Unit 1
Unit 1
Unit 1
Introduction
Multimedia is used to indicate that the information/data being transferred over the network
may be composed of one or more of the following media types:
1. Text: Includes unformatted text - comprising strings of characters from a limited character
set, formatted text strings as used for the structuring, access, and presentation of electronic
documents
2. Images: Include computer generated images, comprising lines, curves, and circles, and
digitized images of documents and pictures
3. Audio: Includes both low fidelity speech - used in telephony, and high fidelity stereophonic
music - used with compact discs
4. Video: Includes short sequences of moving images (also known as video clips) and complete
movies/films
Applications may include - person-to-person communications or
person-to-system communications
In general, two people communicate with each other through suitable
terminal equipment (TE) while a person interacts with a system using either a multimedia
personal computer (PC) or workstation
Personal computers are located either in the home or on a desktop in an office and the system
is a server containing a collection of files or documents each comprising digitized text, images,
audio, and video information either singly or integrated together
The server may also contain a library of digitized movies/videos and the user interacts with
the server by means of a suitable selection device that is connected to the set-top box (STB)
associated with a
television
There are a number of different types of network that are used to provide the networking
infrastructure
For example, public switched telephone networks (PSTNs) – also known as general switched
telephone networks (GSTNs) were designed initially to provide a basic switched telephone
service
But, as a result of advances in digital signal processing hardware and associated software,
they now provide a range of more advanced services involving text, images, and video
Similarly, data networks that were designed initially to support basic data applications such
as electronic mail and file transfers
But, now support a much richer set of applications that involve images, audio, and video
Multimedia information representation
Applications involving text and images comprise blocks of digital data
In a text, a typical unit is a block of characters with each character represented by a fixed
number of binary digits (bits) known as a codeword
Similarly, a digitized image comprises a two-dimensional block of what are called picture
elements with each element represented by a fixed number of bits
Audio and video signals vary continuously with time as the amplitude of the speech, audio,
or video signal varies
This type of signal is known as an analog signal and, typically, the duration of applications
that involve audio and/or video can be relatively long
A typical telephone conversation, for example, can last for several minutes while a movie
(comprising audio and video) can last for a number of hours
In applications that involve just a single type of media, the basic form of representation of
the particular media type is often used
Similarly, in applications that involve either text-and-images or audio-and-video their basic
form is often used since the two media types in these applications have the same form of
representation
However, in applications that involve the different media types integrated together in some
way, it becomes necessary to represent all four media types in a digital form
In the case of text and images, digital data is the standard form of representation
For audio and video, basic forms of representation are analog signals, these must be
converted into a corresponding digital form before they can be integrated with the two other
media types
The digitization of an audio signal produces a digital signal, the amplitude of the signal varies
continuously with time, is of a relatively high bit rate
This is measured in bits per second (bps) and, in the case of a speechsignal, for example, a
typical bit rate is 64 kbps
The same applies to the digitization of a video signal except that much higher bit rates and
longer time durations are involved
The communication networks that are used to support applications that involve audio and
video cannot support the very high bit rates that are required for representing these media types
in a digital form
As a result, a technique known as compression is first applied to the digitized signals in order
to reduce the resulting bit rate to a level which the various networks can support
Compression is also applied to text and images in order to reduce the time delay between a
request being made for some information and the information becoming available on, say, the
screen of a computer
Multimedia networks
There are five basic types of communication network that are used to provide multimedia
communication services:
1. Telephone networks
2. Data networks
3. Broadcast television networks
4. Integrated services digital networks
5. Broadband multiservice networks
The first three network types were initially designed to provide just a single type of service:
telephony, data communications, and broadcast television respectively
The last two network types, however, were designed from the outset to provide multiple
services
Telephone networks
Public switched telephone networks have been in existence for many years and have gone
through many changes during this time
They were designed to provide a basic switched telephone service which is known as a plain
old telephone service or POTS
The term "switched" is used to indicate that a subscriber can make a call to any other
telephone that is connected to the total network
Initially, such networks spanned just a single country but later the telephone networks of
different countries were interconnected so that they now provide an international switched
service
The main components of the network are shown in diagrammatic form in Figure 1.1(a)
Telephones located in the home or in a small business are connected directly to their nearest
local exchange/end office
Those located in a medium or large office/site are connected to a private switching office
known as a private branch exchange or PBX
The PBX provides a (free) switched service between any two telephones that are connected
to it
In addition, the PBX is connected to its nearest local (public) exchange which enables the
telephones that are connected to the PBX also to make calls through a PSTN
More recently, cellular phone networks have been introduced which provide a similar service
to mobile subscribers by means of handsets that are linked to the cellular phone network
infrastructure by radio
The switches used in a cellular phone network are known as mobile switching centers
(MSCs) and these, like a PBX, are also connected to a switching office in a PSTN which
enables both sets of subscribers to make calls to one another
Finally, international calls are routed to and switched by international gateway exchanges
(IGEs) A speech signal is an analog signal since it varies continuously with time according
to the amplitude and frequency variations of the sound resulting from the speech
A microphone is used to convert this into an analog electrical signal
Because of this, telephone networks operate in what is called a circuit mode which means
that, for each call, a separate circuit is set up through the network - of the necessary capacity -
for the duration of the call The access circuits that link the telephone handsets to a PSTN or
PBX were designed, therefore, to carry the two-way analog signals associated with a call
Hence, although within a PSTN all the switches and the transmission circuits that
interconnect them now operate in a digital mode, to carry a digital signal-a stream of binary 1s
and 0s-over the analog access circuits requires a device known as a modem The general
scheme is shown is Figure 1.1(b) Essentially, at the sending side, the modem converts the
digital signal output by the source digital device into an analog signal that is compatible with
a normal speech signal .
This is routed through the network in the same way as a speech signal and, at the receiving
side, the modem converts the analog signal back again into its digital form before relaying this
to the destination digital device
Modems also have the necessary circuits to set up and terminate a call
Hence by using a pair of modems - one at each subscriber access point - a PSTN can also be
used to provide a switched digital service
The early modems supported only a very low bit rate service of 300bps but, as a result of
advances in digital signal processing circuits, modems are now available that support bit rates
of up to 56 kbps
The general scheme is shown in Figure 1.1 (c) and such applications require bit rates in
excess of 1.5 Mbps
It illustrates the PSTN can now support a wide range of other multimedia communication
applications
Data networks
Data networks were designed to provide basic data communication services such as
electronic mail (email) and general file transfers
The user equipment connected to these networks, therefore, is a computer such as a PC, a
workstation, or an email /file server
The two most widely deployed networks of this type are the X.25network and the Internet
X.25 network is restricted to relatively low bit rate data applications and hence is unsuitable
for most multimedia applications
The Internet is made up of a vast collection of interconnected networks all of which operate
using the same set of communication protocols
A communication protocol is an agreed set of rules that are adhered to by all communicating
parties for the exchange of information
The rules define not only the sequence of messages that are exchanged between the
communicating parties but also the syntax of these messages
Hence by using the same set of communication protocols, all the computers that are
connected to the Internet can communicate freely with each other irrespective of their type or
This is also the origin of the term "open systems interconnection"
Figure 1.2 shows a selection of the different types of interconnected network
In the case of a user at home or in a small business, access to the Internet is through an
intermediate Internet service provider (ISP) network
Normally, since this type of user wants access to the Internet intermittently, the user devices
are connected to the ISP networkeither through a PSTN with modems or through an
integratedservices digital network (ISDN), provides access at a higher bit rate
Alternatively, business users obtain access either through a site/campus network if the
business comprises only a single site or, if it comprises multiple sites through an enterprise-
wide private network
In the case of a single site/campus, the network known as a (private) local area network or
LAN
For an enterprise-wide network comprising multiple sites the sites are interconnected
together using an inter-site backbone network to provide a set of enterprise-wide
communication services
The enterprise network is then known as an intranet since internal services are provided
using the same set of communication protocols as those defined for the Internet
The different types of network are all connected to the Internet backbone network through
an interworking unit called a gateway
It is responsible for routing and relaying all messages to and from the connected network, is
also known as a router
All data networks operate in what is called a packet mode
Packet is a container for a block of data and, at its head, is the address of the intended recipient
computer which is used to route the packet through the network
This mode of operation was chosen since the format of the data associated with data
applications is normally in the form of discrete blocks of text or binary data with varying time
intervals between each block
Figure 1.3(b), in the case of satellite and terrestrial broadcast networks, when high-speed PSTN
modem is integrated into the STB
It provides the subscriber with an interaction channel so enhancing the range of services these
network support
This is the origin of the term "interactive television"
The subscriber telephone can be either a digital phone or a conventional analog one
In the case of a digital phone, the electronics that are needed to convert the analog voice and
call setup signals into a digital form are integrated into the phone handset
With an analog phone, the same electronics are located in the network termination equipment
so making the digital mode of operation of the network transparent to the subscriber phone
The digitization of a telephone-quality analog speech signal produces a constant bit rate
binary stream-normal referred to as a bitstream- of 64 kbps
Hence, the basic DSL of the ISDN known as the basic rate access or BRA - supports two
64 kbps channels
They can either be used independently or as a single combined 128kbps channel
Since the two channels were intended for two different calls, this requires two separate
circuits to be set up through the switching network independently
Hence to synchronize the two separate 64 kbps bit streams into a single 128 kbps stream
requires an additional box of electronics to perform, what is known as, the aggregation
function
In addition, a single higher bit rate channel of either 1.5 or 2 Mbps is supported
This is known as the primary rate access or PRA
Service provided has been enhanced and now supports a single switched channel of p x
64kbps where p = 1, 2, 3,.. 30
The various services provided are summarized in Figure 1.4 and an ISDN can support a range
of multimedia applications
Because of the relatively high cost of digitizing the access circuits, in general the cost of the
services associated with an ISDN are higher than the equivalent service provided by a PSTN
Multimedia applications
There are many and varied applications that involve multiple media types
In general, however, they can be placed into one of three categories:
1. Interpersonal communications
2. Interactive applications over the internet
3. Entertainment applications
In many instances the networks that are used to support these applications were initially
designed to provide a service that involved just a single type of medium
As a result of advances in various related technologies that they are now used to support
multimedia applications
Interpersonal communications
Interpersonal communications may involve speech, image, text, or video
In some cases just a single type of medium is involved while inothers two or more media
types are integrated together
Speech only
Interpersonal communications may involve speech telephony - have been provided using
telephones that are connected either to a public switched telephone network
(PSTN/ISDN/cellular phone network) or a PBX
The general scheme is shown in Figure 1.6
Alternatively, by using a multimedia PC equipped with a microphone and speakers, the user
can take part in telephone calls through the PC
This requires, a telephone interface card and associated software andis known as computer
telephony integration or CTI
The added advantages of using a PC instead of a conventional telephone are many
For example, the user can create his or her own private directory of numbers and initiate a
call simply by selecting the desired number onthe PC screen
Generally, providing the access circuit to the network has sufficient capacity - normally
referred to as the circuit's bandwidth - it is possible to integrate telephony with all the other
networked services provided by the PC
In addition to telephony, many public and private networks support additional services
Two examples are voice-mail and teleconferencing
Voice-mail, for example, is used in the event of the called party being unavailable
A spoken message can then be left in the voice mailbox of the called party
This is located in a central repository known as the voice-mail server
The message can be read by the owner of the mailbox the next time he or she contacts the
server
Teleconferencing calls involve multiple interconnected telephones/PCs
Each person can hear and talk to all of the others involved in the call
This type of call is known variously as a conference call or, since it involves a telephone
network, a teleconferencing call or sometimes an audio conferencing call
It requires a central unit known as an audio bridge which provides the necessary support to
set up a conference call automatically
VOIP (Voice over internet protocol)
The Internet is also used to support telephony
Initially, because the Internet was designed to support computer-to- computer
communications, just (multimedia) PC-to-PC telephony was supported
This was subsequently extended so that a standard telephone could also be used as shown in
Figure 1.7
In the case of a PC-to-PC telephone call, the standard addresses that are used to identify
individual computers connected to the Internet are used in the same way as for a data transfer
application
The Internet operates in a packet mode, both PCs must have the necessary hardware and
software to convert the speech signal from the microphone into packets on input and back again
prior to output to the speakers
Telephony over the Internet is also known, therefore, as packet voice or, because the network
protocol associated with the Internet is called the Internet protocol (IP), voice over IP (VoIP)
When a PC connected to the Internet needs to make a call to a telephone that is connected
to a PSTN/ISDN, because these both operate in a circuit mode, an interworking unit known as
a telephony gateway must be used
PC user first sends a request to make a (telephone) call to a pre allocated telephony gateway
using the Internet address
Assuming the user is registered to use this service, the gateway requests the telephone number
of the called party from the source PC
On receipt of this, the source gateway initiates a session (call) with the telephony gateway
nearest to the called party using the Internet address of the gateway
The called gateway then initiates a call to the recipient telephone using its telephone number
and the standard call setup procedure of the PSTN/ISDN
Assuming the called party answers, the called gateway then signals back to the PC user -
through the source gate way - that the call can commence
A similar procedure is followed to clear the call on completion
Image only
An alternative form of interpersonal communications over a PSTNor an ISDN is by the
exchange of electronic images of documents
This type of service is known as facsimile - or simply fax - and is illustrated in Figure 1.8
Normally, this type of communication involves the use of a pair of fax machines, one at each
network termination point
To send a document, the caller keys in the (telephone) number of the intended recipient and
a circuit is set up through the network in the same way as for a telephone call
The two fax machines communicate with each other to establish operational parameters after
which the sending machine starts to scan and digitize each page of the document in turn
Both fax machines have an integral modem within them and, as each page is scanned, its
digitized image is simultaneously transmitted over the network and, as this is received at the
called side, a printed version of the document is produced
Finally, after the last page of the document has been sent and received, the connection through
the network is cleared by the calling machine in the normal way
It is also possible to use a PC instead of a normal fax machine to send an electronic version
of a document that is stored directly within the PC's memory
This mode of operation is known as PC fax
The digital image of each page of the document is sent in the same way as the scanned image
produced by a conventional fax machine
As with telephony, this requires a telephone interface card and associated software
The latter operates in exactly the same way as that in a fax machine and hence the terminal
at the called side can be either a fax machine or another similar PC
In addition, with PC fax it is possible to send the digitized document over other network
types such as an enterprise network
In this case, a LAN interface card and associated software are used
This mode of operation is particularly useful when working withpaper-based documents
such as invoices, and so on
Text only
An example of interpersonal communications involving just text is electronic mail (email)
The user terminal is normally a PC or a workstation and, the most widespread network used
is the Internet
Various operational scenarios are shown in Figure 1.9 (a)
In the case of a user at home, access to the Internet is through a PSTN/ISDN and an
intermediate Internet service provider (ISP) network
Alternatively, business users obtain access either through an enterprise network or a
site/campus network Text only
Associated with each network is a set of one or more server computers
Each is known as an email server and, collectively, these contain a mailbox for each user
connected to that network
A user can both create and deposit mail into his or her mailbox and read mail from it
Both the email servers and the internetwork gateway operate usingthe standard Internet
communication protocols
The format of a typical text-only email message is shown in Figure 1.9 (b) and, at the head
is the unique Internet-wide name of both the sender and recipient of the mail
In addition, a copy of the mail can be sent to multiple recipients eachof whom is listed-in the
cc part of the mail header, the acronym"cc" being the abbreviation for "carbon copy" which
was the original means of making (paper) copies of documents
Normally, the contents of text-only mail comprise unformatted text, typically strings of
ASCII characters
The software associated with CSCW comprises a central program- known as the white board
program - and a linked set of support programs, one in each PC/workstation
The latter is made up of two parts: a change-notification part and an update-control part
Whenever a member of the group updates the contents of his or her whiteboard, the change-
notification part sends details of the changes to the whiteboard program
This relays the changes to the update-control in each of the other PCs/workstations and these
in turn proceed to update the contents of their copy of the whiteboard
The integration of video with speech means that the bandwidth of the access circuits required
to support this type of service is higher than that required for speech only
Moreover, as with telephony, a call may involve not just two persons - and hence
terminals/PCs - but several people each located in their own office
This type of call is then known as a desktop videoconferencing call and is now widely used in
large corporations involving multiple geographically distributed sites in order to minimize
travel between the various locations
Large corporations of this type have an enterprise-wide network to link the sites together
and, in order to support videoconferencing, there is a central unit called a multipoint control
unit (MCU) or sometimes a videoconferencing server - associated with this network
An example is shown in Figure 1.11 (b)
In this way, only a single two-way communication channel between each location and the
MCU is required thereby reducing considerably the communication bandwidth needed
Alternatively, some networks such as LANs and the Internet support what is called
multicasting
This means that all transmissions from any of the PCs/workstations belonging to a
predefined multicast group are received by all the other members of the group
Thus with networks that support multicasting, it is possible to hold a conferencing session
without an MCU
The principle is shown in Figure 1.11 (c) and, as we can deduce fromthis, this is only feasible
when only a limited number of participants are involved owing to the high load it places on the
network
Speech and video
While the application just described involves only a single person at each location, there are
other applications that involve groups of people at one or more of the locations
Two examples are shown in Figure 1.12 Speech and video While the application
In part (a) a person at one location is communicating with a group of people at another
location
This is the case, for example, with the transmission of a live lecture or seminar
Typically, the information stream transferred from the lecturer to the (remote) class would be
integrated speech-find-video together with electronic copies of transparencies and other
documents used in the lecture
In the reverse direction, the information may comprise just speech - for questions - or
integrated speech-and-video to enable the lecturer to both see and hear the members of the class
at the remote location
In terms of communications requirements, these are similar to those for a two-party
videophone call
Alternatively, if the lecture is being relayed to multiple locations, either separate
communications channel is required to each remote site or an MCU is used at the lecturer's site
Because of the relatively high bandwidth that is involved, the network is either an ISDN that
supports multiple 64 kbps channels or a broadband multiservice network if one is available
In the example in Figure 1.12 (b), there is a group of people at each location
This type of application has been in use for many years and was the first example of
videoconferencing
Normally, since a group of people are present at each location, specially equipped rooms
called videoconferencing studios are used which contain all the necessary audio and video
equipment
This comprises one or more video cameras, a large-screen display, and associated audio
equipment, all of which is connected to a unit called a videoconferencing system
A conference can involve just two locations or, more usually, multiple locations as shown in
the figure
In the case of the latter, an MCU is normally used to minimize the bandwidth demands on
the access circuits to the network
In the figure, the MCU is shown as a central facility within the network and hence only a
single two-way communications channel is required for each access circuit of the network
This is the type of arrangement with a telecommunications-provider conference, for example
Alternatively, if a private network is being used, the MCU is normally located at one of the
sites
The communication requirements at that site are then more demanding since it must support
multiple input channels - one from each of the other sites - and a single output channel, the
stream from which must be broadcast to all of the other sites
Multimedia
In Internet-based electronic mail - email - assumed the information content of each email
message consisted of text only
In addition, however, mail containing other media types such as images, audio, and video are
also used
Three examples of electronic mail consisting of media types other than text are voice-mail,
video-mail, and multi- media mail
Voice-mail is similar in principle to that described earlier in relationto telephone networks
With Internet-based voice-mail, however, there is a voicemail server associated with each
network
This is in addition to the email server shown earlier in Figure 1.9(a) In Internet-based
electronic mail - email - assumed the informationcontent of each email message consisted of
text only
The user first enters a voice message addressed to the intended recipient and the local voice-
mail server then relays this to the server associated with the intended recipient's network
The stored voice message is then played out the next time the recipient accesses his or her
voice-mailbox
The same mode of operation is used for video-mail except in this case the mail message
comprises an integrated speech-and-video sequence
Multimedia mail is an extension of text-only mail inasmuch as the basic content of the mail
comprises textual information
With multimedia mail, however, the textual information is annotated with a digitized image,
a speech message, or a video message, as shown in Figure 1.13
In the case of speech-and-video, the annotations can be sent either directly to the mailbox of
the intended recipient together with the original textual message – and hence stored and played
out in the normal way - or they may have to be requested specifically by the recipient when the
textual message is being read
In this way, the recipient can always receive the basic text-only message but the multimedia
annotations can be received only if the terminal being used by the recipient supports voice
and/or video
Anyone can create a new document at a particular server site - providing the server has been
allocated an Internet address – and making hyperlink references from it to any other document
on the Web
Each document has a unique address - known as a uniform resource locator or URL, which
identifies both the location of the server on the Internet where the first page of the document is
stored and also the file reference on that server
The first page of a document is known as the home page and all the hyperlinks on this and
the other pages have similar URLs associated with them
A standard format is used for writing documents
It is known as the Hypertext Markup Language (HTML) and it is also used for writing client
software to explore the total contents of the web
That is, the contents of the linked information on all the web servers
The client function is called a browser and there are a number of user-friendly browsers
available to explore the contents of the web
These allow a user to create a directory of previously visited servers and to open up a dialog
with a particular server at the click of the mouse
Once a desired document has been located, the user simply clicks onan anchor point within
a page of the document to activate the linkage information stored at that point
A key feature of MOD is that a subscriber can initiate the showing of a movie selected from
a large library of movies at any time
Hence, as we can deduce from Figure 1.15(b), this means that the server must be capable of
playing out simultaneously a large number of video streams equal to the number of subscribers
currently watching a movie
This requires the information flow from the server to be extremely high since it must support
not just the transmission of a possibly large number of different movies, but also multiple
copies of each movie
Technically this is very challenging and costly
If the server is supporting a large number of subscribers, then it is common for several
subscribers to request the same movie within a relatively short time interval between each
request
An alternative mode of operation is also used, therefore, in which requests for a particular
movie are not played out immediately but instead are queued until the start of the next play out
time of that movie as shown in Figure 1.15 (c)
In this way, all requests for the same movie which are made during the period up to the next
play out time are satisfied simultaneously by the server outputting a single video stream
This mode of operation, is known as near movie-on-demand or N- MOD
Interactive television
Broadcast television networks include cable, satellite, and terrestrial networks
The basic service provided by these networks is the diffusion of bothanalog and digital
television programs
In addition to the connection with the PSTN, the subscriber is able torespond to the
information being broadcast
The user can vote, participation in games, home shopping and so on
Similar set of services are available through satellite and terrestrial broadcast networks,
except that the STB associated with these networks requires a high speed modem to provide
connections to the PSTN and the internet
Communication modes
The transfer of the information streams associated with anapplication can take place in one
of five modes:
1. Simplex: this means the information associated with the application flows in one direction
only
An example is the transmission of photographic images from a deep- space probe at
predetermined times since this involves just a unidirectional flow of information from the probe
to an earth station
2. Half-duplex: this means that information flows in both directions but alternately
This mode is also known as two-way alternate and an example is a user making a request for
some information from a remote server and the latter returning the requested information
3. Duplex: this means that information flows in both directions simultaneously
It is also known as two-way simultaneous and an example is the two-way flow of digitized
speech and video associated with a videotelephony application
4. Broadcast: this means that the information output by a single source node is received by all
the other nodes - computers, and soon – that connected to the same network
An example is the broadcast of a television program over a cable network as all the television
receivers are connected to the network receive the same set of programs
5. Multicast: this is similar to a broadcast except that the information output by the source is
received by only a specific subset of the nodes that are connected to the network. The latter
form what is called a multicast group and an example application is videoconferencing which
involves a predefined group of terminals/ computers connected to a network exchanging
integrated speech and video streams
In the case of half-duplex and duplex communications, the bit rate associated with the flow
of information in each direction can be either equal or different; if the flows are equal, the
information flow is said to be symmetric and if the flows are different, asymmetric
For example, a video telephone call involves the exchange of an integrated digitized speech
and video stream in both directions simultaneously and hence a symmetric duplex
communications channel is required
Alternatively, in an application involving a browser (program) and a Web server, a low bit
rate channel from the browser to the Webserver is required for request and control purposes
and a higher bit rate channel from the server to the subscriber for the transfer of, say, the
requested file
Hence for this type of application, an asymmetric half-duplexcommunications channel is
sufficient Network types
Network types
In the same way that there are two types of information streamassociated with the different
media types - continuous and block- mode
Two types of communications channel associated with the various network types, one that
operates in a time-dependent way known as circuit-mode the other in a time-varying way
known as packet- mode
The first is known a synchronous communications channel since it provides a constant bit
rate service at a specified rate
The second is known as an asynchronous communications channel since it provides a
variable bit rate service Circuit-mode
A circuit-mode network is shown in Figure 1.19
It comprises an interconnected set of switching offices/exchanges towhich the subscriber
terminals/computers are connected
This type of network is known as a circuit switched network and, prior to sending any
information, the source must first set up a connection through the network
Each subscriber terminal/computer has a unique network-wide number/address associated
with it and, to make a call, the source first enters the number/address of the intended
communication partner Circuit-mode
The local switching office/ exchange then uses this to set up a connection through the network
to the switching office/ exchange to which the destination is connected
Assuming the destination is free and ready to receive a call, a message is returned to the
source indicating that it can now start to transfer/exchange information
Finally, after all the information has been transferred/exchanged, either the source or the
destination requests for the connection to be cleared
The bit rate associated with the connection is fixed and, in general, is determined by the bit
rate that is used over the access circuits that connect the source and destination
terminal/computer to the network
The messages associated with the setting up and clearing of a connection are known as
signaling messages
In a circuit-switched network there is a time delay while a connection is being established
This is known as the call/connection setup delay and two examples of networks that operate
in this way are a PSTN and an ISDN
With a PSTN, the call setup delay can range from a fraction of a second for a local call
through to several seconds for an international call
With an ISDN, however, the delay ranges from tens of milliseconds through to several
hundred milliseconds
Packet mode
There are two types of packet-mode network : connection-oriented(CO) and connectionless
(CL)
The principle of operation of a connection-oriented network is shown in Figure 1.20(a)
It comprises an interconnected set of packet-switching exchange (PSEs)
This type of network is known as a packet-switched network and, as with a circuit-switched
network, each terminal/computer that is connected to the network has a unique network-wide
number/address associated with it
With a connection-oriented network, prior to sending any information, a connection is first
set up through the network using the addresses of the source and destination terminals
However, in a packet switched network, the connection/circuit that is set up utilizes only a
variable portion of the bandwidth of each link and hence the connection is known as a virtual
connection or, more usually, a virtual circuit (VC)
To set up a VC, the source terminal/computer sends a call request control packet to its local
PSE which contains, in addition to the address of the source and destination terminal/computer,
a short identifier known as a virtual circuit identifier (VCI)
With a connection-oriented network, prior to sending any information, a connection is first
set up through the network using the addresses of the source and destination terminals
However, in a packet switched network, the connection/circuit that is set up utilizes only a
variable portion of the bandwidth of each link and hence the connection is known as a virtual
connection or, more usually, a virtual circuit (VC)
To set up a VC, the source terminal/computer sends a call request control packet to its local
PSE which contains, in addition to the address of the source and destination terminal/computer,
a short identifier known as a virtual circuit identifier (VCI)
Each PSE maintains a table that specifies the outgoing link that should be used to reach each
network address
On the receipt of the call request packet, the PSE uses the destination address, within the
packet to determine the outgoing link to be used
The next free identifier (VCI) for this link is then selected and two entries are made in a
routing table
The first specifies the incoming link/VCI and the corresponding out going link/VCI and the
second, in order to route packets in the reverse direction, the inverse of these, as shown in the
figure
The call request packet is then forwarded on the selected outgoing link and the same -procedure
is followed at each PSE along the route until the destination terminal/computer is reached
The VCIs that are used on the various links form the virtual circuit and, at the destination,
assuming the call is accepted, a call accepted packet is returned to the source over the same
route/virtual circuit
Each PSE first uses the incoming link/VCI to determine the outgoing link/VCI from the
routing table
The existing VCI in the packet header is then replaced with that obtained from the routing
table and the packet is forwarded on the identified outgoing link
The same procedure is followed to return information in the reverse direction and, when all
information has been transferred/exchanged, the VC is cleared and the appropriate VCIs are
released by passing a call clear packet along the VC
In a connectionless network, the establishment of a connection is not required and the two
communicating terminals/ computer can communicate and exchange information as and when
they wish
In order to do this, however, as shown in Figure 1.20(b), each packet must carry the full
source and destination addresses in its header in order for each PSE to route the packet onto
the appropriate outgoing link
In a connectionless network, therefore, the term router is normally used rather than packet
switching exchange
In both network types, as each packet is received by a PSE/router on an incoming link, it is
stored in its entirety in a memory buffer
A check is then made to determine if any transmission/bit errors are present in the packet
header - that is, the signal that is used to represent a binary 0 is corrupted and is interpreted by
the receiver as a binary 1 and vice versa - and, if an error is detected, the packet is simply
discarded
The service offered by a packet switched network is said, therefore, to be a best effort service
If a sequence of packets to be received on a number of incoming links all of which need
forwarding on the same outgoing link
Hence a packet may experience an additional delay while it is in the output queue for a link
waiting to be transmitted
Delay will be variable since it depends on the number of packets that are currently present in
the queue when a next packet arrives for forwarding
This mode of operation is known as store and forward and, as we can see, there is a packet
store-and-forward delay in each PSE/router
The sum of the store-and-forward delays in each PSE/router contributes to the overall transfer
delay of the packet across the network
The mean of this delay is known as the mean packet transfer delay and the variation about
the mean the delay variation or jitter
Multipoint conferencing
Multipoint conferencing features in many interpersonal applications including audio- and
videoconferencing, data sharing, and computer- supported cooperative working
Essentially, these involve the exchange of information between three or more terminals/
computers
In practice, because of the different modes of operation of the two network types - circuit-
switched and packet-switched- multipoint conferencing is implemented in one of two ways:
centralized and decentralized
The centralized mode is used with circuit-switched networks such as a PSTN or an ISDN
and, as we show in Figure -1.21(a), with this mode a centralized conference server is used
Prior to sending any information, each terminal/ computer to be involved in the conference
must first set up a connection to the server
Each terminal/computer then sends its own media stream- comprising, say, audio, video, and
data integrated together in some way - to the server using the established connection
The server, in turn, then distributes either the media stream received from a selected terminal/
computer or a mix of the media streams received from several terminals/computers back to all
the other terminals/computers that are involved in the conference
The decentralized mode is used with packet-switched networks that support multicast
communications
Examples include local area networks, intranets, and the Internet
In this mode, as shown in Figure 1.21(b), the output of each terminal/computer is received
by all the other members of the conference/multicast group
Hence a conference server is not normally used and instead it is the responsibility of each
terminal/computer to manage the information streams that it receives from the other members
Circuit-switched network
The QoS parameters associated with a constant bit rate channel that is set up through a circuit-
switched network include:
1.The bit rate
2. The mean bit error rate
3. The transmission delay
The mean bit error rate (BER) of a channel is the probability of a bit being corrupted
during its transmission across the channel in a defined time interval
Hence, for a constant bit rate channel, this equates to the probabilityof a bit being corrupted
in a defined number of bits
A mean BER of 10-3 therefore, means that, on average, for every1000 bits that are
transmitted, 1 of these bits will be corrupted 1. The bit rate 2. The mean bit error rate 3. The
transmission delay
For example, if the application involves speech, then an occasional bit error will go
unnoticed
But in an application involving the transfer of, say, financial information, it is essential that
the received information contains no errors
Hence-with such applications, prior to transmission the source information is normally
divided into blocks
The maximum size of which is determined by the mean BER of the communications channel
For example, if the mean BER is 10-3 , then the number of bits in a block must be
considerably less than 1000 otherwise, on average, every block will contain an error and will
be discarded
Normally, however, bit errors occur randomly and hence, even with a block size of, say, 100
bits, blocks may still contain an error but the probability of this occurring is considerably less
The transmission delay associated with a channel is determined not only by the bit rate that
is used but also delays that occur in the terminal/ computer network interfaces (known as codec
delays), plus the propagation delay of the digital signals as they pass from the source to the
destination across the network
This is determined by the physical separation of the two communicating devices and the
velocity of propagation of a signal across the transmission medium
In free space, for example, the latter is equal to the speed of light (3 xl0 8 ms-1) while it is a
fraction of this in physical media, a typical value being 2 x l0 8 ms-1
Notice that the propagation delay in each case is independent of the bit rate of the
communications channel and, assuming the codec delay remains constant, is the same whether
the bit rate is 1kbps, 1Mbps, or 1Gbps