Professional Documents
Culture Documents
Voip Technology
Voip Technology
INDEX
S No Topic
4. ATM
5. 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.
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.
Protocols
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.
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
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.
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 STM
TS=13 TS=5 TS=13 TS=5 TS=13 TS=5
ATM
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)
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.
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.
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.
Protocol
Presentation Presentation
Layer Layer
Protocol
Session Session
Layer Protocol Layer
Transport Transport
Layer Layer
Network Network Network Network
Layer Layer Layer Layer
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 pre-
allotment 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.
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
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.
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.
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.
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)
M- Terminal 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.
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.
Solution
Data Word 110101010
Divisor 10101
111000111 Quotient
10101 1101010100000 Dividend
10101
11111
10101
10100
11101
11000
10101
11010
10101
11110
10101
1011 Remainder
1101010100000
1011
FCS = Code Word 1101010101011
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
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.
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.
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 :
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:
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 project-
management 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.
• It is vendor independent.
5.0 - Brief :
• 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.
6.0 - A Typical Intranet setup
• 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
• 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
INTERNET
SN
SS7
PSTN/INT. GATE
GATEWAY WAY
CEN.
OFFICE
SCP
MOBILE
OFFICE
IN ISDN
VAS
PINT Reference module consists of following: -
MIB MIB
GDB GDB
2
4
IP
SWITCH SWITCH
I
N/W 1 I
PHONE P P
a PHONE
GATEWAY GATEWAY
3
b HOST
HOST d
c
FIREWALL
CONFERENCE
BRIDGE
WEBSERVER
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 Node
Caller
IP NETWORK Caller
PSTN
PSTN
Node Node
Caller Caller
PSTN
PSTN
VOIP NETWORK WITH DEDICATED IP TRUNK CONNECTIVITY
NODE
PSTN
PSTN
PSTN ATM
ATM NODE
ATM
NODE
ATM PSTN
ATM
NODE
PSTN
NODE
LE LE
PSTN
PSTN
ISP ISP
VOIP
SERVER INTERNET
COMPUTER
LE LE
PSTN
PSTN
VOIP
ISP ISP GATEWAY
INTERNET
PSTN PSTN
PHONE TO PHONE
INTERNET TELEPHONEY
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 Bit Rate (Kbps) Compression Delay (ms)
G.711 PCM 64 5
G.729 CS-ACELP 8 5
G.729a CS-ACELP 8 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 re-
transmission handshake adds more than a round trip delay to transmission. For time-
critical 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
AAL AAL
A
1 2 3
T
B
D
M
C (a)
Fig. 2
A
Comparison of (a) TDM and (b) ATDM 2 3
A 1
B T
D
M (b)
C
(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)
The ATM cell acts as the basic unit of information transfer in the ATM communication
(b) ATM
Octet # Cell Structure
1 2 3 4 5 6 7 8 53
1
2
3
4
Header User information space
5
6
7
8
5 Bytes 48 Bytes
53 Bytes
(ATM Cell)
Bit #
Bit #
1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8
OCTET #
GFC VIP VIP
VIP VCI
VCI VCI
VCI PT CLP VCI PT CLP
HEC HEC
(b) (c)
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
Bit Allocation
Function
UNI NNI
GFC 4 0
VIP 8 12
VCI 16 16
PT 3 3
CLP 1 1
HEC 8 8
NNI NNI
B-ISDN
TT TT TT TT
SWITCH
SWITCH SWITCH SWITCH
ATM CELL A
UNI
UNI
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 Yes
Port Sharing Yes Yes
Sensitive to Protocols Yes No
Efficient with Bursty traffic Yes Yes
High volume of Data No Yes
Transmission Speed Slow/Med Med/Fast
Delay High Low
Error correction Yes No
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:
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 8
DLCI high order C/R EA 0
DLCI low order FECN BECN DE EA 1
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.
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.
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