IMTnotes

Sub[ect: lnLerneL & MulLlmedla 1echnology Sem - 7th Sub[ect Code: 322711(22)
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As we have already studied computer networks, so we can understand various protocols, their working &
requirement.
We also know about operating system, basic hardware and software concepts, and computer architecture.
This knowledge gained from previous semesters will help us here to understand Internet and its internal
working.
The various protocols used to support wired communication as well as wireless communication.
Internet is a boon for modern society communication but it may become curse if do not learn about security
issues for Internet
In our previous semesters we have learn about types of media for communication
Here we have two separate units to deal with two types of communication respectively: wireless and wired
communication and protocols supporting them
The last two units are dedicated to Multimedia: use of multiple media such as audio, video, text, image,
graphics etc. for transfer of information over Internet in more presentable manner.
RELATION BETWEEN INTERNET & MULTIMEDIA:
1. Once upon a time, not so long ago, the words Internet and multimedia were rarely mentioned in the same
sentence. Although you could download GIF images or sound files from an FTP site and then view or listen to
them on your PC, the Internet experience itself was far from a multimedia extravaganza. Indeed, until the
advent of Mosaic and the phenomenal popularity of the World Wide Web, accessing the Internet was like
reading the front page of the Wall Street Journal: lots of good information, but gray, without pictures, and dull
on the eyes.
2. In 1993, a computer program called Mosaic changed all that. Mosaic is a browser -- a program that allows
users to use the Internet's World Wide Web. For the first time, true multimedia -- the mixing of various media
such as text, images, sounds, and movies -- came to the Internet. Today, not only can you download those
sorts of files, but you can also experience them while you are online. And, if you have anything to say, you
can even present your information, complete with mixed media, on your own Web page.
3. The World Wide Web continues to grow in popularity, but most of us have limited bandwidth resources. We
use poky 9600 bps and 14.4 Kbps modems to send and receive data, but in the world of full multimedia we're
going to need much faster access. After all, 14.4 Kbps means 14,400 bits of information every second, and
even with good data compression technology, we're lucky to hit 38,800 bits per second regularly. At these
speeds, video or audio files that are more than a few minutes long can take an hour or more to transfer to a
PC, so if you're waiting to see Gone with the Wind or hear Wagner's entire Ring cycle, forget it. Even users
who are lucky enough to access the Internet with a 28.8 Kbps modem get tired of waiting for things to
download.
4. As a result of this bottleneck, most people get only text and graphics files from the Web. Text and still image
files are generally small, so you don't need to wait too long to view them, but anyone who has waited for a
graphically heavy Web site, such as Time Warner's Pathfinder, soon realizes how frustrating even this
experience can be. Although audio and animation are both possible on the Web, you need a much faster
connection (or the patience of a saint) to send and receive the huge audio and video files that would enable
you to take full advantage of Internet multimedia.
5. On the Internet, and typically in real life, new technologies are first available to a core group of inventors and
experimenters. If the new technology is good enough, or interesting enough, or worthwhile enough, word gets
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out. Other folks begin to hear about the wonders of the new technology, and they want to try it. They find out
what they need, and then they spend whatever time and money is necessary. Slowly, the technology gains
wider and broader acceptance, with more and more people taking part, until at last it becomes so common that
it's practically a household word. Consider, for example, electronic mail. Or the World Wide Web.
6. What's interesting about the Internet these days is that new technologies don't get much of a experimental
period. The Net had more than a decade to shake itself out, from the time of its inception in 1969 to the
beginnings of its widespread use in the early 1980s. After that, especially beginning in the late 1980s, new
technologies started moving from someone's brain to common use in a matter of a couple of years or even
less. Mosaic, the famous World Wide Web browser, is an excellent example. First introduced in February
1993, and solely for UNIX's X Window platform, Mosaic became an important program for thousands of
UNIX, Windows and Macintosh users by the end of 1993, and it became probably the most written-about
computer program in the world by the middle of 1994. If you want to invent an Internet technology today,
you'd better plan to introduce it less than a year after you start working on it, or the technology will likely be
out of date by the time it hits the Net.
7. Nevertheless, important technologies take time to develop. Sometimes, a technology is so complex that years
of research are needed to get it to work at all. Other times, the technology demands so much of related
technologies that only a few people in the world have the equipment necessary to even get an inkling of what
the technology is about. Over the past few years, something called the MBONE has been making its way onto
the Internet slowly and experimentally. But its use is about to increase exponentially, because even those of
us who today rely on mere modems will soon have access to the technologies that are necessary to bring the
MBONE into our homes and offices.
8. Today, "regular Internet users" are at the cusp of a new multimedia revolution. Users who are pushing the
limits of a 14.4 Kbps connection can already use some cool new multimedia tools. In the coming months, true
multimedia will become more commonplace for "regular users" as bandwidth limitations decrease and as
hackers continue to improve compression methods for stuffing more information down that thin 14.4 Kbps
link. Then, too, 28.8 Kbps modems will soon be cheap enough to replace the 14.4's completely, helping to
ease the data bottleneck even further.

Since images, video graphics etc take up more space in memory so compression methods should be learned.
Last unit deals with operating system support to multimedia, MMDBMS and instruction sets.
PR00RAM E00cA1l0NAL 0B1Ec1lVE8 (PE08) AN0 c00R8E 0B1Ec1lVE8:
Introduction to Internet, to understand and learn about evolution of Internet with its applications in daily-life.

It covers wireless and wire-based communication systems & concepts and security issues.

Understand the detailed inner workings of various protocols.

To learn about high quality material over the web, including graphics, animation, sound and movie clips and
all multimedia.

To understand various audio & video formats.

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8YLLAB08 0l8c088l0N:
BOOK FROM BOOK-BANK:
TCP/IP Protocol Suite By Behrouz A. Forouzan. TMH. - UNIT - 1
OTHER BOOKS:
ATM Network by Rainer Handel, Manfred N Huber Stijan Schoder, LPE. - UNIT - 2
Multimedia Concept & Practice, Hartman & Carey, PHI - UNIT - 4&5
Wireless communication, Theodore S. Rappaport - UNIT - 3
UNIT-I Theory of Internet:-
Introduction, Evolution of Internet, Internet applications, Internet Protocol: TCP/IP Protocol, Versions, Class full
addressing, IP data gram, ICMP & IGMP. Functions of ARP and RARP, User Data gram Protocol (UDP),
Transmission Control Protocol (TCP): Flow-Control, Error-Control. Internet Security & Firewalls.
UNIT-II Bounded Media for Internet:-
Cable media, Telephone network, ISDN: Overview, Interfaces & functions, Physical Layer, Data Link Layer,
Network Layer Services, Signaling System Number 7. ATM & B-ISDN: Introduction Services & Applications,
Principles & building blocks of B-ISDN, DIAS network.
UNIT-III Un-Bounded Media for Internet:-
Wireless media: Components and working of Wireless network, IEEE 802.11 standards and WLAN types, Ad-hoc
networks, MACAW Protocol. Features and Goals of Bluetooth, Bluetooth products and security, TCP over Wireless
& Ipv6: Mobile IP, support of Mobility on the Internet, Mobile TCP, Traffic Routing in Wireless Networks, Circuit
switched Data Services, Packet switched Data services. WLL Architecture, WLL Technologies and frequency
spectrum, Local Multipoint Distribution Service (LMDS), Ultra Wideband Technology.
UNIT-IV Introduction to Multimedia:-
Concept of Non-Temporal and Temporal Media. Hypertext and Hypermedia. Presentations: Synchronization, Events,
Scripts and Interactivity, Compression Techniques: Basic concepts of Compression. Still Image Compression: JPEG
Compression. Features of JPEG2000. Video Compression: MPEG- 1&2 Compression Schemes, MPEG-4 Natural
Video Compression. Audio Compression: Introduction to speech and Audio Compression, MP3 Compression
Scheme. Compression of synthetic graphical objects.
UNIT-V Multimedia Systems Technology:-
Architecture for Multimedia Support: Multimedia PC/Workstation Architecture, Characteristics of MMX instruction
set, I/O systems: IEEE 1394 interface, Operating System Support for Multimedia Data: Resource Scheduling with real
time considerations, File System, I/O Device Management. Multimedia Information Management: Multimedia
Database Design, Content Based Information Retrieval: Image Retrieval, Video Retrieval, Overview of MPEG-7,
Design of Video-on demand systems.
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lN1R000c1l0N:
The Internet is a global system of interconnected computer networks that use the standard Internet Protocol
Suite (TCP/IP) to serve billions of users worldwide.
It is a network of networks that consists of millions of private, public, academic, business, and government
networks of local to global scope that are linked by a broad array of electronic and optical networking
technologies.
The Internet carries a vast array of information resources and services, most notably the inter-linked hypertext
documents of the World Wide Web (WWW) and the infrastructure to support electronic mail.
The terms Internet and World Wide Web are often used in everyday speech without much distinction.
However, the Internet and the World Wide Web are not one and the same.
The Internet is a global data communications system. It is a hardware and software infrastructure that
provides connectivity between computers. In contrast, the Web is one of the services communicated via the
Internet. It is a collection of interconnected documents and other resources, linked by hyperlinks and URLs.
The term the Internet, when referring to the Internet has traditionally been treated as a proper noun and
written with an initial capital letter. There is a trend to regard it as a generic term or common noun and thus
write it as "the internet", without the capital.
A network is a group of connected, communicating devices such as computers and printers. An internet
(note the lowercase i) is two or more networks that can communicate with each other. The most notable
internet is called the Internet (uppercase I), composed of hundreds of thousands of interconnected networks.
Private individuals as well as various organizations such as government agencies, schools, research facilities,
corporations, and libraries in more than 100 countries use the Internet. Millions of people are users. Yet this
extraordinary communication system only came into being in 1969.
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The birth of the Internet can be traced to a small government project in the United States of America way
back in 1970s.
It was born from the Advanced Research Projects Agency (ARPA) network called the ARPANET.
The ARPANET had several small computers called Interface Message Processors (IMPs) which were
connected to each other through modems and leased lines that facilitated exchange of data between different
computers via packet switching.
As the news spread about ARPANET, more and more computers got connected to it gradually increasing its
size and laying the seed for the Internet.
Back in those days one could connect to another computer through protocols such as telnet and FTP using a
terminal window. The telnet or FTP commands had to manually typed in at the prompt - there was no user
interface.
To gain access to a remote system one either needed to know the username and password or one was
restricted to only the public directories - directories that were not protected and were thus, open to all. And if
you didn't have an idea of how to locate a file, you had to go through each directory listing and check the file
names (assuming that the file name described its contents)!
The major growth of the Internet came with the development of HTML, the Hyper Text Markup Language,
and programs (browsers) that could read and display those documents.
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This gave rise to the World Wide Web (commonly known as WWW). Nowadays HTML documents, also
called web pages, in addition to text, can also contain images, movie clips, sound clips, animations and much
more.
During its short history, the Internet has grown exponentially. Even at this very moment as you are reading
tons of web pages and web sites are being added to this global virtual web.
With the advent of easy to use WYSIWYG (What You See Is What You Get ) editors the techniques of
creating a web site and putting it online has reached the hands of the common man
People are using the Internet not only for daily tasks such as checking and sending emails (communication)
and searching for information but are also creating their personal and business web sites or writing their hearts
out on a blog.
The Internet is now a global network of networks. Which means it consists of many smaller networks. The
number of computers linked on these smaller networks can range from 2-3 in a small Intranet to thousands of
machines in big organizations.
No one knows the exact number of computers connected to the Internet, because this figure keeps changing
and is increasing with each hour.
Tracing back in time, we can divide the history of the Internet (till the present) into three main parts.
1. FTP: The first stage
2. Gopher: The second stage
3. The World Wide Web: The third stage
In mid 1960s computers were unable to communicate with one another (i.e. they were stand-alone devices).
ARPA of U.S. DoD was interested to find a way to connect computers together to reduce costs and efforts.
In 1967 ARPANET a small network of connected computers was presented as an idea and in 1969 it was
implemented.
In 1972, Vint Cerf and Bob Kahn, part of ARPANET, devised a gateway to serve as an intermediary
hardware to transfer data from a host on one network to a host on another network.
In 1973 development of TCP/IP suite started and in 1977 internet was tested using TCP/IP.

lN1ERNE1 APPLlcA1l0N8 (AN0 08E8):
Internet banking & shopping
Information sharing & retrieval
Online education
Business
Insurance
Shares
Sales & marketing
Entertainment (online gaming)
Mailing system
Multimedia applications
Commercial use
Research work
Security
Ticket booking
Online storage
Social networking
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Email
News
Remote login-telnet
File transfer-FTP
WWW

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The Transmission Control Protocol/Internet Protocol (TCP/IP) protocol suite is the engine for the Internet and
networks worldwide.
The words internetwork and internet are simply a contraction of the phrase interconnected network. However, when
written with a capital I, the Internet refers to the worldwide set of interconnected networks. Therefore, the Internet
is an internet, but the reverse does not apply.
The main design goal of TCP/IP was to build an interconnection of networks, referred to as an internetwork, or
internet, that provided universal communication services over heterogeneous physical networks.
The layered model that dominated data communication and networking literature before 1990 was the Open Systems
Interconnection (OSI) model. Everyone believed that the OSI model would become the ultimate standard for data
communicationsbut this did not happen. The TCP/IP protocol suite became the dominant commercial architecture
because it was used and tested extensively in the Internet; the OSI model was never fully implemented.
THE OSI MODEL:
Established in 1947, the International Standards Organization (ISO) is a multinational body dedicated to
worldwide agreement on international standards. Almost three-fourths of countries in the world are represented in the
ISO. An ISO standard that covers all aspects of network communications is the Open Systems Interconnection (OSI)
model. It was first introduced in the late 1970s. An open system is a set of protocols that allows any two different
systems to communicate regardless of their underlying architecture. The purpose of the OSI model is to show how to
facilitate communication between different systems without requiring changes to the logic of the underlying hardware
and software. The OSI model is not a protocol; it is a model for understanding and designing a network architecture
that is flexible, robust, and interoperable. The OSI model was intended to be the basis for the creation of the protocols
in the OSI stack. The OSI model is a layered framework for the design of network systems that allows communication
between all types of computer systems. It consists of seven separate but related layers, each of which defines a part of
the process of moving information across a network. Understanding the fundamentals of the OSI model provides a
solid basis for exploring data communications:

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Layered Architecture:
The OSI model is composed of seven ordered layers: physical (layer 1), data link (layer 2), network (layer 3),
transport (layer 4), session (layer 5), presentation (layer 6), and application (layer 7). Figure shows the layers involved
when a message is sent from device A to device B. As the message travels from A to B, it may pass through many
intermediate nodes. These intermediate nodes usually involve only the first three layers of the OSI model. In
developing the model, the designers distilled the process of transmitting data to its most fundamental elements. They
identified which networking functions had related uses and collected those functions into discrete groups that became
the layers. Each layer defines a family of functions distinct from those of the other layers. By defining and localizing
functionality in this fashion, the designers created an architecture that is both comprehensive and flexible. Most
important, the OSI model allows complete interoperability between otherwise incompatible systems. Within a single
machine, each layer calls upon the services of the layer just below it. Layer 3, for example, uses the services provided
by layer 2 and provides services for layer 4. Between machines, layer x on one machine logically communicates with
layer x on another machine. This communication is governed by an agreed-upon series of rules and conventions called
protocols.

Layer-to-Layer Communication
In Figure device A sends a message to device B (through intermediate nodes). At the sending site, the message is
moved down from layer 7 to layer 1. At layer 1 the entire package is converted to a form that can be transferred to the
receiving site. At the receiving site, the message is moved up from layer 1 to layer 7.
Interfaces between Layers
The passing of the data and network information down through the layers of the sending device and back up through
the layers of the receiving device is made possible by an interface between each pair of adjacent layers. Each
interface defines what information and services a layer must provide for the layer above it. Well-defined interfaces
and layer functions provide modularity to a network. As long as a layer provides the expected services to the layer
above it, the specific implementation of its functions can be modified or replaced without requiring changes to the
surrounding layers.
Organization of the Layers
The seven layers can be thought of as belonging to three subgroups. Layers 1, 2, and 3physical, data link, and
networkare the network support layers; they deal with the physical aspects of moving data from one device to
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another (such as electrical specifications, physical connections, physical addressing, and transport timing and
reliability). Layers 5, 6, and 7
they allow interoperability among unrelated software systems. Layer 4, the transport layer, links the two subgroups
and ensures that what the lower layers have transmitted is in a form that the upper layers can use.

8ummary of 0
Following figure shows a summary of duties for each layer. Some of these duties are mixed and spread into five
categories in the TCP/IP protocol suite:
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The
suite do not match exactly with those in the OSI model.
software layers built upon the
named similarly to

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8ummary of 0

The TCP/IP protocol suite
named similarly to

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8ummary of 08l Layers

TCP/IP protocol suite
named similarly to the ones in the OSI model.
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reliability). Layers 5, 6, and 7session, presentation, and application
8l Layers:
TCP/IP protocol suite was developed pri
software layers built upon the hardware. Today, however, TCP/IP is thought of a
the ones in the OSI model.
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session, presentation, and application

was developed prior to the OSI model.
hardware. Today, however, TCP/IP is thought of a
the ones in the OSI model.
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session, presentation, and application
or to the OSI model.
suite do not match exactly with those in the OSI model. The original TCP/IP protocol suite was defined as four
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session, presentation, and applicationcan be thought of as th
or to the OSI model. Therefore, the
The original TCP/IP protocol suite was defined as four
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can be thought of as th


Therefore, the layers in the TCP/IP protocol
hardware. Today, however, TCP/IP is thought of as a five-layer model with the layers
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can be thought of as the user support layers;

layers in the TCP/IP protocol
layer model with the layers
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e user support layers;
e user support layers;
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Comparison between OSI and TCP/IP Protocol Suite:

When we compare the two models, we find that two layers, session and presentation, are missing from the TCP/IP
protocol suite. These two layers were not added to the TCP/IP protocol suite after the publication of the OSI model.
The application layer in the suite is usually considered to be the combination of three layers in the OSI model. Two
reasons were mentioned for this decision. First, TCP/IP has more than one transport-layer protocol. Some of the
functionalities of the session layer are available in some of the transport layer protocols. Second, the application layer
is not only one piece of software. Many applications can be developed at this layer. If some of the functionalities
mentioned in the session and presentation are needed for a particular application, it can be included in the
development of that piece of software. TCP/IP is a hierarchical protocol made up of interactive modules, each of
which provides a specific functionality, but the modules are not necessarily interdependent. Whereas the OSI model
specifies which functions belong to each of its layers, the layers of the TCP/IP protocol suite contain relatively
independent protocols that can be mixed and matched, depending on the needs of the system. The term hierarchical
means that each upper level protocol is supported by one or more lower level protocols.
ADDRESSING:

Four levels of addresses are used in an internet employing the TCP/IP protocols: physical (link) address, logical
address, port address, and application-specific address. Each address is related to a one layer in the TCP/IP
architecture.
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TCP/IP and OSI model:

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0lFFERENcE BE1WEEN 1cP/lP PR010c0L 80l1E AN0 08l M00EL:
S.NO. OSI MODEL TCP/IP MODEL
1 7 layers

4 layers

2 Distinguishes and defines between services,
interfaces and protocols very clearly.

Did not clearly distinguish between services, interfaces
and protocols

3 Came after TCP/IP model

Came into existence first.

4 Strict layering

Loosely layered

5 Less credible

More credible

6 Came before the advent of Internet hence does not
support internetworking.

Came after the advent of Internet hence supports
internetworking
7 Protocol dependent standard

Protocol independent standard

8 Protocols are better hidden and hence can be
easily replaced as the technology changes.

Protocols not hidden properly and hence it is not easy
to replace the protocols
9 Supports both CL and CO communication in the
network layer but only CO communication in the
transport layer.

Supports only CL communication in the network layer
but CL and CO communication in the transport layer.

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VER8l0N8:
In May 1974, the Institute of Electrical and Electronic Engineers (IEEE) published a paper entitled "A
Protocol for Packet Network Intercommunication." The paper's authors, Vint Cerf and Bob Kahn, described
an internetworking protocol for sharing resources using packet-switching among the nodes. A central control
component of this model was the "Transmission Control Program" (TCP) that incorporated both connection-
oriented links and datagram services between hosts. The monolithic Transmission Control Program was later
divided into a modular architecture consisting of the Transmission Control Protocol at the connection-
oriented layer and the Internet Protocol at the internetworking (datagram) layer. The model became known
informally as TCP/IP, although formally referenced as the Internet Protocol Suite.
The Internet Protocol is one of the elements that define the Internet. The dominant internetworking protocol in
the Internet Layer in use today is IPv4; the number 4 is the protocol version number carried in every IP
datagram.
The successor to IPv4 is IPv6. Its most prominent modification by IETF from version 4 in the addressing
system. In this version, only network layer protocols are changed. ICMPv4 became ICMPv6, IGMP and ARP
are merged into ICMPv6 and RARP is deleted. IPv6 is also known as IPng. IPv4 uses 32-bit (4 byte)
addresses (with the address space divided into different classes. In future it cannot handle the projected
number of users since the Internet is growing rapidly )(c. 4 billion, or 4.310
9
, addresses) while IPv6 uses
128-bit (16 byte) addresses (c. 340 undecillion, or 3.410
38
addresses) and hence can accommodate a larger
number of users.IPv6 supports authentication, data integrity and confidentiality at the network layer and is
more flexible to allow for the future addition of features.
Although adoption of IPv6 has been slow, as of June 2008, all United States government systems have
demonstrated basic infrastructure support for IPv6 (if only at the backbone level). IP versions 0 to 3 were
development versions of IPv4 and were used between 1977 and 1979. IP Version 5 was used by the Internet
Stream Protocol, an experimental streaming protocol. It was based on the OSI model. It never went beyond
the proposal stage due to extensive layer changes and the projected expense. Version numbers 6 through 9
were proposed for various protocol models designed to replace IPv4: SIPP (Simple Internet Protocol Plus,
known now as IPv6), TP/IX (RFC 1475), PIP (RFC 1621) and TUBA (TCP and UDP with Bigger Addresses,
RFC 1347).
IPv6 is designed to handle the transmission of real-time data such as audio and video and can carry data from
other protocols. It can also handle congestion and route discovery better than the current version.
Other protocol proposals named IPv9 and IPv8 briefly surfaced, but have no support.
Draw header format of IPv4 and IPv6

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IP ADDRESS & ADDRESS SPACE:
The identifier used in the IP layer of the TCP/IP protocol suite to identify each device connected to the Internet is
called the Internet address or IP/logical/network address. An IPv4 address is a 32-bit address that uniquely and
universally defines the connection of a host or a router to the Internet; an IP address is the address of the interface.
IPv4 addresses are unique. They are unique in the sense that each address defines one, and only one, connection to the
Internet. Two devices on the Internet can never have the same address at the same time. However, if a device has two
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connections to the
that the addressing system must be accepted by any host that wants to be
Address Space
A protocol like IPv4 that defines addresses has an
used by the
two different values (0 or 1).
(more tha
the
classful addressing.
supersedes the original architectur
Classes
In classful addressing, the IP address space is divided into five
part of the
Recognizing Classes
We can find t
notation, the first few bits can immediately tell us the class of the address; in the dotted
the first byte can give

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connections to the
Address Space
used by the protocol. If a protocol uses
(more than four billion). Theoretically,
the Internet. IP addresses, when started a few decades ago, used the concept of
classful addressing.
Classes:
part of the whole address space. Figure shows the
Recognizing Classes
We can find the class of an address when the address is given either in binary or dotted decimal notation. In the binary
the first byte can give


connections to the Internet, via two networks, it has two IPv4 addresses. The IPv4 addresses are
Address Space
protocol. If a protocol uses
n four billion). Theoretically,
IP addresses, when started a few decades ago, used the concept of
classful addressing. In the
whole address space. Figure shows the
Recognizing Classes:
he class of an address when the address is given either in binary or dotted decimal notation. In the binary
the first byte can give the class of an address.
Internet, via two networks, it has two IPv4 addresses. The IPv4 addresses are
protocol. If a protocol uses
two different values (0 or 1). IPv4 uses 32
n four billion). Theoretically,
In the mid-1990s, a new architecture,
supersedes the original architecture.
whole address space. Figure shows the
the class of an address.
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A protocol like IPv4 that defines addresses has an address space.
protocol. If a protocol uses b bits to define an
IPv4 uses 32-bit addresses, which means that the address space is 2
if there were no restrictions, more than 4 billion
1990s, a new architecture,
whole address space. Figure shows the class occupation of the address space.
the class of an address.
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address space.
bits to define an address, the address space is 2
bit addresses, which means that the address space is 2
1990s, a new architecture,
In classful addressing, the IP address space is divided into five classes: A, B, C, D,
class occupation of the address space.
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that the addressing system must be accepted by any host that wants to be connected
address space. An address space is
address, the address space is 2
1990s, a new architecture, called classless addressing,
classes: A, B, C, D,
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connected to the Internet.
An address space is the total
address, the address space is 2
b

if there were no restrictions, more than 4 billion devices could be connected to
IP addresses, when started a few decades ago, used the concept of classes. This architecture
classless addressing,
classes: A, B, C, D, and E.
notation, the first few bits can immediately tell us the class of the address; in the dotted-decimal notation, the value of
Internet, via two networks, it has two IPv4 addresses. The IPv4 addresses are universal
to the Internet.
the total number of addresses
because each b
32
or 4,294,967,296
devices could be connected to
. This architecture
classless addressing, was introduced that
E. Each class occupies some
decimal notation, the value of
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universal in the sense
number of addresses
because each bit can have
or 4,294,967,296
. This architecture is called
was introduced that
class occupies some


in the sense
number of addresses
it can have
or 4,294,967,296
is called
was introduced that
class occupies some
LLC1UkL NC1LS
14

Netid and Hostid:
In classful addressing, an IP address in classes A, B, and C is divided into netid and hostid. These parts are of varying
lengths, depending on the class of the address. Figure shows the netid and hostid bytes. Note that classes D and E are
not divided into netid and hostid,

In class A, 1 byte defines the netid and 3 bytes define the hostid. In class B, 2 bytes define the netid and 2 bytes define
the hostid. In class C, 3 bytes define the netid and 1 byte defines the hostid.
Classes and Blocks
One problem with classful addressing is that each class is divided into a fixed number of blocks with each block
having a fixed size.
There are three main problems with class ful addressing, which are somewhat related to each other:
1. Lack of Internal Address Flexibility: Big organizations are assigned large, monolithic blocks of addresses
that don't match well the structure of their underlying internal networks.
2. Inefficient Use of Address Space: The existence of only three block sizes (classes A, B and C) leads to
waste of limited IP address space.
3. Proliferation of Router Table Entries: As the Internet grows, more and more entries are required for routers
to handle the routing of IP datagrams, which causes performance problems for routers. Attempting to reduce
inefficient address space allocation leads to even more router table entries.
Class A
Since only 1 byte in class A defines the netid and the leftmost bit should be 0, the next 7 bits can be changed to find
the number of blocks in this class. Therefore, class A is divided into 2
7
= 128 blocks that can be assigned to 128
organizations (the number is less because some blocks were reserved as special blocks). However, each block in this
class contains 16,777,216 addresses, which means the organization should be a really large one to use all these
addresses. Many addresses are wasted in this class.

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1S

Blocks in class A:

Class B
Since 2 bytes in class B define the class and the two leftmost bit should be 10 (fixed), the next 14 bits can be changed
to find the number of blocks in this class. Therefore, class B is divided into 2
14
= 16,384 blocks that can be assigned to
16,384 organizations (the number is less because some blocks were reserved as special blocks). However, each block
in this class contains 65,536 addresses. Not so many organizations can use so many addresses. Many addresses are
wasted in this class
Blocks in class B:

Class C
Since 3 bytes in class C define the class and the three leftmost bits should be 110 (fixed), the next 21 bits can be
changed to find the number of blocks in this class. Therefore, class C is divided into 2
21
=2,097,152 blocks, in which
each block contains 256 addresses, that can be assigned to 2,097,152 organizations (the number is less because some
blocks were reserved as special blocks). Each block contains 256 addresses. However, not so many organizations
were so small as to be satisfied with a class C block.
Blocks in class C:

Class D
There is just one block of class D addresses. It is designed for multicasting. Each address in this class is used to define
one group of hosts on the Internet. When a group is assigned an address in this class, every host that is a member of
this group will have a multicast address in addition to its normal (unicast) address.
The single block in Class D:

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16

Class E
There is just one block of class E addresses. It was designed for use as reserved Addresses
The single block in Class E:

Two-Level Addressing
The whole purpose of IPv4 addressing is to define a destination for an Internet packet (at the network layer). When
classful addressing was designed, it was assumed that the whole Internet is divided into many networks and each
network connects many hosts. In other words, the Internet was seen as a network of networks. A network was
normally created by an organization that wanted to be connected to the Internet. The Internet authorities allocated a
block of addresses to the organization (in class A, B, or C). The range of addresses allocated to an organization in
classful addressing was a block of addresses in Class A, B, or C. Since all addresses in a network belonged to a
single block, each address in classful addressing contains two parts: netid and hostid. The netid defines the network;
the hostid defines a particular host connected to that network. If n bits in the class defines the net, then 32 n bits
defines the host. However, the value of n depends on the class the block belongs to. The value of n can be 8, 16 or 24
corresponding to classes A, B, and C respectively.
Two-level addressing in classful addressing:

Disadvantages of Classful Addressing:
Many addresses are wasted in class A and B and class C addresses are insufficient for needs of any organization.

Extracting Information in a Block
A block is a range of addresses. Given any address in the block, we normally like to know three pieces of information
about the block: the number of addresses, the first address, and the last address. Before we can extract these pieces of
information, we need to know the class of the address, which we showed how to find in the previous section. After the
class of the block is found, we know the value of n, the length of netid in bits. We can now find these three pieces of
information
1. The number of addresses in the block, N, can be found using N = 2
32-n
.
2. To find the first address, we keep the n leftmost bits and set the (32 n) rightmost bits all to 0s.
3. To find the last address, we keep the n leftmost bits and set the (32 n) rightmost bits all to 1s.

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17

Information extraction in classful addressing:

Network Address
The first address, or beginning address or the network address, is particularly important because it is used
in routing a packet to its destination network. For the moment, let us assume that an internet is made of m networks
and a router with m interfaces. When a packet arrives at the router from any source host, the router needs to know to
which network the packet should be sent; the router needs to know from which interface the packet should be sent
out. When the packet arrives at the network, it reaches its destination host. The network address is actually the
identifier of the network; each network is identified by its network address.

Network Mask
A network mask or a default mask in classful addressing is a 32-bit number with n leftmost bits all set to 1s and (32
n) rightmost bits all set to 0s. Since n is different for each class in classful addressing, we have three default masks
in classful addressing
Network / Default mask:

To extract the network address from the destination address of a packet, a router uses the AND operation When the
destination address (or any address in the block) is ANDed with the default mask, the result is the network
address. The router applies the AND operation on the binary (or hexadecimal representation) of the address and the
mask, the default mask can also be used to find the number of addresses in the block and the last address in the block.

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Finding a network address using the default mask:

Three-Level Addressing: Subnetting (for classes A and B):
The idea of splitting a block to smaller blocks is referred to as subnetting. In subnetting, a network is divided into
several smaller subnetworks (subnets) with each subnetwork having its own subnetwork address.
Subnet Mask:
The network mask is used when a network is not subnetted. When we divide a network to several subnetworks, we
need to create a subnetwork mask (or subnet mask) for each subnetwork. A subnetwork has subnetid and hostid as
shown
Network mask and subnetwork mask:

Subnetting increases the length of the netid and decreases the length of hostid.
Subnet Address:
When a network is subnetted, the first address in the subnet is the identifier of the subnet and is used by the router to
route the packets destined for that subnetwork. Given any address in the subnet, the router can find the subnet mask
using the same procedure we discussed to find the network mask: ANDing the given address with the subnet mask

Supernetting (for class C)
Subnetting could not completely solve address depletion problems in classful addressing because most organizations
did not want to share their granted blocks with others. Since class C blocks were still available but the size of the
block did not meet the requirement of new organizations that wanted to join the Internet, one solution was
supernetting. In supernetting, an organization can combine several class C blocks to create a larger range of
addresses. In other words, several networks are combined to create a supernetwork. By doing this, an organization can
apply for several class C blocks instead of just one. For example, an organization that needs 1000 addresses can
be granted four class C blocks.
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19

Supernet Mask
A supernet mask is the reverse of a subnet mask. A subnet mask for class C has more 1s than the default mask for
this class. A supernet mask for class C has less 1s than the default mask for this class. Unfortunately, supernetting
provided two new problems: First, the number of blocks to combine needs to be a power of 2, which means an
organization that needed seven blocks should be granted at least eight blocks (address wasting). Second, supernetting
and subnetting really complicated the routing of packets in the Internet.
Comparison of subnet, default, and supernet masks:

A subnet mask that divides a block into eight subblocks has three more 1s (2
3
= 8) than the default mask; a supernet
mask that combines eight blocks into one superblock has three less 1s than the default mask.

Disadvantages Of Supernetting And Subnetting:

Unfortunately, supernetting and subnetting provided two new problems: First, the number of blocks to combine needs
to be a power of 2, which means an organization that needed seven blocks should be granted at least eight blocks
(address wasting). Second, supernetting and subnetting really complicated the routing of packets in the Internet.

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LEc10RE # 4:
lP 0A1A0RAM:
HEADER FORMAT:

Version (VER)
This 4-bit field defines the version of the IP protocol. Currently the version is 4. However, version 6 (or IPv6) may
totally replace version 4 in the future. This field tells the IP software running in the processing machine that the
datagram has the format of version 4. All fields must be interpreted as specified in the fourth version of the protocol.
If the machine is using some other version of IP, the datagram is discarded rather than interpreted incorrectly.
Header length (HLEN)
This 4-bit field defines the total length of the datagram header in 4-byte words. This field is needed because the length
of the header is variable (between 20 and 60 bytes). When there are no options, the header length is 20 bytes, and the
value of this field is 5 (5 4 = 20). When the option field is at its maximum size, the value of this field is 15 (15 4 =
60).
Service type:
In the original design of IP header, this field was referred to as type of service (TOS) , which defined how the
datagram should be handled. Part of the field was used to define the precedence of the datagram; the rest defined the
type of service (low delay, high throughput, and so on). IETF has changed the interpretation of this 8-bit field. This
field now defines a set of differentiated services.
In this interpretation, the first 6 bits make up the codepoint subfield and the last 2 bits are not used. The codepoint
subfield can be used in two different ways:
LLC1UkL NC1LS
21

a.
When the 3 right-most bits are 0s, the 3 left-most bits are interpreted the same as the precedence bits in the service
type interpretation. In other words, it is compatible with the old interpretation. The precedence defines the eight-level
priority of the datagram (0 to 7) in issues such as congestion. If a router is congested and needs to discard some
datagrams, those datagrams with lowest precedence are discarded first. Some datagrams in the Internet are more
important than the others. For example, a datagram used for network management is much more urgent and important
than a datagram containing optional information for a group.
b.
When the 3 right-most bits are not all 0s, the 6 bits define 56 (64-8) services based on the priority assignment by the
Internet or local authorities. The first category contains 24 service types; the second and the third each contain 16. The
first category is assigned by the Internet authorities (IETF). The second category can be used by local authorities
(organizations). The third category is temporary and can be used for experimental purposes. Note that these
assignments have not yet been finalized.
Total length:
This is a 16-bit field that defines the total length (header plus data) of the IP datagram in bytes. To find the length of
the data coming from the upper layer, subtract the header length from the total length. The header length can be found
by multiplying the value in the HLEN field by four. Since the field length is 16 bits, the total length of the IP
datagram is limited to 65,535 (2
16
- 1) bytes, of which 20 to 60 bytes are the header and the rest is data from the upper
layer.
Length of data = total length - header length
Identification
This field is used in fragmentation
Flags.
Fragmentation offset.
Time to live:
A datagram has a limited lifetime in its travel through an internet. This field was originally designed to hold a
timestamp, which was decremented by each visited router. The datagram was discarded when the value became zero.
However, for this scheme, all the machines must have synchronized clocks and must know how long it takes for a
datagram to go from one machine to another. Today, this field is mostly used to control the maximum number of hops
(routers) visited by the datagram. When a source host sends the datagram, it stores a number in this field. This value is
approximately two times the maximum number of routes between any two hosts. Each router that processes the
datagram decrements this number by one. If this value, after being decremented, is zero, the router discards the
LLC1UkL NC1LS
22

datagram. This field is needed because routing tables in the Internet can become corrupted. A datagram may travel
between two or more routers for a long time without ever getting delivered to the destination host. This field limits the
lifetime of a datagram. Another use of this field is to intentionally limit the journey of the packet. For example, if the
source wants to confine the packet to the local network, it can store 1 in this field. When the packet arrives at the
first router, this value is decremented to 0, and the datagram is discarded.
Protocol:
This 8-bit field defines the higher-level protocol that uses the services of the IP layer. An IP datagram can encapsulate
data from several higher level protocols such as TCP, UDP, ICMP, and IGMP. This field specifies the final
destination protocol to which the IP datagram should be delivered. In other words, since the IP protocol multiplexes
and demultiplexes data from different higher-level protocols, the value of this field helps in the demultiplexing
process when the datagram arrives at its final destination.
Checksum:
The error detection method used by most TCP/IP protocols is called the checksum. The checksum protects against the
corruption that may occur during the transmission of a packet. It is redundant information added to the packet. The
checksum is calculated at the sender and the value obtained is sent with the packet. The receiver repeats the same
calculation on the whole packet including the checksum. If the result is satisfactory, the packet is accepted; otherwise,
it is rejected.
Source address:
This 32-bit field defines the IP address of the source. This field must remain unchanged during the time the IP
datagram travels from the source host to the destination host.
Destination address:
This 32-bit field defines the IP address of the destination. This field must remain unchanged during the time the IP
datagram travels from the source host to the destination host.
Fragmentation:
For some physical networks, we must divide the datagram to make it possible to pass through these networks. This is
called fragmentation. The source usually does not fragment the IP packet. The transport layer will instead segment
the data into a size that can be accommodated by IP and the data link layer in use. When a datagram is fragmented,
each fragment has its own header with most of the fields repeated, but some changed. A fragmented datagram may
itself be fragmented if it encounters a network with an even smaller MTU. In other words, a datagram can be
fragmented several times before it reaches the final destination. A datagram can be fragmented by the source host or
any router in the path. The reassembly of the datagram, however, is done only by the destination host because each
fragment becomes an independent datagram. Whereas the fragmented datagram can travel through different routes,
and we can never control or guarantee which route a fragmented datagram may take, all of the fragments belonging to
the same datagram should finally arrive at the destination host. So it is logical to do the reassembly at the final
destination. An even stronger objection for reassembling packets during the transmission is the loss of efficiency it
incurs.

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23

When a datagram is fragmented, required parts of the header must be copied by all fragments. The option field may or
may not be copied as we will see in the next section. The host or router that fragments a datagram must change the
values of three fields: flags, fragmentation offset, and total length. The rest of the fields must be copied. Of course, the
value of the checksum must be recalculated regardless of fragmentation.
Fields Related to Fragmentation:
Identification:
This 16-bit field identifies a datagram originating from the source host. The combination of the identification and
source IP address must uniquely define a datagram as it leaves the source host. To guarantee uniqueness, the IP
protocol uses a counter to label the datagrams. The counter is initialized to a positive number. When the IP protocol
sends a datagram, it copies the current value of the counter to the identification field and increments the counter by
one. As long as the counter is kept in the main memory, uniqueness is guaranteed. When a datagram is fragmented,
the value in the identification field is copied into all fragments. In other words, all fragments have the same
identification number, which is also the same as the original datagram. The identification number helps the destination
in reassembling the datagram. It knows that all fragments having the same identification value should be assembled
into one datagram.
Flags:
This is a three-bit field. The first bit is reserved (not used). The second bit is called the do not fragment bit. If its value
is 1, the machine must not fragment the datagram. If it cannot pass the datagram through any available physical
network, it discards the datagram and sends an ICMP error message to the source host If its value is 0, the datagram
can be fragmented if necessary. The third bit is called the more fragment bit. If its value is 1, it means the datagram is
not the last fragment; there are more fragments after this one. If its value is 0, it means this is the last or only
fragment.

Fragmentation offset.
This 13-bit field shows the relative position of this fragment with respect to the whole datagram. It is the offset of the
data in the original datagram measured in units of 8 bytes. Figure 7.8 shows a datagram with a data size of 4000 bytes
fragmented into three fragments. The bytes in the original datagram are numbered 0 to 3999. The first fragment
carries bytes 0 to 1399. The offset for this datagram is 0/8 = 0. The second fragment carries bytes 1400 to 2799; the
offset value for this fragment is 1400/8 = 175. Finally, the third fragment carries bytes 2800 to 3999. The offset value
for this fragment is 2800/8 = 350.
Options
The header of the IP datagram is made of two parts: a fixed part and a variable part. The fixed part is 20 bytes long
The variable part comprises the options, which can be a maximum of 40 bytes. Options, as the name implies, are not
required for a datagram. They can be used for network testing and debugging. Although options are not a required
part of the IP header, option processing is required of the IP software. This means that all implementations must be
able to handle options if they are present in the header.
LLC1UkL NC1LS
24

LEc10RE # 6:
lcMP & l0MP:
lnternet control Message Protocol Verslon 4 (lcMPv4):
Position of ICMP in the network layer:

Messages:
ICMP messages are divided into two broad categories: error-reporting messages and query messages. The error-
reporting messages report problems that a router or a host (destination) may encounter when it processes an IP packet.
The query messages, which occur in pairs, help a host or a network manager get specific information from a router or
another host.

Message Format:
An ICMP message has an 8-byte header and a variable-size data section. Although the general format of the header is
different for each message type, the first 4 bytes are common to all. The first field, ICMP type, defines the type of the
message. The code field specifies the reason for the particular message type. The last common field is the checksum
field. The rest of the header is specific for each message type. The data section in error messages carries information
for finding the original packet that had the error. In query messages, the data section carries extra information based
on the type of the query.
General format of ICMP messages:
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2S

An ICMP message has an 8-byte header and a variable-size data section. Although the general format of the header
is different for each message type, the first 4 bytes are common to all.
Error Reporting Messages:
One of the main responsibilities of ICMP is to report errors. Although technology has produced increasingly reliable
transmission media, errors still exist and must be handled. IP is an unreliable protocol. This means that error checking
and error control are not a concern of IP. ICMP was designed, in part, to compensate for this shortcoming. However,
ICMP does not correct errors, it simply reports them. Error correction is left to the higher-level protocols. Error
messages are always sent to the original source because the only information available in the datagram about the route
is the source and destination IP addresses. ICMP uses the source IP address to send the error message to the source
(originator) of the datagram. Five types of errors are handled: destination unreachable, source quench, time exceeded,
parameter problems, and redirection.
Error-reporting messages:

The following are important points about ICMP error messages:
No ICMP error message will be generated in response to a datagram carrying an ICMP error message.
No ICMP error message will be generated for a fragmented datagram that is not the first fragment.
No ICMP error message will be generated for a datagram having a multicast address.
No ICMP error message will be generated for a datagram having a special address such as 127.0.0.0 or
0.0.0.0.
Always from destination to source.
Destination Unreachable:
When a router cannot route a datagram or a host cannot deliver a datagram, the datagram is discarded and the router
or the host sends a destination-unreachable message back to the source host that initiated the datagram.

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26

Source Quench:
The IP protocol is a connectionless protocol. There is no communication between the source host, which produces the
datagram, the routers, which forward it, and the destination host, which processes it. One of the ramifications of this
absence of communication is the lack of flow control and congestion control.
The source-quench message in ICMP was designed to add a kind of flow control and congestion control to the IP.
When a router or host discards a datagram due to congestion, it sends a source-quench message to the sender of the
datagram. This message has two purposes. First, it informs the source that the datagram has been discarded. Second, it
warns the source that there is congestion somewhere in the path and that the source should slow down (quench) the
sending process.
Time Exceeded:
The time-exceeded message is generated in two cases:
Whenever a router decrements a datagram with a time-to-live value to zero, it discards the datagram and
sends a time-exceeded message to the original source.
When the final destination does not receive all of the fragments in a set time, it discards the received
fragments and sends a time-exceeded message to the original source.

Parameter Problem:
Any ambiguity in the header part of a datagram can create serious problems as the datagram travels through the
Internet. If a router or the destination host discovers an ambiguous or missing value in any field of the datagram, it
discards the datagram and sends a parameter-problem message back to the source.

Redirection:
When a router needs to send a packet destined for another network, it must know the IP address of the next
appropriate router. The same is true if the sender is a host. Both routers and hosts then must have a routing table to
find the address of the router or the next router. Routers take part in the routing update process and are supposed to be
updated constantly. Routing is dynamic. However, for efficiency, hosts do not take part in the routing update process
because there are many more hosts in an internet than routers. Updating the routing tables of hosts dynamically
produces unacceptable traffic. The hosts usually use static routing. When a host comes up, its routing table has a
limited number of entries. It usually knows only the IP address of one router, the default router. For this reason, the
host may send a datagram, which is destined for another network, to the wrong router. In this case, the router that
receives the datagram will forward the datagram to the correct router. However, to update the routing table of the host,
it sends a redirection message to the host.
EXAMPLE:
Host A wants to send a datagram to host B. Router R2 is obviously the most efficient routing choice, but host A did
not choose router R2. The datagram goes to R1 instead. R1, after consulting its table, finds that the packet should have
gone to R2. It sends the packet to R2 and, at the same time, sends a redirection message to host A. Host As routing
table can now be updated.
A host usually starts with a small routing table that is gradually augmented and updated. One of the tools to
accomplish this is the redirection message.

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27

Query Messages:
In addition to error reporting, ICMP can also diagnose some network problems. This is accomplished through the
query messages. A group of five different pairs of messages have been designed for this purpose, but three of these
pairs are deprecated today. Only two pairs are used today: echo request and replay and timestamp request and replay.
In this type of ICMP message, a node sends a message that is answered in a specific format by the destination node.
Echo Request and Reply
The echo-request and echo-reply messages are designed for diagnostic purposes. Network managers and users
utilize this pair of messages to identify network problems. The combination of echo-request and echo-reply messages
determines whether two systems (hosts or routers) can communicate with each other. A host or router can send an
echo-request message to another host or router. The host or router that receives an echo-request message creates an
echo-reply message and returns it to the original sender.
Timestamp Request and Reply
Two machines (hosts or routers) can use the timestamp-request and timestamp-reply messages to determine the
round-trip time needed for an IP datagram to travel between them. It can also be used to synchronize the clocks in two
machines.
The timestamp-request and timestamp-reply messages can be used to compute the one-way or round-trip time
required for a datagram to go from a source to a destination and then back again. The formulas are:
sending time = receive timestamp - original timestamp
receiving time = returned time - transmit timestamp
round-trip time = sending time + receiving time
Deprecated Messages
Three pairs of messages are declared obsolete by IETF:
1. Information request and replay messages are not used today because their duties are done by Address Resolution
Protocol (ARP)
Sub[ect:

2.

3.

lN1ERNE1 0R00P
Position of IGMP in the network laye
IGMP is used by IP hosts to register their dynamic
to discover these group members.
GROUP MANAGEMENT
IGMP is a protocol that manages group membership. The IGMP protocol gives the multic
about the membership status of hosts (routers) connected to the network.
IGMP Messages
IGMP has three types of messages: the query, the membership report, and the leave report. There are two types of
query messages, general and spec
ICMP messages are sent in IP datagrams. The IP header will always have a Protocol number of 2, indicating IGMP
and a Type of Service of zero (routine). The IP data field will contain the 8

28
CSl1/

2. Address mask request and reply
configuration Protocol
3. Router solicitation and advertisement
Configuration Protocol (DHCP).
lN1ERNE1 0R00P
GROUP MANAGEMENT
IGMP Messages


Address mask request and reply
uration Protocol (DHCP).
Router solicitation and advertisement
lN1ERNE1 0R00P MANA0EMEN1 PR010c0L
GROUP MANAGEMENT
IGMP Messages
Address mask request and reply messages are not used today because their duties are
(DHCP).
Router solicitation and advertisement
MANA0EMEN1 PR010c0L
GROUP MANAGEMENT:
query messages, general and special.
LLC1UkL NC1LS
messages are not used today because their duties are
Router solicitation and advertisement messages are not used today because their
MANA0EMEN1 PR010c0L (l0MP
Position of IGMP in the network layer:
IGMP is used by IP hosts to register their dynamic multicast group membership. It
LLC1UkL NC1LS
messages are not used today because their
l0MP):
multicast group membership. It
Sem - 7th
LLC1UkL NC1LS
messages are not used today because their
multicast group membership. It
and a Type of Service of zero (routine). The IP data field will contain the 8-byte IGMP message
Sub[ect Code:
messages are not used today because their duties are done by Dynamic Host
messages are not used today because their duties are done by Dynamic Host
multicast group membership. It is also used by connected routers
byte IGMP message

done by Dynamic Host
duties are done by Dynamic Host

is also used by connected routers
IGMP is a protocol that manages group membership. The IGMP protocol gives the multicast routers information
byte IGMP message

322711(22)
done by Dynamic Host
ast routers information
ast routers information
LLC1UkL NC1LS
29

IGMP MESSAGE FORMAT:

IGMP Operation:
A multicast router connected to a network has a list of multicast addresses of the groups with at least one loyal
member in that network. For each group, there is one router that has the duty of distributing the multicast packets
destined for that group.

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30

Membership Report
Host or a router can join a group.
A host maintains a list of processes that have membership in a group
When a process wants to join a new group, it sends its request to the host
The host then adds the name of the process and the name of the requested group to its list, and sends the
membership report to the router. The report is send twice (in case the first report get lost or damaged).

Leave Report
When a host sees that no process is interested in a specific group G, it sends a leave report. If router receives a leave
report it wont purge the list if there are still other hosts interested in that group. For that purpose the router sends a
special query message with a specified response time for the group in question to see if there is anyone interested in
that group. If there is no response a membership report, it purges the list.

General Query Message
Membership report and leave report are not enough to maintain the membership information. Example: a host that is a
member of a group can shut down and the m/c router would never receive the leave report. Therefore the m/c router
monitors the hosts and routers in LAN by periodically sending (by default every 125 sec) general query message.
Hosts/routers respond by membership report if there is still interest in groups

Delayed Response
In order to keep the traffic low the response to general query message must be done by only one host for a given
group. How can be made sure that only one host answers the query, while the others which have to report the same
group, or groups are not? This is achieved with delayed response:
When a host receives general query message it delays the response: it sets a timer for each group to a different random
value between 0 and 10 seconds, then broadcasts the response(s) according to the timers. If the host receives a
response from another host, whose timer for that group has expired earlier, the host cancels the corresponding timer
and doesnt send the duplicate response for the group. Only one router on the LAN is designated for sending the query
messages the query router. This further reduces the traffic.
LEc10RE # 6:
F0Nc1l0N8 0F ARP AN0 RARP:
ADDRESS MAPPING:
An internet is made of a combination of physical networks connected together by internetworking devices such as
routers. A packet starting from a source host may pass through several different physical networks before finally
reaching the destination host. The hosts and routers are recognized at the network level by their logical addresses.
A logical address is an internetwork address. Its jurisdiction is universal. A logical address is unique universally. It is
called a logical address because it is usually implemented in software. Every protocol that deals with interconnecting
networks requires logical addresses. The logical addresses in the TCP/IP protocol suite are called IP addresses and
are 32 bits long. However, packets pass through physical networks to reach these hosts and routers. At the physical
level, the hosts and routers are recognized by their physical addresses. A physical address is a local address. Its
jurisdiction is a local network. It should be unique locally, but not necessarily universally. It is called a physical
address because it is usually (but not always) implemented in hardware. Examples of physical addresses are 48-bit
MAC addresses in the Ethernet protocol, which are imprinted on the NIC installed in the host or router. The physical
LLC1UkL NC1LS
31

address and the logical address are two different identifiers. We need both of them because a physical network such as
Ethernet can have two different protocols at the network layer such as IP and IPX (Novell) at the same time.
Likewise, a packet at a network layer such as IP may pass through different physical networks such as Ethernet and
LocalTalk (Apple). This means that delivery of a packet to a host or a router requires two levels of addressing: logical
and physical. We need to be able to map a logical address to its corresponding physical address and vice versa. These
can be done using either static or dynamic mapping.
Static Mapping
Static mapping means creating a table that associates a logical address with a physical address. This table is stored in
each machine on the network. Each machine that knows, for example, the IP address of another machine but not its
physical address can look it up in the table. This has some limitations because physical addresses may change in the
following ways:
1.A machine could change its NIC, resulting in a new physical address.
2.In some LANs, such as LocalTalk, the physical address changes every time the computer is turned on.
3. A mobile computer can move from one physical network to another, resulting in a change in its physical address.
To implement these changes, a static mapping table must be updated periodically. This overhead could affect
network performance.
Dynamic Mapping
In dynamic mapping, each time a machine knows the logical address of another machine; it can use a protocol to
find the physical address. Two protocols have been designed to perform dynamic mapping:
Address Resolution Protocol (ARP) and Reverse Address Resolution Protocol (RARP). ARP maps a logical
address to a physical address; RARP maps a physical address to a logical address:

LLC1UkL NC1LS
32

P08l1l0N 0F ARP AN0 RARP lN 1cP/lP PR010c0L 80l1E:

Address Resolutlon Protocol (ARP)
Anytime a host or a router has an IP datagram to send to another host or router, it has the logical (IP) address of the
receiver. But the IP datagram must be encapsulated in a frame to be able to pass through the physical network. This
means that the sender needs the physical address of the receiver. A mapping corresponds a logical address to a
physical address. Figure shows the position of the ARP in the TCP/IP protocol suite. ARP accepts a logical address
from the IP protocol, maps the address to the corresponding physical address and pass it to the data link layer.

ARP associates an IP address with its physical address. On a typical physical network, such as a LAN, each device on
a link is identified by a physical or station address that is usually imprinted on the NIC.
Packet Format
Figure shows the format of an ARP packet. The fields are as follows:

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Hardware type.
This is a 16-bit field defining the type of the network on which ARP is running. Each LAN has been assigned an
integer based on its type. For example, Ethernet is given the type 1. ARP can be used on any physical network.
Protocol type.
This is a 16-bit field defining the protocol. For example, the value of this field for the IPv4 protocol is 0800
16
. ARP
can be used with any higher-level protocol.
Hardware length.
This is an 8-bit field defining the length of the physical address in bytes. For example, for Ethernet the value is 6.
Protocol length.
This is an 8-bit field defining the length of the logical address in bytes. For example, for the IPv4 protocol the value is
4.
Operation.
This is a 16-bit field defining the type of packet. Two packet types are defined: ARP request (1), ARP reply (2).
Sender hardware address.
This is a variable-length field defining the physical address of the sender. For example, for Ethernet this field is 6
bytes long.
Sender protocol address.
This is a variable-length field defining the logical (for example, IP) address of the sender. For the IP protocol, this
field is 4 bytes long.
Target hardware address.
This is a variable-length field defining the physical address of the target. For example, for Ethernet this field is 6 bytes
long. For an ARP request message, this field is all 0s because the sender does not know the physical address of the
target.
Target protocol address.
This is a variable-length field defining the logical (for example, IP) address of the target. For the IPv4 protocol, this
field is 4 bytes long.

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Encapsulation
An ARP packet is encapsulated directly into a data link frame. For example, in Figure an ARP packet is encapsulated
in an Ethernet frame. Note that the type field indicates that the data carried by the frame is an ARP packet.
Operation
Let us see how ARP functions on a typical internet. First we describe the steps involved.
Steps Involved
These are seven steps involved in an ARP process:
1. The sender knows the IP address of the target.
2. IP asks ARP to create an ARP request message, filling in the sender physical address, the sender IP address,
and the target IP address. The target physical address field is filled with 0s.
3. The message is passed to the data link layer where it is encapsulated in a frame using the physical address of
the sender as the source address and the physical broadcast address as the destination address.
4. Every host or router receives the frame. Because the frame contains a broadcast destination address, all
stations remove the message and pass it to ARP. All machines except the one targeted drop the packet. The
target machine recognizes the IP address.
5. The target machine replies with an ARP reply message that contains its physical address. The message is
unicast.
6. The sender receives the reply message. It now knows the physical address of the target machine.
7. The IP datagram, which carries data for the target machine, is now encapsulated in a frame and is unicast to
the destination.
An ARP request is broadcast; an ARP reply is unicast.
Encapsulation of ARP packet:

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3S

Reverse Address Resolutlon Protocol (RARP):
Used when configuration file is not available (e.g. diskless machine).
RARP packet:

08ER 0A1A0RAM PR010c0L (00P):

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Port numbers:

UDP uses port numbers to to create a process-to-process communication. Another responsibility is to provide control
mechanisms at the transport level. UDP does this task at a very minimal level. There is no flow control mechanism
and there is no acknowledgment for received packets. UDP, however, does provide error control to some extent. If
UDP detects an error in the received packet, it silently drops it. UDP is a connectionless, unreliable transport
protocol. It does not add anything to the services of IP except for providing process-to-process communication
instead of host-to-host communication.
Normally servers must have a known port number, while clients can have ephemeral (temporary) port numbers
assigned by UDP. (Ephemeros = lasting a day, Greek).

If UDP is so powerless, why would a process want to use it? With the disadvantages come some advantages. UDP is
a very simple protocol using a minimum of overhead. If a process wants to send a small message and does not care
much about reliability, it can use UDP. Sending a small message using UDP takes much less interaction between the
sender and receiver than using TCP.

Port numbers 0..216-1 = 0..65,535

Socket addresses:

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USER DATAGRAM
UDP packets, called user datagrams, have a fixed-size header of 8 bytes. Figure shows the format of a user
datagram. The fields are as follows:

Source port number.
This is the port number used by the process running on the source host. It is 16 bits long, which means that the port
number can range from 0 to 65,535. If the source host is the client (a client sending a request), the port number, in
most cases, is an ephemeral port number requested by the process and chosen by the UDP software running on the
source host. If the source host is the server (a server sending a response), the port number, in most cases, is a well-
known port number.

Destination port number.
This is the port number used by the process running on the destination host. It is also 16 bits long. If the destination
host is the server (a client sending a request), the port number, in most cases, is a well-known port number. If the
destination host is the client (a server sending a response), the port number, in most cases, is an ephemeral port
number. In this case, the server copies
the ephemeral port number it has received in the request packet.

Length.
This is a 16-bit field that defines the total length of the user datagram, header plus data. The 16 bits can define a total
length of 0 to 65,535 bytes. However, the total length needs to be much less because a UDP user datagram is stored in
an IP datagram with the total length of 65,535 bytes. The length field in a UDP user datagram is actually not
necessary. A user datagram is encapsulated in an IP datagram. There is a field in the IP datagram that defines the total
length. There is another field in the IP datagram that defines the length of the header. So if we subtract the value of
the second field from the first, we can deduce the length of the UDP datagram that is encapsulated in an IP datagram.
However, the designers of the UDP protocol felt that it was more efficient for the destination UDP to calculate the
length of the data from the information provided in the UDP user datagram rather than ask the IP software to supply
this information. We should remember that when the IP software delivers the UDP user datagram to the UDP layer, it
has already dropped the IP header.

Checksum.
This field is used to detect errors over the entire user datagram (header plus data).
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08E 0F 00P:
UDP is suitable for a process that requires simple request-response communication where flow and error
control are not crucial.
UDP is not suitable for transmitting a large amount of data (like FTP) because they are encapsulated into IP
datagrams that are independently routed.
UDP is suitable for processes that have internal flow/error control (like TFTP)
UDP is suitable for multicasting and broadcasting.
UDP is used in network management (like Simple Network Management Protocol (SNMP)
UDP is used for some route updating protocols (like RIP)
UDP is preferred in some real-time applications such as voice and telemetry involving certain degree of
redundancy (temperature measurement, streaming video,)
UDP is also preferred in applications that have short transactions like NSF, DNS, RIP.
LEc10RE # 7:
Vl0E0 LEc10RE - 1ransmlsslon control Protocol (1cP):

1cP 8ERVlcE8:
Process-to-Process Communication
Stream Delivery Service
Full-Duplex Communication
Multiplexing and Demultiplexing
Connection-Oriented Service
Reliable Service

Numbering System
Although the TCP software keeps track of the segments being transmitted or received, there is no field for a segment
number value in the segment header. Instead, there are two fields called the sequence number and the
acknowledgment number using Byte number.

Byte Number
TCP numbers all data bytes (octets) that are transmitted in a connection. Numbering is independent in each direction.
When TCP receives bytes of data from a process, TCP stores them in the sending buffer and numbers them. The
numbering does not necessarily start from 0. Instead, TCP chooses an arbitrary number between 0 and 2
32
-1 for the
number of the first byte. For example, if the number happens to be 1,057 and the total data to be sent is 6,000 bytes,
the bytes are numbered from 1,057 to 7,056.

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

Acknowledgment Number
The value of the acknowledgment field in a segment defines the number of the next byte a party expects to receive.
The acknowledgment number is cumulative.

SEGMENT:
A packet in TCP is called a segment.
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39

TCP segment format:

control fleld:

1cP c0NNEc1l0N:
In TCP, connection-oriented transmission requires three phases: connection establishment, data transfer, and
connection termination.
Connection Establishment
TCP transmits data in full-duplex mode. When two TCPs in two machines are connected, they are able to send
segments to each other simultaneously. This implies that each party must initialize communication and get approval
from the other party before any data are transferred.
Three-Way Handshaking
The connection establishment in TCP is called three-way handshaking. In our example, an application program,
called the client, wants to make a connection with another application program, called the server, using TCP as the
transport layer protocol. The process starts with the server. The server program tells its TCP that it is ready to accept a
connection. This request is called a passive open. Although the server TCP is ready to accept a connection from any
machine in the world, it cannot make the connection itself. The client program issues a request for an active open. A
client that wishes to connect to an open server tells its TCP to connect to a particular server. TCP can now start the
three-way handshaking process:
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A SYN segment cannot carry data, but it consumes one sequence number.
A SYN + ACK segment cannot carry data, but does consume one sequence number.
An ACK segment, if carrying no data, consumes no sequence number.
Data Transfer
After connection is established, bidirectional data transfer can take place. The client and server can send data and
acknowledgments in both directions.
Connection Termination
Any of the two parties involved in exchanging data (client or server) can close the connection, although it is usually
initiated by the client. Most implementations today allow two options for connection termination: three-way
handshaking and four-way handshaking with a half-close option
The FIN segment consumes one sequence number if it does not carry data.
The FIN + ACK segment consumes one sequence number if it does not carry data.
(ACK) consumes no sequence number.

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Connection Reset
TCP at one end may deny a connection request, may abort an existing connection, or may terminate an idle
connection. All of these are done with the RST (reset) flag.
Denying a Connection
Suppose the TCP on one side has requested a connection to a nonexistent port. The TCP on the other side may send a
segment with its RST bit set to deny the request.
Aborting a Connection
One TCP may want to abort an existing connection due to an abnormal situation. It can send an RST segment to close
the connection.
Terminating an Idle Connection
The TCP on one side may discover that the TCP on the other side has been idle for a long time. It may send an RST
segment to end the connection. The process is the same as aborting a connection.
LEc10RE # 8:
FL0W-c0N1R0L:
Flow control balances the rate a producer creates data with the rate a consumer can use the data. TCP separates flow
control from error control
Silly Window Syndrome
A serious problem can arise in the sliding window operation when either the sending application program creates data
slowly or the receiving application program consumes data slowly, or both. Any of these situations results in the
sending of data in very small segments, which reduces the efficiency of the operation. For example, if TCP sends
segments containing only 1 byte of data, it means that a 41-byte datagram (20 bytes of TCP header and 20 bytes of IP
header) transfers only 1 byte of user data. Here the overhead is 41/1, which indicates that we are using the capacity of
the network very inefficiently. The inefficiency is even worse after accounting for the data link layer and physical
layer overhead. This problem is called the silly window syndrome. For each site, we first describe how the problem
is created and then give a proposed solution.

Syndrome Created by the Sender
The sending TCP may create a silly window syndrome if it is serving an application program that creates data slowly,
for example, 1 byte at a time. The application program writes 1 byte at a time into the buffer of the sending TCP. If
the sending TCP does not have any specific instructions, it may create segments containing 1 byte of data. The result
is a lot of 41-byte segments that are traveling through an internet. The solution is to prevent the sending TCP from
sending the data byte by byte. The sending TCP must be forced to wait and collect data to send in a larger block. How
long should the sending TCP wait? If it waits too long, it may delay the process. If it does not wait long enough, it
may end up sending small segments. Nagle found an elegant solution.

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42

Nagles Algorithm Nagles algorithm is simple:
1. The sending TCP sends the first piece of data it receives from the sending application program even if it is only 1
byte.
2. After sending the first segment, the sending TCP accumulates data in the output buffer and waits until either the
receiving TCP sends an acknowledgment or until enough data has accumulated to fill a maximum-size segment. At
this time, the sending TCP can send the segment.
3. Step 2 is repeated for the rest of the transmission. Segment 3 is sent immediately if an acknowledgment is received
for segment 2, or if enough data have accumulated to fill a maximum-size segment.

The elegance of Nagles algorithm is in its simplicity and in the fact that it takes into account the speed of the
application program that creates the data and the speed of the network that transports the data. If the application
program is faster than the network, the segments are larger (maximum-size segments). If the application program is
slower than the network, the segments are smaller (less than the maximum segment size).

Syndrome Created by the Receiver
The receiving TCP may create a silly window syndrome if it is serving an application program that consumes data
slowly, for example, 1 byte at a time. Suppose that the sending application program creates data in blocks of 1
kilobyte, but the receiving
application program consumes data 1 byte at a time. Also suppose that the input buffer of the receiving TCP is 4
kilobytes. The sender sends the first 4 kilobytes of data. The receiver stores it in its buffer. Now its buffer is full. It
advertises a window size of zero, which means the sender should stop sending data. The receiving application reads
the first byte of data from the input buffer of the receiving TCP. Now there is 1 byte of space in the incoming buffer.
The receiving TCP announces a window size of 1 byte,
which means that the sending TCP, which is eagerly waiting to send data, takes this advertisement as good news and
sends a segment carrying only 1 byte of data. The procedure will continue. One byte of data is consumed and a
segment carrying 1 byte of
data is sent. Again we have an efficiency problem and the silly window syndrome. Two solutions have been proposed
to prevent the silly window syndrome created by an application program that consumes data slower than they arrive.

Clarks Solution Clarks solution is to send an acknowledgment as soon as the data arrive, but to announce a
window size of zero until either there is enough space to accommodate a segment of maximum size or until at least
half of the receive buffer is empty.

Delayed Acknowledgment The second solution is to delay sending the acknowledgment. This means that when a
segment arrives, it is not acknowledged immediately. The receiver waits until there is a decent amount of space in its
incoming buffer before acknowledging the arrived segments. The delayed acknowledgment prevents the sending TCP
from sliding its window. After the sending TCP has sent the data in the window, it stops. This kills the syndrome.
Delayed acknowledgment also has another advantage: it reduces traffic. The receiver does not have to acknowledge
each segment. However, there also is a disadvantage in that the delayed acknowledgment may result in the sender
unnecessarily retransmitting the unacknowledged segments. TCP balances the advantages and disadvantages. It now
defines that the acknowledgment should not be delayed by more than 500 ms.

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LEc10RE # 9:
ERR0R-c0N1R0L:

Error-control mechanlsms:

Checksum
Acknowledgment
Retransmission
Out-of-Order Segments
Internet Security & Firewalls:
Means security at the network layer. We talk about security at the transport and application layers, but we also need
security at the network layer for three reasons. First, not all client/server programs are protected at the application
layer. Second, not all client/server programs at the application layer use the services of TCP to be protected by the
transport layer security that we discuss for the transport layer; some programs use the service of UDP. Third, many
applications, such as routing protocols, directly use the service of IP; they need security services at the IP layer.

IP Security (IPSec) is a collection of protocols designed by the Internet Engineering Task Force (IETF) to provide
security for a packet at the network level. IPSec helps create authenticated and confidential packets for the IP layer.

Two Modes
IPSec operates in one of two different modes: transport mode or tunnel mode.

Transport Mode
In transport mode, IPSec protects what is delivered from the transport layer to the network layer. In other words,
transport mode protects the payload to be encapsulated in the network layer
IPSec in transport mode

Tunnel Mode
In tunnel mode, IPSec protects the entire IP packet. It takes an IP packet, including the header, applies IPSec security
methods to the entire packet, and then adds a new IP header,

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IPSec in tunnel mode

Transport mode versus tunnel mode

Two Security Protocols
IPSec defines two protocols the Authentication Header (AH) Protocol and the Encapsulating Security Payload (ESP)
Protocol to provide authentication and/or encryption for packets at the IP level.
Authentication Header (AH) protocol

Encapsulating Security Payload (ESP)

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4S

IPSec services

Security Association
Security Association is a very important aspect of IPSec. IPSec requires a logical relationship, called a Security
Association (SA), between two hosts.
Transport Layer Security:
Two protocols are dominant today for providing security at the transport layer: the Secure Sockets Layer (SSL)
protocol and the Transport Layer Security (TLS) protocol. The latter is actually an IETF version of the former
Location of SSL and TLS in the Internet model

Four Protocols
Four SSL protocols

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Handshake Protocol

APPLICATION LAYER SECURITY
This section discusses two protocols providing security services for e-mails: Pretty Good Privacy (PGP) and
Secure/Multipurpose Internet Mail Extension (S/MIME).

Applications of PGP:
PGP has been extensively used for personal e-mails. It will probably continue to be.
Applications of S/MIME
It is predicted that S/MIME will become the industry choice to provide security for commercial e-mail.
FIREWALLS
A firewall is a device (usually a router or a computer) installed between the internal network of an organization and
the rest of the Internet. It is designed to forward some packets and filter (not forward) others.
Firewall

A firewall is usually classified as
1. packet-filter firewall or
2. proxy-based firewall.
LEc10RE # 10: 1010RlAL # 1: Numerlcals on 8ubnettlng/8upernettlng and classful Addresslng.
LEc10RE # 11: A0VANcE0 8YLLAB08 c00R8E: congestlon control ln 1cP
LEc10RE # 12: REVl8l0N/80MMARY (Lc0), A88l0NMEN1 1

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47

0Nl1 ~ 2: Bounded Medla for lnternet:
LEc10RE # 13: BRl00E c00R8E: Wlred vs. Wlreless Networklng (pros & cons), lnternetworklng
devlces
LEc10RE # 14:
cABLE ME0lA:

There are two basic categories of transmission media
1. Guided
2. Unguided

Guided Transmission media uses a Cabling system that guides the data signals along a specific path. The data
signals are bound by the cabling system. Guided media is also known as Bound media.
Unguided transmission media consists of a means for the data signals to travel but nothing to guide them along a
specific path. The data signals are not bound to a cabling media and as such are after called unbounded media.
There are four basic types of Guided media
(i) Open wire
(ii) Twisted pair
(iii) Coaxial cable
(iv) Optical fiber

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48

cABLE ME0lA:
TRANSMISSION MEDIA:
The means through which data is transformed from one place to another is called transmission or communication
media. There are two categories of transmission media used in computer communications.
BOUNDED/GUIDED MEDIA
UNBOUNDED/UNGUIDED MEDIA
BOUNDED MEDIA:
Bounded media are the physical links through which signals are confined to narrow path. These are also called guide
media. Bounded media are made up o a external conductor (Usually Copper) bounded by jacket material. Bounded
media are great for LABS because they offer high speed, good security and low cast. However, some time they cannot
be used due distance communication. Three common types of bounded media are used of the data transmission. These
are
Coaxial Cable
Twisted Pairs Cable
Fiber Optics Cable
COAXIAL CABLE:

Coaxial cable is very common & widely used commutation media. For example TV wire is usually coaxial.
Coaxial cable gets its name because it contains two conductors that are parallel to each other. The center conductor in
the cable is usually copper. The copper can be either a solid wire or stranded martial.
Outside this central Conductor is a non-conductive material. It is usually white, plastic material used to separate the
inner Conductor form the outer Conductor. The other Conductor is a fine mesh made from Copper. It is used to help
shield the cable form EMI.
Outside the copper mesh is the final protective cover.
The actual data travels through the center conductor in the cable. EMI interference is caught by outer copper mesh.
There are different types of coaxial cable vary by gauge & impedance.
Sub[ect:

Gauge is the measure of the cable thickness. It is measured by the Radio grade measurement, or RG number. The high
the RG number, the thinner the central conductor core,
CHARACTERISTICS OF COAXIAL CABLE
ADVANTAGES COAXIAL CABLE
DISADVANTAGE COAXIAL CABLE
STP
UTP
Twisted Pair Cable
49
CSl1/

Low cost
Easy to install
Up to 10Mbps capacity
Medium immunity form EMI
Medium of attenuation
Inexpensive
Easy to wire
Easy to expand
Moderate level of
Single cable failure can take down an entire network
STP
UTP
Twisted Pair Cable

Low cost
Easy to install
Inexpensive
Easy to wire
Easy to expand
Moderate level of EMI immunity
Twisted Pair Cable
EMI immunity
LLC1UkL NC1LS

EMI immunity

LLC1UkL NC1LS
the RG number, the thinner the central conductor core, the lower the number the thicker the core.
Sem - 7th
LLC1UkL NC1LS
the lower the number the thicker the core.
Sub[ect Code:

322711(22)
Gauge is the measure of the cable thickness. It is measured by the Radio grade measurement, or RG number. The high Gauge is the measure of the cable thickness. It is measured by the Radio grade measurement, or RG number. The high
LLC1UkL NC1LS
S0

The most popular network cabling is Twisted pair. It is light weight, easy to install, inexpensive and support many
different types of network. It also supports the speed of 100 mps. Twisted pair cabling is made of pairs of solid or
stranded copper twisted along each other. The twists are done to reduce vulnerably to EMI and cross talk. The number
of pairs in the cable depends on the type. The copper core is usually 22-AWG or 24-AWG, as measured on the
American wire gauge standard. There are two types of twisted pairs cabling
1. Unshielded twisted pair (UTP)
2. Shielded twisted pair (STP)
1. Unshielded twisted pair (UTP)
UTP is more common. It can be either voice grade or data grade depending on the condition. UTP cable normally has
an impedance of 100 ohm. UTP cost less than STP and easily available due to its many use. There are five levels of
data cabling
Category 1
These are used in telephone lines and low speed data cable.
Category 2
These cables can support up to 4 mps implementation.
Category 3
These cable supports up to 16 mps and are mostly used in 10 mps.
Category 4
These are used for large distance and high speed. It can support 20mps.
Category 5
This is the highest rating for UTP cable and can support up to 100mps.
UTP cables consist of 2 or 4 pairs of twisted cable. Cable with 2 pair use RJ-11 connector and 4 pair cable use RJ-45
connector.
Characteristics of UTP
low cost
easy to install
High speed capacity
High attenuation
Effective to EMI
100 meter limit
LLC1UkL NC1LS
S1

Advantages of UTP
Easy installation
Capable of high speed for LAN
Low cost
Disadvantages of UTP
Short distance due to attenuation
2. Shielded twisted pair (STP)
It is similar to UTP but has a mesh shielding thats protects it from EMI which allows for higher transmission rate.
IBM has defined category for STP cable.
Type 1
STP features two pairs of 22-AWG
Type 2
This type include type 1 with 4 telephone pairs
Type 6
This type feature two pairs of standard shielded 26-AWG
Type 7
This type of STP consist of 1 pair of standard shielded 26-AWG
Type 9
This type consist of shielded 26-AWG wire

Characteristics of STP
Medium cost
Easy to install
Higher capacity than UTP
Higher attenuation, but same as UTP
Medium immunity from EMI
100 meter limit

LLC1UkL NC1LS
S2

Advantages of STP:
Shielded
Faster than UTP and coaxial
Disadvantages of STP:
More expensive than UTP and coaxial
More difficult installation
High attenuation rate
Fiber Optics

Fiber optic cable uses electrical signals to transmit data. It uses light. In fiber optic cable light only moves in one
direction for two way communication to take place a second connection must be made between the two devices. It is
actually two stands of cable. Each stand is responsible for one direction of communication. A laser at one device
sends pulse of light through this cable to other device. These pulses translated into 1s and 0s at the other end.
In the center of fiber cable is a glass stand or core. The light from the laser moves through this glass to the other
device around the internal core is a reflective material known as CLADDING. No light escapes the glass core
because of this reflective cladding.
Fiber optic cable has bandwidth more than 2 gbps (Gigabytes per Second).
Propagation modes

Characteristics Of Fiber Optic Cable:
Expensive
Very hard to install
Capable of extremely high speed
Extremely low attenuation
No EMI interference
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Advantages Of Fiber Optic Cable:
Fast
Low attenuation
No EMI interference
Disadvantages Fiber Optics:
Very costly
Hard to install
LEc10RE # 16:
1ELEPR0NE NE1W0RK:
Public switched telephone network:
The public switched telephone network (PSTN) is the network of the world's public circuit-switched telephone
networks. It consists of telephone lines, fiber optic cables, microwave transmission links, cellular networks,
communications satellites, and undersea telephone cables, all inter-connected by switching centers, thus allowing any
telephone in the world to communicate with any other. Originally a network of fixed-line analog telephone systems,
the PSTN is now almost entirely digital in its core and includes mobile as well as fixed telephones.
Circuit switching Technique:
Communication via circuit switching implies that there is a dedicated communication path between the two stations.
The path is a connected through a sequence of links between network nodes. On each physical link, a logical channel
is dedicated to the connection. Circuit switching is commonly used technique in telephony, where the caller sends a
special message with the address of the callee (i.e. by dialling a number) to state its destination. It involved the
following three distinct steps:

Circuit Establishment:
To establish an end-to-end connection before any transfer of data.
Some segments of the circuit may be a dedicated link, while some other segments may be shared.

Data transfer:
Transfer data is from the source to the destination.
The data may be analog or digital, depending on the nature of the network.
The connection is generally full-duplex.

Circuit disconnect:

Terminate connection at the end of data transfer.

Signals must be propagated to deallocate the dedicated resources

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FIGURE: circuit switching technique:
Thus the actual physical electrical path or circuit between the source and destination host must be established before
the message is transmitted. This connection, once established, remains exclusive and continuous for the complete
duration of information exchange and the circuit becomes disconnected only when the source wants to do so.
Public switched telephone network (PSTN) is an example of circuit-switched network. Its also known as Plain
Old Telephone Service (POTS). The switching centres used for the switching are organised in different levels,
namely: Regional offices (class 1), Section offices (class 2), primary offices (class 3), Toll offices (class 4) and finally
End offices (class 5) as shown in Figure. Level 1 is at the highest level and Level 5 is the lowest level. Subscribers or
the customers are directly connected to these end offices. And each office is connected directly to a number of offices
at a level below and mostly a single office at higher level.
Subscriber Telephones are connected, through Local Loops to end offices (or central offices). A small town may have
only one end office, but large cities have several end offices. Many end offices are connected to one Toll office, which
are connected to primary offices. Several primary offices are connected to a section office, which normally serves
more than one state. All regional offices are connected using mesh topology. Accessing the switching station at the
end offices is accomplished through dialling. In the past, telephone featured rotary or pulse dialling, in which digital
signals were sent to the end office for each dialled digit. This type of dialling was prone to errors due to inconsistency
in humans during dialling. Presently, dialling is accomplished by Touch-Tone technique. In this method the user
sends a small burst of frequency called dual tone, because it is a combination of two frequencies. This combination of
frequencies sent depends on the row and column of the pressed pad.

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Figure: Basic organization of a Public Switched Telephone Network (PSTN):

The connections are multiplexed when have to send to a switching office, which is one level up. For example,
Different connections will be multiplexed when they are to be forwarded from an end-office to Toll office. Figure
shows a typical medium distance telephone circuit:

Figure: Typical medium distance telephone circuit:

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Q-1. What are the three basic steps involved in data communication through circuit switching?
Ans: The steps are:
1. Circuit establishment (before data transfer)
2. Circuit maintenance (When data transfer is going on)
3. Circuit disconnect (When data transfer is over)

Q-2. Mention the key advantages and disadvantages of circuit switching technique.
Ans: Advantages:
1. After path is established, data communication without delay.
2. Very suitable for continuous traffic.
3. It establishes a dedicated path.
4. No overhead after call setup.
5. It is transparent and data passes in order.

Disadvantages:
1. Provide initial delay for setting up the call.
2. Inefficient for bursty traffic.
3. Data rate should be same because of fixed bandwidth.
4. When load increases, some calls may be blocked.

0lA8 NE1W0RK:

The Direct Internet Access System (DIAS) is a technology used to access internet through DSL, jointly developed by
Banyan Networks and TeNeT Group, lIT Madras, allows telecom service providers to provide high
bandwidth Internet access to residential and corporate subscribers, in addition to voice services, without any changes
to the existing cabling infrastructure. In contrast to current residential PSTN (Public Switched Telephone Network)
and ISDN (Integrated Switched Digital Network) dial-up access, the DIAS provides an Always On Internet
Access that is permanently available at the customers premises. Using Digital Subscriber Loop (DSL) techniques,
seamless voice and data connectivity is provided to the customer over the same pair of copper wires. Implementation
of this system is often done using the existing cables. All that is required is the installation of the IAN (Integrated
Access Node) at the Telephone exchange and a DSU (Digital Subscriber Unit) at the customer premises.

DIAS Architecture:
The DIAS has a Digital Subscriber Unit (DSU) that combines voice and Internet data packets on a single twisted-pair
wire at the subscribers premises. At the service providers premises, the Integrated Access Node (IAN) separates
voice and data traffic from a number of subscribers and routes them independently to the PSTN and the Internet
respectively. Apart from this DIAS supports a Lifeline option, which provides a simple telephone service in the case
of power failure or any problem in the customer premises modem.
The IAN is connected to the PSTN via an E1 voice port when V5.2 protocol is running between the DIAS and the
PSTN switch, and to the Internet either through E1 data ports or through an Ethernet port. In the case of V5.2 protocol
the lifeline unit is a separate unit will be catering for 240 Lines. Alternatively the PSTN connectivity can be achieved
through POTS lines with the addition of an optional 2-wire lifeline unit that converts a single E1 line to 30 POTS
lines. This unit is capable of support 120 subscribers per sub rack
The DIAS system as shown in figure below provides two types of voice and data services to the customer.

The BDSU (Basic Digital Subscriber Unit) is designed for the SOHO (Small Office Home Office) and residential
Internet user. It provides a permanent Internet connection at a maximum data rate of 128 kbps, which drops to 64 kbps
when the telephone is in use and transparently, goes back to 128 kbps when the telephone goes on-hook.
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The HDSU (High bit rate Digital Subscriber Unit), which is designed for corporate subscribers, can provide voice
connectivity for a telephone and permanent data connectivity of upto 2 Mbps.

The Basic Rate DSU (BDSU) located at the subscriber's premises has a telephone interface (RJ11) and an Ethernet
port (RJ45) or a Serial Port to provide Internet Access. The High Bit Rate DSU (HDSU) has an Ethernet port and 4/8
Telephone Interfaces (RJ11), thus having the ability to connect to 4 or 8 independent telephones at a corporate office.
The BDSU has local powering off the AC Mains (230V). In the case of Power failure the system will offer simple
POTS connection. The Ethernet port providing Internet access is off during power failure.

The BDSU and the HDSU are connected to the Integrated Access Node, located either at a street corner (curb) or at
the central office, using a twisted pair copper wire. For the BDSU, the maximum length of the copper can be 3.5 km
when 0.4- mm twisted pair copper is used. 128 kbps Internet access can be provided on the BDSU Ethernet port in
such a configuration when the telephone is not being used. The Internet access rate seamlessly drops to 64kbps when
the BDSU telephone is being used providing the telephone a 64kbps circuit-switched access. The HDSU is also
connected using copper to the IAN. The maximum rate at which Internet Access is provided to the HDSU Ethernet
port is 2 Mbps, and this is possible when the length of the copper is less than 2 Km (0.4mm twisted pair copper). The
bit-rate on the HDSUIAN link drops for higher lengths of copper, thus reducing Internet access rate on the HDSU
Ethernet port. Each telephone on the HDSU uses 64kbps when off-hook and reduces the bit-rate of the HDSU
Internet access by the same amount. In on-hook mode, the Internet access rate seamlessly reverts to the original value

Integrated Access Node (IAN):

The IAN is built to cater 60 subscribers in a single 19-inch sub rack. The highly scalable architecture of the DIAS
makes it possible to stack more than one IAN s in a single rack to increase subscriber capacity. The entire stack is
manageable through the local interface or over network using standard SNMP based management tools. The DIAS
has its own SNMP manager. The IAN houses the central processor, the E1uplink and line cards. All line cards have
hot insert/removal capability for easy field maintenance. IAN has powerful set of diagnostics and extensive remote
management capabilities. The IAN provides optional power feed over the network to subscriber that are within 1 km
from IAN.

ADVANTAGES OF DIAS:

Elegant, easy to install state-of-the-art technology, with soft architecture to protect against obsolence.
Instant Internet connection and simultaneous voice
Seamless voice-data connectivity with High Bandwidth on virtual leased line up to2 mbps.
Aggregating long duration data calls at the IAN thereby decongesting the trunk network of Basic Service
Provider.
Flexibility of deployment depending on the subscriber density of area.
Cascadable for more capacity.

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LEc10RE # 16:
l80N: 0VERVlEW, lN1ERFAcE8 & F0Nc1l0N8:
A conceptual Vlew of l80N:
Looking at the concepts of ISDN from the following viewpoints:
1. Principles of ISDN
2. Evolution of the ISDN
3. The User Interface
4. Objectives
5. Benefits
6. Services
7. Architecture

Principles of ISDN
Support of voice and non-voice applications using a limited set of standardized facilities
defines the purpose of ISDN and the means of achieving it
Support for switched and non-switched applications
both circuit-switched and packet-switched connections
also supports non-switched services in the form of dedicated lines
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Reliance on 64-kbps connections
fundamental block of ISDN
chosen because it was the standard rate for digitized voice
Intelligence in the network
sophisticated services beyond simple setup a circuit-switched call
sophisticated network management and maintenance capabilities
use of SS7 and intelligent switching nodes in the network
Layered protocol architecture
user access to ISDN protocol is a layered architecture that can be mapped to OSI model
Already developed standards for OSI may be used for ISDN (e.g. X.25)
New ISDN standard can be based on existing ones (LAPD based on LAPB)
Standards can be developed independently for various layers and functions
Variety of configurations
More than one physical configuration is possible for implementing ISDN
ISDN evolved from IDN but
I in IDN Integration of digital transmission and switching facilities, whereas
I in ISDN Integration of voice and data transmission services.

Evolution of ISDN
Evolution from telephone IDNs
IDN developed for and evolving within existing telephone network.
The telephone networks have the dominant role in ISDN (not PSN, sat.).
Transition of one or more decades
The introduction of ISDN services will be done in the context of existing digital facilities and existing services
(protocol conversion, etc.)
Use of existing networks
Now the interface to packet-switched services is X.25. With the introduction of fast PS and VC control, new
interface will be needed.
Interim user-network arrangements
The lack of digital subscriber loops might delay digital services.
With the use of modem, etc. existing analog facilities support ISDN serv.
Connection at other than 64 kbps
The basic channel for circuit switching
The rate is unnecessarily high with new voice digitizing technologies
The rate is too low for many digital data applications
The User Interface:
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User has access to ISDN via a local interface to a digital pipe.
Pipes of various sizes are available to satisfy different needs.
Pipe to the users promises has a fixed capacity but the traffic on the pipe may be a variable mix up to the capacity
limit.
ISDN requires control signals to instruct how to sort out the time-multiplexed data and provide the required services.
Control signals are multiplexed onto the same digital pipe.
A user may employ less than the maximum capacity of the pipe and will be charged according to the capacity used.
Objectives:
Standardization
to provide universal access to the network
movable ISDN-standard equipment and use of layered protocol
to allow users to select equipment from multiple suppliers
Transparency
digital transmission service is independent of the contents of data
users can develop new applications and protocols
user-provided encryption methods can be employed simply
Separation of competitive functions
ISDN does not preclude competitive functions from basic ones.
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some countries desire certain enhanced services be offered competitively (e.g., videotex, electronic mail)
Leased and switched services
ISDN should provide both leased and switched services
to allow users to optimize on the basis of cost and performance
Cost-related tariff
one type of service does not subsidize others
price distinctions related to the cost of providing specific performance and functional characteristics of a service
Smooth migration
ISDN evolution must coexist with existing equipment and services
specific capabilities required: pre-ISDN terminals to interface to ISDN, inter-network protocols, and protocol
converters

Multiplexed support
accommodating user-owned PBX and LAN equipment
ISDN Benefits and Services:
Benefits:
The principle benefits of ISDN to the customers can be expressed in terms of cost saving and flexibility.
Integrated voice and data means that the user does not have to buy multiple services to meet multiple needs.
Access charges to a single access line only
purchasing services based on actual needs
product diversity, low price, and wide availability of equipment
slow changes in ISDN recommendations; low risk for users

Services:
A variety of voice and data applications are available:
Facsimile: fast digital facsimile standards
Teletex: fast exchange of correspondences between terminals
Videotex: an interactive information retrieval service
Most of these services can be provided with a transmission rate of 64 kbps or less. For higher rates B-ISDN will be
used.
ISDN is an intelligent network.
By use of a flexible signaling protocol, ISDN provides a variety of network facilities for each service.

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ISDN Architecture:

ISDN supports a new physical connector for users, a digital subscriber line, and a variety of transmission services.
Physical interface provides a standardized means of attaching to the network.
The interface supports a basic service consisting of three time multiplexed channels, two at 64 kbps and one at 16
kbps.
In addition, there is a primary service that provides multiple 64- kbps channels.
For both basic and primary service, an interface is defined between the customers equipment (TE) and a device on
The customers premises, known as a network termination (NT).
The subscriber line is the physical path from the subscribers NT to the ISDN central office.
ISDN central office connects subscriber lines to the digital network, providing access to lower-layer transmission
facilities:
Circuit-switched capabilities
same facility provided by other digital-switched telecom. nets (64 kbps)
Non-switched capabilities
a 64 kbps dedicated link, higher rates in B-ISDN using PVC in ATM tx
Switched capabilities
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high speed (>64 kbps) switched connections using ATM in B-ISDN
Packet-switched capabilities
resembles packet-switched service provided by other data networks
Frame-mode capabilities
a service that supports frame relay
Common-channel signaling capabilities
used to control the network and provide call management

Definition of ISDN (1984, 1988):
An ISDN is a network, in general evolving from a telephony IDN, that provides end-to-end digital connectivity to
support a wide range of services, including voice and non-voice services, to which users have access by a limited set
of standard multipurpose user-network interfaces.

ISDN Interfaces and Functions:
Transmission Structure:
Digital pipe between central office and ISDN subscriber carry a number of communication channels, varies from
user to user.
The transmission structure of access links includes channels of:
B channel: 64 kbps
D channel: 16 or 64 kbps
H channel: 384 (H0), 1536 (H11), or 1920 (H12)kbps

B channel
a user channel, carrying digital data, PCM-encoded digital voice, or a mixture of lower-rate traffic at a fraction of 64
kbps
the elemental unit of circuit switching is the B channel
Three kinds of connections which can be set up over a B channel are
Circuit-switched: equivalent to switched digital service, call establishment does not take place over the B channel
but using CCS
Packet-switched: user is connected to PS node, data exchanged via X.25
Semipermanent: equivalent to a leased line, not requiring call establishment protocol, connection to another user by
prior arrangement

D channel
carries CCS information to control circuit-switched calls
may be used for PS or low speed telemetry when no signaling information is waiting.
H channel
provides user information transmission at higher data rates
use the channel as a high-speed trunk or subdivide it based on TDM
examples: fast fax, video, high-speed data, high quality audio.

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ISDN channel structures:
Basic and Primary Channel Structures:

Basic Channel Structure (Basic Access)
consists of two full-duplex 64 B-channs and a full-duplex 16 D-channels.
with overheads such as framing and sync. Total bit rate is 192 kbps
to meet the needs of residential subscribers and small offices
access through a single multifunction terminal or separate terminals simultaneous use of voice and data applications
(PS access)

Primary Channel Structure (Primary Access)
different data rates in different countries
for users with greater capacity needs such as offices w/ LAN, PBX

supporting H channels:
primary rate interface H0 channel structures (3H0 +D, 4H0, and 5H0+D)
primary rate interface H1 channel structures (one H11 and one H12+D)
primary rate interface structures for mixed B and H0 channels
zero or one D+ combination of B and H0 channels
(e.g., 3H0+5B+D or 3H0+6B for 1.544-Mbps interface)

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User-Network Interface Configurations
ITU-T approach for actual users physical configuration
functional grouping: certain arrangements of physical equipment
reference points: conceptual points of separation of group function
using a similar analogy with OSI model, i.e., layering
Equipment has only to conform to relevant interface standards

NT1:
includes functions similar to OSI layer 1
may be controlled by the ISDN provider ( a boundary to network)
performs line maintenance functions
supports multiple channels (e.g., 2B+D) using TDM
might support multiple devices in a multidrop arrangement

NT2:
an intelligent device that may include up to OSI layer 3
examples are digital PBX, a terminal controller, a LAN
example of a switching function:
construction of a private network using semipermanent circuits among a number of sites
Terminal equipment is the subscriber equip. using ISDN

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TE1:
devices that support the standard ISDN interface
examples: digital telephone, integrated voice/data terminals, dig. Fax

TE2
the existing non-ISDN equipment
examples: physical interface RS-232, host computer with X.25
requires a terminal adaptor (TA)

Reference points
Reference point T:
a minimal ISDN network termination at customers premises
separates networks provider equipment from the users one
Reference point S:
the interface of individual ISDN terminals
separates user terminal from network communications functions
Reference point R:
provides a non-ISDN interface between user equipment that is not ISDN compatible and adaptor equipment
comply with X or V series ITU-T recommendation
Reference point U:
describes full-duplex data signal on the subscriber line
Functions of ISDN functional grouping:

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ISDN Protocol Architecture:

It is good if we can fit ISDN protocols into the OSI model, however, the simple 7-layer stack does not capture the
relationship between a control signaling protocol on the D channel being used to set up, maintain, and terminate a
connection on the B or H channel.
Thus, ITU-T defined I.320, a two layered stacks of protocols.
User protocol block: transparent transfer of user information (X.25)
Control protocol block: supporting ISDN signaling
controlling a network connection
controlling multimedia calls
controlling the use of an already established connection
providing supplementary services
The ISDN protocol reference model includes a plane management function that cuts across all the protocol layers.
that enable a network management system to control the parameters and operation of remote systems.

ISDN protocols at the user-network interface:
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ISDN protocols:

Control signaling is a D channel function but user data may also be transferred across the D channel.
ISDN is essentially unconcerned with user layers 4-7.
LAPD (link access protocol, D channel) is based on HDLC but modified for ISDN.
Applications supported: control signaling, PS, and telemetry

ISDN address structure
The address appearing in call setup messages in CCS protocols

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LEc10RE # 17: A0VANcE0 8YLLAB08 c00R8E: types of network protocols

LEc10RE # 18:

ISDN physical layer functions (OSI layer 1):

encoding of digital data for transmission across the interface
full-duplex transmission of B channel data
full-duplex of transmission of D channel data
multiplexing of channels to form basic or primary access
activation and deactivation of physical circuit
power feeding from network termination to the terminal
terminal identification
faulty terminal isolation
D channel contention access (for multipoint configuration in basic access)

ISDN data link layer functions:
In this layer LAPD (Link Access ProtocolD Channel) is used for communication between subscriber &
network.
The basic data link layer functions of ISDN are flow control, error detection & error control are a part of
LAPD
All traffic over D channel uses LAPD
LAPD service will support
multiple terminals at the user-network installation
multiple layer 3 entities (e.g., X.25 level 3, Q.931)
Two types of service to LAPD users by LAPD standard
unacknowledged information-transfer service
acknowledged information-transfer service
unacknowledged information-transfer service
does not guarantee delivery of data nor inform of failure
no flow control, no error control mechanism
supports both point-to-point and broadcast
fast data transfer, e.g. management procedures (alarm messages)

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acknowledged information-transfer service
more common, similar to service of LAPB and HDLC
three phases of connection establishment between two users
connection establishment
data transfer
connection termination
guarantee of frame delivery in the order of transmission

Two types of LAPD operation corresponding to two types of service
unacknowledged operation: layer 3 information is transferred in unnumbered frames, error detection, no
error or flow control
acknowledged operation: layer 3 information is transferred in frames with sequence numbers and
acknowledged, with error and flow control, also referred to as multiple-frame operation
These two types of operation may coexist on a single D channel

LAP0 Frame 8tructure:

FLAG FIELDS
a unique pattern of 01111110 to delimit the frame at both ends
using bit stuffing to allow the presence of arbitrary patterns
Is a data transparency property

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Address Field
a two-part address for dealing with two levels of multiplexing
multiple user devices sharing the same physical interface
multiple types of traffic within each user device (PS data and cont.)
* Terminal endpoint identifier (TEI)
given to each user device manually or automatically
* Service access point identifier (SAPI)
corresponds to a layer 3 protocol entity within a user device
SAPI values are unique within a TEI
SAPI and TEI together identify a logical connection DLCI
at any time, LAPD may maintain multiple logical connection, each with a unique DLCI one log. connec.
per a layer 3 entity
Command/response (C/R) bit shows the type of LAPD message

Control field

three types of frames and three different control field
Information-transfer frames (I-frames)
carrying data to be transmitted for user
piggybacked of flow- and error-control data using go-back-N ARQ
Supervisory frames (S-frames)
providing ARQ mechanism
Unnumbered frames (U-frames)
providing supplemental link-control functions
supporting unacknowledged operation
poll/final (P/F) bit
P=1: in command frame to solicit a response frame from the peer LAPD entity.
F=1: in response frame indicating the response frame transmission

Information Field
presents only in I-frames and some unnumbered frames
max length of information field specified in Q.921=260 bytes

Frame-Check Sequence (FCS) Field
an error-detecting code for all field except flags using CRC

Three phases in acknowledged operation:
Connection Establishment
requested by either network or subscriber by sending a SABME (Set Asynchronous Balanced Mode Extended)
frame (used in HDLC)
SABME contains TEI and SAPI of the layer 3 of requested entity
replied by a UA or DM frame to confirm or reject the request
Data Transfer
sending I-frames with N(S) and N(R) fields in modulo 128
flow- and error-control, sliding window FC and go-back-N ARQ EC
Disconnect
by sending a DISC frame which replied by a UA frame

Sub[ect:

ISDN NETWORK LAYER
Introduction
Q.931: a new ISDN network

DSS 1

ARcRl1Ec10RE 0F 0881

72
CSl1/

ISDN NETWORK LAYER
Introduction
provides out
makes use of the D channel; both for circuit

DSS 1
Digital Subscriber Signaling System Number 1 (DSS 1)
collection of capabilities for providing call control signaling Over D channel
PHYSICAL LAYER
traffic.
At the DATA LINK LAYER
At the NETWORK LAYER,
supported by messages defined in the Q.931 protocol.



ISDN NETWORK LAYER

provides out-of-band call control for B and H channel traffic
PHYSICAL LAYER defined in I.430 (basic access) and I.431 (primary access), supports all B and D channel
DATA LINK LAYER
NETWORK LAYER,
ISDN NETWORK LAYER:
Q.931: a new ISDN network-layer protocol
band call control for B and H channel traffic
defined in I.430 (basic access) and I.431 (primary access), supports all B and D channel
DATA LINK LAYER LAPD supports all D channel traffic, incl
NETWORK LAYER, basic call control procedures and supplementary services procedures are both
(Modeling of basic and supplementary services)
LLC1UkL NC1LS
layer protocol
makes use of the D channel; both for circuit- & packet
LAPD supports all D channel traffic, incl
basic call control procedures and supplementary services procedures are both
LLC1UkL NC1LS
& packet-mode communication
LAPD supports all D channel traffic, incl
Sem - 7th
LLC1UkL NC1LS
mode communication
LAPD supports all D channel traffic, including call control signaling.
Sub[ect Code:
mode communication

uding call control signaling.
uding call control signaling.
322711(22)

Sub[ect:

Q.931 specifies procedures for establishing connections on B and H

Basic functions at the network layer:

Additional functions at the network layer:

73
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BA8lc cALL c0N1R0L: BA8lc cALL c0N1R0L: BA8lc cALL c0N1R0L: BA8lc cALL c0N1R0L:
call control protocol archltecture call control protocol archltecture call control protocol archltecture call control protocol archltecture

same interface to ISDN as the D channel.

interaction with the data link layer to transmit and receive messages
generation and interpretation of layer 3 messages
administration of timers and logical entities used in call control procedures
administration of access resources, including B and packet
verification that services provided are consistent with user requirements

routing and relaying
network connection control
conveying user
network connection mux
segmenting and reassembly
error detection
error recovery
sequencing: mechanisms for sequenced delivery of layer 3 information
congestion control and user data flow control
restart: returning channels and interfaces to an idle condition


with the data link layer to transmit and receive messages
routing and relaying: for end system connected to different subnetworks
network connection control
conveying user-to-network and network
network connection mux
error detection: for procedural errors in layer 3 protocol
error recovery: error recovery mechanisms from detected errors
: mechanisms for sequenced delivery of layer 3 information
: returning channels and interfaces to an idle condition
: for end system connected to different subnetworks
network connection control: mechanisms
network and network
network connection mux: for call control inf. of multiple calls on one LAPD
: for procedural errors in layer 3 protocol
: error recovery mechanisms from detected errors
LLC1UkL NC1LS
call control protocol archltecture call control protocol archltecture call control protocol archltecture call control protocol archltecture: :: :
: mechanisms using data link connections
network and network-to-user information
: for call control inf. of multiple calls on one LAPD
LLC1UkL NC1LS
administration of access resources, including B and packet-layer
using data link connections
user information
Sem - 7th
LLC1UkL NC1LS
Q.931 specifies procedures for establishing connections on B and H channels that share the
layer logical chan
: for end system connected to different subnetworks.
using data link connections
user information
Sub[ect Code:
channels that share the
logical channels
channels that share the
322711(22)

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74

Terminal Types
Two basic types of user terminals
Functional Terminalsintelligent devices employing full range of Q.931 messages and parameters for call control
Stimulus Terminalsdevices with a rudimentary signaling capability
A digital telephone is a stimulus terminal.
transmits signaling information one event at a time
network sends instruction regarding operation to be done by terminal

Types of Messages: (on the basis of application supported):

Circuit-mode connection control: refers to functions to set up, maintain, and clear a circuit-switched connection on a
B channel. This function corresponds to call control in existing circuit switching telecommunications networks.
Packet-mode access connection control: refers to functions to set up a circuit-switched connection (ISDN user) to an
ISDN packet-switching mode (ISDN service provider)
User-to-user signaling not associated with circuit-switched calls:
allows two users to communicate without setting up a circuit switched connection. A temporary signaling
connection is established and cleared in a manner similar to the control of circuit switched connection .Signaling
takes place over the D channel and thus does not consume B channel resources
Messages used with the global call reference: refers to functions that enable the user or network to return one or
more channels to an idle condition.

Types of Message (ON THE BASIS OF FUNCTIONS PERFORMED):

Call establishment services are:
set up a B channel call in response to user request
provide particular network facilities for this call
inform calling user of the progress of the call establishment process
Call information
Call clearing
Miscellaneous (e.g., to negotiate network features, supplementary services)

ISDN SERVICES:
1. Bearer services (based on circuit-switched, packet- switched, frame-switched, or cell-switched networks).
2. Teleservices (telephony, telefax, teletex, telex, teleconferencing).
3. Basic & supplementary services (call waiting, reverse charging, message handling).

LEc10RE # 19

common channel 8lgnallng 8ystem Number 7(887):

lntroductlon:
Network control signaling
transition from inchannel to a common channel approach
Signaling System Number 7 (SS7) was first issued by CCITT in 1980 and revised in 1984, 1988, and 1992.
Designed to be an open-ended common-channel signaling standard for a variety of digital circuit-switched networks.
Specifically designed to be used in ISDN: provides internal control and network intelligence essential to an ISDN.

Sub[ect:

Primary characteristics of SS7
The scope of SS7 is immense, because it must cover all aspects of control

887 Archltecture

Functlonal Archltecture
Control messages in a common channel signaling system are
Although the network is

SS7 functions could be implemented as additional functions in

Or, the network can have separa

7S
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optimized for use in digital telecommunication networks in conjunction with
exchanges, utilizing 64
designed to meet present and future informat
and maintenance
designed to be a reliable means for the transfer of information in the correct
suitable for operation over analog channels
suitable for use on point
including the reliable routing and
887 Archltecture

Although the network is
technology.
mode
mode

exchanges, utilizing 64-kbps digital channels
and maintenance
suitable for use on point-
including the reliable routing and

887 Archltecture:
Although the network is a circuit


Primary characteristics of SS7:
kbps digital channels
-to-point terrestrial and satellite links
including the reliable routing and delivery of control messages and application
Functlonal Archltecture:
a circuit-switched network, the control
Or, the network can have separate switching points for carrying
LLC1UkL NC1LS
kbps digital channels
designed to meet present and future information transfer requirements for call
suitable for operation over analog channels and at speeds below 64 kbps
point terrestrial and satellite links
livery of control messages and application
switched network, the control
te switching points for carrying
LLC1UkL NC1LS
ion transfer requirements for call
and at speeds below 64 kbps
point terrestrial and satellite links
Control messages in a common channel signaling system are short packets routed through the network.
switched network, the control signaling is implemented using packet switching
SS7 functions could be implemented as additional functions in circuit
te switching points for carrying
Sem - 7th
LLC1UkL NC1LS
ion transfer requirements for call
designed to be a reliable means for the transfer of information in the correct sequence without loss or duplication
and at speeds below 64 kbps
The scope of SS7 is immense, because it must cover all aspects of control signalin
short packets routed through the network.
signaling is implemented using packet switching
circuit-switching nodes
control packets only
Sub[ect Code:
digital stored program
ion transfer requirements for call control, remote control, management,
sequence without loss or duplication
signaling for complex digital networks,
livery of control messages and application-oriented content of those
switching nodes associated
control packets only disassociated signaling
digital stored program-control
control, remote control, management,
g for complex digital networks,
oriented content of those

associated signaling
disassociated signaling
322711(22)
control
g for complex digital networks,
oriented content of those messages.

signaling
disassociated signaling

Sub[ect:

Signaling Network Elements
Three functional entities in SS7:

Two planes of o
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signaling points (
(such as a circuit switching node)
signal transfer points (
signaling links

Two planes of o
control plane
information plane

signaling points (SP): any point in signaling network capable of handling SS7 control mes
signal transfer points (STP
signaling links: a data link that connects signaling points
Two planes of operation:
control plane for establishing & managing connections using SPs, STPs.
information plane

Signaling Network Elements:
): any point in signaling network capable of handling SS7 control mes
STP): a signaling point capable of routing control messages, e.g. a pure routing node
: a data link that connects signaling points
peration:
for establishing & managing connections using SPs, STPs.
for information transfer using LEs, TCs
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): a signaling point capable of routing control messages, e.g. a pure routing node
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Sub[ect Code:

): any point in signaling network capable of handling SS7 control messages, e.g. an endpoint
322711(22)
sages, e.g. an endpoint
Sub[ect:

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Sub[ect Code: Sub[ect Code: 322711(22)

322711(22)
Sub[ect:

887 Protocol Archltecture 887 Protocol Archltecture 887 Protocol Archltecture 887 Protocol Archltecture

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887 Protocol Archltecture 887 Protocol Archltecture 887 Protocol Archltecture 887 Protocol Archltecture: :: :
LLC1UkL NC1LS

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322711(22)
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79

SS7 has a layered protocol architecture similar to that of OSI model.
Message transfer part (MTP): the lowest three levels providing a reliable but connectionless (a datagram style)
service for routing messages
signaling data link: a full-duplex physical link dedicated to SS7 (OSI L1)
includes control links between STPs, between an STP and an SP, between SPs
signaling link: a data link control protocol, corresponds to OSI layer 2
provides reliable sequenced delivery of data across signaling data link
signaling network: provides routing data across multiple STPs from control source to control destination.
Signaling connection control part (SCCP): added in 1984 version of SS7
SCCP + MTP = NSP (Network Service Part)
contains different network-layer services to meet needs of NSP users
NSP is a message delivery system
ISDN user part (ISUP): controls signaling needed in an ISDN to deal with ISDN subscriber calls and related
functions
Transaction capabilities application part (TCAP): provides the mechanisms for transaction-oriented (not
connection-oriented) applications and functions
Operation, maintenance, and administration part (O&MAP): specifies network management functions and
messages related to operation & maintenance.
Application service elements (ASEs): additional modules to support new applications.

LEc10RE # 20

A1M & B-l80N: lntroductlon 8ervlces & Appllcatlons:

B BB B- -- -l80N: lN1R000c1l0N l80N: lN1R000c1l0N l80N: lN1R000c1l0N l80N: lN1R000c1l0N: :: :

B-ISDN is a service requiring transmission channels capable of supporting rates greater than the primary rate.
With B-ISDN services, especially video services, requiring data rates orders of magnitudes beyond those that can be
delivered by ISDN will become available.
These includes support for image processing, video, and high capacity workstations and local area networks.
To contrast this new network, the original ISDN network is now referred to as narrowband ISDN.
The key technology developments for B-ISDN are:
Optical fiber transmission systems that can offer low-cost, high-data rate transmission channels for network trunks
and subscriber lines.
Microelectronic circuits that can offer high-speed, low-cost building blocks for switching, transmission, and
subscriber equipment.
High-quality video monitors and cameras that can, with sufficient production quantities, be offered at low cost.
Integration of a wide range of communications facilities:
Worldwide exchange between any two subscribers in any medium or combination of media.
Retrieval and sharing of massive amounts of information from multiple sources, in multiple media, among people in
a shared electronic environment.
Distribution, including switched distribution, of a wide variety of cultural, entertainment, and educational materials
to home or office, virtually on demand.

Sub[ect:

ATM bas

Classification of B
ITU
Services in which there is a two

Services in which the information transfer is primarily one way, from

Broadband Services:

Messaging Services
offer user

Not a real
Analogous narrowband services of X.400 and teletex
Video mail is one of services supported by B

Conversational Services
provide the means for bidir

These services support the general transfer of data specific to a

Conversational services encompa

80
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ATM based/ Broadband Services:

Classification of B
ITU-T classification of B
Interactive Services
two subscribe
Includes: conversational, messaging, and retrieval services
Distribution Services
Includes: broadcast services and cyclical services

Broadband Services:

Messaging Services
offer user-to
Message handling (information editing, processi
Not a real-time service

information transfer between tw
and exchanged between users; not public information).
Real time service.

ed/ Broadband Services:
Classification of B-ISDN services:
T classification of B
Interactive Services
two subscribers or between
: conversational, messaging, and retrieval services
Distribution Services
: broadcast services and cyclical services
Broadband Services:
Messaging Services
to-user communicati
handling (information editing, processi
time service
information transfer between tw
exchanged between users; not public information).
Real time service.
ed/ Broadband Services:
ISDN services:
T classification of B-ISDN services
Services in which there is a two-way exchange of information (other
rs or between a subscriber and a service provider.
user communication between individual users via
handling (information editing, processi

provide the means for bidirectional dialogue communication
information transfer between two users or between a user and a
Conversational services encompass a wide range of applications
LLC1UkL NC1LS
ISDN services
way exchange of information (other
a subscriber and a service provider.
on between individual users via
handling (information editing, processing,
Video mail is one of services supported by B-ISDN
ectional dialogue communication
o users or between a user and a
ss a wide range of applications
LLC1UkL NC1LS
way exchange of information (other
a subscriber and a service provider.
on between individual users via storage units with store
ng, conversion) functions.
ISDN
ectional dialogue communication with bidirec
o users or between a user and a service provider host.
These services support the general transfer of data specific to a given user application (t
ss a wide range of applications
Sem - 7th
LLC1UkL NC1LS
way exchange of information (other than control
Services in which the information transfer is primarily one way, from service provider to B

storage units with store
conversion) functions.
with bidirectional (not store
service provider host.
given user application (t
ss a wide range of applications and data types including video, data, and document.
Sub[ect Code:
than control-signaling information) between
service provider to B
storage units with store-and forward, mailbox, and/or
tional (not store-and
service provider host.
given user application (the information generated by
and data types including video, data, and document.
signaling information) between
service provider to B-ISDN subscriber.
and forward, mailbox, and/or
and-forward), end
he information generated by
322711(22)
signaling information) between
ISDN subscriber.
forward), end-to-end

LLC1UkL NC1LS
81

Video conversational services: the most important service
Video-telephony (would be the most important service of B-ISDN)
Videoconferencing
Video surveillance
Video/audio information transmission service (higher-quality)
Data services in B-ISDN
File transfer in distributed architecture of computers and storage systems
Large-volume or high-speed transmission of measured values or control information
Program downloading
Computer-aided design and manufacturing (CAD/CAM)
Connection of local area networks at different locations
Document services
transfer of very high resolution fax or mixed documents (text, images, voice, video)

Retrieval Services
provide the user with the capability to retrieve information stored in information centers that is available for public
use.
The information is sent to the user on demand only.
The information can be retrieved on an individual basis.
Analogous narrowband service is Videotex.
A general-purpose data-base retrieval system for home or office
Through the public switched telephone network or cable TV system
information in the form of pages of text and simple graphics
Broadband videotex is an enhancement of the existing Videotex
with additional sounds, high-resolution images, short video scenes
Examples are:
Retrieval of encyclopedia
Results of quality tests on consumer goods
computer-supported audiovisual entities
Electronic mail-order catalogs and travel brochures, order, booking

Distribution Services without User Presentation Control
broadcasting of information from a central source to an unlimited number of authorized receivers connected to
network
access to information without any control over it
An example is broadcasting television.
Broadcasting of higher-resolution via B-ISDN rather than radio waves and cable TV distribution systems.
Another example is an electronic newspaper broadcast service.
Transmission of facsimile images of newspaper pages to subscribers who had paid for the service

Distribution Services with User Presentation Control
distributing information from a central source to a large number of users.
Information are in a sequence of frames with cyclic repetition.
User can control start and order of presentation.
Teletext is a narrowband service analogous to cabletext of B-ISDN
a simple one-way system that uses unallocated portions of the bandwidth of a broadcast TV signal.
Transmitter sends pages of text in round-robin fashion.
The user keys in the number of desired page and the decoder reads that page from the incoming signal, stores it, and
displays it.
Sub[ect:

In Cabletext, full digital

BUSINESS AND RESIDENTIAL SERVICES.

B
In B
To support simultaneous services a total subscriber li
Appropriate technology would be the Optical Fiber only.
Circuit

Control of B
The user
B
At user

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Limited to few hundred of pages with a cycle time of 10s seconds.
In Cabletext, full digital
Allowing 10,000 pages with a cycle tim


B-ISDN FUNCTIONAL ARCHITECTURE:
In B-ISDN, to meet the requirements for high
Circuit-switching cannot handle such data rates and hence a fast packet switching at user
is required.
Control of B
The user-network control signaling protocol is an enhanced Q.931.
B-ISDN must also support narrowband ISDN services (64kbps), both circuit switching and packet switching.
At user-network interface these capabilities will be pr

LFC: local function capabilities
TE: terminal equipment


In Cabletext, full digital broadband channel for cyclical
ISDN FUNCTIONAL ARCHITECTURE:
ISDN, to meet the requirements for high
switching cannot handle such data rates and hence a fast packet switching at user
Control of B-ISDN is again based on CCS (SS7).
network control signaling protocol is an enhanced Q.931.
ISDN must also support narrowband ISDN services (64kbps), both circuit switching and packet switching.
network interface these capabilities will be pr
broadband channel for cyclical
ISDN, to meet the requirements for high
ain based on CCS (SS7).
LLC1UkL NC1LS
broadband channel for cyclical
Allowing 10,000 pages with a cycle time of 1 second.
ISDN, to meet the requirements for high-resolution video, an upper channel rate of about 150 Mbps is needed.
ain based on CCS (SS7).
LLC1UkL NC1LS
broadband channel for cyclical transmission of pages with text, video, audio can be used.
e of 1 second.
resolution video, an upper channel rate of about 150 Mbps is needed.
To support simultaneous services a total subscriber line rate of about 600 Mbps is required.
network interface these capabilities will be provided with the connection
Sem - 7th
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transmission of pages with text, video, audio can be used.
ne rate of about 600 Mbps is required.
ovided with the connection
Sub[ect Code:
switching cannot handle such data rates and hence a fast packet switching at user-network interface as ATM
ovided with the connection-oriented ATM facility.
network interface as ATM
oriented ATM facility.

322711(22)
oriented ATM facility.

Sub[ect:

For B
(ATM).
One difference between a packet switching network (e.g., X.25) and

Another difference is that X.25 packets may be of varying length,

Interface and internal switching of B
B
The protocol reference model has three separate planes:

83
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LEc10RE # 21:

Princip

Principles of B

B-ISDN Protocol Reference Model /ATM REFERENCE MODEL:

For B-ISDN, the transfer of information across the user
(ATM).
the same channel as
cells.
B-ISDN also supports
User Plane:

LEc10RE # 21:
Principles & building blocks of B
Principles of B-ISDN:
ISDN Protocol Reference Model /ATM REFERENCE MODEL:
ISDN, the transfer of information across the user
the same channel as data
ISDN also supports circuit
User Plane: for user information transfer with flow
les & building blocks of B
ISDN:
transfer, whereas ATM makes use of CCS.
Interface and internal switching of B-ISDN is packet
circuit-mode applications but over a packet
for user information transfer with flow
LLC1UkL NC1LS
les & building blocks of B-ISDN:
ISDN is packet
mode applications but over a packet
for user information transfer with flow
LLC1UkL NC1LS
ISDN, the transfer of information across the user-network interface
ISDN is packet-based.
mode applications but over a packet
for user information transfer with flow- and error control
Sem - 7th
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network interface uses Asynchronous Transfer Mode
One difference between a packet switching network (e.g., X.25) and ATM is that X.25 includes control signaling on
Another difference is that X.25 packets may be of varying length, whereas ATM packets are of fixed size, called
mode applications but over a packet-based transport mechanism.
and error control
Sub[ect Code:
uses Asynchronous Transfer Mode
ATM is that X.25 includes control signaling on
whereas ATM packets are of fixed size, called
transport mechanism.

transport mechanism.
322711(22)

Sub[ect:

Physical Layer
consists of

timing (synchronization) is one of common functions
Transmission Convergence Sublayer
Responsible for
concerned with generating and maintaining the frame structure appropriate for a given data rate a
packaging ATM cells into a frame (e.g., no frame, sending a flow of cells)
maintaining the cell boundaries so that cells may be recovered after descrambling at the destination
generating and checking cell headers
insertion and suppression of idle cells to adapt the rate of valid ATM cells to the payload capacity of the

ATM Layer
independent of physical medium, with the following functions
having multiple logical connections across an interface similar to X.25 and frame relay

84
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Control Plane:
Management Plane:
a whole and provides coordination between all
functions relating to

Physical Layer
consists of physical medium and transmission convergence sublayers
physical medium
Responsible for
Transmission frame generation and recovery
Transmission frame adaptation
Cell delineation
HEC sequence generation and cell header verification
Cell rate decoupling
transmission system

ATM Layer
Cell multiplexing and demultiplexing
Virtual path identifier and vi

Control Plane: performs call control and connection
Management Plane: includes
functions relating to resour
Physical Layer
physical medium and transmission convergence sublayers
physical medium-dependent functions
Responsible for
Cell delineation
quence generation and cell header verification
Cell rate decoupling
transmission system

Virtual path identifier and vi
performs call control and connection
includes plane management
resources and parameters residing in its protocol entities)
dependent functions
Virtual path identifier and virtual channel identifier translation
LLC1UkL NC1LS
performs call control and connection
plane management
ces and parameters residing in its protocol entities)
thus, depends on the medium
generating and checking cell headers header error control (HEC) code
rtual channel identifier translation
LLC1UkL NC1LS
performs call control and connection control functions
plane management (performs management
a whole and provides coordination between all the planes) and
thus, depends on the medium
header error control (HEC) code
rtual channel identifier translation
Sem - 7th
LLC1UkL NC1LS
control functions
(performs management
the planes) and layer management
physical medium and transmission convergence sublayers Physical Medium Sublayer
thus, depends on the medium used
header error control (HEC) code
Sub[ect Code:
control functions
(performs management functions related to a system as
layer management

Physical Medium Sublayer
functions related to a system as
layer management (performs management
Physical Medium Sublayer includes only
concerned with generating and maintaining the frame structure appropriate for a given data rate at physical layer
322711(22)
includes only
t physical layer
LLC1UkL NC1LS
8S

VPI and VCI have local significance on logical connections and may need to be translated during switching
Cell header generation/extraction
appending cell header to user data from the AAL
Generic flow control
generating flow control information for placement in cell headers

ATM Adaptation Layer
consists of segmentation and reassembly and convergence sublayers
The segmentation and reassembly sublayer is responsible for the segmentation of higher-layer information into a
size suitable for the information field of an ATM cells and the reassembly of the contents of a sequence of ATM cell
information field into higher-layer information on reception.
The convergence sublayer is an interface speecification. It defines the services that AAL provides to higher layers.

PRINCIPLES & BUILDING BLOCKS OF B-ISDN:

BUILDING BLOCKS OF B-ISDN:
1. ATM
2. OPTICAL TRANSMISSION

ATM (Asynchronous Transfer Mode) Protocols OVERVIEW:
ATM combines the benefits of circuit switching (constant transmission delay, guaranteed capacity) with those of
packet switching (flexibility, efficiency for intermittent traffic). To achieve these benefits, ATM uses the following
features:
Fixed-size cells, permitting more efficient switching in hardware than is possible with variable-length packets
Connection-oriented service, permitting routing of cells through the ATM network over virtual connections,
sometimes called virtual circuits, using simple connection identifiers
Asynchronous multiplexing, permitting efficient use of bandwidth and interleaving of data of varying priority and
size
ATM or Cell Relay has a similar concept to frame relay.
Both ATM and frame relay take advantage of the reliability and fidelity of modern digital facilities for fast packet
switching.
ATM was developed as part of the work on broadband ISDN, but is beginning to find application in non-ISDN
environment where very high speed data rates are required.
Physical layer of the ATM protocol architecture involves the specification of a transmission medium and a signal
encoding scheme.

Data rates specified at the physical layer
155.52 Mbps
622.08 Mbps
Possibility of other data rates (higher and lower)

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86

ATM (Asynchronous Transfer Mode) Protocol HIERARCHY/ Overall hierarchy function in ATM network:

Two specific layers for ATM functions
ATM Layer
defines the transmission of data in fixed-sized cells
defines the use of logical connections
ATM Adaptation Layer (AAL)
mapping the higher-layer information into ATM cells to be transport over an ATM network
collecting information from ATM cells for delivery to higher layers

Three functional levels of the physical layer:
Transmission path level
Extends between network elements that assemble and disassemble the payload of a transmission system.
End-to-end communication: payload is the end-user information
User-to-network communication: payload is the signaling information
Digital section level
Extends between network elements that assemble and disassemble a continuous bit or byte stream.
Refers to the exchanges or signal transfer points in a network
Regenerator section level
A portion of a digital section.
For example, a repeater for regeneration (no switching is involved).

ATM Logical Connections:
Logical connections in ATM are referred to as virtual channel connection (VCC).
A VCC is Set up between two end users through the network and a variable-rate, full-duplex flow of fixed-size cells
is exchanged over the connection.
Virtual path connection (VPC) is a bundle of VCCs that have the same endpoints.
Thus, all of the cells flowing over all of the VCCs in a single VPC are switched together.

Sub[ect:

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VC
VP
VCL
VPL
VCI
VPI

Advantages of using virt
Simplified network architecture
separation of the network transport functions to those related to an individual logical connection (virtual
channel) and those related to a group of logical connections (virtual path)
Increased network p
dealing of the network with fewer, aggregated entities
Reduced processing and short connection setup time
by reserving capacity on a virtual path connection in anticipation of later call arrivals, new virtual

Advantages of using virt
Increased network p
Advantages of using virtual paths:
Increased network performance and reliability
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ual paths:
erformance and reliability
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erformance and reliability
Sem - 7th
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Sub[ect Code:


322711(22)
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88

connections can be established by executing simple control functions at the end-points of the virtual path
connections
no call processing required at transit nodes
Enhanced network services
virtual path used internal to the network but visible to the end user thus, user may define closed user group
or closed networks of virtual channel bundles.

Virtual Channel Connection Uses:

Between end users
to carry end-to-end user data
A VPC between end users provides them with an overall capacity
Between an end user and a network entity
to carry user-network control signaling
A user-to-network VPC can be used to aggregate traffic from an end user to a network exchange or network
server
Between two network entities
used for network management and routing functions
A network-to-network VPC can be used to define a common route for the exchange of network
management information

Virtual Path/Virtual Channel Characteristics:

Quality of Service
a user of a VCC is provided with a QoS specified by parameters such as cell loss ration and cell delay
variation
Switched and semi-permanent virtual channel connections
switched connections require call-control signaling
Cell sequence integrity
sequence of transmitted cells within a VCC is preserved
Traffic parameter negotiation and usage monitoring
traffic parameters can be negotiated between a user and the network for each VCC, monitored by the
network
parameters: average rate, peak rate, burstiness, peak duration
Virtual channel identifier restriction within a VPC
This characteristic is only for VPCs
one or more virtual channel identifiers may not be available to the user of the VPC, but may be reserved for
network use

Sub[ect:

LEc10RE # 22:

LEc10RE # 23:

LEc10RE # 24:
LEc10RE # 26:

0Nl1

Wlreless Medla
Unguided media relates to data transmission through the air and is commonly referred to as wireless. The transmission
and reception of data is carried out usin
There are two main ways that antenna work:
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Call establishment using virtual paths:

LEc10RE # 22:
LEc10RE # 23:
LEc10RE # 24:

LEc10RE # 26:

0Nl1 ~ 3: 0N-
Wlreless Medla

LEc10RE # 22: 1010RlAL # 2: A1M AN0 B
LEc10RE # 24: BRl00E c00R8E: Wlreless securlty: WEP, WPA, WlP8, Wl08
LEc10RE # 26:
-B00N0E0 ME0lA F0R lN1ERNE1
Wlreless Medla: components And Worklng 0f Wlreless Network
1010RlAL # 2: A1M AN0 B
REVl8l0N/80MMARY (Lc0), A88l0NMEN1 2
BRl00E c00R8E: Wlreless securlty: WEP, WPA, WlP8, Wl08
B00N0E0 ME0lA F0R lN1ERNE1
: components And Worklng 0f Wlreless Network
and reception of data is carried out using antenna.
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1010RlAL # 2: A1M AN0 B-l80N, Meanlng of l80N.
B00N0E0 ME0lA F0R lN1ERNE1:
g antenna.
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l80N, Meanlng of l80N.

Sem - 7th
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l80N, Meanlng of l80N.
: components And Worklng 0f Wlreless Network:
Sub[ect Code:

322711(22)
Unguided media relates to data transmission through the air and is commonly referred to as wireless. The transmission Unguided media relates to data transmission through the air and is commonly referred to as wireless. The transmission
LLC1UkL NC1LS
90

Directional (in a focused beam)
Omnidirectional (all around)
The unguided media is the wireless media. It simply transports electromagnetic waves without using any physical
conductor. Signals are normally broadcast through the air and thus are available to anyone who has the device
capable of receiving them.
Guided mediums are used for point-to-point connections between two devices while unguided mediums allow signals
to be sent out informally across multiple areas reaching a wide variety of systems.
Example of Unguided (wireless) media:
(i) Radio wave
(ii) Microwave
Terrestrial microwave
Satellite microwave
(iii) Infrared.
Wireless Transmission Frequencies:
1GHz to 40GHz
microwave
highly directional
point to point
satellite
30MHz to 1GHz
omnidirectional
broadcast radio
3 x 10
11
to 2 x 10
14

infrared
local
Antennas:
An antenna can be defined as an electrical conductor or system of conductors used either for radiating electromagnetic
energy or for collecting electromagnetic energy. For transmission of a signal, electrical energy from the transmitter is
converted into electromagnetic energy by the antenna and radiated into the surrounding environment (atmosphere,
space, water). For reception of a signal, electromagnetic energy impinging on the antenna is converted into electrical
energy and fed into the receiver.

Sub[ect:

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0lfference between


0lfference between
0lfference between
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Wireless technology is a famous source of networking for different technologies of computer networking of present
era. Wireless network which is commonly called as WiFi, is a type of that technology in which radio signals of high
frequency are used to transmit data from one device to another. It only operates on several hundred feet.
Types:
Wireless LAN or WiFi is divided into three main parts on which its whole working depends and all of its
applications also depend on these parts. These types are as follows;
1. Infra structure mode.
2. Ad hoc network mode
3. Mixed network mode.

Infra structure Mode:
Any kind of machine that can communicate with every type of work station of wireless LAN or WiFi with the help of
access points is called as infra structure network mode.

Ad hoc Network Mode:
A type of network in which all the work stations are linked together with other work stations without any obstacle is
referred to as ad hoc network mode.

Mixed network mode:
It is form of network which is developed by mixing infra structure and ad hoc network and the work stations can work
simultaneously in it, is known as mixed network mode.

How it Works?

Basically the working of WiFi is based on transmission process. In this process the data which is going to be
transferred has to be transmitted in the form of radio signals, and then the device transmits these signals to antenna
which is used for transferring of data. This antenna is which is used for transmission purpose is connected with a
wired LAN or DSL. A router is also connected with the device which is able to receive the signals and also help in
decoding them. Then this information is spread on internet.

Wireless LAN Architecture:
The process of assembling the parts of computer hardware in computer networking is called as computer architecture.
Similarly if we use this architectural technique in Wireless LAN or WiFi is called as Wireless LAN Architecture. It is
a technique of designing and arrangement of different components in Wireless local area networking device in a
specific way. Special type of device which is the combination of transmitter and receiver called d as transceiver which
is an essential part for standard Wireless LAN architecture that is known as Access points.

Sub[ect:

Components of Wireless Architecture:

Wireless LAN architecture is composed of differ
between different operating systems. These components are very essential for WiFi architecture.

1. Access point
2. Clients
3. Bridge

Access Points
A special type of routing device that is used to
called as AP. It is often connected with the help of wired devices such as Ethernet. It only transmits or transfers the
data between wireless LAN and wired network by using infra structure
support a small group of networks and works more efficiently. It is operated less than hundred feet. It is denoted by
AP.
Clients
Any kind of device such as personal computers, Note books, or any kind of mobile de
wireless network area referred as a client of wireless LAN architecture.
Bridge
A special type of connectors which is used to establish connections between wired network devices such as Ethernet
and different wireless netw
LAN architecture. Two components are also some time play an important role in Wireless LAN architecture i.e.
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Access point
Clients
Bridge

Access Points
AP.
Clients
Bridge


Access Points
and different wireless networks such as wireless LAN. It is called as bridge. It acts as a point of control in wireless
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Wireless LAN architecture is composed of different components which help in establishing the local area network
A special type of routing device that is used to transmit the data between wired and wireless networking device is
orks such as wireless LAN. It is called as bridge. It acts as a point of control in wireless
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transmit the data between wired and wireless networking device is
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data between wireless LAN and wired network by using infra structure mode of network. One access point can only
Sub[ect Code:
mode of network. One access point can only
Any kind of device such as personal computers, Note books, or any kind of mobile devices which are inter linked with

vices which are inter linked with
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1. Basic Service Set (BSS)
2. Extended Service Set (ESS)

LEc10RE # 26: A0VANcE0 8YLLAB08 c00R8E: Wlreless communlcatlons 1ransfer Protocol (Wc1P)

LEc10RE # 27:
lEEE 802.11 standards and WLAN types:
IEEE 802.11 is standard use world widely for Wireless Local Area network, WLAN, it stands for Institute of
Electrical and Electronics Engineers. In simple words it is standard of communication between computers and
wireless devices. The standards were set by IEEE LAN/MAN Standard committee in 5GHZ and 2.4GHz public
spectrum bands.
The IEEE 802.11 is consisting of of numerous mechanism
and services that interrelate to present station mobility translucent to the higher layers of the network mass. IEEE
802.11 is a combination of wireless LAN. We can use IEEE 802.11 and WiFi often as interchangeably depends on
market demand. It is a basic protocol of any high speed wireless network from 1997 to 2008. There are lots of change
occurred in it but the basics high performance" never change. The frequency of it is from 2.4 GHz to 5 GHz and
range is from 100 meter to 5000 meter. It specifies a single medium access control as MAC and three physical layers
and offering different service such as Authentication of system, Deauthentication of system, data privacy, MSDU
delivery, Association with files and system, Disassociation, Distribution of data, Integration of information, and
Reassociation. You can configure a station with IEEE 802.11 in different ways as independent configuration, and
infra-structure configuration. The IEEE 802.11 starts from the need to connect more than one computer with each
other wirelessly especially where the wire cannot go.
IEEE 802.11 required accepting device to activate in a peer-to-peer fashion within the possibly of overlie same as
access control level and data transformation services to allow top layers .There are several physical layer represent
signaling methods and interfaces. Solitude and safety of user data is being moved over the wireless media via IEEE
802.11.According to OSI model an IEEE 802.11 protocol having direct frequency, hopping sequence, spread
spectrum PHY and infrared PHY. IEEE 802.11 specifies a single medium access sub layers called MAC and physical
layers with different specifications. The first is physical layers having two radio and infrared. The physical layer
contain Frequency Hopping Spread Spectrum Radio PHY offering 1 Mbit/s with two level and 2 Mbit/s with 4 level,
As it is Direct Sequence Spread Spectrum Radio PHY provides DBPSK and DQPSK with 1 and 2 Mbit/s
operation,Infrared PHY offer 1 Mbit/s with16-PPM and 2 Mbit/s with PPM.
These days 802.11a, 802.11b, and 802.11g are mostly in use to create a network home, offices and commercial
locations. The architecture of IEEE 802.11 consists of 802.1X for verification, RSN for maintenance track of links,
and AES-based CCMP to provide privacy, veracity and basis. IEEE 802.11with the propagation of modems and DSL
services makes it reliable. People always desire to set up small networks in their homes to distribute their elevated rate
Internet connection. There are many free networks commonly permit anyone within specific range, counting
passersby outer, to connect to the Internet. There is also hard work by unpaid helper groups to launch wireless
community networks to provide free wireless connectivity to the public. Networking without wire has a talented
prospect with 802.11.The dynamic nature of IEEE 802.11 increasing radically in the near future.

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WLAN: Wireless LAN by IEEE 802.11, 802.11a, 802.11b, 802.11g, 802.11n
The Wireless Local Area Networl (WLAN) technology is defined by the IEEE 802.11 family of
specifications. There are currently four specifications in the family: 802.11, 802.11a, 802.11b, and
802.11g. All four use the Ethernet protocol and CSMA/CA (carrier sense multiple access with collision
avoidance instead of CSMA/CD) for path sharing.
802.11 -- applies to wireless LANs and provides 1 or 2 Mbps transmission in the 2.4 GHz band
using either frequency hopping spread spectrum (FHSS) or direct sequence spread spectrum
(DSSS).
802.11a -- an extension to 802.11 that applies to wireless LANs and provides up to 54 Mbps in
the 5GHz band. 802.11a uses an orthogonal frequency division multiplexing (OFDM) encoding
scheme rather than FHSS or DSSS. The 802.11a specification applies to wireless ATM systems
and is used in access hubs.
802.11b (also referred to as 802.11 High Rate or Wi-Fi) -- an extension to 802.11 that applies
to wireless LANS and provides 11 Mbps transmission (with a fallback to 5.5, 2 and 1 Mbps) in
the 2.4 GHz band. 802.11b uses only DSSS. 802.11b was a ratification to the original 802.11
standard, allowing wireless functionality comparable to Ethernet.
802.11g -- offers wireless transmission over relatively short distances at 20 - 54 Mbps in the
2.4 GHz band. The 802.11g also uses the OFDM encoding scheme.
802.11n - builds upon previous 802.11 standards by adding MIMO (multiple-input multiple-
output). IEEE 802.11n offers high throughput wireless transmission at 100Mbps - 200 Mbps.

The modulation used in 802.11 has historically been phase-shift keying (PSK). The modulation method
selected for 802.11b is known as complementary code keying (CCK), which allows higher data speeds
and is less susceptible to multipath-propagation interference. 802.11a uses a modulation scheme
known as orthogonal frequency-division multiplexing (OFDM) that makes possible data speeds as high
as 54 Mbps, but most commonly, communications takes place at 6 Mbps, 12 Mbps, or 24 Mbps.
For short range and low power wireless (less than 10 meters) communications among personal devices such as PDA,
Bluetooth and subsequent IEEE standards (802.15) are taking effects. For long range wireless communications in the
metropolitan areas, WiMax as defined in the IEEE 802.16 is the standard.
Types of Wireless LAN:
A wireless local area network (or LAN) is a data connection which uses radio waves rather than
physical Ethernet cables to connect a network together. It enables any devices connected to the LAN
to freely roam within the area of the wireless broadcast, making it ideal for use with laptops or
mobile devices. There are a few types of wireless LAN configurations available.
Private Wireless LANs
This is the most common area to see a wireless LAN. Home WLANs are meant to be used for small
areas, usually with up to a 200-foot radius, and as such are suitable for homes or small businesses.
The private WLAN subscribes to the 802.11 technology standard for wireless networking, with the
most common being the 802.11b standard, often referred to as Wi-Fi.
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Private WLANS will frequently be based around a single router/access point which will connect to
the Internet and then allow computer terminals to share the network. Any device compatible with
802.11 will be able to connect to a private WLAN.
Wide Area WLAN
These types of WLAN work in the same way as a private WLAN, however they offer a much further
reach in terms of access to the network: up to a 10-mile radius. The technology still uses an 802.11
connection, but to transmit a wider radius more powerful antennas and transceivers are used.
This type of WLAN is useful for connecting a network over a long distance, over a large industrial
estate for example, where a wired network would be costly and unfeasible.
Business WLAN
A business network is an expansion of the private WLAN in terms of technology, except that this
network will use multiple access points rather than one. All of the access points will connect to a
central "hub" which splits the connection into multiple streams, creating a signal which can cover
many different areas. The user will only connect to one network but will have access to all of the
connected access points.
Enterprise WLAN
This WLAN is an expansion on the business WLAN. It covers a much larger area and is found
within governmental, academic or large corporate organizations that need to share a network over
multiple locations. The network is controlled by a central computer, and the access points all connect
wirelessly to this computer. Changes can be made to each access point from this central computer,
and this setup makes it easy for the network to be changed from one central location rather than
multiple locations.
0lfference between WLAN, Bluetooth technology and lR:
Bluetooth is a technology for short range wireless data and realtime two-way voice transfer providing data rates up to
3 Mb/s. WLAN (Wireless Local Area Network) is a technology for wireless data transfer providing data rates up to
54 Mb/s. IrDA (Infrared Data Asso-ciation) is a technology for very short range wireless data transfer providing data
rates up to 16 Mb/s. Bluetooth and WLAN are wireless RF (Radio Frequency) communication systems, while IrDA is
wireless infrared communication system. Bluetooth, WLAN and IrDA are wireless communications technologies,
which differ in terms of their features and data security solutions. These three wireless communication technologies
have been chosen for comparison, because they are widely used all over the world. Bluetooth and WLAN represent
the new and promising generation of wireless communication, while IrDA can be considered as old and impractical
technology.

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LEc10RE # 28:

Ad-hoc networks:
An ad-hoc network is a local area network (LAN) that is built spontaneously as devices connect. Instead of
relying on a base station to coordinate the flow of messages to each node in the network, the individual
network nodes forward packets to and from each other. In Latin, ad hoc literally means "for this," meaning "for
this special purpose" and also, by extension, improvised
An ad hoc neLwork ls one where Lhere are no access polnLs passlng lnformaLlon beLween parLlclpanLs.
lnfrasLrucLure neLworks pass lnformaLlon Lhrough a cenLral lnformaLlon hub whlch can be a hardware devlce
or sofLware on a compuLer. Cfflce neLworks, for example, generally use a server Lo whlch company
worksLaLlons connecL Lo recelve Lhelr lnformaLlon. Ad hoc neLworks, on Lhe oLher hand, do noL go Lhrough a
cenLral lnformaLlon hub.
Ad hoc networks are generally closed in that they do not connect to the Internet and are typically created
between participants. But, if one of the participants has a connection to a public or private network, this
connection can be shared among other members of the ad hoc network. This will allow other users on the
spontaneous ad hoc network to connect to the Internet as well.
Ad hoc networks are common for portable video game systems like the Sony PSP or the Nintendo DS
because they allow players to link to each other to play video games wirelessly. Some retail stores even create
networks within them to allow customers to obtain new game demos via the store's own ad hoc network.
An ad hoc network can be thought of as a peer-to-peer network for the wireless age. Peer-to-peer or
workgroup style networks were used to create a network environment for early Windows computers. This
allowed these early computers to connect to each other to exchange information, usually in a smaller office
environment without the need for domains and the additional management and overhead that comes with
them.
The possibilities with ad hoc networks are quite endless. With connection sharing and other methods to access
public or private wired or wireless networks, the reach of an ad hoc network can extend quite far.
LEc10RE # 29:

MAcAW Protocol:
MACAW: A Media Access Protocol for Wireless LANs:
The media in a wireless network is a shared and scarce resource; as a result, controlling access to this shared
media to maximize performance is one of the key issues in wireless networks
MACAW is a media access protocol, for single channel wireless LAN built upon pre-existing MACA (Multiple
Access, Collision Avoidance) protocol with an aim to increase performance and fairness among contending
wireless nodes. The design of MACAW is based on four observations:
1. Relevant contention occurs at the receiver; sensing carrier at the sender (as in CSMA) is inappropriate
2. Congestion is location dependent
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3. For fair allocation, collision (congestion) information must be shared among devices
4. Information related to contention period must be synchronized among devices to promote fair contention
MACA Briefing
MACA uses two types of short, fixed-length (32 byte) signaling packets: RTS (Request-to-Send) and CTS
(Clear-to-Send). When a station wants to send something, it sends an RTS first. If the receiver is free, it replies
with an CTS. Upon receiving CTS, the sender immediately sends data. Any station overhearing RTS waits long
enough for CTS to pass through. Anyone, overhearing CTS waits for the length of data (length of data is carried
in signaling headers). As a result, hidden-terminal problem is completely avoided and exposed-terminal
problem is simplified. If the sender does not receive CTS within some period, it will time out and schedule for
retransmission using a binary exponential backoff (BEB).
MACAW
MACAW design is based on the observations mentioned before as well as the limitations of MACA. A brief
summary follows:
Backoff algorithm: MACAW replaces BEB with MILD (multiplicative increase and linear decrease) to
ensure that backoff interval grows a bit slowly (1.5x instead of 2x) and shrinks really slowly (linearly to
minimum value). To enable better congestion detection, MACAW shares backoff timers among stations
by putting this info in headers.
Multiple stream model: MACAW uses separate queues for each stream in each node for increased
fairness. In addition, each queue runs independent backoff algorithms. However, all stations attempting to
communicate with the same receiver should use the same backoff value.
Basic exchange: MACAW replaces RTS-CTS-DATA to RTS-CTS-DS-DATA-ACK with the following
extensions:
1. ACK: An extra ACK at the end ensures that errors can be recovered in the link layer, which is
much faster than transport layer recovery. If an ACK is lost, next RTS can generate another ACK
for the previous transmission.
2. DS: This signal ensures a 3-way handshake between sender and receiver (similar to TCP) so that
everyone within hearing distance of the two stations know that a data transmission is about to
happen. Without the DS packet, stations vying for the shared media cannot compete properly and
one is always starved due to the lack of its knowledge of the contention period. In short, DS
enables synchronization.
3. RRTS: RRTS is basically a proxy RTS, when the actual RTS sender is too far away to fight for
the contention slot. However, there is one scenario where even RRTS cannot guarantee fair
contention.
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4. Multicast: Multicast is handled by sending data right away after the RTS packet, without waiting
for CTS. It suffers from the same problems as in CSMA, but the authors leave it as an open
challenge.
MACAW is fairer and gives higher throughput than MACA.

Principles of operation
Assume that node A has data to transfer to node B.
Node A initiates the process by sending a Request to Send frame (RTS) to node B. The destination node (node B) replies with
a Clear To Send frame (CTS). After receiving CTS, node A sends data. After successful reception, node B replies with an
acknowledgement frame (ACK). If node A has to send more than one data fragment, it has to wait a random time after each
successful data transfer and compete with adjacent nodes for the medium using the RTS/CTS mechanism.
Any node overhearing an RTS frame (for example node F or node E in the illustration) refrains from sending anything until a
CTS is received, or after waiting a certain time. If the captured RTS is not followed by a CTS, the maximum waiting time is the
RTS propagation time and the destination node turnaround time.
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Any node (node C and node E) overhearing a CTS frame refrains from sending anything for the time until the data frame and
ACK should have been received (solving the hidden terminal problem), plus a random time. Both the RTS and CTS frames
contain information about the length of the DATA frame. Hence a node uses that information to estimate the time for the data
transmission completion.
Before sending a long DATA frame, node A sends a short Data-Sending frame (DS), which provides information about the length
of the DATA frame. Every station that overhears this frame knows that the RTS/CTS exchange was successful. An overhearing
station (node F), which might have received RTS and DS but not CTS, defers its transmissions until after the ACK frame should
have been received plus a random time.
To sum up, a successful data transfer (A to B) consists of the following sequence of frames:
1. Request To Send frame (RTS) from A to B
2. Clear To Send frame (CTS) from B to A
3. Data Sending frame (DS) from A to B
4. DATA fragment frame from A to B, and
5. Acknowledgement frame (ACK) from B to A.
MACAW is a non-persistent slotted protocol, meaning that after the medium has been busy, for example after a CTS message, the
station waits a random time after the start of a time slot before sending an RTS. This results in fair access to the medium. If for
example nodes A, B and C have data fragments to send after a busy period, they will have the same chance to access the medium
since they are in transmission range of each other.
RRTS
Node D is unaware of the ongoing data transfer between node A and node B. Node D has data to send to node C, which is in the
transmission range of node D. D initiates the process by sending an RTS frame to node C. Node C has already deferred its
transmission until the completion of the current data transfer between node A and node B (to avoid co-channel interference at
node B). Hence, even though it receives RTS from node D, it does not reply back with CTS. Node D assumes that its RTS was
not successful because of collision and hence proceeds to back off (using an exponential backoff algorithm).
If A has multiple data fragments to send, the only instant when node D successfully can initiate a data transfer is during small
gaps in between that node A has completed data transfer and completion of node B next CTS (for node A next data transfer
request). However, due to the node D backoff time period the probability to capture the medium during this small time interval is
not high. To increase the per-node fairness, MACAW introduces a new control message called "Request for Request to Send"
(RRTS).
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Now, when node C, which cannot reply earlier due to ongoing transmission between node A and node B, sends an RRTS message
to node D during next contention period, the recipient of the RRTS (node D) immediately responds with an RTS and the normal
message exchange is commenced. Other nodes overhearing an RRTS defer for two time slots, long enough to hear if a successful
RTSCTS exchange occurs.
To summarize, a transfer may in this case consist of the following sequence of frames between node D and C:
1. Request To Send frame (RTS) from D to C
2. Request for Request to send frame (RRTS) from C to D (after a short delay)
3. Request To Send frame (RTS) from D to C
4. Clear To Send frame (CTS) from C to D
5. Data Sending frame (DS) from D to C
6. DATA fragment frame from D to C,
7. Acknowledgement frame (ACK) from C to D.
MACAW does not generally solve the exposed terminal problem. Assume that node G has data to send to node F in our
example. Node G has no information about the ongoing data transfer from A to B. It initiates the process by sending an
RTS signal to node F. Node F is in the transmission range of node A and cannot hear the RTS from node G, since it is
exposed to co-channel interference. Node G assumes that its RTS was not successful because of collision and hence
backs off before it tries again. In this case, the solution provided by the RRTS mechanism will not improve the situation
much since the DATA frames sent from B are rather long compared to the other frames. The probability that F is
exposed to transmission from A is rather high. Node F has no idea about any node interested in initiating data transfer to
it, until G happens to transmit an RTS in between transmissions from A.Furthermore, MACAW might not behave
normally in multicasting.
Exposed node problem
In wireless networks, the exposed node problem occurs when a node is prevented from sending packets to other nodes due to a
neighboring transmitter. Consider an example of 4 nodes labeled R1, S1, S2, and R2, where the two receivers are out of range of
each other, yet the two transmitters in the middle are in range of each other. Here, if a transmission between S1 and R1 is taking
place, node S2 is prevented from transmitting to R2 as it concludes after carrier sense that it will interfere with the transmission
by its neighbor S1. However note that R2 could still receive the transmission of S2 without interference because it is out of range
of S1.
IEEE 802.11 RTS/CTS mechanism helps to solve this problem only if the nodes are synchronized and packet sizes and data rates
are the same for both the transmitting nodes. When a node hears an RTS from a neighboring node, but not the corresponding
CTS, that node can deduce that it is an exposed node and is permitted to transmit to other neighboring nodes.
Sub[ect:

If the nodes are not synchronised (or if the packet sizes are different or the data rates ar
sender will not hear the CTS or the ACK during the transmission of data of the second sender.

Hidden node problem
In
wireless
control
Hidden nodes
physical
communication range of the AP, but the nodes cannot communicate with each other, as they do not have a physical connection to
each other. In a
the access point, but it is unlikely that the same node can see a node on the opposite end of the access point's range,
nodes a
node
collisions occur, scrambling data. To overcome this problem, handshaking is implemented in conjunction with the CSMA/CA
scheme. The same problem exists in a
IEEE 802.11
RTS/CTS is not a complete solution and may decrease throughput even further, but adaptive acknowledgments from the base
station can help too.
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Hidden node problem
In wireless networking
wireless access point
control.
Hidden nodes
physical star topology
each other. In a
nodes are known as
node A and B can not sense
IEEE 802.11 uses

Hidden node problem
ess networking, the
access point (AP), but not from other nodes communicating with said AP. This leads to difficulties in
Hidden nodes in a wireless network
star topology with an access point with many nodes surrounding it in a circular fashion: Each node is within
each other. In a wireless network, it is likely that the node at the far edge of the access point's range, which is known as
re known as hidden. The problem is when nodes
can not sense the carrier,
uses 802.11 RTS/CTS
Hidden node problem
, the hidden node problem
(AP), but not from other nodes communicating with said AP. This leads to difficulties in
wireless network refer to nodes that are out of range of other nodes or a collection of nodes. Take a
with an access point with many nodes surrounding it in a circular fashion: Each node is within
wireless network, it is likely that the node at the far edge of the access point's range, which is known as
. The problem is when nodes
the carrier, Carrier sense multiple access with collision avoidance
scheme. The same problem exists in a MANET
802.11 RTS/CTS acknowledgment and handshake packets to pa
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hidden node problem or

refer to nodes that are out of range of other nodes or a collection of nodes. Take a
. The problem is when nodes
Carrier sense multiple access with collision avoidance
MANET.
acknowledgment and handshake packets to pa
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hidden terminal problem
A and B start to send
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hidden terminal problem
start to send packets
Sub[ect Code:
If the nodes are not synchronised (or if the packet sizes are different or the data rates are different) the problem may occur that the
hidden terminal problem occurs when a
packets simultaneously to the access point. Since
Carrier sense multiple access with collision avoidance (CSMA/CA
acknowledgment and handshake packets to partly overcome the
e different) the problem may occur that the
occurs when a node is visible from a
simultaneously to the access point. Since
CSMA/CA) does not work, and
rtly overcome the hidden node problem
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is visible from a
(AP), but not from other nodes communicating with said AP. This leads to difficulties in media access
wireless network, it is likely that the node at the far edge of the access point's range, which is known as A, can see
the access point, but it is unlikely that the same node can see a node on the opposite end of the access point's range, B. These
) does not work, and
hidden node problem
is visible from a
media access
, can see
. These
) does not work, and
hidden node problem.
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Co-channel interference or CCI is crosstalk from two different radio transmitters using the same frequency. There can be
several causes of co-channel radio interference; four examples are listed here:
1. Cellular Mobile Networks
2. Adverse weather conditions
3. Poor frequency planning
4. Overly-crowded radio spectrum
5. Daytime vs Nighttime
6. Cancellation of signal
Exponential backoff is an algorithm that uses feedback to multiplicatively decrease the rate of some process, in
order to gradually find an acceptable rate.
Binary exponential backoff / truncated exponential backoff:
In a variety of computer networks, binary exponential backoff or truncated binary exponential backoff refers to
an algorithm used to space out repeated retransmissions of the same block of data, often as part of network
congestion avoidance.
Examples are the retransmission of frames in carrier sense multiple access with collision avoidance (CSMA/CA)
and carrier sense multiple access with collision detection (CSMA/CD) networks, where this algorithm is part of
the channel access method used to send data on these network. In Ethernet networks, the algorithm is commonly
used to schedule retransmissions after collisions. The retransmission is delayed by an amount of time derived from
the slot time and the number of attempts to retransmit.
After c collisions, a random number of slot times between 0 and 2
c
- 1 is chosen. For the first collision, each sender
will wait 0 or 1 slot times. After the second collision, the senders will wait anywhere from 0 to 3 slot times (inclusive).
After the third collision, the senders will wait anywhere from 0 to 7 slot times (inclusive), and so forth. As the number
of retransmission attempts increases, the number of possibilities for delay increases exponentially.
The 'truncated' simply means that after a certain number of increases, the exponentiation stops; i.e. the
retransmission timeout reaches a ceiling, and thereafter does not increase any further. For example, if the ceiling is
set at i = 10 (as it is in the IEEE 802.3 CSMA/CD standard

), then the maximum delay is 1023 slot times.
Because these delays cause other stations that are sending to collide as well, there is a possibility that, on a busy
network, hundreds of people may be caught in a single collision set. Because of this possibility, after 16 attempts at
transmission, the process is aborted.

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LEc10RE # 30:
Compare 8|uetooth, Infrared & 802.11 WLAN.
Features and 0oals of Bluetooth, Bluetooth products and securlty:
Cverv|ew of 8|uetooth 1echno|ogy
Bluetooth is a low cost, low power, short-range radio technology intended to replace cable connections between
cellphones, PDAs and other portable devices. It can cleanup your desk considerably, making wires between
your workstation, mouse,
laptopcomputer etc. obsolete. Ericsson Mobile Communications started developing theBl ue t oot h s y
s t e m i n 1994, l ooki ng f or a r e pl a c e me nt t o t he c a bl e s c onne c t i ngcellphones and their
accessories. The Bluetooth system is named after a tenth-
centuryDa n i s h Vi k i n g k i n g , Ha r a l d B l t a n d , wh o u n i t e d a n d c o n t r o l l e d No r wa y a
n d Denmark. The first Bluetooth devices hit the market around 1999.
Bluetooth is an open standard for short-range radio frequency (RF) communication. Bluetooth
technology is used primarily to establish wireless personal area networks (WPAN), commonly referred to
as ad hoc or peer-to-peer (P2P) networks. Bluetooth technology has been integrated into many types of
business and consumer devices, including cellular phones, personal digital assistants (PDA), laptops,
automobiles, printers, and headsets. This allows users to form ad hoc networks between a wide variety of
devices to transfer voice and data. Bluetooth is a low-cost, low-power technology that provides a
mechanism for creating small wireless networks on an ad hoc basis, known as piconets.

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10S

A piconet is composed of two or more Bluetooth devices in close physical proximity that operate on the
same channel using the same frequency hopping sequence. An example of a piconet is a Bluetooth-based
connection between a cellular phone and a Bluetooth-enabled ear bud.
Bluetooth piconets are often established on a temporary and changing basis, which offers communication
flexibility and scalability between mobile devices.
Some key benef|ts of 8|uetooth techno|ogy are:
Cable replacement. Bluetooth technology replaces a variety of cables, such as those traditionally used for
peripheral devices (e.g., mouse and keyboard connections), printers, and wireless headsets and ear buds that
interface with personal computers (PC) or mobile telephones.
Ease of file sharing. A Bluetooth-enabled device can form a piconet to support file sharing
capabilities with other Bluetooth devices, such as laptops.
Wireless synchronization. Bluetooth provides automatic synchronization between Bluetooth enabled
devices. For example, Bluetooth allows synchronization of contact information contained in electronic
address books and calendars.
Internet connectivity. A Bluetooth device with Internet connectivity can share that access with other
Bluetooth devices. For example, a laptop can use a Bluetooth connection to have a mobile phone establish a
dial-up connection, so that the laptop can access the Internet through the phone.
Bluetooth wireless technology is a short-range communications technology intended to replace the cables
connecting portable and/or fixed devices while maintaining high levels of security. The key features of
Bluetooth technology are robustness, low power, and low cost. The Bluetooth specification defines a
uniform structure for a wide range of devices (up to 8) to connect and communicate with each other.
How Bluetooth Works
Bluetooth can be used to connect almost any device to another device. .Bluetooth can beused to form ad hoc networks
of several (up to eight) devices, called piconets. When Bluetooth devices first connect, there is a piconet master that
initiates the connection, and the others are slave devices. .One piconet can have a maximum of seven active slave
devices and one master device. All communication within a piconet goes through the piconet master. Two or more
piconets together form a scatternet, which can be used to eliminate Bluetooth range restrictions. It is not possible to be
a master of two different piconets because a piconet is a group of devices all synchronized on a hopping sequence set
by the master. For that reason, any devices that share a master must be on the same piconet. Scatternet environment
requires that different piconets must have a common device (so-called scatternet member) to relay data between the
piconets.. Bluetooth devices within a 10 to 100 meters (or 30 to 300 feet) range can share data with a throughput of
1Mbps for Version 1.2 and up to 3 Mbps for Version 2.0 + Enhanced Data Rate (EDR).

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Bluetooth Security
SecurlLy has played a ma[or role ln Lhe lnvenLlon of 8lueLooLh. 1he 8lueLooLh SlC has puL much efforL lnLo maklng
8lueLooLh a secure Lechnology and has securlLy experLs who provlde crlLlcal securlLy lnformaLlon. ln general,
8lueLooLh securlLy ls dlvlded lnLo Lhree modes: (1) non-secure, (2) servlce level enforced securlLy, and (3) llnk level
enforced securlLy. ln non-secure, a 8lueLooLh devlce does noL lnlLlaLe any securlLy measures. ln servlce-level
enforced securlLy mode, Lwo 8lueLooLh devlces can esLabllsh a nonsecure Asynchronous ConnecLlon-Less (ACL) llnk.
SecurlLy procedures, namely auLhenLlcaLlon, auLhorlzaLlon and opLlonal encrypLlon, are lnlLlaLed when a L2CA
(Loglcal Llnk ConLrol and AdapLaLlon roLocol) ConnecLlon-CrlenLed or ConnecLlon-Less channel requesL ls
made.1he dlfference beLween servlce level enforced securlLy and llnk level enforced securlLy ls LhaL ln Lhe laLLer, Lhe
8lueLooLh devlce lnlLlaLes securlLy procedures before Lhe channel ls esLabllshed. As menLloned above, 8lueLooLh's
securlLy procedures lnclude auLhorlzaLlon, auLhenLlcaLlon and opLlonal encrypLlon. AuLhenLlcaLlon lnvolves provlng
Lhe ldenLlLy of a compuLer or compuLer user, or ln 8lueLooLh's case, provlng Lhe ldenLlLy of one plconeL member Lo
anoLher. AuLhorlzaLlon ls Lhe process of granLlng or denylng access Lo a neLwork resource. LncrypLlon ls Lhe
LranslaLlon of daLa lnLo secreL code. lL ls used beLween 8lueLooLh devlces so LhaL eavesdroppers can noL read lLs
conLenLs. Powever, even wlLh all of Lhese defense mechanlsms ln place, 8lueLooLh has shown Lo have some
securlLy rlsks.
The following are the three basic security services specified in the Bluetooth standard:
Authentication: verifying the identity of communicating devices. User authentication is not provided natively by
Bluetooth.
Confidentiality: preventing information compromise caused by eavesdropping by ensuring that only authorized
devices can access and view data.
Authorization: allowing the control of resources by ensuring that a device is authorized to use a service before
permitting it to do so.
8lueLooLh does noL address oLher securlLy servlces such as audlL and non-repudlaLlon, lf such servlces are needed,
Lhey musL be provlded Lhrough addlLlonal means.
8|uetooth n|story
1. 8egan as a prlvaLe developmenL efforL aL Lrlcsson ln 1994
2. 3 companles [olned Lo form Lhe 8lueLooLh Speclal lnLeresL Croup (SlC) ln 1998
3. llrsL speclflcaLlon released ln !uly 1999
4. CurrenL speclflcaLlon ls verslon 1.1
8|uetooth Goa|s
1. Cpen SpeclflcaLlon
2. volce and uaLa CapablllLy
3. Worldwlde usablllLy
4. ShorL-8ange Wlreless SoluLlons

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Goa|s of 8|uetooth techno|ogy
- romote new usage mode|s
- 3-ln-1 phone
- 8rlefcase Lrlck
- AuLomaLlc synchronlzer
- Cvercome system cha||enges
- Pandle boLh volce and daLa
- Ad hoc connecLlons
- WlLhsLand lnLerference (2.4 CPz)
- Worldwlde use
- SecurlLy slmllar Lo wlred connecLlons
- Small slze
- negllglble power consumpLlon
- ublqulLous ueploymenL
- Draft a spec|f|cat|on
Bluetooth security issues are an important factor with any Bluetooth device or system. As with any device these days
that provide connectivity, security is an important issue.
There are a number of Bluetooth security measures that can be incorporated into Bluetooth devices to prevent various
security threats that can be posed.
One of the main requirements for Bluetooth is that it should be easy to connect to other devices. However Bluetooth
security needs to be balanced against the ease of use and the anticipated Bluetooth security threats.
Much work has been undertaken regarding Bluetooth security, however it remains high on the agenda so that users
can use their Bluetooth devices with ease while keeping the security threats to a minimum.

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Bluetooth security basics
There are also three different security modes that a device can implement. A device can be only in one security mode
at a time:
1. Nonsecure: Bluetooth device does not initiate any security measures.
2. Service-level enforced security mode: Two Bluetooth devices can establish a nonsecure Asynchronous
Connection-Less (ACL) link. Security procedures, namely authentica-tion, authorization and optional encryption, are
initiated when a L2CAP (Logical Link Control and Adaptation Protocol) Connection-Oriented or Connection-Less
channel request is made.
3. Link-level enforced security mode: Security procedures are initiated when an ACL link is established. Security
within Bluetooth technology covers three major areas: authentication, authorization and encryption. Authentication
is used for proving the identity of one piconet member to another. The results of authentication are used for
determining the client's authorization level. Encryption is used for encoding the information being exchanged
between Bluetooth devices in the way that eavesdroppers can not read its contents.

Bluetooth security is of paramount importance as devices are susceptible to a variety of wireless and networking
attacking including denial of service attacks, eavesdropping, man-in-the-middle attacks, message modification, and
resource misappropriation.
Bluetooth security must also address more specific Bluetooth related attacks that target known vulnerabilities in
Bluetooth implementations and specifications. These may include attacks against improperly secured Bluetooth
implementations which can provide attackers with unauthorized access.
Many users may not believe there is an issue with Bluetooth security, but hackers may be able to gain access to
information from phone lists to more sensitive information that others may hold on Bluetooth enabled phones and
other devices.
There are three basic means of providing Bluetooth security:
Authentication: In this process the identity of the communicating devices are verified. User authentication is
not part of the main Bluetooth security elements of the specification.
Confidentiality: This process prevents information being eavesdropped by ensuring that only authorised
devices can access and view the data.
Authorisation: This process prevents access by ensuring that a device is authorised to use a service before
enabling it to do so.

Security measures provided by the Bluetooth specifications
The various versions of the specifications detail four Bluetooth security modes. Each Bluetooth device must operate
in one of four modes:
Bluetooth Security Mode 1: This mode is non-secure. The authentication and encryption functionality is
bypassed and the device is susceptible to hacking. Bluetooth devices operation in Bluetooth Security Mode 1.
Devices operating like this do not employ any mechanisms to prevent other Bluetooth-enabled devices from
establishing connections. While it is easy to make connections, security is an issue. It may be applicable to
short range devices operating in an area where other devices may not be present. Security Mode 1 is only
supported up to Bluetooth 2.0 + EDR and not beyond.
Bluetooth Security Mode 2: For this Bluetooth security mode, a centralised security manager controls
access to specific services and devices. The Bluetooth security manager maintains policies for access control
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and interfaces with other protocols and device users.

It is possible to apply varying trust levels and policies to restrict access for applications with different security
requirements, even when they operate in parallel. It is possible to grant access to some services without
providing access to other services. The concept of authorisation is introduced in Bluetooth security mode 2.
Using this it is possible to determine if a specific device is allowed to have access to a specific service.

Although authentication and encryption mechanisms are applicable to Bluetooth Security Mode 2, they are
implemented at the LMP layer (below L2CAP).

All Bluetooth devices can support Bluetooth Security Mode 2; however, v2.1 + EDR devices can only support
it for backward compatibility for earlier devices.
Bluetooth Security Mode 3: In Bluetooth Security Mode 3, the Bluetooth device initiates security
procedures before any physical link is established. In this mode, authentication and encryption are used for all
connections to and from the device.

The authentication and encryption processes use a separate secret link key that is shared by paired devices,
once the pairing has been established.

Bluetooth Security Mode 3 is only supported in devices that conform to Bluetooth 2.0 + EDR or earlier.
Bluetooth Security Mode 4: Bluetooth Security Mode 4 was introduced at Bluetooth v2.1 + EDR.

In Bluetooth Security Mode 4 the security procedures are initiated after link setup. Secure Simple Pairing uses
what are termed Elliptic Curve Diffie Hellman (ECDH) techniques for key exchange and link key generation.

The algorithms for device authentication and encryption algorithms are the same as those defined in Bluetooth
v2.0 + EDR.

The security requirements for services protected by Security Mode 4 are as follows:
o Authenticated link key required
o Unauthenticated link key required
o No security required
Whether or not a link key is authenticated depends on the Secure Simple Pairing association model used.
Bluetooth Security Mode 4 is mandatory for communication between v2.1 + EDR devices.
Common Bluetooth security issues
There are a number of ways in which Bluetooth security can be penetrated, often because there is little security in
place. The major forms of Bluetooth security problems fall into the following categories:
Bluejacking: Bluejacking is often not a major malicious security problem, although there can be issues with
it, especially as it enables someone to get their data onto another person's phone, etc. Bluejacking involves the
sending of a vCard message via Bluetooth to other Bluetooth users within the locality - typically 10 metres.
The aim is that the recipient will not realise what the message is and allow it into their address book.
Sub[ect:

In
upgrading he security to ensure that these Bluetooth security lapses do not arise with their products.
Bluetooth protocol stack:

Bluetoo
protocols, telephony control protocols, and adopted protocols.
stacks are: LMP, L2CAP and SDP. In addition, devices that communicate with Bluetooth almost universally
can use these protocols:
IMP
The
Implemented on the controller.
AVRCP
A/V Remote Control Pro
audio.
I2CAP
The
between two devices using different higher level protocols. Provides segmentation and reassembly of on
packets.
110
CSl1/

Thereafter
contact
Bluebugging:
access a phone and use its features. This may include plac
does not realise that the phone has been taken over.
Car Whispering:
a Bluetooth enabled car stereo system

In order to protect against these and other forms of vulnerability, the manufacturers of Bluetooth enabled devices are

Bluetooth is defined as a layer protocol architecture consisting of core protocols, cable replacement
can use these protocols:
IMP
The Link Management Protocol
AVRCP
audio. Adopted versions
I2CAP
The Logical Link Control and Adaptation Protocol
packets.

Thereafter messages might be automatically opened because they have come from a supposedly known
contact
Bluebugging: This more of an issue. This form of Bluetooth security issue allows hackers to remotely
Whispering:
order to protect against these and other forms of vulnerability, the manufacturers of Bluetooth enabled devices are
th is defined as a layer protocol architecture consisting of core protocols, cable replacement
can use these protocols: HCI
Link Management Protocol
Adopted versions
Logical Link Control and Adaptation Protocol
messages might be automatically opened because they have come from a supposedly known
This more of an issue. This form of Bluetooth security issue allows hackers to remotely
This involves the use of software that allows hackers to send and receive audio to and from
HCI and RFCOMM.
Link Management Protocol (LMP) is used for control of the radio link between two devices.
A/V Remote Control Profile. Commonly used in car navigation systems to control streaming Bluetooth
1.0, 1.3 & 1.4
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and RFCOMM.
(LMP) is used for control of the radio link between two devices.
file. Commonly used in car navigation systems to control streaming Bluetooth
1.0, 1.3 & 1.4
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Logical Link Control and Adaptation Protocol (L2CAP)
Sem - 7th
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access a phone and use its features. This may include placing calls and sending text messages while the owner
[49]
Mandatory protocols for all Bluetooth
(L2CAP) Used to multiplex multiple logical connections
Sub[ect Code:
ing calls and sending text messages while the owner
Used to multiplex multiple logical connections

322711(22)
between two devices using different higher level protocols. Provides segmentation and reassembly of on-air
air
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In Basic mode, L2CAP provides packets with a payload configurable up to 64kB, with 672 bytes as the
default MTU, and 48 bytes as the minimum mandatory supported MTU.
In Retransmission and Flow Control modes, L2CAP can be configured either for isochronous data or
reliable data per channel by performing retransmissions and CRC checks.
Bluetooth Core Specification Addendum 1 adds two additional L2CAP modes to the core specification.
These modes effectively deprecate original Retransmission and Flow Control modes:
Lnhanced ketransm|ss|on Mode (L81M): 1hls mode ls an lmproved verslon of Lhe orlglnal reLransmlsslon mode.
1hls mode provldes a rellable L2CA channel.
Stream|ng Mode (SM): 1hls ls a very slmple mode, wlLh no reLransmlsslon or flow conLrol. 1hls mode provldes
an unrellable L2CA channel.
Reliability in any of these modes is optionally and/or additionally guaranteed by the lower layer Bluetooth
BDR/EDR air interface by configuring the number of retransmissions and flush timeout (time after which
the radio will flush packets). In-order sequencing is guaranteed by the lower layer.
Only L2CAP channels configured in ERTM or SM may be operated over AMP logical links.
SP
The Service Discovery Protocol (SDP) allows a device to discover services offered by other devices, and
their associated parameters. For example, when you use a mobile phone with a Bluetooth headset, the phone
uses SDP to determine which Bluetooth profiles the headset can use (Headset Profile, Hands Free
Profile, Advanced Audio Distribution Profile (A2DP) etc.) and the protocol multiplexer settings needed for
the phone to connect to the headset using each of them. Each service is identified by a Universally Unique
Identifier(UUID), with official services (Bluetooth profiles) assigned a short form UUID (16 bits rather than
the full 128).
RFCOMM
Radio Frequency Communications (RFCOMM) is a cable replacement protocol used to create a virtual
serial data stream. RFCOMM provides for binary data transport and emulatesEIA-232 (formerly RS-232)
control signals over the Bluetooth baseband layer, i.e. it is a serial port emulation.
RFCOMM provides a simple reliable data stream to the user, similar to TCP. It is used directly by many
telephony related profiles as a carrier for AT commands, as well as being a transport layer for OBEX over
Bluetooth.
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112

Many Bluetooth applications use RFCOMM because of its widespread support and publicly available API
on most operating systems. Additionally, applications that used a serial port to communicate can be quickly
ported to use RFCOMM.
BNFP
The Bluetooth Network Encapsulation Protocol (BNEP) is used for transferring another protocol stack's
data via an L2CAP channel. Its main purpose is the transmission of IP packets in the Personal Area
Networking Profile. BNEP performs a similar function to SNAP in Wireless LAN.
AVC1P
The Audio/Video Control Transport Protocol (AVCTP) is used by the remote control profile to transfer
AV/C commands over an L2CAP channel. The music control buttons on a stereo headset use this protocol
to control the music player.
AV1P
The Audio/Video Distribution Transport Protocol (AVDTP) is used by the advanced audio distribution
profile to stream music to stereo headsets over an L2CAP channel. Intended to be used by video distribution
profile in the bluetooth transmission.
1CS
The Telephony Control Protocol Binary (TCS BIN) is the bit-oriented protocol that defines the call
control signaling for the establishment of voice and data calls between Bluetooth devices. Additionally,
"TCS BIN defines mobility management procedures for handling groups of Bluetooth TCS devices."
TCS-BIN is only used by the cordless telephony profile, which failed to attract implementers. As such it is
only of historical interest.
AJopteJ protocolx
Adopted protocols are defined by other standards-making organizations and incorporated into Bluetooths
protocol stack, allowing Bluetooth to create protocols only when necessary. The adopted protocols include:
o|nt-to-o|nt rotoco| ()
lnLerneL sLandard proLocol for LransporLlng l daLagrams over a polnL-Lo-polnL llnk.
1C]I]UD
loundaLlon roLocols for 1C/l proLocol sulLe

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Cb[ect Lxchange rotoco| (C8Lk)
Sesslon-layer proLocol for Lhe exchange of ob[ecLs, provldlng a model for ob[ecL and operaLlon represenLaLlon
W|re|ess App||cat|on Lnv|ronment]W|re|ess App||cat|on rotoco| (WAL]WA)
WAL speclfles an appllcaLlon framework for wlreless devlces and WA ls an open sLandard Lo provlde moblle users
access Lo Lelephony and lnformaLlon servlces.

Baseband error correction
Depending on packet type, individual packets may be protected by error correction, either 1/3 rate forward
error correction (FEC) or 2/3 rate. In addition, packets with CRC will be retransmitted until acknowledged
by automatic repeat request (ARQ)

LEc10RE # 31: c1 - 1 0l8c088l0N
LEc10RE # 32:
1cP 0ver Wlreless & lpv6: Moblle lP, Moblle 1cP:
MC8ILL I:
Mob||e I (or l moblllLy) ls an lnLerneL Lnglneerlng 1ask lorce (lL1l) sLandard communlcaLlons proLocol LhaL ls
deslgned Lo allow moblle devlce users Lo move from one neLwork Lo anoLher whlle malnLalnlng a permanenL l
address.
The Mobile IP protocol allows location-independent routing of IP datagrams on the Internet. Each mobile
node is identified by its home address disregarding its current location in the Internet. While away from its
home network, a mobile node is associated with a care-of address which identifies its current location and
its home address is associated with the local endpoint of a tunnel to its home agent. Mobile IP specifies how
a mobile node registers with its home agent and how the home agent routes datagrams to the mobile node
through the tunnel.
Mobile IP provides an efficient, scalable mechanism for roaming within the Internet. Using Mobile IP,
nodes may change their point-of-attachment to the Internet without changing their home IP address. This
allows them to maintain transport and higher-layer connections while roaming. Node mobility is realized
without the need to propagate host-specific routes throughout the Internet routing fabric.
Applications
Mobile IP is most often found in wired and wireless environments where users need to carry their mobile
devices across multiple LAN subnets. It may for example be used in roaming between overlapping wireless
systems, for example IP over DVB, WLAN, WiMAX and BWA. Currently, Mobile IP is not required
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within cellular systems such as 3G, to provide transparency when Internet users migrate between cellular
towers, since these systems provide their own data link layer handover and roaming mechanisms. However,
it is often used in 3G systems to allow seamless IP mobility between different Packet Data Serving Node
(PDSN) domains. In many applications (e.g., VPN, VoIP), sudden changes in network connectivity and IP
address can cause problems.
How Mobile IP works:
A mobile node can have two addresses - a permanent home address and a care of address (CoA), which is
associated with the network the mobile node is visiting. There are two kinds of entities in Mobile IP:
A home agent stores information about mobile nodes whose permanent home address is in the home
agent's network.
A foreign agent stores information about mobile nodes visiting its network. Foreign agents also
advertise care-of addresses, which are used by Mobile IP.
A node wanting to communicate with the mobile node uses the permanent home address of the mobile node
as the destination address to send packets to. Because the home address logically belongs to the network
associated with the home agent, normal IP routing mechanisms forward these packets to the home agent.
Instead of forwarding these packets to a destination that is physically in the same network as the home
agent, the home agent redirects these packets towards the foreign agent through an IP tunnel by
encapsulating the datagram with a new IP header using the care of address of the mobile node.
When acting as transmitter, a mobile node sends packets directly to the other communicating node through
the foreign agent, without sending the packets through the home agent, using its permanent home address as
the source address for the IP packets. This is known as triangular routing. If needed, the foreign agent could
employ reverse tunneling by tunneling the mobile node's packets to the home agent, which in turn forwards
them to the communicating node. This is needed in networks whose gateway routers have ingress filtering
enabled and hence the source IP address of the mobile host would need to belong to the subnet of the
foreign network or else the packets will be discarded by the router.
The Mobile IP protocol defines the following:
an authenticated registration procedure by which a mobile node informs its home agent(s) of its care-
of-address(es);
an extension to ICMP Router Discovery, which allows mobile nodes to discover prospective home
agents and foreign agents; and
the rules for routing packets to and from mobile nodes, including the specification of one mandatory
tunneling mechanism and several optional tunneling mechanisms.
M0BlLE 1cP (M-1cP):
Special handling of lengthy and/or frequent disconnections
M-TCP splits as I-TCP does
o unmodified TCP fixed network to supervisory host (SH)
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11S

o optimized TCP SH to MH
Supervisory host
o no caching, no retransmission
o monitors all packets, if disconnection detected
set sender window size to 0
sender automatically goes into persistent mode
o old or new SH reopen the window
Advantages
o maintains end to end semantics, supports disconnection, no buffer forwarding
Disadvantages
o loss on wireless link propagated into fixed network
o adapted TCP on wireless link.
o noL a good ldea for heavy Lrafflc.

LEc10RE # 33:
support of Moblllty on the lnternet, 1rafflc Routlng ln Wlreless Networks:
Traffic Routing in Wireless Networks:
The amount of traffic capacity required in a wireless network is highly dependent upon the type of traffic carried. For
example, a subscribers telephone call (voice traffic) requires dedicated network access to provide real-time
communications, whereas control and signaling traffic may be bursty in nature and may be able to share network
resources with other bursty users. Alternatively, some traffic may have an urgent delivery schedule while some may
have no need to be sent in real-time. The type of traffic carried by a network determines the routing services,
protocols, and call handling techniques which must be employed.
Two general routing services are provided by networks. These are connection-oriented services (virtual circuit
routing), and connectionless services (datagram services). In connection-oriented routing, the communications path
between the message source and destination is fixed for the entire duration of the message, and a call set-up procedure
is required to dedicate network resources to both the called and calling parties. Since the path through the network is
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fixed, the traffic in connection-oriented routing arrives at the receiver in the exact order it was transmitted. A
connection-oriented service relies heavily on error control coding to provide data protection in case the network
connection becomes noisy. If coding is not sufficient to protect the traffic, the call is broken, and the entire message
must be retransmitted from the beginning.
LEc10RE # 34:
clrcult swltched 0ata 8ervlces, Packet swltched 0ata servlces.
The old telephone system (PSTN) uses circuit switching to transmit voice data whereas VoIP uses packet-switching
to do so. The difference in the way these two types of switching work is the thing that made VoIP so different and
successful.
To understand switching, you need to realize that the network in place between two communicating persons is a
complex field of devices and machines, especially if the network is the Internet. Consider a person in Mauritius
having a phone conversation with another person on the other side of the globe, say in the US. There are a large
number of routers, switches and other kinds of devices that take the data transmitted during the communication
from one end to the other.
Switching and routing
Switching and routing are technically two different things, but for the sake of simplicity, let us
takeswitches and routers (which are devices that make switching and routing respectively) as devices doing one
job: make a link in the connection and forward data from the source to the destination.
Paths or circuits
The important thing to look for in transmitting information over such a complex network is the pathor circuit. The
devices making up the path are called nodes. For instance, switches, routers and some other network devices, are
nodes.
In circuit-switching, this path is decided upon before the data transmission starts. The system decides on which
route to follow, based on a resource-optimizing algorithm, and transmission goes according to the path. For the
whole length of the communication session between the two communicating bodies, the route is dedicated and
exclusive, and released only when the session terminates.
Packets
To be able to understand packet-switching, you need to know what a packet is. The Internet Protocol(IP), just like
many other protocols, breaks data into chunks and wraps the chunks into structures called packets. Each packet
contains, along with the data load, information about the IP address of the source and the destination nodes,
sequence numbers and some other control information. A packet can also be called a segment or datagram.

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Once they reach their destination, the packets are reassembled to make up the original data again. It is therefore
obvious that, to transmit data in packets, it has to be digital data.
In packet-switching, the packets are sent towards the destination irrespective of each other. Each packet has to
find its own route to the destination. There is no predetermined path; the decision as to which node to hop to in
the next step is taken only when a node is reached. Each packet finds its way using the information it carries,
such as the source and destination IP addresses.
As you must have figured it out already, traditional PSTN phone system uses circuit switching while VoIP uses
packet switching.
Brief comparison
Circuit switching is old and expensive, and it is what PSTN uses. Packet switching is more modern.
When you are making a PSTN call, you are actually renting the lines, with all it implies. See why international
calls are expensive? So if you speak for, say 10 minutes, you pay for ten minutes of dedicated line. You normally
speak only when your correspondent is silent, and vice versa. Taking also into consideration the amount of time no
one speaks, you finally use much less than half of what you are paying for. With VoIP, you actually can use a
network or circuit even if there are other people using it at the same time. There is no circuit dedication. The cost is
shared.
Circuit-switching is more reliable than packet-switching. When you have a circuit dedicated for a session, you are
sure to get all information across. When you use a circuit which is open for other services, then there is a big
possibility of congestion (which is for a network what a traffic jam is for the road), and hence the delays or even
packet loss. This explains the relatively lower quality of VoIP voice compared to PSTN. But you actually have
other protocols giving a helping hand in making packet-switching techniques to make connections more reliable.
An example is the TCP protocol. Since voice is to some extent tolerant to some packet loss (unless text - since a
comma lost can mean a big difference), packet-switching is finally ideal for VoIP.
LEc10RE # 36:
WLL, LM08, 0W1:
W|re|ess Loca| Loop:
- Wlreless Local Loop (WLL)
- 1wo-way wlreless communlcaLlon servlces Lo sLaLlonary or near-sLaLlonary users ln a small area
- Local Loop
- 1he clrculL connecLlng a subscrlbers's Lelephone wlLh llne-LermlnaLlng equlpmenL aL a CenLral Cfflce.
- 1hese clrculLs are separaLed lnLo drops" for lndlvldual subscrlbers
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WLL vs. WirelineSystem

Advantages of WLL over Wire Line Loop

Concentration of Resources
Flexibility to planning and deployment of the system
Fast network deployment and change
Lower maintenance and capital costs
Local mobile feature
Temporary Installation

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Wireless Local Loop Architecture

Wireless Access Network Unit (WANU)
Components
BTSsor Radio Ports (RPs)
A Radio Port Control Unit (RPCU)
Access Manager (AM) and HLR
Functions provided by WANU
Authentication, Air Interface Privacy
Over-the-Air Registration of Subscriber Units
Radio Resource Management
Interworking Function (IWF)
Operation and Maintenance (OAM)
Routing, Billing, and Switching
Protocol conversion and transcoding of voice and data

Wireless Access Subscriber Unit (WASU)
Functions provided by WASU
Air Interface UWLL toward the network
A traditional interface TWLL toward the subscriber
This Interfaces include
Protocol conversion
Transcoding
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Authentication Function
OAM
Signaling Functions
Modem Function to support voice-band data
Switching Function (SF)
Wireless Local Loop Technologies
Satellite-Based Systems
Provide Telephony services for rural communities and isolated areas
Cellular-Based Systems
Provide high-power, wide-range, median subscriber density and median circuit-quality WLL services
Offers both mobility and fixed wireless access via the same platform as cellular
Low-tier PCS or Microcellular-Based Systems
Provide low-power, narrow-range, high subscriber density and high circuit-quality WLL services
Fixed Wireless Access (FWA) Systems
Proprietary radio systems
Disadvantage of the cellular approach
Limitation on toll-quality voice and signaling transparency
Disadvantage of low-tier PCS and microcellular approaches
Narrow radio coverage range
FWA addresses these issues
LMDS:
Local Multlpolnt 0lstrlbutlon 8ervlces (LM08)
LMDS is a broadband wireless line-of-sight point-to-multipoint communication system that operates typically in the
24-31GHz bands. LMDS systems use a cellular-like network architecture similar to mobile networks, except that
LMDS delivers network connectivity to stationary buildings and not to mobile users. Current LMDS systems are able
to offer data rates of up to 622Mbps at the expense of dedicating a large chunk of allocated spectrum (100-112MHz)
to a single subscriber. However, in general, service providers will offer data rates of around 10Mbps because of the
need to maximise the number of subscribers given a finite band of spectrum.
Because of its Point-to-Multipoint nature, LMDS has been touted as a cost-effective last mile solution that service
providers can adopt to connect their subscribers to a high speed Internet backbone. Not only can it deliver relatively
high data rates, but it can also be deployed more rapidly and cheaply than fibre. In addition, LMDS systems are
scalable - service coverage can be extended by adding more base stations or by subdividing an existing cell to deal
with increasing customer demand. However, network planning and expansion is complicated by frequency
interference issues, which affect the placement of base stations and the range of frequencies usable in adjacent and
nearby cells.
There are two ways of separating the uplink connection (from the subscriber to the base station) from the downlink
connection (from the base station to the subscriber). With Time Division Duplexing (TDD), the subscriber and the
base station take turns talking to each other. At any time, both parties will use the entire spectrum allocated for that
link. On the other hand, with Frequency Division Duplexing (FDD), the uplink and the downlink use different
frequency bands separated by a large guard band to avoid interference (e.g. ETSI recommends a separation of
1008MHz for the 24.5-26.5GHz band that Singapore tried to auction).
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Now, since one base station needs to communicate with several sets of Consumer Premises Equipment (CPE), there
has to be a way of partitioning the uplink or the downlink frequency band (for the FDD case) / transmission duration
(for the TDD case) among all the subscribers served by the base station. The Time Division Multiple Access (TDMA)
approach separates the transmissions to the various CPEs in time such that at any instance the base station
communicates with only one CPE. Alternatively, using Frequency Division Multiple Access (FDMA), each CPE is
allocated a small slice of the spectrum allocated to the uplink or downlink, and transmits simultaneously along with
the other CPEs, i.e. their transmissions are separated in frequency.
The radius of an LMDS cell can range from 2.5km to 10km depending on the modulation scheme used and the
climate of the region in which it is deployed. The modulation schemes available differ primarily in how efficient they
are at using available spectrum. Quadrature Phase Shift Keying (QPSK) provides only 1.5 bits per second per Hz,
whereas 64-QAM (Quadrature Amplitude Modulation) pumps out 5 bits per second per Hz. However, higher
bandwidth efficiencies require correspondingly higher carrier-to-interference (C/I) ratios to work, which mean smaller
cell sizes. Furthermore, one must realise that LMDS transmission is most severely attenuated by rain. In a heavy rain
region like Singapore, carrier-grade reliability is achievable up to a range of only about 1.5km using QPSK. 16-QAM
and 64-QAM require cell sizes that are too small to be practical.
UWB
Introduction:
UWB technology stands for the Ultra Wide Band technology. It is a type of technology that
completely based upon the radio waves for the sake of communication among the small distances
but at a very high rate or speed is called as the ultra wide band technology. This technology
generally consists of high bandwidth and most of the time personal area networks are involved
with such technology. All the operations of the UWB technology work wirelessly. The processes for
which the ultra wide band technology is designed are the multimedia processes. The old or the
traditional devices on which UWB technology is applicable are the radar systems etc.

How UWB Technology Works?
The working of the ultra wide band technology is very specific and calculated because it generally
governs the working of those processes in which multimedia processes are involved for example
mp3 players, Mp4, different type of mobile phones in which PDA technology is involved. Ultra
wide band technology working can be applicable on the places where the distance is shorter and
needs transmission of higher speed. First of all, as we know that this technology is completely
depends upon the radio waves and that is why it is some times also called as the UWB radio
transmitter. The ultra wideband technology starts its working when the radio waves from the
UWB transmitter emit and send short signals as to the spectrum. The signals are in such a way
they move around the actual setting for example if the centre point is set on 6 GHz then it will
move around 5 or 7. With the help of such rate UWB provide high sped data transfer wirelessly
between two places but it only operates on some meters. If the users try to use it on large
distance it will not work but some times if it works it drops the data during transmission
Applications of UWB Technology:
There are number of applications of the ultra wide band technology but most of the applications
of ultra wide band technology depend upon the multimedia or the wireless media. Most of the
applications of the ultra wide band technology are of the indoor appliances or the devices that
operates on the small distances. With the help of short distances it enhances the speed of data

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transmission between the applications. Some of the common or the important applications of the
ultra wide band technology as follows
1. Wireless technologies such as monitors, wireless printers, attach different type of devices with the
help of wireless media to the computer system.
2. With the help of application of the ultra wide band technology users can develop the photographs
without attaching with the computer also.
3. RADAR is also an important application of the UWB technology. Generally this technology is used
in those radar systems that are used to detect the presence of the people under the ground or
behind the walls. Basically they are useful for the rescue teams. It also used in repairing of
industries by detecting the materials like steel or wiring etc in the walls or in the ground.
Generally such radar systems are called as ground penetrating radar GPR.
4. One of the important applications of the ultra wide technology in the engineering field is to
measure the accurate distances between the points and locate the actual position of the working.
In this case UWB work in both circumstances such as indoor and outdoor. Such position
applications of the ultra wide technology can also used as security and for navigation purposes

Advantages of UWB:
1. It can transfer more data at low power.
2. It does not disturbed by normal radio frequency.
3. Power saver for portable devices.

LECTURE # 36: TUTORIAL # 3: Circuit Switching vs. Packet Switching.

LECTURE # 37: REVISION/SUMMARY (LCD), ASSIGNMENT 3

LECTURE # 38: BRIDGE COURSE: Introduction to Multimedia: Definition, Categorization of multimedia,
Major characteristics of multimedia
LEc10RE # 39:
0Nl1-lV lntroductlon to Multlmedla
concept of Non-1emporal and 1emporal Medla. Rypertext and Rypermedla. Presentatlons:
8ynchronlzatlon, Events, 8crlpts and lnteractlvlty.

MULTIMEDIA APPLICATIONS
In recent times, multimedia applications use a collection of multiple media sources e.g. text, graphics,
images, sound/audio, animation and/or video. Some examples of multimedia applications are: business
presentations, online newspapers, distance education, and interactive gaming, advertisements, art,
entertainment, engineering, medicine, mathematics, business, scientific research and spatial temporal
applications.

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Other examples of Multimedia Applications include:
World Wide Web (WWW)
Hypermedia courseware
Video-on-demand
Interactive TV
Computer Games
Virtual reality
Digital video editing and production systems
Multimedia Database systems
Video conferencing and Teleconferencing
Groupware
Home shopping
Games
Virtual reality
Digital video editing and production systems
Multimedia Database systems

Multimedia definition
multi
~ many, numerous (lat.)
media
1 .an object on which data can represented and/or stored,
e.g. paper, internet, CD
2. form of communicating spatial information,
e.g. graphics, text, animation, audio etc

Multimedia definition
the interaction of multiple forms of media supported by the computer."
Cartwright et al. (1999)
the use of computers to present text, graphics, video, animation, and sound in an integrated way.
Multimedia characteristics:
1. interactivity, i.e. user controls in real time the course of the presentation
2. diversity of media to represent information
3. concurrent presentation of different media, i.e. user perceives different media at the same time
Temporal and Non-Temporal Media:
Media types can be divided in two types depending on their behavior with respect to time.
1emporal Lypes change wlLh Llme, whereas non-Lemporal Lypes remaln Lhe same.
Temporal
o Audio
o Video
o Music
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o Animation

Non-temporal
o Text
o Images

o Graphics
Also term-pairs dynamic/static time-based/non-time-based and continuous/discrete can be used
here.
Hypertext and Hypermedia
Hypertext - Hypertext is basically the same as regular text - it can be stored, read,
searched, or edited - with an important exception: hypertext is text with pointers to other
text. The browsers let you deal with the pointers in a transparent way -- select the pointer,
and you are presented with the text that is pointed to.
Hypermedia - Hypermedia is a superset of hypertext. Hypermedia documents contain
links not only to other pieces of text, but also to other forms of media - sounds, images,
and movies. Images themselves can be selected to link to sounds or documents. This
means that browsers might not display a text file, but might display images or sound or
animations. Hypermedia simply combines hypertext and multimedia
LEc10RE # 40:
compresslon 1echnlques: Baslc concepts of compresslon.
Lossless data compression
Lossless data compression make use of data compression algorithms that allows the exact original data to be
reconstructed from the compressed data. This can be contrasted to lossy data compression, which does not allow
the exact original data to be reconstructed from the compressed data. Lossless data compression is used in many
applications. For example, it is used in the popular ZIP file format and in the Unix tool gzip. It is also often used as
a component within lossy data compression technologies.

Lossless compression is used when it is important that the original and the decompressed data be identical, or
when no assumption can be made on whether certain deviation is uncritical. Typical examples are executable
programs and source code. Some image file formats, notably PNG, use only lossless compression, while others
like TIFF and MNG may use either lossless or lossy methods. GIF uses a lossless compression method, but most
GIF implementations are incapable of representing full color, so they quantize the image (often with dithering) to
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256 or fewer colors before encoding as GIF. Color quantization is a lossy process, but reconstructing the color
image and then re-quantizing it produces no additional loss.
Lossless compression techniques
Lossless compression methods may be categorized according to the type of data they are designed to compress.
Some main types of targets for compression algorithms are text, executables, images, and sound. Whilst, in
principle, any general-purpose lossless compression algorithm (general-purpose means that they can handle all
binary input) can be used on any type of data, many are unable to achieve significant compression on data that is
not of the form that they are designed to deal with. Sound data, for instance, cannot be compressed well with
conventional text compression algorithms.

Most lossless compression programs use two different kinds of algorithms: one which generates a statistical model
for the input data, and another which maps the input data to bit strings using this model in such a way that
"probable" (e.g. frequently encountered) data will produce shorter output than "improbable" data. Often, only the
former algorithm is named, while the second is implied (through common use, standardization etc.) or
unspecified.

Statistical modelling algorithms for text (or text-like binary data such as executables) include:

Burrows-Wheeler transform (BWT; block sorting preprocessing that makes compression more efficient)
LZ77 (used by Deflate)
LZW
PPM
Encoding algorithms to produce bit sequences are:
Huffman coding (also used by Deflate)
Arithmetic coding
Many of these methods are implemented in open-source and proprietary tools, particularly LZW and its variants.
Some algorithms are patented in the USA and other countries and their legal usage requires licensing by the
patent holder. Because of patents on certain kinds of LZW compression, some open source activists encouraged
people to avoid using the Graphics Interchange Format (GIF) for compressing image files in favor of Portable
Network Graphics PNG, which combines the LZ77-based deflate algorithm with a selection of domain-specific
prediction filters. However, the patents on LZW have now expired.

Many of the lossless compression techniques used for text also work reasonably well for indexed images, but there
are other techniques that do not work for typical text that are useful for some images (particularly simple
bitmaps), and other techniques that take advantage of the specific characteristics of images (such as the common
phenomenon of contiguous 2-D areas of similar tones, and the fact that colour images usually have a
preponderance to a limited range of colours out of those representable in the colour space).

As mentioned previously, lossless sound compression is a somewhat specialised area. Lossless sound compression
algorithms can take advantage of the repeating patterns shown by the wave-like nature of the data - essentially
using models to predict the "next" value and encoding the (hopefully small) difference between the expected
value and the actual data. If the difference between the predicted and the actual data (called the "error") tends to
be small, then certain difference values (like 0, +1, -1 etc. on sample values) become very frequent, which can be
exploited by encoding them in few output bits. This is called "Delta coding".

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Lossy data compression
A lossy data compression method is one where compressing data and then decompressing it retrieves data that
may well be different from the original, but is "close enough" to be useful in some way. Lossy data compression is
used frequently on the Internet and especially in streaming media and telephony applications. These methods are
typically referred to as codecs in this context. Most lossy data compression formats suffer from generation loss:
repeatedly compressing and decompressing the file will cause it to progressively lose quality. This is in contrast
with lossless data compression.

Types of lossy compression:
There are two basic lossy compression schemes:
In lossy transform codecs, samples of picture or sound are taken, chopped into small segments, transformed
into a new basis space, and quantized. The resulting quantized values are then entropy coded.
In lossy predictive codecs, previous and/or subsequent decoded data is used to predict the current sound sample
or image frame. The error between the predicted data and the real data, together with any extra information
needed to reproduce the prediction, is then quantized and coded.

In some systems the two techniques are combined, with transform codecs being used to compress the error
signals generated by the predictive stage.

Lossless vs. lossy compression
The advantage of lossy methods over lossless methods is that in some cases a lossy method can produce a much
smaller compressed file than any known lossless method, while still meeting the requirements of the application.

Lossy methods are most often used for compressing sound, images or videos. The compression ratio (that is, the
size of the compressed file compared to that of the uncompressed file) of lossy video codecs are nearly always far
superior to those of the audio and still-image equivalents. Audio can be compressed at 10:1 with no noticeable
loss of quality, video can be compressed immensely with little visible quality loss, eg 300:1. Lossily compressed
still images are often compressed to 1/10th their original size, as with audio, but the quality loss is more
noticeable, especially on closer inspection.

When a user acquires a lossily-compressed file, (for example, to reduce download-time) the retrieved file can be
quite different from the original at the bit level while being indistinguishable to the human ear or eye for most
practical purposes. Many methods focus on the idiosyncrasies of the human anatomy, taking into account, for
example, that the human eye can see only certain frequencies of light. The psycho-acoustic model describes how
sound can be highly compressed without degrading the perceived quality of the sound. Flaws caused by lossy
compression that are noticeable to the human eye or ear are known as compression artifacts.
Lossless compression algorithms usually exploit statistical redundancy in such a way as to represent the sender's
data more concisely, but nevertheless perfectly. Lossless compression is possible because most real-world data
has statistical redundancy. For example, in English text, the letter 'e' is much more common than the letter 'z',
and the probability that the letter 'q' will be followed by the letter 'z' is very small.

Another kind of compression, called lossy data compression, is possible if some loss of fidelity is acceptable. For
example, a person viewing a picture or television video scene might not notice if some of its finest details are
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removed or not represented perfectly. Similarly, two clips of audio may be perceived as the same to a listener
even though one is missing details found in the other. Lossy data compression algorithms introduce relatively
minor differences and represent the picture, video, or audio using fewer bits.

Lossless compression schemes are reversible so that the original data can be reconstructed, while lossy schemes
accept some loss of data in order to achieve higher compression. However, lossless data compression algorithms
will always fail to compress some files; indeed, any compression algorithm will necessarily fail to compress any
data containing no discernible patterns. Attempts to compress data that has been compressed already will
therefore usually result in an expansion, as will attempts to compress encrypted data.

In practice, lossy data compression will also come to a point where compressing again does not work, although an
extremely lossy algorithm, which for example always removes the last byte of a file, will always compress a file up
to the point where it is empty.
LEc10RE # 41
8tlll lmage compresslon: 1PE0 compresslon, Features of 1PE02000
The JPEG Algorithm
The Joint Photographic Experts Group developed the JPEG algorithm in the late 1980s and early
1990s. They developed this new algorithm to address the problems of that era, specifically the fact that
consumer-level computers had enough processing power to manipulate and display full color photographs.
However, full color photographs required a tremendous amount of bandwidth when transferred over a
network connection, and required just as much space to store a local copy of the image. Other compression
techniques had major tradeoffs. They had either very low amounts of compression, or major data loss in the
image. Thus, the JPEG algorithm was created to compress photographs with minimal data loss and high
compression ratios.
Due to the nature of the compression algorithm, JPEG is excellent at compressing full-color (24-bit)
photographs, or compressing grayscale photos that include many different shades of gray. The JPEG
algorithm does not work well with web graphics, line art, scanned text, or other images with sharp
transitions at the edges of objects. The reason this is so will become clear in the following sections. JPEG
also features an adjustable compression ratio that lets a user determine the quality and size of the final
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image. Images may be highly compressed with lesser quality, or they may forego high compression, and
instead be almost indistinguishable from the original.
JPEG compression and decompression consist of 4 distinct and independent phases. First, the image is
divided into 8 x 8 pixel blocks. Next, a discrete cosine transform is applied to each block to convert the
information from the spatial domain to the frequency domain. After that, the frequency information is
quantized to remove unnecessary information. Finally, standard compression techniques compress the final
bit stream. This report will analyze the compression of a grayscale image, and will then extend the analysis
to decompression and to color images.
Phase One: Divide the Image
Attempting to compress an entire image would not yield optimal results. Therefore, JPEG divides the image
into matrices of 8 x 8 pixel blocks. This allows the algorithm to take advantage of the fact that similar
colors tend to appear together in small parts of an image. Blocks begin at the upper left part of the image,
and are created going towards the lower right. If the image dimensions are not multiples of 8, extra pixels
are added to the bottom and right part of the image to pad it to the next multiple of 8 so that we create only
full blocks. The dummy values are easily removed during decompression. From this point on, each block
of 64 pixels is processed separately from the others, except during a small part of the final compression step.
Phase one may optionally include a change in colorspace. Normally, 8 bits are used to represent one pixel.
Each byte in a grayscale image may have the value of 0 (fully black) through 255 (fully white). Color
images have 3 bytes per pixel, one for each component of red, green, and blue (RGB color). However,
some operations are less complex if you convert these RGB values to a different color representation.
Normally, JPEG will convert RGB colorspace to YCbCr colorspace. In YCbCr, Y is the luminance, which
represents the intensity of the color. Cb and Cr are chrominance values, and they actually describe the color
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itself. YCbCr tends to compress more tightly than RGB, and any colorspace conversion can be done in
linear time. The colorspace conversion may be done before we break the image into blocks; it is up to the
implementation of the algorithm.
Finally, the algorithm subtracts 128 from each byte in the 64-byte block. This changes the scale of the byte
values from 0255 to 128127. Thus, the average value over a large set of pixels will tend towards zero.
The following images show an example image, and that image divided into an 8 x 8 matrix of pixel blocks.
The images are shown at double their original sizes, since blocks are only 8 pixels wide, which is extremely
difficult to see. The image is 200 pixels by 220 pixels, which means that the image will be separated into
700 blocks, with some padding added to the bottom of the image. Also, remember that the division of an
image is only a logical division, but in figure, lines are used to add clarity.
Before: After:
Phase Two: Conversion to the Frequency Domain
At this point, it is possible to skip directly to the quantization step. However, we can greatly assist that
stage by converting the pixel information from the spatial domain to the frequency domain. The conversion
will make it easier for the quantization process to know which parts of the image are least important, and it
will de-emphasize those areas in order to save space.
Currently, each value in the block represents the intensity of one pixel (remember, our example is a
grayscale image). After converting the block to the frequency domain, each value will be the amplitude of a
unique cosine function. The cosine functions each have different frequencies. We can represent the block
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by multiplying the functions with their corresponding amplitudes, then adding the results together.
However, we keep the functions separate during JPEG compression so that we may remove the information
that makes the smallest contribution to the image.
Human vision has a drop-off at higher frequencies, and de-emphasizing (or even removing completely)
higher frequency data from an image will give an image that appears very different to a computer, but looks
very close to the original to a human. The quantization stage uses this fact to remove high frequency
information, which results in a smaller representation of the image.
There are many algorithms that convert spatial information to the frequency domain. The most obvious of
which is the Fast Fourier Transform (FFT). However, due to the fact that image information does not
contain any imaginary components, there is an algorithm that is even faster than an FFT. The Discrete
Cosine Transform (DCT) is derived from the FFT, however it requires fewer multiplications than the FFT
since it works only with real numbers. Also, the DCT produces fewer significant coefficients in its result,
which leads to greater compression. Finally, the DCT is made to work on one-dimensional data. Image
data is given in blocks of two-dimensions, but we may add another summing term to the DCT to make the
equation two-dimensional. In other words, applying the one-dimensional DCT once in the x direction and
once in the y direction will effectively give a two-dimensional discrete cosine transform.
The 2D discrete cosine transform equation is given in figure, where C(x) = 1/2 if x is 0, and C(x) = 1 for
all other cases. Also, f (x, y) is the 8-bit image value at coordinates (x, y), and F (u, v) is the new entry in
the frequency matrix.
( ) ( ) ( ) ( )
( ) ( )
(
+ +
=

= =
7
0
7
0
16
1 2
cos
16
1 2
cos ,
4
1
,
x y
v y u x
y x f v C u C v u F

Figure: DCT Equation

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We begin examining this formula by realizing that only constants come before the brackets. Next, we
realize that only 16 different cosine terms will be needed for each different pair of (u, v) values, so we may
compute these ahead of time and then multiply the correct pair of cosine terms to the spatial-domain value
for that pixel. There will be 64 additions in the two summations, one per pixel. Finally, we multiply the
sum by the 3 constants to get the final value in the frequency matrix. This continues for all (u, v) pairs in
the frequency matrix. Since u and v may be any value from 07, the frequency domain matrix is just as
large as the spatial domain matrix.
The frequency domain matrix contains values from -10241023. The upper-left entry, also known as the
DC value, is the average of the entire block, and is the lowest frequency cosine coefficient. As you move
right the coefficients represent cosine functions in the vertical direction that increase in frequency.
Likewise, as you move down, the coefficients belong to increasing frequency cosine functions in the
horizontal direction. The highest frequency values occur at the lower-right part of the matrix. The higher
frequency values also have a natural tendency to be significantly smaller than the low frequency coefficients
since they contribute much less to the image. Typically the entire lower-right half of the matrix is factored
out after quantization. This essentially removes half of the data per block, which is one reason why JPEG is
so efficient at compression.
Computing the DCT is the most time-consuming part of JPEG compression. Thus, it determines the worst-
case running time of the algorithm. The running time of the algorithm is discussed in detail later. However,
there are many different implementations of the discrete cosine transform. Finding the most efficient one
for the programmers situation is key. There are implementations that can replace all multiplications with
shift instructions and additions. Doing so can give dramatic speedups, however it often approximates
values, and thus leads to a lower quality output image. There are also debates on how accurately certain
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DCT algorithms compute the cosine coefficients, and whether or not the resulting values have adequate
precision for their situations. So any programmer should use caution when choosing an algorithm for
computing a DCT, and should be aware of every trade-off that the algorithm has.
Phase Three: Quantization
Having the data in the frequency domain allows the algorithm to discard the least significant parts of the
image. The JPEG algorithm does this by dividing each cosine coefficient in the data matrix by some
predetermined constant, and then rounding up or down to the closest integer value. The constant values that
are used in the division may be arbitrary, although research has determined some very good typical values.
However, since the algorithm may use any values it wishes, and since this is the step that introduces the
most loss in the image, it is a good place to allow users to specify their desires for quality versus size.
Obviously, dividing by a high constant value can introduce more error in the rounding process, but high
constant values have another effect. As the constant gets larger the result of the division approaches zero.
This is especially true for the high frequency coefficients, since they tend to be the smallest values in the
matrix. Thus, many of the frequency values become zero. Phase four takes advantage of this fact to further
compress the data.
The algorithm uses the specified final image quality level to determine the constant values that are used to
divide the frequencies. A constant of 1 signifies no loss. On the other hand, a constant of 255 is the
maximum amount of loss for that coefficient. The constants are calculated according to the users wishes
and the heuristic values that are known to result in the best quality final images. The constants are then
entered into another 8 x 8 matrix, called the quantization matrix. Each entry in the quantization matrix
corresponds to exactly one entry in the frequency matrix. Correspondence is determined simply by
coordinates, the entry at (3, 5) in the quantization matrix corresponds to entry (3, 5) in the frequency matrix.
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A typical quantization matrix will be symmetrical about the diagonal, and will have lower values in the
upper left and higher values in the lower right. Since any arbitrary values could be used during
quantization, the entire quantization matrix is stored in the final JPEG file so that the decompression routine
will know the values that were used to divide each coefficient.
Figure 1 shows an example of a quantization matrix.

Figure 1: Sample Quantization Matrix

The equation used to calculate the quantized frequency matrix is fairly simple. The algorithm takes a value
from the frequency matrix (F) and divides it by its corresponding value in the quantization matrix (Q). This
gives the final value for the location in the quantized frequency matrix (F
quantize
). Figure 2 shows the
quantization equation that is used for each block in the image.
( )
( )
( )
5 . 0
,
,
, +
|
|
.
|
\
|
=
v u Q
v u F
v u F
Quantize

Figure 2: Quantization Equation

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By adding 0.5 to each value, we essentially round it off automatically when we truncate it, without
performing any comparisons. Of course, any means of rounding will work.
Phase Four: Entropy Coding
After quantization, the algorithm is left with blocks of 64 values, many of which are zero. Of course, the
best way to compress this type of data would be to collect all the zero values together, which is exactly what
JPEG does. The algorithm uses a zigzag ordered encoding, which collects the high frequency quantized
values into long strings of zeros.
To perform a zigzag encoding on a block, the algorithm starts at the DC value and begins winding its way
down the matrix, as shown in figure 3. This converts an 8 x 8 table into a 1 x 64 vector.

Figure 3: Zigzag Ordered Encoding

All of the values in each block are encoded in this zigzag order except for the DC value. For all of the other
values, there are two tokens that are used to represent the values in the final file. The first token is a
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combination of {size, skip} values. The size value is the number of bits needed to represent the second
token, while the skip value is the number of zeros that precede this token. The second token is simply the
quantized frequency value, with no special encoding. At the end of each block, the algorithm places an end-
of-block sentinel so that the decoder can tell where one block ends and the next begins.
The first token, with {size, skip} information, is encoded using Huffman coding. Huffman coding scans the
data being written and assigns fewer bits to frequently occurring data, and more bits to infrequently
occurring data. Thus, if a certain values of size and skip happen often, they may be represented with only a
couple of bits each. There will then be a lookup table that converts the two bits to their entire value. JPEG
allows the algorithm to use a standard Huffman table, and also allows for custom tables by providing a field
in the file that will hold the Huffman table.
uC values use delLa encodlng, whlch means LhaL each uC value ls compared Lo Lhe prevlous value, ln zlgzag order.
noLe LhaL comparlng uC values ls done on a block by block basls, and does noL conslder any oLher daLa wlLhln a
block. 1hls ls Lhe only lnsLance where blocks are noL LreaLed lndependenLly from each oLher. 1he dlfference
beLween Lhe currenL uC value and Lhe prevlous value ls all LhaL ls lncluded ln Lhe flle. When sLorlng Lhe uC values,
!LC lncludes a slze fleld and Lhen Lhe acLual uC delLa value. So lf Lhe dlfference beLween Lwo ad[acenL uC values ls
-4, !LC wlll sLore Lhe slze 3, slnce -4 requlres 3 blLs. 1hen, Lhe acLual blnary value 100 ls sLored. 1he slze fleld for
uC values ls lncluded ln Lhe Puffman codlng for Lhe oLher slze values, so LhaL !LC can achleve even hlgher
compresslon of Lhe daLa.
Steps in JPEG compression:
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ILG 2000 |mproved the fo||ow|ng Def|c|enc|es
- oor sub[ecLlve performance aL raLes below 0.23 blLs per plxel (bpp)
- Lack of ablllLy Lo provlde lossy and lossless compresslon ln Lhe same codesLream
- Lack of robusLness Lo blL errors ln Lhe compressed lmage
- oor performance wlLh compuLer-generaLed lmagery
- oor performance wlLh compound documenLs (LexL and lmage)

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LEc10RE # 42 A0VANcE0 8YLLAB08 c00R8E: L2W codlng 1echnlque:
LEc10RE # 43
Vldeo compresslon: MPE0- 1&2 compresslon 8chemes, MPE0-4 Natural Vldeo compresslon:
MPEG-4 Standard:
Created by Motion Picture Experts Group, a working group of ISO/IEC
Started in 1995
International Standard in 1999
Version 2 International Standard in 2000
MPEG-1, MPEG-2, MHEG-5
MPEG-1: a standard for compressed digital video, bit rates 1,5Mbits/s
MPEG-2: a family of compression profiles and levels up to HDTV
MHEG-5: extended multimedia and interaction Functionality
MPEG-4
MPEG-4 defines coding of all types of multimedia information
1) allows more interaction by the user
2) includes some new video compression methods
3) includes methods for audio and voice encoding
4) scalable
5) object based, allows scenes to be composed of natural and synthetic objects
Very different from MPEG-1&2, more than MHEG-5
Why MPEG4
Mpeg-1&2 are used well beyond their intended scope
New areas where a/v content is needed
Communications, computing and entertainment are constantly converging
Target of MPEG-4
to prevent proprietary standards
to supply uniform (and top-level) a/v encoders and decoders
to act as frame work for different kinds of a/v content and viewers

Goals of MPEG-4:
reusable and reconfigurable content
synthetic and natural content combined
high level of user interaction
scalability
content-based Qos
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MPEG-4 Systems: Objects
A scene consists of audio-visual objects (AVOs)
Objects can be natural (images, video, voice) or synthetic (text, animation, video)
Objects are given a position in 2-D or 3-D space
Objects can be encoded and decoded separately much better compression rates and different quality
Objects can be dynamically added and removed new kind of scalability
Objects can be dynamically moved, changed and Created
MPEG-4 Natural Video Coding:
Object based,allows scenes to be composed of natural and synthetic objects. Objects can be natural (images, video,
voice) or synthetic (text, animation, video)
MPEG4 targets a wide range of av apps (from Digital TV to mobile multimedia)
Provides interface points to existing systems
Progressive and interlaced video formats
Originally targeted for three bit-rate ranges:
< 64Kbit/s
64 - 384 Kbit/s
384kbit - 4Mbit/s
Also higher bit-rates are supported
Natural Video Coding:
Improved compression efficiency over MPEG- 1&2
Contains features needed in low bit-rate systems:
error resiliance
unequal error protection
selectable object update rates
temporal and spatial scalability
LEc10RE # 44: LEc10RE # 44: LEc10RE # 44: LEc10RE # 44:
Audlo and 8peech codlng:
Several different audio and speech encoding methods
Audio is also represented as objects that can be dynamically edited
Audio objects may be given a location in 3-D space
Environmental spatialization

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Natural audio coding
MPEG-4 audio tool is compatible with MPEG- 2's Advanced Audio Coding (AAC)
Target bit-rates:
16Kbit/s for AM-radio quality
64Kbit/s for transparent quality
Now, bit-rates are considerably lower than in MP3
Natural speech coding:
Harmonic Vector Excitation Coder (HVXC)
handles operation at 2 and 4Kbit/s
sampling frequency 8KHz
bandwidth 300-3400Hz
Code-Exited Linear Predictive Coder (CELP)
4-24Kbit/s
Sampling frequency 8khz or 16khz
bandwidth 50-7000Hz
bit-rate scalability of 200 bit/s

LEc10RE # 46:
compresslon of synthetlc graphlcal objects:
[ ALREA0Y c0VERE0 ]
LEc10RE # 46: TUTORIAL # 4: A PROGRAM FOR COMPRESSING A JPEG FILE.

LEc10RE # 47: OPEN ASSIGNMENT

LEc10RE # 48: REVISION/SUMMARY (LCD), ASSIGNMENT 4

LEc10RE # 49: BRIDGE COURSE: Examples of Multimedia Systems
LEc10RE # 60:
Archltecture for Multlmedla 8upport: Multlmedla Pc/Workstatlon Archltecture,

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LEc10RE # 61

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Multimedia workstation components
In view of the requirements imposed on a multimedia system by various media streams, a typical
system should contain at least the following components:

general-purpose microprocessor
large primary storage (mernory)
huge permanent secondary storage (disks)
dedicated media processor
graphics, video, and sound equipment
communication adapters

The general-purpose processor is responsible for the system management, operating system
tasks, and for standard (non multimedia) data processing. The primary storage is needed for
processing, copying, and temporary storage of multimedia information; multimedia objects tend
to be of very large size. The secondary storage in the fom of hard disks. disk arrays, and read
only and rewritable optical disks, provides permanent archiving and distribution of multimedia
data.
The need for real-time processing of isochronous data which imposes a severe constraint on
processing times, entails the use of secondary, dedicated real-time processors (DSPs) to ensure
delay and delay jitter guarantees. Finally, the communication adapters are needed to provide
connectivity to extemal networks at the speeds required by the time-critical multimedia
applications. Multimedia information is displayed and presented to the end user by means of the
audio and video adapters, speakers, monitors, and other multimedia input/output equipment.
LEc10RE # 61:
characterlstlcs of MMX lnstructlon set, l/0 systems: lEEE 1394 lnterface:
MMX (lnstructlon set):
MMX instruction set
What is MMX?
The acronym "MMX" stands for MultiMedia eXtensions. MMX technology is something which Intel
developed. Its goal was to increase the speed at which certain "multimedia" operations are performed. And,
in fact, MMX technology improves the performance of current and future graphics and communications
applications while maintaining compatibility with the existing Intel Architecture (IA) software base.
So in other words, Intel added enhancements to their processors which could speed up multimedia
instructions while remaining compatible with everything already out there. MMX is an extension of IA. In
fact, MMX is IA's most significant enhancement since 1985, when the Intel 386 processor was released. The
386 extended the architecture to 32 bits.
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MMX includes new instructions and data types to achieve increased levels of performance on the host CPU
by exploiting the parallelism inherent in a great deal of the algorithms in these applications.
MMXcan deliver 50%-100% performance gains for multimedia and communications applications over the
same applications run on the same type of processor but without MMX technology.
It's worth noting that Intel, smart people that they are, designed MMX technology so that it would scale well
with processor operating frequencies and future architecture generations. It has been integrated into the
Pentium, and will soon be integrated into the P6 (aka Pentium Pro) processors, which give these processors
an extra boost. MMX will also appear on all future IA processors. And although Intel did create MMX
technology, both AMD and Cyrix are incorporating it into their next generation processors, the K6 and M2,
respectively.
How Does MMX Work?
To implement MMX technology, Intel created 57 new instructions.
MMX technology's potential target applications share several characteristics:
Small native data types (such as 8-bit pixels, 16-bit audio samples)
Compute-intensive recurring operations performed on these data types
A lot of inherent parallelism
These things pointed the MMX definition team in the direction of a single-instruction multiple-data (SIMD)
architecture, in which one instruction performs the same operation on multiple data elements in parallel.
This parallel operation on relatively small data elements is the main source of MMX's performance boost.
The benefits of a SIMD architecture have been identified by other processor architectures, such as the Sun
Microsystems SPARC-Visual Instruction Set (VIS) and the Hewlett-Packard PA-RISC 2.0 MAX-2
instruction set. SIMD architecture has been used for years to provide high performance in a wide range of
systems; it's a proven technology.
Before now, when processing 8- or 16-bit data, the existing 32- or 64-bit CPU bandwidth and processing
resources on Intel processors were underutilized. Only the lower order 8 or 16 bits were manipulated,
leaving the remaining bits unusued. MMX allows full utilization of the wide processing capabilities of the
CPU.
For this paper, like in my primary sources, a data width of 64 bits was chosen. This was for two reasons:
first, the authors' studies showed that using 64 bits of packed elements would enable a fairly substantial
performance boost, and also because the Pentium and Pentium Pro processors use 64-bit wide data busses.
MMX had a couple of design goals which are very important. For the most part they were listed earlier, but
I'm going to list them again, since they really are important.
1. MMX had to substantially improve the performance of multimedia, communications, and other
numeric intensive applications
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2. MMX had to be kept independent of the current microarchitectures, so that it would scale easily with
future advanced microarchitecture techniques and higher processor frequencies in future Intel
processors.
3. MMX processors had to retain backwards compatibility with non-MMX processors. Software must
run without modification on a processor with MMX technology.
4. They had to ensure the coexistence of of existing applications and new applications using MMX
technology.
This last point is important. Modern processors and operating systems can run multiple applications
simultaneously (aka multitasking). New applications which used the new MMX instructions had to be able
to multitask with any other applications. This put some constraints on the MMX technology definition. They
couldn't create a new MMX state or mode (in other words, no new registers) because then operating systems
would have needed to be modified to take care of these new additions.
The main technique for maintaining compatibility of MMX technology was to "hide" it inside the existing
floating-point state and registers (current operating systems and applications are designed to work with the
floating-point state). An operating system doesn't need to know if MMX technology is present, since it's
hidden in the floating-point state. Applications have to check for the presence of MMX technology, and if
it's built into the processor they use the new instructions.
The MMX technology definition process was unusual. In what seems a rare event in the modern computer
industry, it was the engineers and not the managers who led the way. A group of architects and software
engineers analyzed the potential performance of existing and future applications, including graphics, MPEG
video, speech systhesis, speech compression, speech recognition, image processing, 3D graphics, video
conferencing, modems, and audio. They met with external software developers to learn what they would
need from a new IA processor in order to enhance their multimedia and communications applications. The
applications were analyzed to identify the most compute-intensive parts, which were then analyzed in detail
using computer-aided engineering tools. These studies (and the performance potential they showed)
convinced Intel of the need to integrate the new technology ASAP, and to fully convert all IA processors to
use MMX technology.

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Main Features
The table below is a summary of the MMX instruction set.
Opcode Options Description
Padd[b/w/d]
Psub[b/w/d]
Wrap around and
saturate
Parallel Add and Subtract of packed eight bytes, four 16-bit
words, or two 32-bit doublewords.
Pcmpeq[b/w/d]
Pcmpgt[b/w/d]
Equal or greater than
Parallel Compare of eight bytes, four 16-bit words, or two 32-bit
doublewords. Result is mask of 1s if true or 0s if false.
Pmullw
Pmulhw
Result is low or high
order bits
Parallel Multiply of four signed 16-bit words. Low-order or
high-order 16-bits of the 32-bit result are chosen.
Pmaddwd
Word to doubleword
conversion
Parallel Multiply-Add of four signed 16-bit words. Adjacent
pairs of 32-bit results are added together. Result is a doubleword.
Psra[w/d]
Psll[w/d/q]
Psrl[w/d/q]
Shift count in
register or
immediate
Parallel Shift of 4 words, 2 doublewords, or the full 64 bits are
shifted arithmetic right, logical right and left.
Punpckl[bw/wd/dq]
Punpckh[bw/wd/dq]

Parallel Unpacking (interleaved merge) of eight bytes, four 16-
bit words, or two 32-bit doublewords.
Packss[wb/dw] Always saturate Parallel packing of doublewords to words or words to bytes.
Pand
Pandn
Por
Pxor

64-bit bitwise logical operations.
Mov[d/q]

Moves 32 or 64 bits to and from memory to MMX registers or
between MMX registers. 32-bits can be moved between MMX
and integer registers.
Emms Empty FP registers tag bits.

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A lot of multimedia applications execute the same instructions on many pieces of data in a large data set.
Standard processors can only process one piece of data with each instruction. MMX technology processes
several pieces of data with each instruction. It's a simple type of parallelism which provides a big
performance boost for a lot of multimedia applications. Typically, data elements are small: 8 bits per
element for pixels or 8 bits for each pixel component (red, green, and blue) used in graphics and video, 16
bits per element for audio samples or as a higher-precision backup for 8-bit operations, and 32 bits per
element for general computing and some 3D graphics algorithms.
Because of this, MMX defines new data types, which are 64 bits in total size, and are composed of
independent smaller-size data elements. They're called "packed data types". Each element within a packed
data type is a fixed-point integer. The programmer controls the place of the fixed point within each element,
and is responsible for its placement throughout the calculation. This control means an extra burden for
programmers, but it also gives them a lot of flexibility to choose and change fixed-point formats during the
application to fully control the dynamic range of their values.
The four data types defined by MMX are:
1. Packed byte - Eight bytes packed into one 64-bit quantity
2. Packed word - Four words packed into one 64-bit quantity
3. Packed doubleword - Two doublewords packed into one 64-bit quantity
4. Quadword - One 64-bit quantity
MMX instructions are defined to perform the parallel operations on the multiple data elements packed into
the new 64-bit data types. MMX technology extends the basic integer instructions into SIMD versions.
These instructions include the standard add, subtract, multiply, compare, and shift, data-type conversion
functions (to facilitate converting between the new data types), instructions to support 64-bit operations (64-
bit memory moves, 64-bit logical operations), and a multiply-add operation (because a lot of multimedia
applications perform multiply-accumulate operations).
For packed data types, MMX has its most complete instruction support for packed-word (16-bit) data types,
since they found that 16-bit data was the most general and useful for the category of multimedia
applications. It also acts as higher-precision backup for operations on byte data. As stated earlier, a total of
57 new MMX instructions were added overall to the IA.
Now, we can easily notice that the MMX instructions differ from one another by a few characteristics.
Different instructions are supplied to do the same operation on different data types. One operation may work
on a on a packed-byte, while another will work on a packed-word. Some instructions also differ because
they treat a value as signed or unsigned.
A major feature of MMX instructions is saturation arithmetic. Saturation arithmetic is important to many
graphics routines. As an example, assume you add together two medium-red pixels. Saturating arithmetic
ensures the result is a dark red or black. It's certainly different than regular integer math, where you could
perform the above operation and end up with a light-colored result. In other words, saturation arithmetic
handles "wrap-around" problems. This is a very handy thing to have.
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MMX supports both signed and unsigned saturating arithmetic.
The parallelism and saturating arithmetic in MMX are useful in some video conferencing compression
schemes. Instead of directly encoding each frame in a video sequence, it's better to first compute the
differences between the current frame and a recent previous frame. If the two frames are very similar (which
is usually the case in video) it's easy to see from the difference frame that results can be represented with
less information than the original. So for all the pixels in the frame, a pixel-to-pixel difference is computed.
What's really nice about this is that all the differences can be computed in parallel, since they're independent
operations.
This can cause a problem, though. Subtracting two 8-bit unsigned pixels can result in a 9-bit negative
number. You can get around this using saturating arithmetic, though. What you do is use unsigned
saturating subtraction to subtract pixel A from pixel B, and then do the reverse. One will be positive, and the
other will be 0. We don't know which is which, but that's ok, because we can just use a logical OR operation
to combine the results. This operation can be done in parallel on 8 bytes at a time, which provides really
good performance (even better considering this operation is used a lot).
Saturating arithmetic also has value in traditional graphics. Gouraud shading is a standard way to render 3D
images so that they look more realistic. It works by shading polygons by interpolating color values across
scan lines during rendering. Somewhere along a scan line calculations can start to overflow. Unless
precautions are taken, overflow can occur and generate a completely different result from that expected.
Saturation makes sure this doesn't happen.
Changing Data Types
MMX uses two sets of instructions (Pack and Unpack) to convert between the different MMX
data types. Unpack takes small data types and produce large ones (ie, converting 16-bit to 32-bit
words). Unpack takes two operands and interleaves them. If you want to Unpack 16-bit into 32-
bit words, you can take one operand of 16-bit information and interleave it with another operand
filled with 0s. The result of this is a 32-bit word with 0 in the most significant bits.
The Pack instructions convert data from a larger data type to a smaller one. If you expand 8-bit
bytes into 16-bits to do some calculation, you could pack the final result back into 8-bit bytes
before storing them back in memory.
The Unpack instructions are pretty powerful when data organized in one format in memory
needs to be rearranged while executing some algoithm in order to expose the parallelism that
MMX works with. An example of this is the Inverse Discrete Cosine Transform (ICDT) used in
the JPEG image decompression algorithm. This algorithm takes a 2D array of data and operates
first on rows of data, and then on columns of data. In memory, an array is usually laid out one
row after another. MMX lets you manipulate rows in this type of memory organization really
easily, since row elements are in subsequent addresses. But this organization doesn't work well
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on columns. In order to do this in parallel, the array has to be transposed, so that the columns
become rows.
But guess what? The MMX Unpack instruction can be used to transpose an array! It's a two-step
process: in phase 1 the Unpack instruction is used to interleave the word (16-bit) elements of
adjacent rows, and in phase 2 the results of the first phase are unpacked again, this time using
doubleword (32-bit) Unpack instructions to create the desired outputs.
lEEE 1394 lN1ERFAcE:
The IEEE 1394 interface, developed in late 1980s and early 1990s by Apple as FireWire, is a serial bus interface
standard for high-speed communications and isochronous real-time data transfer. The 1394 interface is comparable
with USB and often those two technologies are considered together, though USB has more market share. Apple first
included FireWire in some of its 1999 models, and most Apple computers since the year 2000 have included FireWire
ports, though, as of 2012, nothing beyond the 800 version (IEEE-1394b). The interface is also known by the
brand i.LINK (Sony), and Lynx (Texas Instruments). IEEE 1394 replaced parallel SCSI in many applications,
because of lower implementation costs and a simplified, more adaptable cabling system. The 1394 standard also
defines a backplane interface, though this is not as widely used.
IEEE 1394 is the High-Definition Audio-Video Network Alliance (HANA) standard connection interface for A/V
(audio/visual) component communication and control.
[3]
FireWire is also available in wireless, fiber optic,
and coaxial versions using the isochronous protocols.

1394 0bjectlves:
Digital Interface
Physically Small
Easy to Use
Hot Pluggable
Inexpensive
Scalable Architecture
Flexible Topology
Fast
Open Standard

Definition - What does IEEE 1394 Interface mean?
The IEEE 1394 interface is an electronic standard that is used to connect computers. It includes a plug-and-
socket connection with a serial bus interface. Up to 63 devices can be connected at the same time with
(relatively) high data transfer speeds.

This IEEE 1394 standard has proved very popular for communication between computers and their peripheral
devices. The IEEE 1394 interface is commonly known as FireWire.
Techopedia explains IEEE 1394 Interface
The standard IEEE 1394 was developed by Apple in 1986. Most consumers simply refer to it as "FireWire." It is
widely available in the market in the forms of coaxial, wireless and fiber optic cable.
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IEEE 1394 interface has the following features:
Real-time data transferring for multimedia applications
100-800 Mb/s data transfer rate
Hot plugging is allowed (at a minimum ratio)
Line terminators are not required
Plug-and-play
Auto-configuration for many peripheral devices
Same connectors for multiple devices and components

The FireWire system is commonly used to connect digital cameras and other storage devices. FireWire has an
edge over USB in terms of power distribution, reliability and data transfer speed. Moreover, FireWire system has
adopted the functionalities and characteristics of SCSI.

LEc10RE # 62: A0VANcE0 c00R8E: lP1V and VolP
LEc10RE # 63:
0peratlng 8ystem 8upport for Multlmedla 0ata: Resource 8chedullng wlth realtlme conslderatlons:
Operating System Support

1. Real-time Operation
2. Scheduling Algorithms
Requirements for Multimedia:
Processing of continuous data streams
The data to be processed appears in periodic time-intervals.
Typical operations on multimedia data are:
Creation and playout of audio and video packets,
Transmission of audio and video packets,
Compression and decompression of audio and video packets.
Real-time requirements:
Processing must be completed by a specific time (deadline)
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Processing of a multimedia data packet takes approximately the same amount of resources in each time period.

Reservation of Resources

Pessimistic (deterministic) reservation
Takes the worst case into consideration
Reserves too many resources most of the time and thus leads to sub-optimal use of the resources
Optimistic (stochastic) reservation
Reservation is made for the expected value (mean value) of the processing resources
Leads to a good utilization of the resources
But can lead to an overload of the resources (blocking or sub-optimal execution of a process)
Role of a run-time monitor:
Monitors resource usage
In case of overload initiates suitable action, such as scaling (QoS adaptation) or
blocking of the process and withdrawal of the resource.
Real-time Process Specification for Multimedia

Traditional systems usually have
either a scheduling mechanism for time-sharing applications
or a scheduling mechanism for real-time applications
In multimedia systems, a scheduler for both types of applications is required.

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Real-tlme Appllcatlons vs. Multlmedla Appllcatlons

Data in traditional real-time applications (e.g., in process automation and control) typically have hard real-time
requirements.
Multimedia data is generally intended for presentation to the human being as the data sink. This means:
Rare violations of deadlines are acceptable. For example, a macro block decoding failure in an MPEG video that
is caused by late arrival of the data packet can be tolerated; the decoder repeats the corresponding macro block of the
previous frame.
In traditional real-time systems, operations are not always periodic. Multimedia data is periodic. Scheduling and
dispatching algorithms can take advantage of the periodicity.
Resource requirements are easier to predict.
Process Scheduling Goals:
The main goal of process scheduling and dispatching is to ensure that all deadlines of all processes are satisfied.
Additional optimization goals are:
A high average utilization of the resources, leading to high throughput
fast computation of the resource allocation (an efficient algorithm).
The allocation does not necessarily have to be optimal. Complexity theory tells us that the computation of optimal
schedules is NP-complete.

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Scheduling Algorithms:

Requirements
All deadlines must be met.
Resource utilization should be high.
Best-effort requests should not starve.
The algorithm should be efficient, i.e., not use too much CPU time itself. The scheduler can take advantage of the
periodicity of continuous data streams.

Preemptive vs. Non-preemptive Scheduling
Preemptive scheduling
If a process with higher priority requests a resource, the currently active (running)
process is moved to the READY state, the high-priority process gets the CPU, i.e.,
is moved to the RUNNING state.
There are many more process switches, process management overhead is high.

Non-preemptive scheduling
Active processes are not interrupted by high-priority requests.
Process switching occurs less frequently, overhead is lower.
Non-preemptive scheduling generally performs well for processes with short execution times.

Algorithm 1: Earliest Deadline First (EDF)
The process with the earliest deadline is assigned the highest priority.

Process priorities vary over time.
One can show that the EDF algorithm always finds a valid schedule if there is one.
Resources can be used up to 100%.

EDF Scheduling
In most cases of periodic scheduling, the deadline of a request is identical to the end
of the period since the data buffer is needed again.

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Performance
Preemptive scheduling :
maximum possible throughput:

delay of a packet <= 1/Ri
Non-preemptive scheduling (Nagarajan/Vogt, 1992)
same throughput as above
delay of a packet <= 1/Ri + Pi

Algorithm 2: Rate-Monotonic Scheduling (RM):

The process with the highest packet rate is assigned the highest priority.

While streams are running priorities do not change; only when a new stream starts or when a stream ends, priorities
are re-computed.
RM is an algorithm for periodic processes only.

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Rate-Monotonic Scheduling:
Again, we set the deadline to the end of the period:

Performance
Preemptive scheduling (Liu/Layland, 1973):
maximum possible throughput:

delay of a packet <= 1/Ri

Non-preemptive scheduling:
highly sophisticated computation
guaranteed throughput is much less

LEc10RE # 66:
Flle 8ystem, l/0 0evlce Management:

Why Multlmedla Flle 8ystems7

Heterogeneous data types including digital audio, animations and video consume enormous space and bandwidth and
are delay-sensitive

Multlmedla Flle 8ystems :
contaln addltlonal buffers
Traditional file systems perform an open, several reads and close at the end
During read operations, processes wait until I/O is finished but timing is not all that important. The data
eventually comes.
That is, the user pulls the data in one block at a time by repeately calling read calls to get one block after the
other
File servers of this type are often called pull servers (user pulls the data)

For multimedia,
read calls must be at fairly specified times
and
the video server must be able to supply data blocks without a delay
Multimedia file servers, after a start call, begin sending out frames at the required rate. It is up to the user to
handle them at the rate they come in
File servers of this nature are called push servers because they push data at the user

Pull(a) and Push(b) Servers:
Sub[ect:

File Placement

C

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File Placement
Multimedia files

Contiguous Movie Storage

Video, audio, text in single contiguous file per movie
Read one frame in one disk operation and transmit only relevant parts to the user
This organization is not efficient
servers with multiple concurrent output streams (accessing the desired frame from a movie is not easy in a
contiguous file)


File Placement:
Multimedia files
Are very large
Written once but read many times
Accessed sequ
ontiguous Movie Storage
contiguous file)
Are very large
Accessed sequentialy
ontiguous Movie Storage:
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This organization is not efficient when random access is needed (say for a movie editing system) or in video
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Video, audio, text in single contiguous file per movie instead of separate files for each component
when random access is needed (say for a movie editing system) or in video
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instead of separate files for each component
Sub[ect Code:
322711(22)

Sub[ect:

Noncontiguous Movie Storage
Trade

1SS
CSl1/


a) Small disk bloc
b) Large disk blocks

Trade-offs between small, large blocks

Frame index
-
-
Block index (no splitting frames over blocks)
-
-
Block index (splitting frames over blocks allowed)
-
-
-


mall disk blocks -
a frame index for the whole movie
each index points to one frame data (variable frame size)
arge disk blocks
multiple frames in one block (constant block size)
a block index for the whole movie
offs between small, large blocks
Frame index
heavier RAM usage during movie play
little disk wastage
low RAM usage
major disk wastage
low RAM usage
no disk wastage
extra seeks
-

offs between small, large blocks
little disk wastage
low RAM usage
major disk wastage
low RAM usage
no disk wastage
ra seeks
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offs between small, large blocks:

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heavier RAM usage during movie play (due to variable frame sizes )
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(due to variable frame sizes )
Sub[ect Code:
(due to variable frame sizes )
Sub[ect Code: 322711(22) 322711(22)

Sub[ect:

Placing Files for Near Video on Demand

Placing Multiple files on a Single Disk

1S6
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30 frames/sec with a new stream starting every 5minutes
Stream 24 is just starting (stream repeating on the hour every 2 hours)
Frames needed for all 24 streams at that time are in track 1 as
operation
Double buffering is used (playback from one buffer while reading the next 24 frames from the next track)


Organ
This strategy is based on statistical analysi

For a 1000 movie server, top 5 movies represent a total probability of .307, which means that the disk arm
will sta


operation
Organ-pipe distribution of files on server
most popular movie in middle of disk
next most popular either on either side, etc.
will stay in the cylinders allocated to the top five movies about 30% of the time
pipe distribution of files on server
popular movie in middle of disk
y in the cylinders allocated to the top five movies about 30% of the time
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Placing Multiple files on a Single Disk:
pipe distribution of files on server
This strategy is based on statistical analysis of popularity
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s of popularity
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Frames needed for all 24 streams at that time are in track 1 as a single record which can be read in one read
s of popularity
Sub[ect Code:

a single record which can be read in one read
322711(22)


Sub[ect:

Placing Files on Multiple Disks

(a) No striping
(b) Same striping pattern for all files
(c) Staggered striping
(d) Random striping

Caching

Block Caching

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Organize multimedia files on multiple disks
(a) No striping
(d) Random striping
This organization is not a RAID (no error correction is required

Caching

Block Caching
a) Two users, same movie 10 sec out of sync
b) Merging two streams into one


(a) No striping one disk holds all frames of a movie
(d) Random striping
Block Caching
Two users, same movie 10 sec out of sync
Merging two streams into one
Placing Files on Multiple Disks:
one disk holds all frames of a movie
(b) Same striping pattern for all files all movies start from the same disk
Merging two streams into one by running the first movie a bit slower and the other a bit faster for a while
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one disk holds all frames of a movie
all movies start from the same disk
by running the first movie a bit slower and the other a bit faster for a while
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to balance the load on disks
one disk holds all frames of a movie - popular films may cause a strain on the relevant hard disk
keep the blocks in cache, but this wastes memory
Sem - 7th
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balance the load on disks
films may cause a strain on the relevant hard disk
This organization is not a RAID (no error correction is required but high per
Sub[ect Code:
balance the load on disks
but high performance definitely)
formance definitely)
322711(22)

formance definitely)

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File Caching:
Most movies are stored on DVD or tape to save disk space
copy to disk when needed
results in large startup time
keep most popular movies on disk
Can keep first few minutes of all movies on disk
start movie from this while remainder is fetched

l/0 0evlce Management:
In future computer system design, I/O systems will have to support continuous media such as video
and audio, whose system demands are different from those of data such as text. Multimedia
computing requires us to focus on designing I/O systems that can handle real-time demands. Video-
and audio-stream playback and teleconferencing are real-time applications with different I/O demands.
We primarily consider playback applications which require guaranteed real-time I/O throughput. In a
multimedia server, different service phases of a real-time request are disk, small computer systems
interface (SCSI) bus, and processor scheduling. Additional service might be needed if the request must
be satisfied across a local area network. When requests have to be satisfied within deadlines,
traditional real-time systems use scheduling algorithms such as earliest deadline first (EDF) and least
slack time first. However, EDF makes the assumption that disks are preemptable, and the seek-time
overheads of its strict real-time scheduling result in poor disk utilization. We can provide the constant
data rate necessary for real-time requests in various ways that require trade-offs. We can analyze how
trade-offs that involve buffer space affect the performance of scheduling policies. We can also show
that deferred deadlines, which increase buffer requirements, improve system performance
significantly.

LEc10RE # 66

Multlmedla lnformatlon Management: Multlmedla 0atabase 0eslgn:

At the heart of multimedia information systems lies the multimedia database management system. Traditionally, a
database consists of a controlled collection of data related to a given entity, while a database management system, or
DBMS, is a collection of interrelated data with the set of programs used to define, create, store, access, manage, and
query the database. Similarly, we can view a multimedia database as a controlled collection of multimedia data items,
such as text, images, graphic objects, sketches, video, and audio. A multimedia DBMS provides support for
multimedia data types, plus facilities for the creation, storage, access, query, and control of the multimedia database.
The different data types involved in multimedia databases might require special methods
for optimal storage, access, indexing, and retrieval. The multimedia DBMS should accommodate these special
requirements by providing high-level abstractions to manage the different data types, along with a suitable interface
for their presentation. The data types found in a typical multimedia database include
z text;
z images: color, black and white, photographs, maps, and paintings;
z graphic objects: ordinary drawings, sketches, and illustrations, or 3D objects;
z animation sequences: images or graphic objects, (usually) independently generated;
z video: also a sequence of images (called frames), but typically recording a real-life event and usually produced by a
video recorder;
z audio: generated from an aural recording device; and
z composite multimedia: formed from a combination of two or more of the above data types, such as an intermix of
audio and video with a textual annotation.
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Some multimedia data types such as video, audio, and animation sequences also have temporal requirements, which
have implications on their storage, manipulation, and presentation. The problems become more acute when various
data types from possibly disparate sources must be presented within or at a given time. Similarly, images, graphics,
and video data have spatial constraints in terms of their content. Usually, individual objects in an image or a video
frame have some spatial relationship between them. Such relationships usually produce some constraints when
searching for objects in a database

Purpose of a multimedia DBMS:
A multimedia database management system provides a suitable environment for using and managing multimedia
database information. Therefore, it must support the various multimedia data types, in addition to providing facilities
for traditional DBMS functions like database definition and creation, data retrieval, data access and organization, data
independence, privacy, integration, integrity control, version control, and concurrency support.
The functions of a multimedia DBMS basically resemble those of a traditional DMBS. However, the nature of
multimedia information makes new demandsincluding determining what is needed and how to provide that
functionality. Using the general functions provided by a traditional DBMS as a guide, we can describe the purposes of
a multimedia DBMS as follows:

z Integration. Ensures that data items need not be duplicated during different program invocations requiring the data.
z Data independence. Separation of the database and the management functions from the application programs.
z Concurrency control. Ensures multimedia database consistency through rules, which usually impose some form of
execution order on concurrent transactions.
z Persistence. The ability of data objects to persist (survive) through different transactions and program invocations.
z Privacy. Restricts unauthorized access and modication of stored data.
z Integrity control. Ensures consistency of the database state from one transaction to another through constraints
imposed on transactions.
z Recovery. Methods needed to ensure that results of transactions that fail do not affect the persistent data storage.
z Query support. Ensures that the query mechanisms are suited for multimedia data.
z Version control. Organization and management of different versions of persistent objects, which might be required
by applications.

In concurrency control, a transaction is a sequence of instructions executed either completely or not at all. In the latter
case, the database is restored to its previous state. Defining the appropriate granularity for concurrency is a problem in
multimedia databases. Traditional databases use a single record or table as the unit of concurrency; multimedia
databases typically use a single object (or composite object) as the logical unit of access. Thus the single multimedia
object could form the unit of concurrency. In achieving persistence, a simple method is to store the multimedia files in
some operating system files. However, the huge data volumes make this approach costly to implement. Moreover, the
system also needs to store the multimedia metadata and possibly composite multimedia objects. Thus, most
multimedia DBMSs classify the data as either persistent or transient and store only persistent data after transaction
updates. Transient data are used only during program or transaction execution and are removed afterwards.
Traditionally, a query selects a subset of the data objects based on the users description (usually some form of query
language) of what data to access. A query usually involves various attributes, possibly keyword-based or content-
oriented, and is usually interactive. Thus, functions for relevance feedback and query formulation, similarity (rather
than exact) matches, and mechanisms for displaying ranked results are important in a multimedia DBMS.
Version control becomes important when a persistent multimedia object is updated or modified, as some applications
might need to access previous states of the object. A DBMS provides such access through versions of the persistent
objects. For a multimedia DBMS, the huge volumes of data reinforces the importance of efficiently organizing such
versions. Moreover, the available storage might limit the provision of versions. In addition, version management may

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involve not only versions of single objects, but also versions of the complex objects that make up the multimedia
database.

The special nature of multimedia data also makes it important to support new special functions. These include object
composition and decomposition, management of huge volumes of multimedia data, effective storage management,
and information retrieval and handling of spatial and temporal data objects.

Requirements for the multimedia DBMS
For the multimedia DBMS to serve its expected purpose, it must meet certain special requirements.The requirements
are divided into the following broad categories:
z Traditional DBMS capabilities
z Huge capacity storage management
z Information retrieval capabilities
z Media integration, composition, and presentation
z Multimedia query support
z Multimedia interface and interactivity
z Performance

Figure: A high-level architecture for a multimedia DBMS that meets the requirements for multimedia data:

LEc10RE # 66:

content Based lnformatlon Retrleval: lmage Retrleval, Vldeo Retrleval:

Multimedia information retrieval (MIR) is about the search for knowledge in all its forms, everywhere. Content-based
methods are necessary when text annotations are nonexistent or incomplete. Furthermore, content-based methods can
potentially improve retrieval accuracy even when text annotations are present by giving additional insight into the
media collections.

Multimedia Information Retrieval (MMIR) is a research discipline of computer science that aims at extracting
semantic information from multimedia data sources.
[1]
Data sources include directly perceivable media such
as audio, image and video, indirectly perceivable sources such as text, biosignals as well as not perceivable sources
such as bioinformation, stock prices, etc. The methodology of MMIR can be organized in three groups:
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1. Methods for the summarization of media content (feature extraction). The result of feature extraction is a
description.
2. Methods for the filtering of media descriptions (for example, elimination of redundancy)
3. Methods for the categorization of media descriptions into classes.

content-based lmage Retrleval:
Content-based image retrieval (CBIR), also known as query by image content (QBIC) and content-based visual
information retrieval (CBVIR) is the application of computer vision techniques to the image retrieval problem, that is,
the problem of searching for digital images in large databases. Content based image retrieval is opposed to concept
based approaches (concept based image indexing).
"Content-based" means that the search will analyze the actual contents of the image rather than the metadata such
as keywords, tags, and/or descriptions associated with the image. The term 'content' in this context might refer to
colors, shapes, textures, or any other information that can be derived from the image itself. CBIR is desirable because
most web based image search engines rely purely on metadata and this produces a lot of garbage in the results.

Also
having humans manually enter keywords for images in a large database can be inefficient, expensive and may not
capture every keyword that describes the image.Thus a system that can filter images based on their content would
provide better indexing and return more accurate results.
Potential uses for CBIR include:
Art collections
Photograph archives
Retail catalogs
Medical diagnosis
Crime prevention
The military
Intellectual property
Architectural and engineering design
Geographical information and remote sensing systems

cBlR 1EcRNlq0E8:
Many CBIR systems have been developed, but the problem of retrieving images on the basis of their pixel content
remains largely unsolved.
1. Query techniques
Different implementations of CBIR make use of different types of user queries.
Query by example is a query technique that involves providing the CBIR system with an example image that it will
then base its search upon. The underlying search algorithms may vary depending on the application, but result
images should all share common elements with the provided example.
Options for providing example images to the system include:
A preexisting image may be supplied by the user or chosen from a random set.
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The user draws a rough approximation of the image they are looking for, for example with blobs of color or
general shapes.
This query technique removes the difficulties that can arise when trying to describe images with words.
Semantic retrieval
The ideal CBIR system from a user perspective would involve what is referred to as semantic retrieval, where the user
makes a request like "find pictures of dogs" or even "find pictures of Abraham Lincoln". This type of open-ended task
is very difficult for computers to perform - pictures of chihuahuas and Great Danes look very different, and Lincoln
may not always be facing the camera or in the same pose. Current CBIR systems therefore generally make use of
lower-level features like texture, color, and shape, although some systems take advantage of very common higher-
level features like faces (see facial recognition system). Not every CBIR system is generic. Some systems are
designed for a specific domain, e.g. shape matching can be used for finding parts inside a CAD-CAM database.
Other query methods
Other query methods include browsing for example images, navigating customized/hierarchical categories, querying
by image region (rather than the entire image), querying by multiple example images, querying by visual sketch,
querying by direct specification of image features, and multimodal queries (e.g. combining touch, voice, etc.).
CBIR systems can also make use of relevance feedback, where the user progressively refines the search results by
marking images in the results as "relevant", "not relevant", or "neutral" to the search query, then repeating the search
with the new information.
2. Content comparison using image distance measures
The most common method for comparing two images in content based image retrieval (typically an example image
and an image from the database) is using an image distance measure. An image distance measure compares the
similarity of two images in various dimensions such as color, texture, shape, and others. For example a distance of 0
signifies an exact match with the query, with respect to the dimensions that were considered. As one may intuitively
gather, a value greater than 0 indicates various degrees of similarities between the images. Search results then can
be sorted based on their distance to the queried image.
Color
Computing distance measures based on color similarity is achieved by computing a color histogram for each image
that identifies the proportion of pixels within an image holding specific values (that humans express as colors). Current
research is attempting to segment color proportion by region and by spatial relationship among several color regions.
Examining images based on the colors they contain is one of the most widely used techniques because it does not
depend on image size or orientation. Color searches will usually involve comparing color histograms, though this is
not the only technique in practice.
Texture
Texture measures look for visual patterns in images and how they are spatially defined. Textures are represented
by texels which are then placed into a number of sets, depending on how many textures are detected in the image.
These sets not only define the texture, but also where in the image the texture is located.
Texture is a difficult concept to represent. The identification of specific textures in an image is achieved primarily by
modeling texture as a two-dimensional gray level variation. The relative brightness of pairs of pixels is computed such
that degree of contrast, regularity, coarseness and directionality may be estimated. However, the problem is in
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identifying patterns of co-pixel variation and associating them with particular classes of textures such as silky,
or rough.
Shape
Shape does not refer to the shape of an image but to the shape of a particular region that is being sought out. Shapes
will often be determined first applying segmentation or edge detection to an image. Other methods like use shape
filters to identify given shapes of an image. In some case accurate shape detection will require human intervention
because methods like segmentation are very difficult to completely automate.

CONTENT BASED VIDEO RETRIEVAL:

Recent advances in multimedia technologies allow the capture and storage of video data with relatively inexpensive
computers. Furthermore, the new possibilities offered by the information highways have made a large amount of
video data publicly available. However, without appropriate search techniques all these data are hardly usable. Users
are not satisfied with the video retrieval systems that provide analogue VCR functionality. They want to query the
content instead of the raw video data. For example, a user analysing a soccer video will ask for specific events such as
goals. Content-based search and retrieval of video data becomes a challenging and important problem. Therefore, the
need for tools that can manipulate the video content in the same way as traditional databases manage numeric and
textual data is significant.

The architecture of content-based video retrieval system:

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The process of database population is shown with dashed lines, while querying is shown with solid lines. The raw
video data is stored in the file system, while the storage server is used to store video content meta data and indexes. In
the process of the database population, the features, objects, and events that are specified by system administrator, are
extracted. Indexes and metadata are put in the storage server and videos in the file system. Most queries are resolved
directly in the storage server, but if the query comprises something that has not been already extracted the extractors
do that dynamically.

LEc10RE # 67:

0vervlew of MPE0-7:

MPEG 7:
Started in 1999
International standard in 2001
Formally called Multimedia Content Description Interface.
A standard for describing the multimedia content data "that supports some degree of interpretation of the
informations meaning,which can be passed onto, or accessed by, a device or a computer code.
Application examples:
digital libraries, multimedia directories, broadcast media selection
Standard specifies
a set of descriptors
a set of description schemes (DS)
A language to specify description schemes (description definition language DDL)
Only specifies the format of the descriptions, no algorithms.

Architecture:

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0eslgn of Vldeo-on demand systems:
With the explosion of Internet, people have endless hype, opinions, forecasts, and beliefs about it. Interactive
Television, they feel, is the vision to their beliefs: people will soon be able to purchase products, view movies, play
video games, browse Internet, participate video-conferences without leaving their houses . Of all the new things that
people can do with television, video-on-demands is highly supported by Hollywood since it can lead to new markets
and can bring them unpredictable profits.
People have been passive participants in receiving what TV service providers offer since television was introduced.
Interactive Video On Demand (IVOD), unlike traditional television delivery, provides users with flexibility in
choosing the kinds of information they like to receive . An IVOD system is capable of serving a large number of end
users to concurrently access large number of repositories of stored data, often movies . IVOD is basically an extension
of Video On Demand (VOD). In addition to the freedom of choosing movies, users can interact with movies and
decide the viewing schedule. In other words, IVOD system supports VCR-like functions, such as fast forward,
rewind, pause. The enormous communication bandwidth and disk bandwidth required, and the Quality of Service
(QoS) demanded necessitate a careful design of the system in order to maximize the number of concurrent users while
minimizing the cost.
An IVOD system comprises of 3 major components: the "set-top box" at the client's site, the distribution network, and
the server. There are many design issues to consider in building each of these components.
Video on demand is a technology that provides entertainment on demand to all the subscribers of the service. Video
on demand provides customers with informative and entertaining streams of multimedia and video information. Some
of the services that can be offered by video on demand technology are:
Movies-on-demand: Movies, TV shows, special interest programs and music videos can be watched by home
users at their convenience.
E-commerce: Customers can shop from home for some of their favourite items such as books and software
from various web sites.
Interactive advertising: Customers can interact directly with full motion video advertisements and order the
product on-line while they watch the advertisement.
There are two distinct types of video on demand applications, each requiring different technology.
1. Near-Video-on-demand: This technology delivers the same content using multiple video streams with staggered
start times. For example, twelve video streams each starting at ten minute intervals can deliver a single, two-hour
video. Users wishing to watch the video may have to wait. The waiting time is no longer than ten minutes. In this
technology many users share a single video stream.
2. True-Video-on-demand: This technology provides the users with the requested content immediately. In response
to a user request, the video server delivers the content in a video stream immediately, without any waiting time.
Generally video streams are dedicated to users unless multiple users request a single content title at exactly the
same time. Using this technology, it is possible to provide interactive control of the video stream to the users.
For example, users can be provided with VCR controls such as play, fast forward, rewind, pause. True video-
on-demand will be the focus of this report.
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2 Components Required in a Video-On-Demand System
A detailed explanation of the important components is listed in this section. The components which are important
from Telstras point of view have been analysed and the different techniques that can be used to improve the
performance of the system are discussed.
2.1 Video server
Video server plays a major role in video on demand systems and is a critical component of the system. A video server
has access to the video content and is responsible for the delivery of the video content in continuous streams that are
free of unacceptable video artifacts. A video server therefore has to store and manage large and complex video files.
The important features of a video server are listed below:
Video server capacity: This is the number of video streams to be delivered simultaneously. The type of the video
stream supported by the video server is also important. An MPEG-2 encoded video at 6Mbps requires more video
server capacity than a video that is encoded with MPEG-1 at 1.5Mbps. Ideally a video server should be independent
of the video streams (eg. MPEG1, 2, 4, RealVideo). This allows the content to be stored and delivered in the most
cost-effective manner.
Input/Output performance: Input/output performance is an important video server feature to consider. Normal file
systems are optimized for short and random data access rather than large sequential access. Video files can be stored
as large contiguous files for sequential access or as small clusters for random access. Which is a better approach
depends on the network configuration and is discussed in section 4.2.
Optimized access to large video files is also a desirable property of video servers. The input/output process can be
buffered (the data to be read is transferred to a disk buffer and then transferred to the user memory in smaller chunks)
or direct (data is transferred from the disk directly to the user memory). The direct I/O method is a better approach
because for large video files, the buffered approach slows the data throughput.
Dynamic content management: The number of streams that can be supported by a single video file is limited by the
data throughput characteristics of the content storage device. If a particular video title needs more streams than the
device can support, multiple copies of the video file are needed. One way to achieve this is to store multiple copies of
the most popular video titles. This is not a very cost effective approach. A much better approach is to have a video
server which supports dynamic content management. If a particular video title requires more streams, the video server
duplicates the video file. When the demand for the video title drops, the extra copies are deleted.
Data storage: If only disk-based secondary storage systems are used to store and manage the huge amount of video
data the system cost would be extensively high. A tape-based tertiary storage system seems to be a reasonable
solution to lowering the cost of storage and management of this data. This approach introduces several issues to be
considered. The important issues are the replacement policy on disks, the decomposition and the placement of
continuous data chunks on tapes and the scheduling of multiple requests for materializing objects from tapes to disks.
Replacement policy is concerned with freeing space on disks for materializing a video object that has been requested
and is partially or fully not residing on the disk. Several policies have been proposed and implemented which support
single stream environment and multiple stream environment. A policy which is cost effective and provides a good
response time should be considered when designing the VOD server.

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If a video object is distributed on multiple tapes, it reduces the transfer time as data is transferred simultaneously
from the different tapes at the same time. This strategy increases the time spent on exchanging tapes and also seeking
the location on the tapes. Therefore there is a tradeoff between seek time and transfer time. The video object should be
distributed in such a manner that the total time (seek + transfer) is reduced.
Fault tolerance: Failure of the storage system is a major concern. If the storage system fails, content cannot be
accessed and delivered causing disruption in the services. The video server should have a mechanism to replicate the
content so that if a certain section of the disk fails, content can be transferred from an alternate location. In the middle
of transmission, if the entire video server fails, the VOD system should be able to transfer the connection to another
server without the knowledge of the user. This is discussed in sec 4.2.
Open and scalable architecture: The video server should be able to support a variety of network distribution options.
The video server should be easily and economically upgraded to support the increase in content and number of users.
2.2 Back office software/hardware
This component is required to do the following tasks:
Keeping track of the content: where is it coming from, royalty issues, advertising services.
Content management.
User account management.
Standard billing systems and customer management systems.
There is software/hardware available in the market that can perform the above tasks.
2.3 Other components
Digital set-top boxes, application server and the network configuration are the other components of a video-on-
demand system. The next section discusses the network configuration which is the most important component of a
video-on-demand system.
3 Network Configuration
It is very important to design the network configuration so as to minimize the cost of content delivery. There are two
major costs associated with the delivery of content. The cost of video servers (non-recurring cost) and the cost of
transferring the content (network bandwidth which is a recurring cost). The VOD system should be configured in such
a way that the total cost is minimized. There are two major ways to implement the network architecture: centralised
system architecture and networked system architecture.

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3.1 Centralised system architecture
In this model, there is one large central server which stores all the digital content. This server controls and manages all
content storage and supports all video streams. Therefore if there are 1000 requests at a given time, the server should
be able to transfer 1000 streams of video simultaneously. This also means that the network bandwidth required has to
be very high in order to support 1000 streams. If we are sending MPEG-1 video encoded at 1.5Mbps, then in order to
send 1000 video streams, a total bandwidth of 1.5Gbps is required. Also, as the number of subscribers increase, so
will the number of streams needed, which will require additional channel bandwidth. This is a very expensive
alternative given the costs involved in maintaining high speed networks and is therefore not a feasible option.
Figure 1 configuration for a centralized system architecture.

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Figure 2 typical layout for a networked system architecture

3.2 Networked system architecture
In a networked system architecture, multiple video servers are distributed throughout the network infrastructure. Each
video server controls and manages a subset of the content storage and is responsible for a subset of the video
streams. For example, a video server connected to a particular exchange will be responsible for all the video streams
sent from that exchange.. Ideally all the high demand content is replicated at the video servers connected to each
exchange. This reduces a lot of traffic between these servers and therefore relaxes the bandwidth requirements
between the major hubs. If a local server does not have a requested video title, it searches through a list of all the
video servers which have that title and selects the one with the least network costs. Networked architecture is a
feasible option as it relaxes the bandwidth requirements on the network but has to be combined with other bandwidth
reduction techniques which are discussed in the next section.

4 Bandwidth Reduction Techniques
The key problem in deploying large-scale video on demand applications is economy rather than technology. In order
to have an efficient and cost effective video-on-demand system it is important to reduce the bandwidth requirements
as much as possible. This section explains some of the bandwidth reduction techniques and analyzes their suitability
to video-on-demand services.

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4.1 Multicasting
A typical video-on-demand system uses dedicated channels to service each user request. This increases the bandwidth
requirements as more and more streams are requested. Multicasting is an approach whereby various customers can
share a single movie stream resulting in reduced system cost per customer and improved system scalability.
Multicasting has some limitations associated with it. For example, in order to share a single channel with a group of
customers, all the customers have to request the video at the same time and watch it without any interaction (no pause,
fast forward, rewind, etc). This defeats the purpose of a true and interactive video-on-demand system. There are
certain multicasting techniques which can be used to provide interactive video-on-demand services.

Batching: In this approach, the requested video is intentionally delayed by some amount of time, called a batching
interval, so that subsequent request for the same video arriving during the current batching interval may be serviced
using a single I/O stream. This trades off reduced I/O stream requirements for increased latency. Therefore, large
batching intervals would seem to be incompatible with the notion of a ture VOD system. Batching can only be used
with popular videos since unpopular videos are unlikely to receive multiple requests during the (short) delay interval.

Patching: In this approach, also called as a dynamic multicast, the multicasting stream is not static but can expand
dynamically to accommodate new requests. For example, if there is a video being streamed through a local video
server and there is a new request for the same video, then the video server starts caching the video data in its local
disk. The server therefore needs an initial patching stream to transmit the leading portion of the video. When the
leading portion of the video is transmitted, the rest of the video is transmitted from the data already buffered in the
local disk. Using the patching technique, channels are used only briefly to broadcast the first few minutes of the
video, instead of being held up for the entire duration of the playback. Patching can be seen as a multicasting
technique for interactive video-on-demand systems where a single channel can be shared with multiple
users. Patching also saves the storage costs as only a small amount of the video stream has to be buffered at the local
disk ( depending on the requests).

There are many variations to patching, which are more efficient. Transient patching is similar to patching but the
patching stream is longer than used in the traditional patching scheme by a fixed time slice (t). This helps to reduce
the length of any future video streams if there are any new requests in that time slice (t). Split and merge
protocol divides the video stream into a service stream for regular playback and an interactive stream for
user interaction (pause, fast forward, rewind, etc). If there are new requests for the same video title then patching is
used for that time period.
Patching is very effective in reducing the bandwidth and storage requirements if the number of interactive requests
from the users is within a certain limit. Beyond that, patching looses its efficiency as it results in starting multiple
patches of the same video and increases the bandwidth requirements.
Piggybacking: This approach tries to alter the playback rates of the on-going video streams by 5% (an amount
supposedly undetectable to human observers) in order to merge the respective video streams into a single stream
which can then serve the entire group of requests. The idea is to display the leading stream at a slower rate, and the
trailing stream at a faster rate. Then, assuming this interval is sufficiently small, the faster stream will eventually catch
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up with the slower stream. At this point the streams can be piggybacked or merged. There are different piggybacking
policies that can be used to achieve maximum savings in bandwidth. The three basic types of piggybacking policies
aresimple merging policy, the odd-even policy and the greedy policy. The first two are elementary as they involve at
most a single change of speed for each video stream. Therefore they can achieve a maximum of 50% improvement in
the number of streams saved, because they pair of subsequent streams. The greedy policy changes the speed of the
different streams in such a way that the streams are merged at the earliest possible time in order to achieve maximum
savings in bandwidth.
There are some limitations associated with piggybacking. The first is that it takes a long time for two streams to
merge. Therefore if the streams are very far apart, it is possible that they will never merge at all. The second drawback
is that, unless the VOD server is powerful enough to make the video rate changes on the fly, it will have to store at
least two versions of each video, doubling the storage requirements. The quality of service may also be affected when
the playback rates are altered. Piggybacking assumes that the number of interactive requests is not very high. This is
because, each interactive request causes the merged stream to start a new stream. This is therefore not a
recommended option due to the level of interaction present in an interactive video-on-demand service.
Volatile storage: In this approach, every single video title that is transferred to a video server is cached in the servers
local disk completely. Apart from its local storage, each video server has a volatile storage where the video content is
stored for a short period of time. This procedure enables any future requests to be served from the local server without
any additional video streams required from the distant server (the only exception is when a future request is for a part
of the video which is still not present in the local volatile storage). There are no restrictions in the number of
interactive requests as all the requests are satisfied from the local video server. The only drawback with this approach
is that it increases the storage cost as additional volatile storage is required at each video server connected to an
exchange.
Ideally, this approach can be combined with patching to achieve a hybrid solution which will be cost effective in
terms of storage and bandwidth costs. The storage costs and network costs have to be compared and a topology has to
be designed accordingly.
4.2 Server selection
In a networked system configuration the server selection process is very important. When a local video server does
not have the requested video title, a remote server which is the most cost effective has to be selected for transferring
the video data. There are two methods of server selection: static and dynamic server selection. In static server
selection, the process of selecting the server is done only once when a particular video is requested. Once the server
is selected, the entire video is streamed from this server. Dynamic server selection is a process which constantly keeps
looking for the best available server. If, after a certain video stream has been started from a server, a different server
which is more cost effective is available, the connection is transferred to that video server. This process is transparent
to the user. Dynamic server selection is a more efficient process as it tries to reduce the transfer costs as much as
possible. Dynamic server selection also makes the video-on-demand system fault tolerant. For example, in the middle
of a video transmission, if the video server fails, the connection can be transferred to the next best video server
without any interruption to the user.
In order to make the server selection process very efficient, the video server has to store the video data in an
appropriate manner. For example, if a static server selection process is used then the video file can be stored as one
large continuous file as this will expedite the data access which is sequential from a fixed server. In case of a
dynamic server selection process, the video files can be divided into clusters and stored on the disk. This will enable
faster access to any particular cluster when a connection is transferred to a different video server.
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4.3 Video content distribution
The method of distributing the content is also very important. Depending on the likes and dislikes of the users
connected to a particular exchange, appropriate content should be stored at that local video server. This would depend
on the borrowing habits of the customers. This process can be static or dynamic. In the static approach, the borrowing
habits of all the users over a fixed period of time can be studied and the content can be distributed accordingly. This
process can be repeated every 6/12 months (or a fixed period of time) and the content can be updated accordingly. In a
dynamic model a resident program constantly monitors the borrowing habits of the users and accordingly transfers the
content between the video servers dynamically. The dynamic content distribution process has to be based on a smart
algorithm which considers borrowing habits, locality information and many other factors.

5 Video-on-Demand Costs
The primary costs associated with a video-on-demand system are:

1. Cabling costs: If a video-on-demand system is evolving from a networked system which is scalable, the cables
are largely in place and would not be affected by the additional requirements of a video-on-demand system.
Cabling costs would be dependent on the bandwidth requirements of the various parts of the network.
2. Network bandwidth costs: Although difficult to estimate, network bandwidth costs are a substantial component
of the operating costs. The actual cost depends on the network topology and distances involved, but must attempt
to minimise these costs by appropriate network design. Similar work has been done in our section in
the Vista project.
3. Set-top-box costs: This is a fixed cost and doesn't affect the network architecture of a video-on-demand system.
4. Video server costs: A video server is a massive I/O device capable of producing multiple video streams from
content stored at many sources. The cost of a video server depends on the number of streams produced and the
amount of content (number of videos) stored in the server.

From the above listed costs, the cabling and set-top-box costs are fixed whereas the network bandwidth and video
server costs are variable and would depend on the network architecture. The network topology should be designed in
such a way that the combined cost of the network bandwidth and the video servers (storage and number of
independent streams) is minimised. The network topology would in turn depend on various other factors such as
server selection strategy, bandwidth reduction techniques, content distribution strategy. Therefore all these factors
have to be considered in order to design an efficient video-on-demand system.

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6 Protecting the Video Content
If the video data is cached at the client end, it is possible that the data can be replicated and resold by some of the
users. Some mechanism has to be devised to stop the replication process. For the wide-spread sale of a multimedia
content, the use of caching is necessary because neither the network infrastructure nor any content providers servers
can support an arbitrary number of unicast connections with content that is encrypted in real time. This conflict
between copyright protection and scalability inhibits effectively the deployment of caching and pre-fetching of
commercially sold video content. There are two main techniques used to stop resale and illegal copying of video data:
digital watermarking and data encryption.
Digital watermarking technology enables the possibility to add copyright or customer information into digital data,
like video, to protect the content from resale by an original customer. Usually digital watermarks are additional
hidden tags that are holographically inlaid in the multimedia data. This hidden information is perceptually
undetectable. The information is embedded with a secret key and can be retrieved using the same secret key. This
technique de-motivates the illegal re-use of distributed digital data.

The video encryption approach uses encrypted point-to-point delivery to ensure that only paying users receive the
service. To ensure that copyright violations can be proved, the delivered content is further watermarked with
information about the contents seller as well as the customer.

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Sub[ect: lnLerneL & MulLlmedla 1echnology Sem - 7th Sub[ect Code: 322711(22)

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