ATME COLLEGE OF ENGINEERING

13th KM Stone, Bannur Road, Mysore - 560 028

DEPARTMENT OF COMPUTER SCIENCE AND ENGINEERING

(ACADEMIC YEAR 2020-21)

LESSON NOTES

SUBJECT: DATA COMMUNICATION

SUB CODE: 18CS46
SEMESTER: IV
 INSTITUTIONAL MISSION AND VISION
Objectives
• To provide quality education and groom top-notch professionals, entrepreneurs and leaders for
different fields of engineering, technology and management.
• To open a Training-R & D-Design-Consultancy cell in each department, gradually introduce doctoral
and postdoctoral programs, encourage basic & applied research in areas of social relevance, and develop
the institute as a center of excellence.
• To develop academic, professional and financial alliances with the industry as well asthe academia at
national and transnational levels.
• To develop academic, professional and financial alliances with the industryas well as the academia at
national and transnational levels.
• To cultivate strong community relationships and involve the students and the staff in localcommunity
service.
• To constantly enhance the value of the educational inputs with the participation ofstudents, faculty,
parents and industry.
Vision
• Development of academically excellent, culturally vibrant, socially responsible and globally competent
human resources.
Mission

• To keep pace with advancements in knowledge and make the students competitive and capable at the
global level.
• To create an environment for the students to acquire the right physical, intellectual, emotionaland
moral foundations and shine as torch bearers of tomorrow's society.
• To strive to attain ever-higher benchmarks of educational excellence.
Data Communication(18CS46)

Department of Computer Science &

Engineering
Vision of the Department

➢
To develop highly talented individuals in Computer Science and Engineering to
deal with realworld challenges in industry, education, research and society.
Mission of the Department
➢
To inculcate professional behaviour, strong ethical values, innovative research
capabilities and
leadership abilities in the young minds & to provide a teaching environment that
emphasizes depth, originality and critical thinking.
➢
Motivate students to put their thoughts and ideas adoptable by industry or to
pursue higher studies leading to research

Program Educational Objectives (PEO'S):

1. Empower students with a strong basis in the mathematical, scientific and engineering
fundamentals to solve computational problems and to prepare them for employment,
higher learning and R&D.
2. Gain technical knowledge, skills and awareness of current technologies of computer
science engineering and to develop an ability to design and provide novel engineering
solutions for software/hardware problems through entrepreneurial skills.
3. Exposure to emerging technologies and work in teams on interdisciplinary
projects with effective communication skills and leadership qualities.
4. Ability to function ethically and responsibly in a rapidly changing environment by
applying innovative ideas in the latest technology, to become effective professionals in
Computer Science to bear a life-long career in related areas.

Program Specific Outcomes (PSOs)

1. Ability to apply skills in the field of algorithms, database design, web design, cloud
computing and data analytics.
2. Apply knowledge In the field of computer networks for building network and internet
based applications.

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DATA COMMUNICATION
MODULE – 1 08 Hours
Introduction: Data Communications, Networks, Network Types, Internet History, Standards and
Administration Networks Models: Protocol Layering, TCP/IP Protocol suite, The OSI model,
Introduction toPhysical Layer-1: Data and Signals, Digital Signals, Transmission Impairment, Data
Rate limits, Performance

MODULE – 2 08 Hours

Digital Transmission: Digital to digital conversion (Only Line coding: Polar, Bipolar and
Manchester coding)
Physical Layer-2: Analog to digital conversion (only PCM), Transmission Modes
Analog Transmission: Digital to analog conversion
Bandwidth Utilization: Multiplexing and Spread Spectrum
Switching: Introduction, Circuit Switched Networks and Packet switching

MODULE – 3 08 Hours
Error Detection and Correction: Introduction, Block coding, Cyclic codes, Checksum, Forward error
correction. Data link control: DLC services, Data link layer protocols, HDLC, and Point to Point
protocol (Framing, Transition phases only).

MODULE – 4 08 Hours
Media Access control: Random Access, Controlled Access and Channelization
Wired LANs Ethernet: Ethernet Protocol, Standard Ethernet, Fast Ethernet, Gigabit Ethernet and 10
Gigabit
Ethernet
Wireless LANs: Introduction, IEEE 802.11 Project and Bluetooth

MODULE – 5 08Hours
Other wireless Networks: WIMAX, Cellular Telephony, Satellite networks
Network layer Protocols: Internet Protocol, ICMPv4, Mobile IP
Next generation IP: IPv6 addressing, The IPv6 Protocol, The ICMPv6 Protocol and Transition from
IPv4 to IPv6.

Text Book:
Behrouz A. Forouzan, Data Communications and Networking 5E, 5th Edition, Tata McGraw-Hill,
2013. (Chapters 1.1 to 1.5, 2.1 to 2.3, 3.1, 3.3 to 3.6, 4.1 to 4.3, 5.1, 6.1, 6.2, 8.1 to 8.3, 10.1 to 10.5,
11.1 to 11.4, 12.1 to 12.3, 13.1 to 13.5, 15.1 to 15.3, 16.1 to 16.3, 19.1 to 19.3, 22.1 to 22.4)

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MODULE 1: TABLE OF CONTENTS

DATA COMMUNICATIONS
Components
Data Representation
Direction of Data Flow
NETWORKS
Network Criteria
Physical Structures
Type of Connection
Physical Topology
Bus Topology
Star Topology
Ring Topology
Mesh Topology
NETWORK TYPES
Local Area Network
Wide Area Network
Internetwork
LAN vs. WAN
Switching
Circuit-Switched Network
Packet-Switched Network
The Internet
Accessing the Internet
STANDARDS AND ADMINISTRATION
Internet Standards
Maturity Levels
Requirement Levels
Internet Administration
PROTOCOL LAYERING
Scenarios
Protocol Layering
Principles of Protocol Layering
Logical Connections
TCP/IP PROTOCOL SUITE
Layered Architecture
Layers in the TCP/IP Protocol Suite
Description of Each Layer
Encapsulation and Decapsulation
Addressing 42
Multiplexing and Demultiplexing
THE OSI MODEL
OSI versus TCP/IP
Lack of OSI Model’s Success
DATA AND SIGNALS
Analog and Digital Data
Analog and Digital Signals
Periodic and Nonperiodic
DIGITAL SIGNALS
Bit Rate
Bit Length
Digital Signal as a Composite Analog Signal

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Transmission of Digital Signals

Baseband Transmission
Broadband Transmission (Using Modulation)
TRANSMISSION IMPAIRMENT
Attenuation
Decibel
Distortion
Noise
Signal-to-Noise Ratio (SNR)
DATA RATE LIMITS
Noiseless Channel: Nyquist Bit Rate
Noisy Channel: Shannon Capacity
PERFORMANCE
Bandwidth
Throughput
Latency (Delay)
Bandwidth-Delay Product
Jitter
DIGITAL-TO-DIGITAL CONVERSION
Line Coding
Characteristics
Line Coding Schemes
Unipolar Scheme
Polar Schemes
Bipolar Schemes (or Multilevel Binary)

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MODULE 1
INTRODUCTION
STRUCTURE

Introduction
Objectives
Data Communications
Networks
Network Types
Standards And Administration
Protocol Layering
TCP/IP Protocol Suite
Data and Signals
Transmission Impairment
1.12Data Rate Limits
Performance
Digital To Digital Conversion
Assignment Questions
Expected Outcomes
Further Reading

Introduction
This module provides the introductory concepts of the data communications and the brief descriptions
of the different types of network and topologies.

Objectives
• DATA COMMUNICATIONS and its Components
• Differences between NETWORK topologies & NETWORK TYPES
• Internet Standards and ADMINISTRATION
• PROTOCOL LAYERING,TCP/IP PROTOCOL SUITE & THE OSI MODEL
• DIGITAL SIGNALS
• TRANSMISSION IMPAIRMENT
• DIGITAL-TO-DIGITAL& DIGITAL TO ANALOG Conversion Techniques

Data Communications
Data Communication is defined as exchange of data between 2 devices over a transmission-medium.
A communication-system is made up of hardware (physical equipment) and software (programs)
For data-communication, the communicating-devices must be part of a communication-system.

Four attributes of a communication-system:

1) Delivery
The system must deliver data to the correct destination.
2) Accuracy
The system must deliver the data accurately.
Normally, the corrupted-data are unusable.
3) Timeliness
The system must deliver audio/video data in a timely manner.
This kind of delivery is called real-time transmission.

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Data delivered late are useless.

4) Jitter
Jitter refers to the variation in the packet arrival-time.
In other words, jitter is the uneven delay in the delivery of audio/video packets.

Components of Communication System

Five components of a communication-system (Figure 1.1):
Message
Sender
Receiver
Transmission-Medium
Protocol

1) Message
Message is the information (or data) to be communicated. Message may consist of number/text
picture or audio/video

2) Sender
Sender is the device that sends the data-message. Sender can be computer and mobile phone

3) Receiver
Receiver is the device that receives the message. Receiver can be computer and mobile phone

4) Transmission Medium
Transmission-medium is physical-path by which a message travels from sender to receiver.
Transmission-medium can be wired or wireless.
Examples of wired medium:
twisted-pair wire (used in landline telephone)
coaxial cable (used in cable TV network)
fiber-optic cable
Examples of wireless medium:
radio waves
microwaves
infrared waves (ex: operating TV using remote control)
5) Protocol
A protocol is a set of rules that govern data-communications.
In other words, a protocol represents an agreement between the communicating-devices.
Without a protocol, 2 devices may be connected but not communicating.

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Data Representation

• Five different forms of information:

1) Text
Text is represented as a bit-pattern. (Bit-pattern sequence of bits: 0s or 1s).
Different sets of bit-patterns are used to represent symbols (or characters).
Each set is called a code.
The process of representing symbols is called encoding.
Popular encoding system: ASCII, Unicode.
2) Number
Number is also represented as a bit-pattern.
ASCII is not used to represent number. Instead, number is directly converted to binary-form.
3) Image
Image is also represented as a bit-pattern.
An image is divided into a matrix of pixels (picture-elements). A
pixel is the smallest element of an image. (Pixel Small dot)
The size of an image depends upon number of pixels (also called resolution). For
example: An image can be divided into 1000 pixels or 10,000 pixels.
Two types of images:
i) Black & White Image

¤ If an image is black & white, each pixel can be represented by a value either 0 or 1. ¤ For
example: Chessboard
ii) Color Image
There are many methods to represent color images.
RGB is one of the methods to represent color images.
RGB is called so called ‘.’ each color is combination of 3 colors: red, green & blue.

4) Audio
0
Audio is a representation of sound.
1
By nature, audio is different from text, numbers, or images. Audio is continuous, not discrete.

5) Video
Video is a representation of movie. Video can either be produced as a continuous entity (e.g., by a
TV camera), or be a combination of images arranged to convey the idea of motion.

Direction of Data Flow

Three ways of data-flow between 2 devices (Figure 1.2):

Simplex
Half-duplex
Full-duplex

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1) Simplex
The communication is unidirectional
(For ex: The simplex mode is like a one-way street).
On a link, out of 2 devices:
Only one device can transmit.
Another device can only receive.
For example (Figure 1.2a):
The monitor can only accept output.
Entire-capacity of channel is used to send the data in one direction.
2) Half-Duplex
Both the stations can transmit as well as receive but not at the same time.
(For ex: The half-duplex mode is like a one-lane road with 2 directional traffic).
When one station is sending, the other can only receive and vice-versa.
For example (Figure 1.2b): Walkie-talkies
Entire-capacity of a channel is used by one of the 2 stations that are transmitting the data.
3) Full-Duplex
Both stations can transmit and receive at the same time.
(For ex: The full-duplex is like a 2-way street with traffic flowing in both directions at the
same time).
For example (Figure 1.2c):
Mobile phones (When 2 people are communicating by a telephone line, both can listen and
talk at the same time)
Entire-capacity of a channel is shared by both the stations that are transmitting the data.

Networks
A network is defined as a set of devices interconnected by communication-links.
This interconnection among computers facilitates information sharing among them.
Computers may connect to each other by either wired or wireless media.
Often, devices are referred to as nodes.
A node can be any device capable of sending/receiving data in the network.
For example: Computer & Printer
The best-known computer network is the Internet.

Network Criteria
A network must meet following 3 criteria’s:
1) Performance
Performance can be measured using i) Transit-time or ii) Response-time.
Transit Time is defined as time taken to travel a message from one device to another.
Response Time is defined as the time elapsed between enquiry and response.
The network-performance depends on following factors:
Number of users
Type of transmission-medium
Efficiency of software
Often, performance is evaluated by 2 networking-metrics: i) throughput and ii) delay.
Good performance can be obtained by achieving higher throughput and smaller delay times
2) Reliability
Reliability is measured by
frequency of network-failure
time taken to recover from a network-failure
network's robustness in a disaster
More the failures are, less is the network's reliability.
3) Security
Security refers to the protection of data from the unauthorized access or damage.

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It also involves implementing policies for recovery from data-losses.

Physical Structures
Type of Connection
Two types of connections (Figure 1.3):
Point-to-Point
Only two devices are connected by a dedicated-link (Figure 1.3a).
Entire-capacity of the link is reserved for transmission between those two devices.
For example: Point-to-Point connection b/w remote-control & TV for changing the channels.
2) Multipoint (Multi-drop)
Three or more devices share a single link.
The capacity of the channel is shared, either spatially or temporally (Figure 1.3b).
If link is used simultaneously by many devices, then it is spatially shared connection.
If user takes turns while using the link, then it is time shared (temporal) connection.
(spatially space or temporally time)

Physical Topology
The physical-topology defines how devices are connected to make a network.
Four basic topologies are:
Mesh
Star
Bus and
Ring

Bus Topology
All the devices are connected to the single cable called bus (Figure 1.4).
Every device communicates with the other device through this bus.
A data from the source is broadcasted to all devices connected to the bus.
Only the intended-receiver, whose physical-address matches, accepts the data.

Devices are connected to the bus by drop-lines and taps.

A drop-line is a connection running between the device and the bus.
A tap is a connector that links to the bus or

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Advantages:
Easy installation.
Cable required is the least compared to mesh/star topologies.
Redundancy is eliminated.
Costs less (Compared to mesh/star topologies).
Mostly used in small networks. Good for LAN.
Disadvantages:
Difficult to detect and troubleshoot fault.
Signal reflection at the taps can cause degradation in quality.
A fault/break in the cable stops all transmission.
There is a limit on
Cable length
Number of nodes that can be connected.
Security is very low because all the devices receive the data sent from the source.

Star Topology
All the devices are connected to a central controller called a hub (Figure 1.5).
There exists a dedicated point-to-point link between a device & a hub.
The devices are not directly linked to one another. Thus, there is no direct traffic between devices.
The hub acts as a junction:
If device-1 wants to send data to device-2, the
device-1 sends the data to the hub,
then the hub relays the data to the device-2.

Advantages:
Less expensive: Each device needs only one link & one I/O port to connect it to any devices.
Easy installation & reconfiguration: Nodes can be added/removed w/o affecting the network.
Robustness: If one link fails, it does not affect the entire system.
Easy to detect and troubleshoot fault.
Centralized management: The hub manages and controls the whole network.
Disadvantages:
Single point of failure: If the hub goes down, the whole network is dead.
Cable length required is the more compared to bus/ring topologies.
Number of nodes in network depends on capacity of hub.

Ring Topology
Each device is connected to the next, forming a ring (Figure 1.6).
There are only two neighbors for each device.
Data travels around the network in one direction till the destination is reached.
Sending and receiving of data takes place by the help of token.
Each device has a repeater.
A repeater
receives a signal on transmission-medium &
regenerates & passes the signal to next device.

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Advantages:
Easy installation and reconfiguration.
To add/delete a device, requires changing only 2 connections. 3)
Fault isolation is simplified.
If one device does not receive a signal within a specified period, it can issue an alarm. The
alarm alerts the network-operator to the problem and its location.
Congestion reduced: Because all the traffic flows in only one direction.
Disadvantages:
Unidirectional traffic.
A fault in the ring/device stops all transmission.
The above 2 drawbacks can be overcome by using dual ring.
There is a limit on
Cable length &
Number of nodes that can be connected.
Slower: Each data must pass through all the devices between source and destination.

Mesh Topology
All the devices are connected to each other (Figure 1.7).
There exists a dedicated point-to-point link between all devices.
There are n(n-1) physical channels to link n devices.
Every device not only sends its own data but also relays data from other nodes.
For ‘n’ nodes,
there are n(n-1) physical-links
there are n(n-1)/2 duplex-mode links
Every device must have (n–1) I/O ports to be connected to the other (n-1) devices.

Advantages:
Congestion reduced: Each connection can carry its own data load.
Robustness: If one link fails, it does not affect the entire system.
Security: When a data travels on a dedicated-line, only intended-receiver can see the data.
Easy fault identification & fault isolation: Traffic can be re-routed to avoid problematic links.
Disadvantages:

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Difficult installation and reconfiguration.

Bulk of wiring occupies more space than available space.
Very expensive: as there are many redundant connections.
Not mostly used in computer networks. It is commonly used in wireless networks.
High redundancy of the network-connections.

Network Types
Two popular types of networks:
LAN (Local Area Network) &
WAN (Wide Area Network)

LAN
LAN is used to connect computers in a single office, building or campus (Figure 1.8).
LAN is usually privately owned network.
A LAN can be simple or complex.
Simple: LAN may contain 2 PCs and a printer.
Complex: LAN can extend throughout a company.
Each host in a LAN has an address that uniquely defines the host in the LAN.
A packet sent by a host to another host carries both source host’s and destination host’s addresses.
LANs use a smart connecting switch.
The switch is able to
recognize the destination address of the packet &
guide the packet to its destination.
The switchreduces the traffic in the LAN &
allows more than one pair to communicate with each other at the same time.
Advantages:
1) Resource Sharing
Computer resources like printers and hard disks can be shared by all devices on the network.
2) Expansion
Nowadays, LANs are connected to WANs to create communication at a wider level.

WAN
• WAN is used to connect computers anywhere in the world.
• WAN can cover larger geographical area. It can cover cities, countries and even continents.
• WAN interconnects connecting devices such as switches, routers, or modems.

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• Normally, WAN is
→ created & run by communication companies (Ex: BSNL, Airtel)
→ leased by an organization that uses it.
• A WAN can be of 2 types:
1) Point-to-point WAN
A point-to-point WAN is a network that connects 2 communicating devices through a transmission
media (Figure 1.9).

2) Switched WAN
A switched WAN is a network with more than two ends.
The switched WAN can be the backbones that connect the Internet.
A switched WAN is a combination of several point-to-point WANs that are connected by switches
(Figure 1.10).

1.3.2.1 Internetwork
A network of networks is called an internet. (Internet inter-network) (Figure 1.12).
For example (Figure 1.11):
Assume that an organization has two offices,
First office is on the east coast &
Second office is on the west coast.
Each office has a LAN that allows all employees in the office to communicate with each other.
To allow communication between employees at different offices, the management leases a
point-to-point dedicated WAN from a ISP and connects the two LANs.
(ISP Internet service provider such as a telephone company ex: BSNL).

When a host in the west coast office sends a message to another host in the same office, the router
blocks the message, but the switch directs the message to the destination.
On the other hand, when a host on the west coast sends a message to a host on the east coast, router
R1 routes the packet to router R2, and the packet reaches the destination.

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LAN vs. WAN

Parameters LAN WAN
Expands to Local Area Network Wide Area Network
Meaning LAN is used to connect computers WAN is used to connect computers
in a single office, building or in a large geographical area such
campus as countries
Ownership of network Private Private or public

Range Small: up to 10 km Large: Beyond 100 km

Speed High: Typically 10, 100 and 1000 Low: Typically 1.5 Mbps
Mbps
Propagation Delay Short Long
Cost Low High
Congestion Less More
Design & maintenance Easy Difficult
Fault Tolerance More Tolerant Less Tolerant
Media used Twisted pair Optical fiber or radio waves

Used for College, Hospital Internet

Interconnects LAN interconnects hosts WAN interconnects connecting
devices such as switches, routers,
or modems

Switching
An internet is a switched network in which a switch connects at least two links together.
A switch needs to forward data from a network to another network when required.
Two types of switched networks are 1) circuit-switched and 2) packet-switched networks.
1.3.4.1 Circuit-Switched Network
A dedicated connection, called a circuit, is always available between the two end systems.
The switch can only make it active or inactive.

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As shown in Figure 1.13, the 4 telephones at each side are connected to a switch.
The switch connects a telephone at one side to a telephone at the other side.
A high-capacity line can handle 4 voice communications at the same time.
The capacity of high line can be shared between all pairs of telephones.
The switch is used for only forwarding.
Advantage:
A circuit-switched network is efficient only when it is working at its full capacity.
Disadvantage:
Most of the time, the network is inefficient because it is working at partial capacity.
Packet-Switched Network
In a computer network, the communication between the 2 ends is done in blocks of data called
packets.
The switch is used for both storing and forwarding because a packet is an independent entity
that can be stored and sent later.

As shown in Figure 1.14, the 4 computers at each side are connected to a router.
A router has a queue that can store and forward the packet.
The high-capacity line has twice the capacity of the low-capacity line.
If only 2 computers (one at each site) need to communicate with each other, there is no
waiting for the packets.
However, if packets arrive at one router when high-capacity line is at its full capacity, the
packets should be stored and forwarded.
Advantages:
A packet-switched network is more efficient than a circuit switched network.
Disadvantage:
The packets may encounter some delays.

1.4.5 The Internet Today

• A network of networks is called an internet. (Internet inter-network)
• Internet is made up of (Figure 1.15) 1)
Backbones
2) Provider networks &
3) Customer networks
1) Backbones
Backbones are large networks owned by communication companies such as BSNL and Airtel.
The backbone networks are connected through switching systems, called peering points.
2) Provider Networks
Provider networks use the services of the backbones for a fee.
Provider networks are connected to backbones and sometimes to other provider networks.
3) Customer Networks

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Customer networks actually use the services provided by the Internet.

Customer networks pay fees to provider networks for receiving services.
Backbones and provider networks are also called Internet Service Providers (ISPs).
The backbones are often referred to as international ISPs.
The provider networks are often referred to as national or regional ISPs.

Accessing the Internet

• The Internet today is an internetwork that allows any user to become part of it.
• However, the user needs to be physically connected to an ISP.
• The physical connection is normally done through a point-to-point WAN.
1) Using Telephone Networks
Most residences have telephone service, which means they are connected to a telephone network.
Most telephone networks have already connected themselves to the Internet.
Thus, residences can connect to the Internet using a point-to-point WAN.
This can be done in two ways:
A) Dial-up service
A modem can be added to the telephone line.
A modem converts data to voice.
The software installed on the computer
dials the ISP &
imitates making a telephone connection.
Disadvantages:
The dial-up service is very slow.
When line is used for Internet connection, it cannot be used for voice connection.
It is only useful for small residences.
B) DSL Service
DSL service also allows the line to be used simultaneously for voice & data communication.
Some telephone companies have upgraded their telephone lines to provide higher speed
Internet services to residences.
Using Cable Networks
A residence can be connected to the Internet by using cable service.
Cable service provides a higher speed connection.
The speed varies depending on the number of neighbors that use the same cable.
3) Using Wireless Networks
A residence can use a combination of wireless and wired connections to access the Internet.
A residence can be connected to the Internet through a wireless WAN.
4) Direct Connection to the Internet
A large organization can itself become a local ISP and be connected to the Internet.
The organization
leases a high-speed WAN from a carrier provider and
connects itself to a regional ISP.

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Standards and Administration

Internet Standards
An Internet standard is a thoroughly tested specification useful to those who work with the Internet.
The Internet standard is a formalized-regulation that must be followed.
There is a strict procedure by which a specification attains Internet standard status.
A specification begins as an Internet draft.
An Internet draft is a working document with no official status and a 6-month lifetime.
Upon recommendation from the Internet authorities, a draft may be published as a RFC.
Each RFC is edited, assigned a number, and made available to all interested parties.
RFCs go through maturity levels and are categorized according to their requirement level.
(working document a work in progress RFC Request for Comment)
Maturity Levels
• An RFC, during its lifetime, falls into one of 6 maturity levels (Figure 1.16):

1) Proposed Standard
Proposed standard is specification that is stable, well-understood & of interest to Internet
community.
Specification is usually tested and implemented by several different groups.
2) Draft Standard
A proposed standard is elevated to draft standard status after at least 2 successful independent and
interoperable implementations.
3) Internet Standard
A draft standard reaches Internet standard status after demonstrations of successful
implementation.
4) Historic
The historic RFCs are significant from a historical perspective.
They either
→ have been superseded by later specifications or
→ have never passed the necessary maturity levels to become an Internet standard.
5) Experimental
An RFC classified as experimental describes work related to an experimental situation.
Such an RFC should not be implemented in any functional Internet service.
6) Informational
An RFC classified as informational contains general, historical, or tutorial information related to the
Internet.
Usually, it is written by a vendor.
(ISOC Internet Society IAB Internet Architecture Board)
(IETF Internet Engineering Task Force IRTF Internet Research Task Force)

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(IESG Internet Engineering Steering Group IRSG Internet Research Steering Group)

Requirement Levels
• RFCs are classified into 5 requirement levels:
1) Required
An RFC labeled required must be implemented by all Internet systems to achieve minimum
conformance.
For example, IP and ICMP are required protocols.
2) Recommended
An RFC labeled recommended is not required for minimum conformance.
It is recommended because of its usefulness.
For example, FTP and TELNET are recommended protocols.
3) Elective
An RFC labeled elective is not required and not recommended.
However, a system can use it for its own benefit.
4) Limited Use
An RFC labeled limited use should be used only in limited situations.
Most of the experimental RFCs fall under this category.
5) Not Recommended
An RFC labeled not recommended is inappropriate for general use.
Normally a historic RFC may fall under this category.
1.4.2 Internet Administration
1) ISOC
ISOC is a nonprofit organization formed to provide support for Internet standards process (Fig 1.17).
ISOC maintains and supports other Internet administrative bodies such as IAB, IETF, IRTF, and IANA.
2) IAB
IAB is the technical advisor to the ISOC.
Two main purposes of IAB:
To oversee the continuing development of the TCP/IP Protocol Suite
To serve in a technical advisory capacity to research members of the Internet community.
Another responsibility of the IAB is the editorial management of the RFCs.
IAB is also the external liaison between the Internet and other standards organizations and forums.
IAB has 2 primary components: i) IETF and ii) IRTF.
IETF
IETF is a forum of working groups managed by the IESG.
IETF is responsible for identifying operational problems & proposing solutions to the problems
IETF also develops and reviews specifications intended as Internet standards.
The working groups are collected into areas, and each area concentrates on a specific topic.
Currently 9 areas have been defined. The areas include applications, protocols, routing,
network management next generation (IPng), and security.
ii) IRTF
IRTF is a forum of working groups managed by the IRSG.
IRTF focuses on long-term research topics related to Internet protocols, applications, architecture,
and technology.

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MODULE 1(CONT.): NETWORK MODELS

Protocol Layering
A protocol defines the rules that both the sender and receiver and all intermediate devices need to follow to
be able to communicate effectively.
When communication is simple, we may need only one simple protocol.
When communication is complex, we need to divide the task b/w different layers. We need a
protocol at each layer, or protocol layering.

Scen
arios
First
Scen
ario
In the first scenario, communication is so simple that it can occur in only one layer (Figure 2.1).
Assume Maria and Ann are neighbors with a lot of common ideas.
Communication between Maria and Ann takes place in one layer, face to face, in the same language

Second Scenario
Maria and Ann communicate using regular mail through the post office (Figure 2.2).
However, they do not want their ideas to be revealed by other people if the letters are intercepted.
They agree on an encryption/decryption technique.
The sender of the letter encrypts it to make it unreadable by an intruder; the receiver of the letter decrypts it
to get the original letter.

Protocol Layering
Protocol layering enables us to divide a complex task into several smaller and simpler tasks.
Modularity means independent layers.

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A layer (module) can be defined as a black box with inputs and outputs, without concern about how inputs
are changed to outputs.

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If two machines provide the same outputs when given the same inputs, they can replace each other.
Advantages:
It allows us to separate the services from the implementation.
There are intermediate systems that need only some layers, but not all layers.
Disadvantage:
Having a single layer makes the job easier. There is no need for each layer to provide a service to the
upper layer and give service to the lower layer.

Principles of Protocol Layering 1) First

Principle
• If we want bidirectional communication, we need to make each layer able to perform 2 opposite tasks, one
in each direction.
• For example, the third layer task is to listen (in one direction) and talk (in the other direction).
2) Second Principle
• The two objects under each layer at both sites should be identical.
• For example, the object under layer 3 at both sites should be a plaintext letter.

Logical Connections
We have layer-to-layer communication (Figure 2.3).
There is a logical connection at each layer through which 2 end systems can send the object created from
that layer.

TCP/IP Protocol Suite

TCP/IP is a protocol-suite used in the Internet today.
Protocol-suite refers a set of protocols organized in different layers.
It is a hierarchical protocol made up of interactive modules, each of which provides a specific functionality.
The term hierarchical means that each upper level protocol is supported by the services provided by one or
more lower level protocols.

Layered Architecture

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Let us assume that computer A communicates with computer B (Figure 2.4).

As the Figure 2.5 shows, we have five communicating devices:
1) Source host(computer A) 2) Link-layer switch in link 1
3) Router 4) Link-layer switch in link 2
Destination host (computer B).
Each device is involved with a set of layers depending on the role of the device in the internet.
The two hosts are involved in all five layers.
The source host creates a message in the application layer and
sends the message down the layers so that it is physically sent to the destination host.
The destination host
receives the message at the physical layer and
then deliver the message through the other layers to the application layer.
The router is involved in only three layers; there is no transport or application layer.
A router is involved in n combinations of link and physical layers.
where n = number of links the router is connected to.
The reason is that each link may use its own data-link or physical protocol.
A link-layer switch is involved only in two layers: i) data-link and ii) physical.

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Layers in the TCP/IP Protocol Suite

As shown in the figure 2.6, the duty of the application, transport, and network layers is end-to-end.
However, the duty of the data-link and physical layers is hop-to-hop. A hop is a host or router.
The domain of duty of the top three layers is the internet.
The domain of duty of the two lower layers is the link.
• In top 3 layers, the data unit should not be changed by any router or link-layer switch.
In bottom 2 layers, the data unit is changed only by the routers, not by the link-layer switches.

Identical objects exist between two hops. Because router may fragment the packet at the network layer and
send more packets than received (Figure 2.7).
The link between two hops does not change the object.

Description of Each
Layer Physical Layer
The physical layer is responsible for movements of individual bits from one node to another node.
Transmission media is another hidden layer under the physical layer.
Two devices are connected by a transmission medium (cable or air).
The transmission medium does not carry bits; it carries electrical or optical signals.
The physical layer
receives bits from the data-link layer &
sends through the transmission media.

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Data Link Layer

Data-link-layer (DLL) is responsible for moving frames from one node to another node over a link.
The link can be wired LAN/WAN or wireless LAN/WAN.
The data-link layer
gets the datagram from network layer
encapsulates the datagram in a packet called a frame.
sends the frame to physical layer.
TCP/IP model does not define any specific protocol.
DLL supports all the standard and proprietary protocols.
Each protocol may provide a different service.
Some protocols provide complete error detection and correction; some protocols provide only error
correction.
Network Layer
The network layer is responsible for source-to-destination transmission of data.
The network layer is also responsible for routing the packet.
The routers choose the best route for each packet.
Why we need the separate network layer?
The separation of different tasks between different layers.
The routers do not need the application and transport layers.
TCP/IP model defines 5 protocols:
1) IP (Internetworking Protocol) 2) ARP (Address Resolution Protocol)
3) ICMP (Internet Control Message Protocol) 4) IGMP (Internet Group Message Protocol)
1) IP
IP is the main protocol of the network layer.
IP defines the format and the structure of addresses.
IP is also responsible for routing a packet from its source to its destination.
It is a connection-less & unreliable protocol.
Connection-less means there is no connection setup b/w the sender and the receiver.
Unreliable protocol means
IP does not make any guarantee about delivery of the data.
Packets may get dropped during transmission.
It provides a best-effort delivery service.
Best effort means IP does its best to get the packet to its destination, but with no guarantees.
IP does not provide following services
flow control
error control
congestion control services.
If an application requires above services, the application should rely only on the transport-layer
protocol.
2) ARP
ARP is used to find the physical-address of the node when its Internet-address is known.
Physical address is the 48-bit address that is imprinted on the NIC or LAN card.
Internet address (IP address) is used to uniquely & universally identify a device in the internet.
3) ICMP
ICMP is used to inform the sender about datagram-problems that occur during transit.
4) IGMP
IGMP is used to send the same message to a group of recipients.

Transport Layer
TL protocols are responsible for delivery of a message from a process to another process.
The transport layer
gets the message from the application layer
encapsulates the message in a packet called a segment and
sends the segment to network layer.
TCP/IP model defines 3 protocols:1) TCP (Transmission Control Protocol)
UDP (User Datagram Protocol) &

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SCTP (Stream Control Transmission Protocol)

1) TCP
TCP is a reliable connection-oriented protocol.
A connection is established b/w the sender and receiver before the data can be transmitted.
TCP provides
→ flow control
→ error control and
→ congestion control
2) UDP
UDP is the simplest of the 3 transport protocols.
It is an unreliable, connectionless protocol.
It does not provide flow, error, or congestion control.
Each datagram is transported separately & independently.
It is suitable for application program that
→ needs to send short messages &
→ cannot afford the retransmission.
3) SCTP
SCTP provides support for newer applications such as voice over the Internet.
It combines the best features of UDP and TCP.

Application Layer
The two application layers exchange messages between each other.
Communication at the application layer is between two processes (two programs running at this layer).
To communicate, a process sends a request to the other process and receives a response.
Process-to-process communication is the duty of the application layer.
TCP/IP model defines following protocols:
SMTP is used to transport email between a source and destination.
TELNET is used for accessing a site remotely.
FTP is used for transferring files from one host to another.
DNS is used to find the IP address of a computer.
SNMP is used to manage the Internet at global and local levels.
HTTP is used for accessing the World Wide Web (WWW).
(FTP File Transfer Protocol SMTP Simple Mail Transfer Protocol)
(DNS Domain Name System HTTP Hyper Text Transfer Protocol)
(SNMP Simple Network Management Protocol TELNET Terminal Network)
Encapsulation and Decapsulation

A) Encapsulation at the Source Host

At the source, we have only encapsulation (Figure 2.8).
At the application layer, the data to be exchanged is referred to as a message. A
message normally does not contain any header or trailer.
The message is passed to the transport layer.
The transport layer takes the message as the payload.
TL adds its own header to the payload.

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The header contains

identifiers of the source and destination application programs
information needed for flow, error control, or congestion control.
The transport-layer packet is called the segment (in TCP) and the user datagram (in UDP).
The segment is passed to the network layer.
3) The network layer takes the transport-layer packet as payload.
NL adds its own header to the payload.
The header contains
→ addresses of the source and destination hosts
→ some information used for error checking of the header &
→ fragmentation information.
The network-layer packet is called a datagram.
The datagram is passed to the data-link layer.
4) The data -link layer takes the network-layer packet as payload.
DLL adds its own header to the payload.
The header contains the physical addresses of the host or the next hop (the router). The
link-layer packet is called a frame.
The frame is passed to the physical layer for transmission
B) Decapsulation and Encapsulation at the Router
• At the router, we have both encapsulation & encapsulation and because the router is connected to two or
more links.
Data-link layer
receives frame from physical layer
decapsulates the datagram from the frame and
passes the datagram to the network layer.
The network layer
inspects the source and destination addresses in the datagram header and
consults forwarding table to find next hop to which the datagram is to be delivered.
The datagram is then passed to the data-link layer of the next link.
The data-link layer of the next link
encapsulates the datagram in a frame and
passes the frame to the physical layer for transmission.
C) Decapsulation at the Destination Host
At the destination host, each layer
decapsulates the packet received from lower layer
removes the payload and
delivers the payload to the next-higher layer

Addressing
We have logical communication between pairs of layers.
Any communication that involves 2 parties needs 2 addresses: source address and destination address.
We need 4 pairs of addresses (Figure 2.9):
At the application layer, we normally use names to define
site that provides services, such as vtunotesbysri.com, or
e-mail address, such as [email protected].
At the transport layer, addresses are called port numbers.
Port numbers define the application-layer programs at the source and destination.
Port numbers are local addresses that distinguish between several programs running at the
same time.
3) At the network-layer, addresses are called IP addresses.
IP address uniquely defines the connection of a device to the Internet.
The IP addresses are global, with the whole Internet as the scope. 4) At
the data link-layer, addresses are called MAC addresses
The MAC addresses defines a specific host or router in a network (LAN or WAN).
The MAC addresses are locally defined addresses.

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Multiplexing and Demultiplexing

Multiplexing means a protocol at a layer can encapsulate a packet from several next-higher layer protocols
(one at a time) (Figure 2.10).
Demultiplexing means a protocol can decapsulate and deliver a packet to several next-higher layer protocols
(one at a time).
At transport layer, either UDP or TCP can accept a message from several application-layer protocols.
At network layer, IP can accept
a segment from TCP or a user datagram from UDP.
a packet from ICMP or IGMP.
At data-link layer, a frame may carry the payload coming from IP or ARP.

OSI Model
OSI model was developed by ISO.
ISO is the organization, OSI is the model.
Purpose: OSI was developed to allow systems with diff. platforms to communicate with each other.
Platform means hardware, software or operating system.
OSI is a network-model that defines the protocols for network communications.
OSI has 7 layers as follows (Figure 2.11):
Application Layer
Presentation Layer
Session Layer
Transport Layer
Network Layer
Data Link Layer
Physical Layer
Each layer has specific duties to perform and has to co-operate with the layers above & below it.

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OSI vs. TCP/IP

The four bottommost layers in the OSI model & the TCP/IP model are same (Figure 2.12).
However, the Application-layer of TCP/IP model corresponds to the Session, Presentation &
Application Layer of OSI model.
Two reasons for this are:
TCP/IP has more than one transport-layer protocol.
Many applications can be developed at Application layer
The OSI model specifies which functions belong to each of its layers.
In TCP/IP model, the layers contain relatively independent protocols that can be mixed and
matched depending on the needs of the system.

Lack of OSI Model’s Success

OSI was completed when TCP/IP was fully in place and a lot of time and money had been spent on the
suite; changing it would cost a lot.
Some layers in the OSI model were never fully defined.
When OSI was implemented by an organization in a different application, it did not show a high enough
level of performance

LAYERS IN THE OSI MODEL (Detailed OSI layers not in syllabus, it’s for your reference)
Physical Layer
• Main Responsibility:
Physical-layer (PL) is responsible for movements of individual bits from one node to another node.

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• Other responsibilities of Physical-layer (Figure 2.5):

1) Physical Characteristics of Interfaces and Medium
PL defines the mechanical/electrical characteristics of the interface & transmission-medium i.e.
Mechanical cable, plugs, pins
Electrical modulation, signal strength, voltage levels
PL also defines the type of transmission-medium. (Wired or wireless).
2) Representation of Bits
PL defines the type of encoding i.e. how 0s and 1s are changed to signals.
Data consists of a stream of bits: 0s or 1s.
Bits must be encoded into signals for transmission.
3) Data Rate
PL defines the transmission-rate.
Transmission-rate refers to the number of bits sent per second.
4) Synchronization of Bits
PL deals with the synchronization of the transmitter and receiver.
The sender and receiver are synchronized at bit-level.
5) Line Configuration
PL defines the nature of the connection.
In a point-to-point configuration, a dedicated-link is used to connect between 2 devices
In a multipoint configuration, a shared-link is used to connect between 2 or more devices.
6) Physical Topology
PL defines the type of topology used for connecting the devices in the network.
Topologies can be mesh, star, ring or bus.
7) Transmission Mode
PL defines the direction of data-transfer between 2 devices.
Simplex: Only one device can send; the other device can only receive.
Half-duplex: Two devices can send and receive, but not at the same time.
Full-duplex: Two devices can send and receive at the same time.

Data Link Layer

• Main Responsibility:
Data-link-layer (DLL) is responsible for moving frames from one node to another node.

Other responsibilities of data-link-layer (Figure 2.6 & 2.7):

Framing
DLL receives & divides the stream of bits from network-layer into frames.
2) Physical-addressing
DLL appends a header to the frame coming from the network-layer.
Header contains the physical-address of sender & receiver of the frame.

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3) Flow Control
DLL provides flow-control.
Flow-control ensures that source sends the data at a speed at which destination can receive it
If there is an overflow at the receiver-side, the data will be lost.
4) Error Control
DLL provides error-control.
Error-control is process of identification or correction of error occurred in the transmitted data
Error-control uses mechanisms to
→ detect damaged-frames
→ retransmit lost-frames
→ recognize duplicate frames.
Normally, error control information is present in the trailer of a frame.
5) Access Control
DLL provides access-control.
Access-control determines which device has right to send the data in a multipoint connection.

Network Layer
• Main Responsibility:
Network-layer (NL) is responsible for source-to-destination delivery of a packet, possibly across
multiple-networks.
Data-link-layer vs. Network-layer:
The data-link-layer ensures the delivery of the packet between 2 systems on the same link.
The network-layer ensures that each packet gets from the source to the final destination.
If 2 systems are connected to the same link, there is no need for a network-layer.
However, if the 2 systems are attached to different links, there is often a need for the network-layer
to accomplish source-to-destination delivery.

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• Other responsibilities of network-layer (Figure 2.8 & 2.9):

1) Logical Addressing
NL appends a header to the packet coming from the transport-layer.
The header contains the IP addresses of the sender and receiver.
An IP address is a universally unique address in the network.
NL uses IP address to recognize devices on the network.
2) Routing
NL provides routing of packets.
Routing is the process of finding the best path from a source to a destination.
Routers/gateways are used for routing the packets to their final destination.
NL is concerned with circuit, message or packet switching.

Transport Layer
• Main Responsibility:
Transport-layer (TL) is responsible for process-to-process delivery of the entire message.
Process-to-process delivery means delivery from a specific process on one computer to a specific process on
the other computer.
A process is an application program running on a host.
Network-layer vs. Transport-layer:
Network-layer ensures source-to-destination delivery of individual packets.
Transport-layer ensures that the whole message arrives in order

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Other responsibilities of transport-layer (Figure 2.10 & 2.11):

Service Point Addressing
NL appends a header to the segments coming from the network-layer.
Header contains the port-address of the sender and receiver.
Network-layer vs. Transport-layer:
The network-layer gets each packet to the correct computer.
The transport-layer gets the entire message to the correct process on that computer.
2) Segmentation & Reassembly
A message is divided into segments.
Each segment contains a sequence-number.
At receiver, the sequence-numbers are used to
rearrange the segments in proper order
identify lost/duplicate segments

3) Connection Control
TL can be either i) connectionless or ii) connection-oriented.
i) In connectionless, TL
→ treats each segment as an independent packet and
→ delivers the segment to the transport-layer at the destination-machine.
ii) In connection-oriented, TL
→ first, makes a connection with the destination-machine.
→ then, delivers the packets to the destination-machine.
4) Flow Control & Error Control
Like DLL, TL is responsible for flow-control & error-control.
However, flow-control & error-control are performed end-to-end rather than node-to-node.

Session Layer
• Main Responsibility:
Session-layer (SL) establishes, maintains, and synchronizes the interaction between 2 systems.
Other responsibilities of session-layer (Figure 2.12):
Dialog Control
SL allows 2 systems to start communication with each other in half-duplex or full-duplex.
2) Synchronization
SL allows a process to add checkpoints into stream of data.
The checkpoint is a way of informing the status of the data transfer.
For example:
A checkpoint after first 500 bits of data will ensure that those 500 bits are not sent again in

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case of retransmission at 650th bit. (Checkpoints Synchronization Points)

Presentation Layer
• Main Responsibility:
Presentation-layer (PL) is concerned with syntax & semantics of the info. exchanged b/w 2 systems.

Other responsibilities of presentation-layer (Figure 2.13):

Translation
PL translates data between
format the network requires and
format the computer understands.
PL is responsible for interoperability between encoding methods as different computers use different
encoding-methods.
2) Encryption
PL performs
encryption at the sender and
decryption at the receiver.
Encryption means the sender transforms the original information to another.
Decryption means the receiver transforms the encrypted-message back to its original form.
3) Compression
PL carries out data compression to reduce the size of the data to be transmitted.
Data compression reduces the number of bits contained in the information.
Data compression ensures faster data transfer.
Data compression is important in transmitting multimedia such as audio, video, etc.

Application Layer
Main Responsibility: The application-layer (AL)
provides services to the user
enables the user to access the network.

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Other responsibilities of application-layer (Figure 2.14):

Mail Services
Directory Services
File Transfer, Access, and Management

MODULE 1(CONT.): DATA AND SIGNALS

Data and Signals

Analog & Digital Data
To be transmitted, data must be transformed to electromagnetic-signals.
Data can be either analog or digital.
Analog Data refers to information that is continuous.
For example:
The sounds made by a human voice.
2) Digital Data refers to information that has discrete states.
For example:
Data are stored in computer-memory in the form of 0s and 1s.

Analog & Digital Signals

Signals can be either analog or digital (Figure 3.2).
Analog Signal has infinitely many levels of intensity over a period of time.
Digital Signal can have only a limited number of defined values.

Periodic & Non-Periodic Signals

• The signals can take one of 2 forms: periodic or non-periodic.
1) Periodic Signal
Signals which repeat itself after a fixed time period are called Periodic Signals.
The completion of one full pattern is called a cycle.
2) Non-Periodic Signal
Signals which do not repeat itself after a fixed time period are called Non-Periodic Signals.

DIGITAL SIGNALS
Information can be represented by a digital signal.
For example:
1 can be encoded as a positive voltage.
0 can be encoded as a zero voltage (Figure 3.17a).

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2) A digital signal can have more than 2 levels (Figure 3.17b).

Example 1.1

Bit Rate
The bit rate is the number of bits sent in 1s.
The bit rate is expressed in bits per second (bps).

Example 1.2

Example 1.3

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Example 1.4

Bit Length
• The bit length is the distance one bit occupies on the transmission medium.

Digital Signal as a Composite Analog Signal

A digital signal is a composite analog signal.
A digital signal, in the time domain, comprises connected vertical and horizontal line segments.
A vertical line in the time domain means a frequency of infinity (sudden change in time);
A horizontal line in the time domain means a frequency of zero (no change in time).
Fourier analysis can be used to decompose a digital signal.
If the digital signal is periodic, the decomposed signal has a frequency domain representation with an
infinite bandwidth and discrete frequencies (Figure 3.18a).
If the digital signal is non-periodic, the decomposed signal has a frequency domain representation with an
infinite bandwidth and continuous frequencies (Figure 3.18b).

Transmission of Digital Signals

• Two methods for transmitting a digital signal:
1) Baseband transmission 2) Broadband transmission (using modulation).

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Baseband Transmission
• Baseband transmission means sending a digital signal over a channel without changing the digital signal to
an analog signal (Figure 3.19).

Baseband transmission requires that we have a low-pass channel.

Low-pass channel means a channel with a bandwidth that starts from zero.
For example, we can have a dedicated medium with a bandwidth constituting only one channel.

• Two cases of a baseband communication:

Case 1: Low-pass channel with a wide bandwidth (Figure 3.20a)
Case 2: Low-pass channel with a limited bandwidth (Figure 3.20b)
Case 1: Low-Pass Channel with Wide Bandwidth
ﾇ㩿%籖
To preserve the shape of a digital signal, we need to send the entire spectrum i.e. the
continuous range of frequencies between zero and infinity.
ﾇ㩿%籖
This is possible if we have a dedicated medium with an infinite bandwidth between the sender
and receiver.
㕵%ᬟ
If we have a medium with a very wide bandwidth, 2 stations can communicate by using
digital signals with very good accuracy (Figure 3.21).
㕵%ᬟ
Although the output signal is not an exact replica of the original signal, the data can still be
deduced from the received signal.

Case 2: Low-Pass Channel with Limited Bandwidth

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첽궉% 㶔
In a low-pass channel with limited bandwidth, we approximate the digital signal with an
analog signal.
첽궉% 㶔
The level of approximation depends on the bandwidth available.
﹅鿝%혱 Rough Approximation
Assume that we have a digital signal of bit rate N (Figure 3.22).
If we want to send analog signals to roughly simulate this signal, we need to consider the
worst case, a maximum number of changes in the digital signal.
This happens when the signal carries the sequence 01010101 . . or 10101010. . . .
To simulate these two cases, we need an analog signal of frequency f = N/2.
Let 1 be the positive peak value and 0 be the negative peak value.
We send 2 bits in each cycle; the frequency of the analog signal is one-half of the bit rate, or
N/2.
This rough approximation is referred to as using the first harmonic (N/2) frequency. The required
bandwidth is

뱼翽%쫷 Better Approximation

To make the shape of the analog signal look more like that of a digital signal, we need to add
more harmonics of the frequencies (Figure 3.23).
We can increase the bandwidth to 3N/2, 5N/2, 7N/2, and so on.
In baseband transmission, the required bandwidth is proportional to the bit rate;
If we need to send bits faster, we need more bandwidth.
Example 1.5

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Example 1.6

Broadband Transmission (Using Modulation)

淪纖%⪰ Broadband transmission or modulation means changing the digital signal to an analog signal
for transmission.
淪纖%⪰ Modulation allows us to use a bandpass channel (Figure 3.24).
淪纖%⪰ Bandpass channel means a channel with a bandwidth that does not start from zero.
淪纖%⪰ This type of channel is more available than a low-pass channel.

• If the available channel is a bandpass channel,

We cannot send the digital signal directly to the channel;
We need to convert the digital signal to an analog signal before transmission (Figure 3.25).

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Transmission Impairment
Signals travel through transmission media, which are not perfect.
The imperfection causes signal-impairment.
This means that signal at beginning of the medium is not the same as the signal at end of medium.
What is sent is not what is received.
Three causes of impairment are (Figure 3.26):
Attenuation
Distortion &
Noise.

Attenuation

As signal travels through the medium, its strength decreases as distance increases. This is called attenuation
(Figure 3.27).
As the distance increases, attenuation also increases.
For example:
Voice-data becomes weak over the distance & loses its contents beyond a certain distance.
• To compensate for this loss, amplifiers are used to amplify the signal.

1.10.1.1 Decibel
The decibel (dB) measures the relative strengths of
2 signals or
one signal at 2 different points.
The decibel is negative if a signal is attenuated. The
decibel is positive if a signal is amplified.

Variables P1 and P2 are the powers of a signal at points 1 and 2, respectively.

To show that a signal has lost or gained strength, engineers use the unit of decibel.

Example 1.7

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1-45

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Example 1.8

Example 1.9

Example 1.10

Distortion
Distortion means that the signal changes its form or shape (Figure 3.29).
Distortion can occur in a composite signal made of different frequencies.
Different signal-components
have different propagation speed through a medium.
have different delays in arriving at the final destination.
Differences in delay create a difference in phase if delay is not same as the period-duration.
Signal-components at the receiver have phases different from what they had at the sender.
The shape of the composite signal is therefore not the same.

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1-46

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Noise
Noise is defined as an unwanted data (Figure 3.30).
In other words, noise is the external energy that corrupts a signal.
Due to noise, it is difficult to retrieve the original data/information.
Four types of noise:
Thermal Noise
It is random motion of electrons in wire which creates extra signal not originally sent by
transmitter.
ii) Induced Noise
Induced noise comes from sources such as motors & appliances.
These devices act as a sending-antenna.
The transmission-medium acts as the receiving-antenna.
iii) Crosstalk
Crosstalk is the effect of one wire on the other.
One wire acts as a sending-antenna and the other as the receiving-antenna.
iv) Impulse Noise
Impulse Noise is a spike that comes from power-lines, lightning, and
so on. (spike a signal with high energy in a very short time)

1.10.3.1 Signal-to-Noise Ratio (SNR)

SNR is used to find the theoretical bit-rate limit.
SNR is defined as

SNR is actually the ratio of what is wanted (signal) to what is not wanted (noise).
A high-SNR means the signal is less corrupted by noise.
A low-SNR means the signal is more corrupted by noise.
• Because SNR is the ratio of 2 powers, it is often described in decibel units, SNRdB, defined as

Example 1.11

Data Rate Limits

Data-rate depends on 3 factors:
Bandwidth available
Level of the signals
Quality of channel (the level of noise)
Two theoretical formulas can be used to calculate the data-rate:

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Nyquist for a noiseless channel and

Shannon for a noisy channel.

Noiseless Channel: Nyquist Bit-rate

For a noiseless channel, the Nyquist bit-rate formula defines the theoretical maximum bit-
rate

where bandwidth = bandwidth of the channel

L = number of signal-levels used to
represent data BitRate = bitrate of channel
in bps
According to the formula,
By increasing number of signal-levels, we can increase the bit-rate.
Although the idea is theoretically correct, practically there is a limit.
When we increase the number of signal-levels, we impose a burden on the receiver.
If no. of levels in a signal is 2, the receiver can easily distinguish b/w 0 and 1.
If no. of levels is 64, the receiver must be very sophisticated to distinguish b/w 64
different levels.
In other words, increasing the levels of a signal reduces the reliability of the
system.

Example 1.12

Example 1.13

Example 1.14

Noisy Channel: Shannon Capacity

In reality, we cannot have a noiseless channel; the channel is always noisy.
For a noisy channel, the Shannon capacity formula defines the theoretical maximum bit-
rate.

where bandwidth = bandwidth of channel

in bps. SNR = signal-to-noise
ratio and
Capacity = capacity of channel in bps.
This formula does not consider the no. of levels of signals being transmitted (as done in the
Nyquist bit rate).
This means that no matter how many levels we have, we cannot achieve a data-rate
higher than the capacity of the channel.

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In other words, the formula defines a characteristic of the channel, not the method of
transmission.

Example 1.15

Example 1.16

Example 1.17

Performance
Bandwidth
One characteristic that measures network-performance is bandwidth.
Bandwidth of analog and digital signals is calculated in separate ways:
Bandwidth of an Analog Signal (in hz)
Bandwidth of an analog signal is expressed in terms of its frequencies.
Bandwidth is defined as the range of frequencies that the channel can carry.
It is calculated by the difference b/w the maximum frequency and the minimum
frequency.

In figure 3.13, the signal has a minimum frequency of F1 = 1000Hz and

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maximum frequency of F2 = 5000Hz.

Hence, the bandwidth is given by F2 – F1= 5000 - 1000 = 4000 Hz
(2) Bandwidth of a Digital Signal (in bps)
Bandwidth refers to the number of bits transmitted in one second in a channel (or link).
For example:
The bandwidth of a Fast Ethernet is a maximum of 100 Mbps. (This means that this
network can send 100 Mbps).
Relationship between (1) and (2)
There is an explicit relationship between the bandwidth in hertz and bandwidth in bits per seconds.
Basically, an increase in bandwidth in hertz means an increase in bandwidth in bits per second.
The relationship depends on
baseband transmission or
transmission with modulation.

Throughput
The throughput is a measure of how fast we can actually send data through a network.
Although, bandwidth in bits per second and throughput seem the same, they are actually different.
A link may have a bandwidth of B bps, but we can only send T bps through this link with T always
less than B.
In other words,
The bandwidth is a potential measurement of a link.
The throughput is an actual measurement of how fast we can send data.
For example:
We may have a link with a 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.

Example 1.18

Latency (Delay)
• The latency defines how long it takes for an entire message to completely arrive at the
destination from the time the first bit is sent out from the source.

1) Propagation Time
Propagation time is defined as the time required for a bit to travel from source to
destination.
Propagation time is given by
Propagation speed of electromagnetic signals depends on
medium and
frequency of the signal.

Example 1.19

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2) Transmission Time
The time required for transmission of a message depends on
size of the message and
bandwidth of the channel.
The transmission time is given by

Example 1.20

c
Example 1.21

3) Queuing Time
Queuing-time is the time needed for each intermediate-device to hold the message before
it can be processed.
(Intermediate device may be a router or a
switch) The queuing-time is not a fixed factor.
This is because
i) Queuing-time changes with the load imposed on the network.
ii) When there is heavy traffic on the network, the queuing-time
increases. An intermediate-device
→ queues the arrived messages and
→ processes the messages one by one.
If there are many messages, each message will have to wait.
4) Processing Delay
Processing delay is the time taken by the routers to process the packet header.

Bandwidth Delay Product

Two performance-metrics of a link are 1) Bandwidth and 2) Delay
The bandwidth-delay product is very important in data-communications.
Let us elaborate on this issue, using 2 hypothetical cases as
examples. Case 1: The following figure shows case 1
(Figure 3.32).

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Let us assume,
Bandwidth of the link = 1 bps Delay of the link = 5s.
From the figure 3.32, bandwidth-delay product is 1 x 5 = 5. Thus, there can be maximum 5
bits on the line.
There can be no more than 5 bits at any time on the link.
Case 2: The following figure shows case 2 (Figure 3.33).

Let us assume,
Bandwidth of the link = 4 bps Delay of the link = 5s.
From the figure 3.33, bandwidth-delay product is 5 x 5 = 25. Thus, there can be maximum
25 bits on the line.
At each second, there are 5 bits on the line, thus the duration of each bit is 0.20s.
The above 2 cases show that the (bandwidth X delay) is the number of bits that can fill the link.
This measurement is important if we need to
send data in bursts and
wait for the acknowledgment of each burst.
To use the maximum capability of the link
We need to make the burst-size as (2 x bandwidth x delay).
We need to fill up the full-duplex channel (two directions).
Amount (2x bandwidth x delay) is the number of bits that can be in transition at any time (Fig
3.34).

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Jitter
Another performance issue that is related to delay is jitter.
We can say that jitter is a problem
if different packets of data encounter different delays and
if the application using the data at the receiver site is time-sensitive (for ex: audio/video).
For example:
If the delay for the first packet
is 20 ms the delay for the
second is 45 ms and the delay
for the third is 40 ms
then the real-time application that uses the packets suffers from jitter.

MODULE 1(CONT.): DIGITAL TRANSMISSION

Digital To Digital Conversion

Data can be analog or digital, so can be the signal that represents it.
Signal encoding is the conversion from analog/digital data to analog/digital signal.
The possible encodings are:
Digital data to digital signal
Digital data to analog signal
Analog data to digital signal
Analog data to analog signal

LINE CODING
Line-coding is the process of converting digital-data to digital-signals (Figure 4.1).
The data may be in the form of text, numbers, graphical images, audio, or video
The data are stored in computer memory as sequences of bits (0s or 1s).
Line-coding converts a sequence of bits to a digital-signal.
At the sender, digital-data is encoded into a digital-signal.
At the receiver, digital-signal is decoded into a digital-data.

1.13.1.1 Characteristics
• Different characteristics of digital signal are
1) Signal Element Vs Data Element

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2) Data Rate Vs Signal Rate

3) Bandwidth
4) Baseline Wandering
5) DC Components
6) Built-in Error Detection
7) Self-synchronization
8) Immunity to Noise and Interference
9) Complexity

1) Data Element vs. Signal Element

Data Element Signal Element
A data-element is the smallest entity that can A signal-element is shortest unit (timewise) of
represent a piece of information (Figure 4.2). a digital-signal.
A data-element is the bit. A signal-element carries data-elements.
Data-elements are being carried. Signal-elements are the carriers.
Ratio r is defined as number of data-elements carried by each signal-element.

2) Data Rate vs. Signal Rate

Data Rate Signal Rate

The data-rate defines the number of data- The signal-rate is the number of signal-elements
elements (bits) sent in 1 sec. sent in 1 sec.
The unit is bits per second (bps). The unit is the baud.
The data-rate is sometimes called the bit-rate. The signal-rate is sometimes called the pulse
rate, the modulation rate, or the baud rate
Goal in data-communications: increase the Goal in data-communications: decrease the
data-rate. signal-rate.
Increasing the data-rate increases the speed of Decreasing the signal-rate decreases the
transmission. bandwidth requirement.
The relationship between data-rate and signal-rate is given by
where N = data-rate (in bps)
c = case factor, which varies for
each case S = number of signal-
elements and
r = previously defined factor.
This relationship depends on
value of r.
data pattern.
(If we have a data pattern of all 1s or all 0s, the signal-rate may be different from a data
pattern of alternating Os and 1s).

Bandwidth
Digital signal that carries information is non-periodic.
The bandwidth of a non-periodic signal is continuous with an infinite range.
However, most digital-signals we encounter in real life have a bandwidth with finite
values.
The effective bandwidth is finite.
The baud rate, not the bit-rate, determines the required bandwidth for a digital-signal.
More changes in the signal mean injecting more frequencies into the signal.

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(Frequency means change and change means frequency.)
The bandwidth refers to range of frequencies used for transmitting a signal.
Relationship b/w baud rate (signal-rate) and the bandwidth (range of frequencies) is given
as

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where N = data-rate (in bps)

c = case factor, which varies for
each case r = previously defined
factor
Bmin = minimum bandwidth
Baseline Wandering
While decoding, the receiver calculates a running-average of the received signal-power.
This
average is called the baseline.
The incoming signal-power is estimated against this baseline to determine the value of the
data-element.
A long string of 0s or 1s can cause a drift in the baseline (baseline wandering).
Thus, make it difficult for the receiver to decode correctly.
A good line-coding scheme needs to prevent baseline wandering.
DC Components
When the voltage-level in a digital-signal is constant for a while, the spectrum creates
very
low frequencies.
These frequencies around zero are called DC (direct-current) components.
DC components present problems for a system that cannot pass low frequencies.
For example: Telephone line cannot pass frequencies below 200 Hz.
For Telephone systems, we need a scheme with no DC component.
Built-in Error Detection
Built-in error-detecting capability has to be provided to detect the errors that occurred
during
transmission.
Self Synchronization
To correctly interpret the signals received from the sender, the receiver's bit intervals must
correspond exactly to the sender's bit intervals.
If the receiver clock is faster or slower, the bit intervals are not matched and the receiver
might misinterpret the signals.

As shown in figure 4.3, we have a situation where the receiver has shorter bit duration.
The sender sends 10110001, while the receiver receives 110111000011.
A self-synchronizing digital-signal includes timing-information in the data being
transmitted.
This can be achieved if there are transitions in the signal that alert the receiver to
the beginning, middle, or end of the pulse.
If the receiver’s clock is out-of-synchronization, these points can reset the clock.

Immunity to Noise & Interference

The code should be immune to noise and other interferences.

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Complexity
A complex scheme is more costly to implement than a simple one.
For ex: A scheme that uses 4 signal-levels is more difficult to interpret than one that uses
only 2 levels.

Example 1.22

Example 1.23

LINE CODING SCHEMES

• The Line Coding schemes are classified into 3 broad categories (Figure 4.4):

Unipolar Scheme
• All signal levels are either above or below the time axis.
NRZ (Non-Return-to-Zero)
The positive voltage defines bit 1 and the zero voltage defines bit 0 (Figure 4.5).
It is called NRZ because the signal does not return to 0 at the middle of the bit.

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Disadvantages:
Compared to polar scheme, this scheme is very costly.
Also, the normalized power is double that for polar NRZ.
Not suitable for transmission over channels with poor performance around zero
frequency.
(Normalized power power needed to send 1 bit per unit line resistance)

Polar Schemes
The voltages are on the both sides of the time axis.
Polar NRZ scheme can be implemented with two
voltages (V). For example: -V for bit 1
+V for bit 0.
Non-Return-to-Zero (NRZ)
We use 2 levels of voltage amplitude.
Two versions of polar NRZ (Figure 4.6):
NRZ-L (NRZ-Level)
The level of the voltage determines the value of the bit.
For example: i) Voltage-level for 0 can be positive and
Voltage-level for 1 can be negative.
ii) NRZ-I (NRZ-Invert)
The change or lack of change in the level of the voltage determines the value of the
bit.
If there is no change, the bit is 0;
If there is a change, the bit is 1.

Disadvantages:
Baseline wandering is a problem for both variations (NRZ-L NRZ-I).
In NRZ-L, if there is a long sequence of 0s or 1s, the average signal-power
becomes skewed.

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The receiver might have difficulty discerning the bit value.

In NRZ-I, this problem occurs only for a long sequence of 0s.
If we eliminate the long sequence of 0s, we can avoid baseline
wandering.
The synchronization problem also exists in both schemes.
A long sequence of 0s can cause a problem in both schemes.
A long sequence of 1s can cause a problem in only NRZ-L.
In NRZ-L, problem occurs when there is a sudden change of polarity in the
system. ¤ For example:
In twisted-pair cable, a change in the polarity of the wire results in
all 0s interpreted as 1s and
all 1s interpreted as 0s.
NRZ-I does not have this problem.
Both schemes have an average signal-rate of N/2 Bd.
NRZ-L and NRZ-I both have a DC component problem.

Example 1.24

Return-to-Zero (RZ)
In NRZ encoding, problem occurs when the sender-clock and receiver-clock are
not
synchronized.
Solution: Use return-to-zero (RZ) scheme (Figure 4.7).
RZ scheme uses 3 voltages: positive, negative, and zero.
There is always a transition at the middle of the bit. Either
from high to zero (for 1) or
from low to zero (for 0)

Disadvantages:
RZ encoding requires 2 signal-changes to encode a bit & .’. occupies greater
bandwidth.
Complexity: RZ uses 3 levels of voltage, which is more complex to create and
detect.

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Problem occurs when there is a sudden change of polarity in the system. This result
in
all 0s interpreted as 1s &
all 1s interpreted as 0s.

Biphase: Manchester & Differential Manchester

Manchester Encoding
This is a combination of NRZ-L & RZ schemes (RZ transition at the middle of the bit).
There is always a transition at the middle of the bit. Either
from high to low (for 0) or
from low to high (for 1).
It uses only two voltage levels (Figure 4.8).
The duration of the bit is divided into 2 halves.
The voltage
remains at one level during the first half &
moves to the other level in the second half.
The transition at the middle of the bit provides
synchronization. ii) Differential Manchester
This is a combination of NRZ-I and RZ schemes.
There is always a transition at the middle of the bit, but the bit-values are determined at the
beginning of the bit.
If the next bit is 0, there is a transition. If the next bit is 1, there is none.

Advantages:
The Manchester scheme overcomes problems associated with
NRZ-L. Differential Manchester overcomes problems
associated with NRZ-I.
There is no baseline wandering.
There is no DC component ‘.’ each bit has a positive & negative voltage
contribution.
Disadvantage:
Signal-rate: Signal-rate for Manchester & diff. Manchester is double that for NRZ.

Bipolar Schemes (or Multilevel Binary)

This coding scheme uses 3 voltage levels (Figure 4.9):
positive
negative &

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zero.
Two variations of bipolar encoding:
AMI (Alternate Mark Inversion)
Pseudoternary
i) AMI
Binary 0 is represented by a neutral 0 voltage (AMI Alternate 1 Inversion).
Binary 1s are represented by alternating positive and negative voltages.
ii) Pseudoternary
Binary 1 is represented by a neutral 0 voltage.
Binary 0s are represented by alternating positive and negative voltages.

Advantages:
The bipolar scheme has the same signal-rate as NRZ.
There is no DC component ‘.’ each bit has a positive & negative voltage contribution.
The concentration of the energy is around frequency N/2.
Disadvantage:
AMI has a synchronization problem when a long sequence of 0s is present in the data.

Assignment Questions
1. Define data communications. Explain its 4 fundamental characteristics.
2. Explain different components of data communication system.
3. Explain different forms of information.
4. Describe simplex, half-duplex and full duplex methods of data flow.
5. Explain the 3 criteria necessary for an effective and efficient network.
6. Explain point to point and multipoint connection.
7. Explain the following topologies:
i) Meshii) Star iii) Bus iv) Ring
8. Explain in detail LAN & WAN. List the differences between LAN & WAN.
9. Explain circuit-switched and packet-switched networks.
10. Explain components of Internet.
11. Explain different method of accessing the Internet.
12. With respect to RFC, explain the following:
i) Maturity Levels ii) Requirement Levels.
13. What are internet standards? Explain the functions of standard organizations.
14. Explain TCP/IP architecture with a layer diagram.

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15. List the 5 layers and its functionality in TCP/IP model.

16. With respect to in TCP/IP model, explain the following:
i) Encapsulation and decapsulation. ii) Multiplexing and demultiplexing.
17. Explain four levels of addressing employed in TCP/IP protocol.
18. What are the uses of a layered network model? Compare OSI and TCP/IP models.

Expected Outcomes
The students understand about
• DATA COMMUNICATIONS and its Components
• Differences between NETWORK topologies &
NETWORK TYPES
• Internet Standards and ADMINISTRATION
• PROTOCOL LAYERING,TCP/IP PROTOCOL
SUITE & THE OSI MODEL
• DIGITAL SIGNALS
• TRANSMISSION IMPAIRMENT
• DIGITAL-TO-DIGITAL& DIGITAL TO
ANALOG Conversion Techniques

Further Reading
1. http://www.mhhe.com/engcs/compsci/forouzan/frontmat
ter.pdf
2. http://freecomputerbooks.com/Introduction-to-Data-
Communications.html

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MODULE 2

ANALOG-TO-DIGITAL CONVERSION
PCM
Sampling
Sampling Rate
Quantization
Quantization Levels
Quantization Error
Uniform vs. Non Uniform Quantization
Encoding
Original Signal Recovery
PCM Bandwidth
Maximum Data Rate of a Channel
Minimum Required Bandwidth
TRANSMISSION MODES
PARALLEL TRANSMISSION
SERIAL TRANSMISSION
Asynchronous Transmission
Synchronous Transmission
Isochronous
DIGITAL TO ANALOG CONVERSION
Aspects of Digital to Analog Conversion
Amplitude Shift Keying (ASK)
Binary ASK (BASK)
Implementation of BASK
Bandwidth for ASK
Frequency Shift Keying (FSK)
Binary FSK (BFSK)
Implementation of BFSK
Bandwidth for BFSK
Phase Shift Keying (PSK)
Binary PSK (BPSK)
Implementation of BPSK
Bandwidth for BPSK
Quadrature PSK (QPSK)
Constellation Diagram
Quadrature Amplitude Modulation (QAM)
Bandwidth for QAM
MULTIPLEXING
Frequency Division Multiplexing (FDM)
Multiplexing Process
Demultiplexing Process
Applications of FDM
The Analog Carrier System
Wavelength-Division Multiplexing (WDM)
Time Division Multiplexing (TDM)
Synchronous TDM
Time Slots and Frames
Interleaving
Empty Slots
Data Rate Management
Frame Synchronizing
Statistical TDM
SPREAD-SPECTRUM
Frequency Hopping Spread Spectrum (FHSS)
Bandwidth Sharing

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Direct Sequence Spread Spectrum (DSSS)

Bandwidth Sharing
SWITCHING
Three Methods of Switching
Switching and TCP/IP Layers
CIRCUIT SWITCHED NETWORK
Three Phases
Efficiency
Delay
PACKET SWITCHED NETWORK
Datagram Networks
Routing Table
Destination Address
Efficiency
Delay
Virtual Circuit Network
Addressing
Three Phases
Data Transfer Phase
Setup Phase
Setup Request
Teardown Phase
Efficiency
Delay in Virtual Circuit Networks

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MODULE 2
DIGITAL TRANSMISSION (CONT.)
STRUCTURE

Introduction
Objectives
Analog To Digital Conversion
Digital To Analog Conversion
Multiplexing
Spread Spectrum
Switching
Circuit Switched Network
Packet Switched Network
Assignment Questions
Expected Outcomes
2.12Further Reading

Introduction
This module

Analog To Digital Conversion

An analog-signal may created by a microphone or camera.
To change an analog-signal to digital-data, we use PCM (pulse code modulation).
After the digital-data are created (digitization), then we convert the digital-data to a digital-signal.

PCM
PCM is a technique used to change an analog signal to digital data (digitization).
PCM has encoder at the sender and decoder at the receiver.
The encoder has 3 processes (Figure 4.21):
Sampling
Quantization &
Encoding.

Sampling
We convert the continuous time signal (analog) into the discrete time signal (digital).
Pulses from the analog-signal are sampled every Ts sec
where Ts is the sample-interval or period.
The inverse of the sampling-interval is called the sampling-frequency (or sampling-rate).
Sampling-frequency is given by

Three sampling methods (Figure 4.22):

1) Ideal Sampling

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This method is difficult to implement.

2) Natural Sampling
A high-speed switch is turned ON for only the small period of time when the sampling occurs.
The result is a sequence of samples that retains the shape of the analog-signal.
3) Flat Top Sampling
The most common sampling method is sample and hold.
Sample and hold method creates flat-top samples.
This method is sometimes referred to as PAM (pulse amplitude modulation).

2.1.1.1.1 Sampling Rate

• According to Nyquist theorem,
“The sampling-rate must be at least 2 times the highest frequency, not the bandwidth“.
If the analog-signal is low-pass, the bandwidth and the highest frequency are the same value
(Figure 4.23a).
If the analog-signal is bandpass, the bandwidth value is lower than the value of the
maximum frequency (Figure 4.23b).

Quantization
The sampled-signal is quantized.
Result of sampling is a set of pulses with amplitude-values b/w max & min amplitudes of the signal.
Four steps in quantization:
We assume that the original analog-signal has amplitudes between Vmin & Vmax.
We divide the range into L zones, each of height Δ(delta).

where L = number of levels.

We assign quantized values of 0 to (L-1) to the midpoint of each zone.
We approximate the value of the sample amplitude to the quantized values.
• For example: Let Vmin=-20 Vmax =+20 V L = 8 Therefore, = [+20-(-20)]/8= 5 V
In the chart (Figure 4.26),
First row is normalized-PAM-value for each sample.
Second row is normalized-quantized-value for each sample.
Third row is normalized error (which is diff. b/w normalized PAM value & quantized values).
Fourth row is quantization code for each sample.
Fifth row is the encoded words (which are the final products of the conversion).

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Let L = number of levels.

The choice of L depends on
range of the amplitudes of the analog-signal and
how accurately we need to recover the signal.
If the signal has only 2 amplitude values, we need only 2 quantization-levels.
If the signal (like voice) has many amplitude values, we need more quantization-levels.
In audio digitizing, L is normally chosen to be 256.
In video digitizing, L is normally thousands.
Choosing lower values of L increases the quantization-error.

Quantization Error
Quantization-error is the difference b/w normalized PAM value & quantized values
Quantization is an approximation process.
The input values to the quantizer are the real values.
The output values from the quantizer are the approximated values.
The output values are chosen to be the middle value in the zone.
If the input value is also at the middle of the zone,
Then, there is no error.
Otherwise, there is an error.
• In the previous example,
The normalized amplitude of the third sample is 3.24, but the normalized quantized value is
This means that there is an error of +0.26.

Uniform vs. Non Uniform Quantization

Non-uniform quantization can be done by using a process called companding and expanding.
The signal is companded at the sender before conversion.
The signal is expanded at the receiver after conversion.
Companding means reducing the instantaneous voltage amplitude for large values.
Expanding means increasing the instantaneous voltage amplitude for small values.
It has been proved that non-uniform quantization effectively reduces the SNRdB of quantization.

Encoding
The quantized values are encoded as n-bit code word.
In the previous example,
A quantized value 2 is encoded as 010.
A quantized value 5 is encoded as 101.
• Relationship between number of quantization-levels (L) & number of bits (n) is given by
n=log2L or 2n=L
• The bit-rate is given by:

Example 2.1

Example 2.2

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Example 2.3

Example 2.4

Original Signal Recovery

PCM decoder is used for recovery of the original signal.
Here is how it works (Figure 4.27):
The decoder first uses circuitry to convert the code words into a pulse that holds the amplitude until
the next pulse.
Next, the staircase-signal is passed through a low-pass filter to smooth the staircase signal into an
analog-signal.
The filter has the same cut-off frequency as the original signal at the sender.
If the signal is sampled at the Nyquist sampling-rate, then the original signal will be re-created.
The maximum and minimum values of the original signal can be achieved by using amplification.

PCM Bandwidth
The minimum bandwidth of a line-encoded signal is

We substitute the value of N in above formula:

When 1/r = 1 (for a NRZ or bipolar signal) and c = (1/2) (the average situation), the minimum bandwidth is
This means the minimum bandwidth of the digital-signal is nb times greater than the bandwidth of the
analog-signal.

Maximum Data Rate of a Channel

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The Nyquist theorem gives the data-rate of a channel as

We can deduce above data-rate from the Nyquist sampling theorem by using the following arguments.
We assume that the available channel is low-pass with bandwidth B.
We assume that the digital-signal we want to send has L levels, where each level is a signal-element.
This means r = 1/log2L.
We first pass digital-signal through a low-pass filter to cut off the frequencies above B Hz.
We treat the resulting signal as an analog-signal and sample it at 2 x B samples per second and
quantize it using L levels.
The resulting bit-rate is

This is the maximum bandwidth; if the case factor c increases, the data-rate is reduced.

Minimum Required Bandwidth

• The previous argument can give us the minimum bandwidth if the data-rate and the number of signal-
levels are fixed. We can say

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Transmission Modes
• Two ways of transmitting data over a link (Figure 4.31): 1) Parallel mode & 2) Serial mode.

2.1.1 PARALLEL TRANSMISSION

Multiple bits are sent with each clock-tick (Figure 4.32).
„n‟ bits in a group are sent simultaneously.
„n‟ wires are used to send „n‟ bits at one time.
Each bit has its own wire.
Typically, the 8 wires are bundled in a cable with a connector at each end.

Advantage:
Speed: Parallel transmission can increase the transfer speed by a factor of n over serial transmission.
Disadvantage:
Cost: Parallel transmission requires n communication lines just to transmit the data-stream.
Because this is expensive, parallel transmission is usually limited to short distances.

SERIAL TRANSMISSION
• One bit is sent with each clock-tick using only a single link (Figure 4.33).

Advantage:
Cost: Serial transmission reduces cost of transmission over parallel by a factor of n.
Disadvantage:
Since communication within devices is parallel, following 2 converters are required at interface:
Parallel-to-serial converter
Serial-to-parallel converter
Three types of serial transmission: asynchronous, synchronous, and isochronous.

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Asynchronous Transmission
Asynchronous transmission is so named because the timing of a signal is not important (Figure 4.34).
Prior to data transfer, both sender & receiver agree on pattern of information to be exchanged.
Normally, patterns are based on grouping the bit-stream into bytes.
The sender transmits each group to the link without regard to a timer.
As long as those patterns are followed, the receiver can retrieve the info. without regard to a timer.
There may be a gap between bytes.
We send
1 start bit (0) at the beginning of each byte
1 stop bit (1) at the end of each byte.
Start bit alerts the receiver to the arrival of a new group.
Stop bit lets the receiver know that the byte is finished.
• Here, the term asynchronous means “asynchronous at the byte level”.
However, the bits are still synchronized & bit-durations are the same.

Disadvantage:
Slower than synchronous transmission. (Because of stop bit, start bit and gaps)
Advantages:
Cheap & effective.
Useful for low-speed communication.

Synchronous Transmission
We send bits one after another without start or stop bits or gaps (Figure 4.35).
The receiver is responsible for grouping the bits.
The bit-stream is combined into longer "frames," which may contain multiple bytes.
If the sender wants to send data in separate bursts, the gaps b/w bursts must be filled with a special sequence
of 0s & 1s (that means idle).

Advantages:
Speed: Faster than asynchronous transmission. („.‟ of no stop bit, start bit and gaps).
Useful for high-speed applications such as transmission of data from one computer to another.

Isochronous
Synchronization between characters is not enough; the entire stream of bits must be synchronized.
The isochronous transmission guarantees that the data arrive at a fixed rate.
In real-time audio/video, jitter is not acceptable. Therefore, synchronous transmission fails.

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For example: TV images are broadcast at the rate of 30 images per second. The images must be viewed at the
same rate.

MODULE 2(CONT.): ANALOG TRANSMISSION

Digital To Analog Conversion

• Digital-to-analog conversion is the process of changing one of the characteristics of an analog-signal
based on the information in digital-data (Figure 5.1).

A sine wave can be defined by 3 attributes:

Amplitude
Frequency &
Phase.
When anyone of the 3 attributes of a wave is varied, a different version of the wave will be created.
So, by changing one attribute of an analog signal, we can use it to represent digital-data.
Four methods of digital to analog conversion (Figure 5.2):
Amplitude shift keying (ASK)
Frequency shift keying (FSK)
Phase shift keying (PSK)
Quadrature amplitude modulation (QAM).
QAM is a combination of ASK and PSK i.e. QAM combines changing both the amplitude and phase.
QAM is the most efficient of these 4 methods.
QAM is the method commonly used today.

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Aspects of Digital-to-Analog Conversion

1) Data-element vs. Signal-element
A data-element is the smallest piece of information to be exchanged i.e. the bit. A
signal-element is the smallest unit of a signal that is transmitted.
2) Data Rate vs. Signal Rate
Data rate (Bit rate) is the number of bits per second.
Signal-rate (Baud rate) is the number of signal elements per second.
The relationship between data-rate(N) and the signal-rate(S) is

where r = number of data-elements carried in one signal-element.

The value of r is given by
r = log2L or 2r=L
where L = type of signal-element (not the level)
(In transportation,
a baud is analogous to a vehicle, and
a bit is analogous to a passenger.
We need to maximize the number of people per car to reduce the traffic).
3) Carrier Signal
The sender produces a high-frequency signal that acts as a base for the information-signal.
This base-signal is called the carrier-signal (or carrier-frequency).
The receiver is tuned to the frequency of the carrier-signal that it expects from the sender.
Then, digital-information changes the carrier-signal by modifying its attributes (amplitude,
frequency, or phase). This kind of modification is called modulation (shift keying).
4) Bandwidth
In both ASK & PSK, the bandwidth required for data transmission is proportional to the signal-rate.
In FSK, the bandwidth required is the difference between the two carrier-frequencies.

Example 2.5

Example 2.6

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Amplitude Shift Keying (ASK)

•The amplitude of the carrier-signal is varied to represent different signal-elements.
Both frequency and phase remain constant for all signal-elements.
Binary ASK (BASK)
BASK is implemented using only 2 levels. (Figure 5.3)
This is also referred to as OOK (On-Off Keying).

Implementation of BASK
Here, line coding method used = unipolar NRZ (Figure 5.4).
The unipolar NRZ signal is multiplied by the carrier-frequency coming from an oscillator.
When amplitude of the NRZ signal = 0, amplitude of the carrier-signal = 0.
When amplitude of the NRZ signal = 1, the amplitude of the carrier-signal is held.

Bandwidth for ASK

Here, the bandwidth (B) is proportional to the signal-rate (S) (Figure 5.5)
The bandwidth is given by

where d(0<d<1)= this factor depends on modulation and filtering-process.

Example 2.7

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Frequency Shift Keying (FSK)

The frequency of the carrier-signal is varied to represent different signal-elements.
The frequency of the modulated-signal is constant for the duration of one signal-element, but changes for the
next signal-element if the data-element changes.
Both amplitude and phase remain constant for all signal-elements.

Binary FSK (BFSK)

This uses 2 carrier-frequencies: f1 and f2. (Figure 5.6)
When data-element = 1, first carrier frequency(f1) is used.
When data-element = 0, second carrier frequency(f2) is used.

Implementation
Here, line coding method used = unipolar NRZ.
Two implementations of BFSK: i) Coherent and ii) Non-Coherent.
Coherent BFSK Non Coherent BFSK
The phase continues through the boundary of There may be discontinuity in the phase when
two signal-elements (Figure 5.7). one signal-element ends and the next begins.
This is implemented by using one voltage- This is implemented by
controlled oscillator (VCO). → treating BFSK as 2 ASK modulations and
VCO changes frequency according to the input → using 2 carrier-frequencies
voltage.
When the amplitude of NRZ signal = 0, the VCO
keeps its regular frequency.
When the amplitude of NRZ signal = 0, the VCO
increases its frequency.

Bandwidth for BFSK

FSK has two ASK signals, each with its own carrier-frequency f1 or f2. (Figure 5.6)
The bandwidth is given by
where 2∆f is the difference between f1 and f2,
Example 2.8

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Example 2.9

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Phase Shift Keying (PSK)

The phase of the carrier-signal is varied to represent different signal-elements.
Both amplitude and frequency remain constant for all signal-elements.

Binary PSK (BPSK)

We have only two signal-elements:
First signal-element with a phase of 0°.
Second signal-element with a phase of 180° (Figure 5.9).
ASK vs. PSK
In ASK, the criterion for bit detection is the amplitude of the signal.
In PSK, the criterion for bit detection is the phase.
Advantages:
PSK is less susceptible to noise than ASK.
PSK is superior to FSK because we do not need 2 carrier-frequencies.
Disadvantage:
PSK is limited by the ability of the equipment to distinguish small differences in phase.

Implementation
The implementation of BPSK is as simple as that for ASK. (Figure 5.10).
The signal-element with phase 180° can be seen as the complement of the signal-element with phase 0°.
Here, line coding method used: polar NRZ.
The polar NRZ signal is multiplied by the carrier-frequency coming from an oscillator.
When data-element = 1, the phase starts at 0°.
When data-element = 0, the phase starts at 180°.

Bandwidth for BPSK

The bandwidth is the same as that for BASK, but less than that for BFSK. (Figure 5.9b)
No bandwidth is wasted for separating 2 carrier-signals.

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Quadrature PSK (QPSK)

The scheme is called QPSK because it uses 2 separate BPSK modulations (Figure 5.11):
First modulation is in-phase,
Second modulation is quadrature (out-of-phase).
A serial-to-parallel converter
accepts the incoming bits
sends first bit to first modulator and
sends second bit to second modulator.
The bit to each BPSK signal has one-half the frequency of the original signal.
Advantages:
Decreases the baud rate.
Decreases the required bandwidth.

As shown in Figure 5.11, the 2 composite-signals created by each multiplier are 2 sine waves with the same
frequency, but different phases.
When the 2 sine waves are added, the result is another sine wave, with 4 possible phases: 45°, -45°, 135°,
and -135°.
There are 4 kinds of signal-elements in the output signal (L=4), so we can send 2 bits per signal-element
(r=2).

Example 2.10

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Constellation Diagram
A constellation diagram can be used to define the amplitude and phase of a signal-element.
This diagram is particularly useful
when 2 carriers (one in-phase and one quadrature) are used.
when dealing with multilevel ASK, PSK, or QAM.
In a constellation diagram, a signal-element type is represented as a dot.
The diagram has 2 axes (Figure 5.12):
The horizontal X axis is related to the in-phase carrier.
The vertical Y axis is related to the quadrature carrier.
For each point on the diagram, 4 pieces of information can be deduced.
The projection of point on the X axis defines the peak amplitude of the in-phase component.
The projection of point on Y axis defines peak amplitude of the quadrature component.
The length of the line that connects the point to the origin is the peak amplitude of the signal-element
(combination of the X and Y components);
The angle the line makes with the X axis is the phase of the signal-element.

Example 2.11

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Quadrature Amplitude Modulation (QAM)

This is a combination of ASK and PSK.
Main idea: Using 2 carriers, one in-phase and the other quadrature, with different amplitude levels for each
carrier.
There are many variations of QAM (Figure 5.14).
Figure 5.14a shows the 4-QAM scheme using a unipolar NRZ signal. This is same as BASK.
Figure 5.14b shows another QAM using polar NRZ. This is the same as QPSK.
Figure 5.14c shows another 4-QAM in which we used a signal with 2 positive levels to modulate
each of the 2 carriers.
Figure 5.14d shows a 16-QAM constellation of a signal with 8 levels, 4 positive & 4 negative.

2.3.5.1 Bandwidth for QAM

The bandwidth is same as in ASK and PSK transmission.
QAM has the same advantages as PSK over ASK.

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MODULE 2(CONT.): BANDWIDTH UTILIZATION --

MULTIPLEXING AND SPREADING

Multiplexing
When bandwidth of a medium is greater than bandwidth needs of the devices, the link can be shared.
Multiplexing allows simultaneous transmission of multiple signals across a single data-link (Fig 4.21).
The traffic increases, as data/telecommunications use increases.
We can accommodate this increase by
adding individual links, each time a new channel is needed or
installing higher-bandwidth links to carry multiple signals.
Today's technology includes high-bandwidth media such as optical-fiber and satellite microwaves.
Each has a bandwidth far in excess of that needed for the average transmission-signal.
If the bandwidth of a link is greater than the bandwidth needs of the devices connected to it, the bandwidth
is wasted.
An efficient system maximizes the utilization of all resources; bandwidth is one of the most precious
resources we have in data communications.

In a multiplexed-system, „n‟ lines share the bandwidth of one link.

MUX combines transmission-streams from different input-lines into a single stream (many-to-one).
At the receiving-end, that stream is fed into a demultiplexer (DEMUX).
DEMUX
separates the stream back into its component-transmissions (one-to-many) and
directs the transmission-streams to different output-lines.
Link vs. Channel:
The link refers to the physical path.
The channel refers to the portion of a link that carries a transmission between a given pair of lines.
One link can have many channels.
Three multiplexing techniques (Figure 6.2):
Frequency-division multiplexing (FDM)
Wavelength-division multiplexing (WDM) and
Time-division multiplexing (TDM).
The first two techniques are used for analog-signals.
The third one technique is used for digital-signals.

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Frequency Division Multiplexing (FDM)

FDM is an analog multiplexing technique that combines analog signals (Figure 6.3).
FDM can be used when the bandwidth of a link is greater than the combined bandwidths of the signals to be
transmitted. (Bandwidth measured in hertz).

Multiplexing Process
Here is how it works (Figure 6.4):
Each sending-device generates modulated-signals with different carrier-frequencies (f1, f2, & f3).
Then, these modulated-signals are combined into a single multiplexed-signal.
Finally, the multiplexed-signal is transported by the link.

Carrier-frequencies are separated by sufficient bandwidth to accommodate the modulated-signal.

Channels can be separated by strips of unused bandwidth called guard bands.
Guard bands prevent signals from overlapping.
In addition, carrier-frequencies must not interfere with the original data frequencies.
Although FDM is considered as analog multiplexing technique, the sources can produce digital-signal.
The digital-signal can be sampled, changed to analog-signal, and then multiplexed by using FDM.

Demultiplexing Process
Here is how it works (Figure 6.5):
The demultiplexer uses filters to divide the multiplexed-signal into individual-signals.
Then, the individual signals are passed to a demodulator.
Finally, the demodulator
separates the individual signals from the carrier signals and
passes the individual signals to the output-lines.

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Example 2.12

Example 2.13

Example 2.14

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Applications of FDM
To maximize the efficiency of their infrastructure, Telephone-companies have traditionally
multiplexed signals from lower-bandwidth lines onto higher-bandwidth lines.
A very common application of FDM is AM and FM radio broadcasting.
The first generation of cellular telephones (still in operation) also uses FDM.

Analog Carrier System

To maximize the efficiency, telephone-companies have multiplexed-signals from lower-bandwidth lines
onto higher-bandwidth lines.
Many switched or leased lines are combined into bigger channels.
For analog lines, FDM is used.
One of these hierarchical systems used by AT&T is made up of (Figure 6.9):
Groups
Super groups
Master groups, and
Jumbo groups

Group: In the analog hierarchy, 12 voice channels are multiplexed onto a higher-bandwidth line to
create a group.
A group has 48 kHz of bandwidth and supports 12 voice channels.
Super Group: At the next level, up to five groups can be multiplexed to create a composite-signal
called a supergroup.
A supergroup has a bandwidth of 240 kHz and supports up to 60 voice channels.
Supergroups can be made up of either five groups or 60 independent voice channels.
Master Groups: At the next level, 10 supergroups are multiplexed to create a master group.
A master group must have 2.40 MHz of bandwidth, but the need for guard bands
between the supergroups increases the necessary bandwidth to 2.52 MHz.
Master groups support up to 600 voice channels.
Jumbo Group: Finally, six master groups can be combined into a jumbo group.
A jumbo group must have 15.12 MHz (6 x 2.52 MHz) of bandwidth, but the need for
guard bands b/w the master groups increases the necessary bandwidth to 16.984 MHz

Example 2.15

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Wavelength-Division Multiplexing (WDM)

WDM is an analog multiplexing technique that combines analog signals (Figure 6.10).
WDM is designed to use the high-data-rate capability of fiber optical-cable.
The data-rate of optical-cable is higher than the data-rate of metallic-cable.
Using an optical-cable for one single line wastes the available bandwidth.
Multiplexing allows combining several lines into one line.
WDM is same as FDM with 2 exceptions:
Multiplexing & demultiplexing involve optical-signals transmitted through optical-cable.
The frequencies are very high.

Here is how it works (Figure 6.11):

A multiplexer combines several narrow-bands of light into a wider-band of light.
A demultiplexer divides a wider-band of light into several narrow-bands of light.
A prism is used for combining and splitting of light sources
A prism bends a beam of light based on
angle of incidence and
frequency.

Applications of WDM:
SONET network: Multiple optical-fiber lines can be multiplexed and demultiplexed.
Dense WDM (DWDM) can multiplex a very large number of channels by spacing channels very
close to one another. DWDM achieves even greater efficiency
Time-Division Multiplexing (TDM)
TDM is a digital multiplexing technique that combines digital signals (Figure 6.12).
TDM combines several low-rate channels into one high-rate one.
FDM vs. TDM
In FDM, a portion of the bandwidth is shared.
In TDM, a portion of the time is shared.
Each connection occupies a portion of time in the link.
Several connections share the high bandwidth of a line.

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As shown in Figure 6.12, the link is divided by time.

Portions of signals 1, 2, 3, and 4 occupy the link sequentially.
Digital-data from different sources are combined into one timeshared link.
Although TDM is considered as digital multiplexing technique, the sources can produce analog-signal.
The analog data can be sampled, changed to digital-data, and then multiplexed by using TDM.
Two types of TDM:
Synchronous and
Statistical.
Synchronous TDM
Time Slots & Frames
Each input-connection has an allotment in the output-connection even if it is not sending data.
The data-flow of input-connection is divided into units (Figure 6.13).
A unit can be 1 bit, 1 character, or 1 block of data.
Each input-unit occupies one input-time-slot.
Each input-unit
becomes one output-unit and
occupies one output-time-slot.
However, duration of output-time-slot is „n‟ times shorter than duration of input-time-slot.
If an input-time-slot is T s, the output-time-slot is T/n s
where n = No. of connections.
• In the output-connection, a unit has a shorter duration & therefore travels faster.

In Figure 6.14, n = 3.
A set of data-units from each input-connection is grouped into a frame.
For example:
If there are 3 connections, a frame is divided into 3 time-slots.
One slot is allocated for each data-unit.
One data-unit is used for each input-line.

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Example 2.16

Example 2.17

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Example 2.18

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Interleaving
TDM can be seen as 2 fast-rotating switches (Figure 6.15):
First switch on the multiplexing-side and
Second switch on the demultiplexing-side.
The switches are synchronized and rotate at the same speed, but in opposite directions.
On the multiplexing-side (Figure 6.16)
As the switch opens in front of a connection, that connection has the opportunity to send a unit onto
the path. This process is called interleaving.
2) On the demultiplexing-side
As the switch opens in front of a connection, that connection has the opportunity to receive a unit
from the path.

Example 2.19

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Example 2.20

Empty Slots
• Problem: Synchronous TDM is not efficient.
For example: If a source does not have data to send, the corresponding slot in the output-frame is
empty.

• As shown in Figure 6.18,

Second input-line has no data to send
Third input-line has discontinuous data.
The first output-frame has 3 slots filled.
The second frame has 2 slots filled.
The third frame has 3 slots filled.
No frame is full.
Solution: Statistical TDM can improve the efficiency by removing the empty slots from the frame.

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Data Rate Management

Problem in TDM: How to handle differences in the input data-rates?
If data-rates are not the same, three strategies can be used.
Three different strategies: 1) Multilevel multiplexing 2) Multiple-slot allocation and 3) Pulse stuffing
1) Multilevel Multiplexing
This technique is used when the data-rate of an input-line is a multiple of others.

For example:
As shown in Figure 6.19, we have 2 inputs of 20 kbps and 3 inputs of 40 kbps.
The first 2 input-lines can be multiplexed to provide a data-rate of 40 kbps.
2) Multiple Slot Allocation
Sometimes it is more efficient to allot more than 1 slot in a frame to a single input-line.

For example:
Data-rate of multiple input-lines can be data-rate of one input-line.
As shown in Figure 6.20, the input-line with a 50-kbps data-rate can be given 2 slots in the output-
line.
In first input line, serial-to-parallel converter is used. The converter creates two 25 kbps input lines
out of one 50 kbps input line.
3) Pulse Stuffing
Sometimes the bit-rates of sources are not multiple integers of each other. .‟. above 2 techniques
cannot be used.
Solution:
Make the highest input data-rate the dominant data-rate.
Then, add dummy bits to the input-lines with lower rates.
This will increase data rates of input-line.
This technique is called pulse stuffing, bit padding, or bit stuffing.

As shown in Figure 6.21, the input-line with a lower data-rate = 46kbps is pulse-stuffed to increase
the data-rate to 50 kbps.
Now, multiplexing can take place.

Frame Synchronizing
• Problem: Synchronization between the multiplexer and demultiplexer is a major issue.
If the multiplexer and the demultiplexer are not synchronized, a bit belonging to one channel
may be received by the wrong channel.
Solution: Usually, one or more synchronization-bits are added to the beginning of each frame. These bits are
called framing-bits.

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The framing-bits follow a pattern (frame-to-frame) that allows multiplexer and demultiplexer to
synchronize.
As shown in Figure 6.22, the synchronization-information
→ consists of 1 bit per frame and
→ alternates between 0 & 1.

Example 2.21

Example 2.22

Statistical TDM
• Problem: Synchronous TDM is not efficient.
For ex: If a source does not have data to send, the corresponding slot in the output-frame is empty.
Solution: Use statistical TDM.
Slots are dynamically allocated to improve bandwidth-efficiency.
Only when an input-line has data to send, the input-line is given a slot in the output-frame.
The number of slots in each frame is less than the number of input-lines.

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The multiplexer checks each input-line in round robin fashion.

If the line has data to send;
Then, multiplexer allocates a slot for an input-line;
Otherwise, multiplexer skips the line and checks the next line.

In synchronous TDM (Figure 6.26a), some slots are empty because the corresponding line does not have
data to send.
In statistical TDM (Figure 6.26b), no slot is left empty.
Addressing
Synchronous TDM Statistical TDM
An output-slot needs to carry only data of the An output-slot needs to carry both data &
destination (Figure 6.26a). address of the destination (Figure 6.26b).
There is no need for addressing. There is no fixed relationship between the
Synchronization and pre-assigned relationships inputs and outputs because there are no pre-
between the inputs and outputs serve as an assigned or reserved slots.
address. We need to include the address of the receiver
inside each slot to show where it is to be
delivered.
2) Slot Size
Usually, a block of data is many bytes while the address is just a few bytes.
A slot carries both data and address.
Therefore, address-size must be very small when compared to data-size.
This results in efficient transmission.
For example:
It will be inefficient to send 1 bit per slot as data, when the address is 3 bits.
This means an overhead of 300%.
3) No Synchronization Bit
In statistical TDM, the frames need not be synchronized, so synchronization-bits are not needed.
4) Bandwidth
Normally, the capacity of the link is less than the sum of the capacities of each channel.
The designers define the capacity of the link based on the statistics of the load for each
channel.

Spread Spectrum
Spread-spectrum is used in wireless applications (Figure 6.27).
In wireless applications, all stations use air (or a vacuum) as the medium for communication.
Goal: Stations must be able to share the air medium without interception by an attacker.
Solution: Spread-spectrum techniques add redundancy i.e. they spread the original spectrum needed for each
station.
If the required bandwidth for each station is B, spread-spectrum expands it to Bss such that Bss>>B.
The expanded-bandwidth allows the source to place its message in a protective envelope for a more secure
transmission.
(An analogy is the sending of a delicate, expensive gift. We can insert the gift in a special box to

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prevent it from being damaged during transportation).

Spread-spectrum achieves its goal through 2 principles:

The bandwidth allocated to each station needs to be, by far, larger than what is needed. This allows
redundancy.
The spreading process must occur after the signal is created by the source.
After the signal is created by the source, the spreading process
uses a spreading-code and
spreads the bandwidth.
The spreading-code is a series of numbers that look random, but are actually a pattern.
Two types of spread-spectrum:
Frequency hopping spread-spectrum (FHSS) and
Direct sequence spread-spectrum (DSSS).

Frequency Hopping Spread-Spectrum (FHSS)

This technique uses „M‟ different carrier-frequencies that are modulated by the source-signal.
At one moment, the signal modulates one carrier-frequency.
At the next moment, the signal modulates another carrier-frequency.
Although the modulation is done using one carrier-frequency at a time, 'M‟ frequencies are used in the long
run.
The bandwidth occupied by a source is given by

As shown in Figure 6.28.

A pseudorandom code generator (PN) creates a k-bit pattern for every hopping period Th.
The frequency-table
uses the pattern to find the frequency to be used for this hopping period and
passes the frequency to the frequency-synthesizer.

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The frequency-synthesizer creates a carrier-signal of that frequency.

The source-signal modulates the carrier-signal.

As shown in Figure 6.29, assume we have 8 hopping frequencies.

Here, M = 8 and k = 3.
The pseudorandom code generator will create 8 different 3-bit patterns.
These are mapped to 8 different frequencies in the frequency table (see Figure 6.29).
The pattern for this station is 101, 111, 001, 000, 010, 111 & 100.
At hopping-period 1, the pattern is 101.
The frequency selected is 700 kHz; the source-signal modulates this carrier-frequency. 2)
At hopping-period 2, the pattern is 111.
The frequency selected is 900 kHz; the source-signal modulates this carrier-frequency.

If there are many k-bit patterns & the hopping period is short, a sender & receiver can have privacy. If an
attacker tries to intercept the transmitted signal, he can only access a small piece of data because he
does not know the spreading sequence to quickly adapt himself to the next hop.

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The scheme has also an anti-jamming effect.

A malicious sender may be able to send noise to jam the signal for one hopping period (randomly),
but not for the whole period.

Bandwidth Sharing
If the number of hopping frequencies is M, we can multiplex M channels into one by using the same Bss
bandwidth.
This is possible because
A station uses just one frequency in each hopping period.
Other M-1 stations uses other M-1 frequencies.
In other words, M different stations can use the same Bss if a multiple FSK (MFSK) is used.

Direct Sequence Spread-spectrum (DSSS)

This technique expands the bandwidth of the original signal.
Each data-bit is replaced with „n‟ bits using a spreading-code.
Each bit is assigned a code of „n‟ bits called chips.
The chip-rate is „n‟ times that of the data-bit (Figure 6.32).

For example (Figure 6.33):

Consider the Barker sequence used in a wireless LAN. Here n =11.
Assume that the original signal and the chips in the chip-generator use polar NRZ encoding.
The spreading-code is 11 chips having the pattern 10110111000.
If the original signal-rate is N, the rate of the spread signal is 1/N.
This means that the required bandwidth for the spread signal is 11 times larger than the
bandwidth of the original signal.

The spread signal can provide privacy if the attacker does not know the code.
It can also provide immunity against interference if each station uses a different code.

Bandwidth Sharing
Can we share a bandwidth in DSSS?
The answer is no and yes.
If we use a spreading-code that spreads signals that cannot be combined and separated, we cannot
share a bandwidth.
For example:
Some wireless LANs use DSSS and the spread bandwidth cannot be shared.
2) If we use a special spreading-code that spreads signals that can be combined and separated, we
can share a bandwidth.

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For example:
Cellular telephony uses DSSS and the spread bandwidth is shared b/w several users.

MODULE 2(CONT.): SWITCHING

Switching
A network is a set of connected-devices.
Problem: Whenever we have multiple-devices, we have the problem of how to connect them to make one-to-one
communication possible.
Solution: Use Switching.
A switched-network consists of a series of interlinked-nodes, called switches.
Switches are devices capable of creating temporary connections between two or more devices.
In a switched-network,
Some nodes are connected to the end-systems (For example: PC or TP).
Some nodes are used only for routing.

As shown in Figure 8.1,

The end-systems are labeled A, B, C, D, and so on.
The switches are labeled I, II, III, IV, and V. Each switch is connected to multiple links.

Three Methods of Switching

Three methods of Switching are (Figure 8.2):
Circuit Switching
Packet Switching and
Message Switching.
The first two are commonly used today.
The third has been phased out in general communications but still has networking applications.
Packet switching can further be divided into two subcategories—virtual circuit approach and datagram approach

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Switching and TCP/IP Layers

Switching can happen at several layers of the TCP/IP protocol suite.
Switching at Physical Layer
At the physical layer, we can have only circuit switching.
There are no packets exchanged at the physical layer.
The switches at the physical layer allow signals to travel in one path or another.
2) Switching at Data-Link Layer
At the data-link layer, we can have packet switching.
However, the term packet in this case means frames or cells.
Packet switching at the data-link layer is normally done using a virtual-circuit approach.
3) Switching at Network Layer
At the network layer, we can have packet switching.
In this case, either a virtual-circuit approach or a datagram approach can be used.
Currently the Internet uses a datagram approach, but the tendency is to move to a virtual-
circuit approach.
4) Switching at Application Layer
At the application layer, we can have only message switching.
The communication at the application layer occurs by exchanging messages.
Conceptually, we can say that communication using e-mail is a kind of message-switched
communication, but we do not see any network that actually can be called a message-switched
network.

Circuit Switched Network

This is similar to telephone system.
Fixed path (connection) is established between a source and a destination prior to the transfer of packets.
A circuit-switched-network consists of a set of switches connected by physical-links (Figure 8.3).
A connection between 2 stations is a dedicated-path made of one or more links.
However, each connection uses only one dedicated-channel on each link.
Normally, each link is divided into „n‟ channels by using FDM or TDM.
The resources need to be reserved during the setup phase.
The resources remain dedicated for the entire duration of data transfer until the teardown phase.

The virtual-circuit setup procedure

first determines a path through the network &
sets parameters in the switches by exchanging connect-request & connect-confirm messages
If a switch does not have enough resources to set up a virtual circuit, the switch responds with a connect-
reject message and the setup procedure fails (Figure 7.15).
A connection-release procedure may also be required to terminate the connection.

Three Phases
The communication requires 3 phases: 1) Connection-setup
Data-transfer
Connection teardown.
Setup Phase

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Before the 2 parties can communicate, a dedicated-circuit needs to be established.

Normally, the end-systems are connected through dedicated-lines to the switches. So,
connection-setup means creating dedicated-channels between the switches.
For ex: Assume system-A needs to connect to system-M. For this, following events occur:
System-A sends a setup-request to switch-I.
Switch-I finds a channel between itself and switch-IV that can be dedicated for this purpose.
Switch-I then sends the request to switch-IV, which finds a dedicated-channel between itself
and switch-III.
Switch-III informs system-M of system-A's intention at this time.
Finally, an acknowledgment from system-M needs to be sent in the opposite direction to
system-A.
Only after system A receives this acknowledgment is the connection established.
2) Data Transfer Phase
After the establishment of the dedicated-circuit (channels), the two parties can transfer data.
3) Teardown Phase
When one of the parties needs to disconnect, a signal is sent to each switch to release the
resources.
Efficiency
• Circuit-switched-networks are inefficient when compared to other two types of networks because 1)
Resources are allocated during the entire duration of the connection.
2) These resources are unavailable to other connections.

Delay
Circuit-switched-networks have minimum delay when compared to other two types of networks
During data-transfer,
The data are not delayed at each switch.
The resources are allocated for the duration of the connection.

As in the above figure, there is no waiting time at each switch.

The total delay is the time needed to
Create the connection
Transfer-data and
Disconnect the circuit.
The delay caused by the setup is the sum of 4 parts:
The propagation time of the source-computer request.
The request signal transfer time.
The propagation time of the acknowledgment from the destination computer.
The signal transfer time of the acknowledgment.
The delay due to data-transfer is the sum of 2 parts:
The propagation time.
Data-transfer time which can be very long.
Packet Switched Network

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The message is divided into packets of fixed or variable size.

The packet-size is determined by
network and
governing protocol.
There is no resource reservation; resources are allocated on-demand.
Datagram Networks
This is analogous to postal system.
Each packet is routed independently through the network.
Each packet has a header that contains source and destination addresses.
Each switch examines the header to determine the next hop in the path to the destination.
If the transmission line is busy
then the packet is placed in the queue until the line becomes free.
Packets are referred to as datagrams.
Datagram switching is normally done at the network layer.
In Internet, switching is done by using the datagram switching.
Advantage:
High utilization of transmission-line can be achieved by sharing among multiple packets.
Disadvantages:
Packets may arrive out-of-order, and re-sequencing may be required at the destination
Loss of packets may occur when a switch has insufficient buffer

The Figure 8.7 shows how the 4 packets are transferred from station-A to station-X.
The switches are referred to as routers.
All four packets (or datagrams) belong to the same message, but may travel different paths to reach
their destination.
This is so because the links may be involved in carrying packets from other sources and do not have
the necessary bandwidth available to carry all the packets from A to X.
This approach can cause the datagrams of a transmission to arrive at their destination out-of-
order with different delays between the packets.
Packets may also be lost or dropped because of a lack-of-resources.
It is the responsibility of an upper-layer protocol to
reorder the datagrams or
ask for lost datagrams.
The datagram-networks are referred to as connectionless networks. This is because
The switch does not keep information about the connection state.
There are no setup or teardown phases.
Each packet is treated the same by a switch regardless of its source or destination.

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Routing Table
Each switch has a routing-table which is based on the destination-address.
The routing-tables are dynamic & updated periodically.
The destination-addresses and the corresponding forwarding output-ports are recorded in the tables.

Destination Address
Every packet carries a header that contains the destination-address of the packet.
When the switch receives the packet,
This destination-address is examined.
The routing-table is consulted to find the corresponding port through which the packet should be
forwarded.
The destination address in the header of a packet in remains the same during the entire journey of the packet.
Efficiency
Datagram-networks are more efficient when compared to circuit-switched-network. This is because
Resources are allocated only when there are packets to be transferred.
If a source sends a packet and there is a delay of a few minutes before another packet can be sent, the
resources can be re-allocated during these minutes for other packets from other sources.
Delay
• Datagram-networks may have greater delay when compared to circuit-switched-network. This is because
Each packet may experience a wait at a switch before it is forwarded.
Since not all packets in a message necessarily travel through the same switches, the delay is not
uniform for the packets of a message.

The Figure 8.9 gives an example of delay for one single packet.
The packet travels through two switches.
There are three transmission times (3T), three propagation delays (slopes 3t of the lines), and two
waiting times (W1+ W2).

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Virtual-circuit Network (VCN)

This is similar to telephone system.
A virtual-circuit network is a combination of circuit-switched-network and datagram-network.
Five characteristics of VCN:
As in a circuit-switched-network, there are setup & teardown phases in addition to the data transfer
phase.
As in a circuit-switched-network, resources can be allocated during the setup phase.
As in a datagram-network, resources can also be allocated on-demand.
As in a datagram-network, data is divided into packets.
Each packet carries an address in the header.
However, the address in the header has local jurisdiction, not end-to-end jurisdiction.
As in a circuit-switched-network, all packets follow the same path established during the connection.
A virtual-circuit network is implemented in the data link layer.
A circuit-switched-network is implemented in the physical layer.
A datagram-network is implemented in the network layer.

The Figure 8.10 is an example of a virtual-circuit network.

The network has switches that allow traffic from sources to destinations.
A source or destination can be a computer, packet switch, bridge, or any other device that connects
other networks.

Addressing
Two types of addressing: 1) Global and 2) Local (virtual-circuit identifier).
Global Addressing
A source or a destination needs to have a global address.
Global address is an address that can be unique in the scope of the network or internationally if the
network is part of an international network.
2) Virtual Circuit Identifier
The identifier used for data-transfer is called the virtual-circuit identifier (VCI).
A VCI, unlike a global address, is a small number that has only switch scope.
VCI is used by a frame between two switches.
When a frame arrives at a switch, it has a VCI.
When the frame leaves, it has a different VCI.

Figure 8.11 show how the VCI in a data-frame changes from one switch to another.

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Three Phases
A source and destination need to go through 3 phases: setup, data-transfer, and teardown.
In setup phase, the source and destination use their global addresses to help switches make
table entries for the connection.
In the teardown phase, the source and destination inform the switches to delete the
corresponding entry.
Data-transfer occurs between these 2 phases.
Data Transfer Phase
To transfer a frame from a source to its destination, all switches need to have a table-entry for this virtual-
circuit.
The table has four columns.
The switch holds 4 pieces of information for each virtual-circuit that is already set up.

As shown in Figure 8.12, a frame arrives at port 1 with a VCI of 14.

When the frame arrives, the switch looks in its table to find port 1 and a VCI of 14.
When it is found, the switch knows to change the VCI to 22 & send out the frame from port 3.

As shown in Figure 8.13, each switch changes the VCI and routes the frame.
The data-transfer phase is active until the source sends all its frames to the destination.
The procedure at the switch is the same for each frame of a message.
The process creates a virtual circuit, not a real circuit, between the source and destination.

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Setup Phase
A switch creates an entry for a virtual-circuit.
For example, suppose source A needs to create a virtual-circuit to B.
Two steps are required:1) Setup-request and
Acknowledgment.

1) Setup Request
A setup-request frame is sent from the source to the destination (Figure 8.14).

Following events occurs:

Source-A sends a setup-frame to switch-1.
Switch-1 receives the setup-frame.
Switch-1 knows that a frame going from A to B goes out through port 3.
The switch-1 has a routing table.
The switch
creates an entry in its table for this virtual-circuit
is only able to fill 3 of the 4 columns.
The switch
assigns the incoming port (1) and
chooses an available incoming-VCI (14) and the outgoing-port (3).
does not yet know the outgoing VCI, which will be found during the
acknowledgment step.
The switch then forwards the frame through port-3 to switch-2.
Switch-2 receives the setup-request frame.
The same events happen here as at switch-1.
Three columns of the table are completed: In this case, incoming port (1), incoming-
VCI (66), and outgoing port (2).
Switch-3 receives the setup-request frame.
Again, three columns are completed: incoming port (2), incoming-VCI (22), and
outgoing-port (3).
Destination-B
receives the setup-frame
assigns a VCI to the incoming frames that come from A, in this case 77.
This VCI lets the destination know that the frames come from A, and no other
sources.

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2) Acknowledgment
A special frame, called the acknowledgment-frame, completes the entries in the switching-tables
(Figure 8.15).

The destination sends an acknowledgment to switch-3.

The acknowledgment carries the global source and destination-addresses so the switch
knows which entry in the table is to be completed.
The frame also carries VCI 77, chosen by the destination as the incoming-VCI for
frames from A.
Switch 3 uses this VCI to complete the outgoing VCI column for this entry.
Switch 3 sends an acknowledgment to switch-2 that contains its incoming-VCI in the table,
chosen in the previous step.
Switch-2 uses this as the outgoing VCI in the table.
Switch-2 sends an acknowledgment to switch-1 that contains its incoming-VCI in the table,
chosen in the previous step.
Switch-1 uses this as the outgoing VCI in the table.
Finally switch-1 sends an acknowledgment to source-A that contains its incoming-VCI in the
table, chosen in the previous step.
The source uses this as the outgoing VCI for the data-frames to be sent to destination-B.

Teardown Phase
• Source-A, after sending all frames to B, sends a special frame called a teardown request.
• Destination-B responds with a teardown confirmation frame.
• All switches delete the corresponding entry from their tables.

Efficiency
• Resource reservation can be made in 2 cases:
1) During the setup: Here, the delay for each packet is the same. 2)
On demand: Here, each packet may encounter different delays.
• Advantage of on demand resource allocation:
The source can check the availability of the resources, without actually reserving it.

Delay in Virtual Circuit Networks

There is a one-time delay for setup and a one-time delay for teardown (Figure 8.16).
If resources are allocated during the setup phase, there is no wait time for individual packets.
The packet is traveling through two switches (routers).
There are three transmission times (3T), three propagation times (3τ), data transfer delay, a setup delay and a
teardown delay.
The total delay time is

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Assignment Questions

1. Explain the PCM encoder with neat diagram.

2. What do you mean by Sampling? Explain three sampling methods with a neat diagram.
3. Explain non-uniform quantization and how to recover original signal using PCM decoder.
4. Explain different types of transmission modes.
5. What is sampling and quantization? Explain briefly.
6. Define digital to analog conversion. List different types of digital to analog conversion.
7. Describe ASK, FSK and PSK mechanisms and apply them over the digital data 101101.
8. Discuss the bandwidth requirement for ASK, FSK and PSK.
9. Explain different aspects of digital-to-analog conversion?
10. Define ASK. Explain BASK.
11. Define FSK. Explain BFSK.
12. Define PSK. Explain BPSK.
13. Explain QPSK.
14. Explain the concept of constellation diagram.
15. Explain QAM.

Expected Outcomes

• Explain about PCM Encoder

• Learn about Sampling and quantization
• Analyse digital to analog conversion and vice-versa
• Explain ASK, PSK, FSK

Further Reading
1. http://www.mhhe.com/engcs/compsci/forouzan/frontmatter.pdf
2. http://freecomputerbooks.com/Introduction-to-Data-Communications.html

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MODULE 3: TABLE OF CONTENTS

INTRODUCTION
Types of Errors
Redundancy
Detection versus Correction
Coding
BLOCK CODING
Error Detection
Hamming Distance
Minimum Hamming Distance for Error Detection
Linear Block Codes
Minimum Distance for Linear Block Codes
Parity-Check Code
CYCLIC CODES
Cyclic Redundancy Check (CRC)
Polynomials
Cyclic Code Encoder Using Polynomials
Cyclic Code Analysis
Advantages of Cyclic Codes
CHECKSUM
Concept of Checksum
One's Complement
Internet Checksum
Algorithm
Other Approaches to the Checksum
Fletcher Checksum
Adler Checksum
FORWARD ERROR CORRECTION
Using Hamming Distance
Using XOR
Chunk Interleaving
Combining Hamming Distance and Interleaving
Compounding High- and Low-Resolution Packets
DLC SERVICES
Framing
Frame Size
Character-Oriented Framing
Bit-Oriented Framing
Flow and Error Control
Flow-control
Buffers
Error-control
Combination of Flow and Error Control
Connectionless and Connection-Oriented
DATA-LINK LAYER PROTOCOLS
Simple Protocol
Design
FSMs
Stop-and-Wait Protocol
Design
FSMs
Sequence and Acknowledgment Numbers

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Piggybacking
High-level Data Link Control (HDLC)
Configurations and Transfer Modes
Framing
Frame Format
Control Fields of HDLC Frames
POINT-TO-POINT PROTOCOL (PPP)
Framing
Byte Stuffing
Transition Phases

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MODULE 3
ERROR-DETECTION AND CORRECTION
STRUCTURE

Introduction
Objectives
Introduction to Error detection
Block Coding
Cyclic Codes
Checksum
Forward Error Correction
DLC Services
Data Link Layer Protocols
High-Level Data Link Control (HDLC)
Point-To-Point Protocol (PPP)
3.12Assignment Questions
Expected Outcomes
Further Reading

Introduction
This module provides an overview of the data link layer, its functionalities such as error
detection and correction. Also different protocols used at the data link layer.

Objectives
• Understand the types of errors and error correction mechanisms
• Learn about different protocols used in error correction
• Understand about Checksum and advantage of polynomials

Introduction to Error detection

Types of Errors
When bits flow from 1 point to another, they are subject to unpredictable-changes ‘.’ of interference.
The interference can change the shape of the signal.
Two types of errors: 1) Single-bit error 2) Burst-error.

1) Single-Bit Error
Only 1 bit of a given data is changed
from 1 to 0 or
from 0 to 1 (Figure 10.1a).
2) Burst Error
Two or more bits in the data have changed
from 1 to 0 or
from 0 to 1 (Figure 10.1b).

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A burst-error occurs more than a single-bit error. This is because:

Normally, the duration of noise is longer than the duration of 1-bit.
When noise affects data, the noise also affects the bits.
The no. of corrupted-bits depends on
data-rate and
duration of noise.

Redundancy
The central concept in detecting/correcting errors is redundancy.
Some extra-bits along with the data have to be sent to detect/correct errors. These extra bits are called
redundant-bits.
The redundant-bits are
added by the sender and
removed by the receiver.
The presence of redundant-bits allows the receiver to detect/correct errors.

Error Detection vs. Error Correction

• Error-correction is more difficult than error-detection.
1) Error Detection
Here, we are checking whether any error has occurred or not.
The answer is a simple YES or NO.
We are not interested in the number of corrupted-bits.
2) Error Correction
Here, we need to know
exact number of corrupted-bits and
location of bits in the message.
Two important factors to be considered:
1) Number of errors and
2) Message-size.
Coding
Redundancy is achieved through various coding-schemes.
Sender adds redundant-bits to the data-bits. This process creates a relationship between
redundant-bits and
data-bits.
Receiver checks the relationship between redundant-bits & data-bits to detect/correct errors.
Two important factors to be considered:
Ratio of redundant-bits to the data-bits and
Robustness of the process.
Two broad categories of coding schemes: 1) Block-coding and 2) Convolution coding.

Block Coding
The message is divided into k-bit blocks. These blocks are called data-words.
Here, r-redundant-bits are added to each block to make the length n=k+r.
The resulting n-bit blocks are called code-words.
Since n>k, the number of possible code-words is larger than the number of possible data-words.
Block-coding process is 1-to-1; the same data-word is always encoded as the same code-word.
Thus, we have 2n-2k code-words that are not used. These code-words are invalid or illegal.
Error Detection
If the following 2 conditions are met, the receiver can detect a change in the original code-word:
The receiver has a list of valid code-words.
The original code-word has changed to an invalid code-words.

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• Here is how it works (Figure 10.2):

1) At Sender
The sender creates code-words out of data-words by using a generator. The
generator applies the rules and procedures of encoding.
During transmission, each code-word sent to the receiver may change.
2) At Receiver
a) If the received code-word is the same as one of the valid code-words, the
code-word is accepted;
the corresponding data-word is extracted for use.
If the received code-word is invalid, the code-word is discarded.
However, if the code-word is corrupted but the received code-word still matches a valid code-word,
the error remains undetected.
An error-detecting code can detect only the types of errors for which it is designed; other types of errors may
remain undetected.

Example 3.1

Hamming Distance
The main concept for error-control: Hamming distance.
The Hamming distance b/w 2 words is the number of differences between the corresponding bits.
Let d(x,y) = Hamming distance b/w 2 words x and y.
Hamming distance can be found by
applying the XOR operation on the 2 words and
counting the number of 1s in the result.
For example:
The Hamming distance d(000, 011) is 2 because 000⊕011= 011 (two 1s).
The Hamming distance d(10101, 11110) is 3 because 10101⊕11110 = 01011 (three 1s).
Hamming Distance and Error
Hamming distance between the received word and the sent code-word is the number of bits
that are corrupted during transmission.
For example: Let Sent code-word = 00000 Received
word = 01101
Hamming distance = d(00000, 01101) =3. Thus, 3 bits are in error.

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Minimum Hamming Distance for Error Detection

Minimum Hamming distance is the smallest Hamming distance b/w all possible pairs of code-words.
Let dmin = minimum Hamming distance.
To find dmin value, we find the Hamming distances between all words and select the smallest one.
Minimum-distance for Error-detection
If ‘s’ errors occur during transmission, the Hamming distance b/w the sent code-word and
received code-word is ‘s’ (Figure 10.3).
If code has to detect upto ‘s’ errors, the minimum-distance b/w the valid codes must be ‘s+1’ i.e.
dmin=s+1.
We use a geometric approach to define dmin=s+1.

Let us assume that the sent code-word x is at the center of a circle with radius s.
All received code-words that are created by 0 to s errors are points inside the circle or on the
perimeter of the circle.
All other valid code-words must be outside the circle
For example: A code scheme has a Hamming distance dmin = 4.
This code guarantees the detection of upto 3 errors (d = s + 1 or s = 3).

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Linear Block Codes

Almost all block codes belong to a subset of block codes called linear block codes.
A linear block code is a code in which the XOR of 2 valid code-words creates another valid code-word.
(XOR Addition modulo-2)

The code in Table 10.1 is a linear block code because the result of XORing any code-word with any other
code-word is a valid code-word.
For example, the XORing of the 2nd and 3rd code-words creates the 4th one.
Minimum Distance for Linear Block Codes
Minimum Hamming distance is no. of 1s in the nonzero valid code-word with the smallest no. of 1s.
In Table 10.1,
The numbers of 1s in the nonzero code-words are 2, 2, and 2.
So the minimum Hamming distance is dmin = 2.
Parity Check Code
This code is a linear block code. This code can detect an odd number of errors.
A k-bit data-word is changed to an n-bit code-word where n=k+1.
One extra bit is called the parity-bit.
The parity-bit is selected to make the total number of 1s in the code-word even.
Minimum hamming distance dmin = 2. This means the code is a single-bit error-detecting code.

Here is how it works (Figure 10.4):

At Sender
The encoder uses a generator that takes a copy of a 4-bit data-word (a0, a1, a2, and a3) and generates a
parity-bit r0.
The encoder
accepts a copy of a 4-bit data-word (a0, a1, a2, and a3) and
generates a parity-bit r0 using a generator
generates a 5-bit code-word
The parity-bit & 4-bit data-word are added to make the number of 1s in the code-word even.
The addition is done by using the following:
The result of addition is the parity-bit.
If the no. of 1s in data-word = even, result = 0. (r0=0)

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If the no. of 1s in data-word = odd, result = 1. (r0=1)

In both cases, the total number of 1s in the code-word is even.
The sender sends the code-word, which may be corrupted during transmission.
2) At Receiver
The receiver receives a 5-bit word.
The checker performs the same operation as the generator with one exception:
The addition is done over all 5 bits.
The result is called the syndrome bit (so).
Syndrome bit = 0 when the no. of 1s in the received code-word is even; otherwise, it is 1.
The syndrome is passed to the decision logic analyzer.
If s0=0, there is no error in the received code-word. The data portion of the received code-
word is accepted as the data-word.
If s0=1, there is error in the received code-word. The data portion of the received code-word is
discarded. The data-word is not created.

Example 3.2

Cyclic Codes
• Cyclic codes are special linear block codes with one extra property:
If a code-word is cyclically shifted (rotated), the result is another code-word.
For ex: if code-word = 1011000 and we cyclically left-shift, then another code-word = 0110001.
• Let First-word = a0 to a6 and Second-word = b0 to b6, we can shift the bits by using the following:

Cyclic Redundancy Check (CRC)

• CRC is a cyclic code that is used in networks such as LANs and WANs.

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• Let Size of data-word = k bits (here k=4).

Size of code-word = n bits (here n=7).
Size of divisor = n-k+1 bits (here n-k+1=4). (Augmented increased)
• Here is how it works (Figure 10.5):
1) At Sender
n-k 0s is appended to the data-word to create augmented data-word. (here n-k=3). The
augmented data-word is fed into the generator (Figure 10.6).
The generator divides the augmented data-word by the divisor.
The remainder is called check-bits (r2r 1r0).
The check-bits (r2r1r0) are appended to the data-word to create the code-word.
2) At Receiver
The possibly corrupted code-word is fed into the checker.
The checker is a replica of the generator.
The checker divides the code-word by the divisor.
The remainder is called syndrome bits (r2r1r0).
The syndrome bits are fed to the decision-logic-analyzer.
The decision-logic-analyzer performs following functions:
i) For No Error
If all syndrome-bits are 0s, the received code-word is accepted.
Data-word is extracted from received code-word (Figure 10.7a). ii)
For Error
If all syndrome-bits are not 0s, the received code-word is discarded (Figure 10.7b).

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Example 3.3

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Polynomials
A pattern of 0s and 1s can be represented as a polynomial with coefficients of 0 and 1 (Figure 10.8).
The power of each term shows the position of the bit; the coefficient shows the value of the bit.

Cyclic Code Encoder Using Polynomials

Let Data-word = 1001 = x3+1.
Divisor = 1011 = x3+x+1.
In polynomial representation, the divisor is referred to as generator polynomial t(x) (Figure 10.9).

Cyclic Code Analysis

• We define the following, where f(x) is a polynomial with binary coefficients:

Single Bit Error

If the generator has more than one term and the coefficient of x0 is 1, all single-bit errors can be caught.
Two Isolated Single-Bit Errors
If a generator cannot divide xi+1 (t between 0 & n-1), then all isolated double errors can be detected (Figure
10.10).

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Odd Numbers of Errors

• A generator that contains a factor of x+1 can detect all odd-numbered errors.

Standard Polynomials

Advantages of Cyclic Codes

The cyclic codes have a very good performance in detecting
single-bit errors
double errors
odd number of errors and
burst-errors.
They can easily be implemented in hardware and software. They are fast when implemented in hardware.

Checksum
Checksum is an error-detecting technique.
In the Internet,
The checksum is mostly used at the network and transport layer.
The checksum is not used in the data link layer.
Like linear and cyclic codes, the checksum is based on the concept of redundancy.
Here is how it works (Figure 10.15):
1) At Source
Firstly the message is divided into m-bit units.
Then, the generator creates an extra m-bit unit called the checksum.
The checksum is sent with the message.
2) At Destination
The checker creates a new checksum from the combination of the message and sent-
checksum.
i) If the new checksum is all 0s, the message is accepted.
ii) If the new checksum is not all 0s, the message is discarded.

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Concept of Checksum
Consider the following example:
Example 3.4
Our data is a list of five 4-bit numbers that we want to send to a destination.
In addition to sending these numbers, we send the sum of the numbers.
For example:
Let set of numbers = (7, 11, 12, 0, 6).
We send (7, 11, 12, 0, 6, 36), where 36 is the sum of the original numbers.
The receiver adds the five numbers and compares the result with the sum.
If the result & the sum are the same,
The receiver assumes no error, accepts the five numbers, and discards the sum.
Otherwise, there is an error somewhere and the data are not accepted.
Example 3.5
To make the job of the receiver easy if we send the negative (complement) of the sum, called the checksum.
In this case, we send (7, 11, 12, 0, 6, -36).
The receiver can add all the numbers received (including the checksum).
If the result is 0, it assumes no error; otherwise, there is an error.

One's Complement
• The previous example has one major drawback.
All of our data can be written as a 4-bit word (they are less than 15) except for the checksum.
Solution: Use one's complement arithmetic.
We can represent unsigned numbers between 0 and 2n-1 using only n bits.
If the number has more than n bits, the extra leftmost bits need to be added to the n rightmost bits
(wrapping).
A negative number can be represented by inverting all bits (changing 0 to 1 and 1 to 0).
This is the same as subtracting the number from 2n-1.

• Here is how it works (Figure 10.16):

1) At Sender
The sender initializes the checksum to 0 and adds all data items and the checksum.
The result is 36.
However, 36 cannot be expressed in 4 bits.
The extra two bits are wrapped and added with the sum to create the wrapped sum value 6. The
sum is then complemented, resulting in the checksum value 9 (15 - 6 = 9).
The sender now sends six data items to the receiver including the checksum 9.
2) At Receiver
The receiver follows the same procedure as the sender.
It adds all data items (including the checksum); the result is 45.
The sum is wrapped and becomes 15. The wrapped sum is complemented and becomes 0.
Since the value of the checksum is 0, this means that the data is not corrupted. The receiver drops the
checksum and keeps the other data items.
If the checksum is not zero, the entire packet is dropped.

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Internet Checksum
Traditionally, the Internet has been using a 16-bit checksum.
The sender or the receiver uses five steps.

Algorithm

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Other Approaches to the Checksum

If two 16-bit items are transposed in transmission, the checksum cannot catch this error.
The reason is that the traditional checksum is not weighted: it treats each data item equally.
In other words, the order of data items is immaterial to the calculation.
Two approaches have been used to prevent this problem: 1)Fletcher and 2)Adler

Fletcher Checksum
The Fletcher checksum was devised to weight each data item according to its position.
Fletcher has proposed two algorithms: 8-bit and 16-bit (Figure 10.18).
The first, 8-bit Fletcher, calculates on 8-bit data items and creates a 16-bit checksum.
The second, 16-bit Fletcher, calculates on 16-bit data items and creates a 32-bit checksum.
The 8-bit Fletcher is calculated over data octets (bytes) and creates a 16-bit checksum.
The calculation is done modulo 256 (28), which means the intermediate results are divided by 256 and
the remainder is kept.
The algorithm uses two accumulators, L and R.
The first simply adds data items together;
The second adds a weight to the calculation.

Adler Checksum
The Adler checksum is a 32-bit checksum.
It is similar to the 16-bit Fletcher with three differences (Figure 10.19).
Calculation is done on single bytes instead of 2 bytes at a time.
The modulus is a prime number (65,521) instead of 65,536.
L is initialized to 1 instead of 0.
A prime modulo has a better detecting capability in some combinations of data.

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Forward Error Correction

Retransmission of corrupted and lost packets is not useful for real-time multimedia transmission because it
creates an unacceptable delay in reproducing: we need to wait until the lost or corrupted packet is resent.
We need to correct the error or reproduce the packet immediately.
Several schemes have been designed and used that are collectively referred to as forward error correction
(FEC) techniques.

Using Hamming Distance

To detect t errors, we need to have dmin = 2t + 1 (Figure 10.20).
In other words, if we want to correct 10 bits in a packet, we need to make the minimum hamming distance
21 bits, which means a lot of redundant bits need to be sent with the data.

Using XOR
• Use the property of the exclusive OR operation as shown below.

• We divide a packet into N chunks, create the exclusive OR of all the chunks and send N + 1 chunks.
• If any chunk is lost or corrupted, it can be created at the receiver site.
• If N = 4, it means that we need to send 25 percent extra data and be able to correct the data if only one out
of four chunks is lost.

Chunk Interleaving
Another way to achieve FEC in multimedia is to allow some small chunks to be missing at the receiver.
We cannot afford to let all the chunks belonging to the same packet be missing.
However, we can afford to let one chunk be missing in each packet.

In Figure 10.21, each packet is divided into 5 chunks (normally the number is much larger).
Then, we can create data chunk-by-chunk (horizontally), but combine the chunks into packets vertically.
In this case, each packet sent carries a chunk from several original packets.
If the packet is lost, we miss only one chunk in each packet.
Normally, missing of a chunk is acceptable in multimedia communication.

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Combining Hamming Distance and Interleaving

• Hamming distance and interleaving can be combined.
• Firstly, we can create n-bit packets that can correct t-bit errors.
• Then, we interleave m rows and send the bits column-by-column.
• In this way, we can automatically correct burst-errors up to m×t bit errors.

Compounding High- and Low-Resolution Packets

Another solution is to create a duplicate of each packet with a low-resolution redundancy and combine the
redundant version with the next packet.
For example (Figure 10.22):
We can create 4 low-resolution packets out of 5 high-resolution packets and send them (Fig 10.22).
If a packet is lost, we can use the low-resolution version from the next packet.
In this method, if the last packet is lost, it cannot be recovered, but we use the low-resolution version of a
packet if the lost packet is not the last one.
The audio and video reproduction does not have the same quality, but the lack of quality is not recognized
most of the time.

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MODULE 3(CONT.): DATA LINK CONTROL

DLC Services
The data link control (DLC) deals with procedures for communication between two adjacent nodes i.e.
node-to-node communication.
Data link control functions include 1) Framing and 2) Flow control and 3) Error control.

Framing
A frame is a group of bits.
Framing means organizing the bits into a frame that are carried by the physical layer.
The data-link-layer needs to form frames, so that each frame is distinguishable from another.
Framing separates a message from other messages by adding sender-address & destination-address.
The destination-address defines where the packet is to go.
The sender-address helps the recipient acknowledge the receipt.
• Q: Why the whole message is not packed in one frame?
Ans: Large frame makes flow and error-control very inefficient.
Even a single-bit error requires the re-transmission of the whole message.
When a message is divided into smaller frames, a single-bit error affects only that small frame.

(Our postal system practices a type of framing. The simple act of inserting a letter into an envelope separates
one piece of information from another; the envelope serves as the delimiter. In addition, each envelope
defines the sender and receiver addresses since the postal system is a many-to-many carrier facility).

Frame Size
• Two types of frames:
1) Fixed-Size Framing
There is no need for defining boundaries of frames; the size itself can be used as a delimiter.
For example: ATM WAN uses frames of fixed size called cells.
2) Variable-Size Framing
We need to define the end of the frame and the beginning of the next frame.
Two approaches are used: 1) Character-oriented approach
Bit-oriented approach.
Character-Oriented Framing
Data to be carried are 8-bit characters from a coding system such as ASCII (Figure 11.1).
The header and the trailer are also multiples of 8 bits.
Header carries the source and destination-addresses and other control information.
Trailer carries error-detection or error-correction redundant bits.
To separate one frame from the next frame, an 8- bit (I-byte) flag is added at the beginning and the end of a
frame.
The flag is composed of protocol-dependent special characters.
The flag signals the start or end of a frame.

Problem:
Character-oriented framing is suitable when only text is exchanged by the data-link-layers.
However, if we send other type of information (say audio/video), then any pattern used for the flag
can also be part of the information.
If the flag-pattern appears in the data-section, the receiver might think that it has reached the end of
the frame.
Solution: A byte-stuffing is used.
(Byte stuffing character stuffing)

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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 called the escape character (ESC), which has a
predefined bit pattern.
When a receiver encounters the ESC character, the receiver
removes ESC character from the data-section and
treats the next character as data, not a delimiting flag.
Problem:
What happens if the text contains one or more escape characters followed by a flag?
The receiver removes the escape character, but keeps the flag, which is incorrectly interpreted as the
end of the frame.
Solution:
Escape characters part of the text must also be marked by another escape character (Fig 11.2).

• In short, byte stuffing is the process of adding one extra byte whenever there is a flag or escape character
in the text.
Bit-Oriented Framing
The data-section of a frame is a sequence of bits to be interpreted by the upper layer as text, audio, video,
and so on.
However, in addition to headers and trailers, we need a delimiter to separate one frame from the other.
Most protocols use a special 8-bit pattern flag 01111110 as the delimiter to define the beginning and the end
of the frame (Figure 11.3).

Problem:
If the flag-pattern appears in the data-section, the receiver might think that it has reached
the end of the frame.
Solution: A bit-stuffing is used.
• 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. (Figure 11.4).
• This guarantees that the flag field sequence does not inadvertently appear in the frame.

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• In short, 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.
Flow Control and Error Control
One of the responsibilities of the DLC sublayer is flow and error control at the data-link layer.
Flow Control
Whenever an entity produces items and another entity consumes them, there should be a balance between
production and consumption rates.
If the items are produced faster than they can be consumed, the consumer can be overwhelmed and may
need to discard some items.
We need to prevent losing the data items at the consumer site.

At the sending node, the data-link layer tries to push frames toward the data-link layer at the receiving node
(Figure 11.5).
If the receiving node cannot process and deliver the packet to its network at the same rate that the frames
arrive, it becomes overwhelmed with frames.
Here, flow control can be feedback from the receiving node to the sending node to stop or slow down
pushing frames.

3.6.2.1.1 Buffers
Flow control can be implemented by using buffer.
A buffer is a set of memory locations that can hold packets at the sender and receiver.
Normally, two buffers can be used.
First buffer at the sender.
Second buffer at the receiver.
The flow control communication can occur by sending signals from the consumer to the producer.
When the buffer of the receiver is full, it informs the sender to stop pushing frames.

Error Control
Error-control includes both error-detection and error-correction.
Error-control allows the receiver to inform the sender of any frames lost/damaged in transmission.
A CRC is
added to the frame header by the sender and
checked by the receiver.
At the data-link layer, error control is normally implemented using one of the following two methods. 1)
First method: If the frame is corrupted, it is discarded;
If the frame is not corrupted, the packet is delivered to the network layer.
This method is used mostly in wired LANs such as Ethernet.
2) Second method: If the frame is corrupted, it is discarded;
If the frame is not corrupted, an acknowledgment is sent to the sender.
Acknowledgment is used for the purpose of both flow and error control.

3.6.2.2.1 Combination of Flow and Error Control

Flow and error control can be combined.

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The acknowledgment that is sent for flow control can also be used for error control to tell the sender the
packet has arrived uncorrupted.
The lack of acknowledgment means that there is a problem in the sent frame.
A frame that carries an acknowledgment is normally called an ACK to distinguish it from the data frame.
Connectionless and Connection-Oriented
• A DLC protocol can be either connectionless or connection-oriented.
Connectionless Protocol
Frames are sent from one node to the next without any relationship between the frames;
each frame is independent.
The term connectionless does not mean that there is no physical connection (transmission medium)
between the nodes; it means that there is no connection between frames.
The frames are not numbered and there is no sense of ordering.
Most of the data-link protocols for LANs are connectionless protocols.
Connection-Oriented Protocol
A logical connection should first be established between the two nodes (setup phase).
After all frames that are somehow related to each other are transmitted (transfer phase), the
logical connection is terminated (teardown phase).
The frames are numbered and sent in order.
If the frames are not received in order, the receiver needs to wait until all frames belonging
to the same set are received and then deliver them in order to the network layer.
Connection oriented protocols are rare in wired LANs, but we can see them in some point-to-point
protocols, some wireless LANs, and some WANs.

Data Link Layer Protocols

Traditionally 2 protocols have been defined for the data-link layer to deal with flow and error control:
Simple Protocol and 2) Stop-and-Wait Protocol.
The behavior of a data-link-layer protocol can be better shown as a finite state machine (FSM).
An FSM is a machine with a finite number of states (Figure 11.6).
The machine is always in one of the states until an event occurs.
Each event is associated with 2 reactions:
Defining the list (possibly empty) of actions to be performed.
Determining the next state (which can be the same as the current state).
One of the states must be defined as the initial state, the state in which the machine starts when it turns on.

Simplest Protocol
Assumptions:
The protocol has no flow-control or error-control.
The protocol is a unidirectional protocol (in which frames are traveling in only one direction).
The receiver can immediately handle any frame it receives.

Design
Here is how it works (Figure 11.7):

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At Sender
The data-link-layer
gets data from its network-layer
makes a frame out of the data and
sends the frame.
At Receiver
The data-link-layer
receives a frame from its physical layer
extracts data from the frame and
delivers the data to its network-layer.
Data-link-layers of sender & receiver provide transmission services for their network-layers.
Data-link-layers use the services provided by their physical layers for the physical transmission of
bits.
FSMs
Two main requirements:
The sender-site cannot send a frame until its network-layer has a data packet to send.
The receiver-site cannot deliver a data packet to its network-layer until a frame arrives.
These 2 requirements are shown using two FSMs.
Each FSM has only one state, the ready state.

Here is how it works (Figure 11.8):

At Sending Machine
The sending machine remains in the ready state until a request comes from the process in the
network layer.
When this event occurs, the sending machine encapsulates the message in a frame and sends it to the
receiving machine.
2) At Receiving Machine
The receiving machine remains in the ready state until a frame arrives from the sending machine.
When this event occurs, the receiving machine decapsulates the message out of the frame
and delivers it to the process at the network layer.

Example 3.6

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Stop & Wait Protocol

This uses both flow and error control.
Normally, the receiver has limited storage-space.
If the receiver is receiving data from many sources, the receiver may
be overloaded with frames &
discard the frames.
To prevent the receiver from being overloaded with frames, we need to tell the sender to slow down.

Design
At Sender
The sender
sends one frame & starts a timer
keeps a copy of the sent-frame and
waits for ACK-frame from the receiver (okay to go ahead).
Then,
1) If an ACK-frame arrives before the timer expires, the timer is stopped and the sender sends
the next frame.
Also, the sender discards the copy of the previous frame.
2) If the timer expires before ACK-frame arrives, the sender resends the previous frame and
restarts the timer
At Receiver
To detect corrupted frames, a CRC is added to each data frame.
When a frame arrives at the receiver-site, the frame is checked.
If frame’s CRC is incorrect, the frame is corrupted and discarded.
The silence of the receiver is a signal for the sender that a frame was either corrupted or lost.

FSMs

Here is how it works (Figure 11.11):

Sender States
Sender is initially in the ready state, but it can move between the ready and blocking state.
Ready State: When the sender is in this state, it is only waiting for a packet from the network
layer.
If a packet comes from the network layer, the sender creates a frame, saves a copy of
the frame, starts the only timer and sends the frame. The sender then moves to the
blocking state.

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Blocking State: When the sender is in this state, three events can occur:
If a time-out occurs, the sender resends the saved copy of the frame and restarts the
timer.
If a corrupted ACK arrives, it is discarded.
If an error-free ACK arrives, the sender stops the timer and discards the saved copy of
the frame. It then moves to the ready state.
Receiver
The receiver is always in the ready state. Two events may occur:
If an error-free frame arrives, the message in the frame is delivered to the network layer and
an ACK is sent.
If a corrupted frame arrives, the frame is discarded.

Example 3.7

3.7.2.3 Sequence and Acknowledgment Numbers

• Q: How to deal with corrupted-frame?
Ans: If the corrupted-frame arrives at the receiver-site, then the frame is simply discarded.
• Q: How to deal with lost-frames?
Ans: If the receiver receives out-of-order data-frame, then it means that frames were lost. .’. The lost-
frames need to be resent.
Problem in Stop and Wait protocols:
There is no way to identify a frame.
The received-frame could be the correct one, or a duplicate, or a frame out of order.
Solution: 1) Use sequence-number for each data frame.
2) Use Acknowledgment-number for each ACK frame.
Sequence Numbers
Frames need to be numbered. This is done by using sequence-numbers.
A sequence-number field is added to the data-frame.

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Acknowledgment Numbers
An acknowledgment-number field is added to the ACK-frame.
Sequence numbers are 0, 1, 0, 1, 0, 1, . . .
The acknowledgment numbers can also be 1, 0, 1, 0, 1, 0, …
The acknowledgment-numbers always announce the sequence-number of the next frame expected by
the receiver.
For example,
If frame-0 has arrived safely, the receiver sends an ACK-frame with acknowledgment-1
(meaning frame-1 is expected next).

Example 3.8

Piggybacking
A technique called piggybacking is used to improve the efficiency of the bidirectional protocols.
The data in one direction is piggybacked with the acknowledgment in the other direction.
In other words, when node A is sending data to node B, Node A also acknowledges the data received from
node B.

High-Level Data Link Control (HDLC)

HDLC is a bit-oriented protocol for communication over point-to-point and multipoint links.
HDLC implements the ARQ mechanisms.
Configurations and Transfer Modes
HDLC provides 2 common transfer modes that can be used in different configurations:
Normal response mode (NRM)
Asynchronous balanced mode (ABM).
NRM
The station configuration is unbalanced (Figure 11.14).
We have one primary station and multiple secondary stations.
A primary station can send commands, a secondary station can only respond.
The NRM is used for both point-to-point and multiple-point links.

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ABM
The configuration is balanced (Figure 11.15).
Link is point-to-point, and each station can function as a primary and a secondary (acting as peers).
This is the common mode today.

Framing
• To provide the flexibility necessary to support all the options possible in the modes and configurations,
HDLC defines three types of frames:
Information frames (I-frames): are used to transport user data and control information relating to user
data (piggybacking).
Supervisory frames (S-frames): are used only to transport control information.
Unnumbered frames (U-frames): are reserved for system management.
Information carried by U-frames is intended for managing the link itself.
• Each type of frame serves as an envelope for the transmission of a different type of message.

Frame Format

• Various fields of HDLC frame are:

1) Flag Field
This field has a synchronization pattern 01111110.
This field identifies both the beginning and the end of a frame.
2) Address Field
This field contains the address of the secondary station.
If a primary station created the frame, it contains a to-address.
If a secondary creates the frame, it contains a from-address.
This field can be 1 byte or several bytes long, depending on the needs of the network.
3) Control Field
This field is one or two bytes used for flow and error control.
4) Information Field
This field contains the user's data from the network-layer or management information. Its
length can vary from one network to another.
5) FCS Field
This field is the error-detection field. (FCS Frame Check Sequence)

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This field can contain either a 2- or 4-byte standard CRC.

3.8.2.1.1 Control Fields of HDLC Frames
• The control field determines the type of frame and defines its functionality (Figure 11.17).

1) Control Field for I-Frames

I-frames are designed to carry user data from the network-layer.
In addition, they can include flow and error-control information (piggybacking).
The subfields in the control field are:
The first bit defines the type.
If the first bit of the control field is 0, this means the frame is an I-frame.
The next 3 bits N(S) define the sequence-number of the frame. With 3
bits, we can define a sequence-number between 0 and 7
The last 3 bits N(R) correspond to the acknowledgment-number when piggybacking is used.
The single bit between N(S) and N(R) is called the P/F bit.
The P/F field is a single bit with a dual purpose. It can mean poll or final.
It means poll when the frame is sent by a primary station to a secondary (when the address
field contains the address of the receiver).
It means final when the frame is sent by a secondary to a primary (when the address field
contains the address of the sender).

2) Control Field for S-Frames

Supervisory frames are used for flow and error-control whenever piggybacking is either impossible or
inappropriate (e.g., when the station either has no data of its own to send or needs to send a command or
response other than an acknowledgment).
S-frames do not have information fields.
The subfields in the control field are:
If the first 2 bits of the control field is 10, this means the frame is an S-frame.
The last 3 bits N(R) corresponds to the acknowledgment-number (ACK) or negative
acknowledgment-number (NAK).
The 2 bits called code is used to define the type of S-frame itself.
With 2 bits, we can have four types of S-frames:
1) Receive ready (RR) = 00
This acknowledges the receipt of frame or group of frames.
The value of N(R) is the acknowledgment-number.
2) Receive not ready (RNR) = 10
This is an RR frame with 1 additional function:
It announces that the receiver is busy and cannot receive more frames.
It acts as congestion control mechanism by asking the sender to slow down.
The value of N(R) is the acknowledgment-number.
3) Reject (REJ) = 01
It is a NAK frame used in Go-Back-N ARQ to improve the efficiency of the process.
It informs the sender, before the sender time expires, that the last frame is lost or damaged.
The value of N(R) is the negative acknowledgment-number.
4) Selective reject (SREJ) = 11
This is a NAK frame used in Selective Repeat ARQ.
The value of N(R) is the negative acknowledgment-number.

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3) Control Field for U-Frames

Unnumbered frames are used to exchange session management and control information between connected
devices.
U-frames contain an information field used for system management information, but not user data.
Much of the information carried by U-frames is contained in codes included in the control field.
U-frame codes are divided into 2 sections:
A 2-bit prefix before the P/F bit
A 3-bit suffix after the P/F bit.
Together, these two segments (5 bits) can be used to create up to 32 different types of U-frames.

Example 3.9

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Point-To-Point Protocol (PPP)

PPP is one of the most common protocols for point-to-point access.
Today, millions of Internet users who connect their home computers to the server of an ISP use PPP.

Framing
PPP uses a character-oriented (or byte-oriented) frame (Figure 11.20).

Various fields of PPP frame are:

1) Flag
This field has a synchronization pattern 01111110.
This field identifies both the beginning and the end of a frame.
2) Address
This field is set to the constant value 11111111 (broadcast address).
3) Control
This field is set to the constant value 00000011 (imitating unnumbered frames in HDLC).
PPP does not provide any flow control.
Error control is also limited to error detection.
4) Protocol
This field defines what is being carried in the payload field.
Payload field carries either i) user data or ii) other control information.
By default, size of this field = 2 bytes.
5) Payload field
This field carries either i) user data or ii) other control information.
By default, maximum size of this field = 1500 bytes.
This field is byte-stuffed if the flag-byte pattern appears in this field.
Padding is needed if the payload-size is less than the maximum size.
6) FCS
This field is the PPP error-detection field.
This field can contain either a 2- or 4-byte standard CRC.

Byte Stuffing
Since PPP is a byte-oriented protocol, the flag in PPP is a byte that needs to be escaped whenever it
appears in the data section of the frame.
The escape byte is 01111101, which means that every time the flag like pattern appears in the data, this
extra byte is stuffed to tell the receiver that the next byte is not a flag.
Obviously, the escape byte itself should be stuffed with another escape byte.

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3-33

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Transition Phases
The transition diagram starts with the dead state (Figure 11.21).
Dead State
In dead state, there is no active carrier and the line is quiet.
2) Establish State
When 1 of the 2 nodes starts communication, the connection goes into the establish state.
In establish state, options are negotiated between the two parties.
3) Authenticate State
If the 2 parties agree that they need authentication,
Then the system needs to do authentication;
Otherwise, the parties can simply start communication.
4) Open State
Data transfer takes place in the open state.
5) Terminate State
When 1 of the endpoints wants to terminate connection, the system goes to terminate state.

Assignment Questions
1. Explain two types of errors (4*)
2. Compare error detection vs. error correction (2)
3. Explain error detection using block coding technique. (10*)
4. Explain hamming distance for error detection (6*)
5. Explain parity-check code with block diagram. (6*)
6. Explain CRC with block diagram & an example. (10*)
7. Write short notes on polynomial codes. (5*)
8. Explain internet checksum algorithm along with an example. (6*)
9. Explain various FEC techniques. (6)
10. Explain two types of frames. (2)
11. Explain character oriented protocol. (6*)
12. Explain the concept of byte stuffing and unstuffing with example. (6*)
13. Explain bit oriented protocol. (6*)
14. Differentiate between character oriented and bit oriented format for Framing. (6*)
15. Compare flow control and error control. (4)
16. With a neat diagram, explain the design of the simplest protocol with no flow control. (6)
17. Write algorithm for sender site and receiver site for the simplest protocol. (6)
18. Explain Stop-and-Wait protocol (8*)
19. Explain the concept of Piggybacking (2*)
20. Explain in detail HDLC frame format. (8*)
21. Explain 3 type of frame used in HDLC (8*)
22. With a neat schematic, explain the frame structure of PPP protocol. (8*)
23. Explain framing and transition phases in Point-to-Point Protocol. (8*)

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Expected Outcomes
• Explain the types of errors and error correction mechanisms
• Learn about different protocols used in error correction
• Explain about Checksum and advantage of polynomials

Further Reading
1. http://www.mhhe.com/engcs/compsci/forouzan/frontmatter.pdf
2. http://freecomputerbooks.com/Introduction-to-Data-Communications.html

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MODULE 4: TABLE OF CONTENTS

INTRODUCTION
RANDOM ACCESS PROTOCOL
ALOHA
Pure ALOHA
Vulnerable time
Throughput
Slotted ALOHA
Throughput
CSMA
Vulnerable Time
Persistence Methods
CSMA/CD
Minimum Frame-size
Procedure
Energy Level
Throughput
CSMA/CA
Frame Exchange Time Line
Network Allocation Vector
Collision During Handshaking
Hidden-Station Problem
CSMA/CA and Wireless Networks
CONTROLLED ACCESS PROTOCOL
Reservation
Polling
Token Passing
Logical Ring
CHANNELIZATION
FDMA
TDMA
CDMA
Implementation
Chips
Data Representation
Encoding and Decoding
Sequence Generation
ETHERNET PROTOCOL
IEEE Project 802
Ethernet Evolution
STANDARD ETHERNET
Characteristics
Connectionless and Unreliable Service
Frame Format
Frame Length
Addressing
Access Method
Efficiency of Standard Ethernet
Implementation
Encoding and Decoding
Changes in the Standard
Bridged Ethernet

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Switched Ethernet
Full-Duplex Ethernet
FAST ETHERNET (100 MBPS)
Access Method
Physical Layer
Topology
Implementation
Encoding
GIGABIT ETHERNET
MAC Sublayer
Physical Layer
Topology
Implementation
Encoding
TEN GIGABIT ETHERNET
Implementation
INTRODUCTION OF WIRELESS-LANS
Architectural Comparison
Characteristics
Access Control
IEEE 802.11 PROJECT
Architecture
BSS
ESS
Station Types
MAC Sublayer
DCF
Network Allocation Vector
Collision During Handshaking
PCF
Fragmentation
Frame Types
Frame Format
Addressing Mechanism
Exposed Station Problem
Physical Layer
IEEE 802.11 FHSS
IEEE 802.11 DSSS
IEEE 802.11 Infrared
IEEE 802.11a OFDM
IEEE 802.11b DSSS
IEEE 802.11g
BLUETOOTH
Architecture
Piconets
Scatternet
Bluetooth Devices
Bluetooth Layers
Radio Layer
Baseband Layer
TDMA
Links
Frame Types
Frame Format
L2CAP

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MODULE 4
MEDIUM ACCESS
STRUCTURE

Introduction
Objectives
Introduction to Medium Access
Random Access Protocol
Controlled Access Protocols
Channelization Protocols
Ethernet Protocol
Standard-Ethernet
Fast Ethernet (100 Mbps)
Gigabit Ethernet
Ten Gigabit Ethernet
Introduction of Wireless-LANs
4.13 IEEE 802.11
Bluetooth
Assignment Questions
Expected Outcomes
Further Reading

Introduction
This module provides an overview of the sub-layer of Data link layer i.e., Medium Access Control
layer and its functionalities.
Objectives
• Learn about random access protocol
• Explain controlled access protocols
• Learn about channelization protocols
• Learn about Standard Ethernet, Fast Ethernet , Gigabit Ethernet
• Explain about IEEE 802.11, Bluetooth

Introduction to Medium Access

When nodes use shared-medium, we need multiple-access protocol to coordinate access to medium.
Analogy:
This problem is similar to the rules of speaking in an assembly.
We need to ensure
Each people has right to speak.
Two people do not speak at the same time
Two people do not interrupt each other (i.e. Collision Avoidance)
Many protocols have been designed to handle access to a shared-link (Figure 12.1).
These protocols belong to a sublayer in the data-link layer called Media Access Control (MAC).
1) Four random-access protocols (or Contention Methods):
iv)
i) ALOHA ii) CSMA iii) CSMA/CD CSMA/CA
These protocols are mostly used in LANs and WANs.
2) Three controlled-access protocols:
i) Reservation ii) Polling iii) Token-passing

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Some of these protocols are used in LANs.

3) Three channelization protocols:
i) FDMA ii) TDMA iii) CDMA
These protocols are used in cellular telephony.

Random Access Protocol

No station is superior to another station.
No station is assigned control over other station.
To send the data, a station uses a procedure to make a decision on whether or not to send.
This decision depends on the state of the medium: idle or busy.
This is called Random Access because
Transmission is random among the stations.
There is no scheduled-time for a station to transmit.
This is called Contention Method because
Stations compete with one another to access the medium.
If more than one station tries to send,
there is an access-conflict (i.e. collision) and the frames will be destroyed.
Each station follows a procedure that answers the following questions:
When can the station access the medium?
What can the station do if the medium is busy?
How can the station determine the success or failure of the transmission?
What can the station do if there is a collision?
Four random-access protocols (or Contention methods):
ALOHA
CSMA (Carrier Sense Multiple Access)
CSMA/CD (Carrier Sense Multiple Access with Collision-detection)
CSMA/CA (Carrier Sense Multiple Access with Collision Avoidance)
ALOHA
ALOHA was designed for a wireless LAN, but it can be used on any shared medium.
Since the medium is shared between the stations, there is possibility of collisions.
When 2 or more stations send the data simultaneously, there is possibility of collision & data loss.

Pure ALOHA

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Here is how it works (Figure 12.2):

The sender sends a frame & starts the timer.
The receiver receives the frame and responds with an acknowledgment.
If the acknowledgment does not arrive after a time-out period, the sender resends the frame. The
sender assumes that the frame (or the acknowledgment) has been destroyed.
Since the medium is shared between the stations, there is possibility of collisions.
If two stations try to resend the frames after the time-out, the frames will collide again.
Two methods to deal with collision:
Randomness
When the time-out period passes, each station waits a random amount of time before
resending the frame. This time is called back-off time TB.
The randomness will help avoid more collisions.
ii) Limit Maximum Retransmission
This method prevents congestion by reducing the number of retransmitted frames.
After a maximum number of retransmission-attempts Kmax, a station must give up and try later
(Figure 12.3).

Vulnerable time
• The vulnerable-time is defined as a time during which there is a possibility of collision.
where Tfr = Frame transmission time

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In Figure 12.4,
If station B sends a frame between t-Tfr and t, this leads to a collision between the frames
from station A and station B.
If station C sends a frame between t and t+Tfr, this leads to a collision between the frames from
station A and station C.

Example 4.1

Throughput
• The average number of successful transmissions is given by
where G = average no. of frames in one frame transmission time (Tfr)
For G = 1, the maximum throughput Smax = 0.184.
In other words, out of 100 frames, 18 frames reach their destination successfully.

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Example 4.2

Slotted ALOHA
Slotted ALOHA was invented to improve the efficiency of pure ALOHA.
The time is divided into time-slots of Tfr seconds (Figure 12.5).
The stations are allowed to send only at the beginning of the time-slot.

If a station misses the time-slot, the station must wait until the beginning of the next time-slot.
If 2 stations try to resend at beginning of the same time-slot, the frames will collide again (Fig 12.6).

• The vulnerable time is given by:

Throughput
The average number of successful transmissions is given by
For G = 1, the maximum throughput Smax = 0.368.
In other words, out of 100 frames, 36 frames reach their destination successfully.

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Example 4.3

CSMA
CSMA was developed to minimize the chance of collision and, therefore, increase the performance.
CSMA is based on the principle “sense before transmit” or “listen before talk.”
Here is how it works:
Each station checks the state of the medium: idle or busy.
i) If the medium is idle, the station sends the data.
If the medium is busy, the station defers sending.
CSMA can reduce the possibility of collision, but it cannot eliminate it.

The possibility of collision still exists.

For example:
When a station sends a frame, it still takes time
for the first bit to reach every station and
for every station to sense it.
For example: In Figure 12.7,
At time t1, station B senses & finds the medium idle, so sends a frame.
At time t2, station C senses & finds the medium idle, so sends a frame.
The 2 signals from both stations B & C collide and both frames are destroyed.

Vulnerable Time
The vulnerable time is the propagation time Tp (Figure 12.8).
The propagation time is the time needed for a signal to propagate from one end of the medium to the other.
Collision occurs when
a station sends a frame, and

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other station also sends a frame during propagation time

If the first bit of the frame reaches the end of the medium, every station will refrain from sending.

Persistence Methods
• Q: What should a station do if the channel is busy or idle?
Three methods can be used to answer this question:
1) 1-persistent method 2) Non-persistent method and 3) p-persistent method

1) 1-Persistent
Before sending a frame, a station senses the line (Figure 12.10a).
If the line is idle, the station sends immediately (with probability = 1).
If the line is busy, the station continues sensing the line.
This method has the highest chance of collision because 2 or more stations:
may find the line idle and
send the frames immediately.
2) Non-persistent
• Before sending a frame, a station senses the line (Figure 12.10b). i) If
the line is idle, the station sends immediately.
ii) If the line is busy, the station waits a random amount of time and then senses the line again.

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• This method reduces the chance of collision because 2 or more stations: →

will not wait for the same amount of time and
→ will not retry to send simultaneously.
3) P-Persistent
This method is used if the channel has time-slots with a slot-duration equal to or greater than the maximum
propagation time (Figure 12.10c).
Advantages:
It combines the advantages of the other 2 methods.
It reduces the chance of collision and improves efficiency.
After the station finds the line idle, it follows these steps:
With probability p, the station sends the frame.
With probability q=1-p, the station waits for the beginning of the next time-slot and checks the line
again.
If line is idle, it goes to step 1.
If line is busy, it assumes that collision has occurred and uses the back off procedure.
CSM
A/CD
Disadvantage of CSMA: CSMA does not specify the procedure after a collision has occurred.
Solution: CSMA/CD enhances the CSMA to handle the collision.
Here is how it works (Figure 12.12):
A station
sends the frame &
then monitors the medium to see if the transmission was successful or not.
If the transmission was unsuccessful (i.e. there is a collision), the frame is sent again.

In the Figure 12.11,

At time t1, station A has executed its procedure and starts sending the bits of its frame.
At time t2, station C has executed its procedure and starts sending the bits of its frame.
The collision occurs sometime after time t2.
Station C detects a collision at time t3 when it receives the first bit of A's frame.
Station C immediately aborts transmission.
Station A detects collision at time t4 when it receives the first bit of C's frame.
Station A also immediately aborts transmission.
Station A transmits for the duration t4–t1. Station
C transmits for the duration t3-t2.
For the protocol to work:
The length of any frame divided by the bit rate must be more than either of these durations.

Minimum Frame Size

For CSMA/CD to work, we need to restrict the frame-size.
Before sending the last bit of the frame, the sender must
detect a collision and
abort the transmission.
This is so because the sender
does not keep a copy of the frame and

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does not monitor the line for collision-detection.

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Frame transmission time Tfr is given by

Tfr=2Tp where Tp=maximum propagation time

Example 4.4

Procedure
CSMA/CD is similar to ALOHA with 2 differences (Figure 12.13):
Addition of the persistence process.
We need to sense the channel before sending the frame by using non-persistent, 1-persistent
or p-persistent.
Frame transmission.
In ALOHA, first the entire frame is transmitted and then acknowledgment is waited for.
In CSMA/CD, transmission and collision-detection is a continuous process.

Energy Level
• In a channel, the energy-level can have 3 values: 1) Zero 2) Normal and 3) Abnormal. 1)
At zero level, the channel is idle (Figure 12.14).
2) At normal level, a station has successfully captured the channel and is sending its frame. 3) At
abnormal level, there is a collision and the level of the energy is twice the normal level.
• A sender needs to monitor the energy-level to determine if the channel is →
Idle
→ Busy or
→ Collision mode

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Throughput
The throughput of CSMA/CD is greater than pure or slotted ALOHA.
The maximum throughput is based on
different value of G
persistence method used (non-persistent, 1-persistent, or p-persistent) and
‘p‟ value in the p-persistent method.
For 1-persistent method, the maximum throughput is 50% when G =1.
For non-persistent method, the maximum throughput is 90% when G is between 3 and 8.
CSMA/CA
Here is how it works (Figure 12.15):
A station needs to be able to receive while transmitting to detect a collision.
When there is no collision, the station receives one signal: its own signal.
When there is a collision, the station receives 2 signals:
Its own signal and
Signal transmitted by a second station.
To distinguish b/w these 2 cases, the received signals in these 2 cases must be different.

CSMA/CA was invented to avoid collisions on wireless networks.

Three methods to avoid collisions (Figure 12.16):
Interframe space 2) Contention window and 3) Acknowledgments

1) Interframe Space (IFS)

Collisions are avoided by deferring transmission even if the channel is found idle.
When the channel is idle, the station does not send immediately.

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Rather, the station waits for a period of time called the inter-frame space or IFS.
• After the IFS time,
if the channel is still idle,
then, the station waits for the contention-time &
finally, the station sends the frame.
• IFS variable can also be used to prioritize stations or frame types.
For example, a station that is assigned a shorter IFS has a higher priority.
2) Contention Window
The contention-window is an amount of time divided into time-slots.
A ready-station chooses a random-number of slots as its wait time.
In the window, the number of slots changes according to the binary exponential back-off strategy.
For example:
At first time, number of slots is set to one slot and
Then, number of slots is doubled each time if the station cannot detect an idle channel.

3) Acknowledgment
There may be a collision resulting in destroyed-data.
In addition, the data may be corrupted during the transmission.
To help guarantee that the receiver has received the frame, we can use
Positive acknowledgment and
Time-out timer

Frame Exchange Time Line

• Two control frames are used: 1)
Request to send (RTS) 2)
Clear to send (CTS)
• The procedure for exchange of data and control frames in time (Figure 12.17):
1) The source senses the medium by checking the energy level at the carrier frequency.
If the medium is idle,
then the source waits for a period of time called the DCF interframe space (DIFS);
finally, the source sends a RTS.
The destination
receives the RTS
waits a period of time called the short interframe space (SIFS)
sends a control frame CTS to the source.
CTS indicates that the destination station is ready to receive data.
The source
receives the CTS
waits a period of time SIFS
sends a data to the destination
The destination
receives the data
waits a period of time SIFS
sends a acknowledgment ACK to the source.
ACK indicates that the destination has been received the frame.

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Network Allocation Vector

• When a source-station sends an RTS, it includes the duration of time that it needs to occupy the channel.
• The remaining stations create a timer called a network allocation vector (NAV).
• NAV indicates waiting time to check the channel for idleness.
• Each time a station accesses the system and sends an RTS frame, other stations start their NAV.

Collision During Handshaking

• Two or more stations may try to send RTS at the same time.
• These RTS may collide.
• The source assumes there has been a collision if it has not received CTS from the destination.
• The backoff strategy is employed, and the source tries again.

Hidden-Station Problem
• Figure 12.17 also shows that the RTS from B reaches A, but not C.
• However, because both B and C are within the range of A, the CTS reaches C.
• Station C knows that some hidden station is using the channel and refrains from transmitting until that
duration is over.

CSMA/CA and Wireless Networks

CSMA/CA was mostly intended for use in wireless networks.
However, it is not sophisticated enough to handle some particular issues related to wireless networks, such
as hidden terminals or exposed terminals.

Controlled Access Protocols

Here, the stations consult one another to find which station has the right to send.
A station cannot send unless it has been authorized by other stations.
Three popular controlled-access methods are: 1) Reservation 2) Polling 3) Token Passing

Reservation
Before sending data, each station needs to make a reservation of the medium.
Time is divided into intervals.
In each interval, a reservation-frame precedes the data-frames.
If no. of stations = N, then there are N reservation mini-slots in the reservation-frame.
Each mini-slot belongs to a station.
When a station wants to send a data-frame, it makes a reservation in its own minislot.
The stations that have made reservations can send their data-frames.

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For example (Figure 12.18):

5 stations have a 5-minislot reservation-frame.
In the first interval, only stations 1, 3, and 4 have made reservations.
In the second interval, only station-1 has made a reservation.

Polling
• In a network,
One device is designated as a primary station and
Other devices are designated as secondary stations.
Functions of primary-device:
The primary-device controls the link.
The primary-device is always the initiator of a session.
The primary-device is determines which device is allowed to use the channel at a given time.
All data exchanges must be made through the primary-device.
The secondary devices follow instructions of primary-device.
Disadvantage: If the primary station fails, the system goes down.
Poll and select functions are used to prevent collisions (Figure 12.19).
Select
If the primary wants to send data, it tells the secondary to get ready to receive; this is called select
function.
The primary
alerts the secondary about upcoming transmission by sending select frame (SEL)
then waits for an acknowledgment (ACK) from secondary
then sends the data frame and
finally waits for an acknowledgment (ACK) from the secondary.
2) Poll
If the primary wants to receive data, it asks the secondaries if they have anything to send; this is
called poll function.
When the first secondary is approached, it responds either
with a NAK frame if it has no data to send or
with data-frame if it has data to send.
If the response is negative (NAK frame), then the primary polls the next secondary in the
same manner.
When the response is positive (a data-frame), the primary
reads the frame and
returns an acknowledgment (ACK frame).

Token Passing

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In a network, the stations are organized in a ring fashion i.e. for each station; there is a predecessor and a
successor.
The predecessor is the station which is logically before the station in the ring.
The successor is the station which is after the station in the ring.
The current station is the one that is accessing the channel now.
A token is a special packet that circulates through the ring.
Here is how it works:
A station can send the data only if it has the token.
When a station wants to send the data, it waits until it receives the token from its predecessor.
Then, the station holds the token and sends its data.
When the station finishes sending the data, the station
releases the token
passes the token to the successor.
Main functions of token management:
Stations must be limited in the time they can hold the token.
The token must be monitored to ensure it has not been lost or destroyed.
For ex: if a station that is holding the token fails, the token will disappear from the network
Assign priorities
to the stations and
to the types of data being transmitted.
Make low-priority stations release the token to high priority stations.

Logical Ring
In a token-passing network, stations do not have to be physically connected in a ring; the ring can be a
logical one.
Four physical topologies to create a logical ring (Figure 12.20):
Physical ring
Dual ring
Bus ring
Star ring

1) Physical Ring Topology

When a station sends token to its successor, token cannot be seen by other stations. (Figure 12.20a)
This means that the token does not have the address of the next successor.
Disadvantage: If one of the links fails, the whole system fails.
2) Dual Ring Topology
• A second (auxiliary) ring
→ is used along with the main ring (Figure 12.20b).
→ operates in the reverse direction compared with the main ring. →
is used for emergencies only (such as a spare tire for a car).

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• If the main ring fails, the system automatically combines the 2 rings to form a temporary ring.
• After the failed link is restored, the second ring becomes idle again.
• Each station needs to have 2 transmitter-ports and 2 receiver-ports.
• This topology is used in
i) FDDI (Fiber Distributed Data Interface) and
ii) CDDI (Copper Distributed Data Interface).

3) Bus Ring Topology

• The stations are connected to a single cable called a bus (Figure 12.20c).
• This makes a logical ring, because each station knows the address of its successor and predecessor.
• When a station has finished sending its data, the station →
releases the token and
→ inserts the address of its successor in the token.
• Only the station gets the token to access the shared media.
• This topology is used in the Token Bus LAN.

4) Star Ring Topology

The physical topology is a star (Figure 12.20d).
There is a hub that acts as the connector.
The wiring inside the hub makes the ring i.e. the stations are connected to the ring through the 2 wire
connections.
Disadvantages:
This topology is less prone to failure because
If a link goes down,
then the link will be bypassed by the hub and
the rest of the stations can operate.
Also adding and removing stations from the ring is easier.
This topology is used in the Token Ring LAN.

Channelization Protocols
Channelization is a multiple-access method.
The available bandwidth of a link is shared b/w different stations in time, frequency, or through code.
Three channelization protocols:
FDMA (Frequency Division Multiple Access)
TDMA (Time Division Multiple Access) and
CDMA (Code Division Multiple Access)

FDMA
The available bandwidth is divided into frequency-bands (Figure 12.21).
Each band is reserved for a specific station.
Each station can send the data in the allocated band.
Each station also uses a bandpass filter to confine the transmitter frequencies.
To prevent interferences, small guard bands are used to separate the allocated bands from one another.

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• FDM vs. FDMA

1) FDM
FDM is a multiplexing method in the physical layer.
FDM
combines individual-loads from low-bandwidth channels and
transmits aggregated-load by using a high-bandwidth channel.
The channels that are combined are low-pass.
The multiplexer
modulates & combines the signals and
creates a bandpass signal.
The bandwidth of each channel is shifted by the multiplexer.
2) FDMA
FDMA is an access method in the data link layer.
In each station, the data link layer tells the physical layer to make a bandpass signal from the data
passed to it.
The signal must be created in the allocated band.
There is no physical multiplexer at the physical layer.
The signals created at each station are automatically bandpass-filtered.
They are mixed when they are sent to the common channel.
TDMA
The stations share the bandwidth of the channel in time (Figure 12.22).
Each time-slot is reserved for a specific station.
Each station can send the data in the allocated time-slot.

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• Main problem: Achieving synchronization between the different stations.

i.e. each station needs to know the beginning of its slot and the location of its slot. This
may be difficult because of propagation delays introduced in the system.
To compensate for the delays, we can insert guard-times.
Normally, synchronization is accomplished by having some synchronization bits at the beginning of each
slot.
TDMA vs. TDM
1) TDM
TDM is a multiplexing method in the physical layer.
TDM
combines the individual-data from slower channels and
transmits the aggregated- data by using a faster channel.
The multiplexer interleaves data units from each channel.
2) TDMA
TDMA is an access method in the data link layer.
In each station, the data link layer tells the physical layer to use the allocated time-slot.
There is no physical multiplexer at the physical layer.
CDMA
CDMA simply means communication with different codes.
CDMA differs from FDMA because
only one channel occupies the entire bandwidth of the link.
CDMA differs from TDMA because
all stations can send data simultaneously; there is no timesharing.

(Analogy: CDMA simply means communication with different codes.

For example, in a large room with many people, 2 people can talk privately in English if nobody else
understands English. Another 2 people can talk in Chinese if they are the only ones who understand
Chinese, and so on).

Implementation
Let us assume we have four stations 1, 2, 3, and 4 connected to the same channel.
The data from station-1 are d1, from station-2 are d2, and so on.
The code assigned to the first station is c1, to the second is c2, and so on.
We assume that the assigned codes have 2 properties.
If we multiply each code by another, we get 0.
If we multiply each code by itself, we get 4 (the number of stations).

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Here is how it works (Figure 12.23):

Station-1 multiplies the data by the code to get d1.c1.
Station-2 multiplies the data by the code to get d2.c2. And so on.
The data that go on the channel are the sum of all these terms.
The receiver multiplies the data on the channel by the code of the sender.
For example, suppose stations 1 and 2 are talking to each other.
Station-2 wants to hear what station-1 is saying.
Station-2 multiplies the data on the channel by c1 the code of station-1.
(c1.c1)=4, (c2.c1)=0, (c3.c1)=0, and (c4.c1)=0,
Therefore, station-2 divides the result by 4 to get the data from station-1.

Chips
CDMA is based on coding theory.
Each station is assigned a code, which is a sequence of numbers called chips (Figure 12.24).

These sequences were carefully selected & are called orthogonal sequences
These sequences have the following properties:
Each sequence is made of N elements, where N is the number of stations.
Multiplication of a sequence by a scalar:
If we multiply a sequence by a number i.e. every element in the sequence is multiplied by
that element.
For example,

3) Inner product of 2 equal sequences:

If we multiply 2 equal sequences, element by element, and add the results, we get N, where N
is the number of elements in the each sequence.
For example,

4) Inner product of 2 different sequences:

If we multiply 2 different sequences, element by element, and add the results, we get 0. For

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example,

5) Adding 2 sequences means adding the corresponding elements. The result is another sequence.
For example,

Data Representation
We follow the following rules for encoding:
To send a 0 bit, a station encodes the bit as -1
To send a 1 bit, a station encodes the bit as +1
When a station is idle, it sends no signal, which is interpreted as a 0.
Encoding and Decoding
We assume that
Stations 1 and 2 are sending a 0 bit.
Station-4 is sending a 1 bit.
Station-3 is silent.
Here is how it works (Figure 12.26):
At the sender-site, the data are translated to -1, -1, 0, and +1.
Each station multiplies the corresponding number by its chip (its orthogonal sequence).
The result is a new sequence which is sent to the channel.
The sequence on the channel is the sum of all 4 sequences.
Now imagine station-3, which is silent, is listening to station-2.
Station-3 multiplies the total data on the channel by the code for station-2, which is [+1 -1 +1-1], to
get

Sequence Generation
To generate chip sequences, we use a Walsh table (Figure 12.29).
Walsh table is a 2-dimensional table with an equal number of rows and columns.

In the Walsh table, each row is a sequence of chips.

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W1 for a one-chip sequence has one row and one column. We can choose –1 or +1 for the chip for this trivial
table (we chose +1).
According to Walsh, if we know the table for N sequences WN, we can create the table for 2N sequences
W2N (Figure 12.29).
The WN with the overbar WN stands for the complement of WN where each +1 is changed to -1 and vice
versa.
After we select W1, W2 can be made from four W1's, with the last one the complement of W1
After W2 is generated, W4 can be made of four W2's, with the last one the complement of W2.
The number of sequences in a Walsh table needs to be N = 2m.
Example 4.5

Example 4.6

Example 4.7

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MODULE 4(CONT.): WIRED LANS -- ETHERNET

Ethernet Protocol
IEEE Project 802
• The data-link-layer is divided into 2 sublayers (Figure 13.1):
1) LLC
Flow-control, error-control, and framing duties are grouped into one sublayer called LLC.
Framing is handled in both the LLC and the MAC.
LLC vs. MAC
i) LLC provides one single data-link-control protocol for all IEEE
LANs. ii) MAC provides different protocols for different LANs.
A single LLC protocol can provide interconnectivity between different LANs because
→ it makes the MAC sublayer transparent.
2) MAC
This defines the specific access-method for each LAN.
For example:
CSMA/CD is used for Ethernet LANs.
Token-passing method is used for Token Ring and Token Bus LANs.
The framing function is also handled by the MAC layer.
The MAC contains a number of distinct modules.
Each module defines the access-method and the framing-format specific to the corresponding LAN
protocol.

4.5.2 Ethernet Evolution

Four generations of Ethernet (Figure 13.2):
Standard-Ethernet (10 Mbps)
Fast-Ethernet (100 Mbps)
Gigabit-Ethernet (1 Gbps) and
Ten-Gigabit-Ethernet (10 Gbps)

Standard-Ethernet
The original Ethernet technology with data-rate of 10 Mbps are referred to as the Standard Ethernet.
Characteristics
Connectionless and Unreliable Service
Ethernet provides a connectionless service. Thus, each frame sent is independent of another frame.

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Ethernet has no connection establishment or connection termination phases.

The sender sends a frame whenever it has it.
The receiver may or may not be ready for receiving the frame.
The sender may overload the receiver with frames, which may result in dropping frames.
If a frame drops, the sender will not know about it.
If a frame is corrupted during transmission, the receiver drops the frame.
Since IP is also connectionless, it will also not know about frame drops.
If the transport layer is UDP (connectionless protocol), the frame is lost.
If the transport layer is TCP, the sender-TCP does not receive acknowledgment for its segment and
sends it again.
Ethernet is also unreliable like IP and UDP.

Frame Format

• The Ethernet frame contains 7 fields (Figure 13.3):

1) Preamble
This field contains 7 bytes (56 bits) of alternating 0s and 1s.
This field
→ alerts the receiving-system to the coming frame and
→ enables the receiving-system to synchronize its input timing.
The preamble is actually added at the physical-layer and is not (formally) part of the frame.
2) Start frame delimiter (SFD)
This field signals the beginning of the frame.
The SFD warns the stations that this is the last chance for synchronization.
This field contains the value: 10101011.
The last 2 bits (11) alerts the receiver that the next field is the destination-address.
3) Destination-address (DA)
This field contains the physical-address of the destination-station.
4) Source-address (SA)
This field contains the physical-address of the sender-station.
5) Length or type
This field is defined as a i) type field or ii) length field.
In original Ethernet, this field is used as the type field.
Type field defines the upper-layer protocol using the MAC frame.
In IEEE standard, this field is used as the length field.
Length field defines the number of bytes in the data-field.
6) Data
This field carries data encapsulated from the upper-layer protocols.
Minimum data size = 46 bytes. Maximum data size = 1500 bytes.
7) CRC
This field contains error detection information such as a CRC-32.
Frame Length
• Ethernet has imposed restrictions on both minimum & maximum lengths of a frame (Figure 13.5).

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The minimum length restriction is required for the correct operation of CSMA/CD.
Minimum length of frame = 64 bytes.
Minimum data size = 46 bytes.
Header size + Trailer size = 14 + 4 = 18 bytes.
(i.e. 18 bytes 6 bytes source-address + 6 bytes dest-address + 2 bytes length + 4 bytes CRC).
The minimum length of data from the upper layer = 46 bytes.
If the upper-layer packet is less than 46 bytes, padding is added to make up the difference.
Maximum length of frame =1518 bytes.
Maximum data size = 1500 bytes.
Header size + trailer size = 14 + 4 = 18 bytes.
The maximum length restriction has 2 reasons:
Memory was very expensive when Ethernet was designed.
A maximum length restriction helped to reduce the size of the buffer.
This restriction prevents one station from
monopolizing the shared medium
blocking other stations that have data to send.
Addressing
In an Ethernet-network, each station has its own NIC (6-byte 48 bits).
The NIC provides the station with a 6-byte physical-address (or Ethernet-address).
For example, the following shows an Ethernet MAC address:
(NIC network interface card)
Example 4.8

Unicast, Multicast, and Broadcast Addresses

A source-address is always a unicast address i.e. the frame comes from only one station.

However, the destination-address can be 1) Unicast 2) Multicast or 3) Broadcast.

As shown in Figure 13.4,
If LSB of first byte in a destination-address is 0,
Then, the address is unicast;
Otherwise, the address is multicast.
A unicast destination-address defines only one recipient.
The relationship between the sender and the receiver is one-to-one.

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A multicast destination-address defines a group of addresses.

The relationship between the sender and the receivers is one-to-many.
The broadcast address is a special case of the multicast address.
The recipients are all the stations on the LAN.
A broadcast destination-address is 48 1s (6-byte 48 bits).
Standard Ethernet uses a coaxial cable (bus topology) or a set of twisted-pair cables with a hub (star
topology) (Figure 13.5).

• Question: How actual unicast, multicast & broadcast transmissions are distinguished from each other?
Answer: The way the frames are kept or dropped.
In a unicast transmission, all stations will receive the frame, the intended recipient keeps and handles
the frame; the rest discard it.
In a multicast transmission, all stations will receive the frame, the stations that are members of the
group keep and handle it; the rest discard it.
In a broadcast transmission, all stations (except the sender) will receive the frame and all stations
(except the sender) keep and handle it.

Example 4.9

Access-method
Standard-Ethernet uses 1-persistent CSMA/CD.
1) Slot Time
Slot time = round-trip time + time required to send the jam sequence.
The RTT means time required for a frame to travel from one end of a maximum-length network to the other
end (RTT round-trip time).
The slot time is defined in bits.

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The slot time is the time required for a station to send 512 bits.
The actual slot time depends on the data-rate.
For example: 10-Mbps Ethernet has slot time of 51.2 µs.
2) Slot Time and Collision
The choice of a 512-bit slot time was not accidental.
It was chosen to allow the proper functioning of CSMA/CD.
3) Slot Time and Maximum Network Length
There is a relationship between
slot time and
maximum length of the network (collision domain).
This relationship is dependent on the propagation-speed of the signal in the particular medium.
In most transmission media, the signal propagates at 2x108 m/s (two-thirds of the rate for
propagation in air).
For traditional Ethernet, we calculate

MaxLength= (2 x 108) x (51.2 X 10 - 6)/2= 5120m

Efficiency of Standard Ethernet
The efficiency is defined as the ratio of the time used by a station to send data to the time the medium is
occupied by this station.
The practical efficiency of standard Ethernet has been measured to be

where a = number of frames that can fit on the medium. a =

(propagation delay)/(transmission delay)
As the value of parameter a decreases, the efficiency increases.
If the length of the media is shorter or the frame size longer, the efficiency increases.
In the ideal case, a = 0 and the efficiency is 1.

Example 4.10

Implementation
• The Standard-Ethernet defines several physical-layer implementations (Table 13.1).

Encoding and Decoding

All standard implementations use digital-signaling (baseband) at 10 Mbps (Figure 13.6).
At the sender, data are converted to a digital-signal using the Manchester scheme.
At the receiver, the received-signal is
interpreted as Manchester and
decoded into data.

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1) 10Base5: Thick Ethernet

10Base5 uses a bus topology (Figure 13.7).
A external transceiver is connected to a thick coaxial-cable.
(transceiver transmitter/receiver)
The transceiver is responsible for
transmitting
receiving and
detecting collisions.
The transceiver is connected to the station via a coaxial-cable. The
cable provides separate paths for sending and receiving.
The collision can only happen in the coaxial cable.

The maximum-length of the cable must not exceed 500m.

If maximum-length is exceeded, then there will be excessive degradation of the signal.
If a cable-length of more than 500 m is needed, the total cable-length can be divided into up to 5
segments.
Each segment of maximum length 500-meter, can be connected using repeaters.
2) 10Base2: Thin Ethernet
10Base2 uses a bus topology (Figure 13.8).
The cable is much thinner and more flexible than 10Base5.
Flexible means the cable can be bent to pass very close to the stations.
The transceiver is part of the NIC, which is installed inside the station.
The collision can only happen in the coaxial cable.

Advantages:
Thin coaxial-cable is less expensive than thick coaxial-cable.
Tee connections are much cheaper than taps.
Installation is simpler because the thin coaxial cable is very flexible.
Disadvantage:
Length of each segment cannot exceed 185m due to the high attenuation in the cable.
3)10Base- T: Twisted-Pair Ethernet
10Base-T uses a star topology to connect stations to a hub (Figure 13.9).
The stations are connected to a hub using two pairs of twisted-cable.

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Two pairs of twisted cable create two paths between the station and the hub.
First path for sending.
Second path for receiving.
The collision can happen in the hub.
The maximum length of the cable is 100 m. This minimizes the effect of attenuation in the
cable.
4) 10Base-F: Fiber Ethernet
10Base-F uses a star topology to connect stations to a hub (Figure 13.10).
The stations are connected to the hub using two fiber-optic cables.

Changes in the Standard

Bridged Ethernet
Bridges have two effects on an Ethernet LAN:
They raise the bandwidth &
They separate collision domains.
1) Raising the Bandwidth
A bridge divides the network into two or more networks.
Bandwidth-wise, each network is independent.

For example (Figure 13.12):

A network with 12 stations is divided into two networks, each with 6 stations.
Now each network has a capacity of 10 Mbps.
The 10-Mbps capacity in each segment is now shared between 6 stations (actually 7 because
the bridge acts as a station in each segment), not 12 stations.
In a network with a heavy load, each station theoretically is offered 10/7 Mbps instead of 10/12

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Data Communication(18CS46)

Mbps.
2) Separating Collision Domains
Another advantage of a bridge is the separation of the collision domain.
Figure 13.13 shows the collision domains for an un-bridged and a bridged network.
You can see that the collision domain becomes much smaller and the probability of collision is reduced
tremendously.

Switched Ethernet
The idea of a bridged LAN can be extended to a switched LAN (Figure 13.14).
If we can have a multiple-port bridge, we can have an N-port switch.
In this way, the bandwidth is shared only between the station and the switch.
A layer-2 switch is an N-port bridge with additional sophistication that allows faster handling of the packets.

Full-Duplex Ethernet
The full-duplex mode increases the capacity of each domain from 10 to 20 Mbps.
Instead of using one link between the station and the switch, the configuration uses two links: one to
transmit and one to receive.
1) No Need for CSMA/CD
In full-duplex switched Ethernet,
There is no need for the CSMA/CD method.
Each station is connected to the switch via two separate links.
Each station or switch can send and receive independently without worrying about collision.
Each link is a point-to-point dedicated path between the station and the switch.
There is no longer a need for carrier sensing; there is no longer a need for collision-detection.
The job of the MAC layer becomes much easier.
Carrier sensing and collision-detection functionalities of the MAC sublayer can be turned off.
MAC Control Layer
To provide for flow and error control in full-duplex switched Ethernet, a new sublayer, called
the MAC control, is added between the LLC sublayer and the MAC sublayer.

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Fast Ethernet (100 Mbps)

IEEE created Fast-Ethernet under the name 802.3u.
Fast-Ethernet was designed to compete with LAN protocols such as FDDI or Fiber Channel.
Goals of Fast-Ethernet:
Upgrade the data-rate to 100 Mbps.
Make it compatible with Standard-Ethernet.
Keep the same 48-bit address.
Keep the same frame format.
Keep the same minimum and maximum frame-lengths.

Access Method
Access method is same in Standard-Ethernet.
Only the star topology is used.
For the star topology, there are 2 choices:
In the half-duplex approach, the stations are connected via a hub.
CSMA/CD was used as access-method.
In the full-duplex approach, the connection is made via a switch with buffers at each port.
There is no need for CSMA/CD.
Auto negotiation
A new feature added to Fast-Ethernet is called autonegotiation.
It provides a station/hub with a range of capabilities.
It was used for the following purposes:
To allow 2 devices to negotiate the mode or data-rate of operation.
To allow incompatible devices to connect to one another.
For example: a device with a maximum capacity of 10 Mbps can communicate with
a device with a 100 Mbps capacity.
To allow one device to have multiple capabilities.
To allow a station to check a hub's capabilities.
Physical-layer
The physical-layer in Fast-Ethernet is more complicated than the one in Standard-Ethernet.
Some of the features of this layer are as follows. 1) Topology 2) Implementation and 3) Encoding.
Topology
Fast-Ethernet is used to connect two or more stations together (Figure 13.19).
If there are only 2 stations, they can be connected in point-to-point.
If there are 3 or more stations, they can be connected in star topology with a hub at the center.

Implementation
Fast-Ethernet can be classified as either a two-wire or a four-wire implementation (Table 13.2).
The 2-wire implementations use
Category 5 UTP (100Base-TX) or
Fiber-optic cable (100Base-FX)
The 4-wire implementations use category 3 UTP (100Base-T4).

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Encoding
There are 3 different encoding schemes.
100Base-TX
This uses 2 pairs of twisted-pair cable (either category 5 UTP or STP) (Figure 13.16a).
The MLT-3 encoding scheme is used for implementation.
This is because MLT -3 has good bandwidth
performance. However, 4B/5B block-coding is used to provide bit
synchronization.
This is because MLT-3 is not a self-synchronous line coding scheme.
4B/5B coding creates a data-rate of 125 Mbps, which is fed into MLT-3 for encoding.
2) 100Base-FX
This uses 2 pairs of fiber-optic cables (Figure 13.16b).
Optical fiber can easily handle high bandwidth requirements.
The NRZ-I encoding scheme is used for implementation.
However, 4B/5B block-coding is used to provide bit synchronization.
This is because NRZ-I is not a self-synchronous line coding scheme.
4B/5B encoding increases the bit rate from 100 to 125 Mbps, which can easily be handled by fiber-
optic cable.
3) 100Base-T4
This uses 4 pairs of UTP for transmitting 100 Mbps (Figure 13.16c).
Each UTP cannot easily handle more than 25 Mbaud.
One pair switches between sending and receiving.
Three pairs of UTP can handle only 75 Mbaud (25 Mbaud) each.
Encoding/decoding is more complicated.
We need an encoding scheme that converts 100 Mbps to a 75 Mbaud signal. This requirement is
satisfied by 8B/6T.
The 8B/6T encoding scheme is used for implementation.
8 data elements are encoded as 6 signal elements.
This means that 100 Mbps uses only (6/8) x 100 Mbps, or 75 Mbaud.

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Gigabit Ethernet
IEEE created Gigabit-Ethernet under the name 802.3z.
Goals of Gigabit-Ethernet:
Upgrade the data-rate to 1 Gbps.
Make it compatible with Standard or Fast-Ethernet.
Use the same 48-bit address.
Use the same frame format.
Keep the same minimum and maximum frame-lengths.
To support auto-negotiation as defined in Fast-Ethernet.

MAC Sublayer
Gigabit-Ethernet has two distinctive approaches for medium access: half-duplex and full-duplex.
Almost all implementations of Gigabit-Ethernet follow the full-duplex approach.
1) Full-Duplex Mode
There is a central switch connected to all computers or other switches.
Each switch has buffers for each input-port in which data are stored until they are transmitted.
There is no collision. This means that CSMA/CD is not used.
Lack of collision implies that
the maximum length of the cable is determined
→ by the signal attenuation in the cable &
→ not by the collision-detection process.
2) Half-Duplex Mode
• A switch is replaced by a hub, which acts as the common cable in which a collision might occur.
• CSMA/CD is used.
• The maximum length of the network is totally dependent on the minimum frame size.
• Three methods have been defined: traditional, carder extension, and frame bursting. i)
Traditional
Like traditional Ethernet, the minimum length of a frame is 512 bits.
However, because the length of a bit is 1/100 shorter,
Slot time is 512 bits x 1/1000 gs which is equal to 0.512 gs.
The reduced slot time means that collision is detected 100 times earlier.
The maximum length of the network is 25 m.
This length may be suitable if all the stations are in one room. ii)
Carrier Extension
To allow for a longer network, we increase the minimum frame-length.
Minimum length of frame is 512 bytes (4096 bits). Thus, minimum length is 8 times longer.
A station adds extension bits (padding) to any frame that is less than 4096 bits.
The maximum length of the network is 200 m.
A length from the hub to the station is 100 m.
iii) Frame Bursting
Carrier extension is very inefficient if
we have a series of short frames to send
each frame carries redundant data.
To improve efficiency, frame bursting was proposed.
Instead of adding an extension to each frame, multiple frames are sent.
However, to make these multiple frames look like one frame, padding is added between the
frames. Thus, the channel is not idle.
Physical-layer
The physical-layer in Gigabit-Ethernet is more complicated than that in Standard or Fast-Ethernet.
Some of the features of this layer are as follows. 1) Topology 2) Implementation and 3) Encoding.
Topology
Gigabit-Ethernet is used to connect two or more stations together.
If there are only 2 stations, they can be connected in point-to-point.
If there are 3 or more stations, they can be connected in star topology with a hub at center.
Implementation
Gigabit-Ethernet can be classified as either a two-wire or a four-wire implementation (Table 13.3).
The 2-wire implementations use
Fiber-optic cable (1000Base-SX, short-wave, or 1000Base-LX, long-wave) or

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STP (1000Base-CX)
The 4-wire implementations use category 5 twisted-pair cable (1000Base-T).

Encoding
1) Two-wire Implementation
The NRZ encoding scheme is used for two-wire implementation (Figure 13.17a).
However, 8B/10B block-coding is used to provide bit synchronization.
This is because NRZ is not a self-synchronous line coding scheme.
8B/10B coding creates a data-rate of 1.25 Gbps.
One wire (fiber or STP) is used for sending.
Another wire is used for receiving.
2) Four-wire Implementation
In this, it is not possible to have 2 wires for input and 2 for output (Figure 13.17b).
This is „.‟ each wire would need to carry 500 Mbps, which exceeds the capacity for category 5 UTP.
As a solution, 4D-PAM5 encoding is used to reduce the bandwidth.
Thus, all four wires are involved in both input and output.
Each wire carries 250 Mbps, which is in the range for category 5 UTP cable.

Ten Gigabit Ethernet

IEEE created Ten-Gigabit-Ethernet under the name 802.3ae.
Goals of the Gigabit-Ethernet:
Upgrade the data-rate to 10 Gbps.
Make it compatible with Standard, Fast, and Gigabit-Ethernet.
Use the same 48-bit address.
Use the same frame format.
Keep the same minimum and maximum frame-lengths.
Allow the interconnection of existing LANs into a MAN or a WAN .
Make Ethernet compatible with technologies such as Frame Relay and ATM.
4.9.1 Implementation
Ten-Gigabit-Ethernet operates only in full duplex mode.
This means there is no need for contention; CSMA/CD is not used.
Four implementations are the most common (Table 13.4):
10GBase-SR
10GBase-LR

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10GBase-EW and
10GBase-X4

MODULE 4(CONT.): WIRELESS-LANS

Introduction of Wireless-LANs
4.10.1 Architectural Comparison
1) Medium
In a wired LAN, we use wires to connect hosts.
In a switched LAN, with a link-layer switch, the communication between the hosts is point-to-point and
full-duplex (bidirectional).
In a wireless LAN, the medium is air, the signal is generally broadcast.
When hosts in a wireless LAN communicate with each other, they are sharing the same medium (multiple
access).
2) Hosts
In a wired LAN, a host is always connected to its network at a point with a fixed link layer address related
to its network interface card (NIC).
Of course, a host can move from one point in the Internet to another point.
In this case, its link-layer address remains the same, but its network-layer address will change.
In a wireless LAN, a host is not physically connected to the network; it can move freely and can use the
services provided by the network.
Therefore, mobility in a wired network and wireless network are totally different issues.
3) Isolated LANs
A wired isolated LAN is a set of hosts connected via a link-layer switch (Figure 15.1).
A wireless isolated LAN, called an ad hoc network in wireless LAN terminology, is a set of hosts that
communicate freely with each other.
The concept of a link-layer switch does not exist in wireless LANs.

4) Connection to Other Networks

A wired LAN can be connected to another network or the Internet using a router.
A wireless LAN may be connected to a wired infrastructure network, to a wireless infrastructure network,
or to another wireless LAN (Figure 15.2).
In this case, the wireless LAN is referred to as an infrastructure network, and the connection to the wired
infrastructure, such as the Internet, is done via a device called an access point (AP).
An access point is gluing two different environments together: one wired and one wireless.

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Communication between the AP and the wireless host occurs in a wireless environment.
Communication between the AP and the infrastructure occurs in a wired environment.

Characteristics
1) Attenuation
• The strength of electromagnetic signals decreases rapidly because the signal disperses in all directions;
only a small portion of it reaches the receiver.
• The situation becomes worse with mobile senders that operate on batteries and normally have small
power supplies.
2) Interference
• Another issue is that a receiver may receive signals not only from the intended sender, but also from
other senders if they are using the same frequency band.
3) Multipath Propagation
• A receiver may receive more than one signal from the same sender because electromagnetic waves can be
reflected back from obstacles such as walls, the ground, or objects.
• The result is that the receiver receives some signals at different phases (because they travel different
paths). This makes the signal less recognizable.
4) Error
• Error detection is more serious issues in a wireless network than in a wired network.
If SNR is high, it means that the signal is stronger than the noise (unwanted signal), so we may be
able to convert the signal to actual data.
When SNR is low, it means that the signal is corrupted by the noise and the data cannot be
recovered.

Access Control
The CSMA/CD algorithm does not work in wireless LANs for three reasons:
To detect a collision, a host needs to send and receive at the same time which means the host needs
to work in a duplex mode. Wireless hosts do not have enough power to do so (the power is supplied
by batteries).
They can only send or receive at one time.
The distance between stations can be great.
Signal fading could prevent a station at one end from hearing a collision at other end.
Because of the hidden station problem, in which a station may not be aware of another station‟s
transmission due to some obstacles or range problems, collision may occur but not be detected.
Hidden station problem
Figure 15.3 shows an example of the hidden station problem.
Every station in transmission range of Station B can hear any signal transmitted by station
B.
Every station in transmission range of Station C can hear any signal transmitted by station
C.
Station C is outside the transmission range of B;
Likewise, station B is outside the transmission range of C.
¤ However, Station A is in the area covered by both B and C;
Therefore, Station A can hear any signal transmitted by B or C.

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4.13 IEEE 802.11

Architecture
• The standard defines 2 kinds of services: 1) Basic service set (BSS) and 2)
Extended service set (ESS).
BSS
• IEEE 802.11 defines the basic service set (BSS) as the building block of a wireless-LAN.
• A basic service set is made of (Figure 15.4):
→ stationary or mobile wireless stations and
→ optional central base station, known as the access point (AP).
• There are 2 types of architecture:
1) Ad hoc Architecture
The BSS without an AP is a stand-alone network and cannot send data to other BSSs.
Stations can form a network without the need of an AP.
Stations can locate one another and agree to be part of a BSS.
2) Infrastructure Network
A BSS with an AP is referred to as an infrastructure network.

ESS
The ESS is made up of 2 or more BSSs with APs (Figure 15.5).
The BSSs are connected through a distribution-system, which is usually a wired LAN.
The distribution-system connects the APs in the BSSs.
IEEE 802.11 does not restrict the distribution-system;
The distribution-system can be any IEEE LAN such as an Ethernet.
The ESS uses 2 types of stations:
Mobile stations are normal stations inside a BSS.
Stationary stations are AP stations that are part of a wired LAN.

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When BSSs are connected, the stations within reach of one another can communicate without the use of an AP.
However, communication between two stations in two different BSSs usually occurs via two APs.

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

MAC Sublayer
IEEE 802.11 defines 2 MAC sublayers:
Distributed coordination function (DCF) &
Point coordination function (PCF).
The figure 15.6 shows the relationship between
Two MAC sublayers
LLC sublayer &
Physical layer.

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DCF
One of the 2 protocols defined by IEEE at the MAC sublayer is called the distributed coordination function
(DCF).
DCF uses CSMA/CA as the access method.
Wireless-LANs cannot implement CSMA/CD for 3 reasons:
For collision-detection, a station must be able to send data & receive collision-signals at the same
time. This can mean costly stations and increased bandwidth requirements.
Collision may not be detected because of the hidden station problem.
The distance between stations can be great.
Signal fading could prevent a station at one end from hearing a collision at the other end.
• Process Flowchart: Figure 15.7 shows the process flowchart for CSMA/CA as used in wireless-LANs.

Before sending a frame, the source-station senses the medium by checking the energy-level at the
carrier-frequency.
The channel uses a persistence strategy with back-off until the channel is idle.
After the station is found to be idle,
the station waits for a period of time called the DIFS.
then the station sends a control frame called the RTS.
After receiving the RTS and waiting a period of time called the SIFS, the destination-station sends a
control frame, called the CTS, to the source-station.
CTS frame indicate that the destination-station is ready to receive data.
The source-station sends data after waiting an amount of time equal to SIFS.
The destination-station, after waiting an amount of time equal to SIFS, sends an acknowledgment to
show that the frame has been received.
Acknowledgment is needed in this protocol because the station does not have any means to
check for the successful arrival of its data at the destination.
On the other hand, the lack of collision in CSMA/CD is a kind of indication
to the source that data have arrived.
(DIFS distributed inter frame space
(RTS request to send
SIFS short inter frame space) CTS
clear to send)

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Network Allocation Vector

When a station sends an RTS frame, it includes the duration of time that it needs to occupy the channel
(NAV Network Allocation Vector).
The stations that are affected by this transmission create a timer called a NAV.
NAV shows how much time must pass before these stations are allowed to check the channel for
idleness.
Each time a station accesses the system and sends an RTS frame, other stations start their NAV.
In other words, each station, before sensing the medium to see if it is idle, first checks its NAV to see
if it has expired.

Collision During Handshaking

Two or more stations may try to send RTS frames at the same time.
These control frames may collide.
However, because there is no mechanism for collision-detection, the sender assumes there has been a
collision if it has not received a CTS frame from the receiver.
The back-off strategy is employed, and the sender tries again.
PCF
The PCF is an optional access method that can be implemented in an infrastructure network (not in an
ad hoc network) (PCF Point Coordination Function).
The PCF is implemented on top of the DCF.
The PCF is used mostly for time-sensitive transmission.
PCF has a centralized, contention-free polling access method.
The AP performs polling for stations that are capable of being polled.
The stations are polled one after another, sending any data they have to the AP.
To give priority to PCF over DCF, another set of inter-frame spaces has been defined: PIFS and SIFS.
The SIFS is the same as that in DCF &
PIFS (PCF IFS) is shorter than the DIFS.
This means that if, at the same time, a station wants to use only DCF and an AP wants to use PCF, the
AP has priority.
Due to the priority of PCF over DCF, stations that only use DCF may not gain access to the medium.
To prevent this, a repetition interval has been designed to cover both contention-free (PCF) and
contention-based (DCF) traffic.
The repetition interval, which is repeated continuously, starts with a special control frame, called a
beacon frame.
When the stations hear the beacon frame, they start their NAV for the duration of the contention-free
period of the repetition interval.

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During the repetition interval, the PC (point controller) can send a poll frame, receive data, send an
ACK, receive an ACK, or do any combination of these (802.11 uses piggybacking).
At the end of the contention-free period, the PC sends a CF end (contention-free end) frame to allow
the contention-based stations to use the medium.

Fragmentation
The wireless environment is very noisy; a corrupt frame has to be retransmitted.
The protocol, therefore, recommends fragmentation--the division of a large frame into smaller ones. It
is more efficient to resend a small frame than a large one.
Frame Format

• The MAC layer frame consists of nine fields (Figure 15.9):

1) Frame control (FC)
The FC field is 2 bytes long and defines the type of frame and some control information. The
table describes the subfields.

2) D
In all frame types except one, this field defines the duration of the transmission that is used to
set the value of NAV.
In one control frame, this field defines the ID of the frame.
3) Addresses
There are four address fields, each 6 bytes long.
The meaning of each address field depends on the value of the ToDS and FromDS subfields.
4) Sequence control
This field defines the sequence number of the frame to be used in flow control.
5) Frame body
This field contains information based on the type and the subtype defined in the FC
field. This field can be between 0 and 2312 bytes,
6) FCS
The FCS contains a CRC-32 error detection sequence.

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Frame Types
• A wireless-LAN defined by IEEE 802.11 has three categories of frames: 1.management frames,
2.control frames, and 3.data-frames.
1) Management Frames
Management frames are used for the initial communication between stations and access points.
2) Control Frames
Control frames are used for accessing the channel and acknowledging frames (Figure 15.10).

For control frames the value of the type field is 01; the values of the subtype fields for frames
are shown in the table 14.2.

3) Data-frames
Data-frames are used for carrying data and control information.
Addressing Mechanism
The IEEE 802.11 addressing mechanism specifies 4 cases, defined by the value of the 2 flags in the FC
field, To DS and From DS.
Each flag can be either 0 or 1, resulting in 4 different situations.
The interpretation of the 4 addresses (address 1 to address 4) in the MAC frame depends on the value
of these flags, as shown in the Table 15.3.

Address 1 is always the address of the next device.

Address 2 is always the address of the previous device.
Address 3 is the address of the final destination-station if it is not defined by address 1.
Address 4 is the address of the original source-station if it is not the same as address 2.
Case-1:00
In this case, To DS = 0 and From DS = 0 (Figure 15.11a).
This means that the frame is
not going to a distribution-system (To DS = 0) and i
not coming from a distribution-system (From DS = 0).
The frame is going from one station in a BSS to another without passing through the
distribution-system.
The ACK frame should be sent to the original sender.

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Case-2:01
In this case, To DS = 0 and From DS = 1 (Figure 15.11b).
This means that the frame is coming from a distribution-system (From DS = 1).
The frame is coming from an AP and going to a station.
The ACK should be sent to the AP.
The address 3 contains the original sender of the frame (in another BSS).
Case-3:10
In this case, To DS = 1 and From DS = 0 (Figure 15.11c).
This means that the frame is going to a distribution-system (To DS = 1).
The frame is going from a station to an AP. The ACK is sent to the original station.
The address 3 contains the final destination of the frame (in another BSS).
Case-4:11
In this case, To DS = 1 and From DS = 1 (Figure 15.11d).
This is the case in which the distribution-system is also wireless.
The frame is going from one AP to another AP in a wireless distribution-system.
We do not need to define addresses if the distribution-system is a wired LAN because the
frame in these cases has the format of a wired LAN frame (for example: Ethernet,).
Here, we need four addresses to define
original sender
final destination, and
two intermediate APs.

Exposed Station Problem

In this problem, a station refrains from using a channel even when the channel is available for use.
In the figure 14.12, station A is transmitting to station B.
Station C has some data to send to station D, which can be sent without interfering with the
transmission from A to B.
However, station C is exposed to transmission from A i.e. station C hears what A is sending and thus
refrains from sending.
In other words, C is too conservative and wastes the capacity of the channel.

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The handshaking messages RTS and CTS cannot help in this case.
Station C hears the RTS from A, but does not hear the CTS from B.
Station C, after hearing the RTS from A, can wait for a time so that the CTS from B reaches A; it then
sends an RTS to D to show that it needs to communicate with D.
Both stations B and A may hear this RTS, but station A is in the sending state, not the receiving state.
However, Station B responds with a CTS.
The problem is here (Figure 15.12).
If station A has started sending its data, station C cannot hear the CTS from station D because
of the collision; it cannot send its data to D. It remains exposed until A finishes sending its data
as the figure shows.

Physical Layer

All implementations, except the infrared, operate in the ISM band.

ISM band defines 3 unlicensed bands in the 3 ranges.
902-928 MHz,
2.400--4.835 GHz, and
5.725-5.850 GHz. (ISM industrial, scientific, and medical)
IEEE 802.11 FHSS
IEEE 802.11 FHSS uses the FHSS method (Figure 15.13).
FHSS uses the 2.4-GHz ISM band.
The band is divided into 79 subbands of 1 MHz (and some guard bands).
A pseudorandom number generator selects the hopping sequence.
The modulation technique is either two-level FSK or four-level FSK with 1 or 2 bits/baud.
This results in a data-rate of 1 or 2 Mbps
(FHSS frequency-hopping spread spectrum DSSS direct sequence spread spectrum)

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IEEE 802.11 DSSS

IEEE 802.11 DSSS uses the DSSS method (Figure 15.14).
DSSS uses the 2.4-GHz ISM band.
The modulation technique in this specification is PSK at 1 Mbaud/s.
The system allows 1 or 2 bits/baud (BPSK or QPSK).
This results in a data-rate of 1 or 2 Mbps.

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HRDSSS high-rate direct sequence spread spectrum (OFDM orthogonal frequency-division

multiplexing)
CCK complementary code keying)
IEEE 802.11 Infrared
IEEE 802.11 infrared uses infrared light in the range of 800 to 950 nm (Figure 15.15).
The modulation technique is called pulse position modulation (PPM).
For a 1-Mbps data-rate, a 4-bit sequence is first mapped into a 16-bit sequence in which only one bit is set
to 1 and the rest are set to 0.
For a 2-Mbps data-rate, a 2-bit sequence is first mapped into a 4-bit sequence in which only one bit is set to
1 and the rest are set to 0.
The mapped sequences are then converted to optical signals; the presence of light specifies 1, the absence
of light specifies 0.

IEEE 802.11a OFDM

IEEE 802.11a OFDM describes the OFDM method for signal generation in a 5-GHz ISM band.
OFDM is similar to FDM, with 2 major difference:
All the subbands are used by one source at a given time.
Sources contend with one another at the data-link-layer for access.
The band is divided into 52 subbands. Out of which,
48 subbands are used for sending 48 groups of bits at a time.
4 subbands are used for sending control information.
The scheme is similar to ADSL.
Dividing the band into subbands diminishes the effects of interference.
If the subbands are used randomly, security can also be increased.
OFDM uses PSK and QAM for modulation.
The common data-rates are
i) 18 Mbps (PSK) and ii) 54 Mbps (QAM).
IEEE 802.11b DSSS
IEEE 802.11 b DSSS describes the HRDSSS method for signal generation in the 2.4-GHz ISM band.
HR-DSSS is similar to DSSS, with 1 major difference: HR-DSSS uses encoding method called CCK.
CCK encodes 4 or 8 bits to one CCK symbol (Figure 15.16).
To be backward compatible with DSSS, HR-DSSS defines 4 data-rates: 1, 2, 5.5, and 11 Mbps.
The first two versions (1- & 2-Mbps) use the same modulation techniques as DSSS.
The 5.5-Mbps version
uses BPSK and
transmits at 1.375 Mbaud/s with 4-bit CCK encoding.
The 11-Mbps version
uses QPSK and
transmits at 1.375 Mbps with 8-bit CCK encoding.

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4.11.4.6 IEEE 802.11g

This new specification defines forward error correction and OFDM using the 2.4-GHz ISM band.
The modulation technique achieves a 22- or 54-Mbps data-rate.
It is backward compatible with 802.11b, but the modulation technique is OFDM.

Bluetooth
Bluetooth is a wireless-LAN technology designed to connect devices of different functions such as
telephones, notebooks, computers, cameras, printers, coffee makers, and so on.
A Bluetooth LAN is an ad hoc network. This means the network is formed spontaneously.
The devices
find each other and
make a network called a piconet (Usually, devices are called gadgets)
A Bluetooth LAN can even be connected to the Internet if one of the devices has this capability.
By nature, a Bluetooth LAN cannot be large.
If there are many devices that try to connect, there is confusion.
Bluetooth technology has several applications.
Peripheral devices such as a wireless mouse/keyboard can communicate with the computer.
In a small health care center, monitoring-devices can communicate with sensor-devices.
Home security devices can connect different sensors to the main security controller.
Conference attendees can synchronize their laptop computers at a conference.
Today, Bluetooth technology is the implementation of a protocol defined by the IEEE 802.15 standard.
The standard defines a wireless PAN operable in an area the size of a room or a hall.
(PAN personal-area network)
Architecture
• Bluetooth defines 2 types of networks: 1) Piconet and 2) Scatternet.
Piconets
A Bluetooth network is called a piconet, or a small net. (Figure 15.17).
A piconet can have up to 8 stations. Out of which
One of station is called the primary.
The remaining stations are called secondaries.
All the secondary-stations synchronize their clocks and hopping sequence with the primary station.
A piconet can have only one primary station.
The communication between the primary and the secondary can be one-to-one or one-to-many.

Although a piconet can have a maximum of 7 secondaries, an additional 8 secondaries can be in the parked
state.
A secondary in a parked state is synchronized with the primary, but cannot take part in
communication until it is moved from the parked state.
Because only 8 stations can be active in a piconet, activating a station from the parked state means that an
active station must go to the parked state.
Scatternet
Piconets can be combined to form a scatternet (Figure 15.18).
A station can be a member of 2 piconets.
A secondary station in one piconet can be the primary in another piconet. This is called mediator station.
Acting as a secondary, mediator station can receive messages from the primary in the first piconet.
Acting as a primary, mediator station can deliver the message to secondaries in the second piconet.
Bluetooth Devices

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A Bluetooth device has a built-in short-range radio transmitter.

The current data-rate is 1 Mbps with a 2.4-GHz bandwidth.
This means that there is a possibility of interference between the IEEE 802.11b wireless-LANs and
Bluetooth LANs.

Bluetooth Layers
• Bluetooth uses several layers that do not exactly match those of the Internet model (Figure 15.19).

Radio Layer
• The radio layer is roughly equivalent to the physical layer of the Internet model.
• Bluetooth devices are low-power and have a range of 10 m.
1) Band
Bluetooth uses a 2.4-GHz ISM band divided into 79 channels of 1 MHz each.
2) FHSS
Bluetooth uses the frequency-hopping spread spectrum (FHSS) method in the physical layer to
avoid interference from other devices or other networks.
Bluetooth hops 1600 times per second, which means that each device changes its modulation
frequency 1600 times per second.
A device uses a frequency for only 625 µs (1/1600 s) before it hops to another frequency; the dwell
time is 625 µs.
3) Modulation
To transform bits to a signal, Bluetooth uses a sophisticated version of FSK, called GFSK (FSK
with Gaussian bandwidth filtering).
GFSK has a carrier frequency.
Bit 1 is represented by a frequency deviation above the carrier; bit „a‟ is represented by a frequency
deviation below the carrier.
The frequencies, in megahertz, are defined according to the following formula for each channel:
For example,
The first channel uses carrier frequency 2402 MHz (2.402 GHz).
The second channel uses carrier frequency 2403 MHz (2.403 GHz).
Baseband Layer
The baseband layer is roughly equivalent to the MAC sublayer in LANs.
The access method is TDMA.
The primary and secondary communicate with each other using time slots.
The length of a time slot is exactly the same as the dwell time, 625 μs.
This means that during the time that one frequency is used, a sender sends a frame to a secondary, or a
secondary sends a frame to the primary.
The communication is only between the primary and a secondary; secondaries cannot communicate
directly with one another.

TDMA
Bluetooth uses a form of TDMA that is called TDD-TDMA (timedivision duplex TDMA).
TDD-TDMA is a kind of half-duplex communication in which the secondary and receiver send and receive
data, but not at the same time (halfduplex);
However, the communication for each direction uses different hops.
This is similar to walkie-talkies using different carrier frequencies.
Single-Secondary Communication
If the piconet has only one secondary, the TDMA operation is very simple (Fig 15.21).

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The time is divided into slots of 625 μs.

The primary uses even numbered slots (0, 2, 4, ...); the secondary uses odd-numbered slots
(1, 3, 5,...).
TDD-TDMA allows the primary and the secondary to communicate in half-duplex mode.
In slot 0, the primary sends, and the secondary receives; in slot 1, the secondary sends, and
the primary receives. The cycle is repeated.

Links
Two types of links can be created between a primary and a secondary:
SCQ link (Synchronous Connection-oriented Link) and
ACL links (Asynchronous Connectionless Link).
1) SCA
This link is used when avoiding latency is more important than data-integrity.
(Latency delay in data delivery Integrity error-free delivery)
A physical-link is created between the primary and a secondary by reserving specific slots at regular
intervals.
The basic unit of connection is 2 slots. One slot is used for each direction.
If a packet is damaged, it is never retransmitted.
Application: Used for real-time audio where avoiding delay is all-important.
A secondary
can create up to 3 SCQ links with the primary
can send digitized audio (PCM) at 64 kbps in each link.
2) ACL
This link is used when data-integrity is more important than avoiding latency.
If a payload encapsulated in the frame is corrupted, it is retransmitted.
A secondary returns an ACL frame in the available odd-numbered slot if and only if the previous
slot has been addressed to it.
ACL can use one, three, or more slots and can achieve a maximum data-rate of 721 kbps.

Frame Types
• A frame in the baseband layer can be one of 3 types: 1) one-slot 2) three-slot or 3) five-slot.
1) One-slot frame
A slot is 625 μs.
However, in a one-slot frame exchange, 259 μs is needed for hopping & control mechanisms.
This means that a one-slot frame can last only 625 - 259, or 366 μs.
With a 1-MHz bandwidth and 1 bit/Hz, the size of a one-slot frame is 366 bits.
2) Three-slot frame
A three-slot frame occupies 3 slots.
However, since 259 μs is used for hopping, the length of the frame is 3 x 625 - 259 = 1616
μs or 1616 bits.
A device that uses a three-slot frame remains at the same hop (at the same carrier frequency) for 3
slots.
Even though only once hop number is used, 3 hop numbers are consumed.
That means the hop number for each frame is equal to the first slot of the frame.
3) Five-slot frame
A five-slot frame also uses 259 bits for hopping, which means that the length of the frame is 5 x
625 - 259 =2866 bits.

Frame Format

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• The following describes each field (Figure 15.23):

1) Access code
This field contains synchronization bits and the identifier of the primary to distinguish the frame
of one piconet from anothe
2) Header
This field is a repeated I8-bit pattern. Each pattern has the following subfields: i)
Address
This subfield can define up to 7 secondaries (1 to 7).
If the address is zero, it is used for broadcast communication from the primary to all
secondaries.
ii) Type
This subfield defines the type of data coming from the upper layers.
iii) F
This subfield is for flow control.
When set (1), it indicates that the device is unable to receive more frames (buffer is full).
iv) A
This subfield is for acknowledgment.
Bluetooth uses Stop-and-Wait ARQ.
1 bit is sufficient for acknowledgment.
v) S
This subfield holds a sequence number.
Bluetooth uses Stop-and-Wait ARQ
1 bit is sufficient for sequence numbering.
vi) HEC (header error correction)
This subfield is a checksum to detect errors in each 18-bit header section.
The header has three identical 18-bit sections.
The receiver compares these three sections, bit by bit.
If each of the corresponding bits is the same, the bit is accepted; if not, the majority opinion rules.
This is a form of forward error correction (for the header only).
This double error control is needed because the nature of the communication, via air, is very noisy.
There is no retransmission in this sublayer.
3) Payload
This subfield can be 0 to 2740 bits long.
It contains data or control information coming from the upper layers.
L2CAP
The L2CAP is roughly equivalent to the LLC sublayer in LANs (Figure 15.20).
It is used for data exchange on an ACL link. (L2CAP Logical Link Control and Adaptation Protocol)
SCQ channels do not use L2CAP”(Figure 14.25)

The following describes each field:

Length
This field defines the size of the data, in bytes, coming from the upper layers.
Data can be up to 65,535 bytes.

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2) CID (Channel ID)

This field defines a unique identifier for the virtual channel created at this level.
The L2CAP has specific duties: 1) Multiplexing
Segmentation and reassembly
QoS (quality of service) and
Group management.
1) Multiplexing
The L2CAP can do multiplexing.
At the sender site, L2CAP
accepts data from one of the upper-layer protocols
frames the data and
delivers the data to the baseband layer.
At the receiver site, L2CAP
accepts a frame from the baseband layer
extracts the data, and
delivers the data to the appropriate protocol layer.
It creates a kind of virtual channel.
2) Segmentation and Reassembly
In the baseband layer, the maximum size of the payload field is 2774 bits, or 343 bytes.
This includes 4 bytes to define the packet and packet-length.
Therefore, the size of the packet that can arrive from an upper layer can only be 339 bytes.
However, application layers sometimes need to send a data packet that can be up to 65,535 bytes
(for example: an Internet packet).
The L2CAP
divides the large packets into segments and
adds extra information to define the location of the segments in the original packet.
The L2CAP segments the packet at the source and reassembles them at the destination.
3) QoS
Bluetooth allows the stations to define a QoS level.
If no QoS level is defined, Bluetooth defaults to best-effort service; it will do its best under the
circumstances.
4) Group Management
Another functionality of L2CAP is to allow devices to create a type of logical addressing
between themselves.
This is similar to multicasting.
For example:
2 or 3 secondary devices can be part of a multicast group to receive data from the primary.

Assignment Questions
1. Explain random access protocol. (4)
2. Explain pure ALOHA. (6*)
3. Explain slotted ALOHA. (4*)
4. Explain CSMA. (6*)
5. Explain different persistence methods of CSMA. (6*)
6. Explain CSMA/CA. (6*)
7. Explain CSMA/CD. (10*)
8. List & explain different controlled access protocols. (10*)
9. Explain reservation access method. (4*)
10. Explain polling access method. (6*)
11. Explain token passing access method. (6*)
12. List & explain channelization protocols. (10*)
13. Explain FDMA. (6*)
14. Explain TDMA. (6*)
15. Explain CDMA. (8*)
16. Explain frame format of standard ethernet. (8*)
17. Explain frame length of standard ethernet. (4*)
18. Explain addressing in standard ethernet. (6*)
19. Explain encoding in standard ethernet. (8)

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20. Explain changes in the standard ethernet. (6)

21. List out 5 goals of fast-ethernet. Explain autonegotiation. (6*)
22. Explain encoding in fast-ethernet. (8)
23. Explain MAC Sublayer in gigabit-ethernet (6*)
24. Explain encoding in gigabit-ethernet. (4)
25. Differentiate b/w wireless LAN & wired LAN with reference to architectural comparison (6)
26. Explain characteristics of wireless medium. (4)
27. Explain hidden station problem. (6*)
28. Explain architecture of IEEE 802.11 (10*)
29. Explain DCF in IEEE 802.11 (6)
30. Explain PCF in IEEE 802.11 (6)
31. Explain frame format of IEEE 802.11 (8*)
32. Explain frame types of IEEE 802.11 (8*)
33. Explain addressing in IEEE 802.11 (8*)
34. Explain exposed station problem. (6*)
35. Explain physical Layer of IEEE 802.11 (8)
36. Explain architecture of Bluetooth. (6*)
37. Explain layers of Bluetooth. (8*)
38. Explain radio Layer in Bluetooth. (6)
39. Explain baseband Layer in Bluetooth. (6)
40. Explain 2 types of links in Bluetooth. (6)
41. Explain frame format of Bluetooth. (6)
42. Explain L2CAP in Bluetooth. (6)

Expected Outcomes
• Learn about random access protocol
• Explain controlled access protocols
• Learn about channelization protocols
• Learn about Standard Ethernet, Fast Ethernet , Gigabit Ethernet
• Explain about IEEE 802.11, Bluetooth

Further Reading
1. http://www.mhhe.com/engcs/compsci/forouzan/frontmatter.pdf
2. http://freecomputerbooks.com/Introduction-to-Data-Communications.html

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MODULE 5: TABLE OF CONTENTS

WiMAX
Services
IEEE Project 802.16
Layers in Project 802.16
Data Link layer
Physical layer
MAC Sublayer
CELLULAR TELEPHONY
Operation
Frequency-Reuse Principle
Transmitting
Receiving
Handoff
Roaming
First Generation (1G)
AMPS
Second Generation (2G)
D-AMPS
GSM
IS-95
Third Generation (3G)
IMT-2000 Radio Interfaces
Fourth Generation (4G)
SATELLITE NETWORKS
General Issues for Operation of Satellites
GEO Satellites
MEO Satellites
Global Positioning System
LEO Satellites
Network Layer Protocols
INTERNET PROTOCOL (IP)
Internet Protocol (IP)
Datagram Format
Fragmentation
Maximum Transfer Unit (MTU)
Fields Related to Fragmentation
Options
Security of IPv4 Datagrams
IPSec
ICMPv4
MESSAGES
Error Reporting Messages
Query Messages
Debugging Tools
Ping
Traceroute or Tracert
MOBILE IP
Addressing
Stationary Hosts
Mobile Hosts
Agents

5-1

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Three Phases
Agent Discovery
Registration
Request and Reply
Data Transfer
Inefficiency in Mobile IP
Double Crossing
Triangle Routing
IPv6 ADDRESSING
Representation
Address-space
Three Address Types
Address-space Allocation
Global Unicast Addresses
Special Addresses
Other Assigned Blocks
Autoconfiguration
THE IPv6 PROTOCOL
Changes from IPv4 to IPv6 (Advantages of IPv6)
Packet Format
Concept of Flow and Priority in IPv6
Fragmentation and Reassembly
Extension Header
Comparison of Options between IPv4 and IPv6
THE ICMPv6 PROTOCOL
Error Reporting Messages
Informational Messages
Neighbor Discovery Messages
Group Membership Messages
TRANSITION FROM IPv4 TO IPv6
Strategies

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MODULE 5
OTHER WIRELESS NETWORKS
STRUCTURE

Introduction
Objectives
WiMAX
Cellular Telephony
Satellite Networks
Network Layer Protocols
Internet Protocol (IP)
ICMP
Mobile IP
IPv6 Addressing
The IPv6 Protocol
The ICMPv6 Protocol
Transition from IPv4 to IPv6
Assignment Questions
Expected Outcomes
Further Reading

Introduction
This module provides an overview of the wireless networks, protocols at the network layer such as
Internet Protocol, ICMP and mobile protocols.

Objectives
• Understand Services of WiMAX
• Learn about 1G, 2G, 3G generations of cellular telephony
• Understand ICMP
• Understand the IPv4 and IPv6, also their comparisons

WiMAX
WiMAX stands for Worldwide Interoperability for Microwave Access.
Purpose of WiMAX:
People want to have access to the Internet from home or office (fixed) where the wired access to the
Internet is either not available or is expensive.
People need to access the Internet when they are using their cellular phones (mobiles).
WiMAX provides the “last mile” broadband wireless access.

5.3.1 Services
WiMAX provides 2 types of services to subscribers: 1) Fixed and 2) Mobile.
1) Fixed WiMAX
A base-station can use 3 different types of antenna to optimize the performance:
Omni-directional 2) Sector or 3) Panel (Figure 16.1).
WiMAX uses a beam-steering AAS (Adaptive Antenna System).
While transmitting, antenna can focus its energy in the direction of the subscriber-station.
While receiving, antenna can focus in the direction of the subscriber-station to receive maximum
energy sent by the subscriber.

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2) Mobile WiMAX
• The subscribers are mobile-stations that move from one place to another (Figure 16.2).

IEEE Project 802.16

WiMAX is the result of the IEEE 802.16 project.
The standard is also referred to as wireless local loop.

802.11 Projects 802.16 Projects

Standard for a wireless LAN Standard for a wireless WAN
Defines a connectionless communication Defines a connection-oriented service
Distance b/w base-station & host is very limited Distance b/w base-station & host is above 10 km

IEEE 802.16 was revised into 2 new standards:

IEEE 802.16d which concentrates on the fixed WiMAX.
IEEE 802.16e which defines the mobile WiMAX.

Layers in Project 802.16

Here we discuss, following 2 layers (Figure 16.3):
Data-link layer &
Physical layer.
The data-link layer is divided into 3 sublayers:
Service Specific Convergence Sublayer
Security Sublayer &
MAC Sublayer.
The physical layer is divided into 2 sublayers:
Transmission Convergence Sublayer &
Physical Medium Dependent Sublayer.

Data Link layer

1) Service Specific Convergence Sublayer
This is actually the DLC sublayer revised for broadband wireless communication.
It is devised for a connection-oriented service where each connection may benefit from a specific QoS
2) Security Sublayer
This sublayer provides security for communication using WiMAX.
The nature of wireless communication requires security.
Security provided using encryption for information exchanged b/w subscriber-station & base-station.

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3) MAC Sublayer
The MAC sublayer defines the access method and the format of the frame.
This sublayer is designed for connection-oriented service.
The packets are routed from the base-station to the subscriber-station using a connection identifier.
Connection identifier remains same during the duration of the communication.

Physical Layer
1) Transmission Convergence Sublayer
This sublayer uses TDD.
TDD a variation of TDM designed for duplex (bidirectional) communication.
Each frame is made of 2 subframes (Figure 16.5):
Downstream Subframes: carry data from the base-station to the subscribers.
Upstream Subframes: carry data from the subscribers to the base-station.
Each subframe is divided into slots.

2) Physical Medium Dependent Sublayer

This sublayer is in continuous revision.
Originally, 802.16 defined the band 10-66 GHz.
defined following modulations:
QPSK used for long-distance communication.
QAM-16 used for medium-distance communication.
QAM-64 used for short-distance communication.
Later, IEEE defined 802.16d (fixed WiMAX), which added the band 2-11 GHz (compatible with
wireless LANs) using the OFDM.
Sometime later, IEEE defined 802.16e (mobile WiMAX) and added SOFDM.
(TDD Time-Division Duplex FEC forward error correction)
(OFDM Orthogonal Frequency-Division Multiplexing)
(SOFDM scalable orthogonal frequency division multiplexing)

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MAC Sublayer
The MAC sublayer defines the access method and the format of the frame.
This sublayer is designed for connection-oriented service.
The packets are routed from the base-station to the subscriber-station using a connection identifier.
Connection identifier remains same during the duration of the communication.
Here we discuss, following issues: 1) Access Method 2) Frame Format 3) Addressing
1) Access Method
WiMAX uses the reservation (scheduling) access method.
Base-station needs to make a slot-reservation before sending a data to a subscriber-station
Each subscriber-station needs to make a reservation before sending a data to the base-station
2) Frame Format
Two types of frames (Figure 16.4):
Generic Frame is used to send and receive payload.
Control Frame is used only during the connection establishment.
Both frame-types use a 6-byte generic header.
However, some bytes have different interpretations in different frame types.

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• The frame contains following fields:

1) First bit
The first bit in a frame is the frame identifier.
If first bit = 0, the frame is a generic frame.
If first bit = 1, the frame is a control frame.
2) EC (Encryption Control)
This field uses one bit to define whether the frame should be encrypted for security purpose.
If EC = 0, it means no encryption.
If EC = 1, it means the frame needs to be encrypted at the security sublayer.
3) Type
This field is used to define the type of the payload.
This field is only present in the generic frame.
The payload can be a packed-load or a fragmented-load.
4) CI (Checksum ID)
This field defines whether the checksum field should be present or not.
If the payload is multimedia, FEC is applied to the frame and there is no need for checksum.
5) EK (Encryption Key)
This field defines one of the 4 keys for encryption if encryption is required.
6) Length
This field defines the total length of the frame.
This field is only present in the generic frame.
This field is replaced by the bytes needed field in the control frame.
7) Bytes Needed
This field defines the number of bytes needed for allocated slots in the physical layer.
8) Connection ID
This field defines the connection identifier for the current connection.
9) Header CRC
Both types of frames need to have header CRC field.
Header CRC is used to check whether the header itself is corrupted.
This field uses the polynomial (x8 + x2 + x + 1) as the divisor.
10) Payload
This field defines the payload.
Payload is encapsulated in the frame from the service specific convergence sublayer.
This field is not needed in the control frame.
11) CRC
This field is used for error detection over the whole frame.

Addressing
Each subscriber and base-station typically has a 48-bit MAC address.
However, there is no source or destination address field.

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The reason is that the combination of source and destination addresses are mapped to a VCI during the
connection-establishing phase.
This protocol is a connection-oriented protocol that uses a VCI (Virtual Connection Identifier).
Then, each frame uses the same connection identifier for the duration of data transfer

Cellular Telephony
Cellular telephony is designed to provide communications
between two moving units called mobile-stations (MSs) or
between one mobile-station and one stationary unit called a land unit (Figure 16.6).
A service-provider is responsible for
locating & tracking a caller
assigning a channel to the call and
transferring the channel from base-station to base-station as the caller moves out-of-range.
Each cellular service-area is divided into small regions called cells.
Each cell contains an antenna.
Each cell is controlled by AC powered network-station called the base-station (BS).
Each base-station is controlled by a switching office called a mobile-switching-center (MSC).
MSC coordinates communication between all the base-stations and the telephone central office.
MSC is a computerized center that is responsible for
connecting calls
recording call information and
billing.
Cell-size is not fixed; Cell-size can be increased or decreased depending on population of the area.
Cell-radius = 1 to 12 mi.
Compared to low-density areas, high-density areas require many smaller cells to meet traffic demands.
Cell-size is optimized to prevent the interference of adjacent cell-signals.

Operation
Frequency-Reuse Principle
In general, neighboring-cells cannot use the same set of frequencies for communication.
Using same set of frequencies may create interference for the users located near the cell-boundaries.
However,
set of frequencies available is limited and
frequencies need to be reused.
A frequency reuse pattern is a configuration of N cells. Where N = reuse factor
Each cell uses a unique set of frequencies.
When the pattern is repeated, the frequencies can be reused.
There are several different patterns (Figure 16.7).
The numbers in the cells define the pattern.
The cells with the same number in a pattern can use the same set of frequencies. These cells are called the
reusing cells.

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Transmitting
Procedure to place a call from a mobile-station:
The caller
enters a phone number and
presses the send button.
The mobile-station
scans the band to determine setup channel with a strong signal and
sends the data (phone number) to the closest base-station.
The base-station sends the data to the MSC.
The MSC sends the data on to the telephone central office.
If called party is available, a connection is made and the result is relayed back to the MSC.
The MSC assigns an unused voice channel to the call, and a connection is established.
The mobile-station automatically adjusts its tuning to the new channel.
Finally, voice communication can begin.

Receiving
Procedure to receive a call from a mobile-station:
When a mobile phone is called, the telephone central office sends phone number to the MSC.
MSC searches for the location of the mobile-station by sending query-signals to each cell in a process.
This is called paging.
When the mobile-station is found, the MSC transmits a ringing signal.
When the mobile-station answers, the MSC assigns a voice channel to the call.
Finally, voice communication can begin.
Handoff
During a conversation, the mobile-station may move from one cell to another.
Problem: When the mobile-station goes to cell-boundary, the signal becomes weak.
To solve this problem, the MSC monitors the level of the signal every few seconds.
If signal-strength decreases, MSC determines a new cell to accommodate the communication.
Then, MSC changes the channel carrying the call (hands signal off from old channel to a new one).
• Two types of Handoff: 1) Hard Handoff 2) Soft Handoff
1) Hard Handoff
Early systems used a hard handoff.
A mobile-station only communicates with one base-station.
When the MS moves from one cell to another cell,
Firstly, communication must be broken with the old base-station.
Then, communication can be established with the new base-station.
This may create a rough transition.
2) Soft Handoff
New systems use a soft handoff.
A mobile-station can communicate with two base-stations at the same time.
When the MS moves from one cell to another cell,
Firstly, communication must be broken with the old base-station.

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Then, the same communication may continue with the new base-station.

Roaming
Roaming means that the user
can have access to communication or
can be reached where there is coverage.
Usually, a service-provider has limited coverage.
Neighboring service-providers can provide extended coverage through a roaming contract.
First Generation (1G)
The first generation was designed for voice communication using analog signals.
The main system evolved in the first generation: AMPS (Advanced Mobile Phone System).
AMPS
This system is a 1G analog cellular system.
The system uses FDMA to separate channels in a link.
• Here we discuss, two issues: 1) Bands 2) Transmission
1) Bands
The system operates in the ISM 800-MHz band.
The system uses 2 separate channels (Figure 16.8):
First channel is used for forward communication (base-station to mobile-station)
Band range: 869 to 894 MHz
Second channel is used for reverse communication (mobile-station to base-station).
Band range: 824 to 849 MHz

2) Transmission
The system uses FM and FSK for modulation (Figure 16.9).
Voice channels are modulated using FM.
Control channels are modulated using FSK to create 30-kHz analog signals.
The system uses FDMA to divide each 25-MHz band into 30-kHz channels.

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Second Generation (2G)

The second generation was designed for higher-quality voice communication using digital signals.
1G vs. 2G:
The first generation was designed for analog voice communication.
The second generation was mainly designed for digital voice communication.
Three major systems evolved in the second generation:
D-AMPS (digital AMPS)
GSM (Global System for Mobile communication) and
IS-95 (Interim Standard).

D-AMPS
MPS (Digital AMPS) was improved version of analog
AMPS. D-AMPS was backward-compatible with AMPS.
Thus, in a cell,
First telephone may use AMPS and
Second telephone may use D-AMPS.
• Here we discuss, two issues: 1) Bands 2) Transmission
1) Band
The system uses the same bands and channels as AMPS (Figure 16.10).
2) Transmission
Each voice channel is digitized using a very complex PCM and compression technique.

GSM
• Aim of GSM: to replace a number of incompatible 1G technologies.
• Here we discuss, two issues: 1) Bands 2) Transmission
1) Bands
The system uses two bands for duplex communication (Figure 16.11).
Each band is 25 MHz in width.
Each band is divided into 124 channels of 200 kHz.

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2) Transmission
Each voice channel is digitized and compressed to a 13-kbps digital signal (Figure 16.12).
Each slot carries 156.25 bits.
Eight slots share a frame (TDMA).
26 frames also share a multiframe (TDMA).
We can calculate the bit rate of each channel as follows.

Each 270.8-kbps digital channel modulates a carrier using GMSK (a form of FSK); the result is a
200-kHz analog signal.
Finally, 124 analog channels of 200 kHz are combined using FDMA. The result is a 25-MHz
band (Figure 16.13).

5-13

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5.2.3.3 IS-95
• The system is based on CDMA and DSSS.
• Here we discuss, following 6 issues: 1) Bands 2) Transmission 3) Synchronization
4) Two Data-rate Sets 5) Frequency-Reuse Factor 6) Soft Handoff
1) Bands
The system uses two bands for duplex communication.
The bands can be ISM 800-MHz band or ISM 1900-MHz band.
Each band is divided into 20 channels of 1.228 MHz.
Each service-provider is allotted 10 channels.
IS-95 can be used in parallel with AMPS.
Each IS-95 channel is equivalent to 41 AMPS channels (41 × 30 kHz = 1.23 MHz).
2) Transmission
Two types of Transmission:
i) Forward Transmission (base to mobile)
¤ Communications between the base and all mobiles are synchronized.
The base sends synchronized data to all mobiles (Figure 16.14). ii)
Reverse Transmission (mobile to base)
The use of CDMA in the forward direction is possible because the pilot channel sends a
continuous sequence of 1s to synchronize transmission.
The synchronization is not used in the reverse direction because we need an entity to do that,
which is not feasible.
Instead of CDMA, the reverse channels use DSSS (Figure 16.15).

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3) Synchronization
All base channels need to be synchronized to use CDMA.
To provide synchronization, bases use the services of a satellite system (GPS).
4) Two Data Rate Sets
IS-95 defines two data-rate sets:
The first set defines 9600, 4800, 2400, and 1200 bps.
The second set defines 14,400, 7200, 3600, and 1800 bps.
5) Frequency-Reuse Factor
The frequency-reuse factor is normally 1 because the interference from neighboring cells
cannot affect CDMA or DSSS transmission.
6) Soft Handoff
Every base-station continuously broadcasts signals using its pilot channel.
Thus, a mobile-station can detect the pilot signal from its cell and neighboring cells.
This enables a mobile-station to do a soft handoff.
Third Generation (3G)
3G cellular telephony provides both digital data and voice communication.
For example: Using a Smartphone,
A person can talk to anyone else in the world.
A person can download a movie, surf the Internet or play games.
Interesting characteristics: the Smartphone is always connected; we do not need to dial a number to connect to
the Internet. (IMT Internet Mobile Communication)
Some objectives defined by the blueprint IMT-2000 (3G working group):
Voice quality comparable to that of the existing public telephone network.
Data-rate of
144 kbps for access in a moving vehicle (car)
384 kbps for access as the user walks (pedestrians) and
2 Mbps for the stationary user (office or home).
Support for packet-switched and circuit-switched data services.
A band of 2 GHz.
Bandwidths of 2 MHz.
Interface to the Internet

IMT-2000 Radio Interfaces

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Radio interfaces (wireless standards) adopted by IMT-2000 (Figure 16.16):

IMT-DS
This uses a version of CDMA called W-CDMA (wideband CDMA).
W-CDMA uses a 5-MHz bandwidth.
It is compatible with the CDMA used in IS-95.
2) IMT-MC
This was known as CDMA 2000.
It is an evolution of CDMA technology used in IS-95 channels.
It combines
new wideband (15-MHz) spread spectrum &
narrowband (1.25-MHz) CDMA of IS-95.
It is backward-compatible with IS-95.
It allows communication on multiple 1.25-MHz channels up to 15 MHz.
3) IMT-TC
This uses a combination of W-CDMA and TDMA.
It tries to reach the IMT-2000 goals by adding TDMA multiplexing to W-CDMA.
4) IMT-SC
This uses only TDMA.
5) IMT-FT
This uses a combination of FDMA and TDMA.
Fourth Generation (4G)
4G cellular telephony is expected to be a complete evolution in wireless communications.
Some objectives defined by the 4G working group:
A spectrally efficient system.
High network capacity.
Data-rate of
100 Mbps for access in a moving vehicle
1 Gbps for stationary users and
100 Mbps between any two points in the world.
Smooth handoff across heterogeneous networks.
Seamless connectivity and global roaming across multiple networks.
High quality of service for next generation multimedia support.
Interoperability with existing wireless standards.
All IP, packet-switched, networks.
4G is only packet-based networks.
4G supports IPv6.
4G provides better multicast, security, and route optimization capabilities.
• Here we discuss, following issues: 1) Access Scheme 2) Modulation 3) Radio System
4) Antenna 5) Applications

1) Access Scheme
• To increase efficiency,
i) capacity, ii) scalability & iii) new access techniques are being considered for 4G.
• For example:
i) OFDMA and IFDMA are being considered for the downlink & uplink of the next generation UMTS. ii)
MC-CDMA is proposed for the IEEE 802.20 standard.
2) Modulation
More efficient 64-QAM is being proposed for use with the LTE standards.

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3) Radio System
The 4G uses a SDR system.
The components of an SDR are pieces of software and thus flexible.
The SDR can change its program to shift its frequencies to mitigate frequency interference.
4) Antenna
The MIMO and MU-MIMO antenna system is proposed for 4G.
Using this antenna, 4G allows independent streams to be transmitted simultaneously from all the antennas to
increase the data-rate.
MIMO also allows the transmitter and receiver coordinates to move to an open frequency when interference
occurs.
5) Applications
At the present rates of 15-30 Mbps, 4G is capable of providing users with streaming high-definition television.
At 100 Mbps, the content of a DVD-5 can be downloaded within about 5 minutes for offline access.
(OFDMA Orthogonal FDMA IFDMA interleaved FDMA)
(LTE Long Term Evolution SDR Software Defined Radio)
(MIMO multiple-input multiple-output MU-MIMO multiuser MIMO)
(UMTS Universal Mobile Telecommunications System)
(MC-CDMA multicarrier code division multiple access)

Satellite Networks
A satellite network is a combination of nodes that provides communication from one point on the Earth to
another.
A node can be
Satellite
Earth station or
End-user terminal/telephone
Like cellular networks, satellite networks divide the planet into cells.
Satellites can provide transmission capability to and from any location on Earth.
Advantages:
Satellite makes high-quality communication available to undeveloped parts of the world.
Cost effective: A huge investment in ground-based infrastructure is not required.

General Issues for Operation of Satellites

• Three issues related to the operation of satellites:
1) Orbits 2) Footprint 3) Frequency Bands for Satellite Communication
1) Orbits
An artificial satellite needs to have an orbit.
An orbit is the path in which the satellite travels around the Earth.
The orbit can be equatorial, inclined, or polar (Figure 16.17.)

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The period means the time required for a satellite to make a complete trip around the Earth.
The period of a satellite is determined by Kepler‟s law.
Kepler‟s law defines period as a function of the distance of the satellite from the center of the Earth.

Example 5.1

Example 5.2

2) Footprint
• Satellites process microwaves with bidirectional antennas (line-of-sight).
• Normally, the signal from a satellite is aimed at a specific area called the footprint.
• The signal-power at the center of the footprint is maximum.
• The signal-power decreases, as we move out from the footprint-center.
• The boundary of the footprint is the location where the power-level is at a predefined threshold.

3) Frequency Bands for Satellite Communication

For satellite communication, the frequencies reserved are in the GHz range.
Each satellite sends and receives over 2 different bands (Table 16.1):
Uplink: refers to the transmission from the Earth to the satellite.
Downlink: refers to the transmission from the satellite to the Earth.

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4) Three Categories of Satellites

• Three categories of satellites based on the location of the orbit (Figure 16.18):
1) Geostationary Earth orbit (GEO)
There is only one orbit, at an altitude of 36000 km, for the GEO satellite.
2) Low-Earth-orbit (LEO)
LEO satellites are below an altitude of 2000 km.
3) Medium-Earth-orbit (MEO)
MEO satellites are located at altitudes between 5000 and 15,000 km.

GEO Satellites
There is only one orbit at an altitude of 36,000 km (Figure 16.19).
Because orbital speed is based on the distance from the planet, only one orbit can be geostationary.
The orbit occurs at the equatorial plane.
Sending-antenna must have receiving-antenna in LOS (Line-of-sight).
Problem: A satellite that moves faster/slower than Earth‟s rotation is useful only for short periods. Solution:
To ensure constant communication, the satellite must move at same speed as the Earth.
Thus, the satellite seems to remain fixed above a certain spot.

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MEO Satellites
MEO satellites are located at altitudes between 5000 and 15,000 km.
Example: Global Positioning System (GPS)

Global Positioning System

GPS consists of 24 satellites in 6 orbits (Figure 16.20).
GPS is used for land, sea, and air navigation to provide time and location for vehicles and ships.
The orbits and the locations of the satellites in each orbit are designed systematically.
For example: At any time, 4 satellites are visible from any point on Earth.
• A GPS receiver has an almanac (or calendar) that tells the current position of each satellite.
• Here we discuss, 4 issues: 1) Trilateration 2) Measuring the Distance
3) Synchronization 4) Application

1) Trilateration
GPS is based on a principle called trilateration.
Trilateration means using three distances.
For example (Figure 16.21a):
On a plane, if we know our distance from three points, we know exactly where we are.
Assume we are 10 miles away from point A,
12 miles away from point B, and
15 miles away from point C.

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If we draw three circles with the centers at A, B, and C, we must be somewhere on circle A, somewhere
on circle B, and somewhere on circle C.
These three circles meet at one single point; this is our position (Figure 16.21a).

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In three-dimensional space, the situation is different.

Three spheres meet in two points, (Figure 16.21b).
We need at least four spheres to find our exact position in space (longitude, latitude, and altitude).
2) Measuring the Distance
• Trilateration principle can find our location on the Earth if we know →
our distance from 3 satellites and
→ position of each satellite.
• The position of each satellite can be calculated by a GPS receiver.
• Then, the GPS receiver needs to find its distance from at least three GPS satellites.
• The distance is measured using a principle called one-way ranging.

3) Synchronization
• Satellites use atomic clocks, which are precise and can function synchronously with each other.
• The receiver‟s clock is a normal quartz clock.
• However, there is no way to synchronize receiver‟s clock with the satellite‟s clock.
• There is an unknown offset between the satellite-clocks and the receiver-clock.
• The unknown offset introduces a corresponding offset in the distance calculation.
• Because of the offset, the measured distance is called a pseudo-range.

4) Applications
1) GPS is used by military forces.
For example:
Thousands of portable GPS receivers were used during the WW2 by foot soldiers, vehicles, and
helicopters.
2) GPS is used in navigation.
For example:
The driver of a car can find the location of the car. 3)
GPS is used for clock synchronization.
LEO Satellites
LEO satellites have polar orbits.
Usually, a LEO system has a cellular type of access (similar to the cellular telephone system).
Specifications:
Altitude = 500 to 2000 km
Rotation Period = 90 to 120 min
Satellite Speed =20,000 to 25,000 km/h
Footprint Diameter = 8000 km
Round-Trip Time < 20 ms

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Because LEO satellites are close to Earth, 20 ms RTT is normally acceptable for audio communication.
A LEO system is made of a group of satellites that work together as a network.
Each satellite acts as a switch.
Different types of links (Figure 16.22):
1) ISLs (Inter-Satellite Links)
Satellites that are close to each other are connected through ISLs.
2) UML (User Mobile Link)
A mobile system communicates with the satellite through a UML.
3) GWL (Gate Way Link)
A satellite communicates with the Earth station (gateway) through a GWL.

• LEO satellites can be divided into three categories:

1) Little LEO
Operating frequency < 1 GHz.
They are mostly used for low data-rate messaging.
2) Big LEO
Operating frequency = 1 to 3 GHz.
Examples: Globalstar & Iridium
Globalstar Iridium
Orbit-Altitude 1400 km 750 km
No. of Satellites 48 66
No. of Orbits 6 6
Satellites per Orbit 8 11
3) Broadband LEO
Broadband LEO provides communication similar to fiber-optic networks.
Example: Teledesic
Teledesic satellite provides fiber-optic-like communication (broadband channels, low error rate, and
low delay).
Main purpose: To provide broadband Internet access for users all over the world.

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MODULE 5(CONT.): NETWORK LAYER PROTOCOLS

Network Layer Protocols

The network layer contains following 4 protocols (Figure 19.1):
Internet Protocol (IP)
IP is the main protocol responsible for packetizing, forwarding, and delivery of a packet at the network
layer.
2) Internet Control Message Protocol (ICMP)
ICMP helps IP to handle some errors that may occur in the network-layer delivery.
3) Internet Group Management Protocol (IGMP)
IGMP is used to help IPv4 in multicasting.
4) Address Resolution Protocol (ARP)
ARP is used to glue the network and data-link layers in mapping network-layer addresses to
link-layer addresses.

Internet Protocol (IP)

5.7.1 Internet Protocol (IP)
IP is main protocol responsible for packetizing, forwarding & delivery of a packet at network layer.
IP is an unreliable datagram protocol.
IP provides a best-effort delivery service.
The term best-effort means that the packets can
be corrupted
be lost or
arrive out-of-order.
If reliability is important, IP must be paired with a TCP which is reliable transport-layer protocol.
IP is a connectionless protocol.
IP uses the datagram approach.
Each datagram is handled independently.
Each datagram can follow a different route to the destination.
Datagrams may arrive out-of-order at the destination.
Datagram Format
• IP uses the packets called datagrams.
• A datagram consist of 2 parts (Figure 19.2):

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1) Payload 2) Header.

1) Payload
Payload (or Data) is the main reason for creating a datagram.
Payload is the packet coming from other protocols that use the service of IP.
2) Header
Header contains information essential to routing and delivery.
IP header contains following fields:
1) Version Number (VER)
This field indicates version number used by the packet. Current version=4
2) Header Length (HLEN)
This field specifies length of header.
When a device receives a datagram, the device needs to know
when the header stops and
when the data starts.
3) Service Type
This field specifies priority of packet based on delay, throughput, reliability & cost requirements.
4) Total Length
This field specifies the total length of the datagram (header plus data).
Maximum length=65535 bytes.
5) Identification, Flags, and Fragmentation Offset
These 3 fields are used for fragmentation and reassembly of the datagram.
Fragmentation occurs when the size of the datagram is larger than the MTU of the network.
6) Time-to-Live (TTL)
This field is indicates amount of time, the packet is allowed to remain in the network.
If TTL becomes 0 before packet reaches destination, the router
discards packet and
sends an error-message back to the source.
7) Protocol
This field specifies upper-layer protocol that is to receive the packet at the destination-host.
For example (Figure 19.3):
For TCP, protocol = 6 For UDP, protocol = 17
8) Header Checksum
This field is used to verify integrity of header only.
If the verification process fails, packet is discarded.
9) Source and Destination Addresses
These 2 fields contain the IP addresses of source and destination hosts.
10) Options
This field allows the packet to request special features such as
security level
route to be taken by packet and
timestamp at each router.

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This field can also be used for network testing and debugging.
11) Padding
This field is used to make the header a multiple of 32-bit words.

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Example 5.3

Example 5.4

Example 5.5

(Comparing a datagram to a postal package.

Payload is the content of the package.
Header is only the information written on the package).
Fragmentation
Maximum Transfer Unit (MTU)
Each network imposes a restriction on maximum size of packet that can be carried. This is called the
MTU(maximum transmission unit).
For example:
For Ethernet, MTU = 1500 bytes
For FDDI, MTU = 4464 bytes
When IP wants send a packet that is larger than MTU of physical-network, IP breaks packet into smaller
fragments. This is called fragmentation (Figure 19.5).
Designers have decided to make the maximum length of IP datagram = 65,535 bytes. This ensures that the IP
protocol is independent of the physical network,
When a datagram is fragmented, each fragment has its own header.
A fragmented-datagram may itself be fragmented if it encounters a network with an even smaller MTU.
Source host or router is responsible for fragmentation of original datagram into the fragments.
Destination host is responsible for reassembling the fragments into the original datagram.

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Figure 19.5b: Packet Fragmentation

Fields Related to Fragmentation & Reassembly

• Three fields in the IP header are used to manage fragmentation and reassembly:
1) Identification 2) Flags 3) Fragmentation offset.
1) Identification
• This field is used to identify to which datagram a particular fragment belongs to (so that fragments for
different packets do not get mixed up).
• To guarantee uniqueness, the IP protocol uses an up-counter to label the datagrams.
• When the IP protocol sends a datagram, IP protocol
→ copies the current value of the counter to the identification field and →
increments the up-counter by 1.
• When a datagram is fragmented, the value in the identification field is copied into all fragments.
• The identification number helps the destination in reassembling the datagram.
2) Flags
This field has 3 bits.
The leftmost bit is not used.
DF bit (Don't Fragment):
If DF=1, the router should not fragment the datagram. Then, the router
discards the datagram and
sends an error-message to the source host.
If DF=0, the router can fragment the datagram if necessary.
MF bit (More Fragment):
If MF=1, there are some more fragments to come.
If MF=0, this is last fragment.
3) Fragmentation Offset
This field identifies location of a fragment in a packet.
This field is the offset of the data in the original datagram.
Example 5.6

Example 5.7

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Example 5.8

Example 5.9

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Example 5.10

Options
This field allows the packet to request special features such as
security level
route to be taken by packet and
timestamp at each router.
This field can also be used for network testing and debugging.
As the name implies, options are not required for a datagram.
The header is made of two parts: 1) Fixed part and 2) Variable part.
Maximum size of Fixed part = 20 bytes.
Maximum size of Variable part = 40 bytes
Options are divided into two broad categories: 1) Single-byte options and 2) Multiple-byte options.
1) Single Byte Options
i) No Operation
This option is used as filler between options. ii)
End of Option
This option is used for padding at the end of the option field.
Multiple Byte Options
i) Record Route
This option is used to record the routers that handle the datagram.
This option can list up to 9 router-addresses. ii)
Strict Source Route
This option is used by the source to pre-determine a route for the datagram.
Useful purposes: The sender can choose a route with a specific type of service, such as
minimum delay
maximum throughput or
more secure/reliable.
All the defined-routers must be visited by the datagram.
If the datagram visits a router that is not on the list, the datagram is discarded.
iii) Loose Source Route
This option is similar to the strict source route, but it is less rigid.
Each router in the list must be visited, but the datagram can visit other routers as well. iv)
Timestamp
This option is used to record the time of datagram processing by a router.
The time is expressed in milliseconds from midnight GMT (Greenwich Mean Time).
The recorded-time can help the managers to track the behavior of the routers in the Internet.
Security of IPv4 Datagrams
Nowadays, the Internet is not secure anymore.
Three security issues applicable to the IP protocol:
Packet sniffing
Packet modification and
IP spoofing.
1) Packet Sniffing
Attackers may

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capture certain packets

intercept the packets and
make a copy of the packets.
Packet sniffing is a passive attack.
Passive attack means the attacker does not modify the contents of the packet.
The attack is difficult to detect „.‟ sender & receiver may never know that the packet has been copied.
Solution:
Although the attack cannot be stopped, encryption of packet may make the attacker‟s job difficult.
The attacker may still sniff the packet, but the content is not detectable (or understandable).
2) Packet Modification
Attackers may succeed in accessing the content of a packet.
Then, the attacker can
change the address of the packet or
change the data of the packet
Solution:
The attack can be prevented by data integrity mechanism.
Data integrity guarantees that the packet is not modified during the transmission.
3) IP Spoofing
The attacker pretends as a trusted entity and obtains all the secret information.
For example:
An attacker sends an IP packet to a bank pretending as legitimate customers.
• Solution:
The attack can be prevented using an origin-authentication mechanism.

IPSec (IP Security)

• IP packets can be protected from the various network-attacks using a protocol called IPSec.
• IPSec protocol & IP protocol can be used to create a connection-oriented service between 2 entities.
• Four services of IPSec:
1) Defining Algorithms & Keys
To create a secure channel b/w two entities, the two entities can agree on some available algorithms and
keys.
2) Packet Encryption
To provide privacy, the packets exchanged b/w two parties can be encrypted using the
encryption-algorithms and a shared key.
This prevents the packet sniffing attack.
3) Data Integrity
Data integrity guarantees that the packet is not modified during the transmission.
If the received packet does not pass the data integrity test, the packet is discarded.
This prevents the packet modification attack.
4) Origin Authentication
Origin Authentication guarantees that the packet is not created by a pretender.
This prevents the IP Spoofing attack.

ICMP
ICMP is a network-layer protocol.
This is used to handle error and other control messages.

5.8.1 MESSAGES
ICMP messages are divided into 2 broad categories:

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Error-Reporting Messages
These messages report problems that a router or a host may encounter during the processing of
datagram.
2) Query Messages
These messages help a host or a network manager get specific information from a router or
another host. For
example:
Nodes can discover their neighbors.
Hosts can discover and learn about routers on their network.
Routers can help a node redirect the messages.
Fields of ICMP messages (Figure 19.8):
Type: This field identifies the type of message.
Code: This field specifies the reason for the particular message type. For
example,
Type 03 = problem reaching the destinations
Type 11 = problem related to time exceeded.
Checksum: This field is used to detect errors in the ICMP message.
Data section: This field can be used for diagnostic purposes by matching the information in the ICMP
message with the original data in the IP packet.

Error Reporting Messages

Main responsibility of ICMP: To report some errors that may occur during the processing of the datagram
(Figure 19.9).
These messages report problems that a router or a host may encounter during the processing of datagram.
ICMP does not correct errors; ICMP simply reports the errors to the source.
Error correction is left to the higher-level protocols (such as TCP or UDP)

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Rules for reporting messages:

No error-message will be generated for a datagram having a multicast address (or special address).
No error-message will be generated in response to a datagram carrying an ICMP error-message.
No error-message will be generated for a fragmented datagram that is not the first fragment.

1) Destination Unreachable (Type=3)

This message is related to problem reaching the destinations.
This message uses different codes (0 to 15) to define type of error-message.
Possible values for code field:
Code 0 = network unreachable
Code 1 = host unreachable
Code 2 = protocol unreachable
Code 3 = port unreachable
2) Source Quench (Type=4)
This message informs the sender that
network has encountered congestion and
datagram has been dropped.
The source needs to slow down sending more datagrams.
In other words, ICMP adds a kind of congestion control mechanism to the IP protocol.
3) Redirection Message (Type=5)
This is used when the source uses a wrong router to send out its message.
The router
redirects the message to the appropriate router &
informs the source to change its default router in the future.
The IP address of the default router is sent in the message.
TTL prevents a datagram from being aimlessly circulated in the Internet.
When TTL becomes 0,
the datagram is dropped by the visiting router and
a time exceeded message (type 11) is sent to the source.
4) Parameter Problem (Type=12)
This message can be sent when either
there is a problem in the header of a datagram (code 0) or
some options are missing or cannot be interpreted (code 1).
Query Messages
• These messages help a network manager to get specific information from a router or host.
• Two types of query messages: request (type 8) and reply (type 0).
1) Echo Request & Echo Reply
These messages are used to determine whether a remote-host is alive. A
source-host sends an echo request message to destination-host;
If the destination-host is alive, it responds with an echo reply message.

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Type=8 is used for echo request

Type=0 is used for echo reply.
These messages can be used in two debugging tools: ping and traceroute.
2) Timestamp Request & Timestamp Reply
These messages are used to
find the round-trip time between two devices or
check whether the clocks in two devices are synchronized.
The timestamp request sends a number, which defines the time the message is sent.
The timestamp reply resends another number, which defines the time the message is sent.
The timestamp reply also includes 2 new numbers representing
the time the request was received and
the time the response was sent.
Type=13 is used for timestamp request
Type=14 is used for timestamp reply.
Debugging Tools
There are several tools that can be used in the Internet for debugging.
We can determine the viability of a host or router.
We can trace the route of a packet.
Two tools used for debugging: 1) Ping and 2) Traceroute.

Ping
The ping program can be used to find if a host is alive and responding
Here, ping is used to see how it uses ICMP packets
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 by 1 each time a new message is
sent.
Ping 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).

Traceroute
The traceroute program 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.
Traceroute
The traceroute program is different from the ping program.
The ping program gets help from 2 query messages;
The traceroute program gets help from two error-reporting messages: time-exceeded and
destination-unreachable.
The traceroute is an application layer program, but only the client program is needed. In
other words, there is no traceroute server program.
The traceroute application program is encapsulated in a UDP user datagram, but traceroute
intentionally uses a port number that is not available at the destination.

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Mobile IP
Mobile IP is the extension of IP protocol.
Mobile IP allows mobile computers to be connected to the Internet.

Addressing
In Mobile IP, the main problem that must be solved is addressing.
Stationary Hosts
The original IP addressing assumed that a host is stationary.
A router uses an IP address to route an IP datagram.
An IP address has two parts: a prefix and a suffix.
The prefix associates a host with a network.
For example, the IP address 10.3.4.24/8 defines a host attached to the network 10.0.0.0/8.
• The address is valid only when the host is attached to the network.
If the network changes, the address is no longer valid.
Mobile Hosts
When a host moves from one network to another, the IP addressing structure needs to be modified.
The host has two addresses (Figure 19.12):
Home address &
Care-of address
Home Address
Original address of host called the home address.
The home address is permanent.
The home address associates the host with its home network.
Home network is a network that is the permanent home of the host.
2) Care-of-Address
The care-of address is temporary.
The care-of address changes as the mobile-host moves from one network to another.
Care-of address is associated with the foreign network.
Foreign network is a network to which the host moves.

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When a mobile-host visits a foreign network, it receives its care-of address during the agent discovery
and registration phase.

Agents
Two agents are required to make change of address transparent to rest of the Internet (Fig 19.13):
Home-agent and
Foreign-agent.

1) Home Agent
• The home-agent is a router attached to the home network.
• The home-agent acts on behalf of mobile-host when a remote-host sends a packet to mobile-host.
• The home-agent receives and delivers packets sent by the remote-host to the foreign-agent.
2) Foreign Agent
The foreign-agent is a router attached to the foreign network.
The foreign-agent receives and delivers packets sent by the home-agent to the mobile-host.
The mobile-host can also act as a foreign-agent i.e. mobile-host and foreign-agent can be the same.
However, to do this, a mobile-host must be able to receive a care-of address by itself.
In addition, the mobile-host needs the necessary software to allow it to communicate with the home-agent and
to have two addresses: i) its home address and ii) its care-of address.
This dual addressing must be transparent to the application programs.
Collocated Care-of-Address
When the mobile-host and the foreign-agent are the same, the care-of-address is called a collocated
care-of-address.
Advantage:
mobile-host can move to any network w/o worrying about availability of a foreign-agent.
Disadvantage:

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The mobile-host needs extra software to act as its own foreign-agent.

Three Phases
To communicate with a remote-host, a mobile-host goes through 3 phases (Figure 19.14):
Agent Discovery: involves the mobile-host, the foreign-agent, and the home-agent.
Registration: involves the mobile-host, the foreign-agent, and the home-agent.
Data Transfer: Here, the remote-host is also involved.

Agent Discovery
Agent discovery consists of two subphases:
A mobile-host must discover (learn the address of) a home-agent before it leaves its home network.
A mobile-host must also discover a foreign-agent after it has moved to a foreign network.
This discovery consists of learning the care-of address as well as the foreign-agent‟s address.
Two types of messages are used: i) advertisement and ii) solicitation.

1) Agent Advertisement
• When a router advertises its presence on a network using an ICMP router advertisement, it can append an
agent advertisement to the packet if it acts as an agent.

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• Various fields are (Figure 19.15):

1) Type
This field is set to 16.
2) Length
This field defines the total length of the extension message.
3) Sequence Number
This field holds the message number.
The recipient can use the sequence number to determine if a message is lost.
4) Lifetime
This field defines the number of seconds that the agent will accept requests.
If the value is a string of 1s, the lifetime is infinite.
5) Code
This field is a flag in which each bit is set (1) or unset (0) (Table 19.1).

6) Care-of Addresses
This field contains a list of addresses available for use as care-of addresses.
The mobile-host can choose one of these addresses.
The selection of this care-of address is announced in the registration request.

Agent Solicitation
When a mobile-host has moved to a new network and has not received agent advertisements, it can initiate an
agent solicitation.
It can use the ICMP solicitation message to inform an agent that it needs assistance
Registration
After a mobile-host has moved to a foreign network and discovered the foreign-agent, it must register.
Four aspects of registration:
The mobile-host must register itself with the foreign-agent.
The mobile-host must register itself with its home-agent. This is normally done by the foreign-agent on
behalf of the mobile-host.
The mobile-host must renew registration if it has expired.
The mobile-host must cancel its registration (deregistration) when it returns home.

Request & Reply

To register with the foreign-agent and the home-agent, the mobile-host uses a registration request and a
registration reply.
1) Registration Request
A registration request is sent from the mobile-host to the foreign-agent
to register its care-of address and

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to announce its home address and home-agent address.

Foreign-agent, after receiving and registering the request, relays the message to the home-agent.
The home-agent now knows the address of the foreign-agent because the IP packet that is used for relaying has
the IP address of the foreign-agent as the source address.

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Various fields are (Figure 19.16):

Type
This field defines the type of message.
For a request message the value of this field is 1.
2) Flag
This field defines forwarding information.
The value of each bit can be set or unset (Table 19.2).

3) Lifetime
This field defines the number of seconds the registration is valid.
i) If the field is a string of 0s, the request message is asking for deregistration. ii) This
field If the field is a string of 1s, the lifetime is infinite.
4) Home Address
This field contains the permanent (first) address of the mobile-host.
5) Home Agent Address
This field contains the address of the home-agent.
6) Care-of-Address
This field is the temporary (second) address of the mobile-host.
7) Identification
This field contains a 64-bit number that is inserted into the request by the mobile-host.
This field matches a request with a reply.
8) Extensions
This field is used for authentication.
This field allows a home-agent to authenticate the mobile agent.

Registration Reply
A registration reply is sent from home-agent to foreign-agent and then relayed to the mobile-host.
The reply confirms or denies the registration request. (Figure 19.17)
The fields are similar to registration request with the 3 exceptions:
The value of the type field is 3.

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The code field replaces the flag field and shows the result of the registration request (acceptance or
denial).
The care-of address field is not needed.

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Data Transfer
• After agent discovery & registration, a mobile-host can communicate with a remote-host (Fig 19.17).

Here we have 4 cases (Figure 19.18):

1) From Remote-host to Home Agent
When a remote-host wants to send a packet to the mobile-host, the remote-host uses
address of itself as the source address and
home address of the mobile-host as the destination address.
In other words, the remote-host sends a packet as though the mobile-host is at its home network.
The packet is intercepted by the home-agent, which pretends it is the mobile-host.
This is done using the proxy ARP technique (Path 1 of Figure 19.18).
2) From Home Agent to Foreign Agent
After receiving the packet, the home-agent sends the packet to the foreign-agent, using the tunneling concept.
The home-agent encapsulates the whole IP packet inside another IP packet using its address as the source and
the foreign-agent‟s address as the destination. (Path 2 of Figure 19.18).
3) From Foreign Agent to Mobile Host
When the foreign-agent receives the packet, it removes the original packet.
However, since the destination address is the home address of the mobile-host, the foreign-agent consults a
registry table to find the care-of address of the mobile-host. (Otherwise, the would just be sent back to the
home network.)

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The packet is then sent to the care-of address (Path 3 of Figure 19.18).
4) From Mobile Host to Remote Host
When a mobile-host wants to send a packet to a remote-host (for example, a response to the packet it has
received), it sends as it does normally.
The mobile-host prepares a packet with its home address as the source, and the address of the remote-host as
the destination.
Although the packet comes from the foreign network, it has the home address of the mobile-host (Path 4 of
Figure 19.18).
Inefficiency in Mobile IP
Communication involving mobile IP can be inefficient.
The inefficiency can be severe or moderate.
The severe case is called double crossing or 2X.
The moderate case is called triangle routing or dog-leg routing.
Double Crossing
Double crossing occurs when a remote-host communicates with a mobile-host that has moved to the same
network (or site) as the remote-host (Figure 19.19).
When the mobile-host sends a packet to the remote-host, there is no inefficiency; the communication is local.
However, when remote-host sends a packet to mobile-host, the packet crosses the Internet twice.
Since a computer usually communicates with other local computers (principle of locality), the inefficiency
from double crossing is significant.
Triangle Routing
Triangle routing occurs when the remote-host communicates with a mobile-host that is not attached to the same
network (or site) as the mobile-host.
When the mobile-host sends a packet to the remote-host, there is no inefficiency.

However, when the remote-host sends a packet to the mobile-host, the packet goes from the remote-host to the
home-agent and then to the mobile-host.
The packet travels the two sides of a triangle, instead of just one side (Figure 19.20).

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Solution
One solution to inefficiency is for the remote-host to bind the care-of address to the home address of a mobile-
host.
For example, when a home-agent receives the first packet for a mobile-host, it forwards the packet to the
foreign-agent; it could also send an update binding packet to the remote-host so that future packets to this host
could be sent to the care-of address.
The remote-host can keep this information in a cache.
The problem with this strategy is that the cache entry becomes outdated once the mobile-host moves.
In this case, the home-agent needs to send a warning packet to the remote-host to inform it of the change.

MODULE 5(CONT.): NEXT GENERATION IP

IPv6 Addressing
The main reason for migration from IPv4 to IPv6 is the small size of the address-space in IPv4.
Size of IPv6 address =128 bits (four times the address length in IPv4, which is 32 bits).

Representation
• Two notations can be used to represent IPv6 addresses: 1) binary and 2) colon hexadecimal.

Address Space
The address-space of IPv6 contains 2128 addresses.
Three Address Types
Three types of destination address: 1) Unicast 2) Anycast and 3) Multicast.
1) Unicast Address
A unicast address defines a single interface (computer or router).
The packet with a unicast address will be delivered to the intended recipient.
2) Anycast Address
An anycast address defines a group of computers that all share a single address.
A packet with an anycast address is delivered to only one member of the group.
The member is the one who is first reachable.
3) Multicast Address
A multicast address also defines a group of computers.
Difference between anycasting and multicasting.
In any casting, only one copy of the packet is sent to one of the members of the group.
in multicasting each member of the group receives a copy.
Address Space Allocation
The address-space is divided into several blocks of varying size.
Each block is allocated for a special purpose.

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Global Unicast Addresses

The block in the address-space used for unicast communication b/w 2 hosts in the Internet is called global
unicast address block.
CIDR for the block is 2000::/3. This means that the three leftmost bits are the same for all addresses in this
block (001).
The size of this block is 2125 bits, which is more than enough for Internet expansion for many years to come.
An address in the block is divided into 3 parts (Figure 22.1):
Global routing prefix (n bits)
Subnet identifier (m bits) and
Interface identifier (q bits).

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The global routing prefix is used to route the packet through the Internet to the organization site, such as the
ISP that owns the block.
Since the first 3 bits in this part are fixed (001), the rest of the 45 bits can be defined for up to 245 sites (a
private organization or an ISP).
The global routers in Internet route a packet to its destination site based on the value of n.
The next m bits define a subnet in an organization.
The last q bits define the interface identifier.
Two link layer addressing schemes:
64-bit extended unique identifier (EUI-64) defined by IEEE and
48-bit link-layer address defined by Ethernet.
1) Mapping EUI-64
• To map a 64-bit physical address, the global/local bit of this format needs to be changed from 0 to 1 (local to
global) to define an interface address (Figure 22.2).

2) Mapping Ethernet MAC Address

Mapping a 48-bit Ethernet address into a 64-bit interface identifier is more involved.
We need to change the local/global bit to 1 and insert an additional 16 bits.
The additional 16 bits are defined as 15 ones followed by one zero, or FFFE16 (Figure 22.3).

Special Addresses

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Following are different special addresses (Figure 22.4):

1) Unspecified Address
The unspecified address is a subblock containing only one address.
This address is used during bootstrap when a host does not know its own address and wants to send an inquiry
to find it.
2) Loopback Address
The loopback address also consists of one address.
3) Transition Address
• During the transition from IPv4 to IPv6, hosts can use their IPv4 addresses embedded in IPv6 addresses.
• Two formats have been designed for this purpose: compatible and mapped.
1) Compatible Address
A compatible address is an address of 96 bits of zero followed by 32 bits of IPv4 address.
It is used when a computer using IPv6 wants to send a message to another computer using IPv6.
2) Mapped Address
A mapped address is used when a computer already migrated to version 6 wants to send an address to a
computer still using version 4.
Other Assigned Blocks
IPv6 uses 2 large blocks for private addressing and one large block for multicasting (Figure 22.5).
1) Unique Local Unicast Block
A subblock in a unique local unicast block can be privately created and used by a site.
The packet carrying this type of address as the destination address is not expected to be routed.
This type of address has the identifier 1111 110.
The next bit can be 0 or 1 to define how the address is selected (locally or by an authority).
2) Link Local Block
A subblock in link local block can be used as a private address in a network.
This type of address has the block identifier 1111111010.
The next 54 bits are set to zero.
The last 64 bits can be changed to define the interface for each computer.

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Auto configuration
• When a host in IPv6 joins a network, it can configure itself using the following process:
The host first creates a link local address for itself.
This is done by
taking the 10-bit link local prefix (1111 1110 10)
adding 54 zeros and
adding the 64-bit interface identifier.
The result is a 128-bit link local address.
The host then tests to see if this link local address is unique and not used by other hosts.
Since the 64-bit interface identifier is supposed to be unique, the link local address generated
is unique with a high probability.
To check uniqueness, the host
sends a neighbor solicitation message and
waits for a neighbor advertisement message.
If any host in the subnet is using this link local address, the process fails and the host cannot auto-
configure itself.
If the uniqueness of the link local address is passed, the host stores this address as its link local address (for
private communication), but it still needs a global unicast address.
The host then sends a router solicitation message to a local router.
If there is a router running on the network, the host receives a router advertisement message
that includes
global unicast prefix and
subnet prefix that the host needs to add to its interface identifier to generate its
global unicast address.
If the router cannot help the host with the configuration, it informs the host in the router
advertisement message (by setting a flag).

The IPv6 Protocol

5.11.1 Changes from IPv4 to IPv6 (Advantages of IPv6)
Header Format
IPv6 uses a new header format.
Options are
separated from the base-header and
inserted between the base-header and the data.

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This speeds up the routing process (because most of the options do not need to be checked
by routers).
New Options
IPv6 has new options to allow for additional functionalities.
Extension
IPv6 is designed to allow the extension of the protocol if required by new technologies or
applications.
Resource Allocation
In IPv6,
type-of-service (TOS) field has been removed
two new fields: 1) traffic class and 2) flow label, are added to enable the source to request
special handling of the packet.
This mechanism can be used to support real-time audio and video.
Security
The encryption option provides confidentiality of the packet.
The authentication option provides integrity of the packet.
Packet Format

IP header contains following fields (Figure 22.6):

Version
This specifies version number of protocol. For IPv6, version=6.
2) Traffic Class
This field is used to distinguish different payloads with different delivery requirements.
(Traffic class replaces the type-of-service field in IPv4).
3) Flow Label
This field is designed to provide special handling for a particular flow of data.
4) Payload Length
This indicates length of data (excluding header). Maximum length=65535 bytes.
The length of the base-header is fixed (40 bytes); only the length of the payload needs to be defined.
5) Next Header
This identifies type of extension header that follows the basic header.
6) Hop Limit
This specifies number of hops the packet can travel before being dropped by a router. (Hop
limit serves the same purpose as the TTL field in IPv4).
7) Source and Destination Addresses
These identify source host and destination host respectively.

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8) Payload
The payload contains zero or more extension headers (options) followed by the data from other
protocols (UDP, TCP, and so on).
The payload can have many extension headers as required by the situation.
Each extension header has 2 mandatory fields (Figure 22.7):
Next header and
Length
Two mandatory fields are followed by information related to the particular option.
Concept of Flow & Priority in IPv6
To a router, a flow is a sequence of packets that share the same characteristics such as
traveling the same path
using the same resources or
having the same kind of security
A router that supports the handling of flow labels has a flow label table.
The table has an entry for each active flow label.
Each entry defines the services required by the corresponding flow label.
When a router receives a packet, the router consults its flow label table.
Then, the router provides the packet with the services mentioned in the entry.
A flow label can be used to support the transmission of real-time audio/video.
Real-time audio/video requires resources such as
high bandwidth
large buffers or
long processing time
Resource reservation guarantees that real-time data will not be delayed due to a lack of resources.

Fragmentation & Reassembly

Fragmentation of the packet is done only by the source, but not by the routers. The
reassembling is done by the destination.
At routers, the fragmentation is not allowed to speed up the processing in the router.
Normally, the fragmentation of a packet in a router needs a lot of processing. This is because
The packets need to be fragmented.
All fields related to the fragmentation need to be recalculated.
The source will
check the size of the packet and
make the decision to fragment the packet or not.
If packet-size is greater than the MTU of the network, the router will drop the packet.
Then, the router sends an error message to inform the source.
Extension Header
An IP packet is made of

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base-header &
some extension headers.
Length of base header = 40 bytes.
To support extra functionalities, extension headers can be placed b/w base header and payload.
Extension headers act like options in IPv4.
Six types of extension headers (Figure 22.8):
1) Hop-by-hop option 2) Source routing 3) Fragmentation
4) Authentication 5) Encrypted security payload 5) Destination option.

1) Hop-by-Hop Option
This option is used when the source needs to pass information to all routers visited by the datagram.
Three options are defined: i) Pad1, ii) PadN, and iii) Jumbo payload.
i) Pad1
This option is designed for alignment purposes.
Some options need to start at a specific bit of the 32-bit word.
Pad1 is added, if one byte is needed for alignment. ii)
PadN
PadN is similar in concept to Pad1.
The difference is that PadN is used when 2 or more bytes are needed for alignment. iii)
Jumbo Payload
This option is used when larger packet has to be sent. (> 65,535 bytes)
Large packets are referred to as jumbo packets.
Maximum length of payload = 65,535 bytes.
Destination Option
This option is used when the source needs to pass information to the destination only.
Intermediate routers are not allowed to access this information.
Two options are defined: i) Pad1 & ii) PadN
3) Source Routing
This option combines the concepts of
strict source routing and
loose source routing.
4) Fragmentation
In IPv6, only the original source can fragment.
A source must use a “Path MTU Discovery technique” to find the smallest MTU along the path from the
source to the destination.
Minimum size of MTU = 1280 bytes. This value is required for each network connected to the Internet.
If a source does not use a Path MTU Discovery technique, the source fragments the datagram to a size of 1280
bytes.
5) Authentication
This option has a dual purpose:
Validates the message sender: This is needed so the receiver can be sure that a message is from the
genuine sender and not from an attacker.
Ensures the integrity of data: This is needed to check that the data is not altered in transition

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by some attacker.
6) Encrypted Security Payload (ESP)
• This option provides confidentiality and guards against attacker.
Comparison of Options between IPv4 and IPv6
The no-operation and end-of-option options in IPv4 are replaced by Pad1 and PadN options in IPv6.
The record route option is not implemented in IPv6 because it was not used.
The timestamp option is not implemented because it was not used.
The source route option is called the source route extension header in IPv6.
The fragmentation fields in the base-header section of IPv4 have moved to the fragmentation extension
header in IPv6.
The authentication extension header is new in IPv6.
The encrypted security payload extension header is new in IPv6.

The ICMPv6 Protocol

ICMP, ARP & IGMP protocols in IPv4 are combined into one single protocol called ICMPv6 (Fig 22.9).

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Four groups of messages (Figure 22.10):

Error-reporting messages
Informational messages
Neighbor-discovery messages and
Group-membership messages.

Error-Reporting Messages
• Main responsibility of ICMP: Report errors.
• ICMP forms an error packet, which is then encapsulated in the datagram.
• The encapsulated datagram is delivered to the original source.
• Four types of errors:
1) Destination unreachable
2) Packet too big
3) Time exceeded and
4) Parameter problems.
1) Destination-Unreachable Message
Here, a router cannot forward a datagram or a host cannot deliver the datagram to the upper layer
protocol.
So, the router/host
→ discards the datagram and
→ sends a destination-unreachable message to the source.
2) Packet-Too-Big Message
Fragmentation of the packet is done only by the source, but not by the routers. If a
router receives a datagram larger than MTU size of the network, the router
→ discards the datagram and
→ sends a packet-too-big message to the source.
3) Time-Exceeded Message
A time-exceeded error message is generated in 2 cases: i)
When the TTL value becomes zero and
ii) When not all fragments of a datagram have arrived in the time-limit.
4) Parameter-Problem Message
Any missing value in the datagram-header can create serious problems. If a
router discovers any missing value in any field, the router
→ discards the datagram and
→ sends a parameter-problem message to the source.
Informational Messages
Two types of messages: i) echo request and ii) echo reply.
These 2 messages are used to check whether 2 devices can communicate with each other.
A source-host can send an echo-request message to another host.
The destination-host can respond with the echo-reply message to the source-host.

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Neighbor-Discovery Messages
Two new protocols are used:
Neighbor-Discovery (ND) protocol and
Inverse-Neighbor-Discovery (IND) protocol.
These 2 protocols are used by nodes on the same link for 3 main purposes:
Hosts use the ND protocol to find routers in the neighborhood that will forward packets for them.
Nodes use the ND protocol to find the link-layer addresses of neighbors.
Nodes use the IND protocol to find the IPv6 addresses of neighbors.
Seven types of errors:
Router-Solicitation Message
A host/router uses router-solicitation message to find a router in n/w that can forward a datagram.
Physical address of the host/router is included to make the response easier for the router.
2) Router-Advertisement Message
A host/router sends the router-advertisement message in response to a router solicitation message.
3) Neighbor-Solicitation Message
The neighbor solicitation message has the same duty as the ARP request message.
A host uses the neighbor solicitation message when the host has a message to send to a
neighbor.
The sender knows the IP address of the receiver, but needs the physical address of the receiver.
The physical address is needed for the datagram to be encapsulated in a frame.
4) Neighbor-Advertisement Message
A host sends the neighbor-advertisement message in response to a neighbor solicitation message.
5) Redirection Message
The purpose of the redirection message is the same as for version 4.
However, the format of the packet now accommodates the size of the IP address in version 6.
Also, an option is added to let the host know the physical address of the target router.
6) Inverse-Neighbor-Solicitation Message
A host uses inverse-neighbor-solicitation message to know the physical address of a neighbor, but not
the neighbor‟s IP address.
The message is encapsulated in a datagram using a multicast address. The
node must send the following 2 information in the option field:
i) Physical address of the sender and ii)
Physical address of the target node.
The sender can also include its IP address and the MTU value for the link.
7) Inverse-Neighbor-Advertisement Message
A host sends the inverse-neighbor-advertisement message in response to a inverse-neighbor-discovery
message.
Group Membership Messages
The management of multicast delivery handling in IPv4 is given to the IGMPv3 protocol.
In IPv6, this responsibility is given to the Multicast Listener Delivery protocol.
MLDv2 has 2 types of messages:
Membership-query message and
Membership-report message.
The first type can be divided into 3 subtypes: i) General, ii) Group-specific, and iii) Group-and-source specific.

1) Membership-Query Message
A router sends a membership-query message to find active group-members in the network.
The format of the membership-query in MLDv2 is exactly the same as the one in IGMPv3 three exceptions:
Size of the multicast address & source address has been changed from 32 bits to 128 bits.

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The field size is in the maximum response code field, in which the size has been changed from 8 bits to
16 bits.
The format of the first 8 bytes matches the format for other ICMPv6 packets because MLDv2
is considered to be part of ICMPv6.
2) Membership-Report Message
The format of the membership-report in MLDv2 is exactly the same as the one in IGMPv3 one exception:
Size of the multicast address & source address has been changed from 32 bits to 128 bits.
In particular, the record type is the same as the one defined for IGMPv3 (types 1 to 6).
Transition from IPv4 to IPv6
5.13.1 Strategies
Three strategies have been devised for transition:
Dual stack
Tunneling and
Header translation.
1) Dual Stack
• Recommended: All hosts must run IPv4 and IPv6 (dual stack) simultaneously until all the Internet uses IPv6
(Figure 22.11).

To determine which version to use, the source queries the DNS.

If the DNS returns an IPv4 address, the source sends an IPv4 packet.
If the DNS returns an IPv6 address, the source sends an IPv6 packet.
2) Tunneling
Tunneling is a strategy used when
two computers using IPv6 want to communicate with each other and
the packet must pass through an IPv4 network.
To pass through IPv4 network, the packet must have an IPv4 address (Figure 22.12).
So,
IPv6 packet is encapsulated in an IPv4 packet when the packet enters the IPv4 network.
IPv6 packet is decapsulated from an IPv4 packet when the packet exits the IPv4 network.

3) Header Translation

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Header translation is necessary when the majority of the Internet has moved to IPv6 but some systems still use
IPv4 (Figure 22.13).
The sender wants to use IPv6, but the receiver does not understand IPv6.
Tunneling does not work in this situation because
the packet must be in the IPv4 format to be understood by the receiver.
In this case, the header format must be totally changed through header translation.
The header of the IPv6 packet is converted to an IPv4 header/

Assignment Questions
1. Explain two types of services of WiMAX. (2)
2. Explain layers in Project 802.16. (6)
3. Explain WiMAX MAC frame format. (6*)
4. Discuss the operation of the cellular telephony. (6*)
5. Explain first generation 1G of cellular telephony. (6)
6. Explain second generation 2G of cellular telephony (6)
7. Explain third generation 3G of cellular telephony. (6*)
8. Explain fourth generation 4G of cellular telephony (6*)
9. Explain the following terms with reference to satellite (4*)
i) Orbits ii) Footprint
10. Explain the 3 categories of satellites. (8*)
11. Explain various field of IPv4. (8*)
12. Explain fragmentation. Explain 3 fields related to fragmentation (6*)
13. Explain options of IPv4. (6*)
14. Explain three network attacks to IP protocol. Also, explain four services of IPSec. (8*)
15. With general format, explain various ICMPv4 messages. (6*)
16. Explain two tools that use ICMP for debugging. (6)
17. Explain the following term with reference to Mobile IP: (4*)
i) Home address ii) Care-of address iii) Home-agent iv) Foreign-agent 18.
Explain three phases for communication in Mobile IP. (8*)
18. Explain 3 address types of IPv6. (6)
19. Explain changes from IPv4 to IPv6. (4*)
20. Explain various field of IPv6. (8*)
21. Explain various extension header of IPv6. (8)
22. Explain various ICMPv6 messages. (6)
23. Explain various group membership messages. (6)

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24. Explain 3 ways to make transition from IPv4 to IPv6. (6)

Expected Outcomes
• Explain Services of WiMAX
• Illustrate 1G, 2G, 3G generations of cellular telephony
• Explain ICMP
• Explain the IPv4 and IPv6, also their comparisons

Further Reading

1. http://www.mhhe.com/engcs/compsci/forouzan/frontmatter.pdf
2. http://freecomputerbooks.com/Introduction-to-Data-Communications.html

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