(An P.S.R.

ENGINEERING COLLEGE, SIVAKASI – 626140

Autonomous Institution, Affiliated to Anna University, Chennai)
DEPARTMENT OF COMPUTER SCIENCE AND ENGINEERING

COURSE CODE /NAME : 191CS54 - COMPUTER NETWORKS

SEMSTER :V

ACADEMIC YEAR :2024-2025(ODD SEMESTER)

DEPARTMENT
OF
COMPUTER SCIENCE AND ENGINEERING

PREPARED BY
([Link], AP/CSE)
([Link] Aruna AP/CSE)
 P.S.R. ENGINEERING COLLEGE, SIVAKASI – 626140
(An Autonomous Institution, Affiliated to Anna University, Chennai)
DEPARTMENT OF COMPUTER SCIENCE AND ENGINEERING
COURSE PLAN
Academic Year: 2024-2025 (Odd Semester)
191CS54 COMPUTER NETWORKS L T P C
3 0 2 4
Programme: B.E. . Computer Science and Engineering Sem: 5 Category: PC
Prerequisite: NIL
Aim: To understand the concepts of computer networks, protocols and data communication.
Course Outcomes: The Students will be able to
CO1: Recall the fundamentals of a computer networks.
CO2: Identify and understand various IEEE Standards for LAN.
CO3: Illustrate multi-channel access Routing Protocols.
CO4: Outline the elements and protocols of transport layer.
CO5: Describe congestion in network layer with routing algorithms.
CO6: Discuss various Application Layer Protocols.

Text
[Link] Cum.
[Link] Topic Name Book with
Hrs Hours
page no
UNIT I – COMPUTER NETWORKS FUNDAMENTALS
LEARNING AND TEACHING AID: Chalk & Board, Presentation Slides
1. Building a network T1 1 1
2. Wired and Wireless Networks- Requirements T1 1 2
3. Layering and protocols T1 2 4
4. Internet Architecture T1 1 5
5. Network software - Performance T1 1 6
6. Link Layer Services – Framing T1 1 7
7. Error Detection – Flow control. T1 2 9
UNIT II – INTERNETWORKING AND MEDIA ACCESS CONTROL
LEARNING AND TEACHING AID: Chalk & Board, Presentation Slides
8. Media Access Control T1 1 10
9. Protocol Formats-Ethernet (802.3) T1 2 12
10. Wireless LANs – 802.11 – Bluetooth T1 2 14
11. Switching and Bridging T1 1 15
12. Basic Internetworking-Protocols T1 1 16
13. IP, CIDR, ARP, RARP, DHCP and ICMP T1 1 17
14. RFC for IP and ARP T1 1 18
UNIT III – ROUTING AND ROUTING PROTOCOLS
LEARNING AND TEACHING AID: Chalk & Board, Presentation Slides
15. Routing T1 1 19
16. RIP-OSPF-IGRP- EIGRP-Metrics T1 2 21
17. Switch Basics-Global Internet-Domains- BGP-IPv6 T1 1 22
18. Multicast Communication T1 1 23
19. IP addresses – Address Classes T1 1 24
20. Sub netting- Super netting- Examples T1 1 25
21. Multicast Routing - DVMRP-PIM T1 1 26
22. RFC – IPV6 and OSPF T1 1 27
UNIT IV – TRANSPORT LAYER
LEARNING AND TEACHING AID: Chalk & Board, Presentation Slides
23. Overview of Transport layer – TCP and UDP T1 2 29
24. Reliable byte Stream (TCP)- T1 1 30
25. Connection Management- Flow control – T1
2 32
Retransmission
26. TCP Congestion control T1 1 33
 27. Congestion avoidance (DEC bit, RED) T1 1 34
28. QoS – Application requirements T1 1 35
29. RFC – TCP and UDP T1 1 36
UNIT V – APPLICATION LAYER AND CASE STUDIES
LEARNING AND TEACHING AID: Chalk & Board, Presentation Slides
1. Electronic Mail- SMTP, POP3, IMAP, MIME T1 2 38
2. HTTP and HTTPS – Web Services T1 1 39
3. DNS – SNMP V T1 2 41
4. Wireshark Packet Capturing Tool - Cisco Packet
website 2 43
Tracer Tool
5. Open Source Network Tools-Manage Engine
website 2 45
Tool.
LECTURE: 45, TOTAL: 45 HOURS

Text Books:
1. Larry L. Peterson, Bruce S. Davie, “Computer Networks: A Systems Approach”, Fifth
Edition, Morgan Kaufmann Publishers, 2011.
References:
1. James F. Kurose, Keith W. Ross, “Computer Networking - A Top-Down Approach
Featuring the Internet”, Fifth Edition, Pearson Education, 2009.
2. Nader. F. Mir, “Computer and Communication Networks”, Pearson Prentice Hall
Publishers, 2010.
3. Ying-Dar Lin, Ren-Hung Hwang, Fred Baker, “Computer Networks: An Open Source
Approach”, McGraw Hill Publisher, 2011
4. Behrouz A. Forouzan, “Data communication and Networking”, 4/e, Tata McGraw-Hill,
2011.
Web Links:
 [Link]
 [Link]

Course Program Specific

Program Outcomes (POs)
Outcome Outcomes (PSOs)
s PO1 PO2 PO3 PO4 PO5 PO6 PO7 PO8 PO9 PO10 PO11 PO12 PSO1 PSO2 PSO3 PSO4
CO1 1 2 2 3 3 3
CO2 1 1 2 2 2 1
CO3 2 2 1 2 2
CO4 1 2 1 1 2 3 1 2 2
CO5 2 1 2 2 1
CO6 2 2 1 2 3 3 3
1: Slight (Low) 2: Moderate (Medium) 3: Substantial (High)

PREPARED BY
([Link], AP/CSE)
([Link] Aruna AP/CSE)
 UNIT 1 – COMPUTER NETWORK FUNDAMENTALS

Building a network – Wired and Wireless Networks- Requirements – Layering

and protocols – Internet Architecture – Network software – Performance- Link
Layer Services – Framing – Error Detection – Flow control.

BUILDING A NETWORK
INTRODUCTION TO NETWORKS
 A network is a set of devices (often referred to as nodes) connected bycommunication
links.
 A node can be a computer, printer, or any other device capable of sending or
receiving data generated by other nodes on the network.
 When we communicate, we are sharing information. This sharing can be local orremote.

CHARACTERISTICS OF A NETWORK
The effectiveness of a network depends on three characteristics.
1. Delivery: The system must deliver data to the correct destination.
2. Accuracy: The system must deliver data accurately.
3. Timeliness: The system must deliver data in a timely manner.

CRITERIA NECESSARY FOR AN EFFECTIVE AND EFFICIENT NETWORK

A network must be able to meet a certain number of criteria. The most important of these are
performance, reliability, and security.

Factors that affect the Factors that affect the Factors that affect the
Performance of a network: Reliability of a network: Security of a network:
1. Number of users 1. Efficiency of software. [Link] data from
2. Type of transmission medium 2. Frequency of failure unauthorized access and
3. Capabilities of the connected3. Recovery time of a network viruses.
hardware after a failure

COMPONENTS INVOLVED IN A NETWORK PROCESS

The five components are:
1. Message - It is the information to be communicated. Popular forms of information
include text, pictures, audio, video etc.
2. Sender - It is the device which sends the data messages. It can be a computer,
workstation, telephone handset etc.
3. Receiver - It is the device which receives the data messages. It can be a computer,
workstation, telephone handset etc.
4. Transmission Medium - It is the physical path by which a message travels from sender
to receiver. Some examples include twisted-pair wire, coaxial cable, radiowaves etc.
5. Protocol - It is a set of rules that governs the data communications. It represents an
agreement between the communicating devices. Without a protocol, two devices may be
connected but not communicating.

KEY ELEMENTS OF PROTOCOL

 Syntax: Refers to the structure or format of the data, meaning the order in which they
are presented.
 Semantics: Refers to the meaning of each section of bits.
 Timing: Refers to two characteristics. (1). When data should be sent and (2). Howfast
they can be sent.

TRANSMISSION MODES
o The way in which data is transmitted from one device to another device is known as
transmission mode.
o The transmission mode is also known as the communication mode.
o Each communication channel has a direction associated with it, and transmission media
provide the direction. Therefore, the transmission mode is also known as a directional
mode.
o The transmission mode is defined in the physical layer.

Types of Transmission mode

The Transmission mode is divided into three categories:
o Simplex Mode
o Half-duplex Mode
o Full-duplex mode (Duplex Mode)

SIMPLEX MODE

o In Simplex mode, the communication is unidirectional, i.e., the data flow in one
direction.
o A device can only send the data but cannot receive it or it can receive the data but cannot
send the data.
o This transmission mode is not very popular as mainly communications require the two-
way exchange of data. The simplex mode is used in the business field as in sales that do
 not require any corresponding reply.
o The radio station is a simplex channel as it transmits the signal to the listeners but never
allows them to transmit back.
o Keyboard and Monitor are the examples of the simplex mode as a keyboard can only
accept the data from the user and monitor can only be used to display the data on the
screen.
o In the simplex mode the full capacity of the communication channel can be utilized
during transmission.

Advantage of Simplex mode:

o In simplex mode, the station can utilize the entire bandwidth of the communication
channel, so that more data can be transmitted at a time.

Disadvantage of Simplex mode:

o Communication is unidirectional, so it has no inter-communication between devices.

HALF-DUPLEX MODE

o In a Half-duplex channel, direction can be reversed, i.e., the station can transmit and
receive the data as well.
o Messages flow in both the directions, but not at the same time.
o The entire bandwidth of the communication channel is utilized in one direction at a time.
o In half-duplex mode, it is possible to perform the error detection, and if any erroroccurs,
then the receiver requests the sender to retransmit the data.
o A Walkie-talkie is an example of the Half-duplex mode.
o In Walkie-talkie, one party speaks, and another party listens. After a pause, the other
speaks and first party listens. Speaking simultaneously will create the distorted sound
which cannot be understood.

Advantage of Half-duplex mode:

o In half-duplex mode, both the devices can send and receive the data and also can utilize
the entire bandwidth of the communication channel during the transmission of data.

Disadvantage of Half-Duplex mode:

o In half-duplex mode, when one device is sending the data, then another has to wait, this
causes the delay in sending the data at the right time.
FULL-DUPLEX MODE
o In Full duplex mode, the communication is bi-directional, i.e., the data flow in both the
directions.
o Both the stations can send and receive the message simultaneously.
o Full-duplex mode has two simplex channels. One channel has traffic moving in one
direction, and another channel has traffic flowing in the opposite direction.
o The Full-duplex mode is the fastest mode of communication between devices.
o The most common example of the full-duplex mode is a Telephone network. When two
people are communicating with each other by a telephone line, both can talk and listen
at the same time.

Advantage of Full-duplex mode:

o Both the stations can send and receive the data at the same time.

Disadvantage of Full-duplex mode:

o If there is no dedicated path exists between the devices, then the capacity of the
communication channel is divided into two parts.

COMPARISON - SIMPLEX, HALF-DUPLEX AND FULL-DUPLEX MODE

BASIS FOR SIMPLEX HALF-DUPLEX FULL-DUPLEX

COMPARISON MODE MODE MODE

Direction of Communication is Communication is Communicationis

communication unidirectional. bidirectional, but one at bidirectional.
a time.

Send/Receive A device can only send the Both the devices can Both the devicescan
data but cannot receive it or send and receive the send and receive the
it can only receive the data data, but one at a time. data simultaneously.
but cannotsend it.

Example Radio, Keyboard,and Walkie-Talkie Telephone network.

monitor.
LINE CONFIGURATION / LINE CONNECTIVITY
Line configuration refers to the way two or more communication devices attach to a link. A
link is a communications pathway that transfers data from one device to [Link] are two
possible line configurations:
i. Point to Point : Provides a dedicated Communication link between two devices. It is
simple to establish. The most common example for Point-to-Point connection is a
computer connected by telephone line. We can connect the two devices by means of a
pair of wires or using a microwave or satellite link.

ii. MultiPoint : It is also called Multidrop configuration. In this connection two or more
devices share a single [Link] are two kinds of Multipoint Connections.

 Spatial Sharing: If several devices can share the link simultaneously,it is called
Spatially shared line configuration

 Temporal (Time) Sharing: If users must take turns using the link ,then its called
Temporally shared or Time Shared Line Configuration.

NETWORK TOPOLOGY

Two or more devices connect to a link. Two or more links form a topology. Topology
isdefined as
(1) The way in which a network is laid out physically.
(2) The geometric representation of the relationship of all the links and nodes to one-another.
Network topology refers to the arrangement of different elements like nodes, links, and
devices in a computer network. It defines how these components are connected and interact
with each other.
The various types of topologies are :
 Bus
 Ring
 Tree
 Star
 Mesh
 Hybrid.
BUS TOPOLOGY

 Bus topology is a network type in which every computer and network device is
connected to single cable.
 The long single cable acts as a backbone to link all the devices in a network.
 When it has exactly two endpoints, then it is called Linear Bus topology.
 It transmits data only in one direction.

Advantages of Bus Topology Disadvantages of Bus Topology

1. It is cost effective. 1. Cables fails then whole network fails.
2. Cable required is least compared to 2. If network traffic is heavy or nodes are
other network topology. more, the performance of the network
3. Used in small networks. decreases.
4. It is easy to understand. 3. Cable has a limited length.
5. Easy to expand joining two cables 4. It is slower than the ring topology.
together

RING TOPOLOGY

 In a ring topology, each device has a dedicated point-to-point connection with only
the two devices on either side of it.
 A signal is passed along the ring in one direction, from device to device, until it reaches
its destination.
 Each device in the ring incorporates a repeater.
 When a device receives a signal intended for another device, its repeater regenerates
the bits and passes them along.
 Advantages of Ring Topology Disadvantages of Ring Topology
1. Transmitting network is not affected by 1. Troubleshooting is difficult in ring
high traffic or by adding more nodes, topology.
as only the nodes having tokens can 2. Adding or deleting the computers
transmit data. disturbs the network activity.
2. Cheap to install and expand 3. Failure of one computer disturbs the
whole network

TREE TOPOLOGY
 It has a root node and all other nodes are connected to it forming a hierarchy.
 It is also called hierarchical topology.
 It should at least have three levels to the hierarchy.
 Tree topology is ideal if workstations are located in groups.
 They are used in Wide Area Network.

Advantages of Tree Topology Disadvantages of Tree Topology

1. Extension of bus and star topologies. 1. Heavily cabled.
2. Expansion of nodes is possible and easy. 2. Costly.
3. Easily managed and maintained. 3. If more nodes are added maintenance is
4. Error detection is easily done. difficult.
4. Central hub fails, network fails.

STAR TOPOLOGY
 In a star topology, each device has a dedicated point-to-point link only to a
central controller, usually called a hub.
  The devices are not directly linked to one another.
 The controller acts as an exchange.
 If one device wants to send data to another, it sends the data to the controller,
which then relays the data to the other connected device.

Advantages of Star Topology Disadvantages of Star Topology

1. Fast performance with few nodes and 1. Cost of installation is high.
low network traffic. 2. Expensive to use.
2. Hub can be upgraded easily. 3. If the hub fails, then the whole
3. Easy to troubleshoot. network is stopped.
4. Easy to setup and modify. 4. Performance is based on the hub that
5. Only that node is affected which has is it depends on its capacity
failed, rest of the nodes can work
smoothly

MESH TOPOLOGY

 In a mesh topology, every device has a dedicated point-to-point link to everyother

device.
 The term dedicated means that the link carries traffic only between the two
devices it connects.
 The number of physical links in a fully connected mesh network with n nodes is
given by n (n – 1) / 2.
 Advantages of Mesh Topology Disadvantages of Mesh Topology
1. Each connection can carry its own data 1. Installation and configuration is
load. difficult.
2. It is robust. 2. Cabling cost is more.
3. Fault is diagnosed easily. 3. Bulk wiring is required.
4. Provides security and privacy.

HYBRID TOPOLOGY

 Hybrid Topology is a combination of one or more basic topologies.

 For example if one department in an office uses ring topology, the other departments uses
star and bus topology, then connecting these topologies will result in Hybrid Topology.
 Hybrid Topology inherits the advantages and disadvantages of the topologies included.

Advantages of Hybrid Topology Disadvantages of Hybrid Topology

1. Reliable as Error detecting and 1. Complex in design.
troubleshooting is easy. 2. Costly
2. Effective.
3. Scalable as size can be increased easily.
4. Flexible.

NETWORK TYPES
 A computer network is a group of computers linked to each other that enables the
computer to communicate with another computer and share their resources, data, and
applications.
 A computer network can be categorized by their size.
 A computer network is mainly of three types:
1. Local Area Network (LAN)
2. Wide Area Network (WAN)
3. Metropolitan Area Network (MAN)
LOCAL AREA NETWORK (LAN)
o Local Area Network is a group of computers connected to each other in a small area such
as building, office.
o LAN is used for connecting two or more personal computers through a communication
medium such as twisted pair, coaxial cable, etc.
ss

o It is less costly as it is built with inexpensive hardware such as hubs, network

adapters, and ethernet cables.
o The data is transferred at an extremely faster rate in Local Area Network.
o LAN can be connected using a common cable or a Switch.

Advantages of LAN Disadvantages of LAN

 Resource Sharing  High Setup Cost
 Software Applications Sharing.  Privacy Violations
 Easy and Cheap Communication  Data Security Threat
 Centralized Data.  LAN Maintenance Job
 Data Security  Covers Limited Area
 Internet Sharing

WIDE AREA NETWORK (WAN)

o A Wide Area Network is a network that extends over a large geographical area such
as states or countries.
o A Wide Area Network is quite bigger network than the LAN.
 o A Wide Area Network is not limited to a single location, but it spans over a large
geographical area through a telephone line, fibre optic cable or satellite links.
o The internet is one of the biggest WAN in the world.
o A Wide Area Network is widely used in the field of Business, government, and
education.
o WAN can be either a point-to-point WAN or Switched WAN.

Point-to-point WAN Switched WAN

Advantages of Wide Area Network: Disadvantages of Wide Area Network:

o Large Geographical area o Security issue
o Centralized data o Needs Firewall & antivirus software
o Exchange messages o High Setup cost
o Sharing of software and resources o Troubleshooting problems
o High bandwidth

METROPOLITAN AREA NETWORK (MAN)

o A metropolitan area network is a network that covers a larger geographic area by

interconnecting a different LAN to form a larger network.
o It generally covers towns and cities (50 km)
o In MAN, various LANs are connected to each other through a telephone exchange
line.
 o Communication medium used for MAN are optical fibers, cables etc.
o It has a higher range than Local Area Network(LAN).It is adequate for distributed
computing applications.

INTERNETWORK

o An internetwork is defined as two or more computer network LANs or WAN.

o An Internetwork can be formed by joining two or more individual networks by
means of various devices such as routers, gateways and bridges.
o An interconnection between public, private, commercial, industrial, or government
computer networks can also be defined as internetworking.

Types of Internetwork

Extranet Intranet
 An extranet is used for information sharing.  An intranet belongs to an
The access to the extranet is restricted to only organization which is only
those users who have login credentials. accessible by the organization's
 An extranet is the lowest level of employee or members. The main
internetworking. aim of the intranet is to share the
 It can be categorized as MAN, WAN or other information and resources among
computer networks. the organization employees.
 An extranet cannot have a single LAN,  An intranet provides the facility to
atleast it must have one connection to the work in groups and for
external network. teleconferences.
TRANSMISSION MEDIA
o Transmission media is a communication channel that carries the information from the
sender to the receiver.
o Data is transmitted through the electromagnetic signals.
o The main functionality of the transmission media is to carry the information in the form
of bits (Either as Electrical signals or Light pulses).
o It is a physical path between transmitter and receiver in data communication.
o The characteristics and quality of data transmission are determined by the characteristics
of medium and signal.
o Transmission media is of two types: Guided Media (Wired) and UnGuided Media
(wireless).
o In guided (wired) media, medium characteristics are more important whereas, in
unguided (wireless) media, signal characteristics are more important.
o Different transmission media have different properties such as bandwidth, delay,cost and
ease of installation and maintenance.
o The transmission media is available in the lowest layer of the OSI reference model,
i.e., Physical layer.

FACTORS FOR DESIGNING THE TRANSMISSION MEDIA

o Bandwidth: All the factors are remaining constant, the greater the bandwidth of a
medium, the higher the data transmission rate of a signal.
o Transmission impairment: When the received signal is not identical to the transmitted
one due to the transmission impairment. The quality of the signals will get destroyed
due to transmission impairment.
o Interference: An interference is defined as the process of disrupting a signal when it
travels over a communication medium on the addition of some unwanted signal.

CAUSES OF TRANSMISSION IMPAIRMENT

o Attenuation: Attenuation means the loss of energy, i.e., the strength of the signal
decreases with increasing the distance which causes the loss of energy.
 o Distortion: Distortion occurs when there is a change in the shape of the signal. This type
of distortion is examined from different signals having different frequencies. Each
frequency component has its own propagation speed, so they reach at a different time
which leads to the delay distortion.
o Noise: When data is travelled over a transmission medium, some unwanted signal is
added to it which creates the noise.
TYPES / CLASSES OF TRANSMISSION MEDIA

GUIDED MEDIA
 It is defined as the physical medium through which the signals are transmitted.
 It is also known as Bounded media.
 Types of Guided media: Twisted Pair Cable, Coaxial Cable , Fibre Optic Cable
TWISTED PAIR CABLE

 Twisted pair is a physical media made up of a pair of cables twisted with each other.
 A twisted pair cable is cheap as compared to other transmission media.
 Installation of the twisted pair cable is easy, and it is a lightweight cable.
 The frequency range for twisted pair cable is from 0 to 3.5KHz.
 A twisted pair consists of two insulated copper wires arranged in a regular spiral pattern.
Unshielded Twisted Pair
An unshielded twisted pair is widely used in telecommunication. Following are the categories
of the unshielded twisted pair cable:
o Category 1: Suports low-speed data.
o Category 2: It can support upto 4Mbps.

o Category 3: It can support upto 16Mbps.

o Category 4: It can support upto 20Mbps.
o Category 5: It can support upto 200Mbps.

Advantages :
o It is cheap.
o Installation of the unshielded twisted pair is easy.
o It can be used for high-speed LAN.

Disadvantage:
o This cable can only be used for shorter distances because of attenuation.
Shielded Twisted Pair

A shielded twisted pair is a cable that contains the mesh surrounding the wire that allows the
higher transmission rate.

Advantages :
o The cost of the shielded twisted pair cable is not very high and not very low.
o Installation of STP is easy.
o It has higher capacity as compared to unshielded twisted pair cable.
o It is shielded that provides the higher data transmission rate.
Disadvantages:
o It is more expensive as compared to UTP and coaxial cable.
o It has a higher attenuation rate.
COAXIAL CABLE

o Coaxial cable(Coax) is a very commonly used transmission media, for example,TV

wire is usually a coaxial cable.
o The name of the cable is coaxial as it contains two conductors parallel to each other.
o It has a higher frequency as compared to Twisted pair cable.
o The inner conductor of the coaxial cable is made up of copper, and the outer
conductor is made up of copper mesh.
o The middle core is made up of non-conductive cover that separates the inner
conductor from the outer conductor.
o The middle core is responsible for the data transferring whereas the copper mesh
prevents from the EMI (Electromagnetic interference).
o Common applications of coaxial cable are Cable TV networks and traditional Ethernet
LANs.

Coaxial Cable Standards

 Coaxial cables are categorized by their Radio Government (RG) ratings.
 Each RG number denotes a unique set of physical specifications, including the wire
gauge of the inner conductor, the thickness and type of the inner insulator, the
construction of the shield, and the size and type of the outer casing.
 Each cable defined by an RG rating is adapted for a specialized function.

Types of Coaxial cable :

1. Baseband transmission: It is defined as the process of transmitting a single signal at high
speed.
2. Broadband transmission: It is defined as the process of transmitting multiple signals
simultaneously.
Advantages :
o The data can be transmitted at high speed.
o It has better shielding as compared to twisted pair cable.
o It provides higher bandwidth.

Disadvantages :
o It is more expensive as compared to twisted pair cable.
o If any fault occurs in the cable causes the failure in the entire network.

FIBRE OPTIC CABLE

o Fibre optic cable is a cable that uses electrical signals for communication.
o Fibre optic is a cable that holds the optical fibres coated in plastic that are used to send
the data by pulses of light.
o The plastic coating protects the optical fibres from heat, cold, electromagnetic
interference from other types of wiring.
o Fibre optics provide faster data transmission than copper wires.

Basic elements of Fibre optic cable:

o Core: The optical fibre consists of a narrow strand of glass or plastic known as a core.
A core is a light transmission area of the fibre. The more the area of the core, the
more light will be transmitted into the fibre.
o Cladding: The concentric layer of glass is known as cladding. The main functionality of
the cladding is to provide the lower refractive index at the core interface as to cause the
reflection within the core so that the light waves are transmitted through the fibre.
o Jacket: The protective coating consisting of plastic is known as a jacket. The main
purpose of a jacket is to preserve the fibre strength, absorb shock and extra fibre
protection.

Advantages:
o Greater Bandwidth
o Less signal attenuation
o Immunity to electromagnetic interference
o Resistance to corrosive materials
o Light weight
o Greater immunity to tapping
Disadvantages :
o Requires Expertise for Installation and maintenance
o Unidirectional light propagation.
o Higher Cost.
Propagation Modes of Fibre Optics
 Current technology supports two modes (multimode and single mode) for propagating
light along optical channels, each requiring fiber with different physical characteristics.
 Multimode can be implemented in two forms: step-index or graded-index.

Multimode Propagation
 Multimode is so named because multiple beams from a light source move through the
core in different paths.
 How these beams move within the cable depends on the structure of the core.

Multimode Step-index fiber Multimode Graded-index fiber

In multimode step-index fiber, the density  The multimode graded-index fiber,
of the core remains constant from the decreases this distortion of the signal
center to the edges. through the cable.
A beam of light moves through this  The word index here refers to the index
constant density in a straight line until it of refraction.
reaches the interface of the core and the  The index of refraction is related to
cladding. density.
 At the interface, there is an abrupt change  A graded index fiber, therefore, is one
due to a lower density; this alters the angle with varying densities.
of the beam’s motion.  Density is highest at the center of the
The term step-index refers to the
core and decreases gradually to its
suddenness of this change, which
lowest at the edge.
contributes to the distortion of the signal
as it passes through the fiber.
Single-Mode Propagation

 Single-mode uses step-index fiber and a highly focused source of light that limits beams
to a small range of angles, all close to the horizontal.
 The single-mode fiber itself is manufactured with a much smaller diameter than that of
multimode fiber, and with substantially lower density (index of refraction).
 The decrease in density results in a critical angle that is close enough to 90° to make the
propagation of beams almost horizontal.
 In this case, propagation of different beams is almost identical, and delays are negligible.
All the beams arrive at the destination “together” and can be recombined with little
distortion to the signal.

UNGUIDED MEDIA
o An unguided transmission transmits the electromagnetic waves without using any
physical medium. Therefore it is also known as wireless transmission.
o In unguided media, air is the media through which the electromagnetic energy
can flow easily.

o Unguided transmission is broadly classified into three categories :

Radio Waves, Microwaves , Infrared

RADIO WAVES

o Radio waves are the electromagnetic waves that are transmitted in all the directions of
free space.
o Radio waves are omnidirectional, i.e., the signals are propagated in all the directions.
o The range in frequencies of radio waves is from 3Khz to 1Khz.
o In the case of radio waves, the sending and receiving antenna are not aligned, i.e.,the
wave sent by the sending antenna can be received by any receiving antenna.
o An example of the radio wave is FM radio.
Applications of Radio waves:
o A Radio wave is useful for multicasting when there is one sender and many
receivers.
o An FM radio, television, cordless phones are examples of a radio wave.

Advantages of Radio waves:

o Radio transmission is mainly used for wide area networks and mobile cellularphones.
o Radio waves cover a large area, and they can penetrate the walls.
o Radio transmission provides a higher transmission rate.
MICROWAVES

Microwaves are of two types - Terrestrial microwave & Satellite microwave

Terrestrial Microwave
o Terrestrial Microwave transmission is a technology that transmits the focused beam of a
radio signal from one ground-based microwave transmission antenna to another.
o Microwaves are the electromagnetic waves having the frequency in the range from
1GHz to 1000 GHz.
o Microwaves are unidirectional as the sending and receiving antenna is to be aligned, i.e.,
the waves sent by the sending antenna are narrowly focused.
o In this case, antennas are mounted on the towers to send a beam to another antenna which
is km away.
o It works on the line of sight transmission, i.e., the antennas mounted on the towers
are at the direct sight of each other.

Characteristics of Terrestrial Microwave:

Frequency range: The frequency range of terrestrial microwave is from 4-6 GHzto 21-
23 GHz.
Bandwidth: It supports the bandwidth from 1 to 10 Mbps.
 Short distance: It is inexpensive for short distance.
Long distance: It is expensive as it requires a higher tower for a longer distance.
Attenuation: Attenuation means loss of signal. It is affected by environmental
conditions and antenna size.

Advantages of Terrestrial Microwave:

o Microwave transmission is cheaper than using cables.
o It is free from land acquisition as it does not require any land for the installationof
cables.
o Microwave transmission provides an easy communication in terrains as the
installation of cable in terrain is quite a difficult task.
o Communication over oceans can be achieved by using microwave transmission.

Disadvantages of Terrestrial Microwave:

o Eavesdropping.
o Out of phase signal
o Susceptible to weather condition
o Bandwidth limited

Satellite Microwave
o A satellite is a physical object that revolves around the earth at a known height.
o Satellite communication is more reliable nowadays as it offers more flexibility than
cable and fibre optic systems.
o We can communicate with any point on the globe by using satellite communication.
o The satellite accepts the signal that is transmitted from the earth station, and it
amplifies the signal. The amplified signal is retransmitted to another earth station.

Advantages of Satellite Microwave:

o The coverage area of a satellite microwave is more than the terrestrial microwave.
o The transmission cost of the satellite is independent of the distance from the centre
of the coverage area.
 o Satellite communication is used in mobile and wireless communication applications.
o It is easy to install.
o It is used in a wide variety of applications such as weather forecasting, radio/TVsignal
broadcasting, mobile communication, etc.

Disadvantages of Satellite Microwave:

o Satellite designing and development requires more time and higher cost.
o The Satellite needs to be monitored and controlled on regular periods so that it
remains in orbit.
o The life of the satellite is about 12-15 years. Due to this reason, another launch of the
satellite has to be planned before it becomes non-functional.

INFRARED WAVES

o An infrared transmission is a wireless technology used for communication over short

ranges.
o The frequency of the infrared in the range from 300 GHz to 400 THz.
o It is used for short-range communication such as data transfer between two cell phones,
TV remote operation, data transfer between a computer and cell phone and devices that
resides in the same closed area.

Characteristics of Infrared:
o It supports high bandwidth, and hence the data rate will be very high.
o Infrared waves cannot penetrate the walls. Therefore, the infrared communication in one
room cannot be interrupted by the nearby rooms.
o An infrared communication provides better security with minimum interference.
o Infrared communication is unreliable outside the building because the sun rays will
interfere with the infrared waves.
CONNECTING DEVICES
 Connecting devices are used to connect hosts together to make a network or to connect
networks together to make an internet.
 Connecting devices can operate in different layers of the Internet model.

 Connecting devices are divided into five different categories on the basis of layers
in which they operate in the network.
 1. Devices which operate below the physical layer - Passive hub.
2. Devices which operate at the physical layer - Repeater.
3. Devices which operate at the physical and data link layers - Bridge.
4. Devices which operate at the physical layer, data link layer and network layer –Router.
5. Devices which operate at all five layers - Gateway.
1. HUBS
 Several networks need a central location to connect media segments together. These
central locations are called as hubs.
 The hub organizes the cables and transmits incoming signals to the other media
segments.

The three types of hubs are:

i) Passive hub
 It is a connector, which connects wires coming from the different branches.
 By using passive hub, each computer can receive the signal which is sent
 from all other computers connected in the hub.

ii) Active Hub

 It is a multiport repeater, which can regenerate the signal.
 It is used to create connections between two or more stations in a physical
star topology.

iii) Intelligent Hub

 Intelligent hub contains a program of network management and intelligent path
selection.
2. REPEATERS

 A repeater receives the signal and it regenerates the signal in original bit pattern
before the signal gets too weak or corrupted.
 It is used to extend the physical distance of LAN.
 Repeater works on physical layer.
 A repeater has no filtering capability.

 A repeater is implemented in computer networks to expand the coverage area of the

network, repropagate a weak or broken signal and or service remote nodes.
 Repeaters amplify the received/input signal to a higher frequency domain so that it is
reusable, scalable and available.
 Repeaters are also known as signal boosters or range extender.
 A repeater cannot connect two LANs, but it connects two segments of thesame LAN.

[Link]

 Bridges operate in physical layer as well as data link layer.

 As a physical layer device, they regenerate the receive signal.
 As a data link layer, the bridge checks the physical (MAC) address (of thesource and
the destination) contained in the frame.
  The bridge has a filtering feature.
 It can check the destination address of a frame and decides, if the frameshould be
forwarded or dropped.
 Bridges are used to connect two or LANs working on the same protocol.

Types of Bridges :

 Transparent Bridges
These are the bridge in which the stations are completely unaware of the bridge’s
existence i.e. whether or not a bridge is added or deleted from the network ,
reconfiguration of the stations is unnecessary.
 Source Routing Bridges
In these bridges, routing operation is performed by source station and the frame
specifies which route to follow.

 Translation Bridges
These bridges connect networks with different architectures, such as Ethernet and
Token Ring. These bridges appear as:
– Transparent bridges to an Ethernet host
– Source-routing bridges to a Token Ring host
4. SWITCHES
◾ A switch is a small hardware device which is used to join multiple computers
together with one local area network (LAN).
◾ A switch is a mechanism that allows us to interconnect links to form a large network.
◾ Switch is data link layer device.
◾ A switch is a multi port bridge with a buffer .
◾ Switches are used to forward the packets based on MAC addresses.
◾ It is operated in full duplex mode.
◾ Packet collision is minimum as it directly communicates between source and
destination.
 ◾ It does not broadcast the message as it works with limited bandwidth.
◾ A switch’s primary job is to receive incoming packets on one of its links and to
transmit them on some other link.
◾ A Switch is used to transfer the data only to the device that has been addressed.

 Input ports receive stream of packets, analyzes the header, determines the output
port and passes the packet onto the fabric.
 Ports contain buffers to hold packets before it is forwarded.
 If buffer space is unavailable, then packets are dropped.
 If packets at several input ports queue for a single output port, then only one of them
is forwarded.

Types of Switch

i) Two- Layer Switch

 The two-layer switch performs at the physical and the data link layer.
 It is a bridge with many ports and design allows faster performs.
 A bridge is used to connect different LANs together.
 The two- layer switch can make a filtering decision bases on the MAC address of the
received frame. However, two- layer switch has a buffer which holds the frame for
processing.
ii) Three- Layer Switch
 The three-layer switch is a router.
 The switching fabric in a three-layer allows a faster table lookup andforwarding
mechanism.

[Link]
 A router is a three-layer device.
 It operates in the physical, data-link, and network layers.
 As a physical-layer device, it regenerates the signal it receives.
 As a link-layer device, the router checks the physical addresses (source and destination)
contained in the packet.
 As a network-layer device, a router checks the network-layer addresses.
 A router is a device like a switch that routes data packets based on their IPaddresses.
 A router can connect networks. A router connects the LANs and WANs on the internet.
 A router is an internetworking device.
 It connects independent networks to form an internetwork.

 The key function of the router is to determine the shortest path to the
destination.
 Router has a routing table, which is used to make decision on selecting the route.
 The routing table is updated dynamically based on which they makedecisions on routing
the data packets.

[Link]

 A gateway is a device, which operates in all five layers of the internet or seven layers
of OSI model.
 It is usually a combination of hardware and software.
 Gateway connects two independent networks.

 Gateways are generally more complex than switch or router.

  Gateways basically works as the messenger agents that take data from one system,
interpret it, and transfer it to another system.
 Gateways are also called protocol converters
 A gateway accepts a packet formatted for one protocol and converts it to a packet
formatted to another protocol before forwarding it.
 The gateway must adjust the data rate, size and data format.

[Link]
 Brouter is a hybrid device. It combines the features of both bridge androuter.
 Brouter is a combination of Bridge and Router.
 Functions as a bridge for nonroutable protocols and a router for routableprotocols.
 As a router, it is capable of routing packets across networks.
 As a bridge, it is capable of filtering local area network traffic.
 Provides the best attributes of both a bridge and a router
 Operates at both the Data Link and Network layers and can replace separate bridges and
routers.

PROTOCOL LAYERING
 In networking, a protocol defines the rules that both the sender and receiver and all
intermediate devices need to follow to be able to communicate effectively.
 A protocol provides a communication service that the process use to exchange
messages.
 When communication is simple, we may need only one simple protocol.
 When the communication is complex, we may need to divide the task between different
layers, in which case we need a protocol at each layer, or protocol layering.
 Protocol layering is that it allows us to separate the services from the implementation.
 A layer needs to be able to receive a set of services from the lower layer and to give the
services to the upper layer.
 Any modification in one layer will not affect the other layers.

Basic Elements of Layered Architecture

 Service: It is a set of actions that a layer provides to the higher layer.
 Protocol: It defines a set of rules that a layer uses to exchange the information with peer
entity. These rules mainly concern about both the contents and order of the messages
used.
 Interface: It is a way through which the message is transferred from one layer to another
layer.
Features of Protocol Layering
1. It decomposes the problem of building a network into more manageable components.
2. It provides a more modular design.
Principles of Protocol Layering
1. The first principle dictates that if we want bidirectional communication, we need to make
each layer so that it is able to perform two opposite tasks, one in each direction.
2. The second principle that we need to follow in protocol layering is that the two objects
under each layer at both sites should be identical.

Protocol Graph
 The set of protocols that make up a network system is called a protocol graph.
 The nodes of the graph correspond to protocols, and the edges represent a dependence
relation.
 For example, the Figure below illustrates a protocol graph consists of protocols RRP
(Request/Reply Protocol) and MSP (Message Stream Protocol) implement two different
types of process-to-process channels, and both depend on the HHP (Host-to- Host
Protocol) which provides a host-to-host connectivity service

OSI MODEL
o OSI stands for Open System Interconnection.
o It is a reference model that describes how information from a software application in one
computer moves through a physical medium to the software application in another
computer.
o OSI consists of seven layers, and each layer performs a particular network function.
o OSI model was developed by the International Organization for Standardization (ISO)
in 1984, and it is now considered as an architectural model for the inter- computer
communications.
o OSI model divides the whole task into seven smaller and manageable tasks. Each layer
 is assigned a particular task.
o Each layer is self-contained, so that task assigned to each layer can be performed
independently.

ORGANIZATION OF THE OSI LAYERS

 FUNCTIONS OF THE OSI LAYERS
1. PHYSICAL LAYER

The physical layer coordinates the functions required to transmit a bit stream over a physical
medium.
The physical layer is concerned with the following functions:
 Physical characteristics of interfaces and media - The physical layer defines the
characteristics of the interface between the devices and the transmission medium.
 Representation of bits - To transmit the stream of bits, it must be encoded to signals. The
physical layer defines the type of encoding.
 Signals: It determines the type of the signal used for transmitting the information.
 Data Rate or Transmission rate - The number of bits sent each second –is also defined
by the physical layer.
 Synchronization of bits - The sender and receiver must be synchronized at the bit
level. Their clocks must be synchronized.
 Line Configuration - In a point-to-point configuration, two devices are connected
together through a dedicated link. In a multipoint configuration, a link is shared between
several devices.
 Physical Topology - The physical topology defines how devices are connected to make a
network. Devices can be connected using a mesh, bus, star or ring topology.
 Transmission Mode - The physical layer also defines the direction of transmission
between two devices: simplex, half-duplex or full-duplex.
2. DATA LINK LAYER

It is responsible for transmitting frames from one node to the next node. The other
responsibilities of this layer are
 Framing - Divides the stream of bits received into data units called frames.
 Physical addressing – If frames are to be distributed to different systems on the network
, data link layer adds a header to the frame to define the sender and receiver.
 Flow control- If the rate at which the data are absorbed by the receiver is lessthan the
rate produced in the sender ,the Data link layer imposes a flow ctrl mechanism.
 Error control- Used for detecting and retransmitting damaged or lost frames and to
prevent duplication of frames. This is achieved through a trailer added at the end of the
frame.
 Medium Access control -Used to determine which device has control over the link at any
given time.

3. NETWORK LAYER

This layer is responsible for the delivery of packets from source to destination.
It determines the best path to move data from source to the destination based on the
network conditions, the priority of service, and other factors.
The other responsibilities of this layer are
 Logical addressing - If a packet passes the network boundary, we need another
addressing system for source and destination called logical address. This addressing is
used to identify the device on the internet.
 Routing – Routing is the major component of the network layer, and it determines the
best optimal path out of the multiple paths from source to the destination.

4. TRANSPORT LAYER

It is responsible for Process to Process delivery. That is responsible for source-to- destination
(end-to-end) delivery of the entire message, It also ensures whether the message arrives in
order or not.
The other responsibilities of this layer are
 Port addressing / Service Point addressing - The header includes an address called port
address / service point address. This layer gets the entire message to the correct process
on that computer.
 Segmentation and reassembly - The message is divided into segments and each segment
is assigned a sequence number. These numbers are arranged correctly onthe arrival side
by this layer.
 Connection control - This can either be connectionless or connection oriented.
 The connectionless treats each segment as an individual packet anddelivers to the
destination.
 The connection-oriented makes connection on the destination side before the
 delivery. After the delivery the termination will be terminated.
 Flow control -The transport layer also responsible for flow control but it is performed
end-to-end rather than across a single link.
 Error Control - Error control is performed end-to-end rather than across thesingle
link.

5. SESSION LAYER

This layer establishes, manages and terminates connections between applications. The
other responsibilities of this layer are
 Dialog control - Session layer acts as a dialog controller that creates a dialog between
two processes or we can say that it allows the communication between two processes
which can be either half-duplex or full-duplex.
 Synchronization- Session layer adds some checkpoints when transmitting the data in a
sequence. If some error occurs in the middle of the transmission of data, then the
transmission will take place again from the checkpoint. This process is known as
Synchronization and recovery.

6. PRESENTATION LAYER

It is concerned with the syntax and semantics of information exchanged between two
systems.
The other responsibilities of this layer are
 Translation – Different computers use different encoding system, this layer is
responsible for interoperability between these different encoding methods. It will change
the message into some common format.
 Encryption and decryption-It means that sender transforms the original information to
another form and sends the resulting message over the n/w. and vice versa.
 Compression and expansion-Compression reduces the number of bits contained in the
information particularly in text, audio and video.

7. APPLICATION LAYER
This layer enables the user to access the network. It handles issues such as network
transparency, resource allocation, etc. This allows the user to log on to remote user.
The other responsibilities of this layer are
 FTAM (File Transfer, Access, Management) - Allows user to access files in aremote
host.
 Mail services - Provides email forwarding and storage.
 Directory services - Provides database sources to access information about various
sources and objects.
Internet Architecture
TCP/IP PROTOCOL SUITE
The TCP/IP architecture is also called as Internet architecture. It is developed by the US
Defense Advanced Research Project Agency (DARPA)for its packet switched network
(ARPANET). TCP/IP is a protocol suite used in the Internet today. It is a 4-layer model. The
layers of TCP/IP are

APPLICATION LAYER
 An application layer incorporates the function of top three OSI layers. An application
layer is the topmost layer in the TCP/IP model.
 It is responsible for handling high-level protocols, issues of representation.
 This layer allows the user to interact with the application.
 When one application layer protocol wants to communicate with another application
 layer, it forwards its data to the transport layer.
 Protocols such as FTP, HTTP, SMTP, POP3, etc running in the application layer
provides service to other program running on top of application layer.
TRANSPORT LAYER
 The transport layer is responsible for the reliability, flow control, and correction of data
which is being sent over the network.
 The two protocols used in the transport layer are User Datagram protocol and
Transmission control protocol.
o UDP – UDP provides connectionless service and end-to-end delivery of
transmission. It is an unreliable protocol as it discovers the errors but not specify
the error.
o TCP – TCP provides a full transport layer services to applications. TCP is a
reliable protocol as it detects the error and retransmits the damaged frames.
INTERNET LAYER
 The internet layer is the second layer of the TCP/IP model.
 An internet layer is also known as the network layer.
 The main responsibility of the internet layer is to send the packets from any network,
and they arrive at the destination irrespective of the route they take.
 Internet layer handle the transfer of information across multiple networks through router
and gateway.
 IP protocol is used in this layer, and it is the most significant part of the entire TCP/IP
suite.
NETWORK INTERFACE LAYER
 The network interface layer is the lowest layer of the TCP/IP model.
 This layer is the combination of the Physical layer and Data Link layer defined in the
OSI reference model.
 It defines how the data should be sent physically through the network.
 This layer is mainly responsible for the transmission of the data between two devices
on the same network.
 The functions carried out by this layer are encapsulating the IP datagram into frames
transmitted by the network and mapping of IP addresses into physical addresses.
 The protocols used by this layer are Ethernet, token ring, FDDI, X.25, frame relay.
COMPARISON - OSI MODEL AND TCP/IP MODEL

NETWORK SOFTWARE
When implementing a network application is the interface exported by the network.
Since most network protocols are implemented in software, and nearly all computer systems
implement their network protocols as part of the operating system, when we refer to the
interface “exported by the network,” that the OS provides to its networking subsystem. This
interface is often called the network application programming interface (API). Although each
operating system is free to define its own network API. Socket Interface provided by Berkeley
distribution of unix.
Introduction to Sockets
A socket is one endpoint of a two way communication link between two programs
running on the network. The socket mechanism provides a means of inter-process
communication (IPC) by establishing named contact points between which the communication
take place.
Like ‘Pipe’ is used to create pipes and sockets is created using ‘socket’ system call.
The socket provides bidirectional FIFO Communication facility over the network. A socket
connecting to the network is created at each end of the communication. Each socket has a
specific address. This address is composed of an IP address and a port number.
Socket are generally employed in client server applications. The server creates a
socket, attaches it to a network port addresses then waits for the client to contact it. The client
creates a socket and then attempts to connect to the server socket. When the connection is
established, transfer of data takes place.
The first step is to create a socket, which is done with the following operation:
int socket(int domain, int type, int protocol)
Where:
 domain—parameter specifies the protocol family
o i.e., PF-INET :- Denotes the internet family
o PF-UNIX:- Denotes the UNIX pipe facility
o PF-PACKET:- Denotes direct access to the network interface
 type: the type argument indicates the symounties of the communication
o SOCK-STREAM—Used to denote a byte stream.
o SOCK-DGRAM—denotes the message oriented services provides by UDP.
 Protocol: This argument identifies the specific protocol that is going to be used.
o Ex: UNSPEC—Combination of protocol family internet and socket stream which
implies TCP.
The next step depends on whether you are a client or a server. On a server machine,
the application process performs a passive open—the server says that it is prepared to accept
connections, but it does not actually establish a connection.
The server does this by invoking the following three operations:
int bind(int socket, struct sockaddr *address, int addr len)
int listen(int socket, int backlog)
int accept(int socket, struct sockaddr *address, int *addr len)
The bind operation, as its name suggests, binds the newly created socket to the
specified address. This is the network address of the local participant—the server. When we
use it with the internet protocol, the address is a data structure that includes both the IP address
of the server and a ASP port number. Ex: web serves commonly accept connections on port.
The listen operation then defines how many connections can be pending on the
specified socket.
Finally, the accept operation carries out the passive open. It is a blocking operation
that does not return until a remote participant has established a connection, and when it does
complete it returns a new socket that corresponds to this just-established connection, and the
address argument contains the remote participant’s address.
On the client machine, the application process performs an active open; that is, it
says who it wants to communicate with by invoking the following single operation:
int connect(int socket, struct sockaddr *address, int addr len)
This operation does not return until TCP has successfully established a connection,
at which time the application is free to begin sending data. In this case, address contains the
remote participant’s address. In practice, the client usually specifies only the remote
participant’s address and lets the system fill in the local information. Whereas a server usually
listens for messages on a well-known port, a client typically does not care which port it uses
for itself; the OS simply selects an unused one.
Once a connection is established, the application processes invoke the following two
operations to send and receive data:
int send(int socket, char *message, int msg len, int flags)
int recv(int socket, char *buffer, int buf len, int flags)
The first operation sends the given message over the specified socket, while the
second operation receives a message from the specified socket into the given buffer. Both
operations take a set of flags that control certain details of the operation.
Example Application
CLIENT
#include <stdio.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <netdb.h>
#define SERVER_PORT 5432
#define MAX_LINE 256
int main(int argc, char * argv[]) {
FILE *fp;
struct hostent *hp;
struct sockaddr_in sin;
char *host;
char buf[MAX_LINE];
int s;
int len;
if (argc==2) {
host = argv[1];
}
else {
fprintf(stderr, "usage: simplex-talk host\n");
exit(1);
}
/* translate host name into peer's IP address */
hp = gethostbyname(host);
if (!hp) {
fprintf(stderr, "simplex-talk: unknown host: %s\n", host);
exit(1);
}
/* build address data structure */
bzero((char *)&sin, sizeof(sin));
sin.sin_family = AF_INET;
bcopy(hp->h_addr, (char *)&sin.sin_addr, hp->h_length);
sin.sin_port = htons(SERVER_PORT);
/* active open */
if ((s = socket(PF_INET, SOCK_STREAM, 0)) < 0) {
perror("simplex-talk: socket");
exit(1);
}
if (connect(s, (struct sockaddr *)&sin, sizeof(sin)) < 0){
perror("simplex-talk: connect");
close(s);
exit(1);
}
/* main loop: get and send lines of text */
while (fgets(buf, sizeof(buf), stdin)) {
buf[MAX_LINE-1] = '\0';
len = strlen(buf) + 1;
send(s, buf, len, 0);
}}

SERVER
#include <stdio.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <netdb.h>
#define SERVER_PORT 5432
#define MAX_PENDING 5
#define MAX_LINE 256
int
main()
{
struct sockaddr_in sin;
char buf[MAX_LINE];
int len;
int s, new_s;
/* build address data structure */
bzero((char *)&sin, sizeof(sin));
sin.sin_family = AF_INET;
sin.sin_addr.s_addr = INADDR_ANY;
sin.sin_port = htons(SERVER_PORT);
/* setup passive open */
if ((s = socket(PF_INET, SOCK_STREAM, 0)) < 0) {
perror("simplex-talk: socket");
exit(1);
}
if ((bind(s, (struct sockaddr *)&sin, sizeof(sin))) < 0) {
perror("simplex-talk: bind");
exit(1);
}
listen(s, MAX_PENDING);
/* wait for connection, then receive and print text */
while(1) {
if ((new_s = accept(s, (struct sockaddr *)&sin, &len)) < 0) {
perror("simplex-talk: accept");
exit(1);
}
while (len = recv(new_s, buf, sizeof(buf), 0))
fputs(buf, stdout);
close(new_s);
}}
NETWORK PERFORMANCE
Network performance is measured in using:
o Bandwidth
o Throughput
o Latency
o Jitter
o RoundTrip Time
BANDWIDTH
 The bandwidth of a network is given by the number of bits that can be transmitted
over the network in a certain period of time.
 Bandwidth can be measured in two different values: bandwidth in hertz and
bandwidth in bits per second.
Bandwidth in Hertz
o Bandwidth in hertz refers to the range of frequencies contained in a composite signal
 or the range of frequencies a channel can pass.
o For example, we can say the bandwidth of a subscriber telephone line is 4 kHz.

Bandwidth in Bits per Seconds

o Bandwidth in Bits per Seconds refers to the number of bits transmitted persecond.
o For example, the bandwidth of a network is a maximum of 100 Mbps. This means that
this network can send 100 Mbps.

Relationship
o There is an explicit relationship between the bandwidth in hertz and bandwidth inbits
per second.
o Basically, an increase in bandwidth in hertz means an increase in bandwidthin bits per
second.
THROUGHPUT
 Throughput is a measure of how fast we can actually send data through a network.
 Bandwidth in bits per second and throughput may seem to be same, but they are
different.
 A link may have a bandwidth of B bps, but we can only send T bps through this link. (T
always less than B).
 In other words, the bandwidth is a potential measurement of a link; the throughput is an
actual measurement of how fast we can send data.
 For example, we may have a link with a bandwidth of 1 Mbps, but the devices connected
to the end of the link may handle only 200 kbps. This means that we cannot send more
than 200 kbps through this link.

Problem :
A network with bandwidth of 10 Mbps can pass only an average of 12,000 frames per minute
with each frame carrying an average of 10,000 bits. What is the throughput of this network?
Solution
We can calculate the throughput as

The throughput is almost one-fifth of the bandwidth in this case.

LATENCY (DELAY)
 The latency or delay defines how long it takes for an entire message to travel from one
end of a network to the other.
 Latency is made up of four components: Propagation time, Transmission time, Queuing
time and Processing delay.




Propagation Time
o Propagation time measures the time required for a bit to travel from the source tothe
destination.
o The propagation time is calculated by dividing the distance by the propagationspeed.
o The propagation speed of electromagnetic signals depends on the medium and onthe
frequency of the signal.

Transmission Time
o In data communications we don’t send just 1 bit, we send a message.
o The first bit may take a time equal to the propagation time to reach its destination.
o The last bit also may take the same amount of time.
o However, there is a time between the first bit leaving the sender and the last bit arriving
at the receiver.
o The first bit leaves earlier and arrives earlier.
o The last bit leaves later and arrives later.
o The transmission time of a message depends on the size of the message and the
bandwidth of the channel.

Queuing Time
o Queuing time is the time needed for each intermediate or end device to hold themessage
before it can be processed.
o The queuing time is not a fixed factor. It changes with the load imposed on thenetwork.
When there is heavy traffic on the network, the queuing time increases.
o An intermediate device, such as a router, queues the arrived messages andprocesses
them one by one.
o If there are many messages, each message will have to wait.
Processing Delay
o Processing delay is the time that the nodes take to process the packet header.
o Processing delay is a key component in network delay.
o During processing of a packet, nodes may check for bit-level errors in the packet that
occurred during transmission as well as determining where the packet's next destination
is.
Bandwidth - Delay Product
o Bandwidth and delay are two performance metrics of a link.
o The bandwidth-delay product defines the number of bits that can fill thelink.
o This measurement is important if we need to send data in bursts and wait for the
acknowledgment of each burst before sending the next one.

JITTER

o Another performance issue that is related to delay is jitter.

o Jitter is a problem that if different packets of data encounter different delays and the
application using the data at the receiver site is time-sensitive (audio andvideo data,
for example).
o If the delay for the first packet is 20 ms, for the second is 45 ms, and for the third is 40
ms, then the real-time application that uses the packets endures jitter.

ROUND-TRIP TIME (RTT)

RTT refers to how long it takes to send a message from one end of a network to the
other and back, rather than the one-way latency. This is called as round-trip time (RTT) of
the network.

DLC SERVICES
The data link control (DLC) deals with procedures for communication between two
adjacent nodes—node-to-node communication—no matter whether the link is dedicated or
broadcast.
 Data link control service include
(1) Framing (2) Flow Control (3) Error Control

FRAMING
 The data-link layer packs the bits of a message into frames, so that eachframe is
 distinguishable from another.

 Although the whole message could be packed in one frame, that is notnormally
done.
 One reason is that a frame can be very large, making flow and error control very
inefficient.
 When a message is carried in one very large frame, even a single-bit errorwould
require the retransmission of the whole frame.
 When a message is divided into smaller frames, a single-bit error affectsonly that
small frame.
 Framing in the data-link layer separates a message from one source to adestination
by adding a sender address and a destination address.
 The destination address defines where the packet is to go; the senderaddress helps the
recipient acknowledge the receipt.
Frame Size
 Frames can be of fixed or variable size.
 Frames of fixed size are called cells. In fixed-size framing, there is no need for defining
the boundaries of the frames; the size itself can be used as a delimiter.
 In variable-size framing, we need a way to define the end of one frame and the beginning
of the next. Two approaches were used for this purpose: a character-oriented approach
and a bit-oriented approach.

Character-Oriented Framing
 In character-oriented (or byte-oriented) framing, data to be carried are 8-bitcharacters.
 To separate one frame from the next, an 8-bit (1-byte) flag is added at thebeginning
and the end of a frame.
 The flag, composed of protocol-dependent special characters, signals thestart or end
of a frame.

 Any character used for the flag could also be part of the information.
 If this happens, when it encounters this pattern in the middle of the data,thereceiver
thinks it has reached the end of the frame.
 To fix this problem, a byte-stuffing strategy was added to character-oriented framing.
Byte Stuffing (or) Character Stuffing
 Byte stuffing is the process of adding one extra byte whenever there is a flag or escape
character in the text.
 In byte stuffing, a special byte is added to the data section of the frame when there
is a character with the same pattern as the flag.
 The data section is stuffed with an extra byte. This byte is usually called the escape
character (ESC) and has a predefined bit pattern.
 Whenever the receiver encounters the ESC character, it removes it from the data section
and treats the next character as data, not as a delimiting flag.

Bit-Oriented Framing
 In bit-oriented framing, the data section of a frame is a sequence of bits tobe interpreted
by the upper layer as text, graphic, audio, video, and so on.
 In addition to headers and trailers), we still need a delimiter to separate oneframe from
the other.
 Most protocols use a special 8-bit pattern flag, 01111110, as the delimiter todefine the
beginning and the end of the frame

 If the flag pattern appears in the data, the receiver must be informed thatthis is not the
end of the frame.
 This is done by stuffing 1 single bit (instead of 1 byte) to prevent the patternfrom
looking like a flag. The strategy is called bit stuffing.

Bit Stuffing
  Bit stuffing is the process of adding one extra 0 whenever five consecutive 1s follow a
0 in the data, so that the receiver does not mistake the pattern 0111110 for a flag.
 In bit stuffing, if a 0 and five consecutive 1 bits are encountered, an extra 0 is added.
 This extra stuffed bit is eventually removed from the data by the receiver.
 The extra bit is added after one 0 followed by five 1’s regardless of the
value of the next bit.
 This guarantees that the flag field sequence does not inadvertently appear inthe frame.

FLOW CONTROL
o Flow control refers to a set of procedures used to restrict the amountof data that
the sender can send before waiting for acknowledgment.
o The receiving device has limited speed and limited memory to store thedata.
o Therefore, the receiving device must be able to inform the sending device to stop the
transmission temporarily before the limits are reached.
o It requires a buffer, a block of memory for storing the information until they are
processed.

Two methods have been developed to control the flow of data:

o Stop-and-Wait
o Sliding Window
STOP-AND-WAIT
o The simplest scheme is the stop-and-wait algorithm.
o In the Stop-and-wait method, the sender waits for an acknowledgementafter every
frame it sends.
o When acknowledgement is received, then only next frame is sent.
o The process of alternately sending and waiting of a frame continues untilthe sender
transmits the EOT (End of transmission) frame.

o If the acknowledgement is not received within the allotted time, then the sender
assumes that the frame is lost during the transmission, so it will retransmit the frame.
o The acknowledgement may not arrive because of the following three scenarios :
1. Original frame is lost
2. ACK is lost
3. ACK arrives after the timeout

Advantage of Stop-and-wait
o The Stop-and-wait method is simple as each frame is checked and acknowledged before
the next frame is sent

Disadvantages of Stop-And-Wait
o In stop-and-wait, at any point in time, there is only one frame that is sentand
waiting to be acknowledged.
o This is not a good use of transmission medium.
 o To improve efficiency, multiple frames should be in transition whilewaiting for
ACK.

PIGGYBACKING

A method to combine a data frame

with ACK.
Piggybacking saves bandwidth
Station A and B both have data to
send.
Instead of sending separately,
station A sends a data frame that
includes an ACK.
Station B does the same thing.

SLIDING WINDOW
o The Sliding Window is a method of flow control in which a sender can transmit the
several frames before getting an acknowledgement.
o In Sliding Window Control, multiple frames can be sent one after theanother due to which
capacity of the communication channel can be utilizedefficiently.
o A single ACK acknowledge multiple frames.
o Sliding Window refers to imaginary boxes at both the sender and receiver end.
o The window can hold the frames at either end, and it provides the upper limit on the
number of frames that can be transmitted before the acknowledgement.
o Frames can be acknowledged even when the window is not completely filled.
o The window has a specific size in which they are numbered as modulo-n means that they
are numbered from 0 to n-1.
o For example, if n = 8, the frames are numbered from
0,1,2,3,4,5,6,7,0,1,2,3,4,5,6,7,0,1........
o The size of the window is represented as n-1. Therefore, maximum n-1frames can
be sent before acknowledgement.
o When the receiver sends the ACK, it includes the number of the next framethat it wants
to receive.
o For example, to acknowledge the string of frames ending with framenumber 4, the
receiver will send the ACK containing the number 5.
o When the sender sees the ACK with the number 5, it got to know that theframes from
0 through 4 have been received.
 Sender Window Receiver Window

o At the beginning of a transmission, the o At the beginning of transmission, the receiver

sender window contains n-1 frames. window does not contain n frames, but it
o When a frame is sent, the size of the window contains n-1 spaces for frames.
shrinks. o When the new frame arrives, the size of the
o For example, if the size of the windowis ‘w’ window shrinks.
and if three frames are sent out, then the o For example, the size of the window is w and if
number of frames left out in the sender three frames are received then the number of
window is w-3. spaces available in the windowis (w-3).
o Once the ACK has arrived, then the sender o Once the acknowledgement is sent, the receiver
window expands to the numberwhich will be window expands by the number equal to the
equal to the number of frames acknowledged number of frames acknowledged.
by ACK.

Example of Sliding Window

3. ERROR CONTROL

Data can be corrupted during transmission. For reliable communication, errors must be
detected and corrected. Error Control is a technique of error detection and retransmission.
TYPES OF ERRORS
SINGLE-BIT ERROR
The term Single-bit error means that only one bit of a given data unit (such as byte,character,
data unit or packet) is changed from 1 to 0 or from 0 to 1.

BURST ERROR
The term Burst Error means that two or more bits in the data unit have changedfrom 1 to
0 or from 0 to 1.

ERROR DETECTION TECHNIQUES / METHODS

The basic idea behind any error detection scheme is to add additional informationto a frame
that can be used to determine if errors have been introduced.

PARITY CHECK
 One bit, called parity bit is added to every data unit so that the total number
of 1’s in the data unit becomes even (or) odd.
 The source then transmits this data via a link, and bits are checked andverified at
the destination.
 Data is considered accurate if the number of bits (even or odd) matches thenumber
transmitted from the source.
 This techniques is the most common and least complex method.
 1. Even parity – Maintain even number of 1sE.g., 1011 → 1011 1
2. Odd parity – Maintain odd number of 1s
E.g., 1011 → 1011 0

CYCLIC REDUNDANCY CHECK

 Cyclic codes refers to encoding messages by adding a fixed-length checkvalue.
 CRCs are popular because they are simple to implement, easy to analyze mathematically
and particularly good at detecting common errors caused in transmission channels.

Steps Involved :
 Consider the original message (dataword) as M(x) consisting of ‘k’ bits and
the divisor as C(x) consists of ‘n+1’ bits.
 The original message M(x) is appended by ‘n’ bits of zero’s. Let us call
this zero-extended message as T(x).
 Divide T(x) by C(x) and find the remainder.
 The division operation is performed using XOR operation.
 The resultant remainder is appended to the original message M(x) as CRCand sent by
the sender(codeword).

Example 1:
 Consider the Dataword / Message M(x) = 1001
 Divisor C(x) = 1011 (n+1=4)
 Appending ‘n’ zeros to the original Message M(x).
 The resultant messages is called T(x) = 1001 000. (here n=3)
 Divide T(x) by the divisor C(x) using XOR operation.
Sender Side :

Receiver Side:
(For Both Case – Without Error and With Error)

Polynomials
 A pattern of 0s and 1s can be represented as a polynomial with coefficientsof 0 and 1.
 The power of each term shows the position of the bit; the coefficient showsthe value of
the bit.
INTERNET CHECKSUM

 Checksum is a calculated value that is used to determine the integrity ofdata.

Example : Let the message to be transmitted be 7,11,12,0,6.

ERROR CONTROL

 Error control includes both error detection and error correction.

 Whenever an error is detected, specified frames are retransmitted
 It allows the receiver to inform the sender if a frame is lost or damaged during
transmission and coordinates the retransmission of those frames by the sender.
 Includes the following actions:
o Error detection
o Positive Acknowledgement (ACK): if the frame arrived with noerrors
o Negative Acknowledgement (NAK): if the frame arrived with errors
o Retransmissions after Timeout: Frame is retransmitted after certainamount of time
if no acknowledgement was received
 Error control in the data link layer is based on automatic repeat request(ARQ).

Categories of Error Control

STOP-AND-WAIT ARQ
 Stop-and-wait ARQ is a technique used to retransmit the data in case of damaged or lost
frames.
 This technique works on the principle that the sender will not transmit the next frame
 until it receives the acknowledgement of the last transmitted frame.

Two possibilities of the retransmission in Stop and Wait ARQ:

o Damaged Frame: When the receiver receives a damaged frame(i.e., the frame contains
an error), then it returns the NAK frame. For example, whenthe frame DATA 1 is sent, and
then the receiver sends the ACK 0 frame means that the data 1 has arrived correctly. The sender
transmits the next frame: DATA 0. It reaches undamaged, and the receiver returns ACK 1.
The sender transmits the third frame: DATA 1. The receiver reports an error and returns the
NAK frame. The sender retransmits the DATA 1 frame.

o Lost Frame: Sender is equipped with the timer and starts when the frame istransmitted.
Sometimes the frame has not arrived at the receiving end so that it cannot be acknowledged
either positively or negatively. The sender waits for acknowledgement until the timer goes off.
If the timer goes off, it retransmits the last transmitted frame.
SLIDING WINDOW ARQ

Sliding Window ARQ is a technique used for continuous transmission error control.

Two protocols used in sliding window ARQ:

1. GO-BACK-N ARQ
o In Go-Back-N ARQ protocol, if one frame is lost or damaged, then itretransmits
all the frames after which it does not receive the positive ACK.

o In the above figure, three frames (Data 0,1,2) have been transmitted beforean error
discovered in the third frame.
o The receiver discovers the error in Data 2 frame, so it returns the NAK 2frame.
o All the frames including the damaged frame (Data 2,3,4) are discarded as itis transmitted
after the damaged frame.
o Therefore, the sender retransmits the frames (Data2,3,4).

2. SELECTIVE-REJECT(REPEAT) ARQ

o Selective-Reject ARQ technique is more efficient than Go-Back-n ARQ.

o In this technique, only those frames are retransmitted for which negative
acknowledgement (NAK) has been received.
o The receiver storage buffer keeps all the damaged frames on hold until theframe in error
is correctly received.
o The receiver must have an appropriate logic for reinserting the frames in acorrect order.
o The sender must consist of a searching mechanism that selects only therequested
frame for retransmission.
o In the above figure, three frames (Data 0,1,2) have been transmitted beforean error
discovered in the third frame.
o The receiver discovers the error in Data 2 frame, so it returns the NAK 2frame.
o The damaged frame only (Data 2) is discarded.
o The other subsequent frames (Data 3,4) are accepted.
Therefore, the sender retransmits only the damaged frame (Data2).
 Unit – 2 Internetworking and Media Access Control
Media Access Control – Protocol Formats-Ethernet (802.3) – Wireless LANs
– 802.11 – Bluetooth – Switching and Bridging – Basic Internetworking-
Protocols-IP, CIDR, ARP, RARP, DHCP and ICMP-RFC for IP and ARP.

1. MEDIA ACCESS CONTROL (MAC)

 When two or more nodes transmit data at the same time, their frames will collide and
the link bandwidth is wasted during collision.
 To coordinate the access of multiple sending/receiving nodes to the shared link, we need
a protocol to coordinate the transmission.
 These protocols are called Medium or Multiple Access Control (MAC)Protocols.
MAC belongs to the data link layer of OSI model
 MAC defines rules for orderly access to the shared medium. It tries to ensure that no two
nodes are interfering with each other’s transmissions, and deals with the situation when
they do.

Issues involved in MAC

The key issues involved are –
 Where the control is exercised - refers to whether the control is exercisedin a centralized
or distributed manner
 How the control is exercised - refers to in what manner the control isexercised
Goals of MAC
1. Fairness in sharing
2. Efficient sharing of bandwidth
3. Need to avoid packet collisions at the receiver due to interference
MAC Management
 Medium allocation (collision avoidance)
 Contention resolution (collision handling)
MAC Types
 Round-Robin : – Each station is given opportunity to transmit in [Link] a central
controller polls a station to permit to go, or stations can coordinate among themselves.
 Reservation : - Station wishing to transmit makes reservations for timeslots in advance.
(Centralized or distributed).
 Contention (Random Access) : - No control on who tries; If collision occurs,
retransmission takes place.
MECHANISMS USED
 Wired Networks :
o CSMA / CD – Carrier Sense Multiple Access / Collision Detection
 Wireless Networks :
o CSMA / CA – Carrier Sense Multiple Access / Collision Avoidance
CARRIER SENSE MULTIPLE ACCESS / COLLISION DETECTION(CSMA / CD)
 Carrier Sense in CSMA/CD means that all the nodes sense the medium to check
whether it is idle or busy.
 If the carrier sensed is idle, then the node transmits the entire frame.
 If the carrier sensed is busy, the transmission is postponed.
 Collision Detect means that a node listens as it transmits and can therefore detect when
a frame it is transmitting has collided with a frame transmitted by another node.
Flowchart of CSMA/CD Operation

Transmitter Algorithm in CSMA/CD

 Transmitter Algorithm defines the procedures for a node that senses a busy medium.
 Three types of Transmitter Algorithm exist.
  They are
1. Non-Persistent Strategy
2. Persistent Strategy: 1-Persistent & P-Persistent
Non-Persistent Strategy
 In the non-persistent method, a station that has a frame to send senses the \
line.
 If the line is idle, it sends immediately.
 If the line is not idle, it waits a random amount of time and then senses the line again.

The non-persistent approach reduces the chance of collision because it is unlikely that two or
more stations will wait the same amount of time and retry to send simultaneously.
 However, this method reduces the efficiency of the network because the medium
remains idle when there may be stations with frames to send.
Persistent Strategy
1- Persistent :
 The 1-persistent method is simple and straightforward.
 In this method, after the station finds the line idle, it sends its frameimmediately (with
probability 1).
 This method has the highest chance of collision because two or more stations may find
the line idle and send their frames immediately.

P-Persistent :
 In this method, after the station finds the line idle it follows these steps:
 With probability p, the station sends its frame.
 With probability q = 1 − p, the station waits for the beginning of the next time slot
 and checks the line again.
 The p-persistent method is used if the channel has time slots with a slot duration equal
to or greater than the maximum propagation time.

 The p-persistent approach combines the advantages of the other two strategies. It
reduces the chance of collision and improves efficiency.
EXPONENTIAL BACK-OFF
 Once an adaptor has detected a collision and stopped its transmission, it waits a certain
amount of time and tries again.
 Each time it tries to transmit but fails, the adaptor doubles the amount of time it waits
before trying again.
 This strategy of doubling the delay interval between each retransmission attempt is a
general technique known as exponential back-off.
CARRIER SENSE MULTIPLE ACCESS / COLLISION AVOIDANCE(CSMA/CA)
 Carrier sense multiple access with collision avoidance (CSMA/CA) was invented for
wireless networks.
 Wireless protocol would follow exactly the same algorithm as the Ethernet—Wait until
the link becomes idle before transmitting and back off should a collision occur.
 Collisions are avoided through the use of CSMA/CA’s three strategies: the interframe
space, the contention window, and acknowledgments

Interframe Space (IFS) - First, collisions are avoided by deferring transmission even if the
channel is found idle. When an idle channel is found, the station does not send immediately. It
waits for a period of time called the interframe space or IFS.

Contention Window - The contention window is an amount of time divided into slots. A
station that is ready to send chooses a random number of slots as its wait time. The number of
slots in the window changes according to the binary exponential backoff strategy. This means
that it is set to one slot the first time and then doubles each time the station cannot detect an
idle channel after the IFS time.

Acknowledgment - In addition, the data may be corrupted during the transmission. The
positive acknowledgment and the time-out timer can help guarantee that the receiver has
received the frame.

2. WIRED LAN : ETHERNET (IEEE 802.3)

 Ethernet was developed in the mid-1970’s at the Xerox Palo Alto Research Center (PARC),
 IEEE controls the Ethernet standards.
 The Ethernet is the most successful local area networking technology, that uses bus
topology.
 The Ethernet is multiple-access networks that is set of nodes send and receive frames
over a shared link.
 Ethernet uses the CSMA / CD ( Carrier Sense Multiple Access with Collision
Detection) mechanism.

EVOLUTION OF ETHERNET
 Standard Ethernet (10 Mbps)
The original Ethernet technology with the data rate of 10 Mbps as the Standard Ethernet.
Standard Ethernet types are
1. 10Base5: Thick Ethernet,
2. 10Base2: Thin Ethernet ,
3. 10Base-T: Twisted-Pair Ethernet
4. 10Base-F: Fiber Ethernet.

10Base5: Thick Ethernet 10Base2: Thin Ethernet

 The second implementation is called

 The first implementation is called 10Base5, 10Base2, thin Ethernet, or Cheapernet.
thick Ethernet, orThicknet.  10Base2 also uses a bus topology, but the
 10Base5 was the first Ethernet cable is much thinner and more flexible.
specification to use a bus topologywith  In this case, the transceiver is normally part
an external transceiver (transmitter / of the networkinterface card (NIC), which is
receiver) connected via a tap to a thick coaxial installed inside the station.
cable.
10Base-T: Twisted-Pair Ethernet 10Base-F: Fiber Ethernet

 The third implementation is called

 Although there are several types of optical
10Base-T or twisted-pair Ethernet.
fiber 10-Mbps Ethernet, the most common is
 10Base-T uses a physical star topology. The called 10Base-F.
stations are connected to a hub via two pairs of
 10Base-F uses a star topology to connect
twisted cable.
stations to a hub.
 The stations are connected to the hub using
two fiber-optic cables.
Fast Ethernet (100 Mbps)
Fast Ethernet or 100BASE-T provides transmission speeds up to 100 megabits per
second and is typically used for LAN backbone systems.
The 100BASE-T standard consists of three different component specifications –
1. 100 BASE-TX
2. 100BASE-T4
3. 100BASE-FX

100 BASE-TX 100BASE-T4 100BASE-FX

100Base-TX uses two pairs A new standard, called 100Base-FX uses two pairs of fiber-
of twisted-pair cable either 100Base-T4, was optic cables. Optical fiber can easily
UTP or STP. A 100Base- designed to use four pairs of handle high bandwidth requirements.
TX network can provide a UTP for transmitting 100
data rate of100 Mbps. Mbps.

Gigabit Ethernet (1 Gbps)

 The Gigabit Ethernet upgrades the data rate to 1 Gbps(1000 Mbps).
 Gigabit Ethernet can be categorized as either a two-wire or a four-wire
implementation.
 The two-wire implementations use fiber-optic cable (1000Base-SX, short-wave, or
1000Base-LX, long-wave), or STP (1000Base-CX).
 The four-wire version uses category 5 twisted-pair cable (1000Base-T).

10 Gigabit Ethernet(10 Gbps)

 10 Gigabit Ethernet is an upcoming Ethernet technology that transmits at 10Gbps.
 10 Gigabit Ethernet enables a familiar network technology to be used inLAN,
MAN and WAN architectures.
 10 Gigabit Ethernet uses multimode optical fiber up to 300 meters and single
mode fiber up to 40 kilometers.
 Four implementations are the most common: 10GBase-SR, 10GBase-LR,10GBase-
EW, and 10GBase-X4.

COLLISION DETECTION IN ETHERNET

 As the Ethernet supports collision detection, senders are able to determine acollision.
  At the moment an adaptor detects that its frame is colliding with another, it first makes
sure to transmit a 32-bit jamming sequence along with the 64-bit preamble (totally 96
bits) and then stops the transmission.
 These 96 bits are sometimes called Runt Frame.

FRAME FORMAT OF ETHERNET

The Ethernet frame is defined by the format given in the Fig.

 The 64-bit preamble allows the receiver to synchronize with the signal; it is a sequence
of alternating 0’s and 1’s.
 Both the source and destination hosts are identified with a 48-bit address.
 The packet type field serves as the demultiplexing key.
 Each frame contains up to 1500 bytes of data(Body).
 CRC is used for Error detection
Ethernet Addresses
 Every Ethernet host has a unique Ethernet address (48 bits – 6 bytes).
 Ethernet address is represented by sequence of six numbers separated bycolons.
 Each number corresponds to 1 byte of the 6 byte address and is given by pair of
hexadecimal digits.
 Eg: 8:0:2b:e4:b1:2 is the representation of
00001000 00000000 00101011 11100100 10110001 00000010
 Each frame transmitted on an Ethernet is received by every adaptor connected to the
Ethernet.
 In addition to unicast addresses an Ethernet address consisting of all 1s istreated as
broadcast address.
 Similarly the address that has the first bit set to 1 but it is not the broadcast address is
called multicast address.
ADVANTAGES OF ETHERNET
Ethernets are successful because
 It is extremely easy to administer and maintain. There are no switches thatcan fail, no
routing or configuration tables that have to be kept up-to-date, and it is easy to add a new
host to the network.
 It is inexpensive: Cable is cheap, and the only other cost is the networkadaptor on each
host.
3. WIRELESS LAN ( IEEE 802.11)
 Wireless communication is one of the fastest-growing technologies.
 The demand for connecting devices without the use of cables is increasing everywhere.
 Wireless LANs can be found on college campuses, in office buildings, and in many
public areas.

ADVANTAGES OF WLAN / 802.11

1. Flexibility: Within radio coverage, nodes can access each other as radiowaves can
penetrate even partition walls.
2. Planning : No prior planning is required for connectivity as long as devices follow
standard convention
3. Design : Allows to design and develop mobile devices.
4. Robustness : Wireless network can survive disaster. If the devices survive,
communication can still be established.

DISADVANTAGES OF WLAN / 802.11

1. Quality of Service : Low bandwidth (1 – 10 Mbps), higher error rates due to interference,
delay due to error correction and detection.
2. Cost : Wireless LAN adapters are costly compared to wired adapters.
3. Proprietary Solution : Due to slow standardization process, many solution are proprietary
that limit the homogeneity of operation.
4. Restriction : Individual countries have their own radio spectral policies. This restricts the
development of the technology
5. Safety and Security : Wireless Radio waves may interfere with other devices. Eg; In a
hospital, radio waves may interfere with high-tech equipment.
TECHNOLOGY USED IN WLAN / 802.11
 WLAN’s uses Spread Spectrum (SS) technology.
 The idea behind Spread spectrum technique is to spread the signal over a wider
frequency band than normal, so as to minimize the impact of interference from other
devices.
 There are two types of Spread Spectrum:
 Frequency Hopping Spread Spectrum (FHSS)
 Direct Sequence Spread Spectrum (DSSS)
Frequency Hopping Spread Spectrum (FHSS)
 Frequency hopping is a spread spectrum technique that involves transmitting the signal
over a random sequence of frequencies.
 That is, first transmitting at one frequency, then a second, then a third, and so on.
  The random sequence of frequencies is computed by a pseudorandom number generator.
 The receiver uses the same algorithm as the sender and initializes it with the same seed
and hence is able to hop frequencies in sync with the transmitter to correctly receive the
frame.

Direct Sequence Spread Spectrum (DSSS)

 Each bit of data is represented by multiple bits in the transmitted signal.
 DSSS takes a user data stream and performs an XOR operation with apseudo –
random number.
 This pseudo random number is called as chipping sequence.

TOPOLOGY IN WLAN / 802.11

WLANs can be built with either of the following two topologies /architecture:
 Infra-Structure Network Topology
 Ad Hoc Network Topology
 Infra-Structure Topology Ad-Hoc Topology
(AP based Topology) (Peer-to-Peer Topology)

An infrastructure network is the An adhoc network is the architecture

network architecture for providing that is used to support mutual
communication between wireless communication between wireless
clients and wired network resources. clients.
The transition of data from the Typically, an ad- hoc network is
wireless to wired medium occurs via created spontaneously and does not
a Base Station called AP(Access support access to wired networks.
Point). An adhoc network does not require
An AP and its associated wireless an AP.
clients define the coverage area.

ARCHITECURE OF WLAN / 802.11

The standard defines two kinds of services: the Basic Service Set (BSS) and the
Extended Service Set (ESS).
Basic Service Set (BSS)
 IEEE 802.11 defines the basic service set (BSS) as the building blocks of a wireless
LAN.
 A basic service set is made of stationary or mobile wireless stations and an optional
central base station, known as the access point (AP).
Extended Service Set (ESS)
 An extended service set (ESS) is made up of two or more BSSs with APs.
 In this case, the BSSs are connected through a distribution system, which is awired or a
wireless network.
 The distribution system connects the APs in the BSSs. The extended service set uses two
types of stations: mobile and stationary.
 The mobile stations are normal stations inside a BSS.
 The stationary stations are AP stations that are part of a wired LAN.

Station Types
IEEE 802.11 defines three types of stations based on their mobility in a wireless
LAN:
1. No-transition - A station with no-transition mobility is either stationary (not moving)
or moving only inside a BSS.
2. BSS-transition - A station with BSS-transition mobility can move from one BSS
to another, but the movement is confined inside one ESS
3. ESS-transition - A station with ESS-transition mobility can move from one ESS to
another.

COLLISION AVOIDANCE IN WLAN / 802.11

Wireless protocol would follow exactly the same algorithm as the Ethernet—Wait until
the link becomes idle before transmitting and back off should a collision occur.
Hidden Node Problem
 Consider the situation shown in the Figure.
 Here A and C are both within range of B but not with each other.
  Suppose both A and C want to communicate with B and so they each send a frame to B.
 A and C are unaware of each other since their signals do not carry that far.
 These two frames collide with each other at B, but neither A nor C is aware of this
collision.
 A and C are said to be hidden nodes with respect to each other.
Exposed Node Problem

 Each of the four nodes is able to send and receive signals that reach just the nodes to its
immediate left and right.
 For example, B can exchange frames with A and C but it cannot reach D,while C can
reach B and D but not A.
 Suppose B is sending to A. Node C is aware of this communication because it hears B’s
transmission.
 If at the same time, C wants to transmit to node D.
 It would be a mistake, however, for C to conclude that it cannot transmit to anyone just
because it can hear B’s transmission.
 This is not a problem since C’s transmission to D will not interfere with A’s ability to
receive from B.
 This is called exposed problem.
 Although B and C are exposed to each other’s signals, there is no interference if
B transmits to A while C transmits to D.
MULTIPLE ACCESS WITH COLLISION AVOIDANCE (MACA)
 MACA is used to avoid collisions caused by the hidden terminal problem and exposed
terminal problem.
 MACA uses short signaling packets called RTS and CTS for collision avoidance.
 The RTS and CTS signals helps us to determine who else is in the transmission range
or who is busy.
 When a sender wants to transmit, it sends a signal called Request-To-Send(RTS).
 If the receiver allows the transmission, it replies to the sender a signal called Clear-To-
Send (CTS).
 Any node that sees the CTS frame knows that it is close to the receiver, and therefore
cannot transmit for the period of time.
 Any node that sees the RTS frame but not the CTS frame is not close enough to the
receiver to interfere with it, and so is free to transmit.
 The Signaling packets RTS and CTS contains information such as
 sender address
 receiver address
 length of the data to be sent/received

 The receiver sends an ACK to the sender after successfully receiving aframe.
 All nodes must wait for this ACK before trying to transmit.
 When two or more nodes detect an idle link and try to transmit an RTS frame at the same
time, their RTS frames will collide with each other.
 802.11 do not support collision detection, but instead, the senders realize the collision
has happened when they do not receive the CTS frame after a period of time.
 Each node waits for a random amount of time before trying again.
 The amount of time a given node delays is defined by exponential back-off algorithm.
DISTRIBUTION SYSTEM IN WLAN / 802.11
In wireless network, nodes can move freely. Some nodes are allowed to roam and some
are connected to a wired network infrastructure called access points (AP), and they are
connected to each other by a so-called distribution system.
 Two nodes can communicate directly with each other if they are within reach of each
other,
 When the nodes are at different range, for example when node A wish to communicate
with node E, A first sends a frame to its access point (AP-1), which forwards the frame
across the distribution system to AP-3, which finally transmits the frame to E.

Scanning Process in Distribution System

 The technique for selecting an Access Point is called scanning.
 Scanning will take place whenever a node joins the network as well aswhen it is
not satisfied with the current access point signal.
 It involves the following four steps:
 The node sends a Probe Request frame.
 All AP’s within reach reply with a Probe Response frame.
 The node selects one of the access points and sends that AP an Association
Request frame.
 The AP replies with an Association Response frame.
 There are two types of Scanning. They are
 Active Scanning
 Passive Scanning
Active Scanning
When node C moves from the cell serviced by AP-1 to the cell serviced by AP-2. As it
moves, it sends Probe frames, which eventually result in Probe Response. Since the node is
actively searching for an access point it is called active scanning.
Passive Scanning
AP’s periodically send a Beacon frame to the nodes that advertises the capabilities of
the access point which includes the transmission rates supported by the AP. This is called
passive scanning and a node can change to this AP based on the Beacon frame simply by
sending it an Association Request frame back to the access point.

FRAME FORMAT OF WLAN / 802.11

 Control field - contains three subfields :

 Type field - Indicates whether the frame carries data, RTS or CTS frame
 To DS - Data frame sent to DS
 From DS – ACK sent from DS
When both the DS bits are set to 0, it indicates that one node is sending directly
to another. Addr l identifies the target node and Addr2 identifies the source node.
When both the DS bits are set to 1, it indicates that one node is sending the message to
another indirectly using the distribution system.

 Duration - contains the duration of time the medium is occupied by the nodes.
 Addr l - identifies the final original destination
 Addr 2 - identifies the immediate sender (the one that forwarded the frame from the
distribution system to the ultimate destination)
 Addr 3 - identifies the intermediate destination (the one that accepted the frame from a
wireless node and forwarded it across the distribution system)
 Addr 4 - identifies the original source
  Sequence Control - to avoid duplication of frames sequence number is assigned to each
frame
 Payload - Data from sender to receiver
 CRC - used for Error detection of the frame
4. BLUETOOTH (IEEE 802.15.1 )
 A Bluetooth is an ad hoc network, which means that the network is formed
spontaneously.
 Bluetooth is a wireless LAN technology designed to connect devices of different
functions such as telephones, notebooks, computers (desktop and laptop), cameras,
printers, when they are at a short distance from each other.
 Bluetooth technology is the implementation of a protocol defined by the IEEE
802.15 standard.
 The standard defines a wireless personal-area network (PAN)
 Bluetooth operates in the 2.4 GHz Unlicensed ISM band.
 The range for Bluetooth communication is 0-30 feet (10 meters).
 This distance can be increased to 100 meters by amplifying the power.
 Bluetooth links have typical bandwidths around 1 to 3 Mbps.
 Bluetooth is specified by an industry consortium called the Bluetooth Special Interest
Group.
 Upto eight devices can be connected through Bluetooth.
 One device will function as a Master and the other seven devices will function as
slaves.
 Bluetooth uses Frequency Hopping Spread Spectrum (FHSS) to avoid any
interference.
 Bluetooth supports two kinds of links:
 Asynchronous Connectionless (ACL) links - for data
 Synchronous Connection oriented (SCO) links - for audio/voice
BLUETOOTH ARCHITECTURE
Bluetooth defines two types of networks: Piconet and Scatternet.
PICONET
 The basic Bluetooth network configuration is called a Piconet
 A Piconet is a collection of eight bluetooth devices which are synchronized.
 One device in the piconet can act as Primary (Master), all other devicesconnected to
the master act as Secondary (Slaves).
 All the secondary stations synchronize their clocks and hopping sequencewith the
primary.
  Any communication is between the primary/master and a secondary/slave.
 The communication between the primary and secondary stations can be one-to-one or
one-to-many.
 The slaves do not communicate directly with each other.
 The devices in a piconet can be in any one of the three types/states.
 They are
 Active Device / State
1. Connected to the piconet and participates in the communication.
2. Can be a Master or a Slave device.
3. All active devices are assigned a 3-bit address (AMA).
 Parked Device / State
1. Connected to the piconet, but does not actively participate in the communication.
2. More than 200 devices can be parked.
3. All parked devices use an 8-bit parked member address (PMA).
 Stand-by Device / State
1. Not connected to the piconet.
2. They do not participate in the piconet currently but may take part at a later time.
3. Devices in stand-by do not need an address.

 If a parked device wants to communicate and there are already seven active slaves, one
slave has to switch to park state to allow the parked device to switch to active state.

SCATTERNET
 Piconets can be combined to form what is called a scatternet.
 Many piconets with overlapping coverage can exist simultaneously, called Scatternet.
  A secondary station in one piconet can be the primary in another piconet.
 This station can receive messages from the primary in the first piconet (as a secondary)
and, acting as a primary, deliver them to secondaries in the second piconet.
 A station can be a member of two piconets.

 In the example given below, there are two piconets, in which one slave participates
in two different piconets.
 Master of one piconet cannot act as the master of another piconet.
 But the Master of one piconet can act as a Slave in another piconet

BLUETOOTH LAYERS
Radio Layer
 The radio layer is roughly equivalent to the physical layer of the Internet model.
 Bluetooth uses the frequency-hopping spread spectrum (FHSS) method in the
physical layer to avoid interference from other devices or othernetworks.
 Bluetooth hops 1600 times per second, which means that each device changes its
modulation frequency 1600 times per [Link] transform bits to a signal, Bluetooth
uses a sophisticated version of FSK,called GFSK.
Baseband Layer
 The baseband layer is roughly equivalent to the MAC sublayer in LANs.
 The access method is TDMA.
 The primary and secondary stations communicate with each other using time slots.
The length of a time slot is exactly 625 µs.
 During that time, a primary sends a frame to a secondary, or a secondary sends a
frame to the primary.
L2CAP
 The Logical Link Control and Adaptation Protocol, or L2CAP (L2 heremeans LL)
is equivalent to the LLC sublayer in LANs.
 It is used for data exchange on an ACL link.
 The L2CAP functions are: multiplexing, segmentation and reassembly, quality of
service (QoS), and group management.
5. SWITCHING
o The technique of transferring the information from one computer network to another
network is known as switching.
o Switching in a computer network is achieved by using switches.
o A switch is a small hardware device which is used to join multiple computers
together with one local area network (LAN).
o Switches are devices capable of creating temporary connections between two ormore
devices linked to the switch.
o Switches are used to forward the packets based on MAC addresses.
o A Switch is used to transfer the data only to the device that has been addressed. It verifies
the destination address to route the packet appropriately.
o It is operated in full duplex mode.
o It does not broadcast the message as it works with limited bandwidth.
Advantages of Switching:
o Switch increases the bandwidth of the network.
o It reduces the workload on individual PCs as it sends the information to only that device
which has been addressed.
o It increases the overall performance of the network by reducing the traffic on the
network.
o There will be less frame collision as switch creates the collision domain for each
connection.
Disadvantages of Switching:
o A Switch is more expensive than network bridges.
o A Switch cannot determine the network connectivity issues easily.
o Proper designing and configuration of the switch are required to handle multicast
packets.
Types of Switching Techniques

CIRCUIT SWITCHING

o Circuit switching is a switching technique that establishes a dedicated path between

sender and receiver.
o In the Circuit Switching Technique, once the connection is established then the
 dedicated path will remain to exist until the connection is terminated.
o Circuit switching in a network operates in a similar way as the telephone works.
o A complete end-to-end path must exist before the communication takes place.
o In case of circuit switching technique, when any user wants to send the data, voice, video,
a request signal is sent to the receiver then the receiver sends back the acknowledgment
to ensure the availability of the dedicated path. After receiving the acknowledgment,
dedicated path transfers the data.
o Circuit switching is used in public telephone network. It is used for voice transmission.
o Fixed data can be transferred at a time in circuit switching technology.

Phases in Circuit Switching

Communication through circuit switching has 3 phases:

1. Connection Setup / Establishment - In this phase, a dedicated circuit is established from

the source to the destination through a number of intermediate switching centres. The
sender and receiver transmits communication signals to request and acknowledge
establishment of circuits.

2. Data transfer - Once the circuit has been established, data and voice are transferred from
the source to the destination. The dedicated connection remains as long as the end parties
communicate.

3. Connection teardown / Termination - When data transfer is complete, the connection

is relinquished. The disconnection is initiated by any one of the user. Disconnection
involves removal of all intermediate links from the sender to the receiver.

Advantages
 It is suitable for long continuous transmission, since a continuous transmission route
is established, that remains throughout the conversation.
 The dedicated path ensures a steady data rate of communication.
  No intermediate delays are found once the circuit is established. So, they are
suitable for real time communication of both voice and data transmission.

Disadvantages
 Circuit switching establishes a dedicated connection between the end parties. This
dedicated connection cannot be used for transmitting any other data, even if the data load
is very low.
 Bandwidth requirement is high even in cases of low data volume.
 There is underutilization of system resources. Once resources are allocated to a particular
connection, they cannot be used for other connections.
 Time required to establish connection may be high.
 It is more expensive than other switching techniques as a dedicated path isrequired for
each connection.

PACKET SWITCHING
o The packet switching is a switching technique in which the message is sent in onego,
but it is divided into smaller pieces, and they are sent individually.
o The message splits into smaller pieces known as packets and packets are given aunique
number to identify their order at the receiving end.
o Every packet contains some information in its headers such as source address,
destination address and sequence number.
o Packets will travel across the network, taking the shortest path as possible.
o All the packets are reassembled at the receiving end in correct order.
o If any packet is missing or corrupted, then the message will be sent to resend the
message.
o If the correct order of the packets is reached, then the acknowledgment message will
be sent.
Advantages of Packet Switching:
o Cost-effective: In packet switching technique, switching devices do not require massive
secondary storage to store the packets, so cost is minimized to some extent. Therefore,
we can say that the packet switching technique is a cost-effective technique.
o Reliable: If any node is busy, then the packets can be rerouted. This ensures that the
Packet Switching technique provides reliable communication.
o Efficient: Packet Switching is an efficient technique. It does not require any established
path prior to the transmission, and many users can use the same communication channel
simultaneously, hence makes use of available bandwidth very efficiently.

Disadvantages of Packet Switching:

o Packet Switching technique cannot be implemented in those applications that require
low delay and high-quality services.
o The protocols used in a packet switching technique are very complex and requires high
implementation cost.
o If the network is overloaded or corrupted, then it requires retransmission of lost packets.
It can also lead to the loss of critical information if errors are norrecovered.

APPROACHES OF PACKET SWITCHING

There are two approaches to Packet Switching:
o Datagram Packet switching
o Virtual Circuit Switching

Datagram Packet switching

o It is a packet switching technology in which packet is known as a datagram, is
considered as an independent entity.
o Each packet contains the information about the destination and switch uses this
information to forward the packet to the correct destination.
o The packets are reassembled at the receiving end in correct order.
o In Datagram Packet Switching technique, the path is not fixed.
o Intermediate nodes take the routing decisions to forward the packets.
o Datagram Packet Switching is also known as connectionless switching.
o There are no setup or teardown phases.
o Each packet is treated the same by a switch regardless of its source or destination.
 In this example, all four packets (or datagrams) belong to the same message, but may
travel different paths to reach their destination.

Routing Table
In this type of network, each switch (or packet switch) has a routing table which is based
on the destination address. The routing tables are dynamic and are updated periodically. The
destination addresses and the corresponding forwarding output ports are recorded in the tables.

Delay in a datagram network

 The packet travels through two switches.

 There are three transmission times (3T),three propagation delays (slopes 3t of thelines),
and two waiting times (w1 + w2).
 We ignore the processing time in each switch.
Total delay = 3T + 3t + w1 + w2
Virtual Circuit Switching
o Virtual Circuit Switching is also known as connection-oriented switching.
o In the case of Virtual circuit switching, a virtual connection is established before the
messages are sent.
o Call request and call accept packets are used to establish the connection betweensender
and receiver.
o In this case, the path is fixed for the duration of a logical connection.
Virtual Circuit Identifier (VCI)
A virtual circuit identifier (VCI) that uniquely identifies the connection at this switch.
AVCI, unlike a global address, is a small number that has only switch scope; it is used by a
frame between two switches. When a frame arrives at a switch, it has a VCI; when it leaves, it
has a different VCI.

Virtual Circuit Table

Every Virtual Circuit (VC) maintains a table called Virtual Circuit [Link] entry in
the VC table on a single switch contains the following :
 An incoming interface on which packets for this VC arrive at the switch
 An outgoing interface in which packets for this VC leave the switch
 A outgoing VCI that will be used for outgoing packets
Example :
Source A sends a frame to Source B through Switch 1, Switch 2 and Switch 3.
Types of Virtual Circuits
There are two broad classes of Virtual [Link] are

1. PVC – Permanent Virtual Circuit

 Network Administrator will configure the state
 The virtual circuit is permanent (PVC)

2. SVC – Switched Virtual Circuit

 A host can send messages into the network to cause the state to be established. This
is referred as signaling.
 A host may set up and delete such a VC dynamically without the involvement of a
network administrator

Delay in Virtual-Circuit Networks

 The packet is traveling through two switches (routers).

 There are three transmission times (3T ), three propagation times (3t), data transfer
depicted by the sloping lines, a setup delay (which includes transmission and
propagation in two directions), and a teardown delay (which includes transmission and
propagation in one direction).
Total delay = 3T + 3t + Setup delay + Teardown delay
COMPARISON – CIRCUIT SWITCHING AND PACKET SWITCHING

PACKET SWITCHING
CIRCUIT
SWITCHING Virtual Circuit Switching Datagram Switching

Connection oriented Connection oriented Connection less

Packets may be delivered out

Ensures in order delivery Ensures in order delivery of order

No reordering is required No reordering is required Reordering is required

A dedicated path exists for A dedicated path exists for data No dedicated path exists for
data transfer transfer data transfer

All the packets take the All the packets take the same All the packets may not take
same path path the same path

Resources are allocated Resources are allocated on

before data transfer demand using 1st packet No resources are allocated

Stream oriented Packet oriented Packet oriented

Fixed bandwidth Dynamic Bandwidth Dynamic bandwidth

Reliable Reliable Unreliable

No overheads Less overheads Higher overheads

Implemented at physical
layer Implemented at data linklayer Implemented at network layer

Inefficient in terms of Provides better efficiency than Provides better efficiency than
resource utilization circuit switched systems message switched systems
Example- Telephone
systems Examples- X.25, Framerelay Example- Internet
MESSAGE SWITCHING

o Message Switching is a switching technique in which a message is transferred as a

complete unit and routed through intermediate nodes at which it is stored and forwarded.
o In Message Switching technique, there is no establishment of a dedicated path between
the sender and receiver.
o The destination address is appended to the message. Message Switching provides a
dynamic routing as the message is routed through the intermediate nodes based on the
information available in the message.
o Message switches are programmed in such a way so that they can provide the most
efficient routes.
o Each and every node stores the entire message and then forward it to the next node. This
type of network is known as store and forward network.
o Message switching treats each message as an independent entity.

BRIDGES

 Bridges operate in physical layer as well as data link layer.

 As a physical layer device, they regenerate the receive signal.
 As a data link layer, the bridge checks the physical (MAC) address (of thesource and
the destination) contained in the frame.
 The bridge has a filtering feature.
 It can check the destination address of a frame and decides, if the frameshould be
forwarded or dropped.
 Bridges are used to connect two or LANs working on the same protocol.
Types of Bridges :

 Transparent Bridges
These are the bridge in which the stations are completely unaware of the bridge’s
existence i.e. whether or not a bridge is added or deleted from the network , reconfiguration of
the stations is unnecessary.
 Source Routing Bridges
In these bridges, routing operation is performed by source station and the frame
specifies which route to follow.

 Translation Bridges
These bridges connect networks with different architectures, such as Ethernet and Token
Ring. These bridges appear as:
– Transparent bridges to an Ethernet host
– Source-routing bridges to a Token Ring host
6. IP - INTERNET PROTOCOL
 IP runs on all the nodes (both hosts and routers) in a collection of networks
 IP defines the infrastructure that allows these nodes and networks to function as a single
logical internetwork.

IP SERVICE MODEL
 Service Model defines the host-to-host services that we want to provide
 The main concern in defining a service model for an internetwork is that we can provide
a host-to-host service only if this service can somehow be provided over each of the
underlying physical networks.
 The Internet Protocol is the key tool used today to build scalable, heterogeneous
internetworks.
  The IP service model can be thought of as having two parts:
 A GLOBAL ADDRESSING SCHEME - which provides a way to
identify all hosts in the internetwork
 A DATAGRAM DELIVERY MODEL – A connectionless model of datadelivery.

IP PACKET FORMAT / IP DATAGRAM FORMAT

 A key part of the IP service model is the type of packets that can be carried.
 The IP datagram consists of a header followed by a number of bytes of data.

FIELD DESCRIPTION

Version Specifies the version of IP. Two versions exists – IPv4 and IPv6.
HLen Specifies the length of the header
TOS An indication of the parameters of the quality of servicedesired
(Type of Service) such as Precedence, Delay, Throughput and Reliability.
Length Length of the entire datagram, including the header. The maximum
size of an IP datagram is 65,535(210 )bytes
Ident Uniquely identifies the packet sequence [Link] for
(Identification) fragmentation and re-assembly.

Flags Used to control whether routers are allowed to fragment a [Link] a

packet is fragmented , this flag value is [Link] not, flag value is0.

Offset Indicates where in the datagram, this fragment belongs.

(Fragmentation The fragment offset is measured in units of 8 octets(64 bits). The
offset) first fragment has offset zero.
TTL Indicates the maximum time the datagram is allowed to
(Time to Live) remain in the network. If this field contains the value zero, thenthe
datagram must be destroyed.
Protocol Indicates the next level protocol used in the data portion of the
datagram
Checksum Used to detect the processing errors introduced into the packet

Source Address The IP address of the original sender of the packet.

Destination The IP address of the final destination of the packet.

Address
Options This is optional field. These options may contain values foroptions
such as Security, Record Route, Time Stamp, etc
Pad Used to ensure that the internet header ends on a 32 bit boundary.
The padding is zero.

IP DATAGRAM - FRAGMENTATION AND REASSEMBLY

Fragmentation :
 Every network type has a maximum transmission unit (MTU), which is the largest
IP datagram that it can carry in a frame.

 Fragmentation of a datagram will only be necessary if the path to thedestination

includes a network with a smaller MTU.
 When a host sends an IP datagram,it can choose any size that it wants.
 Fragmentation typically occurs in a router when it receives a datagram that it wants
to forward over a network that has an MTU that is smaller than thereceived
datagram.
 Each fragment is itself a self-contained IP datagram that is transmitted over asequence
of physical networks, independent of the other fragments.
 Each IP datagram is re-encapsulated for each physical network over which it travels.

 For example , if we consider an Ethernet network to accept packets up to 1500bytes

long.
  This leaves two choices for the IP service model:
 Make sure that all IP datagrams are small enough to fit inside one packet on any
network technology
 Provide a means by which packets can be fragmented and reassembled when they
are too big to go over a given network technology.
 Fragmentation produces smaller, valid IP datagrams that can be readily reassembled into
the original datagram upon receipt, independent of the order of their arrival.

Example:

 The original packet starts at the client; the fragments are reassembled at theserver.
 The value of the identification field is the same in all fragments, as is the value of the
flags field with the more bit set for all fragments except the last.
 Also, the value of the offset field for each fragment is shown.
 Although the fragments arrived out of order at the destination, they can becorrectly
reassembled.
  The value of the offset field is always relative to the original datagram.
 Even if each fragment follows a different path and arrives out of order, the final
destination host can reassemble the original datagram from thefragments received
(if none of them is lost) using the following strategy:
1) The first fragment has an offset field value of zero.
2) Divide the length of the first fragment by 8. The second fragment has an offset value
equal to that result.
3) Divide the total length of the first and second fragment by 8. The third fragment
has an offset value equal to that result.
4) Continue the process. The last fragment has its M bit set to 0.
5) Continue the process. The last fragment has a more bit value of 0.
Reassembly:
 Reassembly is done at the receiving host and not at each router.
 To enable these fragments to be reassembled at the receiving host, they allcarry the
same identifier in the Ident field.
 This identifier is chosen by the sending host and is intended to be uniqueamong all
the datagrams that might arrive at the destination from this source over some
reasonable time period.
 Since all fragments of the original datagram contain this identifier, thereassembling
host will be able to recognize those fragments that go together.
 For example, if a single fragment is lost, the receiver will still attempt to reassemble the
datagram, and it will eventually give up and have to garbage-collect the resources that
were used to perform the failed reassembly.
 Hosts are now strongly encouraged to perform “path MTU discovery,” a process by
which fragmentation is avoided by sending packets that are smallenough to traverse
the link with the smallest MTU in the path from sender toreceiver.

IP SECURITY
There are three security issues that are particularly applicable to the IP protocol:
(1) Packet Sniffing (2) Packet Modification and (3) IP Spoofing.
Packet Sniffing
 An intruder may intercept an IP packet and make a copy of it.
 Packet sniffing is a passive attack, in which the attacker does not change the contents
of the packet.
 This type of attack is very difficult to detect because the sender and the receiver may
never know that the packet has been copied.
 Although packet sniffing cannot be stopped, encryption of the packet can make the
attacker’s effort useless.
  The attacker may still sniff the packet, but the content is not detectable.

Packet Modification
 The second type of attack is to modify the packet.
 The attacker intercepts the packet, changes its contents, and sends the new packet
to the receiver.
 The receiver believes that the packet is coming from the original sender.
 This type of attack can be detected using a data integrity mechanism.
 The receiver, before opening and using the contents of the message, can use this
mechanism to make sure that the packet has not been changed during the transmission.
IP Spoofing
 An attacker can masquerade as somebody else and create an IP packet that carries
the source address of another computer.
 An attacker can send an IP packet to a bank pretending that it is coming from one of
the customers.
 This type of attack can be prevented using an origin authentication mechanism
IP Sec
 The IP packets today can be protected from the previously mentioned attacks using a
protocol called IPSec (IP Security).
 This protocol is used in conjunction with the IP protocol.
 IPSec protocol creates a connection-oriented service between two entities in which
they can exchange IP packets without worrying about the three attacks such as Packet
Sniffing, Packet Modification and IP Spoofing.
 IP Sec provides the following four services:
1) Defining Algorithms and Keys : The two entities that want to create a secure channel
between themselves can agree on some available algorithms and keys to be used for
security purposes.
2) Packet Encryption : The packets exchanged between two parties can be encrypted
for privacy using one of the encryption algorithms and a shared key agreed upon in
the first step. This makes the packet sniffing attack useless.
3) Data Integrity : Data integrity guarantees that the packet is not modified during the
transmission. If the received packet does not passthe data integrity test, it is
[Link] prevents the second attack,packet modification.
4) Origin Authentication : IPSec can authenticate the origin of the packet to be sure
that the packet is not created by an imposter. This canprevent IP spoofing attacks.
7. ADDRESS RESOLUTION PROTOCOL (ARP)
o ARP stands for Address Resolution Protocol.
o ARP is the most important protocol of the Data Link Layer.
o ARP is a network layer protocol used to convert a IP address (Network/Logical
address) into a MAC Address (Hardware /Physical address).

o The computer programs/applications use logical address (IP address) to send/receive

messages, however the actual communication happens over the physical address
(MAC address).
o To send a datagram over a network, we need both the logical and physical address.
o IP addresses are made up of 32 bits whereas MAC addresses are made up of 48 bits.
o ARP enables each host to build a table of IP address and correspondingphysical address.
o ARP relies on broadcast support from physical networks.
o The Address Resolution Protocol is a request and response protocol.
o The types of ARP messages are:
1. ARP request
2. ARP reply
ARP Operation
o ARP maintains a cache table in which MAC addresses are mapped to IP addresses.
o If a host wants to send an IP datagram to a host,it first checks for a mapping in the cache
table.
o If no mapping is found, it needs to invoke the Address Resolution Protocol over the
network.
o It does this by broadcasting an ARP query onto the network.
o This query contains the target IP address.
o Each host receives the query and checks to see if it matches its IP address.
o If it does match, the host sends a response message that contains its link-layer address
(MAC Address) back to the originator of the query.
o The originator adds the information contained in this response to its ARP table.
o For example,
To determine system B’s physical (MAC) address, system A broadcasts an ARP request
containing B’s IP address to all machines on its network. All nodes except the destination
discard the packet but update their ARP table.

o Destination host (System B) constructs an ARP Response packet

o ARP Response is unicast and sent back to the source host (System A).
o Source stores target Logical & Physical address pair in its ARP table from ARP
Response.
o If target node does not exist on same network, ARP request is sent to default
router.
ARP Packet

RARP – Reverse ARP

o Reverse Address Resolution protocol (RARP) allows a host to convert its MAC
address to the corresponding IP address.
8. DHCP – DYNAMIC HOST CONFIGURATION PROTOCOL
 The dynamic host configuration protocol is used to simplify the installation and
maintenance of networked computers.
 DHCP is derived from an earlier protocol called BOOTP.
 Ethernet addresses are configured into network by manufacturer and they are
unique.
 IP addresses must be unique on a given internetwork but also must reflect the
structure of the internetwork
 Most host Operating Systems provide a way to manually configure the IP information
for the host

 Drawbacks of manual configuration :

1. A lot of work to configure all the hosts in a large network
2. Configuration process is error-prune
 It is necessary to ensure that every host gets the correct network number and that no
two hosts receive the same IP address.
 For these reasons, automated configuration methods are required.
 The primary method uses a protocol known as the Dynamic Host Configuration
Protocol (DHCP).
 The main goal of DHCP is to minimize the amount of manual configuration
required for a host.
 If a new computer is connected to a network, DHCP can provide it with all the
necessary information for full system integration into the network.
 DHCP is based on a client/server model.
 DHCP clients send a request to a DHCP server to which the server responds with an IP
address
 DHCP server is responsible for providing configuration information to hosts.
 There is at least one DHCP server for an administrative domain.
 The DHCP server can function just as a centralized repository for hostconfiguration
information.
  The DHCP server maintains a pool of available addresses that it hands out to hosts on
demand.

 A newly booted or attached host sends a DHCPDISCOVER message to a special IP

address ([Link]., which is an IP broadcast address.
 This means it will be received by all hosts and routers on that network.
 DHCP uses the concept of a relay agent. There is at least one relay agent on each network.
 DHCP relay agent is configured with the IP address of the DHCP server.
 When a relay agent receives a DHCPDISCOVER message, it unicasts it to the DHCP
server and awaits the response, which it will then send back to the requesting client

DHCP Message Format

 A DHCP packet is actually sent using a protocol called the User Datagram Protocol
(UDP).
9. ICMPV4 - INTERNET CONTROL MESSAGE PROTOCOL VERSION 4
 ICMP is a network-layer protocol.
 It is a companion to the IP protocol.
 Internet Control Message Protocol (ICMP) defines a collection of error messages that
are sent back to the source host whenever a router or host isunable to process an IP
datagram successfully.
ICMP MESSAGE TYPES
 ICMP messages are divided into two broad categories: error-reporting messages and
query messages.
 The error-reporting messages report problems that a router or a host(destination)
 may encounter when it processes an IP packet.
 The query messages help a host or a network manager get specific information from a
router or another host.

ICMP Error – Reporting Messages

 Destination Unreachable―When a router cannot route a datagram, the datagram is

discarded and sends a destination unreachable message to source host.
 Source Quench―When a router or host discards a datagram due to congestion, it sends
a source-quench message to the source host. This message acts as flowcontrol.
 Time Exceeded―Router discards a datagram when TTL field becomes 0 and a time
exceeded 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 problemmessage to source.
 Redirection―Redirect messages 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.
ICMP Query Messages

 Echo Request & Reply―Combination of echo request and reply messages

determines whether two systems communicate or not.
 Timestamp Request & Reply―Two machines can use the timestamp requestand reply
messages to determine the round-trip time (RTT).
 Address Mask Request & Reply―A host to obtain its subnet mask, sends an address
mask request message to the router, which responds with an address mask reply
message.
 Router Solicitation/Advertisement―A host broadcasts a router solicitation message to
know about the router. Router broadcasts its routing information with router
advertisement message.
ICMP MESSAGE FORMAT
 An ICMP message has an 8-byte header and a variable-size data section.


Type Defines the type of the message
Code Specifies the reason for the particular message type
Checksum Used for error detection
Rest of the header Specific for each message type
Data Used to carry information
Identifier Used to match the request with the reply
Sequence Number Sequence Number of the ICMP packet
ICMP DEBUGGING TOOLS
Two tools are used for debugging purpose. They are (1) Ping (2) Traceroute

Ping
 The ping program is used to find if a host is alive and responding.
 The source host sends ICMP echo-request messages; the destination, if alive, responds
with ICMP echo-reply messages.
 The ping program sets the identifier field in the echo-request and echo-reply message
and starts the sequence number from 0; this number is incremented by1 each time a new
message is sent.
 The ping program can calculate the round-trip time.
 It inserts the sending time in the data section of the message.
 When the packet arrives, it subtracts the arrival time from the departure time to get the
round-trip time (RTT).
$ ping [Link]

Traceroute or Tracert
 The traceroute program in UNIX or tracert in Windows can be used to trace the path
of a packet from a source to the destination.
 It can find the IP addresses of all the routers that are visited along the path.
 The program is usually set to check for the maximum of 30 hops (routers) to be visited.
 The number of hops in the Internet is normally less than this.
$ traceroute [Link]
 Unit – 3 Routing and Routing Protocols
Routing - RIP - OSPF - IGRP - EIGRP - Metrics - Switch Basics - Global
Internet - Domains - BGP - IPv6 - Multicast Communication -IP addresses
- Address Classes – Subnetting – Supernetting – Examples - Multicast
Routing – DVMRP - PIM. RFC-IPV6 and OSPF.
Routing
 Routing is the process of selecting best paths in a network.
 Routing a packet from its source to its destination means routing the packet from
a source router (the default router of the source host) to a destination router (the
router connected to the destination network).
 There are two processes inside router:
a) One of them handles each packet as it arrives, looking up the outgoing line to
use for it in the routing table. This process is forwarding.
b) The other process is responsible for filling in and updating the routing tables.
That is where the routing algorithm comes into play. This process is routing.
 The source host needs no forwarding table because it delivers its packet to the default
router in its local network.
 The destination host needs no forwarding table either because it receives the packet
from its default router in its local network.
 Only the intermediate routers in the networks need forwarding tables.
 The basic problem of routing is to find the lowest-cost path between any two nodes,
where the cost of a path equals the sum of the costs of all the edges thatmake up the
path.
 This static approach has several problems:
• It does not deal with node or link failures.
• It does not consider the addition of new nodes or links.
• It implies that edge costs cannot change.
 For these reasons, routing is achieved by running routing protocols among the nodes.
 These protocols provide a distributed, dynamic way to solve the problem of finding
the lowest-cost path in the presence of link and node failures andchanging edge
costs.
 The forwarding table is used when a packet is being forwarded and so must contain
enough information to accomplish the forwarding function.
 This means that a row in the forwarding table contains the mapping from a network
prefix to an outgoing interface and some MAC information, such as the Ethernet
address of the next hop.
 The routing table, on the other hand, is the table that is built up by the routing
algorithms as a precursor to building the forwarding table.
  It generally contains mappings from network prefixes to next hops.
 It may also contain information about how this information was learned, so that the
router will be able to decide when it should discard some information.

 Whether the routing table and forwarding table are actually separate data structures
is something of an implementation choice, but there are numerous reasons to keep
them separate.
 For example, the forwarding table needs to be structured to optimize the process of
looking up an address when forwarding a packet, while the routing table needs to
be optimized for the purpose of calculating changes in topology.
 In many cases, the forwarding table may even be implemented in specialized
hardware, whereas this is rarely if ever done for the routing table.
NETWORK AS A GRAPH
 The Figure below shows a graph representing a network.

 The nodes of the graph, labeled A through G, may be hosts, switches, routers,or
networks.
 The edges of the graph correspond to the network links.
 Each edge has an associated cost , which gives some indication of the desirability
 of sending traffic over that link.
 The basic problem of routing is to find the lowest-cost path between any two nodes,
where the cost of a path equals the sum of the costs of all the edges that make up
the path. Such a static approach has several shortcomings:
o It does not deal with node or link failures.
o It does not consider the addition of new nodes or links.
o It implies that edge costs cannot change, even though we might reasonably
wish to have link costs change over time (e.g., assigning high cost to a link
that is heavily loaded).
o For these reasons, routing is achieved in most practical networks by running
routing protocols among the nodes. These protocols provide a distributed,
dynamic way to solve the problem of finding the lowest-cost path in the
presence of link and node failures and changing edge costs.
Unicast Routing Algorithms
There are three main classes of routing protocols:
1) Distance Vector Routing Algorithm – Routing Information Protocol
2) Link State Routing Algorithm – Open Shortest Path First Protocol
3) Path-Vector Routing Algorithm - Border Gateway Protocol
DISTANCE VECTOR ROUTING
 Distance vector routing is distributed, i.e., algorithm is run on all nodes.
 Each node knows the distance (cost) to each of its directly connected neighbors.
 Nodes construct a vector (Destination, Cost, NextHop) and distributes to its
neighbors.
 Nodes compute routing table of minimum distance to every other node via
NextHop using information obtained from its neighbors.
Initial State

 In given network, cost of each link is 1 hop.

 Each node sets a distance of 1 (hop) to its immediate neighbor andcost to itself as 0.
 Distance for non-neighbors is marked as unreachable with value ∞ (infinity).
 For node A, nodes B, C, E and F are reachable, whereas nodes D and G are
unreachable.
 The initial table for all the nodes are given below

 Each node sends its initial table (distance vector) to neighbors and receivestheir
estimate.
 Node A sends its table to nodes B, C, E & F and receives tables from nodes B,C, E
& F.
 Each node updates its routing table by comparing with each of its neighbor's table
 For each destination, Total Cost is computed as:
 Total Cost = Cost (Node to Neighbor) + Cost (Neighbor to Destination)
 If Total Cost < Cost then
 Cost = Total Cost and NextHop = Neighbor
 Node A learns from C's table to reach node D and from F's table to reach node G.
 Total Cost to reach node D via C = Cost (A to C) + Cost(C to D)
Cost = 1 + 1 = 2.
 Since 2 < ∞, entry for destination D in A's table is changed to (D, 2, C)
 Total Cost to reach node G via F = Cost(A to F) + Cost(F to G) = 1 + 1 = 2
 Since 2 < ∞, entry for destination G in A's table is changed to (G, 2, F)
 Each node builds complete routing table after few exchanges amongst its neighbors.
  System stabilizes when all nodes have complete routing information, i.e.,
convergence.
 Routing tables are exchanged periodically or in case of triggered update.
 The final distances stored at each node is given below:

Updation of Routing Tables

There are two different circumstances under which a given node decides to
send arouting update to its neighbors.
Periodic Update
 In this case, each node automatically sends an update message every so often,even if
nothing has changed.
 The frequency of these periodic updates varies from protocol to protocol, butit is
typically on the order of several seconds to several minutes.
Triggered Update
 In this case, whenever a node notices a link failure or receives an update from one of
its neighbors that causes it to change one of the routes in its routingtable.
 Whenever a node’s routing table changes, it sends an update to its neighbors, which
may lead to a change in their tables, causing them to send an update totheir neighbors.
ROUTING INFORMATION PROTOCOL (RIP)
 RIP is an intra-domain routing protocol based on distance-vector algorithm.

Example

 Routers advertise the cost of reaching networks. Cost of reaching each link is 1
hop. For example, router C advertises to A that it can reach network 2, 3 at cost0
(directly connected), networks 5, 6 at cost 1 and network 4 at cost 2.
 Each router updates cost and next hop for each network number.
 Infinity is defined as 16, i.e., any route cannot have more than 15 hops.
 Therefore RIP can be implemented on small-sized networks only.
 Advertisements are sent every 30 seconds or in case of triggered update.

 Command - It indicates the packet type. Value 1 represents a request packet. Value 2
represents a response packet.
 Version - It indicates the RIP version number. For RIPv1, the value is 0x01.
 Address Family Identifier - When the value is 2, it represents the IP protocol.
 IP Address - It indicates the destination IP address of the route. It can be the
addresses of only the natural network segment.
 Metric - It indicates the hop count of a route to its destination.
Count-To-Infinity (or) Loop Instability Problem
 Suppose link from node A to E goes down.
 Node A advertises a distance of ∞ to E to its neighbors
 Node B receives periodic update from C before A’s update reaches B
 Node B updated by C, concludes that E can be reached in 3 hops via C
 Node B advertises to A as 3 hops to reach E
 Node A in turn updates C with a distance of 4 hops to E and so on
 Thus nodes update each other until cost to E reaches infinity, i.e., no convergence.
 Routing table does not stabilize.
 This problem is called loop instability or count to infinity
Solution to Count-To-Infinity (or) Loop Instability Problem :
 Infinity is redefined to a small number, say 16.
 Distance between any two nodes can be 15 hops maximum. Thus distancevector
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.
 Split horizon with poison reverse allows nodes to advertise routes it learnt from a
node back to that node, but with a warning message
LINK STATE ROUTING (LSR)
 Each node knows state of link to its neighbors and cost.
 Nodes create an update packet called link-state packet (LSP) that contains:
 ID of the node
 List of neighbors for that node and associated cost
 64-bit Sequence number
 Time to live
 Link-State routing protocols rely on two mechanisms:
 Reliable flooding of link-state information to all other nodes
 Route calculation from the accumulated link-state knowledge

Reliable Flooding
 Each node sends its LSP out on each of its directly connected links.
 When a node receives LSP of another node, checks if it has an LSP already for that
node.
 If not, it stores and forwards the LSP on all other links except the incoming one.
 Else if the received LSP has a bigger sequence number, then it is stored and
forwarded. Older LSP for that node is discarded.
 Otherwise discard the received LSP, since it is not latest for that node
 Thus recent LSP of a node eventually reaches all nodes, i.e., reliable flooding.

 Flooding of LSP in a small network is as follows:

 When node X receives Y’s LSP (fig a), it floods onto its neighbors A and C (fig
b)
 Nodes A and C forward it to B, but does not sends it back to X (fig c).
 Node B receives two copies of LSP with same sequence number.
 Accepts one LSP and forwards it to D (fig d). Flooding is complete.
 LSP is generated either periodically or when there is a change in the topology.
Route Calculation
 Each node knows the entire topology, once it has LSP from every other node.
 Forward search algorithm is used to compute routing table from the receivedLSPs.
 Each node maintains two lists, namely Tentative and Confirmed with entries ofthe
form (Destination, Cost, NextHop).

OPEN SHORTEST PATH FIRST PROTOCOL (OSPF)

 OSPF is a non-proprietary widely used link-state routing protocol.
 OSPF Features are:
 Authentication―Malicious host can collapse a network by advertising to
reach every host with cost 0. Such disasters are averted by authenticating
routing updates.
 Additional hierarchy―Domain is partitioned into areas, i.e., OSPF is more
scalable.
 Load balancing―Multiple routes to the same place are assigned same cost.
Thus traffic is distributed evenly.
Link State Packet Format

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

 Type ― represents the type (1–5) of OSPF message.
Type 1 - “hello” message,Type 2 - request,Type 3 – send , Type 4 - acknowledge the receipt
of link state messages , Type 5 - reserved
 SourceAddr ― identifies the sender
 AreaId ― 32-bit identifier of the area in which the node is located
 Checksum ― 16-bit internet checksum
 Authentication type ― 1 (simple password), 2 (cryptographic authentication).
 Authentication ― contains password or cryptographic checksum
Difference Between Distance-Vector And Link-State Algorithms

Distance vector Routing Link state Routing

Each node talks only to its directly Each node talks to all other nodes, but it
connected neighbors, but it tells them tells them only what it knows for sure
everything it has learned (i.e., only the state of its directly
(i.e., distance to all nodes). connected links).

PATH VECTOR ROUTING (PVR)

 Path-vector routing is an asynchronous and distributed routing algorithm.
 The Path-vector routing is not based on least-cost routing.
 The best route is determined by the source using the policy it imposes on theroute.
 In other words, the source can control the path.
 Path-vector routing is not actually used in an internet, and is mostly designed toroute a
packet between ISPs.
Spanning Trees
 In path-vector routing, the path from a source to all destinations is determinedby
the best spanning tree.
 The best spanning tree is not the least-cost tree.
 It is the tree determined by the source when it imposes its own policy.
 If there is more than one route to a destination, the source can choose the routethat
meets its policy best.
 A source may apply several policies at the same time.
 One of the common policies uses the minimum number of nodes to be visited.
Another common policy is to avoid some nodes as the middle node in a route.
 The spanning trees are made, gradually and asynchronously, by each node. When
a node is booted, it creates a path vector based on the information it canobtain about
its immediate neighbor.
 A node sends greeting messages to its immediate neighbors to collect these
pieces of information.
 Each node, after the creation of the initial path vector, sends it to all itsimmediate
neighbors.
 Each node, when it receives a path vector from a neighbor, updates its path
vector using the formula

 The policy is defined by selecting the best of multiple paths.

 Path-vector routing also imposes one more condition on this equation.
 If Path (v, y) includes x, that path is discarded to avoid a loop in the path.
 In other words, x does not want to visit itself when it selects a path to y.
Example:
 The Figure below shows a small internet with only five nodes.
 Each source has created its own spanning tree that meets its policy.
 The policy imposed by all sources is to use the minimum number of nodes toreach
a destination.
 The spanning tree selected by A and E is such that the communication does notpass
through D as a middle node.
 Similarly, the spanning tree selected by B is such that the communication doesnot pass
through C as a middle node.

Path Vectors made at booting time

 The Figure below shows all of these path vectors for the example.
 Not all of these tables are created simultaneously.
 They are created when each node is booted.
 The figure also shows how these path vectors are sent to immediate neighbors after
they have been created.

Updating Path Vectors

 The Figure below shows the path vector of node C after two events.
 In the first event, node C receives a copy of B’s vector, which improves itsvector:
now it knows how to reach node A.
 In the second event, node C receives a copy of D’s vector, which does notchange
its vector.
 The vector for node C after the first event is stabilized and serves as itsforwarding
table.
BORDER GATEWAY PROTOCOL (BGP)
 The Border Gateway Protocol version (BGP) is the only interdomain routing
protocol used in the Internet today.
 BGP4 is based on the path-vector algorithm. It provides information about the
reachability of networks in the Internet.
 BGP views internet as a set of autonomous systems interconnected arbitrarily.

 Each AS have a border router (gateway), by which packets enter and leave that AS.
In above figure, R3 and R4 are border routers.
 One of the router in each autonomous system is designated as BGP speaker.
 BGP Speaker exchange reachability information with other BGP speakers, known
as external BGP session.
 BGP advertises complete path as enumerated list of AS (path vector) to reach a
particular network.
 Paths must be without any loop, i.e., AS list is unique.
 For example, backbone network advertises that networks 128.96 and 192.4.153 can
be reached along the path <AS1, AS2, AS4>.

 If there are multiple routes to a destination, BGP speaker chooses one based on
policy.
  Speakers need not advertise any route to a destination, even if one exists.
 Advertised paths can be cancelled, if a link/node on the path goes down. This
negative advertisement is known as withdrawn route.
 Routes are not repeatedly sent. If there is no change, keep alive messages are sent.

iBGP - interior BGP

A Variant of BGP
 Used by routers to update routing information learnt from other speakers to routers
inside the autonomous system.
 Each router in the AS is able to determine the appropriate next hop for all prefixes.
IPV6 - NEXT GENERATION IP
 IPv6 was evolved to solve address space problem and offers rich set of services.
 Some hosts and routers will run IPv4 only, some will run IPv4 and IPv6 and some
will run IPv6 only.
DRAWBACKS OF IPV4
 Despite subnetting and CIDR, address depletion is still a long-term problem.
 Internet must accommodate real-time audio and video transmission that requires
minimum delay strategies and reservation of resources.
 Internet must provide encryption and authentication of data for some applications
FEATURES OF IPV6
1. Better header format - IPv6 uses a new header format in which options are
separated from the base header and inserted, when needed, between the base
header and the data. This simplifies and speeds up the routing process because most
of the options do not need to be checked by routers.
2. New options - IPv6 has new options to allow for additional functionalities.
3. Allowance for extension - IPv6 is designed to allow the extension of the
protocol if required by new technologies or applications.
4. Support for resource allocation - In IPv6, the type-of-service field has been
removed, but two new fields, traffic class and flow label, have been added to
enable the source to request special handling of the packet. This mechanism can
be used to support traffic such as real-time audio and video.
Additional Features :
1. Need to accommodate scalable routing and addressing
 2. Support for real-time services
3. Security support
4. Auto configuration -The ability of hosts to automatically configure themselves with
such information as their own IP address and domain name.
5. Enhanced routing functionality, including support for mobile hosts
6. Transition from ipv4 to ipv6
ADDRESS SPACE ALLOCATION OF IPV6
 IPv6 provides a 128-bit address space to handle up to 3.4 × 10 38 nodes.
 IPv6 uses classless addressing, but classification is based on MSBs.
 The address space is subdivided in various ways based on the leading bits.
 The current assignment of prefixes is listed in Table

 A node may be assigned an “IPv4-compatible IPv6 address” by zero-extending a

32-bit IPv4 address to 128 bits.
 A node that is only capable of understanding IPv4 can be assigned an “IPv4-
mapped IPv6 address” by prefixing the 32-bit IPv4 address with 2 bytes of all 1s
and then zero-extending the result to 128 bits.

GLOBAL UNICAST
 Large chunks (87%) of address space are left unassigned for future use.
 IPv6 defines two types of local addresses for private networks.
o Link local - enables a host to construct an address that
need not beglobally unique.
o Site local - allows valid local address for use in a isolated site
withseveral subnets.
 Reserved addresses start with prefix of eight 0's.
o Unspecified address is used when a host does not know its
address
o Loopback address is used for testing purposes before
connecting
o Compatible address is used when IPv6 hosts uses IPv4
network
o Mapped address is used when a IPv6 host communicates with a IPv4 host
 IPv6 defines any cast address, assigned to a set of interfaces.
 Packet with any cast address is delivered to only one of the nearest interface.
ADDRESS NOTATION OF IPV6
 Standard representation of IPv6 address is x : x : x : x : x : x : x : x where x is a16-
bit hexadecimal address separated by colon (:).
For example,
47CD : 1234 : 4422 : ACO2 : 0022 : 1234 : A456 : 0124

 IPv6 address with contiguous 0 bytes can be written compactly.

For example,
47CD : 0000 : 0000 : 0000 : 0000 : 0000 : A456 : 0124 → 47CD : : A456 : 0124

 IPv4 address is mapped to IPv6 address by prefixing the 32-bit IPv4 address
with 2 bytes of 1s and then zero-extending the result to 128 bits.
For example,
128. 96.33.81 → : : FFFF : [Link]
This notation is called as CIDR notation or slash notation.
ADDRESS AGGREGATION OF IPV6
 IPv6 provides aggregation of routing information to reduce the burden onrouters.
 Aggregation is done by assigning prefixes at continental level.
 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.

 Prefix - All addresses in the same continent have a common prefix

 RegistryID ― identifies the continent
 ProviderID ― identifies the provider for Internet access, i.e., ISP.
 SubscriberID ― specifies the subscriber identifier
 SubnetID ― contains subnet of the subscriber.
 InterfaceID ―contains link level or physical address.

PACKET FORMAT OF IPV6

IPv6 base header is 40 bytes long.
 Version — specifies the IP version, i.e., 6.
 Traffic Class — defines priority of the packet with respect to trafficcongestion.
It is either congestion-controlled or non-congestion controlled
 Flow Label — provides special handling for a particular flow of data. Router
handles different flows with the help of a flow table.
 Payload Len — gives length of the packet, excluding IPv6 header.
  Next Header — Options are specified as a header following IP header. NextHeader
contains a pointer to optional headers.
 Hop Limit — Gives the TTL value of a packet.
 Source Address / Destination Address — 16-byte addresses of source and
destination host
Extension Headers
 Extension header provides greater functionality to IPv6.
 Base header may be followed by six extension headers.
 Each extension header contains a NextHeader field to identify the header following
it.

 Hop-by-Hop — source host passes information to all routers visited by the packet
 Destination — source host information is passed to the destination only.
 Source Routing — routing information provided by the source host.
 Fragmentation — In IPv6, only the source host can fragment. Source uses a path
MTU discovery technique to find smallest MTU on the path.
 Authentication — used to validate the sender and ensures data integrity.
 ESP (Encrypted Security Payload) — provides confidentiality against eavesdropping
ADVANCED CAPABILITIES OF IPV6
 Auto Configuration — Auto or stateless configuration of IP address to hosts without
the need for a DHCP server, i.e., plug and play.
 Advanced Routing — Enhanced routing support for mobile hosts is provided.
 Additional Functions ― Enhanced routing functionality with support for mobile
hosts.
 Security ― Encryption and authentication options provide confidentiality and
integrity.
  Resource allocation ― Flow label enables the source to request special handling of
real-time audio and video packets
ADVANTAGES OF IPV6
 Address space ― IPv6 uses 128-bit address whereas IPv4 uses 32-bit address.
Hence IPv6 has huge address space whereas IPv4 faces address shortage problem.
 Header format ― Unlike IPv4, optional headers are separated from base header in
IPv6. Each router thus need not process unwanted addition information.
 Extensible ― Unassigned IPv6 addresses can accommodate needs of future
technologies.
Dual-Stack Operation and Tunneling
 In dual-stack, nodes run both IPv6 and IPv4, uses Version field to decide which stack
should process an arriving packet.
 IPv6 packet is encapsulated with an IPv4 packet as it travels through an IPv4
network. This is known as tunneling and packet contains tunnel endpoint as its
destination address.
Network Address Translation
 NAT enables hosts on a network to use Internet with local addresses.
 Addresses reserved for internal use range from [Link] to [Link]
 Organization must have single connection to the Internet through a router that runs
the NAT software.
MULTICAST COMMUNICATION
There are, however, applications that need a broader multicasting capability
that is effective at the scale of internet works. For example, when a radio station is
broadcast over the Internet, the same data must be sent to all the hosts where a user has
tuned in to that station. In that example, the communication is one-to-many. Other
examples of one-to-many applications include transmitting the same news, current stock
prices, or software updates to multiple hosts.
There are also applications whose communication is many-to-many, such as
multimedia teleconferencing, online multiplayer gaming, or distributed simulations. In
such cases, members of a group receive data from multiple senders, typically each other.
From any particular sender, they all receive the same data.
Normal IP communication, in which each packet must be addressed and sent
to a single host, is not well suited to such applications. If an application has data to send to
a group, it would have to send a separate packet with the identical data to each member of
the group. This redundancy consumes more bandwidth than necessary.
The redundant traffic is not distributed evenly but rather is focused around the
sending host, and may easily exceed the capacity of the sending host and the nearby
networks and routers.
To better support many-to-many and one-to-many communication, IP provides
an IP-level multicast analogous to the link-level multicast provided by multi-access
networks like Ethernet The basic IP multicast model is a many-to-many model based on
multicast groups, where each group has its own IP multicast address. The hosts that are
members of a group receive copies of any packets sent to that group’s multicast address. A
host can be in multiple groups, and it can join and leave groups freely by telling its local
router using a protocol.
The multicast addresses are associated with an abstract group, the membership
of which changes dynamically over time. Further, the original IP multicast service model
allows any host to send multicast traffic to a group; it doesn’t have to be a member of the
group, and there may be any number of such senders to a given group. Using IP multicast
to send the identical packet to each member of the group, a host sends a single copy of the
packet addressed to the group’s multicast address. The sending host doesn’t need to know
the individual unicast IP address of each member of the group. Similarly, the sending host
doesn’t need to send multiple copies of the packet because the routers will make copies
whenever they have to forward the packet over more than one link.
IP multicast is more scalable because it eliminates the redundant traffic
(packets) that would have been sent many times over the same links, especially those near
to the sending host.
IP’s original many-to-many multicast has been supplemented with support for
a form of one-to-many multicast. In this model of one-to-many multicast, called Source-
Specific Multicast (SSM), a receiving host specifies both a multicast group and a specific
sending host. The receiving host would then receive multicasts addressed to the specified
group, but only if they are from the specified sender. Many Internet multicast applications
(e.g., radio broadcasts) fit the SSM model. To contrast it with SSM, IP’s original many-to-
many model is sometimes referred to as Any Source Multicast (ASM).
A host signals its desire to join or leave a multicast group by communicating
with its local router using a special protocol for just that purpose. In IPv4, that protocol is
the Internet Group Management Protocol (IGMP); in IPv6, it is Multicast Listener
Discovery (MLD).
IPV4 ADDRESSES
 The identifier used in the IP layer of the TCP/IP protocol suite to identify the
connection of each device to the Internet is called the Internet address or IP address.
 Internet Protocol version 4 (IPv4) is the fourth version in the development of the
Internet Protocol (IP) and the first version of the protocol to be widely deployed.
 IPv4 is described in IETF publication in September 1981.
 The IP address is the address of the connection, not the host or the router. An IPv4
address is a 32-bit address that uniquely and universally defines the connection.
 If the device is moved to another network, the IP address may be changed.
 IPv4 addresses are unique in the sense that each address defines one, and only one,
connection to the Internet.
 If a device has two connections to the Internet, via two networks, it has two IPv4
addresses.
 IPv4 addresses are universal in the sense that the addressing system must be
accepted by any host that wants to be connected to the Internet.
IPV4 ADDRESS SPACE
 IPv4 defines addresses has an address space.
 An address space is the total number of addresses used by the protocol.
 If a protocol uses b bits to define an address, the address space is 2b because each
bit can have two different values (0 or 1).
 IPv4 uses 32-bit addresses, which means that the address space is 232 or
4,294,967,296 (more than four billion).
 4 billion devices could be connected to the Internet.
IPV4 ADDRESS NOTATION
There are three common notations to show an IPv4 address:
(i) binary notation (base 2),
(ii) dotted-decimal notation (base 256), and
(iii) hexadecimal notation (base 16).

In binary notation, an IPv4 address is displayed as 32 bits. To make the address

morereadable, one or more spaces are usually inserted between bytes (8 bits).
In dotted-decimal notation,IPv4 addresses are usually written in decimal form with
a decimal point (dot) separating the bytes. Each number in the dotted-decimal notation is
between 0 and 255.
In hexadecimal notation, each hexadecimal digit is equivalent to four bits. This
means that a 32-bit address has 8 hexadecimal digits. This notation is often used in network
programming.
HIERARCHY IN IPV4 ADDRESSING
 In any communication network that involves delivery, the addressing system is
hierarchical.
 A 32-bit IPv4 address is also hierarchical, but divided only into two parts.
 The first part of the address, called the prefix, defines the network(Net ID); the second
part of the address, called the suffix, defines the node (Host ID).
 The prefix length is n bits and the suffix length is (32- n) bits.
  A prefix can be fixed length or variable length.
 The network identifier in the IPv4 was first designed as a fixed-length prefix.
 This scheme is referred to as classful addressing.
 The new scheme, which is referred to as classless addressing, uses a variable-length
network prefix.
CATEGORIES OF IPV4 ADDRESSING
 There are two broad categories of IPv4 Addressing techniques.
 They are
 Classful Addressing
 Classless Addressing
CLASSFUL ADDRESSING
 An IPv4 address is 32-bit long(4 bytes).
 An IPv4 address is divided into sub-classes:
CLASSFUL NETWORK ARCHITECTURE
Class A
 In Class A, an IP address is assigned to those networks that contain a large
number of hosts.
 The network ID is 8 bits long.
 The host ID is 24 bits long.
 In Class A, the first bit in higher order bits of the first octet is always set to 0 and
the remaining 7 bits determine the network ID.
 The 24 bits determine the host ID in any network.
 The total number of networks in Class A = 27 = 128 network address
 The total number of hosts in Class A = 224 - 2 = 16,777,214 host address

Class B
 In Class B, an IP address is assigned to those networks that range from small-sized
to large-sized networks.
 The Network ID is 16 bits long.
 The Host ID is 16 bits long.
 In Class B, the higher order bits of the first octet is always set to 10, and the
remaining14 bits determine the network ID.
 The other 16 bits determine the Host ID.
 The total number of networks in Class B = 214 = 16384 network address
 The total number of hosts in Class B = 216 - 2 = 65534 host address

Class C
 In Class C, an IP address is assigned to only small-sized networks.
 The Network ID is 24 bits long.
 The host ID is 8 bits long.
 In Class C, the higher order bits of the first octet is always set to 110, and the
remaining 21 bits determine the network ID.
 The 8 bits of the host ID determine the host in a network.
 The total number of networks = 2 21 = 2097152 network address
 The total number of hosts = 28 - 2 = 254 host address
Class D

 In Class D, an IP address is reserved for multicast addresses.

 It does not possess subnetting.
 The higher order bits of the first octet is always set to 1110, and the remaining bits
determines the host ID in any network.

Class E

 In Class E, an IP address is used for the future use or for the research and
development purposes.
 It does not possess any subnetting.
 The higher order bits of the first octet is always set to 1111, and the remaining bits
determines the host ID in any network.

Address Depletion in Classful Addressing

 The reason that classful addressing has become obsolete is address depletion.
 Since the addresses were not distributed properly, the Internet was faced with the
problem of the addresses being rapidly used up.
 This results in no more addresses available for organizations and individuals that
needed to be connected to the Internet.
 To understand the problem, let us think about class A.
 This class can be assigned to only 128 organizations in the world, but each
organization needs to have a single network with 16,777,216 nodes.
 Since there may be only a few organizations that are this large, most of the
addresses in this class were wasted (unused).
 Class B addresses were designed for midsize organizations, but many of the
addresses in this class also remained unused.
 Class C addresses have a completely different flaw in design. The number of
addresses that can be used in each network (256) was so small that most
companies were not comfortable using a block in this address class.
 Class E addresses were almost never used, wasting the whole class.
Advantage of Classful Addressing
 Although classful addressing had several problems and became obsolete, it had one
advantage.
 Given an address, we can easily find the class of the address and, since the
 prefix length for each class is fixed, we can find the prefix length immediately.
 In other words, the prefix length in classful addressing is inherent in the address;
no extra information is needed to extract the prefix and the suffix.

Subnetting and Supernetting

To alleviate address depletion, two strategies were proposed and implemented:
(i) Subnetting and (ii) Supernetting.
Subnetting
 In subnetting, a class A or class B block is divided into several subnets.
 Each subnet has a larger prefix length than the original network.
 For example, if a network in class A is divided into four subnets, each subnet has
a prefix of nsub = 10.
 At the same time, if all of the addresses in a network are not used, subnetting
allows the addresses to be divided among several organizations.
CLASSLESS ADDRESSING
 In 1996, the Internet authorities announced a new architecture called classless
addressing.
 In classless addressing, variable-length blocks are used that belong to no classes.
 We can have a block of 1 address, 2 addresses, 4 addresses, 128 addresses, and so
on.
 In classless addressing, the whole address space is divided into variable length
blocks.
 The prefix in an address defines the block (network); the suffix defines the
node (device).
 Theoretically, we can have a block of 2 0, 21, 22, ……., 232 addresses.
 The number of addresses in a block needs to be a power of 2. An organization can
be granted one block of addresses.

 The prefix length in classless addressing is variable.

 We can have a prefix length that ranges from 0 to 32.
 The size of the network is inversely proportional to the length of the prefix.
 A small prefix means a larger network; a large prefix means a smaller network.
 The idea of classless addressing can be easily applied to classful addressing.
 An address in class A can be thought of as a classless address in which the
prefix length is 8.
 An address in class B can be thought of as a classless address in which the
prefix is 16, and so on. In other words, classful addressing is a special case of
classless addressing.
Notation used in Classless Addressing
 The notation used in classless addressing is informally referred to as slash notation
and formally as classless interdomain routing or CIDR.

 For example , [Link] /24 represents the IP address [Link] and, its
subnet mask [Link], which has 24 leading 1-bits.
Address Aggregation
 One of the advantages of the CIDR strategy is address aggregation (sometimes
called address summarization or route summarization).
 When blocks of addresses are combined to create a larger block, routing can be done
based on the prefix of the larger block.
 ICANN assigns a large block of addresses to an ISP.
 Each ISP in turn divides its assigned block into smaller sub blocks and grants the
sub blocks to its customers.

Special Addresses in IPv4

 There are five special addresses that are used for special purposes: this-host address,
limited-broadcast address, loopback address,private addresses, and multicast address.
This-host Address
 The only address in the block [Link]/32 is called the this-host address.
 It is used whenever a host needs to send an IP datagram but it does not know its own
address to use as the source address.
Limited-broadcast Address
 The only address in the block [Link]/32 is called the limited-broadcast
address.
 It is used whenever a router or a host needs to send a datagram to all devices in a
network.
 The routers in the network, however, block the packet having this address as the
destination; the packet cannot travel outside the network.
Loopback Address
 The block [Link]/8 is called the loopback address.
 A packet with one of the addresses in this block as the destination address never
leaves the host; it will remain in the host.
Private Addresses
 Four blocks are assigned as private addresses: [Link]/8, [Link]/12,
 [Link]/16, and [Link]/16.
Multicast Addresses
The block [Link]/4 is reserved for multicast addresses
MULTICASTING
 In multicasting, there is one source and a group of destinations.
 Multicast supports efficient delivery to multiple destinations.
 The relationship is one to many or many-to-many.
 One-to-Many (Source Specific Multicast)
Radio station broadcast
Transmitting news, stock-price
Software updates to multiple hosts
 Many-to-Many (Any Source Multicast)
Multimedia teleconferencing
Online multi-player games
Distributed simulations
 In this type of communication, the source address is a unicast address, but the
destination address is a group address.
 The group address defines the members of the group.

 In multicasting, a multicast router may have to send out copies of the same
datagram through more than one interface.
 Hosts that are members of a group receive copies of any packets sent to that group’s
multicast address
 A host can be in multiple groups
 A host can join and leave groups
 A host signals its desire to join or leave a multicast group by communicating with
its local router using a special protocol.
 In IPv4, the protocol is Internet Group Management Protocol (IGMP)
 In IPv6, the protocol is Multicast Listener Discovery (MLD)
IGMP OR MLD PROTOCOL
 Hosts communicate their desire to join / leave a multicast group to a router using
Internet Group Message Protocol (IGMP) in IPv4 or Multicast Listener Discovery
(MLD) in IPv6.
 Provides multicast routers with information about the membership status of hosts
connected to the network.
 Enables a multicast router to create and update list of loyal members for each
group.
MULTICAST ADDRESSING
 Multicast address is associated with a group, whose members are dynamic.
 Each group has its own IP multicast address.
 IP addresses reserved for multicasting are Class D in IPv4 (Class D 224.0.0.1to
[Link]), 1111 1111 prefix in IPv6.

 Hosts that are members of a group receive copy of the packet sent when
destination contains group address.
MULTICASTING VERSUS MULTIPLE UNICASTING
 Multicasting starts with a single packet from the source that is duplicated by the
routers. The destination address in each packet is the same for all duplicates.
 Only a single copy of the packet travels between any two routers.
  In multiple unicasting, several packets start from the source.
 If there are three destinations, for example, the source sends three packets, each with
a different unicast destination address.
 There may be multiple copies traveling between two routers

NEED FOR MULTICAST

Without support for multicast
 A source needs to send a separate packet with the identical data to each member of
the group
 Source needs to keep track of the IP address of each member in the group
Using IP multicast
 Sending host does not send multiple copies of the packet
 A host sends a single copy of the packet addressed to the group’s multicast
address
 The sending host does not need to know the individual unicast IP address of
each member
TYPES OF MULTICASTING
 Source-Specific Multicast - In source-specific multicast (one-to-many model),
receiver specifies multicast group and sender from which it is interested to
receive packets. Example: Internet radio broadcasts.
 Any Source Multicast - Supplements any source multicast (many-to-many
model).

MULTICAST APPLICATIONS
 Access to Distributed Databases
 Information Dissemination
 Teleconferencing.
 Distance Learning
MULTICAST ROUTING
 To support multicast, a router must additionally have multicast forwarding tables
that indicate, based on multicast address, which links to use to forward the multicast
packet.
 Unicast forwarding tables collectively specify a set of paths.
 Multicast forwarding tables collectively specify a set of trees - Multicast
distribution trees.
 Multicast routing is the process by which multicast distribution trees are
determined.
 To support multicasting, routers additionally build multicast forwarding tables.
 Multicast forwarding table is a tree structure, known as multicast distribution
 trees.
 Internet multicast is implemented on physical networks that support broadcasting
by extending forwarding functions.
MULTICAST DISTRIBUTION TREES
There are two types of Multicast Distribution Trees used in multicast routing.
They are
 Source-Based Tree: (DVMRP)
For each combination of (source, group), there is a shortest path spanning tree.
 Flood and prune
 Send multicast traffic everywhere
 Prune edges that are not actively subscribed to group
 Link-state
 Routers flood groups they would like to receive
 Compute shortest-path trees on demand
 Shared Tree (PIM)
 Single distributed tree shared among all sources
 Does not include its own topology discovery mechanism, but instead uses
routing information supplied by other routing protocols
 Specify rendezvous point (RP) for group
 Senders send packets to RP, receivers join at RP
 RP multicasts to receivers; Fix-up tree for optimization
 Rendezvous-Point Tree: one router is the center of the group and therefore the
root of the tree.
MULTICAST ROUTING PROTOCOLS
 Internet multicast is implemented on physical networks that support broadcasting
by extending forwarding functions.
 Major multicast routing protocols are:
1. Distance-Vector Multicast Routing Protocol (DVMRP)
2. Protocol Independent Multicast (PIM)
Distance Vector Multicast Routing Protocol

 The DVMRP, is a routing protocol used to share information between routers to

facilitate the transportation of IP multicast packets among networks.
 It formed the basis of the Internet's historic multicast backbone.
 Distance vector routing for unicast is extended to support multicast routing.
 Each router maintains a routing table for all destination through exchange of
distance vectors.
 DVMRP is also known as flood-and-prune protocol.
 DVMRP consists of two major components:
 A conventional distance-vector routing protocol, like RIP
 A protocol for determining how to forward multicast packets, based on
the routing table
 DVMRP router forwards a packet if
  The packet arrived from the link used to reach the source of the packet
 If downstream links have not pruned the tree

 DVMRP protocol uses the basic packet types as follows:

 The forwarding table of DVMRP is as follows:

 Multicasting is added to distance-vector routing in four stages.

 Flooding
 Reverse Path Forwarding (RPF)
 Reverse Path Broadcasting (RPB)
 Reverse Path Multicast (RPM)
Flooding
 Router on receiving a multicast packet from source S to a Destination from
NextHop, forwards the packet on all out-going links.
 Packet is flooded and looped back to S.
 The drawbacks are:
o It floods a network, even if it has no members for that group.
o Packets are forwarded by each router connected to a LAN, i.e., duplicate
flooding

Reverse Path Forwarding (RPF)

 RPF eliminates the looping problem in the flooding process.
 Only one copy is forwarded and the other copies are discarded.
 RPF forces the router to forward a multicast packet from one specific interface: the
one which has come through the shortest path from the source to the router.
  Packet is flooded but not looped back to S.

Reverse-Path Broadcasting (RPB)

 RPB does not multicast the packet, it broadcasts it.
 RPB creates a shortest path broadcast tree from the source to each destination.
 It guarantees that each destination receives one and only one copy of the
packet.
 We need to prevent each network from receiving more than one copy of the
packet.
 If a network is connected to more than one router, it may receive a copy of the
packet from each router.
 One router identified as parent called designated Router (DR).
 Only parent router forwards multicast packets from source S to the attached
network.
 When a router that is not the parent of the attached network receives a multicast
packet, it simply drops the packet.

Reverse-Path Multicasting (RPM)

 To increase efficiency, the multicast packet must reach only those networks that
have active members for that particular group.
  RPM adds pruning and grafting to RPB to create a multicast shortest path tree that
supports dynamic membership changes.

Pruning:
 Sent from routers receiving multicast traffic for which they have no active group
members
 “Prunes” the tree created by DVMRP
 Stops needless data from being sent

Grafting:
 Used after a branch has been pruned back
 Sent by a router that has a host that joins a multicast group
 Goes from router to router until a router active on the multicast group is
reached
 Sent for the following cases
 A new host member joins a group

A new dependent router joins a pruned branch

A dependent router restarts on a pruned branch

Protocol Independent Multicast (PIM)

 PIM divides multicast routing problem into sparse and dense mode.
 PIM sparse mode (PIM-SM) is widely used.
 PIM does not rely on any type of unicast routing protocol, hence protocol
independent.
 Routers explicitly join and leave multicast group using Join and Prune messages.
 One of the router is designated as rendezvous point (RP) for each group in a
domain to receive PIM messages.
  Multicast forwarding tree is built as a result of routers sending Join messages to
RP.
 Two types of trees to be constructed:
 Shared tree - used by all senders
 Source-specific tree - used only by a specific sending host
 The normal mode of operation creates the shared tree first, followed by one or more
source-specific trees

Shared Tree
 When a router sends Join message for group G to RP, it goes through a set of routers.
 Join message is wildcarded (*), i.e., it is applicable to all senders.
 Routers create an entry (*, G) in its forwarding table for the shared tree.
 Interface on which the Join arrived is marked to forward packets for that group.
 Forwards Join towards rendezvous router RP.
 Eventually, the message arrives at RP. Thus a shared tree with RP as root isformed.
Example
 Router R4 sends Join message for group G to rendezvous router RP.
 Join message is received by router R2. It makes an entry (*, G) in its table and forwards
the message to RP.
 When R5 sends Join message for group G, R2 does not forwards the Join. It adds an
outgoing interface to the forwarding table created for that group.

 As routers send Join message for a group, branches are added to the tree, i.e., shared.

 Multicast packets sent from hosts are forwarded to designated router RP.

 Suppose router R1, receives a message to group G. oR1 has no state for group G.
o Encapsulates the multicast packet in a Register message.
o Multicast packet is tunneled along the way to RP.
 RP decapsulates the packet and sends multicast packet onto the shared
tree, towards R2.
 R2 forwards the multicast packet to routers R4 and R5 that have members
for group G.
Source-Specific Tree
 RP can force routers to know about group G, by sending Join message to the sending
host, so that tunneling can be avoided.
 Intermediary routers create sender-specific entry (S, G) in their tables. Thus a
source-specific route from R1 to RP is formed.
 If there is high rate of packets sent from a sender to a group G, then shared-tree is
replaced by source-specific tree with sender as root.

Example

 Rendezvous router RP sends a Join message to the host router R1.

 Router R3 learns about group G through the message sent by RP.
 Router R4 send a source-specific Join due to high rate of packets from sender.
 Router R2 learns about group G through the message sent by R4.
 Eventually a source-specific tree is formed with R1 as root.

Analysis of PIM
 Protocol independent because, tree is based on Join messages via shortest path.
 Shared trees are more scalable than source-specific trees.
 Source-specific trees enable efficient routing than shared trees.

Metrics
The ARPANET was the testing ground for a number of different approaches to link-cost
calculation. The original ARPANET routing metric measured the number of packets that
were queued waiting to be transmitted on each link, meaning that a link with 10 packets
queued waiting to be transmitted was assigned a larger cost weight than a link with 5
packets queued for transmission.
Using queue length as a routing metric did not work well, however, since queue length
is an artificial measure of load—it moves packets toward the shortest queue rather than
toward the destination.
The original ARPANET routing mechanism suffered from the fact that it did not take
either the bandwidth or the latency of the link into consideration.
 A second version of the ARPANET routing algorithm, sometimes called the new
routing mechanism, took both link bandwidth and latency into consideration and used
delay, rather than just queue length, as a measure of load. This was done as follows. First,
each incoming packet was timestamped with its time of arrival at the router (ArrivalTime);
its departure time from the router (DepartTime) was also recorded. Second, when the link-
level ACK was received from the other side, the node computed the delay for that packet
as
Delay = (DepartTime − ArrivalTime) + TransmissionTime + Latency
where TransmissionTime and Latency were statically defined for the link.
DepartTime − ArrivalTime represents the amount of time the packet was delayed (queued)
in the node due to load. the ACK did not arrive, but instead the packet timed out, then
DepartTime was reset to the time the packet was retransmitted. In this case, DepartTime –
ArrivalTime captures the reliability of the link—the more frequent the retransmission of
packets, the less reliable the link. Under light load, it worked reasonably well, since the
two static factors of delay dominated the cost. Under heavy load, however, a congested
link would start to advertise a very high cost. This caused all the traffic to move off that
link, leaving it idle.
A third approach, called the “revised ARPANET routing metric,” addressed these
problems. The major changes were to compress the dynamic range of the metric
considerably, to account for the link type, and to smooth the variation of the metric with
time.
The smoothing was achieved by several mechanisms. First, the delay measurement
was transformed to a link utilization. Second, there was a hard limit on how much the
metric could change from one measurement cycle to the next. By smoothing the changes
in the cost, the likelihood that all nodes would abandon a route at once is greatly reduced.
The compression of the dynamic range was achieved by feeding the measured
utilization, the link type, and the link speed into a function. Observe the following:
 A highly loaded link never shows a cost of more than three times its cost when idle.
 The most expensive link is only seven times the cost of the least expensive.
 A high-speed satellite link is more attractive than a low-speed terrestrial link.
 Cost is a function of link utilization only at moderate to high loads.
The majority of real-world network deployments at the time of writing, metrics change
rarely if at all and only under the control of a network administrator. The reason for this is
partly that conventional wisdom now holds that dynamically changing metrics are too
unstable, even though this probably need not be true. Many networks today lack the great
disparity of link speeds and latencies that prevailed in the ARPANET. Thus, static metrics
are the norm. One common approach to setting metrics is to use a constant multiplied by
(1/link bandwidth).
THE GLOBAL INTERNET
Today’s Internet has hundreds of thousands of networks connected to it.
The global Internet is not just a random interconnection of Ethernets, but instead it takes
on a shape that reflects the fact that it interconnects many different organizations. Figure
4.1gives a simple depiction of the state of the Internet in 1990. Since that time, the
Internet’s topology has grown much more complex than this.
 One of the salient features of this topology is that it consists of end-user sites
(e.g., Stanford University) that connect to service provider networks (e.g., BARRNET was
a provider network that served sites in the San Francisco Bay Area). In 1990, many
providers served a limited geographic region and were thus known as regional networks.
The regional networks were, in turn, connected by a nationwide backbone. In 1990, this
backbone was funded by the National Science Foundation (NSF) and was therefore called
the NSFNET backbone.
The provider networks are typically built from a large number of point-to-point links
that connect to routers; similarly, each end-user site is typically not a single network but
instead consists of multiple physical networks connected by routers and bridges.
Each provider and end-user is likely to be an administratively independent entity.
This has some significant consequences on routing. Because of this independence, each
provider’s network is usually a single autonomous system (AS).
we need to deal with two related scaling issues. The first is the scalability of routing.
We need to find ways to minimize the number of network numbers that get carried around
in routing protocols and stored in the routing tables of routers. The second is address
utilization—that is, making sure that the IP address space does not get consumed too
quickly.
Routing Areas
Internet has a million networks. Routing table entries per router should be
minimized. Link state routing protocol is used to partition domain into areas.
An routing area is a set of routers configured to exchange link-state information.
Area introduces an additional level of hierarchy. Thus domains can grow without
burdening routing protocols.

There is one special area—the backbone area, also known as area 0.

 Routers R1, R2 and R3 are part of backbone area.
  Routers in backbone area are also part of non-backbone areas. Such routers are
known as Area Border Routers (ABR).
 Link-state advertisement is exchanged amongst routers in a non-backbone area.
 They do not see LSAs of other areas. For example, area 1 routers are not aware of
area 3 routers.
 ABR advertises routing information in their area to other ABRs.
 For example, R2 advertises area 2 routing information to R1 and R3, which in turn
pass onto their areas.
 All routers learn how to reach all networks in the domain.
 When a packet is to be sent to a network in another area, it goes through backbone
area via ABR and reaches the destination area.
 Routing Areas improve scalability but packets may not travel on the shortest path.
When dividing a domain into areas, the network administrator makes a tradeoff
between scalability and optimality of routing. The use of areas forces all packets traveling
from one area to another to go via the backbone area, even if a shorter path might have
been available.
INTER DOMAIN ROUTING
Internet is organized as autonomous systems (AS) each of which is under the control
of a single administrative entity.
 A corporation’s complex internal network might be a single AS, as may the network
of a single Internet Service Provider (ISP).
 Interdomain routing shares reachability information between autonomous systems.
 The basic idea behind autonomous systems is to provide an additional way to
hierarchically aggregate routing information in a large internet, thus improving scalability.
 Internet has backbone networks and sites. Providers connect at a peering point.

Traffic on the internet is of two types:

 Local Traffic - Traffic within an autonomous system is called local.
 Transit Traffic - Traffic that passes through an autonomous system is called transit.
Autonomous Systems (AS) are classified as:
 Stub AS - is connected to only one another autonomous system and carries local
traffic only (e.g. Small corporation).
 Multihomed AS - has connections to multiple autonomous systems but refuses to
carry transit traffic (e.g. Large corporation).
  Transit AS - has connections to multiple autonomous systems and is designed to
carry transit traffic (e.g. Backbone service provider).
Policies Used By Autonomous Systems :
 Provider-Customer―Provider advertises the routes it knows, to the customer and
advertises the routes learnt from customer to everyone.
 Customer-Provider―Customers want the routes to be diverted to them. So they
advertise their own prefixes and routes learned from customers to provider and
advertise routes learned from provider to customers.
 Peer―Two providers access to each other’s customers without having to pay.
CHALLENGES IN INTER-DOMAIN ROUTING PROTOCOL
 Each autonomous system has an intra-domain routing protocol, its own policy and
metric.
 Internet backbone must be able to route packets to the destination that complies with
policies of autonomous system along a loopless path.
 Service providers have trust deficit and may not trust advertisements by other AS,
or may refuse to carry traffic from other AS.
TYPES OF ROUTING PROTOCOLS

Two types of Routing Protocols are used in the Internet:

1) Intradomain routing
Routing within a single autonomous system
 Routing Information Protocol (RIP) - based on the distance-vector algorithm -
(REFER distance-vector routing algorithm)
 Open Shortest Path First (OSPF) - based on the link-state algorithm - (REFER link-
state routing algorithm)
2) Interdomain routing
Routing between autonomous systems.
 Border Gateway Protocol (BGP) - based on the path-vector algorithm - (REFER
Path Vector routing algorithm)
 UNIT – IV : TRANSPORT LAYER
Overview of Transport layer – TCP and UDP – Reliable byte stream
(TCP) – Connection management – Flow control – Retransmission –
TCP Congestion control – Congestion avoidance (DECbit, RED) – QoS –
Application requirements. RFC-TCP and UDP.

1. INTRODUCTION
 The transport layer is the fourth layer of the OSI model and is the core of the Internet
model.
 It responds to service requests from the session layer and issues service requests
tothe network Layer.
 The transport layer provides transparent transfer of data between hosts.
 It provides end-to-end control and information transfer with the quality of service
needed by the application program.
 It is the first true end-to-end layer, implemented in all End Systems (ES).

TRANSPORT LAYER FUNCTIONS / SERVICES

 The transport layer is located between the network layer and the application layer.
 The transport layer is responsible for providing services to the application layer; it
receives services from the network layer.
 The services that can be provided by the transport layer are
1. Process-to-Process Communication
2. Addressing : Port Numbers
3. Encapsulation and Decapsulation
4. Multiplexing and Demultiplexing
5. Flow Control
6. Error Control
7. Congestion Control
Flow Control
 Flow Control is the process of managing the rate of data transmission between two
nodes to prevent a fast sender from overwhelming a slow receiver.
 It provides a mechanism for the receiver to control the transmission speed, so that the
receiving node is not overwhelmed with data from transmitting node.

Congestion Control
 Congestion in a network may occur if the load on the network (the number of
packets sent to the network) is greater than the capacity of the network (the number
of packets a network can handle).
 Congestion control refers to the mechanisms and techniques that control the
congestion and keep the load below the capacity.
 Congestion Control refers to techniques and mechanisms that can either prevent
congestion, before it happens, or remove congestion, after it has happened
 Congestion control mechanisms are divided into two categories,
1. Open loop - prevent the congestion before it happens.
2. Closed loop - remove the congestion after it happens.

2. TRANSPORT LAYER PROTOCOLS

 Three protocols are associated with the Transport layer.

 They are
(1) UDP –User Datagram Protocol
(2) TCP – Transmission Control Protocol
(3) SCTP - Stream Control Transmission Protocol

 Each protocol provides a different type of service and should be used appropriately.

UDP - UDP is an unreliable connectionless transport-layer protocol used for its simplicity and
efficiency in applications where error control can be provided by the application-layer process.

TCP - TCP is a reliable connection-oriented protocol that can be used in any application where
reliability is important.

SCTP - SCTP is a new transport-layer protocol designed to combine some features of UDP and
TCP in an effort to create a better protocol for multimedia communication.
[Link] DATAGRAM PROTOCOL (UDP)

 User Datagram Protocol (UDP) is a connectionless, unreliable transport protocol.

 UDP adds process-to-process communication to best-effort service provided by IP.
 UDP is a very simple protocol using a minimum of overhead.
 UDP is a simple demultiplexer, which allows multiple processes on each host to
communicate.
 UDP does not provide flow control , reliable or ordered delivery.
 UDP can be used to send small message where reliability is not expected.
 Sending a small message using UDP takes much less interaction between the sender
and receiver.
 UDP allow processes to indirectly identify each other using an abstract locator called
port or mailbox

UDP PORTS
 Processes (server/client) are identified by an abstract locator known as port.
 Server accepts message at well known port.
 Some well-known UDP ports are 7–Echo, 53–DNS, 111–RPC, 161–SNMP, etc.
 < port, host > pair is used as key for demultiplexing.
 Ports are implemented as a message queue.
 When a message arrives, UDP appends it to end of the queue.
 When queue is full, the message is discarded.
 When a message is read, it is removed from the queue.
 When an application process wants to receive a message, one is removed from the
front of the queue.
  If the queue is empty, the process blocks until a message becomes available.
UDP DATAGRAM (PACKET) FORMAT
 UDP packets are known as user datagrams .
 These user datagrams, have a fixed-size header of 8 bytes made of four fields, each
of 2 bytes (16 bits).

Source Port Number

 Port number used by process on source host with 16 bits long.
 If the source host is client (sending request) then the port number is an temporary
one requested by the process and chosen by UDP.
 If the source is server (sending response) then it is well known port number.

Destination Port Number

 Port number used by process on Destination host with 16 bits long.
 If the destination host is the server (a client sending request) then the
port number is a well known port number.
 If the destination host is client (a server sending response) then port number
is an temporary one copied by server from the request packet.
Length
 This field denotes the total length of the UDP Packet (Header plus data)
 The total length of any UDP datagram can be from 0 to 65,535 bytes.

Checksum
 UDP computes its checksum over the UDP header, the contents of the message
body, and something called the pseudoheader.
 The pseudoheader consists of three fields from the IP header—protocol number,
source IP address, destination IP address plus the UDP length field.
Data
 Data field defines tha actual payload to be transmitted.
 Its size is variable.
UDP SERVICES
Process-to-Process Communication
 UDP provides process-to-process communication using socket addresses, a
combination of IP addresses and port numbers.

Connectionless Services
 UDP provides a connectionless service.
 There is no connection establishment and no connection termination .
 Each user datagram sent by UDP is an independent datagram.
 There is no relationship between the different user datagrams even if they are
 coming from the same source process and going to the same destination program.
 The user datagrams are not numbered.
 Each user datagram can travel on a different path.

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

Error Control
 There is no error control mechanism in UDP except for the checksum.
 This means that the sender does not know if a message has been lost or duplicated.
 When the receiver detects an error through the checksum, the user datagram is
silently discarded.

 The lack of error control means that the process using UDP should provide for this
service, if needed.

Checksum
 UDP checksum calculation includes three sections: a pseudoheader, the UDP header,
and the data coming from the application layer.
 The pseudoheader is the part of the header in which the user datagram is to be
encapsulated with some fields filled with 0s.

Optional Inclusion of Checksum

 The sender of a UDP packet can choose not to calculate the checksum.
 In this case, the checksum field is filled with all 0s before being sent.
 In the situation where the sender decides to calculate the checksum,
but it happens that the result is all 0s, the checksum is changed to all 1s
before the packet is sent.
 In other words, the sender complements the sum two times.

Congestion Control
 Since UDP is a connectionless protocol, it does not provide congestion control.
  UDP assumes that the packets sent are small and sporadic(occasionally or at irregular
intervals) and cannot create congestion in the network.
 This assumption may or may not be true, when UDP is used for interactive real-time
transfer of audio and video.

Encapsulation and Decapsulation

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

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

Multiplexing and Demultiplexing

 In a host running a transport protocol suite, there is only one UDP but possibly
several processes that may want to use the services of UDP.
 To handle this situation, UDP multiplexes and demultiplexes.

APPLICATIONS OF UDP
 UDP is used for management processes such as SNMP.
 UDP is used for route updating protocols such as RIP.
 UDP is a suitable transport protocol for multicasting. Multicasting capability is
embedded in the UDP software
 UDP is suitable for a process with internal flow and error control mechanisms such
as Trivial File Transfer Protocol (TFTP).
 UDP is suitable for a process that requires simple request-response communication
with little concern for flow and error control.
 UDP is normally used for interactive real-time applications that cannot tolerate
uneven delay between sections of a received message.

[Link] CONTROL PROTOCOL (TCP)

 TCP is a reliable, connection-oriented, byte-stream protocol.

 TCP guarantees the reliable, in-order delivery of a stream of bytes. It is a full-duplex
protocol, meaning that each TCP connection supports a pair of byte streams, one
flowing in each direction.
 TCP includes a flow-control mechanism for each of these byte streams that allow the
receiver to limit how much data the sender can transmit at a given time.
 TCP supports a demultiplexing mechanism that allows multiple application programs
on any given host to simultaneously carry on a conversation with their peers.
 TCP also implements congestion-control mechanism. The idea of this mechanism is
to prevent sender from overloading the network.
 Flow control is an end to end issue, whereas congestion control is concerned with how
 host and network interact.

TCP SERVICES
Process-to-Process Communication
 TCP provides process-to-process communication using port numbers.
Stream Delivery Service
 TCP is a stream-oriented protocol.
 TCP allows the sending process to deliver data as a stream of bytes and allows the
receiving process to obtain data as a stream of bytes.
 TCP creates an environment in which the two processes seem to be connected by an
imaginary ―tube that carries their bytes across the Internet.
 The sending process produces (writes to) the stream and the receiving process
consumes (reads from) it.

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

Multiplexing and Demultiplexing

TCP performs multiplexing at the sender and demultiplexing at the receiver.
Connection-Oriented Service
 TCP is a connection-oriented protocol.
 A connection needs to be established for each pair of processes.
 When a process at site A wants to send to and receive data from another
process at site B, the following three phases occur:
1. The two TCP’s establish a logical connection between them.
2. Data are exchanged in both directions.
3. The connection is terminated.
Reliable Service
 TCP is a reliable transport protocol.
 It uses an acknowledgment mechanism to check the safe and sound arrival of data.
TCP SEGMENT
 A packet in TCP is called a segment.
 Data unit exchanged between TCP peers are called segments.
 A TCP segment encapsulates the data received from the application layer.
 The TCP segment is encapsulated in an IP datagram, which in turn is encapsulated in
a frame at the data-link layer. 11
  TCP is a byte-oriented protocol, which means that the sender writes bytes into a TCP
connection and the receiver reads bytes out of the TCP connection.
 TCP does not, itself, transmit individual bytes over the Internet.
 TCP on the source host buffers enough bytes from the sending process to fill a
reasonably sized packet and then sends this packet to its peer on the destination host.
 TCP on the destination host then empties the contents of the packet into a receive
buffer, and the receiving process reads from this buffer at its leisure.
 TCP connection supports byte streams flowing in both directions.
 The packets exchanged between TCP peers are called segments, since each one
carries a segment of the byte stream.

TCP PACKET FORMAT

 Each TCP segment contains the header plus the data.
 The segment consists of a header of 20 to 60 bytes, followed by data from the
application program.
 The header is 20 bytes if there are no options and up to 60 bytes if it contains
options.

SrcPort and DstPort―port number of source and destination process.

SequenceNum―contains sequence number, i.e. first byte of data segment.
Acknowledgment― byte number of segment, the receiver expects next.
HdrLen―Length of TCP header as 4-byte words.
Flags― contains six control bits known as flags.
1. URG — segment contains urgent data.
 3. ACK — value of acknowledgment field is valid.
4. PUSH — sender has invoked the push operation.
5. RESET — receiver wants to abort the connection.
6. SYN — synchronize sequence numbers during connection establishment.
7. FIN — terminates the TCP connection.

Advertised Window―defines receiver’s window size and acts as flow control.

Checksum―It is computed over TCP header, Data, and pseudo header containing IP fields
(Length, SourceAddr & DestinationAddr).
UrgPtr ― used when the segment contains urgent data. It defines a value that must beadded
to the sequence number.
Options - There can be up to 40 bytes of optional information in the TCP header.

TCP CONNECTION MANAGEMENT

 TCP is connection-oriented.
 A connection-oriented transport protocol establishes a logical path between the
source and destination.
 All of the segments belonging to a message are then sent over this logical path.
 In TCP, connection-oriented transmission requires three phases:
Connection Establishment, Data Transfer and Connection Termination.

Connection Establishment
 While opening a TCP connection the two nodes(client and server) want to agree on a
set of parameters.
 The parameters are the starting sequence numbers that is to be used for their
respective byte streams.
 Connection establishment in TCP is a three-way handshaking.

1. Client sends a SYN segment to the server containing its initial sequence number (Flags
= SYN, SequenceNum = x)
2. Server responds with a segment that acknowledges client’s segment and specifies its
initial sequence number (Flags = SYN + ACK, ACK = x + 1 SequenceNum = y).
3. Finally, client responds with a segment that acknowledges server’s sequence number
(Flags = ACK, ACK = y + 1).
  The reason that each side acknowledges a sequence number that is one larger than
the one sent is that the Acknowledgment field actually identifies the ―next
sequence number expected
 A timer is scheduled for each of the first two segments, and if the expected
response is not received, the segment is retransmitted.

Data Transfer
 After connection is established, bidirectional data transfer can take place.
 The client and server can send data and acknowledgments in both directions.
 The data traveling in the same direction as an acknowledgment are carried on the
same segment.
 The acknowledgment is piggybacked with the data.

Connection Termination
 Connection termination or teardown can be done in two ways :
Three-way Close and Half-Close

Three-way Close—Both client and server close simultaneously.

 Client sends a FIN segment.

 The FIN segment can include last
chunk of data.
 Server responds with FIN + ACK
segment to inform its closing.
 Finally, client sends an ACK
segment
Half-Close—Client stops sending but receives data.
 Client half-closes the
connection by sending a FIN
segment.
 Server sends an ACK segment.
 Data transfer from client to the
server stops.
 After sending all data, server sends
FIN segment to client, which is
acknowledged by the client.
STATE TRANSITION DIAGRAM
 To keep track of all the different events happening during connection establishment,
connection termination, and data transfer, TCP is specified as the finite state machine(FSM).
 The transition from one state to another is shown using directed lines.
 States involved in opening and closing a connection is shown above and below
ESTABLISHED state respectively.
 States Involved in TCP :
Opening a TCP Connection
1. Server invokes a passive open on TCP, which causes TCP to move to LISTEN state
2. Client does an active open, which causes its TCP to send a SYN segment to the server
and move to SYN_SENT state.
3. When SYN segment arrives at the server, it moves to SYN_RCVD state and responds
with a SYN + ACK segment.
4. Arrival of SYN + ACK segment causes the client to move to ESTABLISHED state
and sends an ACK to the server.
5. When ACK arrives, the server finally moves to ESTABLISHED state.

Closing a TCP Connection

 Client / Server can independently close its half of the connection or simultaneously.
Transitions from ESTABLISHED to CLOSED state are:
 One side closes:
ESTABLISHED → FIN_WAIT_1 → FIN_WAIT_2 → TIME_WAIT → CLOSED
Other side closes:
ESTABLISHED → CLOSE_WAIT → LAST_ACK → CLOSED
Simultaneous close:
ESTABLISHED → FIN_WAIT_1 → CLOSING → TIME_WAIT → CLOSED

TCP FLOW CONTROL

 TCP uses a variant of sliding window known as adaptive flow control that:
o guarantees reliable delivery of data
o ensures ordered delivery of data
oenforces flow control at the sender
 Receiver advertises its window size to the sender using AdvertisedWindow field.
 Sender thus cannot have unacknowledged data greater than AdvertisedWindow.

16




Send Buffer
 Sending TCP maintains send buffer which contains 3 segments
(1) acknowledged data
(2) unacknowledged data
(3) data to be transmitted.
 Send buffer maintains three pointers
(1) LastByteAcked, (2) LastByteSent, and (3) LastByteWritten
such that:
LastByteAcked ≤ LastByteSent ≤ LastByteWritten
 A byte can be sent only after being written and only a sent byte can be
acknowledged.
 Bytes to the left of LastByteAcked are not kept as it had been acknowledged.
Receive Buffer
 Receiving TCP maintains receive buffer to hold data even if it arrives out-of-order.
 Receive buffer maintains three pointers namely
(1) LastByteRead, (2) NextByteExpected, and (3) LastByteRcvd
such that:
LastByteRead ≤ NextByteExpected ≤ LastByteRcvd + 1
 A byte cannot be read until that byte and all preceding bytes have been received.
 If data is received in order, then NextByteExpected = LastByteRcvd + 1
 Bytes to the left of LastByteRead are not buffered, since it is read by the application.

Flow Control in TCP

 Size of send and receive buffer is MaxSendBuffer and MaxRcvBuffer respectively.
 Sending TCP prevents overflowing of send buffer by maintaining
LastByteWritten − LastByteAcked ≤ MaxSendBuffer
 Receiving TCP avoids overflowing its receive buffer by maintaining
LastByteRcvd − LastByteRead ≤ MaxRcvBuffer
 Receiver throttles the sender by having AdvertisedWindow based on free space

available for buffering.

AdvertisedWindow = MaxRcvBuffer − ((NextByteExpected − 1) – LastByteRead)
 Sending TCP adheres to AdvertisedWindow by computing EffectiveWindow that
limits how much data it should send.
EffectiveWindow = AdvertisedWindow − (LastByteSent − LastByteAcked)
 When data arrives, LastByteRcvd moves to its right and AdvertisedWindow shrinks.
 Receiver acknowledges only, if preceding bytes have arrived.
 AdvertisedWindow expands when data is read by the application.
o If data is read as fast as it arrives then
AdvertisedWindow = MaxRcvBuffer
o If data is read slowly, it eventually leads to a AdvertisedWindow of size 0.
 AdvertisedWindow field is designed to allow sender to keep the pipe full.

TCP TRANSMISSION
 TCP has three mechanism to trigger the transmission of a segment.
  They are
o Maximum Segment Size (MSS) - Silly Window Syndrome
o Timeout - Nagle’s Algorithm

Silly Window Syndrome

 When either the sending application program creates data slowly or the receiving
application program consumes data slowly, or both, problems arise.
 Any of these situations results in the sending of data in very small segments, which
reduces the efficiency of the operation.
 This problem is called the silly window syndrome.
 The sending TCP may create a silly window syndrome if it is serving an application
program that creates data slowly, for example, 1 byte at a time.
 The application program writes 1 byte at a time into the buffer of the sending TCP.
 The result is a lot of 1-byte segments that are traveling through an internet.
 The solution is to prevent the sending TCP from sending the data byte by byte.
The sending TCP must be forced to wait and collect data to send in a larger block.

Nagle’s Algorithm
 If there is data to send but is less than MSS, then we may want to wait some amount
of time before sending the available data
 If we wait too long, then it may delay the process.
 If we don’t wait long enough, it may end up sending small segments resulting in
Silly Window Syndrome.
 The solution is to introduce a timer and to transmit when the timer expires
 Nagle introduced an algorithm for solving this problem.
TCP CONGESTION CONTROL
 Congestion occurs if load (number of packets sent) is greater than capacity of the
network (number of packets a network can handle).
 When load is less than network capacity, throughput increases proportionally.
 When load exceeds capacity, queues become full and the routers discard some
packets and throughput declines sharply.
 When too many packets are contending for the same link
o The queue overflows
o Packets get dropped
o Network is congested
 Network should provide a congestion control mechanism to deal with such a
situation.
 TCP maintains a variable called Congestion Window for each connection.
 TCP Congestion Control mechanisms are:

1. Additive Increase / Multiplicative Decrease (AIMD)

2. Slow Start
3. Fast Retransmit and Fast Recovery

Additive Increase / Multiplicative Decrease (AIMD)

 TCP source initializes CongestionWindow based on congestion level in the network.
 Source increases CongestionWindow when level of congestion goes down and
decreases the same when level of congestion goes up.
 TCP interprets timeouts as a sign of congestion and reduces the rate of transmission.
 On timeout, source reduces its CongestionWindow by half, i.e., multiplicative
decrease. For example, if CongestionWindow = 16 packets, after timeout it is 8.
 Value of CongestionWindow is never less than maximum segment size (MSS).
 When ACK arrives CongestionWindow is incremented marginally, i.e., additive
increase.
Increment = MSS × (MSS/CongestionWindow)
CongestionWindow += Increment
 For example, when ACK arrives for 1 packet, 2 packets are sent. When ACK for both
packets arrive, 3 packets are sent and so on.
 CongestionWindow increases and decreases throughout lifetime of the connection.
  When CongestionWindow is plotted as a function of time, a saw-tooth pattern
results.

Slow Start
 Slow start is used to increase CongestionWindow exponentially from a cold start.
 Source TCP initializes CongestionWindow to one packet.
 TCP doubles the number of packets sent every RTT on successful transmission.
 When ACK arrives for first packet TCP adds 1 packet to CongestionWindow and
sends two packets.
 When two ACKs arrive, TCP increments CongestionWindow by 2 packets and sends
four packets and so on.
 Instead of sending entire permissible packets at once (bursty traffic), packets are sent
in a phased manner, i.e., slow start.
 Initially TCP has no idea about congestion, henceforth it increases
CongestionWindow rapidly until there is a timeout. On timeout:
CongestionThreshold = CongestionWindow/ 2
CongestionWindow = 1
 Slow start is repeated until CongestionWindow reaches CongestionThreshold and
thereafter 1 packet per RTT.
  The congestion window trace will look like

Fast Retransmit And Fast Recovery

 TCP timeouts led to long periods of time during which the connection went dead while
waiting for a timer to expire.
 Fast retransmit is a heuristic approach that triggers retransmission of a dropped packet
sooner than the regular timeout mechanism. It does not replace regular timeouts.
 When a packet arrives out of order, receiving TCP resends the same acknowledgment
(duplicate ACK) it sent last time.
 When three duplicate ACK arrives at the sender, it infers that corresponding packet may
be lost due to congestion and retransmits that packet. This is called fast retransmit
 before regular timeout.
 When packet loss is detected using fast retransmit, the slow start phase is replaced by
additive increase, multiplicative decrease method. This is known as fast recovery.
 Instead of setting CongestionWindow to one packet, this method uses the ACKs that
are still in pipe to clock the sending of packets.
 Slow start is only used at the beginning of a connection and after regular timeout. At
other times, it follows a pure AIMD pattern.


 For example, packets 1 and 2 are received whereas packet 3 gets lost.
o Receiver sends a duplicate ACK for packet 2 when packet 4 arrives.
o Sender receives 3 duplicate ACKs after sending packet 6 retransmits packet 3.
o When packet 3 is received, receiver sends cumulative ACK up to packet 6.
 The congestion window trace will look like

TCP CONGESTION AVOIDANCE

 Congestion avoidance mechanisms prevent congestion before it actually occurs.
 These mechanisms predict when congestion is about to happen and then to reduce the
rate at which hosts send data just before packets start being discarded.
 TCP creates loss of packets in order to determine bandwidth of the connection.
 Routers help the end nodes by intimating when congestion is likely to occur.
 Congestion-avoidance mechanisms are:
o DEC bit - Destination Experiencing Congestion Bit
o RED - Random Early Detection

Dec Bit - Destination Experiencing Congestion Bit

 The first mechanism developed for use on the Digital Network Architecture (DNA).
 The idea is to evenly split the responsibility for congestion control between the
routers and the end nodes.
 Each router monitors the load it is experiencing and explicitly notifies the end nodes
when congestion is about to occur.
 This notification is implemented by setting a binary congestion bit in the packets that
flow through the router; hence the name DECbit.
  The destination host then copies this congestion bit into the ACK it sends back to the
source.
 The Source checks how many ACK has DEC bit set for previous window packets.
 If less than 50% of ACK have DEC bit set, then source increases its congestion
window by 1 packet
 Otherwise, decreases the congestion window by 87.5%.
 Finally, the source adjusts its sending rate so as to avoid congestion.
 Increase by 1, decrease by 0.875 rule was based on AIMD for stabilization.
 A single congestion bit is added to the packet header.
 Using a queue length of 1 as the trigger for setting the congestion bit.
 A router sets this bit in a packet if its average queue length is greater than or equal to1
at the time the packet arrives.
Computing average queue length at a router using DEC bit

 Average queue length is measured over a time interval that includes the
last busy + last idle cycle + current busy cycle.
 It calculates the average queue length by dividing the curve area with time interval.

Red - Random Early Detection

 The second mechanism of congestion avoidance is called as Random Early
Detection (RED).
 Each router is programmed to monitor its own queue length, and when it detects that
there is congestion, it notifies the source to adjust its congestion window.
 RED differs from the DEC bit scheme by two ways:
a. In DECbit, explicit notification about congestion is sent to source, whereas RED
implicitly notifies the source by dropping a few packets.

b. DECbit may lead to tail drop policy, whereas RED drops packet based on
drop probability in a random manner. Drop each arriving packet with some drop
probability whenever the queue length exceeds some drop level. This idea is
called early random drop.
Computation of average queue length using RED

 AvgLen = (1 − Weight) × AvgLen + Weight × SampleLen

where 0 < Weight < 1 and
SampleLen – is the length of the queue when a
sample measurement is made.
 The queue length is measured every time a new packet arrives at the gateway.

 RED has two queue length thresholds that trigger certain activity: MinThreshold and
MaxThreshold
 When a packet arrives at a gateway it compares Avglen with these two values
according to the following rules.
 Unit- 5
Application Layer and Case Studies
Electronic Mail-SMTP, POP3, IMAP, MIME– HTTP and HTTPS – Web
Services – DNS – SNMP- Wireshark Packet Capturing Tool - Cisco Packet
Tracer Tool - Open Source Network Tools - Manage Engine Tool.

HTTP (HYPERTEXT TRANSFER PROTOCOL)

 The HyperText Transfer Protocol (HTTP) is used to define how the client-
server programs can be written to retrieve web pages from the Web.
 It is a protocol used to access the data on the World Wide Web (WWW).
 The HTTP protocol can be used to transfer the data in the form of plain text,
hypertext, audio, video, and so on.
 HTTP is a stateless request/response protocol that governs client/server
communication.
 An HTTP client sends a request; an HTTP server returns a response.
 The server uses the port number 80; the client uses a temporary port number.
 HTTP uses the services of TCP , a connection-oriented and reliable protocol.
 HTTP is a text-oriented protocol. It contains embedded URL known as links.
  When hypertext is clicked, browser opens a new connection, retrieves file from the
server and displays the file.
 Each HTTP message has the general form
START_LINE <CRLF>
MESSAGE_HEADER <CRLF>
<CRLF> MESSAGE_BODY <CRLF>
where <CRLF> stands for carriage-return-line-feed.

Features of HTTP
o Connectionless protocol:
HTTP is a connectionless protocol. HTTP client initiates a request and waits
for a response from the server. When the server receives the request, the
server processes the request and sends back the response to the HTTP client
after which the client disconnects the connection. The connection between
client and server exist only during the current request and response time only.

o Media independent:
HTTP protocol is a media independent as data can be sent as long as both the
client and server know how to handle the data content. It is required for both
the client and server to specify the content type in MIME-type header.

o Stateless:
HTTP is a stateless protocol as both the client and server know each other
only during the current request. Due to this nature of the protocol, both the
client and server do not retain the information between various requests of
the web pages.

HTTP REQUEST AND RESPONSE MESSAGES

 The HTTP protocol defines the format of the request and response messages.

 Request Message: The request message is sent by the client that consists of a
request line, headers, and sometimes a body.
 Response Message: The response message is sent by the server to the client that
consists of a status line, headers, and sometimes a body.
HTTP REQUEST MESSAGE

 The first line in a request message is called a request line.

 After the request line, we can have zero or more request header lines.
 The body is an optional one. It contains the comment to be sent or the file to be
published on the website when the method is PUT or POST.

Request Line
 There are three fields in this request line - Method, URL and Version.
 The Method field defines the request types.
 The URL field defines the address and name of the corresponding web page.
 The Version field gives the version of the protocol; the most current version of
HTTP is 1.1.
 Some of the Method types are

Request Header
 Each request header line sends additional information from the client to the
server.
 Each header line has a header name, a colon, a space, and a header value.
 The value field defines the values associated with each header name.
 Headers defined for request message include
Body
 The body can be present in a request message. It is optional.
 Usually, it contains the comment to be sent or the file to be published on the
website when the method is PUT or POST.

Conditional Request
 A client can add a condition in its request.
 In this case, the server will send the requested web page if the condition is met or
inform the client otherwise.
 One of the most common conditions imposed by the client is the time and date the
web page is modified.
 The client can send the header line If-Modified-Since with the request to tell the
server that it needs the page only if it is modified after a certain point in time.

HTTP RESPONSE MESSAGE

 The first line in a request message is called a status line.

 After the request line, we can have zero or more response header lines.
 The body is an optional one. The body is present unless the response is an error
message

Status Line
 The Status line contains three fields - HTTP version , Status code, Status
phrase
 The first field defines the version of HTTP protocol, currently 1.1.
 The status code field defines the status of the request. It classifies the HTTP
result. It consists of three digits.
1xx–Informational, 2xx– Success, 3xx–Redirection,
4xx–Client error, 5xx–Server error
 The Status phrase field gives brief description about status code in text form.
 Some of the Status codes are
Response Header
 Each header provides additional information to the client.
 Each header line has a header name, a colon, a space, and a header value.
 Some of the response headers are:

Body
 The body contains the document to be sent from the server to the client.
 The body is present unless the response is an error message.

HTTP CONNECTIONS
 HTTP Clients and Servers exchange multiple messages over the same TCP
connection.
 If some of the objects are located on the same server, we have two choices: to
retrieve each object using a new TCP connection or to make a TCP connection and
retrieve them all.
 The first method is referred to as a non-persistent connection, the second as a
persistent connection.
 HTTP 1.0 uses non-persistent connections and HTTP 1.1 uses persistent
connections .

NON-PERSISTENT CONNECTIONS

 In a non-persistent connection, one TCP

connection is made for each
request/response.
 Only one object can be sent over a single
TCP connection
 The client opens a TCP connection and sends
a request.
 The server sends the response and closes the
connection.
 The client reads the data until it
encounters an end-of-file marker.
 It then closes the connection.
PERSISTENT CONNECTIONS

 HTTP version 1.1 specifies a persistent connection by

default.
 Multiple objects can be sent over a single TCP
connection.
 In a persistent connection, the server leaves the
connection open for more requests after sending a
response.
 The server can close the connection at the request of a
client or if a time-out has been reached.
 Time and resources are saved using persistent
connections. Only one set of buffers and variables needs
to be set for the connection at each site.
 The round trip time for connection establishment and
connection termination is saved.

HTTP COOKIES
 An HTTP cookie (also called web cookie, Internet cookie, browser cookie, or
simply cookie) is a small piece of data sent from a website and stored on the user's
computer by the user's web browser while the user is browsing.
 HTTP is stateless , Cookies are used to add State.
 Cookies were designed to be a reliable mechanism for websites to remember
stateful information (such as items added in the shopping cart in an online store) or
to record the user's browsing activity (including clicking particular buttons, logging
in, or recording which pages were visited in the past).
 They can also be used to remember arbitrary pieces of information that the user
previously entered into form fields such as names, addresses, passwords, and credit
card numbers.

Components of Cookie
A cookie consists of the following components:
1. Name
2. Value
3. Zero or more attributes (name/value pairs). Attributes store information such as
the cookie's expiration, domain, and flags
 Creating and Storing Cookies
The creation and storing of cookies depend on the implementation; however, the principle
is the same.
1. When a server receives a request from a client, it stores information about
the client in a file or a string. The information may include the domain name
of the client, the contents of the cookie (information the server has gathered
about the client such as name, registration number, and so on), a timestamp,
and other information depending on the implementation.
2. The server includes the cookie in the response that it sends to the client.
3. When the client receives the response, the browser stores the cookie in the
cookie directory, which is sorted by the server domain name.

Using Cookies
 When a client sends a request to a server, the browser looks in the cookie
directory to see if it can find a cookie sent by that server.
 If found, the cookie is included in the request.
 When the server receives the request, it knows that this is an old client, not a new
one.
 The contents of the cookie are never read by the browser or disclosed to the
user. It is a cookie made by the server and eaten by the server.

Types of Cookies
1. Authentication cookies
These are the most common method used by web servers to know whether the user is logged
in or not, and which account they are logged in with. Without such a mechanism, the site
would not know whether to send a page containing sensitive information, or require the user
to authenticate themselves by logging in.
2. Tracking cookies
These are commonly used as ways to compile individuals browsing histories.
3. Session cookie
A session cookie exists only in temporary memory while the user navigates the website. Web
browsers normally delete session cookies when the user closes the browser.
4. Persistent cookie
Instead of expiring when the web browser is closed as session cookies do, a persistent
cookie expires at a specific date or after a specific length of time. This means that, for the
cookie's entire lifespan , its information will be transmitted to the server every time the user
visits the website that it belongs to, or every time the user views a resource belonging to that
website from another website.
 HTTP CACHING
 HTTP Caching enables the client to retrieve document faster and reduces load on
the server.
 HTTP Caching is implemented at Proxy server, ISP router and Browser.
 Server sets expiration date (Expires header) for each page, beyond which it is not
cached.
 HTTP Cache document is returned to client only if it is an updated copy by
checking against If-Modified-Since header.
 If cache document is out-of-date, then request is forwarded to the server and
response is cached along the way.
 A web page will not be cached if no-cache directive is specified.

HTTP SECURITY
 HTTP does not provide security.
 However HTTP can be run over the Secure Socket Layer (SSL).
 In this case, HTTP is referred to as HTTPS.
 HTTPS provides confidentiality, client and server authentication, and data
integrity.

FTP (FILE TRANSFER PROTOCOL)

 FTP stands for File transfer protocol.
 FTP is a standard internet protocol provided by TCP/IP used for
transmitting the files from one host to another.
 It is mainly used for transferring the web page files from their creator to the
computer that acts as a server for other computers on the internet.
 It is also used for downloading the files to computer from other servers.
 Although we can transfer files using HTTP, FTP is a better choice to
transfer large files or to transfer files using different formats.

FTP OBJECTIVES
 It provides the sharing of files.
 It is used to encourage the use of remote computers.
 It transfers the data more reliably and efficiently.

FTP MECHANISM
 The above figure shows the basic model of the FTP.
 The FTP client has three components:
o user interface, control process, and data transfer process.
 The server has two components:
o server control process and server data transfer process.

FTP CONNECTIONS
 There are two types of connections in FTP -
Control Connection and Data Connection.
 The two connections in FTP have different lifetimes.
 The control connection remains connected during the entire interactive FTP
session.
 The data connection is opened and then closed for each file transfer activity.
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.
 FTP uses two well-known TCP ports:
o Port 21 is used for the control connection
o Port 20 is used for the data connection.

 Control Connection:
o The control connection uses very simple rules for communication.
o Through control connection, we can transfer a line of command or line of
response at a time.
o The control connection is made between the control processes.
o The control connection remains connected during the entire interactive FTP
session.

 Data Connection:
o The Data Connection uses very complex rules as data types may vary.
o The data connection is made between data transfer processes.
o The data connection opens when a command comes for transferring the
files and closes when the file is transferred.
 FTP COMMUNICATION
 FTP Communication is achieved through commands and responses.
 FTP Commands are sent from the client to the server
 FTP responses are sent from the server to the client.
 FTP Commands are in the form of ASCII uppercase, which may or may not be
followed by an argument.
 Some of the most common commands are

 Every FTP command generates at least one response.

 A response has two parts: a three-digit number followed by text.
 The numeric part defines the code; the text part defines needed parameter.

FTP FILE TYPE

 FTP can transfer one of the following file types across the data connection:
ASCII file, EBCDIC file, or image file.
 FTP DATA STRUCTURE
 FTP can transfer a file across the data connection using one of the following data
structure : file structure, record structure, or page structure.
 The file structure format is the default one and has no structure. It is a continuous
stream of bytes.
 In the record structure, the file is divided into records. This can be used only with
text files.
 In the page structure, the file is divided into pages, with each page having a page
number and a page header. The pages can be stored and accessed randomly or
sequentially.

FTP TRANSMISSION MODE

 FTP can transfer a file across the data connection using one of the following
three transmission modes: stream mode, block mode, or compressed mode.
 The stream mode is the default mode; data are delivered from FTP to TCP as a
continuous stream of bytes.
 In the block mode, data can be delivered from FTP to TCP in blocks.
 In the compressed mode, data can be compressed and delivered from FTP to
TCP.

FTP FILE TRANSFER

 File transfer occurs over the data connection under the control of the
commands sent over the control connection.
 File transfer in FTP means one of three things:
o retrieving a file (server to client)
o storing a file (client to server)
o directory listing (server to client).

FTP SECURITY
 FTP requires a password, the password is sent in plaintext which is
unencrypted. This means it can be intercepted and used by an attacker.
 The data transfer connection also transfers data in plaintext, which is insecure.
 To be secure, one can add a Secure Socket Layer between the FTP application
layer and the TCP layer.
 In this case FTP is called SSL-FTP.

EMAIL (SMTP, MIME, IMAP, POP)

 One of the most popular Internet services is electronic mail (E-mail).
 Email is one of the oldest network applications.
 The three main components of an Email are
1. User Agent (UA)
2. Messsage Transfer Agent (MTA) – SMTP
3. Messsage Access Agent (MAA) - IMAP , POP
  When the sender and the receiver of an e-mail are on the same system, we need only
two User Agents and no Message Transfer Agent
 When the sender and the receiver of an e-mail are on different system, we need two
UA, two pairs of MTA (client and server), and two MAA (client and server).

WORKING OF EMAIL

 When Alice needs to send a message to Bob, she runs a UA program to prepare the
message and send it to her mail server.
 The mail server at her site uses a queue (spool) to store messages waiting to be sent.
The message, however, needs to be sent through the Internet from Alice’s site to
Bob’s site using an MTA.
 Here two message transfer agents are needed: one client and one server.
 The server needs to run all the time because it does not know when a client will
ask for a connection.
 The client can be triggered by the system when there is a message in the queue to
be sent.
 The user agent at the Bob site allows Bob to read the received message.
 Bob later uses an MAA client to retrieve the message from an MAA server
running on the second server.
USER AGENT (UA)
 The first component of an electronic mail system is the user agent (UA).
 It provides service to the user to make the process of sending and receiving a
message easier.
 A user agent is a software package that composes, reads, replies to, and
forwards messages. It also handles local mailboxes on the user computers.

 There are two types of user agents: Command-driven and GUI-based.

Command driven
o Command driven user agents belong to the early days of electronic mail.
o A command-driven user agent normally accepts a one character command from
the keyboard to perform its task.
o Some examples of command driven user agents are mail, pine, and elm.

GUI-based
o Modern user agents are GUI-based.
o They allow the user to interact with the software by using both the keyboard and
the mouse.
o They have graphical components such as icons, menu bars, and windows that
make the services easy to access.
o Some examples of GUI-based user agents are Eudora and Outlook.

MESSAGE TRANSFER AGENT (MTA)

 The actual mail transfer is done through message transfer agents (MTA).
 To send mail, a system must have the client MTA, and to receive mail, a
system must have a server MTA.
 The formal protocol that defines the MTA client and server in the Internet is
called Simple Mail Transfer Protocol (SMTP).

MESSAGE ACCESS AGENT (MAA)

 MAA is a software that pulls messages out of a mailbox.
 POP3 and IMAP4 are examples of MAA.
ADDRESS FORMAT OF EMAIL
 E-mail address is userid @ domain where domain is hostname of the mail
server.

MESSAGE FORMAT OF EMAIL

 Email message consists of two parts namely header and body.
 Each header line contains type and value separated by a colon (:).
 Some header contents are:
o From: identifier sender of the message.
o To: mail address of the recipient(s).
o Subject: says about purpose of the message.
o Date: timestamp of when the message was transmitted.
 Header is separated from the body by a blank line.
 Body contains the actual message.

 Email was extended in 1993 to carry many different types of data: audio,
video, images, Word documents, and so on.
 This extended version is known as MIME(Multipurpose Mail Extension).

SIMPLE MAIL TRANSFER PROTOCOL (SMTP)

 SMTP is the standard protocol for transferring mail between hosts in the
TCP/IP protocol suite.
 SMTP is not concerned with the format or content of messages themselves.
 SMTP uses information written on the envelope of the mail (message header), but
does not look at the contents (message body) of the envelope.
 SMTP clients and servers have two main components
o User Agents(UA) – Prepares the message, encloses it in an envelope.
o Mail Transfer Agent (MTA) – Transfers the mail across the internet

 SMTP also allows the use of Relays allowing other MTAs to relay the mail.
 SMTP MAIL FLOW

 To begin, mail is created by a user-agent program in response to user input.

 Each created message consists of a header that includes the recipient's email address
and other information, and a message body containing the message to be sent.
 These messages are then queued in some fashion and provided as input to an SMTP
Sender program.

SMTP COMMANDS AND RESPONSES

 The operation of SMTP consists of a series of commands and responses
exchanged between the SMTP sender and SMTP receiver.
 The initiative is with the SMTP sender, who establishes the TCP connection.
 Once the connection is established, the SMTP sender sends commands over the
connection to the receiver.
 The command is from an MTA client to an MTA server; the response is from an
MTA server to the MTA client.

SMTP Commands
 Commands are sent from the client to the server. It consists of a keyword
followed by zero or more arguments. SMTP defines 14 commands.
 SMTP Responses
 Responses are sent from the server to the client.
 A response is a three digit code that may be followed by additional textual
information.
 SMTP OPERATIONS
Basic SMTP operation occurs in three phases:
1. Connection Setup
2. Mail Transfer
3. Connection Termination

Connection Setup
 An SMTP sender will attempt to set up a TCP connection with a target host
when it has one or more mail messages to deliver to that host.
 The sequence is quite simple:
1. The sender opens a TCP connection with the receiver.
2. Once the connection is established, the receiver identifies itself with
"Service Ready”.
3. The sender identifies itself with the HELO command.
4. The receiver accepts the sender's identification with "OK".
5. If the mail service on the destination is unavailable, the destination host
returns a "Service Not Available" reply in step 2, and the process is
terminated.

Mail Transfer
 Once a connection has been established, the SMTP sender may send one or
more messages to the SMTP receiver.
 There are three logical phases to the transfer of a message:
1. A MAIL command identifies the originator of the message.
2. One or more RCPT commands identify the recipients for this
message.
3. A DATA command transfers the message text.
 Connection Termination
 The SMTP sender closes the connection in two steps.
 First, the sender sends a QUIT command and waits for a reply.
 The second step is to initiate a TCP close operation for the TCP connection.
 The receiver initiates its TCP close after sending its reply to the QUIT
command.

LIMITATIONS OF SMTP
 SMTP cannot transmit executable files or other binary objects.
 SMTP cannot transmit text data that includes national language characters, as these
are represented by 8-bit codes with values of 128 decimal or higher, and SMTP is
limited to 7-bit ASCII.
 SMTP servers may reject mail message over a certain size.
 SMTP gateways that translate between ASCII and the character code EBCDIC do
not use a consistent set of mappings, resulting in translation problems.
 Some SMTP implementations do not adhere completely to the SMTP standards
defined.
 Common problems include the following:
1. Deletion, addition, or recording of carriage return and linefeed.
2. Truncating or wrapping lines longer than 76 characters.
3. Removal of trailing white space (tab and space characters).
4. Padding of lines in a message to the same length.
5. Conversion of tab characters into multiple-space characters.

MULTIPURPOSE INTERNET MAIL EXTENSION (MIME)

 SMTP provides a basic email service, while MIME adds multimedia capability to
SMTP.
 MIME is an extension to SMTP and is used to overcome the problems and limitations of
SMTP.
 Email system was designed to send messages only in ASCII format.
 Languages such as French, Chinese, etc., are not supported.
 Image, audio and video files cannot be sent.
 MIME adds the following features to email service:
 Be able to send multiple attachments with a single message;
Unlimited message length;
Use of character sets other than ASCII code;
Use of rich text (layouts, fonts, colors, etc)
Binary attachments (executables, images, audio or video files, etc.), which
may be divided if needed.
 MIME is a protocol that converts non-ASCII data to 7-bit NVT(Network
Virtual Terminal) ASCII and vice-versa.

MIME HEADERS
 Using headers, MIME describes the type of message content and the encoding
used.
 Headers defined in MIME are:
 MIME-Version- current version, i.e., 1.1
 Content-Type - message type (text/html, image/jpeg, application/pdf)
 Content-Transfer-Encoding - message encoding scheme (eg base64).
 Content-Id - unique identifier for the message.
 Content-Description - describes type of the message body.
MIME CONTENT TYPES
 There are seven different major types of content and a total of 14 subtypes.
 In general, a content type declares the general type of data, and the subtype
specifies a particular format for that type of data.
 MIME also defines a multipart type that says how a message carrying more
than one data type is structured.
 This is like a programming language that defines both base types (e.g., integers
and floats) and compound types (e.g., structures and arrays).
 One possible multipart subtype is mixed, which says that the message contains a
set of independent data pieces in a specified order.
 Each piece then has its own header line that describes the type of that piece.
 The table below lists the MIME content types:

ENCODING FORMATS OF MIME

 MIME uses various encoding formats to convert binary data into the ASCII
character set.
 To transfer binary data, MIME offers five encoding formats which can be used in
the header transfer-encoding:
 7-bit : 7-bit text format (for messages without accented characters);
 8-bit : 8-bit text format;
 quoted-printable : Quoted-Printable format, recommended for messages
which use a 7-bit alphabet (such as when there are accent marks);
 base-64 : Base 64, for sending binary files as attachments;
 binary : binary format; not recommended.
 Since MIME is very open, it can use third-party encoding formats such as:
 BinHex : A proprietary format belonging to Apple
 Uuencode : for UNIX-to-UNIX encoding
 Xencode : for binary-to-text encoding
MESSAGE TRANSFER IN MIME

 MTA is a mail daemon (sendmail) active on hosts having mailbox, used to send an
email.
 Mail passes through a sequence of gateways before it reaches the recipient mail
server.
 Each gateway stores and forwards the mail using Simple mail transfer protocol
(SMTP).
 SMTP defines communication between MTAs over TCP on port 25.
 In an SMTP session, sending MTA is client and receiver is server. In each
exchange:
 Client posts a command (HELO, MAIL, RCPT, DATA, QUIT, VRFY, etc.)
 Server responds with a code (250, 550, 354, 221, 251 etc) and an explanation.
 Client is identified using HELO command and verified by the server
 Client forwards message to server, if server is willing to accept.
 Message is terminated by a line with only single period (.) in it.
 Eventually client terminates the connection.

IMAP (INTERNET MAIL ACCESS PROTOCOL)

 IMAP is an Application Layer Internet protocol that allows an e-mail client to
access e-mail on a remote mail server.
 It is a method of accessing electronic mail messages that are kept on a possibly
shared mail server.
 IMAP is a more capable wire protocol.
 IMAP is similar to SMTP in many ways.
 IMAP is a client/server protocol running over TCP on port 143.
 IMAP allows multiple clients simultaneously connected to the same mailbox, and
through flags stored on the server, different clients accessing the same mailbox at
the same or different times can detect state changes made by other clients.
 In other words, it permits a "client" email program to access remote message stores
as if they were local.
 For example, email stored on an IMAP server can be manipulated from a desktop
computer at home, a workstation at the office, and a notebook computer while
travelling, without the need to transfer messages or files back and forth between
these computers.
 IMAP can support email serving in three modes:
  Offline
 Online
Users may connect to the server, look at what email is available, and
access it online. This looks to the user very much like having local
spool files, but they’re on the mail server.
 Disconnected operation
A mail client connects to the server, can make a “cache” copy of
selected messages, and disconnects from the server. The user can then
work on the messages offline, and connect to the server later and
resynchronize the server status with the cache.

OPERATION OF IMAP
 The mail transfer begins with the client authenticating the user and identifying the
mailbox they want to access.
 Client Commands
LOGIN, AUTHENTICATE, SELECT, EXAMINE, CLOSE, and LOGOUT
 Server Responses
OK, NO (no permission), BAD (incorrect command),
 When user wishes to FETCH a message, server responds in MIME format.
 Message attributes such as size are also exchanged.
 Flags are used by client to report user actions.
SEEN, ANSWERED, DELETED, RECENT

IMAP4
 The latest version is IMAP4. IMAP4 is more powerful and more complex.
 IMAP4 provides the following extra functions:
 A user can check the e-mail header prior to downloading.
 A user can search the contents of the 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.

ADVANTAGES OF IMAP
 With IMAP, the primary storage is on the server, not on the local machine.
 Email being put away for storage can be foldered on local disk, or can be
foldered on the IMAP server.
 The protocol allows full user of remote folders, including a remote folder
hierarchy and multiple inboxes.
 It keeps track of explicit status of messages, and allows for user-defined status.
 Supports new mail notification explicitly.
 Extensible for non-email data, like netnews, document storage, etc.
 Selective fetching of individual MIME body parts.
 Server-based search to minimize data transfer.
 Servers may have extensions that can be negotiated.

POST OFFICE PROTOCOL (POP3)

 Post Office Protocol (POP3) is an application-layer Internet standard protocol used
by local e-mail clients to retrieve e-mail from a remote server over a TCP/IP
connection.
 There are two versions of POP.
• The first, called POP2, became a standard in the mid-80's and requires SMTP
to send messages.
• The current version, POP3, can be used with or without SMTP. POP3 uses
TCP/IP port 110.
 POP is a much simpler protocol, making implementation easier.
 POP supports offline access to the messages, thus requires less internet usage time
 POP does not allow search facility.
 In order to access the messages, it is necessary to download them.
 It allows only one mailbox to be created on server.
 It is not suitable for accessing non mail data.
 POP mail moves the message from the email server onto the local computer,
although there is usually an option to leave the messages on the email server as well.
 POP treats the mailbox as one store, and has no concept of folders.
 POP works in two modes namely, delete and keep mode.
 In delete mode, mail is deleted from the mailbox after retrieval. The delete mode
is normally used when the user is working at their permanent computer and can
save and organize the received mail after reading or replying.
 In keep mode, mail after reading is kept in mailbox for later retrieval. The keep
mode is normally used when the user accesses her mail away from their primary
computer .
  POP3 client is installed on the recipient computer and POP server on the mail
server.
 Client opens a connection to the server using TCP on port 110.
 Client sends username and password to access mailbox and to retrieve
messages.

POP3 Commands
POP commands are generally abbreviated into codes of three or four letters The
following describes some of the POP commands:
1. UID - This command opens the connection
2. STAT - It is used to display number of messages currently in the mailbox
3. LIST - It is used to get the summary of messages
4. RETR -This command helps to select a mailbox to access the messages
5. DELE - It is used to delete a message
6. RSET - It is used to reset the session to its initial state
7. QUIT - It is used to log off the session

DIFFERENCE BETWEEN POP AND IMAP

SNo. POP IMAP

1 Generally used to support single Designed to handle multiple clients.

client.
2 Messages are accessed offline. Messages are accessed online
although it also supports offline mode.
3 POP does not allow search facility. IMAP offers ability to search emails.
4 All the messages have to be It allows selective transfer of messages to
downloaded. the client.
5 Only one mailbox can be created on Multiple mailboxes can be created on the
the server. server.
6 Not suitable for accessing non-mail Suitable for accessing non-mail data i.e.
data. attachment.
 POP commands are generally IMAP commands are not abbreviated,
7 abbreviated into codes of three or four they are full. Eg. STATUS.
letters. Eg. STAT.
8 It requires minimum use of server Clients are totally dependent on server.
resources.
9 Mails once downloaded cannot be Allows mails to be accessed from
accessed from some other location. multiple locations.
10 The e-mails are not downloaded Users can view the headings and sender
automatically. of e-mails and then decide to download.
11 POP requires less internet usage time. IMAP requires more internet usage time.

Advantages of IMAP over POP

 IMAP is more powerful and more complex than POP.
 User can check the e-mail header prior to downloading.
 User can search e-mail for a specific string of characters prior to downloading.
 User can download partially, very useful in case of limited bandwidth.
 User can create, delete, or rename mailboxes on the mail server.

DNS (DOMAIN NAME SYSTEM)

 Domain Name System was designed in 1984.

 DNS is used for name-to-address mapping.
 The DNS provides the protocol which allows clients and servers to communicate
with each other.
 Eg: Host name like [Link] is translated into numerical IP addresses like
[Link]
 Domain Name System (DNS) is a distributed database used by TCP/IP applications
to map between hostnames and IP addresses and to provide electronic mail routing
information.
 Each site maintains its own database of information and runs a server program that
other systems across the Internet can query.
WORKING OF DNS
The following six steps shows the working of a DNS. It maps the host name to an IP
address:
1. The user passes the host name to the file transfer client.
2. The file transfer client passes the host name to the DNS client.
3. Each computer, after being booted, knows the address of one DNS server. The DNS
client sends a message to a DNS server with a query that gives the file transfer server
name using the known IP address of the DNS server.
4. The DNS server responds with the IP address of the desired file transfer server.
5. The DNS server passes the IP address to the file transfer client.
6. The file transfer client now uses the received IP address to access the file transfer
server.

NAME SPACE
 To be unambiguous, the names assigned to machines must be carefully selected
from a name space with complete control over the binding between the names and
IP address.
 The names must be unique because the addresses are unique.
 A name space that maps each address to a unique name can be organized in
two ways: flat (or) hierarchical.

Flat Name Space

 In a flat name space, a name is assigned to an address.
 A name in this space is a sequence of characters without structure.
 The main disadvantage of a flat name space is that it cannot be used in a
large system such as Internet because it must be centrally controlled to avoid
ambiguity and duplication.

Hierarchical Name Space

 In a hierarchical name space, each name is made of several parts.
 The first part can define the organization, the second part can define the name, the
third part can define departments, and so on.
 In this case, the authority to assign and control the name spaces can be
decentralized.
 A central authority can assign the part of the name that defines the nature of the
organization and the name.
 The responsibility for the rest of the name can be given to the organization
itself. Suffixes can be added to the name to define host or resources.
 The management of the organization need not worry that the prefix chosen for a
host is taken by another organization because even if part of an address is the same,
the whole address is different.
 The names are unique without the need to be assigned by a central authority.
 The central authority controls only part of the name, not the whole name.
DOMAIN NAME SPACE
 To have a hierarchical name space, a domain name space was designed. In this
design, the names are defined in an inverted-tree structure with the root at the top.
 Each node in the tree has a label, which is a string with a maximum of 63 characters.
 The root label is a null string.
 DNS requires that children of a node have different labels, which guarantees the
uniqueness of the domain names.

 Each node in the tree has a label, which is a string with a maximum of 63 characters.
 The root label is a null string (empty string). DNS requires that children of a node
(nodes that branch from the same node) have different labels, which guarantees the
uniqueness of the domain names.

Domain Name
 Each node in the tree has a label called as domain name.
 A full domain name is a sequence of labels separated by dots (.)
 The domain names are always read from the node up to the root.
 The last label is the label of the root (null).
 This means that a full domain name always ends in a null label, which
means the last character is a dot because the null string is nothing.
 If a label is terminated by a null string, it is called a fully qualified domain
name (FQDN).
 If a label is not terminated by a null string, it is called a partially qualified
domain name (PQDN).
Domain
 A domain is a subtree of the domain name space.
 The name of the domain is the domain name of the node at the top of the sub-
tree.
 A domain may itself be divided into domains.

DISTRIBUTION OF NAME SPACE

 The information contained in the domain name space must be stored.
 But it is very inefficient and also not reliable to have just one computer store such
a huge amount of information.
 It is inefficient because responding to requests from all over the world, places a
heavy load on the system.
 It is not reliable because any failure makes the data inaccessible.
 The solution to these problems is to distribute the information among many
computers called DNS servers.

HIERARCHY OF NAME SERVERS

 The way to distribute information among DNS servers is to 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 first level
nodes.
 Because a domain created this way could be very large,
 DNS allows domains to be divided further into smaller domains.
 Thus we have a hierarchy of servers in the same way that we have a hierarchy of
names.
ZONE
 What a server is responsible for, or has authority over, is called a zone.
 The server makes a database called a zone file and keeps all the information for
every node under that domain.
 If a server accepts responsibility for a domain and does not divide the domains into
smaller domains, the domain and zone refer to the same thing.
 But if a server divides its domain into sub domains and delegates parts of its
authority to other servers, domain and zone refer to different things.
 The information about the nodes in the sub domains is stored in the servers at the
lower levels, with the original server keeping some sort of references to these lower
level servers.
 But still, the original server does not free itself from responsibility totally.
 It still has a zone, but the detailed information is kept by the lower level servers.

ROOT SERVER
 A root sever is a server whose zone consists of the whole tree.
 A root server usually does not store any information about domains but
delegates its authority to other servers, keeping references to those servers.
 Currently there are more than 13 root servers, each covering the whole domain
name space.
 The servers are distributed all around the world.

PRIMARY AND SECONDARY SERVERS

 DNS defines two types of servers: primary and secondary.
 A Primary Server is a server that stores a file about the zone for which it is an
authority.
 Primary Servers are responsible for creating, maintaining, and updating the
zone file.
 Primary Server stores the zone file on a local disc.
 A secondary server is a server that transfers the complete information about a zone
from another server (Primary or Secondary) and stores the file on its local disc.
 If updating is required, it must be done by the primary server, which sends the
updated version to the secondary.
  A primary server loads all information from the disk file; the secondary server
loads all information from the primary server.

DNS IN THE INTERNET

 DNS is a protocol that can be used in different platforms.
 In the Internet, the domain name space (tree) is divided into three different
sections - Generic domains, Country domains, and Inverse domain.

Generic Domains
 The generic domains define 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.
 The first level in the generic domains section allows seven possible three
character levels.
 These levels describe the organization types as listed in following table.

Country Domains
 The country domains section follows the same format as the generic domains but
uses two characters for country abbreviations
 E.g.; in for India, us for United States etc) in place of the three character
organizational abbreviation at the first level.
 Second level labels can be organizational, or they can be more specific,
national designation.
 India for example, uses state abbreviations as a subdivision of the country
domain us. (e.g., [Link].)

Inverse Domains
 Mapping an address to a name is called Inverse domain.
  The client can send an IP address to a server to be mapped to a domain name and it
is called PTR(Pointer) query.
 To answer queries of this kind, DNS uses the inverse domain

DNS RESOLUTION
 Mapping a name to an address or an address to a name is called name address
resolution.
 DNS is designed as a client server application.
 A host that needs to map an address to a name or a name to an address calls a DNS
client named a Resolver.
 The Resolver accesses the closest DNS server with a mapping request.
 If the server has the information, it satisfies the resolver; otherwise, it either refers
the resolver to other servers or asks other servers to provide the information.
 After the resolver receives the mapping, it interprets the response to see if it is a real
resolution or an error and finally delivers the result to the process that requested it.
 A resolution can be either recursive or iterative.

Recursive Resolution

 The application program on the source host calls the DNS resolver (client) to find
the IP address of the destination host. The resolver, which does not know this
address, sends the query to the local DNS server of the source (Event 1)
 The local server sends the query to a root DNS server (Event 2)
 The Root server sends the query to the top-level-DNS server(Event 3)
 The top-level DNS server knows only the IP address of the local DNS server at the
destination. So it forwards the query to the local server, which knows the IP address
of the destination host (Event 4)
 The IP address of the destination host is now sent back to the top-level DNS
server(Event 5) then back to the root server (Event 6), then back to the source DNS
server, which may cache it for the future queries (Event 7), and finally back to the
source host (Event 8).
Iterative Resolution

 In iterative resolution, each server that does not know the mapping, sends the IP
address of the next server back to the one that requested it.
 The iterative resolution takes place between two local servers.
 The original resolver gets the final answer from the destination local server.
 The messages shown by Events 2, 4, and 6 contain the same query.
 However, the message shown by Event 3 contains the IP address of the top-
level domain server.
 The message shown by Event 5 contains the IP address of the destination local
DNS server
 The message shown by Event 7 contains the IP address of the destination.
 When the Source local DNS server receives the IP address of the destination, it
sends it to the resolver (Event 8).

DNS CACHING
 Each time a server receives a query for a name that is not in its domain, it needs to
search its database for a server IP address.
 DNS handles this with a mechanism called caching.
 When a server asks for a mapping from another server and receives the response, it
stores this information in its cache memory before sending it to the client.
 If the same or another client asks for the same mapping, it can check its cache
memory and resolve the problem.
 However, to inform the client that the response is coming from the cache memory
and not from an authoritative source, the server marks the response as
unauthoritative.
 Caching speeds up resolution. Reduction of this search time would increase
efficiency, but it can also be problematic.
 If a server caches a mapping for a long time, it may send an outdated mapping to
the client.
 To counter this, two techniques are used.
 First, the authoritative server always adds information to the mapping called
time to live (TTL). It defines the time in seconds that the receiving server
can cache the information. After that time, the mapping is invalid and any
query must be sent again to the authoritative server.
  Second, DNS requires that each server keep a TTL counter for each mapping it
caches. The cache memory must be searched periodically and those mappings with
an expired TTL must be purged.

DNS RESOURCE RECORDS (RR)

 The zone information associated with a server is implemented as a set of
resource records.
 In other words, a name server stores a database of resource records.
 A resource record is a 5-tuple structure :
(Domain Name, Type, Class, TTL, Value)
 The domain name identifies the resource record.
 The type defines how the value should be interpreted.
 The value defines the information kept about the domain name.
 The TTL defines the number of seconds for which the information is valid.
 The class defines the type of network

Types of Resource Records

DNS MESSAGES
 DNS has two types of messages: query and response.
 Both types have the same format.
 The query message consists of a header and question section.
 The response message consists of a header, question section, answer section,
authoritative section, and additional section .

 Header
 Both query and response messages have the same header format with
some fields set to zero for the query messages.
 The header fields are as follows:
  The identification field is used by the client to match the response with the
query.
 The flag field defines whether the message is a query or response. It also includes
status of error.
 The next four fields in the header define the number of each record type in
the message.
 Question Section
 The question section consists of one or more question records. It is
present in both query and response messages.
 Answer Section
 The answer section consists of one or more resource records. It is
present only in response messages.
 Authoritative Section
 The authoritative section gives information (domain name) about one or
more authoritative servers for the query.
 Additional Information Section
 The additional information section provides additional information that
may help the resolver.

DNS CONNECTIONS
 DNS can use either UDP or TCP.
 In both cases the well-known port used by the server is port 53.
 UDP is used when the size of the response message is less than 512 bytes
because most UDP packages have a 512-byte packet size limit.
 If the size of the response message is more than 512 bytes, a TCP connection is
used.
DNS REGISTRARS
 New domains are added to DNS through a registrar. A fee is charged.
 A registrar first verifies that the requested domain name is unique and then
enters it into the DNS database.
 Today, there are many registrars; their names and addresses can be found at
[Link]
 To register, the organization needs to give the name of its server and the IP
address of the server.
 For example, a new commercial organization named wonderful with a server
named ws and IP address [Link], needs to give the following
information to one of the registrars:
Domain name: [Link] IP address: [Link]
DDNS (DYNAMIC DOMAIN NAME SYSTEM)
 In DNS, when there is a change, such as adding a new host, removing a host, or
changing an IP address, the change must be made to the DNS master file.
 The DNS master file must be updated dynamically.
 The Dynamic Domain Name System (DDNS) is used for this purpose.
 In DDNS, when a binding between a name and an address is determined, the
information is sent to a primary DNS server.
 The primary server updates the zone.
 The secondary servers are notified either actively or passively.
 In active notification, the primary server sends a message to the secondary servers
about the change in the zone, whereas in passive notification, the secondary servers
periodically check for any changes.
 In either case, after being notified about the change, the secondary server requests
information about the entire zone (called the zone transfer).
 To provide security and prevent unauthorized changes in the DNS records, DDNS
can use an authentication mechanism.

DNS SECURITY
 DNS is one of the most important systems in the Internet infrastructure; it
provides crucial services to Internet users.
 Applications such as Web access or e-mail are heavily dependent on the proper
operation of DNS.
 DNS can be attacked in several ways including:
 Attack on Confidentiality - The attacker may read the response of a DNS server
to find the nature or names of sites the user mostly accesses. This type of
information can be used to find the user’s profile. To prevent this attack, DNS
messages need to be confidential.
 Attack on authentication and integrity - The attacker may intercept the response
of a DNS server and change it or create a totally new bogus response to direct
the user to the site or domain the attacker wishes the user to access. This type of
attack can be prevented using message origin authentication and message
integrity.
 Attack on denial-of-service - The attacker may flood the DNS server to
overwhelm it or eventually crash it. This type of attack can be prevented using
the provision against denial-of-service attack.

 To protect DNS, IETF has devised a technology named DNS Security (DNSSEC)
that provides message origin authentication and message integrity using a security
service called digital signature.
 DNSSEC, however, does not provide confidentiality for the DNS messages.
 There is no specific protection against the denial-of-service attack in the
specification of DNSSEC. However, the caching system protects the upper- level
servers against this attack to some extent.
 SNMP (SIMPLE NETWORK MANAGEMENT PROTOCOL)
 The Simple Network Management Protocol (SNMP) is a framework for
managing devices in an internet using the TCP/IP protocol suite.
 SNMP is an application layer protocol that monitors and manages routers,
distributed over a network.
 It provides a set of operations for monitoring and managing the internet.
 SNMP uses services of UDP on two well-known ports: 161 (Agent) and 162
(manager).
 SNMP uses the concept of manager and agent.

SNMP MANAGER
 A manager is a host that runs the SNMP client program
 The manager has access to the values in the database kept by the agent.
 A manager checks the agent by requesting the information that reflects the
behavior of the agent.
 A manager also forces the agent to perform a certain function by resetting
values in the agent database.
 For example, a router can store in appropriate variables the number of packets
received and forwarded.
 The manager can fetch and compare the values of these two variables to see if the
router is congested or not.
SNMP AGENT
 The agent is a router that runs the SNMP server program.
 The agent is used to keep the information in a database while the manager is used
to access the values in the database.
 For example, a router can store the appropriate variables such as a number of packets
received and forwarded while the manager can compare these variables to determine
whether the router is congested or not.
 Agents can also contribute to the management process.
 A server program on the agent checks the environment, if something goes wrong,
the agent sends a warning message to the manager.
SNMP MANAGEMENT COMPONENTS
 Management of the internet is achieved through simple interaction between a
manager and agent.
 Management is achieved through the use of two protocols:
 o Structure of Management Information (SMI)
o Management Information Base (MIB).

Structure of Management Information (SMI)

 To use SNMP, we need rules for naming objects.
 SMI is a protocol that defines these rules.
 SMI is a guideline for SNMP
 It emphasizes three attributes to handle an object: name, data type, and
encoding method.
 Its functions are:
 To name objects.
 To define the type of data that can be stored in an object.
 To show how to encode data for transmission over the network.

Name
 SMI requires that each managed object (such as a router, a variable in a router, a
value,etc.) have a unique name. To name objects globally.
 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
using a sequence of integers separated by dots.
 The tree structure can also define an object using a sequence of textual names
separated by dots.
Type of data
 The second attribute of an object is the type of data stored in it.
 To define the data type, SMI uses Abstract Syntax Notation One (ASN.1)
definitions.
 SMI has two broad categories of data types: simple and structured.
 The simple data types are atomic data types. Some of them are taken directly
from ASN.1; some are added by SMI.
 SMI defines two structured data types: sequence and sequence of.
 Sequence - A sequence data type is a combination of simple data types, not
necessarily of the same type.
 Sequence of - A sequence of data type is a combination of simple data types
all of the same type or a combination of sequence data types all of the same
type.
Encoding data
 SMI uses another standard, Basic Encoding Rules (BER), to encode data to be
transmitted over the network.
 BER specifies that each piece of data be encoded in triplet format (TLV): tag, length,
value
 Management Information Base (MIB)
The Management Information Base (MIB) is the second component used in network
management.
 Each agent has its own MIB, which is a collection of objects to be managed.
 MIB classifies objects under groups.

MIB Variables
MIB variables are of two types namely simple and table.
 Simple variables are accessed using group-id followed by variable-id and 0
 Tables are ordered as column-row rules, i.e., column by column from top to
bottom. Only leaf elements are accessible in a table type.

SNMP MESSAGES/PDU
SNMP is request/reply protocol that supports various operations using PDUs.
SNMP defines eight types of protocol data units (or PDUs):
GetRequest, GetNext-Request, GetBulkRequest, SetRequest, Response,
Trap,InformRequest, and Report

GetRequest
 The GetRequest PDU is sent from the manager (client) to the agent (server) to
retrieve the value of a variable or a set of variables.

GetNextRequest
 The GetNextRequest PDU is sent from the manager to the agent to retrieve the
value of a variable.
GetBulkRequest
 The GetBulkRequest PDU is sent from the manager to the agent to retrieve a large
amount of data. It can be used instead of multiple GetRequest and GetNextRequest
PDUs.

SetRequest
 The SetRequest PDU is sent from the manager to the agent to set (store) a
value in a variable.

Response
 The Response PDU is sent from an agent to a manager in response to GetRequest or
GetNextRequest. It contains the value(s) of the variable(s) requested by the manager.

Trap
 The Trap PDU is sent from the agent to the manager to report an event. For example,
if the agent is rebooted, it informs the manager and reports the time of rebooting.

InformRequest
 The InformRequest PDU is sent from one manager to another remote manager to get
the value of some variables from agents under the control of the remote manager. The
remote manager responds with a Response PDU.

Report
 The Report PDU is designed to report some types of errors between managers.