M.Sc.

(CS)
SEMESTER - I (CBCS)

ADVANCED NETWORKING
CONCEPTS

SUBJECT CODE : PSCS102

 © UNIVERSITY OF MUMBAI
Prof. Suhas Pednekar
Vice-Chancellor,
University of Mumbai,

Prof. Ravindra D. Kulkarni Prof. Prakash Mahanwar

Pro Vice-Chancellor, Director,
University of Mumbai, IDOL, University of Mumbai,

Programme Co-ordinator : Shri Mandar Bhanushe

Head, Faculty of Science and Technology,
IDOL, University of Mumbai, Mumbai
Course Co-ordinator : Mr. Sumedh Shejole
Asst. Professor, B.Sc. I.T.
IDOL, University of Mumbai, Mumbai

Course Writers : Mr. Milind Thorat

Assistant Professor,
K.J. Somaiya Institute of Engineering &
Information Technology,
Sion, Mumbai

: Mr. Kashafurrehman Ansari

Assistant Professor,
Sanpada College of Commerce & Technology,
Sanpada, Navi Mumbai

: Ms. Diana Fernandes

Assistant Professor,
B. K. Birla College of Arts, Science and
Commerce (Autonomous)

October 2021, Print - I

Published by : Director
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 CONTENTS
Unit No. Title Page No.

Module - I
1. Networking - I 01
2. Networking - II 26
3. Networking - III 44
4. Networking - IV 62

Module - II
5. Network Virtualization 75

Module - III

6. ADHOC Networking 102

Module - IV

7. Wireless Sensor Networks 123


 Syllabus Advanced Networking Concepts
Semester I
Unit I: Networking
Internet and Intranet, Protocol layer and their services, Network
Applications like Web, HTTP, FTP and Electronic Mail in the Internet,
Domain Name System, Transport-Layer Services, Multiplexing and
Demultiplexing, UDP, TCP, TCP Congestion Control, Network Layer,
Virtual Circuit and Datagram Networks, Need of Router, The Internet
Protocol (IP), Routing Algorithms, Routing in the Internet.
Unit II: Network Virtualization
Need for Virtualization, The Virtual Enterprise, Transport Virtualization-
VNs, Central Services Access: Virtual Network Perimeter, A
Virtualization Technologies primer: theory, Network Device
Virtualization, Data-Path Virtualization, Control-Plane Virtualization,
Routing Protocols.
Unit III: Adhoc Networking
Introduction, application of MANET, challenges, Routing in Ad hoc
networks, topology & position based approaches, Routing protocols:
topology based, position based, Broadcasting, Multicasting, & Geocasting,
Wireless LAN, Transmission techniques, MAC protocol issues, Wireless
PANs, The Bluetooth technology.
Unit IV: Wireless Sensor networks:
Need and application of sensor networks, sensor networks design
considerations, empirical energy consumption, sensing and
communication range, design issues, localization scheme, clustering of
SNs, Routing layer, Sensor networks in controlled environment and
actuators, regularly placed sensors, network issues, RFID as passive
sensors.
Text book:
 Computer Networking: A Top-Down Approach 6th edition, James F.
Kurose, Keith W. Ross, Pearson (2012).
 Network Virtualization, Victor Moreno, Kumar Reddy, Cisco Press
(2006).
 Ad Hoc and Sensor Networks: Theory and Applications 2nd edition;
Carlos de Morais Cordeiro, Dharma Prakash Agrawal, World
Scientific Publishing Company; 2 edition (2011)
Reference book:
 TCP/IP Protocol Suite 4 edition, Behrouz Forouzan, McGraw-Hill
Science ( 2009)
 Mobile Ad Hoc Networks: Current Status and Future Trends, Jonathan
Loo, Jaime Lloret Mauri, Jesús Hamilton Ortiz, CRC Press(2011)
 Fundamentals of Sensor Network Programming: Applications and
Technology, S. Sitharama Iyengar, Nandan Parameshwaran, Vir V.
Phoha, N. Balakrishnan, Chuka D. Okoye, Wiley-IEEE Press (2010).


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 Module I

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NETWORKING - I

Unit Structure

1.1 Objective
1.2 Introduction
1.3 Internet and Intranet
1.4 Protocol layer and their services,
1.5 Network Applications like web, HTTP, FTP and Electronic Mail in the
Internet
1.5.1 Overview of HTTP
1.5.2 Non-Persistent and Persistent Connections
1.5.3 HTTP Message Format
1.5.4 HTTP Response Message
1.5.5 User-Server Interaction: Cookies
1.6 FTP
1.7 Electronic mail in the internet
1.7.1 SMTP
1.7.2 Comparison with HTTP
1.7.3 Mail message formats
1.7.4 Mail access protocols
1.8 Domain Name System,
1.8.1 Services provided by DNS
1.8.2 Overview of how DNS works
1.8.3 DNS records and messages
1.9 Summary
2.0 Reference for further reading
2.1 Unit End Exercises

1.1 OBJECTIVE

1. To help to get a grounding of basic network components and

architecture.
2. To explore basic networking models.
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3. To learn the way protocols are used in networks.
4. Understanding of the fundamental concepts of computer networking.
5. To understand the basic taxonomy and terminology of the computer
networking area.
6. To understand the Internet and Intranet, Protocol layer and their
services, Network Applications like web, HTTP, FTP and Electronic
Mail in the Internet, Domain Name System, and Transport-Layer
Services.

1.2 INTRODUCTION

Computer Network is potentially the largest skillfully arranged

system ever created by people, with hundreds of millions of connected
systems, communication links, and switches with billions of users who
connect via laptops, tablets, and smart phones and with an array of new
Internet-connected devices such as sensors, webcams, game consoles,
picture frames, and even washing machines etc. The systems are
connected to each other’s by a network communication links and packet
switches; those are many types of communication links, which are made
up of different types of physical media, like coaxial cable, copper wire,
optical fiber, and radio spectrum. Various links can transmit data at
different rates, with the transmission rate of a link measured in bits per
second. When one system has data to send to another system, the sending
system segments the data and adds header of bytes to each segment.

1.3 INTERNET AND INTRANET

The Internet is a worldwide system of interconnected computer

networks. It uses the standard Internet Protocol (TCP/IP), set of rules.
Each and Every computer on the Internet is identified by a unique IP
address. IP Address is a unique set of numbers (such as 110.22.33.114)
which identifies a computer’s physical location. The Internet is accessible
to every user all over the world.

A special computer Domain Name Server is used to provide a

name to the IP Address so that the user can locate & identify a computer
by a name. For example, a DNS server will resolve a name
https://www.mu.ac.in to a particular IP address to uniquely identify the
computer on which this website is hosted.

Intranet is the system in which multiple computers are connected

to each other’s through physical media. Systems on the intranet are not
available to the world outside the intranet. Normally each organization
has its own Intranet network and members of that organization can access
the computers in their intranet. Each computer in Intranet is also identified
by an IP Address which is unique in between the computers in that
Intranet.

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 Fig.1 Internet & Intranet

Similarities between Internet and Intranet

● Intranet uses the internet protocols (IP) such as TCP/IP and FTP.
● Intranet sites are accessible via the web browser in a similar way as
websites on the internet. However, only members of Intranet network
can permit access to intranet hosted sites.
● In Intranet, own instant messengers can be used over the internet.

Differences between Internet and Intranet

● The Internet is general to Computers all over the world whereas
Intranet is distinct to few computers.
● The Internet provides a wider and better access to websites to a large
population, whereas Intranet is restricted.
● The Internet is not as safe as the Internet. Intranet can be safely
privatized as per the need.

1.4 PROTOCOL LAYER AND THEIR SERVICES

A protocol is required when two entities need to communicate with

each other’s. When communication is not easy or simple, it may divide the
complex task of communication into different layers. On this occasion, we
may need various protocols, one for each layer. Let us use a scenario in
communication in which the role of protocol layering may be better
accepted.

A layered architecture allows us to talk over a well explained,

different part of a large and complex system. This simplification itself is
of extraordinary value by providing modularity, making it much easier to
change the implementation of the service provided by the layer. As long as
the layer provides the same service to the layer above it, and uses the
identical services from the layer below it, the balance of the system
perseveres unchanged when a layer’s implementation is changed.
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 A protocol layer such as HTTP and SMTP are almost implemented
in software in the end systems, so are transport-layer protocols. The
physical layer and data link layers are responsible for administering
communication over a distinct link; they are typically implemented in a
network interface card associated with a given link. The network layer is
generally an associated implementation of hardware and software parts.

Application Layer
The application layer is a network application and their
application-layer protocols reside on this layer. The Internet’s application
layer includes different protocols like, the HTTP protocol which provides
for web document request and transfer, SMTP which administers for the
transfer of e-mail messages, and FTP which provides for the transfer of
files between two systems.

Transport Layer
The Internet’s transport layer carries application-layer messages
between application endpoints. In

The two transport protocols, TCP and UDP, any one of them can
transport application layer messages to each other’s. TCP provides a
connection-oriented service to its applications. This includes guaranteed
delivery of message or packets to the destination and flow control.

Network Layer
Network layer is responsible for moving network-layer packets
known as datagrams from one host to another host. The Internet transport-
layer protocol such as TCP & UDP in a source host passes a transport-
layer segment and a destination address to the network layer, like the
postal service a letter with a destination address. The network layer
provides the service of delivering the chunk of message to the transport
layer in the destination host.

Link Layer
The network layer directs a datagram through a list of routers
between the source and destination. To move a packet from one router to
the next route in the route, the network layer confides on the services of
the link layer. At every node the network layer sends the datagram packet
down to the link layer, which delivers the datagram packet to the next
node along the route information. At this point next node, the link layer
passes the datagram up to the network layer.

Physical Layer
In Physical Layer, a period of time the function of the link layer is
to move entire frames from one network element to an adjacent network
element; the job of the physical layer is to shift the individual bits within
the frame from one node to the next node. The protocol link is dependent
and furthers relies on the actual transmission medium of the protocol link,
for example, twisted-pair copper wire, single-mode fiber optics. For e.g.

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Ethernet has numerous physical-layer protocols, one for twisted-pair
copper wire, another for coaxial cable, another for fiber, and so on.

1.5 NETWORK APPLICATIONS LIKE WEB, HTTP, FTP

Web
The web was the first Internet application that was used
everywhere in today's world. It drastically & continuously changes how
people collaborate inside and outside their work environment. It raised the
Internet from just one of many data networks to essentially the one and
only one data network. Maybe what appeals the most to users is that the
web operates on requirement. Web users receive what they need, when
they want it at any time. This is far from traditional broadcast radio and
television systems, which force users to tune in when the content provider
makes the content available to all users in the world. Web available on
user demand. It is extremely simple for any individual to make
information available over the web for every user can be a publishes at too
little cost. Hyperlinks and search engines help us guide through
information. Forms, JavaScript, Java applets, and many others devices
empower us to relate with different web pages and websites. The web and
its protocols assist as a platform for YouTube, web-based email, and most
mobile Internet applications, including Instagram and Google Maps.

1.5.1 Overview of HTTP

.
The Hypertext Transfer Protocol (HTTP) is the web’s application-
layer protocol, transmitting hypermedia documents and is at the heart of
the web. HTTP is implemented in two programs:
a. a client program and
b. A server program.

The client and server program, executing on different machines,

talk to each other’s by exchanging HTTP messages. HTTP defines the
structure of these messages and how the client and server exchange the
messages & information.

A web page consists of objects. An object is a file like HTML file,

a JPEG image, a Java applet, or a video clip that is addressable by a single
URL. Many web pages consist of a base HTML file and several
referenced objects. For example, the web page consists of a HTML text
file and around five JPEG image files, and then the web page has a total
six objects, the base HTML file and the five image files. The HTML file
relates to the others objects in the page with the objects URLs. Every URL
has two parts: the host name of the server and the object’s path name.

For e.g. the URL has www.SmallSchool.edu for a hostname and

/Department/picture.gif for a path name. Because web browsers, such as
Internet Explorer and Firefox implement the client side of HTTP, in the
context of the web. The words used for browser and client conversely as a

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web server, which implement the server side of HTTP, web objects, each
addressable by a URL. Well liked web servers include Apache and
Microsoft Internet Information Server.

HTTP states how web clients request web pages from web servers
and how servers transfer web pages to clients. When a user requests a
particular web page, the browser sends HTTP request messages and
requests objects which are available on the server side. The server receives
the requests via HTTP and responds with HTTP response messages that
include the objects. HTTP uses TCP/IP as its underlying transport
protocol. The HTTP client first begins a TCP connection with the server.
After the connection is formed, the browser and the server processes
access TCP through their socket interfaces.

Fig.2 HTTP request & response

This inferred that each HTTP request message sent by a client

process in time arrives intact at the server; similarly, each HTTP response
message sent by the server process in time arrives intact at the client. The
loss of data in HTTP need not doubt or TCP recovers from loss or
reordering of data within the network system. That is the task of TCP and
the protocols in the bottom layers of the protocol stack. It is compulsory to
note that the server sends requested files to clients without storing any
state information about the client. If a specific client asks for the same
object twice in a time of a few seconds, the server does not respond by
saying that it just provides the object to the client; instead, the server
resends the object, as it has completely unknown what it did earlier.
Because an HTTP server maintains no information about the clients,
HTTP is said to be a stateless protocol in the network.

1.5.2 Persistent & Non-Persistent Connections

In most Internet applications, the client and server communicate

for an extended period of time, with the client making a series of requests
and the server responding to each of the requests on time one by one.
based on & use of the application, the sequence of requests may be made
one after the others back-to-back, from period to period, at regular
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intervals of time, or asymmetrical to gain a deep understanding of this
design issue, let’s examine the advantages and disadvantages of persistent
connections in the context of a specific application, namely, both non-
persistent connections and persistent connections use in HTTP. Even
though in its default mode HTTP uses persistent connections, HTTP
clients and servers can be well organized to use non-persistent connections
instead.

1.5.3. HTTP with Non-Persistent Connections

Transferring of a web page from server machine to client machine

in the case of no persistent connection, the web page contains a base
HTML file and 10 JPEG images, and that all 11 of these objects are on the
same server machine. For example the URL for the base HTML file is
https://www.SmallSchool.edu/Department/home.index

a. The HTTP client process begins a TCP connection to the server

www.SmallSchool.edu on port number 80, this is the default port
number 80 used for HTTP. This TCP connection, those will be a
socket both at the client and at the server.
b. Through socket, HTTP client sends an HTTP request message to the
server. The request message includes the path name
/Department/home. index which is shown above example.
c. The HTTP server processes the request message via its socket,
retrieves the object /Department/home. index from its storage device
i.e. from RAM or disk, enclose the object in an HTTP response
message, and sends the response message to the client via socket.
d. The HTTP server process acknowledges TCP to close the TCP
connection.
e. The HTTP client receives the response message from HTTP server.
Then the TCP connection is terminated. The response message
indicates that the encapsulated object is an HTML file. The client
extracts this file from the response message, reads the HTML file, and
finds sources to the 10 JPEG objects.
f. The first four steps i.e. a, b, c & d are repeated for each of the
referenced JPEG objects. As the browser receives the web page at the
client machine, it displays the page to the user on the client. Two
different browsers may interpret a web page in somewhat different
techniques. HTTP has neglected how a web page is interpreted by a
client.

The steps above explain the use of non-persistent connections,

whose each TCP connection is closed,

After the server sends the object, the connection does not persist
for others objects. Each TCP connection transports exactly one request

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message and one response message. Thus, in this example, when a user
requests the web page, 11 TCP connections are generated.

The Round Trip Time (RTT) includes packet-propagation delays,

packet stream delays in between routers and switches, and packet-
processing delays. In Figure 3, shows that to initiate a TCP connection
between the client browser and the web server, this process requires a
“three-way handshake” method. The client sends a small TCP segment to
the server, the server acknowledges that segment and responds with a
small TCP segment back, and finally the client sends acknowledgement
back to the server. The first two sections of the three-way handshake take
one RTT. After completing the first two parts of the handshake method,
the client sends the HTTP request message combined with the third part of
the three-way handshake into the TCP connection on the internet. Once
the request message reaches the server, the server sends the HTML file
into the TCP connection to the client browser. This HTTP
request/response eats up another RTT. Thus, approximate, the total
response time is two RTTs plus the transmission time at the server of the
HTML file.

Fig. 3 Back-of-the-envelope calculation for the time needed to request and

receive an HTML file

HTTP with Persistent Connections

After sending a response a server left TCP connection left open.

Forthcoming requests and responses between the same client and server
can be sent over the same connection. In some others cases, an entire web
page can be sent over a single persistent TCP connection in HTTP. On the
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others hand, multiple web pages residing on the same server can be sent
from the server machine to the same client machine over a single
persistent TCP connection. These object requests can be made one after
the others, without waiting for replies to pending requests. The HTTP
server closes a connection when it isn’t used for a few times. When the
server receives the one after the others requests, it sends the objects back-
to-back. The default mode of HTTP uses persistent connections with
pipelining.

1.5.3 HTTP Message Format

The HTTP specifications include the definitions of the HTTP

message formats. There are two types of HTTP messages, request
messages and response messages.

HTTP request message:

GET /somedir/page.html HTTP/1.1

Host: www.SmallSchool.edu
Connection: close
User-agent: Mozilla/5.0
Accept-language: french

This message is written in ASCII text, the human can easily read
this. The second division of the message has five lines, each followed by a
carriage return and a line feed. The end line of the message is followed by
an additional carriage return and line feed. Even though this particular
request message has five lines, a request message can have many more
lines. The first line of an HTTP request message is called the request line
and the future lines are called the header lines.

The three fields of request line:

1. Method field,
2. URL field,
3. HTTP version field.

The method field has various different values, including GET,

POST, HEAD, PUT, and DELETE. The great most part of HTTP request
messages have the GET method. The GET method is used when the
browser requests an object from the server application, with the requested
object is identified in the URL field.

The header line Host: www.SmallSchool.edu specifies the host

on which the object resides. The host header line provides the information
which is required by web proxy caches, including the connection: close
header line, the browser instructs the server that it doesn’t want to disturb
persistent connections, and then the server closes the connection after
sending the requested object to the client.

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 The User agent specifies the header line, that the browser type
that is building the request to the server, at this moment the user agent is
mozilla, a firefox browser. This header line is helpful because the server
sends different varieties of the same object to different types of user
agents.

The Accept-language header indicates that the user selects to

receive a French version of the object, if such an object exists on the
server; or else, the server should send its default version. The
Accept-language header is just numerous content negotiation headers
available in HTTP.

The general format of the request message, as shown in Figure 4.

Beside the header lines those is an “entity body.” The entity body is empty
with the GET method, but is used with the POST method. HTTP clients
generally use the POST method when the user fills out a form for e.g.,
when a user provides search words to a search engine. Using the POST
message, the user is still requesting a web page from the server, but the
specific contents of the web page depend on what the user entered into the
form fields. The value of the POST method field is POST, and then the
entity body contains what the user entered into the form fields.

Fig. 4 General format of an HTTP request message

1.5.4 HTTP Response Message

HTTP response message. This response message could be the response to

the example request message explained in HTTP Request Message.
HTTP/1.1 200 OK
Connection: close
Date: Tue, 30 Aug 2020 15:44:04 GMT
Server: Apache/2.2.3 (CentOS)
Last-Modified: Tue, 30 Aug 2020 15:11:03 GMT
Content-Length: 6821
Content-Type: text/html
(data data data data data ...)

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 The response Message has three sections: an initial status line, six
header lines, and then the entity body. The entity body is the meat of the
message; it contains the requested object itself.

The status line has three fields: the protocol version field, a status
code, and a corresponding status message. In this example, the status line
indicates that the server is using HTTP/1.1.

The header lines.

The server uses this connection for a close header line to inform the
client that it is going to close the TCP connection after sending the
message.
a. The Date header line shows the time and date when the HTTP
response was created and sent by the server.
b. The Server header line shows that the message was generated by
an apache web server.
c. User-agent the header line shows the HTTP request message.
d. Last- Modified the header line shows the time and date when the
object was created or last modified.
e. Content-Length The header line shows the number of bytes of data
in the object being sent.
f. Content-Type the header line shows that the object in the entity
body is HTML text.

The general format of a response message:

The general format of HTTP response message which is shown in

Figure 5 This general format of the response message is the same as the
previous example of a response message format. The status code and
associated phrase indicate the result of the request message. Some
common status codes and associated phrases include:
● 200 OK status code shows that request succeeded and the
information is returned in the response.
● 301 Moved Permanently status code requested object has been
permanently moved & the new URL is specified in Location:
header of the response message format. The client software will
automatically fetch the new URL.
● 400 Bad Request is a generic error code status indicating that the
request could not be known by the server.
● 404 Not found status code shows that the requested document does
not exist on this server.
● 505 HTTP Version Not Supported by this message format: The
requested HTTP protocol version is not supported by the server.

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 Fig. 5 General format of an HTTP response message

1.5.5 User-Server Interaction: Cookies

HTTP cookie is a small block of data stored on the user's computer

or machine by the web browser while browsing a website. This clarifies
server design and has permitted engineers to develop high-performance
web servers that can handle thousands of concurrent TCP connections
simultaneously. However, it is often advantageous for a web site to
identify users, ethics because the server wishes to restrain user access or
because it wants to accept content as a function of the user identity. For
this reason cookies allow sites to keep track of all users. Most of the
commercial web sites use cookies today.

Cookie technology has four components: (As shown in Figure 6)

(1) A cookie header line in the HTTP response message;
(2) A cookie header line in the HTTP request message;
(3) A cookie file kept on the user’s end system and managed by the
user’s browser; and
(4) A back-end database at the web site.

As per the figure 6, an example of how cookies work. Assume

amit, who always accesses the web using Internet Explorer from his home
machine, visits Amazon.com sites for the first time. Let us suppose that in
the past he has already visited the eBay website. When the request comes
into the amazon web server, the server creates a unique identification
number (UIN) and creates an entry in its backend database that is indexed
by the identification number. The amazon web server then responds to
amit’s web browser, also including in the HTTP response message a Set-
cookie header, which consist of the identification number. For example,
the header line is:

Set-cookie: 1678

When Amit’s browser receives the HTTP response message, it sees

the Set-cookie is header. The browser then includes a line to the special
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cookie file that it manages. This line adds the host name of the server and
the identification number in the Set-cookie header. Remember that the
cookie file already has an entry for eBay, since Amit has visited that site
in the past days. As Amit continues to browse the Amazon site, each time
he requests a web page, his browser consults his cookie file, extracts his
identification number for this site and places an identification number in
the cookie header line in the HTTP request. Specifically, each of user
HTTP requests to the Amazon server includes the header line:

Fig. 6 keeping user state with cookies

Cookie: 1678

The amazon server is able to track Amit’s activity at the Amazon

site. Although the Amazon web site does not necessarily know Amit’s
name and its details, it knows exactly which pages user 1678 visited, in
which order, and at what times. amazon uses cookies to provide its
shopping cart service. Amazon can maintain a list of all of Amit’s
intended purchases, so that he can pay for them collectively at the end of
the session. If amit returns to amazon’s site, one week after, his browser
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will continue to put the header line Cookie: 1678 in the request messages.
Amazon also recommends products to amit based on web pages he has
visited at amazon in the past. If Amit also registers himself with amazon
providing full name, e-mail address, postal address, and credit card
information amazon can then include this information in its database,
thereby associating Amit’s name with his identification number. This is
how Amazon and others e-commerce sites provide “one-click shopping”
when Amit chooses to purchase an item during a subsequent visit, he
doesn’t need to re-enter his name, credit card number, or address.

1.6 FTP

In a typical FTP session the user is sitting in front of one host i.e.
the local host and remote host. If the user to access the remote account, the
user must provide a user identification and a password during file transfer.
After providing authorization information, the user can transfer files from
the local file system to the remote file system and vice versa. As shown in
Figure 7, the user interacts with FTP through an FTP user agent. The user
must provide the hostname of the remote host, causing the FTP client
process in the local host to setup a TCP connection with the FTP server
process in the remote host. The user then provides the user recognition and
password, which are sent over the TCP connection as part of FTP. Once
the server has authenticated the user, the user copies one or more files
stored in the local file system into the remote file system or vice versa.

Fig. 7 FTP moves files between local and remote file systems

Fig. 8 Control and data connections

HTTP and FTP are both file transfer protocols and have many
common characteristics:
1. They both run on top of TCP.
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 2. The most striking difference is that FTP uses two parallel TCP
connections to transfer a file, a control connection and a data
connection.

The FTP control and data connections are illustrated in Figure 8.

FTP Commands and Replies

The commands, from client to server, and replies, from server to

client, are sent across the control connection in 7-bit ASCII format. Thus,
like HTTP commands, FTP commands are readable by people. In order to
delineate successive commands, a carriage return and line feed end each
command. Each command consists of four uppercase ASCII characters,
some with optional arguments. Some of the more common commands are
given below:

● USER username: Used to send the user identification to the server.

● PASS password: Used to send the user password to the server.
● LIST: Used to ask the server to send back a list of all the files in
the current remote directory. The list of files is sent over a (new
and non-persistent) data connection rather than the control TCP
connection.
● RETR filename: Used to retrieve a file from the current directory
of the remote host. This command causes the remote host to
initiate a data connection and to send the requested file over the
data connection.
● STOR filename: Used to store a file into the current directory of
the remote host.

Each command is followed by a reply, sent from server to client.

The replies are three-digit numbers, with an optional message following
the number. This is similar in structure to the status code and phrase in the
status line of the HTTP response message. Some typical replies, along
with their possible messages, are as follows:

● 331 Username OK, password required

● 125 Data connection already open; transfer starting
● 425 Can’t open data connection
● 452 Error writing file

1.7 ELECTRONIC MAIL IN THE INTERNET

Electronic mail has been around since the birth of the internet. It
was the most popular application, and has become more improved and
powerful over the years. It is the most important and utilized application.
As it is the same like postal mail service, e-mail is an asynchronous
communication medium people send and read messages when it is

15
convenient for them, without having to interrelate with others people’s
schedules. Electronic mail is fast, easy to distribute, and cheap. The
current email application has many powerful features, including messages
with attachments, hyperlinks, html-formatted text, embedded photos and
much more. His application-layer protocols are the heart of internet e-
mail.

A high-level view of the internet mail system and its key components:

Figure 1 shows a high-level view of the internet mail system. It has

three major components: user agents, mail servers, and the simple mail
transfer protocol (SMTP).

Example: amity, sending an e-mail message to a recipient, sham. User

agents allow users to read, reply to, forward the message, save the
message, and compose messages. Microsoft Outlook and Google Mail are
examples of user agents for email applications. When it is finished
composing a message, the user agent sends the message to amit mail
server, whose message is placed in the mail server’s queue for outgoing
messages. When shyam wants to read a message, his user agent recovers
the message from his mailbox in his mail server. Each recipient, like
shyam, has a mailbox located in one of the mail servers. Shyam's mailbox
manages and maintains the messages that have been sent to him. A typical
message starts its migration in the sender’s user agent, travels to the
sender’s mail server, and travels to the recipient’s mail server, whose it is
at stake in the recipient’s mailbox. When shyam wants to access the
messages in his mailbox, the mail server containing his mailbox
authenticates shyam. amit’s mail server must also deal with failures in
shyam’s mail server. if amit’s server cannot deliver mail to shyam’s
server, amit’s server holds the message in a message queue and attempts
to transfer the message after some time. Reattempts for sending messages
are often done every 30 minutes, if there is no success after several days,
the server removes the message and notifies the sender i.e. to amit with an
e-mail message SMTP is the principal application-layer protocol for
internet electronic mail. It uses the reliable data transfer service of TCP to
transfer mail from the sender’s mail server to the recipient’s mail server.
In most application-layer protocols, SMTP has two sides, a client side,
which executes on the sender’s mail server and a server side which
executes on the recipient’s mail server. Both the client and server sides of
SMTP run on every mail server. When a mail server sends mail to other
mail servers, it acts as an SMTP client. When a mail server receives mail
from others mail servers, it acts as an SMTP server.

16
 Fig. 9 a High-Level view of the internet E-Mail System

1.7.1 SMTP

SMTP is the heart of internet electronic mail. SMTP function is to

move messages from senders’ mail servers to the recipients’ mail servers.
The origin of SMTP is in 1982, SMTP is much older than HTTP. SMTP
comes before HTTP. SMTP restricts the body of all mail messages to
simple 7-bit ASCII code. When transmission capacity was scarce and no
one was emailing large attachments or large image, audio, or video files.
But nowadays, in the multimedia era, before being sent over SMTP the 7-
bit ASCII restriction is a bit. It requires binary multimedia data to be
encoded to ASCII format and it requires the corresponding ASCII
message to be decoded back to binary after SMTP transport. HTTP does
not require multimedia data to be ASCII encoded before transmitting.

To illustrate the basic operation of an SMTP server, let’s walk

through a common example: suppose Amit wants to send shyam a simple
ASCII message.
1. Amit invokes his user agent for e-mail, provides shyam’s e-mail
address (for example, shyam@someschool.edu), composes a message,
and instructs the user agent to send the message.
2. amit’s user agent sends the message to his mail server, whose it is
placed in a message queue.
3. The client side of SMTP, running on amit’s mail server, finds the
message in the message queue. it opens a TCP connection to an SMTP
server, running on shyam’s mail server.

17
4. After some initial SMTP handshaking, the SMTP client sends Amit's
message into the TCP connection.
5. At shyam’s mail server, the server side of SMTP receives the message.
shyam’s mail server then places the message in shyam’s mailbox.
6. Shyam calls on his user agent to read the message at his
convenience.

fig. 10Amit sends a message to Shyam

1.7.2 Comparison with HTTP (SMTP with HTTP)

1. HTTP & SMTP protocols are used to transfer files from one host to
another host
2. HTTP transfers files from a web server to a web client.
3. SMTP transfers e-mail messages from one mail server to another mail
server.
4. In SMTP & HTTP, when transferring the files, both persistent HTTP
and SMTP use persistent connections. Both two protocols have
common characteristics.
5. HTTP is mainly a pull protocol whose someone loads information on
a web server and users use HTTP to pull the information from the
server at their benefit.
6. SMTP is primarily a push protocol; the sending mail server pushes the
file to the receiving mail server. On the other hand the TCP connection
is initiated by the machine that wants to send the file.
7. SMTP requires each message, including the body of each message, to
be in 7-bit ASCII format. If the message contains characters that are
not 7-bit ASCII then the message has to be encoded into 7-bit ASCII.
HTTP data does not impose this constraint.
8. Internet mail places all of this message’s objects into one message.

1.7.3 Mail message formats

When Amit writes a simple mail or letter to shyam, he may include

all kinds of peripheral header information at the top of this letter, like
shyam’s address, his own return address, and the date. Similarly, when an
18
email message is sent from one person to another, a header containing
peripheral information precedes the body of the message itself. This
information is contained in a series of header lines, which are defined in
RFC 5322. the header lines and the body of the message are divided by a
blank line. RFC 5322 specifies the exact format for mail header lines as
well as their semantic explanation. As with HTTP, each header line
contains human readable text, consisting of a keyword followed by a:
followed by a value. Some of the keywords are required and others are
optional.

Each and every header must have a from: header line and a to:
header line. a header may include a Subject: header line as well as others
optional header lines. it is important to note that these header lines are
different from the SMTP commands, the commands were part of the
SMTP handshaking protocol. The header lines checked in this section are
part of the mail message itself. a typical message header shown below:

from: amit@crepes.fr
to: shyam@hamburger.edu
subject: searching for the meaning of life.

After the message header, a blank line follows, and then the
message body follows. It must use telnet to send a message to a mail
server that contains some header lines, including the subject: header line.

1.7.4 Mail access protocols

Once SMTP server delivers the message from Amit's mail server
to Shyam's mail server, the message is placed in Shyam's mailbox. Then
shyam reads his mail by logging onto the server host and then executing a
mail reader that runs on that host. In the early 1990s this was the standard
way of doing things. But today, mail access uses client-server architecture.
The typical user reads email with a client that executes on the user’s end
system, for example, on an office pc, a laptop, or a smartphone. By
executing a mail client on a local pc, users enjoy a rich set of features,
including the ability to view multimedia messages and attachments

1.8 DOMAIN NAME SYSTEM

Human beings can be identified in many ways. For example, we

can be identified by the names that appear on our birth certificates or ID
card and much more. Humans can be identified by our driver’s license
numbers. Even though each of these identifiers can be used to identify
people, within a given context one identifier may be more appropriate than
another. For e.g., the computers at the IRS prefer to use fixed-length social
security numbers rather than birth certificate names. On the other side,
ordinary people prefer the more mnemonic birth certificate names rather
than social security numbers.

19
 Internet hosts can be identified in many ways like humans. One
identifier for a host is its hostname. Hostnames such as ibn.com,
www.yahoo.com, somaiya.edu, and mu.ac.in are mnemonic and are
therefore appreciated by humans. However, hostnames provide little, if
any, information about the location within the internet of the host. (A
hostname such as www.mu.ac.in, which ends with the country code .in,
tells us that the host is probably in India, but doesn’t say much more.)
Furthermore, because hostnames can consist of variable length
alphanumeric characters, they would be difficult to process by routers. For
these reasons, hosts are also identified by IP addresses.

An IP address consists of four bytes and has a rigid hierarchical

structure. An IP address looks like 192.168.3.1, which’s each period
separates one of the bytes expressed in decimal notation from 0 to 255. An
IP address is hierarchical because this is scanning the address from left to
right, obtaining more and more specific information about who's the host
is located on the internet. Similarly, when we scan a postal address from
bottom to top, this obtains more and more specific information about
who’s the addressee is located.

1.8.1 Services provided by DNS

There are two ways to identify a host by a hostname and by an IP

address. People prefer the more mnemonic hostname identifier, while
routers prefer fixed-length, hierarchically structured IP addresses. In order
to reconcile these preferences, we need a directory service that translates
hostnames to IP addresses. This is the main task of the internet’s domain
name system (DNS). The DNS is
1. A distributed database implemented in a hierarchy of DNS servers,
and
2. An application-layer protocol that allows hosts to query the
distributed database.

The DNS protocol runs over UDP and uses PORT 53.

DNS is commonly employed by these application layer protocols

including HTTP, SMTP, and FTP to translate user supplied hostnames to
IP addresses. As an example, consider what happens when a browser (that
is, an HTTP client), running on some user’s host, requests the url
www.somaiya.edu/index.html. in order for the user’s host to be able to
send an HTTP request message to the web server www.somaiya.edu, the
user’s host must first obtain the IP address of www.somaiya.edu.

This is done as follows.

1. The same user machine runs the client side of the DNS application.
2. The browser extracts the hostname, www.somaiya.edu, from the URL
and passes the hostname to the client side of the DNS application.

20
3. The DNS client sends a query containing the hostname to a DNS
server.
4. The DNS client eventually receives a reply, which includes the IP
address for the hostname.
5. Once the browser receives the IP address from DNS, it can initiate a
TCP connection to the HTTP server process located at PORT 80 at
that IP address.

DNS provides a few other important services in addition to translating

hostnames to IP addresses:

a. Host aliasing. a host with a cumbersome hostname can have one or

more alias names. For example, a hostname like relay1.west-
coast.enterprise.com could have, say, two aliases such as
enterprise.com and www.enterprise.com. in this case, the hostname
relay 1.west coast. Enterprise.com is said to be an authorized
hostname. Alias hostnames, when available, are often more mnemonic
than authorized hostnames. DNS can be requested by an application to
obtain the authorized hostname for a supplied alias hostname as well
as the IP address of the host.

b. Mail server aliasing. It is highly advisable that e-mail addresses be

mnemonic. For example, if shyam has an account with hotmail,
shyam’s e-mail address might be as simple as shyam@hotmail.com.
However, the hostname of the hotmail mail server is more complicated
and much less mnemonic than simply hotmail.com, for example, the
authorized hostname might be something like relay1.west-
coast.hotmail.com. DNS can be invoked by a mail application to
obtain the canonical hostname for a supplied alias hostname as well as
the IP address of the host.

c. Load distribution. DNS is also used to perform load distribution

among replicated servers, such as replicated web servers. Busy sites,
such as ibn.com, are replicated over multiple servers, with each server
running on a different end system and each having a different IP
address. For replicated web servers, a set of IP addresses is thus
connected with one canonical hostname. The DNS database contains
this set of IP addresses. When clients make a DNS query for a name
mapped to a set of addresses, the server responds with the entire set of
IP addresses, but rotates the ordering of the addresses within each
reply. Because a client typically sends its HTTP request message to the
IP address that is listed first in the set, DNS rotation distributes the
traffic among the different copies of servers. DNS rotation is also used
for email so that multiple mail servers can have the same alias name.

1.8.2 Overview of how DNS works

It presents a high-level overview of how DNS works. Suppose that

some application running in a user’s host needs to translate a hostname to
21
an IP address. The application will invoke the client side of DNS,
specifying the hostname that needs to be translated. DNS in the user’s host
then takes over, sending a query message into the network. All DNS query
and reply messages are sent within UDP datagrams to PORT 53. After a
delay, ranging from milliseconds to seconds, DNS in the user’s host
receives a DNS reply message that provides the desired mapping. This
mapping is then passed to the invoking application. Thus, from the
perspective of the invoking application in the user’s host, DNS is a black
box providing a simple, straightforward translation service. but in fact, the
black box that implements the service is complex, consisting of a large
number of DNS servers distributed around the globe, as well as an
application-layer protocol that specifies how the DNS servers and
querying hosts communicate.

A simple design for DNS would have one DNS server that
contains all the mappings. In this centralized design, clients simply direct
all queries to the single DNS server, and the DNS server responds directly
to the querying clients. Although the simplicity of this design is attractive,
it is inappropriate for today’s internet, with its vast (and growing) number
of hosts. The problems with a centralized design include:

a. A single point of failure. If the DNS server crashes, so does the entire
internet!
b. Traffic volume. A single DNS server would have to handle all DNS
queries.
c. Distant centralized database. A single DNS server cannot be “close
to” all the querying clients. If we put the single DNS server in Mumbai
city, then all queries from Australia must travel to the other side of the
globe, maybe over slow and congested links. This can lead to
significant delays.
d. Maintenance. The single DNS server would have to keep records for
all internet hosts. Not only would this centralized database be large,
but it would have to be updated frequently to account for every new
host.

1.8.3 DNS Records and Messages

The DNS servers that together implement the DNS distributed

database store resource records (RRS); including RRS that provide
hostname-to-IP address mappings. Each DNS reply message carries one or
more resource records.

A resource record is a four-tuple that contains the following fields:

(Name, Value, Type, TTL)

TTL is the time to live off the resource record; it determines when
a resource should be removed from a cache. In the example records given

22
below, we ignore the TTL field. The meaning of name and value depend
on type:

● If Type=A, then name is a hostname and value is the IP address for

the hostname. Thus, a type record provides the standard hostname-
to-IP address mapping. As an example, (relay1.bar.foo.com,
145.37.93.126, a) is a type of record.
● If Type=NS, then name is a domain (such as foo.com) and value is
the hostname of an authoritative DNS server that knows how to
obtain the IP addresses for hosts in the domain. This record is used
to route DNS queries further beside in the query chain. As an
example, (foo.com, DNS.foo.com, NS) is a type NS record.
● If Type=CNAME, then value is a valid or canonical hostname for
the alias hostname name. This record can provide querying hosts
the canonical name for a hostname. As an example, (foo.com,
relay1.bar.foo.com, CNAME) is a CNAME record.
● If Type=MX, the value is the valid or canonical name of a mail
server that has an alias hostname name. As an example, (abc.com,
mail.bar.abc.com, MX) is an MX record. MX records allow the
hostnames of mail servers to have simple aliases. Note that by
using the MX record, a company can have the same aliased name
for its mail server and for one of its other servers (such as its web
server). To obtain the canonical name for the mail server, a DNS
client would query for an MX record; to obtain the canonical name
for this server, the DNS client would query for the CNAME
record.

DNS messages

There are only two types of DNS messages. Both query and reply
messages have the same format, as shown in figure 3.the semantics of the
various fields in a DNS message are as follows:

a. The first 12 bytes is the header section of message, which has a

number of fields. In the message the first field is a 16-bit number that
identifies the query. This identifier is copied into the reply message to
a query, allowing the client to match received replies with sent queries.
These are a number of flags in the flag field. A 1-bit reply flag
indicates, the message is a query (0) or a reply (1). A 1-bit recursion-
desired flag is set when a client (host or DNS server) capable that the
DNS server perform recursion when it doesn’t have the record. A 1-bit
recursion available field is set in a reply if the DNS server supports
recursion. In the header, those are also four number-of fields. Which
indicate the number of occurrences of the four types of data parts that
follow the header.

23
 Fig. 11 DNS message format

b. The question part of the message contains information about the query
that is being made. This part of message includes a name field that
contains the name that is being queried, and a type field that indicates
the type of question being asked about the name for example, a host
address associated with a name (Type A) or the mail server for a name
(Type MX).
c. In a reply from a DNS server, the answer part of the message contains
the resource records for the name that was originally queried.
Recollect that in each resource record those is the type (for example,
A, NS, CNAME, and MX), the value, and the TTL. A reply can return
multiple RRS in the answer part of the message, since a hostname can
have multiple IP addresses.
d. The authority section of the message contains records of other
authoritative servers.
e. The additional section of the message contains other helpful records.
For example, the answer field in reply to an MX query contains a
resource record providing the canonical hostname of a mail server.

1.7 SUMMARY

1. In this chapter, we’ve studied the conceptual and the implementation

aspects of network applications.
2. We’ve learned about the ubiquitous client-server architecture adopted
by many Internet applications and seen its use in the HTTP, FTP
protocols.
3. We’ve studied these important application-level protocols, and their
corresponding associated applications (the web, file transfer) in some
detail.

24
4. We have also learn difference between Non-Persistent and Persistent
Connections
5. We have seen the HTTP Message format & HTTP Response Message.

1.8 REFERENCE FOR FURTHER READING

1. A Computer Networking, A Top-Down Approach Kurose & ross

2. TCP/IP Protocol Suite 4 edition, Beerhouse Frozen, McGraw-Hill
Science

1.9 UNIT END EXERCISES

1. What is meant by a handshaking protocol?

2. Why is it said that FTP sends control information “out-of-band”?
3. Explain the Protocol layer and their services?
4. Write a short note on:
a. Non-Persistent and Persistent Connections
b. HTTP Message Format
c. HTTP Response Message
d. User-Server Interaction: Cookies



25
 2
NETWORKING - II
Unit Structure

2.1 Objective
2.2 Transport-Layer Services,
2.2.1 Relationship Between Transport And Network Layers
2.2.2 Overview Of The Transport Layer In The Internet
2.3 Multiplexing And Demultiplexing,
2.4 Udp
2.4.1 User Datagram
2.4.2 Udp Services
2.4.3 Udp Applications
2.5 Tcp
2.5.1 Tcp Features
2.5.2 Segment
2.6 Tcp Congestion Control
2.7 Summary
2.8 Reference For Further Reading
2.9 Unit End Exercises

2.1 OBJECTIVE

1. To understand the major components of electronic mail in the

internet
2. To understand the services provided by DNS
3. To learn the mail message formats & access protocols.
4. To understand the transport-layer services

2.2 TRANSPORT-LAYER SERVICES

A transport-layer protocol provides for rational communication

between application processes running on different hosts. In logical
communication, we mean that from an application’s viewpoint, it is as if
the hosts running the processes were directly connected. In the real world,
the hosts may be on opposite sides of the planet, connected via numerous
routers and a wide range of link types. The application processes use the
logical communication provided by the transport layer to send messages to
each other, free from the worry of the details of the physical infrastructure
used to carry these messages. Figure 4 shows the notion of logical
communication.
26
As shown in figure 4, function of transport layer
1. Transport-layer protocols are implemented in the end systems.
2. The transport layer converts the application-layer messages it
receives from a sending application process into transport-layer
packets, known as transport-layer segments in internet
terminology.
3. The transport layer then passes the segment to the network layer at
the sending end system, whose segment is encapsulated within a
network-layer packet (a datagram) and sent to the destination.
4. The network routers act only on the network-layer fields of the
datagram.
5. At the receiving side, the network layer extracts the transport-layer
segment from the datagram and passes the segment up to the
transport layer. the transport layer then processes the received
segment, making the data in the segment available to the receiving
application.

2.2.1 Relationship between Transport and Network layers

The transport layer is just above the network layer in the protocol
stack. Whereas a transport-layer protocol provides logical communication
between processes running on different hosts, a network-layer protocol
provides logical communication between hosts.

2.2.2 Overview of the Transport Layer in the Internet

In TCP/IP, the two distinct transport-layer protocols available to
the application layer. Among two one of these protocols is UDP (user
datagram protocol), which provides an unreliable, connectionless service
to the invoking application. The second protocol is TCP (transmission
control protocol), which provides a reliable, connection-oriented service to
the invoking application. When designing a network application, the
application developer selects between UDP and TCP when creating
sockets.

27
Fig. 4 the transport layer provides logical rather than physical
communication between application processes

2.3 MULTIPLEXING AND DEMULTIPLEXING

In the transport-layer protocol multiplexing and demultiplexing, is

extending the host-to-host delivery service provided by the network layer
to a process-to-process delivery service for applications running on the
hosts. A multiplexing & demultiplexing service is needed for all computer
networks. at the destination host, the transport layer receives segments
from the network layer. the transport layer has the responsibility of
delivering the data in these segments to the appropriate application process
running on the host. Example. User sitting in front of the computer and
user are downloading web pages while running one FTP session and two
telnet sessions. These four network application processes run two telnet
processes, one FTP process, and one HTTP process. When the transport
layer in the user computer receives data from the network layer below, it
needs to direct the received data to one of these four processes.

In the network application, a process can have one or more

sockets, doors through which data passes from the network to the process
and through which data passes from the process to the network back.
Thus, as shown in figure 4, the transport layer in the receiving host does
28
not actually deliver data directly to a process. Each socket has a unique
identifier. The format of the identifier depends on whether the socket is a
UDP or a TCP socket.

Delivering the data in a transport-layer segment to the correct

socket is called demultiplexing. The job of gathering data chunks at the
source host from different sockets, encapsulating each data chunk with
header information to create segments, and passing the segments to the
network layer is called multiplexing. The transport layer in the middle host
must also gather outgoing data from these sockets, form transport-layer
segments, and pass these segments down to the network layer.

Fig. 5 transport-layer multiplexing and demultiplexing

The roles of transport-layer multiplexing and demultiplexing:

The sockets have unique identifiers, and that each segment has
special fields that indicate the socket to which the segment is to be
delivered. These special fields, Shown in figure 6, are the source port
number field and the destination port number field, The 16-bit port
number, between 0 to 65535. The port numbers ranging from 0 to 1023
are called well-known port numbers and are restricted, which means that
they are reserved for use by well-known application protocols such as
HTTP and FTP.

Fig. 6 Source and destination port-number fields in a transport-layer

segment

29
Connectionless Multiplexing and Demultiplexing
The program running in a host can create a UDP socket with the line

clientsocket = socket(socket.af_inet, socket.sock_dgram)

When a UDP socket is created in this way, the transport layer

automatically assigns a port number to the socket. The transport layer
allots a port number in the range 1024 to 65535 that is currently not being
used by any other UDP port in the host. In python program after creating
the socket to associate a specific port number (say, 19157) to this UDP
socket via the socket bind() method:

clientsocket.bind((‘’, 19157))

Connection-Oriented multiplexing and demultiplexing

One best difference between a TCP socket and a UDP socket is that a
TCP socket is identified by a four-tuple:
1. sourceip address.
2. source port number.
3. destinationip address.
4. destination port number.
when a TCP segment arrives from the network to a host, the host uses all
four values to direct the segment to the appropriate socket. In contrast with
udp, two arriving TCP segments with different source IP addresses or
source port numbers will be directed to two different sockets. to gain
further insight, let’s rethink the TCP client-server programming.

Fig. 7 The inversion of source and destination port numbers

❏ The TCP server application has a “welcome socket,” that waits for
connection establishment requests from TCP clients on port
number 12000.
❏ The TCP client creates a socket and sends a connection
establishment request message with the lines:

30
 clientsocket = socket(af_inet, sock_stream)
clientsocket.connect((servername,12000))
❏ A connection initiation request is nothing more than a TCP
segment with destination port number 12000 and a special
connection-establishment bit set in the TCP header. The message
also includes a source port number that was chosen by the user.
❏ The host OS of the computer running the server process receives
the incoming connection-request segment with destination port
12000, it locates the server process that is waiting to accept a
connection on port number 12000. the server process then creates a
new socket:
connectionsocket, addr = serversocket.accept()
❏ The transport layer at the server notes the following four values in
the connection-request segment:
a. the source port number in the segment
b. the ip address of the source host
c. the destination port number in the segment
d. its own ip address.

The recently created connection socket is identified by these four

values; all subsequently arriving segments whose source port, source IP
address, destination port, and destination IP address match these four
values will be demultiplexed to this socket. with the TCP connection now
in place, the client and server can now send data to each other.

2.4 UDP

Figure 8 shows the relationship of the User Datagram Protocol

(UDP) to the other protocols and layers of the TCP/IP protocol suite: UDP
is located between the application layer and the IP layer, and serves as the
intermediary between the application programs and the network
operations.

Fig. 8 Position of UDP in the TCP/IP protocol suite

A transport layer protocol usually has several responsibilities. One

is to create a process-to-process communication. UDP uses port numbers.

31
Second responsibility is to provide control mechanisms at the transport
level. UDP doesn't have a flow control mechanism and no
acknowledgement for received packets. UDP does provide error control to
some extent. If UDP detects an error in the received packet, it drops it.

UDP is a connectionless, unreliable transport protocol. UDP

provides process-to-process communication instead of host-to-host
communication.

UDP is a very simple protocol using a minimum of cost. If a

process wants to send a small segment and does not look much about
reliability, it can use UDP. Sending a small segment using UDP takes
much less interaction between the sender and receiver than using TCP.

2.4.1 USER DATAGRAM

UDP packets, called user datagrams. UDP has a fixed-size header

of 8 bytes. Figure 9 shows the format of a user datagram in detail.

Fig. 9 User datagram format

● Source port number.

1. This is the port number used by the process running on the source
host.
2. It is 16 bits long, port number can range from 0 to 65,535.
3. If the source host is the client, the port number, in most cases, is a
transitory. port number requested by the process.
4. If the source host is the server, the port number is a well-known
port number.

● Destination port number.

1. This is the port number, running on the destination host.
2. 16 bits long.
3. If the destination host is the server, is a well-known port number.
4. If the destination host is the client, is a transitory port number.

32
● Length.
1. Length is a 16-bit field that defines the total length of the user
datagram, header plus data.
2. It defines a total length of 0 to 65,535 bytes.
3. A user datagram is enclosed in an IP datagram.
4. There is a field in the IP datagram that defines the total length.
5. There is another field in the IP datagram that defines the length of
the header.

UDP datagram that is encapsulated in an IP datagram.

UDP length= IP length− IP header’s length

● Checksum. This field is used to detect errors over the entire user
datagram (header
plus data). The checksum is discussed in the next section.

Example
The following is a dump of a UDP header in hexadecimal format.

CB84000D001C001C

a. The source port number is the first four hexadecimal digits

(CB8416), which means that the source port number is 52100.
b. The destination port number is the second four hexadecimal digits
(000D16), which means that the destination port number is 13.
c. The third four hexadecimal digits (001C16) define the length of the
whole UDP packet as 28 bytes.
d. The length of the data is the length of the whole packet minus the
length of the header, or 28 – 8 = 20 bytes.
e. Since the destination port number is 13, the packet is from the
client to the server.
f. The client process is the Daytime.

2.4.2 UDP SERVICES

UDP provides process-to-process communication using sockets, a

combination of IP addresses and port numbers. Several port numbers used
by UDP are shown in Table 1

33
 Table 1. Well-known Ports used with UDP

Connectionless Services
UDP provides a connectionless service. The user datagram sent by
UDP is an independent datagram. There is no connection between the
different user datagrams even if they are coming from the same source
process and going to the same destination program. The user datagrams
are not numerical. There is no connection establishment and no connection
termination as is the case for TCP. This means that each user datagram can
travel on a different path.

Flow Control
UDP is a very simple protocol. There is no flow control, and hence
no window mechanism. The receiver may overflow with incoming
messages. The lack of flow control means that the process using UDP
should provide for this service, if needed.

Error Control
There is no error control mechanism in UDP except for the
checksum. The sender does not realize that if a message has been lost or
duplicated. When the receiver found an error through the checksum, the
user datagram is discarded.

Checksum
UDP checksum calculation is different from the one for IP. Here
the checksum includes three sections:
1. A pseudo header,
2. UDP header, and
3. Data coming from the application layer

34
 Fig. 10 UDP Header

If the checksum does not include the pseudo header, a user

datagram may arrive safe and sound. However, if the IP header is
corrupted, it may be delivered to the wrong Host. The protocol field is
added to make sure that the packet belongs to UDP, and not to TCP. The
value of the protocol field in the datagram for UDP is 17. If this value is
changed during transmission, the checksum calculation at the receiver will
detect it and UDP drops the packet. It is not delivered to the wrong
protocol.

Congestion Control
UDP is a connectionless protocol; it does not provide congestion
control. UDP assumes that the packets sent are small and sporadic, and
cannot create congestion in the network.

Encapsulation and Decapsulation

To send a message from one process to another, the UDP protocol
encapsulates and
decapsulates messages.

Fig. 11Encapsulation and decapsulation

35
Encapsulation
In Encapsulation, it passes the message to UDP along with a pair
of socket addresses and the length of data. UDP receives the data and adds
the UDP header. UDP then passes the user datagram to IP with the socket
addresses. IP adds its own header, using the value 17 in the protocol field,
indicating that the data has come from the UDP protocol. The IP datagram
is then passed to the data link layer. The data link layer receives the IP
datagram, adds its own header and passes it to the physical layer. The
physical layer encodes the bits into electrical or optical signals and sends it
to the remote machine.

Decapsulation
When the message arrives at the destination host, the physical
layer decodes the signals into bits and passes it to the data link layer. The
data link layer uses the header to check the data. If there is no error, the
header and trailer are dropped and the datagram is passed to IP. The IP
software does its own checking. If there is no error, the header is dropped
and the user datagram is passed to UDP with the sender and receiver IP
addresses. UDP uses the checksum to check the entire user datagram. If
there is no error, the header is dropped and the application data along with
the sender socket address is passed to the process. The sender socket
address is passed to the process in case it needs to respond to the message
received.

Queuing
In UDP, queues are associated with ports At the client site, when a
process starts, it requests a port number from the operating system. Some
implementations create both an incoming and an outgoing queue
associated with each process. Other implementations create only an
incoming queue associated with each process

Fig. 12 Queues in UDP

2.4.3 UDP APPLICATIONS

● UDP is suitable for a process that requires simple request-response

communication with flow and error control mechanisms. It is not
usually used for a process such as FTP that needs to send bulk data.
36
 ● UDP is suitable for a process with internal flow and error-control
mechanisms.
● UDP is a suitable transport protocol for multicasting. Multicasting
capability is embedded in the UDP software but not in the TCP
software.
● UDP is used for management processes such as SNMP.
● UDP is used for some route updating protocols such as Routing
Information Protocol (RIP).
● UDP is normally used for real-time applications that cannot
tolerate uneven delay between sections of a received message.

2.5 TCP

Figure 13 shows the relationship of TCP to the other protocols in

the TCP/IP protocol suite. TCP lies between the application layer and the
network layer, and serves as the intermediary between the application
programs and the network operations.

Fig. 13 TCP/IP protocol suite

Process-to-Process Communication
As with UDP, TCP provides process-to-process communication
using port numbers.
Table 2 & 3 lists some well-known port numbers used by TCP.

Table 2. Well-known Ports used by TCP

37
 Table 3. Well-known Ports used by TCP

Stream Delivery Service

TCP, unlike UDP, is a stream-oriented protocol. In UDP, a process
sends messages with predefineborder to UDP for delivery. UDP adds its
own header to each of these messages and delivers to the destination or to
IP for transmission. Each message from this process is called a user
datagram.

TCP, permits the sending process to deliver data as a stream of

bytes and allows the receiving process to obtain data as a stream of bytes.
TCP creates adomain in which the two processes seem to be connected by
an imaginary “tube” that carries their bytes across the Internet. This
imaginary environment is depicted in Figure 14. The sending process
writes to the stream of bytes and the receiving process reads from them.

Fig. 14.Stream delivery

Sending and Receiving Buffers

The sending and the receiving processes may not definitely write
or read data at the same rate, TCP required buffers for storage. There are
two buffers, the sending buffer and the receiving buffer, one for each
direction. On sided to implement a buffer is to use a circular array of 1-
byte locations as shown in Figure 15shows two buffers of 20 bytes each.

Fig. 15 Sending and receiving buffers

38
Segments
At the transport layer, TCP groups a number of bytes together into
a packet called a segment. TCP adds a header to each segment and
delivers the segment to the IP layer for transmission. The segments are
encapsulated in an IP datagram and transmitted. This entire operation is
transparent to the receiving process. Figure 16 shows how segments are
created from the bytes in the buffers.

Fig. 16 TCP segments

Full-Duplex Communication
TCP offers full-duplex service, where data can flow in both
directions at the same time. Each TCP endpoint then has its own sending
and receiving buffer, and segments move in both directions.

Multiplexing and Demultiplexing

Like UDP, TCP performs multiplexing at the sender and
demultiplexing at the receiver. However, since TCP is a connection-
oriented protocol, a connection needs to be established for each pair of
processes.

Connection-Oriented Service
TCP, unlike UDP, is a connection-oriented protocol. when a
process at site A wants to send to and receive data from another process at
site B, the following three phases occur:
1. The two TCPs establish a virtual connection between them.
2. Data is exchanged in both directions.
3. The connection is terminated.

Reliable Service
TCP is a reliable transport protocol. It uses an acknowledgment
mechanism to check the safe and sound arrival of data. We will discuss
this feature further in the section on error control.

2.5.1 TCP FEATURES

Numbering System
TCP software keeps track of the segments being transmitted or
received, there is no field for a segment number value in the segment
header. Instead, there are two fields called the sequence number and the
acknowledgment number. These two fields refer to a byte number.
39
Byte Number
The bytes of data being transferred in each connection are
numbered by TCP. The numbering starts with an arbitrarily generated
number.

Sequence Number
After the bytes have been numbered, TCP assigns a sequence
number to each segment that is being sent. The sequence number for each
segment is the number of the first byte of data carried in that segment.

Acknowledgment Number
When a connection is established, both parties can send and
receive data at the same time. Each party numbers the bytes, usually with
a different starting byte number. The sequence number in each direction
shows the number of the first byte carried by the segment. Each party also
uses an acknowledgment number to confirm the bytes it has received.

The value of the acknowledgment field in a segment defines the

number of the next byte a party expects to receive. The acknowledgment
number is cumulative.

Flow Control
UDP provides flow control. The sending TCP controls how much
data can be accepted from the sending process; the receiving TCP controls
how much data can be sent by the sending TCP. This is done to prevent
the receiver from being overwhelmed with data. The numbering system
allows TCP to use a byte oriented flow control.

Error Control
To provide reliable service, TCP implements an error control
mechanism. error control considers a segment as the unit of data for error
detection, error control is byte-oriented.

Congestion Control
TCP, unlike UDP, takes into account congestion in the network.
The amount of data sent by a sender is not only controlled by the receiver
(flow control), but is also determined by the level of congestion, if any, in
the network.

2.5.2 SEGMENT

A packet in TCP is called a segment.

Format
The format of a segment is shown in Figure 17. The segment
consists of a header of 20 to 60 bytes, followed by data from the
application program. The header is 20 bytes if there are no options and up
to 60 bytes if it contains options.

40
 Fig. 17 TCP segment format

● Source port address. This is a 16-bit field that defines the port
number of the application program in the host that is sending the
segment.
● Destination port address. This is a 16-bit field that defines the port
number of the application program in the host that is receiving the
segment.
● Sequence number. This 32-bit field defines the number assigned to
the first byte of data contained in this segment.
● Acknowledgment number. This 32-bit field defines the byte number
that the receiver of the segment is expecting to receive from the other
party.
● Header length. This 4-bit field indicates the number of 4-byte words
in the TCP header. The length of the header can be between 20 and 60
bytes. Therefore, the value of this field is always between 5 (5× 4= 20)
and 15 (15× 4= 60).
● Reserved. This is a 6-bit field reserved for future use.
● Control. This field defines 6 different control bits or flags as shown in
Figure 18 One or more of these bits can be set at a time. These bits
enable flow control, connection establishment and termination,
connection abortion, and the mode of data transfer in TCP.

Fig. 18 Control field

41
 ● Window size. This field defines the window size of the sending
TCP in bytes. Note that the length of this field is 16 bits, which
means that the maximum size of the window is 65,535 bytes.
● Checksum. This 16-bit field contains the checksum.

Fig. 19 Pseudoheader added to the TCP datagram

● Urgent pointer. This 16-bit field, which is valid only if the urgent
flag is set, is used when the segment contains urgent data.
● Options. There can be up to 40 bytes of optional information in
the TCP header.

2.6 TCP CONGESTION CONTROL

Congestion control in TCP is based on both open-loop and closed-

loop mechanisms. TCP uses a congestion window and a congestion policy
that avoids congestion and detects and alleviates congestion after it has
occurred.

Congestion Window
If the network cannot deliver the data as fast as it is created by the
sender, it must tell the sender to slow down. In other words, in addition to
the receiver, the network is a second entity that determines the size of the
sender’s window.

The sender has two pieces of information: the receiver-advertised

window size and the congestion window size. The actual size of the
window is the minimum of these two.

Actual window size = minimum (rwnd, cwnd)

Congestion Policy
TCP’s general policy for handling congestion is based on three
phases: slow start, congestion avoidance, and congestion detection. In the
slow start phase, the sender starts with a slow rate of transmission, but
increases the rate rapidly to reach a threshold. When the threshold is
reached, the rate of increase is reduced. Finally if ever congestion is

42
detected, the sender goes back to the slow start or congestion avoidance
phase, based on how the congestion is detected.

2.7 SUMMARY

1. We learned that a transport-layer protocol can provide reliable data

transfer even if the underlying network layer is unreliable
2. We learned that TCP is complex, involving connection management,
flow control, and round-trip time estimation, as well as reliable data
transfer.
3. we took a close look at TCP, the Internet’s connection-oriented and
reliable transport-layer protocol.
4. We examined congestion control from a broad perspective, we showed
how TCP implements congestion control.
5. we learned that TCP implements an end-to-end congestion-control
mechanism.

2.8 REFERENCE FOR FURTHER READING

1. Computer Networking: A Top-Down Approach 6th edition, James F.

Kurose, Keith W. Ross, Pearson (2012).
2. TCP_IP Protocol Suite 4th ed. - B. Forouzan (McGraw-Hill, 2010)
BBS

2.9 UNIT END EXERCISES

1. Describe why an application developer might choose to run an

application over UDP rather than TCP.
2. What is the difference between Multiplexing and Demultiplexing?
3. Explain Difference TCP & UDP.
4. What mechanism is used in TCP Congestion Control.
5. Explain & List the UDP Applications.



43
 3
NETWORKING - III
Unit Structure

3.1 Objective
3.2 Introduction
3.3 Network Layer
3.3.1 Switching
3.3.2 Packet Switching At Network Layer
3.4 Virtual Circuit and Datagram Networks
3.5 Need of Router
3.6 The Internet Protocol (IP),
3.7 Summary
3.8 Reference for further reading
3.9 Unit End Exercises

3.1 OBJECTIVE

1. To learn exactly how the network layer implements the host-to-host

communication service.
2. To examine two broad approaches towards structuring network-layer
packet delivery the datagram and the virtual-circuit model—and see
the fundamental role that addressing plays in delivering a packet to its
destination host.
3. To understand an important distinction between the forwarding and
routing functions of the network layer.

3.2 INTRODUCTION

For example a simple network with two hosts, X1 and X2, and
several routers on the path between X1 and X2. Suppose that X1 is
sending data to X2, and consider the role of the network layer in these
hosts and in the intervening routers. The network layer in X1 takes
messages from the transport layer in X1, encapsulates each message into a
datagram and then sends the datagrams to its nearby router, R1. At the
receiving host, X2, the network layer receives the datagrams from its
nearby router R2, extracts the transport-layer segments, and delivers the
segments up to the transport layer at X2. The primary role of the routers is
to forward datagrams from input links to output links.

44
 Fig. 1 Network Layer

Forwarding and Routing:

The role of the network layer is simple to move data packets from
a sending host to a receiving host.
Two network-layer functions:
1. Forwarding. When a packet comes at a router’s input link, the
router must move the packet to the appropriate output link. For
example, a packet arriving from Host X1 to Router R1 must be
forwarded to the next router on a path to X2.

2. Routing. The network layer must determine the route or path taken
by data packets as they flow from a sender to a receiver. The
algorithms that calculate these ride are referred to as routing
algorithms. A routing algorithm would determine, for example, the
path along which packets flow from X1 to X2.

Forwarding refers to the router-local action of transferring a packet

from an input link interface to the appropriate output link interface.
Routing refers to the network-wide process that finds the end-to-end paths
that packets take from source to destination.

Network Service Models:

The network service model defines the characteristics of end-to-
end transport of packets between sending and receiving systems. In the
sending host, when the transport layer passes a packet to the network
layer, specific services that could be provided by the network layer
include:
45
 a. Guaranteed delivery. This service gives assurance that the packet
will eventually arrive at its destination.
b. Guaranteed delivery with bounded delay. This service not only
guarantees delivery of the packet, but delivery within a specified
host-to-host delay bound.

The following services provided to a flow of packets between a source and

destination:

a. In-order packet delivery. This service guarantees that packets

arrive at the destination in the order that they were sent.
b. Guaranteed minimum bandwidth. This network-layer service
emulates the behavior of a transmission link of a specified bit rate
between sending and receiving hosts. As long as the sending host
transmits bits at a rate below the specified bit rate, then no packet
is lost and each packet arrives within a pre specified host-to-host
delay.
c. Guaranteed maximum jitter. This service guarantees that the
amount of time between the transmission of two successive
packets at the sender is equal to the amount of time between their
receipt at the destination.
d. Security services. Using a secret session key known only by a
source and destination host, the network layer in the source host
could encrypt the payloads of all datagrams being sent to the
destination host. In addition to confidentiality, the network layer
could provide data integrity and source authentication services.

The Internet’s network layer provides a single service, known as

best-effort service. From Table 1, it might appear that best-effort service is
a substitute for no service at all. With best-effort service, timing between
packets is not guaranteed to be preserved, packets are not guaranteed to be
received in the order in which they were sent, nor is the eventual delivery
of transmitted packets guaranteed. Given this definition, a network that
delivered no packets to the destination would satisfy the definition of best-
effort delivery service.

Table 1. Internet, ATM CBR, and ATM ABR service models

46
Constant bit rate (CBR) ATM network service. This was the first ATM
service model to be standardized, reflecting early interest by the telephone
companies in ATM and the suitability of CBR service for carrying real-
time, constant bit rate audio and video traffic. The goal of CBR service is
conceptually simple to provide a flow of packets with a virtual pipe whose
properties are the same as if a devoted fixed-bandwidth transmission link
existed between sending and receiving hosts. With CBR service, a flow of
ATM cells is carried across the network in such a way that a cell’s end-to-
end delay, the variability in a cell’s end-to-end delay, and the selection of
cells that are lost or delivered late are all guaranteed to be less than
specified values. These values are acknowledged upon by the sending host
and the ATM network when the CBR connection is first established.

Available bit rate (ABR) ATM network service. Internet offering so

called best-effort service, ATM’s ABR might best be characterized as
being a slightly-better-than-best-effort service. As with the Internet service
model, cells may be lost under ABR service. Cells cannot be reordered,
and a minimum cell transmission rate (MCR) is guaranteed to a
connection using ABR service. If the network has enough free resources at
a given time, a sender may also be able to send cells successfully at a
higher rate than the MCR.

3.3 NETWORK LAYER

3.3.1 Switching

Circuit Switching

Circuit switching, in which a physical circuit is established

between the source and destination of the message before the delivery of
the message. After the circuit is established, the entire message is
transformed from the source to the destination. The source can then inform
the network that the transmission is complete, which allows the network to
open all switches and use the links and connecting devices for another
connection.

In circuit switching, the whole message is sent from the source

to the destination without being divided into packets.

Example: A good example of a circuit-switched network is the early

telephone systems in which the path was established between a caller and
a callee when the telephone number of the caller was dialed by the caller.
When the callee responded to the call, the circuit was established. The
voice message could now flow between the two parties, in both directions,
while all of the connecting devices maintained the circuit. When the caller
or callee hung up, the circuit was disconnected. The telephone network is
not totally a circuit-switched network today.

47
Packet Switching
The network layer in the The Internet today is a packet-switched
network. In this type of network, a message from the upper layer is
divided into manageable packets and each packet is sent through the
network. The source of the message sends the packets one by one; the
destination of the message receives the packets one by one. The
destination waits for all packets belonging to the same message to arrive
before delivering the message to the upper layer. The connecting devices
in a packet-switching network still need to decide how to route the packets
to the final destination. Today, a packet-switched network can use two
different approaches to route the packets: the datagram approach and the
virtual circuit approach.

In packet switching, the message is first divided into

manageable packets at the source before being transmitted. The
packets are assembled at the destination

3.3.2 Packet Switching At Network Layer

The network layer is designed as a packet-switched network. This
means that the packet at the source is divided into manageable packets,
normally called datagrams. Individual datagrams are then transferred from
the source to the destination. The received datagrams are assembled at the
destination before recreating the original message. The packet-switched
network layer of the Internet was originally designed as a connectionless
service, but recently there is a tendency to change this to a connection
oriented service. We first discuss the dominant trend and then briefly
discuss the new one.

Connectionless Service
The network layer was designed to provide a connectionless
service, in which the network layer protocol treats each packet
independently, with each packet having no relationship to any other
packet. The packets in a message may or may not travel the same path to
their destination. When the Internet started, it was decided to make the
network layer a connectionless service to make it simple. The idea was
that the network layer is only responsible for delivery of packets from the
source to the destination.

Fig. 2 A connectionless packet-switched network

48
Connection-Oriented Service

In a connection-oriented service, there is a connection between all

packets belonging to a message. a virtual connection should be set up to
define the path for the datagrams after that all datagrams in a message can
be sent. After connection established, the datagrams can follow the same
path. In this connection oriented service, the packet contains the source
and destination addresses, it must also contain a flow label, a virtual
circuit identifier that defines the virtual path the packet should follow.
Figure 3 shows the concept of connection-oriented service.

Fig. 3 connection-oriented packet switched network

3.4 VIRTUAL CIRCUIT AND DATAGRAM NETWORKS

Computer networks that provide only a connection service at the

network layer are called virtual circuit networks, computer networks that
provide only a connectionless service at the network layer are called
datagram networks.

Characteristics:
1. As in a circuit-switched network, there are setup and teardown
phases in addition to the data transfer phase.
2. Resources can be assign 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 is packetized and each packet
carries an address in the header. The reader may ask how the
intermediate switches know where to send the packet if there is no
final destination address carried by a packet.
4. Circuit-switched network, all packets follow the same path formed
during the connection.
5. A virtual-circuit network is implemented in the data link layer,
while a circuit-switched network is implemented in the physical
layer and a datagram network in the network layer.

49
 Figure 4 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, bridge, or any other device
that connects other networks.

Fig. 4 A simple virtual circuit network

Virtual-Circuit Networks

ATM and frame relay are virtual-circuit networks and therefore,

use connections at the network layer. These network-layer connections are
called virtual circuits.

A virtual circuit consists of

1. a path between the source and destination hosts.
2. virtual circuit numbers, one number for each link as well the path,
and
3. Updating the forwarding table in each router along the path.

A packet belonging to a virtual circuit will carry a virtual circuit

number in its header. Because a virtual circuit may have a different virtual
circuit number on each link, each intervening router must replace the
virtual circuit number of each traversing packet with a new virtual circuit
number. The new virtual circuit number is obtained from the forwarding
table.

Fig. 5 A simple virtual circuit network

The numbers next to the links of R1 in Figure 5 are the link

interface numbers. A requests that the network establish a virtual circuit
between itself and Host B. Suppose also that the network chooses the path
50
A-R1-R2-B and assigns virtual circuit numbers 12, 22, and 32 to the three
links in this path for this virtual circuit. In this case, when a packet in this
virtual circuit leaves Host A, the value in the virtual circuit number field
in the packet header is 12 when it leaves R1 Virtual Circuits, the value is
22; and when it leaves R2, the value is 32.

There are three identifiable phases in a virtual circuit:

1. Virtual circuit setup. during the setup phase, the sending transport
layer contacts the network layer, specifies the receiver’s address, and
waits for the network to set up the virtual circuit . The network layer
determines the path between sender and receiver, that is, the series of
links and routers through which all packets of the virtual circuit will
travel. The network layer also determines the virtual circuit number for
each link along the path. At last, the network layer adds an entry in the
forwarding table in each router along the path details.

2. Data transfer. As shown in Figure 6, once the virtual circuit has been
established, packets can begin to flow along the virtual circuit .

3. Virtual circuit teardown. This is initiated when the sender informs

the network layer of its desire to terminate the virtual circuit. The
network layer will then typically inform the end system on the other
side of the network of the call termination and update the forwarding
tables in each of the packet routers on the path to indicate that the
virtual circuit no longer exists.

Fig. 6 Virtual-circuit setup

The messages that the end systems send into the network to initiate
or terminate a Virtual Circuits, and the messages passed between the
routers to set up the Virtual Circuits are known as signaling messages, and
the protocols used to exchange these messages are often referred to as
signaling protocols. Virtual Circuits setup is shown pictorially in Figure 6.

Datagram Networks:
In a datagram network, a packet is sent to an end system, it stamps
the packet with the address of the destination end system and then pops
the packet into the network. As shown in Figure 7, there is no Virtual
Circuits setup and routers do not maintain any Virtual Circuits state
information.
51
 As a packet is transmitted from source system to destination
system, it passes through a series of routers. All of these routers use the
packet’s destination address to forward the packet. especially.. Each router
has a forwarding table that maps destination addresses to link interfaces.
when a packet arrives at the router, the router uses the packet’s destination
address to look up the appropriate output link interface in the forwarding
table. The router then purposely forwards the packet to that output link
interface.

Fig. 7 Datagram network

3.5 NEED OF ROUTER

A router is a three layer device that routes packets according to

their logical addresses. A router usually connects LANs and WANs on the
Internet and has a routing table that is used for making decisions about the
route. The routing tables are dynamic and are updated using routing
protocols. Figure 1 shows a part of the Internet that uses routers to connect
LANs and WANs.

Fig. 1 Routers connecting independent LANs and WANs

Forwarding function, the actual transfer of packets from a router’s

incoming links to the appropriate outgoing links at that router. The terms
forwarding and switching are often used interchangeably by computer-
networking researchers and practitioners

A high-level view of a generic router architecture is shown in Figure 2

Four router components can be identified:

52
● Input ports
An input port performs several key functions. It executes the physical
layer function of terminating an incoming physical link at a router; this
is shown in the leftmost box of the input port and the rightmost box of
the output port in Figure.

● Switching fabric
The switching fabric connects the router’s input ports with its output
ports. This switching fabric is totally contained within the router a
network inside of a network router.

● Output ports
An output port stores packets received from the switching fabric and
sends these packets on the outgoing link by performing the necessary
link-layer and physical-layer functions.

● Routing processor
The routing processor accomplish the routing protocols, maintains
routing tables and attached link state information, and computes the
forwarding table for the router. It also performs the network
management functions.

Fig. 2 Router architecture

3.6 THE INTERNET PROTOCOL (IP)

The function of IP, how addressing and forwarding are done on the
Internet. The important components of the Internet Protocol (IP) is internet
addressing and forwarding. There are two versions of IP. IP protocol
version 4, which is usually referred to simply as IPv4. IP version 6, which
has been proposed to replace IPv4.

53
Fig. 3 A look inside the Internet’s network layer
.
As shown in Figure 3, the Internet’s network layer has three major
components.

IP protocol: The first component is the IP protocol.

Routing Component: The second major component is the routing

component, which determines the path a datagram follows from source to
destination. routing protocols calculate the forwarding tables that are used
to forward packets through the network.

ICMP Protocol: The final component of the network layer is a facility to

report errors in datagrams and respond to requests for certain network-
layer information.

3.6.1 Datagram Format

Network-layer packet is referred to as a datagram. The datagram

plays a central role in the IPv4 datagram format is shown in Figure 4. The
fields in the IPv4 datagram are the following:

54
 Fig. 4 IPv4 datagram format

● Version number. These 4 bits mentioned the IP protocol version of

the datagram. The version number, the router can determine how to
interpret the remainder of the IP datagram. Different versions of IP use
different datagram formats.
● Header length. An IPv4 datagram can contain a variable number of
options, these 4 bits are needed to determine where in the IP datagram
the data actually begins. Most IP datagrams do not contain options, so
the typical IP datagram has a 20-byte header.
● Type of service. The type of service, TOS bits included in the IPv4
header to allow different types of IP datagrams to be distinguished
from each other. For example, it might be useful to distinguish real-
time datagrams from non-real-time traffic. The certain level of service
to be provided is a policy issue determined by the router’s
administrator.
● Datagram length. This is the total length of the IP datagram,
measured in units of bytes. After all this field is 16 bits long, the
theoretical maximum size of the IP datagram is 65,535 bytes.
However, datagrams are rarely larger than 1,500 bytes.
● Identifier, flags, fragmentation offset. The Identification field (16
bits) is populated with an ID number unique for the combination of
source & destination addresses and Protocol. the first, reserved bit of
the Flags field (3 bits) will be 0 and the second bit, Don’t Fragment,
The Fragment Offset field (13 bits) is used to indicate the begin
position of the data in the fragment in relation to the start of the data in
the original packet.
● Time-to-live. When the TTL field is included to ensure that datagrams
do not circulate forever (due to, for example, a long-lived routing
55
 loop) in the network. This field is reduced by one each time the
datagram is processed by a router. When the TTL field reaches 0, the
datagram must be dropped.
● Protocol. This field is used only when an IP datagram reaches its end
destination. The value of this field indicates the specific transport-layer
protocol. For example, a value of 6 indicates that the data portion is
passed to TCP, a value of 17 indicates that the data is passed to UDP.
● Header checksum. The header checksum supports a router in
detecting bit errors in a received IP datagram. The header checksum is
calculated by treating each 2 bytes in the header as a number and
summing these numbers using 1s complement arithmetic.
● Source and destination IP addresses. When a source creates a
datagram, it inserts its IP address into the source IP address field and
inserts the address of the ultimate destination into the destination IP
address field.
● Options. The options fields allow an IP header to be extended. Header
options were meant to be used rarely, hence the decision to save
overhead by not including the information in options fields in every
datagram header. However, the minimal existence of options does
complicate matters since datagram headers can be of variable length,
one cannot determine a priori where the data field will start.
● Data (payload). the data field of the IP datagram contains the
transport-layer segment (TCP or UDP) to be delivered to the
destination. However, the data field can carry other types of data, such
as ICMP messages.

IP Datagram Fragmentation

All link-layer protocols can carry network-layer packets of the

same size. different protocols can carry big datagrams, whereas other
protocols can carry only little packets. For example, Ethernet frames can
carry up to 1,500 bytes of data, whereas frames for some wide-area links
can carry no more than 576 bytes. The maximum amount of data that a
link-layer frame can transfer is called the maximum transmission unit
(MTU).

The solution is to fragment the data in the IP datagram into two or

more smaller IP datagrams, encapsulate each of these smaller IP
datagrams in a separate link-layer frame and send these frames over the
outgoing link. These smaller datagrams are referred to as a fragment.

Figure 5 an example. A datagram of 4,000 bytes data, 20 bytes of

IP header and 3,980 bytes of IP payload. when it comes at a router and
must be forwarded to a link with an MTU of 1,500 bytes of data. This
understood that the 3,980 data bytes in the original datagram must be
allocated to three separate fragments

56
 Fig. 5 IP fragmentation and reassembly

3.6.2 IPv4 Addressing

A host has only a single link into the network, when the IP in the
host wants to send a datagram, it does so over this link. The wall between
the host and the physical link is called an interface. Now consider a router
and its interfaces. The function of a router is to receive a datagram on one
link and forward the datagram on some other link, a router necessarily has
two or more links to which it is connected to each other. The partition
between the router and any one of its links is also called an interface. A
router has multiple interfaces, one for each of its links. Every host and
router is capable of sending and receiving IP datagrams, IP requires each
host and router interface to have its own IP address. An IP address is
technically connected with an interface, rather than with the host or router
containing that interface. Each IP address is 32 bits long (4 bytes), and
there are thus a total of 232 possible IP addresses. By approximating 210
by 103, it is easy to see that there are about 4 billion possible IP addresses.
These addresses are written in dotted-decimal notation, in which each byte
of the address is written in its decimal form and is separated by a period
(dot) from other bytes in the address. For example, consider the IP address
193.32.216.9. The 193 is the decimal point equivalent of the first 8 bits of
the address; the 32 is the decimal equivalent of the second 8 bits of the
address, and so on. Thus, the address 193.32.216.9 in binary notation is

11000001 00100000 11011000 00001001

Figure 6 provides an example of IP addressing and interfaces. Here

one router (with three interfaces) is used to interconnect seven hosts. The
three hosts in the upper-left portion of Figure 6, and the router interface to
which they are connected, all have an IP address of the form 223.1.1.xxx.
This means that they all have the same leftmost 24 bits in their IP address.

57
The four interfaces are interconnected to each other by a network that
contains no routers at all.

Fig 6. Interface addresses and subnets

3.6.3 Internet Control Message Protocol (ICMP)

ICMP is used by hosts and routers to transfer network layer

information to each other. The use of ICMP is for error reporting. For
example, when running a Telnet, FTP, or HTTP session, you may have
encountered an error message such as “Destination network unreachable.”
This message had its origins in ICMP. At some point, an IP router was
unable to find a path to the host specified in your Telnet, FTP, or HTTP
application. That router generated and sent a type-3 ICMP message to
your host indicating the error. ICMP often examines part of IP but
architecturally it lies just above IP, as ICMP messages are carried inside
IP datagrams.

That is, ICMP messages are carried as IP payload, just as TCP or

UDP segments are carried as IP payload. Similarly, when a host receives
an IP datagram with ICMP specified as the upper-layer protocol, it
demultiplexes the datagram’s contents to ICMP, just as it would
demultiplex the datagram's content to TCP or UDP. ICMP messages
consist of a type and a code field, and contain the header and the first 8
bytes of the IP datagram that caused the ICMP message to be generated in
the first place.

ICMP message types are shown in Figure 7 Note that ICMP

messages are used not only for signaling error conditions. The ping
program sends an ICMP type 8 code 0 message to the specified host. The
destination host, glimpse the echo request, sends back a type 0 code 0

58
ICMP echo reply. Most TCP/IP implementations support the ping server
directly in the operating system; that is, the server is not a process.

Fig. 7 ICMP message types

3.6.4 IPv6

Internet Protocol version 6 (IPv6) is the most recent version of

thehttps://en.wikipedia.org/wiki/Internet_ProtocolInternet Protocol (IP),
thehttps://en.wikipedia.org/wiki/Communication_protocolcommunications
protocol that provides an identification and location system for computers
on networks and routes traffic across the
https://en.wikipedia.org/wiki/InternetInternet.

IPv6 Datagram Format

The format of the IPv6 datagram is shown in Figure 6. The most

important updates introduced in IPv6 are evident in the datagram format:

● Expanded addressing capabilities.

IPv6 increases the size of the IP address from 32 to 128 bits. In
addition to unicast and multicast addresses, IPv6 has introduced a new
type of address, called an anycast address, which allows a datagram to
be delivered to any one of a group of hosts.

● A streamlined 40-byte header.

A number of IPv4 fields have been released or made optional. The
resulting 40 byte fixed length header allows for faster processing of
the IP datagram message.

● Flow labeling and priority.

labeling of packets belonging to particular flows for which the sender
requests special handling, such as a non default quality of service or

59
 real-time service. For example, audio and video transmission might
likely be treated as a flow. On the other hand, the more traditional
applications, such as file transfer and e-mail, might not be treated as
flows.

The following fields are defined in IPv6:

Fig 8. IPv6 header format

1. Version. This 4-bit field identifies the IP version number. IPv6 carries
a value of 6 in this field. Note that putting a 4 in this field does not
create a valid IPv4 datagram.
2. Traffic class. This 8-bit field is similar in spirit to the TOS field we
saw in IPv4.
3. Flow label. this 20-bit field is used to identify a flow of datagrams.
4. Payload length. This 16-bit value is treated as an unsigned integer
giving the number of bytes in the IPv6 datagram following the fixed-
length, 40-byte datagram header.
5. Next header. This field identifies the protocol to which the contents
(data field) of this datagram will be delivered (for example, to TCP or
UDP). The field uses the same values as the protocol field in the IPv4
header.
6. Hop limit. The contents of this field are decremented by one by each
router that forwards the datagram. If the hop limit count reaches zero,
the datagram is discarded.
7. Source and destination addresses. The various formats of the IPv6
128-bit address are described in RFC 4291
8. Data. This is the payload portion of the IPv6 datagram. When the
datagram reaches its destination, the payload will be removed from the
IP datagram and passed on to the protocol specified in the next header
field.

60
3.7 SUMMARY

1. learned that a router may need to process millions of flows of packets

between different source-destination pairs at the same time.
2. To permit a router to process such a large number of flows, network
designers have learned over the years that the router’s tasks should be
as simple as possible.
3. A datagram network layer rather than a virtual-circuit network layer,
using a streamlined and fixed-sized header, eliminating fragmentation
and providing the one and only best-effort service.

3.8 REFERENCE FOR FURTHER READING

1. TCP/IP Protocol Suite 4 edition, Behrouz Forouzan, McGraw-Hill

Science ( 2009)
2. Computer Networking: A Top-Down Approach 6th edition, James F.
Kurose, Keith W. Ross, Pearson (2012).

3.9 UNIT END EXERCISES

1. Explain the services provided by Network Service Models?

2. Explain the difference between circuit switching and packet
switching?
3. Write a short note on Virtual Circuit and Datagram Networks



61
 4
NETWORKING - IV

Unit Structure
4.1 Objective
4.2 Introduction
4.3 Routing Algorithms & Routing in the Internet.
4.3.1 Distance Vector Routing: RIP
4.3.2 Link State Routing: OSPF
4.3.3 Path Vector Routing: BGP
4.4 Summary
4.5 Reference for further reading
4.6 Unit End Exercises

4.1 OBJECTIVE

1. To help learners get an important distinction between the forwarding

and routing functions of the network layer.
2. To understand the packet forwarding, a router at its hardware
architecture and organization.
3. To learn the job of a routing algorithm is to determine good paths from
senders to receivers.
4. To study the theory of routing algorithms.
5. To understand the broadcast and multicast routing.

4.2 INTRODUCTION

Routing is the process of selecting a pathway for traffic in a

network or between or across multiple networks. In General, routing is
performed in many types of networks, including circuit-switched
networks, such as the public switched telephone network (PSTN), and
computer networks, such as the Internet.

In packet switching networks, routing is the higher-level decision

making that handles network packets from their source toward their
destination through intermediate network nodes by specific packet
forwarding mechanisms. Packet forwarding is the movement of network
packets from one network interface to another. In-between nodes are
typically network hardware devices such as routers, gateways, firewalls, or

62
switches. Common-purpose computers also forward packets and perform
routing, they have no specially optimized hardware for the task.

4.3 ROUTING ALGORITHMS & ROUTING IN THE

INTERNET

In Today internet world, an internet can be so large that one routing

protocol cannot manage the task of updating the routing tables of all
routers. For this reason, an internet is divided into autonomous systems.
An autonomous system (AS) is a group of networks and routers under the
authority of a single administration.

Intra-domain routing:
Routing inside an autonomous system is referred to as intra-
domain routing.

Inter-domain routing
Routing between two or more autonomous systems is referred to as
inter-domain routing.

Each autonomous system can choose one or more intradomain

routing protocols to handle routing inside the autonomous system.
Anyhow, only one interdomain routing protocol handles routing between
autonomous systems.

Fig. 9 Intra and inter domain routing

Intra-domain routing protocols:

1. Distance vector and
2. Link state.

Routing Information Protocol (RIP) is the execution of the distance

vector protocol. Open Shortest Path First (OSPF) is the implementation of
the link state protocol. Border Gateway Protocol (BGP) is the
implementation of the path vector protocol. RIP and OSPF are interior
routing protocols; BGP is an exterior routing protocol.

63
Fig. 10 Classification of routing protocol

Classify routing algorithms is according to whether they are global or

decentralized.

 A global routing algorithm calculates the least-cost path between a

source and destination using complete, global knowledge about the
network. This algorithm takes the connectivity between all nodes and
all link costs as inputs parameter. This algorithm require somehow
obtain this information before actually performing the calculation. The
calculation itself can be run at one site or replicated at multiple sites.
The feature is that a global algorithm has complete information about
connectivity and link costs. In exercise, algorithms with global state
information are frequently referred to as link-state algorithms, after all
the algorithm must be aware of the cost of each link in the network.

 In a decentralized routing algorithm, the calculation of the least-cost

path is carried out in an iterative, share out manner. No one node has
complete information about the costs of all network links. Rather than,
each node begins with only the knowledge of the costs of its own
directly attached links. Then, through an iterative process of
calculation and interchange of information with its neighboring nodes,
a node gradually calculates the least-cost path to a destination or set of
destinations.

 A second broad way to classify routing algorithms is according to

whether they are static or dynamic. Static routing algorithms, routes
change very slowly over time, often as a result of human mediation
dynamic routing algorithms change the routing paths as the network
traffic loads or change of topology. A dynamic algorithm can be run
either betimes or in direct response to topology or link cost changes.
Dynamic algorithms are more responsive to network changes; they are
also more susceptible to problems such as routing loops and oscillation
in routes.
64
4.3.1 Distance Vector Routing

In distance vector routing, each node shares its routing table with
its instant neighbors periodically and when there is a change. In distance
vector routing, the least cost route between any two nodes is the route with
minimum distance. In this protocol, as the name implies, each node
maintains a vector table of minimum distances to every node. The table
which at each node also guides the packets to the desired node by showing
the next stop in the route (next-hop routing).Example, nodes as the cities
in an area and the lines as the roads connecting them. A vector table can
show a tourist the minimum distance between cities. In Figure 11, we
show a system of five nodes with their corresponding tables.

Fig. 11 Distance vector routing tables

1. Initialization of DV table

This state is stable, each node knows how to reach any other node
and the cost of that node. At the beginning, anyhow this is not the case.
Each node can know only the distance between itself and its immediate
neighbors, those directly connected to it through proper link. Let us
assume that each node can send a message to the immediate neighbors and
find the distance between itself and these neighbors’ nodes. Figure 11
shows the initial vector tables for each node. The distance for any entry
that is not a neighbor is marked as an infinite loop.

65
 Fig. 12 Initialization of tables in distance vector routing

2. Sharing the vector table

The distance vector routing is the sharing of information (vector

table) between neighbors nodes. Even through node A does not know
about node E, node C does. So if node C shares its routing table with node
A, node A can also know how to reach node E. On the other side, node C
does not know how to reach node D, but node A does. If node A shares its
routing table with node C, node C also knows how to reach node D. In
other words, nodes A and C, as immediate neighbor’s node, can improve
their routing tables if they help each other. A node is not aware of a
neighbor's node table. The best solution for each node is to send its entire
table to the neighbor node and let the neighbor node decide what part to
use and what part to discard.The third column of a routing table (next
stop) is not useful for the neighbor node. When the neighbor node receives
a table, this column needs to be replaced with the sender's name. If any of
the rows can be used, the next node is the sender of the routing table. A
node therefore can send only the first two columns of its table to any
neighbor node.

3. Updating the vector table

When a node receives a two-column table from a neighbor node, it

needs to update its routing table. Updating of routing tables takes three
steps:
1. The receiving node or routing table needs to add the cost between
itself and the sending node to each value in the second column.
The logic is clear. If node C hold that its distance to a destination is
x mi, and the distance between A and C is y mi, then the distance
between A and that destination, via C, is x + y mi.

66
 2. The receiving node requires to add the name of the sending node to
each row as the third column if the receiving node uses
information from any row. The sending/dispatch node is the next
node in the route.
3. The receiving node requires comparing each row of its old table
with the corresponding row of the modified version of the received
table.
a. If the next-node entry is different from one, the receiving node
chooses the row with the smaller cost. If there is a bind, the old
one is kept.
b. If the next node appearance is the same, the receiving node
chooses the new row. For example, suppose node C has
previously advertised a route to node X with distance between.
c. For example that now there is no path between C and X; node
C now advertises this route with a distance of infinity. Node A
must not ignore this value even though its old entry is smaller
than the new one. The old route does not exist anymore. The
new route has a distance of infinity.
Figure 13 shows how node A updates its routing table after receiving the
partial table from node C.

Fig. 13 Updating in distance vector routing

Periodic Update A node table sends its routing table, normally

every 30 s, in a periodic update. The period depends on the protocol that is
using distance vector routing algorithm. Triggered Update A node sends
its two-column routing table to its neighbor’s node anytime there is a
change in its routing table. This is called a triggered update. The change
can result from the following.

67
 1. A node receives a table from a neighbor, resulting in changes in its
own table after updating.
2. A node detects some failure in the neighboring links which results
in a distance change to infinity.

RIP
The Routing Information Protocol (RIP) is an intradomain routing
protocol used inside an autonomous system. It is a very straightforward
protocol based on distance vector routing. RIP implements distance vector
routing directly with some considerations:
1. In an autonomous system, dealing with routers and networks or
links. The routers have routing tables, networks do not.
2. The destination in a routing table is a network, which means the
first column defines a network address.
3. The metric used by RIP is very simple, the distance is defined as
the number of links (networks) to reach the destination. For this
reason, the metric in RIP protocol is called a hop count.
4. Infinity is defined as 16, which means that any route in an
autonomous system using RIP cannot have more than 15 hops.
5. The next-node tables column defines the address of the router to
which the packet is to be sent to reach its destination.

4.3.2 Link State Routing

Link state routing has a different ideology from that of distance
vector routing. In link state routing, if each node in the domain has the
whole topology of the domain the list of nodes and links, they are
connected each other including the type, cost (metric), and condition of
the links (up or down) the node can use Dijkstra's algorithm to build a
routing table. Figure 14 shows the concept.

Fig. 14 Concept of link state routing

The above figure shows a simple domain with five nodes. Each
node uses the same topology to create a routing table, but the routing table

68
for each node is distinctive because the calculations are based on different
interpretations of the topology. This is comparable to a city map. While
each person may have the same map, each needs to take a different route
to reach her specified destination.

Link state routing is based, the global knowledge about the

topology is not clearly mentioned, and each node has partial knowledge
about the same: it knows the state (type, condition, and cost) of its links.
the whole topology can be compiled from the partial knowledge of each
node. Figure 15 shows the same domain as in Figure 14, indicating the
part of the knowledge belonging to each node.

Fig. 15 Link state knowledge

Node A knows that it is connected to node B with metric 5, to node

C with metric 2, and to node D with metric 3. Node C knows that it is
connected to node A with metric 2, to node B with metric 4, and to node E
with metric 4. Node D realizes that it is connected only to node A with
metric 3. And so on. There is an overlap in the knowledge, the overlap
assurance the creation of a common topology-a picture of the whole
domain for each node.

Building Routing Tables

In link state routing, four sets of steps are required to confirm that
each node has the routing table showing the least-cost node to every other
node.
1. Design of the states of the links by each node, called the link state
packet (LSP).
2. Circulation of LSPs to every other router, called flooding, in an
efficient and reliable way.
3. Origin of a shortest path tree for each node.
4. Computation of a routing table based on the shortest path tree.

OSPF
The Open Shortest Path First OSPF protocol is an intradomain
routing protocol based on the concept of link state routing. Its domain is
also called an autonomous system. OSPF protocol divides an autonomous
69
system into areas and subsections. An area is a group of networks, hosts,
and routers all contained within an autonomous system. An autonomous
system can be split into many different areas. All networks inside an area
must be connected to each other through a link. Routers inside an area
flood the area with the help of routing information. At the border of an
area, special routers called area border routers. the areas inside an
autonomous system is a special area called the backbone AS. All the areas
inside an autonomous system must be connected to the backbone. The area
identification of the backbone is zero. Figure 16 shows an autonomous
system and its areas.

Fig. 16 Areas in an autonomous system

Types of Links in OSPF:

A connection is called a link. Four types of links have been defined:
a. point-to-point,
b. transient,
c. stub, and
d. virtual

a. A point-to-point link connects two routers without any help of any

other host or router in between. An example of this type of link is
two routers connected to each other by a telephone line or a T line.
There is no need to assign a network address to this type of link.

Fig. 17 Point-to-point link

70
 b. A transient link is a network link with several routers attached with
each other. The data can come into the network through any of the
routers and leave through any router. For example, consider the
Ethernet in Figure 18. Router A has routers B, C, D, and E as
neighbor’s node. Router B has routers A, C, D, and E as neighbor’s
nodes.

Fig. 18 Transient link

c. A stub link is a network that is connected to only one router in the

network. The data packets come into the network through this
single router and leave the network through this same router.

Fig.19 Stub link

d. When the link between two routers is broken, the management

may create a virtual link between two routers, using a longer path
that probably goes through several routers.

4.3.3 Path Vector Routing

Distance vector routing is subject to uncertainty if there are more

than a few hops in the domain of operation. Link state routing needs a
large amount of resources to calculate routing tables. It also creates heavy
traffic due to flooding. There is a need for a third routing protocol which
we call path vector routing. Path vector routing demonstrates to be useful
for interdomain routing. The idea of path vector routing is similar to that
of distance vector routing.

BGP
Border Gateway Protocol is an interdomain routing protocol, It
uses path vector routing. For example, a large business that manages its
own network and has full control over it is an autonomous system. A local
ISP that provides services to local customers is called an autonomous
system. This divides autonomous systems into three categories: stub,
multihomed, and transit.

71
Stub AS. A stub AS has only one connection to another AS. The
interdomain data traffic in a stub AS can be either created or terminated in
the AS.

Multihomed AS. A multihomed AS has more than one connection to

other ASs, but it is still only a source or sink for data traffic.

Transit AS. A transit AS is a multihomed AS that also allows temporary

traffic. Good examples of transit ASs are national and international ISPs
(Internet backbones).

External and Internal BGP

BGP has two types of sessions: external BGP (E-BGP) and internal
BGP (I-BGP) sessions. The E-BGP session is used to interchange
information between two speaker nodes. It belongs to two different AS.
The I-BGP session is used to exchange routing information between two
routers inside an AS. Figure 21 shows the details

Fig. 21 Internal and external BGP sessions

BGP squint advertises routes to each other on the network. Two of

the more important attributes are AS-PATH and NEXT-HOP:

AS-PATH:
This attribute contains the ASs through which the advertisement
for the prefix has passed. When a prefix is passed into an AS, the AS adds
its ASN to the ASPATH attribute.

NEXT-HOP:
Providing the critical link between the inter-AS and intra-AS
routing protocols, the NEXT-HOP attribute has a subtle but important use.
The NEXT-HOP is the router interface that begins the AS-PATH.

BGP Route Selection

In AS BGP uses eBGP and iBGP to distribute routes to all the

routers. From this issue, a router may learn about more than one route to
any one prefix, in which case the router must select one of the
possibleroutes. The input into route selection process is the set of all
routes that have been acquire and accepted by the router. If there are more
than two routes to the same prefix, then BGP sequentially invokes the
following elimination rules until one route remains:

72
1. Routes are assigned a local priority value as one of their attributes. The
local priority of a route could have been set by the router. This is a
policy decision that is boost up to the AS’s network administrator. The
routes with the giant local preference values are selected.

2. The route with the shortest path (AS-PATH) is selected. If this rule
applies the only rule for route selection, then BGP would be using a
DV algorithm for path determination, where the distance metric uses
the number of AS hops preferably than the number of router hops.

3. From the remaining routes the route with the closest NEXT-HOP
router is selected. Here, closest means cost of the least-cost path, set
on by the intra-AS algorithm, is the smallest

4. If more than one route still leftover, the router uses BGP identifiers to
select the route.

4.4 SUMMARY

1. A router may need to process millions of flows of packets between

different source-destination pairs at the same time.
2. We learned how routing algorithms abstract the computer network to a
graph with nodes and links.
3. We learn that there are two broad approaches: a centralized approach,
in which each node obtains a complete map of the network and
independently applies a shortest-path routing algorithm; and a
decentralized approach, in which individual nodes have only a partial
picture of the entire network, yet the nodes work together to deliver
packets along the shortest routes.
4. We learned how centralized, decentralized, and hierarchical
approaches are embodied in the principal routing protocols in the
Internet: RIP, OSPF, and BGP.

4.5 REFERENCE FOR FURTHER READING

1. TCP/IP Protocol Suite 4 edition, BehrouzForouzan, McGraw-Hill

Science ( 2009)
2. Computer Networking: A Top-Down Approach 6th edition, James F.
Kurose,Keith W. Ross, Pearson (2012).

73
4.6 UNIT END EXERCISES

1. Why are different inter-AS and intra-AS protocols used in the

Internet?
2. Why are policy considerations as important for intra-AS protocols,
such as OSPF and RIP, as they are for an inter-AS routing protocol
like BGP?
3. How does BGP use the NEXT-HOP attribute? How does it use the
AS-PATH attribute?
4. What is an important difference between implementing the broadcast
abstraction via multiple unicasts, and a single network- (router-)
supported broadcast?



74
 Module - II

5
NETWORK VIRTUALIZATION
Unit Structure

5.0 Objectives
5.1 Introduction
5.2 An Overview
5.3 Need for Virtualization
5.4 The virtual Enterprise
5.5 Network Device Virtualization
5.6 Summary

5.0 OBJECTIVES:

This is an introductory tutorial, which covers the basics of

Virtualization and explains how to deal with its various components and
sub-components.

Introduction:
Virtualization is utilization of computer resources which is not in
used 100%.

Virtualization is technology that allows you to create multiple

simulated environments or dedicated resources from a single, physical
hardware system. Software called a hypervisor connects directly to that
hardware and allows you to split 1 system into separate, distinct, and
secure environments known as virtual machines (VMs).

Virtualization is technology that lets you create useful IT services

using resources that are traditionally bound to hardware. It allows you to
use a physical machine’s full capacity by distributing its capabilities
among many users or environments.

5.2 AN OVERVIEW:

Virtualization is a technology that helps users to install different

Operating Systems on a hardware. They are completely separated and
independent from each other.

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5.3 NEED FOR VIRTUALIZATION

Sometimes it’s necessary to make a virtual machine of any

operating system like (Linux or windows) into an existing and running
base operating system on a standalone hardware. Creation of VMs may
differ as per need and requirements.

The most important function of virtualization is the capability of

running multiple operating systems and applications on a single computer
or server. This means increased productivity achieved by fewer servers.

Following figure (i) illustrate the need of the virtualization.

Fig (1-1)

The virtual Enterprise

"A virtual enterprise is a temporary alliance of enterprises that
come together to share skills or core competencies and resources in order
to better respond to business opportunities, and whose cooperation is
supported by computer networks. "

Many business functions of your organization can be outsourced.

What traditionally were considered core functions are no longer a sacred
territory and are available for outsourcing. The difference in cost and
efficiency between an "on demand" or pay per usage outsourced service
and an on-premises and self-manned typical function could be significant
and hard to ignore.

This presents a problem requiring a solution for an Enterprise that

outsources most or all its business functions but retains governance for
planning, coordinating operations, budgeting, and making all key
decisions. In a Wikipedia definition, "a virtual organization is a firm that
outsources the majority of its functions." The Virtual Enterprise (VE) can
be successful, assuming it employs best of breed outsourced services in a
"virtual" Value Chain implementation consisting of company and partner
links.

76
 A VE operates over a virtual Value Chain, i.e., a chain whose
links are owned by a company and its partners, blurring the borders
between the Value Chain of the firm and the Value Network it is a part of.

The Governance is the business function that defines and identifies

the Virtual Enterprise, since most or all other functions of the Enterprise
(primary and secondary in Porter's definition) could be outsourced.

The VE is defined by a new operating model promoting

collaboration and B2B to take advantage of best of breed applications on
the market. This VE business model is increasingly achievable by the
adoption of business process outsourcing (BPO), application outsourcing –
Software as a Service (SaaS) – and, in general, by the fast adoption of
infrastructure virtualization technologies, Web Services, SOA, and
collaborative technologies of the Web2.0.

The "Virtual" Enterprise could be the darling of the entrepreneurial

world, specializing in management and governance skills while
outsourcing most of the Functions of the Enterprise today.

Transport Virtualization-VNs
The authors of Network Virtualization define the technical
requirements posed by the need to virtualize the network. Based on these
requirements, they propose an architectural framework comprised of the
functional areas necessary to successfully support concurrent virtual
networks (VNs) over a shared enterprise physical network.

When segmenting the network pervasively, all the scalability,

resiliency, and security functionality present in a non-segmented network
must be preserved and in many cases improved. As the number of groups
sharing a network increases, the network devices must handle a much
higher number of routes. Any technologies used to achieve virtualization
must therefore provide the necessary mechanisms to preserve resiliency,
enhance scalability, and improve security.

Central Services Access: Virtual Network Perimeter

The default state of a VN is to be totally isolated from other VNs.
In this respect, VNs could be seen as physically separate networks.
However, because VNs actually belong to a common physical network, it
is desirable for these VNs to share certain services such as Internet access,
management stations, DHCP services, Domain Name System (DNS)
services, or server farms. These services will usually be located outside of
the different VNs or in a VN of their own. So, it is necessary for these
VNs to have a gateway to connect to the "outside world." The outside
world is basically any network outside the VN such as the Internet or other
VNs. Because this is the perimeter of the VN, it is also desirable for this
perimeter to be protected by security devices such as firewalls and
intrusion detection systems (IDSs). Typically, the perimeter is deployed at
a common physical location for most VNs. Hence, this location is known

77
as the central services site, and the security devices here deployed can be
shared by many VNs.

The creation of VNs could be seen as the creation of security

zones, each of which has a unique and controlled entry/exit point at the
VN perimeter. Routing within the VNs should be configured so that traffic
is steered to the common services site as required. Figure illustrates a
typical perimeter deployment for multiple VNs accessing common
services. Because the services accessed through the VN perimeter are
protected by firewalls, we refer to these as "protected services.

fig-(1-2)

As shown in above Figure, each VN is head ended by a dedicated

firewall. This allows the creation of security policies specific to each VN
and independent from each other. To access the shared services, all
firewalls are connected to a "fusion" router. The fusion router can provide
the VNs with connectivity to the common services, the Internet, or even
inter-VN connectivity. The presence of this fusion router should raise two
main concerns:
 The potential for traffic leaking between VNs
 The risk of routes from one VN being announced to another VN

The presence of dedicated per-VN firewalls prevents the leaking of

traffic between VNs through the fusion router by only allowing
established connections (connections initiated from "inside" the firewall)
to return through the VN perimeter. It is key to configure the routing on
the fusion device so that routes from one VN are not advertised to another
through the fusion router. The details of the routing configuration at the
central site are discussed in Chapter 8, "Traffic Steering and Service
Centralization."

78
 Figure shows an additional firewall separating the fusion area from
the Internet. This firewall is optional. Whether to use it or not depends on
the need to keep common services or transit traffic in the fusion area
protected from the Internet.

A Virtualization Technologies primer: theory

 Devices—How is traffic separation maintained internally to a
device? What are the primitives used for Layer 2, Layer 3, or
Layer 4 traffic?
 Data path—How is traffic separation enforced across a network
path? What tools are available to maintain the separation across a
network?
 Control plane—Because data-path virtualization essentially builds
an overlay topology, what changes are needed for routing protocols
to function correctly?

5.4 NETWORK DEVICE VIRTUALIZATION

Network Virtualization (NV) refers to abstracting network

resources that were traditionally delivered in hardware to software. NV
can combine multiple physical networks to one virtual, software-based
network, or it can divide one physical network into separate, independent
virtual networks.

Network virtualization software allows network administrators to

move virtual machines across different domains without reconfiguring the
network. The software creates a network overlay that can run separate
virtual network layers on top of the same physical network fabric.

One of the characteristics of a VN is that it provides what are

essentially private communication paths between members of a group over
a shared infrastructure. This creates two requirements for the network
infrastructure:

 Traffic from one group is never mixed with another—For

sending and receiving traffic over shared links, tunnels (many
borrowed from existing virtual private network [VPN] solutions)
can guarantee data separation. Network devices need to enforce
group separation in their internal memory (for example, during
routing table lookups, access lists processing or NetFlow statistics
gathering).

 Each VN has a separate address space—This requirement is

derived from the fact that VNs offer the same characteristics as a
physical network. Address space and forwarding within it are two
of the most basic aspects of any network.

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Why Network Virtualization?
Network virtualization is rewriting the rules for the way services
are delivered, from the software-defined data center (SDDC), to the cloud,
to the edge. This approach moves networks from static, inflexible, and
inefficient to dynamic, agile, and optimized. Modern networks must keep
up with the demands for cloud-hosted, distributed apps, and the increasing
threats of cybercriminals while delivering the speed and agility you need
for faster time to market for your applications. With network
virtualization, you can forget about spending days or weeks provisioning
the infrastructure to support a new application. Apps can be deployed or
updated in minutes for rapid time to value.

How does network virtualization work?

Network virtualization decouples network services from the
underlying hardware and allows virtual provisioning of an entire network.
It makes it possible to programmatically create, provision, and manage
networks all in software, while continuing to leverage the underlying
physical network as the packet-forwarding backplane. Physical network
resources, such as switching, routing, firewalling, load balancing, virtual
private networks (VPNs), and more, are pooled, delivered in software, and
require only Internet Protocol (IP) packet forwarding from the underlying
physical network.

Network and security services in software are distributed to a virtual layer

(hypervisors, in the data center) and “attached” to individual workloads,
such as your virtual machines (VMs) or containers, in accordance with
networking and security policies defined for each connected application.
When a workload is moved to another host, network services and security
policies move with it. And when new workloads are created to scale an
application, necessary policies are dynamically applied to these new
workloads, providing greater policy consistency and network agility.

Benefits of network virtualization

Network virtualization helps organizations achieve major advances
in speed, agility, and security by automating and simplifying many of the
processes that go into running a data center network and managing
networking and security in the cloud. Here are some of the key benefits of
network virtualization:
 Reduce network provisioning time from weeks to minutes
 Achieve greater operational efficiency by automating manual
processes
 Place and move workloads independently of physical topology
 Improve network security within the data center

Example:
One example of network virtualization is virtual LAN (VLAN). A
VLAN is a subsection of a local area network (LAN) created with
software that combines network devices into one group, regardless of

80
physical location. VLANs can improve the speed and performance of busy
networks and simplify changes or additions to the network.

Another example is network overlays. There are various overlay

technologies. One industry-standard technology is called virtual extensible
local area network (VXLAN). VXLAN provides a framework for
overlaying virtualized layer 2 networks over layer 3 networks, defining
both an encapsulation mechanism and a control plane. Another is generic
network virtualization encapsulation (GENEVE), which takes the same
concepts but makes them more extensible by being flexible to multiple
control plane mechanisms.

VMware NSX Data Center – Network Virtualization Platform

VMware NSX Data Center is a network virtualization platform that
delivers networking and security components like firewalling, switching,
and routing that are defined and consumed in software. NSX takes an
architectural approach built on scale-out network virtualization that
delivers consistent, pervasive connectivity and security for apps and data
wherever they reside, independent of underlying physical infrastructure.
The first problem to solve is how to virtualize the forwarding plane in a
way that meets the requirements for address and traffic flow separation.
Depending on the type of device, the virtual separation can go by the
following names:
 Virtual LAN (VLAN)
 Virtual routing and forwarding (VRF)
 Virtual forwarding instance (VFI)
 Virtual firewall context

Layer 2: VLANs
VLANs are a good example of a piece of the virtualization puzzle that
has been around for quite some time. A VLAN is a logical grouping of
ports on a switch that form a single broadcast domain. Ports in a
VLAN can communicate only with other ports in the same VLAN.
How a given switch does this is implementation dependent, but a
common solution is for the switch to tag each frame with a VLAN
number as it arrives on a port. When a frame is sent to other ports, the
output hardware copies the packet only if it is configured with the
VLAN number carried in the frame.

The summary effect of the VLANs is to partition the switch into

logical Layer 2 domains. Each domain has its own address space and
packets from one domain are kept separate from those of another.

Layer 3: VRF Instances

VRFs are to Layer 3 as VLANs are to Layer 2 and delimit the domain of
an IP network within a router. The Cisco website has a more formal
definition:

81
VRFA VPN Routing/Forwarding instance. A VRF consists of an IP
routing table, a derived forwarding table, a set of interfaces that use the
forwarding table, and a set of rules and routing protocols that determine
what goes into the forwarding table.

Unlike the VLAN scenario, where an extra column in the MAC table is
adequate, a VRF partitions a router by creating multiple routing tables and
multiple forwarding instances. Dedicated interfaces are bound to each
VRF.

Following diagram shows a simple logical representation of a router with

two VRFs: RED and GREEN. The RED table can forward packets
between interfaces E1/0, E1/2, and S2/0.102. The GREEN table, on the
other hand, forwards between interfaces E4/2, S2/0.103, and S2/1.103. An
interface cannot be in multiple VRFs at the same time.

Fig (2-1). Multiple VRFs on a Router

FIBs and RIBs

Before looking at the routing information for a VRF, we need to
introduce the routing table's two main data structures, which are used to
find the egress interface for a given packet: theForwarding Information
Base (FIB) and the Routing Information Base (RIB). Long gone are the
days when a router maintained a single routing table on which it did linear,
longest-prefix searches against destination IP addresses.

The FIB is a database of information used to forward packets.

When a packet is received on a routed interface, the router looks up the
destination address in the FIB to find the next hop for the packet.

The FIB structure is particularly efficient for resolving longest-

prefix matches, and Cisco IOS resolves all route redirections so that a
single lookup can yield the entry for the next hop of a packet. Cisco
literature often mentions an adjacency concept when presenting the FIB.
An adjacency is any node in the network that is reachable with a single
Layer 2 hop. It so happens that Cisco IOS also maintains a data structure
82
of adjacencies, which contains, among other things, interface and MAC
layer rewrite information for all possible next hops. FIB entries point to
the adjacency table, and in the remainder of this chapter, we group the two
together and refer simply to the FIB. Hardware-based forwarding paths
use the FIB concept, as does Cisco Express Forwarding (CEF).

Because it contains both Layer 2 and Layer 3 information, the FIB

can be updated by several sources, such as routing protocol and Address
Resolution Protocol (ARP) updates.

The RIB is the memory structure that contains classic routing data. The
RIB can contain recursive routes. If a packet destination is not in the FIB,
the router "punts" the packet to a slow processing path and resolves the
destination next hop using the RIB.

Traffic processing happens according to the same rules as on a

device with no VRFs:
1. Traffic enters the router.
2. The ingress policy is applied.
3. Routing and forwarding lookup occurs.
4. The egress policy is applied.
5. Traffic is forwarded.

Obviously, the ingress and egress policies can include QoS

statements that prioritize traffic to or from a particular interface or address,
but the fact that a packet belongs to a particular VRF has no impact on
those policies. It simply alters what happens in Step 3 of the preceding list.
It is far more common to want to bind an interface or packet flow to a
particular VRF based on policy criteria. For example, all interfaces from a
certain user domain are bound to a single company VRF, or packets with a
10.0.0.0/8 source address are bound to a guest VRF. We look at this in
great detail in some of the design chapters.

Virtual and Logical Routers

A VRF is not the same thing as a completely virtualized device,
even if they are sometimes confused as such (it is true that some
marketing literature encourages such confusion). They simply allow
routers to support multiple address spaces. This is some distance from a
fully virtualized device, where resources can be more or less arbitrarily
allocated to tasks.

Virtualized devices do exist, however, and, to cut through the fog

of confusion, it is helpful to have a taxonomy of terms to start with:
 A logical router (LR) uses hardware partitioning to create multiple
routing entities on a single device. An LR can run across different
processors on different cards of a router. All the underlying
hardware and software resources are dedicated to an LR. This
includes network processors, interfaces, and routing and
83
 forwarding tables. LRs provide excellent fault isolation but do
require abundant hardware to implement.

 A virtual router (VR) uses software emulation to create multiple

routing entities. The underlying hardware is shared between
different router processes (note that we mean an entire instance of
something like the nonkernel parts of IOS, not a single router
process). In a well-implemented virtual router, users can see and
change only the configuration and statistics for "their" router.

Fig (2-2) Logical and Virtual Routers

From the preceding list and mk:@MSITStore:F:\ \Idol\M.Sc-
CS%20IDOL%20COURCE%20WRITING%20CONTENT%20FOR%20ANC\Network%
20Virtualization%20by%20Victor%20Moreno,%20Kumar%20Reddy%20(z-
lib.org).chm::/1587052482/ch04lev1sec1.html - ch04fig02, which gives a
pictorial idea of the difference between VRs and LRs, you can see that
only the LR is completely virtualized. Because of the cost involved of
having all that extra hardware and device management, LRs tend to be
high-end systems. A VR is a software-based virtualization solution, where
all the tasks share the same hardware resources.

Layer 2 Again: VFIs

VFI is a service-specific partition on a switch that associates
attachment circuits in the form of VLANs with virtual switched interfaces
(VSIs).

84
 If that did not make much sense, it is useful to have some
background on the service itself, namely Virtual Private LAN Services
(VPLS), to understand VFIs.

VPLS is a Layer 2 LAN service offered by service providers (SPs)

to connect Ethernet devices over a WAN. The customer devices (call them
customer edges [CEs] for now, are all Ethernet switches. However, the SP
uses a Layer 3 network running Multiprotocol Label Switching (MPLS) to
provide this service. The device on the edge of the SP network is called a
provider edge (PE). Its role is to map Ethernet traffic from the customer
LAN to MPLS tunnels that connect to all the other PEs that are part of the
same service instance. The PEs are connected with a full mesh of tunnels
and behave as a logical switch, called a VSI. Another way to think about
this is to see the VPLS service as a collection of Ethernet ports connected
across a WAN. A VSI is a set of ports that forms a single broadcast
domain.

In many ways, a VSI behaves just as you would expect a regular

switch to. When a PE receives an Ethernet frame from a customer device,
it first learns the source address, as would any switch, before looking at
the destination MAC address and forwarding the frame. If the port
mapping for the destination MAC address is unknown, or is a broadcast,
the frame is sent to all PEs that are part of the VSI. The PEs use split
horizon to avoid creating loops, which in turn means that no spanning tree
is needed across the SP network.

Obviously, the previous explanation hides a fair amount of detail,

but it should be enough to give a high-level view of what is going on.

Once again, there is a need to define and manage groups of

isolated ports and tunnels on a switch. The VLAN construct is too limited,
and a VRF is strictly a Layer 3 affair, so it is necessary to come up with a
new virtual device structure for VPLS, called a VFI.

Fig (2-3).VPLS Topology

85
Virtual Firewall Contexts
Device virtualization is not limited to switches and routers. As a
final example, consider a firewall device. For essentially economic
reasons, you might want to share a single firewall between multiple
different customers or network segments. Each logical firewall needs to
have a complete set of policies, dedicated interfaces for incoming and
outgoing traffic, and users authorized to manage the firewall.

Many vendors provide this capability today and undoubtedly have

their own, well-chosen name for it, but on Cisco firewalls the term context
is used to refer to a virtual firewall. Unlike VRFs, VFIs, or VLANs, a
context is an emulation of a device.

Firewall contexts are a little unusual in the way they assign a

packet to a context. All the partitions we have seen up to now have static
assignment of interfaces (you can assign IP packets to a VRF dynamically.
We cover that later). A firewall module looks at an incoming packet's
destination IP address or Ethernet VLAN tag to decide which context a
packet belongs to. All the firewall needs for one of the two fields to be
unique. So, either each context has a unique IP address space on its
interfaces or the address space is shared, but each context is in a different
VLAN.

Fig (2-4).VRF on Switch Connected to Firewall Contexts Across

VLANs

Network Device Virtualization Summary

True device virtualization allows resources to be allocated to tasks,
or applications. We looked at four different primitives that virtualize the
forwarding paths on switches or routers: VLAN and VFI for Layer 2, VRF
for Layer 3, and contexts for firewalls. Each of these functions slightly
differently. VRFs have the most extensive tie-ins with other features,
which we use extensively in the design sections. Before covering data-
path virtualization, one word about data center designs. We are focusing
on network devices exclusively in this book and do not address the details
of server and storage virtualization, which are two important topics in their
own right.
Data-Path Virtualization
86
 It refers to the virtualization of the interconnection between
devices. This could be a single-hop or multiple-hop interconnection. For
example, an Ethernet link between two switches provides a single-hop
interconnection that can be virtualized by means of 802.1q VLAN tags;
for Frame Relay or ATM transports, separate virtual circuits provide data-
path virtualization. An example of a multiple-hop interconnection would
be that provided by an IP cloud between two devices. This interconnection
can be virtualized through the use of multiple tunnels (generic routing
encapsulation [GRE] for example) between the two devices.

Layer 2: 802.1q Trunking

You probably do not think of 802.1q as a data-path virtualization
protocol. But, the 802.1q protocol, which inserts a VLAN tag on Ethernet
links, has the vital attribute of guaranteeing address space separation on
network interfaces.

Obviously, this is a Layer 2 solution, and each hop must be

configured separately to allow 802.1q connectivity across a network.
Because a VLAN is synonymous with a broadcast domain, end-to-end
VLANs are generally avoided.

Generic Routing Encapsulation

GRE provides a method of encapsulating arbitrary packets of one
protocol type in packets of another type (the RFC uses the expression X
over Y, which is an accurate portrayal of the problem being solved). The
data from the top layer is referred to as the payload. The bottom layer is
called the delivery protocol. GRE allows private network data to be
transported across shared, possibly public infrastructure, usually using
point-to-point tunnels.

Although GRE is a generic X over Y solution, it is mostly used to

transport IP over IP (a lightly modified version was used in the Microsoft
Point-to-Point Tunneling Protocol [PPTP] and, recently, we are seeing
GRE used to transport MPLS). GRE is also used to transport legacy
protocols, such as Internetwork Packet Exchange (IPX) and AppleTalk,
over an IP network and Layer 2 frames.

GRE is purely an encapsulation mechanism. How packets arrive at

tunnel endpoints is left entirely up to the user. There is no control
protocol, no session state to maintain, no accounting records, and so forth;
and this conciseness and simplicity allows GRE to be easily implemented
in hardware on high-end systems. The concomitant disadvantage is that
GRE endpoints have no knowledge of what is happening at the other end
of the tunnel, or even whether it is reachable.

The time-honored mechanism for detecting tunnel reachability

problems is to run a dynamic routing protocol across the tunnel. Routing
Protocol (RP) keepalives are dropped if the tunnel is down, and the RP
itself will declare the neighbor as unreachable and attempt to route around

87
it. You can lose a lot of data waiting for an RP to detect a problem in this
way and reconverge. Cisco added a keepalive option to its GRE
implementation. This option sends a packet through the tunnel at a
configurable period. After a certain number of missed keepalives (the
number is configurable), the router declares the tunnel interface as down.
A routing protocol would detect the interface down event and react
accordingly.

GRE's lack of control protocol also means that there is essentially

no cost to maintaining a quiescent tunnel active. The peers exchange no
state information and must simply encapsulate packets as they arrive.
Furthermore, like all the data-path virtualization mechanisms we discuss,
the core network is oblivious of the number of tunnels traversing it. All the
work is done on the edge.

We do not want to suggest that GRE is the VPN equivalent of a universal

solvent. There is a cost to processing GR Eencapsulation / decapsulation,
route lookup, and so forth but it's in the data path.

GRE IOS Configuration

On Cisco devices, GRE endpoints are regular interfaces. This
seemingly innocuous statement is replete with meaning, because anything
in Cisco IOS that needs to see an interface (routing protocols, access lists,
and many more) will work automatically on a GRE tunnel.

Fig (2-5).GRE Topology

The tunnel source and tunnel destination addresses are part of the
transport network address space. They need to match on both endpoints so
that a source address on one router is the destination address on the remote
device. The router must also have a path in its routing table to the tunnel
destination address. The next hop to the tunnel destination must point to a
real interface and not the tunnel interface.

In this case, the router has a tunnel interface with tunnel

destination of 192.168.2.1 on the public network. The 40.0.0.0/24 network
used for the tunnel IP's address, however, is part of the private address
space used on Sites 1 and 2.

IPsec
88
 IPsec provides a comprehensive suite of security services for IP
networks. IPsec was originally conceived to provide secure transport over
IP networks. The security services include strong authentication
(Authentication Header [AH]) and Encryption (Header [EH]) protocols
and ciphers and key-exchange mechanisms. IPsec provides a way for
peers to interoperate by negotiating capabilities and keys and security
algorithms.

IPsec peers maintain a database of security associations. A security

association (SA) is a contract between peers, which defines the following:
 The specific encryption and authentication algorithms used, such
as Triple DES (Triple Data Encryption Standard)
 The IPsec protocol service (Encapsulating Security Payload [ESP]
or AH)
 Key material needed to communicate with the peer

The SA is negotiated when an IPsec session is initiated. Each IPsec

header contains a unique reference to the SA for this packet in a Security
Parameter Index (SPI) field, which is 32-bit numeric reference to the SA
needed to process the packet. Peers maintain a list of SAs for inbound and
outbound processing. The value of the SPI is shared between peers. It is
one of the things exchanged during IPsec session negotiation.

At the protocol level, there are two IPsec headers:

 AH Offers nonrepudiatable authentication between two parties.
The authentication service also provides for message integrity and
certain instances of (identity) spoofing.
 ESP Offers encrypted communication between two parties. The
encryption service allows message confidentiality, integrity,
nonrepudiation, and protection against spoofing and replay attacks.

It is possible to use authentication and encryption services separately

or together. If used in combination, the AH header precedes the ESP
header.

In a normal routing scenario, when a router needs to forward a

packet, it finds the outgoing interface by looking for a matching IP address
prefix in the routing table. The actual interface used for forwarding
corresponds to the shortest path to the IP destination, as defined by the
routing policy. Other administrative policies, such as QoS and security,
may affect the choice of interface. This collection of criteria used for
forwarding decisions is more generally referred to as a Forward
Equivalency Class (FEC). The classification of a packet to FEC is done on
each router along the IP path and happens independently of the other
routers in the network.

MPLS decouples packet forwarding from the information in the IP

header. An MPLS router forwards packets based on fixed-length labels
89
instead of matching on a variable-length IP address prefix. The label is a
sort of shortcut for an FEC classification that has already happened.
Where the label comes from is discussed later in this section, but for now,
it is enough to say that the labels are calculated based on the topology
information in the IP routing table. RFC 3031 puts it like this:

In MPLS, the assignment of a particular packet to a particular FEC

is done just once, as the packet enters the network. The FEC to which the
packet is assigned is encoded as a short fixed length value known as a
"label." When a packet is forwarded to its next hop, the label is sent along
with it; that is, the packets are "labeled" before they are forwarded.

In the MPLS forwarding paradigm, once a packet is assigned to a FEC, no

further header analysis is done by subsequent routers; all forwarding is
driven by the labels.

Before looking at this in more detail, we need to introduce some

definitions:
 Label switching router (LSR) A router that switches based on
labels. An LSR swaps labels. Unlike a traditional router, an LSR
does not have to calculate where to forward a packet based on the
IP packet header (which is a simplified way of saying it does not
do FEC classification when it receives a packet). An LSR uses the
incoming label to find the outgoing interface (and label). LSRs are
also called provider (P) routers.
 Edge LSR A router that is on the edge of an MPLS network. The
edge LSR adds and removes labels from packets. This process is
more formally called imposition and disposition (and also pushing
and popping, because labels are said to go on a stack). Edge LSRs
are often referred to as provider edge (PE) routers.

Customer edge (CE) An IP router that connects to the PE device.

The CE performs IP forwarding. The PE and CE form routing protocol
adjacencies.

90
 Fig (2-6).MPLS Forwarding

As a packet flows across the network shown in

mk:@MSITStore:F:\ \Idol\M.Sc-
CS%20IDOL%20COURCE%20WRITING%20CONTENT%20FOR%20ANC\Network%
20Virtualization%20by%20Victor%20Moreno,%20Kumar%20Reddy%20(z-
lib.org).chm::/1587052482/ch04lev1sec2.html - ch04fig10, it is processed by
each hop as follows:

1 At the edge of the network, as shown in mk:@MSITStore:F:\ \Idol\M.Sc-

. CS%20IDOL%20COURCE%20WRITING%20CONTENT%20FOR%20ANC\Network
%20Virtualization%20by%20Victor%20Moreno,%20Kumar%20Reddy%20(z-
lib.org).chm::/1587052482/ch04lev1sec2.html - ch04fig10, edge LSR A
classifies a packet to its FEC and assigns (or imposes) label 17 to the
packet. A label is of local significance on that interface just like an ATM
VPI/VCI or a Frame Relay DLCI.
2 In the core, LSRs, such as LSR C and LSR B, swap label values. LSR C
. removes the old label, 17 in the example shown in
mk:@MSITStore:F:\ \Idol\M.Sc-
CS%20IDOL%20COURCE%20WRITING%20CONTENT%20FOR%20ANC\Network
%20Virtualization%20by%20Victor%20Moreno,%20Kumar%20Reddy%20(z-
lib.org).chm::/1587052482/ch04lev1sec2.html - ch04fig10, and imposes the
new one, 22. The values of the ingress label and interface are used to
find the values of the egress label and interface.
Note
Not all MPLS forwarding modes use incoming interface. Frame mode,
used in certain L2VPN services, just uses the incoming label as the same
label value is advertised to all peers
3 LSR B, as the second-last hop in the MPLS network, removes the
. outermost label from the label stack, which is called penultimate hop
popping (PHP). So, packets arrive at edge LSR D without any label, and
standard IP routing is used to forward the packet. The process of

91
 removing a label is also called disposition. PHP avoids recursive
lookups on edge LSR D.
4 After the label is removed, the packet is forwarded using standard IP
. routing.

Now the difference with standard IP forwarding should be clearer.

FEC classification is done when a packet enters the MPLS network, not at
every hop. An LSR needs to look only at the packet's label to know which
outgoing interface to use. There can be different labels on an LSR for the
same IP destination. Saying the same thing in a different way, there can be
multiple LSPs for the same destination.

A key point to understand is that the control plane is identical in

both the IP and MPLS cases. LSRs use IP routing protocols to build
routing tables, just as routers do. An LSR then goes the extra step of
assigning labels for each destination in the routing table and advertising
the label/FEC mapping to adjacent LSRs. ATM switches can also be
LSRs. They run IP routing protocols, just as a router LSR does, but label
switch cells rather than packets.

What is missing from this description is how label information is

propagated around the network. How does LSR A in
mk:@MSITStore:F:\ \Idol\M.Sc-
CS%20IDOL%20COURCE%20WRITING%20CONTENT%20FOR%20ANC\Network%
20Virtualization%20by%20Victor%20Moreno,%20Kumar%20Reddy%20(z-
lib.org).chm::/1587052482/ch04lev1sec2.html - ch04fig10 know what label to
use? MPLS networks use a variety of signaling protocols to distribute
labels:

 LDP Used in all MPLS networks

 iBGP Used for L3 VPN service
 RSVP Used for Traffic Engineering
 Directed LDP Used for L2VPN service, such as VPLS

Label Distribution Protocol (LDP), which runs over tcp/646, is used in

all MPLS networks to distribute labels for all prefixes in the nodes routing
table. Referring again to mk:@MSITStore:F:\ \Idol\M.Sc-
CS%20IDOL%20COURCE%20WRITING%20CONTENT%20FOR%20ANC\Network%
20Virtualization%20by%20Victor%20Moreno,%20Kumar%20Reddy%20(z-
lib.org).chm::/1587052482/ch04lev1sec2.html - ch04fig10, LSR D and LSR B
would bring up a LDP session (LSR B would have another session with
LSR C and so forth). LSR D is connected to the customer 192.168.2.0/24
network and advertises this prefix to all its routing peers. LSR D also
sends a label to LSR B for the 192.168.2.0 network. When LSR B's
routing protocol converges and it sees 192.168.2.0 as reachable, it sends
label 22 to LSR C. This process continues until LSR A receives a label
from LSR C.

92
 The complete end-to-end set of labels from LSR A to LSR D form an
LSP. An LSP is unidirectional. There is another LSP, identified by a
different set of labels, for return traffic from LSR D to LSR A.

Understand that two operations must complete for the LSP from
LSR A to 192.168.2.0 to be functional:
 The backbone routing protocol must converge so that LSR A has a
route to 192.168.2.0.
 LDP must converge so that labels are propagated across the
network.

Fig (2-6) does not show a numeric value for the label between LSR
B and LSR D. In fact, as already discussed, the packet on this link has no
label at all, because of PHP. Nevertheless, LSR D does still advertise a
special value in LDP, called an implicit null (which has a reserved value
of 3), so that LSR B performs PHP.

Fig-(2-7) Network Virtualization

Above diagram simply shows the concept of Data path
virtualization which comes under network virtualization technology. It
provides virtualize communication path between network access points as
shown in the above diagram.

Data-Path Virtualization Summary

We presented several different protocols that can be used for data-
path virtualization. Two of them are suitable for Layer 2 traffic only:
802.1q, which is configured on each hop, and L2TPv3 which is configured
end to end. IPsec is suitable for IP transport. Finally, GRE and MPLS
LSPs can be used for either Layer 2 or Layer 3. GRE is another IP tunnel

93
protocol, configured only on endpoints. MPLS creates a new forwarding
path and is configured on all hops in a network.

Control-Plane Virtualization
Refers to all the protocols, databases, and tables necessary to make
forwarding decisions and maintain a functional network topology free of
loops or unintended blackholes. This plane could be said to draw a clear
picture of the topology for the network device. A virtualized device must
posses a unique picture of each VN it is to handle, hence the requirement
to virtualize the control-plane components.

Data-path virtualization essentially creates multiple separate

logical networks over a single, shared physical topology. To move packets
across these VNs, you need to need a routing protocol.

The most familiar virtualized control plane is probably Per VLAN

Spanning Tree (PVST), which has a separate spanning-tree instance for
each VLAN running on a switch. Even through PVST has been around
longer than the term virtualization, it illustrates the central point we are
making here very crisply. Different logical networks have different
topologies and, therefore, different optimal paths. Switches have to run
different spanning-tree calculations for each such network.

The remainder of this section deals with extensions to routing

protocols to allow them to run with multiple routing instances. However,
we will return to the topic of control-plane virtualization, because many
different router and switch functions, such as NetFlow, DHCP, RADIUS,
and so on, need to receive the same treatment and become VRF aware.

94
 Fig-(2-7)

VRF-Aware Routing
Cisco's major interior gateway protocol (IGP) routing protocol
implementations are VRF aware. This means that they understand that
certain routes may be placed only in certain routing tables. The routing
protocols manage this by peering within a constrained topology, where a
routing protocol instance in a VRF peers with other instances in the same
VN. No special information is added to the route advertisements to
identify VRF names, so routing instances must communicate over private
links.

With some protocols (for example, BGP), a single routing instance

can manage multiple VRF tables; with others (for example, OSPF), a
different routing process runs for every VRF. Remember that in both
cases, every VRF requires a route optimization calculation, so increasing
the number of VRFs does have a computational impact on a network
device.

Multi-Topology Routing
Multi-Topology Routing (MTR) is a recent innovation at Cisco. As
the name suggests, it creates multiple routing topologies across a shared,
common infrastructure. However, MTR does not try to be yet another
VPN solution. Instead, it creates paths through a network that you can map
to different applications or classes of applications, with the understanding
that, by separating traffic in this way, you can provide better performance
characteristics to certain critical applications.

MTR bases its operation on the creation of separate RIBs and FIBs
for each topology. The separate RIBs and FIBs are created within a
common address space. Thus, MTR creates smaller topologies that are a
subset of the full topology (also known as the base topology). The main
difference between MTR and a VPN technology is that, with MTR, a
single address space is tailored into many topologies that could overlap;
whereas VPNs create totally separate and independent address spaces.

Thus, MTR must carry out two distinct functions:

 At the control planeColor the routing updates, so that the different
topology RIBs are populated accordingly. Based on these RIBs,
the corresponding FIBs are to be written.
 At the forwarding plane Identify the topology to which each packet
belongs and use the correct FIB to forward the packet.

At each hop, there will be a set of prefixes and routes in the RIB for
each topology. The contents of these RIBs are dynamically updated by
routing protocol colored updates. Based on this RIB information, a
separate FIB is built for each topology.

95
 To forward traffic over different topologies, the router looks for a
code point in each packet and chooses an FIB based on this code point. A
first implementation of MTR uses differentiated services (DiffServ) code
point (DSCP) as such a code point, but other code points could be used by
future implementations. The DSCP value is used as a pointer to the correct
forwarding table, and the packet's destination address is used to make a
forwarding decision based on the information in the topology's FIB. MTR
uses the terminology of color to refer to separate topologies. So, a RED
value in a packet's DSCP field is recognized by the router, which will
forward the packet using the RED forwarding table (FIB).

MTR must run contiguously across a network, and the color

mappings must be consistent (that is, you cannot use DSCP X as Green on
one hop but as Red on the next). MTR does not allow you to double dip: If
the destination route is not in the routing table of the color a packet is
using, the packet can either be dropped or forwarded over the base
topology there are no lookups in "backup topologies" other than the base
topology (which is equivalent to regular routing).

MTR does not change how routing works; it just runs across
multiple topologies (using a single process with colored updates).

Control-Plane Virtualization Summary

Control-plane virtualization refers to adaptations made to routing
protocols to be able to operate on virtualized devices. We concentrated on
per-VRF routing because that is the main tool we use for design. However,
VRs and LRs also run separate routing instances in a similar manner to the
one shown here. In all cases, there are no changes to the protocol "on the
wire." MTR is an interesting new development that can also be
categorized in the virtualized control-plane bucket.

Routing Protocols
Routing Protocols are the set of defined rules used by the routers
to communicate between source & destination. They do not move the
information to the source to a destination, but only update the routing table
that contains the information.

Network Router protocols help you to specify way routers

communicate with each other. It allows the network to select routes
between any two nodes on a computer network.

Types of Routing Protocols

There are mainly two types of Network Routing Protocols
 Static
 Dynamic

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 Routing Protocols

Static Routing Protocols

Static routing protocols are used when an administrator manually
assigns the path from source to the destination network. It offers more
security to the network.

Advantages
 No overhead on router CPU.
 No unused bandwidth between links.
 Only the administrator is able to add routes

Disadvantages
 The administrator must know how each router is connected.
 Not an ideal option for large networks as it is time intensive.
 Whenever link fails all the network goes down which is not
feasible in small networks.

Dynamic Routing Protocols

Dynamic routing protocols are another important type of routing
protocol. It helps routers to add information to their routing tables from
connected routers automatically. These types of protocols also send out
topology updates whenever the network changes’ topological structure.

Advantage:
 Easier to configure even on larger networks.
 It will be dynamically able to choose a different route in case if a
link goes down.
 It helps you to do load balancing between multiple links.

Disadvantage:
97
  Updates are shared between routers, so it consumes bandwidth.
 Routing protocols put an additional load on router CPU or RAM.

Distance Vector Routing Protocol (DVR)

Distance Vector Protocols advertise their routing table to every
directly connected neighbor at specific time intervals using lots of
bandwidths and slow converge.

In the Distance Vector routing protocol, when a route becomes

unavailable, all routing tables need to be updated with new information.

Advantages:
 Updates of the network are exchanged periodically, and it is
always broadcast.
 This protocol always trusts route on routing information received
from neighbor routers.

Disadvantages:
 As the routing information are exchanged periodically,
unnecessary traffic is generated, which consumes available
bandwidth.

Internet Routing Protocols:

The following are types of protocols which help data packets find
their way across the Internet:

Routing Information Protocol (RIP)

RIP is used in both LAN and WAN Networks. It also runs on the
Application layer of the OSI model. The full form of RIP is the Routing
Information Protocol. Two versions of RIP are
1. RIPv1
2. RIPv2

The original version or RIPv1 helps you determine network paths

based on the IP destination and the hop count journey. RIPv1 also
interacts with the network by broadcasting its IP table to all routers
connected with the network.

RIPv2 is a little more sophisticated as it sends its routing table on to a

multicast address.

Interior Gateway Protocol (IGP)

IGRP is a subtype of the distance-vector interior gateway protocol
developed by CISCO. It is introduced to overcome RIP limitations. The
metrics used are load, bandwidth, delay, MTU, and reliability. It is widely
used by routers to exchange routing data within an autonomous system.
98
 This type of routing protocol is the best for larger network size as
itbroadcasts after every 90 seconds, and it has a maximum hop count of
255. It helps you to sustain larger networks compared to RIP. IGRP is also
widely used as it is resistant to routing loop because it updates itself
automatically when route changes occur within the specific network. It is
also given an option to load balance traffic across equal or unequal metric
cost paths.

Link State Routing Protocol

Link State Protocols take a unique approach to search the best
routing path. In this protocol, the route is calculated based on the speed of
the path to the destination and the cost of resources.

Routing protocol tables:

Link state routing protocol maintains below given three tables:
 Neighbor table: This table contains information about the
neighbors of the router only. For example, adjacency has been
formed.
 Topology table: This table stores information about the whole
topology. For example, it contains both the best and backup routes
to a particular advertised network.
 Routing table: This type of table contains all the best routes to the
advertised network.

Advantages:
 This protocol maintains separate tables for both the best route and
the backup routes, so it has more knowledge of the inter-network
than any other distance vector routing protocol.
 Concept of triggered updates are used, so it does not consume any
unnecessary bandwidth.
 Partial updates will be triggered when there is a topology change,
so it does not need to update where the whole routing table is
exchanged.

Exterior Gateway Protocol (EGP)

EGP is a protocol used to exchange data between gateway hosts
that are neighbors with each other within autonomous systems. This
routing protocol offers a forum for routers to share information across
different domains. The full form for EGP is the Exterior Gateway
Protocol. EGP protocol includes known routers, network addresses, route
costs, or neighboring devices.

Enhanced Interior Gateway Routing Protocol (EIGRP)

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 EIGRP is a hybrid routing protocol that provides routing protocols,
distance vector, and link-state routing protocols. The full form routing
protocol EIGRP is Enhanced Interior Gateway Routing Protocol. It will
route the same protocols that IGRP routes using the same composite
metrics as IGRP, which helps the network select the best path destination.

Open Shortest Path First (OSPF)

Open Shortest Path First (OSPF) protocol is a link-state IGP tailor-
made for IP networks using the Shortest Path First (SPF) method.

OSPF routing allows you to maintain databases detailing

information about the surrounding topology of the network. It also uses
the Dijkstra algorithm (Shortest path algorithm) to recalculate network
paths when its topology changes. This protocol is also very secure, as it
can authenticate protocol changes to keep data secure.

Here are some main difference between these Distance Vector and
Link State routing protocols:

Distance Vector Link State

Distance Vector protocol sends the Link State protocol sends only link-
entire routing table. state information.
It is less susceptible to routing
It is susceptible to routing loops.
loops.
Updates are sometimes sent using Uses only multicast method for
broadcast. routing updates.
It is hard to configure this routing
It is simple to configure.
protocol.
Does not know network topology. Know the entire topology.
Example RIP, IGRP. Examples: OSPF IS-IS.

Intermediate System-to-Intermediate System (IS-IS)

ISIS CISCO routing protocol is used on the Internet to send IP
routing information. It consists of a range of components, including end
systems, intermediate systems, areas, and domains.

The full form of ISIS is Intermediate System-to-Intermediate

System. Under the IS-IS protocol, routers are organized into groups called
areas. Multiple areas are grouped to make form a domain.

Border Gateway Protocol (BGP)

BGP is the last routing protocol of the Internet, which is classified
as a DPVP (distance path vector protocol). The full form of BGP is the
Border Gateway Protocol.

100
 This type of routing protocol sends updated router table data when
changes are made. Therefore, there is no auto-discovery of topology
changes, which means that the user needs to configure BGP manually.

What is the purpose of Routing Protocols?

Routing protocols are required for the following reasons:
 Allows optimal path selection
 Offers loop-free routing
 Fast convergence
 Minimize update traffic
 Easy to configure
 Adapts to changes
 Scales to a large size
 Compatible with existing hosts and routers
 Supports variable length

Classful Vs. Classless Routing Protocols

Here are some main difference between these routing protocols:
Classful Routing Protocols Classless Routing Protocols
Classful routing protocols never send Classless routing protocols can
subnet mask detail during routing send IP subnet mask information
updates. while doing routing updates.
RIPv1andIGRP are classful RIPv2, OSPF, EIGRP, and IS-IS
protocols. These two are classful are all types of class routing
protocols as they do not include protocols which has subnet mask
subnet mask information. information within updates.

5.5 SUMMARY:

Features RIP V1 RIP V2 IGRP OSPF EIGRP

Classful/
Classful Classless Classful Classless Classless
Classless

Composite,
Composite
Metric Hop Hop Bandwidth Bandwidth,
Bandwidth, Delay.
Delay.

Periodic 30 seconds 30 seconds 90 seconds None 30 seconds

Advertising 224.0.0.5
255.255.255.255.255 223.0.0.9 255.255.255.255.255 224.0.0.10
Address 224.0.0.6

Distance
Category Distance Vector Distance Vector Link State Hybrid
Vector

Default 120 120 200 110 170

101
Features RIP V1 RIP V2 IGRP OSPF EIGRP
Distance



102
 Module - III

6
ADHOC NETWORKING

Unit Structure
6.1 Introduction
6.2 Application of MANET
6.3 Challenges:
6.4 Broadcasting
6.5 Fundamentals of WLANs
6.6 Infrared vs Radio Transmission

6.1 INTRODUCTION

Recent trends in compact computing and wireless technologies are

expansion of ad hoc network. Ad hoc network consists of versatile flat
forms which are free to move expeditiously. Ad hoc networks are multi-
hop network that use wireless communication for transmission without
any fixed infrastructure.

Adhoc network is an autonomous system node connected with

wireless link The node in the ad hoc network communicates with other
node without any physical representation. The nodes in the ad hoc
organization instantly form the network whenever the communication is
established. Each node in the network communicates with other node
using radio waves. The entire network is distributed and nodes are
collaborated with each other without fixed station access point (AP) or
base station. An ad hoc network is local area network that builds an
automatic connection to the nodes in the network

In the Windows operating system, ad hoc is a communication

mode (setting) that allows computers to directly communicate with each
other without a router. Wireless mobile ad hoc networks are self-
configuring, dynamic networks in which nodes are free to move.

In the Windows operating system, ad hoc is a communication

mode (setting) that allows computers to directly communicate with each
other without a router. Wireless mobile ad hoc networks are self-
configuring, dynamic networks in which nodes are free to move.

102
Following figures shows the difference between infrastructure based
wireless network and Ad hoc wireless network.

As you can see in infrastructure based wireless network there is

using of a network device called a router of a switch but those network
devices are absent into Ad hoc wireless network.

6.2 APPLICATION OF MANET

Mobile ad hoc network (MANET) is a decentralized type of

wireless network.The network is said to be ad hoc because it does not rely
on a pre-existing infrastructure, such as routers in wired networks or
access points in managed (infrastructure) wireless networks.

With the increased number of lightweight devices as well as

evolution in wireless communication, the ad hoc networking technology is
gaining effort with the increasing number of widespread applications.

Ad hoc networking can be used anytime, anywhere with limited or

no communication infrastructure. The preceding infrastructure is fancy or
annoying to use. The ad hoc network architecture can be used in real time
business applications, corporate companies to increase the productivity
and profit.

The ad hoc networks can be classified according to their

application as Mobile Ad hoc Network (MANET) which is a self-
arranging infrastructure less network of mobile devices communicated
through wireless link.

103
 Vehicular Ad hoc Network (VANET) uses travelling cars as nodes
in a network to create a mobile network. Wireless Sensor Network (WSN)
consists of autonomous sensors to control the environmental actions. The
importance of ad hoc network has been highlighted in many fields which
are described below:

Military arena: An ad hoc networking will allow the military battleground

to maintain an information network among the soldiers, vehicles and
headquarters.

Provincial level: Ad hoc networks can build instant link between

multimedia network using notebook computers or palmtop computers to
spread and share information among participants (e.g. Conferences).

Personal area network: A personal area network is a short range,

localized network where nodes are usually associated with a given range.

Industry sector: Ad hoc network is widely used for commercial

applications. Ad hoc network can also be used in emergency situation such
as disaster relief. The rapid development of non-existing infrastructure
makes the ad hoc network easily to be used in emergency situation.

Bluetooth: Bluetooth can provide short range communication between the

nodes such as a laptop and mobile phone.

6.3 CHALLENGES:

The ad hoc networks are self-forming, self-maintaining, self-

healing architecture. The challenges are, no fixed access point, dynamic
network topology, contrary environment and irregular connectivity. Ad
hoc network immediately forms and accommodate the modification and
limited power. Finally, ad hoc have no trusted centralized authority. Due
to the dynamic changing property, the ad hoc faces some challenges which
are listed in the below sections.

Quality of Service (QoS) The ad hoc network is dynamically

creating the organization whenever the node wants to communicate with
their neighbour node. Due the dynamic changing topology in ad hoc
network, providing QoS is a tedious task.

QoS are essential because of rapid development in mobile

technology and real time applications like multimedia, voice. Providing
QoS in ad hoc network is necessary to maintain best-effort-of service.

The QoS metric are bandwidth, latency, jitter and delivery

guarantee. The bandwidth is used to denote the data rate carried in the
network. Latency ensures the delay occur from origin to target. Jitter
denotes the variation of delay. Reliability demonstrate the percentage of
deny to access the network service.
104
 Wireless channels are varying rapidly and it severely affects the
multi-hop flows. In ad hoc networks, the peer-to-peer channel quality may
alter rapidly. So, the link quality may affect the peer-to-peer QoS metrics
in the multi-hop path.

Limited Bandwidth
The wireless networks have a limited bandwidth in comparison to
the wired networks. Wireless link has lower capacity as compare to
infrastructure networks. The effect of fading, multiple accesses,
interference condition is very low in ADHOC networks in comparison to
maximum radio transmission rate.

Dynamic topology
Due to dynamic topology the nodes has less trust between them. I
some settlement are found between the nodes then it also makes trust level
questionable.

High Routing
In ADHOC networks due to dynamic topology some nodes change
their position which affects the routing table.

Problem of Hidden terminal

The Collision of the packets are held due to the transmission of
packets by those nodes which are not in the direct transmission range of
sender side but are in range of receiver side.

Transmission error and packet loss

By increasing in collisions, hidden terminals, interference, uni-
directional links and by the mobility of nodes frequent path breaks a
higher packet loss has been faced by ADHOC networks.

Mobility
Due to the dynamic behaviour and changes in the network
topology by the movement of the nodes. ADHOC networks faces path
breaks and it also changes in the route frequently.

Security threats
New security challenges bring by Ad hoc networks due to its
wireless nature. In Ad hoc networks or wireless networks, the trust
management between the nodes leads to the numerous security attacks.

Routing in Ad hoc networks

In ad hoc networks, nodes are not familiar with the topology of
their networks. Instead, they have to discover it: typically, a new node
announces its presence and listens for announcements broadcast by its
neighbours. Each node learns about others nearby and how to reach them,
and may announce that it too can reach them.

105
 There are basic 4 types of routing in ad hoc network is present
listed below.
1) Table-driven (proactive) routing
2) On-demand (reactive) routing
3) Hybrid (both proactive and reactive) routing
4) Hierarchical routing protocols

1) Table-driven (proactive) routing

 This type of protocols maintains fresh lists of destinations and their
routes by periodically distributing routing tables throughout the
network.
 In proactive routing (table-driven routing), the routing tables are
created before packets are sent – Link-state (e.g. OSPF) –
Distance-vector (e.g. RIP)
 Each node knows the routes to all other nodes in the network
 Problems in Ad-Hoc networks
 Maintenance of routing tables requires much bandwidth
 Dynamic topology ˇ much of the routing information is never used
ˇWaste of capacity
 Flat topology ˇNo aggregation

2) On-demand (reactive) routing

 This type of protocol finds a route on demand by flooding the
network with Route Request packets.
 In reactive routing the routes are created when needed.
 Before a packet is sent, a route discovery is performed.
 The results are stored in a cache.
 When intermediate nodes move, a route repair is required.
 Advantages – Only required routes are maintained
 Disadvantages
 Delay before the first packet can be sent
 Route discovery usually involves flooding

3) Hybrid (both proactive and reactive) routing

 This type of protocol combines the advantages of proactive and
reactive routing.
 The routing is initially established with some proactively
prospected routes and then serves the demand from additionally
activated nodes through reactive flooding.
 The choice of one or the other method requires predetermination
for typical cases.

106
4) Hierarchical routing protocols
 With this type of protocol, the choice of proactive and of reactive
routing depends on the hierarchic level in which a node resides.
 The routing is initially established with some proactively
prospected routes and then serves the demand from additionally
activated nodes through reactive flooding on the lower levels.
 The choice for one or the other method requires proper
attributation for respective levels.

Routing Protocols: Topology based

An ad hoc wireless multi-hop network (AHWMNs) is a collection
of mobile devices which form a communication network with no pre-
existing wiring or infrastructure. Routing in AHWMNs is challenging
since there is no central coordinator that manage routing decisions.
AHWMN routing protocols are classified as topology-based, position-
based.

Fig. 3.Classification of routing protocol

Topology-based routing protocols use the information about the

links that exist in the network to perform packet forwarding. They can be
further divided into proactive, reactive and hybrid approaches.

Proactive algorithms employ classical routing strategies such as

distance- vector routing (e.g. DSDV) or link-state routing (e.g. OLSR).
They maintain routing information about the available paths in the
network even if these paths are currently not used. The main drawback of
these approaches is the maintenance of unusual path may occupy a
significant part of the available bandwidth if the topology of the network
changes frequently.

Reactive routing protocols such as AODV and DSR maintain only

the routes that are currently in use and hence reduce the burden on the
network. However, they still have some inherent limitations. First, since
the routes are maintained only while in use, it is required to perform a

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route discovery before packets are exchanged between communication
nodes. Second, even though route discovery is restricted to the routes
currently in use, it may still generate a significant amount of network
traffic when the topology of the network changes frequently.

Position-based routing algorithms eliminate some of the limitations

of topology-based routing by using additional information. Position based
routing based on idea that the source sends a message to the geographic
location of destination instead of using the network address. Position
based routing requires information about the physical position of
participating nodes. Commonly, each node determines its own position
through the use of Global Positioning System (GPS). Decisions made
based on destination position and position of forwarding nodes neighbors.
A location service is used by the sender of packet to determine the
position of the destination and to include it in the packet destination
address.

Greedy Perimeter Stateless Routing (GPSR) protocol is an efficient

and scalable routing protocol in MANETs. In GPSR protocol, a node route
the data packet using the locations of its one hop neighbors. When the
node needs to send a data packet to destination node, it transmits the data
packet to the neighbour who has the shortest distance to the destination
node among all its neighbors within its transmission range. GPSR protocol
uses two forwarding strategies to route the data packet to the destination.
They are greedy forwarding and perimeter forwarding. GPSR makes
greedy forwarding decisions using only information about a router’s
immediate neighbors in the network topology. When a packet reaches a
region where greedy forwarding is impossible, the algorithm recovers by
routing around the perimeter of the region. By keeping state only about the
local topology, GPSR scales better in per-router state than shortest-path
and ad-hoc routing protocols as the number of network destinations
increases. Under mobility’s frequent topology changes, GPSR can use
local topology information to find correct new routes quickly.

In GPSR, packets are marked by their originator with their

destinations’ locations. As a result, a forwarding node can make a locally
optimal, greedy choice in choosing a packet’s next hop. Specifically, if a
node knows its radio neighbors’ positions, the locally optimal choice of
next hop is the neighbor geographically closest to the packet’s destination.
Forwarding in this regime follows successively closer geographic hops,
until the destination is reached.

Position based
Position based routing are as follows:
1) GFG
2) GOAFR
3) GOAFR+
4) AFR
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5) OGPR
6) GPVFR
7) LAR

1) GFG
Nearly Stateless Routing with Guaranteed Delivery is schemes
where nodes maintain only some local information to perform routing.
The face routing and Greedy-Face-Greedy (GFG) schemes were described
in. In order to ensure message delivery, the face routing (called perimeter
algorithm) constructs a planar and connected so-called Gabriel sub graph
of the unit graph, and then applies routing along the faces of the sub graph
(e.g. by using the right hand rule) that intersect the line between the source
and the destination. If a face is traversed using the right hand rule then a
loop will be created, since a face will never exist. Forwarding in the right
hand rule is performed using the directional approach.

To improve the efficiency of the algorithm in terms of routing

performance, face routing can be combined with algorithms that usually
find shorter routes, such as the greedy algorithm to yield GFG algorithm.
Routing is mainly greedy, but if a mobile host fails to find a neighbor
closer than itself to the destination, it switches the message from ‘greedy’
state to ‘face’ state.

2) GOAFR
A greedy routing approach is not only worth being considered due
to its simplicity in both concept and implementation. Above all in dense
networks such an algorithm can also be expected to end paths of good
quality efficiently here, the straightforwardness of a greedy strategy
contrasts highly the inexible exploration of faces inherent to face routing.
For practical purposes it is inevitable to improve the performance of a face
routing variant by leveraging the potential of a greedy approach. Such a
combination of greedy routing and our OAFR algorithm forms Greedy
Other Adaptive Face Routing GOAFR. In principle greedy routing is used
as long as possible. Local minima potentially met under ways are escaped
from by use of OAFR.

3) GOAFR+
The GOAFR+ algorithm is a combination of greedy routing and
face routing. Whenever possible the algorithm tries to route greedily, that
is by forwarding the message at each intermediate node to the neighbour
located closest to the destination. Doing so, however, the algorithm can
reach a local minimum with respect to the distance from destination that is
a node um none of whose neighbours is located closer to destination than
itself. In order to overcome such a local minimum, GOAFR+ applies a
face routing technique, borrowing from the Face Routing algorithm.

Face Routing proceeds towards the destination by exploring the

boundaries of the faces of a planarized network graph, employing the local
right hand rule (in analogy to following the right hand wall in a maze).
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Additionally the algorithm restricts itself to a searchable area occasionally
being resized during algorithm execution.

4) AFR
The basis of this algorithm is formed by Face Routing. At the heart
of Face Routing lies the exploration of the boundaries of faces in a planar
graph, employing the local right hand rule (in analogy to following the
right hand wall in a maze). On its way around a face, the algorithm keeps
track of the points where it crosses the line connecting the source and the
destination. Having completely surrounded a face, the algorithm returns to
the one of these intersections lying closest to the destination, where it
proceeds by exploring the next face closer to destination. If the source and
the destination are connected, Face Routing always ends a path to the
destination.

5) OGPR
OGPR is an efficient and scalable routing protocol, that inherits the
well-known techniques for routing,
1) Greedy forwarding
2) Reactive route discovery
3) Source routing

In this, protocol source node utilizes the geographic topology

information obtained during the location request phase to establish
geographic paths to their respective destinations.

6) GPVFR
In this section we describe Greedy PVFR, a non-oblivious routing
algorithm that does not require the participating nodes to have complete
face information. GPVFR is designed as a tri-modal algorithm with the
following modes
 Greedy: greedy forwarding using neighbour information,
 OPVFR: greedy forwarding using face information, and
 Perimeter: perimeter traversal (as in GPSR).

Under GPVFR, packets are first routed in Greedy mode. When

greedy forwarding to an immediate neighbour fails, a node may find that it
knows of another node along its planar faces that is nearer to the
destination than itself.

7) LAR
The Location Aided Routing proposal does not define a location-
based routing protocol but instead proposes the use of position information
to enhance the route discovery phase of reactive ad hoc routing
approaches. Reactive ad hoc routing protocols frequently use flooding as a
means of route discovery.

Under the assumption that nodes have information about other

nodes’ positions, this position information can be used by LAR to restrict
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the flooding to a certain area. This is done in a fashion similar to that of
the DREAM approach. When node S wants to establish a route to node D,
S computes an expected zone for D based on available position
information. If no such information is available LAR is reduced to simple
flooding. If location information is available (e.g., from a route that was
established earlier), a request zone is defined as the set of nodes that
should forward the route discovery packet.

The request zone typically includes the expected zone. The first is
a rectangular geographic region. In this case, nodes will forward the route
discovery packet only if they are within that specific region. The second is
defined by specifying (estimated) destination coordinates plus the distance
to the destination. In this case, each forwarding node overwrites the
distance field with its own current distance to the destination. A node is
allowed to forward the packet again only if it is at most some δ (system
parameter) farther away than the previous node.

6.4 BROADCASTING

Multicasting
In multicasting routing, the data are transmitted from one source to
multiple destinations. Multicast protocols can be categorized into two
types, namely tree-based multicast and mesh based multicast. The tree
based multicast routing protocols utilize the network resource in efficient
manner. Mesh based protocols are robust due to formation of many
redundant paths between the nodes and in high packet delivery ratio.

Ad hoc multicast routing protocol (AMRoute): Xie et al. [75]

developed AMRoute, with main design objective are: scalability and
robustness. In ad hoc network with highly dynamic mobile nodes, the
control packets overhead are high due to maintenance of multi cast tree.

Adaptive demand-driver multicast routing (ADMR): ADMR, on-

demand multicast routing algorithm, developed by Jetchera and Johnson
[76]. This protocol does not support any non on-demand components.
ADMR, uses a source based forwarding trees and monitors the traffic
pattern and rate of the source. ADMR navigates back to the normal mode,
when the mobility of the node is reduced.

Differential destination multicast (DDM): Ji and Corson [77]

proposed the DDM algorithm. DDM has two important characteristics
features: 1. the sender node will have full control over the members of
group nodes. 2. Source node, encodes the address within each data packets
header on an in-band fashion.

Dynamic core based multicast routing (DCMP): Das et al. [78]

proposed DCMP source initiated multicast protocol with an objective to
increase the scalability and efficiency as well as to decrease the overhead.
In this protocol the source as been classified into active, core active and
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passive. A core active source can support up to maximum of MaxPassSize
passive resource and the hop distance between them is limited by the
MaxHop parameter.

AdhocQoS multicasting (AQM): AQM protocols developed by

Bur and Ersoy [79]. In this protocol QoS of the neighboring node
monitored and maintained as well as used for efficient multicast routing.
Node announces the QoS status during the session initiation phase to join
a session, the nodes executes request-reply–reverse procedure, ensures the
QoS information is updated and a possible route is chosen session is
initiated by a session initiator node.

Content based multicast (CBM): CBM developed by Zhou and

Singh [80]. In CBM the nodes collect information about threats and
resource at a time period t and distance d away from the location of the
node.

Energy efficient multicast routing: Li et al. [81] proposed an

energy efficient multicast routing protocol. The authors constructed a
weighted network graph by considering the transmission power of each
node as a weight between edges. Each node has only information
regarding their neighbors. The objective of minimum energy multicast
(MEM), problem is to develop the multicast tree with a minimum total
energy cost. In this approach, multicast tree is formed by nodes within the
highest energy efficiency.

QoS multicast routing protocols for clustering mobile ad hoc

networks (QMRPCAH): QMRPCAH, QoS aware multicast routing
protocol for clustered ad hoc network was developed by Layuan and
Chunlin [82]. It enhances scalability and flexibility.

Epidemic-based reliable and adaptive multicast for mobile ad hoc

networks (Eramobile): Eramobile, highly reliable and an adaptive
multicast protocol proposed by Ozkasap et al. [83]. In this protocol bio-
inspired epidemic methods are utilized in multicast operation in order to
support dynamic and topology changes due the unpredictable mobility of
the nodes in the network. Table 7 illustrates the comparative analysis of
multicast routing protocol.

Geocasting
Geocast routing protocols have the combined features of both
geographical and multicast routing protocols. The major advantage of
Geocast routing protocols are performance improvement and minimizing
the control overhead.

Geocasting in mobile ad hoc networks (GeoTORA): Ko and

Vaidya [8] proposed the GeoTORA protocol, is based upon the unicast
TORA routing protocol.

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 Geocast protocol for mobile ad hoc network based on GRID
(GEOGRID): GeoGRID routing protocol was developed by Liao et al. [7]
GeoGrid extends on the unicasting routing protocol GRID. GeoGRID
exploit location information in route discovery to define the forwarding
zone or geographical area.

Direction guided routing (DGR): An and Papavassilliou [94]

designed DGR algorithm based on clustering mechanism. In DGR, the
nodes in the network are grouped into clusters and the cluster head is
elected using the techniques such a mobile clustering algorithm (MCA).

Geocast adaptive mesh environment for routing (GAMER):

GAMER protocol developed by Camp and Liu [95] is based on the
mobility nature of nodes. This protocol exploits the mesh creation
approach. Table 9 illustrates the Geocast routing protocols comparative
analysis.

RS = routing structure; H = hierarchical; F = flat; SP = shortest path;

RC = route cache; RT = route table.

Wireless Lan
Wireless LAN stands for Wireless Local Area Network. It is also
called LAWN (Local Area Wireless Network). WLAN is one in which a
mobile user can connect to a Local Area Network (LAN) through a
wireless connection.

The IEEE 802.11 group of standards defines the technologies for

wireless LANs. For path sharing, 802.11 standard uses the Ethernet
protocol and CSMA/CA (carrier sense multiple access with collision
avoidance). It also uses an encryption method i.e. wired equivalent privacy
algorithm.

Wireless LANs provide high speed data communication in small

areas such as building or an office. WLANs allow users to move around in
a confined area while they are still connected to the network.

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In some instance wireless LAN technology is used to save costs and avoid
laying cable, while in other cases, it is the only option for providing high-
speed internet access to the public. Whatever the reason, wireless solutions
are popping up everywhere.

Examples of WLANs that are available today are NCR's waveLAN

and Motorola's ALTAIR.

Advantages of WLANs
 Flexibility: Within radio coverage, nodes can communicate without
further restriction. Radio waves can penetrate walls, senders and
receivers can be placed anywhere (also non-visible, e.g., within
devices, in walls etc.).
 Planning: Only wireless ad-hoc networks allow for communication
without previous planning, any wired network needs wiring plans.
 Design: Wireless networks allow for the design of independent, small
devices which can for example be put into a pocket. Cables not only
restrict users but also designers of small notepads, PDAs, etc.
 Robustness: Wireless networks can handle disasters, e.g.,
earthquakes, flood etc. whereas, networks requiring a wired
infrastructure will usually break down completely in disasters.
 Cost: The cost of installing and maintaining a wireless LAN is on
average lower than the cost of installing and maintaining a traditional
wired LAN, for two reasons. First, after providing wireless access to
the wireless network via an access point for the first user, adding
additional users to a network will not increase the cost. And second,
wireless LAN eliminates the direct costs of cabling and the labor
associated with installing and repairing it.
 Ease of Use: Wireless LAN is easy to use and the users need very
little new information to take advantage of WLANs.

Disadvantages of WLANs
 Quality of Services: Quality of wireless LAN is typically lower than
wired networks. The main reason for this is the lower bandwidth due
to limitations is radio transmission, higher error rates due to
interference and higher delay/delay variation due to extensive error
correction and detection mechanisms.
 Proprietary Solutions: Due to slow standardization procedures, many
companies have come up with proprietary solutions offering
standardization functionality plus many enhanced features. Most
components today adhere to the basic standards IEEE 802.11a or
802.11b.
 Restrictions: Several govt. and non-govt. institutions world-wide
regulate the operation and restrict frequencies to minimize
interference.
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 Global operation: Wireless LAN products are sold in all countries so,
national and international frequency regulations have to be considered.
 Low Power: Devices communicating via a wireless LAN are typically
power consuming, also wireless devices running on battery power.
Whereas the LAN design should take this into account and implement
special power saving modes and power management functions.
 License free operation: LAN operators don't want to apply for a
special license to be able to use the product. The equipment must
operate in a license free band, such as the 2.4 GHz ISM band.
 Robust transmission technology: If wireless LAN uses radio
transmission, many other electrical devices can interfere with them
(such as vacuum cleaner, train engines, hair dryers, etc.). Wireless
LAN transceivers cannot be adjusted for perfect transmission is a
standard office or production environment.

6.5 FUNDAMENTALS OF WLANS

1. HiperLAN
 HiperLAN stands for High performance LAN. While all of the
previous technologies have been designed specifically for an adhoc
environment, HiperLAN is derived from traditional LAN
environments and can support multimedia data and asynchronous data
effectively at high rates (23.5 Mbps).
 A LAN extension via access points can be implemented using standard
features of the HiperLAN/1 specification. However, HiperLAN does
not necessarily require any type of access point infrastructure for its
operation.
 HiperLAN was started in 1992, and standards were published in 1995.
It employs the 5.15GHz and 17.1 GHz frequency bands and has a data
rate of 23.5 Mbps with coverage of 50m and mobility< 10 m/s.
 It supports a packet-oriented structure, which can be used for networks
with or without a central control (BS-MS and ad-hoc). It supports 25
audio connections at 32kbps with a maximum latency of 10 ms, one
video connection of 2 Mbps with 100 ms latency, and a data rate of
13.4 Mbps.
 HiperLAN/1 is specifically designed to support adhoc computing for
multimedia systems, where there is no requirement to deploy a
centralized infrastructure. It effectively supports MPEG or other state
of the art real time digital audio and video standards.
 The HiperLAN/1 MAC is compatible with the standard MAC service
interface, enabling support for existing applications to remain
unchanged.

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 HiperLAN 2 has been specifically developed to have a wired
infrastructure, providing short-range wireless access to wired networks
such as IP and ATM.

The two main differences between HiperLAN types 1 and 2 are

as follows:
 Type 1 has a distributed MAC with QoS provisions, whereas type
2 has a centralized schedule MAC.
 Type 1 is based on Gaussian minimum shift keying (GMSK),
whereas type 2 is based on OFDM.
 HiperLAN/2 automatically performs handoff to the nearest access
point. The access point is basically a radio BS that covers an area
of about 30 to 150 meters, depending on the environment.
MANETs can also be created easily.

The goals of HiperLAN are as follows:

 QoS (to build multiservice network)
 Strong security
 Handoff when moving between local area and wide areas
 Increased throughput
 Ease of use, deployment, and maintenance
 Affordability
 Scalability

One of the primary features of HiperLAN/2 is its high speed

transmission rates (up to 54 Mbps). It uses a modulation method called
OFDM to transmit analog signals. It is connection oriented, and traffic is
transmitted on bidirectional links for unicast traffic and unidirectional
links toward the MSs for multicast and broadcast traff

This connection oriented approach makes support for QoS easy, which
in turn depends on how the HiperLAN/2 network incorporates with the
fixed network using Ethernet, ATM, or IP.

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 The HiperLAN/2 architecture shown in the figure allows for
interoperation with virtually any type of fixed network, making the
technology both network and application independent.

Hiper LAN/2 networks can be deployed at "hot spot" areas such as

airports and hotels, as an easy way of offering remote access and internet
services.

2. Home RF Technology
 A typical home needs a network inside the house for access to a public
network telephone and internet, entertainment networks (cable
television, digital audio and video with the IEEE 1394), transfer and
sharing of data and resources (printer, internet connection), and home
control and automation.
 The device should be able to self-configure and maintain connectivity
with the network. The devices need to be plug and play enabled so that
they are available to all other clients on the network as soon as they are
switched on, which requires automatic device discovery and
identification in the system.
 Home networking technology should also be able to accommodate any
and all lookup services, such as Jini. Home RF products allow you to
simultaneously share a single internet connection with all of your
computers - without the hassle of new wires, cables or jacks.
 Home RF visualizes a home network as shown in the figure:

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 A network consists of resource providers, which are gateways to
different resources like phone lines, cable modem, satellite dish, and
so on, and the devices connected to them such as cordless phone,
printers and fileservers, and TV.
 The goal of Home RF is to integrate all of these into a single network
suitable for all applications and to remove all wires and utilize RF
links in the network suitable for all applications.
 This includes sharing PC, printer, fileserver, phone, internet
connection, and so on, enabling multiplayer gaming using different
PCs and consoles inside the home, and providing complete control on
all devices from a single mobile controller.
 With Home RF, a cordless phone can connect to PSTN but also
connect through a PC for enhanced services. Home RF makes an
assumption that simultaneous support for both voice and data is
needed.

Advantages of Home RF
 In Home RF all devices can share the same connection, for voice or
data at the same time.
 Home RF provides the foundation for a broad range of interoperable
consumer devices for wireless digital communication between PCs and
consumer electronic devices anywhere in and around the home.
 The working group includes Compaq computer corp. Ericson
enterprise network, IBM Intel corp., Motorola corp. and other.
 A specification for wireless communication in the home called the
shared wireless access protocol (SWAP) has been developed.

3. IEEE 802.11 Standard

IEEE 802.11 is a set of standards for the wireless area network
(WLAN), which was implemented in 1997 and was used in the industrial,
scientific, and medical (ISM) band. IEEE 802.11 was quickly
implemented throughout a wide region, but under its standards the
network occasionally receives interference from devices such as cordless
phones and microwave ovens. The aim of IEEE 802.11 is to provide
wireless network connection for fixed, portable, and moving stations
within ten to hundreds of meters with one medium access control (MAC)
and several physical layer specifications. This was later called 802.11a.
The major protocols include IEEE 802.11n; their most significant
differences lie in the specification of the PHY layer.

4. Bluetooth
Bluetooth is one of the major wireless technologies developed to
achieve WPAN (wireless personal area network). It is used to connect
devices of different functions such as telephones, computers (laptop or
desktop), notebooks, cameras, printers, and so on.

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Architecture of Bluetooth
 Bluetooth devices can interact with other Bluetooth devices in several
ways in the figure. In the simplest scheme, one of the devices acts as
the master and (up to) seven other slaves.
 A network with a master and one or more slaves associated with it is
known as a piconet. A single channel (and bandwidth) is shared among
all devices in the piconet.

 Each of the active slaves has an assigned 3-bit active member address.
many other slaves can remain synchronized to the master though
remaining inactive slaves, referred to as parked nodes.
 The master regulates channel access for all active nodes and parked
nodes. Of two piconets are close to each other, they have overlapping
coverage areas.
 This scenario, in which nodes of two piconets intermingle, is called a
scatternet. Slaves in one piconet can participate in another piconet as
either a master or slave through time division multiplexing.
 In a scatternet, the two (or more) piconets are not synchronized in
either time or frequency. Each of the piconets operates in its own
frequency hopping channel, and any devices in multiple piconets
participate at the appropriate time via time division multiplexing.
 The Bluetooth baseband technology supports two link types.
Synchronous connection oriented (SCO) types, used primarily for
voice, and asynchronous connectionless (ACL) type, essentially for
packet data.

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6.6 INFRARED VS RADIO TRANSMISSION

Infrared Transmission
 Infrared technology uses diffuse light reflected at walls, furniture etc.
or a directed light if a line of sight (LOS) exists between sender and
receiver.
 Infrared light is the part of the electromagnetic spectrum, and is an
electromagnetic form of radiation. It comes from the heat and thermal
radiation, and it is not visible to the naked eyes.
 In infrared transmission, senders can be simple light emitting diodes
(LEDs) or laser diodes. Photodiodes act as receivers.
 Infrared is used in wireless technology devices or systems that convey
data through infrared radiation. Infrared is electromagnetic energy at a
wave length or wave lengths somewhat longer than those of red light.
 Infrared wireless is used for medium and short range communications
and control. Infrared technology is used in instruction detectors; robot
control system, medium range line of sight laser communication,
cordless microphone, headsets, modems, and other peripheral devices.
 Infrared radiation is used in scientific, industrial, and medical
application. Night vision devices using active near infrared
illumination allow people and animals to be observed without the
observer being detected.
 Infrared transmission technology refers to energy in the region of the
electromagnetic radiation spectrum at wavelength longer than those of
visible light but shorter than those of radio waves.
 Infrared technology allows computing devices to communicate via
short range wireless signals. With infrared transmission, computers
can transfer files and other digital data bidirectional.

Advantages of infrared
 The main advantage of infrared technology is its simple and
extremely cheap senders and receivers which are integrated into
nearly all mobile devices available today.
 No licenses are required for infrared and shielding is very simple.
 PDAs, laptops, notebooks, mobile phones etc. have an infrared
data association (IrDA) interface.
 Electrical devices cannot interfere with infrared transmission.

Disadvantages of Infrared
 Disadvantages of infrared transmission are its low bandwidth
compared to other LAN technologies.

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  Limited transfer rates to 115 Kbit/s and we know that even 4
Mbit/s is not a particular high data rate.
 Their main disadvantage is that infrared is quite easily shielded.
 Infrared transmission cannot penetrate walls or other obstacles.
 Typically, for good transmission quality and high data rates a LOS
(Line of site), i.e. direct connection is needed.

Radio Transmission
 Almost all networks use radio waves for data transmission, e.g.,
GSM at 900, 1800, and 1900 MHz, DECT at 1880 MHz etc. Radio
transmission technologies can be used to set up ad-hoc connections
for work groups, to connect, e.g., a desktop with a printer without a
wire, or to support mobility within a small area.
 The two main types of radio transmission are AM (Amplitude
Modulation) and (FM) Frequency Modulation.
 FM minimizes noise and provides greater reliability. Both AM and
FM process sounds in patterns that are always varying of electrical
signals.
 In an AM transmission the carrier wave has a constant frequency,
but the strength of the wave varies. The FM transmission is just the
opposite; the wave has constant amplitude but a varying frequency.
 Usually the radio transmission is used in the transmission of
sounds and pictures. Such as, voice, music and television.
 The images and sounds are converted into electrical signals by a
microphone or video camera. The signals are amplified, and
transmitted. If the carrier is amplified it can be applied to an
antenna.
 The antenna converts the electrical signals into electromagnetic
waves and sends them out or they can be received. The antenna
consists commonly of a wire or set of wires.

Advantages of Radio Transmission

 Advantages of radio transmission include the long-term
experiences made with radio transmission for wide area networks
(e.g. microwave links) and mobile cellular phones.
 Radio transmission can cover larger areas and can penetrate
(thinner) walls, plants, furniture etc.
 Additional coverage is gained by reflection.
 Radio typically does not need a LOS (Line of Site) if the
frequencies are not too high.
 Higher transmission rates (e.g. 54 Mbit/s) than infrared (directed
laser links, which offer data rates well above 100 Mbit/s).
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Disadvantages of Radio Transmission
 Radio transmission can be interfered with other senders, or
electrical devices can destroy data transmitted via radio.
 Bluetooth is simple than infrared.
 Radio is only permitted in certain frequency bands.
 Shielding is not so simple.
 Very limited ranges of license free bands are available worldwide
and those that are available are not the same in all countries.
 A lot harmonization is going on due to market pressure.

Transmission techniques

MAC Protocol issues

Wireless PANs
The Bluetooth technology



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 Module IV

7
WIRELESS SENSOR NETWORKS
Unit Structure
7.0 Objective
7.1 Need and application of sensor networks
7.2 sensor networks design considerations
7.3 empirical energy consumption
7.4 sensing and communication range
7.5 localization scheme
7.6 clustering of SNs
7.7 Routing layer
7.8 Sensor networks in controlled environment and actuators
7.9 regularly placed sensors
7.10 network issues
7.11 RFID as passive sensors
7.12 Unit End Questions

7.0 OBJECTIVES

This chapter would make you understand the following concepts

● Implementation and use of wireless sensor networks
● To understand placement of sensor nodes to promote productivity
● Implement and evaluate new ideas for solving wireless sensor
network design issues
● Various applications of wireless sensor networks

7.1 NEED AND APPLICATION OF SENSOR

NETWORKS

7.1.1 Introduction
A sensor is a device that measures a change in a physical or
environmental condition

In other words, a sensor is a device that responds to any thing or

reaction, for example, any environmental parameters such as light, heat,
humidity, or pressure, and generates a signal that can be interpreted and
measured to produce output and display to the users for information
purposes.

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 The Sensor Network community oftenly states a sensor node as a
small, wireless sensing device, which has the ability to respond to the
action, then process the captured data and transmit the data over the
wireless connection using radio link.

Mostly sensors are implemented for measuring the

environmental parameters such as light, heat, humidity, pressure,
temperature. But it also has the ability to measure other factors too such as
the vibrations, electromagnetic fields to predict any natural calamities.

The information captured by these small units called the

sensors can be transmitted using wireless links so as to reduce the
configuration complexity and make the information process simple and
dynamic.

Once we have an idea about what are sensors, now let's look
into what is a sensor network.

Sensor Network

A Sensor Network is an ad hoc wireless network which is

made of a large number (hundreds or thousands) of sensor nodes, which
are positioned randomly.
A sensor network communications consist of a protocol stack model.

Figure: Protocol stack of Sensor Network

Protocol stack model is broadly classified into:

1. Communication protocol
2. Management protocol

1. Communication Protocol
It consists of a physical layer, data link layer, network layer, transport
layer and application layer

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2. Management Protocol
It consists of power management, mobility management and task
management planes.

Wireless sensor networks (WSNs)

Figure: Wireless Sensor Network

Wireless Sensor Networks (WSNs) can be identified as a self-

configured and infrastructure-less wireless networks which is formed by
hundreds or thousands of sensor nodes which monitors physical or
environmental conditions, such as sound, pressure, temperature, vibration,
motion or pollutants and passes their data through the network to a base
station (sink node) where the data is basically collected, observed and
analysed.

A base or sink station acts like an interface (mediator) between the

network and the user.

Every sensor node is equipped with computing and sensing

devices, power components and radio transceivers.

After the sensor nodes are configured, they self-organize a

considerable network infrastructure with multi-hop communication.

Local Positioning algorithms and Global Positioning System

(GPS) are used to obtain information of location and position of sensor
nodes.

These networks are also known as Actuator Networks because they

consist of actuators.

7.1.2 Application of Sensor Network

WSNs has been implemented in various application domains to

solve problems and make a change in our lives in many different ways
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Military applications:
WSNs have a vital role to play in military control, communication,
command, computing, battlefield surveillance, intelligence,
reconnaissance and targeting systems.

Area monitoring:
The sensor nodes are installed sparsely over an area and the
phenomenon is monitored. The monitored parameters (humidity,
temperature) are then sent to the base stations to take appropriate actions.

Transportation:
WSN helps in collecting traffic information to alert the drivers
regarding the traffic and congestion problems in their path.

Health applications:
WSN also contributes in health applications in monitoring patients,
conducting diagnosis, drug administration in hospitals, monitoring
patient’s physiological data, and tracking & monitoring patients, doctors.

Environmental sensing:
WSN is popularly developed to cover many applications related to
earth science research. This includes sensing earthquakes, glaciers,
volcanoes, oceans, etc.

Some of the other areas are as follows:

● Air pollution monitoring
● Greenhouse monitoring
● Forest fires detection
● Landslide detection

Structural monitoring:
Sensors can be used to inspect the building structures, monitor the
movement within buildings and durability of the infrastructure such as
bridges, flyovers, tunnels etc. The result of the monitoring helps the civil
architect to take preventive measures against any hazard

Industrial monitoring:
WSN has been developed for implementing machinery on
condition-based maintenance (CBM) as it promotes cost savings and
enables new features and increases productivity.

Agricultural sector:
Wireless sensor network measures the humidity of the soil and
indicates the need of irritation of crops thus increasing the productivity. It
also promotes irrigation automation thus enabling use of water efficiently
and reducing wastage of water.

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7.2 SENSOR NETWORKS DESIGN CONSIDERATION

WSN consists of a large number of sensor nodes that have less

processing capability and limited power. The number of sensor nodes in
the network are not constant i.e. there can be addition of new sensor nodes
in the network or nodes may be deleted also.

While designing the sensor network, the factors that are to be

considered are listed below.

Fault Tolerance:
There is a possibility that a node fails, thus changing the topology
of the network. In such a case, the network must be made robust to adopt
the changes so that the functioning of the topology is not disrupted and the
network functions efficiently.

Lifetime:
It is assumed that WSN should work for at least 6 months to 1 year
using a 3 V battery providing good performance and with good energy.
The designer should keep in mind that it consumes less energy thus
making the network to last for longer.

Scalability:
The WSN must be able to support additional nodes at any given
point of time without interfering with the other’s performance. Also some
applications require more number of sensor nodes so the design should be
made in such a way that it supports a large design of network.

Date Aggregation:
The sensor nodes are placed close to each other due to which
similar data can be generated by the nodes which are next to each other.
The data can be collected and duplicate data can be extracted at different
levels.

Cost:
The cost of each sensor node is very expensive and as we all know
the Sensor network is made up of a large number of sensor nodes and
hence the cost will be a major concern. So, the design must be such that all
the data is monitored by optimal usage of the number of nodes.

Environment:
The sensor nodes deployed in WSN must be survivable under all
conditions as the environment may be demanding.

Heterogeneity Support:
The design of the sensor network must support different types of
sensor nodes based on its functionality.

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Autonomous Operations:
The WSN should be able to operate, organize , and reorganize on
its own without human intervention.

Limited Memory and Processing Capability:

Depending on the functionality, each sensor node has restricted
power, memory, and processing capabilities, so the design should be such
that additional memory or power is not needed.

7.3 EMPIRICAL ENERGY CONSUMPTION

A challenge for the design of WSN is minimizing the energy

consumption of Wireless Sensors.

The energy consumption in WSN involves three components:

1. Sensing Unit (Sensing transducer and A/D Converter)
2. Communication Unit (transmission and receiver radio)
3. Computing/Processing Unit.

If we want to conserve energy then we’ll have to put some SNs

to sleep mode.

● Sensing Unit:
○ The Sensing transducer captures the physical parameters of the
environment.
○ It does physical signal sampling and then converts it into electrical
signals.
○ Using this component, the energy consumption depends on the
hardware, the application used and the sensing energy spent.

● An AD Converter for sensor consumes only 3.1 JLlW , in 31 pJ/8-bit

sample at lVolt supply.

The standby power consumption at IV supply is 4lpW.

Transmission Energy:
Energy consumed to transmit information is given as follows

Receiver Energy:
Energy consumed to receive k bits.

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● Commutation Unit:

The computation unit consists of a microcontroller/ processor

with memory which can monitor, control and operate the sensing,
computing and communication unit.

The energy consumption of this unit is categorised into two parts:

● switching energy
● leakage energy

Switching energy is calculated as

where Ctotal is the total capacitance switched by the computation and Vdd
is the supply voltage.

When no computation is carried out, the energy consumed is

called leakage energy. It can be calculated as:

where VT is the thermal voltage, I0 are the parameters of the processor

Sleeping: To save energy, sensors can be put to sleep-active cycles. When
a sensor is put to sleep, thus saving energy.

7.4 SENSING AND COMMUNICATION RANGE

A wireless sensor network (WSN) consists of a large number of sensor

nodes (SNs) The main objective of a SN is to monitor physical and
environmental parameters. In a given area, the sensors need to be
deployed so that the complete area sensing can be done, without leaving
any area not monitored. The SNs can be distributed randomly or placed at
a preferred location.

Every SN has its own sensing range and to sense the complete area,
the neighboring SNs have to be installed close to each other and at
maximum 2rs distance from each other.

If the Sensor Nodes are uniformly distributed with the node having
density X, then the probability of having ‘m’ Sensor nodes within the area
of S is Poisson distributed as

This is basically the probability that the monitored area is not

covered by any Sensor node and therefore the probability pcovcr of the
coverage by at least one SN is:

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 It gives us information about the coverage of the area so that we
understand how many more sensor nodes are needed to be deployed.

Figure: Sensing range of the sensor nodes

When there is at least one sensor node within the

communication range then the transmission between neighboring Sensor
Nodes is achievable

Information from one sensor node is not sufficient. We need a

set of SNs to monitor the area and provide information so for this the
major concern is how far should the sensor nodes be placed. The distance
between two sensor nodes is estimated as 2rs from the sensing coverage
point.

Let us consider SN1 and SN2, to establish a communication

between these two SNs the minimum distance between them should be 2rs
i.e. the communication coverage range should be at least twice the sensing
distance

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7.5 LOCALIZATION SCHEME

In Wireless Sensor Networks (WSNs), Localization is to

discover the current location of the sensor nodes. Localization is
calculated by the communication between localized and unlocalized
sensor nodes for deciding their geometrical position. Location is nothing
but distance and angle between nodes.

The many concepts used in localization are stated as follows.

● Lateration - Distance between nodes is measured.
● Angulation - Angle between nodes is measured
● Trilateration - It is the distance measurement from three nodes.
Position of an unlocalized node is calculated by the intersection of
three circles and the single point is the position.
● Multilateration - To determine the location, more than three nodes are
needed
● Triangulation - Minimum two angles has to be measured of an
unlocalized node from localized sensor nodes.

Localization schemes are classified as:

● anchor based or anchor free
● centralized or distributed
● GPS based or GPS free
● fine grained or coarse grained
● stationary or mobile sensor nodes
● range based or range free.

● Anchor Based and Anchor Free

In anchor-based mechanisms, the few node positions are known.
Location of unlocalized nodes are determined by the position of the
known nodes. Accuracy is highly dependent on the number of anchor
nodes.

In Anchor-free mechanism, algorithms estimate relative positions of

nodes instead of computing absolute node positions

● Centralized and Distributed

In a centralized based algorithm, one central point known as sink node
collects all the information from the nodes. This sink node or base station
calculates the position of nodes and forwards the information to others. It
consumes less energy as well as the Computation cost is less. It promotes
clustering.

In distributed based algorithms, individually the sensors calculate

and estimate their positions and they directly communicate with the sink
node or the base station. It does not promote clustering.

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● GPS Based and GPS Free
In GPS-based schemes, every node has a GPS receiver which makes it
very costly. And also the localization accuracy is very high.

GPS-free algorithms are less expensive as they do not have GPS, and
they calculate the distance between the nodes relative to the local network

● Coarse Grained and Fine Grained

For coarse-grained localization schemes, the result is achieved using
received signal strength.

For Fine-grained localization schemes, the result is achieved using

the received signal strength

● Stationary and Mobile Sensor Nodes

Localization algorithms also depend on the field of sensor nodes on
which they are deployed. Some nodes are fixed at one place i.e. they are
static in nature because many applications prefer to use static nodes.
Because of which, most of the localization algorithms are designed for
static nodes. There are few applications that use mobile sensor nodes.

7.6 CLUSTERING OF SENSOR NETWORKS

Basically, clustering means grouping. So clustering of Sensor

nodes indicates collecting the sensed data and limits the transmission of
sensed data within the cluster so as to reduce traffic and congestion in
network
For clustering the sensor nodes, we first need to discover the
neighbors by sending the Beacon signals and the cluster head (CH) is also
selected.
The major concern here is how to group neighboring Sensor
nodes and how many clusters need to be formed for optimized
performance.
One of the approaches is to partition the Wireless Sensor
Network into clusters in such a manner that all members of the clusters are
connected to the Cluster Head (CH) directly.

Figure: Clustering of SNs in WSN

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 The above figure shows randomly deployed SNs. The Sensor
nodes in a cluster can transmit data to CH directly thus reducing the
energy consumption.

The CHs also transmits information among themselves

The energy consumed in any wireless transmission is
proportional to the square of the distance between the Sensor nodes and
the Cluster head.

It can be convinient to partition SNs in a WSN into a d-cluster.

Each SN within a cluster is expected to maintain a list of all members of a
cluster.

Figure: Clustering with d clusters

It is unrealistic to assume that hundreds or thousands of Sensor

node will have the information about the whole WSN connectivity

How to select the CH of a cluster?

A simplest technique is by selecting the largest weight in the cluster.
Another technique is to use the Sensor node with the highest
degree i.e. a node having the largest number of neighbors in the cluster.
We need to select the CH calculatively because the data collected by the
cluster members is trusted with the transmission of that data. The CH does
most of the work so the CH may run out of energy. In such cases, a
dynamic change in CH must be implemented.

The CH may allot different time slots to the cluster member for
data transmission so as to avoid collision of data. So it is preferable for
each SN to use one specific channel. Sufficient number of SNs are
required to be deployed so as to monitor information of every corner of
the area.

7.7 ROUTING LAYER

The question arises that how the nodes will communicate

among themselves The routing protocols define how nodes will
communicate with each other so as to share information throughout the
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network. The routing protocols in WSN can be classified in the following
approach:

Figure: WSN Routing Protocol approaches

1. Node centric approach

In this type of approach, every destination node is identified as
a numeric identifier. Low energy adaptive clustering hierarchy (LEACH)
is one of the protocols that implements a node centric approach .

● Low energy adaptive clustering hierarchy (LEACH)

Figure: Low energy adaptive clustering hierarchy (LEACH)

Low energy adaptive clustering hierarchy (LEACH) routing

protocol first organizes the cluster sensor nodes so that the energy is
equally divided among all the sensor nodes in the network thus increasing
the lifetime of the nodes. Using the LEACH protocol, we can form
clusters such that a cluster head(CH) is selected and the remaining are
sensor nodes before the communication begins. The Cluster head (CH) is
assumed as a routing node for all the other member nodes in the specific
cluster.

In LEACH, cluster head (CH) is either selected randomly from

the cluster or some sensor nodes can volunteer themselves as a cluster
head and inform the other nodes

2. Data-centric approach

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 In some wireless sensor networks, the transmission of sensed data
is more important than how the data was collected collecting data from the
nodes.

In a data centric routing approach, the sink node instructs the

nodes to collect some specific characteristics.

Two of the protocol which follow this approach is as follows:

● Sensor protocols for information via negotiation (SPIN)

This protocol implements three messages namely
1. ADV
2. REQ
3. DATA

Figure: Working of SPIN Protocol

● The figure above shows the transmission of data using SPIN protocol.
● Firstly the node which has some information broadcasts an ADV
packet to all its neighboring nodes.
● If any node is interested in knowing the data, then that node sends an
REQ message to the advertising node.
● When the receiving node receives the REQ message from a specific
node, it sends the actual DATA to that node who has shown an
interest.
● Once the interested node receives the DATA. It further broadcast the
ADV message to its neighboring nodes. In this way, the data is
transmitted through the network.

● Directed diffusion (DD)

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 Figure: Directed diffusion

Directed diffusion is another data centric routing technique

where importance is given to the data. It uses this approach for
information gathering and transmission of data among the nodes. The
Working is quite similar to that of the SPIN protocol.

As shown in the diagram, this approach has three steps:

● Propagate interest
● Set up the route
● Send data to the interested nodes

This routing protocol provides energy saving and efficiency

thus increasing the lifetime of the network.

3. Source-initiated (Src-initiated)
In this approach, if the source node has data to share, it initiates
a route from the source to the destination node.

Source-initiated can be implemented using SPIN Protocol

4. Destination-initiated (Dst-initiated)
Mostly, the route is generated by the source node but sometimes, the
route generation is initiated by the destination node. If this has to be
achieved then there are protocols needed to set up route generation.

Directed Diffusion (DD) & LEACH are the two protocols that
implement the Destination-initiated approach.

7.8 SENSOR NETWORKS IN CONTROLLED

ENVIRONMENT & ACTUATORS

Sensor networks in controlled environments and actuators is a

group of sensors that are deployed at certain areas and gather information
about their environment and actuators, such as motors or servos that
interact with them. All elements communicate wirelessly in a human-
controlled or autonomous manner.
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 It is also referred as wireless sensor and actor networks because
there can be more than one actuator which will be involved on an actor
point. An actor point can consist of a combination of multi-geared electric
motors and servos that are arranged together to accomplish more complex
functionalities. It supports automated measurement of environmental
variables and can also have a control on various aspects of the
environment directly through autonomous or controllable sensors and
actors.

As they are built of multiple nodes whose involvement range

differs from hundreds to thousands connected with more than one sensor
with sensor hubs, individual actuators or actors.

Nowadays, It is mostly used where the measurements need to

be accurate and precise such as telemedicine, monitoring industrial
settings, entire population and scientific development.

Initially, Sensor networks in controlled environments and

actuators were deployed by government and military agencies to monitor
the persons, battlefields and other organizations and environments.

It is also contributing to the increasing trend of IoT (Internet of

Things).

7.9 REGULARLY PLACED SENSOR

A simple strategy is to place the sensors in the form of a two

dimensional grid as such a cross-point and such configuration may be very
useful for uniform coverage if the area is easily accessible and the sensor
can be placed anywhere. Such symmetric allocation of sensor nodes
allows best possible regular coverage and clustering also is very easy to
form with the neighboring SNs.

There are three samples of Sensor Networks namely rectangular,

triangular and hexagonal clustering as shown in the above figure

The first figure is of the rectangular clustering of size 5x5 where a

Sensor node is placed at each intersection of lines.

The rectangle, triangle, or hexagonal placement of the Sensor nodes

indicate the minimum sensing area that needs to be covered by each
sensor.

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Figure: Two Dimensional grid format for sensor placement

Detailed representation of Sensor nodes in three different

cluster samples, are shown in the figure above. The sensing area covered
by rectangular allocation of sensor nodes is represented in a rectangular
format, while sensing by triangular and hexagon placement is represented
as triangular and hexagonal respectively.

Figure: Placement of sensor and covered sensing range

The table above specifies the placement configuration of

sensors and sensing range covered by the sensor nodes.

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 It may be noted that the radio transmission distance between
adjacent SNs need to be such that the sensors can receive data from
adjacent sensors using wireless radio. The three placements also promote
clustering of the Sensor nodes and the size of each cluster can be fixed as
per our requirements. If the sensing and radio transmission ranges are set
to the minimum value, then all the SNs need to be active all the time to
cover the area and function properly. If range is widened , then each sub-
region will require to deploy more than one sensor node to monitor that
range and some selected Sensor nodes can be put to sleep to save energy.

Figure: Detailed representation of SNs configuration

7.10 RFID AS PASSIVE SENSORS

In this era of technology, there is an evolution of many

technologies that are providing support to the needs of the business in a
cost-effective way.

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 One of the most popular technologies is RFID or radio-
frequency identification, or RFID.
RFID sensors are categorised as active and passive.
Passive RFID systems can operate in either low frequency (LF) or
high frequency (HF) or ultra-high frequency (UHF) radio bands
RFID tags are embedded in our day-to-day applications, such as
employee badges, inventory control, retail security tags, pay terminals,
and so.

Passive RFID Tags

Passive RFID are implemented using high-power readers that transmit
low-frequency, high-power RF signals to battery-free tags. The circuit is
activated by the antenna in the tag which in turn is activated by the amount
of energy flowing to it.

Further, the reader receives a coded message by the tag at a different

frequency. Passive RFID technology is used for determining theft and
inventory tracking.

The range of Passive RFID is roughly around 1-5 meters from the
Passive RFID reader, so a large number of readers will be required to track
the location of an item.

How does Passive RFID work?

The RFID system is made up of three parts namely – an RFID reader
or interrogator, an RFID antenna, and RFID tags. But passive RFID differs
slightly which includes two main components namely the tag's antenna
and the integrated circuit (IC) or microchip.

Initially, the Passive tags wait for a signal from an RFID reader. The
reader then sends energy to an antenna which converts that energy into an
Radio Frequency wave that is sent into the read zone. The RFID tag’s
internal antenna draws in energy from the RF waves, once the tag is read
within the read zone. The energy is then moved to the Integrated Circuit
(IC) from the tag’s antenna and powers the chip which generates a signal
back to the RF system. This is known as backscatter. The backscatter is
nothing but a change in the electromagnetic or Radio Frequency wave,
which is detected by the reader through the antenna which interprets the
information.

The most basic structure is referred to as an RFID inlay.

There are many other types of passive RFID but they are categorized
into two types-
1. hard tags
2. inlays tags.

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Hard RFID tags -
Hard RFID tags are made of plastic, metal, ceramic and even rubber
and are durable. They also come in all kinds of shapes and sizes and are
designed for a unique function, or application.

Some of these tags will be supported within two or more groups.

● High Temperature – Some specific passive RFID tags are designed
to resist extreme temperatures and accommodate different types of
applications among themselves.
● Rugged – In outdoor environments, applications require a tag that can
monitor dust, snow, ice, and debris in the environment.
● Size – When choosing an RFID tag, size is one of the major
considerations. Applications tracking small or large items have
specific size constraints when tracking small or large items.
● Materials – UHF metal-mount tags are used if an application requires
tracking metal assets as these tags reduce the problems faced by UHF
RFID around metal.
● Embeddable – An embeddable tags can be fit in small crevices and be
covered to safeguard the RFID tag from harm.

Figure: A roll of Passive RFID inlays

Inlays tags are the RFID tags having high volumes, but of low cost.
These inlays are mainly grouped into three types:
1. Dry Inlays – An antenna and RFID microchip (IC) is attached to a
material or a layer called a web. These inlays look like they have been
glazed (coated) with no adhesive.
2. Wet Inlays – A RFID microchip and an antenna is attached to a
substrate, usually PVT or PET, with an adhesive. These inlays can
then be peeled off from their roll and stuck on an item.
3. Paper Face Tags – These tags are basically wet inlays with a poly
face or a white paper. These tags are used for applications that need
printed logos or numbers.
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All Passive RFID tags do not operate at the same frequency.
Passive RFID tags operate at three different frequencies depending on
parameters such as the attachment materials, read range, and the
application options.

● Low Frequency (LF) : 125KHz to 134 KHz –

A Long wavelength with a short read range of about 1 - 10 cms.
As it is not affected much by water or metal, this frequency is used with
animal tracking.

● High Frequency (HF): 13.56 MHz –

Also called Near-Field Communication (NFC)
A medium wavelength with a medium read range of about 1 cm to 1m.
Data transmissions, DVD kiosks, access control applications, and passport
security are done using this type of frequency.

● Ultra High Frequency (UHF): 865 - 960 MHz –

A high-energy wavelength beyond 1m having a long read
range. Passive UHF tags can be read from an average distance of about 5 -
6 meters, but larger UHF tags can achieve up to 30+ meters of read range
in ideal conditions. Race timing, file tracking, IT asset tracking, file
tracking, and so on uses this type of frequency which needs more distance
of read range.

7.11 UNIT END QUESTIONS

1. What is WSN? What are the applications of WSN?

2. State and explain the design consideration in SNs?
3. What is the sensing and communication range between two SNs?
4. What do you mean by localization?
5. How is clustering done in WSN? How is the Cluster Head selected?
6. Explain RFID as a passive sensor



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