Voip Technology

Voice Over Internet Protocol Technology
INDEX S No 1. 2. 3. 4. 5. Topic Data Communication Concepts OVERVIEW OF INTERNET & INTRANET SERVICES OVERVIEW OF VOICE OVER INTERNET PROTOCOL ATM Frame Relay

DATA COMMUNICATIONS CONCEPTS

DATA COMMUNICATIONS CONCEPTS
1.0 Introduction When we communicate, we are sharing information. This sharing can be local or remote. Between individuals, local communication usually occurs face to face, while remote communication takes place over distance. The term telecommunications, which includes telephony, telegraphy, and television, means communication at a distance (tele is Greek for far). The word data refers to facts, concepts, and instructions presented in whatever form is agreed upon by the parties creating and using the data. In the context of computer information systems, data are represented by binary information units (or bits) produced and consumed in the form of 0s and 1s. Data communication is the exchange of data (in the form of 0s and 1s) between two devices via some form of transmission medium (such as a wire cable). Data communication is considered local if the communicating devices are in the same building or a similarly restricted geographical area, and is considered remote if the devices are farther apart. For data communication to occur, the communicating devices must be part of a communication system made up of a combination of hardware and software. The effectiveness of a data communication system depends on three fundamental characteristics: 1. Delivery: The system must deliver data to the correct destination. Data must be received by the intended device or user and only by that device or user. 2. Accuracy: The system must deliver data accurately. Data that have been altered in transmission and left uncorrected are unusable. 3. Timeliness: The system must deliver data in a timely manner. Data delivered late are useless. In the case of video, audio, and voice data, timely delivery means delivering data as they are produced, in the same order that they are produced, and without significant delay. This kind of delivery is called real-time transmission. 1.1 Components of Data Communication A data communication system is made up of five components. 1. Message: The message is the information (data) to be communicated. It can consists of text, numbers, pictures, sound, or video – or any combination of these. 2. Sender: The sender is the device that sends the data message. It can be a computer, workstation, telephone handset, video camera, and so on. 3. Receiver: The receiver is the device that receives the message. It can be a computer, workstation, telephone handset, television, and so on. 4. Medium: The transmission medium is the physical path by which a message travels from sender to receiver. It can consist of twisted pair wire, coaxial cable, fiber optic cable, laser, or radio waves (terrestrial or satellite microwave). 5. Protocol: A protocol is a set of rules that govern data communication. It represents an agreement between the communicating devices. Without a protocol, two devices may be connected but not communicating, just as a person speaking Malayalam cannot be understood by a person who speaks only Telugu.

1.2

Networks A network is a set of devices (often referred to as nodes) connected by media links. A node can be a computer, printer, or any other device capable of sending and / or receiving data generated by other nodes on the network. The links connecting the devices are often called communication channels. Distributed processing Networks use distributed processing, in which a task is divided among multiple computers. Instead of a single large machine being responsible for all aspects of process, each separate computer (usually a personal computer or workstation) handles a subset. Applications In the short term they have been around, data communication networks have become an indispensable part of business, industry, and entertainment. Some of the network applications in different fields are the following: Marketing and sales: Computer networks are used extensively in both marketing and sales organizations. Marketing professionals use them to collect, exchange and analyze data relating to customer needs and product development cycles. Sales applications include teleshopping, which uses order-entry computers or telephones connected to an order-processing network, and on-line reservation services for hotels, airlines, and so on. Financial Services: Today’s financial services are totally dependent on computer networks. Applications include credit history searches; foreign exchange and investment services, and electronic funds transfer (EFT), which allows a user to transfer money without going into a bank (an automated teller machine). Manufacturing: Computer networks are used today in many aspects of manufacturing, including the manufacturing process itself. Two applications that use networks to provide essential services are computer-assisted design (CAD) and computer-assisted manufacturing (CAM), both of which allow multiple users to work on a project simultaneously. Electronic messaging: electronic mail (e-mail). Probably the most widely used network application is

Directory services: Directory services allow lists of files to be stored in a central location to speed worldwide search operations. Information services: Network information services include bulletin boards and data banks. A World Wide Web site offering the technical specifications for a new product is an information service. Electronic data interchange (EDI): EDI allows business information (including documents such as purchase orders, and invoices) to be transferred without using paper. Teleconferencing: Teleconferencing allows conferences to occur without the participants being in the same place. Applications include simple text conferencing

(where participants communicate through their keyboards and computer monitors), voice conferencing (where participants at a number of locations communicate simultaneously over the phone), and video conferencing (where participants can see as well as talk to one another). Cellular telephone: Cellular networks are to maintain wireless phone connections even while traveling over large distances. Cable television: Future services provided by cable television networks may include video on demand, as well as the same information, financial, and communications services currently provided by the telephone companies and computer networks. Protocols and Standards Protocols In Computer networks, communication occurs between entities in different systems. An entity is anything capable of sending or receiving information. Examples include application programs, file transfer packagers, browsers, database management systems, and electronic mail software. A system is a physical object that contains one or more entities. Examples include computers and terminals. But two entities cannot just send bit streams to each other and expect to be understood. For communication to occur, the entities must agree on a protocol A protocol is a set of rules that govern data communication. A protocol defined what is communicated, how it is communicated, and when it is communicated. The key elements of a protocol are syntax, semantics, and timing. Syntax Syntax refers to the structure or format of the data, meaning the order in which they presented. For example, a simple protocol might expect the first eight bits of data to be the address of the sender, the second eight bits to be the address of the receiver, and the rest of the stream to be the message itself. Semantics Semantics refers to the meaning of each section of bits. How is a particular pattern to be interpreted, and what action is to be taken based on that interpretation? For example,, does an address identify the route to be taken or the final destination of the message? Timing Timing refers to two characteristics: when data should be sent and how fast it can be sent. For example, if a sender produces data at 100 Mbps but the receiver can process data at only 1 Mbps, the transmission will overload the receiver and data will be largely lost. Standards With so many factors to synchronize, a great deal of coordination across the nodes of a network is necessary if communication is to occur at all, let alone accurately or efficiently. A single manufacturer can build all its products to work well together, but

what if some of the best components for your needs are not made by the same company? What good is a television that can pick up only one set of signals if local stations are broadcasting another? Where there are no standards, difficulties arise. Automobiles are an example of nonstandrdized products. A steering wheel from one make or model of car will not fit into another model without modification. A standard provides a model for development that makes it possible for a product to work regardless of the individual manufacturer. Standards are essential in creating and maintaining an open and competitive market for equipment manufacturers and in guaranteeing national and international interoperability of data and telecommunications technology and processes. They provide guidelines to manufacturers, vendors, government agencies, and other service providers to ensure the kind of interconnectivity necessary in today’s marketplace and in international communications.

Data Networks – An overview
The author has vast experience in teaching the subjects of Data Communication, Data Networks, INET, X.25, Frame Relay, ISDN, Broadband ISDN, ATM, SDH mobile communication, GSM etc. in ALTTC, Ghaziabad and BRBRAITT Ghaziabad. The author was invited by Department of P&T, Govt. of China for delivering lectures on these topics for about 2 months. Since he intends to write series of articles on these topics in future, I would be happy if a feedback is given whether the presentation level is okay or too simple or too difficult. Based upon the feedback, articles in future would be modulated by the author. -Editor Data Networks – An overview 1. INTRODUCTION: Now that we have entered into BSNL and going to face competitive environment sooner or later even in Himachal Pradesh, our first task would be to earn more and more revenues from our services. World over, the trend of non-voice services are gaining momentum and Data Networks are becoming more and more popular and have in fact started dominating growth of Internet in last 3-4 years even in India. We in Himachal have many plans of such network in near future like expansion of Internet even to the Block HQtrs, voice on Internet, Mobile GSM, WLL, SDH and other value added services etc. Basic philosophy of these modern networks is very different from that of our conventional telephone networks. Predominantly the non-voice Data Networks work on the principle of packet transportation and information in pieces rather than continuous information as a whole as on circuit networks. Even CCS#7 and the D-Channel of ISDN are primarily packet networks only. I, Therefore, through of making the very basic concepts of these types of networks clear to our friends. I am aiming of writing these articles keeping

in mind the needs of our colleagues in the Department who unfortunately could not get opportunity of training and exposure to these modern technologies I, would, therefore, put forward my arguments in a very simple language so that our friends could understand the basic concepts easily and build up the subject of their own later. Those who know may please therefore bear with me. Any further technical details that may be required by anyone may freely discuss with me at any time. In the first part, I would cover the very basic fundamental concepts of these network including hand shaking done at various levels between the nodes as the information proceeds and also of the aspects of the management, by giving the example of X.25 networks which is considered to be the mother protocol for such type of networks. In the 3rd installment I would prefer to cover the concepts of frame relay and ATM technologies. 2. DATA NETWORKS VS VOICE NETWORKS There exist basic differences between voice communication and data communication, which is predominantly information exchange between the machines. In data communication, therefore, the issue relating to human psyche does not exist. On one side where the voice communication is delay sensitive (recall the irritation of long distance communication vis satellite), machines in Data networks are designed to tolerate these delays. On the other hand whereas the voice communication is not sensitive towards the loss of information (we can still make out the meaning even if some words are lost), data communication is very-very sensitive towards these aspects (Only One thousand rupees would get credited in your account, when you actually transferred one million rupees because of change of just one bit during transmission). In data communication, machines can talk to one another and adjust according to one another’s needs i.e. mismatch in speeds and other various protocol which may also include conversion from one language to another. These protocols are standard protocols and are inbuilt in such networks. To take care of mismatch of speeds of the machines and Data communications being basically bur sty in nature (and also for other reason, which I would discuss later. Data networks are designed for variable bit rate (VBR) whereas the voice networks are predominantly constant bit rate (CBR) networks with fixed basic speed of 64k/bps (Kilo bit per second) (Recall in PCM sampling is done @ of 8 K/sec. Frame of 125 micro sec with 32 T.S. and each time slot with 8 bits). We have been living for decades together with the basic telephone switch operating every 3.9 micro seconds in real time whether there exists or there does not exist any speech information at the input of the switch, (Remember the time switch, where all zeros are switched even there is no speech on information to be switched). I say real time why because voice communication error correction is not carried out and therefore error checking is never done. As against this since the data network are very sensitive for errors. The information to be switched must be first checked for any error (which may have been caused during transmission from the previous node) prior to switching. Therefore in data networks the information at the input of the switch is required to be stored for some time for detection of error and its correction and unless it is ensured fully that the information has been received correct with respect to the previous nose, the information is never pushed further by

the network towards its destination. For this both nodes handshake and information may have to be re transmitted many times between the two nodes unless both nodes are fully satisfied. This error correction takes time and therefore data networks are never Real-Time networks in true sense. In this process of error correction it is therefore just possible that the piece of information arrived at the input of a node at a later stage but received uncorrupted may get switched earlier than the information received earlier but received corrupted and is still waiting for error correction. In voice communication, where the sequence of various information pieces of the various subscribers is never changed w.r.t. time, in Data networks sequence of information pieces w.r.t. time may get changed from node to node. Since voice networks are Real-Time networks with time slops called STM (Synchronous Transfer Mode) networks. Multiplexing of information of many subscribers in STM is done w.r.t. their allotted time (TDM) and are dependent.

STM TS=13 TS=5 TS=13 TS=5

STM TS=13 TS=5

(Position of information w.r.t. time remains same)

ATM

(Position of information pieces may change)

Contrary to this, because data networks are error sensitive, and information is stored for error detection and correction, therefore the position of the information pieces w.r.t time cannot be fixed. In other words the information is switched and transported Asynchronously and information transferred in Asynchronous node (ATM). STM is therefore fore time dependent. What I mean by “time- independent” is that switching at the input of switch. Data information is switched only as and when it arrives and no switching will take place for empty time slots as is done in time switch of voice networks.

B.W. wastage

ATM (No. B.W. wastage) (Statistical Multiplexing)

Multiplexing therefore in Data networks is not Time Division Multiplexing but only statistical multiplexing. It is exactly (in Voice networks) as if gate across a river opens every minute whether there is a car standing or car not standing to cross the river. On the other hand in data communication the gate will open on when a car is waiting to cross. To make my point more clear, if for one hour there is no traffic to cross the river, in data networks; gate will remain closed for one hour and in voice network. the gate will open and close sixty times @ one minute. Logically if there is no traffic, why should the gate open and pass “No Traffic” equivalent to all zeros, which unnecessarily occupy the precious bandwidth. The gate should in fact remain closed, to conserve the resources of switch transmission media and the bandwidth. That is what exactly is done in Data network. Voice networks are bandwidth waste network and with very low efficiency 50% of the bandwidth is virtually lost because when some one is talking he not listening and when he is listening, he just cannot talk. Even when he talking there are many pauses and period of silences in between during which series of zeros are transmitted to fill the gap. Because of this, bandwidth wasted further. On an average 65 to 70% of bandwidth gets lost because of nature of such communications. Since data networks cannot afford to loose to bandwidth of this sort (high cost involved), the data switch and data transmission has to take place only when the actual information is available for transportation. Since in Voice Networks, subscribers are pre-allotted their time for sharing the resources and by no means, some body’s time, if he is not using telephone can be dynamically re-allotted to another person who is waiting. Voice Networks are called ‘Fixed Bandwidth’ Networks. In contrast to this in Data Networks because Multiplexing is done statistically, resulting in saving of lot of bandwidth anybody who wants to hook his machine on the network, can be accommodated. Data Networks are therefore, ‘Bandwidth on Demand”’ Networks. There exists another important difference between the two networks. The size of information pieces in Voice Networks whether it is speech, signaling, control or information for the purpose of administration/management is always fixed (eight bits in 3.9 micro sec.). In Data Networks, however the sizes of information pieces for the actual Data to be transported, for signaling and control information associated with Data etc. will vary. It is the same way as if

the gate in Voice network is designed to open only for the size of a car. In Data networks however gate is designed to open for all the sizes of vehicles, it can be Cycle, Scooter, Maruti, and Fiat or for that matter a TATA Truck or even a large size Railway Container Truck. 3. ISO’S-SEVEN LAYER OSI PROTOCOL Having understood the basic characteristic of Data Communications, let us move ahead with the components of Data Network. Basically the Data Networks constitute three components manufactured/operated by three different agencies. The wide area public network consisting of nodes (data Exchanges) and transmission media are operated by agencies like DOT/VSNL etc., the machines connected on the networks are provided by computer/terminal manufactures and the applications etc. which run on computers and which help in transporting the data information from one machine to another machine depending upon the requirement of users are written by the software experts. Historically there had been very rapid progress in computer hardware and application software, but comparatively long distance public data networks could not keep that pace and could not develop that fast to transport effectively the data information from one place to another. Therefore, smaller Data Networks confined to few kms. Area and within the building like LANs etc. came into existence very rapidly. Interconnecting of these LANs however could not be done for many years. Main reason behind this mismatch was that no international standards were available for connecting one side very fast computer and application software (with fancy and innovative attractive packages for subscribers) and on the other hand long distance public data networks. To bridge this gap during late 70’s International Standard Organization (ISO) came up with very broad and strong specifications which were happily accepted by industries and telecom operators. These set of rules to match computer with the networks so that data could transparently flow from one machine to another machine is known as ISO’s-OSI (Open System Interconnection) seven layer protocols. Once these specifications were accepted, many matching long distance public Data Networks were designed to transport computer informations. The most popular network of 70’s and 80’s was the X.25 network, which I will describe in more detail later. To understand this seven layer OSI protocol, following example should be understood. Suppose a very Senior Chinese Officer, with many staff members including Secretaries, Pas and well organized Central Registry Section for receiving and dispatching the letters, wants to invite an Indian counterpart Officer to his country. What he or his secretary will do to get a piece of paper, type the invitation letter in India and pass it on to the central registry section of his office for dispatch. The in charge of the central registry section will drop the letter box/office of P&T China which in turn will give it to Deptt. Of Post, India at a common meeting point and depending upon the address, the Deptt. Of India will deliver the letter to the central registry section of the office of Indian officer. This central registry section which may also receive hundreds of such letters from various other directors will sort out the mail division-wise and deliver all the letters meant for the officer including the invitation letter to his secretary. It is just possible that this central registry section, by mistake (and often it happens) may have also receive one two

letters not actually meant for this office. The central registry will discard such misspent letters and if possible re-direct them to their destinations. Secretary of the officer on receipt of bunch of letters from central registry will remove the envelops of all the letters including the Chinese invitation and initiate the process of taking actions of his own on some of the letters as per the instructions of Boss personally. In between, however if the secretary finds some difficulties in understanding any of this invitations letter, he would contract the secretary of the Chinese Officer on telephone and giving the reference of the letter, will seek from him the clarifications. After seeking the clarifications, he would like to give to his Boss this invitation letter at the right time of the officer hours (as per standing direction of his boss) along with any other previous correspondence relating to this invitation. Since the letter is in Chinese language the secretary calls the interpreter who will convert the Chinese language into English language and explain the contents to the Boss. This is how a meaningful communication is established between officers of China inviting an officer of India through letter correspondence. In machine-to-machine communication, exactly same analogy is followed. The two offices one in China and other in India are the two machines and the Postal Deptt. Is equivalent to the wide area Data Networks. The piece of paper and writing the words on it is exactly the process of putting the electronic bits property formatted and coded on the physical media of the network. This exercise in Data Networking is done by a layer, which is called PHYSICAL LAYER. Why we put the invitation letter in an envelope is that, its contents remain secret and to ensure that the paper and the information do not get mutilated enrooted adverse weather conditions. Exactly the same way in Data Communication, the actual data is protected by few extra bits, which are purposely put on the both sides of the data as an envelope so that the data as protected and not damaged during transmission. Putting such electronic envelop for protection is done by a layer known as DATALINK LAYER. Writing the address on the envelop is to transport the envelop to the direction of its destination. Exactly in the same fashion electronic address is written along with actual data on the electronic envelope. The data moves from node to node and each node reads the address and pushes forward the data towards to next destination node and data finally reaches the destination machine. This function of routing is done by a layer called NETWORK LAYER. The Data Network has now delivered the electronic invitation letter to the Computer of Indian Officer. Just like the role of Postal Deptt. Is over once the letter is delivered to the customer, exactly the same way once the data is handed over to the destination machine, the role of Data Network is completed. The function of central registry section i.e. sorting out letters coming from many directions, discarding the misspent letters and ensuring that letters meant only for the office have been received, is done by a layer in the computer. This layer is known as TRANSPORT LAYER. Transport layer depending upon the destinations etc. and ensures the correctness. The role of secretary is performed by the next layer known as SESSION LAYER, which initiates the process of handing over the data to the user of the computer. The session layer of one computer will talk to Session Layer of the other computer that has sent the information exactly the same way the secretary of the Indian Boss had talked to the secretary of Chinese Boss for clarifications. Session Layer will then pass on

the date to the layer, which actually presents the information to the user. This layer is known as PRESENTATION LAYER and works exactly like an interpreter who converts the Chinese language to English and English language to Chinese language. The main function of presentation Layer therefore, is to provide services to the next layer the ways it is understand.
End system
Application Layer Presentation Layer Session Layer Transport Layer
Network Layer Data Link Layer Network Layer Data Link Layer Network Layer Data Link Layer

End system Protocol Protocol Protocol Protocol
Application Layer Presentation Layer Session Layer Transport Layer
Network Layer Data Link Layer

Physical Layer

Physical Layer Transmission Media

Physical Layer

Physical Layer Transmission Media

Layered architecture of the OSI Reference Model Like the role various sections of the office of CGM like Planning and Development, Operation & Maintenance, CGM himself, Administration etc. are well defined and are headed by CGM, GM(Dev.), DGM(Opn.), DGM (Admin), exactly the same way the function of a computer for which it is being used are well defined. These functions are stored in the computers as software packages and the usages of these packages and their controlled flow is done in various APPLICATION LAYER. These packages will include programs for handling. File transfer, for handling, for handling Electronic mail (X.400), handling Directory enquiry (X.500) and many more. Appropriate package from the layer will be picked up to handle the invitation letter and is finally given to user at printer. These seven layers i.e. PHYSICAL LAYER, DATA LAYER, NETWORK LAYER, TRANSPORT LAYER, SESSION LAYER, PRESENTATION LAYER and the APPLICATION LAYER are responsible for controlling and ensuring the correctness of the information flow in computer to computer networking. The first three layers i.e. PHYSICAL, DATA and NETWORK layers are known as the low level OSI layers and belong to network operator like DOT, MTNL and VSNL etc. and the remaining four layers i.e. transport, session, presentation AND application LAYER BELONG TO COMPUTER MACHINES AND ARE KNOWN AS UPPER LAYERS. The functions and behavior of these layers are fully well defined and their boundaries are perfectly earmarked. No layer trespassers the boundaries of other layers. Each layer provides service to the upper layer and receives service from the lower layer.

In the Trans side from China to India, the information (invitation letter) will travel, making use of the programs of the Application Layer of Chinese computer, vertically down ward to the Physical Layer passing through all the remaining layers. Similarly the information after traveling on the networks and reaching up to Indian computer will move upwards from Physical Layer to the Application Layer. Each one of the layers of one computer if fell necessary will talk to its under part on the other side for handshaking and for any other requirements, about clarifications etc. This type of communication between layers of same status like session layer to session layer (secretary to secretary discussion as was explained earlier) is called peer-to-peer horizontal communication. For this peer-to- peer communication, some reference is therefore required. Whenever therefore information is vertically moving and is received from the upper layer and handed over to the next layer, each layer will add to the information received, its own particular characteristic and parameters for the purpose of recognition and as reference point. Therefore, by the time the actual i.e. PRESENTATION, SESSION, TRANSPORT, NETWORK and DATA LINK layer would have added the extra bits as over heads. These overheads are known as Headers. Header will, provide details of handshaking control information, address information, information regarding the correctness of data, packet size, packet number etc. On the network therefore, not only the actual data (here the invitation letter) but also lot more in formations as the reference/characteristic informations as Headers of various layers will also travel. Reverse process will takes place at the receiving end computer where the whole information will travel upwards from Physical Layer to Application Layer. Each layer for performing its assigned duty will go on peeling off and removing the corresponding Header information’s, read the content of the header and perform actions as needed.
4. CONCEPTS OF PACKETISATION

Before I make you understand the need of packetisation, I wish, you should clearly make difference between time division multiplexing and statistical multiplexing. In voice communication, keep in mind that most of the time, most of the people do not communicate. On an average, if one subscriber make use of telephone in a day for one hour, he is wasting, the band width of the network for 23 hours because in TDM, pre-allotted time slots of subscribers can not be dynamically re-allotted to other subscribers who are in need of services. Therefore, on a PCM link at any moment with the above scenario speech information would be available in time slot No, 5 and time slot No.13 and all other times slots in between 5 and 13 would be empty wasting a bandwidth to the tune of 64 x 7 = 448 Kbps (time slot No. 5 and 13 have been taken arbitrarily just to explain the point). It is exactly the same as if one car is at the bus stand and the other car is as a distance at the tunnel point and in between the whole of the road is empty, because the people who have been permitted to ply their vehicles on this road are sleeping at that time. In Data networks, since there is no preallotment of time and multiplexing is not time development whosoever is coming, his car will be permitted to ply if there is space available on the road. Efforts would therefore be made in Data Network scenario, that as many cars as possible are cascaded one after another on the road between tunnel point and bus stand to fully utilize the road. Similarly more and more information pieces are accommodated between time slot.5 and slot No.13 in the above example. Therefore, for achieving above goal of putting as many cars as possible on the road more and more cars are diverted to this road from other congested routes.

In voice network, it is our experience that when some body shimla wants to talk to Mumbai, circuits are not available during busy hours, and when circuits are available during nighttime, people just do not prefer to make calls. If you imagine the telecom network of the country from above sitting on an Aero plane you will find that in the network, in many directions, many circuits remain idle as there is not sufficient traffic in those directions and on the other hand circuits towards the popular directions like Delhi, Mumbai, Calcutta, and Hyderabad are always busy with traffic and circuits are always felt short. This Type of scenario as given above cannot be accepted in Data network and bandwidth in Data networks cannot be tolerated to be wasted. Therefore some body though of distributing the data traffic more event on the network as against the scenario of more circuits less circuits towards Mumbai side and less traffic comparatively more circuits towards Bhubneshwar side to make best use of the available bandwidth of the network. People therefore, thought why not to route some part of the call meant for Mumbai vis lesser-used directions Bhubneshwar, Srinagar, Guwahati, Vishakhapattnam and even via Rome if possible. So what is done in Data networks, a message of 30 minutes meant for Mumbai from Delhi is broken into 30 pieces of one minute each (for example) and routed towards Mumbai via many directions which are available at that time from Delhi. Each packet will be an independent information piece carrying the address of Mumbai. Packet may take any free route available may be via Bhubneswar, via Guwahati, via Vishakhapattnam and even in Rome or London and try to reach Mumbai. The disadvantage however would be that the packet coming via Rome or London would reach Mumbai later than the packet via Bhubneswar. Therefore in Mumbai al the 30 packets will reach not in that sequence in which they were broken in pieces. In this case therefore, the responsibility of the network (i.e. collectively responsibility of Physical, Data and Network Layer) would thus be limited only in transporting all the packets from various directions to Mumbai and hand over the packets to called party computer. Delivering the packets in correct sequencings is not the responsibility of Data networks. Once all the 30 packets are handed over by the Network to Transport Layer of the computer, it is the responsibility of computer to sequence them property. In this scene, since no formal circuit as established between Delhi and Mumbai and the thirty packets whereat their liberty to reach Mumbai taking any direction available, this approach of communication is called CONNECTIONLESS. Internet AND ALL l/Ns work on this principle and are connectionless type of networks. Connectionless approach is very simple and very effective for smaller networks. However, on the other hand as the public networks deliver the call from one place to another in full Toto and do not handover the call to the called party computer half cooked to be sequenced by themselves as explained above. What therefore we do in these networks is to establish a virtual/logical circuit between the calling party and and transfer all packets in sequence on the circuit upto the computer of the called subscriber. This circuit is only the logical one, and therefore called virtual circuit and is not a physical circuit or a channel working on time division multiplexing. What exactly is the logical/virtual circuit in data network. I would explain in the following paragraphs. This approach of communication is called CONNECTION ORIENTED X.25 network. Frame relay and the ATM networks are all CONNECTION ORIENTED NETWORKS.

In this approach unlike in connectionless, all 30 packets will not come in the address of Mumbai. Since all packets will follow the same route one after their only the first packet will contain the address of Mumbai computer and will ask the route from Shimla to Mumbai in the network which is available at that time and all remaining 29 packets will follow the first packet. The network in the fashion that I will explain and the route is disconnected only when calling/called party disconnects the call remembers the route. The difference between a circuit in Voice network and a Virtual Channel in the Data network is that, that if a party has established a circuit in Voice network for 20 minutes and he actually talks only for 12 minutes remaining 8 minutes he does other jobs while holding the circuit the bandwidth of the network to the tune of 8 minutes is wasted. In Data network however will virtual connection, bandwidth for 12 minutes will be used for this call and bandwidth of 8 minutes when his person is not sending any Data will be allotted to some other subscriber and is not wasted. Network will however hold the first connection for 20 minutes. 5. CONCEPT FO VIRTUAL VCHANNEL In time division multiplexing since the position of information piece at the output of node is always fixed with respect to time, network knows the movement of information and as has been mentioned many times above the position of a particular packet is a particular call at the output is changing from node to node. Question therefore arises that in time independent statistical multiplexing scenario as in Data Networks. If the position of the packet goes on changeling from node to node how does the network keep track pf information [piece? It is very simple. Suppose a subscriber ‘A’ of Shimla wants to establish a Data call call to subscriber ‘B’ at Mumbai. Computer of ‘A’ will pocketsize the whole Data into pieces and start transmitting them to the node at Shimla. As mentioned above only the first packet will contain the address of computer ‘B’ of Mumbai and depending upon the availability to the channels enroots suppose the first packet takes the route Shimla-Delhi-Gwalior-Bhopal-Nasil-Mumbai-B computer. Supposing if computer ‘A’ assigns any arbitrary number say 21 to the first channels number 21 between computer ‘A’ and Sheila node. This packet when enters to Shimla node is ready by the node for the Delhi and also assigns a new number to the packet say 420. This number is available at that moment with the node and is arbitrary picked up by the node from bunch of many numbers that and at his disposal and stored in memory. Node maintains a table where he is keeping the total record of ‘input number vs output number’ for all the packets arriving from all the directions. Node also keeps updating the table. While out putting the packet towards Delhi route the content of the header will also be changed from 21 to 420 and packet will assumed floating between Shimla and Delhi on the virtual channel 420. At Delhi the node reads again the address and takes an output route to Gwalior. The new number say 840 which is available with the Delhi node at that time is given to the packet and the packet will float between Delhi and Gwalior on virtual channel number 840 Similarly from Gwalior to Bhopal it may float on channel number 1111, between Bhopal and Nasik on virtual channel 2222, between Nasik and Mumbai on virtual channel number 3333 and from Mumbai node to ‘B’ computer on virtual channel number 4444. From the above explanation it is clear that the same packet will assume different numbers from node to node as it proceeds in the

network. Each node knows of the packet at its input, which was given by the previous node and its changed output number given by himself. We therefore, say that virtual circuit is established between subscriber ‘A’ of Shimla and subscriber ‘B’ of Mumbai with a set of numbers 21-420-840-11112222-3333-4444. All 30 packets of this call will float on this virtual channel with these numbers written in their headers. This set of number belong to this call only and will remain with this call, till the call is disconnected and these numbers cannot be given to any other call by the network. Or conversely, all those packets in the network, which bear these numbers in their header, must belong to this call. Once the call is disconnected, these numbers are also wiped cut from the tables of the nodes and these number can now be allotted again to any other call by the network. Building of virtual circuits in data networks is therefore nothing but allotment of numbers to the packets and writing these numbers in the header of the packet for the purpose of locating them in the network. Unlike in Voice networks where time slots are directly function of bandwidth i.e. if time slot are lost bandwidth is lost, in Data networks the chain of numbers allotted to a call is nothing to do with bandwidth allocation. If subscriber ‘A’ transmits data only for 12 minutes out of his call of 20 minutes, bandwidth to this call will be a for allotted 12 minutes only on these numbers and networks can allot bandwidth of 8 minutes to some other subscriber on another set of numbers. Or you may say that these numbers will get wanted for 8 minutes but not the bandwidth. The above set of numbers will however remain operational for 20 minutes for call ‘A’ till it is disconnected. Virtual manual are of two types. Permanent Virtual Channel (PVC) and Switched Virtual Channel (SVC). PVC is just hot line and the number resource in the network are pre-prepared and these number just cannot be allotted to any body else. For residence these numbers, however subscriber has to pay in the above example once the Shimla node reads 21 it must assign the number 480 to the packet at the output if the virtual channel of the above example is a PVC. Similarly the number 840, 1111 etc. are reserved. If the computer ‘A’ transmits first packet on channel 21 network immediately comes to know that subscriber ‘A’ of Shimla wants to get connected to subscriber ‘B’ of Mumbai. Add any of subscriber ‘B’ of Mumbai is not required in PVC. On the hand SVCs are built up as and when required depending upon the availability of the route and the vitality of number resources if at any stage routes are not available because of any reason, SVCs can not be build and call is disconnected.

Data Network an overview Part-II
By S.C. Chandock CGM BRBRAITT

In my last article, the following concepts were developed: 1. Voice Networks are delay sensitive but not sensitive to loss information whereas Data Networks are very sensitive to the loss can tolerate some delay. 2. Information must proceed in the Data Network from node to node error free and for that any error occurred during transmission is corrected. 3. The processing of information is carried out in Data Networks as and when it arrives unlike in Voice Communication where ever ‘no information’ is filled with Dummy 4. Therefore, in Data Networks, Multiplexing is the independent and is done statistically. Bandwidth is therefore lot of bandwidth is 5. The position of information piece on the Data Network is not fixed and vary from node to node. Voice Network works on the principle of STM whereas Data Networks are primarily ATM networks. 6. For evenly distributing the traffic on the whole of network, the call is broken into pieces called “Packets” and packet in many directions and reach destination approach is connection less. 7. The virtual channel in connection oriented mode is the logical circuit between point ‘A’ and ‘B’ which is established the time of origination of a call and is cleared on disconnection of call. In the conventional circuit switching, bandwidth is lost whereas in Packet Networking with local channel, bandwidth during ‘no-use’ is saved. Virtual Channels are of two types ‘PVC’ and ‘SVC’. 8. In connection oriented scenario, the channel is established between calling party and called party and all Data Packets are floated on this virtual channel. The route is determined by the first packet and all other packets follow the same route. 9. However in the connectionless scene, each Packet is an independent information from any route that is available at that particular moment. Internet and LANs work on this principle. Having understood the above basic philosophies, I will now try to take you inside the Network to make you understand how exactly the calls are established how does the Packet look like, how the error is detected and corrected and how does the Network manages its affairs. There exist many Data Networks like INTERNET, INET, ISDN, ATM, Frame Relay, X-25, SDH, CCS#7, SMDS and many more which have been designed keeping in view their specific uses. However, PLEASE KEEP IN MIND THAT ALL PACKET NETWORKS MENTIONED HERE, WORK ON THE BASIC FUNDAMENTAL PRINCIPLES AS HAS BEEN SUMMARISED ABOVE AND INMY LAST ARTICLE AND IN THE FOLLOWING PARAGRAPHS. HOWEVER, TO MAKE MY POINTS MORE CLEAR, I WILL GIVE THE EXAMPLE OF THE NETWORK BASED UPON X-25 SPECIFICATIONS. When I say ‘X.25’ specifications, I mean to say the rules defined in the ‘ITU-T (CCITT)’ documents of series. Protocol is nothing but set of rules under which the signal flows are controlled and exchange the information they have.

For any Network, there has to be a finite number of logical channels between any two Nodes or between the terminal and the connecting Node. In X-25 the maximum no. of virtual channels on a link is 40. (The example of VCC 4444 from the Node to called party terminal of Mumbai of the last article was therefore wrong). You can visualize this concept as if there exist a road between to Nodes with 4096 lanes on which various types of traffic (Packets) are lying. Therefore, we require 12 nos. of bits (2 to the power 12 is 4096) to define a particular virtual channel for the packets of a particular call. Since virtual channel no. is assigned by the Nodes randomly, there is possibility of collision of packets coming from opposite directions on a particular channel for the outgoing and for the incoming calls (if you remember for any call, the virtual channel no. is given by previous Node and the next Node is nor aware of it. If by chance the same channel no. is given by the next nos. for an incoming call, coming from other direction. Packets on the same channel will ) Therefore, a discipline of allotting the virtual channel nos. for the outgoing and or incoming calls has to maintain. Even there due to any reasons, if a collision does take place, incoming call is destroyed. The scheme allocation of channels is shown in the diagram. Boundaries (Low & High) can be assigned by the Admn depending upon needs. If you remember, in my last article, I had mentioned that the lower three lawyers i.e. Physical Data and Network Layers are the responsibility of Data Network whereas upper Layer i.e. Transport, Presentation, Session and Application Layers are the responsibility of terminal. When I am talking of ‘Packets’ assume that I am referring to Network Layer, similarly word ‘Frame’ is associated with Data Layer and ‘Bit’ is associated with Physical Layer. The ‘Frame’ envelops the ‘Packet’ and the ‘Packet’ resides inside a ‘Frame’. Frame is therefore, made up of Packet plus some ‘Overheads’. How does a Packet and Frame look like, I will explain later. Any terminal in technical language is known as Data Terminal Equipment (DTE) and to the equipment where this DTE is connected on the communication network side is called DCE (Data Communication Equipment). Therefore a modem is PSTN is a DCE and a PC is a DTE. There exist a standard protocol of hand shaking between DTE and DCE depending upon the type of network. In X-25, a PAD (Packet Assembler and Dissembler) is used whose output is compatible with X-25 network. At the input of the PAD, is the character mode asynchronous information coming from the out put of the PCs. PAD works as a multiplexer with many PCs at the input and a single X-25 packet mode stream at the output. The character mode information generated by PCs is converted by PAD into packet formal information, which is accepted by the X-25 network. PAD therefore, removes the ‘Start’ and ‘Stop’ bits of the characters while outputting into the network and adds these bits in the opposite direction before information is given to PC. There are about 22 parameters associated with a PAD, which can be controlled and modified by the PCs connected to PAD depending upon their needs. Details of these parameters are given in the specification of X-3. Just to have and idea of these parameters, two of them have been shown in the diagram. Any character mode PC when interacting with X-25 Network must therefore, take help of a PAD. Such Hardware/Software may be terminals or may be situated

within a node. When the functions of PAD are within a terminal, the terminal can be directly connected to the Network and such type of connection is called X-25 connection. X-25 connection works at a higher speed and on this link theoretically a subscriber can generate 4096 calls simultaneously on all logical channels available on the. In contrast to this, X-28 is a single connection with only one virtual channel at the disposal. What I said above in these paragraphs is shown in the diagram. Please remember that in Packet Network, all information exchanged between the terminals via the Network is in the form of pieces and not continuous as in Circuit Network. For example for establishing a call, calling terminal at Shimla will generate a piece of information called ‘Call Request Packet’ which will travel up to the called terminal at Mumbai and if call is accepted by the called terminal, a ‘Call Accept Packet’ will generated by the Mumbai terminal which will again travel all across the Network from Mumbai to Shimla will give to the calling terminal, this packet as ‘Call Connect Packet’) called terminal could have not accepted the call due to any reasons and in that event a ‘Call Reject Packet’ is sent to calling party). After receipt of Call Connect Packet, the call between Shimla and Mumbai is established (by the Network as it keeps record of all Packets) on the virtual channel on the route taken by the ‘Call Request Packet’. Data Packets will then be exchanged between the two terminals. After exchange of all the Data a ‘Call Clear Packet’ is generated by either party and the call is released and the virtual channel (chain of numbers) held till now is made free by the Network. In between, for the purpose of management and administration and error correction and also for smooth flow of information, additional Packets are generated by the Terminal/Networks. Broadly therefore, there exist four category of Packets i.e. packets for Establishing/Cleaning the Call, Data Packets actually carrying the Data to be exchanged between terminals, packets required for Management/Administration and packets required for controlled smooth flow of information. Each Packet type is identified by one byte called ‘PTI’ i.e. ‘Packet Type Identified’. The above-explained points are shown in the diagram. Reset, restart and diagnostic interrupt is the Management/Administration Packets. The RR, RNR, REJ type of Packet of Packet is the flow control Packets. Please also remember that in Data Network, working on connection oriented mode as in (X-25, Frame Relay, ATM etc.), it is the responsibility of the Network deliver the Packets in sequence to the called terminal. Therefore, Packet moving from Node to Node has to be properly accounted and ensured that no Packet is lost or duplicated. Network has also to ensure that all Packets must be delivered error free. How this is done, I will explain in the following paragraphs. However, even after taken all precautions that a Network can do, it is just possible (probability is too small) that some Packets may be lost during transmission, duplicated or may be delivered to the terminals will talk to each other and ask for re-transmission of the lost Packets/ eroded Packet. Re-transmission may take several times if required till Packets are fully recovered and received un-corrupted and both terminals are satisfied. The surest way that the Packet is not lost or not duplicated during transmission from Node to Node or from terminals to Node is that each

Packet when received at the next Node must be acknowledged. Therefore, to keep track of the Packets, each packet is numbered and accounted. Counting can be from 0 to 7 in Moudulo 8 or can be from 0 to 127 for bigger Network in Moudulo 127. The problem in acknowledging on 1:1 basis (i.e. each Packet is acknowledged by the received Node.) is that, that we are adding unwanted 100% overheads on the Network by way of acknowledgements pulling load on the Network and thereby, reducing its efficiency. Therefore to reduce these overheads, a concept of ‘WINDOW’ has been adopted. When I say the window of live, I mean to say that sender can transmit five Packets continuously without waiting for any acknowledgement from the receiver. The sender must however wait for the result of all the five Packets transmitted and till the result is known, the sender is not permitted to transmit the 6th Packet. The maximum size of the window, the sender is not permitted to transmit the 6th Packet. The maximum size of the window is 7. Generally both Nodes inform each other in a piggyback manner simultaneously the status of receipts of Packets. The information is sent in the form of (P(R), P(S) where-P(R) means-I have already received R number of Packets of Packets from you error-free and P(S) means-This is my 6th Packet that I am sending to you. Both Nodes should go on either acknowledging the Packets continuously within window size or after the expiry of the window, must acknowledge with a Packet known as RR (Receive Ready). Therefore RR six means that five Packets have been received error-free and receiver is waiting for the 6th packet. Only on receipt of RR six the end Node will transmit the 6th Packet. If a Packet is received corrupted, the receive Node will intimate to the send side with a packet known as REJ (Reject) Packet. REJ three, therefore, means that two packets have been received correctly but 3rd Packet is received corrupted and received Node wants all the Packets starting from Packet no. three onwards to be sent again. On the other hand, if the receive Node wants only a particular Packet (and not all the Packets as in the above case), he will send a Packet called SREJ (Select Reject). SREJ2 therefore means that the send side should re-transmit only the Packet No.2 and need not send all the packets beyond 2 as was in the case of REJ. If the receive side at any point of time is not ready due to temporary suspend the transmission of the Packets. RNR nine (Receive not Ready) therefore means that the receive side has send side has already transmitted few Packets before the receipt of RNR, he is supposed to re-transmit all such Packets. The above explanation in the form of RR, REJ, RNR etc. is explained in the diagram. The code of RR, REJ, SPEJ, RNR is given in the PTI byte. The size of the Packet has great influence on the through-put and the of the Network to the size of the packet is small, say 16 bytes-more nos. of Packets be required to transmit a message. That means more processing in the switches more handshaking in the form of acknowledgements and therefore comparatively more delay in reaching destination. However, the advantages is that because of smaller size, the probability of error occurring during transmission is less and error if at all takes place can be corrected easily. On the other hand if the size of Data Packet is large, say 1024 bytes, though less nos. of Packets would be required and less handshaking done but if error is detected in some of the Packets during transmission, it would take more time in re-transmission and handshaking for the recovery of the Packets and therefore more delays. Small and big Packets both therefore have

advantages as well as disadvantages. The compromise trade off is therefore done and invariably in X-25 Network the Packet size is generally 128 bytes. Though the Data Networks are very robust and efficient and as the information moves from Node to Node, corrected at every Node and as each Packet is accounted, in 99.99% of cases the message once handed over to network is correctly given to the called party terminal. However, provision exist that for every sensitive Data (like money transaction data of bank), the calling party may ask for acknowledgement of receipt of Data right from the called party terminal. In all other cases, by default, the acknowledgements are given by the Network itself. The bit ‘D’ in the Head of the Packet defines the type of acknowledgement designed. There also exists another bit called ‘M’ bit (M stands for more) which indicates the last Packet of the message. M bit will therefore be 0 for all the 29 Data packets of a message (if the message is broken into 30 pieces, example of which was given in the last article) but for the last 30th Packet, the M bit will be set to one. Network at this stage will recognize that Data exchange is completed now and Network will then prepare itself to receive Call Clear packets from either of the terminal. One of the biggest advantages of Data Network is that two de-similar terminals at both ends with different prosperities can interact with each other. The parameters of terminal like Speed, Size of the Packet, Size of the Window, and Called Party to Pay, Closed Used Group etc. are negotiable on each call basis. These facilities are included in the ‘Call Request Packet’ and the called party can negotiate these parameters and suggest his own parameters at which he wants to interact. These facilities are included in the ‘Call Accept Packet’. The call will then be established by the network on mutually agreed parameters. Each Packet, therefore, will have a head consisting of three bytes which will give the qualifications of the Packet-like the type of acknowledge desired, the of counting of Data Packets whether moudulo 7 or moudulo 127, identification of the Manuel number from Node to Node which turn will identify the call to which the belong, the type of the Packet-whether it is Data Packet carrying actual Data or Packet carrying Call Establishment/Call details of Flow Control or Packet with Management/Administration details. After identifying the type of Packet, the Node will process the Packet accordingly. For Data Packets, PTI will also contain the PS and PR. The three-byte head of a Packet is shown below. Also the Call request packet and Data Packet carrying actual data is shown in the annexure. LCN- 12 bits logical (virtual) channel which will go on changing from Node to Node to the values of 21, 420, 840, 1111, 2222, 3333 and 4444 (of the example of last article) MTerminal will set M=1 for the last Packet and ‘0’ for all other packets.

To ensure that the content of the Packet is not corrupted during transmission, each Packet is put inside a frame. Frame which is the concept of level two i.e. Data Link Layer ensures safe transmission of packet and in case error is detected at the receive Node, error correction is carried out. High Level Data Link Control (HDLC) protocol is generally used at level two for this purpose. General structure of HDLC frame is given below: Structure of HDLC F lag is always 01111110. Address field is one byte long and therefore can address 256 terminals/Nodes in point to point multi point configuration. Since in X-25 Network, the information moves from ‘A’ to ‘B’ on point to point basis, the address field in X-25 refers only two directions either from ‘A’ to ‘B’ or from ‘B’ to ‘A’. Control field is against one byte long and contains exactly the same as was explained in pti field in case of Packets i.e. it serves for counting of the frame (FR, FS) in a piggy-back manner and also controls the flow of the frames. Please remember counting and flow controlling of the Packets at Layer 3 is totally independent from the counting and floor controlling of the Packets at level 2. Even the window sizes at Data Layer may be different from that of Network Layer. What happens in fact is when a frame reaches a Node, the Node removes all other bits of the Frame and takes out the Packet. Process the Packet as per what is defined in the PTI, changes the LCN (first Node will change LCN from 21 to 420 of last article) puts back the packet in the new Frame envelope after writing address byte (‘A’ to ‘B’ to ‘A’), writing control field (whether Frame contains the Data Packet, or Frame is only the supervising Frame for flow controlling) generates FCS bits and pushes the Frame out to the next Node. The whole of the Data i.e. address + Control + Packet information is divided by a suitable divisor and the remainder is added to the contents of Address + Control + Packet and the result in put into FCS. This exercise is done at the send end. Since the remainder is added to the content of Address + Control + Packet therefore, if the FCS is divided by the same divider at the receive end, the remainder should be 0. This confirms transmission between send and receive has taken place without any error. However confirms transmission between send and receive has taken place without any error. However, instead of 0, if some other value is obtained as remainder at the receive end, that means some error has occurred which requires to be corrected. How the FCS is calculated, is explained in the Annexure. HDLC is a very powerful and comprehensive level two Data Link Protocol. A simpler version of this protocol (address is point to point and not to multi point) is called LAP-B used in X-25. A similar protocol is LAP-D used on the D channel of ISDN. In the connectionless scenario (like in LCN and INTERNET etc.) as each Packet is an independent complete information piece with source and destination address attached (and no logical channels to float as in connection oriented) each Packet fights for accessing the single common media available. In GSM also the mobile units contest for grabbing the common media, as many mobile units may try to originate the call exactly at the same time. Therefore, in such scenario level two Data Link Control protocol is broken into two separate Layers i.e. LLC (Link Layer Control) and MAC (media Access Control). LLC

performs the general functions of error detection and correction whereas MAC arbitrates and regulates in a particular fashion the accessing of medium. The Physical Layer i.e. level one Layer is responsible for transport of bits from one device to the other on physical connection. It converts the bits into electrical signals having characteristic suitable for transmission over the physical medium. It also supports the relay function if required. The Physical Layer also provide synchronization signal necessary for transmission. Modulation and encoding is also carried out by this Layer. Hoverer no error correction is done at level one. The 25 PIN Rs.232 standard falls in this category and converts the computer output to the physical media and provides synchronization. As the details of this standard are available in books, I would not go further into this. X-25 Networks was developed during early 70’s when the reliable media like Optical Fibre was not available and therefore X-25 ran on conventional co-axial or M/W with BER arounf (10 to the Power-3). Therefore, there was necessity that information moved in controlled environment from Node to Node, error corrected and every Packet accounted at each Node. However, with OF coming with BER as low as (10 to the power-9), the necessity of processing every Packet for error correction at each Node was not required because in all probability (almost 100%) each information piece reaches the next Node exactly the same as was transmitted by the previous Node. Therefore, in modem Networks like Frame Relay and ATM etc. error correction and accounting of the Packets are not done at all. If a Packet is found received corrupted at any stage, it is simply discarded by the Network to be recovered by the end terminals at Transport Layers. Keeping this in mind, Frame Relay as the backbone networks have been developed where no switching takes place, no error correction is done and no accounting of packets is carried out. Each Node will have a unique address and they are interconnected in mesh topography with one another with PVCs (since switching in the true sense is not done, Frame relay Network do not have SVCs) hot line type of connections. This increase the speed of transmission may folds. Whereas most of the X-25 Networks work at 256 KB/sec., Frame relay can go up to the speed of 10 Mb/s. Since conventional switching is not done, Frame relay Networks do not have the Network Layer of level 3 to work upon and Frames containing the Data Packets run at high speeds on PVCs with least interruptions from Nodes. In ATM technology, the positive aspects of both circuit switching as well as that of Packet switching is taken advantages of. Unlike in Frame relay, where the transmission is purely on PVC, in ATM both PVCs as well as SVCs are used. Packets are switched as in X-25, but switched very fast. However, error correction is not performed. Errored Packets are simply dropped in the Network. X-25 and Frame relay are designed only for Data Communication. ATM on the other hand can take all the components i.e. Voice, Data/Text and Video. The Packets size in ATM is fixed and is comparatively small of the order of 53 bytes. 5 bytes are for the header and remaining 48 bytes Payload. ATM at present work at the speed of 155Mb/s. However, the ultimate speed of the ATM is designed for 625 Mb/s. ATM is the technologies of tomorrow. In Data Networks, error is detected at Nodes and the Packets are simply dropped in FR and ATM and in X-25, error is corrected. Information pieces

are therefore required to be stored for 0 shot period in the memory. Stay in the memory will depend upon the rate at which the Packet arrive and the rate at which they are processed. Therefore, queuing theory play an important role in such Networks. As per this theory, for an optimum designed rate of arrival and disposal (rate of arrival can be slightly more than rate of processing), the length of queue linearly increased in the initial stages w.r.t arrivals. However, after a particular cut off point the length of queue increases much faster for the same rate of arrivals. And a time would come when the length of queue (packets in the memory) becomes unmanageable, memory capacity falls short and Packets start overflowing. At this stage the Network start collapsing under load. Since in X-25, a very strong flow control mechanism exists, particularly with ATM flow control does not exist there is always possibility of Network collapsing if subscriber’s input of data is not checked beyond a threshed level. Different mechanism is therefore adopted in FR and ATM with which collapse of Network is avoided and if Networks start collapsing immediately recovery procedure are followed. In FR and ATM, therefore, to regulate traffic ‘POLICING’ is resorted to. Example of PAD Parameters. Parameter 2. Echo. The PAD transmits back the characters received from the DTE C for display on the DET screen. OCTU PTI FIELD Bits 87654321 00001011 00001111 00010011 00010111 PACKET TYPES From DTE to DCE From DCE to DTE CALL REQUEST CALL ACCEPTED CLEAR REQUEST DTE CLEAR CONFIRMATION DTE DATA DTE INTERRUPT DTE INTERRUPT CONFIRMATION DTE R R DTE RNR DTE REJ RESET REQUEST DTE RESET CINFIRMATION RESTART REQUEST

INCOMING CALL CALL CONNECTED CLEAR INDICATION DCE CLEAR CONFIRMATION xxxMxxx0 DCE DATA 00100011 DCE INTERRUPT 00100111 DCE INTERRUPT CONFIRMATION xxx00001 DCE R R xxx00101 DCE RNR xxx01001 DOES NOT EXIST 00011011 RESET INDICATION 00011111 DCE RESET CONFIRMATION 11111011 RESTART INDICATION 11111111 DTE RESTART DCE RESTART CONFIRMATION CONFIRMATION 11110001 1 DIAGNOSTIC Table 5-2. PTI Field Encoding Across Packet Types.

The maximum length of the address in X-25 Network is 14. First three digits are the country code, next digit is the Network number within the country (maximum 9) and remaining 10 digits are the terminal address. The facilities are usually CUG, Fast Select, Packet Size, Window Size, Called Party to Pay and the speed of the terminals etc. Example of generating FCS Generate FCS code for the data word 110101010 using the divisor 10101. Solution Data Word Divisor 110101010 10101 Quotient Dividend

111000111 10101 1101010100000 10101 11111 10101 10100 11101 11000 10101 11010 10101 11110 10101 1011 1101010100000 1011 FCS = Code Word 1101010101011 XAdd Zeors

Remainder

= (Length c Data word – Length of Divisor) = (9-5) = 4 Divisor 10101 is called Generating Polynomial. G (X) and is represented. G (x) = 1 x4 + o x3 + 1 x2 + o x1 + 1x0 G (x) = x4 + x2 + 1

OVERVIEW OF INTERNET & INTRANET SERVICES

OVERVIEW OF INTERNET SERVICES 1.0 INTERNET: A WINDOW TO GLOBAL INFORMATION

Internet is an exciting arena where you can find information about almost every topic. On the Internet you have books, encyclopedias, and any type of reference material at your fingertips. In addition you have expert opinions on various topics and can communicate with people. Network of Networks: Networking occurs when two or more individuals, business, or resources joined together to form a union i.e more powerful than each of its separate components. In the computer world, networking or being on a network means the joining together of many computers. A network enables each user to access vast amounts of data that is located on other computers in the network. Computer networks are available in a variety of formats, but there are two primary types: local area networks and wide area networks. A local area network, or LAN, usually occurs with multiple computers that are within close proximity of each other. A LAN uses cables to connect its users. A wide area network, or WAN, uses data send over telephone lines – as opposed to cables – to connect its members. Internet is built through collaborative efforts of people worldwide. Internet is an Electronic link to the world of information and entertainment. It is a system of connected computers allows exchange of information to other computers having connection. The Internet uses a variety of telephone lines that are similar to the ones you currently use in your own home. These networks are joined to form a much larger region of computer networks. When you connect your computer to the Internet, you are actually connecting to a network that connects to other networks through a network backbone. The network backbone is a large network with many connections to other networks. Between these backbones, interconnections called gateways make it possible for a computer on one network to exchange messages and data with another computer connected to another network. 1.1 HISTORY OF INTERNET The development of Internet was necessitated by need to have spread command control in the whole of USA in the event of a nuclear attack. Packet switching was the first technology evolved for this purpose. Thus started ARPANET linking 4 nodes and TCP/IP emerged in 1973. Evolution of domain name server i.e. DNS occurred in 1983 making it easier for people to remember names instead of IP numbers.

1.2 A.

COMPONENTS OF INTERNET World Wide Web

Also referred to as Web or WWW. Using a web browser, a software programme necessary to navigate the World Wide Web, you can experience all that it has to offer. Areas on the web you can “travel to “ are called web sites there are a variety of sites on the web. Each site has its own address, and many sites let you jump to additional sites. Some sites contain only text, but some also contain pictures, sounds, movies or any combination of these. Some sites do nothing but help you find additional information. These sites contain software called a search engine. A search engine behaves like a reference librarian who helps you find

information sources. Search Engines allow you to type one or more keywords so that you can search for the information you are looking for. Search engines use Indexing software called ‘Robots’ and ‘Spiders’. These act as agents that constantly crawl the web to search for new and updated web pages. They visit each and every web site on the Internet. These robots record the entire text of the Home page and the sub pages of the web site, which may get registered even without your knowledge. However, you can submit your URL to a search engine to increase the speed of this process, instead of waiting for the robot to locate your URL address. A genuine search engine will require only the URL of your web site, the rest of the information will be trapped by its spiders. WWW is a system on Internet that supports hypertext to access different Internet protocols on a single interface. Hypertext is a document containing words that connect to other documents. These words are called hyperlinks. B. E-Mail (Electronic Mail) This component allows computer users locally and worldwide to exchange messages. As soon as we get connected to Net generally we click on the E-Mail. It is a system for delivering messages from one person to another. Assuming 2 accounts/user there are ½ billion net users and one billion E-mail accounts. In India more than 80% of dial up accounts are primarily used for sending and receiving emails. It is the most dominant application of the Internet. Constant development has taken place in this direction and now formatted HTML based mail, a facility for file attachments, voice mail, Video mail and WAP (Wireless Application Protocol) enabled etc. are empowering the E-mail. Thus, E-mail is used for conducting meetings and discussions and to carry out work assignments. In order to send Email you need:(i) (ii) (iii) (iv) Ability to compose a message. A way of transferring it to a storage facility. A way for the recipient of your message to get the mail from the storage facility. A way to read it.

E-Mail Environment: This works in a client – server environment. You need software that runs under windows (The client). You need an E-mail Post Office programme running on a host computer (the server). The client programme talks to the server program over the network to transfer mail back and forth. E-mail server is a series of programs running on the operating system (example UNIX) of the host computer. It sends and receives mail from other Electronic Post Offices. Internet E-Mail Addresses: Addresses consist of: user name, an @ sign and the name of an Internet host site. User names are unique to their E–mail Server. E-mail addresses cannot have any spaces in them. Make use of E-mail to fullest:

(i)

When working with web based E-mail applications you can type longer messages in advance on a text editor while you are offline. Attach these files separately. Thus save time Online. Using POP3 click open your inbox, tick of all the messages you want to read, let the pages be down loaded, close the browser and read them while you are offline. Alternately, right click each message and open it in separate window. Send attachments:- If messages to be sent are too large, it can be compressed through any zip program. While sending file attachments be sure that the recipient has the right application on his computer. Secure your mail: It is possible to make sure that confidential mail is delivered to the right person. Mails can be secured with security software like PGP (Pretty Good Privacy) or Sigaba. Some security websites offer to transact mail securely for you ex. www.sigaba.com.

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Multiple Mail Accounts: ISPs do not provide more than a couple of E-mail accounts depending on the plan subscribed. Web based E-mail accounts, however, allow the use of number of accounts by using different user IDs. Spam Mails: Unwanted mail is called Spam mails. You should protect your inbox with unwanted mails. Some special E-mail filter programs like Spam buster or Spam hater are designed to help you get rid of junk E-mails. Hybrid Mail: If you want your E-mail to be delivered in print by traditional postal service or courier this mail facility can be used..

Telnet: This is Internet’s remote login service. It allows the user to log onto computers with the help of on the Internet and use online databases, library catalogs, chat services, and more. Telnet stands for Telecommunications Network. This is a protocol that provides a way for users (client) to connect to multi user computers (servers) on the Internet, whether in the next building or across the other side of the world. Telnet over TCP/IP On the Internet the ability to connect with another machine is made possible by the Transmission Control Protocol (TCP), which enables two machines to transmit data back and forth in a manner coherent to the operating systems of each device and the Internet protocol (IP), which provides a unique 32 bit address for each machine connected to the network. Telecommunication applications built over these capabilities provides the local terminal with the means to emulate a terminal compatible with the remote computer. Connection Establishment:

The TELNET TCP connection is established between the users port U and the server port L. Telnet protocol gives you the ability to connect to a machine, by giving commands and instructions interactively to that machine, thus creating an interactive connection. In Telnet, the local system become transparent to the user, who gets the feeling that he is connected directly to the remote computer. The commands typed by the user are transmitted directly to the remote machine and the response from the remote machine is displayed on the users monitor screen. An interactive connection is also known as remote login. Thus in order to remoter login the users computer must have the ability to : • • • Establish a connection to another machine Emulate a terminal compatible with the remote machine. Regulate the flow of data from the users terminal to remote machine, and vice-a-versa.

Syntex of Telnet Commands : telnet < address of remote host > Many Remote Hosts require you to have an account to login (user-Id and pass word). FTP (File transfer protocol) It transfers files between computers, FTP sites contain books, articles, software etc. Data on most systems is represented in units called files. Data transfer between systems usually involves transfer of files. FTP is the standard for transferring files over the Internet across any platform. As a protocol, FTP is a series of rules that describes for computers the proper way to Establish a connection with a remote computer. For connected users it provides to view directories and files on the remoter computer. Get files and send to the remote computer. It can down load graphic images, Sound files, Drivers, copies of Historical Documents, Demonstrations, etc. This is a way through which quick delivery of products is ensured. FTP enables the user to interactively access files and directories on remote hosts and perform directory operations such as: • Listing of files in the remote directory. • Renaming and deleting the files ( If you have permission) • Transferring files from Remote Host to Local Host. • Transferring files from Local Host to Remote Host. FTP Components: There are two components: • FTP client FTP client initiates the control connection and data connection to the FTP server • FTP Server.

FTP server is a computer on the Internet that offers files for down loading. In order to have access to FTP the client must have FTP account. Types of FTP Account • Private Account It allows you to FTP files to and from your personal account with your ISP. The ISP provides the login ID and password for a private account and only you can FTP files back and forth. Public Account. With this account anyone can login and transfer files. These also require login ID and password. These are known as anonymous accounts. Using the browser as an FTP client the address will be : ftp://ftp.microsoft.com

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USENET This is the Service that allows an individual to join one or more discussion groups This is a forum where people exchange ideas or it can thought of as a network of large bulletin boards at the heart of the UseNet system lie various news groups or bulletin boards where messages are posted. The strength of this system lies in the fact that there is a news group for virtually every conceivable topic under the sun. New groups are classified by hierarchical names of two or more parts separated by dots. Hierarchies of news groups: Rec. :Recreational Soc. :Social Comp. :Related to Computer Sci. :Related to Science News :Related to net news Talk. :Long arguments fairly political Misc. :Misc Alt. :Alternate Biz. :Business. News groups can either be open that is anybody can post any article to the group and it is immediately displayed for everybody in the group to read. In a closed news groups it is to be sent to the moderator of the group who decides its suitability for the particular group and then may decide to post it or rejected. INTERNET TELEPHONY It is a cheaper way of globally reaching out to people. There are at present four different ways of making a call over the Internet: • • Computer to Computer Computer to Telephone • • Using an internet appliance Telephone to telephone.

For making computer to computer call you need an Internet telephony program example Microsoft net meeting. These offer features such as video, voice mail, call waiting, call holding, text check, file transfer and data sharing.

For making computer to telephone calls you would have to avail of the facility provided by an Internet Telephony Service Provider example Net2 phone. This means you would have to pay a small fee. The call goes to your PC over the Internet and then to the Internet Telephone Gateway at the calls destination which connects the Internet to the Telephone System. To make calls through the Internet appliance method you would need a separate piece of hardware. Connect this device between your telephone and an answering machine and configure them with your ISP information. When you call the other partner the Internet appliances learn each other’s IP address and then connect to each other over the Internet. For making telephone to telephone calls. This is same as normal PSTN call. The differences the call is routed over the Internet for most or part of the way thus reducing long distance charges. Call originates from a telephone and is routed to an Internet telephone Gateway that connects the telephone network to the Internet. The call then travels across the Internet to its destination where it is routed to another Internet Telephony Gateway that connects the Internet back to the telephone network. Then the call is directed to the called telephone.

OVERVIEW OF INTRANET SERVICES
2.0 - Introduction Intranet is a smaller private version of Internet. It uses Internet protocols to create enterprise-wide network which may consists of interconnected LANs. It may or may not include connection to Internet. Intranet is an internal information system based on Internet technology and web protocols for implementation within a corporate organization. This implementation is performed in such a way as to transparently deliver the immense informational resources of an organization to each individual’s desktop with minimal cost, time and effort. Every organization can constantly refer to the central messages and develop their own supporting sites accordingly. Use the Web as an information, communications, and projectmanagement tool across the organization. 2.1 Need of Intranet

In an Intranet environment is used to communicate over two or more networks across different locations. Users having multi-locations with multi-networks. Users having single locations with multi-networks. Users having single locations with single networks. 2.2 Advantages of Intranet • • • • From a technology point of view, an Intranet is simply beautiful. because : It is scaleable. It is Interchangeable. It is platform independent. It is Hardware independent.

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It is vendor independent.

2.4

Applications of Intranet Intranet may be used in various ways in the corporate and any organisation. Following are the few applications of Intranet: •

Publishing Corporate documents Corporate documents such as newsletters, annual reports, maps, company facilities, price lists, product information literature can be easily published and propagated across an organization. Intranet technology facilitates efficient, timely and accurate communication across the entire corporate organization and cuts down on the cost of publishing these information on paper now and then. Access into searchable directories Intranet provides rapid access to corporate phone books and the like . By using this technology, information can be made more widely available. Excellent Mailing Facilities With Intranet mail products mailing attachment of documents, sound, vision and other multimedia is facilitated. With the evolution of this web technology one-tomany communication has become more effective. Proper Sharing of Information Using Intranet technology, applications such as Bulletin Board Services can help every individual in an organization to put forth his views on various topics and discuss it with others in the organization. Developing Groupware Applications The flow of documents can be automated by incorporating intranet in an organization. Thus the overall efficiency of an organization increases, as less manual and paper involvement will be required. Typical examples are sanctioning of expense reports/travel reports, Conference room booking, etc.

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Organizational and personnel changes can be immediately communicated on the intranet. Mergers , new ventures, new projects ,product releases can be immediately communicated. Instant availability of the latest organizational information. Conference type online interaction. Employees can view benefits programs, Company policy and procedures online. Distribution of software and manuals centrally. Reduce paper work with the organization. In manufacturing units all products details and company standards can be put centrally on the Intranet.

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6.0 - A Typical Intranet setup

A typical Intranet implementation involves a high end machine called a server which can be accessed by individual PCs commonly referred to as clients, through the network. The Intranet site setup can be quite inexpensive, especially if users are already connected by a LAN. Most popular Intranet web servers can run on any platforms widely found in most organizations. Basic requirements for setting up an intranet site are: • Software Server and clients software Server OS can be Windows NT, Unix, OS/2 .Web Server s/w should be installed Client OS can be Windows 3.x, Windows 95,Windows NT workstation, OS/2 .Web Browser software
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Hardware

Server and Client PCs, Networking elements like Router, Switches and Hub etc.

OVERVIEW OF VOICE OVER INTERNET PROTOCOL
Contents
• • • • • • Introduction Inter working between PSTN and Internet PINT Reference Model IP Telephony Architecture VOIP Network architecture Phone to Phone Internet Telephony Objectives • 1.0 To understand the concepts of Voice Over Internet

OVERVIEW OF VOIP
Introduction During the beginning era of Telecommunication manual switchboards were used for putting through the telephone calls. Each telephone handset was connected to a pair of copper wires that ran to a switchboard. To make a phone call, the operator plugged one end of that pair of copper wires into a socket that connected to another pair of wires to another telephone handset. The two pairs of copper wires for the two telephones stayed plugged together for the duration of the phone call, forming a electrical circuit dedicated to that call. When the call gets over, the operator unplugged the wires and disconnects the circuit for human voice. Telephone networks were extended worldwide. With the rapid changes in the technology operators were quickly replaced by mechanical switches and later by electronic switches. The Telephone switching system used for the set up of voice calls, which connects the calls on the principle of circuit switching. Scenario changes with the evolution of technologies, handsets got dials and eventually keypads. Analog switching system changes to digital ones. Multiple phone calls are multiplexed over a single wire, microwave towers replaced copper wires, and wires and towers are giving way to fiber-optic cables. On the great trunk lines of the phone system, calls are transmitted as packets of bits. But when you make a phone call today you still have a dedicated circuit connecting your telephone to the telephone you are calling. The circuit-switched telephone network establishes a circuit for the entire duration of a phone call. The copper wires connected to each phone are dedicated to that call. These switching systems also used for the connectivity of facsimile and other kind of data through Circuit switching. With the evolution of technology packet switching technique developed for the transmission of data. The Internet, on the other hand, is a packet-switched network. Unlike the telephone system, the Internet does not have dedicated circuits or paths. Instead, any device attached to the network can hand off packets of data to the

network. The network consists of specialized computers called routers and communication links between the routers. The routers pass each packet from one router to another until it reaches the destination. If there is a large amount of data, the sender will break it up into multiple packets. The receiver then must reassemble the packets to recreate the original message. Everything like e-mail, Web pages, MP3 files, pictures, and movies sent over the Internet as packets. All the machines and devices on the Internet use the Internet Protocol, a standard method to communicate with each other. That standard method defines the format of data packets and the form of the numeric addresses that identify sources and destinations. It defines how to reassemble packets into a complete message. The Internet Protocol is the key to the fantastic growth of the Internet. It is simple and it is standard. Any device that follows the Internet Protocol can send and receive messages over the Internet. The human voice, like text, pictures or any other information, can be digitized and turned into bits, and these bits can be sent over the Internet using the Internet Protocol. IP telephony uses the Internet to send audio between two or more computer users in real time, so the users can converse. Vocal Tec introduced the first IP telephony software product in early 1995. Running a multimedia PC, the Vocal Tec Internet lets users speak into their microphone and listen via their speakers. Within a year of its birth, IP telephony technology had caught the world's attention. The technology has improved to a point where conversations are easily possible. And it continues to get better. In March of 1996, Vocal Tec announced it was working with Dialogic Corporation to produce the first IP telephony gateway. The original Internet telephone products based on multimedia PCs are tremendous - offering the ability to combine voice and data on one network. They also offer low-cost long distance "telephone" service. It is popularized as Voice Over Internet Protocol (VOIP). VoIP is a new low-quality and inexpensive service and does not match the existing phone system in terms of voice quality, reliability and security. Like other disruptive innovations, it has the potential to improve faster than its established competitor and eventually displace the existing higher-quality telephone service in the mainstream market. The circuit-switched telephone network establishes a circuit for the entire duration of a phone call. The copper wires connected to each phone are dedicated to the call; no other voice calls or faxes call reach those phones. The electronic switches maintain the connection as an actual physical connection or as an entry in a database. A conversation between two people contains lots of silence, times when neither person is talking, but the telephone system transmits this silence, just as it transmits the voice signal when either person is talking. Keeping a circuit open uses some measurable portion of the very expensive telephone network, and the telephone company charges for the use of that capital equipment. VoIP, in contrast, does not keep a circuit open, it simply transmits the voice signal as packets whenever one person is talking. This technique is much more efficient and makes better use of the communication network. It does not dedicate lines or switches or table entries for extended time periods, but uses existing Internet connections. It uses the packet switching facilities of the Internet to send only the data needed to recreate the human voice at the destination. This lets VoIP calls be made at a much lower cost than conventional phone calls. VOIP technology has been developed to condense telephone voice data and transfer it along the signaling data lines now present for Internet and Intranet communication. These advances aspire to considerably reduce the amount of

hardware and physical space needed to support available technology, the supervision necessary to maintain both networks and the cost of the call. An IP system can seamlessly integrate all forms of office communication. As a result, the complete telephony service architecture will be deployed over the Internet and this is known as IP telephony. Functions made available by IP technology include: The convergence of voice mail and e-mail into one multimedia system, accessible from remote phone, office phone or PC. Multimedia Video conferencing delivered to any end point in an office. Remote location access to all of the physical and data resources of an office. 1.1 Inter working between PSTN and Internet

There are two tendencies toward bringing together the advantages of the Internet and the Telecommunication world. • Use the PSTN as the medium transport infrastructure, but the devolves part of the request mechanisms to be run on the Internet. The tendency is exemplified by the work going on within the PSTN/Internet Inter working (PINT) group at the Internet Task Force (IETF). • PSTN as the principal infrastructure f or the transport of user data and promotes the Internet as a serious alternative for conveying this data. 1.2 PINT Reference Model A working group, called PSTN/Internet Inter working (PINT) group, was created at the IETF in 1997 with the objective of opening up the IN architecture to user requests issued from the Internet In this scenario, the user makes the call request from the Internet and later communicates through the PSTN. The reference model laid out by the PINT group is depicted in Fig. 1.

INTERNET

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PINT Reference module consists of following: Service Node (SN) Service Management System (SMS) Web Server PSTN/Internet Gateway Service control Point (SCP) Central Office Mobile Switching Center (SMC) ISDN and other Value added services Requests received from the user on the Web are sent to a Web server that processes and forwards them to the gateway between the Internet and the PSTN. This gateway then communicates with legacy IN components such as the Service Node (SN), the SSP (both the mobile switching center and the classical telephone central office), the SCP, and the Service Management System (SMS). An SN can be regarded as a hybrid component that provides, among other activities, the combined functionality of an SSP and an SCP. The SMS is a system that embeds the functions necessary for the management of the IN infrastructure. There is another trend, toward the interweaving of the IN and the Internet, where nearly the entire IN control system runs on the Internet. In this trend, the role of the SCP is reduced to finding a Web server that contains the logic and data to be used for the services, unlike in the conventional IN system where the SCP controls the whole service software. By running much of the IN control system on the Internet, service providers can enable the customers to create their services as easily as create their Web pages. Interoperability, among IN system providers, can also be achieved. 1.3 IP Telephony Architecture

The promoters of telephony take different approach over the Internet. In this scheme, all the control as well as user information transfer is achieved using the Internet. The main ideas behind the Architecture are continuous data flow through the Internet and Inter operability among legacy telephone equipments and computers. Following four groups of possible interactions are identified – • • • • Gateway to Gateway signaling and Information transport Interaction between the SCP and the Gateway Data Base. PC to network signaling Management Information flow.

Technical Implementation Frame work – Two relevant proposals for VOIP/IP Telephony are – • ITU-T Recommendation H.323 • The session Initiation protocol (SIP) being worked at the IETF H.323 primary goal is to enable Audio-Visual interactions among terminals situated in various environments.

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CONFERENCE BRIDGE WEBSERVER

MIB : management Information Base GDB: Gateway Database • Gateway Trunk signalling & Transport • Gateway Database & SCP links • PC(HOST) to network siganalling • SNMP MIB & Network Management. 1.4 H.323 Components • • • Terminal Gateway Gate Keeper

Terminals These are LAN client endpoints that provide real-time, two-way communications. All H.323 terminals are required to support H.245, H.225, Q.931, Registration Admission Status (RAS) and real-time transport (RTP) protocols. An H.323 terminal can communicate with either another H.323 terminal, a H.323 gateway . H.323 Gateway. It is in charge of translating call signaling, control messages and multiplexing techniques across the network technologies involved in the call.

H.323 Gate Keeper It performs network administration Band width renegotiation and address translation. H.323 terminals must get permission from gate keeper before they can place or accept a call. The gate keeper empower the N/W administrator with much control over the Network and provides users with the possibility to define alias addresses that are independent of the Network addresses. The gatekeeper is also helpful component that can be used to implement many supplementary services. 1.5 Session Initiation Protocol ( SIP)

SIP is a protocol that enables the invitation of users to participate in Multi media sessions. It also enables user mobility by forwarding invitation to the user’s current location. 1.6 VOIP Network Architecture Most of the Voice Over Internet Protocol services which offered for long distance Voice and FAX services over low cost IP Network. With minimal upgrade, the existing IP network can carry voice traffic over packet N/W. Installing a VOIP gateway with a voice/FAX card enables the IP N/W to directly interface with PSTN, making the process of placing calls over the IP N/W transparent to the users. Advantages:• Reduced cost • Increased revenue from the existing points of presence. • Ability to expand customer • Service bundling opportunity across voice and data services. • Positioned for value added applications and services. • Lower cost IP infrastructure leveraging voice compression & silence suppression.
Caller Node Node Caller IP NETWORK PSTN Node Caller Node Caller PSTN Node Caller

PSTN PSTN VOIP NETWORK WITH DEDICATED IP TRUNK CONNECTIVITY

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1.7

Typical voice call handling in a VoIP application.

It is useful to understand what happens at an application level when a call is placed using VoIP. The diagram below describes the general flow of a two-party voice call using VoIP.

1.8

Table - Typical VoIP Call Handling

1.9

Issues related to Voice Over Internet Protocol • • • • • • • Technical Efficiency Switching Costs Charging for VOIP Security Inter Operate ability Absence of IP Access at Subs. Level End to end speech performance. Transmission errors & packet loss Echo Cancellation Effect of Bandwidth limitation in IP N/W

1.10

Network Convergence and VoIP

Delay A very important design consideration in implementing voice communications networks is minimizing one-way, end-to-end delay. Voice traffic is real-time traffic and if there is too long a delay in voice packet delivery, speech will be unrecognizable. An acceptable delay is less than 200 milliseconds. Delay is inherent in voice networking and is caused by a number of different factors. There are basically two kinds of delay inherent in today's telephony networks: Propagation delay – caused by the characteristics of the speed of light traveling via a fiber-optic-based or copper-based medium of the underlying network. Handling delay (also called serialization delay) – caused by the devices that handle voice information and have a significant impact on voice quality in a packet network. This delay includes the time it takes to generate a voice packet. DSPs may take 5ms to 20ms to generate a frame and usually one or more frames are placed in a voice packet. Another component of this delay is the time taken to move the packet to the output queue. Some devices expedite this process by determining packet destination and getting the packet to the output queue quickly. The actual delay at the output queue, in terms of time spent in the queue before being serviced, is yet another component of this handling delay and is normally around 10ms. A CODEC-induced delay is considered a handling delay. The table below shows the delay introduced by different CODECs. Table CODEC-Induced Delays CODEC G.711 PCM G.729 CS-ACELP G.729a CS-ACELP Bit Rate (Kbps) 64 8 8 Compression Delay (ms) 5 5 15

Serialization delay Serialization delay is the amount of time a router takes to place a packet on a wire for transmission. Fragmentation helps to eliminate serialization delay, but fragmentation, such as FRF.12, doesn't help without a queuing mechanism in place. For example, if a 1000-byte packet enters a router's queue and is fragmented into ten 100-byte packets, without a queuing mechanism in place, a router will still send all 1000-bytes before it starts to send another packet. Conversely, if there is a queuing mechanism in place, but no fragmentation, voice traffic can still fail. If a router receives a 1000-byte packet in its queue and begins sending this packet in an instant before it receives a voice packet, the voice packet will have to wait until all 1000 bytes are sent across the wire, before entering the queue, because once a router starts sending a packet, it will continue to do so until the full packet is processed. Therefore, it is essential that there is a method for a router to break large data packets into smaller ones, and a queuing strategy in place to help voice packets jump to the front of a queue ahead of data packets for transmission. End-to-End delay End-to-end delay depends on the end-to-end signal paths/data paths, the CODEC, and the payload size of the packets. Jitter Jitter is variation in the delay of arrivals of voice packets at the receiver. This causes a discontinuity of the voice stream. It is usually compensated for, by using a play-out buffer for playing out the voice smoothly. Play-out control can be exercised both in adaptive or non-adaptive play-out delay mode. Echo Cancellation Echo is hearing your own voice in the telephone receiver while you are talking. When timed properly, echo is reassuring to the speaker. If the echo exceeds approximately 25ms, it can be distracting and cause breaks in the conversation. In a traditional telephony network, echo is normally caused by a mismatch in impedance from the four-wire network switch conversion to the two-wire local loop and is controlled by echo cancellers. In voice over packet-based networks or VoIP, echo cancellers are built into the low bit-rate CODECs and are operated on each DSP. Echo cancellers are limited by design by the total amount of time they will wait for the reflected speech to be received, which is known as an echo trail. The echo trail is normally 32ms. Reliability Traditional data communication strives to provide reliable end-to-end communication between two peers. They use checksum and sequence numbering for error control and some form of negative acknowledgement with a packet retransmission handshake for error recovery. The negative acknowledgement with subsequent retransmission handshake adds more than a round trip delay to transmission. For timecritical data, the retransmitted message/packet might therefore be entirely useless. Thus, VoIP networks should leave the proper error control and error recovery scheme to higher communication layers. They can thus provide the level of reliability required, taking into account the impact of the delay characteristics. Therefore, UDP is the transport level protocol of choice for voice and like communications. Reliability is built into higher layers. Audio data is delay-sensitive and requires the transmitted voice packets to reach the destination with minimum delay and minimum delay jitter. Although TCP/IP provides reliable connection, it is at the cost of packet delay or higher network latency. On the other hand, UDP is faster compared to TCP.

However, as packet sequencing and some degree of reliability are required over UDP/IP, RTP over UDP/IP is usually used for voice and video communication. Interoperability In a public network environment, in order for products from different vendors to interoperate with each other, they need to conform to standards. These standards are being devised by the ITU-T and the IETF. H.323 from ITU-T is by far the more popular standard. However, SIP/MGCP standards from IETF are rapidly gaining more acceptance as relatively light weight and easily scalable protocols. Security On the Internet, since anybody can capture packets meant for someone else, security of voice communication becomes an important issue. Some measure of security can be provided by using encryption and tunneling. Usually, the common tunneling protocol used is Layer 2 Tunneling protocol, and the common encryption mechanism used is Secure Sockets Layer (SSL). Integration with PSTN and ISDN IP Telephony needs to co-exist with traditional PSTN for still some more time. It means that both PSTN and IP telephony networks should appear as a single network to users. This is achieved through the use of gateways between the Internet on the one hand and PSTN or ISDN on the other. Scalability As succeeding VoIP products strive to provide Telco-grade voice quality over IP as is true for PSTN, but at a progressively lower cost, there is a potential for high growth rates in VoIP systems. In such a scenario, it is essential that these systems be flexible enough to grow into large user markets. 1.11 Applications of VOIP • • • • • • • Internet Voice Telephony Intranet and Enterprise N/W voice Telephony Internet FAX Service Internet Video Conferencing Multimedia Internet Collaboration Internet Call Centre Public Switched Telephone Network Interconnection

VOIP on the Intranet VOIP is possible over Intranet through VPN utility. Each of the offices at different places will have a PBX system through a gateway they get connected to the corporate LAN. The LAN is connected to Internet through the router, but the network becomes an Intranet by usage of VPN. Thus calls from different PBX, are possible through VOIP in this configuration. New Services • • • VOIP for free phone facility of IN Virtual Second Call Mobile Services.

1.12

Commercial issue related to Internet Telephony Service

• By pass the PSTN Companies using the VOIP can circumvent the long distance carrier N/W. As calls are made, a hybrid PSTN/IP gateway combination residing on a corporate Intranet will convert voice signals into packets. These packets will then make their way across the corporate Intranet rather than the PSTN. The packets go to the receiving gateway and are translated and the call is received. This is an impressive cost saving on voice calls & Faxes to the users at the expense of PSTN operators. • Integration of Cable TV, Internet and Telephone A unit is placed at the subscriber’s end which has to wires linked to the telephone and computer, where both voice and data are multiplexed. For the home subscriber, this unit provides a connectivity of 144 Kbps. When the telephone handset is lifted, the internet connectivity is not interrupted as voice requires 64 Kbps and 80 Kbps is still left for internet use. This is more than adequate for the home user who rarely gets even 33.6 Kbps connectivity. In most cases, he uses 14.4 Kbps or 9.6 Kbps. By Passing of International Accounting The use of VOIP enables real time communications to occur between IP network connected PCs and Telephones on the PSTN. Thus VOIP by passes the PSTN and International circuits and thus avoids the International Accounting Rate System which may be substantially higher as compared to voice over IP calls.

•

Asynchronous Transfer Mode Asynchronous Transfer Mode
1. Introduction Telecommunications now demand an integration of various services, namely, voice, data and video. Asynchronous Transfer Mode (ATM) is conceived as a means to realize the basic goal of this Broad Band Integrated Services Digital Network (BISDN). The B-ISDN’s fundamental goal is the integrated provision of various types of services with a great number of disparate characteristics and ATM is a means of accommodating this requirement of versatility. ATM is emerging technology promising highly increased bandwidth greater flexibility and manageability. 2. Packet and Synchronous Transfer Mode In a switching network transfer mode is the organization of information prior to its transmission. The public networks conventionally employ packet transfer mode and synchronous transfer mode. In packet transfer mode, data is transported in varying sized formatted packets to its destination within network. It is used for data communication in local area network (LANs) and Metropolitan area networks (MANs). Synchronous transfer mode is most common in public Switched Telephone Network (PSTN) because of its dedicated circuit oriented topology. It is used for transmission of voice and images due to its capability of providing constant bandwidth. Packet transfer mode guarantees flexible use of available bandwidth and has proved to be ideal for data communication, whereas synchronous transfer mode supports use of voice and images. With increase in the number of network users, the need was felt for a single network capable of supporting these two types of services.

3. The need for ATM:In future the bulk of information handled by public and private networks will be in a digital domain. As observed from table-1, voice will shrink to only small percentage of total traffic offered in broadband communication network. ATM has ability to increase the amount of bandwidth without changing the information structure. Table - 1 Traffic offered by various services Service Transmission speed Requirement Voice Low speed LAN Real Time Video High speed LAN

Public Switched Telephone 64 kb/s Network (PSTN) Ethernet Animation Graphics Fibre Distributed Interface (FDDI) Data 10Mb/s 10 Mb/s 100 Mb/s

High Definition Television 150 Mb/s (HDTV) (SDH Based) 4.

High Quality High Resolution TV.

Asynchronous Transfer Mode Communication ATM is a packet mode transfer technique with a special format that employs asynchronous time division multiplexing (ATDM). In the B-ISDN, service information is transferred by way of a continuous flow of packets of a fixed size, and these packet are called ATM cells (analogous to cell in conventional biology, the basic unit of life). Accordingly, service information is first partitioned down to a fixed size and subsequently mapped into an ATM cell and this cell is asynchronous time division multiplexed with other ATM Cells to from the basic units of BISDN transmission. ATM is a type of statistical multiplexing technique that time division multiplexing mutually asynchronous ATM cells coming from several different channels. When ATM is employed the number of ATM cells measures the capacity of a service channel. Consequently the amount of transmitted information is reflected in the corresponding number of ATM cells and the bursitiness of the service information is indicated by the degree of ATM cell crowding. Here, the transmission capacity is assigned at the user’s request at the time of call set up, and this scheme endows a versatile transfer capacity on all the services. ATM is a connection-oriented method that transfers service information through the establishment of virtual channels (Vcs). A connection identifier is assigned whenever a VC is established and when the connection is released, the identifier is removed. The order of ATM cells inside a VC is maintained by an ATM layer function. Signalling information for setting up a call is delivered via dedicated ATM cells. ATM allows the integration of various BISDN services possessing different characteristics. The broadband and narrowband services can co-exist within the same communications network by using the ATM cells of the same format, differing only in the number of cells that each type requires constant bit rate (CBR) services are made up of ATM cells with a uniform distribution, variable bit rate (VBR) services which are widely distributed, are made-up of the same ATM cells. Also the delay problem associated with real time services is solved through the use of Vcs thus making their provision a possibility. The ATM communications technique can be said to be on integration of the existing circuit mode digital communications technique with the packet mode communications technique. ATM communication system uses ATM cells as its basic unit of transmission it has a close connection with packet mode communication. But there is difference in that the packet mode was developed to support non-real time VBR data signals, where as ATM can manage real time CBR signals as well. Also packet mode communication is designed for use in regional LANs, whereas ATM is to be used for public networks, thus differences arise in terms of address assignment, access and flow control, switching and transmission. On the other hand, the circuit mode has a fundamental difference from ATM is that in the former, the information signal is transmitted in continuous bit streams by allocating a separate channel for this purpose, whereas in the latter, segmented service information is fitted into ATM cells and transmitted through a VC.

For a systematic and flexible information transfer, ATM prescribes a threelayer protocol reference model. The associated layers include the physical layer, ATM layer and ATM adaptation layer (AAL). The AAL performs the function of mapping service signals into ATM cells payload space and the ATM layer executes the ATM cell header related functions for the transport delivery of the ATM payload space. The physical layers function is to transfer ATM cells by converting them into transmission bit streams. The layered architecture for ATM communication shown in Fig. 1. A comparison can be made among the circuit mode, packet mode and ATM as listed in Table-2. As is evident from the table-2, the ATM can be stated in one phrase as, “a packet mode that can provide real time multimedia services by adopting fixed size packets (i.e., cell) and ATDM, together with a simplified header function (i.e., without flow control and error control and connection oriented virtual circuit”.

AAL ATM Physical
layer End System Intermediate Node

AAL ATM Physical
layer End System

ATM Physical
layer

Fig. 1 Layered architecture for ATM Communication Table – 2 Comparison of Communication Modes Circuit Real time CBR Circuit (physical connection) TDM (STM) Continuous bit stream (no delay) Circuit Switching Public Network ATM Mode Real/non real time CBR/VBBR Virtual Circuit (C.O.) ATDM (ATM) Continuous cell stream (cell processing delay) ATM Switching Public/local Network Packet Mode Non real time VBR Virtual circuit/data (CO/CL) Statistical packet multiplexing Intermittent packet stream (large delay) Packet Switching Local Network

Service type Connection type Multiplexing Transmission Switching Network

Other ♦ Protocol Structure ♦ Packet type ♦ Flow Control ♦ Error Control

-

3 layer architecture Fixed length Not included Not included

7 layer architecture Variable length Included Included

(a) Asynchronous Time-Division Multiplexing The TDM, which is widely used for multiplexing existing plesiochronous digital tributaries, is essentially synchronous multiplexing technique as far as the system clock is concerned. This is because the TDM signals are constructed through the repletion of multiplexed frame created using the multiplexer system clock. This results in the appearance of low speed signals at fixed locations inside the frame as depicted in (fig.2 (a). In other words, low speed signals always exist at location that are synchronized with the system clock.
A B C T D M 1 2 3

(a)

Fig. 2
A B C

Comparison of (a) TDM and (b) ATDM
A T D M 1

2

3

(b)

(a) TDM and (b) ATDM Exist at locations that are synchronized with the system clock. ATDM is a type of multiplexing technique that stores each of the incoming low speed signals inside a buffer, then retrieves and inserts the stored signals one by one into a multiplexing slot according to priority scheduling principle. The simplest example of the priority scheduling principle would be first-in/first-out (FIFO) and here the input signals are ATM cells in the case of ATM communications system. Therefore, as shown in Fig. 2(b), the low speed input signals do not occupy locations inside the ATDM signals is a well-regulated manner, and thus behave asynchronously compared to their TDM equivalent. ATDM is superior to TDM in that it has higher channel utilization factor. TDM assigns an exclusive channel to each of the incoming service signals; thus even when a given channel is in a vacant state containing no effective information. It is not possible to pass other service information through it. But since there is no exclusive channel allocation in ATDM, a blank channel can be taken by any incoming signal, resulting in a higher channel utilization factor. Such channel utilization relationships are illustrated in fig.3. In the Fig.3., the length in the vertical direction denotes the channel capacity of the multiplexed signals, while the horizontal length corresponds to the time duration. Also, the parts in slanted lines

or those that are darkened indicate the presence of effective information corresponding to the size of an ATM cell.
A B C (a)

A+B+C

(b) Fig. 3. Comparison of channel use (a) TDM (dedicated) & ATDM (shared) 1.5 ATM cello structure:The ATM cell acts as the basic unit of information transfer in the ATM communication

(b) ATM Cell Structure Octet #
1 2 3 4 5 1 2 3 4 5 6 7 8

6 7

8

53

Header

User information space

5 Bytes (ATM Cell) Bit #

48 Bytes 53 Bytes Bit #
OCTET # VIP

1

2 3
GFC

4

5

6 7
VIP

8

1

2 3

4
VIP

5

6 7

8

VCI VCI

VCI VCI HEC PT CLP VCI

PT HEC

CLP

(c) (b) Fig. 4. ATM cell structure (a) cell structure, (b) Header structure at UNI and © Header structure at NNI

In case of TDM, since the multiplexed signals is no more than a combination of several independent channels, it can be seen that any vacant space in each channel is maintained as it is. But in ATDM the multiplexed signal consists of just a single channel; hence, any information vacancy can be collected and used for providing a new service, consequently increasing the channel utilization factor. Fig.4. (a) The ATM cell is composed of 53 bytes. The first 5 bytes are for the cell header field and the remaining 48 bytes from the user information field. The cell header field is divided into generic flow control (GFC), Virtual path identifier (VPI), Virtual channel Identifier (VCI), pay load type (PT), cell loss priority (CLP) and header error control (HEC) fields. The associated bit sizes differ at the NNI and the UNI. The bit sizes for the two interfaces are as shown in TAVLES-3 and fig.4(b) & 4(c) The main function of GFC header is the physical access control, it can also be used for reduction of cell jitter for constant bit rate (CBR) services, fair capacity allocation for variable bit are (VBR) services, and traffic control for VBR flows. Such a function requires the capability to control any UNI structure, weather it be a ring, a star, a bus configuration or any combination of them. Table - 3
Function GFC VIP VCI PT CLP HEC Bit Allocation UNI 4 8 16 3 1 8 NNI 0 12 16 3 1 8

(c)

The role of the VIP/VCI field is to indicate VC or VP identification numbers in order to distinguish cells belonging to the same connection. Virtual path identifier (VPI) is intended to be used by switches for routing ATM cells. These bits will be updated each time the cell is switched by virtual path switch or cross connect. Generic flow control (GFC) is used to identify congestion between various terminals of access. Virtual channel identifier (VCI) identifies individual channels used within a cell. This permits one virtual path connection for data and another for voice within the same cell. Paryload type (PT) indicates type of data carried by cell. Cell loss priority (CLP) determines priority of cell. In case of overload it may be necessary to discard cells, CLP bits are used to ensure that important cells are not discarded. Finally header error control (HEC) is used to detect and correct bit errors in cell header. Being the most vital function eight bits are reserved for this. Routing of ATM Cells When an ATM cell arrives at ATM switch, values of VPI and VCI are read from the header. These values are compared with dual port memory of this switch (containing VPI/VCI translation tables and outgoing port is determined. VPI/VCI slots are updated with these new values, and the cell is routed to

next switch as shown in fig.5. Next cell is similarly routed with the information that is placed in the same location of cells.

NNI TT TT

B-ISDN

NNI TT TT SWITCH

SWITCH ATM CELL A UNI

SWITCH

SWITCH

UNI

UNI - USER NETWORK INTERFACE NNI - NETWORK NODE INTERFACE TT - TRANSMISSION TABLE USER EQUIPMENT BISDN - BROADBAND INTEGNATED SERVICES SIGNAL NETWORK USER EQUIPMENT

Fig. 5 Signaling architecture for ATM In a typical case when ISDN is carrying HDTV signals at a rate of 150Mb/s. inserting time for ATM cells (53bytes) is only 2.8 msec. (53 x 8/150), which is very small. This routing of ATM cells is extremely fast, which makes ATM network ideal to support real-time traffic. Several classes of services are defined by ITU-T depending on the user bit rate (constant/variable) and the type of data (stream/packet), are placed over it. These services may involve data, voice, image or a combination of these services (multimedia) because of inherent nature of ATM (low delay and high bandwidth) it supports multimedia services offered.

FRAME RELAY

RELAY
Learning Objectives: Understand what is Frame Relay Comparison of the performance of Frame Relay vs. X.25 Understand the frame structure Understand Frame Relay Multimedia Applications Introduction Frame Relay is a packet switching technology that relies on low error rates, digital transmission links and high performance processors. It was intended to be an intermediate solution for the rapid increase in the demand of high bandwidth communication (e.g. LANs) and was originally conceived of as a protocol for use over ISDN interfaces. Frame Relay technology was designed with the following features in mind: 1. Low latency and higher throughput - To achieve this, a simple link layer protocol is used. 2. Bandwidth on demand - It is highly desirable that bandwidth should be assigned to the user based on the actual demands. The bandwidth can be allocated to users at call setup. However, to make it more effective, a user can renegotiate the requested bandwidth whenever traffic bursts are to be transferred with certain peak rates throughout the call. 3. Dynamic sharing of bandwidth - Increased sharing of resources would yield a better utilization of the network bandwidth. The bursty nature of data traffic could be exploited by allowing some users to consume the bandwidth during other users’ idle periods. 4. Backbone network - A backbone network is needed to maximize user connectivity, to accommodate a variety of end system technologies, and to be less vendor dependent. Applications in Use Applications which are particularly suited to use the Frame Relay protocol are applications that: 1. Require the consolidated transport of several protocols. 2. LAN-to-LAN interconnections and other applications that generate bursty traffic. 3. Support Large Host computers by providing a cost effective multiplexed communications interface, e.g. SNA transport. Frame Relay vs. X.25 Frame relay is a streamlined packet transfer method of X.25. It is a switching and statistical multiplexing technology without the error control of the X.25, therefore being much faster. While X.25 is only implemented at speeds below 64 Kbps, frame relay is implemented up to T1 (24 times faster) and some carriers may implement it at T3 rates (672 times faster) - therefore it’s sometime called fast packet. X.25 was created with the intention to operate up to the 3rd level of the OSI model. Frame Relay only operates at the first two layers of the model. Frame Relay is basically “dumb” and relies upon customer equipment. Frame Relay typically operates over WAN facilities that offer more reliable connection services. This means that frame relay has significantly less

processing to do at each node, which improves throughput by order of magnitude. More Comparison points of X.25 and Frame Relay: X25 FR Multiplexing of virtual circuits Yes Port Sharing Sensitive to Protocols Efficient with Bursty traffic High volume of Data Transmission Speed Delay Yes Yes Yes No High Yes Yes No Yes Yes Low

Slow/Med Med/Fast

Yes No Error correction Frame Relay Virtual Circuits: Frame Relay provides connection-oriented data link layer communication. This service is implemented by using a Frame Relay virtual circuit, which is a logical connection created between two data terminal equipment (DTE) devices across a Frame Relay packet-switched network (PSN). A number of virtual circuits can be multiplexed into a single physical circuit for transmission across the network. A virtual circuit can pass through any number of intermediate DCE devices (switches) located within the Frame Relay PSN. Frame Relay virtual circuits fall into two categories: • • switched virtual circuits (SVCs) permanent virtual circuits (PVCs).

A commn. session across an SVC consists of four operational states: Call Setup The virtual circuit between two Frame Relay DTE devices is established Data Transfer Data is transmitted between the DTE devices over the virtual circuit. Idle The connection between DTE devices is still active, but no data is transferred. If an SVC remains in an idle state for a defined period of time, the call can be terminated.
Call Termination The virtual circuit between DTE devices is terminated. After the virtual circuit is terminated, the DTE devices must establish a new SVC if there is additional data to be exchanged. It is expected that SVCs will be established, maintained, and terminated using the same signaling protocols used in ISDN.

PVCs always operate in one of the following two operational states: Data Transfer Data is transmitted between the DTE devices over the virtual circuit. Idle: The connection between DTE devices is active, but no data is transferred. Unlike SVCs, PVCs will not be terminated under any circumstances due to being in an idle state. DTE devices can begin transferring data whenever they are ready because the circuit is permanently established. Frame Relay - The Protocol Frame Relay is a packet switching protocol designed for high quality and high-speed transmission line. It offers cost saving implementations, compared to other data transfer methods. Frame Relay adds relay and routing functions to the data link layer (layer 2 of the OSI reference model). Some of the functions associated with packet transport, such as error correction, flow control, etc., are still formed, but on an end-to-end basis by the end-user devices, instead of by the network. How does Frame Relay work? As described in the Protocol Architecture Diagram below, Application A initiates the communication process by sending a request for session establishment to the transport layer via the presentation and session layers. The transport layer forwards call control information through the ISDN via the D channel using Q.931 procedures. The signaling message is routed through the network and is used to define the virtual path and calls parameters that will be used during the data transfer stage. Once the call is established, data is transferred through the network between applications A and B on a hop-by-hop basis by using the DLCI in the frame header and routing information at each node as determined during call setup. One of the characteristics of Frame Relay is that it minimizes the amount of processing performed on each frame by the network and allows for very fast transfer of information.

The frame structure
Frame Relay’s frame structure is essentially identical to that defined for LapD. The frame relay format can be distinguished from Lap-D by its absence of a control field.

Frames are constructed by encapsulating layer 2 messages (excluding the CRC and flags), with a two-byte header, a CRC, and a flag delimiter Each frame relay PDU consists of the following fields:

Flag Field : The flag is used to perform high-level data link synchronization, which indicates the beginning and end of the frame with the unique pattern 01111110. To ensure that the 01111110 patterns does not appear somewhere inside the frame, bit stuffing and destuffing procedures are used. Address Field: The address field - 2 octets by default 1 2 3 4 5 6 7 C/R FECN BECN DE 8 EA 0 EA 1 DLCI high order DLCI low order

The address field can vary from 2 to 4 octets in size. One of the reasons why this field is variable is due to the fact that there is the possibility that 1024 (1022 if LMI (1023) is excluded) DLCI’s may not be enough. DLCI (Data Link Circuit Identifier) This function of the address field allows multiple connections to be carried over a single channel (multiplexed) and enables the network to route each frame on a hop-by-hop basis along a virtual path defined either at call setup or subscription time. It is used to identify the virtual connection so that the receiving end knows which connection this frame belongs to. It should be pointed out that this DLCI has only local significance. Several virtual connections can be multiplexed over the same physical channel. The Command/Response bit (C/R) This is used by higher end applications to manage end-to-end connections. It is not used at OSI level 2 (Frame Relay).

EA bits This allows the address to be extended in size from 10 bits to 17 or 24 bits respectively. Thus the address field has built in scalability. Forward Explicit Congestion Notification (FECN) This is activated when the frame is switched onto a link where the Frame Relay interface has become congested. Thus the receiving device knows that the PVC is congested. Backward explicit Congestion Notification (BECN) This is activated to tell the switch to engage in congestion avoidance where traffic is in the opposite direction of the received frame. i.e. user-transmitted frames will encounter congestion. Discard Eligibility (DE) This is set to indicate that a certain frame should be dropped in preference to other frames without the bit set when and if the link becomes congested. Information/Data Field The information field contains actual data from applications that operate at higher levels of the OSI model. It can also be used for call controlling. The maximum number of data bytes that may be put in a frame is a system parameter. The actual maximum frame length may be negotiated at call set-up time. The standard specifies that the maximum information field size (to be supported by any network) is at least 262 octets. Since end-to-end protocols typically operate on the basis of larger information units, it is recommended that the network support the maximum value of at least 1600 octets, to avoid the need for segmentation and reassembling by end users. Frame Check Sequence (FCS) It is necessary to implement error detection at each switching node in order to avoid wasting bandwidth due to the transmission of error frames. The error detection mechanism used in frame relay is based on the Cyclic Redundancy Check (CRC). There is no more end-to-end packet level error control; only the node-to-node frame level error control is kept. No error recovery through retransmission is performed and frames received with errors are simply discarded

Link Management Interface -LMI
An optional extension to the frame relay interface standards, which ensures link management functions. This extension allows “keep-alive” messages, configuration information, and congestion status to be exchanged across the UNI between the access device and the network device. The LMI message begins with a DLCI of 1023 or a DLCI of 0 (according to the LMI version used). This DLCI together with an unnumbered information frame type and protocol discriminator of 00001001 identifies it as an LMI frame. The frame includes a message type and some information elements (IEs), which carry the data itself

Error Checking & Handling
Frame Relay does not implement error correction. However, it does implement an error-checking mechanism known as the cyclic redundancy check (CRC). Typically Frame Relay is implemented on reliable network media and error correction may be left to higher-layer protocols running on top of Frame Relay. That way, data integrity is not sacrificed. Frame relay is being used increasingly for transport of traditional; highly bursty LAN interconnection and for constant bit rate data types such as voice and video. In addition, many high profile networks are adopting frame relay not only as the network interface but also as a backbone switching technology. The main reason for this wide adoption is the simplicity and uniquity of frame relay which assures low cost. Frame Relay and Multimedia Applications Multimedia networks are designed to carry a mixture of information types. Although Frame Relay was intended to carry bursty frame-type data, it recently shows the ability to carry other types of information simultaneously as a part of a multimedia application. Traditional and bursty data: traditional data includes all forms of computer data that have been carried by leased lines. In many cases it means bursty data and in many other cases it means traffic that is carried by circuits since delay sensitivity and delay variation must be minimized. As frame relay achieves higher speeds, the delay that will cause problems for circuit type data will not be a problem and frame relay will find wider utilization. Increasingly frame relay users will move from traditional leased line connections over to frame relay, and the carriers and service providers will be positioned to handle this type of traffic.

Voice: This is and will continue to be an important traffic type for frame relay. Depending on the availability, cost and utilization, the suitable network (telephony, frame relay or internet) will be selected for voice. Much of the research and development done on Internet and intranet voice has direct applicability to frame relay and both areas will continue to flourish. Image: Since images are actually very large files, image application on frame relay networks will benefit from standardized frame fragmentation. With no fragmentation a large frame could monopolize resources and cause delays for other applications. Therefore, in many cases image applications use separate dedicated leased lines or other transport capability besides frame relay service. Improving image compression technologies together with fragmentation will make image application perform better. Video: Video is known to be a difficult application for frame relay due to the real time, delay sensitive, constant bit rate nature of current video transmission techniques. The appearance of real time video compression and use of other technologies has given the video applications (accept the highest bandwidth ones) a bursty aspect and has improved their sensitivity to delay.
From a transmission point of view, video can be divided into 3 categories:

• •

High bandwidth broadcast quality or near broadcast quality real time videoThis is not a very good application. Real time or near real time video that can be compressed or modified to require a smaller amount of bandwidth. As a result of variations in compression it exhibits a bursty traffic profile and may be multiplexed with other information. This type of video has great promise for use on frame relay and there are a number of products that use this method. Stored video - In this method , video "clips" are being compressed, stored and played back across the network many times. The method doesn’t necessarily require the most powerful compression tool available on the market (there are many innovative video compression algorithms but most of them are not usable for real time because they take too much time to perform the compression) since the data/image stored is not needed immediately.

•

Multimedia: since it is a mixture of information types, multimedia takes into account all of the information transport issues. The multimedia nature of the communication enriches it's information content, therefore we will be seeing more multimedia applications using frame relay.