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CS1302 COMPUTER NETWORKS

UNIT – I
DATA COMMUNICATION

Need for Communication

A communication service enables the exchange of information between users at

different locations.
Communication services & applications are everywhere. Some examples are given below

Need for Computer Communication over Distances

Computer communication has become essential for the following reasons:

(a) Computers can send data at a very fast speed over long distances using satellite and
microwave links. Therefore, the cost of transfer of documents using computers is cheaper than
Other conventional means like telegrams.
(b) Computers can have fax system integrated with them. This allows them to send pictures along
with the text. Thus the newspaper reports can be prepared and sent all over the world at a very
high speed by composing and publishing them from different centers.
(c) The time taken to transmit the messages is very small. Hence different computers can be
connected together and the users can work together as a group. Software packages have
been developed for group working in Data Base Management (DBMS) and graphic works.
(d) Different departments of an organization may be separated physically being at distant places
but their data could be stored on a central computer. This data is accessed by computers

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located in different departments. The data at the central computer may be updated from time to
time and accessed by all users. This prevents any bottlenecks in the smooth functioning of the
organization. The latest data (say for inventory) will be easily available at all times to all the
users.
(e) Fluctuations of prices in foreign exchange and shares/equities can be communicated
instantaneously
using the medium of computer communications only. The transfer can be accelerated and verified
at any instant of time.

1.1 Data Communication:

Data Communication is defined as the exchange of data between two devices via some
form of transmission medium such as a wire cable. The communicating devices must be a
part of a communication system made up of a combination of hardware ( physical
equipment) and software (programs).

Characteristics of data Communication:

The effectiveness of a data communication depends on three characteristics
1.Delivery
2.Accuracy
3.Timeliness

Delivery : The system must deliver data to correct destination.

Accuracy: The system must deliver data accurately.

Timeliness: The system must deliver data in a timely manner. Data delivered late are
useless. Timely delivery means delivering data as they are produced, in the same order
that they are produced. and without significant delay. This kind of delivery is called real
–time transmission.

Components:
The components of a data communication are

Message
Sender
Receiver
Medium
Protocol

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Message : The message is the information to be communicated. It can consist of text

,pictures, numbers, sound, video or audio .
Sender. The sender is the device that sends the data message. It can be a computer or
workstation telephone handset, video camera and so on..
Receiver. The receiver is the device that receives the message. It can be a computer or
workstation telephone handset, video camera and so on..
Medium. The transmission medium is the physical path by which a message travels from
sender to receiver. It could be a twisted pair wire , coaxial cable, fiber optic cable, or
radio waves.
Protocol. A protocol is a set of rules that governs data communications. It represents an
agreement between the communicating devices.

Data representation.
Information comes in different forms such as text, numbers, images, audio and video.

Text.
Text is represented as a bit pattern,
The number of bits in a pattern depends on the number of symbols in the language.
Different sets of bit patterns have been designed to represent text symbols. Each set is
called a code. The process of representing the symbols is called coding.
ASCII : The American National Standards Institute developed a code called the
American Standard code for Information Interchange .This code uses 7 bits for each
symbol.
Extended ASCII : To make the size of each pattern 1 byte(8 bits),the ASCII bit patterns
are augmented with an extra 0 at the left.
Unicode : To represent symbols belonging to languages other than English,a code with
much greater capacity is needed. Unicode uses 16 bits and can represent up to 65,536
symbols.
ISO:The international organization for standardization known as ISO has designed a
code using a 32 – bit pattern. This code can represent up to 4,294,967,296 symbols.
Numbers
Numbers are also represented by using bit patterns. ASCII is not used to represent
numbers. The number is directly converted to a binary number.
Images
Images are also represented by bit patterns. An image is divided into a matrix of
pixels,where each pixel is a small dot. Each pixel is assigned a bit pattern. The size and
value of the pattern depends on the image.The size of the pixel depends on what is called
the resolution.
Audio
Audio is a representation of sound. Audio is by nature different from text, numbers or
images. It is continuous not discrete
Video
Video can be produced either a continuous entity or it can be a combination of images.

Direction of data flow

Communication between two devices can be simplex, half-duplex or full-duplex

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Simplex
In simplex
mode, the
communication
is unidirectional.
Only one of the
devices on a link
can transmit; the
other can only
receive.
Ex.Keyboards
and monitors.
The keyboard
can only
introduce input.
The monitor can
only accept output.

Half-duplex
In half-duplex mode ,each station can both transmit and receive but not at the same time.
When one device is sending ,the other can only receive.
The half-duplex mode is like a one-lane road with two directional traffic. The entire
capacity of a channel is taken over by whichever of the two devices is transmitting at the
time.
Ex. Walkie-talkies and CB(citizen band radios.

Full-duplex
In full-duplex mode, both stations can transmit and receive simultaneously. It is like a
two-way street with traffic flowing in both directions at the same time. Signals going in
either direction share the capacity of the link.
Ex. Telephone network
When two people are communicating by a telephone line, both can listen and talk at the
same time.

1.2 Network:
Definition :

An interconnected collection of autonomous computers‖ interconnected = able to

exchange information

A set of nodes connected by communication links .A node can be any device capable of
sending &/or receiving data to &/or from other nodes in the network

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A connected collection of hardware and software that permits information exchange and
resource sharing.
Information = data, text, audio, video, images, …
Resources = printers, memory, link bandwidth

Uses of networks

•companies & organizations •resource sharing: programs, equipment, data…

•high reliability: multiple processors/links/file copies/...
•scalability: gradually improve system performance
•rapid communications & remote cooperation •saving money
•access to remote & diverse information sources •communicating with other people
•entertainment •education, healthcare, access to government...

Distributed Processing

Networks use distributed processing which is termed as a task divided among multiple
computers. Instead of a single machine responsible for all aspects of a process, separate
computers handle a subset.

Network Criteria
Performance
Performance can be measured by means of transit time, response time, number of users,
type of transmission medium, and capabilities of the connected hardware and the
efficiency of the software.
Transit time The amount of time required for a message to travel from one device to
another.
Response time: The elapsed time between an inquiry and a response.
Reliability: Reliability is measured by the frequency of failure ,the time it takes a link to
recover from a failure.
Security: Network security is protecting data from unauthorized access.

Physical Structures
Type of connection
There are two possible type of connections
Point-to-point
Multipoint
A point-to-point connection provides a dedicated link between two devices. The entire
link is reserved for transmission between those two devices.
Ex. Change of television channel by infrared remote control. A point-to-point connection
is established between the remote control and the televisions control system.
A multipoint (also called multidrop) connection is one in which more than two specific
devices share a single link. The capacity of the channel is shared either spatially or
temporally.

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Physical Topology
Physical Topology refers to the way in which network is laid out physically. Two or more
links form a topology. The topology of a network is the geometric representation of the
relationship of all the links and the linking devices to one another.
The basic topologies are
Mesh
Star
Bus and
Ring

Mesh
In a mesh topology each device has a dedicated point to point link to every other device.
The term dedicated means that the link carries traffic only between the two devices it
connects.
A fully connected mash network therefore has n(n-1)/2 physical channels to link n
devices. To accommodate that many links every device on the network has (n-1) I/O
ports.

Merits.

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Dedicated link guarantees that each connection can carry its own data load. This
eliminates the traffic problems that occur when links shared by multiple devices.
If one link becomes unusable, it does not incapacitate the entire system.
Privacy or security: When every message travels along a dedicated line only the
intended recipient

Demerits

The amount of cabling and the I/O ports required

Installation and reconnection are difficult
The sheer bulk of the wires accommodate more space than available.

The hardware required to connect each link can be prohibitively expensive.

(Mesh)

Star topology

Each device has a dedicated point to point link only to a central controller usually called a
hub. If one device has to send data to another it sends the data to the controller, which
then relays the data to the other
connected device.

Merits

Less expensive than a mesh

topology. Each device needs
only one link and I/O port to
connect it to any number of
others.
Installation and reconfigure is
easy.
Robustness. If one link fails
only that link is affected.
Requires less cable than a mesh.

Demerits
Require more cable compared to bus and ring topologies.

Bus

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One long cable acts as a backbone to link all the devices in a network Nodes are
connected to the bus cable by drop lines and taps. A drop line is a connection running
between the device and the main cable. A tap is a connector that either splices into the
main cable or punctures the sheathing of a cable to create a contact with a metallic core.
As the signal travels farther and farther, it becomes weaker .So there is limitation in the
number of taps a bus can support and on the distance between those taps.

Merits Demerits

Ease of installation. Difficult reconnection and isolation.

Bus uses less cabling than mesh or Signal reflection at the taps can cause
star topologies. degradation in quality.

A fault or break in the bus cable stops all

transmission. It also reflects signals back in
the direction of origin creating noise in both
directions.
Ring

Each device has a dedicated point to point connection only with the two devices on either
side of it.

A signal is passed along the ring in one direction from device to device until it reaches
the destination
Each device in the ring incorporates a repeater. It regenerates the bits and passes them
along, when it receives the signal intended for another device.

Merits:

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Easy to install and reconfigure.

To add or delete a device requires changing only two connections.
The constraints are maximum ring length and the number of devices.
If one device does not receive the signal within a specified period, it issue an
alarm that alerts the network operator to the problem and its location

Demerits
A break in the ring disables the entire network. It can be solved by using a dual ring
or a switch capable of closing off the break.

Networks Models
Categories of Network

The three primary categories are of network are Local Area Network (LAN),
Metropolitan Area Network (MAN), and Wide Area Network (WAN). The category into
which a network fall is determined by its size, ownership, the distance it covers and its
physical architecture.

LAN

A LAN is usually privately owned

and links the devices in a single office,
building or campus.
A LAN can be as simple as two PCs or
it can extend throughout a company.
LAN size is limited to a few kilometers.
The most widely used LAN system is
the Ethernet system developed by the
Xerox Corporation.
It is designed to allow resources
(hardware, software or data) to be
shared between PC’s or workstations.

It may be used to provide a (shared) access to remote organizations through a router

connected to a Metropolitan Area Network (MAN) or a Wide Area Network (WAN)
One of the computers may be given a large capacity disk drive and may become a
server to other clients. Software can be stored on this server and used by the whole
group.
The size of the LAN may be determined by the licensing restrictions on the
numbers per copy of software or the number of users licensed to access the
operating system.
Also differentiated from other types of network by transmission media and
topology.LAN use only one type of transmission medium. The common LAN
topologies are bus, ring and star.

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LANs have data rates in the 4 to 10 megabits per second. Can also reach 100
Mbps with gigabit systems in development.
Intermediate nodes (i.e. repeaters, bridges and switches) allow LANs to be
connected together to form larger LANs. A LAN may also be connected to another
LAN or to WANs and MAN’s using a "router‖

Metropolitan Area Network

A MAN is designed to extend over an entire city.
May be a single network such as cable TV network
May be a means of connecting a number of LANs into a larger network
Resources may be shared LAN to LAN as well as device to device
Example: Company can use a MAN to connect the LANs in all its offices throughout
a city.
A MAN can be owned by a private company or it may be a service provided by a
public company ,such as local telephone company
Telephone companies provide a popular MAN service called (SMDS) Switched
Multi-megabit Data Services.

Wide Area Network (WAN)

A WAN provides long distance transmission of data, voice, image and video
information over large geographic areas.
It may comprise a country, continent or even the whole world. Transmission rates
are typically 2 Mbps, 34 Mbps, 45 Mbps, 155 Mbps, 625 Mbps (or sometimes
considerably more).
WAN utilize public, leased, or private communication equipment usually in
combinations and therefore span an unlimited number of miles.
A WAN that is wholly owned and used by a single company is referred to as an
Enterprise Network. The figure represents the comparison of the different types of
networks

Internetwork
When two or more networks are connected they become an internetwork or internet

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1.3 Protocols
A protocol is a set of rules that governs data communication. It defines what is
communicated, how it is communicated, and when it is communicated. The key elements
of a protocol are syntax, semantics and timing
Syntax: It refers to the structure or format of the data. This refers the order in which the
data are presented.
Example
The first 8 bits of data to be the address of the sender.
The second 8 bits to be the address of the receiver.
The rest of the stream may be the message itself
Semantics: It refers to the meaning of each section of bits. How a particular pattern to be
interpreted? What action is to be taken based on that interpretation?
Example
An address specifies the route to be taken or the final destination of the
message.
Timing: It refers to two characteristics
When data should be sent and how fast they can be sent.
Example
If a sender produces data at 100 Mbps and the receiver process data at only
1 Mbps, it will overload the receiver and data will be lost.

Standards
Why do we need standards?
To create and maintain an open and competitive market for equipment
manufacturers
To guarantee national and international interoperability of data,
telecommunication technology and process
To give a fixed quality and product to the customer
To allow the same product to be re used again elsewhere
To aid the design and implementation of ideas

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To provide guidelines to manufacturers, vendors, government agencies and other

service providers to ensure kind of interconnectivity.

Data communication standards are divided into two categories

De facto(from the fact):
Standards that have not been approved by an organized body.
It have been adopted as standards through widespread use.
This is often established originally by manufacturers to define the functionality of
a new product or technology.
De jure (by law):
Those that have been legislated by an officially recognized body.

Standards organizations
Standards are developed through the cooperation of standards creation committees,
forums, and government regulatory agencies.

Standards Creation Committees

ITU - International Telecommunications Union formerly the (CCITT):
It a standard for telecommunication in general and data systems in particular.
ISO - International Standards Organization:
It is active in developing cooperation in the realms of scientific, technological and
economic activity.
ANSI - American National Standards Institute:
It is a private nonprofit corporation and affiliated with the U.S federal
government.
IEEE - Institute of Electrical and Electronics Engineers:
It aims to advance theory, creativity, and product quality in the fields of electrical
engineering , electronics radio and in all related branches of Engineering.
It oversees the development and adoption of international standards for computing
and communications.
EIA - Electronic Industries Association:
It is a nonprofit organization devoted to the promotion of electronics
manufacturing concerns.
Its activities include public awareness education and lobbying efforts in addition
to standards development.
It also made significant contributions by defining physical connection interfaces
and electronic signaling specifications for data communication.
Forums
It work with universities and users to test, evaluate ,and standardize new
technologies.
The forums are able to speed acceptance and use of those technologies in the
telecommunications community.
It presents their conclusions to standard bodies.
Regulatory Agencies:

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Its purpose is to protect the public interest by regulating radio, television and wire
cable communications.
It has authority over interstate and international commerce as it relates to
communication.
Internet Standards
It is a thoroughly tested specification that is useful to and adhered to by those who
work with the internet.
It is a formalized regulation that must be followed.
A specification begins as an internet draft and attains Internet standard status.
An Internet draft is a working document and it may be published as Request for
Comment (RFC).RFC is edited, assigned a number, and made available to all
interested parties.

2.2OSI
The Open Systems Interconnection (OSI) architecture has been developed by the
International Organization for Standardization (ISO) to describe the operation and design
of layered protocol architectures. This forms a valuable reference model and defines
much of the language used in data communications.

Layered Architecture

>Each layer provides services to the next higher

layer and shields the upper layer from the details
implemented in the lower layers.

>Each layer appears to be in direct (virtual)

communication with its associated layer on the
other computer. Actual communication between
adjacent layers takes place on one computer only.

>Layering simplifies design, implementation, and

testing. Only the lowest level (Physical layer) can
directly communicate with its peer
communications process into parts.

Peer-to-Peer Processes

The processes on each machine that communicate at a given layer are called peer-
to-peer processes.
At higher layers communication must move down through the layers on device A
aver to device B and then back up through the layers.

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Each layer in the sending device adds its own information to the message it
receives from the layer just above it and passes the whole package to the layer
just below and transferred to the receiving device.

Interfaces between layers

The passing of data and network information down through the layers of the
sending device and back up through the layers of the receiving device is made
possible by an interface between each pair of adjacent layers.
Each interface defines what information and services a layer must provide for
the layer above it.
Well defined interfaces and functions provide modularity to a network.

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2.3 Layers in the OSI model

Physical Layer
It coordinates the functions required to transmit a bit stream over a physical
medium.
It deals with the mechanical and electrical specifications of the interface and
transmission media.
Mechanical: cable, plugs, pins...
Electrical/optical: modulation, signal strength, voltage levels, bit times
It also defines the procedures and functions that physical devices and interfaces
have to perform for transmission to occur.

(Information flows from top to bottom at the sender and bottom to top at the
receiver.)

Major responsibilities of Physical layer are

Physical characteristics of interfaces and media: It defines the characteristics
of the interface between the devices and the transmission media. Also defines the
type of transmission medium.
Representation of bits: To transmit the bits, it must be encoded into electrical or
optical signals. It defines the type of representation how 0s and 1s are
changed to signals.
Data rate: The number of bits sent each second is also defined by the physical
layer.
Synchronization of bits: Sender and the receiver must be synchronized at the bit
level .i.e the sender and the receiver clocks must be synchronized.

Data link layer

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The data link layer is responsible for hop-to-hop (node-to-node) delivery. It transforms
the physical layer a raw transmission facility to a reliable link. It makes physical layer
appear error free to the network layer.

The duties of the data link layer are

Framing: The data link layer divides the stream of bits received from the network
layer into manageable data units called frames.
Physical Addressing: If the frames are to be distributed to different systems on
the network the data link layer adds a header to the frame to define the receiver or
sender of the frame. If the frame is intended for a system located outside the
senders network then the receiver address is the address of the connecting device
that connects the network to the next one.
Flow Control: If the rate at which the data absorbed by the receiver is less than
the rate produced in the sender, the data link layer imposes a flow control
mechanism to overwhelming the receiver.
Error control Reliability is added to the physical layer by data link layer to
detect and retransmit loss or damaged frames. and also to prevent duplication of
frames. This is achieved through a trailer added to the end of the frame
Access control When two or more devices are connected to the same link it
determines which device has control over the link at any given time.

Network Layer
The network layer is responsible for source-to-destination delivery of a packet across
multiple networks. It ensures that each packet gets from its point of origin to its final
destination. It does not recognize any relationship between those packets. It treats each
one independently as though each belong to separate message.

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The functions of the network layer are

Logical Addressing If a packet has to cross the network boundary then the
header contains information of the logical addresses of the sender and the receiver.
Routing when independent networks or links are connected to create an
internetwork or a large network the connective devices route the packet to the final
destination.

Transport Layer
The network layer is responsible for process-to-process delivery that is source to
destination delivery of the entire message.

The responsibilities of Transport layer are

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Service-point (port) addressing: Computers run several programs at the same

time. Source-to-destination delivery means delivery from a specific process on one
computer to a specific process on the other. The transport layer header therefore
includes a type of address called a service – point address.
Segmentation and reassembly: A message is divided into segments and each
segment contains a sequence number. These numbers enable the Transport layer to
reassemble the message correctly upon arriving at the destination. The packets lost in
the transmission is identified and replaced.
Connection control: The transport layer can be either connectionless or
connection-oriented. A connectionless transport layer treats segment as an independent
packet and delivers it to the transport layer. A connection-oriented transport layer
makes a connection with the transport layer at the destination machine and delivers the
packets. After all the data are transferred the connection is terminated.
Flow control: Flow control at this layer is performed end to end .
Error Control: Error control is performed end to end. At the sending side ,the
transport layer makes sure that the entire message arrives at the receiving transport
layer with out error. Error correction is achieved through retransmission.

Session Layer
Session layer is the network dialog controller. It establishes, maintains, and
synchronizes the interaction between communicating systems.

Specific responsibilities of the layer are

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Dialog Control: Session layer allows two systems to enter in to a dialog.

Communication between two processes takes place either in half-duplex or full-duplex.
Example: The dialog between terminals connected to a mainframe. Can be half-duplex.
Synchronization. The session layer allows a process to add checkpoints into a
stream of data. Example If a system is sending a file of 2000 pages, check points may
be inserted after every 100 pages to ensure that each 100 page unit is advised and
acknowledged independently. So if a crash happens during the transmission of page
523, retransmission begins at page 501, pages 1 to 500 need not be retransmitted.

Presentation layer

It is concerned with the syntax and semantics of the information exchanged between
two systems.

Responsibilities of the presentation layer are

Translation: The processes in two systems are usually exchanging information
in the form of character strings, numbers, and so on. The since different computers use
different encoding systems, the presentation layer is responsible for interoperability
between these different encoding methods. At the sender, the presentation layer
changes the information from its sender-dependent format into a common format. The
presentation layer at the receiving machine changes the common format into its
receiver dependent format.
Encryption: The sender transforms the original information from to another
form and sends the resulting message over the entire network. Decryption reverses the
original process to transform the message back to its original form.
Compression: It reduces the number of bits to be transmitted. It is important in
the transmission of text, audio and video.

Application Layer
It enables the user (human/software) to access the network. It provides user interfaces
and support for services such as electronic mail, remote file access and transfer, shared
database management and other types of distributed information services.

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Services provided by the application layer are

Network Virtual terminal: It is a software version of a physical terminal and
allows a user to log on to a remote host.
File transfer, access and management. This application allows a user to
access files in a remote computer, to retrieve files from a remote computer and to
manage or control files in a remote computer.
Mail services. This application provides the basis for e-mail forwarding and
storage.
Directory services. It provides distributed database sources and access for
global information about various objects and services.

Summary of layers

7 Transmission Media
Transmission media are actually located below the physical layer and directly
controlled by the physical layer.
Transmission media can be divided into two broad categories
Guided &
Unguided

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Guided media: It includes twisted-pair cable, coaxial cable, and fiber-optic cable
Unguided media: It is usually air.

7.1 Guided media

Guided media, which are those that provide a conduit from one device to another,
include twisted-pair cable, coaxial cable, and fiber-optical cable.

Coaxial cable
Coaxial cable carries
signals of higher
frequency ranges
than twisted pair
cable.

It has a central core conductor of solid or stranded wire enclosed in an insulating

sheath. This in turn encased in an outer conductor of metal foil, braid or a
combination of the two.
The metallic wrapping serves both as a shield against noise and as the second
conductor completes the circuit.
The outer conductor is also enclosed in an insulating sheath and the whole cable is
protected by a plastic cover.

Coaxial cable Standards

Coaxial cables are categorized by their radio government (RG) ratings.Each RG
number denotes a set of physical specifications such as,
wire gauge of the inner conductor

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thickness and type of the inner insulator

the construction of the shield
the size and type of outer casing
Categories of coaxial cables

Category Impedance Use

RG-59 75 Cable TV
RG-58 50 Thin Ethernet
RG-11 50 Thick Ethernet

Coaxial Cable Connectors

Coaxial Cable Connectors are used to connect coaxial cable to devices. The most
common type of connector is the Bayone Neill-concelman or BNC connectors. There
are three popular types of connectors
BNC connector
BNC T connector &
BNC terminator

BNC connector BNC T connector

It is used to connect the end of the cable It is used in Ethernet networks to branch out a
to a device such as a TV set. cable for connection to a computer or other
devices.
BNC terminator
It is used at the end of the cable to prevent the reflection of the signal.
Performance
Attenuation is much higher in coaxial cables than in twisted pair cable.
Coaxial cable has a much higher bandwidth the signal weakens rapidly and needs the
frequent use of repeaters.

Basic definitions

Signal Attenuation is the phenomenon whereby the amplitude of a signal

decreases as it propagates along a transmission line.
Attenuation is a function of distance and frequency of signal
Repeaters are used to increase the power of the signal at appropriate intervals
Skin effect, which increases attenuation as the bit rate of the transmitted signal
increases

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Applications
Coaxial cable is used in analog telephone network where a single coaxial
cable could carry 10,000 voice signals.
It is also used in digital telephone network where a cable could carry
digital data up to 600 Mbps.
Cable TV networks also used RG-59 coaxial cables.
It is also used in traditional Ethernets.

Fiber Optic Cable.

A fiber optic cable is made of glass or plastic and transmits signals in the form of light.

Properties of light
Light travels in a straight line as long as it moves through a single uniform
substance. If array traveling through one substance suddenly enters another the ray
changes direction.

Refraction:
If the angle of incidence (the angle the ray makes with the line perpendicular to the
interface between the two substances) is less than the critical angle the ray refracts and
moves closer to the surface.
Reflection:
If the angle of incidence is greater than the critical angle the ray reflects and travels
again in the denser substance.

Optical fibers use reflection to guide light through a channel.

A glass or plastic core is surrounded by a cladding of less dense glass or plastic. The
difference in the density of the two materials must be such that a beam of light moving
through the core is reflected off the cladding.

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Propagation Modes

There are two modes for propagating light along optical channels; each requires fiber
with different physical characteristics
Multimode
Single mode
Multimode
Multiple beams from a light source move through the core in different paths.
Multimode can be implemented in two forms
Step-index
Graded index
Multimode Step –index fiber
In Multimode Step –index fiber
the density of the fiber remains
constant from the center to the
edges
A beam of light moves through
this constant density in a
straight line.
When it reaches the interface of
the core and the cladding, there
is an abrupt change to a lower
density that alters the angle of
the beams motion.
Step-index -> the suddenness of
this change.

Multimode Graded-index fiber

It decreases the distortion of the
signal through the cable.
Density is highest at the center
of the core and decreases
gradually to its lowest at the
edge.
Single-Mode
It uses step-index fiber and a highly focused source of light that limits beams to a
small range of angles, all close to the horizontal

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The Single-Mode fiber itself is manufactured with a smaller diameter than that of
multimode fiber and with lower density.
This results in a critical angle that is close enough to 90. To make it horizontal.
All the beams arrive at the destination together and can be recombined with little
distortion to the signal.

Fiber Sizes
Optical fibers are defined by the ratio of the diameter of their core to the diameter of their
cladding expressed in micrometers.

Type Core Cladding Mode

50/125 50 125 Multimode, graded-index
62.5/125 62.5 125 Multimode, graded-index
100/125 100 125 Multimode, graded-index
7/125 7 125 Single-mode

Fiber-optic cable connectors.

Fiber construction Fiber-optic cable connectors

Three different types of connectors are used by fiber –optic cable.

SC ( subscriber channel) Connector:
It is used in cable TV.
ST(Straight-tip) Connector:
It is used for connecting cable to networking devices .
Performance :
Attenuation is flatter than in the case of twisted pair cable and coaxial cable.
Few repeaters are needed when we use fiber optic cable.
Application
It is used in cable TV and LAN (Fast Ethernet and 100Base –X.
Advantages
Higher bandwidth: It can support higher bandwidth than twisted pair or coaxial cable.

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Less signal attenuation: Transmission distance is greater than that of other guided
media. Signals can be transmitted for 50 km without requiring regeneration.
Immunity to electromagnetic Interference : Electromagnetic noise can not affect fiber-
optic cables
Resistance to corrosive materials: glass is more resistant to corrosive materials.
Light-weight: It is of less weight than the copper cables.
More Immune to taping: Fiber-optic cables are more immune to taping than copper
cables.
Disadvantages :
Installation/Maintenance. Installation/Maintenance need expertise since it is a new
technology.
Unidirectional: Propagation of light is unidirectional. Bidirectional communication is
achieved by means of two optical fibers.
Cost: It is more expensive and the use of optical fiber cannot be justified if the need for
bandwidth is not high.

4.1 Line Coding

Line Coding is the process of converting binary data , a sequence of bits , to a digital
signal.

Characteristics of Line coding

Some characteristics of line coding are
Signal level versus data level
Pulse rate vs. bit rate
Dc components and
Self-synchronization
Signal level: The number of values allowed in a particular signal is termed as signal
level.
Data level: The number of values used to represent data are termed as data level.
DC component (zero frequency):
If the positive voltages are not get cancelled by the negative voltages then it is called a dc
component. This component is undesirable for 2 reasons They are
If the signal is to pass through a system that does not allow the passage of a dc
component, the signal is distorted and may create errors in the output.
This component is an extra energy residing on the line and is useless.

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Self-synchronization:
Need: To correctly interpret the signals received from the sender, the receivers bit
intervals must correspond exactly to the senders bit intervals. If the receiver clock is
faster or slower, the bit intervals are not matched and the receiver might interpret the
signals differently than the sender intended.

A Self-synchronizing digital signal includes timing information in the data being

transmitted. This can be achieved if there are transitions in the signal that alert the
receiver to the beginning, middle or end of the pulse. If the receiver’s clock is out of
synchronization, these alerting points can reset the clock

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Line coding schemes(digital to digital transmission)

Line coding schemes are divided in to three categories.
1. Unipolar
2. Polar
3. Bipolar

Unipolar
Unipolar encoding uses only one polarity.0 is represented by zero voltage and 1 is
represented by positive voltage.It are inexpensive to implement. Unipolar encoding has
two problems
1. Lack of synchronization
2. A dc component

Polar encoding:
It uses two voltage levels
1. Positive
2. Negative
The types of polar encoding are
1. Non return to zero(NRZ)
2. Return to zero(RZ)
3. Biphase

NRZ

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The level of the signal is always either positive or negative.

NRZ-L
The level of the signal depends on the type of bit it represents.
The bit 0 is represented by positive voltage
The bit 1 is represented by negative voltage.
Demerits
Problem arises when there is a long stream of 0s or 1s in the data.
If the receiver receives a continuous voltage ,it should determine how many bits are sent
by relying on its clock.
The receiver may or may not be synchronized with the sender clock

NRZ-I
The 1 bit is represented by an inversion (transition between a positive and a negative
voltage) of the voltage level.
The existence of 1’s in the data stream allows the receiver to resynchronize its timer to
the actual arrival of the transmission.
A string of 0’s can still cause problems.

RZ
It uses three values
Positive
Negative &
Zero
In RZ the signal changes during each bit.
1. A 1 bit is actually represented by positive-to-zero and
2. A 0 bit is actually represented by negative-to-zero
Demerits
It requires two signal changes to encode one bit.
It occupies more bandwidth.

Biphase
The signal changes at the middle of the bit interval and does not return to zero.
There are two types of biphase encoding
Manchester
Differential Manchester

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Manchester
It uses the inversion at the middle of each bit interval for both synchronization
and bit representation.
The bit 1 is represented by negative -to-positive transition.
The bit 0 is represented by positive-to-negative transition.
Merits
A single transition achieves the level of synchronization but with only two levels of
amplitude

Differential Manchester
Inversion at the middle of the bit interval is used for synchronization.
Presence or absence of additional transition at the beginning of the interval is used to
identify the bit.
A bit 0 is represented by a transition.
A bit 1 means no transition.
It requires two signal changes to represent binary 0,but only one to represent binary 1.

Bipolar
It uses three voltage levels
Positive

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Negative and
Zero
The bit 0 is represented by zero level
The 1s are represented by alternate positive and negative voltages.If the first 1 bit
is represented by positive amplitude,the second will be represented by the
negative amplitude, and so on.
There are three types of bipolar encoding
1. AMI
2. B8ZS
3. HDB3
Bipolar Alternate Mark Inversion
A binary 0 is represented by zero voltage.
A binary 1s are represented by alternate positive and negative voltages.
Merits
By inverting on each occurrence of 1,
The dc component is zero
A long sequence of 1s stays synchronized.
Pseudoternary
A binary 0 alternate between positive and negative voltages.

Comparison
The comparison of the different encoding schemes of the following NRZ, Polar
NRZ,NRZ Inverted, Bipolar, Manchester, Differential Manchester are given.

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MODEMS

The term modem is a composite word that refers to the two functional entities that make
up the device; a signal modulator and a signal demodulator. A modulator creates a band-
pass analog signal from binary data. A demodulator recovers the binary data from the
modulated signal.

Modem stands for modulator and demodulator.

TELEPHONE MODEMS

Traditional telephone lines can carry frequencies between 300 and 3300 HZ, giving them
BW of 3000 Hz; All this range is used for transmitting voice, where a great deal of
interference and distortion can be accepted without loss of intelligibility.

The effective BW of a telephone line being used for data Transmission is 2400 Hz,
covering the range from 600 to 3000 Hz.

Figure shows the relationship of modems to a communication link. The computer on the
left sends binary data to the modulator portion of the modem; the data is sent as an analog
signal on the telephone lines. The modem on the right receives the analog signal,
demodulates it through its demodulator, and delivers data to the computer on the right.

The communication can be bidirectional, which means the computer on the right can also
send data to the computer on the left using the same modulation and demodulation
processes.
Modem standards
V-series standards published by the ITU-T.
V.32 V.32bis V.34bis

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V.90 V.92
V.32
This modem uses a combined modulation and demodulation encoding technique
called trellis-coded modulation. Trellis is essentially QAM plus a redundant bit. The Data
stream is divided into 4-bit sections. Instead of a quad bit, however, a pentabit is
transmitted. The value of the extra bit is calculated from the values of the data bits.
In any QAM system, the receiver compares each received signal point to all valid
points in the constellation and selects the closest point as the intended value. A signal
distorted by transmission noise can arrive closer in value to an adjacent point than to the
intended point, resulting in a misidentification of the point and an error in the received
data.
By adding a redundant bit to each quad bit, trellis-coded modulation increases the
amount of information used to identify each bit pattern thereby reduces the number of
possible matches.
The V.32 calls for 32-QAM with a baud rate of 2400. Because only 4 bits of each
pentabit represents data, the resulting speed is 4*2400=9600.
V.32 bis
The V.32 bis modem support 14,400-bps transmission. The V.32 uses
128-QAM transmission.
V.34 bis
The V.34 bis modem support 28,800-bps transmission with a 960-point
constellation to a bit rate of 33,600 with a 1664-point constellation.

V.90
Traditional modems have a limitations on the data rate.V.90 modems with
a bit rate of 56,000 bps, called 56Kmodems, are available. Downloading rate is 56K,
while the uploading rate is a maximum of 33.6 kbps.

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After modulation by the modem, an analog signal reaches the telephone company
switching station. Where it is sampled and digitized to be passed through the digital
network. The quantization noise introduced in the signal at the sampling point limits the
data rate according to the capacity. This limit is 33.6 Kbps.
V.92
The standard above V.92 is called V.92. These modems can adjust their speed,
and if the noise allows, they can upload data at the rate of 48 Kbps. The modem has
additional features. For example, the modem can interrupt the internet connection when
there is an incoming call if the line has call-waiting service.

RS 232 INTERFACE

RS 232 is a standard interface by EIA and RS232C is the latest version of this
interface.
INTERFACING WITH RS232

It expects a modem to be connected to both receiving and transmitting end.

The modem is termed as DCE (Data Communication Equipment) And the
computer with which modem is interfaced is called DTE (Data Terminal
Equipment).
The DCE and DTE are linked via a cable whose length does not exceed 50 feet.
The DTE has 35 pins male connector and DCE has 25 pins Female connector.

FEATURES OF RS232 INTERFACE

1. RS232 Signal LEVEL

RS232 standard follows –ve logic, Logic1 is represented by negative
voltage. Logic0 is represented by +ve voltage.
Level 1 varies from -3 to -15v and level 0 varies from 3 to 15v

2. RS232 SIGNALS

SL NO PIN NUMBER SIGNAL SIGNAL NAME

1 1 --- Frame ground
2 2 TXD Transmit data
3 3 RXD Receive data
4 4 RTS Request to send
5 5 CTS Clear to send
6 6 DSR Data Set Ready
7 7 SG Signal Ground
8 8 RLSD or CD Received line signal detect or carrier detect
9 20 DTR Data Terminal Ready
10 22 RI Ring Indicator

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COMMUNICATION BETWEEN DCE AND DTE

Before sending data to the other end the DTE requests the permission from the
modem by issuing RTS signal.
The modem has a method to find out if any telephone line is free and if the other
end of modem is ready.
When the modem finds the communication path is ready for communication it
issues CTS signal to DTE as an acknowledgement.
The DTE issues DTR signal when it is powered on, error free and ready for
logical connection through the modem.
The modem issues a DSR signal to indicate that it is powered on and it is error
free.
The data is transferred by TXD signal from DTE to DCE and RXD signal
receives data from DCE to DTE.
The RI and RLSD signals are used with the dialed modem, when the telephone
link is shared.

Communication

Frame ground
1 1
RTS
4 4
CTS
5 5
MODEM DTR TERMINAL
20 20
DCE DSR DTE
6 6
TXD
2 2
RXD
3 3
RI
22 22
RLSD or CD
8 8
Ground
7 7

25 pin female connector 25 pin male connector

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UNIT II

DATA LINK LAYER

ERROR DETECTION AND CORRECTION

ERROR:

Data can be corrupted during transmission. For reliable communication, errors

must be detected and corrected. Signals flows from one point to another. This is subjected
to unpredictable interferences from heat, magnetism and other forms of electricity.

TYPES OF ERRORS:

Single bit Error:

The term single bit error means that only one bit of a given data unit is changed
from 1 to 0 or 0 to 1. 010101 is changed to 110101 here only one bit is changed
by single bit error.

Burst Error:

A burst error means that 2 or more bits in the data unit have changed.

Example:

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Three kinds of errors can occur:

the bits in the frame can be inverted, anywhere within the frame including the
data bits or the frame's control bits,
additional bits can be inserted into the frame, before the frame or after the frame
and
Bits can be deleted from the frame.

DETECTION

Redundancy

Error detection use the concept of redundancy, which means adding extra bits for
detecting errors at the destination .i.e., instead of repeating the entire data stream, a
shorter group of bits may be appended to the end of each unit.

 To detect or correct errors, we need to send extra (redundant) bits with data.
 The receiver will be able to detect or correct the error using the extra information.
 Detection
◦ Looking at the existence of any error, as YES or NO.
◦ Retransmission if yes. (ARQ)
 Correction
◦ Looking at both the number of errors and the location of the errors in a
message.
◦ Forward error correction. (FEC)

Coding

 Encoder vs. decoder

 Both encoder and decoder have agreed on a detection/correct method in priori.

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Modulo Arithmetic

 In modulo-N arithmetic, we use only the integers in the range 0 to N−1, inclusive.
 Calculation
◦ If a number is greater than N−1, it is divided by N and the remainder is the
result.
◦ If it is negative, as many N’s as needed are added to make it positive.
 Example in Modulo-12
◦ 1512 = 312
◦ -312 = 912

Modulo-2 Arithmetic

 Possible numbers are {0, 1}

 Arithmetic
◦ Addition
 0+0=0, 0+1=1, 1+0= 1, 1+1=2=0
◦ Subtraction
 0-0=0, 0-1=-1=1, 1-0=1, 1-1=0
◦ Surprisingly, the addition and subtraction give the same result.
◦ XOR (exclusively OR) can replace both addition and subtraction.

Detection methods

Parity check
Cyclic redundancy check
checksum

Parity check

A redundant bit called parity bit, is added to every data unit so that the total number of
1’s in the unit becomes even (or odd).

SIMPLE PARITY CHECK

 A simple parity-check code is a single-bit error-detecting code in which n = k + 1

with dmin = 2.
 A simple parity-check code can detect an odd number of errors.

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In a simple parity check a redundant bit is added to a string of data so that total number of
1’s in the data become even or odd.
The total data bit is then passed through parity checking function. For
even parity, it checks for even number of 1’s and for odd parity it checks even number of
1’s. If an error is detected the data is rejected.

Encoder and decoder for simple parity-check code

 In modulo,
◦ r0 = a3+a2+a1+a0
◦ s0 = b3+b2+b1+b0+q0
 Note that the receiver adds all 5 bits. The result is called the syndrome.

Example 1: data to be transmitted = 10110101

5 1’s in the data
Parity bit is 1
Transmitted codeword = 101101011
If receiver gets 101101011, parity check ok ---accept (OK)
If receiver gets 101100011, parity check fails ---reject (OK), ask for frame to be re-
transmitted

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If receiver gets 101110011, parity check ok ---accept (NOT OK: even number of
errors undetected)
If receiver gets 001100011, parity check ok ---accept (NOT OK: even number of
errors undetected)

Let us look at some transmission scenarios. Assume the sender sends the dataword 1011.
The codeword created from this dataword is 10111, which is sent to the receiver. We
examine five cases:
1. No error occurs; the received codeword is 10111. The syndrome is 0. The
dataword 1011 is created.
2. One single-bit error changes a1. The received codeword is 10011. The
syndrome is 1. No dataword is created.
3. One single-bit error changes r0. The received codeword is 10110. The syndrome is
1. No dataword is created.
4. An error changes r0 and a second error changes a3 . The received codeword is
00110. The syndrome is 0. The dataword 0011 is created at the receiver. Note that
here the dataword is wrongly created due to the syndrome value.
5. Three bits—a3, a2, and a1—are changed by errors. The received codeword is
01011. The syndrome is 1. The dataword is not created. This shows that the
simple parity check, guaranteed to detect one single error, can also find any odd
number of errors.

CYCLIC REDUNDANCY CHECK

CRC is based on binary division. In CRC, instead of adding bits to achieve the
desired parity, a sequence of redundant bits, called the CRC or the CRC remainder, is
appended to the end of the data unit so that the resulting data unit becomes exactly
divisible by a second, predetermined binary number. At its destination, the incoming data
unit is assumed to be intact and is therefore accepted. A remainder indicates that the data
unit has been damaged in transit and therefore must be rejected.

STEP BY STEP PROCEDURE

Dividing the data unit by a predetermined divisor derives the redundancy bits
used by CRC; the remainder is CRC.
First a starting of n 0’s is appended to the data unit. The number n is one less than
the number of bits in the predetermined divisor, which is n+1 bits.
The newly elongated data unit is divided by the divisor, using a process called
binary division. The remainder resulting from this division is the CRC.
The CRC of n bits derived in step 2 replaces the appended 0s at the end of the
data unit. Note that the CRC may consist of all 0s.
The data unit arrives at the receiver data first, followed by the CRC. The receiver
treats the whole string as unit and divides it by the same divisor that was used to
find the CRC remainder.

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If the string arrives without error, the CRC checker yields a remainder of zero ad
the data unit passes. If the string has been changed in transit, the division yields a
non zero remainder and the data does not pass.

Architecture of CRC

CRC GENERATOR AND CHECKER

CRC GENERATOR

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Division in the CRC decoder for two cases

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Hardwired design of the divisor in CRC

Simulation of division in CRC encoder

The CRC encoder design using shift registers

General design of encoder and decoder of a CRC code

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HAMMING CODE:

1 •A minimum number of redundancy bits needed to correct any single bit error in
2 the data
3
4 •A minimum of 4 redundancy bits is needed if the number of data bits is 4.
5
6 •Redundancy bits in the Hamming code are placed in the codeword bit positions
7 that are a power of 2
8
9 •Each redundancy bit is the parity bit for a different combination of data bits
10
11 •Each data bit may be included in more than one parity check.

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Easy way to compute the redundancy bit values: write down binary
representations for positions of data bits which contain a 1; compute parity
bits for each ―column‖; put parity bits into codeword in correct order.

• Here: data is 1001101 so codeword will look like 100x110x1xx (where x denotes
redundancy bits) ⇒1’s in positions 3, 6, 7, and 11
11: 1 0 1 1
7: 0 1 1 1
6: 0 1 1 0
3: 0 0 1 1
1001
parity bits
r1 r2 r4 r8
So codeword is 10011100101(as before)
suppose that the bit in position 7 is received in error:

1 • If the transmitted codeword is received error-free, the ―new‖ parity bits the
receiver computes will all be 0 ,the receiver knows no bit errors occurred.
1 • This simple form of Hamming code can be used to provide some protection
against burst errors, by transmitting 1st bit from every codeword to be transmitted, then
2nd bit from every one of these codeword, and so on…In some cases, burst errors can be
corrected

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FLOW CONTROL AND ERROR CONTROL

The two main features of data kink layer are flow control and error control.

FLOW CONTROL
Flow control coordinates that amount of data that can be sent before receiving ACK It is
one of the most important duties of the data link layer.

ERROR CONTROL
Error control in the data link layer is based on ARQ (automatic repeat request), which
is the retransmission of data.
The term error control refers to methods of error detection and retransmission.
Anytime an error is detected in an exchange, specified frames are retransmitted. This
process is called ARQ.

FLOW AND ERROR CONTROL MECHANISMS

1. STOP-AND WAIT ARQ.
2. GO-BACK-N ARQ.
3. SELECTIVE-REPEAT ARQ.

STOP-AND- WAIT ARQ

This is the simplest flow and error control mechanism. It has the following
features.
The sending devise keeps the copy of the last frame transmitted until it receives
an acknowledgement for that frame. Keeping a copy allows the sender to re-
transmit lost or damaged frames until they are received correctly.
Both data and acknowledgement frames are numbered alternately 0 and 1. A data
frame 0 is acknowledged by an ACK 1.
A damaged or lost frame is treated in the same manner by the receiver. If the
receiver detects an error in the received frame, it simply discards the frame and
sends no acknowledgement.
The sender has a control variable, which we call S, that holds the number of
recently sent frame. The receiver has a control variable, which we call R that
holds the number of the next frame expected.
The sender starts a timer when it sends a frame. If an ACK is not received within
an allotted time period the sender assumes that the frame was lost or damaged and
resends it.
The receivers send only positive ACK for frames received safe and sound; it is
silent about the frames damaged or lost.

OPERATION:

The possible operations are

Normal operation
lost frame
ACK lost
delayed ACK.

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The sender sends frame 0 and wait to receive ACK 1. when ACK 1 is received it sends
frame 1 and then waits to receive ACK 0, and so on.

The ACK must be received before the time out that is set expires. The following figure
shows successful frame transmission.

Sender Receiver

S=0 Frame 0
R=0
ACK 1
S=1 Frame 1
R=1
ACK 0
S=0

Time Time

Lost or damaged acknowledgement

o When the receiver receives the damaged frame it discards it, which essentially
means the frame is lost. The receiver remains silent about a lost frame and
keeps its value of R.
o For example in the following figure the sender transmits frame 1, but it is lost.
The receiver does nothing, retaining the value of R (1). After the timer at the
sender site expires, another copy of frame 1 is sent.

Sender Receiver

S=0 Frame 0
R=0
ACK 1
S=1 Frame 1
R=1
Lost
S=1 Frame 1 R=1
Time-out
ACK0 R=0
S=0

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Lost acknowledgement
o A lost or damaged ACK is handle in the same by the sender; if the sender
receives a damaged ACK, it discards it.
o The following figure shows a lost ACK 0.the waiting sender does not know if
frame 1 has been received. When the timer for frame 1 expires the sender
retransmits frame 1.
o Note that the receiver has already received frame 1 and is expecting to receive
frame 0. Therefore, its silently discards the second copy of frame 1.

Sender Receiver

S=0 Frame 0
R=0
ACK 1
S=1 Frame 1
R=1
ACK 0
Time-out S=1 Lost
Frame 1
R=0
ACK0
S=0 Expecting frame 0
Frame 1 is discarded

Time Time

Delayed acknowledgement

o An ACK can be delayed at the receiver or by some problem with the link. The
following figure shows the delay of ACK 1; it ids received after the timer for
frame 0 as already expired.

o The sender has already retransmitted a copy of frame 0. The receiver expects
frame 1 so its simply discards the duplicate frame 0.

o The sender has now received two ACK’s, one that was delayed and one that
was sent after the duplicate frame 0 arrived. The second ACK 1 is discarded.

Sender Receiver

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S=0 Frame 0
R=0
Time-out ACK1
S=0 Frame0 R=1

S=1 Frame 1 Expecting frame 0

ACK1 Frame 1 is discarded
Discarded

S=1 Frame1
Time-out R=1
ACK0

Time Time
BIDIRECTIONAL TRANSMISSION
The stop – and – wait mechanism is unidirectional. We can have bi-directional
transmission if the two parties have two separate channels for full duplex communication
or share the same channel for off duplex transmission. In this case, each party needs both
S and R variables to track frames sent and expected.
PIGGYBACKING
It’s a method to combine a data frame with an ACK. In following figure both the
sender and the receiver have data to send. Instead of sending separate data and ACK
frames. It can save bandwidth because the overhead from a data frame and an ACK frame
can be combined into just one frame

Sender Receiver

R=0
S=0 Frame 0,ACK0 R=0
S=0
R=0 Frame0,ACK 1
S=1
Frame 1,ACK1 R=1
S=1
R=1 Frame1,ACK 0
S=0

Time Time

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GO-BACK-N ARQ
As in Stop-and-wait protocol senders has to wait for every ACK then next frame
is transmitted. But in GO-BACK-N ARQ number of frames can be transmitted
without waiting for ACK. A copy of each transmitted frame is maintained until
the respective ACK is received.

Features of GO-BACK-N ARQ

1.sequence numbers.
Sequence numbers of transmitted frames are maintained in the header of
frame. If k is the number of bits for sequence number, then the numbering can
range from 0 to 2k-1. Example: if k=3 means sequence numbers are 0 to 7.
2. sender sliding window:
Window is a set of frames in a buffer waiting for ACK. This window keeps on
sliding in forward direction, the window size is fixed. As the ACK is received, the
respective frame goes out of window and new frame to sent come into window.
Figure illustrates the sliding window.
If Sender receives. ACK 4, then it knows Frames upto
and including Frame 3 were correctly received

Window size=7

3. Receiver sliding window:

In the receiver side size of the window is always one. The receiver is expecting to
arrive frame in specifies sequence. Any other frame is received which is out of order is
discarded. The receiver slides over after receiving the expected frame. The following
figure shows the receiver side-sliding window.

6 7 0 1 2 3 4 5 6 7
4. Control variables:

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Sender variables and Receiver variables:

Sender deals with three different variables
S -> sequence number of recently sent frame
SF -> sequence number of first frame in the window.
SL -> sequence number of last frame in the window.
The receiver deals with only one variable
R -> sequence number of frame expected.
5. Timers
The sender has a timer for each transmitted frame. The receivers don’t have any
timer.
6. Acknowledgement:
The receiver responds for frame arriving safely by positive ACK. For damaged or
lost frames receiver doesn’t reply, the sender has to retransmit it when timer of that frame
elapsed. The receiver may ACK once for several frames.
7. resending frames:
if the timer for any frame expires, the sender sender has to
resend that frame and the subsequent frame also, hence the protocol is called GO-
BACK-N ARQ.
Operation
Normal operation:Following diagram shows this mechanism. The sender keeps track of
the outstanding frames and updates the variables and windows as acknowledgements
arrive.

SF SL Sender Receiver

0 1 2 3 0 1 Frame 0
0 1 2 3 0 1
111 111
S Frame 1
0 1 2 3 0 1
0 1 2 3 0 1 ACK 2 111
111
S Frame 2
0 1 2 3 0
0 1 2 3 0 1 1111 111
ACK3
111
S Frame 3 0 1 2 3 0 1
111
0 1 2 3 0
111 1 111
S
Time Time
Damaged or lost frame:

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Figure shows that frame 2 is lost. Note that when the receiver receives frame 3, it is
discarded because the receiver is expecting frame 2, not frame3. after the timer for frame
2 expires at the sender site, the sender sends frame 2 and 3.

SF SL Sender Receiver

0 1 2 3 0 1 Frame 0
0 1 2 3 0 1
111 111
S Frame 1
0 1 2 3 0 1
0 1 2 3 0 1 ACK 2 111
111
S Frame 2 lost
0 1 2 3 0
0 1 2 3 0 1 1111 111
111
Frame 3
0 1 2 3 0 1 Frame 3 is discarded, not
111 in the window
Timeout
resent Frame 2
0 1 2 3 0 1
111 0 1 2 3 0 1
Frame 3 111
resent

0 1 2 3 0 0 1 2 3 0 1
111 1 111 111
S R

Time Time

Damaged or lost acknowledgement:

If an ACK is lost, we can have two situations. If the next ACK arrives before the
expiration of timer, there is no need for retransmission of frames because ACK are
cumulative in this protocol.. if the next ACK arrives after the timeout, the frame and all
the frames after that are resent. The receiver never resends an ACK.
For diagrams refer your class work notes.

Delayed Acknowledgement:
A delayed ACK also triggers the resending of frames.
SELECTIVE REPEAT ARQ:
The configuration and its control variables for this are same as those
selective repeat ARQ.

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The size of the window should be one half of the value 2m.
The receiver window size must also be the size. In this the receiver is
looking for a range of sequence numbers.
The receiver has control variables RF and RL to denote the boundaries of
the window.

selective repeat also defines a negative ACK NAK that reports the sequence
number of a damaged frame before the timer expires.

Operation

Normal operation
Normal operations of the selective repeat ARQ is same as GO-BACK-N ARQ
mechanism.
Lost or damaged frame
The following figure shows operation of the mechanism with an example of a lost
frame.
Frame 0 and 1 are accepted when received because they are in the range specified
by the receiver window. When frame 3 is received, it is also accepted for the same
reason. However the receiver sends a NAK 2 to show that frame 2 has not been
received. When the sender receives the NAK 2, it resends only frame 2, which is then
accepted because it is in the range of the window.

Sender Receiver

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0 1 2 3 0 1 Frame 0
0 1 2 3 0 1
111 111
S Frame 1
0 1 2 3 0 1
0 1 2 3 0 1 ACK 2 111
111
S Frame 2 lost

0 1 2 3 0 1
111
Frame 3
0 1 2 3 0 1
111 NAK 2 0 1 2 3 0 1
111
Frame 2
0 1 2 3 0 1
111 resent 0 1 2 3 0 1
111

Time Time

Lost and delayed ACKs and NAKs

In this sender also sets a timer for each frame sent. The remaining operations are
same as GO-BACK-N ARQ.

High-level Data Link Control (HDLC) protocol

HDLC standardized ISO in 1979 and accepted by most other standards bodies
(ITU-T, ANSI)
3 types of end-stations:
Primary–sends commands
Secondary–can only respond to Primary’s commands
Combined–can both command and respond
3 types of configuration
(Note: no balanced multipoint)

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TRANSFER MODE

Mode = relationship between 2 communicating devices;

Describes who controls the link
o NRM = Normal Response Mode
o ABM = Asynchronous Balanced Mode
NRM:
only difference is that secondary needs permission from the Primary in NRM, but
doesn’t need permission from the Primary in ARM.

FRAMES:
3 types of Frames are
1 I-Frame – transports user data and control info about user data.
1 S-Frame – supervisory Frame, only used for transporting control information

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1 U-Frame – unnumbered Frame, reserved for system management(managing the

link itself)

FRAME FORMAT

U-Frames:

U-frames are used for functions such as link setup. They do not contain any
sequence numbers.
Five code bits denote the frame type (but there are not 32 different possibilities):
Set Asynchronous Balanced Mode (SABM).Used in the link set up to indicate
ABM mode will be used.
Set Normal Response Mode (SNRM).Used for asymmetric mode (master/slave).
SABME and SNMRE—extended format.
Disconnect (DISC).Used to disconnect the logical connection.
Frame Reject (FRMR)—reject frame with incorrect semantics.
Unnumbered Acknowledgement (UA).Used to acknowledge other frames in this
class.
Unnumbered Information (UI)–initialisation, poling and status information
needed by the data link layer.
U-frames may carry data when unreliable connectionless service is called for.

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S-Frames:

S-frames are similar to unnumbered frames, the main difference being that they
do carry sequence information.
Some supervisory frames function as positive and negative acknowledgements,
they therefore play a very important role in error and flow control.
Two bits indicate the frame type, so that there are four possibilities.

Receiver Ready -RR(Positive Acknowledgement)

Receiver Not Ready -RNR
Reject -REJ(NAK go-back-N)
Selective Reject -SREJ(NAK selective retransmit)

Control Field:

IEEE 802.4 TOKEN BUS

IEEE 802.4 describes a token bus LAN standard.

In token passing method stations, connected on a bus are arranged in a logical
ring. When the logical ring is initiated, the highest number station may send the
first frame. After this it passes permission to its immediate neighbor by sending a
special frame called a token.

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The token propagates around the logical ring, with only the token holder being
permitted to transmit frames. Since only one station at a time holds the token,
collision do not occur.
There is no relation between the physical location of the station on the bus and its
logical sequence number..

The following figure shows the operation of the token bus.

Physical topology

90 50 120

400 75

Logical sequence of token passing

50 75

400

120 90

Token passing in a bus

802.4 cable standards

The token bus standard specifies three physical layer options in terms of
transmission medium, signaling technique, data rate and maximum electrical
cable segment length.

Medium options
1. Broadband: Transmission medium is co-axial cable and its uses AM/PSK
as a signaling techniques, data rate is 1,5,10 mbps.

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2. Carrier band: Transmission medium is co-axial cable and its uses KSK as
a signaling techniques, data rate is 1,5,10Mbps.

3. Optical fiber: Transmission medium is optical fiber and its uses ASK with
Manchester encoding as a signaling techniques, data rate is 5,10,20Mbps.

IEEE 802.4 Frame format

Token bus frame format is shown in the following figure.

1 1 1 2-6 2-6 0-8182 4

Preamble
pr SD FC DA SA DATA FCS ED

Preamble: the preamble is an at least one byte long pattern to establish bit
synchronization
SD: Start frame delimiter: Its also one byte unique bit pattern, which marks the
start of the frame.
FC: Frame control: The frame control field is used to distinguish data frames from
control frames. For data frame, it carries the frames priority. The frame control
field indicates the type of the frame data frame or control frame.
DA: Destination address: The destination address field is 2 or 6 bytes long.
SA: Source address: The destination address field is 2 or 6 bytes long.
DATA: Data field
FCS: Frame check sequence: frame check sequence is 4 bytes long and contains
CRC code. It is used to detect transmission errors on DA, SA, FC and data fields.
ED: End delimiter: It is a unique bit pattern, which marks the end of the frame. It
is one byte long.
The total length of the frame is 8191 bytes.

Performance:
For token ring, the slightly higher delay compared to CSMS/CD bus occurs. For
higher transmission loads the token ring performs well.

IEEE 802.5 TOKEN RING

IEEE 802.4 describes a token ring LAN standard.

In a token ring a special bit pattern, called the token circulates around the ring
when all stations are idle.
When a station transmits, it breaks the ring and inserts its own frame with source
and destination address.

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When the frame eventually returns to the originating station after completing the
round, the station removes the frame and closes the ring. Because there is only
one token, only one station can transmit at a given instant, thus solving the
channel access problem.
Each station is connected to the ring through a Ring Interface Unit (RIU). The
sequence of token is determined by the physical locations of the stations on the
ring.

The following figure shows the operation and arrangement of the Token Ring.

stations

Unidirectional bus

Ring interface

802.5 cable standards

Its uses two types of transmission medium.

1. Shielded twisted pair cable: (STP)

It uses differential Manchester encoding technique. Data rate is 4 or
16 Mbps. Maximum number of repeaters allowed is 250.
2. Unshielded twisted pair cable: (UTP)
It uses differential Manchester encoding technique. Data rate is
4Mbps. Maximum number of repeaters allowed is 250.

IEEE 802.5 Frame format

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Token ring frame format is shown in the following figure.

Token frame format

1 1 1 byte
SD AC ED

Data Frame

1 1 1 2-6 2-6 No limit 4 1

1

SDpr AC FC DA SA DATA FCS ED FS

SD: Start frame delimiter: Its also one byte unique bit pattern, which marks the
start of the frame.
AC: Access control: It is one byte long field containing priority bits(P), Token
bit(T),
monitoring bit(M), and reservation bir(R).
FC: Frame control: The frame control field is used to distinguish data frames from
control frames. For data frame, it carries the frames priority. The frame control
field indicates the type of the frame data frame or control frame.
DA: Destination address: The destination address field is 2 or 6 bytes long.
SA: Source address: The destination address field is 2 or 6 bytes long.
DATA: Data field
FCS: Frame check sequence: frame check sequence is 4 bytes long and contains
CRC code. It is used to detect transmission errors on DA, SA, FC and data fields.
ED: End delimiter: It is a unique bit pattern, which marks the end of the frame. It
is one byte long.
FS: Frame status: This field is none byte long and contains a unique bit pattern
marking the end of a token or a data frame.

Performance:
When traffic is light, the token will spend most of its time idly circulating around
the ring. When traffic is heavy, there is a queue at each station. Network efficiency is
more.

Disadvantages:

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A break in a link or repeater failures disturbs the entire network.

Installation of new repeaters requires identification of two topologically
adjacent repeaters.
Since the ring is closed loop, a packet will circulate indefinitely unless it is
removed.
Each repeater adds an increment of delay.
There is practical limit to the number of repeaters.

Fiber Distributed Data Interface

Introduction

The Fiber Distributed Data Interface (FDDI) specifies a 100-Mbps token-passing, dual-
ring LAN using fiber-optic cable. FDDI is frequently used as high-speed backbone
technology because of its support for high bandwidth and greater distances than copper. It
should be noted that relatively recently, a related copper specification, called Copper
Distributed Data Interface (CDDI), has emerged to provide 100-Mbps service over
copper. CDDI is the implementation
of FDDI protocols over twisted-pair copper wire. This chapter focuses mainly on FDDI
specifications and operations, but it also provides a high-level overview of CDDI.

FDDI uses dual-ring architecture with traffic on each ring flowing in opposite directions
(called counter-rotating). The dual rings consist of a primary and a secondary ring.
During normal operation, the primary ring is used for data transmission, and the
secondary ring remains idle. As will be discussed in detail later in this chapter, the
primary purpose of the dual rings is to provide superior reliability and robustness. Figure
8-1 shows the counter-rotating primary and secondary FDDI rings.

FDDI Specifications

FDDI specifies the physical and media-access portions of the OSI reference model. FDDI
is not actually a single specification, but it is a collection of four separate specifications,
each with a specific function. Combined, these specifications have the capability to
provide high-speed connectivity between upper-layer protocols such as TCP/IP and IPX,
and media such as fiber-optic cabling.

FDDI's four specifications are the Media Access Control (MAC), Physical Layer
Protocol (PHY), Physical-Medium Dependent (PMD), and Station Management (SMT)
specifications. The MAC specification defines how the medium is accessed, including
frame format, token handling, addressing, algorithms for calculating cyclic redundancy
check (CRC) value, and error-recovery mechanisms. The PHY specification defines data
encoding/decoding procedures, clocking requirements, and framing, among other
functions. The PMD specification defines the characteristics of the transmission medium,
including fiber-optic links, power levels, bit-error rates, optical components, and
connectors. The SMT specification defines FDDI station configuration, ring

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configuration, and ring control features, including station insertion and removal,
initialization, fault isolation and recovery, scheduling, and statistics collection.

FDDI is similar to IEEE 802.3 Ethernet and IEEE 802.5 Token Ring in its relationship
with the OSI model. Its primary purpose is to provide connectivity between upper OSI
layers of common protocols and the media used to connect network devices. Figure 8-3
illustrates the four FDDI specifications and their relationship to each other and to the
IEEE-defined Logical Link Control (LLC) sublayer. The LLC sublayer is a component of
Layer 2, the MAC layer, of the OSI reference model.

Figure 8-3: FDDI Specifications Map to the OSI Hierarchical Model

FDDI Frame Format

The FDDI frame format is similar to the format of a Token Ring frame. This is one of the
areas in which FDDI borrows heavily from earlier LAN technologies, such as Token
Ring. FDDI frames can be as large as 4,500 bytes. Figure 8-10 shows the frame format of
an FDDI data frame and token.

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Figure 8-10: The FDDI Frame Is Similar to That of a Token Ring Frame

FDDI Frame Fields

The following descriptions summarize the FDDI data frame and token fields illustrated in
Figure 8-10.

Preamble—Gives a unique sequence that prepares each station for an upcoming

frame.
Start delimiter—Indicates the beginning of a frame by employing a signaling
pattern that differentiates it from the rest of the frame.
Frame control—Indicates the size of the address fields and whether the frame
contains asynchronous or synchronous data, among other control information.
Destination address—Contains a unicast (singular), multicast (group), or
broadcast (every station) address. As with Ethernet and Token Ring addresses,
FDDI destination addresses are 6 bytes long.
Source address—Identifies the single station that sent the frame. As with Ethernet
and Token Ring addresses, FDDI source addresses are 6 bytes long.
Data—Contains either information destined for an upper-layer protocol or control
information.
Frame check sequence (FCS)—Is filed by the source station with a calculated
cyclic redundancy check value dependent on frame contents (as with Token Ring
and Ethernet). The destination address recalculates the value to determine whether
the frame was damaged in transit. If so, the frame is discarded.
End delimiter—Contains unique symbols; cannot be data symbols that indicate
the end of the frame.
Frame status—Allows the source station to determine whether an error occurred;
identifies whether the frame was recognized and copied by a receiving station.

Dual Ring

FDDI's primary fault-tolerant feature is the dual ring. If a station on the dual ring fails or
is powered down, or if the cable is damaged, the dual ring is automatically wrapped
(doubled back onto itself) into a single ring. When the ring is wrapped, the dual-ring
topology becomes a single-ring topology. Data continues to be transmitted on the FDDI

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ring without performance impact during the wrap condition. Figure 8-6 and Figure 8-7
illustrate the effect of a ring wrapping in FDDI.

Figure 8-6: A Ring Recovers from a Station Failure by Wrapping

Figure 8-7: A Ring also Wraps to Withstand a Cable Failure

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When a single station fails, as shown in Figure 8-6, devices on either side of the failed (or
powered-down) station wrap, forming a single ring. Network operation continues for the
remaining stations on the ring. When a cable failure occurs, as shown in Figure 8-7,
devices on either side of the cable fault wrap. Network operation continues for all
stations.

It should be noted that FDDI truly provides fault tolerance against a single failure only.
When two or more failures occur, the FDDI ring segments into two or more independent
rings that are incapable of communicating with each other.

The IEEE 802.3 Logical Relationship to the ISO Reference Model

Figure 7-4 shows the IEEE 802.3 logical layers and their relationship to the OSI
reference model. As with all IEEE 802 protocols, the ISO data link layer is divided into
two IEEE 802 sublayers, the Media Access Control (MAC) sublayer and the MAC-client
sublayer. The IEEE 802.3 physical layer corresponds to the ISO physical layer.

Figure 7-4 Ethernet's Logical Relationship to the ISO Reference Model

The MAC-client sublayer may be one of the following:

• Logical Link Control (LLC), if the unit is a DTE. This sublayer provides the interface
between the Ethernet MAC and the upper layers in the protocol stack of the end station.
The LLC sublayer is defined by IEEE 802.2 standards.

• Bridge entity, if the unit is a DCE. Bridge entities provide LAN-to-LAN interfaces
between LANs that use the same protocol (for example, Ethernet to Ethernet) and also
between different protocols (for example, Ethernet to Token Ring). Bridge entities are
defined by IEEE 802.1 standards.

Because specifications for LLC and bridge entities are common for all IEEE 802 LAN
protocols, network compatibility becomes the primary responsibility of the particular

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network protocol. Figure 7-5 shows different compatibility requirements imposed by the
MAC and physical levels for basic data communication over an Ethernet link.

Figure 7-5 MAC and Physical Layer Compatibility Requirements for Basic Data
Communication

The MAC layer controls the node's access to the network media and is specific to the
individual protocol. All IEEE 802.3 MACs must meet the same basic set of logical
requirements, regardless of whether they include one or more of the defined optional
protocol extensions. The only requirement for basic communication (communication that
does not require optional protocol extensions) between two network nodes is that both
MACs must support the same transmission rate.

The 802.3 physical layer is specific to the transmission data rate, the signal encoding, and
the type of media interconnecting the two nodes. Gigabit Ethernet, for example, is
defined to operate over either twisted-pair or optical fiber cable, but each specific type of
cable or signal-encoding procedure requires a different physical layer implementation.

The Ethernet MAC Sublayer

The MAC sub layer has two primary responsibilities:

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• Data encapsulation, including frame assembly before transmission, and frame

parsing/error detection during and after reception

• Media access control, including initiation of frame transmission and recovery from
transmission failure

The Basic Ethernet Frame Format

The IEEE 802.3 standard defines a basic data frame format that is required for all MAC
implementations, plus several additional optional formats that are used to extend the
protocol's basic capability. The basic data frame format contains the seven fields shown
in Figure 7-6.

• Preamble (PRE)—Consists of 7 bytes. The PRE is an alternating pattern of ones and

zeros that tells receiving stations that a frame is coming, and that provides a means to
synchronize the frame-reception portions of receiving physical layers with the incoming
bit stream.

• Start-of-frame delimiter (SOF)—Consists of 1 byte. The SOF is an alternating pattern

of ones and zeros, ending with two consecutive 1-bits indicating that the next bit is the
left-most bit in the left-most byte of the destination address.

• Destination address (DA)—Consists of 6 bytes. The DA field identifies which

station(s) should receive the frame. The left-most bit in the DA field indicates whether
the address is an individual address (indicated by a 0) or a group address (indicated by a
1). The second bit from the left indicates whether the DA is globally administered
(indicated by a 0) or locally administered (indicated by a 1). The remaining 46 bits are a
uniquely assigned value that identifies a single station, a defined group of stations, or all
stations on the network.

• Source addresses (SA)—Consists of 6 bytes. The SA field identifies the sending

station. The SA is always an individual address and the left-most bit in the SA field is
always 0.

• Length/Type—Consists of 2 bytes. This field indicates either the number of MAC-

client data bytes that are contained in the data field of the frame, or the frame type ID if
the frame is assembled using an optional format. If the Length/Type field value is less
than or equal to 1500, the number of LLC bytes in the Data field is equal to the
Length/Type field value. If the Length/Type field value is greater than 1536, the frame is
an optional type frame, and the Length/Type field value identifies the particular type of
frame being sent or received.

• Data—Is a sequence of n bytes of any value, where n is less than or equal to 1500. If
the length of the Data field is less than 46, the Data field must be extended by adding a
filler (a pad) sufficient to bring the Data field length to 46 bytes.

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• Frame check sequence (FCS)—Consists of 4 bytes. This sequence contains a 32-bit

cyclic redundancy check (CRC) value, which is created by the sending MAC and is
recalculated by the receiving MAC to check for damaged frames. The FCS is generated
over the DA, SA, Length/Type, and Data fields.

Figure 7-6 The Basic IEEE 802.3 MAC Data Frame Format

Note Individual addresses are also known as unicast addresses because they refer to a
single MAC and are assigned by the NIC manufacturer from a block of addresses
allocated by the IEEE. Group addresses (a.k.a. multicast addresses) identify the end
stations in a workgroup and are assigned by the network manager. A special group
address (all 1s—the broadcast address) indicates all stations on the network.

Frame Transmission

Whenever an end station MAC receives a transmit-frame request with the accompanying
address and data information from the LLC sublayer, the MAC begins the transmission
sequence by transferring the LLC information into the MAC frame buffer.

• The preamble and start-of-frame delimiter are inserted in the PRE and SOF fields.

• The destination and source addresses are inserted into the address fields.

• The LLC data bytes are counted, and the number of bytes is inserted into the
Length/Type field.

• The LLC data bytes are inserted into the Data field. If the number of LLC data bytes
is less than 46, a pad is added to bring the Data field length up to 46.

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• An FCS value is generated over the DA, SA, Length/Type, and Data fields and is
appended to the end of the Data field.

After the frame is assembled, actual frame transmission will depend on whether the MAC
is operating in half-duplex or full-duplex mode.

The IEEE 802.3 standard currently requires that all Ethernet MACs support half-duplex
operation, in which the MAC can be either transmitting or receiving a frame, but it
cannot be doing both simultaneously. Full-duplex operation is an optional MAC
capability that allows the MAC to transmit and receive frames simultaneously.

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UNIT III

Network Layer
• Transport segment from sending to receiving host.
• On sending side encapsulates segments into datagrams.
• On receiving side, delivers segments to transport layer.
• network layer protocols in every host, router.
• Router examines header fields in all IP datagrams passing through it.

Network-Layer Functions
• forwarding: move packets from router’s input to appropriate router
Output.
• routing: determine route taken by packets from source to destination.

Internetworking

Internetworking is a scheme for interconnecting multiple networks of dissimilar

technologies
Uses both hardware and software
o Extra hardware positioned between networks
o Software on each attached computer
System of interconnected networks is called an internetwork or an internet

Switching Schemes

(1) Circuit Switching

(2) Message Switching (Store-and-Forward)
(3) Packet Switching (Store-and-Forward)

Circuit Switching

• Provides service by setting up the total path of connected lines hop-by-hop from the
origin to the destination
• Example: Telephone network

1. Control message sets up a path from origin to destination

2. Return signal informs source that data transmission may proceed
3. Data transmission begins
4. Entire path remains allocated to the transmission (whether used or not)
5. When transmission is complete, source releases the circuit.

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VC implementation

A VC consists of:
1. Path from source to destination
2. VC numbers, one number for each link along path
3. Entries in forwarding tables in routers along path
 Packet belonging to VC carries a VC number.
 VC number must be changed on each link.
 New VC number comes from forwarding table

Packet Switching
• Messages are split into smaller pieces called packets.
• These packets are numbered and addressed and sent through the network one at a time.
• Allows Pipelining
– Overlap sending and receiving of packets on multiple links.

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IP Addresses

Each network interface on the Internet as a unique global address, called the
IP address. An IP address- is 32 bits long. It encodes a network number and a host
number.IP addresses are written in a dotted decimal notation:
128.238.42.112 means
10000000 in 1st Byte
11101110 in 2nd Byte
00101010 in 3rd Byte
01110000 in 4th Byte

Internet Address Classes:

IP distinguishes 5 classes of addresses.

Internet Address Classes

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IP Address classes

• Class A:
– For very large organizations
– 16 million hosts allowed
• Class B:
– For large organizations
– 65 thousand hosts allowed
• Class C
– For small organizations
– 255 hosts allowed
• Class D
– Multicast addresses
– No network/host hierarchy

Internet= a collection of connected networks which share a common set of rules for
communication
IP Address Hierarchy

• Note that Class A, Class B, and Class C addresses only support two levels
of hierarchy
• Each address contains a network and a host portion, meaning two levels of
Hierarchy.
• However, the host portion can be further split into ―subnets‖ by the address
class owner
• This allows for more than 2 levels of hierarchy.

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IP Subnetting

Subnetting is a technique used to allow a single IP network address to

Span multiple physical networks
IP hosts should support subnetting.
Subnetting is done by using some of the bits of the host-id part of the IP address
physical layer network identifier
The subnet mask is used to determine the bits of the network identifier.
All hosts on the same network should have the same subnet mask.
IP address is composed of a Netid part and a Hostid part ⇒2-level
hierarchy.

Sometimes a 2-level hierarchy is insufficient for an organisation’s needs.

1
2
3

4
5
6
7
8
9
10
11
12
13
Subnetting

An IP packet from some other network destined for host 141.14.2.21 still reaches
router R1, since the destination address is still a Class B address with Netid141.14
and Hostid 2.21 as far as the rest of the Internet is concerned.
when the packet reaches router R1, the interpretation of the IP address changes

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R1 knows that there are 3 levels of hierarchy within the organization, and that in
this case, the Netid is 141.14,the Subnetid is 2, and the Hostid is 21.
How is this knowledge of the internal network hierarchy implemented in the
organization’s routers?
Masking of IP addresses during the packet-forwarding process.
Masking is done whether or not subnetting is being used with subnetting,

the Netid defines the site, the Subnetid defines the physical network, and the Hostid
defines the actual machine.
14

15
16
17
Subnet Masks
Subnet masks allow hosts to determine if another IP address is on the same
subnet or the same network.
18
19

20
21
22
23 Router
24
A router is a hardware component used to interconnect networks
A router has interfaces on multiple networks

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Networks can use different technologies

Router forwards packets between networks
Transforms packets as necessary to meet standards for each network
25
26
Routing issues:

Scalability: must be able to support large numbers of hosts, routers, networks

Adapt to changes in topology or significant changes in traffic, quickly and efficiently

self-healing: little or not human intervention

Route selection may depend on different criteria

Performance: "choose route with smallest delay"

Policy : "choose a route that doesn't cross a government network" (equivalently: "let no
non-government traffic cross this network")
27
Classification of Routing algorithms

Centralized versus decentralized

Centralized: central site computes and distributed routes (equivalently:

information for computing routes known globally, each router makes same
computation)

Decentralized: each router sees only local information (itself and physically-
connected neighbors) and computes routes on this basis. pros and cons?

Static versus adaptive

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Static: routing tables change very slowly, often in response to human

intervention
Dynamic: routing tables change as network traffic or topology change

Two basic approaches adopted in practice:

Link-state routing: centralized, dynamic (periodically run)

Distance vector: distributed, dynamic (in direct response to changes)
28
29
30 Routers
31
Routers are distinguished by the functions they perform
Internal routers
• Only route packets within one area
Area border routers
• Connect to areas together
Backbone routers
• Reside only in the backbone area
AS boundary routers
• Routers that connect to a router outside the AS
32
33
34 Routing
35 The most common routing algorithms are distance-vector and link-
state routing.

Distance-vector:
Each router exchanges information about the entire network with
neighboring routers at regular intervals.
Neighboring routers = connected by a direct link (e.g. a LAN)
Regular intervals: e.g. every 30 seconds

Link-state:
Each router exchanges information about its neighborhood with all
routers in the network when there is a change.
Neighborhood of a router = set of neighbor routers for this router.
Each router’s neighborhood information is flooded through the
network.

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Change: e.g. if a neighboring router does not reply to a status

message.

―Clouds‖ represent LANs; number in cloud represents network ID

A, B, C, D, E, F are routers (or gateways)

Each router sends its information about the entire network only to its
neighbors

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How do non-neighboring routers learn about each other and share

information?
•A router sends its information to its neighbors
•Each neighbor router adds this information to its own, and sends the
updated information to its neighbors; the first router learns about its
neighbors’ neighbors.

Routing table update algorithm (distributed Bellman-Ford algorithm):

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Add 1 to cost of each incoming route (since each neighbor is 1hop

away)
If a new destination is learned, add its information to the routing table
If new information received on an existing destination:
If Next Hop field is the same, replace existing entry with the new
information even if the cost is greater(―new information invalidates
old‖)
If Next Hop field is not the same, only replace existing entry with the
new information if the cost is lower

Example of routing table update algorithm

Final routing tables :

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Problem with distance-vector routing:

Slow convergence of distance vector routing algorithms under some
conditions
Slow reaction to link/router failure because information only comes
from neighboring routers and it may be out-of-date (e.g. it may not
properly reflect the impact of the failure on route costs)

Link-State routing

•Each router sends information about its neighborhood to every other router

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Link cost is usually a weighted sum of various factors

e.g. traffic level, security level, packet delay
Link cost is from a router to the network connecting it to another
router.
when a packet is in a LAN (which is typically a broadcast network),
every node –including the router –can receive it
No cost assigned when going .

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Routers share information by advertising, which means sending link-

state packets•.

Every router builds a link-state packet and floods it through the

network, so when all such packets have been received at a router, it
can build its link-state database.
Assuming that every router receives the same set of link-state packets
(as if the routers were synchronized),every router builds the same
link-state database. Using this database, each router can then calculate
its routing table.

To calculate its routing table, a router uses Dijkstra’s Shortest-Path

algorithm
First, identify all link costs in the network: either from the link-state
database, or using the fact that the cost of any link from a network to a
router is 0
This algorithm builds a shortest-path spanning tree for the router such
a tree has a route to all possible destinations, and no loops.
The router running the algorithm is the root of its shortest-path
spanning tree.
Even if all routers’ link-state databases are identical, the trees
determined by the routers are different (since the root of each tree is
different)
A node is either a network or a router; nodes are connected by arcs.

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The algorithm keeps track of 2 sets of nodes and arcs –Temporary and
Permanent.
Initially, the Temporary set contains all neighbor nodes of the router
itself, and the arcs connecting them to the router; only the router is
initially Permanent.
When all nodes and arcs are in the Permanent set, the algorithm has
terminated.
Identify the Temporary node whose arc has the lowest cumulative cost
from the root: this node and arc are moved into the Permanent set.
Any nodes which are connected to the new Permanent node and are
not already in the Temporary set, along with the connecting arcs, are
made Temporary.
Also, if any node already in the Temporary set has a lower cumulative
cost from the root by using a route passing through the new
Permanent node, then this new route replaces the existing one
Repeat until all nodes and arcs are Permanent.

1
2
Let’s follow the steps of the algorithm run by router A.

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Once a router has found its shortest-path spanning tree, it can build its
routing table.
To complete the Example, here is router A’s link-state routing table

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In large networks, the memory required to store the link-state database

and the computation time to calculate the link-state routing table can
be significant.
in practice, since the link-state packet receptions are not synchronized,
routers may be using different link-state databases to build their
routing tables.
Note:

Link-state routing algorithms have several desirable properties, e.g.

rapid convergence; small amount of traffic generated; rapid response
to topology changes.

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UNIT - IV
The Transport Layer is responsible for end-to-end data transport
Primary functions include:
Provision of connection oriented or connectionless
service.
Disassembling and reassembling data.
Setup and release of connections across the network.

Services provided by Internet transport protocols

TCP service:
• connection- oriented: setup required between client, server
• reliable transport between sending and receiving process
• flow control: sender won’t overwhelm receiver
• congestion control: throttle sender when network overloaded
• does not provide: timing, minimum bandwidth

UDP service:
• unreliable data transfer between sending and receiving process
• does not provide: connection setup, reliability, flow control, congestion
control, timing, or bandwidth guarantee guarantees

UDP

•UDP is a connectionless transport protocol–extends IP’s host-to-host

delivery service into a process-to-process communication service
can have multiple application processes on a single host,
each with their own port number.
A process is uniquely addressed by a < port, host > pair
Common services are available at well-known (and
reserved) ports on each host; user applications must
choose their ports from the set of non-reserved ports.

•UDP doesn’t support flow control or reliable/in-order delivery, but it does

support error detection by computing an ―optional‖ checksum over the UDP
header, UDP data, and IP pseudo header(includes source and destination
address fields from the IP header)

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•New: Reliable UDP –provides reliable in-order delivery (up to a maximum

number of retransmissions), with simple window flow control, for virtual
connections.

Addressing

An address at the transport layer is typically a tuple (Station, Port) where

Station is the network address of the host, and
Port identifies the application

UDP Data Packet

The source port, much like the source port in TCP, identifies the
process on the originating system. TCP ports and UDP ports are not
the same. There is no relationship between the two.
The destination port identifies the receiving process on the receiving
machine. Whereas the IP address identifies which machine should get
the packet, the port identifies which machine should get the data.
The length field contains the length of the UDP datagram. This
includes the length of the UDP header and UDP data. It does not
include anything added to the packet in-transit by other protocols --
but these are stripped away before UDP sees the datagram at the other
side.

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The checksum field is used by UDP to verify the correctness of the

UDP header and data. If the checksum indicates an error, the packet is
dropped. UDP is unreliable, so it makes no attempt to mitigate the
loss.

Application
Datagram oriented
unreliable, connectionless
simple
unicast and multicast
Useful only for few applications, e.g., multimedia applications
Used a lot for services
– network management(SNMP), routing (RIP),naming(DNS), etc.

Port Numbers
UDP (and TCP) use port numbers to identify applications
A globally unique address at the transport layer (for both UDP and
TCP) is a tuple <IP address, port number>
There are 65,535 UDP ports per host.

TCP: Transmission Control Protocol

TCP is a reliable, point-to-point, connection-oriented, full-duplex protocol.

Reliable: A reliable protocol ensures that data sent from one machine to
another will eventually be communicated correctly. It does not guarantee
that this data will be transmitted correctly within any particular amount of

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time -- just that given enough time, it will arrive. Life isn't perfect, and it is
possible for corrupted data to be thought correct by a reliable protocol -- but
the probability of this occurring is very, very, very low Point-to-point: Point-
to-point protocols are those protocols that communicate information
between two machines. By contrast, broadcast and multicast protocols
communicate information from one host to many hosts.

Connection-oriented :A connection oriented protocol involves a

connection or session between the endpoints. In other words, each
host is aware of the other and can maintain information about the state
of communication between them. The connection needs to be
initialized and destroyed. The shared state that is possible with a
connection-oriented protocol is essential to a reliable protocol. In
particular, the notion of a sequence number or serial number is a
practical necessity, if not a theoretical necessity.
Full-duplex:By full-duplex we mean a mode of communication such
that both sides can send and receive concurrently

TCP Data Packet

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TCP header fields

Flag bits:

– URG: Urgent pointer is valid

If the bit is set, the following bytes contain an urgent message in the
sequence number range ―SeqNo <= urgent message <= SeqNo +
urgent pointer‖
ACK: Segment carries a valid acknowledgement
PSH: PUSH Flag
Notification from sender to the receiver that the receiver should pass
all data that it has to the application.
Normally set by sender when the sender’s buffer is empty
RST: Reset the connection
The flag causes the receiver to reset the connection.
Receiver of a RST terminates the connection and indicates higher
layer application about the reset
SYN: Synchronize sequence numbers
Sent in the first packet when initiating a connection
FIN: Sender is finished with sending
Used for closing a connection
Both sides of a connection must send a FIN.

TCP Connection Establishment

TCP uses a three-way handshake

(1) ACTIVE OPEN: Client sends a segment with

SYN bit set *
port number of client
initial sequence number (ISN) of client

(2) PASSIVE OPEN: Server responds with a segment

SYN bit set *
initial sequence number of server
ACK for ISN of client

(3) Client acknowledges by sending a segment with

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ACK ISN of server (* counts as one byte)

Connection Termination :

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Congestion Control

Open Loop Congestion Control

-To prevent congestion before it happens

Retransmission policy :
Good Retransmission policy & Retransmission timer.

Window policy
-Selective Repeat Window.

Acknowledgement policy:
-Does not acknowledge every packet.

Discarding Policy:
-Good discarding Policy.

Admission Policy
-Switches check the resource requirement of flow.

Closed Loop Congestion control

- To alleviate congestion after it happens
Back Pressure(router con):
-Inform the previous upstream router.

Choke point:
-packet sent by router to Source.
Implicit Signaling:
-Source can detect
Explicit Signaling:
-Routers inform sender
Backward Signaling:
-Warn the Source (opp dir)
Forward Signaling:
-Warn the Destination

Slow Start :
Set cwnd size to max. seg size. Increases exponentially.

Additive Increase:
After it reaches threshold increase by 1 seg.If it reaches time-out then multiplicative
decrease.

Multiplicative decrease:

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Set the threshold to one half of last cwnd size. Each time it is reduced to one half of last
cwnd size if a time –out occurs.
Traffic Shaping
• Traffic shaping controls the rate at which packets are sent (not just how many)

• At connection set-up time, the sender and carrier negotiate a traffic pattern
(shape)

• Two traffic shaping algorithms are:

– Leaky Bucket
– Token Bucket

The Leaky Bucket Algorithm

• The Leaky Bucket Algorithm used to control rate in a network. It is
implemented as a single-server queue with constant service time. If the bucket
(buffer) overflows then packets are discarded.
•
• The leaky bucket enforces a constant output rate regardless of the burstiness of
the input. Does nothing when input is idle.

• The host injects one packet per clock tick onto the network. This results in a
uniform flow of packets, smoothing out bursts and reducing congestion.

• When packets are the same size (as in ATM cells), the one packet per tick is okay.
For variable length packets though, it is better to allow a fixed number of bytes
per tick.

Token Bucket Algorithm

• In contrast to the LB, the Token Bucket (TB) algorithm, allows the output rate to
vary, depending on the size of the burst.

• In the TB algorithm, the bucket holds tokens. To transmit a packet, the host must
capture and destroy one token.

• Tokens are generated by a clock at the rate of one token every t sec.

• Idle hosts can capture and save up tokens (up to the max. size of the bucket) in
order to send larger bursts later.
Token bucket operation
• TB accumulates fixed size tokens in a token bucket

• Transmits a packet (from data buffer, if any are there) or arriving packet if the
sum of the token sizes in the bucket add up to packet size

• More tokens are periodically added to the bucket (at rate t). If tokens are to be
added when the bucket is full, they are discarded

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Token bucket properties

• Does not bound the peak rate of small bursts, because bucket may contain enough
token to cover a complete burst size

• Performance depends only on the sum of the data buffer size and the token bucket
size

Domain Name System

Introduction
 A sys that can map a name to an address or an add to a name.
 Mapping was done using a host file
It has 2 columns
Name and address
 Every host could store the host file on its disk and should be updated from master
file.
 If a program or a user wanted to map a name to an add. ,host consulted the host
file and found mapping

 Divide the huge amt of info into smaller parts

 Store each part on a different computer
 The host that needs mapping can contact the closest computer holding the needed
information
 i.e. DNS
Name Space

 Names assigned to machines must be selected from name space with control over
the binding between names and IP addresses.
 A name space that maps each address to a unique name can be organised in two
ways.
Flat Name Space
Hierarchical Name Space

Flat Name Space :

• Name is assigned to an address.
• Name in this space is a sequence of
characters without structure.
Demerit:
• Can’t be used in large system such as Internet
• It must be centrally controlled to avoid ambiguity and duplication
Hierarchical Name Space
• Each name is made of several parts
• First part – nature of the organization
• Second part – Name
• Third part – department
• So the authority to assign and control the name space can be decentralized

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• Suffixes can be added to the name to define the host or resources

• To have a hierarchical name
Space, DNS was designed
• Names are defined in an inverted tree structure with root at the top.
• Tree can have only 128 levels
Label :
Each node in the tree has a label(a string with a maximum of 63 characters)

 Root label is a null string

 Children of a node have different labels which guarantees the uniqueness of
domain names.
Domain Name:
 Each node in the tree has a domain name.
 DN is a sequence of labels separated by dots
 Always read from the node up to the root

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FQDN

 If label is terminated by a null string it is called a FQDN

 A domain name that contains the full name of a host
 Contains all labels from specific to general
 Uniquely define the name of the host.
challenger.atc.fhda.edu.

PQDN

 If label is not terminated by a null string it is called PQDN.

 Starts from a node but does not reach the root.
 Used when the name to be resolved belongs to the same site as client.
 Resolver can supply the missing part called suffix ,to create an FQDN.

 DNS client adds suffix atc.fhda.deu before passing the address to the DNS server.
Domain:
 A domain is a sub-tree of the domain space.
 Domain may itself be divided into sub domains.

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Distribution of Name Space

 The info contained in the domain name space must be stored.

 Not reliable to have info in 1 computer.

 So distribute the information among many computers called servers

Hierarchy of Name Servers

 Distribute the info among many computers called DNS servers

 Divide the whole space into many domains based on the first level.
 Let the root stand alone and create as many domains as there are

 Allows domains to be divided further into smaller domains

 Each server can be responsible for either a large or a small domain

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Zone

 What a Server is responsible for or has authority over is called zone.

 If a server accepts responsibility for a domain and does not divide the domain into
smaller domains.
 The domain and the zone refer to the same thing.
 Server makes a database called a zone file.
 It keeps all the information for every node under that domain

Root server

 Is a server whose zone consists of whole tree

 It does not store any info about domain but delegates authority to other servers
Primary server:
 A server that stores a file about the zone for which it is an authority.
 Responsible for creating, maintaining the and updating the zone file
 It stores the zone file on a local disk

Secondary server

 A server that transfers the complete information about a zone from another server
and stores the file on its local disk
 It neither creates nor updates the zone files.
 Updating is done by a primary server, which sends the updated version to
secondary

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A primary server loads all information from the disk file; the secondary server
loads all information from the primary server

DNS In The Internet

 In the Internet, Domain name space (tree) is divided into 3 sections.

Generic Domain

 It defines registered hosts according to their generic behavior.

 Each node in the tree defines a domain, which is an index to the domain name
space database.
 Seven labels describe three organization types.

Generic Domain Labels

Label Description

com Commercial organizations

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edu Educational institutions

gov Government institutions

int International organizations

mil Military groups

net Network support centers

org Nonprofit organizations

Country Domain
Follows the same format as the generic domain but uses two character country
abbreviations.

Inverse domain

It is used to map an address to a name

Ex:
 When a server has received a request from a client to do a task
 Whereas the server has a file that contains a list of authorized clients, the server
lists only the IP address of the client

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 To determine if the client is on the authorized list ,server send a query

 To inverse DNS server and ask for a mapping of address to a name
 This query is called inverse or pointer (PTR) query
 To handle this ,inverse domain is added to the domain space with the first level
node called arpa
 Second level is also one single node named in-addr
 Rest of the domain defines the IP address.

Resolution
 Mapping a name to an address or an address to a name is called name-address
resolution.

Resolver
 A host that needs to map an address to a name or a name to an address calls a
DNS client named a resolver.
 It accesses the closest DNS server with a mapping request
 If the server has the information , it satisfies the resolver.
 Otherwise it refers the resolver to other servers or ask other servers to provide
information.
 After the resolver receives the mapping ,it interprets to see if it is a real resolution
or an error and finally delivers the result to the process that requested it.

Mapping Names to addresses

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 The resolver gives a domain name to the server and asks for the corresponding
address
 In this ,server checks the generic domain or the country domain to find the
mapping
 If from the generic domain the resolver receives a domain name such as
chal.atc.fhda.edu.

 Query is sent by the resolver to the local DNS server for resolution
 If cant refers the resolver to other servers or ask other servers directly
 If from the country domain, the resolver receives a domain name such as
ch.fhda.cu.ca.us.
Mapping addresses to names
 Client can send an IP address to a server to be mapped to a domain name – called
PTR query
 To answer this uses inverse domain
 In the request IP address is reversed and 2 labels in-addr & arpa are appended to
create a domain acceptable by the inverse domain section
 132.34.45.121 ,121.45.34.132.in-addr.arpa.

Recursive Resolution

 The resolver expects the server to supply the final answer

 If server is the authority for the domain name ,it checks the database and responds
 If not the authority ,sends the request to another server (parent) and waits for
response

 If the parent is the authority respond otherwise sends the query to yet another
server
 If resolved, response travels back until it reaches the requesting client
 This is recursive resolution

Iterative Resolution

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DNS Messages

Header format
 Identification
Used by the client to match the response with the query.
Uses a diff id no. each time it sends a query.
Server duplicates this no. in the corresponding response.
 Flag
Collection of fields that define the
Type of msg
Type of answers requested
Type of desired resolution (recursive or iterative)

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No. of question records

Contains the number of queries in the question section of the msg
No. of answer records
Contains the number of answer records in the answer section of the response msg.
value is zero in the query msg
No. of authoritative records
 Contains the number of authoritative records in the authoritative section of a
response msg
 value is zero in query msg
No. of additional records
 Contains the number of additional records in the additional section of a response
msg
 value is zero in query msg

Question Section
 Consist of one or more question records
 Present on both query and response msg

Answer Section
 Consist of one or more resource records
 Present only on response msg
 It includes the answer from the server to the client (resolver)
Authoritative Section
 Consist of one or more resource records
 Present only on response msg
 It gives info (domain name ) about one or more authoritative servers for the query

Additional Information Section

 Consist of one or more resource records
 Present only on response msg
 It gives additional info (domain name ) that help the resolver

DNS can use the services of UDP or TCP, using the well-known port 53.

SMTP

Mail : Exchanges info between people

Format of an email

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Addresses

 To deliver mail, an addressing system used by SMTP consists of 2 parts

 Local part : defines the name of a specific file called mail box
 All the mail received for a user is stored in the mail box for retrieval by the user
agent
 Domain Name : comes from the DNS database or is a logical name (name of the
organization)

Email address

User agent
A s/w package that composes, reads, replies to, and forward messages.

Some examples of command-driven user agents are mail, pine, and elm
Some examples of GUI-based user agents are Eudora, Outlook, and Netscape.

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MIME:
It converts a Non-ASCII code to ASCII code.

MIME Header:

Data types and subtypes in MIME

Type Subtype Description

Text Plain Unformatted text
Body contains ordered
Mixed
parts of different data types
Same as above, but no
Parallel
order
Multipart
Similar to mixed, but the
Digest
default is message/RFC822
Parts are different versions
Alternative
of the same message
Body is an encapsulated
RFC822
message
Body is a fragment of a
Message Partial
bigger message
Body is a reference to
Ext. Body
another message
JPEG Image is in JPEG
Image
GIF Video is in GIF format

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Video MPEG Video is in MPEG format

Single-channel encoding of
Audio Basic
voice at 8 KHz
PostScript Adobe PostScript
Application Octet- General binary data (8-bit
Stream bytes)

Content-transfer encoding

Category Description
Type ASCII characters and short lines
7bit ASCII characters and short lines
8bit Non-ASCII characters and short lines
Non-ASCII characters with unlimited-
Binary
length lines
6-bit blocks of data are encoded into
Base64
8-bit ASCII characters

Quoted Printable : Non-ASCII characters are encoded as an equal sign followed by an

ASCII code.

Base64

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Base64 encoding table

Value Code Value Code Value Code Value Code Value Code Value Code

0 A 11 L 22 W 33 h 44 s 55 3
1 B 12 M 23 X 34 i 45 t 56 4
2 C 13 N 24 Y 35 j 46 u 57 5
3 D 14 O 25 Z 36 k 47 v 58 6
4 E 15 P 26 a 37 l 48 w 59 7
5 F 16 Q 27 b 38 m 49 x 60 8
6 G 17 R 28 c 39 n 50 y 61 9
7 H 18 S 29 d 40 o 51 z 62 +
8 I 19 T 30 e 41 p 52 0 63 /
9 J 20 U 31 f 42 q 53 1 䦋㌌㏒䦋좈 琰茞䦋䦋Ü 䦋㌌㏒䦋좈 琰
10 K 21 V 32 g 43 r 54 2 䦋㌌㏒䦋좈 琰茞䦋䦋Ü 䦋㌌㏒䦋좈 琰

Quoted Printable:

MTA

 Actual mail transfer is done through MTA

 To send mail a system must have a client MTA and to receive a mail a server
MTA
 Mail transfer occurs between two mail servers

MTA client and server

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Commands and responses

 Uses commands and responses to transfer messages between an MTA client and
an MTA server
 Command or reply is terminated by a two character end–of–line token
Commands
 Sent from client to server
 Consist of a keyword followed by zero or more arguments
Responses
 Sent from server to the client
 Response is a three digit code that may be followed by additional textual
information
Mail transfer
 Transferring a mail message occurs in 3 phases
Connection establishment
 After a client has made a TCP connection to the well known port 25 ,SMTP
server starts the connection phase.
Message transfer
Message between a sender and one or more recipients can be exchanged.

 Connection Termination
After the message is transferred ,the client terminates the connection

Mail Delivery
Consists of 3 stages
Ist stage
 Email goes from user agent to the local server.
 Mail does not go directly to the remote server.
 Mail is stored in the local server until it can be sent.
 User agent uses SMTP client s/w and the local server uses SMTP server s/w.

Second stage

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 Email is relayed by local server, which now acts as SMTP client to the remote
server, which is the SMTP server in this stage
 Email is delivered to the remote server ,not to the remote user agent

Third stage
 The remote user agent uses a mail access protocol such as POP3 or IMAP4 to
access the mailbox and the mail

Mail Access Protocols

 SMTP is involved in the first and second stages but not in third stage, since it is a
push protocol (pushes msgs from sender to receiver).
 The third stage needs a pull protocol
 Operation must start with the recipient
 Mail must stay in the mail server mailbox until the recipient receives it
 Third stage uses a mail access protocol(POP3,IMAP4)

POP3
 Simple but limited in functionality
 Mail access starts with the client when the user needs to download email from the
mailbox on the mail server
 Client (user agent opens a connection with the server on TCP port 110.
 It sends its user name and password to access the mailbox
 User can then list and retrieve the mail messages one by one
 POP3 has two modes
Delete mode& Keep mode
Delete mode
 Mail is deleted from the mail box after each retrieval

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 Normally used when the user is working at permanent computer and save and
organize the received mail after reading or replying
Keep mode
 Normally used when the user accesses mail away from primary computer. Mail is
read but kept in the system for later retrieval and organizing.

 Assumes that each time a client accesses the server, the whole mailbox will be
cleared out
 Not convenient when access their mailboxes from different clients (home or
hotel)
IMAP4
 Powerful and more complex.
 User can check the email header prior to downloading.
 User can check the contents of email for a specific string of characters prior to
downloading.
 Can partially download email.
 User can create, delete or rename mailboxes on the mail server.
 Can create a hierarchy of mailboxes in a folder for email storage.

FTP
 For copying a file from one host to another
FTP uses the services of TCP. It needs two TCP connections. The well-known port 21
is used for the control connection, and the well-known port 20 is used for the data
connection

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 When a user starts an FTP session control connection opens

 While the control connection is open, the data connection can be opened and close
multiple times if several files are transferred
 Conn remains open during the entire process
 Service type used is minimize delay
 User types commands and expects to receive responses without significant delay
Data connection
 Uses the well known port 20 at the server site
 Connection open when data ready to transfer
 Closed when it is not needed
 Service type used is maximize throughput
Communication
 FTP client and server run on different computers
 Must communicate with each other
 May use different operating system, diff character sets, diff file structures and diff
file formats
 FTP make this compatible
 FTP has 2 diff approaches. one for ctrl conn & the other for data conn
Communication over ctrl conn
 Same approach as SMTP
 Uses the ASCII character set.
 Communication is achieved through commands and responses
 Each line is terminated with a two-character (carriage return and line feed) end-
of-line token

Communication over data conn

 To transfer files through data conn, client must

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 Define the type of file

 Structure of the data
 Transmission mode
Heterogeneity solved by
File type ,data structure and transmission mode
Using the data connection

File transfer

HTTP
 Used mainly to access data on the www
 The protocol transfers data in the form of plain text, hyper text, audio and video
and so on.
 A client sends a request ,which looks like mail to the server
 The server sends the response which looks like a mail reply to the client
The request and response messages carry data in the form of a letter with a MIME-like
format

HTTP uses the services of TCP on well-known port 80.

Request Message

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Request Line:

Request msg

Request type:
Several request types are defined
RT categorizes the request msgs into several methods
URL
A client that wants to access a web page needs an address.
To facilitate access of documents ,it uses URL.
It defines 4 things
 Method: a protocol used to retrieve the document (FTP and HTTP)
 Host : a computer where info is located
 Port number of server
 Path name of file where info is located
 Current version is HTTP 1.1

Methods
 Request type defines several kinds of messages referred as methods
 Request method is the actual command or request that a client issues to the server
 GET :if the client wants to retrieve the document from the server

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 HEAD: if the client wants some info about the document

 POST: used by the client to provide some info to the server

 PUT: used by the client wants to provide a new or replacement document to be

stored on the server
 PATCH : similar to PUT with some differences to be implemented with the
existing file.
 COPY : copies a file to another location
 MOVE: moves a file to another location
 DELETE : removes a document on the server
 LINK : creates a link or links from a doc to another location
 UNLINK: deletes the link created by the link

Response Message:

Status Line:

Header Format:

 Exchange additional info between the client and the server

 Example:

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Client can request that the doc be sent in a special format

Server can send extra info about the document
 General header: gives info about the msg
 Request Header: can only be in response msg
Specifies the servers configuration and special info about the request
 Entity header :gives info about the body of the document

Other features
 Persistent Connection: the server leaves the connection open for more requests
after sending a response.
 Non-Persistent Connection: one TCP connection is made for each request and
response.
HTTP version 1.1 specifies a persistent connection by default

WWW
 Repository of info spread all over the world and linked together
 It has a unique combination of flexibility, portability and user-friendly features .
 It is a distributed client-server service.
 A client using a browser can access a service using a server.
 The service provided is distributed over many locations called websites.

Hypertext and hypermedia

 Info is stored in a set of documents that are linked using the concept of pointers
 An item can be associated with another document by a pointer
 Hypermedia: It can contain pictures , graphics and sound
 A unit of Hypertext or hypermedia available on the web is called a page

Hypertext:

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Browser Architecture:

Categories of Web documents

Static Document :

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HTML
 A language for creating web pages.
 Allows to embed formatting instructions in the file itself.

Boldface tags

Effect of boldface tags

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Structure of a Web page

 A web page is made of 2 parts

 Head and Body
 Head : contains the title of the page

Beginning and ending tags

Common tags

Beginning Ending
Meaning
Tag Tag

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Skeletal Tags

<HTML> </HTML> Defines an HTML document

<HEAD> </HEAD> Defines the head of the document

<BODY> </BODY> Defines the body of the document

Title and Header Tags

<TITLE> </TITLE> Defines the title of the document

<Hn> </Hn> Defines the title of the document

Beginning Ending
Meaning
Tag Tag

Text Formatting Tags

<B> </B> Boldface

<I> </I> Italic
<U> </U> Underlined
<SUB> </SUB> Subscript
<SUP> </SUP> Superscript
Data Flow Tag

<CENTER> </CENTER> Centered

<BR> </BR> Line break

Beginning Ending
Meaning
Tag Tag
List Tags
<OL> </OL> Ordered list
<UL> </UL> Unordered list

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<LI> </LI> An item in a list

Image Tag
<IMG> 䦋㌌㏒䦋좈 琰茞䦋䦋ÜDefines an image

Hyperlink Tag

<A> </A> Defines an address (hyperlink)

Executable Contents

<APPLET> </APPLET> The document is an applet

This example shows how tags are used to let the browser format the appearance of the
text

Dynamic Document

 Do not exist in a predefined format

 It is created by a web server whenever a browser requests the document

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Active document

Skeleton of an applet

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Instantiation of the object defined by an applet

Creation and compilation

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HTML document carrying an applet

In this example, we first import two packages, java.awt and java.applet. They
contain the declarations and definitions of classes and methods that we need. Our
example uses only one publicly inherited class called First. We define only one
public method, paint. The browser can access the instance of First through the

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public method paint. The paint method, however, calls another method called
drawString, which is defined in java.awt.*.

import java.applet.*;
import java.awt.*;

public class First extends Applet

{
public void paint (Graphics g)
{
g.drawString ("Hello World", 100, 100);
}
}

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UNIT II

DATA LINK LAYER

ERROR DETECTION AND CORRECTION

ERROR:
Data can be corrupted during transmission. For reliable communication, errors must be detected and corrected. Signals
flows from one point to another. This is subjected to unpredictable interferences from heat, magnetism and other forms of
electricity.

TYPES OF ERRORS:
 Single bit Error:
The term single bit error means that only one bit of a given data unit is changed from 1 to 0 or 0 to 1. 010101 is changed
to 110101 here only one bit is changed by single bit error.

 Burst Error:
A burst error means that 2 or more bits in the data unit have changed.
Example:

Three kinds of errors can occur:

 the bits in the frame can be inverted, anywhere within the frame including the data bits or the frame's control bits,
 additional bits can be inserted into the frame, before the frame or after the frame and
 Bits can be deleted from the frame.

DETECTION
Redundancy
Error detection use the concept of redundancy, which means adding extra bits for detecting errors at the destination .i.e.,
instead of repeating the entire data stream, a shorter group of bits may be appended to the end of each unit.

 To detect or correct errors, we need to send extra (redundant) bits with data.
 The receiver will be able to detect or correct the error using the extra information.
 Detection
◦ Looking at the existence of any error, as YES or NO.
◦ Retransmission if yes. (ARQ)
 Correction
◦ Looking at both the number of errors and the location of the errors in a message.
◦ Forward error correction. (FEC)
Coding
 Encoder vs. decoder
 Both encoder and decoder have agreed on a detection/correct method in priori.

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Modulo Arithmetic

 In modulo-N arithmetic, we use only the integers in the range 0 to N−1, inclusive.
 Calculation
◦ If a number is greater than N−1, it is divided by N and the remainder is the result.
◦ If it is negative, as many N’s as needed are added to make it positive.
 Example in Modulo-12
◦ 1512 = 312
◦ -312 = 912

Modulo-2 Arithmetic

 Possible numbers are {0, 1}

 Arithmetic
◦ Addition
 0+0=0, 0+1=1, 1+0= 1, 1+1=2=0
◦ Subtraction
 0-0=0, 0-1=-1=1, 1-0=1, 1-1=0
◦ Surprisingly, the addition and subtraction give the same result.
◦ XOR (exclusively OR) can replace both addition and subtraction.

Detection methods

 Parity check
 Cyclic redundancy check
 checksum

Parity check

A redundant bit called parity bit, is added to every data unit so that the total number of 1’s in the unit becomes even (or odd).

SIMPLE PARITY CHECK

 A simple parity-check code is a single-bit error-detecting code in which n = k + 1 with dmin = 2.

 A simple parity-check code can detect an odd number of errors.
In a simple parity check a redundant bit is added to a string of data so that total number of 1’s in the data become even or odd.
The total data bit is then passed through parity checking function. For even parity, it checks for even number of
1’s and for odd parity it checks even number of 1’s. If an error is detected the data is rejected.

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Encoder and decoder for simple parity-check code

 In modulo,
◦ r0 = a3+a2+a1+a0
◦ s0 = b3 +b2+b1 +b0+q0
 Note that the receiver adds all 5 bits. The result is called the syndrome.

Example 1: data to be transmitted = 10110101

 5 1’s in the data
 Parity bit is 1
 Transmitted codeword = 101101011
 If receiver gets 101101011, parity check ok ---accept (OK)
 If receiver gets 101100011, parity check fails ---reject (OK), ask for frame to be re-transmitted
 If receiver gets 101110011, parity check ok ---accept (NOT OK: even number of errors undetected)
 If receiver gets 001100011, parity check ok ---accept (NOT OK: even number of errors undetected)

Let us look at some transmission scenarios. Assume the sender sends the dataword 1011. The codeword created from this
dataword is 10111, which is sent to the receiver. We examine five cases:
1. No error occurs; the received codeword is 10111. The syndrome is 0. The dataword 1011 is created.
2. One single-bit error changes a1. The received codeword is 10011. The syndrome is 1. No dataword is created.
3. One single-bit error changes r0. The received codeword is 10110. The syndrome is 1. No dataword is created.
4. An error changes r0 and a second error changes a3 . The received codeword is 00110. The syndrome is 0. The dataword
0011 is created at the receiver. Note that here the dataword is wrongly created due to the syndrome value.
5. Three bits—a3, a2, and a1—are changed by errors. The received codeword is 01011. The syndrome is 1. The dataword is
not created. This shows that the simple parity check, guaranteed to detect one single error, can also find any odd number
of errors.

CYCLIC REDUNDANCY CHECK

CRC is based on binary division. In CRC, instead of adding bits to achieve the desired parity, a sequence of redundant
bits, called the CRC or the CRC remainder, is appended to the end of the data unit so that the resulting data unit becomes exactly
divisible by a second, predetermined binary number. At its destination, the incoming data unit is assumed to be intact and is
therefore accepted. A remainder indicates that the data unit has been damaged in transit and therefore must be rejected.

STEP BY STEP PROCEDURE

 Dividing the data unit by a predetermined divisor derives the redundancy bits used by CRC; the remainder is CRC.
 First a starting of n 0’s is appended to the data unit. The number n is one less than the number of bits in the
predetermined divisor, which is n+1 bits.
 The newly elongated data unit is divided by the divisor, using a process called binary division. The remainder resulting
from this division is the CRC.
 The CRC of n bits derived in step 2 replaces the appended 0s at the end of the data unit. Note that the CRC may consist
of all 0s.
 The data unit arrives at the receiver data first, followed by the CRC. The receiver treats the whole string as unit and
divides it by the same divisor that was used to find the CRC remainder.
 If the string arrives without error, the CRC checker yields a remainder of zero ad the data unit passes. If the string has
been changed in transit, the division yields a non zero remainder and the data does not pass.

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Architecture of CRC

CRC GENERATOR AND CHECKER

CRC GENERATOR

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Division in the CRC decoder for two cases

Hardwired design of the divisor in CRC

Simulation of division in CRC encoder

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The CRC encoder design using shift registers

General design of encoder and decoder of a CRC code

HAMMING CODE:

•A minimum number of redundancy bits needed to correct any single bit error in the data
•A minimum of 4 redundancy bits is needed if the number of data bits is 4.
•Redundancy bits in the Hamming code are placed in the codeword bit positions that are a power of 2
•Each redundancy bit is the parity bit for a different combination of data bits
•Each data bit may be included in more than one parity check.

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 Easy way to compute the redundancy bit values: write down binary representations for positions of data bits which
contain a 1; compute parity bits for each “column”; put parity bits into codeword in correct order.
 Here: data is 1001101 so codeword will look like 100x110x1xx (where x denotes redundancy bits) 1’s in positions 3, 6,
7, and 11
11: 1 0 1 1
7: 0 1 1 1
6: 0 1 1 0
3: 0 0 1 1
1001
parity bits
r1 r2 r4 r8
So codeword is 10011100101(as before)
suppose that the bit in position 7 is received in error:

• If the transmitted codeword is received error-free, the “new” parity bits the receiver computes will all be 0 ,the receiver knows
no bit errors occurred.
• This simple form of Hamming code can be used to provide some protection against burst errors, by transmitting 1st bit from
every codeword to be transmitted, then 2nd bit from every one of these codeword, and so on…In some cases, burst errors can be
corrected

FLOW CONTROL AND ERROR CONTROL

The two main features of data kink layer are flow control and error control.

FLOW CONTROL
Flow control coordinates that amount of data that can be sent before receiving ACK It is one of the most important duties
of the data link layer.

ERROR CONTROL
 Error control in the data link layer is based on ARQ (automatic repeat request), which is the retransmission of data.
 The term error control refers to methods of error detection and retransmission.
 Anytime an error is detected in an exchange, specified frames are retransmitted. This process is called ARQ.

FLOW AND ERROR CONTROL MECHANISMS

1. STOP-AND WAIT ARQ.
2. GO-BACK-N ARQ.
3. SELECTIVE-REPEAT ARQ.

STOP-AND- WAIT ARQ

This is the simplest flow and error control mechanism. It has the following features.
 The sending devise keeps the copy of the last frame transmitted until it receives an acknowledgement for that frame.
Keeping a copy allows the sender to re- transmit lost or damaged frames until they are received correctly.
 Both data and acknowledgement frames are numbered alternately 0 and 1. A data frame 0 is acknowledged by an ACK
 A damaged or lost frame is treated in the same manner by the receiver. If the receiver detects an error in the received
frame, it simply discards the frame and sends no acknowledgement.
 The sender has a control variable, which we call S, that holds the number of recently sent frame. The receiver has a
control variable, which we call R that holds the number of the next frame expected.
 The sender starts a timer when it sends a frame. If an ACK is not received within an allotted time period the sender
assumes that the frame was lost or damaged and resends it.

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OPERATION:

The possible operations are

Normal operation
lost frame
ACK lost
delayed ACK.

The sender sends frame 0 and wait to receive ACK 1. when ACK 1 is received it sends frame 1 and then waits to receive ACK 0,
and so on.

The ACK must be received before the time out that is set expires. The following figure shows successful frame transmission.

Sender Receiver

S=0 Frame 0
R=0
ACK 1
S=1 Frame 1
R=1
ACK 0
S=0





Time Time

Lost or damaged acknowledgement

o When the receiver receives the damaged frame it discards it, which essentially means the frame is lost. The receiver
remains silent about a lost frame and keeps its value of R.
o For example in the following figure the sender transmits frame 1, but it is lost. The receiver does nothing, retaining
the value of R (1). After the timer at the sender site expires, another copy of frame 1 is sent.

Sender Receiver

S=0 Frame 0
R=0
ACK 1
S=1 Frame 1
R=1
Lost
S=1 Frame 1 R=1
Time-out
ACK0 R=0
S=0

Lost acknowledgement
o A lost or damaged ACK is handle in the same by the sender; if the sender receives a damaged ACK, it discards it.
o The following figure shows a lost ACK 0.the waiting sender does not know if frame 1 has been received. When the
timer for frame 1 expires the sender retransmits frame 1.
o Note that the receiver has already received frame 1 and is expecting to receive frame 0. Therefore, its silently
discards the second copy of frame 1.

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Sender Receiver

S=0 Frame 0
R=0
ACK 1
S=1 Frame 1
R=1
ACK 0
Time-out S=1 Lost
Frame 1
R=0
ACK0
S=0 Expecting frame 0
Frame 1 is discarded

Time Time

 Delayed acknowledgement

o An ACK can be delayed at the receiver or by some problem with the link. The following figure shows the delay of
ACK 1; it ids received after the timer for frame 0 as already expired.

o The sender has already retransmitted a copy of frame 0. The receiver expects frame 1 so its simply discards the
duplicate frame 0.

o The sender has now received two ACK’s, one that was delayed and one that was sent after the duplicate frame 0
arrived. The second ACK 1 is discarded.

Sender Receiver

S=0 Frame 0
R=0
Time-out ACK1
S=0 Frame0 R=1

S=1 Frame 1
ACK1 Expecting frame 0
Frame 1 is discarded
Discarded
S=1 Frame1
Time-out R=1
ACK0

Time Time

BIDIRECTIONAL TRANSMISSION
The stop – and – wait mechanism is unidirectional. We can have bi-directional transmission if the two parties have two
separate channels for full duplex communication or share the same channel for off duplex transmission. In this case, each party
needs both S and R variables to track frames sent and expected.
PIGGYBACKING
It’s a method to combine a data frame with an ACK. In following figure both the sender and the receiver have data to
send. Instead of sending separate data and ACK frames. It can save bandwidth because the overhead from a data frame and an
ACK frame can be combined into just one frame

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Sender Receiver

R=0
S=0 Frame 0,ACK0 R=0
S=0
R=0 Frame0,ACK 1
S=1
Frame 1,ACK1 R=1
S=1
R=1 Frame1,ACK 0
S=0

Time Time

GO-BACK-N ARQ
 As in Stop-and-wait protocol senders has to wait for every ACK then next frame is transmitted. But in GO-BACK-N
ARQ number of frames can be transmitted without waiting for ACK. A copy of each transmitted frame is maintained
until the respective ACK is received.

Features of GO-BACK-N ARQ

1. Sequence numbers.
Sequence numbers of transmitted frames are maintained in the header of frame. If k is the number of bits for sequence
number, then the numbering can range from 0 to 2k-1. Example: if k=3 means sequence numbers are 0 to 7.
2. Sender sliding window:
 Window is a set of frames in a buffer waiting for ACK. This window keeps on sliding in forward direction, the window
size is fixed. As the ACK is received, the respective frame goes out of window and new frame to sent come into window.
Figure illustrates the sliding window.
 If Sender receives. ACK 4, then it knows Frames up to and including Frame 3 were correctly received

Window size=7

4. Receiver sliding window:

In the receiver side size of the window is always one. The receiver is expecting to arrive frame in specifies sequence. Any other
frame is received which is out of order is discarded. The receiver slides over after receiving the expected frame. The following
figure shows the receiver side-sliding window.

6 7 0 1 2 3 4 5 6 7

4. Control variables:
Sender variables and Receiver variables: The receiver deals with only one variable
Sender deals with three different variables R -> sequence number of frame expected.
S -> sequence number of recently sent frame
SF -> sequence number of first frame in the window.
S -> sequence number of last frame in the window.

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5. Timers
The sender has a timer for each transmitted frame. The receivers don’t have any timer.
6. Acknowledgement:
The receiver responds for frame arriving safely by positive ACK. For damaged or lost frames receiver doesn’t reply, the
sender has to retransmit it when timer of that frame elapsed. The receiver may ACK once for several frames.
7. Resending frames:
If the timer for any frame expires, the sender sender has to resend that frame and the subsequent frame also, hence the
protocol is called GO-BACK-N ARQ.
Operation
Normal operation: Following diagram shows this mechanism. The sender keeps track of the outstanding frames and updates the
variables and windows as acknowledgements arrive.

SF SL Receiver
Sender

Frame 0
0 1 2 3 0 1 0 1 2 3 0 1

S Frame 1
0 1 2 3 0 1
0 1 2 3 0 1 ACK 2

S Frame 2
0 1 2 3 0
0 1 2 3 0 1 ACK3

S Frame 3 0 1 2 3 0 1
0 1 2 3 0 

S
Time Time
Damaged or lost frame:
Figure shows that frame 2 is lost. Note that when the receiver receives frame 3, it is discarded because the receiver is expecting
frame 2, not frame3. after the timer for frame 2 expires at the sender site, the sender sends frame 2 and 3.

SF SL
Sender Receiver

Frame 0
0 1 2 3 0 1 0 1 2 3 0 1

S Frame 1
0 1 2 3 0 1
0 1 2 3 0 1 ACK 2

S Frame 2 lost
0 1 2 3 0
0 1 2 3 0 1
Frame 3
0 1 2 3 0 1 Frame 3 is discarded, not
Timeout in the window
resent Frame 2
0 1 2 3 0 1
0 1 2 3 0 1
Frame 3
resent

0 1 2 3 0  0 1 2 3 0 1

S R

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Damaged or lost acknowledgement:
If an ACK is lost, we can have two situations. If the next ACK arrives before the expiration of timer, there is no need for
retransmission of frames because ACK are cumulative in this protocol.. if the next ACK arrives after the timeout, the frame and
all the frames after that are resent. The receiver never resends an ACK.
For diagrams refer your class work notes.

Delayed Acknowledgement:
A delayed ACK also triggers the resending of frames.
SELECTIVE REPEAT ARQ:
 The configuration and its control variables for this are same as those selective repeat ARQ.
 The size of the window should be one half of the value 2m.
 The receiver window size must also be the size. In this the receiver is looking for a range of sequence numbers.
 The receiver has control variables RF and RL to denote the boundaries of the window.

Selective repeat also defines a negative ACK NAK that reports the sequence number of a damaged frame before the timer expires.

Operation

Normal operation
Normal operations of the selective repeat ARQ are same as GO-BACK-N ARQ mechanism.
Lost or damaged frame
The following figure shows operation of the mechanism with an example of a lost frame.
Frame 0 and 1 are accepted when received because they are in the range specified by the receiver window. When frame 3
is received, it is also accepted for the same reason. However the receiver sends a NAK 2 to show that frame 2 has not been
received. When the sender receives the NAK 2, it resends only frame 2, which is then accepted because it is in the range of
the window.

Sender Receiver

Frame 0
0 1 2 3 0 1 0 1 2 3 0 1

S Frame 1
0 1 2 3 0 1
0 1 2 3 0 1 ACK 2

S Frame 2 lost

0 1 2 3 0 1
Frame 3
0 1 2 3 0 1 NAK 2 0 1 2 3 0 1
Frame 2
0 1 2 3 0 1 resent 0 1 2 3 0 1

Time Time

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Lost and delayed ACKs and NAKs

In this sender also sets a timer for each frame sent. The remaining operations are same as GO-BACK-N ARQ.
High-level Data Link Control (HDLC) protocol

 HDLC standardized ISO in 1979 and accepted by most other standards bodies (ITU-T, ANSI)
 3 types of end-stations:
Primary–sends commands
Secondary–can only respond to Primary’s commands
Combined–can both command and respond
 3 types of configuration
(Note: no balanced multipoint)

TRANSFER MODE
 Mode = relationship between 2 communicating devices;
 Describes who controls the link
o NRM = Normal Response Mode
o ABM = Asynchronous Balanced Mode
NRM:
Only difference is that secondary needs permission from the Primary in NRM, but doesn’t need permission from the
Primary in ARM.

FRAMES:
Three types of Frames are
I-Frame – transports user data and control info about user data.
S-Frame – supervisory Frame, only used for transporting control information
U-Frame – unnumbered Frame, reserved for system management (managing the link itself)

FRAME FORMAT

U-Frames:

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 U-frames are used for functions such as link setup. They do not contain any sequence numbers.
 Five code bits denote the frame type (but there are not 32 different possibilities):
 Set Asynchronous Balanced Mode (SABM).Used in the link set up to indicate ABM mode will be used.
 Set Normal Response Mode (SNRM).Used for asymmetric mode (master/slave).
 SABME and SNMRE—extended format.
 Disconnect (DISC).Used to disconnect the logical connection.
 Frame Reject (FRMR)—reject frame with incorrect semantics.
 Unnumbered Acknowledgement (UA).Used to acknowledge other frames in this class.
 Unnumbered Information (UI)–initialisation, poling and status information needed by the data link layer.
 U-frames may carry data when unreliable connectionless service is called for.

S-Frames:

 S-frames are similar to unnumbered frames, the main difference being that they do carry sequence information.
 Some supervisory frames function as positive and negative acknowledgements, they therefore play a very important role
in error and flow control.
 Two bits indicate the frame type, so that there are four possibilities.
 Receiver Ready -RR(Positive Acknowledgement)
 Receiver Not Ready -RNR
 Reject -REJ(NAK go-back-N)
 Selective Reject -SREJ(NAK selective retransmit)

Control Field:

IEEE 802.4 TOKEN BUS

 IEEE 802.4 describes a token bus LAN standard.
 In token passing method stations, connected on a bus are arranged in a logical ring. When the logical ring is initiated, the
highest number station may send the first frame. After this it passes permission to its immediate neighbor by sending a
special frame called a token.
 The token propagates around the logical ring, with only the token holder being permitted to transmit frames. Since only
one station at a time holds the token, collision do not occur.
 There is no relation between the physical location of the station on the bus and its logical sequence number..
The following figure shows the operation of the token bus.
Physical topology

90 50 120

Logical sequence of token passing 400 75

50 75

400
Token passing in a bus

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802.4 cable standards
 The token bus standard specifies three physical layer options in terms of transmission medium, signaling technique, data
rate and maximum electrical cable segment length.
Medium options
1. Broadband: Transmission medium is co-axial cable and its uses AM/PSK as signaling techniques, data rate is 1,5,10 mbps.
2. Carrier band: Transmission medium is co-axial cable and its uses KSK as a signaling techniques, data rate is 1, 5,10Mbps.
3. Optical fiber: Transmission medium is optical fiber and its uses ASK with Manchester encoding as a signaling techniques,
data rate is 5,10,20Mbps.

IEEE 802.4 Frame format

 Token bus frame format is shown in the following figure.
1 1 1 2-6 2-6 0-8182 4

pr
Preamble SD FC DA SA DATA FCS ED

 Preamble: the preamble is an at least one byte long pattern to establish bit synchronization
 SD: Start frame delimiter: Its also one byte unique bit pattern, which marks the start of the frame.
 FC: Frame control: The frame control field is used to distinguish data frames from control frames. For data frame, it
carries the frames priority. The frame control field indicates the type of the frame data frame or control frame.
 DA: Destination address: The destination address field is 2 or 6 bytes long.
 SA: Source address: The destination address field is 2 or 6 bytes long.
 DATA: Data field
 FCS: Frame check sequence: frame check sequence is 4 bytes long and contains CRC code. It is used to detect
transmission errors on DA, SA, FC and data fields.
 ED: End delimiter: It is a unique bit pattern, which marks the end of the frame. It is one byte long.
 The total length of the frame is 8191 bytes.

Performance:
For token ring, the slightly higher delay compared to CSMS/CD bus occurs. For higher transmission loads the token ring
performs well.

IEEE 802.5 TOKEN RING

 IEEE 802.4 describes a token ring LAN standard.
 In a token ring a special bit pattern, called the token circulates around the ring when all stations are idle.
 When a station transmits, it breaks the ring and inserts its own frame with source and destination address.
 When the frame eventually returns to the originating station after completing the round, the station removes the frame
and closes the ring. Because there is only one token, only one station can transmit at a given instant, thus solving the
channel access problem.
 Each station is connected to the ring through a Ring Interface Unit (RIU). The sequence of token is determined by the
physical locations of the stations on the ring.

The following figure shows the operation and arrangement of the Token Ring.

stations

Unidirectional bus

Ring interface

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802.5 cable standards

Its uses two types of transmission medium.

1. Shielded twisted pair cable: (STP)

It uses differential Manchester encoding technique. Data rate is 4 or
16 Mbps. Maximum number of repeaters allowed is 250.
2. Unshielded twisted pair cable: (UTP)
It uses differential Manchester encoding technique. Data rate is 4Mbps. Maximum number of repeaters
allowed is 250.

IEEE 802.5 Frame format

 Token ring frame format is shown in the following figure.

Token frame format

1 1 1 byte
SD AC ED

Data Frame
1 1 1 2-6 2-6 No limit 4 1 1

SDpr AC FC DA SA DATA FCS ED FS

 SD: Start frame delimiter: Its also one byte unique bit pattern, which marks the start of the frame.
 AC: Access control: It is one byte long field containing priority bits(P), Token bit(T), monitoring bit(M), and reservation
bir(R).
 FC: Frame control: The frame control field is used to distinguish data frames from control frames. For data frame, it
carries the frames priority. The frame control field indicates the type of the frame data frame or control frame.
 DA: Destination address: The destination address field is 2 or 6 bytes long.
 SA: Source address: The destination address field is 2 or 6 bytes long.
 DATA: Data field
 FCS: Frame check sequence: frame check sequence is 4 bytes long and contains CRC code. It is used to detect
transmission errors on DA, SA, FC and data fields.
 ED: End delimiter: It is a unique bit pattern, which marks the end of the frame. It is one byte long.
 FS: Frame status: This field is none byte long and contains a unique bit pattern marking the end of a token or a data
frame.

Performance:
When traffic is light, the token will spend most of its time idly circulating around the ring. When traffic is heavy, there is a
queue at each station. Network efficiency is more.

Disadvantages:
 A break in a link or repeater failures disturbs the entire network.
 Installation of new repeaters requires identification of two topologically adjacent repeaters.
 Since the ring is closed loop, a packet will circulate indefinitely unless it is removed.
 Each repeater adds an increment of delay.
 There is practical limit to the number of repeaters.

Fiber Distributed Data Interface

Introduction

The Fiber Distributed Data Interface (FDDI) specifies a 100-Mbps token-passing, dual-ring LAN using fiber-optic cable.
FDDI is frequently used as high-speed backbone technology because of its support for high bandwidth and greater distances than
copper. It should be noted that relatively recently, a related copper specification, called Copper Distributed Data Interface (CDDI),
has emerged to provide 100-Mbps service over copper. CDDI is the implementation
of FDDI protocols over twisted-pair copper wire. This chapter focuses mainly on FDDI specifications and operations, but it also
provides a high-level overview of CDDI.

FDDI uses dual-ring architecture with traffic on each ring flowing in opposite directions (called counter-rotating). The dual rings
consist of a primary and a secondary ring. During normal operation, the primary ring is used for data transmission, and the

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secondary ring remains idle. As will be discussed in detail later in this chapter, the primary purpose of the dual rings is to provide
superior reliability and robustness. Figure 8-1 shows the counter-rotating primary and secondary FDDI rings.

FDDI Specifications

FDDI specifies the physical and media-access portions of the OSI reference model. FDDI is not actually a single specification, but
it is a collection of four separate specifications, each with a specific function. Combined, these specifications have the capability
to provide high-speed connectivity between upper-layer protocols such as TCP/IP and IPX, and media such as fiber-optic cabling.

FDDI's four specifications are the Media Access Control (MAC), Physical Layer
Protocol (PHY), Physical-Medium Dependent (PMD), and Station Management (SMT) specifications. The MAC specification
defines how the medium is accessed, including frame format, token handling, addressing, algorithms for calculating cyclic
redundancy check (CRC) value, and error-recovery mechanisms. The PHY specification defines data encoding/decoding
procedures, clocking requirements, and framing, among other functions. The PMD specification defines the characteristics of the
transmission medium, including fiber-optic links, power levels, bit-error rates, optical components, and connectors. The SMT
specification defines FDDI station configuration, ring configuration, and ring control features, including station insertion and
removal, initialization, fault isolation and recovery, scheduling, and statistics collection.

FDDI is similar to IEEE 802.3 Ethernet and IEEE 802.5 Token Ring in its relationship with the OSI model. Its primary purpose is
to provide connectivity between upper OSI layers of common protocols and the media used to connect network devices. Figure 8-
3 illustrates the four FDDI specifications and their relationship to each other and to the IEEE-defined Logical Link Control (LLC)
sublayer. The LLC sublayer is a component of Layer 2, the MAC layer, of the OSI reference model.

Figure 8-3: FDDI Specifications Map to the OSI Hierarchical Model

FDDI Frame Format

The FDDI frame format is similar to the format of a Token Ring frame. This is one of the areas in which FDDI borrows heavily
from earlier LAN technologies, such as Token Ring. FDDI frames can be as large as 4,500 bytes. Figure 8-10 shows the frame
format of an FDDI data frame and token.

Figure 8-10: The FDDI Frame Is Similar to That of a Token Ring Frame

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FDDI Frame Fields

The following descriptions summarize the FDDI data frame and token fields illustrated in Figure 8-10.

 Preamble—Gives a unique sequence that prepares each station for an upcoming frame.
 Start delimiter—Indicates the beginning of a frame by employing a signaling pattern that differentiates it from the rest of
the frame.
 Frame control—Indicates the size of the address fields and whether the frame contains asynchronous or synchronous
data, among other control information.
 Destination address—Contains a unicast (singular), multicast (group), or broadcast (every station) address. As with
Ethernet and Token Ring addresses, FDDI destination addresses are 6 bytes long.
 Source address—Identifies the single station that sent the frame. As with Ethernet and Token Ring addresses, FDDI
source addresses are 6 bytes long.
 Data—Contains either information destined for an upper-layer protocol or control information.
 Frame check sequence (FCS)—Is filed by the source station with a calculated cyclic redundancy check value dependent
on frame contents (as with Token Ring and Ethernet). The destination address recalculates the value to determine
whether the frame was damaged in transit. If so, the frame is discarded.
 End delimiter—Contains unique symbols; cannot be data symbols that indicate the end of the frame.
 Frame status—Allows the source station to determine whether an error occurred; identifies whether the frame was
recognized and copied by a receiving station.

Dual Ring

FDDI's primary fault-tolerant feature is the dual ring. If a station on the dual ring fails or is powered down, or if the cable is
damaged, the dual ring is automatically wrapped (doubled back onto itself) into a single ring. When the ring is wrapped, the dual-
ring topology becomes a single-ring topology. Data continues to be transmitted on the FDDI ring without performance impact
during the wrap condition. Figure 8-6 and Figure 8-7 illustrate the effect of a ring wrapping in FDDI.

Figure 8-6: A Ring Recovers from a Station Failure by Wrapping

Figure 8-7: A Ring also wraps to withstand a Cable Failure

When a single station fails, as shown in Figure 8-6, devices on either side of the failed (or powered-down) station wrap,
forming a single ring. Network operation continues for the remaining stations on the ring. When a cable failure occurs, as shown
in Figure 8-7, devices on either side of the cable fault wrap. Network operation continues for all stations.

It should be noted that FDDI truly provides fault tolerance against a single failure only. When two or more failures occur, the
FDDI ring segments into two or more independent rings that are incapable of communicating with each other.

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The IEEE 802.3 Logical Relationship to the ISO Reference Model

Figure 7-4 shows the IEEE 802.3 logical layers and their relationship to the OSI reference model. As with all IEEE 802 protocols,
the ISO data link layer is divided into two IEEE 802 sublayers, the Media Access Control (MAC) sublayer and the MAC-client
sublayer. The IEEE 802.3 physical layer corresponds to the ISO physical layer.

Figure 7-4 Ethernet's Logical Relationship to the ISO Reference Model

The MAC-client sublayer may be one of the following:

• Logical Link Control (LLC), if the unit is a DTE. This sublayer provides the interface between the Ethernet MAC and the
upper layers in the protocol stack of the end station. The LLC sublayer is defined by IEEE 802.2 standards.

• Bridge entity, if the unit is a DCE. Bridge entities provide LAN-to-LAN interfaces between LANs that use the same protocol
(for example, Ethernet to Ethernet) and also between different protocols (for example, Ethernet to Token Ring). Bridge entities are
defined by IEEE 802.1 standards.

Because specifications for LLC and bridge entities are common for all IEEE 802 LAN protocols, network compatibility becomes
the primary responsibility of the particular network protocol. Figure 7-5 shows different compatibility requirements imposed by
the MAC and physical levels for basic data communication over an Ethernet link.

Figure 7-5 MAC and Physical Layer Compatibility Requirements for Basic Data Communication

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The MAC layer controls the node's access to the network media and is specific to the individual protocol. All IEEE 802.3 MACs
must meet the same basic set of logical requirements, regardless of whether they include one or more of the defined optional
protocol extensions. The only requirement for basic communication (communication that does not require optional protocol
extensions) between two network nodes is that both MACs must support the same transmission rate.

The 802.3 physical layer is specific to the transmission data rate, the signal encoding, and the type of media interconnecting the
two nodes. Gigabit Ethernet, for example, is defined to operate over either twisted-pair or optical fiber cable, but each specific
type of cable or signal-encoding procedure requires a different physical layer implementation.

The Ethernet MAC Sublayer

The MAC sub layer has two primary responsibilities:

 Data encapsulation, including frame assembly before transmission, and frame parsing/error detection during and after
reception
 Media access control, including initiation of frame transmission and recovery from transmission failure
The Basic Ethernet Frame Format
The IEEE 802.3 standard defines a basic data frame format that is required for all MAC implementations, plus several
additional optional formats that are used to extend the protocol's basic capability. The basic data frame format contains the seven
fields shown in Figure 7-6.

• Preamble (PRE)—Consists of 7 bytes. The PRE is an alternating pattern of ones and zeros that tells receiving stations that a
frame is coming, and that provides a means to synchronize the frame-reception portions of receiving physical layers with the
incoming bit stream.

• Start-of-frame delimiter (SOF)—Consists of 1 byte. The SOF is an alternating pattern of ones and zeros, ending with two
consecutive 1-bits indicating that the next bit is the left-most bit in the left-most byte of the destination address.

• Destination address (DA)—Consists of 6 bytes. The DA field identifies which station(s) should receive the frame. The left-
most bit in the DA field indicates whether the address is an individual address (indicated by a 0) or a group address (indicated by a
1). The second bit from the left indicates whether the DA is globally administered (indicated by a 0) or locally administered
(indicated by a 1). The remaining 46 bits are a uniquely assigned value that identifies a single station, a defined group of stations,
or all stations on the network.

• Source addresses (SA)—Consists of 6 bytes. The SA field identifies the sending station. The SA is always an individual
address and the left-most bit in the SA field is always 0.

• Length/Type—Consists of 2 bytes. This field indicates either the number of MAC-client data bytes that are contained in the
data field of the frame, or the frame type ID if the frame is assembled using an optional format. If the Length/Type field value is
less than or equal to 1500, the number of LLC bytes in the Data field is equal to the Length/Type field value. If the Length/Type
field value is greater than 1536, the frame is an optional type frame, and the Length/Type field value identifies the particular type
of frame being sent or received.

• Data—Is a sequence of n bytes of any value, where n is less than or equal to 1500. If the length of the Data field is less than
46, the Data field must be extended by adding a filler (a pad) sufficient to bring the Data field length to 46 bytes.

• Frame check sequence (FCS)—Consists of 4 bytes. This sequence contains a 32-bit cyclic redundancy check (CRC) value,
which is created by the sending MAC and is recalculated by the receiving MAC to check for damaged frames. The FCS is
generated over the DA, SA, Length/Type, and Data fields.

Figure 7-6 The Basic IEEE 802.3 MAC Data Frame Format

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Note: Individual addresses are also known as unicast addresses because they refer to a single MAC and are assigned by the NIC
manufacturer from a block of addresses allocated by the IEEE. Group addresses (a.k.a. multicast addresses) identify the end
stations in a workgroup and are assigned by the network manager. A special group address (all 1s—the broadcast address)
indicates all stations on the network.

Frame Transmission

Whenever an end station MAC receives a transmit-frame request with the accompanying address and data information from the
LLC sublayer, the MAC begins the transmission sequence by transferring the LLC information into the MAC frame buffer.

• The preamble and start-of-frame delimiter are inserted in the PRE and SOF fields.

• The destination and source addresses are inserted into the address fields.

• The LLC data bytes are counted, and the number of bytes is inserted into the Length/Type field.

• The LLC data bytes are inserted into the Data field. If the number of LLC data bytes is less than 46, a pad is added to bring
the Data field length up to 46.

• An FCS value is generated over the DA, SA, Length/Type, and Data fields and is appended to the end of the Data field.

After the frame is assembled, actual frame transmission will depend on whether the MAC is operating in half-duplex or full-
duplex mode.

The IEEE 802.3 standard currently requires that all Ethernet MACs support half-duplex operation, in which the MAC can be
either transmitting or receiving a frame, but it cannot be doing both simultaneously. Full-duplex operation is an optional MAC
capability that allows the MAC to transmit and receive frames simultaneously.

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3.1

What is a switch and its function?

A switch is a multi-input, multi-output device, which transfers packets from an
input to one or more outputs.
Large networks can be built by interconnecting a number of switches. Hosts are
connected to the switch using point-to-point link.
A switch receives packets on one of its links and transmits them on one or more
other links. This is known as switching or forwarding.
List the different types of switched networks.
Circuit switched networks
Packet switched networks
o Datagram networks
o Virtual-circuit networks
Message switched networks
Bring out the differences between circuit and packet switching.

Circuit switching Packet switching

Source and destination host are physically No such physical connection exists
connected
Switching takes place at the physical layer Switching takes place at network (datagram) or
data link layer (VCN)
Resources such as bandwidth, switch buffer & Resources are allocated on demand
processing time, are allocated in advance.
Resources remain allocated for the entire Resources can be reallocated when idle.
duration of data communication.
There is no delay during data transfer. Delay exists at each switch during data transfer
Data transferred between the two stations is a Data is transferred as discrete packets
continuous flow of signal
Example: Telephony Example: Internet

Explain packet switched networks in detail.

Datagram networks
Datagram network is referred to as connectionless network.
In connectionless, the switch does not keep information about the connection state. In
a datagram network, the message is divided into packets of fixed or variable size.
The sender does not know whether the network will deliver or destination host is alive. There
is no resource allocation for a packet. There is no reserved bandwidth on the links,
and no scheduled processing time.
Resources are allocated on demand. The allocation is primarily done on a FCFS basis.
When a packet arrives at a switch, it must wait if there are other packets being
processed. o The lack of reservation creates delay.
Each packet is treated independently of all others regardless of its source or destination.
Packets belonging to the same message may travel different paths to reach their
destination.
Packets can arrive out of order at the destination

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3.2

Packets may also be dropped due to lack of resources.

A switch or link failure may not have any adverse effect, if an alternate path is available.
It is the responsibility of an upper-layer protocol to reorder the datagrams or ask for lost
datagrams before passing them on to the application.
Datagram switching is done at the network layer.

Datagram network Routing table

Routing table
Each switch has a routing table that contains destination address and the
corresponding output port.
When the switch receives a packet:
o destination address is examined
o The routing table is consulted to find the corresponding port through which
the packet should be forwarded.
The routing table is dynamic and is updated periodically.
Analysis
The efficiency of a datagram network is better than that of a circuit-switched
network, since resources are allocated only when there are packets to be sent.
The resources can be reallocated if idle, for other packets.
Each packet experiences a wait at a switch if there are packets queued up,
before it is forwarded.
Virtual Circuit Switching networks
A virtual-circuit network (VCN) is a connection-oriented
model. It is implemented in the data link layer.
A virtual connection from the source to the destination is established before any
data is sent. This is known as connection setup phase
o Each switch should contain an entry in VC table that has four columns.
o The entry contains incoming port, incoming VCI, outgoing port and outgoing
VCI A Virtual Circuit Identifier (VCI) uniquely identifies a connection at a switch.
o A VCI is a small number that has only link local scope.
o A frame arrives at a switch with a VCI and when it leaves, it has a different VCI.
o VCI configured by the administrator is known as permanent virtual circuit (PVC)
o In large networks, VCI is set by a host through signaling, known as
switched virtual circuit (SVC)
In the teardown phase, the source requests the switches to delete corresponding entries.
Permanent Virtual Circuit
The figure shows a frame from source A on its way to destination B and how its
VCI changes during the trip as configured by the network administrator.

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3.3

The frame arrives at port 1 with a VCI of 14.

The switch looks in its table to find port 1 and a VCI of 14. When it is found, the
switch knows to change the VCI to 66 and send out the frame from port 3.
Each switch changes the VCI and routes the frame.
The data transfer phase is active until source sends all its frames to the destination.
This process creates a virtual circuit between the source and destination.
Switched Virtual Circuit
Two steps are involved in the setup phase namely setup request and acknowledgment.
Setup Request—A setup request frame is sent from the source to the destination.
a) Source A sends a setup frame to switch1.
b) Switch1 receives the setup request frame.
o It knows that a frame going from A to B goes out through port 3.
o The switch creates an entry in its table for this virtual circuit.
o The switch assigns the incoming port (1), chooses an available incoming
VCI (14) and the outgoing port (3).
o It does not know the outgoing VCI and is left blank.
o Forwards frame to switch2.
c) Switch2 receives the setup request frame.
o The table entries made are incoming port (l), incoming VCI (66), and
outgoing port (2). The frame is forwarded to switch3.
d) Similarly for switch3 the entries are incoming port (2), incoming VCI (22), and
outgoing port (3).
e) Destination B receives the setup frame, and if it is ready to receive frames from host A,
o It assigns an unused VCI to the incoming frames that come from A, say 77.
o When a frame comes with VCI 77, host B know that frames come from A.
Acknowledgment—A special ACK frame is used to complete entry in the
switching table. a) The destination host B sends an acknowledgment to switch3.
o The ACK frame carries source and destination addresses so the switch
knows which entry in the table is to be completed.
o The frame also carries VCI 77, chosen by the destination as the incoming
VCI for frames from host A.
o Switch 3 uses this VCI to complete the outgoing VCI column for this entry. o
77 is incoming VCI for destination B whereas for switch3 it is outgoing VCI.

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3.4

b) Switch3 sends an acknowledgment to switch2 that contains its incoming VCI in

the table. Switch 2 uses this as the outgoing VCI in the table.
c) Similarly Switch 2 sends an acknowledgment to switch1. The process is the
same and outgoing VCI is updated.
d) Finally switch1 sends an acknowledgment to source A.
e) The source uses the incoming VCI from switch 1 as its outgoing VCI for the data
frames to be sent to destination B.

Analysis
There is at least one RTT delay before data is sent due to setup request and
acknowledgement
The per-packet overhead is reduced since VCI is a small number.
If a switch or link in a connection fails, the connection is teardown and a new one is setup

What is source routing?

All the information about network topology that is required to switch a packet
across the network to the destination is provided by the source host.
The header contains an ordered list of intermediate hosts through which the
packet must traverse.
For each packet that arrives on an input, the switch reads the port number in the
header and transmits the packet on that output.
Source routing can be used in both datagram and virtual circuit networks
Explain Internetworking Protocol in detail.
Internetwork is interconnection of different physical networks to provide host-to-
host packet delivery service.
o Internet is a logical network built on a collection of physical network.
The node that interconnects different networks is known as router.
Internet Protocol (IP) is used to build scalable, heterogeneous internetworks.
The ability of IP to run over any networking technology is its strength.
IP provides neither error control nor flow control.
IP does nothing when a packet gets lost or corrupted. It is an unreliable service.
o If reliability is needed, IP must be paired with a reliable protocol such
as TCP. The IP Service model has two parts
o Datagram (connectionless) model of data delivery
o Addressing scheme that identifies all hosts in the internetwork uniquely.

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3.5

The above diagram shows an internetwork comprising different physical network

such as Ethernet, FDDI ring and point-to-point link.
Each network has a set of hosts Hi and the networks are connected by set of routers Rj.
Datagram Forwarding
All IP datagram contain destination address to enable the network forward
packets correctly in a connectionless manner.
Each router compares network id of the destination address with the network id
of each of its interfaces.
o If a match occurs, then the destination lies on the same physical
network as the interface, and the packet is directly delivered.
o Otherwise, the packet is forwarded to the next hop router after
consulting its forwarding table.
o In case of no match, then the packet is forwarded to the default router.
Example
If H1 sends a datagram to H8, then forwarding is as follows:
o H1 sends datagram to its default router, say R1, since it cannot deliver directly.
o R1 sends datagram to its default router, say R2, since H8 network id does
not match any of its interface.
o R2 forwards the datagram to R3 based on its forwarding table shown below.

o R3 forwards the datagram directly to H8, since both are on the same network.
Packet Format
IPv4 datagram is a variable-length packet consisting of two parts, header and data.
The header is 20–60 bytes and contains information essential to routing and
delivery The minimum packet length is 20 bytes and maximum is 65,535 bytes.

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3.6

Version specifies version of the IPv4 protocol, i.e. 4.

HLen defines length of the datagram header in 4-byte words. When there are no
options, the value is 5 (5 × 4 = 20).
TOS allows packets to be treated differently based on application needs. The
parameters used to distinguish are delay, throughput, reliability and cost
Length specifies the total packet length (header + data). The total length of the
16
IPv4 datagram is restricted to 65,535 bytes (2 - 1).
o If length is large for any lower layer protocols then fragmentation is
done. Ident a 16-bit identifier that uniquely identifies a datagram.
Flags It is a 3-bit field. The first bit is reserved. The second bit (D) is called the do
not fragment bit. The third bit (M) is called the more fragment bit.
Offset shows relative position of this fragment with respect to the whole datagram. It
is offset of the data in the original datagram measured in units of 8 bytes.
TTL defines lifetime of the datagram (default value 64) in hops.
o Each router decrements TTL by 1 before forwarding.
o If the value is zero, the datagram is discarded.
Protocol specifies the higher-level protocol such as (6–TCP, 17–UDP, 1–
ICMP). Checksum contains 16-bit checksum for the packet header.
SourceAddr 32-bit address of the source host.
DestinationAddr 32-bit address of the destination host.
Options If HLen > 5 then options are specified (up to 40 bytes). Some options
are: o Record Route used to record the routers that handle the datagram.
o Strict Source Route used by the source to predetermine a route for the datagram.
Fragmentation
Each physical network has Maximum Transmission Unit (MTU), the largest IP datagram
contained in a frame. MTU for FDDI is 4500, Ethernet is 1500, point-to-point is 532, etc.
The IP datagram is encapsulated in the physical network's frame through which it travels. If
the datagram payload is greater than MTU, then it is fragmented to fit the link-layer
frame. The fragmented packets are each of size MTU, except the last one.
If D flag bit is set, then datagram is not fragmented. If no alternate path is
available, then it is discarded.

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3.7

The router usually fragments the datagram, when it has to forward the packet
over a network that has a smaller MTU. Each fragment is routed independently.
o A fragmented datagram may be further fragmented, if it encounters a
network with a smaller MTU.
When a datagram is fragmented, the Ident field is copied to all fragments. The
identification number helps the destination in reassembling the datagram.
On fragmentation the router changes three fields: Flags, Offset and Length.
The router sets the M bit in the flags field sets the Offset to 0 for the first
fragment. For the last fragment M bit is not set.
IP does not attempt to recover from missing fragments and discards all other
fragments. Reassembly is done at the receiving host and not at each router.
Example
Suppose host H1 sends a datagram to host H8 with a payload of 1400 bytes.
The datagram goes through the ETH and FDDI network without any fragmentation.
When the packet arrives at router R2, which has an MTU of 532 bytes, it is
fragmented with a maximum payload of 512 (plus 20 bytes for IP header)
The three fragments are forwarded by router R3 through Ethernet to the destination host.

th
The data carried in the second fragment starts with 513 byte, so the Offset field
in this header is set to 64 (count of 8-byte chunks)
The third fragment contains the last 376 bytes of data, and Offset is set to 128.

Before fragmentation After fragmentation at R2

Global Addressing
IP addresses are hierarchical, i.e., it corresponds to hierarchy in the
internetwork. IP addresses consist of two parts, network id and host id.
The network id identifies the network to which the host is attached.
o Hosts attached to the same network have the same network id in their IP
address. The host id is used to uniquely identify a host on a network.
The routers have an address on each network, one for each interface.
32
IPv4 uses 32-bit addresses, i.e., address space is 2 (more than 4 billion)
IPv4 address is expressed compactly as four octets (each in the range 0–
255) in dotted decimal notation.

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3.8

IPv4 Classful Addressing

In classful addressing, the address space is divided into five classes: A, B, C, D, and E.
The class of an IP address is identified by seeing the MSBs in binary notation or
first byte in decimal notation.
Class Binary Decimal Application
A 0 0–127 Unicast
B 10 128–191 Unicast
C 110 192–223 Unicast
D 1110 224–239 Multicast
E 1111 240–255 Reserved
Classes A, B and C are used for unicast addressing.
Class D was designed for multicasting and class E is reserved.
Classes A, B, C have certain bits for the network part and rest for the host part
i.e., networks belonging to a class and number of hosts attached to it are fixed.

Class A Class B Class C

Class No. of networks No. of hosts per network Designed for
24
A 126 2 –2 WAN
B 16,382 65,534 Campus networks
C 221 254 LAN
In c1assful addressing, a large part of the available addresses were wasted,
since Class A and B were too large for most organizations.
Class C is suited only for small organization and reserved addresses were sparingly used.

State the components of a router

The control processor is responsible for implementing the routing

protocols. The switching fabric transfers packets from one port to another
Routers are designed to handle variable-length packets
packetsize × pps = linerate, i.e, packet size at which the router can forward at line rate.

Detail the process of determining the physical address of a destination host (ARP).
A host or router to send an IP datagram, needs to know both the logical and
physical address of the destination.
The destination IP address can be obtained from DNS host or forwarding table.
The physical address of the receiver is needed to pass through the physical network.

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3.9

The Address Resolution Protocol (ARP) enables a source host to know the
physical address of another node when the logical address is known.
ARP relies on broadcast support provided by physical networks such as
Ethernet, Token ring, etc.
ARP enables each host on a network to build up a table of mapping between IP
address and physical address.
Header Format
0 8 16 31
Hardware Type Protocol Type
HLen PLen Operation
Sender Hardware address
Sender Protocol address
Target Hardware address
Target Protocol address

Hardware Type defines type of the physical network (1 for ethernet).

Protocol Type specifies the value of upper-layer protocol (8 for IPv4).
HLen specifies length of the physical address in bytes (6 for Ethernet).
PLen specifies length of the logical address in bytes (4 for IPv4).
Operation defines the type of ARP (1 for ARP request, 2 for ARP reply).
Sender Hardware address variable-length field contains physical address of the sender.
Sender Protocol address variable-length field contains logical address of the sender.
Target Hardware address variable-length field contains physical address of the target.
Target Protocol address variable-length field contains logical address of the target.
Address Translation
The host checks its ARP table with the logical address.
o If an entry exists, then corresponding physical address is used to send a
datagram. o Otherwise it finds physical address using ARP.

1. ARP request packet is created with Operation field set to 1.

2. The Target Physical address field is not known and filled with 0 (broadcast address).
3. The ARP request is encapsulated in IP packet and broadcasted on the physical network.
4. Each host takes note of sender's logical and physical address.
5. All nodes except the target node discard the packet.

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3.10

6. The target node constructs an ARP reply packet with Operation set to 2.
7. ARP reply is unicast, sent back to the sender.
8. The sender receives the reply packet and stores target logical-physical address
pair in its ARP table for sending future packets.
9. If target node does not exist on the same network, then ARP request is sent to the
default router, which then forwards it to the next hop router and so on till destination.
ATMARP
ARP relies on broadcasting, whereas ATM network does not support
broadcasting. ATMARP or Classical IP over ATM uses Logical IP Subnet (LIS).
The ATM network is divided into several subnets.
All nodes on the same subnet have the same network id.
Two nodes on the same subnet can communicate directly, whereas nodes on
different subnets communicate via one or more routers.
Each node in the LIS is configured with ATM address of the ARP server to
establish a virtual circuit to the ARP server when it boots.
The node sends a registration message that includes its IP and ATM address to
the ARP server.
Thus ARP server builds the database of all node as <IP address, ATM address> pair.
Any node that wants to send a packet to some IP address requests the ARP
server to provide the corresponding ATM address.
The ARP server performs a lookup operation and returns the ATM address.
The node can also maintain a cache of IP-to-ATM address mappings.
The source node establishes VC with the destination node and sends packets.

The above ATM network has two subnets.

o Host H1 is connected to the router interface that connects
to LIS10. o Similarly H2 is connected to interface LIS12.
o For hosts on different subnets (say from H1 to H2), both hosts have to
establish a virtual circuit to the router.
What is RARP? List the disadvantage.
A host knows its IP address from configuration file and physical address from its NIC.
A diskless workstation booted from its ROM or newly booted workstation does
not know its IP address as it is assigned by the network administrator.
In such cases, Reverse Address Resolution protocol (RARP) allows the host to
broadcast its physical address in a RARP request at the link-layer level.
The RARP request has a destination address of all 1s.
The RARP server receives the request, looks up the physical address in its
configuration file and sends the IP address in a RARP reply.
RARP enables a host to know its logical address using its physical address.

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3.11

If an administrator has several networks/subnets, then a RARP server is required

for each network/subnet, since RARP broadcast is not forwarded by routers.
RARP is replaced by protocols such as BOOTP and DHCP.
Discuss the configuration of IP address to hosts automatically using DHCP.
Operating systems allow system administrator to manually configure IP address. o
Manual configuration is tedious and error-prone on any network.
Dynamic Host Configuration Protocol (DHCP) enables auto configuration of IP
address to hosts using DHCP.
The drawback is it is difficult to identify a malfunctioning host.
DHCP is derived from Bootstrap Protocol (BOOTP) and is connectionless.
The UDP port for sending data to server is 67 and port 68 for sending data to client.
DHCP provides both static (manual) and dynamic (automatic) address allocation.
For static allocation, a DHCP server has a manually created static database that
binds physical address to IP address.
Header Format

Operation specifies type of DHCP packet.

HType value for type of the physical network (1 for
ethernet). HLen length of the physical address in bytes (6
for ethernet). Xid specifies the transaction id.
ciaddr specifies client IP address in case of DHCPREQUEST
yiaddr this field is known as your IP address, to be filled by DHCP
server. siaddr contains IP address of the DHCP server.
giaddr contains IP address of the Gateway or relay agent.
chaddr contains hardware (physical) address of the client.
Dynamic Address Allocation
Dynamic allocation is required when a host moves from one network to another
or else is connected / disconnected from a network.
The administrator provides DHCP server a range of unassigned addresses to be
assigned to hosts on demand.
To contact DHCP server, a booted/attached host broadcasts a DHCPDISCOVER
message with IP address 255.255.255.255 encapsulated in a UDP packet.
The DHCP server checks its static database first.
o If the lookup is successful, the corresponding IP address is returned.
Otherwise, the server selects an unassigned IP address based on client's MAC address.
o Fills the selected address in yiaddr field and adds an entry to dynamic database.

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3.12

o DHCP Server sends DHCPOFFER message containing Client IP and MAC address,
server IP address and options (lease duration, default route, DNS server, etc.)
There can be multiple DHCP server on a network but the client accepts only one
offer. The client broadcasts a DHCPREQUEST message requesting the offered address.
Based on transaction id, the corresponding DHCP server sends an
acknowledgement as a DHCPACK containing the requested configuration.
When the lease expires, the client renews the lease.
o The server either agrees or disagrees with the renewal.
DHCP relay
DHCP is an application layer protocol.
o Both the server and client need not exist on the same network.
A DHCP relay agent receives broadcast message from the client.
o Stores it's address in giaddr and is sent as unicast to DHCP server.
o The DHCP server's response is sent to the relay agent, which is
retransmitted back to the client.

Write short notes on error reporting using ICMP.

The IP protocol is a best-effort delivery service.
o It has no error-reporting or error-correcting mechanism and also lacks
mechanism for host and management queries.
Internet Control Message Protocol (ICMP) is designed to handle these lacunae.
ICMP control messages are either used to report errors to the source host or
used to diagnose network problems.
An ICMP message is encapsulated within an IP packet.
Debugging tools such as ping and traceroute use ICMP messages internally.
Header Format

Type 8 bit field that specifies type of the ICMP message.

Code 8-bit field that specifies the sub-type for the given
type. Checksum contains 16-bit checksum sequence.
Rest of Header varies based on Type and Code field.
Control Messages
Destination Unreachable When a router cannot route a datagram, the datagram
is discarded and sends a destination unreachable (Type = 3) message.

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3.13

Source Quench When a router or host discards a datagram due to congestion, it

sends a source-quench (Type = 4) message. This message acts as flow control.
Time Exceeded When TTL field becomes 0, the router discards the datagram and
a time-exceeded (Type = 11) message is sent to the source host.
Parameter Problem If a router discovers ambiguous or missing value in any field of the
datagram, it discards the datagram and sends parameter problem (Type = 12) message.
Redirection Redirect messages (Type = 5) are sent by the default router to inform the
source host to update its forwarding table when the packet is routed on a wrong path.
Echo Request & Reply The combination of echo-request (Type = 8) and echo-reply (Type
= 0) messages determines whether two systems can communicate at the IP level.
Timestamp Request & Reply Two machines can use the timestamp request (Type = 13)
and timestamp reply messages (Type = 14) to determine the round-trip time (RTT).
Address Mask Request & Reply A host to obtain its subnet mask, sends an
address mask request (Type = 17) message to the router, which responds with an
address mask reply (Type = 18) message.
Router Solicitation & Advertisement For the host to know if the routers are functioning, it
can broadcast a router solicitation (Type = 10) message. The router then broadcast its
routing information using the router advertisement (Type = 9) message.
Discuss the various queuing methods in detail.
Routers have finite buffer space.
When a packet arrives, it is placed at rear end of queue at the router's buffer
space. The packet at front of the queue is taken out of the queue for forwarding.
The common queuing algorithms are:
o First-In-First-Out (FIFO) Queuing
o Priority Queuing
o Fair Queuing
o Waited Fair Queuing
FIFO Queuing
The first packet that arrives at a router is the first packet to be forwarded i.e., FIFO.
The router discards any packet that arrives when the queue is full. This is known
as Tail drop since packets arriving at tail end of queue is dropped.
Thus FIFO queuing is a combination of FIFO scheduling discipline and Tail drop policy.

FIFO queuing Tail Drop

Analysis
Simple to implement and is widely used.
Packets are dropped without regard to its flow type or importance.
Does not help in congestion control and it is left to TCP at the end hosts.

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3.14

Priority Queuing
Priority queuing is a variation of FIFO queuing
Each packet is marked with a priority. The priority can be set in TOS field of IP
header. Routers have a FIFO queue, one for each type of priority.
The router always forwards packets out of the highest priority queue. If that queue is
empty, then packets in the next high priority queue is taken for processing.
Packets in the lowest priority queue are processed last.
The network can charge more to deliver high-priority packets than low-priority ones.

Analysis
A priority queue can provide better QoS than FIFO queue because high priority
traffic such as multimedia, can reach the destination with less delay.
Routing updates after a topological change is marked in TOS field, helps in
stabilization of routing tables.
The potential drawback is that packets in lower-priority queues may never be processed,
if there is a continuous flow in high-priority queues. This condition is called starvation.
Fair Queuing (FQ)
Fair Queuing addresses the problems of FIFO queuing such as non-
discrimination of traffic sources and lack of congestion-control.
In fair queuing, a separate queue is maintained for each type of
flow. Router services these queues in a round-robin manner.
When a flow's queue gets filled up, further packets are
discarded. All flows have a fair share of the bandwidth.
FQ segregates traffic so that ill-behaved traffic sources do not interfere with the
legitimate traffic sources.
FQ enforces fairness among a collection of flows managed by a well-behaved
congestion control algorithm.

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3.15

Round-robin servicing
Round-robin servicing needs to be done in terms of bit-by-bit, but interleaving bits
from different packets is not feasible.
FQ simulates bit-by-bit RR by first determining when a given packet would finish being
transmitted and then using it to sequence the packets for transmission as follows:
o Let Pi denote the length of packet i
o Let Si denote the time when the router starts to transmit packet i
o Let Fi denote the time when the router finishes transmitting packet i (Fi = Si + Pi)
A packet can be transmitted after its arrival time Ai and not before its predecessor i-1 has
been transmitted. Hence, Si = max (Fi-1 , Ai) and Fi = max (Fi-1 , Ai) + Pi
The packet with the lowest Fi timestamp is the next to be transmitted.
A newly arriving packet cannot preempt a packet that is currently being transmitted.

Shorter packets sent first Non-preemptive of packet in progress

Analysis
The link is never idle as long as there is at least one packet in any of the flow.
This characteristic is known as work-conserving.
If there are n flows, then a flow cannot use more than 1/n of the total bandwidth.
If some flows are empty, then their bandwidth is shared amongst the available flows.
Weighted Fair Queuing (WFQ)
Weighted Fair Queuing is a variation of fair queuing.
In WFQ, each flow is assigned a weight, whereas FQ gives each queue a weight of 1.
The weight specifies how many bits to transmit each time the router services that
queue The weight also implies the percentage of the link’s bandwidth that flow will get.
Packets are assigned different classes and admitted to different queues based
on their priority.
The system processes packets in each queue in a round-robin fashion with the
number of packets selected from each queue based on the corresponding weight.

In above example, three packets are processed from the first queue, two from
the second queue, and one from the third queue.

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3.16

Classify routing protocols

Intra-domain routing
o Distance vector routing (eg. RIP)
o Link state routing (eg.
OSPF) Inter domain routing
o Path vector (eg. BGP)
Explain distance vector routing in detail with an example.
Each node knows the distance (cost) to each of its directly connected neighbors.
Hosts that are not directly connected or if link is down, is assigned infinite cost.
Each node constructs a vector containing (Destination, Cost, Next Hop) to all
other nodes and distributes to its neighbors.
Each node computes a vector (table) of minimum distance (cost) to every other
node using the information from its neighbors.
Thus the table at each node guides a packet to the desired node by showing the Next Hop.
Initial State
For the given network, each node sets a distance of 1 (hops) to its immediate neighbors.
The distance for non-neighbors is marked as unreachable with value (infinity).
The initial routing table stored at A is
Destination Cost Next Hop
B 1 B
C 1 C
D —
E 1 E
F 1 F
G
Sharing & Updation
Each node shares its cost list (distance) to all of its directly connected
neighbors. Node A receives distance vectors from B, C, E and F.
o For example the tables received by A from C and F are:
Destination Cost Next Hop Destination Cost Next Hop
A 1 A A 1 A
B 1 B B
D 1 D C
E D
F E
G G 1 G
Now node A can use information from its neighbors to reach other unreachable
nodes. For example, node F tells node A that it can reach node G at a cost of 1.
Each node updates its routing table by comparing with its neighbor tables as follows
o For each destination Total Cost is computed as:
Total Cost = Cost(Node, Neighbor) + Cost(Neighbor, Destination).
o If Total Cost < NodeCost(Destination) then
NodeCost(Destination) = Total Cost and Next Hop(Destination) = Neighbor

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3.17

For example, A compares its table with C's table

o Total Cost for B = Cost(A, B) + Cost(B, C)
= 2 Since 2 > 1, there is no change
o Total Cost for D = Cost (A, C) + Cost(C, D) = 1 + 1 = 2.
Since 2 < , entry for destination D in A's table is changed to (D,
2, C) o Similarly other entries are checked and there is no change.
In a similar manner, A updates its routing table using information from B, E
and F. The final routing table at A is
Destination Cost Next Hop
B 1 B
C 1 C
D 2 C
E 1 E
F 1 F
G 2 F
Each node builds complete routing table after few exchanges with its neighbors.
The process of obtaining complete routing information to all nodes is called
convergence. The sharing & updation process take place periodically and in case
of triggered update. Periodic updation is normally done every 30 seconds.
Triggered Update
A node can test link status by using hello (control) packets.
Alternatively a link or node failure is presumed, if it does not receive periodic
updates from its neighbor for a while.
This forces the node to update its neighbors, neighbors update their neighbors
and so on. Assume that F detects that its link to G has failed.
o F sets its new distance to G as and shares its table
with A. o Node A updates its distance to G as .
o Node A also receives periodic update from C with distance to
G as 2. o Node A updates its distance to G as 3 through C.
Loop Instability
Suppose link from node A to E goes down.
A advertises a distance of in nity to E, meanwhile B and C advertise a distance of 2 to E. o
B using information from C, concludes that E can be reached in 3 hops through C. o
B advertises this to A, and A in turn updates C with a distance of 4 hops to E.
o Now node C advertises with a distance of 5 to E and so on.
Thus the nodes update each other until cost to E reaches a large number, say infinity.
Thus convergence does not occur. This problem is known as loop instability.
Solutions
Infinity is redefined to a small number. Most implementations define 16 as
infinity. o Distance between any two nodes should not exceed 15 hops.
o Thus distance vector routing cannot be used in large networks.
When a node updates its neighbors, it does not send those routes it learned from
each neighbor back to that neighbor. This is known as split horizon.

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3.18

o For example, if B has the route (E, 2, A) in its table, then it does not
include the route (E, 2) in its update to A.
o Continued absence of route update for a destination leads to deletion of its entry.
In split horizon with poison reverse, Node B can still advertise the value of (E, 2) to A,
but with a warning message.
o This approach delays the convergence process and does not work well for
large number of nodes.
Routing Information Protocol (RIP)
RIP is an intra-domain routing protocol used inside an autonomous system
based on distance-vector algorithm.
It is extremely simple and widely used, since it was distributed with Unix BSD.
The routers advertise the cost of reaching networks, instead of reaching other
routers. RIP takes the simplest approach, with all link costs being equal to 1.
The distance is defined as the number of links to reach the destination.
o The metric in RIP is called a hop count.

In example, Router C advertises to A that it can reach

o Networks 2 and 3 at a cost of 0 (directly connected),
o Networks 5 and 6 at cost 1 and network 4 at cost 2.
As in distance vector routing algorithm, a router updates cost and next hop
information for each network number.
Infinity is defined as 16, i.e., any route in an AS using RIP cannot have more than
15 hops. It is limited to run on only smaller networks.
Routers running RIP send their advertisements every 30 seconds or when it is
updated by another router
Packet Format
RIP packet format contains mostly (network address, distance) pair as
shown. o RIP supports multiple address families that include IP.

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3.19

Explain link state routing with an example.

Each node knows the state of link to its neighbors and the cost
involved. Link-state routing protocols rely on two mechanisms:
o Reliable dissemination of link-state information
o Route calculation from the accumulated link-state knowledge
Reliable Flooding
Reliable flooding is the process of ensuring all nodes having a copy of the link-
state information from all other nodes
Each node creates an update packet called the link-state packet (LSP) containing
o ID of the node
o List of directly connected neighbors of that node and cost to each one
o Sequence number
o Time to live
Each node sends its LSP out on each of its directly connected links.
Transmission of LSPs between adjacent routers is made reliable using acknowledgment.
When a node receives LSP from a neighbor, it checks to see whether it has a copy.
o If not, store and forward the LSP on all other links except the incoming one.
o Otherwise, if the received LSP has a bigger sequence number, then it is
stored and forwarded. The older one is discarded.
Since a node passes the recent LSP to its neighbors, which in turn forwards to
their neighbors, the recent LSP eventually reaches all nodes.
LSP is generated either periodically or when there is a change in the topology.
Example

LSP arrives at node X, which sends it to neighbors A and

C. A and C do not send it back to X, but send it to B.
Since B receives two identical copies of the LSP, it accepts one that comes first
and discards the other.
B passes the LSP on to D. Since D has no neighbors to flood, the process is complete.
Reducing Overhead
Flooding creates traffic and is overhead on the network. Mechanisms to reduce are:
1. Timer using long timers, in terms of hours for periodic generation.
2. Sequence number 64-bit sequence numbers do not wrap around soon and is
used to discard old LSPs.
3. Time to live A router decrements TTL before forwarding a LSP. When TTL reaches
0, the node refloods the LSP. All nodes delete their stored LSP for that ID.
Route Calculation
Once a node has copy of the LSP from every other node, it knows the entire network.
Each node computes its routing table directly from the LSPs using a variation of
Dijkstra algorithm called forward search algorithm
Each node maintains two lists namely Tentative and Confirmed.
Each of these lists contains a set of entries of the form (Destination, Cost, NextHop)

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3.20

Forward Search algorithm

1. Initialize the Confirmed list with an entry for the Node with a cost of 0.
2. For the node just added to Confirmed list, call it node Next and select its LSP.
3. For each neighbor of Next:
a. Calculate cost to reach Neighbor as Cost(Node, Next) + Cost(Next, Neighbor).
b. If Neighbor is currently on neither Confirmed nor Tentative list, then add
(Neighbor, Cost, NextHop) to Tentative list.
c. If Neighbor is currently on Tentative list, and the Cost is less than currently
listed cost for Neighbor, then replace the current entry with (Neighbor,
Cost, NextHop), where NextHop is the direction to reach Next.
4. If the Tentative list is empty then Stop.
a. Otherwise, pick the entry from the Tentative list with the lowest cost, move
it to the Confirmed list, and go to Step 2.

For the given network, the process of building routing table for node D is tabulated
Step Confirmed Tentative Comment
1 (D, 0, –) D is moved to Confirmed list initially
2 (D, 0, –) (B, 11, B) Based on D's LSP, its immediate neighbors B and C are
(C, 2, C) added to Tentative list
3 (D, 0, –) (B, 11, B) The lowest-cost member C of Tentative list is moved onto
(C, 2, C) Confirmed list. C's LSP is to be examined next.
4 (D, 0, –) (B, 5, C) Cost to reach B through C is 5, so the entry (B,11,B) is
(C, 2, C) (A, 12, C) replaced. C's neighbor A is also added to Tentative list
5 (D, 0, –) (A, 12, C) The lowest-cost member B is moved to the Confirmed list.
(C, 2, C) B's LSP is to be examined next
(B, 5, C)
6 (D, 0, –) (A, 10, C) Since A could be reached B at a lower cost than the existing
(C, 2, C) one, the Tentative list entry (A,12,C) is replaced to (A,12,C).
(B, 5, C)
7 (D, 0, –) The lowest-cost and only member A is moved to Confirmed
(C, 2, C) list. Processing is over.
(B, 5, C)
(A, 10, C)
Analysis
Link-state routing stabilizes quickly without generating much traffic and
responds to changes in topology dynamically.
The amount of information stored (a LSP for each node) is large.

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3.21

Open Shortest Path First Protocol (OSPF)

OSPF is one of the most widely used link-state routing protocols.
Authentication of routing messages Misconfigured hosts are capable of bringing
down a network by advertising to reach every host with the lowest cost 0. Such
disasters are averted by mandating routing updates to be authenticated.
Additional hierarchy In OSPF, a domain is partitioned into areas, i.e., a router
need not know the complete network, instead only its area.
Load balancing OSPF allows multiple routes to the same place to be assigned
the same cost and will cause traffic to be distributed evenly over those routes.
OSPF Header

Version represents the current version, i.e., 2.

Type represents the type value (1–5) of OSPF message.
o Type 1 also known as hello message to find out whether its neighbors are alive.
o Other types are used to request, send and acknowledge link-state
messages. SourceAddr identifies the sender of the message
AreaId 32-bit identifier of the area in which the node is
located Checksum 16-bit checksum
Authentication type has value 0 if no authentication is used, 1 for simple password
and 2 for cryptographic authentication checksum.
Authentication contains password or cryptographic checksum
Link State Advertisement (LSA)
The basic of OSPF message is the Link State Advertisement.
o A message can contain multiple LSAs.
LSA is used by routers to advertise the networks that are directly connected to it.
It is also used by routers to advertise the cost of reaching it over a link from other routers.

LS Age is incremented at each node until it reaches a maximum

Type defines type of LSA. Type1 LSAs advertise the cost of links between routers.

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Link-state ID 32-bit identifier that identifies the router.

Advertising router For type 1 LSA, it is same as Link-state ID
LS sequence number used to detect old or duplicate
packets LS checksum covers all fields except LS Age
Length length of the LSA in bytes
Link ID and Link Data identify a link
Metric specifies cost of the link.
Link Type specifies type of link (for example, point-to-point)
TOS allows OSPF to choose different routes based on the value in TOS field
What are subnets and why are they required? Explain routing using subnets.
Classful Address Depletion
The network part of IPv4 is used to identify a single physical
network. A smaller network requires a class C address.
For networks with more than 255 hosts, class B address is required.
Class B address are sought after in anticipation of more hosts to be added in the future.
IPv4 address space is exhausted in the process of assigning one per physical network.
At most 253 addresses can go unused in a class C network whereas over 64,000
addresses can go unused in a class B network.
This results in inefficient usage of the available address space.
Increase in network numbers, increases size of forwarding tables and degrade
router performance.
Subnetting
Subnetting reduces the total number of network numbers by assigning a single
network number to many adjacent physical networks.
Each physical network is referred to as subnet.
For subnetting, the subnets must be close to each other.
o For example, each department having a network within a college campus
with a router connecting to the external world.
All nodes on a subnet is configured with a subnet mask.
Subnet mask introduces a level of hierarchy into IP address. It is written like a IP
address (for example 255.255.255.0)
The bitwise AND of IP address and its subnet mask gives the subnet number.
Thus all nodes have the same subnet number, i.e., hosts on different physical
network share a single network number.

Thus subnetting provides efficient usage of available address space by assigning

a single network number amongst multiple adjacent physical networks.

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3.23

Routing

When the host wants to send a packet to another host, it performs a bitwise AND
between its own subnet mask and the destination IP address.
o If the result equals its own subnet number, then the packet is delivered
directly over the subnet.
o Otherwise, the packet is sent to a router to be forwarded to another subnet.
For example, when H1 sends a packet to H2 in the above given network, then:
o H1 performs bitwise AND (255.255.255.128, 128.96.34.139) which is equal
to 128.96.34.128
o This does not match the H1's subnet number 128.96.34.0
o Therefore H1 sends the packet to the default router R1
Routing Table
To support subnetting, entries in routing table are of the form (SubnetNumber,
SubnetMask, NextHop)
To perform a lookup, the router performs a AND (destination address,
SubnetMask) for each entry in the table.
If the result matches the SubnetNumber for an entry, then the packet is forward to
the corresponding NextHop router
The outer world sees the collection of subnets as a single network and has only
one entry in the forwarding table for all the subnets.
Routers within the campus must be able to route packets to the right subnet.
R1's forwarding table is as follows.

SubnetNumber SubnetMask NextHop

128.96.34.0 255.255.255.128 Interface 0
128.96.34.128 255.255.255.128 Interface 1
128.96.33.0 255.255.255.0 R2
When H1 sends a datagram to H2, R1 receives the datagram from H1.
o It ANDs the H2 address 128.96.34.139 with SubnetMask and compares the
result with SubnetNumber for each entry in the table.

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3.24

o The result matches for the second entry. Thus the packet is delivered to
H2 through Interface 1
Forwarding Algorithm
D = destination IP address
for each forwarding table entry (SubnetNumber, SubnetMask, NextHop)
D1 = SubnetMask & D
if D1 = SubnetNumber
if NextHop is an interface
deliver datagram directly to destination
else
deliver datagram to NextHop (a router)
Write short notes on CIDR.
Subnetting helps in address assignment, but does not prevents an organization
go for a class B address, anticipating number of hosts could go beyond 255.
Exhaustion of address space centers on exhaustion of class B address.
If class C addresses were given, then number of entries in the routing table gets larger.
The address efficiency in class C can be as low as 0.78% (2/55) and in class B
can be as low as 0.39% (256/65535).
Classless Interdomain Routing (CIDR) tries to balance between minimize the
number of routing table entries and handling addresses space efficiently.
CIDR aggregates routes, by which an entry in the forwarding table is used to
reach multiple networks.
Example1
Consider an autonomous system (AS) with 16 class C networks.
Instead of providing 16 class addresses at random, a block of contiguous class C
address is given. For example, from 192.4.16 to 192.4.31
The bitwise analysis shows 20 MSBs (11000000 00000100 0001) are the same for
that block, i.e., a 20-bit network id.
The 20-bit network number supports hosts that range between class B and C address.
Thus higher address efficiency is achieved by providing small chunks of address,
smaller than class B network and a single network prefix to be used in forwarding table.
Restrictions
The addresses in a block must be contiguous.
The number of addresses in a block must be a power of 2.
The first address must be evenly divisible by the number of addresses.
A protocol such as BGP is required to support classless addressing.
o The network number is represented as <length, value> pairs
Route Aggregation
Consider the case of an ISP to provide internet connectivity to a large number of
corporation and campuses.
In example, two corporations served by the ISP is assigned adjacent 20-bit
network prefixes.

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3.25

Since both of them are reachable through ISP, the ISP advertises a 19-bit
common prefix that both share.

What is an autonomous system?

Internet is so large that no one routing protocol can handle the task of updating
the routing tables of all routers. Internet is divided into autonomous systems.
An autonomous system is a group of networks and routers under the authority of
a single administration.
Routing inside an autonomous system is referred to as intra-domain routing.
Routing between autonomous systems is referred to as inter-domain routing.
What is interdomain routing?
The interdomain routing involves AS sharing their reachability information with
each other AS.
The goal of interdomain routing is reachability and not optimality.
The two major interdomain routing protocols are Exterior Gateway Protocol
(EGP) and Border Gateway Protocol (BGP).

What are the problems in interdomain routing?

An internet backbone must be able to route packets to any destination, i.e., there
should be a match in the routing/forwarding table.
Each AS has its own intradomain routing protocols and chooses the metric
assigns to path. This varies from one AS to another.
Autonomous systems may not trust each other.
Write short notes on BGP.
Border Gateway Protocol (BGP) is an inter-domain routing protocol using path
vector routing
Traffic on the internet can be classified into two types:
o local traffic that starts/ends on nodes within an AS
o transit traffic that passes through an
AS AS can be classified into three types
o Stub AS has only a single connection to one other AS. This AS can carry
local traffic only, such as Small corporation.
o Multihomed AS has connections to more than one other AS but refuses to
carry transit traffic, such as Large corporation.
o Transit AS has connections to more than one other AS and is designed to
carry both transit and local traffic, such as the backbone providers
Each AS selects one of its nodes to be the BGP speaker.
Speaker node creates a routing table for that AS and advertises it to other BGP
speakers in the neighboring ASs.

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3.26

Network of autonomous systems Multi backbone internet

Each AS also has a border gateway through which packets enter and leave the AS.
BGP advertises complete paths as an enumerated list of ASs to reach a
particular network. BGP ensures that paths are loop-free.
The attributes in a path can be well known or optional. The well known attributes
are recognized by all routers.
If there are different routes to a destination, the BGP speaker chooses the best
one according to local policies, and then advertises.
A BGP speaker need not advertise any route to a destination, even if it has one.
Example
The BGP speaker for provider A (AS2) advertises that the networks 128.96,
192.4.153, 192.4.32, and 192.4.3 can be reached directly from AS2.
The backbone network, on receiving this advertisement, advertises that networks
128.96, 192.4.153, 192.4.32, and 192.4.3 can be reached along the path (AS1, AS2).

BGP speakers can cancel previously advertised paths if a critical link or node on
a path goes down. This negative advertisement is known as withdrawn route.
The format of BGP-4 update message that carries advertisement is shown below

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3.27

BGP Sessions
The exchange of routing information between two routers takes place in a BGP
session. To create a reliable environment, BGP uses the services of TCP.
The routes need not be repeatedly sent, if there is no change. This is done by
sending keep alive messages.
Two types of BGP session are external BGP (E-BGP) and internal BGP (I-BGP).
o E-BGP is used to exchange routing information between two speaker
nodes belonging to two different ASs.
o I-BGP is used to exchange routing information between two routers inside an AS.

Discuss the notation, representation and address space of IPv6.

CIDR and subneting could not solve the address exhaustion faced by IPv4.
IPv6 was evolved to solve this problem.
The striking features of IPv6 are:
o support for real-time
services o security support
o auto configuration
o enhanced routing functionality, including support for mobile hosts
Addresses Space
38
IPv6 provides a 128-bit address space to handle up to 3.4 ×10 nodes.
IPv6 addresses do not have classes, but classification is based on the leading
bits. The IPv4's classes A, B and C start with 001 prefix (unicast addresses).
Multicast address (start with a byte of 1s) serves the purpose of class D address.
Large chunks of address space are left unassigned to allow for new features in the future.
Common Prefixes
Prefix Usage
0000 0000 Reserved
0000 001 Reserved for ISO protocol
0000 010 Reserved for Novell network layer
001 Aggregated Global Unicast Addresses (Class A, B and C)
010 Provider-based unicast addresses
100 Geographic-based unicast addresses
1111 1110 10 Link local use addresses
1111 1110 11 Site local use addresses
1111 1111 Multicast addresses
Address Notation
The standard representation is x:x:x:x:x:x:x:x where x is a hexadecimal representation of
a 16-bit address separated by colon (:) as shown below
47CD:1234:4422:ACO2:0022:1234:A456:0124
An IPv6 address with a large number of contiguous 0s is written compactly by omitting
the 0s (47CD:0000:0000:0000:0000:0000:A456:0124 is written as 47CD::A456:0124)
An IPv4 address can be mapped to IPv6 address by prefixing the 32-bit IPv4 address with
2 bytes of all 1s and then zero-extending the result to 128 bits.
128. 96.33.81 is written as ::FFFF:128.96.33.81

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3.28

Address Aggregation
The goal of the IPv6 address allocation plan is to provide aggregation of routing
information to reduce the burden on intradomain routers.
Aggregation is done by assigning prefixes at continental level.
Continental boundaries form natural divisions in the Internet topology
o For example, if all addresses in Europe have a common prefix, then routers in
other continents would need one routing table entry for all networks in Europe.
The format for provider-based unicast address aggregation is shown below.

o RegistryID contains identifier assigned to the continent. It is either

INTERNIC (North America), RIPNIC (Europe) or APNIC (Asia and Pacific)
o ProviderID variable-length field identifies the provider for Internet access
such as an ISP.
o SubscriberID specifies the assigned subscriber identifier
o SubnetID defines a specific subnet under the territory of
subscriber. o InterfaceID contains the link level or physical address.
Addressing
Multicast address as in IPv4 is used to address a group of hosts.
IPv6 also defines Anycast addresses. A packet destined for an anycast address
is delivered to only one member of the anycast group (the nearest one).
Reserved addresses start with prefix of eight 0s. It is classified into
o unspecified address is used when a host does not know its address
o loopback address is used for testing purposes before connected to network
o compatible address is used when IPv6 hosts communicate through IPv4 network
o mapped address is used when a IPv6 host communicates with a IPv4 host.
IPv6 header defines Local addresses for private networks. It is classified into
o Site local address for use in a isolated site with several subnets.
o Link local address for use in a isolated subnet
Packet Format
The IPv6 base header is always 40 bytes long. The packet format is:

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3.29

Version—specifies the IP version, i.e., 6.

TrafficClass—defines the priority of the packet with respect to traffic congestion. It
is either congestion-controlled or non-congestion controlled
FlowLabel—is designed to provide special handling for a particular flow of data.
The router handles flow with the help of a flow table.
PayloadLen—gives the length of the packet, excluding the IPv6 header
NextHeader—If options are required, then it is specified in one or more special
headers following the IP header, its value is contained in NextHeader field.
Otherwise, it identifies the higher-level protocol (TCP/UDP).
HopLimit—This field serves the same purpose as TTL field in IPv4.
SourceAddress and DestinationAddress—contains 16-byte address of the source
and destination host respectively.
Extension Header
To provide greater functionality to IP datagram, the base header can be followed
by up to six extension headers. They are:
1. Hop-by-Hop—used when the source needs to pass information to all routers
visited by the datagram.
2. Source Routing—accounts for both strict and loose source routing.
3. Fragmentation—In IPv6, only the original source can fragment. A source must
use a path MTU discovery technique to find the smallest MTU on the path.

4. Authentication—used to validate the sender and ensures the integrity of data

5. Encrypted Security Payload—provides confidentiality and guards against
eavesdropping.
6. Destination—used when source needs to pass information to destination only.
Intermediate routers cannot access this information.
Auto configuration
IPv6 provides a new form of autoconfiguration called stateless auto-configuration,
which allows a host to be connected without the help of a DHCP server.
List the advantages of IPv6.
Large address space An IPv6 address is 128 bits long. Compared with the 32-bit
96
address of IPv4, this is a huge (2 ) increase in the address space.
Better header format IPv6 uses a new header format in which options are
separated from the base header and inserted, when needed.
New options IPv6 has new options to allow for additional functionalities.
Allowance for extension IPv6 is designed to allow the extension of the protocol if
required by new technologies or applications.
Support for resource allocation In IPv6, flow label has been added to enable the
source to request special handling of the packet such as real-time audio and video.
Support for more security The encryption and authentication options in IPv6
provide confidentiality and integrity of the packet.

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3.30

State the drawbacks of IPv4?

Despite all short-term solutions, such as subnetting, classless addressing, and
NAT, address depletion is still a long-term problem in the Internet.
The Internet must accommodate real-time audio and video transmission that requires
minimum delay strategies and reservation of resources, which are not provided in IPv4.
The Internet must provide encryption and authentication of data for some
applications. No encryption or authentication is provided by IPv4.
How NAT helps to solve address space depletion?
The idea behind Network Address Translation (NAT) is that all hosts that uses
Internet do not need to have globally unique addresses.
NAT enables a organization to have a large set of addresses internally and one
address or a small set of addresses externally.
Three sets of addresses are reserved for internal use (10.0.0.0 – 10.255.255.255,
172.16.0.0 – 172.31.255.255 and 192.168.0.0 – 192.168.255.255).
The organization must have only one single connection to the Internet through a
router that runs the NAT software.
Briefly explain IGMP?
Internet Group Message Protocol (IGMP) is a protocol that manages group membership.
Provides the multicast routers information about the membership status of hosts (routers)
connected to the network.
Enables a multicast router to create and update list of loyal members related to
each router interface.
The operations are:
o Joining a group
o Leaving a group
o Monitoring membership
Explain Multicast routing protocols in detail.
A host places a multicast address in the destination address field to send packets
to a set of hosts belonging to a group.
Internet multicast can be implemented on physical networks that support
broadcasting by extending forwarding functions. The extended ones are:
o Link-State Multicast
o Distance-Vector Multicast
o Protocol Independent Multicast (PIM)
Link-State Multicast
Multicasting is added to the existing link-state routing.
o Each router knows entire topology by way of update messages.
o Dijkstra's algorithm is used to compute shortest path spanning tree to
reach all destinations.
Each router determines which groups have members on which LAN by
monitoring the periodical announcements.
o If a host does not advertise periodically, then it has left the group.
Equipped with group and membership knowledge, each router computes shortest
path multicast tree from any source to any group using Dijkstra's algorithm.

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3.31

Link-state routing is expensive as each router must store a multicast tree from
every source to every group.
Distance-Vector Multicast
Multicasting is added to existing distance-vector routing in two stages.
o Each router maintains a table of (Destination, Cost, NextHop) for all
destination through exchange of distance vectors.
o Reverse Path Broadcast mechanism that floods packets to other networks
o Reverse Path Multicasting that prunes end networks that do not have
hosts belonging to a multicast group.
Reverse-Path Broadcasting
A router when it receives a multicast packet from source S to a Destination from
NextHop, then it forwards the packet on all out-going links.
The drawbacks are:
o It floods a network, even if it has no members for that group
o Duplicate flooding, i.e., packets are forwarded over the LAN by each
router connected to that LAN.
Duplicate flooding is avoided by
o Designating a router on the shortest path as parent router.
o Only parent router is allowed to forward multicast packets from source S
to that LAN.
Reverse-Path Multicasting
Multicasting is achieved by pruning networks that do not have members for a group G.
Pruning is achieved by identifying a leaf network, which has only one router (parent). The
leaf network is monitored to determine if it has any members for group G.
The router then decides whether or not to forward packets addressed to G over that
LAN. The information "no members of G here" is propagated up the shortest path tree.
Thus routers can come to know for which groups it should forward multicast
packets. Including all this information in a routing update is expensive.
Protocol Independent Multicast (PIM)
The above two multicast routing did not scale well.
PIM divides multicast routing into sparse and dense mode.
In PIM sparse mode (PIM-SM), routers leave and join multicast group using PIM
Join and Prune messages.
PIM designates a rendezvous point (RP) for each group in a domain to receive
PIM messages.
All routers in the domain know the IP address of RP for each group.
A multicast forwarding tree is built as a result of routers sending Join messages to the
RP. The tree may be either shared by multiple senders or source-specific to a sender.
Shared Tree
When a router sends Join message for group G to RP, it goes through a
sequence of routers.
Each router along the path creates an entry (*, G) in its forwarding table for the
shared tree before forwarding the Join message.
Eventually, the message arrives at RP. Thus a shared tree with RP as root is formed.

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3.32

Join from R4 Join from R5 Multicast message from Host

The above figure shows router R4 sending Join message for group G to RP.
It goes through R2. R2 makes an entry (*, G) in its table and forwards the message to RP.
Later when R5 sends Join message for group G, it shares the tree. Therefore R2 does not
forwards the Join message.
When a host attached to router R1, sends a message to group G, which is received by
R1. R1 does not know about group G, therefore it encapsulates the multicast packet with
unicast address and is tunneled along the way to RP.
RP decapsulates the packet and sends the multicast packet to R2, which
forwards it to routers R4 and R5 that have members for group G.
Source-specific tree.
RP has the option of forcing about group G, onto other routers by sending a
source-specific Join message to sending host, so that tunneling can be avoided.
The intermediary routers create an entry (S, G) for source-specific tree.
If more packets are sent from source S to group G, then other routers switch to
source-specific tree with source host as root.

Source-specific Join from RP Routers switch to Source tree

Analysis
PIM is protocol independent because, tree formation is based on path that Join
messages follows based on shortest path.
Shared trees are more scalable than source-specific trees.
Source-specific trees enable efficient routing than shared trees.
Briefly explain the mechanisms to avoid and control congestion in the network
Congestion control refers to techniques that can either prevent congestion before
it happens or remove congestion after it has happened thereby keeping the load
below capacity.

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3.33

Retransmission Policy
Retransmission increases congestion in the network. But, a good retransmission
policy can prevent congestion.
The retransmission policy and the retransmission timers must be designed to
optimize efficiency and at the same time prevent congestion.
Window Policy
The Selective Repeat window is better than Go-Back-N for congestion control,
since it tries to send specific packets that have been lost or corrupted.
Acknowledgment Policy
Sending fewer ACK means imposing less load on the network.
A receiver may send an acknowledgment only if it has a packet to be
sent. A receiver may decide to acknowledge only N packets at a time.
Discarding Policy
A good discarding policy by the routers may prevent congestion and at the same
time may not harm the integrity of the transmission.
For example, in audio transmission, if the policy is to discard less sensitive
packets when congestion is likely to happen, the quality of sound is still
preserved and congestion is prevented or alleviated.
Admission Policy
An admission policy, which is a quality-of-service mechanism, can also prevent
congestion in virtual-circuit networks.
Switches in a flow first check the resource requirements of a flow before
admitting it to the network.
A router can deny establishing a virtual circuit connection if there is congestion in
the network or if there is a possibility of future congestion.
Backpressure
In backpressure mechanism, a congested node stops receiving data from the
immediate upstream node or nodes.
This may cause the upstream node to become congested, and it in turn rejects
data from upstream node, and so on.
Backpressure is a node-to-node congestion control that starts with a node and
propagates, in the opposite direction of data flow to the source.
This technique is used in virtual circuit networks.

Choke Packet
A choke packet is a packet sent by a node to the source to inform it of congestion.
In choke packet method, warning is from the router which has encountered
congestion, to the source station directly.

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3.34

Implicit Signaling
In implicit signaling, the source guesses that there is congestion somewhere in
the network from other symptoms.
For example, when a source sends several packets and there is no acknowledgment
for a while, it assumes that network is congested. Therefore, the source slows down.
Explicit Signaling
The node that experiences congestion can explicitly send a signal to the source
or destination by setting a bit that can be set in a packet
This bit can warn the source that there is congestion and that it needs to slow
down to avoid discarding of packets.
The receiver use policies such as slowing down acknowledgments to alleviate congestion.

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4.1

Illustrate and explain UDP and its packet format.

User Datagram Protocol (UDP) is a connectionless, unreliable transport protocol.
It does not add anything to the services of IP except process-to-process
communication.
UDP is a simple multiplexer/demultiplexer that allow multiple processes on
each host to share the network.
UDP does not implement flow control or reliable/ordered delivery.
UDP ensures delivering of message to the intended recipient by the use of checksum. If a
process wants to send a small message and does not require reliability, UDP is
used.
Port Number
Each process is assigned a unique 16-bit port number on
that host. Processes are identified by (host, port) pair.
Processes can be classified as either as client / server.
o Client process usually initiates exchange of information with the server o
Server process is identified by a well-known port number (0 – 1023).
o Client process is assigned an ephemeral port number (49152 –
65,535) by operating system.
o Some well known UDP ports are:
Port Protocol
7 Echo
13 Daytime
53 DNS
111 RPC
161 SNMP
Ports are usually implemented as a message queue.
o When a message arrives, UDP appends the message to the end of
the queue. o When queue is full, the message is discarded.
o When a message is read, it is removed from the
queue. o When queue is empty the process is blocked

UDP Header
UDP packets, called user datagrams, have a fixed-size header of 8 bytes.

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4.2

SrcPort and DstPort—Contains port number for both the sender (source) and
receiver (destination) of the message.
Length—This 16-bit field defines total length of the user datagram, header plus data.
The total length is less than 65,535 bytes as it is encapsulated in an IP datagram.
UDP length = IP length - IP header's length
Checksum—It is computed over pseudo header, UDP header and message
content to ensure that message is correctly delivered to the exact recipient.
o The pseudo header consists of three fields from the IP header (protocol
number (17), source and destination IP address), plus the UDP length field.
Applications
UDP is used for management processes such as SNMP.
UDP is used for some route updating protocols such as
RIP. UDP is a suitable transport protocol for multicasting.
UDP is suitable for a process with internal flow and error control
mechanisms such as Trivial File Transfer Protocol (TFTP).
Bring out the classification of port numbers.
Well-known ports range from 0 to 1023 are assigned and controlled by IANA.
Registered ports range from 1024 to 49,151 are not assigned or controlled
by IANA. They can only be registered with IANA to prevent duplication.
Ephemeral (dynamic) ports range from 49,152 to 65,535 is neither controlled nor
registered. It is usually assigned to a client process by the operating system.
Distinguish between network and transport layer
Network layer Transport layer
The network layer is responsible for host-to- The transport layer is responsible for
host delivery process-to-process delivery of a packet
Host address is required for delivery Host address and port number is required for
delivery
Error detection is not offered Error detection is done using checksum
Flow control is not done Flow control is not done
Multicasting capability is not inbuilt Multicasting is embedded into UDP

With a neat architecture, explain TCP in detail.

Transmission Control Protocol (TCP) offers a reliable, connection-
oriented, byte-stream service
TCP guarantees the reliable, in-order delivery of a stream
of bytes It is a full-duplex protocol
TCP supports demultiplexing mechanism for process-to-process communication.
TCP has built-in congestion-control mechanism, i.e., sender is prevented
from overloading the network.

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4.3

Process-to-Process Communication
Like UDP, TCP provides process-to-process communication. A TCP
connection is identified a 4-tuple (SrcPort, SrcIPAddr, DstPort, DstIPAddr).
Some well-known port numbers used by TCP are
Port Protocol
23 TELNET
25 SMTP
80 HTTP
Segment Format
TCP is a byte-oriented protocol, i.e. the sender writes bytes into a TCP
connection and the receiver reads bytes out of the TCP connection.
TCP groups a number of bytes together into a packet called segment and adds a header
onto each segment. Segment is encapsulated in a IP datagram and transmitted.

SrcPort and DstPort fields identify the source and destination ports.
SequenceNum field contains sequence number, i.e. first byte of data in that segment.
Acknowledgment defines byte number of the segment, the receiver expects next.
HdrLen field specifies the number of 4-byte words in the TCP header. Flags
field contains six control bits or flags. They are set to indicate:
o URG—indicates that the segment contains urgent data. o
ACK—the value of acknowledgment field is valid.
o PSH—indicates sender has invoked the push operation.
o RESET—signifies that receiver wants to abort the connection.
o SYN—synchronize sequence numbers during connection establishment.
o FIN—terminates the connection
AdvertisedWindow field defines the receiver window and acts as flow control.
Checksum field is computed over the TCP header, the TCP data, and pseudoheader.
UrgPtr field indicates where the non-urgent data contained in the segment begins.
Optional information (max. 40 bytes) can be contained in the header.
Connection Establishment
The connection establishment in TCP is called three-way handshaking
1. The client (active participant) sends a segment to the server (passive participant)
stating the initial sequence number it is to use (Flags = SYN, SequenceNum = x)
2. The server responds with a single segment that both acknowledges the
client’s sequence number (Flags = ACK, Ack = x + 1) and states its own
beginning sequence number (Flags = SYN, SequenceNum = y).

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4.4

3. Finally, the client responds with a segment that acknowledges the server’s
sequence number (Flags = ACK, Ack = y + 1).

State Transition Diagram

The states involved in opening and closing a connection is shown above and
below ESTABLISHED state respectively.
The operation of sliding window (i.e., retransmission) is not
shown. The two events that trigger a state transition is:
o a segment arrives from its peer.
o the local application process invokes an operation on TCP
TCP’s state transition diagram defines the semantics of both its peer-to-peer
interface and its service interface.

Opening
1. The server first invokes a passive open on TCP, which causes TCP to move
to LISTEN state
2. Later, the client does an active open, which causes its end of the connection
to send a SYN segment to the server and to move to the SYN_SENT state.
3. When the SYN segment arrives at the server, it moves to SYN_RCVD state and
responds with a SYN + ACK segment.
4. The arrival of this segment causes the client to move to the ESTABLISHED
state and to send an ACK back to the server.

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4.5

5. When this ACK arrives, the server finally moves to the ESTABLISHED state.
a. Even if the client's ACK gets lost, sever will move to ESTABLISHED state
when the first data segment from client arrives.
Closing
In TCP, the application process on both sides of the connection can
independently close its half of the connection or simultaneously.
Three combinations of transitions from ESTABLISHED to CLOSED state are possible.
Connection Termination
Three-way Handshaking
1. The client TCP after receiving a Close command from the client process
sends a FIN segment. FIN segment can include the last chunk of data.
2. The server TCP responds with FIN + ACK segment to inform its closing.
3. The client TCP finally sends an ACK segment.
Four-way Half-Close
In TCP, one end can stop sending data while still receiving data, known as half-close. For
instance, submit its data to the server initially for processing and close its connection.
At a later time, the client receives the processed data from the server.
1. The client TCP half-closes the connection by sending a FIN segment.
2. The server TCP accepts the half-close by sending the ACK segment. The
data transfer from the client to the server stops.
3. The server can send data to the client and acknowledgement can come from the client.
4. When the server has sent all the processed data, it sends a FIN segment to the client.
5. The FIN segment is acknowledged by the client.

Three-way close Four-way close

What is urgent data in TCP?

At times an application may need to send urgent data, i.e., sending process
wants a piece of data to be read out of order by the receiving process. For
example, to abort a process by issuing Ctrl+C keystroke.
The sending TCP inserts the urgent data at beginning of the segment and
sets URG flag.
The urgent pointer field in the header defines start of normal data.
When the receiving TCP receives a segment with the URG bit set, it delivers
urgent data out of order to the receiving application.

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4.6

What is push operation in TCP?

The receiving TCP buffers the data and delivers them to the application
program when it is convenient for the receiving process.
In case of interactive applications, delayed delivery of data is not acceptable. The
application program at the sending site can request a Push operation.
This instructs the sending TCP not to wait for the window to be filled. It must
create a segment and send it immediately.
The sending TCP also sets the push bit (PSH) to let the receiving TCP know that
the segment includes data that must be delivered to the receiving application
program as soon as possible and not to wait for more data to come.

Explain TCPs adaptive control and its uses.

TCP uses a variant of sliding window known as adaptive flow control
that: o guarantees the reliable delivery of data in ordered manner
o enforces flow control at the sender
The receiver advertises a window size to the sender using AdvertisedWindow
field in the TCP header
The sender cannot have unacknowledged data greater than value of
AdvertisedWindow
Reliable and Ordered Delivery

Sender Receiver
TCP on the sending side maintains a send buffer that is divided into 3 segments
namely acknowledged data, unacknowledged data and data to be transmitted
Similarly TCP on the receiving side maintains a receive buffer to hold data
even if it arrives of order.
The send buffer maintains three variables namely LastByteAcked, LastByteSent,
and LastByteWritten as shown above. The relation between them is obvious
LastByteAcked LastByteSent and LastByteSent LastByteWritten
The bytes to the left of LastByteAcked are not kept as it had been acknowledged.
The receive buffer maintains three variables namely LastByteRead,
NextByteExpected, and LastByteRcvd. The relation between them is
LastByteRead < NextByteExpected and NextByteExpected LastByteRcvd + 1
If data are received in order, NextByteExpected is the next byte after LastByteRcvd Bytes to
the left of LastByteRead is not buffered as it has been read by the application

Flow Control
The capacity of send and receiver buffer is MaxSendBuffer and MaxRcvBuffer
respectively.
The sending TCP prevents overflowing of its buffer by maintaining
LastByteWritten LastByteAcked MaxSendBuffer

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4.7

The receiving TCP avoids overflowing its receive buffer by maintaining

LastByteRcvd LastByteRead MaxRcvBuffer
The receiver throttles the sender by advertising a window that is no larger
than the amount of free space that it can buffer as
AdvertisedWindow = MaxRcvBuffer ((NextByteExpected 1) LastByteRead)
When data arrives, the receiver acknowledges it as long as preceding bytes
have arrived.
o LastByteRcvd moves to its right (incremented), and the advertised
window shrinks
The advertised window expands when the data is read by the application
o It data is read as fast as it arrives then AdvertisedWindow = MaxRcvBuffer
o If it is read slow, it eventually leads to a AdvertisedWindow of size 0.
The sending TCP adheres to the advertised window by computing effective
window, that limits how much data it should send as
EffectiveWindow = AdvertisedWindow (LastByteSent LastByteAcked)
When a acknowledgement arrives for x bytes, LastByteAcked is incremented
by x and the buffer space is freed accordingly
Fast Sender vs. Slow Receiver
A slow receiver prevents being swamped with data from a fast receiver by using
AdvertisedWindow field
Initially the fast sender transmits at a higher rate.
The receiver's buffer gets filled up. Hence, AdvertisedWindow shrinks,
eventually to 0.
When the receiver advertises window of size 0, sender cannot transmit any
further data. Therefore, the TCP at the sender blocks the sending process.
When the receiving process reads some data, those bytes are acknowledged. Thus the
AdvertisedWindow expands.
The LastByteAcked is incremented and buffer space is freed to that extent,
The sending process becomes unblocked and is allowed to fill up the free space.
Checking AdvertisedWindow status
TCP always sends a segment in response that contains the latest values for the
Acknowledge and AdvertisedWindow fields, even if these values have not changed.
Thus the sender can come to know the status of AdvertisedWindow even
after the receiver advertises a window of size 0.
AdvertisedWindow
The TCP's AdvertisedWindow field is 16 bits long, half the size of
SequenceNum The length of 16-bits ensures that it does not wrap around
The length of AdvertisedWindow is designed such that it allows the sender
to keep the pipe full.
The 16-bit length also accounts for product of delay × bandwidth.
What is adaptive retransmission? Explain the algorithms used?
TCP guarantees reliability through retransmission.
o Retransmission due to timeout before ACK.
o Timeout is a function of RTT.
o RTT is highly variable between any two hosts on the internet. o
Appropriate timeout is chosen using adaptive retransmission.

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4.8

Original Algorithm
TCP estimates SampleRTT by computing the duration between sending of
a packet and arrival of its ACK.
TCP then computes EstimatedRTT as a weighted average between the
previous and current estimate as
EstimatedRTT = × EstimatedRTT + (1 ) × SampleRTT
where is the smoothening factor and its value is in the range
0.8–0.9 Timeout is twice the EstimatedRTT
TimeOut = 2 × EstimatedRTT
Karn/Partridge Algorithm
The flaw discovered in original algorithm after years of use is
o whether ACK should be associated with the original or retransmission segment o If
ACK is associated with original one, then SampleRTT becomes too large
o If ACK is associated with retransmission, then SampleRTT becomes too small

Karn/Partridge proposed a solution to the above by making changes to the

timeout mechanism.
Each time TCP retransmits, it sets the next timeout to be twice the last timeout.
o Loss of segments is mostly due to congestion and hence TCP source
does not react aggressively to a timeout.
Jacobson/Karels Algorithm
The main problem with original algorithm is that variance of the sample
RTTs is not taken into account.
o if variation among samples is small, then EstimatedRTT can be trusted o
otherwise timeout should not be tightly coupled with the EstimatedRTT
In this new approach, the sender measures a new SampleRTT as
before. The Deviation amongst RTTs is computed as follows:
Difference = SampleRTT EstimatedRTT
EstimatedRTT = EstimatedRTT + ( × Difference)
Deviation = Deviation + ( |Difference| Deviation)
where is a fraction between 0 and 1
TCP now computes TimeOut as a function of both EstimatedRTT and Deviation
as listed:
TimeOut = × EstimatedRTT + × Deviation
where = 1 and = 4 usually
When variance is small, difference between TimeOut and EstimatedRTT is negligible.
When variance is larger, Deviation plays a greater role in deciding TimeOut.
What is silly window syndrome? When should TCP transmit a segment?
When an ACK arrives, the window enlarges for transmission.

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4.9

Even if window size is less than one MSS, TCP decides to go ahead and
transmit a half-full segment.
The strategy of aggressively taking advantage of any available window leads
to a situation now known as the silly window syndrome.
If the sender aggressively fills, then any small segments introduced into the
system remains in the system indefinitely as it does not combine with
adjacent segments to create larger ones as shown.

Nagle’s Algorithm
Nagle's suggests a solution as to what the sending TCP should do when there is
data to send and window size is less than one MSS. The algorithm is listed below:
When the application produces data to send
if both the available data and the window MSS
send a full segment
else
if there is unACKed data in flight
buffer the new data until an ACK arrives
else
send all the new data now
It’s always OK to send a full segment if the window allows.
It’s also OK to immediately send a small amount of data if there are currently
no segments in transit, but if there is anything in flight, the sender must wait
for an ACK before transmitting the next segment.
Explain TCP congestion control techniques in detail.
In TCP congestion control, each source has to determine the available
capacity in the network, so that it can send packets without loss.
By using ACKs to pace transmission of packets, TCP is said to be self-clocking. TCP
maintains a state variable CongestionWindow for each connection. Therefore
MaxWindow = MIN(CongestionWindow, AdvertisedWindow)
EffectiveWindow = MaxWindow (LastByteSent LastByteAcked)
Thus, a TCP source is allowed to send no faster than network or destination host
The problem is that available bandwidth changes over time. The three
congestion control mechanism are:
o Additive Increase/Multiplicative Decrease
o Slow Start
o Fast Retransmit and Fast Recovery
Additive Increase/Multiplicative Decrease (AIMD)
TCP source sets the CongestionWindow based on the level of congestion it
perceives to exist in the network.
The additive increase/multiplicative decrease (AIMD) mechanism works as follows: o The
source increases CongestionWindow when level of congestion goes down
and decreases CongestionWindow when level of congestion goes up.
TCP interprets timeouts as a sign of congestion and reduces the rate at
which it is transmitting.

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4.10

o Each time a timeout occurs, the source sets CongestionWindow to half

of its previous value. This is known as multiplicative decrease.
o For example, if CongestionWindow is set to 16 packets, after a packet
loss, it is set to 8.
o The CongestionWindow is not allowed to fall below one packet size or
MSS, irrespective of the level of congestion.
Every time, the source successfully sends one packet, CongestionWindow is
increased by a fraction (additive increase).
o An ACK acknowledges receipt of MSS bytes, the increment is computed as
Increment = MSS × (MSS/CongestionWindow)
CongestionWindow += Increment

This pattern of continually increasing and decreasing the congestion window

continues throughout the lifetime of the connection

When the current value of CongestionWindow as a function of time, it results

as a saw-tooth pattern.
AIMD decreases its CongestionWindow aggressively but increases conservatively.
o Having small CongestionWindow only results in less probability of
packets being dropped.
o Thus congestion control mechanism becomes stable.
Since timeout is an indication of congestion that triggers multiplicative
decrease, TCP needs the most accurate timeout mechanism.
AIMD is appropriate only when source is operating close to network capacity.
Slow Start
Slow start increases the congestion window exponentially, rather than
linearly. It is usually used from cold start.
The source starts by setting CongestionWindow to one packet.
o When ACK arrives, TCP adds 1 to CongestionWindow and sends two packets. o
Upon receiving two ACKs, TCP increments CongestionWindow by 2 and
sends four packets.
o Thus TCP doubles the number of packets every RTT as shown.

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4.11

Slow start provides exponential growth and is designed to avoid bursty nature of
TCP. Initially TCP has no idea about congestion, henceforth it increases
CongestionWindow rapidly until there is a packet
loss. When a packet is lost:
o TCP immediately decreases CongestionWindow by half (multiplicative
decrease).
o It stores the current value of CongestionWindow as CongestionThreshold
and resets to CongestionWindow one packet
o The CongestionWindow is incremented one packet for each ACK arrived
until it reaches CongestionThreshold and thereafter one packet per RTT.
In initial stages, TCP loses more packets because it attempts to learn the
available bandwidth quickly through exponential increase

In example, initial slow start causes increase in CongestionWindow up to

34KB. Congestion occurs at 2secs and loss of packets results.
o CongestionThreshold is set to 17KB and CongestionWindow to 1 packet.
o Thereafter additive increase is followed
Fast Retransmit and Fast Recovery
Fast retransmit is a heuristic that triggers the retransmission of a dropped packet
sooner than the regular timeout mechanism. It does not replace regular timeouts.
When a packet arrives out of order, the receiving TCP resends the same
acknowledgment (duplicate ACK) it sent the last time.
The sending TCP waits for three duplicate ACK, to confirm that the packet is
lost before retransmitting the lost packet.
This is known as fast retransmit and it signals congestion.
Instead of setting CongestionWindow to one packet, this method uses the ACKs
that are still in pipe to clock the sending of packets. This is called fast recovery.
The fast recovery mechanism removes slow start phase and follows additive
increase. The fast retransmit/recovery results increase in throughput by 20%.

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4.12

In example the third packet gets lost. The sender on receiving three
duplicate ACKs (ACK 2) retransmits the third packet.

In graph shown, fast recovery avoids slow start from 3.8 to 4 sec. Therefore
congestion window is reduced by half from 22 KB to 11 KB.
Slow start is only used at the beginning of a connection and after regular timeout.
At other times, the congestion window follows a pure additive
increase/multiplicative decrease pattern
TCP's fast retransmit can detect up to three dropped packets per window.
Explain in detail about TCP congestion avoidance algorithms.
Congestion avoidance mechanisms prevent congestion before it actually occurs.
When congestion is likely to occur, TCP decreases load on the network.
TCP creates loss of packets in order to determine bandwidth of the
connection The three congestion-avoidance mechanisms are:
1. DECbit
2. Random Early Detection (RED)
3. Source-based congestion avoidance
DECbit
Was developed for use on Digital Network Architecture
In DEC bit, each router monitors the load it is experiencing and explicitly
notifies the end node when congestion is about to occur by setting a binary
congestion bit called DECbit in packets that flow through it.
The destination host copies the DECbit onto the ACK and sends back to the source.
Eventually the source reduces its transmission rate and congestion is avoided.

Algorithm
A single congestion bit is added to the packet header.
A router sets this bit in a packet if its average queue length is 1.
The average queue length is measured over a time interval that spans the
last busy + last idle cycle + current busy cycle.
Router calculates average queue length by dividing the curve area by time interval

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4.13

The source computes how many ACK has DEC bit set for the previous
window packets it has sent.
1. If it is less than 50% then source increases its congestion window by 1 packet.
2. Otherwise, source decrease the congestion window by 87.5%.
Random Early Detection (RED)
Proposed by Floyd and Jackson
In RED, router implicitly notifies the source that congestion is likely to occur
by dropping one of its packets.
The source is notified by timeout or duplicate ACK.
The router drops a few packets earlier before it runs out of space, so that
it need not drop more packets later.
Each incoming packet is dropped with a probability known as drop probability when
the queue length exceeds drop level.
Algorithm
RED computes average queue length using a weighted running average as follows:
AvgLen = (1 Weight) × AvgLen + Weight × SampleLen
o where 0 < Weight < 1 and SampleLen is length of the queue when a sample
measurement is made.
o The weighted running average detects long-lived congestion.
RED has two queue length thresholds MinThreshold and MaxThreshold. When a
packet arrives at the gateway, RED compares the current AvgLen with these
thresholds and decides whether to queue or drop the packet as follows:
if AvgLen MinThreshold
queue the packet
if MinThreshold < AvgLen < MaxThreshold
calculate probability P
drop the arriving packet with probability
P if MaxThreshold AvgLen
drop the arriving packet
P is a function of both AvgLen and how long it has been since the last packet
was dropped. It is computed as
TempP = MaxP × (AvgLen MinThreshold)/(MaxThreshold MinThreshold) P
= TempP/(1 count × TempP)
The probability of drop increases slowly when AvgLen is between the two thresholds,
reaching MaxP at the upper threshold, at which point it jumps to unity.
MaxThreshold is set to twice of MinThreshold as it works well for the Internet traffic.
Because RED drops packets randomly, the probability that RED decides to drop a
flow’s packet(s) is roughly proportional to share of the bandwidth for that flow.

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4.14

RED thresholds Drop probability function

Source-Based Congestion Avoidance

The source looks for signs of congestion on the network, for example, a
considerable increase in the RTT, indicate queuing at a router.
Some mechanisms
1. Every two round-trip delays, it checks to see if the current RTT is greater
than the average of the minimum and maximum RTTs.
a. If it is, then the algorithm decreases the congestion window by one-eighth.
b. Otherwise the normal increase as in TCP.
2. Every RTT, it increases the window size by one packet and compares the
throughput achieved to the throughput when the window was one packet smaller.
a. If the difference is less than one-half the throughput achieved when only
one packet was in transit, it decreases the window by one packet.
TCP Vegas
In standard TCP, it was observed that throughput increases as congestion
window increases, but not beyond the available bandwidth.
Any further increase in the window size only results in packets taking up
buffer space at the bottleneck router
TCP Vegas uses this idea to measure and control the right amount of extra
data in transit.
If a source is sending too much extra data, it will cause long delays and
possibly lead to congestion.
TCP Vegas’s congestion-avoidance actions are based on changes in the
estimated amount of extra data in the network.
A flow’s BaseRTT is set to the minimum of all RTTs and is mostly the first
packet sent.
The expected throughput is given by ExpectedRate = CongestionWindow/BaseRTT
The sending rate, ActualRate is computed by dividing number of bytes
transmitted during a RTT by that RTT.
The difference between two rates is computed, say Diff = ExpectedRate –
ActualRate Two thresholds and are defined such that <
o When Diff < , congestion window is linearly increased during the next RTT o
When Diff > , congestion window is linearly decreased during the next RTT
o When < Diff < , the congestion window is unchanged
When actual and expected output varies significantly, the congestion
window is reduced as it indicates congestion in the network.
When actual and expected output is almost the same, the congestion
window is increased to utilize the available bandwidth.
The overall goal is to keep between and extra bytes in the network.

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4.15

The expected & actual throughput with thresholds and (shaded region) is shown.

What is the need for QoS?

Certain applications are not satisfied with best-effort service offered by the
network. o Multimedia applications require minimum bandwidth.
o Real-time applications require timeliness rather than ensuring
correctness of the message.
Therefore the network needs to support different level of service for
different set of application.
Classify the real-time applications based on QoS.
Real Time Tolerance is based on
response to loss of packets
Adaptability is based
Intolerant Tolerant on response to delay
Eg FTP Eg. Audio experienced.
Bit rate vs quality
Non-adaptive Adaptive

Rate Adaptive Delay Adaptive

Define QoS.
QoS is defined as a set of attributes pertaining to the performance of a
connection. The attributes may be either user or network oriented.
QoS on the Internet can be broadly classified into
o Integrated Services (IntSrv)
o Differentiated Services
Explain how QoS is provided through integrated services.
Integrated Services (IntSrv) is a flow-based QoS model, i.e., user creates flow
from source to destination and informs all routers of the resource requirement.
Service Classes
The two classes of service defined are Guaranteed and Controlled load service.
Guaranteed service in which the network assures that delay will not be beyond
some maximum if flow stays within TSpec. It is designed for intolerant applications.
Controlled load service meets the need of tolerant, adaptive applications
which requests low-loss or no-loss such as file transfer, e-mail, etc.
Flowspec
The set of information given to the network for a given flow is called
flowspec. It has two parts namely
o Tspec defines the traffic characterization of the flow
o Rspec defines resources that the flow needs to reserve (buffer, bandwidth, etc.)

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4.16

TSpec
The bandwidth of real-time application varies constantly for most application.
The average rate of flows cannot be taken into account as variable bit rate applications
exceed the average rate. This leads to queuing and subsequent delay/loss of packets.
Token Bucket
The solution to manage varying bandwidth is to use token bucket filter that
can describe bandwidth characteristics of a source/flow.
The two parameters used are token rate r and a bucket
depth B A token is required to send a byte of data.
A source can accumulate tokens at rate r/second, but not more than B tokens.
Bursty data of more than r bytes per second is not permitted. Therefore
bursty data should be spread over a long interval.
The token bucket provides information that is used by admission control
algorithm to determine whether or not to consider the new request for service.
Example
Flow A generates data at a steady rate of 1 Mbps, which is described using
a token bucket filter with rate r = 1 Mbps and a bucket depth B = 1 byte.
Flow B sends at rate of 0.5 Mbps for 2 seconds and then at 2 Mbps for 1
second, which is described using a token bucket filter with rate r = 1 Mbps
and a bucket depth B = 1 MB.
The additional depth allows it to accumulate tokens when it sends 0.5 Mbps
(2 × 0.5 = 1 MB) and uses the same to send for bursty data of 2 Mbps.

Admission Control
When a flow requests a level of service, admission control examines TSpec
and RSpec of the flow.
It checks to see whether the desired service can be provided with currently available
resources, without causing any worse service to previously admitted flows.
o If it can provide the service, the flow is admitted otherwise denied.
The decision to allow/deny a service can be heuristic such as "currently
delays are within bounds, therefore another service can be admitted."
Admission control is closely related to policy. For example, a network admin will
allow CEO to make reservations and forbid requests from other employees.
Reservation Protocol (RSVP)
The Resource Reservation Protocol (RSVP) is a signaling protocol to help IP
create a flow and make a resource reservation.
RSVP provides resource reservations for all kinds of traffic including multimedia
which uses multicasting. RSVP supports both unicast and multicast flows.
RSVP is a robust protocol that relies on soft state in the routers.

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4.17

o Soft state unlike hard state (as in ATM, VC), times out after a short
period if it is not refreshed. It does not require to be deleted.
o The default interval is 30 ms.
Since multicasting involves large number of receivers than senders, RSVP follows receiver-
oriented approach that makes receivers to keep track of their requirements.

RSVP Messages
To make a reservation, the receiver needs to know:
o What traffic the sender is likely to send so as to make an appropriate
reservation, i.e., TSpec.
o Secondly, what path the packets will travel.
The sender sends a PATH message to all receivers (downstream) containing TSpec.
A PATH message stores necessary information for the receivers on the way.
PATH messages are sent about every 30 seconds.
The receiver sends a reservation request as a RESV message back to the sender
(upstream), containing sender's TSpec and receiver requirement RSpec.
Each router on the path looks at the RESV request and tries to allocate
necessary resources to satisfy and passes the request onto the next router.
o If allocation is not feasible, the router sends an error message to the receiver
If there is any failure in the link a new path is discovered between sender and the
receiver. The RESV message follows the new path thereafter.
A router reserves resources as long as it receives RESV message, otherwise
released. If a router does not support RSVP, then best-effort delivery is followed.
Reservation Merging
In RSVP, the resources are not reserved for each receiver in a flow, but merged.
When a RESV message travels from receiver up the multicast tree, it is likely to come
across a router where reservations have already been made for some other flow.
If new resource requirements can be met using existing allocations, then
new allocation is not done.
o For example, receiver B has already made a request for 3 Mbps. If A
comes with a new request for 2 Mbps, then no new reservations are made.
A router that handles multiple requests with one reservation is known as
merge point. This is because, different receivers require different quality.
Reservation merging meets the needs of all receivers downstream of the merge point.

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4.18

Packet classification is done by examining the fields source address, destination

address, protocol number, source port and destination port in the packet header.
Weighted fair queuing or a combination of queuing disciplines is used.
List the disadvantages of integrated services
Scalability IntSrv requires router to maintain information for each flow, which
is not feasible for today's internet growth
Service type limitation Only two types of services are provided. Certain
applications may require more than the offered services.
Explain how QoS is provided through differentiated services
Differentiated Services (DiffServ) is a class-based QoS model designed for IP.
Premium class
The default best-effort model is enhanced as a new class called premium.
The premium packets have bits set (marked) in the header by the gateway
router or by the ISP router.
IETF has defined a set of behaviors for routers known as per-hop behaviors (PHB).
IETF has replaced the existing TOS field in IPv4 or Class field in IPv6 with 6-
bit DiffServ code points (DSCP) and remaining 2 bits unused.

6-bit DSCP can be used to define 64 PHB that could be applied to a packet.
The three PHBs defined are default PHB (DE PHB), expedited forwarding
PHB (EF PHB) and assured forwarding PHB (AF PHB).
The DE PHB is the same as best-effort delivery and is compatible with TOS.
Expedited Forwarding (EF PHB)
Packets marked for EF treatment should be forwarded by the router with
minimal delay (latency) and loss by ensuring required bandwidth.
A router guarantees EF, only if arrival rate of EF packets is less than forwarding rate
The rate limiting of EF packets is achieved by configuring routers at the edge of an
administrative domain to ensure that it is less than bandwidth of the slowest link.
Queuing can be either using strict priority or weighted fair queuing.
o In strict priority, EF packets are preferred over others, leaving less
chance for other packets to go through.
o In weighted fair queuing, other packets are given a chance, but there is a
possibility of EF packets being dropped, if there is excessive EF traffic.
Assured Forwarding
The AF PHB is based on RED with In and Out (RIO) algorithm.
In RIO, the drop probability increases as the average queue length increases. The
following example shows RIO with two classes named in and out.

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4.19

The out curve has a lower MinThreshold than in curve, therefore under low
levels of congestion, only packets marked out will be discarded.
If the average queue length exceeds Minin, packets marked in are also dropped.
The terms in and out are explained with the example "Customer X is allowed
to send up to y Mbps of assured traffic".
o If the customer sends packets less than y Mbps then packets are marked in. o
When the customer exceeds y Mbps, the excess packets are marked out.
Thus combination of profile meter at the edge router and RIO in all routers, assures (but
does not guarantee) the customer that packets within the profile will be delivered
RIO does not change the delivery order of in and out packets.
If weighted fair queuing is used, then weight for the premium queue is
chosen using the formula. It is based on the load of premium packets.
Bpremium = Wpremium / (Wpremium + Wbest-effort)
o For example, if weight of premium queue is 1 and best-effort is 4, then
only 20% of the link is reserved for premium packets.
How differentiated services overcome the limitations of integrated services?
1. The main processing was moved from the core of the network to edge of the network
(scalability). Thus routers need not store information about flows. The applications
define the type of service they need each time when a packet is sent.
2. The per-flow service is changed to per-class service. The router routes the
packet based on class of service defined in the packet, not the flow.
Different types of classes (services) based on the needs of applications.
Write short notes on ATM QoS.
The five ATM service classes are:
1. constant bit rate (CBR)
2. variable bit rate—real-time (VBR-rt)
3. variable bit rate—non-real-time (VBR-nrt)
4. available bit rate (ABR)
5. unspecified bit rate (UBR)
Constant Bit Rate
Sources of CBR traffic are expected to send at a constant rate.
The source’s peak rate and average rate of transmission are equal.
CBR class is designed for customers who need real-time audio or video
services. CBR is a relatively easy service for implementation
Variable Bit Rate
The VBR class is divided into two subclasses: real-time (VBR-rt) and non-
real-time (VBR-nrt).
VBR-rt is designed for users who need real-time services (such as voice and video
transmission) and use compression techniques to create a variable bit rate.
The traffic generated by the source is characterized by a token bucket, and
the maximum total delay required through the network is specified.
VBR-nrt bears some similarity to IP’s controlled load service. The source
traffic is specified by a token bucket.
VBR-nrt is designed for users who do not need real-time services but use
compression techniques to create a variable bit rate
Unspecified Bit Rate

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4.20

UBR class is a best-effort delivery service that does not guarantee anything. UBR
allows the source to specify a maximum rate at which it will send.
o Switches may make use of this information to decide whether to
admit or reject or negotiate with the source for a less peak rate.
Available Bit Rate
ABR apart from being a service class also defines a set of congestion-
control mechanism.
The ABR mechanisms operate over a virtual circuit by exchanging special ATM
cells called resource management (RM) cells between the source and destination.
RM cells work as explicit congestion feedback mechanism as shown below.

ABR allows a source to increase or decrease its allotted rate as conditions dictate.
ABR class delivers cells at a minimum rate. If more network capacity is
available, this minimum rate can be exceeded.
ABR is suitable for applications that are bursty in nature.
What is equation based congestion control?
TCP’s congestion-control algorithm is not appropriate for real-time applications.
A smooth transmission rate is obtained by ensuring that flow’s behavior
adheres to an equation that models TCP’s behavior.

To be TCP-friendly, the transmission rate must be inversely proportional to

the round-trip time (RTT) and the square root of the loss rate ( ).

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5.1

Discuss the components of an email system and the protocols used.

The e-mail system involved in Alice sending a message to Bob is shown.

User Agent
A user agent (UA) is software that is either command (eg. pine, elm) or GUI
based (eg. Microsoft Outlook, Netscape). It facilitates:
o Compose helps to compose messages by providing template with built-in editor.
o Read checks mail in the incoming box and provides information such as sender,
size, subject and flag (read, new).
o Reply allows user to reply (send message) back to sender
o Forward facilitates forwarding message to a third party.
o Mailboxes creates two mailboxes for each user, namely inbox (to store
received emails) and outbox (to keep all sent mails).
Message Format
RFC822 defines message to have two parts namely header and a body.
The message header is a series of <CRLF> terminated lines. Each header line contains a
type and value separated by a colon (:). It is filled by the user/system. Some of them are:
o From user who sent the message
o To identifies the message recipient(s).
o Subject says something about the purpose of the
message o Date when the message was transmitted
o E-mail address consists of user_name@domain_name where
domain_name is hostname of the mail server.
The body of the message contains the actual information
o The header is separated from the message body by a blank line.
Initially email system was designed to send messages only in NVT 7-bit ASCII format.
o Languages such as French, German, Chinese, Japanese were not supported.
o Image, audio and video files cannot be sent.
Multipurpose Internet Mail Extensions (MIME)
MIME is a supplementary protocol that allows non-ASCII data to be sent through e-mail.

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5.2

MIME transforms non-ASCII data to NVT ASCII and delivers to client MTA. The
NVT ASCII data is converted back to non-ASCII form at the recipient mail server.
MIME defines five headers. They are:
o MIME-Version specifies the current version 1.1
o Content-Type specifies message type such as text (plain, html), image
(jpeg, gif), audio, video and application (postscript, msword). If more than
one type exists, then it is termed as multipart (mixed).
o Content-Transfer-Encoding defines how data in the message body is
encoded such as binary, base64, 7-bit, etc.
o Content-Id unique identifier the whole message in a multiple message
type. o Content-Description describes type of the message body.
Example
MIME-Version: 1.0
Content-Type: multipart/mixed; boundary="-------
417CA6E2DE4ABCAFBC5" From: Alice Smith <[email protected]>
To: [email protected]
Subject: promised material
Date: Mon, 07 Sep 1998 19:45:19 -0400
---------417CA6E2DE4ABCAFBC5 Content-
Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit
…
---------417CA6E2DE4ABCAFBC5
Content-Type: image/jpeg Content-
Transfer-Encoding: base64
Message Transfer Agent (MTA): SMTP
Message Transfer Agent (MTA) is a mail daemon that helps to transmit/receive
message over the network.
To send mail a system must have the client MTA, and to receive mail a system
must have a server MTA.
Simple Mail Transfer Protocol (SMTP) defines communication between
client/server MTA.
SMTP uses TCP connection on port 25 to forward the entire message and store at
intermediate mail servers/mail gateways until it reaches the recipient mail server.

SMTP defines how commands and responses must be sent back and forth.

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5.3

Command Description
MAIL FROM Sender of the message
RCPTTO Recipient of the message
DATA Body of the mail
QUIT Terminate
VRFY Name of recipient to be verified before forwarding
EXPN Mailing list to be expanded

Common responses sent from server MTA are:

Code Description
220 Service ready
250 Request completed
354 Start mail input
450 Mailbox not available
500 Syntax error; unrecognized command
551 User not local
Example
HELO cs.princeton.edu
250 Hello [email protected]
[128.12.169.24] MAIL FROM:<[email protected]>
250 OK
RCPT TO:<[email protected]>
250 OK
DATA
354 Start mail input; end with <CRLF>.<CRLF>
Blah blah blah...
...etc. etc. etc.
<CRLF>.<CRLF>
250 OK
QUIT
221 Closing connection
In each exchange, the client posts a command and the server responds with a
code. and a human-readable explanation for the code.
After the commands and responses, client sends the message which is ended by
a period (.) and terminates the connection.
Message Access Agent (MAA)/Mail Reader: POP and IMAP
MAA or mail reader allows user to retrieve messages from the mailbox, so that
user can perform actions such as reply, forwarding, etc.
The two message access protocols are:
o Post Office Protocol, version 3 (POP3)
o Internet Mail Access Protocol, version 4 (IMAP4)
SMTP is a push type protocol whereas POP3 and IMAP4 are pop type protocol.

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5.4

POP3
POP3 is simple and limited in functionality
POP3 works in two modes namely, delete and keep mode.
o In delete mode, mail is deleted from the mailbox after retrieval
o In keep mode, mail after reading is kept in mailbox for later retrieval.
POP3 client is installed on the recipient computer and POP3 server on the
mail server. The client opens a connection to the server on TCP port 110.
The client sends username and password to access the mailbox and retrieve the messages.

IMAP4
IMAP is a client/server protocol running over TCP. The client issues commands
and the mail server responds.
The exchange begins with the client authenticating itself to access the mailbox.
This is represented as a state transition diagram.

When the user asks to FETCH a message, server returns it in MIME format and
the mail reader decodes it.
IMAP also defines message attributes such as size and flags such as Seen, Answered,
Deleted and Recent.

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5.5

Explain HTTP protocol in detail.

WWW is a distributed client/server service, in which a client (web browser) can access a
service through a server, where the service is distributed over many locations called sites.
Web browsers allow users to access files using uniform resource locator (URL).
When user enters URL, the browser forms a request message and sends to the server.
The server retrieves the requested URL and sends it as a response message.
The browser displays the response in HTML / appropriate format.
HTTP is a stateless request/response protocol that governs client/server communication.
HTTP uses TCP connection on port 80 to transfer data between client and server.

Message Format
START_LINE <CRLF>
MESSAGE_HEADER
<CRLF> <CRLF>
MESSAGE_BODY <CRLF>
Request Message
Request Line
Request Header : Value
Blank Line
Body (optional)
Request Line
The Request line contains three fields as shown.
Request Type URL HTTP version
o HTTP version specifies current version of the protocol i.e., 1.1 o
The Request type specifies methods that operate on the URL are:
Method Description
GET retrieve document specified as URL
HEAD retrieve meta-information about the URL document
POST send information from client to the server
PUT store document under specified URL
TRACE echoes the incoming request
OPTION request information about available options
DELETE delete specified URL
For example, the request line to retrieve file index.html on host cs.princeton.edu is
GET http://www.cs.princeton.edu/index.html HTTP/1.1

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5.6

Request Header
Request Header specifies client's configuration and preferred document format:
Request Header Description
Accept-charset specifies the character set the client can handle
Authorization specifies what permissions the client has
From specifies e-mail address of the user
Host specifies host name and port number of the server
If-modified-since server sends the URL if it is newer than specified date
Referrer specifies URL of the linked document
User-agent specifies name of the browser
The above example using request header is specified as
GET index.html HTTP/1.1
Host: www.cs.princeton.edu
Response Messages
Status Line
Response Header : Value
Blank Line
Body
Status line
The Status line contains three fields as shown.
HTTP version Code Status Phrase
The status code field consists of three digits (1xx–Informational, 2xx–Success,
3xx– Redirection, 4xx–Client Error, 5xx–Server Error)
The status phrase explains the status code in text form. Some of them are:
Code Phrase Description
100 Continue Initial request received, client to continue process
200 OK Request is successful
201 Created A new URL is created.
204 No content There is no content in the body.
301 Moved permanently The requested URL is no longer in use
304 Not modified The document has not been modified
401 Unauthorized The request lacks proper authorization
404 Not found The document is not found
500 Internal server error There is an error, such as a crash, at the server site
For example, the server reports as follows, if the requested file is not found
HTTP/1.1 404 Not Found

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5.7

Response Header
Response Header Description
Content-encoding specifies the encoding scheme
Content-length shows length of the document
Content-type specifies the medium type
Expires gives date and time up to which the document is valid
Last-modified gives date and time when the document was last updated
Location specifies location of the created or moved document
The response for a moved page is given below.
HTTP/1.1 301 Moved Permanently
Location: http://www.princeton.edu/cs/index.html.
Persistent vs non-persistent connection
In non-persistent, a TCP connection is required for each request/response
o Imposes high overhead on the server because the server needs N buffers
for N URL pointers and TCP overhead for each connection
In persistent, Client and server can exchange multiple request/response
messages over the same TCP connection
o Eliminates the connection setup overhead and load on the server
o TCP’s congestion window mechanism is able to operate more efficiently.
o The server times out, if there is no request from the client for a specified period
Caching
Caching enables the client to retrieve document faster and reduces load on
the server. Caching can be implemented at different places
o For example, the ISP router can cache pages. Further such request
coming from its clients, the ISP responds.
o Proxy server is a host that keeps copies responses to recent requests.
The client sends request to the proxy server. The proxy server either
responds to client or forwards the request to the server.
o The browser also can cache pages.
Server assigns expiration date (using Expires header field) to each page, beyond
which the page should not be cached.
Therefore prior to caching a page, its expiration date is checked. If a cached
page reaches its expiration, then the page is deleted.
The proxy verifies whether it has the latest document by using If-Modified-Since
header. A page must not be cached if no-cache directive is specified.
Explain the role of DNS on a computer network.
Domain-names are easily remembered than IP address of a host, since it is user-
friendly. Thus, need for a system to map domain name to an IP address that includes:
o A namespace to define domain names without
conflict. o Binding of domain names to IP address
o A name server that returns IP address for a given name

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5.8

During early days of internet, there were only few hundred hosts
o A central authority called the Network Information Center (NIC) maintained
name-to-address bindings in a flat-file called hosts.txt
o A new host that joins the internet would mail its name and IP address
to NIC. o NIC updates hosts.txt and mails to all hosts.
o Name server resolved domain names using a simple lookup on hosts.txt
As hosts grew to thousands and millions, the flat file approach failed, leading to
evolution of DNS in mid 1980s.
Name Hierarchy
DNS uses hierarchical name space for domains in the Internet.
Hierarchical naming permits use of same sub-domain name in different
domains. Domain names are case insensitive and can be up to 63 characters
DNS names are processed from right to left and use periods (.) as separator.
DNS can be used to map names to values, not necessarily domain names to IP address.
DNS hierarchy can be visualized as a tree, where each node in the tree
corresponds to a domain and the leaves relate to hosts.
Six big domains are .edu (education) .com (commercial) .gov (US government) .mil
(US military) .org (non-profitable organization) and .net (network providers).
Top level domain exist one for each country .fr (france) .in (india), etc.

Name Servers
The domain hierarchy is partitioned into zones. Topmost domains are managed by
NIC. Each zone acts as central authority for that part of the sub-tree.

In the .edu hierarchy, princeton is a zone.

o Each zone can be further sub-divided that manage using their own name
servers such as CS department under princeton university.

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5.9

Each zone information is implemented on at least two name servers.

Clients send queries to name servers, and name servers respond to it.
The response contains either the host IP address or address of another
name server Each name server contains a collection of resource records.
A resource record is a name-to-value binding and is a 5-tuple with the following fields
Name Value Type Class TTL
Name tells the domain to which this record applies. It is the primary search key,
used to satisfy queries
The Type field tells what kind of record it is. Some commonly used types are:
o NS Value field contains a name server
o CNAME Value field contains canonical name for the host. Used to define aliases.
o MX Value field contains a mail server that accepts messages for the domain.
o A Value field contains an IP address
The Value field can be a number, a domain name, or an ASCII string. The
semantics depend on the record type
For internet information, the Class field is always IN.
The TTL field gives an indication of how long the resource record is valid.
Root name server
The root name server contains an NS record for each second-level server.
It also has an A record that translates this name into corresponding IP
address. The following shows part of .edu root name server
(princeton.edu, cit.princeton.edu, NS, IN)
(cit.princeton.edu, 128.196.128.233, A, IN)
…
Zone name server
The zone name server princeton.edu has a name server available on host
cit.princeton.edu that contains the following records.
Some records contain A records, whereas others point to next level name servers.
(cs.princeton.edu, gnat.cs.princeton.edu, NS,
IN) (gnat.cs.princeton.edu, 192.12.69.5, A, IN)
…

Eventually, third-level name server, such as the domain cs.princeton.edu,

contains A records for all of its hosts.
(cs.princeton.edu, gnat.cs.princeton.edu, MX, IN)
(cicada.cs.princeton.edu, 192.12.69.60, A, IN)
(cic.cs.princeton.edu, cicada.cs.princeton.edu, CNAME,
IN) (gnat.cs.princeton.edu, 192.12.69.5, A, IN)
Name Resolution for cicada.cs.princeton.edu
1. The client first sends a query containing cicada.cs.princeton.edu to the root server.
2. The root server, does not finds an exact match, but locates the NS record for
princeton.edu

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5.10

3. The root returns the A record for princeton.edu back to the client.
4. The client sends the same query to 128.196.128.233 and receives the A record for
cs.princeton.edu
5. Finally the client sends the query to 192.12.69.5 and gets the A record for
cicada.cs.princeton.edu
Drawbacks
All hosts should know the root name server, which is not feasible.
Instead, the client can send query to the local name server that it knows
The local name server can query the root name server on behalf of the client.
Once the local NS gets the required response, it caches the A record based on
TTL and sends the record to the client.

Why is POP not preferred?

It does not allow the user to organize their mail on the
server The user cannot have different folders on the server
It does not allow the user to partially check the contents of the mail before downloading
List the advantages of IMAP over POP
IMAP4 is more powerful and more complex than POP3. The additional
features are: A user can check the e-mail header prior to downloading.
A user can search e-mail for a specific string of characters prior to downloading.
A user can partially download e-mail. This is especially useful if bandwidth is
limited and the e-mail contains multimedia with high bandwidth requirements.
A user can create, delete, or rename mailboxes on the mail server.
A user can create a hierarchy of mailboxes in a folder for e-mail storage.
Explain how SNMP is used to manage nodes on the network
Simple Network Management Protocol (SNMP) is a framework for managing
devices in an internet using TCP/IP.
It provides a set of fundamental operations for monitoring and maintaining an
internet SNMP uses the concept of manager and agent.
o A manager is a host that runs the SNMP client program.

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5.11

o A managed station called an agent, is a router that runs the SNMP server program
SNMP is an application layer protocol, therefore it can monitor devices of different
manufacturers installed on different physical
networks. SNMP management includes:
o A manager that checks an agent by requests information on behavior of the agent.
o A manager forces an agent to perform a task by setting/resetting values in
the agent database.
o An agent warns the manager of an unusual situation.
SNMP uses services of UDP on two well-known ports, 161 (agent) and 162 (manager).
SNMP is supported by two other protocols in Internet Network management. They are:
o Structure of Management Information (SMI)
o Management Information Base
(MIB) The role of SNMP is to
o Define format of the packet to be sent from a manager to an agent and vice versa.
o Interprets the result and creates statistics
o Responsible for reading and setting object
values The role of SMI is to
o Define rules for naming objects and object types.
o Uses Basic Encoding Rules to encode data to be transmitted over the
network. The role of MIB is to
o creates a collection of named objects, their types, and their relationships
to each other in an entity to be managed
Object Identifier
SMI uses an object identifier, which is a hierarchical identifier based on a tree
structure The tree structure starts with an unnamed root.
Each object can be defined by using a sequence of integers separated by dots.

The objects that are used in SNMP are located under the mib-2 object, so their
identifiers always start with 1.3.6.1.2.1
Object identifiers follow lexographic ordering.

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5.12

MIB Groups
Each agent has its own MIB2 (version 2), which is a collection of all the objects
that the manager can manage.
The objects in MIB2 are categorized under 10 different groups namely system,
interface, address translation, ip, icmp, tcp, udp, egp, transmission, and snmp.
o sys (system defines general information about the node such as the
name, location, and lifetime.
o if (interface defines information about all the interfaces of the node such as physical
address and IP address, packets sent and received on each interface, etc.
o at (address translation defines information about the ARP table
o ip defines information related to IP such as the routing table, statistics on
datagram forwarding, reassembling and drop, etc.
o tcp defines general information related to TCP, such as the connection table,
time-out value, number of ports, and number of packets sent and received.
o udp information on UDP traffic such as total number of UDP packets sent
and received.

MIB variables
MIB variables are of two types namely simple and table.
To access any of the simple variable content, use id of the group (1.3.6.1.2.1.7)
followed by the id of the variable and an instance suffix, which is 0.
o For example, variable udpInDatagrams is accessed as 1.3.6.1.2.1.7.1.0

In case of table, only leaf elements are accessible.

o In this case, the group id is followed by table id and so on up to the leaf element.
o To access a specific instance (row) of the table, add the index to the above ids.
o The indexes are based on the value of one or more fields in the entries.
o Tables are ordered according to column-row rules, i.e one should go
column by column from top to bottom.

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5.13

SNMPv3 PDU
SNMP is request/reply protocol that defines PDUs GetRequest, GetNextRequest,
GetBulkRequest, SetRequest, Response and Trap.

o GetRequest used by manager to retrieve value of agent's variable(s)

o GetNextRequest used by manager to retrieve next entries in a agent's
table o SetRequest used by manager to set value of an agent's variable
o Response sent from an agent to manager in response to GetRequest/
GetNextRequest that contains value of variables
o Trap sent from an agent to the manager to report an event such as reboot.
PDU Format

The SNMP client puts the identifier for the MIB variable it wants to get into the
request message, and sends this message to the server.
The server then maps this identifier into a local variable, retrieves the current value
held in this variable, and uses BER to encode the value it sends back to the client.
Discuss Telnet in detail
TErminaL NETwork (TELNET) is a general-purpose client/server application
program. TELNET is the standard TCP/IP protocol for virtual terminal.
TELNET enables connection to a remote system in such a way that the local
terminal appears to be a terminal at the remote system.
TELNET was designed during days of time-sharing environment in which a large
computer supported multiple users.
Interaction between user and computer occurs through a terminal (keyboard +
monitor + mouse).
Each user has an identification name and a password.
To access, user logs into the system with a user id / log-in name.
The user is authenticated using password and hence unauthorized access is prevented.
Remote Logon
The process of remote login using TELNET client and server program is shown.

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5.14

The user keystrokes are sent to the terminal driver, where the local operating
system accepts the characters but does not interpret them.
The characters are sent to the TELNET client, which transforms the characters to
a universal character set called Network Virtual Terminal (NVT) characters and
puts it over the network.
The commands/text in NVT form reaches the remote host.
The TELNET server at well-known port 23, converts NVT characters onto remote
character set.
Since the operating system is not designed to receive data from TELNET server,
data is redirected via a pseudo terminal driver to the remote operating system.
The remote operating system passes the data to the corresponding applications.
NVT Character Set
Every operating system use a special combination of characters as tokens
o For example, the end-of-file token in DOS is Ctrl+z, whereas in UNIX it is Ctrl+d.
TELNET solves the problem of heterogeneity, by defining a universal interface called the
network virtual terminal (NVT) character set.
Data transmitted over the network is NVT, whereas at the host level data is
processed using its own character set.
NVT uses two sets of 8-bit characters, one for data and the other for control.
o For data, the MSB is 0 and for control it
is 1. Some NVT control characters are:
Character Purpose
EOF End of file
EOR End of record
IP Interrupt process
AYT Are you there
EC Erase character
EL Erase line
IAC Interrupt as control

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5.15

TELNET uses the same connection to send both data and control characters.
To distinguish data from control characters, each sequence of control characters
is preceded by a special control character called IAC.
For example, to display file1, the command is cat file1, by mistake the user types
cat filea<backspace>1.

Options
TELNET lets the client and server negotiate options before or during the session.
Options are extra features available with a more sophisticated terminal whereas
simple terminals use default features. Some options are
Options Purpose
Echo Echo the received data to the sender
Status Request the status of TELNET
Line mode Change to line mode.
The control characters used for option negotiation are WILL, WONT, DO and DONT.
Modes
TELNET operate in three modes namely default, character and line mode.
o In default mode, the client sends characters only after the line is typed.
o In character mode, each character typed is sent by the client to the server. o
In line mode, line editing is done by the client and sends after a line is typed
Briefly explain the transfer of file contents using FTP.
File Transfer Protocol (FTP) is the standard provided by TCP/IP for copying a file
from one host to another.
FTP establishes two connections between hosts
o Data connection is used for data transfer
o Control connection is used for control information.
o FTP uses two well-known TCP ports, 21 for control and 20 for data connection.

Control Connection
FTP uses 7-bit NVT ASCII character set to communicate across the control
connection. Communication is achieved through commands and responses.
Each command or response is only one short line terminated with <CRLF>

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5.16

When a user starts an FTP session, the control connection opens.

While the control connection is open, the data connection can be opened and
closed multiple times if several files are transferred.
Data Connection
File transfer occurs over the data connection under the control of the commands
sent over the control connection.
A file transfer in FTP means one of the following:
o A file is to be copied from the server to the client. This is called retrieving a file. It is
done under the supervision of the RETR command
o A file is to be copied from the client to the server. This is called storing a file. It
is done under the supervision of the STOR command.
o A list of directory or file names is to be sent from the server to the client. This
is done under the supervision of the LIST command.
The client defines the type of file to be transferred, the structure of the data, and
the transmission mode.
Before sending the file through the data connection it is prepared for
transmission through the control connection.
File Type
FTP can transfer either an ASCII file, EBCDIC file, or
image file. o ASCII file is the default format for
transferring text files. o IBM uses EBCDIC encoding.
o The image file is the default format for transferring binary files.
Data Structure
FTP interprets file's data structure as either file, record or page
structure. o In file structure, the file is a continuous stream of bytes.
o In record structure, the file is divided into records (used only for text files)
o In page structure, the file is divided into pages. Each page has a page
number and header. Page access can be random or sequential.
Transmission Mode
FTP uses stream (default), block or compressed mode of transmission.
o In stream mode, data is delivered to TCP as a continuous stream of bytes. If it's a file
structure, end-of-file (EOF) is not needed. In case of record structure, each record is
marked by a end-of-record (EOR) and the end of the file has a EOF character.
o In block mode, data is delivered to TCP in blocks, where each block is preceded by a
3-byte header. The first byte is the block descriptor and next 2 bytes define the size.
o In compressed mode, the compression used is run-length encoding. Consecutive
appearance of character is replaced by an occurrence and count of repetitions.
Example
$ ftp voyager.deanza.tbda.edu
Connected to voyager.deanza.tbda.edu.
220 (vsFTPd 1.2.1)
530 Please login with USER and
PASS. Name: forouzan
331 Please specify the password.

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5.17

Password:
230 Login successful.
ftp> Is reports
150 Here comes the directory listing.
drwxr-xr-x 23027 411 4096 Sep 24 2002 business
drwxr-xr-x 23027 411 4096 Sep 24 2002 school
226 Directory send OK.
What is anonymous FTP?
To use FTP, a user should know user name and password on the remote server.
Some sites have a set of files available for public access, to enable anonymous
FTP. To access these files, a user does not need to have an account.
User access to the system is very limited. For example, most sites allow the user
to download files.
Write short notes on PGP.
Pretty Good Privacy (PGP) is a popular approach in providing encryption and
authentication capabilities for e-mail.
PGP takes note that each user has his own set of criteria by which he/she wants
to trust the keys certified by someone else.
o For example, one may trust signed certificates of co-workers than a
renowned politician and vice-versa.
PGP provides tools needed to manage the level of trust put in these certificates.
PGP allows certification relationships to form an arbitrary mesh and not a rigid
hierarchy as in Privacy Enhanced Mail (PEM).
PGP allows each user to decide for themselves how much trust they wish to
place in a given certificate
o As the number of trust-worthy signatures for a public key increase, validity
for the same and the user's confidence level increases.
PGP key-signing parties are a regular feature of network community meetings
such as IETF. The activities include:
o Collect public keys from known persons.
o Share their public key with others
o Get their public key signed by
others o Sign public key of others
o Collect certificate from trust-worthy persons.
PGP stores the set of collected certificates in a file called key ring.
PGP allows a wide variety of different cryptographic algorithms to be used
o The actual algorithms used in a message are specified in header fields
PGP allows a user to list his preferred algorithms in the file that contains his/her
public key.
Integrity and Authentication
1. Integrity and authentication refers to A sending message to B and proves that it came from
A.
2. A creates a cryptographic checksum over the message body, such as MD5 and
then encrypts the checksum using A’s private key.

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5.18

3. On receipt of the message, B uses PGP’s key management software to search

his key ring for A’s public key.
4. If key is found
a. Checksum of the received message is calculated
b. Encrypted checksum is decrypted using A’s public key,
c. The two checksums are compared. If both are same, then it confirms that
A has sent the message and its integrity.
5. If key is not found, the sender and authenticity of the message cannot be verified.
6. Apart from signature verification, PGP also tells B the level of trust previously
assigned to this public key.

Encryption
1. A randomly picks a per-message key k to encrypt the message using a symmetric
algorithm such as DES
2. The per-message key k is encrypted using B's public key
3. PGP obtains B's public key from A’s key ring and notifies A of the level of trust
assigned to this key.
4. On receipt, B uses its private key to decrypt the per-message key k.
5. The same algorithm is applied to decrypt the message using per-message key k.

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5.19

Write short notes on SSH.

Secure Shell (SSH) provides a remote login service in a secure
manner SSH uses well-known port 22.
SSH is used to provide strong client/server authentication
o Passwords are not sent as clear text over the network. It is sent in encrypted form.
o Thus sending password through un-trusted network is not a
problem Unlike Telnet and rlogin, SSH supports message integrity
and confidentiality SSH version 2 consist of the following protocols
o Transport layer protocol SSH-TRANS
o Authentication protocol SSH-AUTH
o Connection protocol SSH-CONN
SSH-TRANS
SSH-TRANS provides an encrypted channel for
communication. It runs on top of a TCP connection.
Client and server establish secure channel by having the client authenticate the
server using RSA.
o Server informs the client of its public key at the time of connection
o Client warns the user when it tries to connect to the server for the first time,
since it does not know the server
Once authenticated, the client and server establish a session key that they will
use to encrypt any data sent over the channel.
o Client remembers the server's public key
o For future connection, the client compares server's response with the saved key
SSH-TRANS includes a negotiation of the encryption algorithm the two sides are
going to use. For example, 3DES is commonly selected.
SSH-TRANS includes a message integrity check of all data exchanged over the channel.
SSH-AUTH
Server is authenticated during setup of SSH-TRANS channel by
default User can authenticate using any of the three mechanism
1) Login with username and password. Password is sent in encrypted form
2) Public key encryption by asking the user to store user’s public key on the server
3) Host based authentication requires the client to be authenticated when it
connects to server for the first time. Further connection from a trusted host
is believed to be from the same user.
In UNIX,
o /.ssh/known_hosts records the keys for all the hosts the user has logged into
o /.ssh/authorized_keys contains the public keys needed to authenticate the
user when he or she logs into this machine
o /.ssh/identity contains the private keys for authenticating user on remote machine
SSH-CONN
SSH can be extended to support insecure TCP applications such as X Windows,
IMAP mail readers, etc using SSH-CONN.
Insecure applications are run by tunneling through SSH, known as port forwarding.
o Client on host A communicates with server on host B using
SSH. o Client data sent through SSH is encrypted at sender side

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5.20

o The receiving SSH at well-known port decrypts the contents

o Content is forwarded to the actual port on which the server is listening

What is Web-based mail?

E-mail is such a common application that some websites today provide this
service to anyone who accesses the site such as Hotmail, Yahoo, etc.
Mail transfer from Alice's browser to her mail server is done through HTTP
The message transfer from sending mail server to receiving mail server is through SMTP
Finally, the message from the receiving Web server to Bob's browser is done using HTTP
The website sends a form to be filled in by Bob, which includes log-in id and password.
If the credentials match, the e-mail is transferred from Web server to Bob's
browser in HTML format.

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