Multimedia Communication Overview
Multimedia Communication Overview
Module 1: Introduction, multimedia information representation, multimedia networks, multimedia applications, media types,
communication modes, network types, multipoint conferencing, network QoS application QoS.
Introduction:
Multimedia communication includes a range of applications and networking infrastructures.
Definition1: The term "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:
Text: Includes both Unformatted Text - comprising strings of characters from a limited character set and Formatted Text -
comprises strings as used for the structuring, access, and presentation of electronic documents.
Images: Includes Computer Generated Image - comprising lines, curves, and circles, and Digitized Images of documents
and pictures.
Audio: Includes both low-fidelity speech - as used in telephony and high- fidelity speech - stereophonic music as used with
compact discs.
Video: Includes short sequences of moving images (also known as video clips) and complete movies/films.
Definition2: Multimedia is any combination of text, art, sound, animation, and video. It is delivered to the user by electronic
or digitally manipulated means. A multimedia project development requires creative, technical, organizational, and business
skills.
Definition3: Multimedia is the presentation of a (usually interactive) computer application, incorporating media elements such
as text, graphics, video, animation and sound on computer.
Multimedia applications may involve either of the following:
Person-to-Person communications or
Person-to-Systemcommunications
Person-to-Person communicates using suitable Terminal Equipment (TE)
Person-to-System communications:
Person interacts with the system using suitable Digital device like workstation or multimedia personal computer (PC).
These Digital device are located either in homes or offices.
Basically 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 in some way alternatively It may also contain - a library of digitized
movies/videos.
User interacts with the server by means of a suitable selection device connected to the Set-top box (STB) associated with a
television or modem used with the computers.
Networking infrastructure: provided using a number of different types of network
Networks: Two types
Designed initially to provide just a single type of service due to advances in various technologies these networks can now
provide a range of different other services.
Ex 1: PSTN (Public Switched Telephone Network) or GSTN (General Switched Telephone Network) – designed initially
to provide the basic switched telephone service but due to the Advances in digital signal processing hardware and associated
software PSTNs/GSTNs now provide a range of more advanced services involving - text, images, and video.
Ex 2: Data network: designed initially to support basic data applications - e-mail, file transfers, and others now support a
much richer set of applications - which involve images, audio, and video.
Designed from the outset to provide multimedia communication services.
Ex 1: ATM networks.
Multimedia Information Representation:
Applications involving text and images - comprise blocks of digital data units.
Text data - typical unit is block of characters with each character represented by, fixed number of Binary digits (bits) or
Codeword.
Digitized image data - comprises a 2-D block of pixels (picture elements) with each pixel represented by a fixed number of
bits.
Applications involving text and images: comprise the short request for a file.
Ex.: file contents being returned, the duration of the overall transaction is relatively short.
Applications involving Audio and Video Signals: Vary continuously with time as the amplitude of the speech, audio, or video
signal varies.
Ex.: Typical telephone conversation can last for several minutes and Movie (comprising audio and video) can last for a number
of hours.
Applications involves of single type of media: Basic form of representation of the particular media type is often used.
Applications involving 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.
Applications involving of different media types: We integrated together in some way as it's necessary to represent all 4 media
types in a digital form.
For text and images: This (digital) is their standard form of representation.
For audio and video: since, their basic forms of representations are analog signals
- these must be converted into a corresponding digital form - before they can be integrated with the two other media types.
Digitization of an audio signal: produces a digital signal with amplitude of the signal varies continuously with time and is of
relatively high bit rate, is measured by bps (bits per second) and for speech signal a typical bit rate of 64 kbps.
Applications involving audio can be of a long duration: this bit rate must be sustained for an equally long time period.
Digitization of video signal: the same applies as that of audio signals but, except that the much higher bit rates and longer time
durations are involved.
In general, 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 hence we go for compression.
Compression: It's a technique first applied to the digitized signals in order to reduce the resulting bit rate to a level which can
support be supported by various networks.
Compression to text and images: To reduce the time delay between a requests being made for some information and the
information becoming available on the screen of a computers or over others
Multimedia Networks:
Five basic types of communication networks 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.
1,2, and 3 networks are initially designed to provide just a single type of service as listed as below:
Telephone networks: telephony
Data networks: data communications
Broadcast television networks: broadcast television
Technological developments enabled these networks to provide additional services.
4, and 5 networks: Designed from the outset to provide multiple services.
Telephone networks:
Public Switched Telephone network (PSTNs) has been in existence for many years and have gone through many changes over
the time.
Designed to provide a basic switched telephone service which, with the advent of the other network types has become
known as POTS (Plain Old Telephone Service).
'Switched': term is used to indicate that the subscriber can make a call to any other telephone that is connected to the
total network.
Initially such networks spanned just a single country later, telephone networks of different countries were interconnected so,
that they now provide an international switched service.
Main components of the network are shown in the Fig below.
Local Exchange/End Office: telephones located in the home or in a small business are connected directly to their nearest
LEs/Eos.
Private Branch Exchange (PBX):
Telephones located in the medium or large office/site are connected to a PBX or Private switching Office.
Provides a (free) switched service between any two telephones - that are connected to it.
Connected to its nearest LE (public), which enables the telephone that are connected to the PBX also to make calls
through a PSTN.
Cellular Phone Networks: Been introduced which provide the similar service to the mobile subscribes by means of the
handsets that are linked to the cellular phone network infrastructure by radio.
MSC (Mobile Switching Center): it's the switch used in the cellular phone network Like the PBXs also, connected to a
switching office in a PSTN which, enables both sets of subscribers to make calls to one another.
IGE (International Gateway Exchange): route and switch the international calls.
General scheme of modem is shown in the Fig below.
Speech signal: is an analog signal varies continuously with time according, to the amplitude and frequency variations of the
sound resulting from the speech.
Microphone: used to convert this into an analog electrical signal. Telephone networks operate in circuit mode which means,
for each call a separate circuit is set up through the network of the necessary capacity for the duration of the call.
Access circuits: link the telephone handsets to a PSTN or PBX were designed to carry the 2-way analog signals associated with
a call.
Hence, within the PSTN all the switches and the transmission circuits that interconnect them operate in digital mode to carry a
digital signal a stream of binary 1s and 0s over the analog access circuits require the device modem.
Modem:
At the sending side: modem converts the digital signal output by the source digital device into an analog signal which is,
compatible with a normal speech signal it is routed through the network in the same way as a speech signal.
At the receiving side: modem converts the analog signal back again into its digital form before, relaying this to the destination
digital device.
Have the necessary circuits to set up and terminate the call.
Using a pair of modems: at each subscriber access point a PSTN can also be used to provide a switched digital service.
Early modems: supported only a very low bit rate service of 300bps.
Modems now support, bit rates of up to 56kbps as the result of advances in digital signal processing circuits and is sufficient, to
support various applications comprising of text and images integrated together and also services that comprise speech and low
resolution video modems are now available to use with same access circuits that provide a high bit rate channel which is in
addition to the speech channel used for telephony the bit rate of this second channel, typically is such that it can support high
resolution audio and video hence, they are used to provide access to servers that support a range of entertainment related
applications.
Figure below shows the general scheme of this, and such applications need bit rates in excess of 1.5 Mbps.
Technological advances in modems area have been made PSTNs can now support speech applications and also a wide range of
other multimedia communication applications.
Data networks:
Designed to provide basic data communication services such as e-mail and general file transfers.
User equipments - connected to data networks: are the computers such as a PC, a workstation, or an e-mail/file server.
Two widely deployed types of data networks:
X.25 network and
Internet.
X.25 network: operational mode is restricted to relatively low bit rate data applications. Hence, unsuitable for most multimedia
applications.
Internet: Made up of a vast collection of interconnected networks all of which operate using the same set of communication
protocols.
Communication protocol: an agreed set of rules that are adhered to by all communicating parties for the exchange of
information.
Rules define the sequence of messages that are exchanged between the communication parties and the syntax of these messages.
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 manufacturer this is the origin of the term "open systems interconnection".
Figure below shows a selection of the different types of interconnected network.
User at home or in a small business access to Internet is through an intermediate: ISP (Internet Service Provider) network
normally, this type of user wants access to the Internet intermittently the user devices are connected to the ISP network either
through a PSTN with modems or through an ISDN (Integrated Services Digital Network which provide access at a higher bit
rate).
Business user - obtain access through a site/campus network if, the business comprises only a single site or obtain access
through an enterprise-wide private network if, it comprises multiple sites.
Colleges and Universities In the case of a single site/campus: network is known as a (private) LAN (Local Area Network), In
the case of sites that are interconnected together using an inter-site backbone network to provide a set of enterprise-wide
communication services network is known as an enterprise-wide private network Providing communication protocols used by
all the computers connected to the network are the same as those defined for use with Internet.
Enterprise network (Intranet): all internal services are provided by using the same set of communication protocols, as those
defined for the Internet.
IBN (Internet Backbone Network): different types of network are all connected to it through an interworking unit called
gateways.
Gateways (Router): an interworking unit connects IBN and the different types of network responsible for routing and relaying
all messages to and from the connected network hence, also called as a router.
Packet mode: all data networks operate in this mode.
Packet: container for a block of data and has head in which, address of the intended recipient computer (which is used to route
the packet through the network).
This mode of operation is 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.
Multimedia PCs: have become available that support a range of other applications.
Ex.: with the addition of microphone and a pair of speakers with sound card and associated software to digitize the speech PCs
now are used to support telephony and other speech-related applications with the addition of video camera and associated
hardware and software a range of other applications involving video can be supported.
Due to those availability above of higher bit rate transmission circuits and routing nodes have become available, and also more
efficient algorithms to represent speech, audio and video in a digital form.
Packet-mode networks and the Internet in particular: support general data communication applications and also a range of
other multimedia communication applications involving speech, audio, and video currently.
ISDN can support a range of multimedia applications Due to the relatively high cost of digitizing the access circuits: cost of the
services associated with an ISDN is higher than the equivalent service provided by a PSTN.
Broadband multiservice networks:
Designed in mid-1980s for use, as public switched networks to support a wide range of multimedia communication
applications.
Broadband: term used to indicate the circuits associated with a call could have bit rates in excess of the maximum bit rate of
2Mbps 30X64kbps provided by an ISDN.
B-ISDN (Broadband Integrated Services Digital Networks): alternate names for broadband multiservice networks since,
were designed to be an enhanced ISDN.
N-ISDN (Narrow Integrated Services Digital Networks): alternate name for ISDN.
B-ISDN: when in first technology associated with the digitization of the video signal using were, in general, an ISD could not
support services that included video.
Due to considerable advances in the field of compression from ISDN now support multimedia communication applications that
includes video, and also can the other 3 types of network combined effect, the slow down considerably the deployment of B-
ISDN.
Number of the basic design features associated with the B-ISDN: have been used as the basis of other broadband multiservice
networks.
Ex.: A multiservice network implies that the network must support multiple services.
Different multimedia applications require different bit rates the rate being determined by the types of media that are involved
hence, switching and transmission methods that are used within these networks must be more flexible than those used in
networks such as a PSTN or ISDN which were initially designed to provide a single type of service.
To have this flexibility:
All the different media types associated with a particular application are first converted in the source equipment into a
digital form.
These to be integrated together.
Resulting binary stream is divided into multiple fixed-size packets called cells.
Information streams: of this type provides a more flexible way of both transmitting and switching the multimedia information
associated with a the different types of application.
Ex.: Transmission terms in: cells relating to the different applications can be integrated together more flexibly.
Use of fixed-sized cells: means the switching of cells can be carried out much faster than, if variable-length packets were used.
Different multimedia applications generate cell streams of different rates: this mode of operation in rate of transfer of cells
through the network also varies hence, the name: ATM (Asynchronous Transfer Mode) ATM networks (Broadband
multiservice networks) - alternate name: Cell-Switching Networks.
Ex.: ATM LANs - span a single site, ATM MANs – span large town or city.
Ex.: For broadband multiservice network is shown in the Figure below
Being used as a high-speed backbone network to interconnect a number of LANs distributed around a large town or city.
Note: Two of the LANs are ATM LANs and other two are simply higher-speed versions of older data only LANs. It's the
typical of ATM networks which must often interwork with older (legacy) networks.
Multimedia Applications:
Many and varied applications involving of multiple media types present.
Major categories of multimedia applications:
Interpersonal communications.
Interactive applications over the Internet.
Entertainment applications.
In many instances networks used to support applications were initially designed to provide the service which involves just the
single type of medium and with advances in technology, made multimedia applications support possible along with initial
designed of basic services being from those possible and in some applications basic designed applications become - still more
enhanced form is of possible.
Multimedia PC with microphone and speakers, if using user can make telephone calls through PC.
This requires the telephone interface card and associated software called CTI (Computer Telephony Integration).
The advantages of using PC, instead of conventional telephone for calls are:
User can create his or her own private directory of numbers and initiate a call simply by selecting the desired number on
the PC screen.
Circuit’s bandwidth is more (providing access circuits to the network has sufficient capacity).
Integration of telephony with all the other networked services are possible by PC.
In addition to Telephony many public and private networks support additional services.
Ex.: Voice-mail and Teleconferencing
Voice-mail: Used when the called party being unavailable Spoken message is then be left in the voice mail box of the called
party Voice mail server, located in the central repository had voice mail boxes, Message can be read by owner of the mailbox
the next time he, or she contact the server.
Teleconferencing: Calls involve multiple interconnected telephones/PCs. Person can hear and talk to all of the others involved
in the call called the conference call/teleconference
call since, it involves a telephone network or audio conference call which require an
audio bridge - a central unit which supports to set up a conference call automatically.
Internet was used to support telephony. Initially, designed to support computer-to- computer communications Just (multimedia)
PC-to-PC telephony was supported subsequently, extended so that a standard telephone could be used.
Figure below shows telephone over the Internet below
PC-to-PC telephone call: Standard addresses are used to identify individual computers connected to the internet are used same
way as for a data transfer application.
Internet: operates in the 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.
Thus Telephony over the Internet is known as Packet voice as the network protocol associated with the internet is called the
Internet Protocol (IP), Voice over IP (VoIP).
Telephony gateway: It’s a Interworking unit to connect the PC connected to the Internet and a telephone connected to the
PSTN/ISDN - since both operate in the circuit mode PC user sends a request to make a telephone call to a preallocated telephony
gateway using the latter’s internet Address Gateway requests from the source PC the telephone number of the called party
assuming user is registered for this service. Source gateway on receipt of above initiates the session (call) with the telephony
gateway nearest to the called party using the Internet address of the gateway. Called party then, initiates a call to the recipient
telephone using its telephone number and the standard call procedure of the PSTN/ISDN.
Assuming the called party answers called gateway signals back to the PC user through the source gateway that the call can
commence. Similar procedure followed to clear the call on completion.
Image Only:
Exchange of electronic images of documents is an alternate form of interpersonal communications over PSTN/ISDN
known as Facsimile (Simply, fax).
Figure below illustrates facsimile
Communication involves use of the pair of fax machines one at each network termination point.
Document sending: caller keys in the telephone number of the intended recipient, a circuit is set up through the network in the
same way as for a telephone call 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 it’s 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 after the last page of the document has been sent and
received connection through the network is cleared by the calling machine in the normal ways.
PC fax:
PC can be used instead of the normal fax machine to send an electronic version of document stored directly within the PCs
memory. Digital image of each page of the document is sent in the same way as the scanned image produced by a conventional
fax machine.
With Telephony this requires a telephone interface card and associated software, latter operates in the same way as like the fax
machine so, and terminal at the called side can be either a fax machine /another similar PC.
It Is Possible to send (by using LAN interface card and associated software) the digitized document over other network types
such as an enterprise network particularly, this mode of operation useful when working with paper-based documents, such as
invoices.
Text only:
Ex.: E-mail (Electronic mail).
User terminal is normally are normallya PC or a workstation.
Figure below shows various operational scenarios
User at home access to the Internet through the PSTN/ISDN, and through an intermediate ISP network.
Business users obtain access either through an enterprise network/site or campus network.
Email servers: One or more associated with each network Collectively contain a mailbox for each user connected to that
network User can both create and deposit mail his/her mailbox read mail from it. Standard Internet communication protocol
used by e-mail servers and internetwork gateway.
Figure Below. shows the format of the text-only e-mail message
At the head: unique Internet-wide name of both the sender and recipient of the mail, In addition present mail copy can be sent
to multiple recipients each of whom is listed in the cc part of the mail header ‘cc’ acronym for the carbon copy the original
means of making (paper) copies of documents Text only mails content: comprise unformatted text typically, strings of ASCII
characters.
Text and Images:
CSCV (Computer-Supported Cooperative Working) application: involves – text and images integrated.
Network used: enterprise network/LAN/Internet.
Figure below. shows the general scheme
Typically distributed group of people each in the place of work are all working on the same project.
User terminal is either a PC or a workstation.
Shared whiteboard: Window on each person’s display is used as the shared workspace, display comprises integrated text and
images.
Software associates comprises of whiteboard program, a central program and a linked set of support programs, one in each
PC/workstation.
Linked set of supported programs made up of change-notification part and update-control part.
Change-notification part: Sends details of the changes in the whiteboard program whenever, a member of the group
updates the contents of whiteboard.
Update-control part: Present in each of the other PCs/workstations obtain above change information in turn, proceed to update
the contents of their copy of the whiteboard.
In Cases of Home use: Terminals used normally dedicated to providing the videophone service.
In Cases of Office Use: Single multimedia PC/workstation is used to provide videophone service together with a range of other
services.
In both the cases: video camera, microphone and speaker used for telephony by the terminals/PCs.
Dedicated terminal using a separate screen is used for the display Multimedia PC or workstation using a window of the
PC/workstation screen to display the moving image of the called party.
Network must provide two-way communication channel between 2 parties of sufficient BW to support the integrated speech-
and-video generated by each terminal/PC.
Integration of video and speech: Bandwidth of the access circuits required to support is higher than that require for speech
only.
Desktop videoconferencing call: Telephony like: call may involve not just 2 persons and so, terminals/PCs several people each
located in their own office.
Used widely in large corporations involving multiple geographically distributed sites to minimize the travel between the
various locations. Large corporations of this type have enterprise-wide network to link the sites together MCU (Multipoint
control unit) is a Central unit to support the videoconferencing. Videoconferencing server associated with the network used in
few cases.
Figure above shows separate window on screen of each participant’s PC/workstation should be used to display video image of
all the other participants.
Needed to implement displaying of video image of all the other participants on screen of each participant this requires:
Multiple integrated speech-and-video communication channels, one for each participant, being sent to each of the other
participants needed to do this which Require more bandwidth than is available.
Integrated speech-and-video information stream: from each participant is sent to the MCU which then, selects just a single
information stream to send to each participant.
Ex.: voice-activated MCU
MCU whenever detects a participant speaking it relays the information stream from the participants to all the other participants
so, a single 2-way communication channel is needed between each location and the MCU is needed thereby reducing the
communication bandwidth needed considerably.
Some Networks such as LANs and the Internet supports Multicasting where all transmissions from any of the PCs/workstations
belonging to a predefined multicast group are received by all the other members of the group Possible to hold a conferencing
session without an MCU possible with networks that support multicasting.
Figure below shows the principle of this is only feasible when only a limited number of participants are involved owing to the
high load it places on the network.
In Figure below a person at one location is communicating with a group of people at another location.
Ex.: for this case transmission of a live lecture or seminar, typically information stream, transferred from the lecturer to the
remove class would be integrated speech-and video together with electronic copies of transparencies, and other documents used
in the lecture In reverse direction 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.
Communication requirements in terms, these are similar to those for a two-party videophone call. If the lecturer is relayed to
multiple locations a separate communication channel is required to each remote site or MCU is used at the lecturer’s site.
Relatively high BW that is involved network is either an ISDN (supports of multiple 64kbps channels) or a broadband
multiservice network if one is available.
In Figure above there is a group of people at each location. This type is in use from many years was the first example of
videoconferencing. Normally, a group of people are present at each location.
Videoconferencing studios: Specially equipped rooms are used – which contain all the necessary audio and video equipment,
comprising of one or more video cameras, a large- screen display, and associated audio equipment, all of which are connected to
a unit called videoconferencing system.
Conference can involve just 2 locations or more usually, multiple locations (in this latter case an MCU is normally, used to
minimize the BW demands on the access circuits to the network) as Figure in MCU is shown. as the central facility within the
network and hence, only a single 2-way communications channel is required for each access circuit of the network. Ex.: this
type of arrangement, with a telecommunications-provider conference.
If a private network alternately used MCU is normally located at one of the sites Communication requirements, are then more
demanding since, it must support multiple input channels one for each of the other sites and a single output channel, the stream
from which must be broadcast to all of the other sites.
Multimedia:
Assumption: The information content of each e-mail message consisted of text only used in the earlier discussed.
Ex.: In addition an mail containing, other media types such as images, audio, and video are also used like voice-mail, video-mail,
and multimedia mail.
Voice-mail: Similar in principle to earlier discussed telephone networks. Internet-based voice-mail there is a voice-mail
server associated with each network, in addition to e-mail server.
Figure below Shows this
User first enters the voice message addressed to the intended recipient local voice-mail server then, relays this to the server
associated with the intended recipient’s network stored voice message is then, played out the next time the recipient accesses
voice-mailbox.
Same mode of operation is used for video-mail except, the mail message comprises an integrated speech-and video sequence.
Multimedia mail: An extension of text-only mail in as much as the basic content of the mail comprises textual information.
Textual information is annotated with a digitized image, a speech message, or a video message, as in Figure.
Speech-and-video case in 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.
Recipient can always receive the basic text-only message but, the multimedia annotations can be received only if the terminal
being use by the recipient supports voice and/or video.
Interactive Applications Over The Internet:
Internet is used to support a range of interactive applications along with interpersonal communication applications.
Ex.: WWW (World Wide Web) or simply Web server comprises the linked set of multimedia information servers that
are geographically distributed around the Internet.
Total information stored on all the servers is equivalent to a vast library of document.
Figure below shows the general principle
Each document comprises a linked set of pages and linkages between the pages are known as hyperlinks.
Hyperlinks are pointers also known as references to other pages of the same document or to any other document within the
total web so, a reader of the document has the option at well-defined points throughout the pages that make up a document to
jump either to a different page of the same document or, to a different document. Also, to return subsequently to a specific
point on a page at a later time.
Optional linkage points within documents are defined by the creator of the document and are known as anchors for which the
necessary linkage information is attached.
Hypertext are documents comprising only texts and are created using hypertext.
Hypermedia are documents comprising multimedia information and are created using hypermedia.
Figure below Shows general structure of this type of document.
There is no central authority for the introduction of new documents into the web. On side in anyone create a new document
providing the server has been allocated an Internet address, and make hyperlink references from it to any other document on the
web.
URL (Uniform Resource Locator) is a Document’s unique address which identifies both location of the server on the Internet,
where the first page of the document is stored and also the file reference on the server.
Home page is the First page of the document all the hyperlinks on this and other pages have similar URLs associated with
them physical location of a page is transparent, to the user and in theory can be located anywhere on the web.
A Standard format is used for writing documents is known as HTML (Hyper Text Markup Language) and is also used for
writing client software to explore the total contents of the web, i.e., the contents of the linked information on all the web servers.
A Browser is a Client function and there are number of user-friendly browsers available to explore visited servers and to open
up a dialog with a particular server at the click of the mouse. Once the desired document has been located, the user simply clicks
on an anchor point within a page of the document to activate the linkage information stored at that point Possible to return to the
previous anchor at any time.
With the hypertext document: Anchor is usually, an underlined word or phrase.
With the hypermedia document: Anchor is usually, an icon of an appropriate shape.
Ex.: Loudspeaker for a sound annotation for a video camera for a video clip.
In Some applications client simply wishes to browse through the information stored at a particular site. Ex.: Browsing through
sales literature, product information, application notes periodicals, newspapers, and so on. In general, no charge for accessing
this information however, access to books, journals, and similar documents may be by subscriptions only.
Teleshopping (home shopping)/ Telebanking (home banking) applications: A client may wish not only to browse through
the information at a site but also to initiate an additional transaction Server must provide additional transaction processing
support for, say, ordering and purchasing since, this will also often involve financial transaction, more rigorous security
procedures are required for access and authentication purposes.
Entertainment Applications:
Entertainment applications can be of 2 types:
Movie/video-on-demand
Interactive television
Movie/Video-On-Demand:
The video and audio associated with entertainment applications must be of a much higher quality/resolution. Since, wide-screen
televisions and stereophonic sound are often used.
Digitized movie/video with sound requires a minimum channel bit rate (bandwidth) of
1.5Mbps. Hence, network used to support this application, must be either a PSTN with a high bit rate modem or a cable
network of this type.
For PSTN: high bit rate channel provided by the modem used only over the access circuit and provides additional services to
the other switched services that the PSTN supports.
Figure below. Shows – the general operating scheme in both the cases.
Information stored on the server: collection of digitized movies/videos. Normally, subscriber terminal comprises a conventional
television with device for interaction purposes. User interactions are relayed through the server through a set-top box which also
contains the high bit rate modem.
MOD (Movie-On-Demand)/VOD (Video-On-Demand): From suitable menu subscriber is able to browse through the set of
movies/videos available and initiate the showing of a selected movie. Subscriber can control the showing of the movie by using
similar controls to those used on a conventional VCR (Video Cassette Recorder) i.e., pause, fast-forward, and so on.
Key feature of MOD: a subscriber can initiate showing of a movie selected from a large library of movies at any time of the day
or night.
From Figure below, 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 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 it
is very challenging and costly since, the cost of the server is directly related to the aggregate information flow rate from it.
Server: if, supporting a large number of subscribers it is common for several subscribers to request the same movie within a
relatively short time interval between each request.
Alternative mode 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 below.
N-MOD (Near Movie-On-Demand): in this mode of operation all request 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 clearly, the viewer
is unable to control the play out of the movies.
Similar applications as above been made use in Business environment except, the stored information in the server is typically,
training and general educational material, company news, and so on and, thus the number of stored videos is normally much
less as is the number of simultaneous users so, video servers required are less sophisticated than those used in public MOD/N-
MOD systems.
Stored video streams/programs are often in a different format is as that of CD-ROMs since, the received video stream can then be
displayed directly on the screen of a multimedia PC or workstation.
Communication requirements of the private networks are the same as those identified for use with a public networks.
Interactive Television:
Broadcast television networks: include cable, satellite, and terrestrial networks.
Basic service of this network is diffusion of both analog and digital television (and radio) programs.
STB (Set-Top Box): associated with these networks has a modem within it.
For cable networks as in Figure below. , STB provides both a low bit rate connection to the PSTN and a high bit rate
connection to the Internet.
By connecting appropriate TE to the STB a keyboard, telephone, and so on subscriber is able to gain access to all the services
provided through the PSTN and the Internet. Through the connection to the PSTN subscriber is able to actively respond to the
information being broadcast it’s the origin of the term interaction television.
Typical uses of the return channel are for voting, participation in games, home shopping, and so on.
As in Figure a 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 the connections to the PSTN and the Internet.
A conference server is used output of each terminal/computer is sent to the server either over individual circuits terminals A and
B or using multicasting terminals C and D Server that determines output stream(s) to be sent to each terminal.
Types of conferencing based on media Type:
1. Data conferencing
2. Audio conferencing
3. Videoconferencing
4. Multimedia conferencing
Data conferencing: Involves data only. Ex.: include data sharing and computer-supported cooperative working.
Audio conferencing: Involves audio (speech) only.
Videoconferencing: Involves speech and video synchronized and integrated together.
Multimedia conferencing: Involves speech, video, and data integrated together.
Data conferencing:
Information flow between the various parties is relatively infrequent; Conference server is a general-purpose
computer with the conference function implemented in software.
With the other 3 types of conferencing the information flows demand the use of special purpose units.
Audio conferencing:
Audio bridge is the unit.
Typical units supporting 6 through 48 conference participants.
Video and multimedia conferencing:
MCU (Multipoint Control Unit) is the unit.
Due to volume and rate of the information being exchanged centralized mode of working is used with both network types.
MCU consists of 2 parts:
Multipoint Controller part (MC): Concerned with the establishment of connections to each of the conference participants and
with the negotiation of an agreed set of operational parameters screen resolution, refresh rate, and others.
Multipoint Processor (MP): Concerned with the distribution of the information streams generated during the conference.
Include such functions as the mixing of the various media streams into an integrated stream, voice-activated switching and
continuous presence.
Audio bridge:
When using a audiobridge,acall is scheduled for a particular date, time, and duration.
Everyone who is to take part in the call is assigned a user ID and password. At appropriate time all participants call in and after
verified, they can hear and speak to the other participants.
In the same way MCU when using a call is scheduled as for an audio bridge.
Once the conference starts each participant can hear, see, and share data with the other participants MCU with
Dial-in mode: The participants calling in.
Dial-out mode: MCU calls the participants provides better security.
Voice-activated switching mode: Face of the participant is displayed in a window on the screen of the participant’s
terminal/computer, in the second window the face of the remote participant who is currently talking. When another participant
starts to talk face of the new speaker replaces the face of the current remote participant. In the event two or more participants
starting to talk at the same time MCU normally selects person who speaks the loudest.
Continuous -presence mode: Remote window is divided into a number of smaller windows each of which displays the face of
the last set of participants who spoke or who are currently speaking. With both modes of speech from all participants are
normally mixed into a single stream and hence, each participant can always hear what is said by all the other participants.
Network QoS:
Definition: “Operational parameters associated with a communications channel through a network collectively determine the
suitability of the channel in relation to its use for a particular application”.
QoS parameters are different for circuit-switched and packet-switched networks.
Circuit-Switched Network:
QoS parameters associated with a constant bit rate channel that is set up through a
Circuit-switched network are
Bit rate
Mean bit error rate
Transmission delay
Bit rate:
In digital telecommunication, the bit rate is the number of bits that pass a given point in a telecommunication network in a
given amount of time, usually a second.
A bit rate is usually measured in some multiple of bits per second.
The term bit rate is a synonym for data transfer rate (or simply data rate).
In practice, both circuit-switched and packet-switched provide unreliable service (best- try or best-effort service).
Unreliable service (Best-try/Best-effort service) is a type of service where any blocks containing bit errors will be discarded
either with the network (packet-switched network) or in the network interface at the destination (both packet-switched and
circuit switched networks).
Application dictates that only error-free blocks are acceptable, it is necessary for sending terminal/computer to divide the source
information into blocks of a defined maximum size and for the destination to detect when a block is missing.
When the above occurs destination must request source send another copy of the missing block. This type of service is called
Reliable service.
Due to above case delay is introduced so, that retransmission procedure should be invoked relatively infrequently which dictates
a small block size. This leads to high overheads since, each block must contain additional information that is associated with
the transmission procedure.
So, choice of block size compromise between increased delay resulting from a large block size hence, retransmission and the
loss of transmission BW from the high overheads of using a small block size.
Transmission delay:
Associated with the channel is determined by:
Bit rate used
Delays occur in the terminal/computer network interfaces (codec delays) + propagation delay of the digital, as they pass from
source to destination, across the network.
The above is determined by physical separation of 2 communicating devices and velocit y of propagation of a signal, across the
transmission medium (free space: 3*108 m/s and a fraction of this in physical media, a typical value 2*108 m/s).
Propagation delay: in each case is independent of the bit rate of the communication channel. Assuming codec delay remains
constant; propagation delay remains same whether bit rate is 1kpbs, 1Mbps, or 1Gbps.
PROBLEM
Drive the maximum block size that should be used over a channel which has a mean BER probability of 10-4 if the probability
of a block containing an error –and hence being discarded – is to be 10-1.
ANSWER:
PB=1-(1-P)N
Hence 0.1=1-(1-10-4)N and N=950bits Alternatively, PB=N x P
Hence 0.1=Nx10-4 and N=1000bits
Packet-switched network:
QoS parameters – associated with a packet-switched network include
1. Maximum packet size
2. Mean packet transfer rate
3. Mean packet error rate
4. Mean packet transfer delay
5. Worst-case jitter
6. Transmission delay
Packet-switched network: rate of packets transfer across the network influenced strongly by bit rate of the interconnecting links
due to, variable store-and-forward delays in each PSE/router. Actual rate of transfer of packets across the network is also
variable.
Mean packet transfer rate: Measure of the average number of packets transferred across the network/second coupling with
packet size being used determines equivalent mean bit rate of the channel.
Mean PER (Mean Packet Error Rate): Probability of a received packet containing one or more bit errors. It is same as block
error rate, associated with a circuit-switched network. Thus related to both maximum packet size and worst-case BER of the
transmission links which interconnects PSEs/routers that make up the network.
Mean packet transfer delay: Summation of the mean store-and-forward delay that a packet experiences in each PSE/router
which, it encounters along a rout.
Jitter: Worst-case variation in the mean packet transfer delay.
Transmission delay: Same for network operates in the packet mode or a circuit mode Includes: Codec delay, in each of the
two-communicating computers and Signal propagation delay.
PROBLEM:
Determine the propagation delay associated with the following communication channels:
Assume that the velocity of propagation of a signal in the case of (1) and (2) is 2x108ms-1 and in the case of (3) 3x108ms-1
ANSWER:
Physical separation
Propagation delay Tp =
velocity of propagation
Application QoS
Network QoS parameters define what the particular network being used provides rather what application requires.
Application has its own QoS parameters requirement associated with it: Application involving images: Ex.: parameters may
include minimum image resolution and size. Application involving video: Ex.: parameters may include digitization format and
refresh rate.
Effect of jitter is overcome by retaining a defined number of packets in a memory buffer at the destination before play out of the
information bitstream is started. Memory buffer operates using a first-in, first-out (FIFO) discipline. Number of packets
retained in the
buffer before output starts is determined by the worst-cast jitter and the bit rate of the information stream.
Figure shows when using the packet-switched network for this type of application additional delay is incurred at the source as
the information bitstream is converted into packets.
Packetization delay: Additional delay, incurred at the source as the information bitstream is converted into packets adds to the
transmission delay of the channel.
To minimize overall input-to-output delay, packet size used for application is kept small, but of sufficient size to overcome the
effect of the worst-case jitter.
To simplify determining a particular network which can meet the QoS requirement of an application: Number of standard
application service classes have been defined.
Each service class with specific set of QoS parameters associated – for network, can either meet this or not.
Networks support a number of different service classes.
Ex.: Internet – to ensure, the QoS parameters associated with each class is met – packets relating to each class are given a
different priority then, each class packets can be differently treated.
Internet packets relating to multimedia applications involving real-time streams are given higher priority than, packets relating
to applications such as e-mail.
Packets containing real-time streams such as radio and video are more sensitive to delay and jitter, than the packets containing
textual information. Hence, during periods of congestion packets containing real-time streams are transmitted first packets
containing video are more sensitive to packet loss than, packets containing audio hence are given more priority.
PROBLEM:
A packet-switch network with a worst-case jitter of 10ms is to be used for a number of applications each of which
involves a constant bit rate information stream. Determine the minimum amount of memory that is required at the destination
and a suitable packet size for each of the following input bit rate. It can be assumed that the main packet transfer network
exceeds the equivalent input bit rate in each case.
(1) 64kbps
(2) 256kbps
(3) 1.5Mbps
ANSWER:
(1) At 64kbps, 10ms=640bits
Hence chose a packet size of, say, 800 bits with a FIFO buffer of 1600 bits -2 packet –and start play out of the bitstream
after the first packet has been received.
(2) At 256kbps, 10ms=2560 bits
Hence chose a packet size of, say, 2800 bits with a FIFO buffer of 4800 bits.
All types of multimedia information are process and store within the computer in a digital form.
Textual information: contains strings of characters entered through keyboard. Codeword: each character represented by a
unique combination of fixed number of bits. Complete text hence, can be represented by strings of codewords.
Image: computer-generated graphical images made up of a mix of lines, circles, squares, and so on each represented in a
digital form. Ex.: line - represented by start and end co-ordinates of the line, each coordinate being defined in the form of a
pair of digital values relative to the complete image.
Audio and video: microphone and video cameras produce electrical signals, whose amplitude varies continuously with
time amplitude indicating the magnitude of the sound wave/image-intensity at that instant.
Analog Signal: signal whose amplitude varies continuously with time. In order to store and process analog signal type of
media in a computer we should convert any time-varying analog signals into a digital form is necessary.
for speech and audio - in like, loud speakers, and for display of digitized images in like, computer monitors - digital values
of media types must be converted back again into a corresponding time-varying analog form on output – from the computer.
for a particular media type:
Conversion of analog signal into digital signal is carried out using an electrical circuit known as Signal Encoder, it
includes following steps:
Sampler: It samples the amplitude of analog signals at repetitive time intervals.
Quantization: converting amplitude of each sample into a corresponding digital value.
Conversion of stored digital sample relating to a particular media type into their corresponding time-varying analog form is
performed by a electrical circuit is known as a signal decoder.
All media types associated with the various multimedia applications stored and processed within a computer in an all-digital
form so, different media types can be readily integrated together resulting integrated bitstream can be transmitted over a
single all-digital communication network.
Digitization principles:
Analog signals:
Figure below shows general properties relating to any time-varying analog signal. In figure amplitude of signals varies
Speech is a humans produce sounds, which are converted into electrical signals by a microphone are made up of a range of
sinusoidal signals varying in frequency between 50Hz and 10kHz and for music range of signals is wider varies between
15kHz to 20kHz being comparable with the limits of the sensitivity of the ear.
Analog signal when, being transmitted through a network BW of transmission channel (range of frequencies, channel will
pass) ≥ BW of the signal if BW of channel < BW of signal some low and/or high frequency components will be lost,
thereby degrading the quality of the received signal. Such, a channel is called the bandlimiting channel as in Figure below.
Encoder design:
Signal encoder is a electronic circuit converts, time-varying analog signals to digital form.
Bandlimitng filter: remove selected higher-frequency components from the source signal (A).
Sample-And-Hold: got output of bandlimiting filter, (B) signal used to sample amplitude of the filtered signal at regular
time intervals (C) and to hold the sample amplitude constant between samples (D) signal Quantizer circuit got signal (D)
which converts each sample amplitude into a binary value known as a codeword like (E) signal.
Polarity (sign) of sample: positive or negative relative to the zero level indicated by most significant bit of each codeword.
A binary 0 indicates a positive value and a binary 1 indicates a negative value. To represent the amplitude of a time-varying
analog signal precisely require 2 things:
Signal should be sampled at a rate > maximum rate of change of signal amplitude.
Number of quantization levels used to be as large as possible.
Sampling Rate:
Nyquist Sampling Theorem: states that for an accurate representation of a time- varying analog signal it's amplitude must
be sampled at a minimum rate that is equal to or greater than twice the highest sinusoidal frequency component that is present
in the signal known as Nyquist rate, normally represented as either Hz or, or correctly, samples per second (sps). sampling
signal at a rate < Nyquist rate results in additional frequency components being generated that are not present in the original
signal which, in turn cause original signal to become distorted.
Figure below, shows effect of under sampling single-frequency sinusoidal signal caused by sampling a signal at a rate
lower than the Nyquist rate.
Ex.: original signal is assumed to 6kHz sine wave sampling rate(8ksps)< Nyquist rate (12ksps, 2*6ksps) results in a lower
frequency 2kHz signal being created in place of the original 6kHz signal such, signals called alias signals (since, they
replace the corresponding original signals).
In general, all frequency components present in the original signal higher in frequency than half the sampling frequency
being used (in Hz) generate related lower-frequency alias signals which will simply add to those making up the original
source signal thereby causing it to become distorted.
Bandlimiting filter/Antialiasing filter: source signal is passed into the bandlimiting filter to pass only those frequency
components up to that determined by Nyquist rate any higher-frequency components in the signal which are higher than this
are removed before the signal is sampled.
In practice transmission channel, used/available has a lower bandwidth than that of source signal to avoid distortion
bandwidth and hence, frequency range of the transmission channel that determines the sampling rate used rather than the
BW of the source signal in such cases, source signal may have higher frequency component, than those dictated
by the Nyquist rate of the transmission channel so, it is necessary to pass the source signal through a bandlimiting filter
(designed to pass only those sinusoidal frequency components which are within the BW of the transmission channel) so,
generation of any alias signals caused by under sampling source signal is avoided.
Quantization intervals:
To represent in the digital form the amplitudes of the set of analog samples would require an infinite number of binary
digits. When finite numbers of digits are used, each sample can be represented by a corresponding number of discrete levels.
Figure below. Shows effect of using a finite number of bits
Ex.: here, 3 bits to represent each sample including a sign bit results in 4 positive and 4 negative quantization intervals, the
two magnitude bits - being determined by the particular quantization interval the analog input signal is in at the time of each
sample.
Fig a shows: if Vmax, is the maximum positive and negative signal amplitude.
n: number of binary bits used
q: magnitude of each quantization interval is given by
Signal - anywhere within a Quantization intervals will be represented by the same binary codeword. Thus each codeword:
corresponds to a nominal amplitude level - which, is at the center of the corresponding quantization interval. Thus, Actual
signal level - may different from this by: up to + or -q/2.
Quantization Error: It is the difference between the actual signal amplitude and the corresponding nominal amplitude.
Figure below. shows quantization error values shown expanded.
Usually the error values vary randomly from sample to sample thus quantization error is also known as Quantization
noise.
Noise: This term used in electrical circuits to refer to a signal whose, amplitude varies randomly with time.
Smallest amplitude relative to its peak amplitude of the signal is the influencing factor for the choice of the number of
quantization intervals for a particular signal.
With high-fidelity music: It is important to be able to hear very quite passages without any distortion created by
quantization noise.
Dynamic range, D (of the signal) is the ratio of the peak amplitude of a signal to its minimum amplitudes.
decibels (dB): D is normally quantified using logarithmic scale.
Determining the quantization intervals, and number of bits to be used it is necessary to ensure level of quantization noise
relative to the smallest signal amplitude is acceptable.
TEXT:
Definition: “In information technology, text is a human-readable sequence of characters and the words they form that can
be encoded into computer-readable formats such as ASCII. Text is usually distinguished from non-character encoded data,
such as graphic images in the form of bitmap s and program code, which is sometimes referred to as being in "binary" (but
is actually in its own computer- readable format)”.
There are 3 Types of text used to produce pages of documents:
Unformatted Text: alternative name plaintext and it enables pages to be created comprises of strings of fixed-sized
characters from a limited character set.
Formatted Text: alternative name rich text and it enables pages and complete documents to be created which, comprise of
strings of characters of different styles, size and shape with tables, graphics, and images inserted at appropriate points.
Hypertext: It enables an integrated set of documents (each comprising formatted text) to be created which have defined
linkages between them.
Unformatted text:
Two examples of character sets widely used to create pages consisting of unformatted text strings are:
ASCII character set
Mosaic character set
Table below shows set of characters, available in the ASCII character set.
ASCII - American Standard Code for Information Interchange is one of the most widely used character sets.
Table shows binary codewords used to represent each character. Each character is represented by a unique 7-bit binary
codeword.
Use of 7 bits means there are 128 (27) alternate characters and codeword used to identify each character and is obtained by
combining the corresponding column (bits 7-5) and row (bits 4-1) bits together.
Bit 7 is MSB and Bit 0 is LSB thus codeword for uppercase M is 1001101.
Printable Characters: It is a collection of normal alphabetic, numeric and punctuation characters but ASCII total characters
also includes a number of control characters including:
Format control characters: BS (backspace), LF (Linefeed), CR (Carriage Return), SP (Space), DEL (Delete), ESC (Escape),
and FF (Formfeed).
Information separators: FS (File Separator), RS (Record Separator).
Transmission control characters: SOH (Start-Of-Heading), STX (Start-Of- Text), ETX (End-Of-Text), ACK
(Acknowledge), NAK (Negative Acknowledge), SYN(Synchronous Idle), and DLE (Data Link Escape).
Fig. b tabulates the character set which is the supplementary version of that in Fig. a.
Characters in columns 010/011 and 110/111 are replaced by set of mosaic characters.
Mosaic characters used with uppercase characters to create relatively simple graphical images.
A example application which uses mosaic character set are Videotext and Teletex which are mosaic general broadcast
information services available, through a standard television set, used in number of countries.
Some simple graphical symbols and text of larger sizes can be constructed by the use of groups of the basic symbols.
Formatted text:
It is produced by most word processing packages used extensively in the publishing sector for the preparation of papers,
books, magazines, journals, and so on.
Examples of word processing packages are MS word, libaro office, kingsoft office, office pro etc.
To print a document consisting of formatted text: printer must be first set up, microprocessor within the printer must be
programmed to detect and interpret the format-control character sequences in the defined way and to convert the following text,
table, graphic, or picture into a line-by-line form ready for printing.
Commands such as print preview often provided which cause the page to be displayed on the computer screen in a similar
way, to tell how it will appear when it is printed.
WYSIWYG: what-you-see-is-what-you-get can be achieved as above.
Hypertext:
It is a type of formatted text enables a related set of documents (known as pages) to be created which define linkage points
on pages referred to as hyperlinks between pages.
Ex.: contents of courses, current research projects, staff profiles, or publications these can also implemented as linked set of
pages on a different computer, and providing all the computers at the sites are connected to the same network (and use the
same set of communication protocols), additional hyperlinks between the two sets of ages can be introduced.
Images:
Definition: “An image is an artifact that depicts or records visual perception”.
Ex: 2-D picture.
All objects including the free-form objects made up of a series of lines connected to each other. curved line as what may
appear in practice, is a series of very short lines each made up of a string of pixels which, in the limit, have the resolution of
a pair of adjacent pixels on the screen.
Figure below. shows some examples
Attributes: each object has a number of attributes associated with it they include:
Its shape - a line, a circle, a square, and so on.
Its size - in terms of pixel positions of its border coordinates.
Color of border.
its shadow, and so on
Editing of an object involves simply, changing selected attributes associated with the object.
Ex.: As in Figure below, square can be moved to different location on the screen by simply, changing its border coordinates
and leaving the remaining attributes unchanged.
PCM speech:
Interpersonal applications involving speech uses PSTN for communication purpose.
Initially PSTN operated with analog signal throughput, the source speech signal being transmitted and switched (routed)
unchanged in its original analog form progressively older analog transmission circuits were replaced by digital circuits.
The above was carried out over a number of years and because of the need to interwork between earlier analog and newer
digital equipments during the transition period design of digital equipment was based on operating parameters of the earlier
analog network.
BW of the speech circuit in this network was limited to 200Hz through to 3.4 kHz. Nyquist rate is 6.8 kHz due to poor
quality of the band limiting filters used meant that a sampling rate of 8kHz was required to avoid aliasing.
To minimize the resulting bit rate:
7 bits/sample were selected for use in North America and Japan with bit rate of 56 kbps.
8 bits/sample in Europe (both including a sign bit) with bit rate of 64kbps.
PCM (Pulse Code Modulation) is a digitization procedure and the international standard relating to this is ITU-T
Recommendation G.711.
Figure below. Shows circuits that make up a PCM encoder and decoder consisting of compressor (encoder) and expander
(decoder) circuits.
Uses linear quantization intervals where quantization intervals are equal irrespective of the magnitude of the input signal
same level of quantization noise is produced.
Effect of above is that noise level is the same for both low amplitude (quite) signals and high amplitude (loud) signals.
Ear more sensitive to noise on quite signals than it is on loud signals. To reduce the effect of quantization noise with 8
bits/sample, in a PCM system quantization intervals are made non-linear (unequal) with narrower intervals used for
smaller amplitude signals than, for larger signals these can be achieved by compressor and at the destination, the reverse
operation by the expander circuits overall operation is known as companding.
Figures below. Shows input/output relationship of both circuit’s characteristic curve for compression - compression
characteristics and characteristic curve for expansion
- expansion characteristics.
Prior to input signal being sampled and converted into a digital form by ADC, it is passed through compression circuit,
which effectively compresses the amplitude of the input signal.
Level of compression and hence, quantization intervals increases as the amplitude of the input signal increases resulting
compressed signal is passed through ADC which performs linear quantization of the compressed signal.
At the destination each received codeword is first fed into a linear DAC Analog output from the DAC is then, passed to the
expander circuit which perform the reverse operation of the compressor circuit.
Modern systems perform compression and expansion operations digitally same principles are applied.
2 different compression-expansion characteristics are in use:
µ-law: used in North America and Japan.
A-law: used in Europe and some other countries.
CD-quality audio:
Discs used in CD players and CD-ROMs are digital storage devices for stereophonic music and more general
multimedia information streams.
CD-DA (CD-digital audio) standard associated with these devices.
Music has an audible BW of from 15Hz through 20 kHz. So, minimum sampling rate is 40ksps.
In the standard actual rate used is higher this rate because of following reasons:
Allow for imperfections in the band limiting filter used.
So that resulting bit rate is then, compatible with one of the higher transmission channel bit rates available with
public networks.
Sampling rate used one of is 44.1ksps means, signal is sampled at 23 microsecond intervals.
High number of samples can be used since; BW of a recording channel on a CD is large.
Recording of stereophonic music requires 2 separate channels - so, total bit rate required is double that for mono.
Hence, bit rate/channel = sampling rate x bits/sample
= 44.1 x 103 X 16
= 705.6 kbps (for mono) Total bit rate = 2 X 705.6 = 1.411 Mbps (for stereo)
CD-ROMs also uses same bit rate which, are widely used for distribution of multimedia titles. To reduce the access delay
multiples of this rate are used within a computers.
Not feasible to interactively access a 30s portion of a multimedia title over a 64kbps channel with 1.5Mbps channel the time
is high for interactive purposes.
Problem: Assume the CD-DA standard is being used, derive:
I. The storage capacity of a CD-ROM to store a 60 minute multimedia title.
II. the time to transmit a 30 second portion of the title using a transmission channel of bit rate: a.
64kbps b. 1.5kbps
Solution:
i. The CD-DA digitization procedure yields a bit rate of 1.411Mbps. Thus, storage capacity for 60 minutes is =
= 1.411 x 60 x 60mb
= 5079.6Mbits or 634.95 Mbytes
ii. One 30 second portion of the title =1.411 x 30 = 42.33 Mbits Thus time to transmit this
data :
At 64kbps = ( 42.33 x 106 ÷ 64 x 103)
= 661.4 sec
At 1.5kbps = ( 42.33 x 106 ÷ 1.5 x 106)
= 28.22 sec
Synthesized audio:
Once digitized any form of audio can be stored within a computer, amount of memory required to store a digitized audio
waveform can be very large even for relatively short passages.
Synthesized audio can be defined as sound generated by electronic signals of different frequencies. Sound can be
synthesized by the use of sound synthesizers. The synthesizers use different programmed algorithms to generate sound to
different waveform synthesis.
Synthesized audio is often hence, used in multimedia applications since, the amount of memory required to be between
two and three orders of magnitude less that required to store the equivalent digitized waveform versions.
It is much easier to edit synthesized audio and mix, several passages together.
Figure below. Shows components that make up audio synthesizer.
Video:
Definition: Correlated sequence of images with respect to time.
Futures in range of multimedia application.
Entertainment: broadcast television and vcr/ dvd recordings;
Interpersonal: video telephony and video conferencing;
Interactive: window containing short video clips.
The quality of video required, however varies considerably from one application to another.
Ex.: in case of video telephony, a small window on the screen of a PC is acceptable while for a movie, a large
screen format is preferable.
In practice there is not just a single standard associated with video but set off standards each
targeted at a particular video application.
All the different standards are based on basic principles that are associated with Broadcast television.
Digital video:
In most multimedia applications the video signals need to be digital form since it then becomes possible to store them in the
memory of the computer and to readily edit and integrate them with other media type.
For transmission reasons the three component signals have to be combined for analog television broadcasts, with digital
television it is more usual to digitize the three component signals separately prior to transmission.
The above is done to enable editing and other operations readily performed.
Since the three component signals are treated separately in digital transmission, in principle it is possible simply to digitize
the three RGB signals make up the picture.
PC video:
Number of multimedia applications that involve live video, use a window on the screen of a PC monitor for display
purposes.
Ex.: desk video telephony, videoconferencing, and also video-in-a-window.
for multimedia applications that involve mixing live video with other information on a PC screen, the line sampling rate is
normally modified in order to obtain the required horizontal resolution like 640 (480 x pixels per line with a 525-line PC
monitor and 768 (576 x 4/3) pixels per with a 625-line PC monitor.
To achieve the necessary resolution with a 525-line monitor, the line sampling rate is reduced from 13.5 MHz to
12.2727MHz while for a 625-line monitor, the line sampling rate must be increased from 13.5 MHz to 14.75 MHz
In the case of desktop video telephony and videoconferencing, the video signals from the camera are sampled at this rate
prior to transmission and hence can be displayed directly on a PC screen. In the case of a digital television broadcast a
conversion is necessary before the video is displayed.
It is to be remembered that all PC monitors use progressive scanning rather than interlaced scanning.
Packet-mode networks, like the Internet, originally designed for general data communication applications, now also support multimedia applications involving speech, audio, and video. This expansion was made possible by more efficient algorithms for digitizing multimedia content, which, alongside higher bit rate transmission circuits and improved routing nodes, enable the integrated handling of multimedia alongside data packets .
Digitization involves converting analog signals into digital form using a signal encoder, which includes bandlimiting, sampling, and quantization processes. This is important in multimedia communications to ensure that analog data such as voice, images, and video can be processed, stored, and transmitted efficiently over digital networks. Bandlimiting removes excess frequency components, sampling encodes signal amplitude at discrete intervals, and quantization converts these samples into digital values, allowing integrated multimedia data transmission over all-digital networks .
Home and small business users typically access the Internet intermittently through an ISP network using a PSTN with modems or ISDN for higher bit rates. Larger businesses usually access the Internet through a site/campus network or an enterprise-wide private network if multiple sites are involved. Educational institutions might use a LAN for a single site or an enterprise network for interconnected sites .
X.25 networks support low-bit-rate data applications, making them unsuitable for most multimedia applications, which require higher bandwidth. In contrast, the Internet allows for transmission across a vast network of interconnected systems using standard communication protocols, making it well-suited for multimedia tasks that involve diverse media types such as audio, video, and complex internet applications due to its open communication support and higher bit rate capabilities .
Quantization noise, potentially detracting from the audio quality, occurs when analog signals are converted to digital form with limited bits for each sample. Using a minimum of 12 bits for audio and 16 for music helps mitigate this noise. PCM systems use non-linear quantization with narrower intervals for low amplitude signals, employing compression and expansion (companding) to reduce noise effectively, ensuring higher fidelity and better audio quality in network transmissions .
Modems convert digital signals from digital devices into analog signals suitable for transmission over analog circuits, compatible with voice signals, and vice versa. Initially, early modems supported low bit rates of 300bps. Advances in digital signal processing have increased the supported bit rates to 56kbps, enabling them to transmit higher-resolution audio and video, key for multimedia applications. This is because modems now provide a high-bit-rate channel in addition to the traditional voice channel .
ATM networks use fixed-size cells for fast switching and transfer, contrasting with older network types that rely on variable-length packets, which are slower to switch. This allows ATM networks to efficiently manage streams at varying rates, well-suited for high-speed multimedia applications by acting as a backbone network that can interoperate with older LANs and provide services across large areas, enabling efficient multimedia data handling .
Gateways, also known as routers, connect different network types to the Internet Backbone Network (IBN). They facilitate the routing and relaying of messages between connected networks, thus ensuring interoperability and seamless integration across various network protocols and data formats, effectively acting as the conduits for communication pathways across disparate networking environments .
Fourier analysis is crucial in signal processing as it decomposes any analog signal into a potentially infinite series of sinusoidal components, each with its amplitude and phase. This decomposition helps in understanding and manipulating signal frequency components, which is essential for designing and optimizing digital communication networks, particularly in encoding analog waves into digital formats for efficient transmission without losing fidelity .
"Open systems interconnection" allows computers of different types and manufacturers to communicate over the Internet by adhering to the same set of communication protocols. These protocols define the sequence and syntax of message exchanges, ensuring seamless interoperability and enabling a vast array of interconnected networks to operate cohesively despite their differences, which supports global communication .