrupted during transmission, and 111 is
received (the leftmost bit is corrupted). This is not a valid
codeword and is discarded.
3.The codeword is corrupted during transmission, and 000 is
received (the right two bits are corrupted). This is a validmunotes.in

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137codeword. The receiver incorrectly extracts the dataword 00. Two
corrupted bits have made the error undetectable.
Note: An error -detectin g code can detect only the types of errors for which
it is designed; other types of errors may remain undetected.
Error Correction
As we said before, error correction is much more difficult than
error detection. In error detection, the receiver needs to k now only that the
received codeword is invalid; in error correction the receiver needs to find
(or guess) the original codeword sent. We can say that we need more
redundant bits for error correction than for error detection.
Figure 7.8 shows the role of b lock coding in error correction. We can see
that the idea is the same as error detection but the checker functions are
much more complex.
Figure 7.8: Structure of Encoder and Decoder in Error correction
Example 7.3:
Let us add more redundant bits to Example 7.2 to see if the receiver can
correct an error without knowing what was actually sent. We add 3
redundant bits to the 2 -bit data word to make 5 -bitcodewords. Table 7.2
shows the data words and code words.
Table 7.2: Code for Error correction
Assume the data word is 01.
The sender consults the table (or uses an algorithm) to create the
codeword 01011. The codeword is corrupted during transmission, and
01001 is received (error i n the second bit from the right). First, the
receiver finds that the received codeword is not in the table.munotes.in

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138This means an error has occurred. (Detection must come before
correctio n.) The receiver, assuming that there is only 1 bit corrupted, uses
the following strategy to guess the correct data word.
1.Comparing the received codeword with the first codeword in the table
(01001 versus 00000), the receiver decides that the first codewo rd is
not the one that was sent because there are two different bits.
2.By the same reasoning, the original codeword cannot be the third or
fourth one in the table.
3.The original codeword must be the second one in the table because this
is the only one that differs from the received codeword by 1 bit. The
receiver replaces 01001 with 01011 and consults the table to find the
data word 01.
Hamming Distance
One of the central concepts in coding for error control is the idea
of the Hamming distance. The Hamming distance between two words (of
the same size) is the number of differences between the corresponding
bits. We show the Hamming distance between two words xandyas
d(x, y).
The Hamming distance can easily be found if we apply the XOR
operatio n(⊕) on the two words and count the number of 1s in the result.
Note that the Hamming distance is a value greater than zero.
Example 7.4:
Let us find the Hamming distance between two pairs of words.
The Hamming distance d (000, 011) is 2 because 000 ⊕011 is 011
(two 1s).
The Hamming distance d (10101, 11110) is 3 because 10101 ⊕
11110 is 01011 (three 1s).
Minimum Hamming Distance
Although the concept of the Hamming distance is the central point
in dealing with error detection and correction codes, the measurement that
is used for designing a code is the minimum Hamming distance. In a set of
words, the minimum Hamming distance is the smallest Hamming distance
between all possible pairs. We use d minto define the minimum Hamming
distance in a coding sche me. To find this value, we find the Hamming
distances between all words and select the smallest one.
Example 7.5:
Find the minimum Hamming distance of the coding scheme in Table 7.1.munotes.in

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139Solution:
We first find all Hamming distances.
d(000, 011) = 2d(000, 101) = 2 d(000, 110) = 2
d(011, 101) =2d(011, 110) = 2 d(101,110) = 2
Thedminin this case is 2.
Example 7.6:
Find the minimum Hamming distance of the coding scheme in Table 7.2.
Solution:
We first find all the Hamming distances.
d(00000, 01011 )=3 d(00000, 10101) = 3d(00000, 11110) = 4
d(01011, 10101) = 4d(01011, 11110) = 3 d(10101, 11110) = 3
The d minin this case is 3.
We need to mention that any coding scheme needs to have at least
three parameters: the codeword size n,the data word size k,and the
minimum Hamming distance dmin.A coding scheme C is written as C(n,
k)with a separate expression for dmin-For example, we can call our Table
7.1 coding scheme is C(3, 2) with dmin=2and our Table 7.2 coding
scheme is C(5, 2)with dmin=3.
Hamming Distance and Error
Before we explore the criteria for error detection or correction, let
us discuss the relationship between the Hamming distance and errors
occurring during transmission. When a codeword is corrupted during
transm ission, the Hamming distance between the sent and received code
words is the number of bits affected by the error. In other words, the
Hamming distance between the received codeword and the sent codeword
is the number of bits that are corrupted during tran smission. For example,
if the codeword 00000 is sent and 01101 is received, 3 bits are in error and
the Hamming distance between the two is d(00000, 01101) =3.
Minimum Distance for Error Detection
Now let us find the minimum Hamming distance in a code if we
want to be able to detect up to serrors. If serrors occur during
transmission, the Hamming distance between the sent codeword and
received codeword is s. If our code is to detect up to serrors, the minimum
distance between the valid codes must be s+ 1, so that the received
codeword does not match a valid codeword. In other words, if the
minimum distance between all valid code words is s+1 , the received
codeword cannot be erroneously mistaken for another codeword. The
distances are not enough( s+1 )for the receiver to accept it as valid. The
error will be detected. We need to clarify a point here: Although a code
with dmin=s+1 may be able to detect more than serrors in some special
cases, only sor fewer errors are guaranteed to be detected.munotes.in

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140Example 7.7:
The minimum Hamming distance for Table 7.1 coding scheme is
2. This code guarantees detection of only a single error. For example, if
the third codeword (101) is sent and one error occurs, the received
codeword does not match any valid codewor d. If two errors occur,
however, the received codeword may match a valid codeword and the
errors are not detected.
Example 7.8:
For Table 7.2 coding scheme has d min= 3. This code can detect up
to two errors. Again, we see that when any of the valid code words is sent,
two errors create a codeword which is not in the table of valid code words.
The receiver cannot be fooled. However, some combinations of three
errors change a valid codeword to another valid codeword. The receiver
accepts the received codewo rd and the errors are undetected.
7.6 LINEAR BLOCK CODING
Almost all block codes used today belong to a subset called linear
block codes. The use of nonlinear block codes for error detection and
correction is not as widespread because their structure makes theoretical
analysis and implementation difficult. We therefore concentrate on linear
block codes.
A linear block code is a code in which the exclusive OR (a ddition
modulo -2) of two valid code words creates another valid codeword.
Example 7.9:
Let us see if the two codes we defined in Table 7.1 and Table 7.2
belong to the class of linear block codes.
1.The scheme in Table 7.1 is a linear block code because the result
of XORing any codeword with any other codeword is a valid
codeword. For example, the XORing of the second and third code
words creates the fourth one.
2.The scheme in Table 7.2 is also a linear block code. We can create
all four code words by XORing t wo other code words.
Minimum Distance for Linear Block Codes
It is simple to find the minimum Hamming distance for a linear
block code. The minimum Hamming distance is the number of 1s in the
nonzero valid codeword with the smallest number of 1s.
Example 7.10:
In our first code -Table 7.1, the numbers of 1s in the nonzero code
words are 2, 2, and 2. So the minimum Hamming distance is dmin=2.I n
our second code -Table 7.2, the numbers of 1s in the nonzero code words
are 3, 3, and 4. So in this code we have dmin=3.munotes.in

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141Let us now we can study some Linear block codes
Simple Parity Check code
The most familiar error -detecting code is the simple parity -check
code. In this code, a k-bitdata word is changed to an n -bit codeword
where n=k+ 1. The extra bit, called the parity bit, is selected to make the
total number of 1s in the codeword even.
The minimum Hamming distance for this category is d min=2,
which means that the code is a single -bit error -detecting code; it cannot
correct any error.
Our first cod e-Table 7.1 is a parity -check code with k= 2 and n
=3. The code in Table 7.3 is also a parity -check code with k=4 and n
=5.Figure 7.9 shows a possible structure of an encoder (at the sender) and
a decoder(at the receiver).
Table 7.3: Simple Parity -check code C(5,4)
Figure 7.9: Encoder and Decoder for Simple Parity -check code
The encoder uses a generator that takes a copy of a 4 -bit data word
(a0, a1,a2 anda3)and generates a parity bit r0.The data word bits and themunotes.in

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142parity bit create the 5 -bitcodeword. The parity bit that is added makes the
number of 1s in the codeword even.
r0 = a3 + a2 + a1 + a0 (modulo -2)
This is normally done by adding the 4 bits of the data word
(modulo -2); the result is the parity bit. In other words, if the number of 1s
is even, the result is 0; if the number of 1s is odd, the result is 1.In both
cases, the total number of 1s in the cod eword is even.
The sender sends the codeword which may be corrupted during
transmission. The receiver receives a 5 -bit word. The checker at the
receiver does the same thing as the generator in the sender with one
exception: The addition is done over all 5 bits. The result, which is called
the syndrome, is just 1 bit. The syndrome is 0 when the number of 1s in
the received codeword is even; otherwise, it is 1.
s0 = b3 + b2 + b1 + b0 + q0 (modulo -2)
The syndrome is passed to the decision logic analyze r. If the
syndrome is 0, there is no error in the received codeword; the data portion
of the received codeword is accepted as the data word; if the syndrome is
1, the data portion of the received codeword is discarded. The data word is
not created.
Exampl e 7.11:
Let us look at some transmission scenarios. Assume the sender
sends the data word 1011. The codeword created from this data word is
10111, which is sent to the receiver. We examine five cases:
1.No error occurs; the received codeword is 10111. The sy ndrome is
0. The data word 1011 is created.
2.One single -bit error changes a1.The received codeword is 10011.
The syndrome is 1. No data word is created.
3.One single -bit error changes r0.The received codeword is 10110.
The syndrome is 1. No data word is created. Note that although
none of the data word bits are corrupted, no data word is created
because the code is not sophisticated enough to show the position
of the corrupted bit.
4.An error changes r0and a second error changes a3.The received
codeword i s 00110. The syndrome is 0. The data word 0011 is
created at the receiver. Note that here the data word is wrongly
created due to the syndrome value. The simple parity -check
decoder cannot detect an even number of errors. The errors cancel
each other out a nd give the syndrome a value of 0.
5.Three bits -a3,a2and a1are changed by errors. The received
codeword is 01011. The syndrome is 1. The data word is not
created. This shows that the simple parity check, guaranteed to
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143Note: A simple parity -check code can detect an odd number of errors.
Hamming Codes
Now let us discuss a category of error -correcting codes called
Hamming codes. These codes were originally designed with dmin=3,
which means that they can detect up to two errors or correct one single
error. Although there are some Hamming codes that can correct more than
one error, our discussion focuses on the single -bit error -correcting code.
First let us find the relationship between nandkin a Hamming
code. We need to choose an integer m>= 3. The values of nandkare then
calculated from masn=2m–1andk=n-m.The number of check bits r
=m.
For example, if m=3, then n=7 andk=4 . This is a Hamming
code C(7, 4) with dmin= 3.
Table 7.4 shows the datawords and codewords for this code.
Table 7.4: Hamming code C(7, 4)
Figure 7.10 shows the structure of the encoder and decoder for this
example.
Figure 7.10: Encoder and Decoder for Hamming codemunotes.in

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144A copy of a 4 -bit data word is fed into the generator that creates
three parity checks r0, r1 andr2as shown below:
r0 = a2 + a1 + a0 (modulo -2)
r1 = a3 + a2 + a1 (modulo -2)
r2 = a1 + a0 + a3 (modulo -2)
The checker in the decoder creates a 3 -bit syndrome –s2, s1 ands0
in which each bit is the parity check for 4 out of the 7 bits in the received
codeword:
s0 = b2 + b1 + b0+q0 (modulo -2)
s1 = b3 + b2 + b1+q1 (modulo -2)
s2 = b1 + b0 + b3+q2 (modulo-2)
The 3 -bit syndrome creates eight different bit patterns (000 to 111)
that can represent eight different conditions. These conditions define a
lack of error or an error in 1 of the 7 bits of the received codeword, as
shown in Table 7.5.
Table 7.5: Logical decision made by the correction logic analyzer of
Decoder
Example 7.12:
Let us trace the path of three data words from the sender to the destination:
1.The data word 0100 becomes the codeword 0100011. The codeword
0100011 is received. The syndrome is 000 (no error), the final data
word is 0100.
2.The data word 0111 becomes the codeword 0111001. The codeword
0011001 is received. The syndrome is 011. According to Table 7.5, b2
is in error. After flipping b2 (c hanging the1 to 0), the final data word is
0111.
3.The data word 1101 becomes the codeword 1101000. The codeword
0001000 is received (two errors). The syndrome is 101, which means
that according to Table 7.5, b0 is in error. After flipping b0, we
get0000, th e wrong data word. This shows that our code cannot correct
two errors.
Example 7.13:
We need a data word of at least 7 bits. Calculate values of kandn
that satisfy this requirement.
Solution:
We need to make k =n -mgreater than or equal to 7, or2m−1−m≥7.
1.If we set m =3 , the result is n =23-1andk =7 -3, or4, which is not
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1452.If we set m =4 , then n =24–1=15 andk =15 -4 =11 , which satisfies the
condition. So the code is C(l5, 11) .
7.7 CYCLIC CODES
Cyclic codes are special linear block codes with one extra
property. In a cyclic code, if a codeword is cyclically shifted (rotated), the
result is another codeword. For example, if 1011000is a codeword and we
cyclically left -shift, then 0110001 is also a codeword.
In this case, if we call the bits in the first word a0toa6and the bits
in the second word b0tob6,we can shift the bits by using the following:
b1=a0,b2=a1,b3=a2,b4=a3,b5=a4,b6=a5,b0=a6
In the rightmost equation, the last bit of the first word is wrapped
around and becomes the first bit of the second word.
Cyclic Redundancy Check
A category of cyclic codes called the Cyclic Redundancy Check
(CRC) that is used in networks such as LANs and WANs.
Table 7.6 shows an example of a CRC code. We can see both the linear
and cyclic properties of this code.
Figure 7.11 shows one possible design for the encoder and decoder.
Table 7.6: CRC code with C(7, 4)munotes.in

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146
Figure 7.11: Encoder and Decoder for CRC code
In the encoder, the data word has kbits (4 here); the codeword has
nbits (7 here).The size of the data word is augmented by adding n–k(3
here) 0s to the right -hand side of the word. The n-bitresult is fed into the
generator. The generator uses a divisor of size n-k+1 (4 here),
predefined and agreed upon. The generator divides the augmented data
word by the divisor (modulo -2 division). The quotient of the division is
discarded; the remainder (r2, r1,r0 )is appended to the data word to create
the codeword.
The decoder receives the possibly corrupted codeword. A copy of
allnbits is fed to the checker which is a replica of the generator. The
remainder produced by the chec ker is a syndrome of n–k(3 here) bits,
which is fed to the decision logic analyzer. The analyzer has a simple
function. If the syndrome bits are all as 0s, the 4 leftmost bits of the
codeword are accepted as the data word (interpreted as no error);
otherwise, the 4 bits are discarded (error).
Encoder
Let us take a closer look at the encoder. The encoder takes the data
word and augments it with n-knumber of 0s. It then divides the
augmented data word by the divisor, as shown in Figure 7.12.munotes.in

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147
Figure 7.12: Division in the CRC Encoder
Decoder
The codeword can change during transmission. The decoder does
the same division process as the encoder. The remainder of the division is
the syndrome. If the syndrome is all 0s,there is no error; the data word is
separated from the received codeword and accepted. Otherwise,
everything is discarded. Figure 7.13 shows two cases: The lefth and figure
shows the value of syndrome when no error has occurred; the syndrome
is000. The rig ht-hand part of the figure shows the case in which there is
one single error. The syndrome is not all 0s( i ti s 011).
Figure 7.13: Division in the CRC Decoder for two casesmunotes.in

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148Polynomials
A better way to understand cyclic cod es and how they can be
analyzed is to represent them as polynomials. A pattern of 0s and 1sc a n
be represented as a polynomial with coefficients of 0and1. The power of
each term shows the position of the bit; the coefficient shows the value of
the bit. Fi gure 7.14 shows a binary pattern and its polynomial
representation. In Figure 7.14a we show how to translate a binary pattern
to a polynomial; in Figure 7.14b we show how the polynomial can be
shortened by removing all terms with zero coefficients.
Figure 7.14: A Polynomial to represent a binary word
Degree of a Polynomial
The degree of a polynomial is the highest power in the polynomial.
For example, the degree of the polynomial x6+x+1is6. Note that the
degree of a polynomial is 1 less than the number of bits in the pattern. The
bit pattern in this case has 7bits.
Adding and Subtracting Polynomials
Adding and subtracting polynomials in mathematics are done by
adding or subtracting the co efficients of terms with the same power. In our
case, the coefficients are only 0and1, and adding is in modulo -2. This has
two consequences. First, addition and subtraction are the same. Second,
adding or subtracting is done by combining terms and deletin g pairs of
identical terms. For example, adding x5+x4+x2andx6+x4+x2givesjust
x6+x5.The terms x4andx2are deleted. However, note that if we add, for
example, three polynomials and we get x2three times, we delete a pair of
them and keep the third.
Multiplying or Dividing Terms
In this arithmetic, multiplying a term by another term is very
simple; we just add the powers. For example, x3xx4isx7.For dividing, we
just subtract the power of the second term from the power of the first. For
example, x5/x2isx3.
Dividing One Polynomial by Another
Division of polynomials is conceptually the same as the binary
division we discussed for an encoder. We divide the first term of the
dividend by the first term of the divisor to get the first term of the quotient.
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149from the dividend. We repeat the process until the dividend degree is less
than the divisor degree.
Shifting
A binary pattern is often shifted a number of bits to the right or
left. Shifting to the left means adding extra 0s as rightmost bits; shifting to
the right means deleting some rightmost bits. Shifting to the left is
accomplished by multiplying each term of the polynomial by xm,where m
is the number of shifted bits; shiftin g to the right is accomplished by
dividing each term of the polynomial by xm.The following shows shifting
to the left and to the right. Note that we do not have negative powers in the
polynomial representation.
Shifting left 3 bits: 10011 becomes 10011000 x4+x+1 becomes
x7+x4+x3
Shifting right 3 bits: 10011 becomes 10x4+x+1becomes x
Cyclic Code Encoder Using Polynomials
Now we show the creation of a code word from a data word. The data
word 1001 is represented as x3+1. The divisor 1011 is represented as x3+
x+1 . To find the augmented data word, we have left -shifted the data
word 3bits (multiplying by x3).The result is x6+x3.
Division is straightforward.
We divide the first term of the dividend, x6,by the first term of the
diviso r,x3.The first term of the quotient is then x6/x3,orx3.Then we
multiply x3by the divisor and subtract (according to our previous
definition of subtraction) the result from the dividend. The result is x4,
with a degree greater than the divisor’s degree ; we continue to divide until
the degree of the remainder is less than the degree of the divisor. CRC
division using Polynomials is shown in the Figure 7.15.
Figure 7.15: CRC division using Polynomialsmunotes.in

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150It can be seen that the polynomial representation can easily
simplify the operation of division in this case, because the two steps
involving all -0s divisors are not needed here. (Of course, one could argue
that the all -0s divisor step can also be eliminated in binary divisio n). In a
polynomial representation, the divisor is normally referred to as the
generator polynomial t(x).
We can summarize the criteria for a good polynomial generator:
A good polynomial generator needs to have the following characteristics:
1.It should hav e at least two terms.
2.The coefficient of the term x0should be 1.
3.It should not divide xt+1, for t between 2 and n -1.
4.It should have the factor x+ 1.
Standard Polynomials
Some standard polynomials used by popular protocols for CRC
generation are shown in Table 7.7.
Table 7.7: Standard Polynomials
Advantages of Cyclic Codes
Cyclic codes have a very good performance in detecting single -bit
errors, double errors, an odd number of errors, and burst errors. They can
easily be implemented in hardware and software. They are especially fast
when implemented in hardware. This has mad e cyclic codes a good
candidate for many networks.
7.8 CHECKSUM
The last error detection method we discuss here is called the
checksum. The checksum is used in the Internet by several protocols
although not at the data link layer. However, we briefly dis cuss it here to
complete our discussion on error checking. Like linear and cyclic codes,
the checksum is based on the concept of redundancy.
Concept
The concept of the checksum is not difficult. Let us illustrate it
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151Example 7.14:
Suppose our data is a list of five 4 -bit numbers that we want to send to a
destination. In addition to sending these numbers, we send the sum of the
numbers. For example, if the set of numbers is(7, 11, 12, 0, 6), we send (7,
11, 12,0,6,36), where 36 is th e sum of the original numbers. The receiver
adds the five numbers and compares the result with the sum. If the two are
the same, the receiver assumes no error, accepts the five numbers, and
discards the sum. Otherwise, there is an error somewhere and the d ata are
not accepted.
Example 7.15:
We can make the job of the receiver easier if we send the negative
(complement) of the sum, called as the checksum. In this case, we send (7,
11, 12,0,6, -36). The receiver can add all the numbers received (including
the checksum). If the result is 0, it assumes no error; otherwise, there is an
error.
One’s Complement
The previous example has one major drawback. All of our data can
be written as a 4 -bitword (they are less than 15) except for the checksum.
One solution i s to use one’s complement arithmetic. In this arithmetic, we
can represent unsigned numbers between 0and2n-1using only nbits. If
the number has more than nbits, the extra leftmost bits need to be added
to the nrightmost bits (wrapping). In one’s comp lement arithmetic, a
negative number can be represented by inverting all bits (changing a 0 to a
1 and a 1 to a 0). This is the same as subtracting the number from 2n-1.
Example 7.16:
How can we represent the number 21 in one’ s complement
arithmetic using only four bits?
Solution:
The number 21in binary is 10101 (it needs five bits). We can wrap
the leftmost bit and add it to the four rightmost bits. We have (0101 + 1) =
0110 or6.
Example 7.17:
How can we represent the numb er-6 in one’s complement
arithmetic using only four bits?
Solution:
In one’s complement arithmetic, the negative or complement of a
number is found by inverting all bits. Positive 6is0110; negative 6is1001.
If we consider only unsigned numbers, this is 9. In other words, the
complement of 6is9. Another way to find the complement of a number in
one’s complement arithmetic is to subtract the number from 2n-1(16-1
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152Example 7.18:
Let us redo Exercise 7.15 using one’ s complement arithmetic.
Figure 7.16 shows the process at the sender and at the receiver. The sender
initializes the checksum to 0and adds all data items and the checksum.
The result is 36. However, 36cannot be expressed in 4bits. The extra two
bits are wrapped and added with the sum to create the wrapped sum value
6. In the figure, we have shown the details in binary. The sum is then
complemented, resulting in the checksum value 9 (15 -6 = 9) . The sender
now sends six data items to the receiver includi ng the checksum 9. The
receiver follows the same procedure as the sender. It adds all data items
(including the checksum); the result is 45. The sum is wrapped and
becomes 15. The wrapped sum is complemented and becomes 0. Since the
value of the checksum i s0, this means that the data is not corrupted. The
receiver drops the checksum and keeps the other data items. If the
checksum is not zero, the entire packet is dropped.
Figure 7.16: Example 7.18
Internet Checksum
Traditionally, the Internet has been using a 16 -bit checksum. The
sender calculates the checksum by following these steps.
Sender site:
1.The message is divided into 16 -bit words.
2.The value of the checksum word is set to 0.
3.All words inclu ding the checksum are added using one’s complement
addition.
4.The sum is complemented and becomes the checksum.
5.The checksum is sent with the data.
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153Receiver site:
1.The message (including checksum) i s divided into 16 -bit words.
2.All words are added using one’s complement addition.
3.The sum is complemented and becomes the new checksum.
4.If the value of checksum is 0, the message is accepted; otherwise,
it is rejected.
The nature of the checksum (treating words as numbers and adding and
complementing them) is well -suited for software implementation. Short
programs can be written to calculate the checksum at the receiver site or to
check the validity of the message at the receiver site.
7.9 SUMMARY
Data can be corrupted during transmission. Some applications require
that errors be detected and corrected.
In a single -bit error, only one bit in the data unit has changed. A burst
error means that two or more bits in the da ta unit have changed.
To detect or correct errors, we need to send extra (redundant) bits with
data.
There are two main methods of error correction: forward error
correction and correction by retransmission.
In coding, we need to use modulo -2 arithmetic. O perations in this
arithmetic are very simple; addition and subtraction give the same
results. We use the XOR (exclusive OR) operation for both addition
and subtraction.
In block coding, we divide our message into blocks, each of kbits,
called data words. We add rredundant bits to each block to make the
length n=k+r.The resulting n -bit blocks are called code words.
In block coding, errors be detected by using the following two
conditions:
oThe receiver has (or can find) a list of valid code words.
oThe original codeword has changed to an invalid one.
The Hamming distance between two words is the number of
differences between corresponding bits. The minimum Hamming
distance is the smallest Hamming distance between all possible pairs
in a set of words.
To guarantee the detection of up to s errors in all cases, the minimum
Hamming distance in a block code must be d min= s + 1.
In a linear block code, the exclusive OR (XOR) of any two valid code
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154A simple parity -check code is a single -bit error -detecting code in
which n=k+1 w i t hd min=2. A simple parity -check code can detect an
odd number of errors.
All Hamming codes discussed in this book have d min= 3. The
relationship between mandnin these codes is n=2m-1.
Cyclic codes are special linear block codes with one extra property. In
a cyclic code, if a codeword is cyclically shifted (rotated), the result is
another codeword.
A category of cyclic codes called the cyclic redundancy check (CRC)
is used in networks such as LANs and WANs.
A pattern of 0s and 1s can be represented as a polynomial with
coefficients of 0 and 1.
Traditionally, the Internet has been using a 16 -bit checksum, which
uses one’s complement arithmetic. In this arithmetic, we can represent
unsigned numbers between0 and 2n-1 using only nbits.
7.10 REFERENCE FOR FURTHER READING
For more details about topics discussed in this chapter, we
recommend the following books.
1.Data Communication and Networking by Behrouz A. Forouzan,
McGraw -Hill, 2007.
2.Coding and Information Theory by R. W. Hamming, Prentice Hall,
1980.
3.The Art of Error Correcting Coding by Robert H. Morelos -
Zaragoza, Wesley, 2002.
4.Error Coding Cookbook by C. Britton Rorabaugh, McGraw -Hill,
1996.
7.11 MODEL QUESTIONS
1.How does a single -bit error differ from a burst error?
2.Discuss the concept of redundancy in error detection and correction.
3.Distinguish between forward error correction versus error correction
by retransmission.
4.What is the definition of a linear block code? What is the definition of
a cyclic code?
5.What is the Hamming distance? What is the minimum Hamming
distance?
6.In CRC, show the relationship between the following entities (size
means the number of bits):munotes.in

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155a.The size of the data word and the size of the codeword
b.The size of the divisor and the remainder
c.The degree of the polynomial generator and the size of the divisor
d.The degree of the polynomial generator and the size of the
remainder
7.What kind of arithmetic is u sed to add data items in checksum
calculation?
8.What kind of error is undetectable by the checksum?
Exercises
1.Apply the exclusive -OR operation on the following pair of patterns:
a.(10001)(10000)
b.(10001)(10001) (What do you infer from the result?)
c.(11100)(00000) (What do you infer from the re sult?)
d.(10011)(11111) (What do you infer from the result?)
2.What is the Hamming distance for each of the following code words:
a.d (10000, 00000)
b.d (10101, 10000)
c.d (11111,11111)
d.d (000, 000)
3.Find the minimum Hamming distance for the following cases:
a.Detection of two errors.
b.Correction of two errors.
c.Detection of 3 errors or correction of 2 errors.
d.Detection of 6 errors or correction of 2 errors.
4.Answer the following questions:
a.What is the polynomial representation of 101110?
b.What is the result of shifting 101110 three bits to the left?
c.Repeat part b using polynomials.
d.What is the result of shifting 101110 four bits to the right?
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156Unit -III
8
DATA LINK CONTROL
Unit Structure
8.0 Objectives
8.1 Introduction
8.2 Framing
8.3 Flow and Error Control
8.4 Protocols
8.5 Noiseless Channels
8.6 Noisy Channels
8.7 HDLC
8.8 Point to Point Protocol
8.9 Summary
8.10 Review Your Learning’s
8.11 Sample Questions:
8.12 References for further reading
8.0 OBJECTIVES
1.Define functions of Data Link Layer in data transmission.
2.Describe how Data Link Layer prepares data for transmission on
network media.
3.Describe Protocols used in Data Link Layer.
4.Distinguish Stop -and-Wait and Go -Back -N ARQ protocols.
5.Explain the purpose of encapsulating packets into frames to facilitate
media access.
8.1 INTRODUCTION
The two main functions of the data link layer are data link control
andmedia access control . The first, data link control, deals with the
design and procedures for communication between two adjacent nodes:
node -to-node communication. This we will see in this chapter. The second
function of the data link layer is media access control, or how to share the
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157Data link control functions include framing, flow and error
control, and software implemented protocols that provide smooth and
reliable transmission of frames between nodes. Here, we will first discuss
framing, or how t o organize the bits that are carried by the physical layer,
then we will see flow and error control.
To implement data link control, we n eed protocols. Each protocol
is a set of rules that need to be implemented in software and run by the
two nodes involved in data exchange at the data link layer. There are five
protocols: two for noiseless (ideal)channels and three for noisy (real)
channel s. Those in the first category are not actually implemented but
provide a foundation for understanding the protocols in the second
category.
8.2 FRAMING
Data transmission in the physical layer means moving bits in the
form of a si gnal from the source to the destination. The physical layer
provides bit synchronization to ensure that the sender and receiver use the
same bit durations and timing. The data link layer, on the other hand,
needs to pack bits into frames, so that each fram e is distinguishable from
another. Our postal system practices a type of framing. The simple act of
inserting a letter into an envelope separates one piece of information from
another; the envelope serves as the delimiter. In addition, each envelope
define s the sender and receiver addresses since the postal system is a
many -to-many carrier facility.
Framing in the data link layer separates a message from one source
to a destination, or from other messages to other destinations, by adding a
sender address a nd a destination address. The destination address defines
where the packet is to go; the sender address helps the recipient
acknowledge the receipt.
Although the whole message could be packed in one frame, that is
not normally done. One reason is that a f rame can be very large, making
flow and error control very inefficient. When a message is carried in one
very large frame, even a single -bit error would require the retransmission
of the whole message. When a message is divided into smaller frames, a
singl e-bit error affects only that small frame.
Fixed -Size Framing
Frames can be of fixed or variable size. In fixed -size framing, there
is no need for defining the boundaries of the frames; the size itself can be
used as a delimiter. An example of this type of framing is the ATM wide -
area network, which uses frames of fixed size called cells.
Variable -Size Framing
Our main discussion in this chapter concerns variable -size framing,
prevalent in local area networks. In variable -size framing, we need a way
to define the end of the frame and the beginning of the next. Historically,munotes.in

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158two approaches were used for this purpose: ac h a r a c t e r -oriented approach
and a bit -oriented approach.
Character -Oriented Protocols
In a character -oriented protocol, data to be carried are 8 -bit
characters from a coding system such as ASCII (see Appendix A). The
header, which normally carries the source and destination addresses and
other control information, and the trailer, which carries error detection or
error cor rection redundant bits, are also multiples of 8 bits. To separate
one frame from the next, an 8 -bit (I -byte) flag is added at the beginning
and the end of a frame. The flag, composed of protocol -dependent special
characters, signals the start or end of a f rame. Figure shows the format of a
frame in a character -oriented protocol.
Character -oriented framing was common when only text was
exchanged by the data link layers. The flag could be selected to be any
character not used for text communication. Now, how ever, we send other
types of information such as graphs, audio, and video. Any pattern used
for the flag could also be part of the information. If this happens, the
receiver, when it encounters this pattern in the middle of the data, thinks it
has reached the end of the frame. To fix this problem, a byte -stuffing
strategy was added to character -oriented framing. In byte stuffing (or
character stuffing), a special byte is added to the data section of the frame
when there is a character with the same pattern as the flag. The data
section is stuffed with an extra byte. This byte is usually called the escape
character (ESC), which has a predefined bit pattern. Whenever the
receiver encounters the ESC character, it removes it from the data section
and treats the next character as data, not a delimiting flag.
Byte stuffing by the escape character allows the presence of the
flag in the data section of the frame, but it creates another problem. What
happens if the text contains one or more escape characters followed by a
flag? The receiver removes the escape char acter, but keeps the flag, which
is incorrectly interpreted as the end of the frame. To solve this problem,
the escape characters that are part of the text must also be marked by
another escape character. In other words, if the escape character is part of
the text, an extra one is added to show that the second one is part of the
text. Figure 8.1 shows the situation. Byte stuffing is the process of adding
1 extra byte whenever there is a flag or escape character in the text.munotes.in

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159
Figure 8.1: Byte Stuffing and Unstuffing
Character -oriented protocols present another problem in data
communications.
The universal coding systems in use today, such as Unicode, have
16-bit and 32 -bitcharacters that conflict with 8 -bit character s. We can say
that in general, the tendency is moving toward the bit -oriented protocols
that we discuss next.
Bit-Oriented Protocols
In a bit -oriented protocol, the data section of a frame is a sequence
of bits to be interpreted by the upper layer as text , graphic, audio, video,
and so on. However, in addition to headers (and possible trailers), we still
need a delimiter to separate one frame from the other. Most protocols use
a special 8 -bit pattern flag 01111110 as the delimiter to define the
beginning a nd the end of the frame, as shown in Figure 8.2
Figure 8.2: A frame in bit -oriented protocol
Bit stuffing is the process of adding one extra 0 whenever five
consecutive 18 follow a 0in the data, so that the receiver does not mistake
the pattern 0111110 for a flag.
This flag can create the same type of problem we saw in the byte -
oriented protocols . That is, if the flag pattern appears in the data, we need
to somehow inform the receiver that this is not the end of the frame. We
do this by stuffing 1 single bit (instead of I byte) to prevent the pattern
from looking like a flag. The strategy is calle d bit stuffing.munotes.in

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160In bit stuffing, if a 0 and five consecutive 1 bits are encountered,
an extra 0 is added. This extra stuffed bit is eventually removed from the
data by the receiver. Note that the extra bit is added after one 0 followed
by five 1s regardle ss of the value of the next bit. This guarantees that the
flag field sequence does not inadvertently appear in the frame.
Figure 8.3 shows bit stuffing at the sender and bit removal at the
receiver. Note that even if we have a 0 after five 1s, we still st uff a O. The
0 will be removed by the receiver.
Figure: 8.3: bit Stuffing and Unstuffing
This means that if the flag like pattern 01111110 appears in the
data, it will change to 011111010 (stuffed) and is not mista ken as a flag by
the receiver. The real flag 01111110is not stuffed by the sender and is
recognized by the receiver
8.3 FLOW AND ERROR CONTROL
Data communication requires at least two devices working
together, one to send and thitherto receive. The most important
responsibilities of the data link layer are flow control and error control.
Together these functions are known as data link control .
Flow Control
Flow control refers to a set of procedures used to restrict the
amount of data that the sender can send before waiting for
acknowledgment .
Flow control coordinates the amount of data that can be sent before
receiving an acknowledgment and is one of the most important duties of
the data link layer. In most protocols, flow control is a set of procedures
that tells the sender how much data it can transmit before it must wait for
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161allowed to overwhelm the receiver. Any receiving device has a limi ted
speed at which it can process incoming data and a limited amount of
memory in which to store incoming data. The receiving device must be
able to inform the sending device before those limits are reached and to
request that the transmitting device send fewer frames or stop temporarily.
Incoming data must be checked and processed before they can be used.
The rate of such processing is often slower than the rate of transmission.
For this reason, each receiving device has a block of memory, called a
buffer , reserved for storing incoming data until they are processed. If the
buffer begins to fill up, the receiver must be able to tell the sender to halt
transmission until it is once again able to receive.
Error Control
Error control in the data link layer is based on automatic repeat
request, which is there transmission of data. Error control is both error
detection and error correction . It allows the receiver to inform the sender
of any frames lost or damaged in transmission and coordinates there
transmissio n of those frames by the sender. In the data link layer, the term
error control refers primarily to methods of error detection and
retransmission . Error control in the data link layer is often implemented
simply: Any time an error is detected in an exchang e, specified frames are
retransmitted. This process is called automatic repeat request (ARQ).
8.4 PROTOCOLS
Now we need to combine framing, flow control, and error control
to achieve the delivery of data from one node to another. The protocols are
normally implemented in software by using one of the common
programming languages. Protocols can be used for noiseless (error -free)
channels and those that can be used for noisy (error -creating) channels.
The protocols in the first catego ry cannot be used in real life, but they
serve as a basis for understanding the protocols of noisy channels. Figure
8.4 shows the classifications.
Figure: 8.4: Taxonomy of Data Link Layer Protocols
All these protocols are unidirectional in the sense that the data
frames travel from the sender node to receiver node. Special frames, called
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162in the opposite direction for flow and error contr ol purposes, but data flow
in only one direction. However, in a real -life network, the data link
protocols are implemented as bi -directional. In these protocols the flow
and error control information such as ACKs and NAKs is included in the
data frames in a technique called piggybacking. Because bidirectional
protocols are more complex than unidirectional ones.
8.5 NOISELESS CHANNELS
There are ideal channels in which no frames are lost, duplicated, or
corrupted. There are two prot ocols for this type of channel. The first is a
protocol which does not use flow control and other is which does.
Stop-and-Wait Protocol
If data frames arrive at the receiver site faster than they can be
processed, the frames must be stored until their use . Normally, the receiver
does not have enough storage space, especially if it is receiving data from
many sources. This may result in either the discarding of frames or denial
of service. To prevent the receiver from becoming overwhelmed with
frames, we so mehow need to tell the sender to slow down. There must be
feedback from the receiver to the sender. This protocol is called the Stop -
and-Wait Protocol because the sender sends one frame, stops until it
receives confirmation from the receiver (okay to go ah ead), and then sends
the next frame. It will have unidirectional communication for data frames,
but auxiliary ACK frames (simple tokens of acknowledgment) travel from
the other direction.
Design
Following figure 8.10 illustrates the mechanism. At any time, there is
either one data frame on the forward channel or one ACK frame on the
reverse channel. It uses half -duplex link.
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163Example
Following figure 8.13 shows an example of communication using this
protocol. It is still very simple. The sender sends one frame and waits for
feedback from the receiver. When the ACK arrives, the sender sends the
next frame. Note that sending two frames in the protocol involves the
sender in four events and the receiver in two events.
Figure 8. 6: Flow Diagram of Above Example
8.6 NOISY CHANNELS
Although the Stop -and-Wait Protocol gives us an idea of how to add flow
control to its predecessor, noiseless channels are non -existent. We can
ignore the error or we need to add error control to our protocols.
Stop-and-Wait Automatic Repeat Request
Our f irst protocol, called the Stop -and-Wait Automatic Repeat
Request (Stop -and-Wait ARQ), adds a simple error control mechanism to
the Stop -and-Wait Protocol. To detect and correct corrupted frames, we
need to add redundancy bits to our data frame. When the fr ame arrives at
the receiver site, it is checked and if it is corrupted, it is silently discarded.
The detection of errors in this protocol is manifested by the silence of the
receiver. Lost frames are more difficult to handle than corrupted ones.
In our previous protocols, there was no way to identify a frame.
The received frame could be the correct one, or a duplicate, or a frame out
of order. The solution is to number the frames. When the receiver receives
a data frame that is out of order, this means t hat frames were either lost or
duplicated. Using frame -number we can do sequencing and if any frame is
required then we can ask sender to resend it. To remedy this problem, the
sender keeps a copy of the sent frame. At the same time, it starts a timer. Ifmunotes.in

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164the timer expires and there is no ACK for the sent frame, the frame is
resent, the copy is held, and the timer is restarted. Since an ACK frame
can also be corrupted and lost, it too needs redundancy bits and a sequence
number. The ACK frame for this proto col has a sequence number field. In
this protocol, the sender simply discards a corrupted ACK frame or
ignores anout -of-order one.
1.Sequence Numbers
A field is added to the data frame to hold the sequence number of
that frame. For example, if we decide tha t the field is mbits long, the
sequence numbers start from 0, go to 2m-1, and then are repeated.
2.Acknowledgment Numbers
The acknowledgment numbers always announce the sequence
number of the next frame expected by the receiver. For example, if frame
0 has arrived safe and sound, the receiver sends an ACK frame with
acknowledgment 1 (meaning frame 1 is expected next). If frame 1 has
arrived sa fe and sound, the receiver sends an ACK frame with
acknowledgment 0 (meaning frame 0 is expected)
Figure 8. 7: Design of Stop -and-Wait ARQ Protocol
Example
Assume that, in a Stop -and-Wait ARQ system, the bandwidth of the line is
1 Mbps, and 1 bit takes 20 ms to make a round trip. What is the
bandwidth -delay product? If the system data frames are 1000 bits in
length, what is the utilization percentage of the l ink?
Solution
The bandwidth -delay product is
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165
Figure 8. 8: Flow Diagram for above example
The system can send 20,000 bits during the time it takes for the
data to go from the se nder to the receiver and then back again. However,
the system sends only 1000 bits. We can say that the link utilization is
only 1000/20,000, or 5 percent. For this reason, for a link with a high
bandwidth or long delay, the use of Stop -and-Wait ARQ wastes the
capacity of the link.
Example
What is the utilization percentage of the link in above example if we have
a protocol that can send -up to 15 frames before stopping and worrying
about the acknowledgments?
Solution
The bandwidth -delay product is still 2 0,000 bits. The system can send up
to 15 frames or15,000 bits during a round trip. This means the utilization
is 15,000/20,000, or 75 percent. Of course, if there are damaged frames,
the utilization percentage is much less because frames have to be resent.
Pipelining
In networking and in other areas, a task is often begun before the previous
task has ended. This is known as pipelining. There is no pipelining in
Stop-and-Wait ARQ because we need to wait for a frame to reach the
destination and be acknowledg ed before the next frame can be sent.
However, pipelining does apply to our next two protocols because several
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166Pipelining improves the efficiency of the transmission if the number of
bitsin transition is large with respect to the bandwidth -delay product.
Sliding Window
In this protocol the sliding window is an abstract concept that
defines the range of sequence numbers that is the concern of the sender
and receiver. In other words, the sender and receiver need to deal with
only part of the possible sequence numbers. The range which is the
concern of the sender is called the send sliding window; the range that is
the concern of the receiver is called the receive sliding window. We
discuss both here. The send window is an imaginary box covering the
sequence numbers of the d ata frames which can be in transit. In each
window position, some of these sequence numbers define the frames that
have been sent; others define those that can be sent. The maximum size of
the window is 2m–1. Let the size be fixed and set to the maximum v alue.
Figure shows a sliding window of size 15 (m=4).
The window at any time divides the possible sequence numbers
into four regions.
The first region, from the far left to the left wall of the window,
defines the sequence numbers belonging to frames that are already
acknowledged. The sender does not worry about these frames and
keeps no copies of them.
The second region, colored in Figure defines the range of sequence
numbers belonging to the frames that are sent and have an unknown
status. The sender ne eds to wait to find out if these frames have been
received or were lost. We call these outstanding frames.
The third range, white in the figure, defines the range of sequence
numbers for frames that can be sent; however, the corresponding data
packets hav e not yet been received from the network layer.
The fourth region defines sequence numbers that cannot be used until
the window slides.
The send window is an abstract concept defining an imaginary box
of size 2m~ 1 with three variables: S f,S nand S sizewhere Sf is send
window, i.e., the first outstanding frame, Sn is send window, i.e. the next
frame to be sent and S size is Send window having size.
The send window can slide one or more slots when a valid
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Figure 8 .9: Send Window for Go -Back -NA R Q
The receive window makes sure that the correct data frames are received
and that the correct acknowledgments are sent. The size of the receive
window is always I. The receiver is always looking f or the arrival of a
specific frame. Any frame arriving out of order is discarded and needs to
be resent. Following figure shows the receive window.
Figure 8. 10: Receive Window for Go -Back -NA R Q
The receive window is an abstract concept defining an imaginary
box of size 1 with one single variable Rn• The window slides when a
correct frame has arrived; sliding occurs one slot at a time. Note that we
need only one variable Rn(receive window, next f rame expected) to
define this abstraction. The sequence numbers to the left of the window
belong to the frames already received and acknowledged; the sequence
numbers to the right of this window define the frames that cannot be
received. Any received frame with a sequence number in these two
regions is discarded. Only a frame with a sequence number matching the
value of Rnis accepted and acknowledged. The receive window alsomunotes.in

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168slides, but only one slot at a time. When a correct frame is received (and a
frame is received only one at a time), the window slides.
Acknowledgment
The receiver sends a positive acknowledgment if a frame has arrived safe
and sound and in order. If a frame is damaged or is received out of order,
the receiver is silent and will discard all subsequent frames until it receives
the one it is expecting.
Resending a Frame
When the timer expires, the sender resends all outstanding frames. For
example, suppose the sender has already sent frame 6, but the timer for
frame 3 expires. This means thatframe 3 has not been acknowledged; the
sender goes back and sends frames 3, 4,5, and 6again. That is why the
protocol is called Go-Back -NARQ.
Design
Figure shows the design for this protocol. As we can see, multiple frames
canbe in transit in the forward direction, and multiple acknowledgments
in the reverse direction. The idea is similar to Stop -and-Wait ARQ; the
difference is that the send window allows us to have as many frames in
transition as there are slots in the send window.
Send Window Si ze
We can now show why the size of the send window must be less than 2m.
As an example, we choose m= 2, which means the size of the window can
be2m-1, or 3.
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169
Figure 8. 12: Window Size for Go -Back -N ARQ Protocol
8.7 HDLC
High -level Data Link Control (HDLC) is a bit -oriented protocol
for communication over point -to-point and multipoint links. It implements
the ARQ mechanisms we discussed in this chapter above.
Configurations and Transfer Modes
HDLC provides two common tran sfer modes that can be used in
different configurations: normal response mode (NRM) and asynchronous
balanced mode (ABM).
Normal Response Mode
In normal response mode (NRM), the station configuration is
unbalanced. We have one primary station and multiple secondary stations.
A primary station can send commands; a secondary station can only
respond. The NRM is used for both point -to-point and multiple -point
links, as shown in following figure.
Figure 8. 13: Normal Response Modemunotes.in

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170Asynchronous Balanced Mode
In asynchronous balanced mode (ABM), the configuration is
balanced. The link is point -to-point, and each station can function as a
primary and a secondary (acting as peers), as shown in following figure.
This is the common mode today.
Figure 8. 14: Asynchronous Balanced Mode
Frames
To provide the flexibility necessary to support all the options possible in
the modes and configurations just described, HDLC defines three types of
frames: information frames (I-frames), supervisory frames (S -frames), and
unnumbered frames (V -frames). Each type of frame serves as an envelope
for the transmission of a different type of message. I -frames are used to
transport user data an d control information relating to user
data(piggybacking). S -frames are used only to transport control
information. V -frames are reserved for system management. Information
carried by V -frames is intended for managing the link itself.
Frame Format
Each frame in HDLC may contain up to six fields, as shown in
Figure 11.27: a beginning flag field, an address field, a control field, an
information field, a frame check sequence (FCS) field, and an ending flag
field. In multiple -frame transmissions, the e nding flag of one frame can
serve as the beginning flag of the next frame.
Figure 8. 15: HDLC Frames
Fields
Let us now discuss the fields and their use in different frame types.
Flag field . The flag field of an HDLC frame is an 8 -bit sequence with
the bit pattern01111110 that identifies both the beginning and the end
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171Address field . The second field of an HDLC frame contain s the
address of the secondary station. If a primary station created the frame,
it contains a toaddress. If a secondary creates the frame, it contains a
from address. An address field can be1 byte or several bytes long,
depending on the needs of the netwo rk.
Control field: The control field is a 1 -or 2-byte segment of the frame
used for flow and error control. The interpretation of bits in this field
depends on the frame type.
Information field : The information field contains the user's data from
the net work layer or management information. Its length can vary
from one network to another.
FCS field : The frame check sequence (FCS) is the HDLC error
detection field. It can contain either a 2 -or 4-byte ITU -TC R C .
Control Field : The control field determines the type of frame and
defines its functionality. The formats are shown in following figure.
Figure 8. 16: Control Field Format
8.8 POINT TO POINT PROTOCOL
Although HDLC is a general protocol that can be used for both
point -to-point and multipoint configurations, one of the most common
protocols for point -to-point access is the Point -to-Point Protocol (PPP).
Today, millions of Internet users who need to conne cttheir home
computers to the server of an Internet service provider use PPP. The
majority of these users have a traditional modem; they are connected to
the Internet through a telephone line, which provides the services of the
physical layer. But to cont rol and manage the transfer of data, there is a
need for a point -to-point protocol at the data link layer. PPP is by far the
most common.
PPP provides several services:
1.PPP defines the format of the frame to be exchanged between devices.
2.PPP defines how two devices can negotiate the establishment of the
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1723.PPP defines how network layer data are encapsulated in the data link
frame.
4.PPP defines how two devices can authenticate each other.
5.PPP provides multiple ne twork layer services supporting a variety of
network layer protocols.
6.PPP provides connections over multiple links.
7.PPP provides network address configuration. This is particularly
useful when a home user needs a temporary network address to
connect to the Internet.
On the other hand, to keep PPP simple, several services are missing:
1.PPP does not provide flow control. A sender can send several frames
one after another with no concern about overwhelming the receiver.
2.PPP has a very simple mechanism for erro r control. A CRC field is
used to detect errors. If the frame is corrupted, it is silently discarded;
the upper -layer protocol needs to take care of the problem. Lack of
error control and sequence numbering may cause a packet to be
received out of order.
3.PPP does not provide a sophisticated addressing mechanism to handle
frames in a multipoint configuration.
8.9 SUMMARY
Data link control deals with the design and procedures for
communication between two adjacent nodes: node -to-node
communication.
Framing in the data link layer separates a message from one source to
a destination, or from other messages going from other sources to
other destinations,
Frames can be of fixed or variable size. In fixed -size framing, there is
no need fo rdefining the boundaries of frames; in variable -size
framing, we need a delimiter (flag) to define the boundary of two
frames.
Variable -size framing uses two categories of protocols: byte -oriented
(or character -oriented) and bit -oriented. In a byte -orient ed protocol,
the data section of a frame is a sequence of bytes; in a bit -oriented
protocol, the data section of a frame is a sequence of bits.
In byte -oriented (or character -oriented) protocols, we use byte
stuffing; a special byte added to the data secti on of the frame when
there is a character with the same pattern as the flag.
In bit -oriented protocols, we use bit stuffing; an extra 0 is added to the
data section of the frame when there is a sequence of bits with the
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173Flow contr ol refers to a set of procedures used to restrict the amount of
data that the sender can send before waiting for acknowledgment.
Error control refers to methods of error detection and correction.
For the noiseless channel, we discussed two protocols: the S implest
Protocol and the Stop -and-Wait Protocol. The first protocol has neither
flow nor error control; the second has no error control. In the Simplest
Protocol, the sender sends its frames one after another with no regards
to the receiver. In the Stop -and-Wait Protocol, the sender sends one
frame, stops until it receives confirmation from the receiver, and then
sends the next frame.
For the noisy channel, we discussed three protocols: Stop -and-Wait
ARQ, GO -Back -N,and Selective Repeat ARQ. The Stop -and-Wait
ARQ Protocol, adds a simple error control mechanism to the Stop -and-
Wait Protocol. In the G o-Back -NARQ Protocol, we can send several
frames before receiving acknowledgments, improving the efficiency of
transmission. In the Selective Repeat ARQ protocol weavoid
unnecessary transmission by sending only frames that are corrupted.
High -level Data Link Control (HDLC) is a bit -oriented protocol for
communication
over point -to-point and multipoint links. However, the most common
protocols for point -to-point ac cess is the Point -to-Point Protocol
(PPP), which is a byte -oriented protocol.
8.10 REVIEW YOUR LEARNINGS:
1.Are you able to explain Stop-and-Wait Protocol?
2.Are you able to explain framing? Bit oriented and Byte oriented
Stuffing?
3.Are you able to explain protocols of Data Link Layer?
4.Are you able to explain the functions of Data Link Layer?
5.Can you explain what is Sliding Window in networks?
8.11SAMPLE QUESTIONS:
1.Briefly describe the services provided by the d ata link layer.
2.Define framing and the reason for its need.
3.Compare and contrast byte -oriented and bit -oriented protocols. Which
category
has been popular in the past (explain the reason)? Which category is
popular now(explain the reason)?
4.Compare and cont rast byte -stuffing and bit -stuffing. Which technique
is used inbyte -oriented protocols? Which technique is used in bit -
oriented protocols?
5.Compare and contrast flow control and error control.munotes.in

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1746.What are the two protocols we discussed for noiseless channels i n this
chapter?
7.What are the three protocols we discussed for noisy channels in this
chapter?
8.Explain the reason for moving from the Stop -and-Wait ARQ Protocol
to the GO -Back -N ARQ Protocol.
9.Compare and contrast the Go -Back -N ARQ Protocol with Selective -
RepeatARQ.
10.Compare and contrast HDLC with PPP. Which one is byte -oriented;
which one is bit-oriented?
11.Define piggybacking and its usefulness.
12.Which of the protocols described in this chapter utilize pipelining?
8.12 REFERENCES FOR FURTHER READING
Computer Networks, Andrew S. Tanenbaum,
Data Communication and Networking, Behrouz A. Forouzan, Tata
McGraw Hill Fifth Edition 2013
http://eti2506.elimu.net/Introduction/Books/Data%20Communications
%20and%20Networking%20By%20Behrouz%20A.Forouzan.pdf
https://nptel.ac.in/courses/106/105/106105082/
http://www.nptelvideos.in/2012/11/data -communication.html
https://www.edx.org/learn/data -communications
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1759
MEDIA ACCESS CONTROL
Unit Structure
9.0 Objectives:
9.1 Introduction
9.2 Random Access
9.3 Controlled Access
9.4 Channelization
9.4.1 Frequency -Division Multiple Access (FDMA)
9.4.2 Time -Division Multiple Access (TDMA)
9.4.3 Code -Division Multiple Access (CDMA)
9.5 Wired LANs
9.5.1 IEEE Standards
9.5.2 Fast Ethernet
9.5.3 Gigabit Ethernet
9.5.4 10 Gigabit Ethernet
9.6 Summary
9.7 Review Your Learnings
9.8 Sample Questions:
9.9 References for further reading
9.0 OBJECTIVES
1.Define the functions of Data Link Layer
2.Describe protocols used in MAC layer
3.Understand IEEE standards used in computer networks
4.Differentiate between various Ethernet connectivity
5.Identify communication network type based on its working like
cellular, Bluetooth, Wi -max, etc.
6.Demonstrate use of Virtual LAN
9.1 INTRODUCTION
The medium access control (MAC) is a sub layer of the data link
layer of the open system interconnections (OSI) reference model for data
transmission. It is responsible for flow control and multiplexing for
transmission medium. It controls the transmission of data packets via
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176A media access control is a network data transfer policy that
determines how data is transmitted between two computer terminals
through a network cable. The media access control policy involves sub-
layers of the data link layer 2 in the OSI reference model.
Figure 9.1: Mac Layer in OSI Reference Layer
Functions of MAC Layer
oIt provides an abstraction of the physical layer to the LLC and upper
layers of the OSI network.
oIt is responsible for encapsulating frames so that they are suitable for
transmission via the physical medium.
oIt resolves the addressing of source node as well as the destination
node, or gro ups of destination nodes.
oIt performs multiple access resolutions when more than one data frame
is to be transmitted. It determines the channel access methods for
transmission.
oIt also performs collision resolution and initiating retransmission in
case of collisions. It generates the frame check sequences and thus
contributes to protection against transmission errors.
MAC Addresses
MAC address or media access control address is a unique identifier
allotted to a network interface controller (NIC) of a devic e. It is used as a
network address for data transmission within a network segment like
Ethernet, Wi -Fi, and Bluetooth.
MAC address is assigned to a network adapter at the time of
manufacturing. It is hardwired or hard coded in the network interface card
(NIC). A MAC address comprises of six groups of two hexadecimal
digits, separated by hyphens, colons, or no separators. An example of a
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177The essence of the MAC protocol is to ensure non -collision and
eases the transfer of d ata packets between two computer terminals. A
collision takes place when two or more terminals transmit
data/information simultaneously. This leads to a breakdown of
communication, which can prove costly for organizations that lean heavily
on data transmis sion. When nodes or nodes are connected and use a
common link, called a multipoint or broadcast link, we need a multiple -
access protocol to coordinate access to the link.
9.2 RANDOM ACCESS
In random acce ss or contention methods, no node is superior to another
node, and none is assigned the control over another. At each instance, a
node that has data to send uses a procedure defined by the protocol to
make a decision on whether or not to send. This decisio n depends on the
state of the medium (idle or busy). Two features give this method its
name. First, there is no scheduled time for a node to transmit.
Transmission is random among the nodes. That is why these methods are
called random access. Second, no ru les specify which node should send
next. Nodes compete with one another to access the medium. That is why
these methods are also called contention methods.
Figure 9.2: Multiple Access Protocols
In this, each node has the right to the medium without being
controlled by any other node. However, if more than one node tries to
send, there is an access conflict -collision -and the frames will be either
destroyed or modified. To avoid access c onflict or to resolve it when it
happens, each node follows a procedure that answers the following
questions:
When can the node access the medium?
What can the node do if the medium is busy?
How can the node determine the success or failure of the transmis sion?
What can the node do if there is an access conflict?
The random -access methods are developed from protocol known
as ALOHA, which used a very simple procedure called multiple access
(MA). The method was improved with the addition of a procedure thatmunotes.in

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178forces the node to sense the medium before transmitting. This was called
carrier sense multiple access. This method then developed into two
methods namely 1. Carrier Sense Multiple Access with Collision
Detection (CSMA/CD) and 2. Carrier Sense Multiple Acc esswith
Collision Avoidance (CSMA/CA). CSMA/CD tells the node what to do
when a collision is detected. CSMA/CA tries to avoid the collision.
ALOHA
ALOHA, the earliest random -access method, was designed for a
radio (wireless) LAN, but it can be used on an y shared medium. There can
be potential collisions in this arrangement as the medium is shared
between the nodes.
Pure ALOHA
The original ALOHA protocol is called pure ALOHA and is a
simple yetan elegant protocol. Each node sends a frame whenever it has a
frame to send. However, since there is only one channel to share, there is
the possibility of collision between frames from different nodes. Figure
shows an example of frame collisions in pure ALOHA.
There are four nodes (unrealistic assumption) that contend with
one another for access to the shared channel. The figure shows that each
node sends two frames; there are a total of eight frames on the shared
medium. Some of these frames collide because mul tiple frames are in
contention for the shared channel.
It is obvious that we need to resend the frames that have been
destroyed during transmission. The pure ALOHA protocol relies on
acknowledgments from the receiver. When a node sends a frame, it
expect s the receiver to send an acknowledgment. If the acknowledgment
does not arrive after a time -out period, the node assumes that the frame (or
the acknowledgment) has been destroyed and resends the frame. A
collision involves two or more nodes. If all these nodes try to resend their
frames after the time -out, the frames will collide again. Pure ALOHA
dictates that when the time -out period passes, each node waits a random
amount of time before resending its frame. The randomness will help
avoid more collisions . This time is the back -off time TB.munotes.in

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179Pure ALOHA has a second method to prevent congesting the
channel with retransmitted frames. After a maximum number of
retransmissions attempts Kmax' a node must give up and try later. Figure
shows the procedure for pure ALOHA based on the above strategy.
Figure 9.3: Procedure for Pure ALOHA Protocol
The time -out period is equal to the maximum possible round -trip
propagation delay, which is twice the amount of time required to send a
frame between the two most widely separated nodes (2 x Tp)' The back -
off time TBis a random value that normally depend son K(the number of
attempted unsuccessful transmissions). The formula for TBdepends on the
implementation. One common formula is the binary exponential back -
off.In this method, for each retransmission, a multiplier in the range 0 to
2K-1 is randomly chosen and multiplied by Tp(maximum propagation
time) or Trr(the average time required to send out a frame) to find TB'
Note th at in this procedure, the range of the random numbers increases
after each collision. The value of Kmaxis usually chosen as 15.
Slotted ALOHA
Pure ALOHA has a vulnerable time of 2 x Tfr.This is so bec ause
there is no rule that defines when the node can send. A node may send
soon after another node has started or soon before another node has
finished. Slotted ALOHA was invented to improve the efficiency of pure
ALOHA. In slotted ALOHA we divide the time into slots of Tfr and force
the node to send only at the beginning of the time slot. Figure shows an
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180
Figure 9.4: Frames in Slotted ALOHA
Because a node is allowed t o send only at the beginning of the
synchronized timeslot, if a node misses this moment, it must wait until the
beginning of the next timeslot. This means that the node which started at
the beginning of this slot has already finished sending its frame. Of course,
there is still the possibility of collision if two nodes try to send at the
beginning of the same time slot. However, the vulnerable time is now
reduced to one -half, equal to Tfr. Figure shows the situation. Figure
shows that the vulnerable time f or slotted ALOHA is one -half that of pure
ALOHA.
Slotted ALOHA vulnerable time = Tfr
Throughput It can be proved that the average number of successful
transmissions for slotted ALOHA is S = G x e-G.The maximum
throughput Smax is 0.368, when G = 1.In other words, if a frame is
generated during one frame transmission time, then 36.8percent of these
frames reach their destination successfully. This result can be expected
because the vulnerable time is e qual to the frame transmission time.
Therefore, if a node generates only one frame in this vulnerable time (and
no other node generates a frame during this time), the frame will reach its
destination successfully.
Carrier Sense Multiple Access (CSMA)
To m inimize the chance of collision and increase the performance,
the CSMA method was formed. The chance of collision can be reduced if
a node senses the medium before trying to use it. CSMA is based on the
principle "sense before transmit" or "listen before t alk."CSMA can reduce
the possibility of collision, but it cannot eliminate it. The reason for this is
shown in Figure a space and time model of a CSMA network. Nodes are
connected to a shared channel. The possibility of collision still exists
because of p ropagation delay; when a nodes ends a frame, it still takes
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181Persistence Methods
What should a node do if the channel is busy? What should a node
do if the channel is idle? Three met hods have been devised to answer
these questions: the I -persistent method, the no persistent method, and the
p-persistent method. Figure shows the behaviour of three persistence
methods when a node finds a channel busy.
I-Persistent :TheI-persistent method is simple and straightforward. In
this method, after the node finds the line idle, it sends its frame
immediately (with probability I).This method has the highest chance of
collision because two or more nodes may find the line idle and send their
frames immediately.
No persistent In theno persistent method, a node that has a frame to
send senses the line. If the line is idle, it sends immediately. If the line is
not idle, it waits a random amount of time and then senses the line again .
The no persistent approach reduces the chance of collision because it is
unlikely that two or more nodes will wait the same amount of time and
retry to send simultaneously. However, this method reduces the efficiency
of the network because the medium rem ains idle when there may
be nodes with frames to send.
p-Persistent: The p-persistent method is used if the channel has time
slots with a slot duration equal to or greater than the maximum
propagation time. The p -persistent approach combines the advantage so f
the other two strategies. It reduces the chance of collision and
improves efficiency. In this method, after the node finds the line idle it
follows these steps:1. With probability p,the node sends its frame.
2. With probability q=1-p,the node wa its for the beginning of the next
time slot and checks the line again.
a. If the line is idle, it goes to step 1.
b. If the line is busy, it acts as though a collision has occurred and uses the
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182
Figure 9.5: Behaviour of Three Persistence Methods
Carrier Sense Multiple Access with Collision Detection (CSMA/CD)
The CSMA method does not specify the procedure following a
collision. Carrier sense multiple access with collision detection
(CSMA/CD) aug ments the algorithm to handle the collision. In this
method, a node monitors the medium after it sends a frame to see if the
transmission was successful. If so, the node is finished. If, however, there
is a collision, the frame is sent again. To better understand CSMA/CD, let
us look at the first bits transmitted by the two nodes involved in the
collision. Although each node continues to send bits in the
frame until it detects the collision, we show what happens as the first bits
collide. In Figure nodes A and C are involved in the collision.
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183Carrier Sense Multiple Access with Collision Avoidance
(CSMA/CA) .The basic idea behind CSMA/CD is that a node needs to be
able to receive while transmitting to detect a collision. When there is no
collision, the node receives one signal: its own signal. When there is a
collision, the node receives two signals: its own signal and the signal
transmitt ed by a second node. To distinguish between these two cases, the
received signals in these two cases must be significantly different. In other
words, the signal from the second node needs to add a significant amount
of energy to the one created by the firs t node.
Collisions are avoided through the use of CSMA ICA's three strategies:
the inter frame space, the contention window, and acknowledgments, as
shown in Figure.
Figure 9.7:T i m i n gi nC S M A / C A
Contention Win dow
The contention window is an amount of time divided into slots. A
node that is ready to send chooses a random number of slots as its wait
time. The number of slots in the window changes according to the binary
exponential back -off strategy. This means t hatit is set to one slot the first
time and then doubles each time the node cannot detect an
idle channel after the IFS time. This is very similar to the p -persistent
method except that a random outcome defines the number of slots taken
by the waiting nod e. One interesting point about the contention window is
that the node needs to sense the channel after each time slot. However, if
the node finds the channel busy, it does not restart the process; it just stops
the timer and restarts it when the channel is sensed as idle. This gives
priority to the node with the longest waiting time.
9.3 CONTROLLED ACCESS
Incontrolled access, the nodes consult one another to find which node
has the right to send. A node cannot send unless it has been authorized by
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1841.Reservation
In the reservation method, a no de needs to make a reservation
before sending data. Time is divided into intervals. In each interval, a
reservation frame precedes the data frames sent in that interval.
Ifthere are Nnodes in the system, there are exactly Nreservation
minis lots in the reservation frame. Each minis lot belongs to a node.
When a node needs to send a data frame, it makes a reservation in its own
minis lot. The nodes that have made reservations can send their data
frames after the reservation f rame. Figure shows a situation with five
nodes and a five -minis lot reservation frame. In the first interval, only
nodes 1, 3, and 4 have made reservations. In the second interval, only
node 1 has made a reservation.
2.Polling
Polling works with topologies in which one device is designated as
a primary node and the other devices are secondary nodes. All data
exchanges must be made through the primary device even when the
ultimate destination is a secondary device. The primary device controls the
link; the se condary devices follow its instructions. It is up to the primary
device to determine which device is allowed to use the channel ata given
time. The primary device, therefore, is always the initiator of a session.
Figure 9.8: Select and Poll function
Ifthe primary wants to receive data, it asks the second Aries if they have
anything to send; this is called poll function. If the primary wants to send
data, it tells the secondary to get ready to receive; this is cal led select
function
Select
Theselect function is used whenever the primary device has something to
send. Remember that the primary controls the link. If the primary is
neither sending nor receiving data, it knows the link is available. If it has
something to send, the primary device sends it. What it does not know,
however, is whether the target device is prepared to receive. So the
primary must alert the secondary to the upcoming transmission and wait
for an acknowledgment of the second ary's ready status. Before sending
data, the primary creates and transmits a select (SEL) frame, one field of
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185Poll
The poll function is used by the primary device to solicit transmissions
from the secon dary devices. When the primary is ready to receive data, it
must ask (poll) each device in turn if it has anything to send. When the
first secondary is approached, it responds either with a NAK frame if it
has nothing to send or with data (in the form of adata frame) if it does. If
the response is negative (a NAK frame), then the primary polls the next
secondary in the same manner until it finds one with data to send. When
the response is positive (a data frame), the primary reads the frame and
returns an acknowledgment (ACK frame), verifying its receipt.
3. Token Passing
In the token -passing method, the nodes in a network are organized
in a logical ring. In other words, for each node, there is a predecessor and
asuccessor. The predecessor is the node whi ch is logically before the
node in the ring; the successor is the node which is after the node in the
ring. The current node is the one that is accessing the channel now. The
right to this access has been passed from the predecessor to the current
node. Th e right will be passed to the successor when the current node has
no more data to send.
But how is the right to access the channel passed from one node to
another? In this method, a special packet called a token circulates through
the ring. The possession of the token gives the node the right to access the
channel and send its data. When a node has some data to send, it waits
until it receives the token from its predecessor. It then holds the token and
sends its data. When the node has no more data to send , it releases the
token, passing it to the next logical node in the ring. The node cannot send
data until it receives the token again in the next round. In this process,
when a node receives the token and has no data to send, it just passes the
data to the next node.
Token management is needed for this access method. Nodes must
be limited in the time they can have possession of the token. The token
must be monitored to ensure it has not been lost or destroyed. For
example, if a node that is holding the tok en fails, the token will disappear
from the network. Another function of token management is to assign
priorities to the nodes and to the types of data being transmitted. And
finally, token management is needed to make low -priority nodes release
the token to high priority nodes.
Logical Ring
In a token -passing network, nodes do not have to be physically
connected in a ring; the ring can be a logical one. Figure 12.20 show four
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Figure 9.9: Logical Ring and Physical Topology in token -passing access
method
In the physical ring topology, when a node sends the token to its
successor, the token cannot be seen by other nodes; the successor is the
next one in line. This means that the token does not have to have the
address of the next successor. The problem with this topology is that if one
of the links -the medium between two adjacent nodes fails, the whole
system fails.
In the bus ring topology, also called a token bus, the nodes are
connected to a single cable called a bus. They, however, make a logical
ring, because each node knows the address of its successor (and also
predecessor for token management purposes).When a node has finished
sending its data, it releases the token and inserts the address of its
successor in the token. Only the node with the address matching the
destination address of the token gets the token to access the shared media.
The Token Bus LAN, standard ized by IEEE, uses this topology.
In a star ring topology, the physical topology is a star. There is a
hub, however, that acts as the connector. The wiring inside the hub makes
the ring; the nodes are connected to this ring through the two wire
connection s. This topology makes the net workless prone to failure
because if a link goes down, it will be bypassed by the hub and the rest of
the nodes can operate. Also adding and removing nodes from the ring is
easier.
9.4 CHANNELIZATION
Channelization is a multiple -access method in which the available
bandwidth of a link is shared in time, frequency, or through code, between
different nodes. There are three channelization protocols: FDMA, TDMA,
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1879.4.1 Frequency -Divis ion Multiple Access (FDMA)
In frequency -division multiple access (FDMA), the available
bandwidth is divided into frequency bands. Each node is allocated a band
to send its data. In other words, each band is reserved for a specific node,
and it belongs to the node all the time. Each node also uses a band pass
filter to confine the transmitter frequencies. To prevent node interferences,
the allocated bands are separated from one another by small guard bands.
Figure shows the idea of FDMA.
Figure 9.10: FDMA
FDMA specifies a predetermined frequency band for the entire
period of communication. This means that stream data (a continuous flow
of data that may not be packetized) can easily be used with FDMA. This
can be used in cellular telephone systems.
FDM is a physical layer technique that combines the loads from
low-bandwidth channels and transmits them by using a high-bandwidth
channel. The channels that are combined are low pass. The multiplexer
modulates the signals, combines them, and creates a band pass signal. The
bandwidth of each channel is shifted by the multiplexer. FDMA is an
access method in the data lin k layer. The data link layer in each node tells
its physical layer to make a band pass signal from the data passed to it.
The signal must be created in the allocated band. There is no physical
multiplexer at the physical layer. The signals created at each node are
automatically band pass filtered. They are mixed when they are sent to the
common channel .
9.4.2 Time -Division Multiple Access (TDMA)
In time -division multiple access (TDMA), the nodes share the
bandwidth of the channel in time. Each node is allocated a time slot during
which it can send data. Each node transmits its data in is assigned time
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188
Figure 9.11:T D M A
The main problem with TD MA lies in achieving synchronization
between the different nodes. Each node needs to know the beginning of its
slot and the location of its slot. This may be difficult because of
propagation delays introduced in the system if the nodes are spread over a
large area.
9.4.3 Code -Division Multiple Access (CDMA)
Code -division multiple access (CDMA) was conceived several
decades ago. Recent advances in electronic technology have finally made
its implementation possible. CDMA differs from FDMA because only one
channel occupies the entire bandwidth of the link. It differs from TDMA
because all nodes can send data simultaneously; there is no timesharing.
Example:
CDMA simply means communication with different codes. For example,
in a large room with many people, two people can talk in English if
nobody else understands English. Another two people can talk in Chinese
if they are the only ones who understand Chinese, and so on. In other
words, the common channel, the sp ace of the room in this case, can easily
allow communication between several couples, but in different languages
(codes).
9.5 WIRED LANS
A local area network (LAN) is a computer network that is designed
for a limited g eographic area such as a building or a campus. Although a
LAN can be used as an isolated network to connect computers in an
organization for the sole purpose of sharing resources, most LANs today
are also linked to a wide area network(WAN) or the Internet. The LAN
market has seen several technologies such as Ethernet, Token Ring, Token
Bus, FDDI, and ATM LAN.
9.5.1 IEEE Standards
In 1987, the American National Standards Institute (ANSI) has
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189Organization for Standardization (ISO) as an international standard under
the designation ISO 8802. The relationship of the 802 Stan dard to the
traditional OSI model is shown in following figure. The IEEE has
subdivided the data link layer into two sub -layers: logical link control
(LLC) and media access control (MAC). IEEE has also created several
physical layer standards for different LAN protocols.
Figure 9.12: IEEE Standards for LAN
Data Link Layer
The data link layer in the IEEE standard is divided into two sub-
layers: LLC and MAC.
1.Logical Link Control (LLC)
In IEEE Project 802, flow control, error control, and part of the
framing duties are collected into one sub -layer called the logical link
control. Framing is handled in both the LLC sub layer and the MAC sub
layer. The LLC provides one single data link con trol protocol for all IEEE
LANs. In this way, the LLC is different from the media access control sub
layer, which provides different protocols for different LANs. A single
LLC protocol can provide inter -connectivity between different LANs
because it makes the MAC sub layer transparent.
Media Access Control (MAC)
Media access control layer defines the specific access method for
each LAN. For example, it defines CSMA/CD as the media access method
for Ethernet LANs and the token passing method for Token Ring and
Token Bus LANs. Framing function is also handled by the MAC layer.
MAC sub layer contains a number of distinct modules which defines the
access method and the framing format specific to the corresponding LAN
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190Physical Layer
The physical layer is dependent on the implementation and type of
physical media used. IEEE defines detailed specifications for each LAN
implementation .
Figure 9.13: Ethernet Evolutions
Addressing
Each node on a n Ethernet network (such as a PC, work node, or printer)
has its own network interface card (NIC). The NIC fits inside the node and
provides the node with a 6 -byte physical address. As shown in following
figure, the Ethernet address is 6 bytes(48 bits), no rmally written in
hexadecimal notation, with a colon between the bytes.
Figure: Example of Ethernet address in hexadecimal notation
Figure 9.14: Standard Ethernet Categories
9.5.2 Fast Ethernet
Fast Ethernet is a variation of Ethernet standards that carry data traffic at
100 Mbps (Megabits per second) in local area networks (LAN). It was
launched as the IEEE 802.3u standa rd in 1995and stayed the fastest
network till the introduction of Gigabit Ethernet.
Fast Ethernet is popularly named as 100 -BASE -X. Here, 100 is the
maximum throughput, i.e., 100 Mbps, BASE denoted use of base band
transmission, and X is the type of mediu m used, which is TX or FX.
The common varieties of fast Ethernet are 100 -Base -TX, 100 -BASE -FX
and 100 -Base -T4 as shown in figure below.munotes.in

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Figure 9.15: Categories of Fast Ethernet
Frame Format of IEEE 802.3
The frame format of IEEE 802.3u is same as IEEE 802.3. The
fields in the frame are:
Preamble −It is a 7 -bytes starting field that provides alert and timing
pulse for transmission.
Start of Frame Delimiter (SOF) −It is a 1 -byte field that contains an
alternating pattern of ones and zeros ending with two ones.
Destination Address −It is a 6 -byte field containing physical address
of destination nodes.
Source Address −It is a 6 -byte field containing the physical ad dress
of the sending node.
Length −It a 2 -bytes field that stores the number of bytes in the data
field.
Data −This is a variable sized field carries the data from the upper
layers. The maximum size of data field is 1500 bytes.
Padding −This is added to the data to bring its length to the minimum
requirement of 46 bytes.
CRC −CRC stands for cyclic redundancy check. It contains the error
detection information
9.5.3 Gigabit Ethernet
Gigabit Ethernet (GbE) is the family of Ethernet technologies that
achieve theoretical data rates of 1 gigabit per second (1 Gbps). It was
introduced in 1999 and was defined by the IEEE 802.3ab standard. The
popular varieties of fast Ethernet are 1000Base -SX, 1000Base -LX,
1000BASE -T and 1000Base -CX.
The goals of the Gigabit Ethernet design can be summarized as follows:
1.Upgrade the data rate to 1 Gbps.
2.Make it compatible with Standard or Fast Ethernet.
3.Use the same 48 -bit address.
4.Use the same frame format.
5.Keep the same minimum and maximum frame lengt hs.
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1929.5.4 10 Gigabit Ethernet
10-Gigabit Ethernet is the family of Ethernet technologies that
achieve maximum rates up to 10 gigabits per second (10 Gbps). It is also
known as 10GE, 10GbE or 10 Gig E. It is defined by the IEEE 802.3ae -
2002 standard.
10GE is a thousand times faster than standard Ethernet and
supports only full -duplex communication. Multimode fibre having 0.85 μ
frequency is used for medium distances and single m ode fibre having 1.5 μ
frequency is used for long distances.
The popular varieties of fast Ethernet are 1000Base -SX, 1000Base -
LX, 1000BASE -T and 1000Base -CX.
The goals of the Ten -Gigabit Ethernet design can be summarized
as follows:
1.Upgrade the data rate to 10 Gbps.
2.Make it compatible with Standard, Fast, and Gigabit Ethernet.
3.Use the same 48 -bit address.
4.Use the same frame format.
5.Keep the same minimum and maximum frame lengths.
6.Allow the interconnection of existing L ANs into a metropolitan area
network (MAN)or a wide area network (WAN).
7.Make Ethernet compatible with technologies such as Frame Relay and
ATM
9.6 SUMMARY
We can consider the data link layer as two sub layers. The uppe r sub
layer is responsible for data link control, and the lower sub layer is
responsible for resolving access to the shared media.
Many formal protocols have been devised to handle access to a shared
link. We categorize them into three groups: random acces s protocols,
controlled access protocols, and channelization protocols.
In random access or contention methods, no node is superior to
another node and none is assigned the control over another.
ALOHA allows multiple access (MA) to the shared medium. There are
potential collisions in this arrangement. When a node sends data,
another node may attempt to do so at the same time. The data from the
two nodes collide and become garbled.
Channelization is a multiple -access method in which the available
bandwidth o falink is shared in time, frequency, or through code,
between different nodes. We discussed three channelization protocols:
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193The common Fast Ethernet implementations are 1OOBase -TX (two
pairs of twisted pair cable), lOO Base -FX (two fiber-optic cables), and
100Base -T4 (four pairs of voice -grade, or higher, twisted -pair cable).
Gigabit Ethernet has a data rate of 1000 Mbps
9.7 REVIEW YOUR LEARNINGS:
1.List three categories of multiple access protocols discussed in this
chapter.
2.Define random access and list three protocols in this category
3.Define controlled access and list three protocols in this category.
4.Define channelization and list three protocols in this category.
5.Explain why collision is an issue in a random -access protocol but not
in controlled access or channelizing protocols.
6.Compare and contrast a random -access protocol with a controlled
access protocol.
7.Compare and contrast a random -access protocol with a channelizin g
protocol.
8.Compare and contrast a controlled access protocol with a channelizing
protocol.
9.Do we need a multiple access protocol when we use the 1ocalloop of
the telephone company to access the Internet? Why?
9.8 SAMPLE QUESTIONS:
1.In a CDMAlCD network with a data rate of 10 Mbps, the minimum
frame size is found to be 512 bits for the correct operation of the
collision detection process. What should be the minimum frame size if
we increase the data rate to 100 Mbps? To1 Gbps? To 10 Gbps?
2.One hundred nodes on a pure ALOHA network share a l -Mbps
channel. If frames are 1000 bits long, find the throughput if each node
is sending 10 frames per second?
3.Compare the data rates for Standard Ethernet, Fast Ethernet, Gigabit
Ethernet, and Ten-Gigabit Ethernet
4.What are the common Ten -Gigabit Ethernet implementations?
5.Explain CSMA/CD.
6.Explain Fast Ethernet and its types .munotes.in

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1949.9 REFERENCES FOR FURTHER READING
Data Communication and Networking, Behrouz A. Forouzan , Tata
McGraw Hill Fifth Edition 2013
Computer Networks, Andrew S. Tanenbaum,
https://nptel.ac.in/courses/106/105/106105183/
https://nptel.ac.in/content/storage2/courses/106105080/pdf/M5L2.pdf
https://www.coursera.org/lecture/peer -to-peer-protocols -local -area-
networks/medium -access -control -nWWWd

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19510
WIRELESS LAN, CONNECTING DEVICES
AND VIRTUAL LAN
Unit Structure :
10.0 Objectives
10.1 Connecting Devices
10.1.1 Passive Hubs
10.1.2 Repeaters
10.1.3 Active Hubs
10.1.4 Bridges
10.1.5 Two -Layer Switches
10.1.6 Routers
10.1.7 Three -Layer Switches
10.1.8 Gateway
10.2 Wireless LANs
10.2.1 IEEE 802.11
10.2.2 Bluetooth
10.3 Wi -Max
10.4 Cellular Telephony
10.5 Satellite Networks
10.6 V irtual LAN
10.6.1 Features of VLANs
10.6.2 Types of VLANs
10.6.3 Advantages of VLAN
10.7 Summary
10.8 Review Your Learnings
10.9 Sample Questions
10.10 References for further reading
10.0 OBJECTIVES
1.Explain IEEE 802.11 standards
2.Describe Bluetooth, Wi -max technologies.
3.Explain working of Cellular Networks.
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1965.Analyse connecting devices like routers, gateways and switches which
are used in network connections.
6.Explain Virtual LAN
10.1CONNECTING DEVICES
In this section, we divide connecting devices into five different
categories based on the layer in which they operate in a network, as shown
in figure below:
Figure 10.1: Five Categories of Connecting Devices
The five categories contain devices which can be defined as
1. Those which operate below the physical layer such as a passive hub.
2. Those wh ich operate at the physical layer (a repeater or an active hub).
3. Those which operate at the physical and data link layers (a bridge or a
two-layer switch).
4. Those which operate at the physical, data link, and network layers (a
router or a three -layer switch).
5. Those which can operate at all five layers (a gateway).
10.1.1Passive Hubs
A passive hub is just a connector. It connects the wires coming
from different branches. In a star -topology Ethernet LAN, a passive hub is
justa point where the signals coming from different stations collide; the
hub is the collision point. This type of a hub is part of the media; its
location in the Internet model is below the physical layer.
10.1.2Repeaters
A repeater i s a device that operates only in the physical layer.
Signals that carry information within a network can travel a fixed distance
before attenuation endangers the integrity of the data. A repeater receives
a signal and, before it becomes too weak or corrupt ed, regenerates the
original bit pattern. The repeater then sends the refreshed signal. A
repeater can extend the physical length of a LAN, as shown in figure
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Figure 10.2: Repeater connecting 2 segments of LAN
A repeater does not actually connect two LANs; it connects two
segments of the same LAN. The segments connected are still part of one
single LAN. A repeater is not a device that can connect two LANs of
different pro tocols.
A repeater can overcome the 10Base5 Ethernet length restriction.
In this standard, the length of the cable is limited to 500 m. To extend this
length, we divide the cable into segments and install repeaters between
segments. Note that the whole ne twork is still considered one LAN, but
the portions of the network separated by repeaters are called segments.
The repeater acts as a two -port node, but operates only in the physical
layer. When it receives a frame from any of the ports, it regenerates and
forwards it to the other port.
10.1.3 Active Hubs
An active hub is actually a multipart repeater. It is normally use d
to create connections between stations in a physical star topology. We
have seen examples of hubs in some Ethernet implementations ( 10Base-
T, for example). However, hubs can also be used to create multiple levels
of hierarchy, as shown in figure below. The hierarchical use of hubs
removes the length limitation of 10Base -T (100 m).
Figure 10.3: Hierarchy of Hubs
10.1.4 Bridges
A bridge operates in both the physical and the data link laye r. As a
physical layer device, it regenerates the signal it receives. As a data link
layer device, the bridge can check the physical (MAC) addresses (source
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198What is the difference in functionality between a bridge a nd a
repeater? A bridge has filtering capability. It can check the destination
address of a frame and decide if the frame should be forwarded or
dropped. If the frame is to be forwarded, the decision must specify the
port. A bridge has a table that maps ad dress to ports.
In following figure, two LANs are connected by a bridge. If a
frame destined for station 712B13456142 arrives at port 1, the bridge
consults its table to find the departing port. According to its table, frames
for 7l2B 13456142 leave throu gh port 1; therefore, there is no need for
forwarding, and the frame is dropped. On the other hand, if a frame for
712B13456141 arrives at port 2, the departing port is port 1and the frame
is forwarded. In the first case, LAN 2 remains free of traffic; in the second
case, both LANs have traffic. In this example, we used a two -port bridge;
in reality a bridge usually has more ports.
Figure 10.4: A bridge connecting 2 LANs
10.1.5 Two -Layer Switches
When we use the term switch, we must be careful because a switch
can mean two different things. We must clarify the term by adding the
level at which the device operates. We can have a two -layer switch or a
three -layer switch. A three -layer sw itchis used at the network layer; it is
a kind of router. The two-layer switch performs at the physical and data
link layers. At w o -layer switch is a bridge, a bridge with many ports and a
design that allows better (faster) performance
10.1.6 Routers
A router is a three -layer device that routes packets based on their
logical addresses (host -to-host addressing). A router normally connects
LANs and WANs in the Internet and has a routing table that is used for
making decisions about t he route. The routing tables are normally
dynamic and are updated using routing protocols. Figure shows a part of
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Figure 10.5: Routers Connecting Independent LAN and WAN
10.1.7 Three -Layer Switches
A three -layer switch is a router, but a faster and more
sophisticated. The switching fabric in a three -layer switch allows faster
table lookup and forwarding. In this book, we use the terms router and
three -layer switch interchangeably.
10.1.8 Gateway
Although some textbooks use the terms gateway and router
interchangeably, most of the literature distinguishes between the two. A
gateway is normally a computer that operates in all five layers of the
Internet or seven layers of OSI model. A gateway takes an application
message, reads it, and interprets it. This means that it can be used as a
connecting device between two internet works tha t use different models.
For example, a network designed to use the OSI model can be connected
to another network using the Internet model. The gateway connecting the
two systems can take a frame as it arrives from the first system, move it up
to the OSI ap plication layer, and remove the message.
10.2 WIRELESS LANS
Wireless communication is one of the fastest -growing
technologies. The demand for connecting devices without the use of
cables is increasing everywhere. Wirel ess LANs can be found on college
campuses, in office buildings, and in many public areas.
Here we will concentrate on two promising wireless technologies
for LANs: IEEE 802.11 wireless LANs, sometimes called wireless
Ethernet, and Bluetooth, a technology for small wireless LANs.
10.2.1 IEEE 802.11
IEEE 802.11 standard, popularly known as WiFi, lays down the
architecture and specifications of wireless LANs (WLANs). WiFi or
WLAN uses high -frequency radio waves instead of cables for connecting
the devices in LAN. Users connected by WLANs can move around within
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200IEEE 802.11 Architecture
The components of an IEEE 802.11 architecture are as follows −
Stations (STA) −Stations comprises of all devices and equipment that
are connected to the wireless LAN. A station can be of two types −
Wireless Access Point (WAP) −WAPs or simply access points (AP)
are generally wireless routers that form the base stations or access.
Client. Clients are workstations, computers, laptops, printers, smart
phones, etc.
Each station has a wireless network interface controller.
Basic Service Set (BSS) −A basic service set is a group of stations
communicating at the physical layer level. BS S can be of two
categories depending upon the mode of operation −
oInfrastructure BSS −Here, the devices communicate with other
devices through access points.
oIndependent BSS −Here, the devices communicate in a peer -to-peer
basis in an ad hoc manner.
1.Extended Service Set (ESS) −It is a set of all connected BSS.
2.Distribution System (DS) −It connects access points in ESS.
Frame Format of IEEE 802.11
The main fields of a frame of wireless LANs as laid down by IEEE
802.11 are −
Frame Control −It is a 2 -bytes starting field composed of 11
subfields. It contains control information of the frame.
Duration −It is a 2 -byte field that specifies the time per iod for which
the frame and its acknowledgment occupy the channel.
Address fields −There are three 6 -byte address fields containing
addresses of source, immediate destination, and final endpoint
respectively.
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201Data −This is a variable -sized field that carries the data from the
upper layers. The maximum size of the data field is 2312 bytes.
Check Sequence −It is a 4 -byte field containing error detection
information.
10.2.2 Bluetooth
Bluetooth is a network technology that connects mobile devices
wirelessly over a short -range to form a personal area network (PAN).
They use short -wavelength, ultra -high freque ncy (UHF) radio waves
within the range 2.400 to 2.485 GHz, instead of RS -232 data cables of
wired PANs.
Features of Bluetooth
Bluetooth technology was released in 1999 as Bluetooth 1.0, by
Special Interest Group (SIG) who continues to manage it.
It was initially standardized as IEEE 802.15.1.
Mobile computing devices and accessories are connected wirelessly by
Bluetooth using short -range, low -power, inexpensive radios.
UHF radio waves within the range of 2.400 to 2.485 GHz are using for
data commu nications.
A PAN or a piconet can be created by Bluetooth within a 10 m radius.
Presently, 2 to 8 devices may be connected.
Bluetooth protocols allow devices within the range to find Bluetooth
devices and connect with them. This is called pairing. Once, th e
devices are paired, they can transfer data securely.
Bluetooth has lower power consumption and lower implementation
costs than Wi -Fi. However, the range and transmission speeds are
typically lower than Wi -Fi.
The lower power requirements make it less sus ceptible to interference
with other wireless devices in the same 2.4GHz bandwidth.
Bluetooth version 3.0 and higher versions can deliver a data rate of 24
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202The Bluetooth version 4.0 came in 2010. It is characterized by low
energy consumption, multi vendor interoperability, the economy of
implementation, and greater range.
Bluetooth Devices
A Bluetooth device has a built -in short -range radio transmitter. The
current data rate is1 Mbps with a 2.4 -GHz bandwidth. This means that
there is a possibility of interference between the IEEE 802.11b wireless
LANs and Bluetooth LANs.
Bluetooth Layers:
Bluetooth uses several layers that do not exactly match those of the
Internet model and these are shown in figure below.
Radio Layer
The radio layer is roughly equivalent to the physical layer of the
Internet model. Bluetooth devices are low -power and have a range of 10
m.
Base band Layer
The base band layer is roughly equivalent to the MAC sub layer in
LANs. The access method is TDMA (see Chapter 12). The primary and
secondary communicate with each other using time slots. The length of a
time slot is exactly the same as the dwell time, 625 Ils. This means that
during the time that one frequency is used, a sender sends a frame to a
secondary, or a secondary sends a frame to the primary. Note that the
communication is only between the primary and a secondary; second
Aries cannot communicate directly with one another.
L2CAP
The Logical Link Control and Adaptation Protocol, or L2CAP (L2
here means LL), is roughly equivalent to the LLC sub layer in LANs. It is
used for data exchange on an ACL link; SCQ channels do not use L2CAP.
It uses 16 -bit length field which defines the s ize of the data, in
bytes, coming from the upper layers. Data can be up to 65,535 bytes. The
channel ID (CID) defines a unique identifier for the virtual channel
created at this level. The L2CAP has specific duties: multiplexing,
segmentation and reassembl y, quality of service (QoS), and group
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Figure 10.6: Bluetooth Layers
10.3 WI -MAX
WiMAX is one of the hottest broadband wireless technologies
around today. It is based on IEEE 802.16 specification, and it is expected
to deliver high quality broadband services. This is a brief tutorial that
covers the fundamentals of WiMAX.
Wireless means transmitting signals usin g radio waves as the
medium instead of wires. Wireless technologies are used for tasks as
simple as switching off the television or as complex as supplying the sales
force with information from an automated enterprise application while in
the field. Now co rdless keyboards and mice, PDAs, pagers and digital and
cellular phones have become part of our daily life.
Some of the inherent characteristics of wireless communications
systems which make it at tractive for users, are given below −
Mobility −A wireless communications system allows users to access
information beyond their desk and conduct business from anywhere
without having a wire connectivity.
Reach ability −Wireless communication systems enable people to be
stay connected and be reachable, regardless of the location they are
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204Simplicity −Wireless communication system are easy and fast to
deploy in comparison of cabled network. Initial setup c ost could be a
bit high but other advantages overcome that high cost.
Maintainability −In a wireless system, you do not have to spend too
much cost and time to maintain the network setup.
Roaming Services −Using a wireless network system, you can
provide service anywhere any time including train, buses, aeroplanes
etc.
New Services −Wireless communication systems provide various
smart services like SMS and MMS.
10.4 CELLULAR TELEPHONY
Cellular network is an underlying technology for mobile phones,
personal communication systems, wireless networking etc. The
technology is developed for mobile radio telephone to replace high power
transmitter/receiver systems. Cellular networks use lower power, shorter
range and more transmitters for data transmission.
Each cellular service area is divided into small regions called cells.
Each cell contai ns an antenna and is controlled by a solar or AC powered
network station, called the base station(BS). Each base station, in tum, is
controlled by a switching office, called a mobile switching center (MSC).
The MSC coordinates communication between all the base stations and
the telephone central office. It is a computerized center that is responsible
for connecting calls, recording call information, and billing as shown in
following figure.
Figure 10.7: Cellular Sy stem
Features of Cellular Systems
Wireless Cellular Systems solves the problem of spectral
congestion and increases user capacity. The features of cellular systems
are as follows −
Offer very high capacity in a limited spectrum.
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205Enable a fixed number of channels to serve an arbitrarily large number
of users by reusing the channel throughout the coverage region.
Communication is always between mobile and base station (not
directly between mobiles).
Each cellular base station is allocated a group of radio channels within
a small geographic area called a cell.
Neighbouring cells are assigned different channel groups.
By limiting the coverage area to within the boundary of the cell, the
channe l groups may be reused to cover different cells.
Keep interference levels within tolerable limits.
Frequency reuse or frequency planning.
Organization of Wireless Cellular Network.
Cellular network is organized into multiple low power transmitters
each 100 w or less.
Shape of Cells
The coverage area of cellular networks are divided into cells, each
cell having its own antenna for transmitting the signals. Each cell has its
own frequencies. Data communication in cellular networks is served by
its base statio n transmitter, receiver and its control unit.
Handoff
It may happen that, during a conversation, the mobile station moves from
one cell to another. When it does, the signal may become weak. To solve
this problem, the MSC monitors the level of the signal e very few seconds.
If the strength of the signal diminishes, the MSC seeks a new cell that can
better accommodate the communication. The MSC then changes the
channel carrying the call (hands the signal off from the old channel to a
new one).Hard Handoff Ear ly systems used a hard handoff. In a hard
handoff, a mobile station only communicates with one base station. When
the MS moves from one cell to another, communication must first be
broken with the previous base station before communication can be
establish ed with the new one. This may create a rough transition. Soft
Handoff New systems use a soft handoff. In this case, a mobile station can
communicate with two base stations at the same time. This means that,
during handoff, a mobile station may continue wit h the new base station
before breaking off from the old one.
Roaming
One feature of cellular telephony is called roaming. Roaming means, in
principle, that a user can have access to communication or can be reached
where there is coverage. A service provider usually has limited coverage.
Neigh boring service providers can provide extended coverage through a
roaming contract. The situation is similar to snail
mail between countries. The charge for delivery of a letter between two
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20610.5 SATELLITE NETWORKS
If communication takes place between any two earth stations
through a satellite, then it is called as satellite communication . In this
communication, electromagnetic waves are used as carrier signals. These
signals carry the information such as voice, audio, video or any other data
between ground and space and vice -versa.
In general terms, a satellite is a smaller object that r evolves around
a larger object in space. For example, moon is a natural satellite of earth.
We know that Communication refers to the exchange (sharing) of
information between two or more entities, through any medium or
channel. In other words, it is nothi ng but sending, receiving and
processing of information.
If the communication takes place between any two earth stations
through a satellite, then it is called as satellite communication . In this
communication, electromagnetic waves are used as carrier si gnals. These
signals carry the information such as voice, audio, video or any other data
between ground and space and vice -versa
Need of Satellite Communication
The following two kinds of propagation are used earlier for
communication up to some distance.
Ground wave propagation −Ground wave propagation is suitable for
frequencies up to 30MHz. This method of communication makes use
of the troposphere conditions of the earth.
Sky wave propagation −The suitable bandwidth for this type of
communication is broadly between 30 –40 MHz and it makes use of
the ionosphere properties of the earth.
The maximum hop or the station distance is limited to 1500KM
only in both ground wave propagation and sky wave propagation. Satellite
communication overcomes this limitation. In this method, satellites
provide communication for long distances , which is well beyond the line
of sight.
Since the satellites locate at certain height above earth, the
communication takes place between any two earth stations easily via
satellite. So, it overcomes the limitation of communication between two
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207
Figure 10.8: Satel lite Communication
How a Satellite Works
Asatellite is a body that moves around another body in a
particular path. A communication satellite is nothing but a microwave
repeater station in space. It is helpful in telecommunications, radio and
television a long with internet applications.
Arepeater is a circuit, which increases the strength of the received
signal and then transmits it. But, this repeater works as a transponder .
That means, it changes the frequency band of the transmitted signal from
the re ceived one.
The frequency with which, the signal is sent into the space is called
asUplink frequency . Similarly, the frequency with which, the signal is
sent by the transponder is called as Downlink frequency . The following
figure illustrates this concep tc l e a r l y .
Pros and Cons of Satellite Communication
In this section, let us have a look at the advantages and
disadvantages of satellite communication.
Following are the advantages of using satellite communication:
Area of coverage is more than that of te rrestrial systems
Each and every corner of the earth can be covered
Transmission cost is independent of coverage area
More bandwidth and broadcasting possibilites
Following are the disadvantages of using satellite communication −
Launching of satellites into orbits is a costly process.
Propagation delay of satellite systems is more than that of
conventional terrestrial systems.
Difficult to provide repairing activities if any problem occurs in a
satellite system.
Free space loss is more
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208Applications of Satellite Communication
Satellite communication plays a vital role in our daily life.
Following are the applications of satellite communication −
Radio broadcasting and voice communications
TV b roadcasting such as Direct To Home (DTH)
Internet applications such as providing Internet connection for data
transfer, GPS applications, Internet surfing, etc.
Military applications and navigations
Remote sensing applications
Weather condition monitoring & Forecasting
10.6 VIRTUAL LAN
Virtual Local Area Networks or Virtual LANs (VLANs) are a
logical group of computers that appear to be on the same LAN irrespective
of the configuration of the underlying physical network. Network
administrators partition the networks to match the function al requirements
of the VLANs so that each VLAN comprise of a subset of ports on a
single or multiple switches or bridges. This allows computers and devices
in a VLAN to communicate in the simulated environment as if it is a
separate LAN.
10.6.1 Features of VLANs
A VLAN forms sub -network grouping together devices on separate
physical LANs.
VLAN's help the network manager to segment LANs logically into
different broadcast domains.
VLANs func tion at layer 2, i.e., Data Link Layer of the OSI model.
There may be one or more network bridges or switches to form
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209Using VLANs, network administrators can easily partition a single
switched network into multiple networks dep ending upon the
functional and security requirements of their systems.
VLANs eliminate the requirement to run new cables or reconfiguring
physical connections in the present network infrastructure.
VLANs help large organizations to re -partition devices aim ing
improved traffic management.
VLANs also provide better security management allowing partitioning
of devices according to their security criteria and also by ensuring a
higher degree of control connected devices.
VLANs are more flexible than physical LA Ns since they are formed
by logical connections. This aid is quicker and cheaper reconfiguration
of devices when the logical partitioning needs to be changed.
10.6.2 Types of VLANs
Figure 10.9: Types of VLAN
Protocol VLAN −Here, the traffic is handled based on the protocol
used. A switch or bridge segregates, forwards or discards frames the
come to it based upon the traffics protocol.
Port-based VLAN −This is also called static VLAN. He re, the
network administrator assigns the ports on the switch / bridge to form
a virtual network.
Dynamic VLAN −Here, the network administrator simply defines
network membership according to device characteristics
10.6.3 Advantages of VLAN
There are several advantages to using VLANs:
1.Cost and Time Reduction :VLANs can reduce the migration cost of
stations going from one group to another. Physical reconfiguration
takes time and is costly. Instead of physically moving one station to
another segment or even to another switch, it is much easier and
quicker to move it by using software.
2.Creating Virtual Work Groups: VLANs can be used to create virtual
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210working on the same pr oject can send broadcast messages to one
another without the necessity of belonging to the same department.
This can reduce traffic if the multicasting capability of IP was
previously used.
3.Security :VLANs provide an extra measure of security. People
belon ging to the same group can send broadcast messages with the
guaranteed assurance that users in other groups will not receive these
messages.
10.7SUMMARY
IEEE 802.11 defines several physical layers, with different data rates
and modulating techniques.
Bluetooth is a wireless LAN technology that connects devices (called
gadgets) in a small area.
A Bluetooth network is called a Pico net. Multiple Pico nets fo rm a
network called a scatter net.
A Bluetooth network consists of one primary device and up to seven
secondary devices.
A backbone LAN allows several LANs to be connected.
A repeater is a connecting device that operates in the physical layer of
the Intern etmodel. A repeater regenerates a signal, connects segments
of a LAN, and has no filtering capability.
A bridge is a connecting device that operates in the physical and data
link layers of the Internet model.
A virtual local area network (VLAN) is configu red by software, not by
physical wiring.
Membership in a VLAN can be based on port numbers, MAC
addresses, IP addresses, IP multicast addresses, or a combination of
these features.
VLANs are cost -and time -efficient, can reduce network traffic, and
provide anextra measure of security
10.8 REVIEW YOUR LEARNINGS:
1.Match the layers in Bluetooth and the Internet model.
2.Can you explain various devices like switches, gateways, bridges and
routers used in network connections?
3.What are the two types of links between a Bluetooth primary and a
Bluetooth
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2114.In multiple -secondary communication, who uses the even -numbered
slots and who uses the odd -numbered slots?
5.How much time in a Bluetooth one -slot frame is used for the ho pping
mechanism?
What about a three -slot frame and a five -slot frame?
6.How does a VLAN save a company time and money?
7.Explain the working of Cellular Network Communication.
8.Which one has more overhead, a router or a gateway? Explain your
answer.
10.9 SAMPLE QUESTIONS:
1.How does a VLAN provide extra security for a network?
2.How does a VLAN reduce network traffic?
3.What is the basis for membership in a VLAN?
4.Which one has more overhead, a bridge or a router? Explain your
answer.
5.Which one has more overhead, a repeater or a bridge? Explain your
answer
10.10 REFERENCES FOR FURTHER READING
Data Communication and Networking, Behrouz A. Forouzan, Tata
McGraw Hill Fifth Edition 2013
Computer Networks, Andrew S. Tanenbaum
https://nptel.ac.in/courses/106/105/106105183/
https://nptel.ac.in/content/st orage2/courses/106105080/pdf/M5L2.pdf
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212Unit IV
11
INTRODUCTION TO NETWORK LAYER
Unit Structure
11.0 Objectives
11.1 Introduction
11.2 Network Layer Services
11.2.1 Packetizing
11.2.2 Routing
11.2.3 Forwarding
11.2.4 Error Control
11.2.5 Flow Control
11.2.6 Congestion Control
11.2.7 Quality of Service
11.2.8 Security
11.3 Packet Switching
11.3.1 Datagram Approach
11.3.2 Virtual -Circuit Approach
11.4 Network -Layer Performa nce
11.4.1 Delay
11.4.2 Throughput
11.4.3 Packet Loss
11.4.4 Congestion Control
11.4.4.1 Open -Loop Congestion Control
11.4.4.2 Closed -Loop Congestion Control
11.5 IPv4 Addresses
11.5.1 Address Space
11.5.2 Classful Addressing
11.5.2.1 Subnettin g and Supernetting
11.5.3 Classless Addressing
11.5.3.1 Slash Notation
11.5.4 Block allocation and address aggregation
11.5.5 Dynamic Host Configuration Protocol (DHCP)
11.5.6 Network Address Resolution (NAT)
11.6 Forwarding of IP Packets
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21311.6.2 Forwarding Based on Label
11.7 Summary
11.8 List of References
11.9 Unit End Exercise
11.0 OBJECTIVES
After going through this chapter, you will be able to
Understand the services provided by the net work layer.
Understand different packet switching techniques that occurs at the
network layer.
Understand the factors that affect network layer performance.
Understand how congestion can be controlled at the network layer.
Understand how addressing is mana ged at the network layer.
Understand how packets are forwarded at the network layer.
11.1INTRODUCTION
The network layer is the third layer in the TCP/IP reference suite
model.
It is responsible for host -to-host delivery of packets.
It provides service to its higher layer i.e., transport layer and receives
services from its lower layer i.e., data link layer.
11.2NETWORK LAYER SERVICES
With reference to the figure, we shall discuss the various services
provided by the network layer
The network layer is involved in source, destination and all routers in
the path that govern the transmission of packet.
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21411.2.1 PACKETI ZING
The important service is to create packets from datagram received by
the transport layer at the source and decapsulate the packet at the
destination.
The source host receives the payload from transport layer and adds a
header that contains the source and destination address and extra
information that is required at the network layer and forwards the
packet to the data link layer.
The destination host receives the network layer packet from its datalink
layer, decap sulates the packet and forwards the payload to the transport
layer.
The routers can only fragment the packet and not allowed to change
any information and add fragmentation information to the packet
header.
11.2.2 ROUTING
This service defines that network layer needs to find the best route
to deliver the packet from source to destination.
So, several algorithms and strategies are used by the network layer
to find out the best route.
11.2.3 FORWARDING
Forwarding is an activity individual router takes when the packet
arrives on one of its interfaces.
Routing decides about the entire path from source to destination
but forwarding is from one router to another.
A routing or a forwarding table is maintained at each router for
smooth forwarding of the packets.
The forwarding can be done on basis of destination address
available in the packet header or based on labels assigned
11.2.4 ERROR CONTROL
The network layer does not directly provide error control and it is
handled by the higher layers.
But it includes ch ecksum field that checks for corruption only in
the header but not the entire packet.
However, the ICMP protocol which is an auxiliary network layer
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21511.2.5 FLOW CONTROL
Flow control regulates the a mount of data a source can send
without overwhelming the receiver.
The network layer does not directly provide flow control as there is
no direct error control and upper layer uses the service of network
layer, so double flow control mechanism would increa se the
complexity and reduce the efficiency of the system.
11.2.6 CONGESTION CONTROL
Congestion in the network causes the packet to get flooded at one
area and as router cannot manage that it discards the packet.
So, error control mechanism at higher la yers cause retransmission
of packet and if situation worsens then it ends up no packet
reaching the destination.
The network layer applies several congestion policies to handle
such situation.
11.2.7 QUALITY OF SERVICE (QoS)
Internet allowed new multimedi a communication, audio -video
communication and so QoS has become vital.
The basics of this is started in the network layer but better
implementations are carried out in the higher layer.
11.2.8 SECURITY
When Internet was designed, security was not a need of the hour.
But as data communication grew to a wider scale, the need for
security emerged.
As network layer was already designed, we created another virtual
layer to implement security called IPSec.
11.3PACKET SWITCHING
The network layer implements packet switching as unit of data in
this layer is the packet.
Packet Switching transmits data across the network by breaking it
down into packets for more efficient transfer using various
network devices.
In order for faster transmission, the packets are fragmented and
each packet takes a different route to reach the destination and then
at the destination, reassembly takes place and packets are arranged
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216A packet switched network uses two different approach to r oute
the packets namely the datagram approach and the virtual circuit
approach.
11.3.1 DATAGRAM APPROACH
In this datagram approach, the network layer is responsible for delivery
of packet from source to destination.
Packets dynamically choose route from s ource to destination based on
the destination address.
The packets arrive out of order and hence at the destination it is
rearranged.
In case packet size is large, the intermediate router can further
fragment the packet but reassembly takes place at the de stination only.
The router maintains a simple routing or forwarding table with
destination address and output interface.
Figure 11.2 –A datagram approach –connectionless service
11.3.2 VIRTUAL -CIRCUIT APPROACH
In the virtual circuit approach, a connection -oriented service is
established between the packets and they take a fixed route to
reach from source to destination.
A logical path is established and each packet is routed based on
label or VC identifier.
Thetransfer of packets takes place through three phases.
Figure 11.3 –A virtual circuit approach –connection -oriented servicemunotes.in

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217The first phase is the set -up phase where the router creates an virtual
circuit entry in the routing table and tries to establish a logical path.
It sends a request packet from source to destination informing about its
VC ID.
The packet is moved forward ti ll it reaches the destination and each
router updates it with its unique VC.
The destination now sends the acknowledgment packet and the table is
updated with switching entries.
The second phase is data transfer phase, where all packets belonging to
one me ssage are routed in the logical route created based on the label
or VC ID.
The last phase is the tear down phase, where the source sends a special
packet called tear down packet to indicate the end of packet transfer.
The destination acknowledges with conf irmation packet and all the
routing entries are deleted in the logical path.
11.4NETWORK -LAYER PERFORMANCE
The network layer is not perfect.
The higher layers use the service of network layer and so
performance of network layer is vital.
The performance of a network is measured in terms of delay,
throughput, and packet loss.
We can improve the performance of the network through
congestion control.
11.4.1 DELAY
A packet as it is transmitted from source to destination encounters
delay and d oes not reach instantaneously as expected.
The delay in the network is dependent on transmission delay,
propagation delay, processing delay and queuing delay.
All packets cannot be sent instantaneously. So, the time taken to put all
packets on the link is thetransmission delay .
It is calculated as Delay tr= (Packet length) / (Transmission rate).
The time taken for a bit in a packet to travel from one point to another
is called the propagation delay .
It is calculated as Delay pg= (Distance) / (Propagation s peed).
Theprocessing delay is the time required for a router or a destination
host to receiv e a packet from its input port, remove the header, perform
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218in the case of a router or deliver the packet to the upper -layer protocol
in the case of the destination host.
It is calculat ed as Delay pr= Time required to process a packet in a
router or a destination host.
Thequeuing delay for a packet in a router is measured as the time a
packet waits in the input queue and output queue of a router.
It is calculated as Delay qu= The time a packet waits in input and output
queues in a router.
So, in a network with n routers there are n + 1 links and hence the
delay is calculated as Total delay (n1) (Delay trDelay pgDelay pr)(n)
(Delay qu)
11.4.2 THROUGHPUT
Throughput is the number of bits successfully transmitted per unit
time.
The transmission time defines the bits transmitted from one point to
another but throughput defines the time taken by the entire link.
For example, consider the simple network
Figure 11.4 –Network with different transmission rate
The three links have different transmission time i.e link 1 = 200 kbps,
link 2 = 100 kbps and link 3 = 150 kbps.
The throughput is calculated as the minimum transmission time i.e.,
Throughput minimum {TR1, TR2, TRn}.
So, for the above network the throughput is 100 kbps.
11.4.3 PACKET LOSS
The routers maintain buffers to store packets as they receive.
They process the packet and then decide to forward on correct
interface.
Thebuffer size is limited and if the processing time is longer the buffer
is in capable of storing the packets thereby causing the packet loss.
Loss of packet means the packet has to be retransmitted.
Over a period of time the network gets congested causing more packet
loss and therefore proper queues have to be maintained to prevent
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21911.4.4 CONGESTION CONTROL
Congestion is defined as a state occurring in network layer when
the message traf fic is heavy such that it slows down the response
time in the network.
Congestion can be controlled before it happens called as open loop
congestion control and remove the congestion after it has happened
called closed loop congestion control.
11.4.4.1 OPEN -LOOP CONGESTION CONTROL
Policies are applied such that congestion does not occur as prevention
is a good measure.
Retransmission policy and retransmission timers must be designed to
avoid congestion in the network as packet loss causes the pa cked to be
retransmitted.
So excessive retransmission can cause heavy congestion and this can
be avoided by designing optimized and efficient retransmission policy.
A good window policy also helps in preventing congestion.
A selective repeat strategy woul d be better choice as only corrupt
packed would be retransmitted and in Go -Back N all packets from the
corrupted one will be transmitted making several right packets to be
retransmitted congesting the network more.
Even an acknowledgement policy is vital i n preventing corruption by
not congesting the network for sending the acknowledgement for every
packet received but sending a cumulative acknowledgment.
Thediscarding policy adopted by the routers may prevent congestion
and same time will not harm integri ty of transmission as in case of
audio files where we can decide to discard only less sensitive packet
and thereby preserve the quality of the sound.
Finally, admission policy is used to prevent congestion in virtual
circuit networks where switches can che ck resource requirement before
admitting packets in the link.
11.4.4.2 CLOSED -LOOP CONGESTION CONTROL
Policies are applied after the congestion has occurred and alleviate
the level of effect it has caused.
In the Backpressure technique , the congested rout er informs its
immediate router to stop the sending more packets and this control
is carried till the upstream device.
Inchoke packet technique , the congested router sends a special
packet called choke directly to the source to slow down or stop
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220
Backpressure Technique Choke Packet Technique
Figure 11.5 –Closed -loop Congestion Control Technique
In the implicit signaling technique , the source makes assumptions
that congestion has occurred as it has not received
acknowledgement for long and thereby takes decision to slower the
rate of packet transmission by itself.
In the explicit signaling technique , the source receives a signal or
message that congestion has occurred hence decides to slower the
rate of packet transmission.
11.5IPv4 ADDRESSES
The addressing scheme used in the network layer for identifying each
device is the Internet address or the IP address version 4 (IPv4).
It is a numerical representation that uniquely identifies a specific
interface on the network.
11.5.1 ADDRESS SPACE
IPv4 uses 32 -bit addresses which limits the address
space to4,294,967,296 (232) addresses.
IPv4 address is represented 32 -bit binary value and most
commonly in dotted decimal notation which consist of four octets
of address in decima l format separated by dots.
Each byte (octet) is only 8 bits.
Each number in the dotted -decimal notation is between 0 and 255.
Figure 11.6 –IPv4 address representation
Each octet represents 8 -bit binary; so decimal number has to be
converted to eight bit binary number.
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Figure 11.7 –Format of an IPv 4 address
The first part is prefix which defines the network and the second
part is suffix which defines the host or the node.
The entire IPv4 address in 32 bits long where n bits define the
prefix length and 32 -n bits define the suffix length.
11.5.2 CLASSFUL ADDRESSING
When IPv4 was first designed, the prefix length was fixed and this
scheme is called the classful addressing.
The entire address space is divided into five classes.
Each class have a fixed number of blocks and each block has a fixed
number of hosts.
Class Leading
bitsNet ID
bitsHost ID
bitsNo. of
NetworksAddress per
NetworkStart
AddressEnd Address
Class
A0 8 24 27=1 2 8 224=1 6 , 7 7 7 , 2 1 6 0.0.0.0 127.255.255.255
Class
B10 16 16 214=1 6 , 3 8 4 216=6 5 , 5 3 6 128.0.0.0 191.255.255.255
Class
C110 24 8 221=2 , 0 9 7 , 1 5 2 28=2 5 6 192.0.0.0 223.255.255.255
Class
D1110 Not
DefinedNot
DefinedNot Defined Not Defined 224.0.0.0 239.255.255.255
Class
E1111 Not
DefinedNot
DefinedNot Defined Not Defined 240.0.0.0 255.255.255.255
Table 11.1 –Classful addressing scheme
So, we can easily determine the class by identifying the leading bits in
binary notation and start address in decimal notation.
The disadvantages of classful addressing scheme are
oIn class A, the number of addresses in each block is more than
enough for almost any organization and this results in wastage of
addresses.
oThe same logic applies with class B resulting in wastage of
addresses.
oWhereas a block in class C is too small tofulfil the addresses
requirement of an organization.
oEach address in class D defines a group of hosts that need
tomulticast the address leading to wastage of address.
oThe addresses of class E are reserved for the future purpose which
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222oFinally, we are not assigning address as per user requirements
causing either shortage or excess address being assigned.
So classful addressing leads to address depletion problem and hence
better address classification schemes need to be used.
11.5.2.1 SUBNETTING AND SUPERNETTING
To alleviate the address depletion problem, two strategies are designed
namely subnetting and Super netting.
Subnetting is a technique to divide the large network into small
networks or sub -networks.
In subnetting the netwo rk address bits are increased.
Supernetting is a technique to combine several small networks into a
large network.
In supernetting the host addresses bits are increased.
Figure 11.8 –Subnetting technique
11.5.3 CLASSLESS ADDRESSING
Subnetting and supernetting did not solve all of the address depletion
problem.
With growing internet requirements, more addresses were required.
So immediate short -term solution was to use the classless addressing
scheme.
Here variable -length blocks are used that belong to no classes i.e.,
whole address space is divided into variable length block with block
size being a power of 2 namely 20,21,22…. 232addresses.
Figure 11.9 –Classless addressing scheme with variable blocks
11.5.3.1 SLASH NOTATION
It is also referred as classless interdomain routing (CIDR) technique.
The prefix length is added to the address separated by slash (/) as
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Figure 11.10 –Slash notation representation
So given an address we can extract three information from it.
oThe number of addresses in the block can be found as N = 232−n or
N=N O T( mask) + 1.
oThe first address is found by keeping the n leftmost bits and set the
(32−n) rightmost bits all to 0s or by (Any address in the block)
AND (mask).
oThe last address is found by keeping the n leftmost bits and set the
(32−n) rightmost bits all to 1s or by (Any address in the block)
OR [(NOT (mask)].
The first address is called the network address and routing of packets
in the network is based on this address.
So given any address, we need to find the network address and t hen
router refers the routing or forwarding table to find the corresponding
interface to forward the packet.
11.5.4 BLOCK ALLOCATION AND ADDRESS AGGREGATION
The responsibility of block allocation is governed by Internet
Corporation for Assigned Names and Numbers (ICANN).
As per ICANN, two rules apply for proper operation of CIDR
oThe requested address N must be power of two as N = 232−n or n
= 32 −log2N must result in integer value.
oThe requested block should be from large contiguous location
making the first address a vital parameter and this is obtained as a
value which is divisible by the number of addresses in the block
i.e., first address = (prefix in decimal) × 232−n = (prefix in
decimal) × N.
The major advantage of CIDR notat ion is address aggregation also
known as address summarization or route summarization.
ICANN assigns a large block of addresses to an ISP and each ISP
divides its assigned block into smaller subblocks and grants the
subblocks to its customers.
It can also be coined as many blocks of addresses are aggregated into
one block and granted to one ISP.
IPv4 has certain addresses categorized as special address used for
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Table 11.2 –Special IPv4 Address
11.5.5 DYNAMIC HOST CONFIGURATION PROTOCOL (DHCP)
The Dynamic Host Configuration Protocol (DHCP) is a network
management protocol that is used to automatically assign IP address to
an organization after a block of addresses are assigned to it.
DHCP is application layer protocol using client -server paradigm and
performs the function for the network layer.
A DHCP server dynamically assigns an IP address and other network
configuration parameters such as the subnet mask, default gateway
address, do main name server (DNS) address and other pertinent
configuration parameters to each device on a network so they can
communicate.
The primary reason DHCP is used because it simplifies the
management of IP addresses on networks as no two hosts can have the
same IP address and configuring them manually leads to chances of
errors and wrong allocation.
DHCP protocol works by DHCP client sending the request message
and the DHCP server replying by the response message.
The DHCP message format is illustrated as fol lowsmunotes.in

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Figure 11.11 –DHCP Message Format
There are 8 message types used in DHCP namely
1-DHCPDISCOVER, 2 -DHCPOFFER, 3-DHCPREQUEST,
4-DHCPDECLINE, 5 -DHCPACK, 6 -DHCPNACK,
7-DHCPRELEASE, 8 -DHCPINFORM.
DHCP uses two well -known ports 67 and 68 for communication as the
message is broadcast.
Figure 11.12 –DHCP Operation
In addition to simplified management, the DHCP server provides
benefits such as accurate IP configuration, reduced IP address
conflicts, automation of IP address administration and efficient change
management.
DHCP uses services of UDP and provides error control through the
implementation of checksum and retransmission policy for missing
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22611.5.6 NETWORK ADDRESS RESOLUTION (NAT)
One public address is needed to access the Internet and private
addresses can be used in internal private network.
The private network addresses can route traffic inside the network well
but to access resource outside the Internet and obtain a response a
public or global address is needed.
So, the Network Address Translation (NAT) allows for easy mapping
of the private and public IP address.
This translation is done for both economic and security reasons.
All of the outgoing packets passing the NAT router w ill replaces the
source address in the packet with the global NAT address.
All incoming packets passing through the NAT router will replaces the
destination address in the packet (the NAT router global address) with
the appropriate private address.
Figure 11.13 –Address Translation Process
The translation table has two columns namely private address and
destination address of the packet.
The router translates the sou rce address of outgoing packet to routers
global address and notes it in the routing table.
When it receives a response, it checks the routing table entry and
changes the corresponding destination address which is NAT router
global address with the corresp onding private IP address.
With just one global address, only one internal host can communicate
with external host.
To remove this restriction, a NAT router maintains a pool of IP
addresses so that several internal host can communicate
simultaneously.
This allows many to one relationship but to have a good many to many
relationships then a modified translation table makes the job easier.
This is required when two internal host want to communicate with
same external host, then additional information would he lp in proper
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227Private
AddressPrivate
PortExternal
AddressExternal
PortTransport
Protocol
172.18.3.1 1400 25.8.3.2 80 TCP
172.18.3.2 1401 25.8.3.2 80 TCP
: : : : :
Table 11.3 –Translation Table
When the response from HTTP comes back, the combination of source
address (25.8.3.2) and destination port address (1401) defines the
private network host to which the response should be directed.
This is feasible because the port address defined are unique .
11.6 FORWARDING OF IP PACKETS
IP address play a vital role in the forwarding of the packets in the
network.
Forwarding a packet can be delivering a packet to the next hop which
could be an intermediate router or even the destination host.
IP is a connectionless protocol then forwarding is based on destination
address and if IP works as connection -oriented protocol then
forwarding is based on label.
11.6.1 FORWARDING BASED ON DESTINATION ADDRESS
In classless addressing the entire address space is one entity as blocks
are variable size.
The /n mask along with the destination address helps in determining in
which interface the packet has to be forwarded.
So, in a classless forwarding table four piec es of information helps in
forwarding decision -the mask, the network address, the interface
number, and the IP address of the next router.
Figure 11.14 –Forwarding module in classless addressing
schememunotes.in

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228The tab le is searched row wise and each row has n leftmost bits of
destination address kept and rest is set to 0s.
Several algorithms are proposed to find the prefix.
An easy proposal is to use the address aggregation where larger
addresses are aggregated into on e larger block thus minimizing the
routing table entries as all packets for that network would be
forwarded through that interface only.
Figure 11.15 –Address aggregation Technique
An alteration called longest mask matching is applied when an
organization connected to one router moves to another organization.
According to this principle, the forwarding table is stored as sorted
from longest to shortest mask.
In case it is not sorted then wrong mask might be applied resulting in
packet being forwarded at wrong interface.
Figure 11.16 –Longest Mask Matching Techniquemunotes.in

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229Another technique to solve huge routing tables is to create hierarchy
which d ecreases the size of the forwarding table.
Any number of hierarchies can be created following the basic principle
of classless addressing.
Figure 11.17 –Hierarchical Routing Technique
Further hierarchical routing is extended to geographical routing to
decrease the size of the routing table.
The entire region is geographically divided into large blocks and each
block is assigned an address and mask.
11.6.2 FORWARDING BASED ON LABEL
When IP is used as a connection -oriented protocol, the forwarding of
packets is based on a label attached to a packet.
Here the forwarding is done by accessing a table using an index and
switch helps in this forwarding decision.
Figure 11.18 –Forwarding based on label
The latest edition by IETF is to use Multi -Protocol Label Switching
(MPLS) where a router can forward the packet based on the
destination address; when behaving like a switch, it ca nf o r w a r da
packet based on the label.
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Figure 11.19 –MPLS Header
The MPLS header is actually a stack of sub headers that is used for
multilevel hierarch ical switching.
11.7 SUMMARY
The network layer in the Internet provides services to the transport
layer and receives services from the network layer.
The main services provided by the network layer are packetizing and
routing the packet from th e source to the destination.
One of the main duties of the network layer is to provide packet
switching.
Performance of the network layer is measured in terms of delay,
throughput, and packet loss.
Congestion control is a mechanism that can be used to impr ove the
performance.
IPv4 addressing managed the communication
Some problems of address shortage in the current version can be
temporarily alleviated using DHCP and NAT protocols.
Forwarding helps to understand how routers forward packets.
11.8 LIST OF R EFERENCES
“Data Communications and Networking” by Behrouz A. Forouzan, 5th
Edition, McGraw -Hill Publication
“Computer Networks” by Andrew Tanenbaum, 5th Edition, Pearson
Education
https://www.javatpoint.com/network -layer
https://www.paessler.com/it -explained/ip -address
11.9 UNIT END EXERCISE
1.What are the responsibilities of Network Layer?
2.Explain the three phases in the virtual circuit approach.
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2314.In classless addressing, c an two different blocks can have the same
prefix length? Explain.
5.Rewrite the IP address in binary notation.
a.125.23.65.123
b.226.36.16.244
6.Rewrite the IP address in dotted -decimal notation.
a.11011111 11000000 01110101 11011101
b.11101111 01011101 01110111 10000111
7.Find the class of the following classful IP address
a.130.34.2.1
b.01010111 1000100 10001110 00001111
8.Find the first address, last address and number of address in the block.
a.200.107.16.17/18
b.14.12.72.8/24
9.Combine the f ollowing three blocks of addresses into a single block:
16.27.24.0/26, 16.27.24.64/26 and 16.27.24.128/25
10.An ISP is granted the block 80.70.56.0/21. The ISP needs to allocate
addresses for two organizations each with 500 addresses, two
organizations each w ith 250 addresses, and three organizations each
with 50 addresses.
a.Find the number and range of addresses in the ISP block.
b.Find the range of addresses for each organization and the range of
unallocated addresses.
c.Show the outline of the address distribut ion and the forwarding
table.
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23212
NETWORK LAYER PROTOCOLS
Unit Structure
12.0 Objectives
12.1 Introduction
12.2 Internet Protocol (IPv4)
12.2.1 Datagram Format
12.2.2 Fragmentation
12.2.3 Options
12.2.4 Security of IPv4 Datagram
12.3 Internet Control Message Protocol ( ICMPv4)
12.3.1 Messages
12.3.1.1 Error -reporting Messages
12.3.1.2 Query Messages
12.3.2 Debugging Tools
12.4 Mobile IP
12.4.1 Addressing
12.4.2 Agents
12.4.3 Three Phases
12.4.4 Inefficiency in Mobile IP
12.5 Summary
12.6 List of References
12.7 Unit End Exercise
12.0 OBJECTIVES
After going through this chapter, you will be able to
Identify the roles of various protocols in the network layer
Identify the IPv4 datagram format
Understand the concept of fragmentation
Understand how ICMP4 helps in debugging
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23312.1 INTRODUCTION
The main duty of the network layer protocols is to route the packets
according to unique network device addresses and render flow and
congestion control to prevent network resource depletion.
The network layer has one main protocol namely Internet Protocol (IP)
and three auxiliary protocol namely Internet Control Message Protocol
(ICMP), Internet Group Management Protocol (IGMP) and Address
Resolution Protocol (ARP).
The IP and ICMP are available in version 4 and 6.
The IPv4 protocol is responsible for crea tion of packet, forwarding,
routing and delivering the packet to the destination host.
The ICMPv4 is responsible for error handling at the network layer.
The IGMP is responsible for multicasting in IPv4.
The ARP is responsible for IP address to MAC address mapping and
works for the data link layer though it is a network layer protocol.
We shall learn in this chapter about the IPv4 and ICMPv4 protocol.
Figure 12.1 –Protocols in the TCP/IP reference suite
12.2INTERNET PROTOCOL (IPv4)
IPv4 is a connectionless, unreliable protocol as each datagram is
handled independently and takes a different route to reach the
destination.
The network layer cannot handle large datagrams and so source
fragments the datagram into smaller ones.
The datagrams arrive out of order at destination and reassembly takes
place at the destination.
The intermediate routers can further fragment the datagram but cannot
perform reassembly.
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23412.2.1 DATAGRAM FORMAT
The important service of the network layer is to create the
datagrams.
The data obtained from higher layer are created in datagram by
adding additional information required by the network layer for
forwarding the packet.
The IPv4 datagram is variable length and consist of two parts:
header and data.
Figure 12.2 –IPv4 datagram
The header has minimum length of 20 bytes and maximum length of
60 bytes .
The field VER indicates the version number which is 4 bits and value
is 4 i.e., 0100.
The field HLEN defines the header length which is 4 -bit field and
calculated by multiplying the value of this field by 4.
For example, if HLEN = (0010) 2then 2 x 4 = 8 and since minimum
length is 20 bytes it means this packet is corrupted. If HLEN = (1000) 2
then 8 x 4 = 32 bytes means 20 bytes header and 12 bytes are options
and packet is not corrupted.
The field Service type is now redefined to different ial service and
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Figure 12.3 –Service Type / Differential Service
The total length field is 16 bit and it defines the total length i.e., header
+ data in bytes with maximum value 65535 (all 1s).
With this we can calculate the length of the data as total length –
(HLEN) x 4.
For exampl e, if HLEN = 5 and total length = (0028) 16then length of
data = 40 –5 x 4 = 20 bytes
The 16 -bit identification field identifies uniquely the datagram created
by the source. As large message is divided into smaller datagrams,
each datagram belongs to the same message needs to be identified.
The three -bit flag field has leftmost flag unused, middle flag indicating
do not fragment (D) and last flag indicating if its first or last fragment.
The 13 -bit field fragmentation offset defines relative position of th e
fragment with respect to whole datagram.
The time to live field indicates the number of hops i.e., routers visited
by the datagram.
The protocol field indicates the higher layer payload that is
encapsulated in the datagram. In other words, the data is cr eated from
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Figure 12.4 –Encapsulation of data and protocol values
The header checksum is a 16 -bit field which is 1s complement of the
sum of other fields to manage error checking for corrupted datagram
not reaching right destination, payload reaching wrong protocol or
problems during reassembly.
The source and destination address are 32 bit and uniquely identify the
sender and receiver of the datagram respectively.
Options can be up to 40 bytes and used for network testing and
debugging information.
Data or payload is the main part of the datagram as it is message
obtained for higher layer w hich is encapsulated and header attached
for proper delivery.
12.2.2 FRAGMENTATION
The network layer cannot handle large data and hence the data is
fragmented.
The fragmentation takes place when the when the maximum size of
datagram is greater than maxim um size of data that can be held a
frame i.e., its Maximum Transmission Unit (MTU).
The data flow should not be disrupted and data received from the
transport layer is fragmented.
The IP protocol is independent of physical network and the maximum
length of IP datagram is 65535 bytes.
Datagram is fragmented by source and intermediate routers and
reassembly is taken care by the destination host.
Three fields namely flag, fragmentation offset and total length are
changed for the fragmentation process.
The chec ksum must be calculated each time the datagram passes the
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Figure 12.5 –Fragmentation
In the figure, the source creates a datagram with header size as 20
bytes and payload w ith 500 bytes.
As the datagram reaches the router the, the router fragments into 3
smaller datagrams with header size remaining same for all three
fragments and data divided int three sizes namely 180 bytes, 180
bytes and remaining 140 bytes.
The destinat ion host identifies the datagram with the identification
bits in the IP header as each fragment carries the same identification
number.
The flag values indicate the fragmentation status.
Figure 12.6 –Flags
D means do not fragment bit. If D =1 the router should not fragment
the datagram and in that case if router cannot handle such large
datagram, then it discards it and sends an ICMP error message to
source. If D = 0 then datagram can be fragmented if required.
M means more fragment bit. If M = 1, then it is not the last but first
or intermediate fragment. If M = 0 then it indicates it is the last
fragment,
For the purpose of reassembly at the destinatio n host identifies the
sequence of datagram from the fragmentation offset.
For example, the original datagram of 4000 bytes is fragmented into
three fragments with bytes numbered 0 to 3999.
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Figure 12.7 –Fragmentation Scenario
The first fragment has offset 0/8 = 0 and carries bytes 0 to 1399.
The second fragment has offset 1400/8 = 175 and carries bytes 1400 to
2799.
The last fragment has offset 2800/8 = 350 and carries bytes 2800 to
3999.
So, the basic strategy involved in fragmentation is
oThe offset field for first fragment is always zero.
oDividing the length of first fragment by 8 gives offset of second
fragment.
oDividing the total length of first and second fragment by 8 gives
the offset for the third fragment.
oContinue the same calculation for remaining fragments.
oThe last fragment has M set to 0.
For example, a packet with M = 0 indicates it could be last fragment or
the original packet was not fragmented at all. If M = 1, then it could be
first fragment or intermediate fragment. If M = 1 and fragmentation
offset = 0 then it is first fragment only.
12.2.3 OPTIONS
Options provide additional information in the header for network
testing and debugging and are not compulsory to be the part of the
IPv4 header.
Options can be maximum 40 bytes.
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239AN o -Operation is used as filler between option and End -of-Option is
used as padding at end of field.
A record route option records addresses of up to 9 routers that handled
the datagram.
In strict source route option, the source host predetermines the rou te of
the datagram it has to travel and the datagram has to visit only those
routers mentioned in the route else it gets discarded.
The loose source route option is similar with the variation that
datagram has to visit the router listed but can also visit other routers
not listed.
The timestamp option records the time the datagram was processed by
the router.
12.2.4 SECURITY OF IPv4 DATAGRAM
When IPv4 protocol was introduced, security was not a concern.
But with heavy data transactions and business communi cation,
security is now a major concern.
Three major issues applicable to security related issues are
oPacket Sniffing –While packet is travelling from source to
destination an attacker may intercept the packet, read the content
and create copies of it. We cannot exactly prevent sniffing but
encryption makes the content decipherable and understandable for
the attacker.
oPacket Modification –Here the attacker modifies the content of the
packet and replays the packet. Using data integrity schemes, we
can prev ent the data from modification.
oIP Spoofing –Here the attacker forges as the sender and sends the
packet to receiver. Using data authentication schemes, we can
prevent this masquerade and forgery attack.
A new protocol called IPsec is introduced to handle the missing
security aspect in IP.
It creates a connection -oriented service between sender and receiver
handling the above three attacks.
IPSec provides four services namely defining keys and algorithms,
encryption of data, data integrity and authenticati on.
12.3 INTERNET CONTROL MESSAGE PROTOCOL ( ICMPv4)
The IPv4 Protocol does not handle errors or correct errors.
The router may discard the datagram because it could not find the
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240The IP protocol is inefficient to report and handle such problems.
The ICMP protocol is designed to handle these problems.
It is also a network layer protocol but the ICMP message is
encapsulated into IP datagram before sending it lower layer.
The protocol field of IP datagram is set to 1 to indicate it is an ICMP
message.
12.3.1 MESSAGES
ICMP messages are of two types: error -reporting message and query
message
The error -reporting message report the errors and problems the router
and destination face while processing the datagram.
The query message occurs in pair to place a request and obtain
response of information regarding the router.
The ICMP message has 8 -byte header and variable size data section.
The first 4 bytes of header are common in both types a nd next four
bytes are different in both.
Figure 12.9 –ICMP message structure
The type indicates the type of message and code indicates reason for
message type.
The checksum is calculated over header and data unlike the IPv4
where checksum is calculated for only header content.
The data section in error -reporting message carries the error occurred
while processing the datagram and in query message it contains
information depending on the query.
12.3.1.1 ERROR -REPORTING MESSAGES
The main role of ICMP is to report error during the processing of
datagram as IP is unreliable and incapable of doing so.
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241The higher layers need to take care of the error correction.
Errors are reported to source host using the source IP address available
in the header.
ICMP cannot directly float the error message in the network and so it
forms an error packet and encapsulates it in the IP datagram.
Figure 12.10 –Encapsulation of ICMP packet in IP datagram
The various error reporting message are
oDestination Unreachable –It is sent by a router when it cannot
deliver an IP datagram.
oSource Quench –It is sent by a destination host or router if it is
receiving data too quickly and not able to handle the datagram. The
message is a request that the source slow down datagram
transmission
oRedirection -It is sent by a router to optimize net work traffic by
redirecting the datagram to another router if it receives a datagram
that should have been sent to a different router.
oTime Exceeded –It is sent by a router if the datagram has reached
the maximum limit of routers through which it can tra vel.
oParameter Problem -It is sent by a router if a problem occurs
during the transmission of a datagram such that it cannot complete
processing due to invalid header.
12.3.1.2 QUERY MESSAGES
Query message are independent of IP datagram but again need to be
encapsulated in a datagram as a carrier.
Query message comes in pair and are used to test the availability and
actives ness of router in the network.
The various query messages are
oEcho Request & Echo Reply –It is used to test destination
accessibilit y and status. A host sends an Echo Request and listens
for a corresponding Echo Reply.
oTimestamp Request & Timestamp Reply –It is used to
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242The following query messages are now deprecated a nd replaced by
other messages.
oInformation Request &Information Reply -These messages
were used earlier by hosts to perform the mapping of IP and MAC
address. It is now taken care by the Address Resolution Protocol
(ARP).
oAddress Mask Request& Address Mask Reply –These
messages were used to find the mask of the subnet. A host sends
anAddress Mask Request to a router and receives an Address
Mask Reply in return. It is now taken care by the Dynamic Host
Configuration Protocol (DHCP).
oRouter Adve rtisement and Router Solicitation –These
messages were used to allow hosts to discover the existence of
routers. Routers periodically broadcast their IP addresses
viaRouter Advertisement messages. Hosts may also request a
router address by broadcasting a Router Solicitation message to
which a router replies with a Router Advertisement. It is now taken
care by the Dynamic Host Configuration Protocol (DHCP).
12.3.2 DEBUGGING TOOLS
ICMP makes use of two important tools foe debugging purpose –
ping and trace route
The ping is a computer network administration software utility that
is used to test the reachability of a host on an Internet Protocol
network.
It sends an ICMP echo request packet to the destination.
If the destination is alive, it prompts with the echo reply message.
The syntax is ping and the IP address of the destination.
Figure 12.11 –ping command output
The traceroute command in UNIX and tracert command in
Windows is used to determine the path taken to a destination by
sending ICMP echo request to the destination with incrementally
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243Each router decrements the TTL c ount as it forward the packet.
When TTL = 0, then router throws an ICMP time exceeded
message to the source host.
In case the path is not found then ICMP destination unreachable
message is also sent to the source.
The syntax is tracert and IP address of th e destination.
Figure 12.12 –tracert command output
12.4MOBILE IP
Mobile IP is extension of IP protocol in mobile that are connected to
the Internet.
Itis an Internet Engineering Task Force standard communications
protocol that is designed to allow mobile device users to move from
one network to another.
In the original network the host are stationary and belong to one
specific network.
So, assigning of I P address is simple defining the prefix and suffix and
communication takes place with this address as its valid and host
belongs to that network.
If the host changes the network, then address will become invalid.
As the name implies, the mobile host moves from network to network
and normal addressing scheme will not work as an address valid in one
network will not be valid in another network.
So different scheme is required for mobile host.
12.4.1 ADDRESSING
The mobile host changes its address as it moves from one network to
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244The host makes use of DHCP to obtain new address as it moves to a
new network.
But using DHCP leads to four problems:
oNeed to update the configuration file whenever new address in
obtained.
oThe system needs to be rebooted as it moves to new network.
oThe DNS tables needs to revised to make the Internet aware of the
network changes.
oDuring data transmission, if the host moves from one network to
another then data exchange is interrupted.
So viable solution is to use two addresses –home address and care -of
address.
The original address is called home address and is permanent and host
associates with home network.
The care -of address is temporary and when it associates host with
foreign network i.e., the other network the host moves in.
Figure 12.13 –Home address and Care -of address of Mobile Host
12.4.2 AGENTS
To manage the home address and care -of address, we require the
home agent and foreign agent.
The router attached to home network of the mobile host is the
home agent.
The router attached to foreign network of the mobile host is called
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Figure 12.14 –Home agent and foreign agen t
The mobile host and foreign agent can be same and in that case the
mobile host obtains the care -of address through DHCP.
This care -of address is called collocated care -of address.
The advantage of dual addressing is easy movement from one
network to anot her and disadvantage is extra software required for
processing.
12.4.3 THREE PHASES
The communication of mobile host with remote host goes through
three phases.
Figure 12.15 –Phases in remote communication
The first and second phase involves the mobile host and two agents
whereas the last phase involves even the remote host.
The entire communication is a nine -step process.
Step1 –A mobile host must know its home address and so it requests
the home agent by sending the agent solicitation message.
Step 2 -The home agent responses with the address through the agent
advertisement message.
Step 3 –The same process is to be repeated when the mobile host
moves to foreign network and sends an agent solicitation me ssage to
the foreign agent.
Step 4 –The foreign agent responses with the care -of address through
the agent advertisement message.
Step 5 –The mobile host needs to register to foreign agent via the
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246Step 6 –The mobile host al so needs to register with the home agent.
This is done by foreign agent on behalf of mobile host via the
registration request message.
Step 7 –The home agent replies to the foreign agent via the
registration response message.
Step 8 –This registration re sponse message is relayed back to the
mobile host by the foreign agent.
Step 9 –The mobile host can now transfer data and communicate with
the remote host.
Figure 12.16 –Mobile Host and Remote Host Communication
The format of the various messages exchanges in the communication
are discussed below.
The router solicitation message in ICMP is used in the place of agent
solicitation message.
The agent advertisement message is piggy backed with the router
advertisement packet.
Through this message the router advertises its presence on the
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247
Figure 12.17 –Agent Advertisement Message
The type field has vale 16.
The length field indicates the total length of the extension message.
The sequence number field is used to map in case the message is
lost.
Lifetime field indicates the value in seconds for the agent to accept
request. For infinite lifetime the value is all 1s.
The various value for code field is tabulated.
The last field is used only by the foreign agent where is a list of
care-of address is available and finalized in the registration phase.
The mobile host sends the registration request message to
foreign agent and home agent to register the home address, home
network and care -of address.
Figure 12.18 –Registration Request Message
The Type field has the value 1 and the flag field defines
forwarding information which is tabulated.
The lifetime field defines number of seconds the registration is
valid and all 0s indicate for deregistration and all 1s indicate
infinite lifetime.
Next set of fields indicate the home address, home agent address
and care -of address.
The identification is a unique value set in request message and
repeated in reply message for matching purpose.
The extensions are used for authentica tion purpose.munotes.in

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248Theregistration reply message is sent in response to the request
which the home agent sends to foreign agent and which is relayed
to the mobile host regarding confirming or denying the registration
request.
Figure 12.19 –Registration Response Message
The Type field has the value 3 and code field replaces the flag field
with either acceptance or denial of request. The remaining fields are
same as registration request message.
The registration request and reply message are encapsulated inside a
UDP datagram with agent using well known port no 434 and mobile
host using ephemeral port number.
The data transfer process occurs in four steps.
Figure 12.20 –Data transfer between remote host and mobile
host
Step 1 –When the remote host wants to send an IP packet, it creates a
packet with source address as its address and home address of mobile
host as destination address. The home agent interc epts this packet
suing proxy ARP technique.
Step 2 –The home agent creates a special tunnel and forwards the
packet to the foreign agent by encapsulating the original IP packet into
another packet with its address as source address and foreign agent
addres s as destination.
Step 3 –When the foreign agent receives the packet it checks the table
entry and replaces the home address with care -of address and forwards
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249Step 4 –The mobile host responses directly by creating an IP packe t
with its home address as source address and remote host as destination
address which the foreign agent handles.
The entire data transfer process is transparent.
12.4.4 INEFFICIENCY IN MOBILE IP
Every communication has flaws and Mobile IP can be inefficient
for two reasons.
The severe case of inefficiency is called the double crossing.
In situation where remote host and mobile host are associated with
the same router, so rather than internal communication taking
place, the communication takes place as a long path.
Figure 12.21 -Double Crossing
So rather than having local communication, the communication is
spread across the Internet and same message crosses twice.
The moderate case of inefficiency is the triangle routing.
The remote host could be in closer proximity with the mobile host and
rather than direct communication, the communication is two -fold a s
packet goes first to home agent and then to mobile host.
Figure 12.22 –Triangle Routing
Solution for inefficiency is the remote host needs to bind the care -of
and home address of the mobile host.
Doing so will reduce the traversal of packet depending on the location
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25012.5 SUMMARY
The Internet Protocol version 4 (IPv4) is an unreliable connectionless
protocol responsible for host -to-host delivery of datagrams
An IPv4 datagram can be fragmented by source or intermediate router
one or more times during its path from the source to the destination but
the reassembly of the fragments is done at the destination only.
The Internet Control Message Protocol version 4 (I CMPv4) supports
the unreliable and connectionless Internet Protocol (IP) by handling
errors during transmission.
The error reporting message and query message helps the IPv4 for
smooth transition.
The Mobile IP designed for mobile communication is an enhan ced
version of the Internet Protocol (IP).
12.6 LIST OF REFERENCES
“Data Communications and Networking” by Behrouz A. Forouzan, 5th
Edition, McGraw -Hill Publication.
https://www.javatpoint.com/network -layer -protocols
https://www.geeksforgeeks.org
12.7 U NIT END EXERCISE
1.A host is sending 100 datagrams to another host. If the identification
number of the first datagram is 1024, what is the identification number
of the last?
2.In an IPv4 datagram, the value of the header -length (HLEN) field is
(6)16. How many bytes of options have been added to the packet?
3.What are the source and destination IP addresses in a datagram that
carries the ICMPv4 message reported by a router?
4.An IP datagram has arrived with the following partial information in
the header (in hexadecimal): 45000054 00030000 2006...
a.What is the header size?
b.Are there any options in the packet?
c.What is the size of the data?
d.Is the packet fragmented?
e.How many more routers can the packet travel to?
f.What is the protocol number of the payload bei ng carried by
the packet?
5.Determine if a datagram with the following information is a first
fragment, a middle fragment, a last fragment, or the only fragment (no
fragmentation):
a. M bit is set to 1 and the value of the offset field is zero.
b. M bit is s et to 1 and the value of the offset field is nonzero.
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25113
UNICAST ROUTING
Unit Structure
13.0 Objectives
13.1 Introduction
13.2 Internet as a Graph
13.2.1 Least -cost Routing
13.3Routing Algorithms
13.3.1 Distance -Vector Routing
13.3.2 Link -State Routing
13.3.3 Path -Vector Routing
13.4 Unicast Routing Protocols
13.4.1 Internet Structure
13.4.2 Routing Information Protocol (RIP)
13.4.3 Open Shortest Path First (OSPF)
13.4.4 Border Gateway Protoco l Version 4 (BGP4)
13.5 Summary
13.6 List of References
13.7 Unit End Exercise
13.0 OBJECTIVES
After going through this chapter, you will be able to
Understand the general concept of unicast routing
Understand different unicast routing algorithms
Understand the unicast routing protocols
13.1INTRODUCTION
The network layer is responsible for host -to-host delivery of
packets.
The unicast routing governs the transmission of packet to only one
destination (one to one) whereas multicast routing gover ns the
transmission of packet to several destinations (one to many).
Unicast routing can be implemented using hierarchical routing
where we route in steps.munotes.in

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25213.2 INTERNET AS A GRAPH
The packet is routed hop by hop from source to destination.
The forwardi ng tables are used for the routing process.
Source host requires no forwarding table reference as it passes onto
default router.
Destination host needs no forwarding table as it receives the packet
and it has to reassemble the packets.
The intermediate router uses the forwarding table and needs to decide
the best route to reach the destination.
So, the internet can be modeled as a graph. i.e., weighted graph.
A graph consists of nodes and edges, so the source, destination and
router repr esent nodes and links represent edges.
Consider an internet with 7 routers which has been converted to
weighted graph.
Figure 13.1 -Internet modeled as a Graph
The weight represents the cost taken to travel from o ne node to
another.
If there is no edge then cost represent infinity.
The cost has different interpretation for different routing algorithm.
13.2.1 LEAST -COST ROUTING
Inorder to find the best route from source to destination we can create
least cost trees from a weighted graph.
A least cost tree chooses one node as a root node, and finds the best
path from that node to all other node.
The best path is chosen on the basis of least cost to reach from that
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Figure 13.2 -Least Cost Trees for Internet
The basic principle of least cost trees is that if there are N routers in
the Internet, then there are (N –1) least cost paths from one router to
all routers in the network.
On the whole there are N x (N –1) least cost paths for the entire
network.
A better way to visualize is to create a least cost tree.
We consider one source router and find the shortest pa th to all the
other routers in the network.
Two important properties to be remembered while constructing a least
cost tree are:
oA least cost tree from router X to router Y in X’s tree is inverse in
Y’s tree. For example, route from A to C in A’s tree is A BC=
7 and in C’s tree is C BA = 7.
oThe cost involved in travelling from router X to router Z in X’ s tree
is equal to cost travelling from router X to router Y in X’s tree plus
cost travelling from router Y to router Z in Y’s tree. For example,
the route from A to F in A’s tree is A BEF = 8 and route
from A to E in A’s tree = is A BE = 6 and E to F in E’s tree
EF = 2 i.e., 6 + 2 = 8.
13.3ROUTING ALGORITH MS
Routing algorithms are meant for determining the routing of packets in
a node.
Several routing algorithms have been devised.
The difference in the various algorithms is the interpretation of cost
and construction of least cost treesmunotes.in