Handout Summar Training

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INDEX

CHAPTER CONTENTS PAGE

1 OVERVIEW OF TELECOMMUNICATION NETWORKS-I

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2 OVERVIEW OF TELECOMMUNICATION NETWORKS-II

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3 PCM PRINCIPLES 40

4 DIGITAL SWITCHING 60

5 SIGNALLING IN TELECOMMUNICATION 73

6 FIBRE-OPTIC COMMUNICATION TECHNOLOGY 106

7 OF TRANSMISSION SYSTEMS & THEIR FEATURES 125

8 SDH 133

9 DWDM 145

10 MOBILE COMMUNICATION 160

11 CDMA TECHNOLOGY 194

12 INTRODUCTION TO BROADBAND 213

13 INTELLIGENT NETWORK 248

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Chapter-1

OVERVIEW OF TELECOMMUNICATION NETWORKS-I Introduction

The telephone is a telecommunication device that is used to transmit and receive electronically or digitally encoded speech between two or more people conversing. It is one of the most common household appliances in the world today. Most telephones operate through transmission of electric signals over a complex telephone network which allows almost any phone user to communicate with almost any other user.

Telecommunication networks carry information signals among entities, which are geographically far apart. An entity may be a computer or human being, a facsimile machine, a teleprinter, a data terminal and so on. The entities are involved in the process of information transfer that may be in the form of a telephone conversation (telephony) or a file transfer between two computers or message transfer between two terminals etc.

With the rapidly growing traffic and untargeted growth of cyberspace, telecommunication becomes a fabric of our life. The future challenges are enormous as we anticipate rapid growth items of new services and number of users. What comes with the challenge is a genuine need for more advanced methodology supporting analysis and design of telecommunication architectures. Telecommunication has evaluated and growth at an explosive rate in recent years and will undoubtedly continue to do so.

The communication switching system enables the universal connectivity. The universal connectivity is realized when any entity in one part of the world can communicate with any other entity in another part of the world. In many ways telecommunication will acts as a substitute for the increasingly expensive physical transportation.

The telecommunication links and switching were mainly designed for voice communication. With the appropriate attachments/equipments, they can be used to transmit data. A modern society, therefore needs new facilities including very high bandwidth switched data networks, and large communication satellites with small, cheap earth antennas.

Voice Signal Characteristics

Telecommunication is mainly concerned with the transmission of messages between two distant points. The signal that contains the messages is usually converted into electrical waves before transmission. Our voice is an analog signal, which has amplitude and frequency characteristics.

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Voice frequencies: - The range of frequencies used by a communication device determines the communication channel, communicating devices, and bandwidth or information carrying capacity. The most commonly used parameter that characterizes an electrical signal is its bandwidth of analog signal or bit rate if it is a digital signal. In telephone system, the frequencies it passes are restricted to between 300 to 3400 Hz.

In the field of telecommunications, a Telephone exchange or a Telephone switch is a system of electronic components that connects telephone calls. A central office is the physical building used to house inside plant equipment including telephone switches, which make telephone calls "work" in the sense of making connections and relaying the speech information.

Switching system fundamentals

Telecommunications switching systems generally perform three basic functions: they transmit signals over the connection or over separate channels to convey the identity of the called (and sometimes the calling) address (for example, the telephone number), and alert (ring) the called station; they establish connections through a switching network for conversational use during the entire call; and they process the signal information to control and supervise the establishment and disconnection of the switching network connection.

In some data or message switching when real-time communication is not needed, the switching network is replaced by a temporary memory for the storage of messages. This type of switching is known as store-and-forward switching.

Signaling and control

The control of circuit switching systems is accomplished remotely by a specific form of data communication known as signaling. Switching systems are connected with one another by telecommunication channels known as trunks. They are connected with the served stations or terminals by lines.

In some switching systems the signals for a call directly control the switching devices over the same path for which transmission is established. For most modern switching systems the signals for identifying or addressing the called station are received by a central control that processes calls on a time-shared basis. Central controls receive and interpret signals, select and establish communication paths, and prepare signals for transmission. These signals include addresses for use at succeeding nodes or for alerting (ringing) the called station.

Most electronic controls are designed to process calls not only by complex logic but also by logic tables or a program of instructions stored in bulk electronic memory. The tabular technique is known as translator. The electronic memory is now the most accepted technique and is known as stored program control (SPC). Either type of control

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may be distributed among the switching devices rather than residing centrally. Microprocessors on integrated circuit chips are a popular form of distributed stored program control.

Switching fabrics

Space and time division are the two basic techniques used in establishing connections. When an individual conductor path is established through a switch for the duration of a call, the system is known as space division. When the transmitted speech signals are sampled and the samples multiplexed in time so that high-speed electronic devices may be used simultaneously by several calls, the switch is known as time division.

In the early stages of development in telecommunication, manual switching methods were deployed. But later on to overcome the limitations of manual switching; automatic exchanges, having Electro-mechanical components, were developed. Strowger exchange, the first automatic exchange having direct control feature, appeared in 1892 in La Porte (Indiana). Though it improved upon the performance of a manual exchange it still had a number of disadvantages, viz., a large number of mechanical parts, limited availability, inflexibility, bulky in size etc. As a result of further research and development, Crossbar exchanges,having an indirect control system, appeared in 1926 in Sweden.

The Crossbar exchange improved upon many short- comings of the Strowger system. However, much more improvement was expected and the revolutionary change in field of electronics provided it. A large number of moving parts in Register, marker, Translator, etc., were replaced en-block by a single computer. This made the exchange smaller in size, volume and weight, faster and reliable, highly flexible, noise-free, easily manageable with no preventive maintenance etc.

Network Architecture.

When electronic devices were introduced in the switching systems, a new concept of switching evolved as a consequence of their extremely high operating speed compared to their former counter-parts, i.e., the Electro-mechanical systems, where relays, the logic elements in the electromechanical systems, have to operate and release several times which is roughly equal to the duration of telephone signals to maintain required accuracy.

Research on electronic switching started soon after the Second World War, but commercial fully electronic exchange began to emerge only about 30 years later. However, electronic techniques proved economic for common control systems much earlier. In electromechanical exchanges, common control systems mainly used switches and relays, which were originally designed for use in switching networks. In common controls, they are operated frequently and so wear out earlier. In contrast, the life of an electronic device is almost independent of its frequency of operation. This gave a motivation for developing electronic common controls and resulted in electronic

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replacements for registers, markers, translators etc. having much greater reliability than their electromechanical predecessors.

In electromechanical switching, the various functions of the exchange are achieved by the operation and release of relays and switch (rotary or crossbar) contacts, under the direction of a Control Sub-System. These contracts are hard - wired in a predetermined way. The exchange dependent data, such as subscriber’s class of service, translation and routing, combination signaling characteristics are achieved by hard-ware and logic, by a of relay sets, grouping of same type of lines, strapping on Main or Intermediate Distribution Frame or translation fields, etc. When the data is to be modified, for introduction of a new service, or change in services already available to a subscriber, the hardware change ranging from inconvenient to near impossible, are involved.

In an SPC exchange, a processor similar to a general-purpose computer is used to control the functions of the exchange. All the control functions, represented by a series of various instructions, are stored in the memory. Therefore the processor memories hold all exchange dependent data. such as subscriber date, translation tables, routing and charging information and call records. For each call processing step. e.g. for taking a decision according to class of service, the stored data is referred to, Hence, this concept of switching. The memories are modifiable and the control program can always be rewritten if the behavior or the use of system is to be modified. This imparts and enormous flexibility in overall working of the exchange.

Digital computers have the capability of handling many tens of thousands of instructions every second, Hence, in addition to controlling the switching functions the same processor can handle other functions also. The immediate effect of holding both the control programme and the exchange data, in easily alterable memories, is that the administration can become much more responsive to subscriber requirements. both in terms of introducing new services and modifying general services, or in responding to the demands of individual subscriber. For example, to restore service on payment of an overdue bill or to permit change from a dial instrument to a multi frequency sender, simply the appropriate entries in the subscriber data-file are to be amended. This can be done by typing- in simple instructions from a teletypewriter or visual display unit. The ability of the administration to respond rapidly and effectively to subscriber requirements is likely to become increasingly important in the future.

The modifications and changes in services which were previously impossible be achieved very simply in SPC exchange, by modifying the stored data suitably. In some cases, the subscribers can also be given the facility to modify their own data entries for supplementary services, such as on-demand call transfer, short code (abbreviated) dialing, etc.

The use of a central processor also makes possible the connection of local and remote terminals to carry out man-machine dialogue with each exchange. Thus, the maintenance and administrative operations of all the SPC exchanges in a network can be

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performed from a single centralized place. The processor sends the information on the performance of the network, such as, traffic flow, billing information, faults, to the centre, which carries out remedial measures with the help of commands. Similarly, other modifications in services can also be carried out from the remote centre. This allows a better control on the overall performance of the network.

As the processor is capable of performing operations at a very high speed, it has got sufficient time to run routine test programmes to detect faults, automatically. Hence, there is no need to carry out time consuming manual routine tests.

In an SPC exchange, all control equipment can be replaced by a single processor. The processor must therefore be quite powerful, typically it must process hundreds of calls per second, in addition to performing other administrative and maintenance tasks. However, totally centralized control has drawbacks. The software for such a central processor will be voluminous, complex, and difficult to develop reliably. Moreover, it is not a good arrangement from the point of view of system security, as the entire system will collapse with the failure of the processor. These difficulties can be overcome by decentralizing the control. Some routine functions such as scanning, signal distributing, marking, which are independent of call processing, can be delegated to auxiliary or peripheral processors.

Stored program control (SPC) has become the principal type of control for all types of new switching systems throughout the world, including private branch exchanges, data and Telex systems. Two types of data are stored in the memories of electronic switching systems. One type is the data associated with the progress of the call, such as the dialed

address of the called line. Another type, known as the translation data, contains infrequently changing information, such as the type of service subscribed to by the calling line and the information required for routing calls to called numbers. These translation data, like the program, are stored in a memory, which is easily read but protected to avoid accidental erasure. This information may be readily changed, however, to meet service needs. The flexibility of a stored program also aids in the administration and maintenance of the service so that system faults may be located quickly.

SPC exchanges can offer a wider range of facilities than earlier systems. In addition, the facilities provided to an individual customer can be readily altered by changing the customer’s class-of-service data stored in memory. Moreover, since the processor’s stored data can be altered electronically,some of these facilities can be controlled by customers. Examples include:-

1. Call barring (outgoing or incoming): The customer can prevent unauthorized calls being made and can prevent incoming calls when wishing to be left in peace.

2. Call waiting: The ‘Call waiting’ service notifies the already busy subscriber of a third party calling him.

3. Alarm calls: The exchange can be instructed to call the customer at a pre-arranged

time (e.g. morning alarm).

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4. Call Forwarding: The subscriber having such a feature can enable the incoming calls coming to his telephone to be transferred to another number during his absence.

5. Conference calls: Subscriber can set up connections to more than one subscriber and conduct telephone conferences under the provision of this facility.

6. Dynamic Barring Facility: Subscriber having STD/ISD facilities can dynamically lock such features in their telephone to avoid misuse. Registering and dialing a secret code will extend such such a facility.

7. Abbreviated Dialing: Most subscribers very often call only limited group of telephone numbers. By dialing only prefix digit followed by two selection digits, subscribers can call up to 100 predetermined subscribers connected to any automatic exchange. This shortens the process of dialing all the digits.

8. Malicious call Identification: Malicious call identification is done immediately and the information is obtained in the print out form either automatically or by dialing an identification code.

9. Do Not Disturb: This facility enables the subscriber to free himself from attending his incoming calls. Using this facility the calls coming to the subscriber can be routed to an operator position or to an answering machine. The operator position or the machine can inform the calling subscriber that the called subscriber is temporarily inaccessible. Today SPC is a standard feature in all the electronic exchanges.

Implementation of Switching Network.

In an electronic exchange, the switching network is one of the largest sub-system in terms of size of the equipment. Its main functions are Switching (setting up temporary connection between two or more exchange terminations), Transmission of speech and signals between these terminations, with reliable accuracy. There are two types of electronic switching system. viz. Space division and Time Division. Space Division switching System

In a space Division Switching system, a continuous physical path is set up between input and output terminations. This path is separate for each connection and is held for the entire duration of the call. Path for different connections is independent of each other. Once a continuous path has been established., Signals are interchanged between the two terminations. Such a switching network can employ either metallic or electronic cross points. Previously, usage of metallic cross-points using reed relays and all were favored. They have the advantage of compatibility with the existing line and trunk signaling conditions in the network. Time Division Switching System

In Time Division Switching, a number of calls share the same path on time division sharing basis. The path is not separate for each connection, rather, is shared

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sequentially for a fraction of a time by different calls. This process is repeated periodically at a suitable high rate. The repetition rate is 8 KHz, i.e. once every 125 microseconds for transmitting speech on telephone network, without any appreciable distortion. These samples are time multiplexed with staggered samples of other speech channels, to enable sharing of one path by many calls. The Time Division Switching was initially accomplished by Pulse Amplitude

Modulation (PAM) Switching. However, it still could not overcome the performance limitations of signal distortion noise, cross-talk etc. With the advent of Pulse Code Modulation (PCM), the PAM signals were converted into a digital format overcoming the limitations of analog and PAM signals. PCM signals are suitable for both transmission and switching. The PCM switching is popularly called Digital Switching. Digital Switching Systems

A Digital switching system, in general, is one in which signals are switched in digital form. These signals may represent speech or data. The digital signals of several speech samples are time multiplexed on a common media before being switched through the system.

To connect any two subscribers, it is necessary to interconnect the time-slots of the two speech samples, which may be on same or different PCM highways. The digitalized speech samples are switched in two modes, viz., Time Switching and Space Switching. This Time Division Multiplex Digital Switching System is popularly known as Digital Switching System.

The ESS No.1 system was the first fully electronic switching system but not digital. But later came ESS No.4 system which was digital for trunk portion only. When designed, the cost of A/D conversion (CODEC) on each subscriber line was seen as prohibitive. So the ESS No.4 system was acting as a Trunk/Tandem exchange but not as a local exchange. So the main difficulty for implementing a digital local exchange was the implementation of the subscriber line interface. This was solved by the introduction of Integrated Circuits, which made the digital local exchange economically feasible. This implementation handles the following functions:

B-Battery feed O-Over-voltage protection (from lightning and accidental power line contact) R-Ringing S-Supervisory Signaling C-Coding (A/D inter conversion & low pass filtering) H-Hybrid (2W to 4W conversion) T-Testing the connectivity of Subscriber

Examples of digital exchanges (switching systems) include CDOT, OCB, AXE, EWSD, 5ESS etc.

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The general architecture of a Digital Switching System is depicted in fig2General architecture of Digital Switching System

Figure-2

The next evolutionary step was to move the PCM codec from the exchange end of the customer’s line to the customer’s end. This provides digital transmission over the customer’s line, which can have a number of advantages. Consider data transmission. If there is an analog customer’s line, a modem must be added and data can only be transmitted at relatively slow speeds. If the line is digital, data can be transmitted by removing the codec (instead of adding a modem). Moreover, data can be transmitted at 64 kbit/s instead of at, say, 2.4 kbit/s. Indeed, any form of digital signal can be transmitted whose rate does not exceed 64 kbit/s. This can include high-speed

fax, in addition to speech and data.

Subs interface

Other exchanges

N x 2 Mbps links

CONTROL PROCESSOR

Operation & Maintenance

Trunks interface

Other auxiliary inter faces Such as, (a) Tone generator (b) Frequency receives (c) Conference call facility (d) CCS# 7 Protocol

Manager (e) V 5.2 access manager

Digital Switch

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Control of switching systems Switching systems have evolved from being manually controlled to being

controlled by relays and then electronically. The change from the manual system to the Strowger step-by-step system brought about a change from centralized to distributed control. However, as systems developed and offered more services to customers, it became economic to perform particular functions in specialized equipments that were associated with connections only when required, thus, common control was introduced.

Later, the development of digital computer technology enabled different functions to be performed by the same hardware by using different programs; thus switching system entered the era of stored-program control (SPC).

There are basically two approaches to organizing stored program control: centralized and distributed. Early electronic switching systems (ESS) developed during the period 1970-75 almost invariably used centralized control. Although many present day exchange designs continue to use centralized SPC, with the advent of low cost powerful microprocessors and very large scale integration (VLSI) chips such as programmable logic arrays (PLA) and programmable logic controllers (PLC), distributed SPC is gaining popularity.

The figure below shows the evolution of electronic switching systems from the manual switching systems. The figure also depicts the changing scenario from digital switching to Broadband where the focus will be for high bit rate data transmissions.

Development of exchanges

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Local and trunk Network Trunk Lines

The term Trunk Line in telecommunications refers to the high-speed connection between telephone central offices in the Public Switched Telephone Network (PSTN). Trunk lines are always digital. The wiring between central offices was originally just pairs of twisted copper wire (the twists in the wiring prevented things known as crosstalk and noise). Because it is expensive to string up (or lay trenches for buried cables), the phone company researched ways in which to carry more data over the existing copper lines. This was achieved by using time-division multiplexing. Later, when fiber-optic technology became available, phone companies upgraded their trunk lines to fiber optics and used statistical time-division multiplexing, synchronous digital heirarchy, coarse or dense wave division multiplexing and optical switching to further improve transmission speeds.

The signaling information exchanged between different exchanges via inter exchange trunks for the routing of calls is termed as Inter exchange Signaling. Earlier in band /out of band frequencies were used for transmitting signaling information. Later on, with the emergence of PCM systems, it was possible to segregate the signaling from the speech channel. A trunk line is a circuit connecting telephone switchboards (or other switching equipment), as distinguished from local loop circuit which extends from telephone exchange switching equipment to individual telephones or information origination/termination equipment.

When dealing with a private branch exchange (PBX), trunk lines are the phone lines coming into the PBX from the telephone provider. This differentiates these incoming lines from extension lines that connect the PBX to (usually) individual phone sets. Trunking saves cost, because there are usually fewer trunk lines than extension lines, since it is unusual in most offices to have all extension lines in use for external calls at once. Trunk lines transmit voice and data in formats such as analog, T1, E1, ISDN or PRI. The dial tone lines for outgoing calls are called DDCO (Direct Dial Central Office) trunks. A signal travelling over a trunk line is not actually flowing any faster. The electrical signal on a voice line takes the same amount of time to traverse the wire as a similar length trunk line. What makes trunk lines faster is that the signal has been altered to carry more data in less time using more advanced multiplexing and modulation techniques. If you compared a voice line and a trunk line and put them side by side and observed them, the first pieces of information arrive simultaneously on both the voice and trunk line. However, the last piece of information would arrive sooner on the trunk line. No matter what, you can't break the laws of physics. Electricity over copper or laser light over fiber optics, you cannot break the speed of light--though that has rarely stopped uneducated IT or IS managers from demanding that cabling perform faster instead of upgrading equipment.

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Trunk lines can contain thousands of simultaneous calls that have been combined using time-division multiplexing. These thousands of calls are carried from one central office to another where they can be connected to a de-multiplexing device and switched through digital access cross connecting switches to reach the proper exchange and local phone number.

Local and trunk Network

S : Remote line unit L : Local subscriber exchange TR : Transit exchange CID : Outgoing international exchange CIA : Incoming international exchange CTI : International transit exchange

09 TR TR L S

s

S

L

CI

TR

TR

L S

S

CI

CT

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What is Trunking?

In telecommunications systems, trunking is the aggregation of multiple user circuits into a single channel. The aggregation is achieved using some form of multiplexing. Trunking theory was developed by Agner Krarup Erlang, Erlang based his studies of the statistical nature of the arrival and the length of calls. The Erlang B formula allows for the calculation of the number of circuits required in a trunk based on the Grade of Service and the amount of traffic in Erlangs the trunk needs cater for.

Definition

In order to provide connectivity between all users on the network one solution is to build a full mesh network between all endpoints. A full mesh solution is however impractical, a far better approach is to provide a pool of resources that end points can make use of in order to connect to foreign exchanges. The diagram below illustrates the where in a telecommunication network trunks are used.

A Modern Telephone Network Indicating where trunks are used. SLC - Subscriber

line concentrator

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LE – Local Exchange

TDM TAX –II – Level II Tax

TDM TAX –I – Level –I Tax

Level I Taxs are connected to the Gateway.

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Call routing

Routing in the PSTN is the process used to route telephone calls across the public switched telephone network. This process is the same whether the call is made between two phones in the same locality, or across two different continents.

Relationship between exchanges and operators

Telephone calls must be routed across a network of multiple exchanges, potentially owned by different telephone operators. The exchanges are all are inter-connected together using trunks. Each exchange has many "neighbours", some of which are also owned by the same telephone operator, and some of which are owned by different operators. When neighbouring exchanges are owned by different operators, they are known as interconnect points.

This means that there is really only one virtual network in the world that enables any phone to call any other phone. This virtual network comprises many interconnected operators, each with their own exchange network. Every operator can then route calls directly to their own customers, or pass them on to another operator if the call is not for one of their customers.

The PSTN is not a fully meshed network with every operator connected to every other - that would be both impractical and inefficient. Therefore calls may be routed through intermediate operator networks before they reach their final destination. One of the major problems in PSTN routing is determining how to route this call in the most cost effective and timely manner.

Call routing

Each time a call is placed for routing, the destination number (also known as the called party) is entered by the calling party into their terminal. The destination number generally has two parts, a prefix which generally identifies the geographical location of the destination telephone, and a number unique within that prefix that determines the specific destination terminal. Sometimes if the call is between two terminals in the same local area (that is, both terminals are on the same telephone exchange), then the prefix may be omitted.

When a call is received by an exchange, there are two treatments that may be applied:

• Either the destination terminal is directly connected to that exchange, in which case the call is placed down that connection and the destination terminal rings.

• Or the call must be placed to one of the neighbouring exchanges through a connecting trunk for onward routing.

Each exchange in the chain uses pre-computed routing tables to determine which connected exchange the onward call should be routed to. There may be several

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alternative routes to any given destination, and the exchange can select dynamically between these in the event of link failure or congestion.

The routing tables are generated centrally based on the known topology of the network, the numbering plan, and analysis of traffic data. These are then downloaded to each exchange in the telephone operators network. Because of the hierarchical nature of the numbering plan, and its geographical basis, most calls can be routed based only on their prefix using these routing tables.

Some calls however cannot be routed on the basis of prefix alone, for example non-geographical numbers, such as toll-free or freephone calling. In these cases the Intelligent Network is used to route the call instead of using the pre-computed routing tables.

In determining routing plans, special attention is paid for example to ensure that two routes do not mutually overflow to each other, otherwise congestion will cause a destination to be completely blocked.

According to Braess' paradox, the addition of a new, shorter, and lower cost route can lead to an increase overall congestion[. The network planner must take this into account when designing routing paths.

One approach to routing involves the use of Dynamic Alternative Routing (DAR). DAR makes use of the distributed nature of a telecommunications network and its inherent randomness to dynamically determine optimal routing paths. This method generates a distributed, random, parallel computing platform that minimises congestion across the network, and is able to adapt to take changing traffic patterns and demands into account.

Routing can be loosely described as the process of getting from here to there. Routing may be discussed in the context of telephone networks or computer networks. In telephone networks, routing is facilitated by switches in the network, whereby in computer networks routing is performed by routers in the network.

Definition: Routing in telephone networks

Routing in the context of telephone networks is the selection of a specific circiut group, for a given call or traffic stream, at an exchange in the network . "The objective of routing is to establish a successful connection between any two exchangesin the network" . By selecting routes that meet the constraints set by the user traffic and the network, routing determines which network resources (circuit group) should be used to transport which user traffic.

Different networks employ different routing techniques, but all communication networks share a basic routing functionality based on three core routing functions

Assembling and distributing information on the state of the network and user traffic that is used to generate and select routes.

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Generating and selecting feasible and optimal routes based on network and user traffic state information. Forwarding user traffic along the selected routes. The public switched telephone network (PSTN) architecture is made up of a hierarchy of exchanges (e.g local and regoinal exchanges) with each level of the hierarchy performing different functions . Two adjacent exchanges in the network may be connected by several direct routes consisting of one or more circuits . In circuit-switched networks, such as the PSTN, switching and transmission resources are dedicated to a call along the path from source to destination for the complete duration of the call. Routing decisions are imperative in facilitating this process as they determine the most efficient links to use to connect users for a call . Routing in the PSTN is done using a hop-by-hop approach . When a user wants to make a call, they dial the destination number to which the call should be routed. This destination number is made up of a prefix (area code or national destination network), which identifies the geographical location of the called party, and a unique number (the subscriber number) linked to the prefix that identifies the exact destination to which the call should be routed The end exchange to which the calling party is connected (the originating exchange) uses the area code to identify the outgoing circuit group connecting to the first choice adjacent exchange en-route This circuit group is called the first choice route and is obtained using a routing table at the originating switch . The function of the switch at the originating end exchange is to connect the switch input port to which the calling user is connected to a free outgoing circuit group in the first choice group . If all the circuits along the first choice route are fully occupied, the switch then attempts to use an alternative route circuit group to route the call to the destination exchange . The originating exchange then forwards the address to the adjacent exchange (first choice or alternate route), and the procedure is repeated at the adjacent exchange in order to reach the destination end exchange to which the called party is connected . When the address reaches the destination exchange, it only needs to process the last part of the address to identify the switch input port that the called party is connected . Routing directs forwarding . Forwarding of traffic can be done using connection-oriented or connectionless approaches . In connection-oriented forwrding, forwarding instructions are installed in all the switches along a designated route before the route can be used to transport traffic . Traffic forwarded using the connectionless approach carries its own forwarding information either as precise routing commands for each switch along a route or as hints that may be autonomously interpreted by any switch in the network . In PSTN, forwarding of traffic is based on the connection-oriented approach. Call routing is achieved using pre-computed routing tables, containing all the possible pre-defined routes for a connection, at each switch .The pre-defined routes specified in the routing table include information of a direct route (or routes) to be used under normal traffic and network conditions (e.g no link failure or network congestion) as well as alternative routes that should be used in the event that all circuits along the direct route are fully occupied . An alternative route may be an indirect route consisting of several circuit groups connecting two exchanges via other exchanges . The following example illustrates the use of an alternative route to connect two exchanges in the event of the direct route being congetsed.

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A Typical Telephone Exchange -OCB-283 FUNCTIONAL ARCHITURE

The Alcatel E10 system is located at the heart of the telecommunication networks concerned. It is made up of three independent functional units: - The “Subscriber Access Subsystem” which carries out connection of analogue and

digital subscriber lines,

- “Connection and Control” which carries out connections and processing of calls,

- “Operation and Maintenance” which is responsible for all functions needed by the network operating authority.

Each functional unit is equipped with softwares which are appropriate for handling the functions for which it is responsible.

Synchronization and Time Base Station STS

Time base (BT)

The BT ensures times distribution for LR and PCM to provide the synchronization, and also for working out the exchange clock.Time distribution is tripled.

Time generation can be either autonomous or slaved to an external rhythm with a view to synchronise the system with the network

Auxiliary Equipment Control Station SMA

Auxiliary equipment manager (ETA)

The ETA Supports:

- The tone generators (GT).

- The frequency receiving and generation (RGF) devices,

- Conference circuits (CCF),

- The exchange clock

CCS7 protocol handler (PUPE) and CCS7 controller (PC): CCITT No. 7 protocol processing

For connection of 64 kbit/s signaling channels, semi- permanent connections are established via the connection matrix, to the PUPE which processes the CCITT No. 7 protocol.

More precisely, the PUPE function carries out the following:

- “signaling channel” Level 2 processing,

- the “message routing” function

(Part of Level 3). The PC carries out:

- the “network management” function (part of Level 3),

- PUPE defence,

- Various observation tasks which are not directly linked to CCITT No. 7.

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OCB 283

SUBCRIBER

ACCESS

SUBSYSTE

M

CONNECTION

AND

CONTROL

OPERATION

AND MAINTENANCE

DATA

NETWORK

TELEPHONE

NETWORK

VALUE ADDED NETWORK

CCITT N07 SIGNALLING

NETWORK

NT

PABX

℡℡℡℡

℡℡℡℡

ALCATEL 1000 E10 OCB 283

OPERATION AND MAINTENANCE

NETWORK

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Host switching matrix (SMX)

The SMX is a square connection matrix with a single time stage, T, duplicated in full, which enables up to 2048 matrix links (LR) to be connected.

A matrix link LR is an internal PCM, with 16 bits per channel (32 channels). The MCX can execute the following:

1) an unidirectional connection between any incoming channel and any out going channel. There can be as many simultaneous connections as there are outgoing channels. It should be remembered that a connection consists of allocating the information contained within an incoming channel to an outgoing channel,

2) connection between any incoming channel and any M outgoing channels,

3) connection of N incoming channels belonging to one frame structure of any multiplex onto N outgoing channels which belong to the same frame structure, abiding to the integrity and sequencing of the frame received. This function is referred to as “connection with N x 64 kbit/s”.

The MCX is controlled by the COM function (matrix switch controller) to ensure the:

- set up and breakdown of the connections by access to the matrix command memory. This access is used to write at the output T.S. address the incoming T.S. address

- defense of the connections. Security of the connections in order to assure a good data switching.

Truck Control Station SMT

PCM controller (URM)

The URM provides the interface between external PCMs and the OCB283. These PCM come from either:

- a remote subscriber digital access unit (CSN) or from a remote electronic satellite concentrator CSE,

- another switching centre, on channel-associated signalling or CCITT No.7,

- the digital recorded announcement equipment

In particular, the URM carries out the following functions:

- HDB3 conversion to binary (PCM → matrix link),

- binary conversion to HDB3 (matrix link” → PCM),

- extraction and pre-processing of the channel-associated signalling of T.S.16 (PCM → command),

- transmission of channel-associated signalling in T.S.16 (command →PCM).

Main Control Station SMC

Call handler (MR)

The MR is responsible for the establishment and breaking off of communications.

21

The call handler takes the decisions necessary for processing of communications in terms of the signaling received, after consultation of the subscriber and analysis database manager (TR) if necessary. The call handler processes new calls and handling-up operations, releases equipment, commands switching on and switching off etc.

In addition, the call handler is responsible for different management tasks (control of tests of circuits, sundry observations).

Operation and maintenance function (OM) SMM

The functions of the operation and maintenance subsystem are carried out by the operation and maintenance software OM).

The operating authority accesses all hardware and software equipment of the Alcatel 1000 E10 system via computer terminals belonging to the operation and maintenance subsystem: consoles, magnetic media, intelligent terminal. These functions can be grouped into 2 categories:

- operation of the telephone application,

- operation and maintenance of the system.

In addition, the operation and maintenance subsystem carries out:

- loading of softwares and of data for connection and command and for the subscriber digital access units,

- temporary backup of detailed billing information,

- centralisation of alarm data coming from connection and control stations, via alarm rings,

- central defence of the system.

Finally, the operation and maintenance subsystem permits two-way communication with operation and maintenance networks, at regional or national level (TMN). CSN - digital satellite center The digital satellite center [CSN center satellite numerique) is a subscriber connection unit on which both analogue and digital subscribers can be connected.

Its design and composition enable the CSN to fit into an existing network and can be

connected to time-based systems using the CCITT N° 7 type of semaphore signalling.

The CSN is a connection unit designed to adapt to a variety of geographical situation: it can be either local [CSNL] or distant [CSND] with respect to the connecting switch.

22

A Typical Telephone Exchange -OCB-283

CSN : Digital satellite center

SMC : Main Control Station

SMA : Auxiliary Equipment Control Station

SMT : Truck Control Station

SMX : Matrix Control Station

SMM : Maintenance Station

STS : Synchronization and Time Base Station

SMX STS 1 x 3

CS

CSCS

Circuits and announcement machine

L

SMT ( 1 TO 28) X 2

SMA

( 2 TO 37)

SMC 2 TO 14

1 TO 4 MAS

1

SMM 1 x 2

L

L

AT

23

Chapter-2

Overview of Telecommunications Networks –II Institutional mechanism and role

Introduction: All industries operate in a specific environment which keeps changing and

the firms in the business need to understand it to dynamically adjust their actions for best

results.

Like minded firms get together to form associations in order to protect their common

interests. Other stake holders also develop a system to take care of their issues.

Governments also need to intervene for ensuring fair competition and the best value for

money for its citizens. This handouts gives exposure on the Telecom Environment in

India and also dwells on the role of international bodies in standardizing and promoting

Telecom Growth in the world.

Institutional Framework: It is defined as the systems of formal laws, regulations, and

procedures, and informal conventions, customs, and norms, that broaden, mold, and

restrain socio-economic activity and behaviour. In India, The Indian telegraph act of 1885

amended from time to time governs the telecommunications sector. Under this act, the

government is in-charge of policymaking and was responsible for provisioning of

services till the opening of telecom sector to private participation. The country has been

divided into units called Circles, Metro Districts, Secondary Switching Areas (SSA),

Long Distance Charging Area (LDCA) and Short Distance Charging Area (SDCA).

Major changes in telecommunications in India

began in the 1980s. The initial phase of telecom reforms began in 1984 with the creation

of Center for Department of Telematics (C-DOT) for developing indigenous technologies

and private manufacturing of customer premise equipment. Soon after, the Mahanagar

Telephone Nigam Limited (MTNL) and Videsh Sanchar Nigam Limited (VSNL) were

set up in 1986.

The Telecom Commission was established in 1989. A crucial aspect of the institutional

reform of the Indian telecom sector was setting up of an independent regulatory body in

1997 – the Telecom Regulatory Authority of India (TRAI), to assure investors that the

sector would be regulated in a balanced and fair manner. In 2000, DoT corporatized its

24

services wing and created Bharat Sanchar Nigam Limited. Further changes in the

regulatory system took place with the TRAI Act of 2000 that aimed at restoring

functional clarity and improving regulatory quality and a separate disputes settlement

body was set up called Telecom Disputes Settlement and Appellate Tribunal (TDSAT) to

fairly adjudicate any dispute between licensor and licensee, between service provider,

between service provider and a group of consumers. In October 2003, Unified Access

Service Licenses regime for

basic and cellular services was introduced. This regime enabled services providers to

offer fixed and mobile services under one license. Since then, Indian telecom has seen

unprecedented customer growth crossing 600 million connections. India is the fourth

largest telecom market in Asia after China, Japan and South Korea. The Indian telecom

network is the eighth largest in the world and the second largest among emerging

economies. A brief on telecom echo system and various key elements in institutional

framework is given below:

Department of Telecommunications: In India, DoT is the nodal agency for taking care

of telecom sector on behalf of government. Its basic functions are:

Policy Formulation

Review of performance

25

Licensing

Wireless spectrum management

Administrative monitoring of PSUs

Research & Development

Standardization/Validation of Equipment

International Relations

Main wings within DoT: Telecom Engineering Center (TEC)

USO Fund

Wireless Planning & Coordination Wing (WPC)

Telecom Enforcement, Resource and Monitoring (TERM) Cell

Telecom Centers of Excellence (TCOE)

Public Sector Units

Bharat Sanchar Nigam Limited(BSNL)

Indian Telephone Industries Limited (ITI)

Mahanagar Telephone Nigam Limited(MTNL)

Telecommunications Consultants India Limited(TCIL)

R & D Unit

Center for development of Telematics (C-DoT)

The other key governmental institutional units are TRAI & TDSAT. Important units are

briefed below:

Telecom Engineering Center (TEC): It is a technical body representing the interest of

Department of Telecom, Government of India. Its main functions are:

Specification of common standards with regard to Telecom networkequipment, services

and interoperability.

Generic Requirements (GRs), Interface Requirements (IRs)

Issuing Interface Approvals and Service Approvals

Formulation of Standards and Fundamental Technical Plans

Interact with multilateral agencies like APT, ETSI and ITU etc. for standardisation

Develop expertise to imbibe the latest technologies and results of R&D

Provide technical support to DOT and technical advice to TRAI & TDSAT

Coordinate with C-DOT on the technological developments in the Telecom Sector

26

for policy planning by DOT www.tec.gov.in

Universal Service Obligation Fund (USO): This fund was created in 2002. This fund is

managed by USO administrator. All telecom operators contribute to this fund as per

government policy. The objective of this fund is to bridge the digital divide i.e. ensure

equitable growth of telecom facilities in rural areas. Funds are allocated to operators who

bid lowest for providing telecom facilities in the areas identified by USO administrator.

WIRELESS PLANNING & COORDINATION (WPC ) This unit was created in 1952 and

is the National Radio Regulatory Authority responsible for Frequency Spectrum

Management, including licensing and caters for the needs of all wireless users

(Government and Private) in the country. It exercises the statutory functions of the

Central Government and issues licenses to establish, maintain and operate wireless

stations. WPC is divided into major sections like Licensing and Regulation (LR), New

Technology Group (NTG) and Standing Advisory Committee on Radio Frequency

Allocation (SACFA). SACFA makes the recommendations on major frequency

allocation issues, formulation of the frequency allocation plan, making

recommendations on the various issues related to International Telecom Union (ITU), to

sort out problems referred to the committee by various wireless users, Siting clearance of

all wireless installations in the country etc.

Telecom Enforcement, Resource and Monitoring (TERM) Cell: In order to ensure

that service providers adhere to the licence conditions and for taking care of telecom

network security issues, DoT opened these cells in 2004 and at present 34 cells are

operating in various Circles and big districts in the country. Key functions of these units

are Inspection of premises of Telecom and Internet Service Providers, Curbing illegal

activities in telecom services, Control over clandestine / illegal operation of telecom

networks by vested interests having no license, To file FIR against culprits, pursue the

cases, issue notices indicating violation of conditions of various Acts in force from time

to time, Analysis of call/subscription/traffic data of various licensees, arrangement for

lawful interception / monitoring of all communications passing through the licensee’s

network, disaster management, network performance monitoring, Registration of OSPs

and Telemarketers in License Service Areas etc..

Telecom Centers of Excellence (TCOE): (www.tcoe.in) The growth of Indian

27

Telecommunications sector has been astounding, particularly in the last decade. This

growth has been catalysed by telecommunications sector liberalization and reforms.

Some of the areas needing immediate attention to consolidate and maintain the growth

are:

Capacity building for industry talent pool

Continuous adaptation of the regulatory environment to facilitate induction/

adoptation of high potential new technologies and business models

Bridging of high rural - urban teledensity/digital divide

Faster deployment of broadband infrastructure across the country

Centres of Excellence have been created to work on (i) enhancing talent pool,

(ii)technological innovation, (iii) secure information infrastructure and (iv) bridging of

digital divide. These COEs are also expected to cater to requirements of South Asia as

regional leaders. The main sponsor (one of the telecom operators), the academic institute

where the Centers are located and the tentative field of excellence are enumerated in the

table below:

Field of Excellence in Telecom Associated Institute Sponsor

Next Generation Network & Network Technology

IIT, Kharagpur Vodafone Essar

Telecom Technology & Management

IIT ,Delhi Bharti Airtel

Technology Integration, Multimedia & Computational Maths

IIT, Kanpur BSNL

Telecom Policy, Regulation, Governance, Customer Care & Marketing

IIM, Ahmedabad IDEA Cellular

Telecom Infrastructure & Energy IIT, Chennai Reliance

Disaster Management of Info systems & Information Security

IISc, Bangalore Aircel

Rural Application IIT Mumbai Tata Telecom

Spectrum Management (Proposed) WPC, Chennai Govt with Industry consortium

28

Telecom Regulatory Authority of India (TRAI): TRAI was established under TRAI

Act 1997 enacted on 28.03.1997. The act was amended in 2000. Its Organization setup

consists of One Chairperson, Two full-time members & Two part-time members. Its

primary role is to deals with regulatory aspects in Telecom Sector & Broadcasting and

Cable services.

TRAI has two types of functions as mentioned below:

Mandatory Functions

Tariff policies

Interconnection policies

Quality of Service

Ensure implementation of terms and conditions of license

Recommendatory Functions

New license policies

Spectrum policies

Opening of sector

www.trai.gov.in

Telecom Dispute Settlement Appellate Tribunal (TDSAT): TDSAT was established in

year 2000 by an amendment in TRAI act by transferring the functions of dispute handling

to new entity i.e. TDSAT. The organization setup consists of one Chairperson & two full-

time members. Its functions are:

Adjudicate any dispute between

licensor and licensee

two or more licensees

group of consumers

Hear & dispose off appeal against any direction, decision or order of the Authority

under TRAI Act www.tdsat.nic.in

Key International Standardization Bodies for Telecom sector:

ITU is the leading United Nations agency for information and communication technology

issues, and the global focal point for governments and the private sector in developing

networks and services. For nearly 145 years, ITU has coordinated the shared global use

of the radio spectrum, promoted international cooperation in assigning satellite orbits,

29

worked to improve telecommunication infrastructure in the developing world, established

the worldwide standards that foster seamless interconnection of a vast range of

communications systems and addressed the global challenges of our times, such as

mitigating climate change and strengthening cybersecurity. Vast spectrum of its work

area includes broadband Internet to latest-generation wireless technologies, from

aeronautical and maritime navigation to radio astronomy and satellite-based meteorology,

from convergence in fixed-mobile phone, Internet access, data, voice and TV

broadcasting to next-generation networks. ITU also organizes worldwide and regional

exhibitions and forums, such as ITU TELECOM WORLD, bringing together the most

influential representatives of government and the telecommunications and ICT industry

to exchange ideas, knowledge and technology for the benefit of the global community,

and in particular the developing world. ITU is based in Geneva, Switzerland, and its

membership includes 191 Member States and more than 700 Sector Members and

Associates. On 1 January 2009, ITU employed 702 people from 83 different countries.

The staff members are distributed between the Union's Headquarters in Geneva,

Switzerland and eleven field offices located around the world. www.itu.int

Asia Pacific Telecommunity: Headquartered at Bangkok, the APT is a unique

organization of Governments, telecom service providers, manufactures of communication

equipment,research & development organizations and other stake holders active in the

field of communication and information technology. APT serves as the focal organization

for communication and information technology in the Asia Pacific region. The APT has

34 Members, 4 Associate Members and 121 Affiliate Members. The objective of the

Telecommunity is to foster the development of telecommunication services and

information infrastructure throughout the region with a particular focus on the expansion

thereof in less developed areas. APT has been conducting HRD Programme for

developing the skills of APT Members to meet the objectives of APT. The topics include

Information

Communication Technologies (ICT), Network and Information Security, Finance and

Budget,Telecommunication Management, Mobile Communications, Multimedia,

Satellite Communication, Telecommunications and ICT Policy and Regulation,

Broadband Technologies, e-Applications, Rural Telecommunications Technologies, IP

Networks and Services, Customer Relations, etc. www.aptsec.org

30

The European Telecommunications Standards Institute (ETSI) produces

globallyapplicable standards for Information and Communications Technologies (ICT),

including fixed, mobile, radio, converged, broadcast and internet technologies. It is

officially recognized by the European Union as a European Standards Organization. ETSI

is a not-forprofit organization with more than 700 ETSI member organizations drawn

from 62 countries across 5 continents world-wide. ETSI unites Manufacturers, Network

operators, National Administrations , Service providers, Research bodies, User groups ,

Consultancies. This cooperation has resulted in a steady stream of highly successful ICT

standards in mobile, fixed, and radio communications and a range of other standards that

cross these boundaries, including Security, Satellite, Broadcast, Human Factors, Testing

& Protocols, Intelligent transport, Power-line telecoms, eHealth, Smart Cards,

Emergency communications, GRID & Clouds, Aeronautical etc. ETSI is consensus-based

and conducts its work through Technical Committees, which produce standards and

specifications, with the ETSI General Assembly and Board. www.etsi.org

BSNL: Bharat Sanchar Nigam Limted was formed in year 2000 and took over the service

providers role from DoT. Today, BSNL has a customer base of over 9 crore and is the

fourth largest integrated telecom operator in the country. BSNL is the market leader in

Broadband, landline and national transmission network. BSNL is also the only operator

covering over 5 lakh village with telecom connectivity. Area of operation of BSNL is all

India except Delhi &Mumbai.

MTNL: Mahanagar Telephone Nigam Limited, formed in 1984 is the market leader in

landline and broadband in its area of operation. www.mtnl.net.in

TCIL: TCIL, a prime engineering and consultancy company, is a wholly owned

Government of India Public Sector Enterprise. TCIL was set up in 1978 for providing

Indian telecom expertise in all fields of telecom, Civil and IT to developing countries

around the world. It has its presence in over 70 countries. www.tcil-india.com

ITI: Indian telephone Industries is the oldest manufacturing unit for telephone

instruments. To keep pace with changing times, it has started taking up manufacturing of

new technology equipment such as GSM, OFC equipment, Invertors, Power plants,

Defense equipments, Currency counting machines etc. www.itiltd-india.com

Centre for Development of Telematics (CDoT): This is the R & D unit under DoT setup

in 1984. The biggest contribution of this centre to Indian telecom sector is the

31

development of low capacity (128 port) Rural automatic Exchange (RAX) which enabled

provisioning of telephone in even the smallest village. This was specially designed to suit

Indian environment, capable of withstanding natural temperature and dusty conditions.

Prominent Licenses provided by DoT:

Access Service (CMTS & Unified Access Service): The Country is divided into 23

Service Areas consisting of 19 Telecom Circle Service Areas and 4 Metro Service Areas

for providing Cellular Mobile Telephone Service (CMTS). Consequent upon

announcement of guidelines for Unified Access (Basic& Cellular) Services licenses on

11.11.2003, some of the CMTS operators have been permitted to migrate from CMTS

License to Unified Access Service License (UASL). No new CMTS and Basic service

licenses are being awarded after issuing the guidelines for Unified access Service

Licence(UASL). As on 31st March 2008, 39 CMTS and 240 UASL licenses operated.

o 3G & BWA (Broadband Wireless Access): Department of Telecom started the auction

process for sale of spectrum for 3G and BWA (WiMax) in April 2010 for 22 services

areas in the country. BSNL & MTNL have already been given spectrum for 3G and

BWAand they need to pay the highest bid amount as per auction results. BSNL & MTNL

both are providing 3G services. BSNL has rolled out its BWA service by using WiMax

technology.

Mobile Number Portability (MNP) Service: Licenses have been awarded to two

perators to provide MNP in India. DoT is ensuring the readiness of all mobile operators

and expects to start this service any time after June 2010.

Infrastructure Provider: There are two categories IP-I and IP-II. For IP-I the applicant

company is required to be registered only. No license is issued for IP-I. Companies

registered as IP-I can provide assets such as Dark Fibre, Right of Way, Duct space and

Tower. This was opened to private sector with effect from 13.08.2000. An IP-II license

can lease / rent out /sell end to end bandwidth i.e. digital transmission capacity capable to

carry a message. This was opened to private sector with effect from 13.08.2000. Issuance

of IP-II Licence has been discontinued w.e.f. 14.12.05

INMARSAT : INMARSAT (International Maritime Satellite Organisation) operates a

constellation of geo-stationary satellites designed to extend phone, fax and data

communications all over the world. Videsh Sanchar Nigam Ltd (VSNL) is permitted

to provide Inmarsat services in India under their International Long Distance(ILD)

32

licence granted by Department of Telecommunications(DoT). VSNL has commissioned

their new Land Earth Station (LES) at Dighi, Pune compatible with 4th generation

INMARSAT Satellites (I-4) and INMARSAT-B, M, Mini-M & M-4 services are now

being provided through this new LES after No Objection Certificate (NOC) is issued by

DoT on case by case basis.

National Long Distance: There is no limit on number of operators for this service and

license is for 20 years.

International Long Distance: This was opened to private sector on 1st April 2002 with

no limit on number of operators. The license period is 20 years.

Resale of IPLC: For promoting competition and affordability in International Private

Leased Circuits (IPLC) Segment, Government permitted the “Resale of IPLC” by

introducing a new category of License called as – “Resale of IPLC” Service License

witheffect from 24th September 2008. The Reseller can provide end-to-end IPLC

between India and country of destination for any capacity denomination. For providing

the IPLC service, the Reseller has to take the IPLC from International Long Distance

(ILD) Service Providers licensed and permitted to enter into an arrangement for leased

line with Access Providers, National Long Distance Service Providers and International

Long Distance Service Providers for provision of IPLC to end customers.

Sale of International Roaming SIM cards /Global Calling Cards in India: The

cards being offered to Indian Customers will be for use only outside India. However, if it

is essential to activate the card for making test calls/emergent calls before the departure

of customer and /or after the arrival of the customer, the same shall be permitted for forty

eight (48) hours only prior to departure from India and twenty four (24) hours after

arrival in India.

Internet without Telephony: The Internet Service Provider (ISP) Policy was announced

in November, 98. ISP Licenses , which prohibit telephony on Internet ,are being

issued starting from 6.11.98 on non-exclusive basis. Three category of license exist

namely A,B and C. A is all India, B is telecom Circles, Metro Districts and major districts

where as C is SSA wide.

Internet with Telephony: Only ISP licensees are permitted, within their service

area, to offer Internet Telephony service. The calls allowed are PC to PC in India, PC in

India to PC/Telephone outside India, IP based calls from India to other countries.

33

VPN: Internet Service Providers (ISPs) can provide Virtual Private Network (VPN)

Services. VPN shall be configured as Closed User Group(CUG) only and shall carry only

the traffic meant for the internal use of CUG and no third party traffic shall be carried on

the VPN. VPN shall not have any connectivity with PSTN / ISDN / PLMN except when

the VPN has been set up using Internet access dial-up facility to the ISP node. Outward

dialing facility from ISP node is not permitted.

VSAT & Satellite Communication: There are two types of CUG VSAT licenses : (i)

Commercial CUG VSAT license and (ii) Captive CUG VSAT license. The commercial

VSAT service provider can offer the service on commercial basis to the subscribers by

setting up a number of Closed User Groups (CUGs) whereas in the captive VSAT

service only one CUG can be set up for the captive use of the licensee. The scope of the

service is to provide data connectivity between various sites scattered within territorial

boundary of India via INSAT Satellite System using Very Small Aperture

Terminals (VSATs). However, these sites should form part of a Closed User Group

(CUG). PSTN connectivity is not permitted.

Radio Paging: The bids for the Radio Paging Service in 27 cities were invited in 1992,

the licenses were signed in 1994 and the service was commissioned in 1995. There was

a provision for a fixed license fee for first 3 years and review of the license fee

afterwards.The license was for 10 years and in 2004 Govt offered a extended 10 years

license with certain license fee waivers but with the wide spread use of mobile phones,

this service has lost its utility.

PMRTS: Public Mobile Radio Trunking service allows city wide connectivity through

wireless means. This service is widely used by Radio Taxi operators and companies

whose workforce is on the move and there is need to locate the present position of

employee for best results. PSTN connectivity is permitted.

INSAT MSS: INSAT Mobile Satellite System Reporting Service (INSAT MSS

Reporting Service) is a one way satellite based messaging service available through

INSAT. The basic nature of this service is to provide a reporting channel via satellite to

the group of people, who by virtue of their nature of work are operating from remote

locations without any telecom facilities and need to send short textual message or short

data occasionally to a central station.

Voice Mail/ Audiotex/ UMS (Unified Messaging Service): Initially a seprate license

34

was issued for these services. For Unified Messaging Service, transport of Voice Mail

Messages to other locations and subsequent retrieval by the subscriber must be on a

nonreal time basis. For providing UMS under the licence, in addition to the licence for

Voice Mail/Audiotex/UMS, the licensee must also have an ISP licence. The ISP licence

as well as Voice Mail/Audiotex/ UMS licence should be for the areas proposed to be

covered by UMS service. Since start of NTP-99, all access provider i.e. CMTS, UASL,

Fixed service providers are also allowed to provide these services as Value Added

Service (VAS) under their license conditions.

Telemarketing: Companies intending to operate as Telemarketes need to obtain this

license from DoT. Other Service Provider (including BPO): As per New Telecom Policy

(NTP) 1999, Other Service Providers (OSP), such as tele-banking, tele-medicine, tele-

trading, ecommerce, Network Operation Centers and Vehicle Tracking Systems etc are

allowed to operate by using infrastructure provided by various access providers for non-

telecom services.

Telecom Operators: Interested companies obtain license for various services to get

authorization to provide licensed telecom services in India. While hundreds of license

holders exists in India for various services, major operators are BSNL, Bharti (Airtel),

Vodafone, Reliance, Aircel, Idea and Tata etc. There is a stiff competition in the market

and operators struggle to provide innovative services earlier than others, at rates lower

than rivals, continuously find ways to extend better customer care and improve profit

margins by managing costs. A typical diagram depicting various macro level activities

performed by a telecom service provider is given below:

35

In today’s fast growing customer base in telecom market, rising expectations of

customers for prompt service support, very efficient & powerful software solutions are a

must. For this purpose, over the years, OSS (Operations Support Systems) & BSS

(Business Support Systems) software solutions have been developed to manage these

activities. The term OSS most frequently describes "network systems" dealing with the

telecom network itself, supporting processes such as maintaining network inventory,

provisioning services, configuring network components, and managing faults. Business

Support Systems or BSS typically refers to "business systems" dealing with customers,

supporting processes such as taking orders, processing bills, and collecting payments.

The two systems together are often abbreviated BSS/OSS or simply B/OSS. Many

proprietary software solutions are available from different vendors. A standardization

initiative has been taken up by Telecom Management forum, an international

membership organization of communications service providers and suppliers to the

communications industry. TM Forum is regarded as the most authoritative source for

standards and frameworks in OSS. TM Forum has been active in proving a framework

and discussion forum for advancements in OSS and BSS. A typical architecture of

OSS/BSS application is given below:

Optical-OFC, DWDM etc., Transport-SDH,PDH, ATM,PSTN, DSL etc., IP-MPLS,

Internet, IP TV, Multicast etc., Fixed/Wireless-PSTN, GSM, CDMA, WiMax, 3G etc.,

System-Windows, Unix etc.

36

Sector Specific industry associations:

The Cellular Operators Association of India (COAI) was constituted in 1995 as a

registered, non-profit, non-governmental society dedicated to the advancement of

communication, particularly modern communication through Cellular Mobile Telephone

Services. COAI represents Indian Cellular industry and on its behalf it interacts with the

policy maker, the licensor, the regulator, the spectrum management agency and the

industry (telecom /non-telecom) associations. www.coai.com

Key Objectives of the COAI

To improve standards and competitiveness in the Cellular Industry and attain the

status of world class infrastructure. To facilitate affordable mobile telephony services

for Indians.

To study the best practices & research of the industry as well as to analyse the

Cellular Experience worldwide.

To assist relevant authorities by providing them information about the industry to help

them formulate suitable policies for the industry's growth.

To improve standards and quality of services in consultation with GSM India - the

Indian chapter of the GSM Association.

To maintain and upgrade services in terms of speech transmission, access, coverage,

security etc, to enable expansion of cellular services.

To help address problems of cellular operators relating to operational, regulatory,

financial, or licensing through interaction with the Ministry of Communications & IT,

Ministry of Finance, Ministry of Commerce, Department of telecommunications,

Telecom Regulatory Authority of India, Financial Institutions etc.

Association of Unified Telecom Service Providers of India (AUSPI) is the representative

industry body of Unified Access Service Licensees providing CDMA & GSM Mobile

Services, Fixed Line Services as well as Value Added Services throughout the length and

breadth of the country. AUSPI is a registered society and works as a non-profit

organization with the aim of delivering the promise of improved Access, Coverage and

Teledensity in India. The objectives of the Association include collection and

dissemination of knowledge and information for promotion and healthy growth of

telecom services, enunciating a telecom vision for India, fueling unprecedented domestic

investment, improving teledensity and bringing value for customers. The Association

37

interacts on policy and regulatory issues with various Government bodies such as the

Department of Telecommunications, Telecom Regulatory Authority of India, apex

industry organizations like ASSOCHAM, Confederation of Indian Industry (CII) and

Federation of Indian Chambers of Commerce & Industry (FICCI), technical institutions,

financial analysts and other institutions of world repute. The Association formulates

expert opinion on industry issues and submits whenever necessary, recommendations to

the concerned authorities. www.auspi.in

TEMA Established in 1990, Telecom Equipment Manufacturers’ Association of

India

(TEMA) is recognized by the Government of India as the National Apex body to

represent telecom Technology Providers, Global and Indian, Private and Government

owned companies. TEMA has membership of more than 150 member companies

covering almost 80 per cent of Indian Telecom Equipment Manufacturing. Services

offered to TEMA members include, interaction with Government, Policy makers,

interaction with various National Confederations of Industries, overseas Delegations,

Exhibition Organizers, Market Development Assistance Authorities, Tender Information,

Excise and Customs Departments, Telecom Engineering Center for product specifications

etc. Our members are exporting a variety of Telecom Equipments to South America,

Middle-East, Africa, SAARC, CIS andSouth East Asian Countries. TEMA also has an

Export Promotion Forum set up by theMinistry of Commerce, Government of India to

promote Export of Telecom Equipments and Services. The Forum also make various

recommendations to the Government for making necessary changes in various policies

and procedures for promotion of Exports and Services.

Key Industry/ Trade Associations influencing the Telecom Market

The Confederation of Indian Industry (CII) works to create and sustain an environment

conducive to the growth of industry in India, partnering industry and government alike

through advisory and consultative processes.CII is a non-government, not-for-profit,

industry led and industry managed organisation, playing a proactive role in India's

development process. Founded over 115 years ago, it is India's premier business

association, with a direct membership of over 7800 organisations from the private as well

as public sectors, including SMEs and MNCs, and an indirect membership of over 90,000

companies from around 396 national and regional sectoral associations. With 64 offices

38

in India, 9 overseas in Australia, Austria, China, France, Germany, Japan, Singapore,

UK, and USA, and institutional partnerships with 221 counterpart organisations in 90

countries, CII serves as a reference point for Indian industry and the international

business community. www.cii.in

The Associated Chambers of Commerce and Industry of India (ASSOCHAM), India's

premier apex chamber covers a membership of over 2 lakh companies and professionals

across the country. It was established in 1920 by promoter chambers, representing all

regions of India. As an apex industry body, ASSOCHAM represents the interests of

industry and trade, interfaces with Government on policy issues and interacts with

counterpart international organizations to promote bilateral economic issues.

ASSOCHAM is represented on all national and local bodies and is, thus, able to pro-

actively convey industry viewpoints, as also communicate and debate issues relating to

public-private partnerships for economic development. www.assocham.org

FICCI: Established in 1927, FICCI is the largest and oldest apex business organisation in

India. FICCI plays a leading role in policy debates that are at the forefront of social,

economic and political change. Its publications are widely read for their in-depth research

and policy prescriptions. FICCI works closely with the government on policy issues,

enhancing efficiency, competitiveness and expanding business opportunities for industry

through a range of specialised services and global linkages. It also provides a platform for

sector specific consensus building and networking. www.ficci.com

Job opportunities in Telecom Sector

Government sector: Every year UPSC conducts Indian Engineering Services exam for

recruitment to fill up vacancies notified by various departments such as Broadcasting,

Military Engineering Service, Indian Telecom Service, Indian Railways, Wireless

Planningetc. Numbers of vacancies vary year to year.

Entry level engineers with Telecom Operators: All operators recruit thousands on fresh

engineers every year owing to the high growth in telecom market. BSNL recruits of the

order of thousand fresh graduates every year at Junior Telecom Officer level.

Sales Engineers: Many Telecom solutions are very sophisticated and technical. Such

sales need to be handled by telecom engineers.

39

Manufacturing Sector: Most of the MNCs have set up factories in India for

manufacturing telecom network equipment as well as Customer premises equipment.

There is enough job potential with these firms.

Support jobs in Non-Telecom sector: In today’s scenario, all industries use many

telecom facilities for faster and efficient communication. All such activities require

maintenance professionals. Even in medical sector, growing use of telemedicine has

created a new marketfor telecom professionals.

Research & Development: Many MNCs have outsourced R & D in telecom to Indian

firms.

For example Nokia has outsourced its product design to M/s TCS. All such deals create

jobopportunities for telecom engineers.

IT sector: The core of BPO sector is the telecom network. IT sectors generates huge

telecomjobs.

Education sector: Government of India’s mission mode project on Education such as

Sarva shiksha Abhiyan, connecting all libraries in India, providing broadband to all

schools etc. requires telecom professionals to install and manage this huge network.

National E-Governance Project: The ambitions plan of India to network each nook &

corner of the country and provide a citizen centric, single window service counter

requires creation of vast telecom network across the country. Each State is implementing

State Wide Area Network (SWAN). All such projects create demand for telecom

professionals.

Research executives with Consultancy Firms: Telecom growth impacts a country’s

economy.

Many consultancy firms thrive on generating reports on business models, future potential

andextending guidance to existing and new entrants in telecom market. There is a

significant need for telecom professionals with such firms also.

Pay range: Entry level engineer can get a starting annual package ranging from 2-4 lakh

depending on the nature of job & employer firm.

40

Chapter-3

PCM PRINCIPLES

INTRODUCTION

A long distance or local telephone conversation between two persons could be provided

by using a pair of open wire lines or underground cable as early as early as mid of 19th

century. However, due to fast industrial development and an increased telephone

awareness, demand for trunk and local traffic went on increasing at a rapid rate. To cater

to the increased demand of traffic between two stations or between two subscribers at the

same station we resorted to the use of an increased number of pairs on either the open

wire alignment, or in underground cable. This could solve the problem for some time

only as there is a limit to the number of open wire pairs that can be installed on one

alignment due to headway consideration and maintenance

problems. Similarly increasing the number of open wire pairs that can be installed on one

alignment due to headway consideration and maintenance problems. Similarly increasing

the number of pairs to the underground cable is uneconomical and leads to

maintenance problems.

It, therefore, became imperative to think of new technical innovations which could

exploit the available bandwidth of transmission media such as open wire lines or

underground cables to provide more number of circuits on one pair. The technique used

to provide a number of circuits using a single transmission link is called Multiplexing.

MULTIPLEXING TECHNIQUES

There are basically two types of multiplexing techniques

i. Frequency Division Multiplexing (FDM)

ii Time Division Multiplexing (TDM)

Frequency Division Multiplexing Techniques (FDM)

The FDM techniques is the process of translating individual speech circuits (300-3400

Hz) into pre-assigned frequency slots within the bandwidth of the transmission medium.

The frequency translation is done by amplitude modulation of the audio frequency with

an appropriate carrier frequency. At the output of the modulator a filter network is

connected to select either a lower or an upper side band. Since the intelligence is carried

in either side band, single side band suppressed carrier mode of AM is used. This results

41

in substantial saving of bandwidth mid also permits the use of low power amplifiers.

Please refer Fig. 1.

FDM techniques usually find their application in analogue transmission systems. An

analogue transmission system is one which is used for transmitting continuously varying

signals.

Time Division Multiplexing

Basically, time division multiplexing involves nothing more than sharing

a transmission medium by a number of circuits in time domain by establishing a sequence

of time slots during which individual channels (circuits) can be transmitted. Thus the

entire bandwidth is periodically available to each channel. Normally all time slots1 are

equal in length. Each channel is assigned a time slot with a specific common repetition

period called a frame interval. This is illustrated in Fig. 2.

Each channel is sampled at a specified rate and transmitted for a fixed

duration.All channels are sampled one by,thecycleisrepeatedagainand again. The

channels are connected to individual gates which are opened one by one in a fixed

sequence. At the receivingendalsosimilar gates are opened in unision with the gates at the

transmitting end.

The signal received at the receiving end will be in the form of discrete

samples and these are combined to reproduce the original signal. Thus, at a given instant

of time, onty one channel is transmitted through the medium, and by sequential sampling

42

a number of channels can be staggered in time as opposed to transmitting all the channel

at the same time as in EDM systems. This staggering of channels in time sequence for

transmission over a common medium is called Time Division Multiplexing (TDM).

PULSE CODE MODULATION SYSTEM

It was only in 1938, Mr. A.M. Reaves (USA) developed a Pulse Code Modulation (PCM)

system to transmit the spoken word in digital form. Since then digital speech transmission

has become an alternative to the analogue systems.

PCM systems use TDM technique to provide a number of circuits on the same

transmission medium viz open wire or underground cable pair or a channel provided by

carrier, coaxial, microwave or satellite system.

Basic Requirements For PCM System

To develop a PCM signal from several analogue signals, the following processing steps

are required

• Filtering

• Sampling

43

• Quantisation

• Encoding

• Line Coding

FILTERING

Filters are used to limit the speech signal to the frequency band 300-3400 Hz.

SAMPLING

It is the most basic requirement for TDM. Suppose we have an analogue signal Fig. 3 (b),

which is applied across a resistor R through a switch S as shown in Fig. 3 (a) . Whenever

switch S is closed, an output appears across R. The rate at which S is closed is called the

sampling frequency because during the make periods of S, the samples of the analogue

modulating signal appear across R. Fig. 3(d) is a stream of samples of the input signal

which appear across R. The amplitude of the sample is depend upon the amplitude of the

input signal at the instant of sampling. The duration of these sampled pulses is equal to

the duration for which the switch S is closed. Minimum number of samples are to be sent

for any band limited signal to get a good approximation of the original analogue signal

and the same is defined by the sampling Theorem.

FIG. 3 : SAMPLING PROCESS

44

Sampling Theorem

A complex signal such as human speech has a wide range of frequency

components with the amplitude of the signal being different at different frequencies. To

put it in a different way, a complex signal will have certain amplitudes for all frequency

components of which the signal is made. Let us say that these frequency components

occupy a certain bandwidth B. If a signal does not have any value beyond this bandwidth

B, then it is said to be band limited. The extent of B is determined by the highest

frequency components of the signal.

Sampling Theorem States

" If a band limited signal is sampled at regular intervals of time and at a rate equal to or

more than twice the highest signal frequency in the band, then the sample contains all the

information of the original signal." Mathematically, if fH is the highest frequency in the

signal to be sampled then the sampling frequency Fs needs to be greater than 2 fH.

i.e. Fs>2fH Let us say our voice signals are band limited to 4 KHz and let sampling

frequency be 8 KHz.

Time period of sampling Ts = 1 sec

8000 or Ts = 125 micro seconds If we have just one channel, then this can be sampled every 125 microseconds and

the resultant samples will represent the original signal. But, if we are to sample N

channels one by one at the rate specified by the sampling theorem, then the time available

for sampling each channel would be equal to Ts/N microseconds.

Fig. .4 shows how a number of channels can be sampled and combined.

The channel gates (a, b ... n) correspond to the switch S in Fig. 3. These gates are opened

by a series of pulses called "Clock pulses". These are called gates because, when closed

these actually connect the channels to the transmission medium during the clock period

and isolate them during the OFF periods of the clock pulses. The clock pulses are

staggered so that only one pair of gates is open at any given instant and, therefore, only

one channel is connected to the transmission medium. The time intervals during which

the common transmission medium is allocated to a particular channel is called the Time

Slot for that channel. The width of.this time slot will depend, as stated above, upon the

number of channels to be combined and the clock pulse frequency i.e. the sampling

frequency.

45

FIG. 4: SAMPLING & COMBINING CHANNELS

In a 30 channel PCM system. TS i.e. 125 microseconds are divided into

32 parts. That is 30 time slots are used for 30 speech signals, one time

slot for signalling of all the 30 chls, and one time slot for

synchronization between Transmitter & Receiver.

The time available per channel would be Ts/N = 125/32

= 3.9 microseconds

Thus in a 30 channel PCM system, time slot is 3.9 microseconds and time period

of sampling i.e..the interval between 2 consecutive samples of a channel is 125

microseconds. This duration i.e. 125 microseconds is called Time Frame.

The signals on the common medium (also called the common highway)

of a TDM system will consist of a series of pulses, the amplitudes of

which are proportional to the amplitudes of the individual channels at

their respective sampling instants. This is illustrated in Fig. 5

46

i

FIG 5 : PAM OUTPUT SIGNALS

The original signal for each channel can be recovered at the receive end by

applying gate pulses at appropriate instants and passing the signals through low pass

filters. (Refer Fig. 6)

Fig. 6 : RECONSTRUCTION OF ORIGINAL SIGNAL

QUANTISATION

In FDM systems we convey the speech signals in their analogue electrical form. But

in PCM, we convey the speech in discrete form. The sampler selects a number of points

on the analogue speech signal (by sampling process) and measures their instant values.

The output of the sampler is a PAM signal as shown in Fig. 3; The transmission of PAM

signal will require linear amplifiers at trans and receive ends to recover distortion less

47

signals. This type of transmission is susceptible to all the disadvantages of AM signal

transmission. Therefore, in PCM systems, PAM signals are converted into digital form by

using Quantization Principles. The discrete level of each sampled signal is quantified

with reference to a certain specified level on an amplitude scale.

The process of measuring the numerical values of the samples and giving them a

table value in a suitable scale is called "Quantising". Of course, the scales and the number

of points should be so chosen that the signal could be effectively reconstructed after

demodulation.

Quantising, in other words, can be defined as a process of breaking down a

continuous amplitude range into a finite number of amplitude values or steps.

A sampled signal exists only at discrete times but its amplitude is drawn from a

continuous range of amplitudes of an analogue signal. On this basis, an infinite number of

amplitude values is possible. A suitable finite number of discrete values can be used to

get an. approximation of the infinite set. The discrete value of a sample is measured by

comparing it with a scale having a finite number of intervals and identifying the interval

in which the sample falls. The finite number of amplitude intervals is called the

"quantizing interval". Thus, quantizing means to divide the analogue signal's total

amplitude range into a number of quantizing intervals and assigning a level to each

intervals.

For example, a 1 volt signal can be divided into 10mV ranges like 10-20mV, 30-

40mV and so on. The interval 10-20 mV, may be designated as level 1, 20-30 mV as

level 2 etc. For the purpose of transmission, these levels are given a binary code. This is

called encoding. In practical systems-quantizing and encoding are a combined process.

For the sake of understanding, these are treated separately.

Quantizing Process

Suppose we have a signal as shown in Fig. 7 which is sampled at instants a, b, c, d

and e. For the sake of explanation, let us suppose that the signal has maximum amplitude

of 7 volts.

In order to quantize these five samples taken of the signal, let us say the total amplitude is

divided into eight ranges or intervals as shown in Fig. 7. Sample (a) lies in the 5th range.

Accordingly, the quantizing process will assign a binary code corresponding to this i.e.

48

101, Similarly codes are assigned for other samples also. Here the quantizing intervals

are of the same size. This is called Linear Quantizing.

FIG. 7 : QUANTIZING-POSITIVE SIGNAL

Assigning an interval of 5 for sample 1, 7 for 2 etc. is the quantizing

process.Giving,the assigned levels of samples,th ebinary codeis called coding of the

quantized samples.

Quantizing is done for both positive and negative swings. As shown in

Fig.6,eight quantizing levels are used for each direction of the analogue signal. To

indicate whether a sample is negative with reference to zero or is positive with reference

zero, an extradigitisadded to the binary code. This extra digit is called the "sign bit". In

Fig.8. positive values have a sign bit of '1' and negative values have sign

bit of'0'.

49

FIG. 8 : QUANTIZING - SIGNAL WITH + Ve & - Ve VALUES Relation between Binary Codes and Number of levels. Because the quantized samples are coded in binary form, the quantization

intervals will be in powers of 2. If we have a 4 bit code, then we can have 2" = 16 levels.

Practical PCM systems use an eight bit code with the first bit as sign bit. It means we can

have 2" = 256 (128 levels in the positive direction and 128 levels in the negative

direction) intervals for quantizing.

Quantization Distortion

Practically in quantization we assign lower value of each interval to a sample falling in

any particular interval and this value is given as

50

Table-1 : Illustration of Quantization Distortion

Analogue Signal

Amplitude Range

Quantizing Interval

(mid value)

Quantizing Level Binary Code

0-10 mv 5 mv 0 1000

10-20mv 15mv 1 1001

20-30 mv 25 mv 2 1010

30-40 mv 35 mv 3 1011

40-50 mv 45 mv 4 1100

If a sample has an amplitude of say 23 mv or 28 mv, in either case it will be

assigned \he \eve\ "2". This Is represented in binary code 1010. When this is decoded at

the receiving end, the decoder circuit on receiving a 1010 code will convert this into an

analogue signal of amplitude 25 mv only. Thus the process' of quantization leads to an

approximation of the input signal with the detected signal having some deviations in

amplitude from the actual values. This deviation between the amplitude of samples at the

transmitter and receiving ends (i.e. the difference between the actual value & the

reconstructed value) gives rise to quantization distortion.

If V represent the step size and 'e' represents the difference in amplitude

fe' must exists between - V/2 & + V/2) between the actual signal level and its quantized

equivalent then it can be proved that mean square quantizing error is equal to (V2). Thus,

we see that the error depends upon the size of the step. 12

In linear quantization, equal step means equal degree of error for all input

amplitudes. In other words, the signal to noise ratio for weaker signals will be poorer.

To reduce error, we, therefore, need to reduce step size or in other words,

increase th,e number of steps in the given amplitude range. This would however,

increase the transmission bandwidth because bandwidth B = fm log L. where L is

the number of quantum steps and fm is the highest signal frequency. But as we knows

from speech statistics that the probability of occurrence of a small amplitude is much

greater than large one, it seems appropriate to provide more quantum levels (V = low

value) in the small amplitude region and only a few (V = high value) in the region of

higher amplitudes. In this case, provided the total number of specified levels remains

51

unchanged, no increase in transmission bandwidth will be required. This will also try to

bring about uniformity in signal to noise ratio at all levels of input signal. This type of

quantization is called non-uniform quantization.

In practice, non-uniform quantization is achieved using segmented quantization

(also called companding). This is shown in Fig. 9 (a). In fact, there are equal number of

segments for both positive and negative excursions. In order to specify the location of a

sample value it is necessary to know the following :

1. The sign of the sample (positive or negative excursion)

2. The segment number

3. The quantum level within the segment

As seen in Fig. 9 (b), the first two segment in each polarity are collinear, (i.e. the slope is

the same in the central region) they are considered as one segment. Thus the total number

of segment appear to be 13. However, for purpose of analysis all the 16 segments will be

taken into account.

ENCODING

Conversion of quantised analogue levels to binary signal is called encoding. To

represent 256 steps, 8 level code is required. The eight bit code is also called an eight bit

"word".

52

The 8 bit word appears in the form

P ABC WXYZ

Polarity bit ‘1’ Segment Code Linear encoding

for + ve 'O' for - ve. in the segment

The first bit gives the sign of the voltage to be coded. Next 3 bits gives the segment

number. There are 8 segments for the positive voltages and 8 for negative voltages. Last

4 bits give the position in the segment. Each segment contains 16 positions.

Referring to Fig. 9(b), voltage Vc will be encoded as 1 111 0101.

FIG. 9 (b) : ENCODING CURVE WITH COMPRESSION 8 BIT CODE

The quantization and encoding are done by a circuit called coder. The coder

converts PAM signals (i.e. after sampling) into a 8 bit binary signal. The coding is done

as per Fig. 9 which shows a relationship between voltage V to be coded and equivalent

binary number N. The function N = f(v) is not linear.

The curve has the following characteristics.

53

It is symmetrical about the origins. Zero level corresponds to zero voltage to be encoded.

It is logarithmatic function approximated by 13 straight segments numbered 0 to 7 in

positive direction and 'O' to 7 in the negative direction. However 4 segments 0, 1, 0, 1

lying between levels + vm/64 -vm/64 being colinear are taken as one segment.

The voltage to be encoded corresponding to 2 ends of successive segments are in the

ratio of 2. That is vm, vm/2, vm/4, vm/8, vm/16, vm/32, vm/64, vm/128 (vm being the

maximum voltage).

There are 128 quantification levels in the positive part of the curve and 128 in the

negative part of the curve.

In a PCM system the channels are sampled one by one by applying the sampling

pulsqs to the sampling gates. Refer Fig. 10. The gates open only when a pulse is applied

to them and pass the analogue signals through them for the duration for which the gates

remain open. Since only one gate will be activated at a given instant, a common encoding

circuit is used for all channels. Here the samples are quantized and encoded. The encoded

samples of all the channels and signals etc are combined in the digital combiner and

transmitted.

54

The reverse process is carried out at the receiving end to retreive the original

analogue signals. The digital combiner combines the encoded samples in the form of

"frames". The digital separator decombines the incoming digital streams into individual

frames. These frames are decoded to give the PAM (Pulse Amplitude Modulated)

samples. The samples corresponding to individual channels are separated by operating

the receive sample gates in the same sequence i.e. in synchronism with the transmit

sample gates.

55

CONCEPT OF FRAME

In Fig. 10, the sampling pulse has a repetition rate of Ts sees and a pulse width of

"St". When a sampling pulse arrives, the sampling gate remains opened during the time

"St" and remains closed till the next pulse arrives. It means that a channel is activated for

the duration "St". This duration, which is the width of the sampling puse, is called the

"time slot" for a given channel.

Since Ts is much larger as compared to St. a number of channels can be sampled

each for a duration of St within the time Ts. With reference to Fig. 10, the first sample of

the first channel is taken by pulse 'a', encoded and is passed on the combiner. Then the

first sample of the second channel is taken by pulse 'b' which is also encoded and passed

on to the combiner, Likewise the remaining channels are also sampled sequentially and

are encoded before being fed to the combiner. After the first sample of the Nth channel is

taken and processed, the second sample of the first channel is taken, this process is

repeated for all channels. One full set of samples for all channel taken within the duration

Ts is called a "frame". Thus the set of all first samples of all channels is one frame; the

set of all second samples is another frame and so on.

As already said in para 5.3.5, Ts in a 30 channel PCM system is 125

microseconds and the signalling information of all the channels is transmitted through a

separate time slot. To maintain synchronization between transmit and receive ends,

the synchronization data is transmitted through another time slot. Thus for a 30 chl

PCM system, we have 32 time slots.

Thus the time available per channel would be 3.9 microsecs.

Thus for a 30 chl PCM system,

Frame = 125 microseconds

Time slot per chl = 3.9 microseconds.

Structure of Frame

A frame of 125 microseconds duration has 32 time slots. These slots are

numbered Ts 0 to Ts 31.

56

Information for providing synchronization between trans and receive ends is passed

through a separate time slot. Usually the slot Ts 0 caries the synchronizsation signals.

This slot is also called Frame alignment word (FAW).

The signalling informatiori is transmitted through time slot Ts 16.

Ts 1 to Ts 15 are utilized for voltage signal of channels 1 to 15 respectively.

Ts 17 to Ts 31 are utilized for voltage signal of channels 16 to 30 respectively.

SYNCHRONIZATION

The output of a PCM terminal will be a continuous stream of bits. At the

receiving end, the receiver has to receive the incoming stream of bits and discriminate

between frames and separate channels from these. That is, the receiver has to recognise

the start of each frame correctly. This operation is called frame alignment or

Synchronization and is achieved by inserting a fixed digital pattern called a "Frame

Alignment Word (FAW)" into the transmitted bit stream at regular intervals. The receiver

looks for FAW and once it is detected, it knows that in next time slot, information for

channel one will be there and so on.

The digits or bits of FAW occupy seven out of eight bits of Ts 0 in the following

pattern.

Bit position of Ts 0 B1 B2 B3 B4 B5 B6 B7 B8

FAW digit value X 0 0 1 1 0 1 1

The bit position B1 can be either '1' or '0'. However, when the PCM system is to

be linked to an international network, the B1 position is fixed at '1'.

The FAW is transmitted in the Ts O of every alternate frame.

Frame which do not contain the FAW, are used for transmitting supervisory and

alarm signals.

To distinguish the Ts 0 of frame carrying supervisory/alarm signals from those carrying

the FAW, the B2 bit position of the former are fixed at T. The FAW and alarm signals are

transmitted alternatively as shown in Table - 2.

57

TABLE-2

Frame Remark

Numbers B1 B2 B3 B4 B5 B6 B7 B8

FO X 0 0 1 1 0 1 1 FAW

F1 X 1 Y Y Y 1 1 1 ALARM

F2 X 0 0 1 1 0 1 1 FAW

F3 etc X 1 Y Y Y 1 1 1 ALARM

In frames 1, 3, 5, etc, the bits B3, B4, B5 denote various types of alarms. For example, in

B3 position, if Y = 1, it indicate Frame synchronisation alarm. If Y = 1 in B4, it indicates

high error density alarm. When there is no alarm condition, bits B3 B4 B5 are set 0. An

urgent alarm is indicated by transmitting "all ones". The code word for an urgent alarm

would be of the form.

X 111 1111 SIGNALLING IN PCM SYSTEMS

In a telephone network,-the signalling information is used for proper routing of

a call between two subscribers, for providing certain status information like dial tone,

busy tone, ring back. NU tone, metering pulses, trunk offering signal etc. All these

functions are grouped under the general terms "signalling" in PCM systems. The

signaling information can be transmitted in the form of DC pulses (as in step by step

exchange) or multifrequency pulses (as in cross bar systems) etc.

The signalling pulses retain their amplitude for a much longer period than

the pulses carrying speech information. It means that the signalling information

is a slow varying signal in time compared to the speech signal which is fast changing in

the time domain. Therefore, a signalling channel can be digitized with less number of bits

than a voice channel.

In a 30 chl PCM system, time slot Ts 16 in each frame is allocated for

carrying signalling information.

The time slot 16 of each frame carries the signalling data

corresponding to two VF channels only. Therefore, to cater for 30 channels, we must

transmit 15 frames, each having 125 microseconds duration. For carrying

synchronization data for all frames, one additional frame is used. Thus a group of

58

16 frames (each of 125 microseconds) is formed to make a "multiframe". The duration of

a multiframe is 2 milliseconds. The multiframe has 16 major time slots of 125

microseconds duration. Each of these (slots) frames has 32 time slots carrying, the

encoded samples of all channels plus the signaling and synchronization data. Each

sample has eight bits of duration 0.400 microseconds (3.9/8 = 0.488) each. The

relationship between the bit duration frame and multiframe is illustrated in Fig. 11 (a) &

11 (b).

59

FIG. 11 (B) 2.048 Mb/s PCM MULTIFRAME

We have 32 time slots in a frame, each slot carries an 8 bit word.

The total number of bits per frame = 32 x 8 = 256

The total number of frames per seconds is 8000

The total number of bits per second are 256 x 8000 = 2048 K/bits.

Thus, a 30 chP PCM system has 2048 K bits.

Multiframe Structure

In the time slot 16 of FO, the first four bits (positions 1 to 4) contain the

multiframe alignment signal which enables the receiver to identify a multiframe.

The other four bits (no. 5 to 8) are spare. These may be used for carrying alarm signals.

Time slots 16 of frames F1 to FT5 are used for carrying the signalling information. Each

frame carries signalling, data for two VF channels. For instance, time slot Ts 16 of frame

F1 carries the signal data for VF channel 1 in the first four bits. The next four bits are

used for carrying signalling information

for channel 16. Similarly, time slot Ts16 of F2 carries signalling data of chls 2 .and 17.

Thus in multiframe structure, four signalling bits are provided for each VF channels.

As each multiframe includes 16 frames, each with a sacnqtoq -per sec.,.the.signalling of

each channel will occur at a rate of 500 per sec.

60

CHAPTER - 4

DIGITAL SWITCHING

Introduction

A Digital switching system, in general, is one in which signals are switched in

digital form. These signals may represent speech or data. The digital signals of several

speech samples are time multiplexed on a common media before being switched through

the system.

To connect any two subscribers, it is necessary to interconnect the time-slots of the two

speech samples which may be on same or different PCM highways. The digitalised

speech samples are switched in two modes, viz., Time Switching and Space Switching.

This Time Division Multiplex Digital Switching System is popularly known as Digital

Switching System.

In this handout, general principles of time and space switching are discussed. A

practical digital switch, comprising of both time and space stages, is also explained.

Time and Space Switching

Generally, a digital switching system several time division

multiplexed (PCM) samples. These PCM samples are conveyed on PCM highways (the

common path over which many channels can pass with separation achieved by time

division.). Switching of calls in this environment , requires placing digital samples from

one time-slot of a PCM multiplex in the same or different time-slot of another PAM

multiplex.

For example, PCM samples appearing in TS6 of I/C PCM HWY1 are transferred to TS18

of O/G PCM HWY2, via the digital switch, as shown in Fig1.

FIG 1 DIGITAL SWITCH

61

The interconnection of time-slots, i.e., switching of digital signals can be achieved using

two different modes of operation. These modes are: -

I. Space Switching

ii. Time switching

Usually, a combination of both the modes is used.

In the space-switching mode, corresponding time-slots of I/C and O/G PCM

highways are interconnected. A sample, in a given time-slot, TSi of an I/C HWY, say

HWY1, is switched to same time-slot, TSi of an O/G HWY, SAY HWY2. Obviously

there is no delay in switching of the sample from one highway to another highway since

the sample transfer takes place in the same time-slot of the PCM frame.

Time Switching, on the other hand, involves the interconnection of different time-

slots on the incoming and outgoing highways by re-assigning the channel sequence. For

example, a time-slot TSx of an I/C Highway can be connected to a different time-slot.,

TSy, of the outgoing highway. In other words, a time switch is, basically, a time-slot

changer.

Digital Space Switching

Principle

The Digital Space Switch consists of several input highways, X1, X2,...Xn and

several output highways, Y1, Y2,.............Ym, inter connected by a crosspoint matrix of

n rows and m columns. The individual crosspoint consists of electronic AND gates. The

operation of an appropriate crosspoint connects any channel, a , of I/C PCM highway to

the same channel, a, of O/G PCM highway, during each appropriate time-slot which

occurs once per frame as shown in Fig 2. During other time-slots, the same crosspoint

may be used to connect other channels. This crosspoint matrix works as a normal space

divided matrix with full availability between incoming and outgoing highways during

each time-slot.

Each crosspoint column, associated with one O/G highway, is assigned a column

of control memory. The control memory has as many words as there are time-slot per

frame in the PCM signal. In practice, this number could range from 32 to 1024. Each

crosspoint in the column is assigned a binary address, so that only one crosspoint per

column is closed during each time-slot. The binary addresses are stored in the control

memory, in the order of time-slots. The word size of the control memory is x bits, so that

2x = n, where n is the number of cross points in each column .

62

A new word is read from the control memory during each time-slot, in a

cyclic order. Each word is read during its corresponding time-slot, i.e.,Word 0

(corresponding to TS0), followed by word 1 (corresponding to TS1) and so on.

The word contents are contained on the vertical address lines for the duration of

the time-slot. Thus, the cross point corresponding to the address, is operated

63

during a particular time-slot. This cross point operates every time the particular

time-slot appears at the inlet in successive frames. normally, a call may last for

around a million frames.

As the next time-slot follows, the control memory is also advanced by one

step, so that during each new time-slot new corresponding words are read from

the various control memory columns. This results in operation of a completely

different set of cross points being activated in different columns. Depending upon

the number of time-slots in one frame, this time division action increases the

utilisation of cross point 32 to 1024 times compared with that of conventional

space-divided switch matrix.

Illustration

Consider the transfer of a sample arriving in TS7 of I/C HWY X1 to O/G

HWY Y3. Since this is a space switch, there will be no reordering of time i.e., the

sample will be transferred without any time delay, via the appropriate cross point.

In other words, the objective is to connect TS7 of HWY X1 and TS7 of HWY Y3.

The central control (CC) selects the control memory column

corresponding output highway Y3. In this column, the memory location

corresponding to the TS7 is chosen. The address of the cross point is written in

this location, i.e., 1, in binary, is written in location 7, as shown in fig 2.This cross

point remains operated for the duration of the time-slot TS7, in each successive

frame till the call lasts.

For disconnection of call, the CC erases the contents of the control memory

locations, corresponding to the concerned time-slots. The AND gates, therefore,

are disabled and transfer of samples is halted.

Practical Space Switch

In a practical switch, the digital bits are transmitted in parallel rather than

serially, through the switching matrix.

In a serial 32 time-slots PCM multiplex, 2048 Kb/s are carried on a single wire

sequentially, i.e., all the bits of the various time-slots follow one another. This

single wire stream of bits, when fed to Serial to Parallel Converter is converted

into 8-wire parallel output. For example, all 8 bits corresponding to TS3 serial

input are available simultaneously on eight output wires (one bit on each output

wire), during just one bit period, as shown in fig.3. This parallel output on the

eight wires is fed to the switching matrix. It can be seen that during one full time-

slot period, only one bit is carried on the each output line, whereas 8 bits are

64

carried on the input line during this period. Therefore, bit rate on individual

output wires, is reduced to 1/8th of input bit rate=2048/8=256Kb/s

Due to reduced bit rate in parallel mode, the cross point is required to be

operated only for 1/8th of the time required for serial working. It can, thus, be

shared by eight times more channels, i.e., 32 x 8 = 256 channels, in the same

frame.

However, since the eight bits of one TS are carried on eight wires, each

cross point have eight switches to interconnect eight input wires to eight output

wires. Each cross point (all the eight switches) will remain operated now for the

duration of one bit only, i.e., only for 488 ns (1/8th of the TS period of 3.9 µs)

Fig 3 Serial parallel converter

For example, to connect 40 PCM I/C highways, a matrix of 40x 40 = 1600

cross points each having a single switch, is required in serial mode working.

Whereas in parallel mode working, a matrix of (40/8 x 40/8) = 25 cross point is

sufficient. As eight switches are required at each cross point 25 x 8 = 200

switches only are required. Thus, there is a reduction of the matrix by 1/8th in

parallel mode working, hence reduction in size and cost of the switching matrix.

65

Digital Time Switch

Principle

A Digital Time Switch consists of two memories, viz., a speech or buffer

memory to store the samples till destination time-slots arrive, and a control or

connection or address memory to control the writing and reading of the samples

in the buffer memory and directing them on to the appropriate time-slots.

Speech memory has as many storage locations as the number of time-slots

in input PCM, e.g., 32 locations for 32 channel PCM system.

The writing/reading operations in the speech memory are controlled by the

Control Memory. It has same number of memory locations as for speech memory,

i.e., 32 locations for 32 channel PCM system. Each location contains the address

of one of the speech memory locations where the channel sample is either written

or read during a time-slot. These addresses are written in the control memory of

the CC of the exchange, depending upon the connection objective.

A Time-Slot Counter which usually is a synchronous binary counter, is

used to count the time-slots from 0 to 31, as they occur. At the end of each frame,

It gets reset and the counting starts again. It is used to control the timing for

writing/reading of the samples in the speech memory.

Illustration

Consider the objective that TS4 of incoming PCM is to be connected to

TS6 of outgoing PCM. In other words, the sample arriving in TS4 on the I/C

PCM has to be delayed by 6 - 4 = 2 time-slots, till the destination time-slot, viz.,

TS6 appears in the O/G PCM. The required delay is given to the samples by

storing it in the speech memory. The I/C PCM samples are written cyclically i.e.

sequentially time-slot wise , in the speech memory locations. Thus, the sample in

TS4 will be written in location 4, as shown in fig.4.

The reading of the sample is controlled by the Control Memory. The

Control Memory location corresponding to output time-slot TS6, is 6. In this

location, the CC writes the input time-slot number, viz.,4, in binary. These

contents give the read address for the speech memory, i.e., it indicates the speech

memory locations from which the sample is to be read out, during read cycle.

When the time-slot TS6 arrives, the control memory location 6 is read. Its

content addresses the location 4 of the speech memory in the read mode and

sample is read on to the O/G PCM.

66

In every frame, whenever time-slot 4 comes a new sample will be written in

location 4. This will be read when TS6 occurs. This process is repeated till the

call lasts.

For disconnection of the call, the CC erases the contents of the control

memory location to halt further transfer of samples.

Time switch can operate in two modes, viz.,

I. Output associated control

ii. Input associated control

Output associated control

In this mode of working , 2 samples of I/C PCM are written cyclically in the speech memory locations in the order of time-slots of I/C PCM, i.e., TS1 is written in location 1, TS2 is written in location 2, and so on, as discussed in the example of Sec.4.2. The contents of speech memory are read on output PCM in the order specified by control memory. Each location of control memory is rigidly associated with the corresponding time-slot of the O/G PCM and contains the address of the TS of incoming PCM to be connected to. The control memory is always read cyclically, in synchronism with the occurrence of the time-slot. The entire process of writing and reading is repeated in every frame, till the call is disconnected.

FIG 4 OUTPUT ASSOCIATED CONTROL SWITCH

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It may be noticed that the writing in the speech memory is sequential and independent of the control memory, while reading is controlled by the control memory, i.e., there is a sequential writing but controlled reading.

Input associated control Here, the samples of I/C PCM are written in a controlled way, i.e., in the order specified by control memory, and read sequentially. Each location of control memory is rigidly associated with the corresponding TS of I/C PCM and contains the address of TS of O/G PCM to be connected to. The previous example with the same connection objective of connecting TS4 of I/C PCM to TS6 of O/G PCM may be considered for its restoration. The location 4 of the control memory is associated with incoming PCM TS4. Hence, it should contain the address of the location where the contents of TS4 of I/C PCM are to be written in speech memory. A CC writes the number of the destination TS, viz., 6 in this case, in location 4 of the control memory. The contents of TS4 are therefore, written in location of speech memory, as shown in fig5. The contents of speech memory are read in the O/G PCM in a sequential way, i.e., location 1 is read during TS1, location 2 is read during TS2, and so on. In this case, the contents of location 6 will appear in the output PCM at TS6. Thus the input PCM TS4 is switched to output PCM TS6. In this switch, there is sequential reading but controlled writing.

FIG 5 INPUT ASSOCIATED CONTROLLED TIME SWITCH

68

Time Delay Switching The writing and reading, of all time-slots in a frame, has to be completed within one frame time period (before the start of the next frame). A TS of incoming PCM may, therefore, get delayed by a time period ranging from 1 TS to 31 TS periods, before being transmitted on outgoing PCM. For example, consider a case when TS6 of incoming PCM is to be switched to TS5 in outgoing PCM. In this case switching can be completed in two consecutive frames only, i.e., 121 microseconds for a 32 channel PCM system. However, this delay is imperceptable to human beings.

Non-Blocking feature of a Time Switch In a Time Switch, there are as many memory locations in the control and speech memories as there are time-slots in the incoming and outgoing PCM highways, i.e., corresponding to each time-slot in incoming highway, there is a definite memory location available in the speech and control memories. Similarly, corresponding to each time-slot in the outgoing highway there is a definite memory location available in the control and speech memories. This way, corresponding to free incoming and outgoing time-slots, there is always a free path available to interconnect them. In other words, there is no blocking in a time switch.

Two Dimensional Switching Though the electronic cross points are not so expensive, the cost of accessing and selecting them from external pins in a Space Switch, becomes prohibitive as the switch size increases. Similarly, the memory location requirements rapidly go up as a Time Switch is expanded, making it uneconomical. Hence, it becomes necessary to employ a number of stages, using small switches as building blocks to build a large network. This would result in necessity of changing both the time-slot and highway in such a network. Hence, the network, usually, employs both types of switches viz., space switch and time switch, and. therefore, is known as two dimensional network. These networks can have various combinations of the two types of switches and are denoted as TS, STS, TSST,etc. Though to ensure full availability, it may be desirable to use only T stages. However, the networks having the architecture of TT, TTT, TTTT, etc., are uneconomical, considering the acceptability of tolerable limits of blocking, in a practical network. Similarly, a two-stage two-dimensional network, TS or ST, is basically suitable for very low capacity networks only. The most commonly used architecture has three stages, viz., STS or TST. However, in certain cases, their derivatives, viz., TSST, TSSST, etc., may also be used.

69

An STS network has relatively simpler control requirements and hence, is still being favoured for low capacity networks, viz., PBX exchanges. As the blocking depends mainly on the outer stages, which are space stages, it becomes unsuitable for high capacity systems. A TST network has lesser blocking constraints as the outer stages are time stages which are essentially non-blocking and the space stage is relatively smaller. It is, therefore, most cost-effective for networks handling high traffic, However, for still higher traffic handling capacity networks, e.g., tandom exchanges, it may be desirable to use TSST or TSSST architecture. The choice of a particular architecture is dependent on other factors also, viz., implementation complexity, modularity, testability, expandability, etc. As a large number of factors favour TST structure, it is most widely used.

TST Network As the name suggests, in a TST network, there are two time stages separated by a space stage. The former carry out the function of time-slot changing, whereas the latter performs highway jumping. Let us consider a network having n input and n output PCM highways. Each of the input and output time stages will have n time switches and the space stage will consist of an n x n cross point matrix. The speech memory as well as the control memory of each time switch and each column of a control memory of the space switch will have m locations, corresponding to m time-slots in each PCM. Thus, it is possible to connect any TS in I/C PCM to any TS in O/G PCM. In the case of a local exchange, the network will be of folded type, i.e., the O/G PCM highways, via a suitable hybrid. Whereas, for a transit exchange, the network will be non-folded, having complete isolation of I/C and O/G PCM highways. However, a practical local exchange will have a combination of both types of networks. For the sake of explanation, let us assume that there are only four I/C and O/G PCM highways in the network. Hence, there will be only four time switches in each of the T-stages and the space switch will consist of 4x4 matrix. let us consider an objective of connecting two subscribers through this switching network of local exchange, assuming that the CC assigns TS4 on HWY0 to the calling party and TS6 on HWY3 to the called party The speech samples of the calling party have to be carried from TS4 of I/C HWY 0 and to TS6 of O/G HWY3 and those of the called party from TS6 of I/C HWY 3 to TS4 of O/G HWY 0 , with the help of the network. The CC establishes the path, through the network in three steps. To introduce greater flexibility, it uses an intermediate time-slot, TSx, which is also known as internal

70

time-slot. The three switching steps for transfer of speech sample of the calling party to the called party are as under: Step 1 Input Time Stage (IT) TS4 HWY0 to TSx HWY0 Step 2 Space stage (S)Tsx HWY0 to Tsx HWY3 Step 3 Output Time Stage (OT)Tsx HWY3 to TS6 HWY3 As the message can be conveyed only in one direction through this path, another independent path, to carry the massage in the other direction is also established by the CC, to complete the connection. Assuming the internal time-slots to be TS10 and TS11, the connection may be established as shown in fig 6.

FIG 6 T S T SWITCH

Let us now consider the detailed switching procedure making some more

assumptions for the sake of simplicity. Though practical time switches can handle

256 time-slots in parallel mode, let us assume serial working and that there are

only 32 time-slots in each PCM. Accordingly, the speech and control memories in

time switches and control memory columns in space switch, will contain 32

locations each.

To establish the connection, the CC searches for free internal time-

slots. Let us assume that the first available time-slots are TS10 and TS11, as

before. To reduce the complexity of control, the first time stage is designed as

output-controlled switch, whereas the second time stage is input-controlled.

71

FIG 7 T S T SWITCH STRUCTURE

For transfer of speech samples from the calling party to the called party of

previous example, CC orders writing of various addresses in location 10 of

control memories of IT-10, OT-3 and column 3 of CM-S of corresponding to O/G

highway, HWY3. Thus, 4 corresponding to I/C TS4 is written in CM-IT-0, 6

corresponding to O/G TS6 is written in CM-OT-3 and 0 corresponding to I/C

HWY 0 is written in column 3 of CM-S, as shown in fig. 7.

As the first time switch is output-controlled, the writing is done

sequentially. Hence, a sample, arriving in TS4 of I/C HWY 0, is stored in

location 4 of SM-IT-0. It is readout on internal HWY 0 during TS10 as per the

72

control address sent by CM-IT-0. In the space switch, during this internal TS10,

the cross point 0 in column 3 is enabled, as per the control address sent by column

3 of CM-S, thus, transferring the sample to HWY3. The second time stage is input

controlled and hence, the sample, arriving in TS10, is stored in location 6 of SM-

OT-3, as per the address sent by the CM-OT-3. This sample is finally, readout

during TS6 of the next frame, thus, achieving the connection objective.

Similarly, the speech samples in the other direction, i.e., from the called

party to the calling party, are transferred using internal TS11. As soon as the call

is over, the CC erases the contents in memory locations 10 and 11 of all the

concerned switches, to stop further transfer of message. These locations and time-

slots are, then, avialable to handle next call.

Switching Network Configuration of some Modern Switches

E10B - T-S-T

EWSD - T-S-S-S-T

AXE10 - T-S-T

CDOT(MBM) - T-S-T

5ESS - T-S-T

OCB 283 - T

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CHAPTER 5

SIGNALLING IN TELECOMMUNICATION

1 Introduction A telecommunication network establishes and realizes temporary connections, in

accordance with the instructions and information received from subscriber lines and inter

exchange trunks, in form of various signals. Therefore, it is necessary to interchange

information between an exchange and it external environment i.e. between subscriber

lines and exchange, and between different exchanges. Though these signals may differ

widely in their implementation they are collectively known as telephone signals.

A signalling system uses a language which enables two switching equipments to converse

for the purpose of setting up calls. Like any other language. it possesses a vocabulary of

varying size and varying precision, ie. a list of signals which may also vary in size and a

syntax in the form of a complex set of rules governing the assembly of these signals.This

handout discusses the growth of signalling and various type of signalling codes used in

Indian Telecommunication.

Telephony started with the invention of magneto telephone which used a magneto to

generate the ringing current, the only signal, sent over a dedicated line between two

subscribers. The need for more signals was felt with the advent of manual switching. Two

additional signals were, therefore, introduced to indicate call request and call release.

The range of signals increased further with the invention of electro-mechanical

automatic exchanges and is still growing further at a very fast pace, after the advent of

SPC electronic exchanges.

The interchange of signaling information can be illustrated with the help of a typical call

connection sequence. The circled number in Fig. 1 correspond to the steps listed below

i. A request for originating a call is initiated when the calling subscriber

lifts the handset.

ii. The exchange sends dial-tone to the calling subscriber to indicate to him to start

dialing.

iii. The called number is transmitted to the exchange, when the calling subscriber

dials the number.

iv. If the number is free, the exchange sends ringing current to him.

v. Feed-back is provided to the calling subscriber by the exchange by sending,

a) Ring-back tone, if the called subscriber is free(shown in fig.1)

b) Busy tone if the called subscriber is busy ( not shown in the figure), or

c)Recorded message, if provision exists, for non completion of call due to

some other constraint ( not shown in figure).

74

vi. The called subscriber indicates acceptance of the incoming call by lifting the

handset

vii. The exchange recognizing the acceptance terminates the ringing current and the

ring-back tone, and establishes a connection between the calling and called

subscribers.

viii. The connection is released when either subscriber replaces the handset.When the

called subscriber is in a different exchange, the following inter-exchange

trunk. signal functions are also involved, before the call can be set up.

ix The originating exchange seizes an idle inter exchange trunk, connected to a digit

register at the terminating exchange.

x. The originating exchange sends the digit. The steps iv to viii are then performed

to set up the call.

2 Types of Signalling

Subscriber Line signalling

Calling Subscriber Line Signaling

In automatic exchanges the power is fed over the subscriber’s loop by the centralized

battery at the exchange. Normally, it is 48 V. The power is fed irrespective of the

state of the subscriber, viz., idle, busy or talking.

Call request

When the subscriber is idle, the line impedance is high. The line impedance falls, as soon

as, the subscriber lifts the hand-set, resulting in increase of line current. This is

detected as a new call signal and the exchange after connecting an appropriate

equipment to receive the address information sends back dial-tone signal to the

subscriber.

Address signal

After the receipt of the dial tone signal, the subscriber proceeds to send the address digits.

The digits may be transmitted either by decade dialing or by multifrequency

pushbutton dialling.

1. Decadic Dialling

The address digits may be transmitted as a sequence of interruption of the DC loop by a

rotary dial or a decadic push-button key pad. The number of interruption (breaks)

indicate the digit, exept0, for which there are 10 interruptions. The rate of such

interruptions is 10 per second and the make/break ration is 1:2. There has to be a

inter-digital pause of a few hundred milliseconds to enable the exchange to

75

distinguish between consecutive digits. This method is, therefore, relatively slow

and signals cannot be transmitted during the speech phase.

2. Multifrequency Push-button Dialling

This method overcomes the constraints of the decadic dialling. It uses two sets of four

voice frequencies. pressing a button (key), generates a signal comprising of two

frequencies. one from each group. Hence, it is also called Dual-Tone Multi-

frequency (DTMF) dialling. The signal is transmitted as long as the key is kept

pressed. This provides 16 different combinations. As there are only 10 digits, at

present the highest frequency, viz., 1633 Hz, is not used and only 7 frequencies

are used, as shown in Fig.2.

By this method, the dialling time is reduced and almost 10 digits can be

transmitted per second. As frequencies used lie in the speech band, information

may be transmitted during the speech phase also, and hence, DTMF telephones

can be used as access teminals to a variety of systems, such as computers with

voice output. The tones have been so selected as to minimize harmonic

interference and probability of simulation by human voice.

�

�

FIGURE 2. TONE-DIALLING FREQUENCY GROUPS.

1

ABC 2

DEF 3

GHI 4

JKL 5

MNO 6

PRS 7

TUV 8

WXY 9

*

OPER 0

#

HIGH FREQUENCY GROUP

1209 Hz 1336 Hz

097 Hz 170 Hz 662 Hz

76

End of selection signal

The address receiver is disconnected after the receipt of complete address. After the

connection is established or if the attempt has failed the exchange sends any one of the

following signals.

1. Ring-back tone to the calling subscriber and ringing current to the called

subscriber, if the called line is free.

2. Busy-tone to the calling subscriber, if the called line is busy or otherwise

inaccessible.

3. Recorded announcement to the calling subscriber, if the provision exists, to

indicate reasons for call failure, other than called line busy.

Ring back, tone and ringing current are always transmitted from the called subscriber

local exchange and busy tone and recorded announcements, if any, by the equipment as

close to the calling subscriber as possible to avoid unnecessary busying of equipment and

trunks.

Answer Back Signal

As soon as the called subscriber lifts the handset, after ringing, a battery reversal signal

is transmitted on the line of the calling subscriber. This may be used to operate special

equipment attached to the calling subscriber, e.g., short-circuiting the transmitter of a

CCB, till a proper coin is inserted in the coin-slot.

Release signal

When the calling subscriber releases i.e., goes on hook, the line impedance goes high.

The exchange recognizing this signal, releases all equipment involved in the call. This

signal is normally of more than 500 milliseconds duration.

Permanent Line (PG) Signal

Permanent line or permanent glow (PG) signal is sent to the calling subscriber if he fails

to release the call even after the called subscriber has gone on-hook and the call is

released after a time delay. The PG signal may also be sent, in case the subscriber takes

too long to dial. It is normally busy tone.

Called subscriber line signals.

Ring Signal

On receipt of a call to the subscriber whose line is free, the terminating exchange sends

the ringing current to the called telephone. This is typically 25 or 50Hz with suitable

interruptions. Ring-back tone is also fed back to the calling subscriber by the terminating

exchange.

77

Answer Signal

When the called subscriber, lifts the hand-set on receipt of ring, the line impedance goes

low. This is detected by the exchange which cuts off the ringing current and ring-back

tone.

Release Signal

If after the speech phase, the called subscriber goes on hook before the calling subscriber,

the state of line impedance going high from a low value, is detected. The exchange sends

a permanent line signal to the calling subscriber and releases the call after a time delay, if

the calling subscriber fails to clear in the meantime.

Register Recall Signal

With the use of DTMF telephones, it is possible to enhance the services, e.g., by dialing

another number while holding on to the call in progress, to set up a call to a third

subscriber. The signal to recall the dialling phase during the talking phase, is called

Register Recall Signal. It consists of interruption of the calling subscriber’s loop for

duration less than the release signal. it may be of 200 to 320 milliseconds duration.

Inter-exchange Signaling

Inter-exchange signaling can be transmitted over each individual inter exchange

trunk. The signals may be transmitted using the same frequency band as for speech

signals (inband signaling), or using the frequencies outside this band (out-of-band

signaling). The signaling may be

i. Pulsed

The signal is transmitted in pulses. Change from idle condition to one of active states for

a particular duration characterizes the signal, e.g., address information

ii. Continuous

The signal consists of transition from one condition to another, a steady state condition

does not characterizes any signal.

iii. Compelled

It is similar to the pulsed mode but the transmission is not of fixed duration but condones

till acknowledgement of the receiving unit is received back at the sending unit. It is a

highly reliable mode of signal transmission of complex signals.

Line signals

DC Signaling

The simplest cheapest, and most reliable system of signaling on

trunks, was DC signaling, also known as metallic loop signaling, exactly the same

as used between the subscriber and exchange, i.e.,

78

i. Circuit seizure/release corresponding to off/on-hook signal of the

subscriber.

ii. Address information in the from of decade pulses.

In-Band and Out-of-Band Signals

Exchanges separated by long distance cannot use any form of DC line signaling.

Suitable interfaces have to be interposed between them, for conversion of the

signals into certain frequencies, to enable them to be carried over long distance. A

signal frequency (SF) may be used to carry the on/off hook information. The

dialing pulses can also be transmitted by pulsing of the states. The number of

signals is small and they can be transmitted in-band or out-of band. The states

involved are shown in Table 1.

TABLE 1. SINGLE FREQUENCY SIGNALING STATES TONE SIGNAL CONDITION

State Forward Backward Idle (On hook)

FORWARD

Seizure(off hook)

Release (on hook)

BACKWARD

Answer(off hook)

Clear Back (on hook)

Blocking (off hook)

On

off

on

off

off

on

On

on

off/on

off

on

off

For in band signaling the tone frequency is chosen to be 2600Hz. or 2400 Hz. As

the frequency lies within the speech band, simulation of tone-on condition

indicating end-of call signal by the speech, has to be guarded against, for pre-

mature disconnection.

Out-of- Band signaling overcomes the problem of tone on condition imitation by

the speech by selecting a tone frequency of 3825 Hz which is beyond the speech

band. However, this adds up to the hard-ware costs.

5.3.1.3 E & M Signals

E & M lead signaling may be used for signaling on per-trunk basis. An additional

pair of circuit, reserved for signaling is employed. One wire is dedicated to the

forward signals ((M-Wire for transmit or mouth) which corresponds to receive or

R-lead of the destination exchange, and the other wire dedicated to the backward

79

signals (E-wire for receive or ear) which corresponds transmit or send wire or S-

Lead of the destination exchange. The signaling states are shown in table2.

TABLE 2. E & M SIGNALING STATES State Outgoing Exchange

M- lead E-lead

Incoming Exchange

M- lead Elead

Idle(On hook)

Earth Open Earth Open

FORWARD

seizure(off hook)

Battery Open Earth Earth

Release(On hook)

BACKWARD

Earth Earth/open Battery/Earth Open

Answer(off hook) battery Earth Battery Earth

Clear Back

(On hook)

battery Open Earth earth

Blocking Earth Earth Battery Open

This type of signaling is normally used in conjunction with an interface to change

the E & M signals into frequency signal to be carried along with the speech.

Register Signals

It was, however felt that the trunk service could not be managed

properly

without the trunk register which basically is an address digit receiver, with such

development, the inter-exchange signaling was sub- divided into two categories.

1. Line signaling in which the signals operate throughout the duration of call, and

2. Register signaling during the relatively short phase of setting up the call,

essentially for transmitting the address information.

80

Fig.3. Compelled signalling procedure In other words, register signals are interchanged between registers during a phase

between receipt of trunk seizure signal and the exchange switching to the speech

phase. These signals are proceed-to-send (PTS) signals, address, signals, and

signals indicating the result of the call attempt.

The register signals may be transmitted in band or out of band. however, in the

latter case, the signaling is relatively slow and only limited range of signals may

be used. For example, a single out-of-band frequency may be selected and

information sent as pulses.

In-band transmission can be used easily as there can be no possible interference

with the speech signals. To reduce transmission time and to increase reliability, a

number of frequencies are used in groups. Normally 2 out of 6 frequencies are

used. To make the system more reliable compelled sequence is used. Hence, this

system is normally called compelled sequence Multi-frequency (CSMF) signaling

as shown in Fig.3. In CCITT terminology it is termed as R2 system. As the

frequencies need be transmitted only for a short duration to convey the entire

information, the post dialling delay is reduced.

When more than two exchanges are involved in setting up the connections the

signaling may be done in either of the two modes

i. End-to-end signaling

outgoing register incomming register

forward signal time signal cessation recognition next forward signal

2-and-2only signal recognition

acknowledgement backward signal and request for next signal compelled signal sequence acknowledgement backward signal

time

signal cessation recognition

Sending

81

The signaling is always between the ends of the connection, as the call progresses.

Considering a three exchanges, A-B-C, connection, initially the signaling is

between A-B, then between A-C after the B-C connection is established.

ii. Link-By-Link signaling

The signaling is always confined to individual links. Hence, initially the signaling

is between A-B, then between B-C after the B-C connection is established.

Generally supervisory (or line) and subscriber signaling is necessarily on link-by-

link basis. Address component may be signalled either by end-to-end or link-by-

link depending upon the network configuration.

R2 Signalling

CCITT standardized the R2 signaling system to be used on national and

international routes. However, the Indian environment requires lesser number of

signals and hence, a slightly modified version is being used.

There is a provision for having 15 combinations using two out of six frequencies

viz., 1380, 1500, 1620, 1740, 1860 and 1980 Hz, for forward signals and another

15 combination using two out of six frequencies viz., 1140,1020, 900, 780, 660

and 540 Hz, for backward signals. In India, the higher frequency in the forward

group i.e., 1980 Hz, and the lower frequency in the backward group, i.e., 540 hz,

are not used. Thus, there are 10 possible combinations in both the directions. The

weight codes for the combinations used are indicated in Table 3 and the

significance of each signal is indicated in Table 4 and 5.

TABLE 3- SIGNAL FREQUENCY INDEX AND WEIGHT CODE

Signal Frequency (Hz)

Forward 1380 1500 1620 1740 1860

Backward 1140 1020 900 780 660

Index f 0 f 1 f 2 f 3 f 4

Weight Code 0 1 2 4 7

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TABLE 4-FORWARD SIGNALS Signal Weight Group I Group II

1 0+1 Digit 1 Ordinary subscriber 2 0+2 Digit2 Subscriber with

priority Test / Mtce, equipment

3 1+2 Digit3 Spare 4 0+4 Digit4 STD Barred 5 1+4 Digit5 Spare 6 2+4 Digit6 CCB 7 0+7 Digit7 Changed Number to

Operator 8 1+7 Digit8 Closed Number 9 2+7 Digit9 Closed Number 10 4+7 Digit0 Spare TABLE 5 -BACKWARD SIGNALS Signal No.

Weight Code

Group A Group B

1 0+1 Send next digit Called line free with out metering

2 0+2 Restart Changed number 3 1+2 Address complete, Changeover to

reception of group B signals Called line busy

4 0+4 Calling line identification for malicious calls

Local congestion

5 1+4 send calling subscribers category Number unobtainable 6 2+4 Set up speech connection called line fee, with

metering 7 0+7 Send last but 1 digit Route congestion 8 1+7 Send last but 2 digit Spare 9 2+7 Send last but 3 digit Route Breakdown 10 4+7 Spare Malicious call blocking Note : Signals A2, and A7 to A9 are used in Tandem working only.

It can be seen from the tables that

1. Forward signals are used for sending the address information of the called

subscriber, and category and address, information of the calling subscriber.

2. Backward signals are used for demanding address information and caller’s

category and for sending condition and category of called line.

R2 signaling is fully compelled and the backward signal is transmitted as an

acknowledgement to the forward signal. This speeds up the interchange of

information, reducing the call set up time. However, the satellite circuits are an

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exception and semi-compelled scheme may only be used due to long propagation

time.

Register signals may be transmitted on end-to-end basis. It is a self checking

system. Each signal is acknowledgement appropriately at the other end after the

receiver checks the presence of only 2 and only 2 out of 5 proper frequencies.

An example of CSMF signaling between two exchanges may be illustrated

by considering a typical case. The various signals interchanged after seizure of the

circuit using DC signaling are

1. originating exchange sends first digit

2. Receipt of the digit is acknowledged by the terminating exchanges by sending A5

(demanding the caller’s category).

3. A5 is acknowledgement by sending any11-1 to 11-5 by the originating exchange

4. Terminating exchange acknowledges this by A1, demanding for next digit.

5. Originating exchange, acknowledges A1 by sending any of 1-1to 1-10 sending the

digit.

6. The digits are sent in succession by interchange of steps v and vi.

7. On receipt of last digit, the terminating exchange carries out group and line

selection and then sends A3, indicating switching over to group B signals.

8. This is acknowledgement by the originating exchange by sending the caller’s

category again.

9. The terminating exchange acknowledgements by sending the called line condition

by sending any of B2 to B6.

10. In response to B6, the originating exchanges switches through the speech path and

the registers are released. Alternatively, in response to B2 to B5, the registers are

released and appropriate tone is fed to the calling subscriber by the originating

exchange.

3 Digital Signalling

All, the systems discussed so far, basically, are on per line or per trunk basis, as

the signals are carried on the same line or trunk. With the emergence of PCM

systems, it was possible to segregate the signaling from the speech channel.

Inter exchange signalling can be transmitted over a channel directly associated

with the speech channel, channel-associated signalling (CAS) , or over a

dedicated link common to a number of channels, common channel signalling

(CCS). The information transmitted for setting up and release of calls is same in

both the cases. Channel associated signalling requires the exchanges, to have

access to each trunk via the equipment which may be decentralised, whereas, in

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common channel signalling, the exchange is connected to only a limited number

of signalling links through a special terminal.

Channel- Associated signalling

In the PCM systems the signalling information is conveyed on a separate channel which

is rigidly associated with the speech channel. Hence, this method is known as channel

associated signalling (CAS). Though the speech sampling rate is 8 Khz, the signals do not

change as rapidly as speech and hence, a lower sampling rate of 500 Hz, for digitisation

of signals can suffice. Based on this concept, TS 16 of each frame of 125 microseconds is

used to carry signals of 2 speech channels, each using 4 bits.

Hence, for a 30 channel PCM system, 15 frames are required to carry all the signals. To

constitute a 2 millisecond multiframe of 16 frames. F 0 to F 15 TS 16 of the frame F 0 is

used for multiframe synchronisation. TS 16 of F1 contains signal for speech channels 1

and 16 being carried in TS 1 and TS 17, respectively, TS16 of F2 contains signals of

speech channels 2 and 17 being carried in TS2 and TS 18, respectively and so on, Both

line signals and address information can be conveyed by this method.

Although four bits per channel are available for signalling only two bits are used. As the

transmission is separate in the forward and backward direction, the bits in the forward

link are called af and bf, and those in the backward link are called ab and bb. Values for

these bits are assigned as shown in Table 6.

As the dialling pulses are also conveyed by these conditions, the line state recognition

time is therefore, above a threshold value. The bit bf is normally kept at 0, and the value

1 indicates a fault.

However, the utilisation of such a dedicated channel for signalling for each speech

channel is highly inefficient as it remains idle during the speech phase. Hence, another

form of signalling known as common-channel signalling evolved.

State Bit Value

Forward Backward. af bf ab bb Idle 1 0 1 0 Seizure 0 0 1 0 Seizure acknowledge 0 0 1 1 Answer 0 0 0 1 Clear Forward 1 0 0/1 1 Clear Back 0 0 1 1

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COMMON CHANNEL SIGNALING SYSTEM No. 7 (CCS#7)

1. Introduction

Communication networks generally connect two subscriber terminating

equipment units together via several line sections and switches for message

exchange (e.g. speech, data, text or images). Control information has to be

transferred between the exchanges for call control and for the use of facilities. In

analog communication networks, channel-associated signaling systems have so far

been used to carry the control information. Fault free operation is guaranteed with

the channel-associated signaling systems in analog communication networks, but

the systems do not meet requirements in digital, processor-controlled

communication network. Such networks offer a considerably larger scope of

performance as compared with the analog communication networks due, for

instance, to a number of new services and facilities. The amount and variety of the

information to be transferred is accordingly larger. The information can no longer

be economically transported by the conventional channel-associated signaling

systems. For this reason, a new, efficient signaling system is required in digital,

processor-controlled communication networks.

The CCITT has, therefore, specified the common channel signalling system no.7

(CCS-7). CCS-7 is optimised for application in digital networks. It is

characterised by the following main features :

• internationally standardized (national variations possible).

• suitable for the national, international and intercontinental network level.

• suitable for various communication services such as telephony, text services, data

services digital network (ISDN).

• high performance and flexibility along with a future-oriented concept which well

meet new requirements.

� high reliability for message transfer.

� processor-friendly structure of messages (signal units of multiples of 8

bits).

� signalling on separate signalling links; the bit rate of the circuits is,

therefore, exclusively for communication.

� signalling links always available, even during existing calls.

� use of the signalling links for transferring user data also.

� used on various transmission media

- cable (copper, optical fiber)

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- radio relay

- satellite (up to 2 satellite links)

� use of the transfer rate of 64 Kbit/s typical in digital networks.

� used also for lower bit rates and for analog signalling links if necessary.

� automatic supervision and control of the signalling network.

2. CC#7 Signalling terminology

2.1 Signalling Network

In contrast to channel-associated signalling, which has been standard practice until

now, in CCS7 the signalling messages are sent via separate signalling links (See

Fig. 1). One signalling link can convey the signalling messages for many circuits

The CCS7 signalling links connect signalling points (SPs) in a communication

network. The signalling points and the signalling links form an independent

signalling network which is overlaid over the circuit network.

Fig 1. Signalling via a Common Channel Signalling l ink

2.2. Signalling Points (SP) A distinction is made between signalling points (SP) and signalling transfer points

(STP).

The SPs are the sources (originating points) and the sinks (destination points) of

signalling traffic. In a communication network these are primarily the exchanges.

The STPs switch signalling messages received to another STP or to a SP on the

basis of the destination address. No call processing of the signalling messages

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occurs in a STP. A STP can be integrated in a SP (e.g. in an exchange) or can form

a node of its own in the signalling network. One or more levels of STPs are

possible in a signalling network, according to the size of the network.

All SPs in the signalling network are identified by means of a code within the

framework of a corresponding numbering plan and, therefore, can be directly

addressed in a signalling message.

2.3. Signalling links

A signalling link consists of a signalling data link (two data channels operating

together in opposite directions at the same date rate) and its transfer control

functions. A channel of an existing transmission link (e.g. a PCM30 link) is used

as the signalling data link. Generally, more than one signalling link exists between

two SPs in order to provide redundancy. In the case of failure of a signalling link,

functions of the CCS7 ensure that the signalling traffic is rerouted to fault-free

alternative routes. The routing of the signalling links between two SPs can differ.

All the signalling links between two SPs are combined in a signalling link set.

2.4. Signalling Modes

Two different signalling modes can be used in the signalling networks for CCS7,

viz. associated mode and quasi-associated mode.

In the associated mode of signalling, the signalling link is routed together with

the circuit group belonging to the link. In other words, the signalling link is

directly connected to SPs which are also the terminal points of the circuit group

(See Fig.2). This mode of signalling is recommended when the capacity of the

traffic relation between the SPs A and B is heavily utilized.

Fig. 2 Associated Mode of Signalling

In the quasi-associated mode of signalling, the signalling link and the speech

circuit group run along different routes, the circuit group connecting the SP A

directly with the SP B. For this mode, the signalling for the circuit group is carried

out via one or more defined STPs (See Fig. 3.3). This signalling mode is

favourable for traffic relations with low capacity utilization, as the same signalling

link can be used for several destinations.

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Fig. 3 Quasi-associated mode

2.5 Signalling Routes

The route defined for the signalling between an originating point and a destination

point is called the signalling route. The signalling traffic between two SPs can be

distributed over several different signalling routes. All signalling routes between

two SPs are combined in a signalling route set.

2.6 Network Structure

The signalling network can be designed in different ways because of the two

signalling modes. It can constructed either with uniform mode of signalling

(associated or quasi-associated) or with a mixed mode (associated and quasi-

associated).

The worldwide signalling network is divided into two levels that are functionally

independent of each other; an international level with an international network and

a national level with many national networks. Each network has its own

numbering plans for the SPs.

3. Planning Aspects

Economic, operational and organizational aspects must be considered in the

planning of the signalling network for CCS7. An administration should also have

discussions with the other administrations at an early stage before CCS7 is

introduced in order to make decisions, for example, on the following points :

(a) Signalling network

- mode of signalling

- selection of the STPs

- signalling type (en block or overlap)

- assignment of the addresses to SPs.

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(b) Signalling data links, e.g. 64 kbit/s digital or 4.8 kbit/s analog

(c) Safety requirements

- load sharing between signalling links

- diverting the signalling traffic to alternative routes in event of faults.

- error correction

(d) Adjacent traffic relations

The signalling functions in CCS7 are distributed among the following parts :

- message transfer part (MTP)

- function – specific user parts (UP)

The MTP represents a user-neutral means of transport for messages between the

users. The term user is applied here for all functional units which use the transport

capability of the MTP.

Each user part encompasses the functions, protocols and coding for the signalling

via CCS7 for a specific user type (e.g. telephone service, data service, ISDN). In

this way, the user parts control the set-up and release of circuit connections, the

processing of facilities as well as administration and maintenance functions for the

circuits.

The functions of the MTP and the UP of CCS7 are divided into 4 levels. Levels to

3 are allotted to the MTP while the UPs form level 4 .

Fig. 4 Functional Levels of CCS7

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The message transfer part (MTP) is used in CCS7 by all user parts (UPs) as a

transport system for message exchange. Messages to be transferred from one UP

to another are given to the MTP (See Fig.5). The MTP ensures that the messages

reach the addressed UP in the correct order without information loss, duplication

or sequence alteration and without any bit errors.

4. Functional Levels

Fig. 5 Message exchange between two Signalling Points with CCS7

4.1 Level I (Signalling Data Link) defines the physical, electrical and functional

characteristics of a signalling data link and the access units. Level 1 represents the

bearer for a signalling link. In a digital network, 64-kbit/s channels are generally

used as signalling data links. In addition, analog channels (preferablywith a bit rate

of 4.8 kbit/s) can also be used via modems as a signalling data

link.

4.2 Level 2 (Signalling Link) defines the functions and procedures for a correct

exchange of user messages via a signalling link. The following functions must be

carried out at level 2 :

- delimitation of the signal units by flags.

- elimination of superfluous flags.

- error detection using check bits.

- error correction by re-transmitting signal units.

- error rate monitoring on the signalling data link.

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- restoration of fault-free operation, for example, after disruption of the signalling

data link.

4.3 Level 3 (Signalling Network) defines the inter-working of the individual

signalling links. A distinction is made between the two following functional areas

:

- message handling, i.e. directing the messages to the desired signalling line, or to

the correct UP.

- signalling network management, i.e. control of the message traffic, for example,

by means of changeover of signalling links if a fault is detected and changeback to

normal operation after the fault is corrected.

The various functions of level 3 operate with one another, with functions of other

levels and with corresponding functions of other signalling of other SPs.

5. CCS#5 Signalling messages – Common terms

5.1 Signal Units (SU)

The MTP transport messages in the form of SUs of varying length. A SU is

formed by the functions of level 2. In addition to the message it also contains

control information for the message exchange. There are three different types of

SUs :

- Message Signal Units (MSU).

- Link Status Signal Units (LSSU).

- Fill-in Signal Units (FISU).

Using MSUs the MTP transfers user messages, that is, messages from UPs (level

4) and messages from the signalling network management (level 3). The structure

of the three types of message units is shown in Fig.6.

The LSSUs contain information for the operation of the signalling link (e.g. of the

alignment).

The FISUs are used to maintain the acknowledgement cycle when no user

messages are to be sent in one of the two directions of the signalling link.

5.2 Protocol Information Bits

Flag (F) : (8 bits) The SUs are of varying length. In order to clearly separate them

from one another, each SU begins and ends with a flag. The closing flat of one

SUs is usually also the opening flag of the next SU. However, in the event of

overloading of the signalling link, several consecutive flags can be sent. The flag

is also used for the purpose of alignment. The bit pattern of a flg is 01111110.

5.3 Backward Sequence Number (BSN) : (7 bits) The BSN is used as an

acknowledgement carrier within the context of error control. It contains the

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forward sequence number (FSN) of a SU in the opposite direction whose

reception is being acknowledged. A series of SUs can also be acknowledged with

one BSN.

5.4 Backward Indicator Bit (BIB) : (1 bit) The BIB is needed during general

error correction. With this bit, faulty SUs are requested to be retransmitted for

error correction.

Fig. 6 Format of Various Signal Units

5.5 Forward Sequence Number (FSN) : (7 bits) A FSN is assigned

consecutively to each SU to be transmitted. On the receive side, it is used for

supervision of the correct order for the SUs and for safeguarding against

transmission errors. The numbers 0 to 127 are available for the FSN.

5.6 Forward Indicator Bit (FIB) : (1 bit) The FIB is needed during general

error correction. It indicates whether a SU is being sent for the first time or

whether it is being retransmitted.

5.7 Length Indicator (LI) : (6 bits) The LI is used to differentiate between

the three SUs. It gives the number of octets between the check-bit (CK) field and

the LI field. The LI field contains different values according to the type of SU; it

is 0 for FISU, 1 or 2 for LISU and is greater than 2 for MSU.

The maximum value in the length indicator fields is 63 even if the signalling

information field (SIF) contains more than 63 octets.

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5.8 Check bits (CK) : (16 bits) The CKs are formed on the transmission side

from the contents of the SU and are added to the SUs as redundancy. On the receive side,

the MTP can determine with the CKs whether the SU was transferred without any errors.

The SUs acknowledged as either positive or faulty on the basis of the check.

5.9 Fields specific to MSUs :

5.9.1 Service Information Octet (SIO) : (8 bits) It contains the Service

Indicator (SI, 4 bits) and Subservice field (SSF, 4 bits) whose last 2 bits are Network

Indicator (NI).

An SI is assigned to each user of the MTP. It informs the MTP which UP has sent

the message and which UP is to receive it. Four SI bits can define 16 UPs (3-SCCP, 4-

TUP, 5-ISUP, 6-DATAUP, 8-MTP test, etc.). The NI indicates whether the traffic is

international (00,01) or national (10,11). In CCS7 a SP can belong to both national and

international network at the same time. So SSF field indicate where the SP belongs.

5.9.2 Signalling Information Fields (SIF) : (2 to 272 octets) It contains the

actual user message. The user message also includes the address (routing label, 40 bits)

of the destination to which the message is to be transferred. The maximum length of the

user message is 62 octets for national and 272 octets for international networks (one octet

= 8 bits). The format and coding of the user message are separately defined for each UP.

5.10 Fields Specific to LSSUs

5.10.1 Status Field (SF) : (1 to 2 octets) It contains status indications for the

alignment of the transmit and receive directions. It has 1 or 2 octets, out of which only 3

bits of first octet are defined by CCITT, indicating out (000), normal (001), Emergency

(010) alignments, out-of-service (011), Local processor outage (100) status, etc.

5.10.2 Addressing of the SUs (in SIF)

A code is assigned to each SP in the signalling network according to a numbering

plan. The MTP uses the code for message routing. The destination of a SU is specified in

a routing label. The routing label is a component of every user message and is transported

in the SIF. The routing label in a MSU consists of the following (See Fig. 7).

Fig. 7 Routing Label of a Message Signal Unit

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5.10.3 Destination Point Code (DPC) : (14 bits) identifies the SP to which this

message is to be transferred.

5.10.4 Originating Point Code (OPC) : (14 bits) specifies the SP from which

the message originates.

The coding of OPC and DPC is pure binary and using 14 bits linear encoding, it is

possible to identify 16,384 exchanges. The number of exchanges in DOT network having

CCS7 capability are expected to be within this limit.

5.10.5 Signalling Link Selection (SLS) field : (4 bits) The contents of the SLS field

determine the signaling route (identifying a particular signalling link within s link set or

link sets) along which the message is to be transmitted. In this way, the SLS field is used

for load sharing on the signalling links between two SPs.

The SIO contains additional address information. Using the SI, the destination

MTP identifies the UP for which the message is intended. The NI, for example, enables a

message to be identified as being for national or international traffic.

LSSUs and FISUs require no routing label as they are only exchanged between

level 2 of adjacent MTPs.

The message sent from a user to the MTP for transmission contains : the user

information, the routing label, the SI, the NI and a LI. The processing of a user message

to be transmitted in the MTP begins in level 3 (See Fig.8).

The MTP is responsible for (a) transmitting, (b) receiving SUs, (c) for correcting

transmission errors, (d) for the signalling network management, and (e) for the alignment.

Its functions are spread over the functional levels 1, 2 and 3.

5.11 The message routing (level 3) determines the signalling link on which the

user message is to be transmitted. To do this, it analyzes the DPC and the SLS field in the

routing label of the user message, and then transfers the message to the appropriate

signalling link (level 2).

5.12 The transmission control (level 2) assigns the next FSN and the FIB to

the user message. In addition, it includes the BSN and the BIB as an acknowledgement

for the last received MSU. The transmission control simultaneously enters the part of the

MSU formed so far in the transmission and retransmission buffers. All MSUs to be

transmitted are stored in the retransmission buffer until their fault-free reception is

acknowledged by the receive side. Only then are they deleted.

5.13 The check bit and flag generator (level 2) generates CKs for

safeguarding against transmission errors for the MUS and sets the flag for separating the

SUs. In order that any section of code identical to the flag (01111110) occurring by

chance is not mistaken for the flag, the user messages are monitored before the flag is

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added to see if five consecutive ones (1) appear in the message. A zero (0) is

automatically inserted after five consecutive 1s. On the receive side, the zero following

the five 1s is then automatically removed and the user message thereby regains its

original coding.

The check-bit and flag generator transfers a complete MSU to level 1. In level 1,

the MUS is sent on the signalling data link.

The bit stream along a signalling data link is received in level 1 and transferred to

level 2. Flag detection (level 2) examines the received bit stream for flags. The bit

sequence between two flags corresponds to one SU. The alignment detection (level 2)

monitors the synchronism of transmit and receive sides with the bit pattern of the flags.

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Using the CKs transmitted, error detection (level 2) checks whether the SU was

correctly received. A fault-free SU is transferred to the receive control, while a

faulty SU is discarded. The reception of a faulty SU is reported to error rate

monitoring, in order to keep a continuous check on the error rate on the receive

side of the signalling link. If a specified error rate is exceeded, this is reported to

the signalling link status control by error rate monitoring. The signalling link status

control then takes the signalling link out of service and sends a report to level 3.

5.14 The receive control (level 2) checks whether the transferred SU contains

the expected FSN and the expected FIB. If this is the case and if it is a MSU, the

receive control transfers the user message to level 3 and causes the reception of

the MSU to be positively acknowledged. If the FSN of the transferred MSU does

not agree with that expected, the receive control detects a transmission error and

causes this and all subsequent MSU to be retransmitted (see subheading

"Correction of Transmission Errors").

5.15 The message discrimination (level 3) accepts the correctly received user

message. It first determines whether the user message is to be delivered to one of

the immediately connected UPs or to be transferred to the another signalling link

(quasi-associated message). This pre selection is achieved in the message

discrimination by evaluation of the DPC. A user message which only passes

through a SP (STP) is transferred by the message discrimination to the message

routing, where it is treated as a user message to be transmitted.

If a received user message is intended for one of the connected UPs (SP), it is

transferred to message distribution (level 3). The message distribution evaluates

the SIO, thereby determining the UP concerned, and delivers the user message

there.

6. Signalling Network Management

The signalling network management is a function of level 3. It controls the

operation and the inter working of the individual signalling links in the signalling

network. To this end, the signalling network management exchanges messages

and control instructions with the signalling links of level 2, sends message to the

UPs and works together with the signalling network management in adjacent SPs.

For the inter working with other SPs the signalling network management uses the

transport function of the MTP.

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Management messages are transferred in MSUs like user messages. For

discrimination, the management messages have their own SI. The signalling

network management contains 3 function blocks :

(a) The signalling link management controls and monitors the individual

signalling links. It receives the messages concerning the alignment and status of

the individual signalling links, or concerning operating irregularities and effects

any changes in status which may be necessary. In addition, the signalling link

management controls the putting into service of signalling links, including initial

alignment and automatic realignment of signalling links after failures or alignment

losses due to persistent faults. If necessary, the signalling link management

transfers messages to the signalling traffic management or receives instructions

from there.

(b) The signalling route mangement controls and monitors the operability of

signalling routes. It exchanges messages with the signalling route management in

the adjacent STPs for this purpose. The signalling route management receives, for

example, messages concerning the failure or re-availability of signalling routes or

the overloading of STPs. In cooperation with the signalling traffic management, it

initiates the appropriate actions in order to maintain the signalling operation to the

signalling destinations involved.

(c) The signalling traffic management controls the diversion of the

signalling traffic from faulty signalling links or routes to fault-free signalling links

or routes. It also controls the load distribution on the signalling links and routes.

To achieve this, it can initiate the following actions :

- changeover; on failure of a signalling link the signalling traffic management

switches the signalling traffic from the failed signalling link to a fault-free

signalling link.

- changeback; when signalling link becomes available again after a fault has been

corrected, the signalling traffic management reverse the effect of the changeover.

- rerouting; when SP can no longer be reached on a normal route, the signalling

traffic management diverts the signalling traffic to a predefined alternative route.

When overloading occurs, the signalling traffic management sends messages to

the users in its own SP in order that they reduce the load. The management also

informs the adjacent SPs of the overloading in its own SP and requests them to

also reduce the load.

The signalling traffic management accomplishes its functions by

- receiving messages from the signalling link and signalling route management.

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- sending control instructions to signalling link and signalling route management.

- directly accessing the signalling links, e.g. during emergency alignment.

- modifying the message routing on failure of signalling routes.

- exchanging management messages with the signalling traffic management in

adjacent SPs.

As discussed earlier, level 4 functions, which include formatting of messages

based on the applications, are allotted to UPs. Each UP provides the functions for

using the MTP for a particular user type. Some of the UPs as currently specified

by the CCITT are :

- telephone user part (TUP)

- integrated services digital network user part (ISDN-UP)

- the signalling connection control part (SCCP)

- the transaction capabilities application part (TCAP)

For Intelligent Network (IN) application, Intelligent Application Part (INAP) and

TCAP are used. SCCP forms the interface between these UPs and MTP.

Fig.9 shows the users of the MTP as well as their relationship to one another and

to the MTP. CCS7 can be adapted to all requirements due to the modular

structure. Expansion for future applications is also possible. Each CCS7 user can

specify its own UP, for example, the mobile user part (MUP) is Siemen's own

specification for the mobile telephone network C450.

Fig. 9 Message Transfer Part Users

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7. MTP users

7.1. Telephone User Part (TUP)

Use of CCS7 for telephone call control signalling requires (i) application of TUP

functions, in combination with (ii) application of an appropriate set of MTP

functions. The TUP is one of level 4 users in CCS7. It is specified with the aim of

providing the same features for telephone signalling as other telephone signalling

systems. It exchanges signalling messages through MTP. Signalling messages

contain information relating to call set up and conditions of speech path. The TUP

message consists of SIF and a SIO. These signalling information are generated by

the TUP of the originating exchange. The label is 40 bits long, comprises DPC,

OPC and CIC. CIC indicates one of the speech circuit connecting the destination

and originating points. Level 3 identifies the user to which a message belongs by

SIO, which comprises a SI and SSF. For TUP SI value is 4. The SSF distinguishes

the signalling message is for national or international network.

7.2 Integrated Services Digital Network User Part

The ISDN-UP covers the signalling functions for the control of calls, for the

processing of services and facilities and for the administration of circuits in ISDN.

The ISDN-UP has interface to the MTP and the SCCP for the transport of MSUs.

The ISDN-UP can use SCCP functions for end-to-end signalling.

CCITT SIGNALLING SYSTEM NO. 7 :

INTEGRATED SERVICES DIGITAL NETWORK USER PART

7.2.1 Overview of the ISDN User Part

The integrated services digital network user part (ISUP) is the protocol which

provides the signalling functions required by CCITT No. 7 signalling to support

basic bearer services and supplementary services for voice and non-voice

applications in an Integrated Services Digital Network (ISDN).

The ISUP is suited for application in dedicated telephone and circuit-switched

data networks and in analogue and moved analogue/digital networks. In particular,

the ISUP meets the requirements defined by the CCITT for world-wide

International semiautomatic and automatic telephone and circuit-switched data

traffic.

The ISUP can be used for national and international applications. The signalling

procedures, information elements and message type specified are for both

applications. Coding space has been reserved to allow national administrations

and recognized private operating agencies to introduce network specific signalling

messages and elements of information within the protocol structure.

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The ISUP makes use of the services provided by the messages transfer part (MTP)

(1) and, in some cases, by the signalling connection control part (SCCP0 of

CCITT No.7 signalling for the transfer of information between ISDN user parts.

7.2.2 Services Supported by the ISDN User Part

The ISUP protocol supports the basic bearer service; that is the establishment,

supervision and release of 64 kbit/s circuit-switched network connections between

customer line exchange terminations.

In addition to the basic bearer service the ISUP is expected to support (in the 1988

Recommendations) the following supplementary services :

� Calling line identification (presentation and restriction).

� Call forwarding,

� Closed user group,

� Direct dialling-in, and

� User-to-user signalling.

Signaling Connection Control Part

Introduction: The SCCP function is covered in ITU-T recommendations Q.711 to

Q.714 and Q.716. The signalling connection control part provides additional functions to

message transfer part for transfer of circuit related and non-circuit related signalling

information and other type of information between exchanges and other specialized

centrals in telecommunications network via SS#7 networks. The overall objective of

SSCP is to provide means for:

A transfer capability for signalling data units with or without the use of logical signalling

connections.

A logical signalling connection between two SCCP users with the SS#7 network.

Enhanced addressing capabilities.

The following figure illustrates the SCCP position in the SS#7 hierarchy:

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The functions of SCCP are used for handling transactions required by TCAP and

also for transfer of circuit related and call related signalling information for ISDN

UP with or without set up of end-to-end logical signalling connections.

The SCCP relies on the MTP to route the signalling information from one node to

another node. For this, it interacts with the user parts and with the MTP.

Primitives are used to convey information between the levels. Primitives are

nothing but set of commands and their respective responses associated with the

services requested of the SCCP.

SCCP and OSI model

The SCCP enhances the services of MTP to provide the functional equivalent of

Network layer (i.e. layer #3 of OSI model). The MTP and the SCCP together is

also referred to as Network Service Part (NSP).

SCCP Addressing

The addressing capability of MTP is limited to delivering the message to a node

(identified by Network indicator and DPC) and to distribute it to a user using four

bit service indicator (octet SIO ). SCCP supplements this capability by providing

an addressing capability that uses DPC + SSN .The SSN is a local addressing

information used by SCCP to identify each of the SCCP users at a node.

SCCP provides enhanced addressing capability to MTP to enable it to address

messages with Global Title (GT). A Global Title is an address that does not

explicitly contain information usable for routing by MTP.

SCCP

Addressing

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SCCP provides enhanced addressing capability to MTP to enable it to address

messages with Global Title (GT). A Global Title is an address that does not

explicitly contain information usable for routing by MTP.

Global Title

The SS#7 signalling method identifies the destination and origination using

signalling point codes. Since a signalling point code has only fourteen bits. It is too

small to be uniquely addressed on a global scale. For this reason, signalling point

codes are always combined with a Network indicator- which means that a code is

only valid in one particular network.

To facilitate unambiguous global addressing, a unique international address or

sender information is necessary. The global address is known as The Global title

and is sent in the SCCP message in SS#7 messages. Since, SPCs are only ever

valid in individual networks, so called Global title translation must be performed at

each relevant network gateway. The SCCP performs the Global title translation,

whereby an internationally unique address (Global Title) is translated to an SPC

and Network indicator in order to be transferred to the network border or to a

destination if it is located in the same network.

Global title translation is always used if no SPC for the destination is available at

all. For example, the HLR is to be identified on the basis of the IMSI.

SCCP Functional Units

The services supported by

SCCP are divided into two groups viz. Connection-oriented services and

Connectionless services. The protocols used for providing these services are

divided in four classes; two for connectionless services and two for connection-

oriented services. Each protocol class defines which level of services SCCP to

provide. The four protocol classes are described below:

Connectionless services

Class 0 – Basic connectionless

Class 1 – Sequenced connectionless

Connection oriented services

Class 2 – Basic connection oriented

Class 3 – Flow control connection oriented

The first two classes 0 and 1, support the connectionless environment, for example,

for use by TCAP. These are particularly suitable for frequent transmission of short

messages. As an example, to check validity of the credit card, an interrogation

message can sent to a data centre and reply received on the same route. The

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connectionless services are all that is used in today’s networks. Classes 2 and 3 are

used for connection-oriented services, for example by ISDN–UP and, even though

well-defined, are not used in today’s network.

The SCCP is divided into four functional units:

� SCCP routing control (SCRC).

� SCCP connectionless control (SCLC).

� SCCP connection-oriented control.

� SCCP management control (SCMG).

SCCP Functional Units

Transaction Capabilities Application Part (TCAP)

The TCAP recommendations are covered in ITU-T Q.771 to Q.775. The TCAP

portion of the CCS#7 protocol is used to transfer non-circuit related information

between two signalling points in the network. It is used to communicate between

the SSP, SCP or other SSPs through an exchange of TCAP messages. There is no

setup of speech/data channel connections.

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Non-circuit related information would be such things as data queries for

services (1600) where there is not a physical end-to-end connection between the

signalling points TCAP supports the exchange of non-circuit related data between applications

across the SS7 network using the SCCP connectionless service. Queries and

responses sent between SSPs and SCPs are carried in TCAP messages. For

example, an SSP sends a TCAP query to determine the routing number associated

with a dialed 1600 number and to check the personal identification number (PIN)

of a calling card user. In mobile networks (IS-41 and GSM), TCAP carries Mobile

Application Part (MAP) messages sent between mobile switches and databases to

support user authentication, equipment identification, and roaming.

Applications for the TCAP

� In mobile networks to report the location of a mobile network subscriber to

the home exchanges.

� In credit card service to check the validity and to execute account

transactions.

Functions of TCAP

• TCAP supports real-time remote operations and is structured in two sub-

layers:

i) Component sub-layer, dealing with individual actions called components.

ii)Transaction sub-layer,dealing with the exchange of messages containing

components

• The component sub-layer is above the transaction sub-layer.

• The TCAP layer interfaces directly with SCCP layer.

• A component consists of a request to invoke an operation.

• An invocation of the operation is identified by a Component ID.

• Components are passed individually between TCAP users.

The originating TC user may send several components to the component

sub-layer before they are transmitted in a single message to the remote end. At the

remote end each one is delivered individually to the destinating TC-user.

Successive component exchanged between TC-users in order to perform an

application constitute a dialogue. The component sub-layer allows several

dialogues to run concurrently between two TC-users each being identified by a

particular ID.

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TCAP serves all ( application specific0 ASEs in a node.To send a message an

ASE passes a series of TC requests to TCAP and TCAP passes the message to

SCCP. When a TCAP receives a message from its SCCP it passes the contents to

the destination ASE in its node.

.

ASE1

SCCP-A

MTP-A MTP-B

SCCP-B

TCAP-B

ASE2

TCAP-A

Component

TCAP messages

MSUs

TC-primitives TC primitives

N-primitive N-primitives

MTP-primitives MTP-primitives

Messages and message paths

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CHAPTER - 6

FIBRE-OPTIC COMMUNICATION TECHNOLOGY

1.0 A Brief History of Fiber-Optic Communications Optical communication systems date back to the 1790s, to the optical semaphore

telegraph invented by French inventor Claude Chappe. In 1880, Alexander Graham Bell

patented an optical telephone system, which he called the Photophone. However, his

earlier invention, the telephone, was more practical and took tangible shape.

By 1964, a critical and theoretical specification was identified by Dr. Charles K.

Kao for long-range communication devices, the 10 or 20 dB of light loss per kilometer

standard. Dr. Kao also illustrated the need for a purer form of glass to help reduce light

loss. By 1970 Corning Glass invented fiber-optic wire or "optical waveguide fibers"

which was capable of carrying 65,000 times more information than copper wire, through

which information carried by a pattern of light waves could be decoded at a destination

even a thousand miles away. Corning Glass developed an SMF with loss of 17 dB/km at

633 nm by doping titanium into the fiber core. By June of 1972, multimode germanium-

doped fiber had developed with a loss of 4 dB per kilometer and much greater strength

than titanium-doped fiber. Prof. Kao was awarded half of the 2009 Nobel Prize in Physics

for "groundbreaking achievements concerning the transmission of light in fibers for

optical communication". In April 1977, General Telephone and Electronics tested and

deployed the world's first live telephone traffic through a fiber-optic system running at 6

Mbps, in Long Beach, California. They were soon followed by Bell in May 1977, with an

optical telephone communication system installed in the downtown Chicago area,

covering a distance of 1.5 miles (2.4 kilometers). Each optical-fiber pair carried the

equivalent of 672 voice channels and was equivalent to a DS3 circuit. Today more than

80 percent of the world's long-distance voice and data traffic is carried over optical-fiber

cables.

2.0 Fiber-Optic Applications FIBRE OPTICS: The use and demand for optical fiber has grown tremendously

and optical-fiber applications are numerous. Telecommunication applications are

widespread, ranging from global networks to desktop computers. These involve the

transmission of voice, data, or video over distances of less than a meter to hundreds of

kilometers, using one of a few standard fiber designs in one of several cable designs.

Carriers use optical fiber to carry plain old telephone service (POTS) across their

nationwide networks. Local exchange carriers (LECs) use fiber to carry this same service

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between central office switches at local levels, and sometimes as far as the neighborhood

or individual home (fiber to the home [FTTH]).

Optical fiber is also used extensively for transmission of data. Multinational firms

need secure, reliable systems to transfer data and financial information between buildings

to the desktop terminals or computers and to transfer data around the world. Cable

television companies also use fiber for delivery of digital video and data services. The

high bandwidth provided by fiber makes it the perfect choice for transmitting broadband

signals, such as high-definition television (HDTV) telecasts. Intelligent transportation

systems, such as smart highways with intelligent traffic lights, automated tollbooths, and

changeable message signs, also use fiber-optic-based telemetry systems.

Another important application for optical fiber is the biomedical industry. Fiber-optic

systems are used in most modern telemedicine devices for transmission of digital

diagnostic images. Other applications for optical fiber include space, military,

automotive, and the industrial sector.

3.0 ADVANTAGES OF FIBRE OPTICS : Fibre Optics has the following advantages :

• SPEED: Fiber optic networks operate at high speeds - up into the gigabits

• BANDWIDTH : large carrying capacity

• DISTANCE: Signals can be transmitted further without needing to be "refreshed" or

strengthened.

• RESISTANCE: Greater resistance to electromagnetic noise such as radios, motors or

other nearby cables.

• MAINTENANCE: Fiber optic cables costs much less to maintain.

4.0 Fiber Optic System : Optical Fibre is new medium, in which information (voice, Data or Video) is transmitted

through a glass or plastic fibre, in the form of light, following the transmission sequence

give below :

(1) Information is Encoded into Electrical Signals.

(2) Electrical Signals are Coverted into light Signals.

(3) Light Travels Down the Fiber.

(4) A Detector Changes the Light Signals into Electrical Signals.

(5) Electrical Signals are Decoded into Information.

- Inexpensive light sources available.

- Repeater spacing increases along with operating speeds because low loss

fibres are used at high data rates.

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Fig. 1

5.0 Principle of Operation - Theory Total Internal Reflection - The Reflection that Occurs when a Ligh Ray Travelling in One Material Hits a Different Material and Reflects Back into the Original Material without any Loss of Light.

Fig. 2

Speed of light is actually the velocity of electromagnetic energy in vacuum such as space. Light travels at slower velocities in other materials such as glass. Light travelling from one material to another changes speed, which results in light changing its direction of travel. This deflection of light is called Refraction. The amount that a ray of light passing from a lower refractive index to a higher one is bent towards the normal. But light going from a higher index to a lower one refracting away from the normal, as shown in the figures.

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ø1

Angle of incidence

n1

n2

ø2

n1

n2

ø1

ø2

n1

n2

ø1 ø2

Angle ofreflection

Light is bent away from normal

Light does not enter second material

Fig. 3

As the angle of incidence increases, the angle of refraction approaches 90o to the normal.

The angle of incidence that yields an angle of refraction of 90o is the critical angle. If the

angle of incidence increases amore than the critical angle, the light is totally reflected back

into the first material so that it does not enter the second material. The angle of incidence

and reflection are equal and it is called Total Internal Reflection.

6.0 PROPAGATION OF LIGHT THROUGH FIBRE

The optical fibre has two concentric layers called the core and the cladding. The inner core

is the light carrying part. The surrounding cladding provides the difference refractive index

that allows total internal reflection of light through the core. The index of the cladding is

less than 1%, lower than that of the core. Typical values for example are a core refractive

index of 1.47 and a cladding index of 1.46. Fibre manufacturers control this difference to

obtain desired optical fibre characteristics. Most fibres have an additional coating around

the cladding. This buffer coating is a shock absorber and has no optical properties affecting

the propagation of light within the fibre. Figure shows the idea of light travelling through a

fibre. Light injected into the fibre and striking core to cladding interface at grater than the

critical angle, reflects back into core, since the angle of incidence and reflection are equal,

the reflected light will again be reflected. The light will continue zigzagging down the

length of the fibre. Light striking the interface at less than the critical angle passes into the

cladding, where it is lost over distance. The cladding is usually inefficient as a light carrier,

and light in the cladding becomes attenuated fairly. Propagation of light through fibre is

governed by the indices of the core and cladding by Snell's law.

Such total internal reflection forms the basis of light propagation through a optical fibre.

This analysis consider only meridional rays- those that pass through the fibre axis each

time, they are reflected. Other rays called Skew rays travel down the fibre without passing

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through the axis. The path of a skew ray is typically helical wrapping around and around

the central axis. Fortunately skew rays are ignored in most fibre optics analysis.

The specific characteristics of light propagation through a fibre depends on many factors,

including

- The size of the fibre. - The composition of the fibre.

- The light injected into the fibre.

Jacket

CladdingCore

Cladding

Angle of reflection

Angle of incidence

Light at less thancritical angle isabsorbed in jacket

Jacket

Light is propagated by total internal reflection

Jacket

Cladding

Core

(n2)

(n2)

Fig. 4 Propagation of light through fiber

7.0 Geometry of Fiber A hair-thin fiber consist of two concentric layers of high-purity silica glass the

core and the cladding, which are enclosed by a protective sheath as shown in Fig. 5. Light

rays modulated into digital pulses with a laser or a light-emitting diode moves along the

core without penetrating the cladding.

Fig. 5 Geometry of fiber

The light stays confined to the core because the cladding has a lower refractive index—a

measure of its ability to bend light. Refinements in optical fibers, along with the

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development of new lasers and diodes, may one day allow commercial fiber-optic networks

to carry trillions of bits of data per second.

The diameters of the core and cladding are as follows.

Core (µm) Cladding (µ m)

8 125

50 125

62.5 125

100 140

125 8 125 50 125 62.5 125 100

Core Cladding

Typical Core and Cladding Diameters Fibre sizes are usually expressed by first giving the core size followed by the cladding size.

Thus 50/125 means a core diameter of 50µm and a cladding diameter of 125µm.

8.0 FIBRE TYPES

The refractive Index profile describes the relation between the indices of the core and

cladding. Two main relationship exists :

(I) Step Index

(II) Graded Index

The step index fibre has a core with uniform index throughout. The profile shows a sharp

step at the junction of the core and cladding. In contrast, the graded index has a non-

uniform core. The Index is highest at the center and gradually decreases until it matches

with that of the cladding. There is no sharp break in indices between the core and the

cladding.

By this classification there are three types of fibres :

(I) Multimode Step Index fibre (Step Index fibre)

(II) Multimode graded Index fibre (Graded Index fibre)

(III) Single- Mode Step Index fibre (Single Mode Fibre)

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8.1 STEP-INDEX MULTIMODE FIBER has a large core, up to 100 microns in

diameter. As a result, some of the light rays that make up the digital pulse may travel a

direct route, whereas others zigzag as they bounce off the cladding. These alternative

pathways cause the different groupings of light rays, referred to as modes, to arrive

separately at a receiving point. The pulse, an aggregate of different modes, begins to

spread out, losing its well-defined shape. The need to leave spacing between pulses to

prevent overlapping limits bandwidth that is, the amount of information that can be sent.

Consequently, this type of fiber is best suited for transmission over short distances, in an

endoscope, for instance.

Fig. 6 STEP-INDEX MULTIMODE FIBER

8.2 GRADED-INDEX MULTIMODE FIBER contains a core in which the

refractive index diminishes gradually from the center axis out toward the cladding. The

higher refractive index at the center makes the light rays moving down the axis advance

more slowly than those near the cladding.

Fig.7 GRADED-INDEX MULTIMODE FIBER

Also, rather than zigzagging off the cladding, light in the core curves helically

because of the graded index, reducing its travel distance. The shortened path and the

higher speed allow light at the periphery to arrive at a receiver at about the same time as

the slow but straight rays in the core axis. The result: a digital pulse suffers less

dispersion.

8.3 SINGLE-MODE FIBER has a narrow core (eight microns or less), and the

index of refraction between the core and the cladding changes less than it does

for multimode fibers. Light thus travels parallel to the axis, creating little pulse

dispersion. Telephone and cable television networks install millions of kilometers

of this fiber every year.

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Fig. 8 SINGLE-MODE FIBER

9.0 OPTICAL FIBRE PARAMETERS Optical fiber systems have the following parameters.

(I) Wavelength.

(II) Frequency.

(III) Window.

(IV) Attenuation.

(V) Dispersion.

(VI) Bandwidth.

9.1 WAVELENGTH

It is a characterstic of light that is emitted from the light source and is measures in nanometers (nm). In the visible spectrum, wavelength can be described as the colour of the light.

For example, Red Light has longer wavelength than Blue Light, Typical wavelength for fibre use are 850nm, 1300nm and 1550nm all of which are invisible.

9.2 FREQUENCY

It is number of pulse per second emitted from a light source. Frequency is measured in units of hertz (Hz). In terms of optical pulse 1Hz = 1 pulse/ sec.

9.3 WINDOW

A narrow window is defined as the range of wavelengths at which a fibre best operates. Typical windows are given below :

Window Operational Wavelength

800nm - 900nm 850nm

1250nm - 1350nm 1300nm

1500nm - 1600nm 1550nm

9.4 ATTENUATION

Attenuation is defined as the loss of optical power over a set distance, a fibre with lower

attenuation will allow more power to reach a receiver than fibre with higher attenuation.

Attenuation may be categorized as intrinsic or extrinsic.

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9.4.1 INTRINSIC ATTENUATION

It is loss due to inherent or within the fibre. Intrinsic attenuation may occur as

Absorption - Natural Impurities in the glass absorb light energy.

Fig. 9 Absorption of Light

(1) Scattering - Light Rays Travelling in the Core Reflect from small Imperfections

into a New Pathway that may be Lost through the cladding.

LightRay

Light is lost

Fig. 10 Scattering

9.4.2 EXTRINSIC ATTENUATION

It is loss due to external sources. Extrinsic attenuation may occur as –

(I) Macrobending - The fibre is sharply bent so that the light travelling down the fibre cannot make the turn & is lost in the cladding.

Fig. 11 Micro and Macro bending

(II) Microbending - Microbending or small bends in the fibre caused by crushing contraction etc. These bends may not be visible with the naked eye.

Attenuation is measured in decibels (dB). A dB represents the comparison between the

transmitted and received power in a system.

9.5 BANDWIDTH

LightRay

115

It is defined as the amount of information that a system can carry such that each pulse of

light is distinguishable by the receiver.

System bandwidth is measured in MHz or GHz. In general, when we say that a system

has bandwidth of 20 MHz, means that 20 million pulses of light per second will travel

down the fibre and each will be distinguishable by the receiver.

9.6 NUMBERICAL APERTURE

Numerical aperture (NA) is the "light - gathering ability" of a fibre. Light injected into

the fibre at angles greater than the critical angle will be propagated. The material NA

relates to the refractive indices of the core and cladding.

NA = n12 - n22

where n1 and n2 are refractive indices of core and cladding respectively.

NA is unitless dimension. We can also define as the angles at which rays will be

propagated by the fibre. These angles form a cone called the acceptance cone, which

gives the maximum angle of light acceptance. The acceptance cone is related to the NA

∅ = arc sing (NA) or

NA = sin ∅

where ∅ is the half angle of acceptance

The NA of a fibre is important because it gives an indication of how the fibre accepts and

propagates light. A fibre with a large NA accepts light well, a fibre with a low NA

requires highly directional light.

In general, fibres with a high bandwidth have a lower NA. They thus allow fewer modes

means less dispersion and hence greater bandwidth. A large NA promotes more modal

dispersion, since more paths for the rays are provided NA, although it can be defined for

a single mode fibre, is essentially meaningless as a practical characteristic. NA in a

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multimode fibre is important to system performance and to calculate anticipated

performance.

Fig. 12 Numerical Aperture of fiber

* Light Ray A : Did not Enter Acceptance Cone - Lost

* Light Ray B : Entered Acceptance Cone - Transmitted through the Core by Total Internal Reflection.

9.7 DISPERSION

Dispersion is the spreading of light pulse as its travels down the length of an optical fibre as shown in figure 13. Dispersion limits the bandwidth or information carrying capacity of a fibre. The bit-rates must be low enough to ensure that pulses are farther apart and therefore the greater dispersion can be tolerated.

There are three main types of dispersion in a fibre -

(I) Modal Dispersion

(II) Material dispersion

(III) Waveguide dispersion

Fig. 13 Dispersion

9.8 BANDWIDTH AND DISPERSION :

A bandwidth of 400 MHz -km means that a 400 MHz-signal can be transmitted for 1 km.

It means that the product of frequency and the length must be 400 or less. We can send a

lower frequency for a longer distance, i.e. 200 MHz for 2 km or 100 MHz for 4 km.

Multimode fibres are specified by the bandwidth-length product or simply bandwidth.

117

Single mode fibres on the other hand are specified by dispersion, expressed in ps/km/nm.

In other words for any given single mode fibre dispersion is most affected by the source's

spectral width. The wider the source spectral width, the greater the dispersion.

Conversion of dispersion to bandwidth can be approximated roughly by the following

equation.

0.187

BW = --------------------------

(Disp) (SW) (L)

Disp = Dispersion at the operating wavelength in seconds/ nm/ km.

SW = Spectral width of the source in nm.

L = Fibre length in km.

So the spectral width of the source has a significant effect on the performance of a single mode fibre.

9.9 OPTICAL WINDOWS :

Attenuation of fibre for optical power varies with the wavelengths of light. Windows are

low-loss regions, where fiber carry light with little attenuation. The first generation of

optical fibre operated in the first window around 820 to 850 nm. The second window is

the zero-dispersion region of 1300 nm and the third window is the 1550 nm region as

shown in figure 14.

Fig. 14 Optical Windows

10.0 CABLE CONSTRUCTION There are two basic cable designs are:

1. Tight Buffer Tube Cable

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2. Loose Buffer Tube Cable

Loose-tube cable, used in the majority of outside-plant installations and tight-buffered cable, primarily used inside buildings.

10.1 Tight-Buffered Cable With tight-buffered cable designs, the buffering material is in direct contact with

the fiber. This design is suited for "jumper cables" which connect outside plant cables to

terminal equipment, and also for linking various devices in a premises network. Single-

fiber tight-buffered cables are used as pigtails, patch cords and jumpers to terminate

loose-tube cables directly into opto-electronic transmitters, receivers and other active and

passive components.

Multi-fiber tight-buffered cables also are available and are used primarily for

alternative routing and handling flexibility and ease within buildings.The tight-buffered

design provides a rugged cable structure to protect individual fibers during handling,

routing and connectorization. Yarn strength members keep the tensile load away from the

fiber.

Fig. 15 Tight Buffer Tube Cable

10.2 Loose-Tube Cable The modular design of loose-tube cables typically holds 6, 12, 24, 48, 96 or even

more than 400 fibers per cable. Loose-tube cables can be all-dielectric or optionally

armored. The loose-tube design also helps in the identification and administration of

fibers in the system.

In a loose-tube cable design, color-coded plastic buffer tubes house and protect

optical fibers. A gel filling compound impedes water penetration. Excess fiber length

(relative to buffer tube length) insulates fibers from stresses of installation and

119

environmental loading. Buffer tubes are stranded around a dielectric or steel central

member, which serves as an anti-buckling element.

The cable core, typically uses aramid yarn, as the primary tensile strength

member. The outer polyethylene jacket is extruded over the core. If armoring is required,

a corrugated steel tape is formed around a single jacketed cable with an additional jacket

extruded over the armor.

Loose-tube cables typically are used for outside-plant installation in aerial, duct

and direct-buried applications.

Here are some common fiber cables types are given below:

10.2. 1 Distribution Cable

Distribution Cable (compact building cable) packages individual 900µm buffered

fiber reducing size and cost. The connectors may be installed directly on the 900µm

buffered fiber at the breakout box location.

Fig. 16 Distribution Cable

10.2.2 Loose Tube Cable

Loose tube cable is designed to endure outside temperatures and high moisture

conditions. The fibers are loosely packaged in gel filled buffer tubes to repel water.

Recommended for use between buildings that are unprotected from outside elements.

Loose tube cable is restricted from inside building use.

120

Fig.17 Loose Tube Cable

10.2.3 Aerial Cable/Self-Supporting

Aerial cable provides ease of installation and reduces time and cost. Figure 8

cable can easily be separated between the fiber and the messenger. Temperature range

(-55ºC to +85ºC)

Fig. 18 Aerial Cable/Self-Supporting

10.2.4 Hybrid & Composite Cable

Hybrid cables offer the same great benefits as our standard indoor/outdoor cables,

with the convenience of installing multimode and single mode fibers all in one pull. Our

composite cables offer optical fiber along with solid 14 gauge wires suitable for a variety

of uses including power, grounding and other electronic controls

Fig. 19 Hybrid & Composite Cable

10.2.5 Armored Cable

Armored cable can be used for rodent protection in direct burial if required. This

cable is non-gel filled and can also be used in aerial applications. The armor can be

121

removed leaving the inner cable suitable for any indoor/outdoor use. (Temperature rating

-40ºC to +85ºC)

Fig. 20 Armored Cable

Fibre Optic Cables (Loose Buffer Tube) have the following parts in common ;

(I) Optical Fibre

(II) Buffer

(III) Strength member

(IV) Jacket

Table-1 Cable Components

Component Function Material

Buffer Protect fibre From Outside Nylon, Mylar, Plastic

Central Member

Facilitate Stranding

Temperature Stability

Anti-Buckling

Steel, Fibreglass

Primary Strength Member Tensile Strength Aramid Yarn, Steel

Cable Jacket Contain and Protect

Cable Core PE, PUR, PVC, Teflon

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Abrasion Resistance

Cable Filling

Compound

Prevent Moisture

intrusion and Migration

Water Blocking

Compound

Armoring Rodent Protection

Crush Resistance Steel Tape

11.0 CABLE DRUM LENGTH :

Cables come reeled in various length, typically 1 to 2 km, although lengths of 5 or 6 kms

are available for single mode fibres. Long lengths are desirables for long distance

applications, since cable must be spliced end to end over the run. Each splice introduce

additional loss into the system. Long cable lengths mean fewer splices and less loss.

12.0 OFC Splicing Splices are permanent connection between two fibres. The splicing involves

cutting of the edges of the two fibres to be spliced.

Splicing Methods

The following three types are widely used :

1. Adhesive bonding or Glue splicing.

2. Mechanical splicing.

3. Fusion splicing.

12.1 Adhesive Bonding or Glue Splicing

This is the oldest splicing technique used in fibre splicing. After fibre end

preparation, it is axially aligned in a precision V–groove. Cylindrical rods or another kind

of reference surfaces are used for alignment. During the alignment of fibre end, a small

amount of adhesive or glue of same refractive index as the core material is set between

and around the fibre ends. A two component epoxy or an UV curable adhesive is used as

the bonding agent. The splice loss of this type of joint is same or less than fusion splices.

But fusion splicing technique is more reliable, so at present this technique is very rarely

used.

12.2 Mechanical Splicing

This technique is mainly used for temporary splicing in case of emergency

repairing. This method is also convenient to connect measuring instruments to bare fibres

for taking various measurements.

The mechanical splices consist of 4 basic components :

(i) An alignment surface for mating fibre ends.

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(ii) A retainer

(iii) An index matching material.

(iv) A protective housing

A very good mechanical splice for M.M. fibres can have an optical performance

as good as fusion spliced fibre or glue spliced. But in case of single mode fibre, this type

of splice cannot have stability of loss.

12.3 Fusion Splicing

The fusion splicing technique is the most popular technique used for achieving

very low splice losses. The fusion can be achieved either through electrical arc or through

gas flame.

The process involves cutting of the fibres and fixing them in micro–positioners on

the fusion splicing machine. The fibres are then aligned either manually or automatically

core aligning (in case of S.M. fibre) process. Afterwards the operation that takes place

involve withdrawal of the fibres to a specified distance, preheating of the fibre ends

through electric arc and bringing together of the fibre ends in a position and splicing

through high temperature fusion.

If proper care taken and splicing is done strictly as per schedule, then the splicing

loss can be minimized as low as 0.01 dB/joint. After fusion splicing, the splicing joint

should be provided with a proper protector to have following protections:

(a) Mechanical protection

(b) Protection from moisture.

Sometimes the two types of protection are combined. Coating with Epoxy resins

protects against moisture and also provides mechanical strength at the joint.

Now–a–days, the heat shrinkable tubes are most widely used, which are fixed on

the joints by the fusion tools.

The fusion splicing technique is the most popular technique used for achieving very low splice losses. The introduction of single mode optical fibre for use in long haul network brought with it fibre construction and cable design different from those of multimode fibres.

The splicing machines imported by BSNL begins to the core profile alignment system, the main functions of which are :

(1) Auto active alignment of the core.

(2) Auto arc fusion.

(3) Video display of the entire process.

(4) Indication of the estimated splice loss.

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The two fibres ends to be spliced are cleaved and then clamped in accurately

machined vee–grooves. When the optimum alignment is achieved, the fibres are fused

under the microprocessor contorl, the machine then measures the radial and angular off–

sets of the fibres and uses these figures to calculate a splice loss. The operation of the

machine observes the alignment and fusion processes on a video screens showing

horizontal and vertical projection of the fibres and then decides the quality of the splice.

The splice loss indicated by the splicing machine should not be taken as a final

value as it is only an estimated loss and so after every splicing is over, the splice loss

measurement is to be taken by an OTDR (Optical Time Domain Reflectometer). The

manual part of the splicing is cleaning and cleaving the fibres. For cleaning the fibres,

Dichlorine Methyl or Acetone or Alcohol is used to remove primary coating.

With the special fibre cleaver or cutter, the cleaned fibre is cut. The cut has to be

so precise that it produces an end angle of less than 0.5 degree on a prepared fibre. If the

cut is bad, the splicing loss will increase or machine will not accept for splicing. The

shape of the cut can be monitored on the video screen, some of the defect noted while

cleaving are listed below :

(i) Broken ends.

(ii) Ripped ends.

(iii) Slanting cuts.

(iv) Unclean ends.

It is also desirable to limit the average splice loss to be less than 0.1 dB.

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CHAPTER - 7 OF TRANSMISSION SYSTEMS & THEIR FEATURES

1.0 INTRODUCTION

With the introduction of PCM technology in the 1960s, communications networks were

gradually converted to digital technology over the next few years. To cope with the

demand for ever higher bit rates, a multiplex hierarchy called the plesiochronous digital

hierarchy (PDH) evolved. The bit rates start with the basic multiplex rate of 2 Mbit/s with

further stages of 8, 34 and 140 Mbit/s. In North America and Japan, the primary rate is

1.5 Mbit/s. Hierarchy stages of 6 and 44 Mbit/s developed from this. Because of these

very different developments, gateways between one network and another were very

difficult and expensive to realize. PCM allows multiple use of a single line by means of

digital time-domain multiplexing. The analog telephone signal is sampled at a bandwidth

of 3.1 kHz, quantized and encoded and then transmitted at a bit rate of 64 kbit/s. A

transmission rate of 2048 kbit/s results when 30 such coded channels are collected

together into a frame along with the necessary signaling information. This so-called

primary rate is used throughout the world. Only the USA, Canada and Japan use a

primary rate of 1544 kbit/s, formed by combining 24 channels instead of 30. The growing

demand for more bandwidth meant that more stages of multiplexing were needed

throughout the world. A practically synchronous (or, to give it its proper name:

plesiochronous) digital hierarchy is the result. Slight differences in timing signals mean

that justification or stuffing is necessary when forming the multiplexed signals. Inserting

or dropping an individual 64 kbit/s channel to or from a higher digital hierarchy requires

a considerable amount of complex multiplexer equipment.

Traditionally, digital transmission systems and hierarchies have been based on

multiplexing signals which are plesiochronous (running at almost the same speed). Also,

various parts of the world use different hierarchies which lead to problems of

international interworking; for example, between those countries using 1.544 Mbit/s

systems (U.S.A. and Japan) and those using the 2.048 Mbit/s system. To recover a 64

kbit/s channel from a 140 Mbit/s PDH signal, it’s necessary to demultiplex the signal all

the way down to the 2 Mbit/s level before the location of the 64 kbit/s channel can be

identified. PDH requires “steps” (140-34, 34-8, 8-2 demultiplex; 2-8, 8-34, 34-140

multiplex) to drop out or add an individual speech or data channel (see Figure 1).

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Fig. 1 Plesiochronous Digital Hierarchies (PDH)

The main problems of PDH systems are:

1. Homogeneity of equipment

2. Problem of Channel segregation

3. The problem cross connection of channels

4. Inability to identify individual channels in a higher-order bit stream.

5. Insufficient capacity for network management;

6. Most PDH network management is proprietary.

7. There’s no standardized definition of PDH bit rates greater than 140 Mb/s.

8. There are different hierarchies in use around the world. Specialized interface equipment is required to interwork the two hierarchies.

1988 SDH standard introduced with three major goals:

– Avoid the problems of PDH

– Achieve higher bit rates (Gbit/s)

– Better means for Operation, Administration, and Maintenance (OA&M)

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SDH is an ITU-T standard for a high capacity telecom network. SDH is a synchronous digital transport system, aim to provide a simple, economical and flexible telecom infrastructure. The basis of Synchronous Digital Hierarchy (SDH) is synchronous multiplexing - data from multiple tributary sources is byte interleaved.

SDH brings the following advantages to network providers:

1.1 High transmission rates

Transmission rates of up to 40 Gbit/s can be achieved in modern SDH systems. SDH is therefore the most suitable technology for backbones, which can be considered as being the super highways in today's telecommunications networks.

1.2 Simplified add & drop function

Compared with the older PDH system, it is much easier to extract and insert low-bit rate channels from or into the high-speed bit streams in SDH. It is no longer necessary to demultiplex and then remultiplex the plesiochronous structure.

1.3 High availability and capacity matching

With SDH, network providers can react quickly and easily to the requirements of their

customers. For example, leased lines can be switched in a matter of minutes. The network

provider can use standardized network elements that can be controlled and monitored

from a central location by means of a telecommunications network management (TMN)

system.

1.4 Reliability

Modern SDH networks include various automatic back-up and repair mechanisms to

cope with system faults. Failure of a link or a network element does not lead to failure of

the entire network which could be a financial disaster for the network provider. These

back-up circuits are also monitored by a management system.

1.5 Future-proof platform for new services

Right now, SDH is the ideal platform for services ranging from POTS, ISDN and mobile

radio through to data communications (LAN, WAN, etc.), and it is able to handle the

very latest services, such as video on demand and digital video broadcasting via ATM

that are gradually becoming established.

1.6 Interconnection

SDH makes it much easier to set up gateways between different network providers and to

SONET systems. The SDH interfaces are globally standardized, making it possible to

combine network elements from different manufacturers into a network. The result is a

reduction in equipment costs as compared with PDH.

2.0 Network Elements of SDH

Figure 2 is a schematic diagram of a SDH ring structure with various tributaries. The

mixture of different applications is typical of the data transported by SDH. Synchronous

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networks must be able to transmit plesiochronous signals and at the same time be capable

of handling future services such as ATM.

Current SDH networks are basically made up from four different types of network

element. The topology (i.e. ring or mesh structure) is governed by the requirements of the

network provider.

2.1 Regenerators

Regenerators as the name implies, have the job of regenerating the clock and amplitude

relationships of the incoming data signals that have been attenuated and distorted by

dispersion. They derive their clock signals from the incoming data stream. Messages are

received by extracting various 64 kbit/s channels (e.g. service channels E1, F1) in the

RSOH (regenerator section overhead). Messages can also be output using these channels.

2.2 Terminal Multiplexer

Terminal multiplexers Terminal multiplexers are used to combine plesiochronous and

synchronous input signals into higher bit rate STM-N signals.

Fig. 2 Schematic diagram of hybrid communications networks

2.3 Add/drop Multiplexers(ADM)

Add/drop multiplexers (ADM) Plesiochronous and lower bit rate synchronous signals can

be extracted from or inserted into high speed SDH bit streams by means of ADMs. This

feature makes it possible to set up ring structures, which have the advantage that

automatic back-up path switching is possible using elements in the ring in the event of a

fault.

2.4 Digital Cross-connect

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Digital cross-connects (DXC) This network element has the widest range of functions. It

allows mapping of PDH tributary signals into virtual containers as well as switching of

various containers up to and including VC-4.

2.5 Network Element Manager

Network element management The telecommunications management network (TMN) is

considered as a further element in the synchronous network. All the SDH network

elements mentioned so far are software-controlled. This means that they can be

monitored and remotely controlled, one of the most important features of SDH. Network

management is described in more detail in the section “TMN in the SDH network”

3.0 SDH Rates SDH is a transport hierarchy based on multiples of 155.52 Mbit/s. The basic unit of SDH

is STM-1. Different SDH rates are given below:

STM-1 = 155.52 Mbit/s

STM-4 = 622.08 Mbit/s

STM-16 = 2588.32 Mbit/s

STM-64 = 9953.28 Mbit/s

Each rate is an exact multiple of the lower rate therefore the hierarchy is synchronous.

4.0 Back-up network switching- Automatic protection switching (APS)

Modern society is virtually completely dependent on communications technology. Trying

to imagine a modern office without any connection to telephone or data networks is like

trying to work out how a laundry can operate without water. Network failures, whether

due to human error or faulty technology, can be very expensive for users and network

providers alike. As a result, the subject of so-called fall-back mechanisms is currently one

of the most talked about in the SDH world. A wide range of standardized mechanisms is

incorporated into synchronous networks in order to compensate for failures in network

elements.

Two basic types of protection architecture are distinguished in APS. One is the linear

protection mechanism used for point-to-point connections. The other basic form is the so-

called ring protection mechanism which can take on many different forms. Both

mechanisms use spare circuits or components to provide the back-up path. Switching is

controlled by the overhead bytes K1 and K2.

4.1 Linear protection The simplest form of back-up is the so-called 1 + 1 APS. Here, each working line is

protected by one protection line. If a defect occurs, the protection agent in the network

elements at both ends switch the circuit over to the protection line. The switchover is

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triggered by a defect such as LOS. Switching at the far end is initiated by the return of an

acknowledgment in the backward channel. 1+1 architecture includes 100% redundancy,

as there is a spare line for each working line. Economic considerations have led to the

preferential use of 1:N architecture, particularly for long-distance paths. In this case,

several working lines are protected by a single back-up line. If switching is necessary, the

two ends of the affected path are switched over to the back-up line. The 1+1 and 1:N

protection mechanisms are standardized in ITU-T Recommendation G.783. The reserve

circuits can be used for lower-priority traffic, which is simply interrupted if the circuit is

needed to replace a failed working line.

Fig 3 Linear protection

4.2 Ring protection

The greater the communications bandwidth carried by optical fibers, the greater the cost

advantages of ring structures as compared with linear structures. A ring is the simplest

and most cost-effective way of linking a number of network elements. Various protection

mechanisms are available for this type of network architecture, only some of which have

been standardized in ITU-T Recommendation G.841. A basic distinction must be made

between ring structures with unidirectional and bi-directional connections.

4.2.1 Unidirectional rings

Figure 4 shows the basic principle of APS for unidirectional rings. Let us assume that

there is an interruption in the circuit between the network elements A and B. Direction y

is unaffected by this fault. An alternative path must, however, be found for direction x.

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Figure 4: Two fiber unidirectional path switched ring

The connection is therefore switched to the alternative path in network elements A and B.

The other network elements (C and D) switch through the back-up path. This switching

process is referred to as line switched. A simpler method is to use the so-called path

switched ring (see figure 4). Traffic is transmitted simultaneously over both the working

line and the protection line. If there is an interruption, the receiver (in this case A)

switches to the protection line and immediately takes up the connection.

4.2.2 Bi-directional rings

In this network structure, connections between network elements are bi-directional. This

is indicated in figure 5 by the absence of arrows when compared with figure 5. The

overall capacity of the network can be split up for several paths each with one bi-

directional working line, while for unidirectional rings, an entire virtual ring is required

for each path. If a fault occurs between neighboring elements A and B, network element

B triggers protection switching and controls network element A by means of the K1 and

K2 bytes in the SOH.

Even greater protection is provided by bi-directional rings with 4 fibers. Each pair of

fibers transports working and protection channels. This results in 1:1 protection, i.e. 100

% redundancy. This improved protection is coupled with relatively high costs.

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Figure 5: Two fiber bi-directional line-switched ring (BLSR)

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CHAPTER - 8

SDH

INTRODUCTION Synchronous digital hierarchy (SDH) is an ITU-T universal standard that defines acommon and reliable architecture for transporting telecommunications services ona worldwide scale. Synchronous optical network (SONET) is today a subset ofSDH, promoted by American National Standards Institute (ANSI) and used in theU.S., Canada, Taiwan, and Korea.From now on we will use the acronym SDH to refer to the generic ITU-T standardthat includes also SONET.

THE EMERGENCE OF SDH/SONET NETWORKS

During the 1980s, progress in optical technologies and microprocessors offered new challenges to telecommunications in terms of bandwidth and data processing .At that time, plesiochronous hierarchies (T-carrier and PDH) dominated transport systems, but a series of limitations and the necessity to introduce new transmission technologies moved to develop a new architecture .Antitrust legislation was the final factor that hastened the development of SONET.It was applied to the telecommunications business and forced the giant, Bell,to be split up into small companies, the regional Bell operating companies(RBOCs). SONET, developed at Bellcore labs in 1984, grew out of the need to interconnect RBOCs using standardized optical interfaces. Telecom liberalization was confirmed around the world during the ‘90s, and this has inevitably led to global competition and interoperation. In 1988, the Comité Consultatif InternationalTélégraphique Et Téléphonique CCITT (now ITU) proposed creating broadband-ISDN (B-ISDN) to simultaneously transport data, voice, video, and multimedia over common transmission infrastructures. Asynchronous transfer mode (ATM) was selected for the switching layer, and SDH for transport at the physical layer.

Limitations of Plesiochronous Networks

Plesiochronous networks have the following limitations: • Their management, supervision, and maintenance capabilities are limited, as there are no overhead bytes to support these functions. One example of this is that if a resource fails, there is no standard function whereby the network can be reconfigured. • Access to 64-Kbps digital channels from higher PDH hierarchical signals requires full demultiplexing, because the use of bit-oriented procedures removes any trace of the channels. • In PDH it was not possible to create higher bit rates directly; one could do so only after following all the steps and hierarchies (see Figure 1). • Plesiochronous ANSI and European Telecommunications Standard Institute (ETSI) hierarchies were not compatible. • There were no standards defined for rates over 45 Mbps in T-carrier, and over 140 Mbps in PDH. • Different manufacturers of plesiochronous equipment could not always be interconnected, because they implemented additional management channels or proprietary bit rates.

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Figure 1.1 SDH and SONET allow for direct multiplexing and demultiplexing. These limitations meant that it was necessary to design a new transmission architecture to increase the flexibility, functionality, reliability, and interoperability of networks.

The SDH/SONET Challenge

What had to be decided first was how to provide smooth migration from legacy installations. Hence a basic frame period of 125 µs was selected, the same of E1 and 1 frames, in order to guarantee compatibility with existing services such as plain old telephone service (POTS), integrated services digital network (ISDN), frame relay (FRL) or any n x 64 Kbps . Note that a byte constantly carried on a 125-µs frame period defines a 64-Kbps channel.

Some of the remarkable features of SDH compared with its predecessors are:

Synchronous versus plesiochronous Plesiochronous means “almost synchronous.” This in its turn means that nodes try to do work in the same frequency, but in fact they do not, because each PDH island use its own clock. In synchronous networks, all digital transitions should occur simultaneously, and all the nodes must be fed with the same master clock . There may, however, be a phase difference between the transitions of the two signals but this must lie within standardized limits.

Bytes versus bits In SDH and SONET, such basic operations as multiplexing, mapping, or alignment are byte oriented, to keep transported elements identified throughout the wholetransmission path

Direct access The main difference between SDH and its predecessors is in synchronization and byte-oriented operations. Synchronization enables us to insert and extract tributaries directly at any point and at any bit rate, without delay or extra hardware. For this reason, PDH/T-carrier must completely demultiplex signals of various megabits per second, to access any embedded channel of n x 64 Kbps.

Full management In SDH and SONET, payload and overheads are always accessible, and there is no need to demultiplex the signal. This drastically improves operation, administration, and maintenance (OA&M) functions, which are essential to enable centralized management independently of the bit rate. SDH and SONET also provide embedded mechanisms to protect the network against link or node failures, to monitor network performance, and to manage network events.

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Providing circuits for public networks The basic function of SDH, just like any transmission network, is that of providing metropolitan or long-haul transport to networks such as POTS, ISDN, FRL, Gigabit Ethernet (10GbE), Universal Mobile Telecommunications System (UMTS) or Internet. Signaling, switching, routing, and billing do not depend on SDH, as it is only in charge of providing bandwidth between two points. Universal standard SDH and SONET standards enable transmission over multiple media, including fiber optics, radio, satellite, and electrical interfaces. They allow internetworking between equipment from different manufacturers by means of a set of generic standards and open interfaces. Scalability is also an important point, as transmission rates of up to 40 Gbps have been defined, making SDH a suitable technology for high-speed trunk networks.

COMPARISON OF SDH AND SONET SDH and SONET are compatible but not identical. SDH is used worldwide except in the U.S., Canada, Japan, and partially in South Korea, and Taiwan. Both define a similar set of structures and functions; however, there are differences in usage.

Table 1.1 Terminology comparison.

The 51.84-Mbps STS-1 is the basic building block of SONET (OC-1 if this signal is transmitted over fiber optics). STS-1 was enough to transport all T-carrier tribu- taries but not to carry the 140Mbps PDH tributary. Then SDH chose a basic transport frame three

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times faster at 155.53 Mbps, in order to allocate the full European hierarchy. SDH and SONET terminology have some differences, in referring to the same concepts, bytes, and structures. Nevertheless, beyond the names, the functionality is equivalent

Internetworking is always possible because the evolution of both technologies has been the same with the new hierarchies up to 40 Gbps and to the last standards like link capacity adjustment scheme (LCAS). The objective is to guarantee universal connectivity.

SDH/SONET Layers In plesiochronous networks, interactions are simple and direct. In synchronous networks they are more sophisticated, so responsibilities have been divided among several layers that communicate with their counterparts by making use of specific overheads, formats, and protocols. This architecture is equivalent to the layered open system interconnection (OSI) model to define and design communication networks

Figure 1.2 SDH standards define a layered client/server model that can be divided into

up to four layers in order to manage transmission services.

Path layers Path layers are the route to transport clients’ information across the synchronous network from its source to its destination, where the multiplexers interface with client equipment At this layer clients’ information is mapped/ demapped into a frame and path overhead is added. There are two specialized path layers

1. Lower-order path (LP), or virtual tributary path (VT Path) in SONET, to transport lower-rate services. Associated overhead is lower-order path overhead (LO-POH) or virtual tributary path overhead (VT-POH) in SONET.

2. Higher-order path (HP), synchronous transport signal path (STS Path) in SONET, to transport higher-rate services or a combination of lower-rate services. Associated overhead is higher-order path overhead HO-POH or synchronous transport signal path overhead (STS-POH) in SONET.

Some of the path layer functions are routing, performance monitoring, anomalies and defect management, security and protection, as well as specific path OAM functions support.

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Figure1.3 SDH and SONET line topology and network layers.

Multiplex section or line layer

Multiplexer section (MS), or line section in SONET, is a route between two adjacent multiplexers. This layer has several capabilities such as bit error detection, and circuit protection when an intermediate link or node collapses. It also carries synchronization reference information and OAM information between nodes. Associated overhead is multiplex section overhead (MSOH) or line overhead (LOH) in SONET.

Regeneration section or section layer

The regeneration section (RS), (the section layer in SONET) is the link between two successive NEs. It reads and writes specific overheads and management functions for each type of transmission media. Its most typical functions are framing, bit error detection, and regenerator OAM functions support. Associated overhead is regeneration section overhead (RSOH) or section overhead(SOH) in SONET

Physical layers

Fiber optics and metallic cable, together with terrestrial radio and satellite links can be used as the physical layer (PL). Fiber optics is the most common medium because of its capacity and reliability. Radio is a cost-effective medium when distance, geographical difficulties, or low-density areas make the optical alternative less practical. Nevertheless, radio has some important weaknesses; for example, noise and frequency allocation, that limit bit rates to 622 Mbps. Electrical cables are also used in some legacy installations

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SDH/SONET FORMATS AND PROCEDURES SDH defines a set of structures to transport adapted payloads over physical

transmission networks (ITU-T Rec. G.707). Five basic procedures are involved here (see

Figure 1.4):

• Mapping: A procedure by which tributaries are adapted into virtual containers at the

boundary of an SDH network.

• Stuffing: This is a mapping procedure to adapt the bit rate of client data streams into

standardized, fixed-size containers.

• Multiplexing: A procedure by which multiple lower-order signals are adapted into a

higher-order path signal, or when the higher-order path layer signals are adapted into a

multiplex section.

• Overhead addition: This procedure is to attach information bytes to a data signal for

internal routing and management.

• Aligning: A procedure by which a pointer is incorporated into a tributary unit (TU) or

an administrative unit (AU). TU and AU pointers are used to find units anywhere in the

transmission network.

Figure 1.4 SDH Multiplexing map.

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SDH/SONET Frame Structure

The basic transport frame in SONET is synchronous transport signal (STS-1), while in

SDH it is synchronous transmission module (STM-1) • STS-1 is a 3 x 9 byte structure

transmitted at 52 Mbps, which is equivalent to STM-0.

• STM-1 is a 9 x 9 byte structure transmitted at 155 Mbps, which is equivalent to optical

carrier 3 (OC-3) and electrical STS-3. Both have the same structure that is based on three

types of information blocks:

1. Overhead blocks: These blocks contain information that is used to manage quality,

anomalies, defects, data communication channels, service channels, and so on. There are

two types of overhead blocks, RSOH (managed by the regenerator section layer) and

MSOH (managed by the multiplex section layer).

2. Payload blocks or virtual containers (VCs): They contain a combination of client

signals and overhead blocks. VC does not have a fixed position in the frame, but it floats

in the frame to accommodate clock mismatches.

3. Pointers: They track the VC position, pointing to its first byte, while moving inside the

frame

Containers as transport interfaces

Containers (C-n) are used to map client bit streams. Adaptation procedures have been

defined to suit most telecom transport requirements. These include PDH, metropolitan

area network (MAN), asynchronous transfer mode (ATM), high-level data link control

(HDLC), internet protocol (IP), and Ethernet streams. Placing signals inside a container

requires a stuffing function to match the client stream with the container capacity. The

justification function is necessary for asynchronous mappings, to adapt clock differences

and fluctuations.

Virtual containers or virtual tributaries

Virtual containers (VC-n) or virtual tributaries (VTs) in SONET, support end-to-end path

layer connections; that is, between the point where the client stream is inserted into the

network and the point where it is delivered. Nobody is allowed to modify the VC

contents across the entire path. VCs consist of a C-n payload and a path overhead (POH).

Fields are organized into a block structure that repeats every 125 or 500 µs. Containers

hold client data, and the POH provides information to guarantee end-to-end data

integrity.

There are two types of VCs:

• The lower-order VC, such as VC-11, VC-12, VC-2, and VC-31. These consist of a

small container (C-11, C-12, C-2, and C-3), plus a 4-byte POH attached to the container

(9 bytes for VC-3).

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• The higher-order VC, such as VC-3 or VC-4. These consist of either a big container (C-

3, C-4) or an assembly of tributary unit groups (TUG-2, TUG-3). In both cases, a 9-byte

POH is attached.

Tributary units and tributary unit groups A tributary unit is a structure for adaptation

between the lower-order and higher-order path layer.

A TUG is an SDH signal made up of byte-interleaved multiplexing of one or more TUs.

In other cases, lower-order TUGs are multiplexed to form a higher-order TUG (for

instance, seven multiplexed TUG-2s form one TUG-3), and in other cases, a TUG is

formed by a single TU (for instance, a single TU-3 is enough to form a TUG-3). TUGs

occupy fixed positions in higher-order VCs.

2.4.1.4 Administrative unit

An administrative unit (AU-n) provides adaptation between the higher-order path layer

and the multiplex section layer. It consists of an HO-VC payload and an AU pointer

indicating the payload offset.

Multiplexing Map

A multiplexing map is a road map that shows how to transport and multiplex a number of

services in STM/OC frames • The client tributary (PDH, T-carrier, ATM, IP, Ethernet,

etc.) needs to be mapped into a C-n container, and a POH added to form a VC-n, or a VT

for SONET.

• The VC/VT is aligned with a pointer to match the transport signal rate. Pointers

together with VCs form TUs or AUs.

• A multiplexing process is the next step, whereby TUG-n and AUG-n groups are

created.

• When it comes to TUGs, they are multiplexed again to fill up a VC, synchronous

payload envelope (SPE) in SONET, and a new alignment operation is performed.

• Finally, an administrative unit group (AUG) is placed into the STM/OC transport frame.

NETWORK ELEMENTS AND TOPOLOGY

Network Elements

SDH systems make use of a limited number of network elements (NEs) within which all the installations are fitted

• Regenerators (REGs) or section terminating equipments (STEs): Every signal sent through any transmission medium (optical, electrical or radio-electrical) experiences attenuation, distortion, and noise. Regenerators supervise the re ceived data and restore the signal’s physical characteristics, including shape and synchronization. They also manage the monitoring and maintenance functions of the regenerator section (RS) or section in

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• Line terminal multiplexers (LTMUX) or path terminal equipment (PTE): they are common in line and access topologies. Their function is to insert and extract data in synchronous frames .

• Add and drop multiplexers (ADMs) can insert or extract data directly into or from the traffic that is passing across them, without demultiplexing/multiplexing the frame. Direct access to the contents of the frame is a key feature of SDH, as it enables us to turn any point of the network into a service node, just by installing an ADM.

• Digital cross-connects (DXCs) configure semipermanent connections to switch traffic between separate networks. The switched traffic can be either SDH streams or selected tributaries. Although it is not common, DXCs can also insert and drop tributaries in transport frames.

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Network Topology

Synchronous multiplexers provide great flexibility for building topologies, which is why point-to-point, linear, ring, hub, meshed, and mixed topologies are all possible:

• Linear point-to-multipoint: this topology follows the basic point-to-point structure, but now includes ADM multiplexers performing add and drop functions at intermediate points.

• Ring: this topology closes itself to cover a specific area, with ADM multiplexers installed at any point. It is flexible and scalable, which makes it very suit able for wide

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area and metropolitan networks. Rings are frequently used to build fault-tolerant architectures.

• Hub or star: this topology concentrates traffic at a central point, to make topology changes easier. A hub can join several networks with different topologies.

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CHAPTER - 9

DENSE WAVELENGTH DIVISION MULTIPLEXING

(DWDM) 1.0 INTRODUCTION The revolution in high bandwidth applications and the explosive growth of the Internet, however, have created capacity demands that exceed traditional TDM limits. To meet growing demands for bandwidth, a technology called Dense Wavelength Division Multiplexing (DWDM) has been developed that multiplies the capacity of a single fiber. DWDM systems being deployed today can increase a single fiber’s capacity sixteen fold, to a throughput of 40 Gb/s. The emergence of DWDM is one of the most recent and important phenomena in the development of fiber optic transmission technology. Dense wavelength-division multiplexing (DWDM) revolutionized transmission technology by increasing the capacity signal of embedded fiber. One of the major issues in the networking industry today is tremendous demand for more and more bandwidth. Before the introduction of optical networks, the reduced availability of fibers became a big problem for the network providers. However, with the development of optical networks and the use of Dense Wavelength Division Multiplexing (DWDM) technology, a new and probably, a very crucial milestone is being reached in network evolution. The existing SONET/SDH network architecture is best suited for voice traffic rather than today’s high-speed data traffic. To upgrade the system to handle this kind of traffic is very expensive and hence the need for the development of an intelligent all-optical network. Such a network will bring intelligence and scalability to the optical domain by combining the intelligence and functional capability of SONET/SDH, the tremendous bandwidth of DWDM and innovative networking software to spawn a variety of optical transport, switching and management related products.

In traditional optical fiber networks, information is transmitted through optical fiber by a single light beam. In a wavelength division multiplexing (WDM) network, the vast optical bandwidth of a fiber (approximately 30 THz corresponding to the low-loss region in a single-mode optical fiber) is carved up into wavelength channels, each of which carries a data stream individually. The multiple channels of information (each having a different carrier wavelength) are transmitted simultaneously over a single fiber. The reason why this can be done is that optical beams with different wavelengths propagate without interfering with one another. When the number of wavelength channels is above 20 in a WDM system, it is generally referred to as Dense WDM or DWDM.

DWDM technology can be applied to different areas in the telecommunication networks, which includes the backbone networks, the residential access networks, and also the Local Area Networks (LANs). Among these three areas, developments in the DWDM-based backbone network are leading the way, followed by the DWDM-based LANs. The development on DWDM-based residential access networks seems to be lagging behind at the current time.

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2.0 DEVELOPMENT OF DWDM TECHNOLOGY

Early WDM began in the late 1980s using the two widely spaced wavelengths in the 1310 nm and 1550 nm (or 850 nm and 1310 nm) regions, sometimes called wideband WDM. The early 1990s saw a second generation of WDM, sometimes called narrowband WDM, in which two to eight channels were used. These channels were now spaced at an interval of about 400 GHz in the 1550-nm window. By the mid-1990s, dense WDM (DWDM) systems were emerging with 16 to 40 channels and spacing from 100 to 200 GHz. By the late 1990s DWDM systems had evolved to the point where they were capable of 64 to 160 parallel channels, densely packed at 50 or even 25 GHz intervals.

As fig. 1 shows, the progression of the technology can be seen as an increase in the number of wavelengths accompanied by a decrease in the spacing of the wavelengths. Along with increased density of wavelengths, systems also advanced in their flexibility of configuration, through add-drop functions, and management capabilities.

Figure 1 Evolution of DWDM

3.0 VARIETIES of WDM

Early WDM systems transported two or four wavelengths that were widely spaced. WDM and the “follow-on” technologies of CWDM and DWDM have evolved well beyond this early limitation.

3.1 WDM

Traditional, passive WDM systems are wide-spread with 2, 4, 8, 12, and 16 channel counts being the normal deployments. This technique usually has a distance limitation of less than 100 km.

3.2 CWDM

Today, coarse WDM (CWDM) typically uses 20-nm spacing (3000 GHz) of up to 18 channels. The CWDM Recommendation ITU-T G.694.2 provides a grid of wavelengths for target distances up to about 50 km on single mode fibers as specified in ITU-T

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Recommendations G.652, G.653 and G.655. The CWDM grid is made up of 18 wavelengths defined within the range 1270 nm to 1610 nm spaced by 20 nm.

3.3 DWDM

Dense WDM common spacing may be 200, 100, 50, or 25 GHz with channel count reaching up to 128 or more channels at distances of several thousand kilometers with amplification and regeneration along such a route.

4.0 DWDM System Function

DWDM stands for Dense Wavelength Division Multiplexing, an optical technology used to increase Band width over existing fiber optic backbones. Dense wavelength division multiplexing systems allow many discrete transports channels by combining and transmitting multiple signals simultaneously at different wavelengths on the same fiber. In effect, one fiber is transformed into multiple virtual fibers. So, if you were to multiplex 32 STM-16 signals into one fiber, you would increase the carrying capacity of that fiber from 2.5 Gb/s to 80 Gb/s. Currently, because of DWDM, single fibers have been able to transmit data at speeds up to 400Gb/s.

A key advantage to DWDM is that it's protocol and bit rate-independent. DWDM-based networks can transmit data in SDH, IP, ATM and Ethernet etc. Therefore, DWDM-based networks can carry different types of traffic at different speeds over an optical channel. DWDM is a core technology in an optical transport network. Dense WDM common spacing may be 200, 100, 50, or 25 GHz with channel count reaching up to 128 or more channels at distances of several thousand kilometers with amplification and regeneration along such a route.

Fig. 2 Block Diagram of a DWDM System

The concepts of optical fiber transmission, loss control, packet switching, network topology and synchronization play a major role in deciding the throughput of the network.

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4.0 TRANSMISSION WINDOWS Today, usually the second transmission window (around 1300 nm) and the third

and fourth transmission windows from 1530 to 1565 nm (also called conventional band) and from 1565 to 1620 nm (also called Long Band) are used. Technological reasons limit DWDM applications at the moment to the third and fourth window.

The losses caused by the physical effects on the signal due by the type of materials used to produce fibres limit the usable wavelengths to between 1280 nm and 1650 nm. Within this usable range the techniques used to produce the fibres can cause particular wavelengths to have more loss so we avoid the use of these wavelengths as well.

5.0 DWDM SYSTEM COMPONENTS

Figure 3 shows an optical network using DWDM techniques that consists of five main components:

1. Transmitter (transmit transponder):

- Changes electrical bits to optical pulses

- Is frequency specific

- Uses a narrowband laser to generate the optical pulse

2. Multiplexer/ demultiplexer:

- Combines/separates discrete wavelengths

3. Amplifier:

- Pre-amplifier boosts signal pulses at the receive side

- Post-amplifier boosts signal pulses at the transmit side (post amplifier) and on the receive side (preamplifier)

- In line amplifiers (ILA) are placed at different distances from the source to provide recovery of the signal before it is degraded by loss.

- EDFA (Eribium Doped Fiber Amplifier) is the most popular amplifier.

4. Optical fiber (media):

- Transmission media to carry optical pulses

- Many different kinds of fiber are used

5. Receiver (receive transponder)

- Changes optical pulses back to electrical bits

- Uses wideband laser to provide the optical pulse

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Figure 3: DWDM System Components

5.1 BENEFITS of DWDM

• Increases bandwidth (speed and distance)

• Does not require replacement or upgrade their existing legacy systems

• Provides "next generation" technologies to meet growing data needs

• Less costly in the long run because increased fiber capacity is automatically available; don't have to upgrade all the time.

6.0 Fibers Supporting DWDM

For transmitting the DWDM signal, the conventional single mode optical fibers i.e. ITU G 652 compliant, are not completely suitable. Due to availability of Optical Amplifier working in 1550 nm region, the operating wavelengths are chosen in the C band i.e. from 1530 to 1565 nm. The ITU G 652 fiber has very high dispersion in 1550 nm region, which limits the distance between repeater stations severely.

ITU G 652 fiber with the high dispersion at 1550 nm, typically 18 ps/nm-km. Although, it is possible to compensate the dispersion by using dispersion compensating fibers (DCF), these DCF adds to additional optical loss.

Conversely, in case of ITU G 653 fibers with zero dispersion at 1550 nm, the nonlinearities such as Four Wave Mixing (FWM) plays dominant role, rendering the fiber unsuitable for long distance transmission. A fiber that has small but non-zero amount of dispersion can minimize the non-linearity effects. The ITU G 655 compliant, Non Zero Dispersion Fibers (NZDF) has dispersion which is carefully chosen to be small enough to enable high speed transmission over long distances, but large enough to suppress FWM. With the proper use and placement of Optical Amplifiers, it is possible to have the repeater less link.

In summary, a future-proof fiber optic network should have combination of ITU G 652 and G 655 fibers. The number of each type of fiber in a cable is generally chosen based on the type of network. The complete fiber optic network can be defined in broader manner in two parts:

(a) High capacity, long haul Backbone network or Transport network and (b) High Speed, Local Access network. The Backbone network connects the major cities of the networks and carries high bit rate signals so the G 655 fibers should be deployed in backbone. The Local Access network is used for carrying the data up to the customer

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premises. Due to smaller spacing between the stations / customer premises equipment, the standard single mode fibers (ITU G 652) can be deployed.

7.0 OPTICAL NE TYPES

Optical Multiplexer/Demultiplexer

Multiplexing and Demultiplexing of different wavelength signals.

(b) Optical Amplifiers

Pure optical 1R regeneration (just amplification) of all transmitted signals.

(c) Transponders

Wavelength “change” and 2R regeneration (reshaping and amplification) or 3 R regeneration (reshaping retiming and amplification).

(d) Regenerators

Real 3 R regeneration (reshaping, retiming and amplification) of the signal. Therefore, the signals have to be demultiplexed, electrically regenerated and multiplexed again. They are necessary if the length to be bridged is too long to be covered only by optical amplifiers, as these only perform reshaping and retiming.

(e) Optical Add/Drop Multiplexer

Adding and Dropping only specific wavelengths from the joint optical signal. This may use complete de-multiplexing or other techniques.

(f) Optical cross-connects

To cater for the huge amount of data expected in an optical network even the cross-connects have to work on a purely optical level.

8.0 DWDM Backbone Networks

We can divide the network structures of DWDM-based backbone networks into three classes:

• Simple point-point DWDM link,

• DWDM wavelength routing with electronic TDM (time domain multiplexing) and switching/routing backbone network, and

• All-optical DWDM network.

8.1 Point-To-Point DWDM links

The simplest application of the DWDM technology in backbone networks is the point-to-point link. Figure 4 shows the architecture of the networks using 4 network switching/routing nodes as an example. In this architecture, the electronic nodes can be SONET/SDH switches, Internet routers, ATM switches, or any other type network nodes. The DWDM node consists of typically a pair of wavelength multiplexer / demultiplexer (lightwave grating devices) and a pair of optical-electrical/ electrical-optical convertors. Each wavelength channel is used to transmit one stream of data individually. The DWDM wavelength multiplexer combines all of the lightwave channels into one light

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beam and pumps it into one single fiber. The combined light of multiple wavelengths is separated by the demultiplexer at the receiving end. The signals carried by each wavelength channel are then converted back to the electrical domain through the O/E convertors (photodetectors). In this way, one wavelength channel can be equivalent to a traditional fiber in which one lightbeam is used to carry information. It is worth noting that the wavelength channels in one fiber can be used for both directions or two fibers are used with each for one direction.

The advantage of the point-to-point DWDM links is that it increases the bandwidth by creating multiple channels with low costs. The limitation of this approach, however, is that the bandwidth of each wavelength channel may not be fully utilized due to the speed of the electrical devices, which is referred to as the well-known electro-optic bottleneck. Also, the use of the wavelength channels may not be optimal due to the fact that the meshes formed by the wavelength channel are all identical, which can be seen in Figure 4.

Figure 4: DWDM point-to-point link backbone network

8.2 Wavelength routing with electronic TDM

Figure 5 depicts the second type of DWDM application in backbone networks, in which wavelength routers are used to configure or reconfigure the network topology within the optical domain and the TDM (Time Domain Multiplexing) network nodes are used to perform multiplexing and switching in the electrical domain. This combined optical and electrical network architecture can be applied in SONET/SDH in which the electrical TDM network nodes would be SONET switches, or in the Internet in which the electrical TDM network nodes would be the Internet routers. The architecture can also be used in an ATM network where the electrical TDM network nodes would be ATM switches.

The advantage of this combined architecture in comparison with the simple DWDM point-to-point links is that it can optimize the use of DWDM wavelength channels by

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reconfiguring the mesh formed by the wavelength channels. The topology of reconfiguration can be dynamic in which network topology is reset periodically according to the traffic with the time period in the order of seconds or milliseconds. The reconfiguration can also be static in which the mesh is set for a longer period of time. The enabling technology for this architecture is the wavelength router in the optical domain. Different types of wavelength routers are available commercially which range from mechanically controlled to thermally controlled and to semiconductor wavelength switches. The advantage of this architecture is its ability to utilize the bandwidth capacity to the level that the electronics can handle, because of the reconfiguration of the mesh by the wavelength routers. The problem, however, is still the electro-optic bottleneck. Nevertheless, it has improved in comparison with the point-to-point DWDM links. Several technical issues are required to be dealt with for this architecture, which include the control system for the mesh reconfiguration, the traffic evaluation among the DWDM channels and among the fibers, as well as the technology for constructing the wavelength routers. These technical issues have been the topics of research for some time and all are still undergoing development at the current time.

Figure 5: Wavelength routing with electronic TDM DWDM networks

8.3 All-Optical DWDM Networks

The goal of all-optical DWDM networks is to eliminate the conversions between electricity and light. The all-optical network is also referred to as the transparent network. Two types of all-optical DWDM backbone networks have been proposed, which are:

• Wavelength switching DWDM networks without TDM, and

• DWDM with optical domain TDM.

These two types of DWDM backbone networks are discussed in the following two subsections.

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8.3.1 Wavelength switching without TDM

Circuit switching can be achieved by using wavelength switches (also called wavelength routers). Figure 6 shows the architecture of the network, in which wavelength switches are used to establish connections between the two communicators. This is quite the same as the old PSTN (Public Switched Telephone Network) system where the crossbar types of electrical switches are used to establish the circuit for the two users. The wavelength switches for this type of networks can switch among the wavelength channels of multiple fiber input and output ports. A wavelength router may have the additional capacity of changing the wavelength of the signal between routers resulting in high utilization of wavelength channels. In the case of a wavelength router without wavelength conversion, the two users involved in the communication are connected by one signal wavelength across all of switches in the light path. However, with the wavelength conversions, the two sides of the communication can be connected by different wavelengths in different fiber links between switches.

Figure 6: Wavelength switching without TDM

Figure 7 shows an example of a wavelength router structure with multiple fiber inputs and outputs.

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Figure 7: A wavelength router

The advantage of this type of DWDM network is its simplicity in the switching mechanism. The major problem for this network is the under utilization of the bandwidth capacity of the wavelength channels. This is because once the light path is established between the two sides of the communication parties; it is up to the user to use the available bandwidth of the wavelength channel which can be tens of GHz or even hundreds of GHz. It is obviously inefficient for voice communications.

Therefore, this type of all-optical DWDM backbone networks will unlikely be used to replace the SONET/SDH. However, it can be useful for Enterprise intranets where different types of communications can share the same network with low costs of building it. Also, because of the transparency of the network, it provides many advantages, such low error rate and low maintenance costs.

8.3.2 Wavelength switching with optical TDM

As it has been mentioned above, the wavelength routing all-optical network has the problem of low efficiency in utilizing the bandwidth of wavelength channels with each having the capacity of hundreds of Gigabits per second. Although the combined wavelength routing and electronic time domain multiplexing can increase the bandwidth utilization to some degree, it introduces the O/E conversions that may restrict the speed and cause packet delays. Therefore, it is natural to implement optical TDM in future optical networks, which eliminates the O/E conversions resulting in a transparent high-speed all-optical network. Replacing the electrical TDM nodes in the DWDM with electrical TDM architecture (Figure 5) by optical TDM nodes, we obtain a DWDM wavelength routing with optical TDM architecture, as shown in Figure 8. As we have seen that the electrical TDM/switching nodes in Figure 5 can be of any kind, such as SONET/SDH switches, Internet routers, and ATM switches. This indicates that the all-optical TDM nodes in the all-optical architecture can be optical SONET/SDH switches, or all-optical ATM switches, or all-optical Internet routers. Different types of all-optical

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TDM/switch nodes can also be in one network, provided the protocol conversions are implemented.

Figure 8: All-optical TDM/switch with wavelength router

In fact, the optical TDM/switch node and the wavelength router in one routing site (Figure 8) can be combined into one all-optical switching node that not only forwards packets through time domain multiplexing but also selects the light path intelligently according to the availability and traffic loads of the links. This architecture is shown in Figure 9.

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Figure 9: All-optical TDM/switch with self wavelength selection

The only question left so far is how can we build such all-optical TDM/ switch nodes. This is the problem that has not been solved. Although some research and proposals have been published, such as using a hybrid mechanism where electrical signals are used to perform the control and light signals are used to carry the data, the system is far from mature. Also it still involves electricity which still has many problems to be dealt with. Looking into the possibilities of building the three major types of the current electrical switches, namely the SONET/SDH switch, the ATM switch, and the Internet router, the SONET/SDH switch is the simplest one among the three since it is for circuit switching. The Internet router is the most complex one among the three, since it requires a digital optical processor and all-optical memories. In any case, digital optical logic devices are required to build these switches. Therefore, it seems clear that we need to have optical digital processing power before we can implement any of these TDM switches, unless other mechanism emerges to revolutionize the existing concept of packet switching networks.

9.0 DWDM Access Optical Networks An overriding belief existed even in the early 1970's that optical fiber would one day make its way into the subscriber loop and be used to connect individual homes. Research on the fiber based residential access network architecture and protocols have since then become one of the major areas in the telecommunication arena. The ATM (Asynchronous Transfer Mode) based B-ISDN (Broadband Integrated Services Digital Network) architecture had been once believed to be the leading candidate for realizing the fiber-to-the-home access network. However, with the technological development of the DWDM, broadband residential access fiber network has taken another turn, which leads to a DWDM-based fiber optical network to deliver both narrowband and broadband services. This section provides an overview of the network architectures that have been developed for the residential access networks based on the DWDM technology. DWDM-based access optical networks can be classified into two categories, passive DWDM access networks and active DWDM networks. The term of active DWDM network here refers as to the DWDM network in which the TDM (time domain multiplexing) is applied in the wavelength channels. These two types of access network architecture are discussed in the following subsections.

9.1 DWDM Passive Optical Networks (PON)

DWDM passive optical networks (PON) use the wavelength channels to connect the users with the central office. Each service uses one wavelength channel. The early PON was developed for narrowband services, such as the PON architecture developed by British Telecom. However, recent PONs are for both broadband and narrowband services. A passive subscriber loop is attractive because it uses no active devices outside the central office (CO), except at the customer premises. Several architectures of passive optical networks have been proposed for WDM or DWDM, which include the single-star, the tree, the doublestar, and the star-bus. Figure 10 shows the single-star architecture in which each household has a dedicated fiber to the central office (CO).

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Figure 10: Single-star PON architecture

The WDM channels in the fiber are used to carry all required services, such as voice and video. This architecture is designed for easy installation and upgrading; however, the cost of dedicated optical fiber between the customer and the CO in this network is still a major concern. Thus, this architecture may not be suitable for widespread deployment in the near term. Figure 11 shows the tree PON architecture, in which the DWDM channels are split in the way of tree branches with each user having one or more wavelength channels. This architecture reduces the fiber use in comparison with the single-star. It is a better architecture, especially for DWDM-based system in which a large number of wavelength channels are available. This architecture can satisfy the customer needs for both narrowband and broadband services. One drawback of this network architecture is its rigidity, in terms of network upgrading. The star-bus architecture can be considered as a variation of the tree architecture, which improves the flexibility of the tree architecture.

Figure 11: Tree PON architecture

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Figure 12 depicts the double-star PON architecture. This architecture provides more flexibility in comparison with the star-bus architecture. It can be considered as the front-runner among the possible architectures of PON for residential access applications.

Figure12: The double-star PON architecture

9.2 DWDM Active Access Optical Networks

In the passive DWDM access networks, each wavelength channel is used to provide one service at a given time regardless of the channel capacity and bandwidth requirement of the service. With the increasing bandwidth capacity of DWDM technology, the bandwidth of one signal channel becomes high enough to carry several or many services even in the access environment. This leads to the thinking of applying TDM in each individual DWDM wavelength channel, resulting in the active DWDM access optical network in which TDM is used within each channel to provide integrated services. The Asynchronous Transfer Mode (ATM) has been proposed as the TDM protocol in the active DWDM access networks. With the ATM coming into the picture, the original B-ISDN (Broadband Integrated Services Digital Network) protocols are again surfacing in the access network arena. But this time, only one wavelength channel replaces the whole optical fiber in the system. The network topologies for the passive DWDM access network discussed in the previous subsection can also be used for the active DWDM access network. Although an active DWDM access network provides high utilization of the wavelength channels and in return reduces the fiber costs, it adds additional costs because of the ATM devices in the system from CO to user premises. It also increases the complexity of system management and maintenance, which leads to high operating costs. Another twist in this hard-to-decide matter is the birth of the very high channel-count DWDM, in which thousands of wavelength channels are created and transmitted with one fiber. This may make the active DWDM access network architecture lose its potential advantages, and make the passive high channel-count DWDM PON become the leader in the race of access network architectures.

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10. Conclusion

The DWDM point-to-point technology has already played an important role in the backbone networks and it will continue to be installed for existing and new fiber links. However, for the all-optical DWDM network to become viable, we may have to wait till the optical processing power becomes available. This may create a time gap between the DWDM point-to-point applications and the all-optical DWDM transparent networks. In the access network case, it is still not clear whether the passive or the active is the leader. Although the cost barrier has been weakened through replacing the fiber by DWDM channels in the active access network architecture, the TDM devices in the system may still be too high at the present time. On the other hand, the fiber cost (along with the costs of the passive devices) in the passive architecture is probably still not cheap enough to make it ahead of the active architecture. However, the very high channel-count DWDM may change the landscape of the access network world, and it may become even cheaper than the combined costs of twisted pair copper and coaxial cables.

Assignment

Qu.1 What is DWDM?

Qu.2 Write down the benefits of DWDM?

Qu.3 Write down the network Elements of DWDM?

Qu.4 Explain the Point to Point DWDM links?

Qu.5 Explain the working of All Optical DWDM Networks?

Qu.6 What is DWDM passive optical Network?

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CHAPTER -10

MOBILE COMMUNICATION 1. Principle of Mobile Communication Introduction In Telecom network conventionally each user is connected to the Telephone exchange individually. This dedicated pair starts from MDF, where it is connected to the appropriate Equipment point and ends at the customer premises Telephone. (With flexibility at cabinet/pillar/ distribution points DPs)

The connectivity from exchange to customer premises is called “Access Network or Local Loop”, and mostly comprises of underground cable from exchange up to DP’s and insulated copper wires (Drop Wires)later on This type of Access Network does not require separate Authentication of customer before extending services. Whenever the cable capacity has reached the maximum additional cable is laid to augment the capacity. Even though there are advantages in introducing wireless connectivity in Subscriber’s loop, we have to tackle certain issues viz,

1. Duplexing methodology. 2. Multiple Access methods. 3. Cellular principle or reuse concept. 4. Techniques to cope with “mobile” environment.

Duplexing Methodology: Duplexing is the technique by which the send and receive paths are separated over the medium, since transmission entities (modulator, amplifiers, demodulators) are involved. There are two types of duplexing:

• Frequency Division Duplexing FDD • Time Division Duplexing TDD

Frequency Division Duplexing FDD Different Frequencies are used for send and receive paths and hence there will be a forward band and reverse band. Duplexer is needed if simultaneous transmission (send)

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and reception (receive) methodology is adopted .Frequency separation between forward band and reverse band is constant Time Division Duplexing (TDD) TDD uses different time slots for transmission and reception paths. Single radio frequency can be used in both the directions instead of two as in FDD. No duplexer is required. Only a fast switching synthesizer, RF filter path and fast antenna switch are needed. It increases the battery life of mobile phones. GSM and CDMA systems use Frequency Division Duplexing and corDECT uses Time Division Duplexing. Multiple Access methodology: The technique of dynamically sharing the finite limited radio spectrum by multiple users is called Multiple Access Technique. By adopting multiple access techniques all users can not get the services simultaneously and some amount of blocking is introduced by the system. This is known as GOS (Grade of Service). Generally there are three different types of multiple access technologies. They are

• Frequency Division Multiple Access (FDMA) • Time Division Multiple Access (TDMA) • Code Division multiple Access (CDMA)

Frequency Division Multiple Access (FDMA): FDMA is a familiar method of allocating bandwidth, where a base station is allowed to transmit on one or more number of preassigned carrier frequencies and a mobile unit transmits on corresponding reverse channels. No other base station within range of the mobile will be transmitting on the same forward channel, and no other mobile within range of the base station should be transmitting on the same reverse channel. Both the base and the mobile usually transmit continuously during a conversation, and fully occupy their assigned forward and reverse channels. No other conversation can take place on these channels until the first conversation is completed.

FDMA Analogy It may be easier to visualize FDMA by imagining a cocktail party where two people wish to converse with each other. Then everyone in the room must be silent except for the speaker. The speaker may talk as long as they wish, and when they finish someone else may start speaking, but again only one at a time. New speakers must wait (or find another party) for the current speaker to finish before starting. Everyone in the room can hear and understand the speaker, unless they are too far away or the speaker's voice is too soft. If the intended listener is close enough, the speaker may decide to whisper. Conversely, if

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the listener is too far away, the speaker may have to shout. Since no one else should be talking, this presents no problem. If someone talks out of turn, the listener will probably be confused and not be able to understand either speaker. Features Of Frequency Division Multiple Access (FDMA)

• No Precise coordination in time domain is necessary in FDMA System. • It is well suited for narrow band analog systems. • Guard spacing between channels causes wastage of frequency resource.

Otherwise good modulation techniques are to be employed to avoid such guard spacing.

• The transmission is simultaneous and continuous and hence duplexers are needed. Continuous transmission leads to shortening of battery life.

Time Division Multiple Access (TDMA)

TDMA is a more efficient, but more complicated way of using FDMA channels. In a TDMA system each channel is split up into time segments, and a transmitter is given exclusive use of one or more channels only during a particular time period. A conversation, then, takes place during the time slots to which each transmitter (base and mobile) is assigned. TDMA requires a master time reference to synchronize all transmitters and receivers.

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TDMA Analogy In TDMA, everyone in the room agrees to watch a clock on the wall, and speak

only during a particular time. Each person wishing to talk is given a set period of time, and each person listening must know what that time period will be. For example, everyone may agree on time slots with duration of ten seconds. Speaker number one may talk for ten seconds starting from the top of the minute. The listener who wishes to hear this speaker must also be made aware of the schedule, and be ready to listen at the top of the minute. Speaker number two may speak only from ten seconds after the minute until twenty seconds after. As with FDMA, only one person at a time may speak, but each speaker's time is now limited and many persons may take their turn. If someone in the room cannot see the clock, they will not be able to speak and will have great difficulty understanding the speakers. Features of TDMA

• There can be only one carrier in the medium at any time, if a simple TDMA scheme is followed.

• Transmission is in bursts and hence is well suited for digital communication. • Since the transmission is in bursts, Battery life is extended. • Transmission rate is very high compared to analog FDMA systems. • Precise synchronization is necessary.

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• Guard time between slots is also necessary Time and Frequency Division Multiple Access

Both methods of FDMA and TDMA are combined to achieve higher capacity in practical systems. A channel gets a certain frequency band for a certain amount of time. The Best example for such system is GSM.

Cellular Concepts: Even though multiple access techniques allowed multiple users to share the medium simultaneously, due to constraints in providing resources, an amount of blocking will exist. The amount of blocking is called “Grade Of Services”(GOS). Based on GOS and resource availability (no. of carriers/no. of timeslots/both) the traffic handling capacity of the system is calculated. If this total traffic is divided by traffic per subscriber, we get number of subscribers supported by the system. For these purposes Erlang B table (Blocking calls cleared) is useful particularly in FDMA-TDMA. Why Cellular? Assuming 30mE traffic per subscriber, sub density of 30 per sq.km, and GOS 1%

Radius Area (KM2) Subs Total Traffic RF Channels

1 3.14 100 3.0E 8

3 28.03 900 27E 38

10 3.14 10000 300E 360

Providing 360 RF channels for 10,000 subscribers in an area of 314 sq.km on a single base station is not feasible and if still either the area of coverage or sub density increases, the system cannot function at all for want of bandwidth. Hence the solution is dividing the service area into small units, called cell, with base stations radiating with low power, and limited number of carriers required as per traffic. The same carriers are again reused at a different cell, which is geographically separated. (Frequency Reuse) In case of CDMA it appears that there is no limitation for simultaneous calls but practically there is a limit to CDMA capacity. And it is essentially the amount of interference a CDMA receiver can tolerate. As more and more units transmit, the amount of noise a receiver sees goes up, since all signals not using the receiver's specific PN code appear as noise. At some point there is so much noise that the receiver can no longer hear

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the transmitter. Boosting the transmitter power won't help overall, since it increases the noise for all the other receivers, who would in turn tell their transmitters to boost power, and the situation remains. In a nutshell, if a unit near a base station is transmitting with too much power, signals from units far from the base station will be lost in the noise. Hence cellular concept is applicable even in the case of CDMA where code used for identification of cell/sector is reused. Advantages of Cellular Principle Base stations can transmit at low power compared to a single high power transmitter. It requires less RF bandwidth to cover a given area. Frequency reuse gives good spectrum efficiency. (FDMA-TDMA) Disadvantage of cellular principle Reuse introduces interference. Established calls should be handed over to next cell to avoid dropping of calls when the customer is in mobility. Mobile Environment BTS is connected to Mobile or Fixed Wireless Terminal by air Interface. This connectivity differs from our earlier UHF/Microwave which is purely Line of Sight (LOS) system. In mobile communication due to the mobility of the user from the BTS LOS to BTS may exist or may not exist. The radio wave is subject to attenuation, reflection, Doppler shift and interference from other transmitter. These effects cause loss of signal strength and distortion which will impact the quality of voice or data. To cope with the harsh conditions, any mobile technology makes use of an efficient and protective signal processing. Proper cellular design must ensure that sufficient radio coverage is provided in the area. Types of signal strength variations The signal strength variation for mobile is due to different types of signal strength fading. There are two types of signal strength variations

• Macroscopic Variations Due to the terrain contour between BTS and MS. The fading effect is caused by shadowing and diffraction (bending) of radio waves.

• Microscopic variations. Due to multipath, Short-term or Rayleigh fading. As the MS moves, radio waves from many different paths will be received.

Macroscopic Variations Macroscopic Variations can be modeled as the addition of two components that make up the path loss between mobile and base station. The first component is the deterministic component (L) that adds loss to the signal strength as the distance(R) increases between base and mobile. This component can be written as

L=1/Rn Where n = typically 4. The other macroscopic component is a Log normal random variable which takes into account the effects of shadow fading caused by variations in terrain and other obstructions in the radio path. Local mean value of path loss=deterministic component +log normal random variable Microscopic Variations Microscopic Variations or Rayleigh Fading occur as the mobile moves over short distances compared to the distance between mobile and base. These short term variations are caused by signal scattering in the vicinity of the mobile unit e.g. by hill, building or

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traffic. The result is that not one but many different paths are followed between transmitter and receiver (Multipath Propagation). The reflected wave will be altered in both phase and amplitude. The signal may effectively disappear if the reflected wave is 180 degree out of phase with the direct path signal. The partial out of phase relationships among multiple received signal produce smaller reduction in received signal strength. Special features of mobile technologies All mobile techniques incorporate some special features to overcome the hazards created by mobile environment. The following are a few to name:

1. Coding. 2. Diversity techniques. 3. Adaptive equalization( in case of GSM) 4. Rake Receiver (in case of CDMA)

1.Coding Coding includes:

• Speech coding, • Convolutional coding or Forward Error Correction coding • Interleaving

Speech Coding Human speech is band limited between 300Hz to 3400Hz and undergoes Frequency Modulation in analog systems. In digital fixed PSTN systems band limited speech is sampled at the rate of 8 KHz and each sampled is encoded into 8 bits leading to 64Kbps (PCM A-Law of encoding).Digital cellular radio cannot handle the high bit rate used for PSTN systems. Smart techniques for signal analysis and processing have been developed for reduction of the bit rate Different mobile communication systems use different bit rates for voice encoding. The following table gives a glimpse. No. Technology Bit rate per voice

chl Voice coding

technique 1 GSM 13Kbps RPE-LTP 2 CDMA IS95A 9.6Kbps/14.4 Kbps QCELP/EVRC 3 Cor-DECT 32Kbps ADPCM

RPE-LTP: Regular Pulse Excited Long Term Prediction QCELP: Qualcomm Code Excited Linear Prediction EVRC: Enhanced Variable Rate Coding ADPCM: Adaptive Differential Pulse Code Modulation Forward Error Correction Coding: Sometimes this process is called Convolutional Coding or Channel Coding. The purpose of this process is to build redundancy in the signal so that even if error occurs, the receiver will be able to recover the lost information. Several methods are available for this purpose and each mobile system uses its own choice. Interleaving: Interleaving is a simple, but powerful, method of reducing the effects of burst errors and recovering bits when burst errors occur. The symbols (output of Forward Error

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Correction Coder) from each group are interleaved in a pattern that the receiver knows. The interleaver is located at the BTS and in the phone. An illustrative example is shown below.

2. Diversity Techniques: To cope up with the mobile environment Diversity techniques are employed .This can be Space Diversity, Polarisation Diversity, Frequency Diversity and Time Diversity. Space and Polarisation Diversity:

It is implemented in the BTS by deploying two antennas, one for Transmitting and receiving, the other for only receiving. Both antennas should be kept with minimal separation (10 times wave length). Space Diversity can be combined with Polarisation Diversity by making the Diversity antenna in an opposite polarization. In modern times the same antenna with dual polarized elements are available so that with single antenna, at least polarisation diversity can be achieved. Space Diversity can be implemented only when sufficient space is available in the tower for mounting the antennas. Frequency Diversity: Signal degradation can be averted by changing the present frequency to another in case of narrow band systems. This avoids frequency selective fading. In a narrow band system like GSM this is achieved by slowly hopping the frequency of transmission of BTS in a predetermined manner. In case of a wide band system like CDMA signal occupies a large bandwidth and frequency diversity is inherently achieved. Time Diversity:

In all the mobile communication systems by employing interleaving time diversity is automatically achieved.

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3. Adaptive Equalisation: The transmitter trains the receiver to adapt to the air environment by sending a known sequence along with the data. Corrections as applied to the known sequence are applied to the data to retrieve it error free. This is used in GSM. 4. Rake Receiver: The rake receiver is multiple receivers in one. There is a rake receiver at both the mobile and BTS. It turns what is a problem in other technologies into an advantage for CDMA. Signals sent over the air can take multi-paths resulting in degradation of signal. The rake receiver identifies the three strongest multi-path signals and combines them to produce one very strong signal. The rake receiver therefore uses multipath to reduce the power the transmitter must send.

Conclusion: Wireless means convenience. However to achieve this certain precautionary measures are taken to overcome the bandwidth scarcity, multipath problems, etc., There are multiple access techniques to share the bandwidth amongst several users. Cell Concepts

Frequency scarcity problem

Consider a city where one lakh mobile subscribers need to be served. Considering that a single RF loop requires a frequency spectrum of 50KHz, total 1,00,000*50KHz = 5GHz of RF bandwidth would be needed to serve these subscribers if an individual dedicated RF loop is needed for every subscriber. Obviously such a huge bandwidth is not available that too for a limited number of subscribers in a city. This clearly shows that an acute problem of frequency scarcity is going to dictate the design and implementation of a mobile radio system.

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In order to handle this problem one of the obvious measures is not to allot a dedicated RF channel to an individual subscriber. Rather a group of a few common RF channels would be available to a relatively large number of subscribers so that RF channel is allocated only when a mobile user wants to make the call.

Even if we consider the number of RF channel required to serve, for example, a city of 10Km radius with a subscriber density of 30 subs/ Sq.Km, the RF resource requirement for 1% grade of service with 30mE traffic per mobile subscriber depicted in the following table.

City Radius (Km) Area (Sq.Km) No. of Subs RF Chls reqd.

1 3.14 100 8

3 28.3 900 38

10 314 10,000 @360

25 960 60,000 @2000

Clearly this is not practical to allot 360 radio channels (360*50KHz = 18 MHz) for only 10,000 subscribers of this single city alone.

� hence frequency reuse is a must to cover the total service area with a limited available rf resources

� hence the need for a cellular principle

What is a Cell?

• A base station (transmitter) having a number of RF channels is called a cell.

• Each cell covers a limited number of mobile subscribers within the cell boundaries (Coverage area)

• Typical Cell Radius Approx = 30Km (Start up), 1Km (Mature)

Concept of a cluster of cells

In a cellular network a concept of frequency reuse is applied wherein RF carrier frequencies used in a given geographical location are simultaneously reused at

1 2

Fig 1. OMNI DIRECTIONAL CELLS

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geographically separated locations (cells. A typical seven cell pattern is depicted in figure 2 wherein the total available frequency resource is divided in seven parts consisting of a number of radio channels. Such a group of cells where the available frequency spectrum is consumed within those cells only is called a cluster of cells.

These seven sets of different frequency (a cluster size N of seven) can be reused after

certain distance as shown in figure 3.

Co-Channel Interference

Co-Channel Interference in a multi-cell environment

1

1

11

11

1

1 2 3 4 5 6 7

Given Freq. Resource

Fig 2. A CLUSTER OF CELLS

1 2 3 4 5 6 7

Given Freq. Resource

1 2 3 4 5 6 7

Given Freq. Resource

Frequency Reuse Pattern “N” = 7 Fig 3

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The cells highlighted as in fig 4 in the adjacent cluster use same set of RF

channels and hence are termed as co-channel cells. The distance between the cells using the same frequency should be sufficient to keep the co-channel interference to an acceptable level.

Co-channel Interference function

Higher Q Reduced Co-Channel Interference Lower Q Increased Co-Channel Interference Co-Channel Interference Q = D / R = √ 3N N= Cluster Size R= Size (Radius of Cell) D= Distance between two Co-Chl Cells

R D

Fig 4. A Multi Cell Environment

R

Co-channel interference is a function of “Q” Q = D / R D

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N Q = D/R

Higher Q Less Interference

1 1.73 Higher N

3 3.00 More Cluster Size

4 3.46 Less RF freq/cell

7 4.58 Less Traffic Handling Capacity of the system

9 5.20 Lower Q Higher Interference

12 6.00 Increased System Handling Capacity

Sectorisation of cells

One way of reducing the level of interference is use directional antenna at base

stations, with each antenna illuminating a sector of the cell, and with a separate channel

set allocated to each sector. There are two commonly used methods of Sectorisation

either using three 120˚ sector or 60˚ sector, both of which reduces the number of prime

interference sources.

The three sector case is generally used with a seven cell pattern, giving an over all

requirement for 21 channel sets as shown in fig 7 & 8.

f3

f2

f1

1

3

2

f3

f2

f1

1

3

2

Omni Directional Three Sectored Cell Three Sectored Cell

Fig 6.

A three sectored cell configuration Single location

Fig 7.

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FREQUENCY REUSE PATTERN

Frequency assignment in cells

Concept of frequency assignment

The cell layout (4-Cell, 3-Sectored) is shown in fig 9.

Frequency Assignment

• 1A – 1, 13, 25, 37, 49, 61, 73, 85, 97, 109, 121

• 2A – 2, 14, 26, 38, 50, 62, 74, 86, 98, 110, 122

• 1B – 5, 17, 29, 41, 53, 65, 77, 89, 101, 113

• 1C – 9, 21, 33, 45, 57, 69, 81, 93

Fig 8 8.

1A 1B 1C 2A

2B 2C

4A 4B 4C

2A 2B 2C

4A 4B 4C

4A 4B 4C

1A 1B 1C

2A 2B 2C

4A 4B 4C

1A 1B 1C

3A 3B 3C

1A 1B 1C

3A 3B

3C

3A 3B

3C

3A 3B

3C

BTS BSC

Fig 9 9.

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1.5.1 Frequency planning aspects

Frequency planning aspects A1 B1 C1 D1 A2 B2 C2 D2 A3 B3 C3 D3

1 2 3 4 5 6 7 8 9 10 11 12

13 14 15 16 17 18 19 20 21 22 23 24

25 26 27 28 29 30 31 32 33 34 35 36

Trunking Efficiency Cells in City / Sub-urban areas. Number of access channel and system capacity

• Start Up Cells with Larger Diameter

• Mature cells with Smaller Diameters

• Cells in city center Smaller Diameters

• Cells in Sub-urban areas Larger

Diameters

Fig 11

D1 D2 D3 C1

C2 C3

B1 B2 B3

A1 A2 A3

Fig 10

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Extract from Traffic Table – Erlang B Model

GoS Number of Access

Channels 0.5%

1.0% 2%

5 1.13 1.36 1.66 10 3.96 4.46 5.08 20 11.1 12.0 13.2 33 21.5 22.9 24.6 50 36.0 37.9 40.3 56 41.2 43.3 45.9 99 80.0 84.1 87.0 100 80.9 85.0 88.0

TRUNKING EFFICIENCY

Sr.No Number of Access Channels Erlang Capacity Number of Users Served

1 5 1.66 16 G 2 10 5.08 50 O 3 20 13.2 132 S 4 50 40.3 403 2% 5 100 88.0 880 TRUNKING EFFICIENCY

• The Number Of Users Served In A Cell Are Directly Proportional To The Access

Channels Allocated In A Cell

• More The Number Of Access Channels In A Cell Further Increase In The

System Handling Capacity

TRUNKING EFFICIENCY

20

10

10

50 Subs 50 Subs 132 Subs

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• It is better to have a single cell than to split into two with half the number of access channels

Capacity Considerations Radio frequency carrier and its allotment to traffic and control.

• TDMA time slots (BP0 to BP7) in one radio frequency carrier are allotted for

traffic and control channels. Normally BP0 only for control channel.

• Traffic channel (TCH) is used to carry speech and data. These are dedicated

channels.

• Common channels are be used by both idle mode and dedicated mode mobiles.

Capacity Considerations.

Capacity consideration with one RF carrier per cell in a site is shown below.

Capacity consideration with more than one RF carrier per cell in a site is shown below

96 Access Channels 9 – Signaling

32 Access Channels 3 – Signaling

8 Access Channels 1 – Signaling 7 - Voice

1 2 8

1 1 1

8 Access Channels 1 – Signaling 7 - Voice

With 2% GoS

2.94 E

2.94 E/25mE=120 Subs

120 Subs/Sector × 3 = 360 Subscribers

1 1 1

4 4 4

12 12 12

120×3 = 360 Subs 840×3 = 2520 Subs 3200×3 = 9600 Subs

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Fixed telephones, using wired access network, are meant to be used at a particular

location only. We can have telephones at our office/business and our residence. The fixed

telephones are linked to a place but the modern day life style demands that we should

have telephone facility while on move also. Mobile communication facilitates telephonic

conversation in a fast moving vehicle. This means that phones moves along with a person

thereby moving telephone is linked to a person and not to a place. In this words our reach

becomes broader and world shrinks into a Global village.

Mobile communication objectives The important objectives of the mobile communication are

� Any time Anywhere communication

� Mobility & Roaming

� High capacity & subs. density

� Efficient use of radio spectrum

� Seamless Network Architecture

� Low cost

� Innovative Services

� Standard Interfaces

Introduction to GSM

The Global System for Mobile Communications (GSM) is a digital cellular

commununication system. It was developed in order to create a common European

mobile telephone standard but it has been rapidly accepted world wide.

Mobile communication Generations

1 G - analog (cellular revolution)

-only mobile voice services

2 G - digital (breaking digital barrier)

- mostly for voice services & data delivery possible

3 G - Voice & data ( breaking data barrier)

- Mainly for data services where voice services will also be possible.

Frequency bands used in GSM

GSM 900 Mhz

DCS 1800 MHz

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GSM 900 Mhz

Mobile to Cell(UP-LINK) - 890 to 915 MHz

Cell to Mobile (DOWN -LIN - 935 to 960 MHz

GSM ARCHITCTURE A GSM network basically consists of four main subsystems: Mobile station &

The SIM, The Base Station Subsystem, The Network and Switching Subsystem and the

Operation and Support Subsystem.

GSM Network Structure Every telephone network needs a well-designed structure in order to route

incoming calls to the correct exchange and finally to the called subscriber. In a mobile

network, this structure is of great importance because of the mobility of all its subscribers

.In the GSM system, the network is divided into the following partitioned areas.

� GSM service area

� PLMN service area

� MSC service area

� Location area

� Cells

The GSM service area is the total area served by the combination of all member

countries where a mobile can be serviced.

The next level is the PLMN service area. A Public Land Mobile Network

(PLMN) is the area served by one network operator. There can be several within a

country, based on its size. All incoming calls for a GSM/PLMN network will be routed to

a gateway MSC (Mobile switching center). A gateway MSC works as an incoming transit

exchange for the GSM/PLMN.

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In a GSM/PLMN network, all mobile-terminated calls will be routed to a gateway

MSC. Call connections between PLMNs, or to fixed networks, must be routed through

certain designated MSCs called a gateway MSC. The gateway MSC contains the

interworking functions to make these connections. They also route incoming calls to the

proper MSC within the network.

The next level of division is the MSC/VLR service area. In one PLMN there can

be several MSC/VLR service areas. MSC/VLR is a role controller of calls within its

jurisdiction. In order to route a call to a mobile subscriber, the path through links to the

MSC in the MSC area where the subscriber is currently located. The mobile location can

be uniquely identified since the MS is registered in a VLR, which is generally associated

with an MSC.

The next division level is that of the LA’s (Location Area) within a MSC/VLR

combination. There are several LA’s within one MSc/VLR combination.

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A LA is a part of the MSC/VLR service area in which a MS (Mobile Station)

may move freely without updating location information to the MSC/VLR exchange that

control the LA. Within a LA a paging message is broadcast in order to find the called

mobile subscriber. The LA can be identified by the system using the location Area

Identity (LAI). The LA is used by the GSM system to search for a subscriber in a active

state.Lastly, a LA is divided into many cells. A cell is an identity served by one BTS.

The MS distinguishes between cells using the Base Station Identification code

(BSIC) that the cell site broadcast over the air.

The architecture of GSM network is shown in the figure below and the functions

of components of the network are briefly explained.

Mobile Station (MS)- A Mobile Station consists of two main elements: The mobile terminal(handset)

and the SIM (Subscriber Identity Module).

The SIM is a smart card that identifies the terminal. By inserting the SIM card

into the terminal, the user can have access to all the subscribed services. Without the SIM

card ,the terminal is not operational.

The SIM card is protected by a four digit Personal Identification Number(PIN).In

order to identify a subscriber to the system, the SIM card contains some parameters of the

user such as its International Mobile Subscriber Identity (IMSI).

The Base Station Subsystem (BSS) The BSS connects the Mobile Station and the Network and Switching Subsystem

(NSS).It is in charge of transmission and reception. The BSS can be divided into two

parts:The Base Transceiver Station (BTS) and The Base Station Controller (BSC).

The BTS corresponds to the Transceivers and the Antennas used in each cell of

the network. A BTS is usually placed in the centre of a cell. Its transmitting power

defines the size of a cell. Each BTS has normally between one and sixteen transceivers

depending on the density of users in the cell.

The BSC controls a group of BTS and manages their radio resources. A BSC is

principally in charge of handovers, frequency hopping, exchange functions and control of

the radio frequency power levels of the BTSs.

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The Network and Switching Subsystem

Its main role is to manage the communication between the mobile users and other

users such as mobile users, ISDN users, landline users etc. It also includes data bases

needed in order to store information about the subscribers and to manage their mobility.

The different components of the NSS are briefed below.

The Mobile services Switching Center (MSC)

It is the central component of the NSS. The MSC performs the switching

functions of the network. It also provides connection to other networks.

The Gateway Mobile services Switching Center (GMSC)

A Gateway is a node interconnecting different networks. The GMSC is the

interface between the mobile cellular network and the PSTN. It is in charge of routing

calls from the fixed network toward a GSM user.

Home Location Register (HLR) The HLR is considered as a very important database that stores information of a

given set of subscribers. It also stores the current location of these subscribers and the

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services to which they have access. The location of the subscriber corresponds to the SS7

address of the Visitor Location Register (VLR) associated to the terminal.

Visitor Location Register (VLR)

The VLR is collocated with an MSC. The VLR contains information obtained

from subscriber’s HLR in order to provide the subscribed services to visiting users. When

a subscriber enters the covering area of a new MSC, the VLR associated to this MSC will

request information about the new subscriber to its corresponding HLR. The VLR will

then have enough information in order to assure the subscribed services without needing

to ask the HLR each time a communication is established.

The Authentication Center (AuC) The AuC register is used for security purposes. It provides the parameters needed

for authentication and encryption functions. These parameters help to verify the user’s

identity.

The Equipment Identity Register (EIR) The EIR is a register containing information about the mobile equipments

Particularly it contains a list of all valid mobile terminals. A terminal is identified by its

International Mobile Equipment Identity (IMEI). The EIR allows then to forbid calls

from stolen or unauthorized terminals.

There are three classes of ME that are stored in the database, and each group has

different characteristics.

� White List White List:- contains those IMEIs that are known to have been assigned

to valid MS’s. This is the category of genuine equipment.

� Black List :- contains IMEIs of mobiles that have been reported stolen.

� Grey List:- contains IMEIs of mobiles that have problems (for example, faulty

software, wrong make of the equipment). This list contains all MEs with faults not

important enough for barring.

The Operation and Support Subsystem (OSS) The OSS is connected to the different components of the NSS and to the BSC, in

order to control and monitor the GSM system. It is also in charge of controlling the traffic

load of the BSS.

The OMC provides alarm-handling functions to report and log alarms generated

by the other network entities. The maintenance personnel at the OMC can define that

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criticality of the alarm. Maintenance covers both technical and administrative actions to

maintain and correct the system operation, or to restore normal operations after a

breakdown, in the shortest possible time.

The fault management functions of the OMC allow network devices to be

manually or automatically removed from or restored to service. The status of network

devices can be checked, and tests and diagnostics on various devices can be invoked. For

example, diagnostics may be initiated remotely by the OMC.

Call Processing, SMS and VMS In this we discuss the call processing aspect and look into specifics case of a mobile originated (MO) call and a mobile terminated (MT) call. We also look into short message (SMS) and voice mail service (VMS) as implemented IMPCS pilot project. RF channel overview: - RF channel play important role in call processing case. These are basically three types of RF control channel. Broadcast control channel : The broadcast channels are points to multi-point channel, which are defined only for down-link direction (BTS to mobile station). They are divided into: BCCH (Broad cast control channel:- BCCH acts as a beacon. It informs the mobile about system configuration parameters (e.g. LAI, CELLIDENTY, NEIGHBOURING cell identify). Using this information MS choose the best cell to attach to. BCCH is always transmitted on full power and it is never frequency hopped. FCCHC frequency correction channel. MS must tune to FCCH to listen to BCCH. FCCH transmits a constant frequency shift of the radio carrier that is used by the MS for frequency correction.

SCH (synchronization channel). . SCH is used to synchronize the MS in time .SCH carries TDMA frame number and BSIC (Base Station Identity Code)

Common control channels : Common control channels are specified as point to multi-point, which operate only in one direction either in up-link or down-link direction.

PCH (Paging Channel): - PCH is used in down-link direction for sending paging message to MS whenever there is incoming call.

RACH (Random Access Channel ) :-RACH is used by the MS to request allocation of a specific dedicated control channel (SDCCH) either in response to a paging message or for call origination /registration from the MS. this is an up-link channel and operate in point to point mode.

AGCH (Access Grant Channel ):- AGCH is a logical control channel which is used to allocated a specific dedicated control channel (SDCCH) to MS when MS request for a channel over RACH. AGCH is used in downlink direction.

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3.Dedicated Control Channel : Dedicated control channel are full duplex, point to point channel. They are used for signalling between the BTS and certain MS. They are divided into: -

SACCH (Slow Associated Control Channel): the SACCH is a duplex channel, which is always allocated to TCH or SDCCH. The SACCH is used for

• Radio link supervision measurements.

• Power control.

• Timing advance information.

In 26 frame traffic multi-frame 13th frame (frame no .12) is used for SACCH.SACCH is used only for non-urgent procedures.

FACCH (Fast Associated Control Channel). FACCH is requested in case the

requirement of signaling is urgent and signaling requirement can not be met by SACCH.

This is the case when hand-over is required during conversation phase. During the call

FACCH data is transmitted over allocated TCH instead of traffic data. This is marked by

a flag known as stealing flag.

SDCCH (Stand Alone Dedicated Control Channel)- The SDCCH is a duplex, point to

point channel which is used for signaling in higher layer. It carries all the signaling

between BTS & MS when no TCH is allocated to MS. The SDCCH is used for service

request, location updates, subscriber authentication, ciphering. equipment validation and

assignment of a TCH.

Mobile originated (MO) call: - There are four distinct phase of a mobile originated call-

-Setup phase.

-Ringing phase.

-Conversation phase.

-Release phase.

Out of these phases the setup phase is the most important phase and includes

authentication of the subscriber, Ciphering of data over radio interface, validation of

mobile equipment, validation of subscriber data at VLR for requests service and

assignment of a voice channel on A-interface by MSC. Whenever MS wants to initiate on

outgoing call or want to send an SMS it requests for a channel to BSS over RACH. On

receiving request from MS, BSS assigns a stand-alone dedicated control channel

(SDCCH) to MS over access grant channel (AGCH). Once a SDCCH has been allocated

to MS all the call set up information flow takes place over SDCCH.

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A connection management (CM) entity initiates a CM Service Request message to

the network. Network tries to establish an MM connections between the MS and the

network and upon successful establishment of MM connection a CM Service Accept

message is received by MS from the network. MS now sends a Call Set up Request to

the network which contains the dialed digits (DD) of the called party. As the call setup

message is received at the MSC/VLR certain check are performed at MSC/VLR like-

whether the requested service is provisioned for the subscriber or not, whether the dialed

digits are sufficient or not, any operator determined barring (ODB) does not allow call to

proceed further etc. As these checks are performed at MSC/VLR a Call Proceeding

Message is sent from the network towards the MS. After all the checks are successfully

passed MSC sends Assignment command to the BSS which contains a free voice channel

on A-interface On getting this message BSS allocates a free TCH to the MS and informs

the MS to attach to it. MS on attaching to this TCH informs the BSS about it. On

receiving a response from the BSS, MSC switches the speech path toward the calling MS.

Thus at the end of Assignment the speech path is through from MS to MSC. It is

important to note that at this stage mobile has not connected user connection as yet. MS

at this stage does not listen anything.

After assignment MSC sends a network set-up message to the PSTN requesting

that a call be set up. Included in the message are the MS dialed digits (DD) and details

specifying which trunk should be used for the call. The PSTN may involve several

switching exchanges before finally reaching the final local exchange responsible for

applying the ringing tone to the destination phone. The local exchange will generate the

ringing tone over the trunk, or series of trunk (if several intermediate switching exchange

are involved), to the MSC. At this point in time MS will hear ringing tone. The PSTN

notifies the MSC with a network-alerting message when this event occurs. MSC informs

the MS that the destination number is being alerted. It is important to note that this is

primarily a status message to the MS. The MS hears the ringing tone from the destination

local exchange through the established voice path.

When the destination party goes off hook, PSTN informs the MSC of this event. At

this point, MS is connected to the destination party and billing is started. MSC informs

the MS that connection has been established and MS acknowledges the receipts of the

connect message.

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Under normal condition, the termination of a call is MS initiated or network

initiated. In this scenario, we have assumed that MS initiates the release of the call by

pressing “end” button and MS send a disconnect message to the MSC. The PSTN party is

notified of the termination of the call by a release message from the MSC. The end- to-

end connection is terminated. When MSC is left with no side task (e.g. charging

indication etc.) to complete a release message is sent to the MS. MS acknowledges with a

release complete message. All the resources between MSC and the MS are released

completely.

Mobile Terminated (MT) call- The different phases of a mobile terminated call are

- Routing analysis

- Paging.

- Call setup.

- Call release.

The phases of mobile terminated (MT) call are similar to a mobile originated (MO)

call except routing analysis and paging phase. Call to a mobile subscriber in a PLMN first

comes to gateway MSC (GMSC). GMSC is the MSC, which is the capable of querying

HLR for subscriber routing information. GMSC need not to be part of home PLMN,

though it is normal practice to have GMSC as part of PLMN in commercially deployed

networks.

GMSC opens a MAP (Mobile Application Part) dialogue towards HLR and Send /

Routing / Info-Request (SRI request) specific service message is sent to HLR. SRI

request contains MSISDN of the subscriber. HLR based on location information of this

subscriber in its database, opens a MAP dialogue towards VLR and sends Provide /

Roaming / Number-request (PRN request)to the VLR. VLR responds to PRN request

with PRN response message, which carries an MSRN (mobile subscriber roaming

number), which can be used for routing toward visiting MSC in the network. HLR returns

MSRN to GMSC (MSC that queried HLR) in SRI response message. On getting MSRN

the GMSC routes the call towards VMSC The purpose of this entire exercise is to locate

where the terminating mobile subscriber is.

The MSRN received at GMSC is in international format (Country Code + Area

Code + subscriber number). Normally, based on the routing info at GMSC, the call may

be routed out of the GMSC towards VMSC of the terminating subscriber, in which case

appropriate signaling protocol (MF or ISUP) depending on the nature of connecting of

GMSC with subsequent exchange along the route will apply. If at VMSC the terminating

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mobile subscriber is found to be free (idle), paging is initiated for terminating mobile

subscriber. MSC uses the LAI provided by the VLR to determine which BSS’s should

page the MS. MSC transmit a message to each of these BSS requesting that a page be

performed. Included in the message is the TMSI of the MS. Each of the BSS’s broadcasts

the TMSI of the mobile in a page message on paging channel (PCH).

When MS detects its TMSI broadcast on the paging channel , it responds with a

channel request message over Random Access Channel (RACH). Once BSS receives a

channel request message , it allocates a stand –alone Dedicated Control

Channel(SDCCH) and forwards this channel assignment information to the MS over

Access Grant Channel (AGCH). It is over this SDCCH that the MS communicates with

the BSS and MSC until a traffic channel assigned to the MS. MS transmits paging

response message to the BSS over the SDCCH. Included in this message is MS TMSI

and LAI. BSS forwards this paging response message to the MSC. Now Authentication

and Ciphering phases are performed to check the authenticity of MS and encrypt data

over radio interface.

On the network side after paging is initiated, while waiting for paging response, a

defensive timer called, ”Early ACM” timer is run at MSC to avoid network timeouts. On

successfully getting paging response, a setup message is constructed to be sent towards

terminating MS. In case paging fails due to authentication failure or when the subscriber

is out of radio-coverage, the call is cleared.

In case CLIP is not subscribed by the terminating mobile subscriber, calling number

is not included in set-up message. In case CLIP is subscribed and PI value in calling

number parameter indicates “presentation allowed” the number is included in the set-up

message. In case CLIP is subscribed but PI received in calling number parameter

indicates “presentation restricted” then number is included only if CLIRO is also

subscribed to.

MS on receiving the set-up message performs compatibility Checking before

responding to the set-up message – it is possible that MS might be incompatible for

certain types of call set-ups. Assuming that MS passes compatibility checking, it

acknowledges the call setup with set-up confirm message. After getting set-up confirm

message from the MS, MSC performs assignment phase (similar to one discussed in MO

call) and a voice path is established from MSC to the MS. MS begins altering the user

after it receives the traffic channel assignment. MS send alerting message to the MSC

.MSC upon receiving the alerting indication from the MS, begins generating an audible

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ringing tone to the calling party and sends a network alerting via GMSC to the PSTN.

Prior to this the calling party heard silence.

At this point in the call, MS is alerting the called party by generating on audible

tone. One of the three events can occur-calling party hangs-up, mobile subscriber answers

the phone, or the MSC times out waiting for the mobile subscriber to the answer the call.

Since radio traffic channel is a valuable resource, GSM does not allow a MS to ring

forever.

In the present scenario we have assumed that the mobile subscriber answers the

phone. The MS in response to this action stops alerting and sends a connect message to

the MSC. MSC removes the audible tone to the PSTN and connects the PSTN trunk to

BSS trunk (terrestrial channel) and sends a connect message via GMSC to the PSTN. The

caller and the called party now have a complete talk path. This event typically marks the

beginning of the call for billing purposes. MSC sends a connect acknowledge message to

the MS.

The release triggered by the land user is done in similar way as the release

triggered by mobile user. MSC receives a release message from the network to terminate

end-to-end connection. PSTN stops billing the calling landline subscriber. MSC sends a

disconnect message towards the MS and MS responds by a Release message. MSC

release the connection to the PSTN and acknowledges by sending a Release Complete

message to PSTN. Now the voice trunk between MSC and BSS is cleared, traffic channel

(TCH) is released and the resources are completely released.

The mobile-to-mobile call scenario is a combination of phases encountered in

mobile originated (MO) and mobile terminated (MT) call.

Short Message Service (SMS)

SMS is a simple bearer service and acts as a bi-directional alphanumeric paging

service, which allows value added service provision as well as management services

provision such as advice of charge. A short message can carry at most 160 characters (it

can be less depending upon the type of characters and their coding scheme). The SMS

could be either in broadcast mode (via CBCH channel) or in a point-to-point mode (via

either SDCCH channel if mobile is in idle state, or SACCH if the mobile is in dedicated

mode).

SMS allows to provide many values added service to individual/ corporate

clients. Individuals may be interested in messaging (transmitting messages in compact

way) or leisure services (weather forecast, road traffic, restaurant booking, movies, TV

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programs etc.). Business users may be interested in corporate information (company

performance, stock value), e-commerce etc.

SMS involves specific entities in the GSM network: first is the SMS Service

Centre (SMS-SC simply SMSC) which can be connected to several networks and many

MSC’s (SMS- GMSC’s or SMS-IWMSC’s) within the same PLMN and which is

addressed by a mobile using a E.164 number of the numbering plan of the PLMN. SMSC

is capable of following functionality’s:

-Transmission of short message towards a mobile, retaining the responsibility of the

message until reception of acknowledgement or expiration of the validity period.

-Reception of the short messages from MS and transmission of acknowledgement to the

PLMN.

-Transferring messages received from Internet to mobile.

The second entity involved by the SMS is the SME (short message entity),

which is responsible for producing or receiving a short message. The SME can be

connected to the SMSC via a data network such as X.25 or IP.

A short message is characterized by its parameters the most significant are the validity

period, the service center time stamp which indicates the SM arrival time at the SC, etc.

In IMPCS (pilot project), the SMS architecture has been implemented by C-

DoT. The hardware architecture of SMSC is similar to HLR and is located on same

physical platform. It services as an inter-working and relaying function of the message

transfer between two MS. The service provided are-

(i) Mobile Originated short message- Enables MS to send an SMS ( up-to 140 bytes) to another MS via SMSC.

(ii)Mobile terminated short message- Enables delivery of an SMS to a particular MS.

(iii)Operator initiated SMS- This facility enables fixed network subscriber to send an SMS to a mobile subscriber through an operator at SMSC.

(iv)SMS Newsletter Service- A group of mobile subscriber can subscribe to SMSC for receiving periodic news regarding sports, weather, traffic etc. The subscription is done through on operator at SMSC. The operator feeds the news segments, which are transferred, to the subscriber periodically.

Voice Mail System (VMS)

VMS offers function of call answering device in the system. It provides personal voice mailbox to the subscribers. VMS redirects/forwards voice calls of a temporarily in accessible subscriber (busy or no reply) to a personal mailbox of the

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subscriber connected to the MSC. Whenever a call is redirected to VMS, it first greets the caller with a personalised greeting message and prompts the caller to leave the message in the mailbox. Later on the called party (mobile subscriber) can access the VMS from PLMN/PSTN phone by means of access code.

VMS interfaces with MSC on E1 lines using R2 MF/CCS#7 signaling protocol. In IMPCS network the VMS consists of Pentium PC equipped with Dialogic card loaded with Windows NT 4.0. Dialogic card provides telephony network interface, voice recording, compression and play. The disk capacity requirement of the PC is totally application dependent. For 10,000 subscribers, if each subscriber stores 10 minutes of voice data then disk storage for subscriber voice information is around 20 GB.

General Packet Radio Service (GPRS) General Packet Radio Service (GPRS) is a mobile data service available to users of GSM

mobile phon. It is often described as "2.5G", that is, a technology between the second

(2G) and third (3G) generations of mobile telephony. It provides moderate speed data

transfer, by using unused TDMA channels in the GSM network. Originally there was

some thought to extend GPRS to cover other standards, but instead those networks are

being converted to use the GSM standard, so that is the only kind of network where

GPRS is in use. GPRS is integrated into GSM standards releases starting with Release 97

and onwards. First it was standardised by ETSI but now that effort has been handed onto

the 3GPP.

GPRS is different from the older Circuit Switched Data (or CSD) connection included in

GSM standards releases before Release 97 (from 1997, the year the standard was feature

frozen). In CSD, a data connection establishes a circuit, and reserves the full bandwidth

of that circuit during the lifetime of the connection. GPRS is packet-switched which

means that multiple users share the same transmission channel, only transmitting when

they have data to send. This means that the total available bandwidth can be immediately

dedicated to those users who are actually sending at any given moment, providing higher

utilization where users only send or receive data intermittently. Web browsing, receiving

e-mails as they arrive and instant messaging are examples of uses that require intermittent

data transfers, which benefit from sharing the available bandwidth.

Usually, GPRS data are billed per kilobytes of information transceived while circuit-

switched data connections are billed per second. The latter is to reflect the fact that even

during times when no data are being transferred, the bandwidth is unavailable to other

potential users.

GPRS originally supported (in theory) IP, PPP and X.25 connections. The latter has been

typically used for applications like wireless payment terminals although it has been

removed as a requirement from the standard. X.25 can still be supported over PPP, or

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even over IP, but doing this requires either a router to do encapsulation or intelligence

built into the end terminal.

Mobile Phone and data standards

• PTT • MTS • IMTS

0G

• AMTS • PALM 0.5G • ARP • NMT • AMPS

1G

• CDPD • GSM • D-AMPS • cdmaOne • PDC

2G

• CSD • GPRS • HSCSD

2.5G

• wiDEN • CDMA2000 1xRTT 2.75G • EDGE • W-CDMA 3G • CDMA2000 1xEV

3.5G • HSDPA 3.75 • HSUPA 4G

How does GPRS work ?

GPRS is packet based, wherein GPRS data is handled as a series of "packets" that can be

routed over several paths through the network, rather than as a continuous bit-stream over

a dedicated dial-up connection. With GPRS, the information is split into separate but

related "packets" before being transmitted and reassembled at the receiving end. The

Internet itself is an example of a packet data network, the most famous of many such

network types.

In second-generation mobile networks, calls are handled using traditional circuit-

switching technology. A dedicated "circuit", or "timeslot", is allocated between two

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points for the duration of a call. No other phone can use this circuit during the call,

regardless of whether any data is being transmitted or not.

The GPRS standard is delivered in a very elegant manner - with network operators

needing only to add a couple of new infrastructure nodes and making a software upgrade

to some existing GSM network elements.

Key User Features of GPRS The General Packet Radio Service (GPRS) is a new non-voice value added service that

allows information to be sent and received across a mobile telephone network. It

supplements today's Circuit Switched Data and Short Message Service. GPRS is NOT

related to GPS (the Global Positioning System), a similar acronym that is often used in

mobile contexts. GPRS has several unique features which can be summarized as:

SPEED

Theoretical maximum speeds of up to 171.2 kilobits per second (kbps) are achievable

with GPRS using all eight timeslots at the same time. This is about three times as fast as

the data transmission speeds possible over today's fixed telecommunications networks

and ten times as fast as current Circuit Switched Data services on GSM networks. By

allowing information to be transmitted more quickly, immediately and efficiently across

the mobile network, GPRS may well be a relatively less costly mobile data service

compared to SMS and Circuit Switched Data.

IMMEDIACY

GPRS facilitates instant connections whereby information can be sent or received

immediately as the need arises, subject to radio coverage. No dial-up modem connection

is necessary. This is why GPRS users are sometimes referred to be as being "always

connected". Immediacy is one of the advantages of GPRS (and SMS) when compared to

Circuit Switched Data. High immediacy is a very important feature for time critical

applications such as remote credit card authorization where it would be unacceptable to

keep the customer waiting for even thirty extra seconds.

NEW APPLICATIONS, BETTER APPLICATIONS

GPRS facilitates several new applications that have not previously been available over

GSM networks due to the limitations in speed of Circuit Switched Data (9.6 kbps) and

message length of the Short Message Service (160 characters). GPRS will fully enable

the Internet applications you are used to on your desktop from web browsing to chat over

the mobile network. Other new applications for GPRS, profiled later, include file transfer

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and home automation- the ability to remotely access and control in-house appliances and

machines.

SERVICE ACCESS

To use GPRS, users specifically need:

(1) a mobile phone or terminal that supports GPRS (existing GSM phones do NOT support GPRS) .

(2) a subscription to a mobile telephone network that supports use of GPRS must be enabled for that user. Automatic access to the GPRS may be allowed by some mobile network operators, others will require a specific opt-in knowledge of how to send and/ or receive GPRS information using their specific model of mobile phone, including software and hardware configuration (this creates a customer service requirement)

(3) a destination to send or receive information through GPRS. Whereas with SMS this was often another mobile phone, in the case of GPRS, it is likely to be an Internet address, since GPRS is designed to make the Internet fully available to mobile users for the first time. From day one, GPRS users can access any web page or other Internet applications- providing an immediate critical mass of uses.

Having looked at the key user features of GPRS, lets look at the key features from a network operator perspective. GPRS networks are able to handle higher bit rates than GSM networks, but the data rates still fall short of what is required to make existing GSM networks deliver services at a speed comparable to that promised by third-generation networks. The delay in the deployment of third-generation systems led to the emergence of a technology known as EDGE. This was capable of delivering services similar to those of third-generation networks, yet with implementation on the existing second-generation networks (e.g. GSM). EDGE stands for 'enhanced data rates for GSM evolution'. The enhancement from GSM was to GPRS (i.e. voice and packet), while further enhancement of GPRS led to EDGE networks, as shown in Figure 1. The fundamental concept remains the same, i.e. voice, CS data and PS data being carried, and the network architecture is the same as in a GPRS network. Enhancement of HSCSD is known as ECSD (enhanced circuit-switched data), while enhancement of GPRS is known as EGPRS. EGPRS implementation has a major effect on protocol structure (e.g. on layer 1 or layer 2). The modulation and coding schemes are quite different in EGPRS (this is explained later in the chapter). In ECSD, though user data rates do not go beyond 64 kbps, fewer time slots are required to achieve this compared HSCSD. The architecture of ECSD is based on HSCSD

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transmission and signalling, thus having minimal impact on existing specifications. In this chapter we will focus on EDGE network planning aspects from the EGPRS perspective.

Figure 1 EDGE evolution

THE EDGE SYSTEM As shown in Figure 2, the EDGE system is quite similar to the GPRS system (compare

with Figure 1), but with the capability for higher data rates. The most important change

the new modulation scheme. In GSM and GPRS, the GMSK modulation scheme was

Used

In GMSK modulation, only one bit per symbol is used. In an EDGE network, octagonal

phase-shift keying (8-PSK) modulation is used which enables a threefold higher gross

data rate of 59.2 kbps per radio time slot by transmitting three bits per symbol. GMSK is

a constant-amplitude modulation while 8-PSK has variations in the amplitude. This

amplitude variation changes the radio performance characteristics, so hardware changes

in the base stations are mandatory.

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CHAPTER - 11

CDMA TECHNOLOGY

Access Network: Access network, the network between local exchange and subscriber, in the Telecom Network accounts for a major portion of resources both in terms of capital and manpower. So far, the subscriber loop has remained in the domain of the copper cable providing cost effective solution in past. Quick deployment of subscriber loop, coverage of inaccessible and remote locations coupled with modern technology have led to the emergence of new Access Technologies. The various technological options available are as follows :

1. Multi Access Radio Relay 2. Wireless In Local Loop 3. Fibre In the Local Loop

Wireless in Local Loop (WILL) Fixed Wireless telephony in the subscriber access network also known as Wireless in Local Loop (WLL) is one of the hottest emerging market segments in global telecommunications today. WLL is generally used as “the last mile solution” to deliver basic phone service expeditiously where none has existed before. Flexibility and expediency are becoming the key driving factors behind the deployment of WILL. WLL shall facilitate cordless telephony for residential as well as commercial complexes where people are highly mobile. It is also used in remote areas where it is uneconomical to lay cables and for rapid development of telephone services. The technology employed shall depend upon various radio access techniques, like FDMA, TDMA and CDMA. Different technologies have been developed by the different countries like CT2 from France, PHS from Japan, DECT from Europe and DAMPS & CDMA from USA. Let us discuss CDMA technology in WLL application as it has a potential ability to tolerate a fair amount of interference as compared to other conventional radios. This leads to a considerable advantage from a system point of view. SPREAD SPECTRUM PRINCIPLE Originally Spread spectrum radio technology was developed for military use to counter the interference by hostile jamming. The broad spectrum of the transmitted signal gives rise to “ Spread Spectrum”. A Spread Spectrum signal is generated by modulating the radio frequency (RF) signal with a code consisting of different pseudo random binary sequences, which is inherently resistant to noisy signal environment. A number of Spread spectrum RF signals thus generated share the same frequency spectrum and thus the entire bandwidth available in the band is used by each of the users using same frequency at the same time.

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Fig-1 CDMA ACCESS – A CONCEPT

On the receive side only the signal energy with the selected binary sequence code is accepted and original information content (data) is recovered. The other users signals, whose codes do not match contribute only to the noise and are not “despread” back in bandwidth (Ref Fig-1) This transmission and reception of signals differentiated by “codes” using the same frequency simultaneously by a number of users is known as Code Division Multiple Access (CDMA) Technique as opposed to conventional method of Frequency Division Multiple Access and Time Division Multiple Access. In the above figure, it has been tried to explain that how the base band signal of 9.6 Kbps is spread using a Pseudo-random Noise (PN) source to occupy entire bandwidth of 1.25 Mhz. At the receiving end this signal will have interference from signals of other users of the same cell, users of different cells and interference from other noise sources. All these signals get combined with the desired signal but using a correct PN code the original data can be reproduced back. CDMA channel in the trans and receive direction is a FDD (Frequency Division Duplexing) channel. The salient features of a typical CDMA system are as follows: Frequency of operation: 824-849Mhz and 869-894 Mhz Duplexing Mehtod: Frequency Division Duplexing (FDD) Access Channel per carrier: Maximum 61 Channels RF Spacing: 1.25 Mhz Coverage: 5 Km with hand held telephones and approx.

20 Km with fixed units. The different types of codes used for identification of traffic channels and users identification etc as follows: Different Codes used in CDMA Walsh Code: In CDMA the traffic channels are separated by unique “Walsh” code. All such codes are orthogonal to each other. The individual subscriber can start communication using one of these codes. These codes are traffic channel codes and are used for orthogonal spreading

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of the information in the entire bandwidth. Orthogonality provides nearly perfect isolation between the multiple signals transmitted by the base station. The basic concept behind creation of the code is as follows: (a) Repeat the function right (b) Repeat the function below Invert function (diagonally) 0 ----- 0 0 -------- 0 0 0 0 0 1 0 1 0 1 0 0 1 1 0 1 1 0 Long Code: The long pseudo random noise (PN) sequence is based on 242 characteristic polynomial. With this long code the data in the forward direction (Base to Mobile) is scrambled. The PN codes are generated using linear shift registers. The long code is unique for the subscribers and is known as users address mask. Short Code: The short pseudo random noise (PN) sequence is based on 215 characteristic polynomial. This short code differentiates the cells & the sectors in a cell. It also consists of codes for I & Q channel feeding the modulator. Advantages: CDMA wireless access provides the following unique advantages. Larger Capacity:

Let us discuss this issue with the help of Shannon’s Theorem. It states that the channel capacity is related to product of available band width and S/N ratio.

C = W log 2 (1+S/N)

Where C = channel capacity

W = Band width available

S/N = Signal to noise ratio.

It is clear that even if we improve S/N to a great extent the advantage that we are expected to get in terms of channel capacity will not be proportionally increased. But instead if we increase the bandwidth (W), we can achieve more channel capacity even at a lower S/N. That forms the basis of CDMA approach, wherein increased channel capacity is obtained by increasing both W & S/N. The S/N can be increased by devising proper power control methods.

Vocoder and variable data rates:

As the telephone quality speech is band limited to 4 Khz when it is digitized with PCM its bit rate rises to 64Kb/s vocoding compress it to a lower bit rate to reduce bandwidth. The transmitting vocoder takes voice samples and generates an encoded speech/packet for transmission to the receiving vocoder. The receiving vocoder decodes the received speech packet into voice samples. One of the important feature of the variable rate vocoder is the use of adaptive threshold to determine the required data rate. Vocoders are variable rate vocoders. By operating the vocoder at half rate on some of the frames the capacity of the system can be enhanced without noticeable degradation in the quality of the speech. This phenomenon helps to absorb the occasional heavy requirement of traffic

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apart from suppression of background noise. Thus the capacity advantage makes spread spectrum an ideal choice for use in areas where the frequency spectrum is congested.

Less (Optimum) Power per cell:

Power Control Methods: As we have already seen that in CDMA the entire bandwidth of 1.25Mhz is used by all the subscribers served in that area. Hence they all will be transmitting on the same frequency using the entire bandwidth but separated by different codes. At the receiving end the noise contributed by all the subscribers is added up. To minimize the level of interfering signals in CDMA, very powerful power control methods have been devised and are listed below:

1. Reserve link open loop power control

2. Reserve link closed loop power control

3. Forward link power control

The objective of open loop power control in the reverse link (Mobile to Base) is that the mobile station should adjust its transmit power according to the changes in its received power from the base. Open loop power control attempts to ensure that the received signal strength at the base station from different mobile stations, irrespective of their distances from the base site, should be same.

In Closed loop power control in reverse link, the base station provides rapid corrections to the mobile stations’ open loop estimates to maintain optimum transmit power by the mobile stations. The base station measures the received signal strength from the mobile connected to it and compares it with a threshold value and a decision is taken by the base every 1.25 ms to either increase or decrease the power of the mobile.

In forward link power control (Base to Mobile) the cell (base) adjusts its power in the forward link for each subscriber, in response to measurements provided by the mobile station so as to provide more power to the mobile who is relatively far away from the base or is in a location experiencing more difficult environment.

These power control methods attempt to have an environment which permits high quality communication (good S/N) and at the same time the interference to other mobile stations sharing the same CDMA channel is minimum. Thus more numbers of mobile station are able to use the system without degradation in the performance. Apart from the capacity advantage thus gained power control extends the life of the battery used in portables and minimizes the concern of ill effects of RF radiation on the human body.

Seamless Hand-off:

CDMA provides soft hand-off feature for the mobile crossing from one cell to another cell by combining the signals from both the cells in the transition areas. This improves the performance of the network at the boundaries of the cells, virtually eliminating the dropped calls.

No Frequency Planning:

A CDMA system requires no frequency planning as the adjacent cells use the same common frequency. A typical cellular system (with a repetition rate of 7) and a CDMA

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system is shown in the following figures which clearly indicates that in a CDMA network no frequency planning is required.

CDMA Frequency

Frequency Reuse of 7 in GSM

High Tolerance to Interference:

The primary advantage of spread spectrum is its ability to tolerate a fair amount of interfering signals as compared to other conventional systems. This factor provides a considerable advantage from a system point of view.

Multiple Diversity:

Diversity techniques are often employed to counter the effect of fading. The greater the number of diversity techniques employed, the better the performance of the system in a difficult propagation environment.

CDMA has a vastly improved performance as it employs all the three diversity techniques in the form of the following:

A .Frequency Diversity: A wide band RF signal of 1.25 Mhz being used.

B. Space Diversity: Employed by way of multipath rake receiver.

C. Time Diversity:Employed by way of symbol interleaving error detection and correction coding.

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Capacity Considerations

Let us discuss a typical CDMA wireless in local loop system consisting of a single base station located at the telephone exchange itself, serving a single “cell”. In order to increase the number of subscribers served the cell is further divided into “sectors”. These sectors are served by directional antennas.

The capacity of a cellular system is claimed to be 20-40 active lines per sector per 1.25 MHz for a single CDMA Radio Channel. In WLL environment assuming an average busy hour traffic of 0.1 Erlang, 400 subscribers can be served per sector over a single 1.25 MHz channel.

Assuming typically six sectors in a cell the total capacity of a CDMA network consisting of 1.25 MHz duplex channels is 2400 (400x6) subscribers.

Capacity can further be increased if we use another frequency on the same base station covering the same geographical area (overlapping cell). Thus in 10 Mhz in the bandwidth we can utilize 5 MHz of bandwidth in the forward link and 5 Mhz in the reverse link. Hence if we have 4 RF carriers in 5 Mhz bandwidth, the network can support 12000 (5x400x6) subscribers per cell.

Conclusion

Hence we see that use of common frequency, multipath rake receiver, power control & variable bit rate vocoding and soft hand-off features of CDMA give us the benefits of no frequency planning, larger capacity, flexibility alongwith high performance quality.

Introduction to CDMA 2000-1X

Network entity description

Base station subsystem (BSS) Base station subsystem is the general term for the wireless devices and wireless channel control devices that serve one or several cells. Generally, a BSS contains one more base station controllers (BSC) and base transmitter stations (BTS).

Mobile switch center (MSC)

MSC is a functional entity that performs control and switching to the mobile stations within the area that it serves, and an automatic connecting device for the subscriber traffic between the CDMA network and other public networks or other MSCs. MSC is the kernel of the CDMA cellular mobile communication system, and it is different from a wired switch in that an MSC must consider the allocation of the wireless resources and the mobility of subscribers, and at least it must implement the follows processing activities:

1. Location Registration processing;

2. Handoff.

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Architecture of CDMA MSC Based WLL system

Gateway MSC (GMSC)

When a non-CDMA subscriber calls a CDMA subscriber, the call will first be routed to an MSC, which will inquires the corresponding HLR and further route the call to the called party’s MSC. This kind of MSC is called Gateway MSC (GMSC). It is up to the network operator to select which MSCs as GMSCs.

Visitor location register (VLR)

VLR is responsible for the storage and updating of the subscriber data of mobile stations that roamed to the service area of this VLR. The VLR is generally configured together with the MSC. When the mobile station enters a new location area, the MSC will notice the VLR, which will initiate registration processing to the HLR to update the subscriber location information. The VLR also stores necessary information for the establishment of calls in the database for the MSC to search. One VLR can cover one or more MSC areas.

Home location register (HLR)

The HLR provides subscriber information storage and management functions for the mobile network, including mobile subscriber subscription and cancellation and service authorization and cancellation. At the same time, it helps in the implementation of subscriber’s call and service operations. A CDMA can contain one or more HLRs based on the number of subscribers, equipment capacity and network organization mode, with multi-HLR mode realized in the form of virtual HLRs. The subscriber information stored in the HLR includes the following two types in information:

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1. Subscription information

2. Subscriber-related information stored in the HLR

Authentication center (AUC)

Authentication center is a function entity for the management of authentication information related to the mobile station. It implement mobile subscriber authentication, stores the mobile subscriber authentication parameters, and is able to generate and transmit the corresponding authentication parameters based on the request from MSC/VLR. The authentication parameters in the AUC can be stored in the encrypted form. The authentication center is generally configured together with the HLR. The authentication parameter stored in the AUC include:

1. Authentication key (A_KEY);

2. Share secret data (SSD);

3. Mobile identification number/international mobile subscriber identity (MIN/IMSI);

4. Authentication algorithm (AAV); 5. Accounting (COUNT).

Short message center (MC or SC)

As an independent entity in the CDMA cellular mobile communication system, the short message center works in coordination with other entities such as MSC, HLR to implement the reception, storing and transfer of the short messages from CDMA cellular mobile communication system subscribers, and store subscriber-related short message data.

Short message entity (SME)

SME is a function entity for synthesis and analysis of short messages. Operation and maintenance Center (OMC) The OMC provides the network operator with network operation and maintenance services, manages the subscriber information and implements network planning, to enhance the overall working efficiency and service quality of the system. There two type of operation and maintenance centers: OMC-S and OMC-R. An OMC-S is mainly used for the maintenance work at the mobile switching subsystem (MSS) side; an OMC-R is mainly used for the maintenance work at the base station subsystem (BSS) side. Third Generation Standards • CDMA2000/FDD-MC — CDMA2000 using Frequency Division Duplexing- Multicarrier (FDD-MC) mode. Here multicarrier implies N x 1.25 MHz channels overlaid on N existing IS-95 carriers or deployed on unoccupiedspectrum. CDMA2000 includes: 1x —using a spreading rate of 1.2288 Mcps 3x —using a spreading rate of 3 x 1.2288 Mcps or 3.6864 Mcps 1xEV-DO (1x Evolution - Data Optimized)—using a spreading rate of 1.2288 Mcps optimized for data • WCDMA/FDD-DS —Wideband CDMA (WCDMA) Frequency Division

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Duplexing-Direct Sequence spreading (FDD-DS) mode. This has a single 5 MHz channel. WCDMA uses a single carrier per channel and employs a spreading rate of 3.84 Mcps. • UTRA TDD/ TD-SCDMA — Universal Mobile Telephone Services Terrestrial Radio Access (UTRA) and TD-SCDMA. These are Time Division Duplexed (TDD) standards aimed primarily at asymmetric services used in unpaired (i.e., no separate uplink and downlink) bands. TD-SCDMA is based on a synchronous Time Division scheme for TDD and wireless local loop applications. The frame and slot structure are the same as W-CDMA. However, in TDD mode each slot can be individually allocated either the uplink or the downlink. Advantages of CDMA2000: CDMA2000 is backward compatible with IS-95. Thus a network that is converted to CDMA2000 from IS-95 will support users with IS-95 handsets. A motivating factor for migration to CDMA2000-enabled handsets is that it permits use of enhanced data service and increases the voice capacity of the network. The voice capacity of a CDMA2000 network increases as the percentage of subscribers with CDMA2000 handsets increases. IS-95 handsets do not contribute to this capacity improvement. It reuses and builds on the full complement of existing CDMA air interface and network standards. Both IS-95 and CDMA2000 equipped mobiles can operate on the same frequency assignment. Existing IS-95 networks can be converted to CDMA2000 without impact to existing IS-95

The following are the new additions in CDMA 2000 from IS95.

_ Spreading Rate 1 (1x) and Spreading Rate 3 (3x) _ Logical Channels _ Radio Configurations _ Many new Physical Channels _ Transmit Diversity Pilot Channels _ Enhanced Access Channel procedures _ Reverse Link Pilot Channel Spreading Rates CDMA2000 supports two different spreading rates: • Spreading Rate 1— also called “1x” Both Forward and Reverse Channels use a single direct-sequence spread carrier with a chip rate of 1.2288 Mcps. • Spreading Rate 3— also called “3x” or MC (Multi-Carrier) Forward Channels use three direct-sequence spread carriers each with a chip rate of 1.2288 Mcps.Reverse Channels use a single direct-sequence spread carrier with a chip rate of 3.6864Mcps. As such SR3 appears to be extinct.

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Physical and Logical Channels:

In IS 95A, in the forward link Pilot, Sync, Paging and Traffic Channels exist where as in reverse link Access and traffic channel are available. All overhead information is carried on the Paging Channel. During conversation or in dedicated mode the signaling info is exchanged by either fully or partially clearing the traffic. CDMA2000 technology defines new Physical and Logical Channels for the transport of user data and signaling information. A Physical Channel is a communication path between the mobile and the Base Station, described in terms of the digital coding and RF characteristics. A Logical Channel is a communication path within the protocol layers of either the Base Station or the mobile.

Radio Configurations: A Radio Configuration (RC) defines the following characteristics of a Forward or Reverse Traffic Channel, Viz Rate Set, Spreading Rate Channel Coding

(Turbo or convolutional), Channel Coding Rate, Modulation (QPSK or BPSK)and Transmit Diversity Allowed. IS-2000 defines Radio Configurations: - RC1 and RC2 correspond to IS-95 A/B Rate Set 1 and Rate Set 2 respectively - RC3 through RC9 on the Forward link - RC3 through RC6 on the Reverse link

Variable Length Walsh Codes: Walsh Code used in IS95 is 64 chips long. CDMA20001x can use Walsh Codes up to 128 Chips long. Higher data rate channels use shorter length Walsh codes to maintain a constant chip rate. Using one of the shorter Walsh codes precludes using all longer codes that contain the bit pattern of the shorter code. New Common Channels: CDMA2000 introduces several new Forward Link Common Channels: • Pilot Channels - If transmit diversity is supported; one or more Pilots may be used. The auxiliary Pilot Channels may be used for smart antenna applications. • Quick Paging Channel - This channel provides for improved slotted mode operation and improved battery life for the mobile. Walsh codes W80, W48 and W112 are reserved for Quick Paging Channels, if the Base Station supports Quick Paging Channels. • Common Control Channel - This channel carries mobile-directed messages for CDMA2000 compatible mobiles. • Broadcast Channel - This channel carries broadcast messages for CDMA2000 compatible mobiles, including overhead messages and broadcast Short Message Service (SMS) messages. • Common Power Control Channel - This channel is used with Enhanced Access Channel Procedures (Reservation Mode), to send power control bits to the mobile so that Access Channel messages may be sent under power control. • New Dedicated Channels: CDMA2000 introduces several new Forward Link Dedicated Channels: • Forward Fundamental Channel - This channel is used for the transmission of user and signaling information to a specific mobile during a call. Each Forward Traffic Channel may contain one Forward Fundamental Channel.

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• Forward Dedicated Control Channel - This channel is used for transmission of user and signaling information to a specific mobile during a call. Each Forward Traffic Channel may contain one Forward Dedicated Control Channel. • Forward Supplemental Channel (valid for Radio Configurations 3 thro 9) This channel is used for the transmission of user information to a specific mobile during a call. This is typically used for high-speed data applications. Each Forward Traffic Channel may contain up to two Supplemental Channels. • Power Control Subchannel - This subchannel is typically associated with the Fundamental Channel, but if the F-FCH is not used for a given call, then it is associated with the Dedicated Control Channel (F-DCCH). • All of the CDMA2000 dedicated channels can be established using the TIA/EIA Paging (F-PCH) and Access (R-ACH) Channels. • Reverse Link Channels: - Access Channel (R-ACH) _ Reverse Pilot Channel (R-PICH) _ Enhanced Access Channel (R-EACH) _ Reverse Common Control Channel (R-CCCH) _ Reversed Dedicated Control Channel (R-DCCH) _ Reverse Fundamental Channel (R-FCH) _ Reverse Supplemental Channel (R-SCH) _ Reverse Supplemental Code Channel (R-SCCH) The Access Channel and Reverse Supplemental Channel are retained for backward compatibility with TIA/EIA-95A/B. For Radio Configurations 1 and 2, the channel structure for the Reverse Fundamental Channel and Reverse Supplemental Channel is the same as the channel structure of Rate Set 1 and Rate Set 2 used in TIA/EIA-95A/B.

EV-DO

EV-DO is a mobile technology that facilitates higher throughput on mobile platform.

The third generation of cellular standards has seen a dominance of CDMA as the underlying access technology. UMTS (Universal Mobile Telecommunication Services) is 3G evolution for GSM world. The standardization work for UMTS is being carried-out by 3GPP. The standardization work for CDMA 2000 and its enhancements is being carried out under the supervision of 3GPP2.

1x Evolution-Data Optimized, abbreviated as EV-DO or 1xEV-DO, is an evolution of CDMA 2000 1x to support higher data rates. It is defined in TIA (Telecommunication Industry Association) standard IS 856. It is commonly referred to as DO. It is officially termed as "CDMA2000, High Rate Packet Data Air Interface". Working on same carrier bandwidth of 1.25 MHz as CDMA 2000 1x systems, 1xEV-DO provides significantly higher data rates to Access Terminals (mobile devices). Downlink data rates supported are up to 2.4576 Mb/s in Rev. 0 and up to 3.1 Mb/s in Rev. A.

Traditional wireless networks create a physical path between receiving and sending devices, much like traditional telephone networks. EVDO instead adopts the same approach used for the internet. IP, the Internet Protocol, breaks data into small pieces called packets. Each packet is sent independently of all the other packets. This saves

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bandwidth for use by other devices; when neither party on a phone call is speaking, the connection consumes no bandwidth because there are no packets to send. Radio resources are allocated only at the time of actual data transfer leading to better spectral efficiency.

EV-DO does not support voice services. In Forward link supports data rates up to 2.4576 Mbps. There is no power control in Forward Link. Peak data rate in Reverse Link is 153.6 kbps.

Generic Model of CDMA 2000 1x EVDO System:

A generic model of a CDMA 2000 1 x EV-DO System typically consists of:

a) Access Network (AN) consisting of Radio Node (RN) & Radio Network Controller

b) Packet Core Network (PCN)

a) Radio Node (RN): It is a multiple circuit transceiver which shall radiate to cover a cell or a sector. It consists of radio modules, base band signal processor, network interface, antenna, feeder etc. It can be co-located with RNC or remotely located. RN shall include the functions related to channel coding/decoding, interleaving, encryption, frame building, modulation/demodulation, RF transceiver, antenna diversity, low noise amplification etc. as per CDMA 2000 1 x EV-DO standards.

The AN obtains the timing reference and positioning reference from the GPS system and hence the GPS receiver shall form an integral part of the RN along with other fixtures such as GOS antenna, cable etc. AN split mounting arrangements with tower mountable RF components such as PAs, LNAs, Filters etc. are also acceptable.

b) Radio Network Controller (RNC): It is responsible for inter connection between the RN and the PCN and it provides control and management for one or more RNs. It assigns traffic channels to individual users, monitors system performance and provides interface between the RN and the PCN. RNC performs the radio processing functions such as management of the radio resources, radio channel management, local connection management etc.

It also processes information required for decision on handover of calls from one RN to another. RNC can be collocated with the PCN or remotely located. The Packet Control Function (PCF) shall form an integral part of RNC.

Packet Core Network (PCN): The packet data core network provides packet data services to Access Terminal (AT) and consists of PDSN, HA, AAA, AN-AAA and FA functionalities. The functional entities AAA and ANAAA may be a single physical entity or two separate physical entities.

Operations and Maintenance Centre (OMC): The Operations and Maintenance Centre (OMC) allow the centralized operation of the various units in the system and the functions needed to maintain the sub systems. The OMC provides the dynamic monitoring and controlling of the network management functions for operation and maintenance.

Call Processing in CDMA

Call processing refers to all the necessary functions that the system needs to carry out in order to set up, maintain, and tear down a call between a mobile and another party.

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Two types of connections are possible:

Mobile-to-Land call or Mobile-to-mobile call.

Call can be either mobile originated or mobile terminated. Since mobile station is the common element in both cases the IS-95 standard specifies the call states from the perspective of the CDMA.

States of mobile:

During normal operation, the mobile can occupy any one of the following states

· Mobile station initialization state;

· Mobile station idle state;

· System access state;

· Mobile station control on the traffic channel state.

• After power-up, the mobile first enters the mobile station initialization state, where the mobile selects and acquires a system. Upon exiting the initialization state, the mobile has fully acquired the system and its timing.

• Then the mobile enters the mobile station idle state where the mobile monitors messages on the paging channel .

• Any one of the following three events will cause the mobile to transition from the idle state to the system access state

1) The mobile receives a paging channel message requiring an acknowledgment or response,

2) The mobile originates a call, or

3) The mobile performs a registration.

• In the access state, the mobile sends messages to the base station on the access channel.

• When the mobile is directed to a traffic channel, it enters the mobile station control on the traffic channel state where the mobile communicates with the base station using the forward and reverse traffic channels. When the call is terminated, the mobile returns to the initialization state.

Initialization State

After power-up, the mobile enters the initialization state. This state contains four sub states, which the mobile sequentially goes through:

1. System determination sub state;

2. Pilot channel acquisition sub state;

3. Sync channel acquisition sub state;

4. Timing change sub state

Idle State:

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Paging Channel Monitoring

In the idle state, the mobile monitors the paging channel on the forward link. In order to receive messages and receive an incoming call, the mobile needs to monitor the paging channel for messages. The paging channel transmission is divided into slots that are 80 ms in length. There are two ways that the mobile can monitor the paging channel:

Nonslotted mode

Slotted mode.

In non-slotted mode, the mobile monitors the paging channel at all times. In slotted mode, the mobile monitors the paging channel only during assigned paging channel slots. Because the mobile doesn’t have to monitor all the slots all the time, the mobile operating in the slotted mode can conserve battery power.

Paging Channel Messages

There are a total of six overhead messages that are sent to the mobile on the paging channel:

• System parameters message;

• Neighbor list message;

• CDMA channel list message;

• Extended system parameters message;

• Global service redirection message;

• Access parameter message.

The first five messages are of configuration parameters and the last message is of access information.

Access State

In the access state, the mobile transmits messages to the base station using the access channel. In addition, the mobile also receives messages from the base station on the paging channel. There are six sub states that the mobile can occupy within the access state.

• Update overhead information sub state;

• Page response sub state;

• Mobile station origination attempt sub state;

• Registration access sub state;

• Mobile station order/message response sub state;

• Mobile station message transmission sub state.

Traffic channels state

The mobile may enter the traffic channel state from 2 sub states within the access state: the page response of state or the mobile state origination attempt sub state. In other words

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after the mobile has successfully originated, the mobile may enter the traffic channel state. In the traffic channel state, the mobile communicates with the base station using the forward and reverse traffic channels. This state consists of 5 sub states

• traffic channel initialization sub state

• waiting for order sub state

• waiting for mobile state answer sub state

• conversation sub state

• release sub state

Hand Offs in CDMA

As the phone moves through a network the system controller transfers the call from one cell to another, this process is called “handoff”. Handoffs maybe done with the assistance of the mobile or the system controller will control the process by itself. Handoffs are necessary to continue the call as the phone travels. Handoffs may also occur in idle state due to mobility.

Types of Handoffs in CDMA: There are primarily three types of Handoffs in CDMA. They are

• Soft

• Hard and

• Idle.

The type of handoff depends on the handoff situation.

To understand this we should know the cellular concept used in CDMA.

CDMA frequency- reuse planning (cellular concept):

Each BTS in a CDMA network can use all available frequencies. Adjacent cells can transmit at the same frequency because users are separated by code channels, not frequency channels. BTSs are separated by offsets in the short PN code This feature of CDMA, called "frequency reuse of one," eliminates the need for frequency planning

Soft Handoff:

A soft handoff establishes a connection with the new BTS prior to breaking the connection with the old one. This is possible because CDMA cells use the same frequency and because the mobile uses a rake receiver. The CDMA mobile assists the network in the handoff. The mobile detects a new pilot as it travels to the next coverage area. The new base station then establishes a connection with the mobile. This new communication link is established while the mobile maintains the link with the old BTS.

Soft handoffs are also called "make-before-break." Soft handoff can take place only when the serving cell and target cell are working in the same frequency.

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TUTORIAL on CDMA Principles

1) MATCH THE FOLLOWING:

COLUMN-A COLUMN-B

1)RF Spacing in CDMA: 1).......... A)Walsh code 0

2)RF Spacing in WCDMA 2).......... B) 869-889MHz

3) Pilot channel 3).......... C) Walsh code 32

4) Sync channel 4).......... D) Walsh codes 1-7

5) Paging channel 5).......... E) 1.25 Mhz

6) Traffic channel 6).......... F) 824-844MHz

7) Reverse link freq 7)………. G) 1.25 Mhz

8) Forward link freq 8)……… I) Walsh codes 8-31, 33-63.

9)Space Diversity 9)…….. J)Employed by way of symbol interleaving

10)Time Diversity 10)…… K)Employed by way of multipath rake receiver

2) SAY TRUE OR FALSE

1) CDMA is a spread spectrum technique.

2) In CDMA all subs can use the same frequency.

3) Walsh codes are orthogonal codes to each other.

4) For soft hand off freq of operation of the cells involved should be same.

5) CDMA system uses FDD technique.

6) CDMA supports two rate sets for voice encoding.

7) In CDMA power control is applicable only for forward link.

8) Hard hand off is a break before make type.

9) No of access channels will be normally equal to no. of paging channels.

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10) User mask is derived from electronic serial number of the hand set/ FWT.

11) Symbol is produced after coding process.

12) Chip is produced after spreading process.

13) CDMA system power control is applied only in the reverse link.

14) All CDMA system need GPS support for their functioning.

15) Long code is used for scrambling in the forward link.

16) Short code gives the BTS ID.

3) CHOOSE THE BEST CORRECT ANSWER:

1) Softer hand off is of ____________ type A) break before make B) make before break

2) Sequence used for spreading at trans end is 1001, then receiver will use ______ for despreading A) 0110 B) 1001 C) 1010 D) 1111

3) Long code is of length A) 2^15-1 B) 2^42-1 C) 2^8-1

4) Long and short codes are: A) orthogonal codes B) pseudo random codes C) none of the above

5) In forward link spreading is done by A) Walsh code B) Long code

6) If there are 4 shift registers in a PN code generator, the length of the code is A) 15 bits B) 16 bits C) 8 bits

7) Forward link is A) BTS to mobile B) Mobile to BTS

8) Reverse link is A) BTS to mobile B) Mobile to BTS

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9) Forward link works in A) 824- 849MHz B) 869-894MHz

10) There are _____ no. of Walsh codes. A) 20 B) 64 C) 50

11) Length of each Walsh code is A) 64 chips B) 23 chips C) 12 chips

12)Space Diversity is achieved in cdma by A) rake receiver B) Two antenna C) Booster antenna

Questions for CDMA2000-1X 1) MATCH THE FOLLOWING:

1) MSC-VLR 1).......... A) E interface

2) MSC-HLR 2).......... B) 3G Standard

3) HLR-VLR 3).......... C) 3.6864Mcps

4) MSC-MSC 4).......... D) N interface

5) HLR- SMC 5).......... E) peak data 2.4Mbps

6) CDMA IS 95 A network 6).......... F) C interface

7) CDMA 2000 1X-EVDO 7).......... G) B interface

8) CDMA 20001X network 8).......... H) D interface

9) Spreading Rate 1 9)……… I)2G Standard

10)Spreading Rate 3 10)……. J) 1.2288 Mcps.

2) SAY TRUE OR FALSE:

1) One BSC can control only one BTS

2) CC for India is 404

3) MCC for India is +91

4) MSIN and MIN are same.

5) ESN is a unique number assigned to the instrument.

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6) CDMA IS 95A is a 2G standard.

7) IMT 2000 is the official name for 3G.

8) CDMA 2000 is backward compatible to IS 95A.

9) CDMA 2000 1x uses single carrier.

10) CDMA 2000 3x is multi carrier system.

11) RC1 and2 of CDMA2000 correspond to rate set 1 and2 of IS95A.

12) Variable length Walsh Codes are used in IS95A

13) In RC1 and RC2 the no. of supplemental channels are seven maximum.

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CHAPTER – 12

INTRODUCTION TO BROADBAND

Objectives

The main objective of this chapter is to build up the following

i) To understand what is Broadband

ii) To understand the need of broadband

iii) To familiarize with Broadband Network

Overview

The confluence of two forces—the globalization of business and the networking of

information technology—has created the Internet economy. Advances in

telecommunications and data technology are creating new opportunities for countries,

businesses and individuals—just as the Industrial Revolution changed fortunes around

the globe. The new economy is defining how people do business, communicate, shop,

have fun, learn, and live on a global basis—connecting anyone to anything. The

evolution of Internet has come into existence & Internet service is expanding rapidly.

The demands it has placed upon the public network, especially the access network, are

great.

The rapid growth of distributed business applications; the proliferation of private

networks, e-commerce, and bandwidth-intensive applications (such as multimedia,

videoconferencing, and video on demand) generate the demand for bandwidth and access

network.

Moreover, an increasing number of consumers in this area further leads to a demand for

carrying these applications faster and reliable. Essentially, the broadband revolution is

about a huge increase in the range of services that can be offered via the Internet and

digital television. It promises a new age in entertainment and communications, as well as

a major boost for e-commerce. To meet this explosive demand for bandwidth and to

capitalize on this growing data opportunity, many data competitive local-exchange

carriers are aggressively targeting small businesses, SOHOs, and teleworkers in the

selected areas of the country in which they are operating.

However, technological advances promise big increases in access speeds, enabling public

networks to play a major role in delivering new and improved telecommunications

services and applications to consumers .The Internet and the network congestion that

followed, has led people to focus both on the first and last mile as well as on creating a

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different network infrastructure to avoid the network congestion and access problems.

The solution to this is Broadband.

As a result, many different companies have worked to develop "broadband" or

high-speed access. These broadband services will allow Internet subscribers to send or

receive video and audio content of digital quality; to download interactive graphic-rich

WebPages; and to allow Internet entrepreneurs to bring new services to market that take

advantage of speeds that will make the Internet truly interactive in real time. In short,

broadband promises to revolutionize the Internet in the same way that the introduction of

the Internet revolutionized communication.

In deploying these broadband services, service providers are developing whole

new ways to access the Internet. The Internet was never designed to handle the amount of

traffic that we are seeing today. The increasing penetration of broadband access and the

demand for multimedia applications is exacerbating this problem. The increasing

importance of the Internet and the importance of delivering Web content quickly and

reliably have put strains on the network.

Broadband indicates a means of connectivity at a high or ‘broad’ bandwidth,

which is capable of delivering multiple services simultaneously. It generally refers to

transmission of data over numerous frequencies. With the evolution of computer

networking and packet switching concept a new era of integrated communication has

emerged in the telecom world.

A concept of broadband services and the means of access technologies to bridge

the need of customer and service provider has emerged through out the world.

"Broadband" refers to high-speed Internet access. Traditionally, residential subscribers

have accessed the Internet by attaching a modem to their phone line and placing a local

call to their ISP. This "dial-up" or "narrow band" service has a number of constraints on

speed. Most commercial modems can achieve a maximum speed over the phone line of

56 kilobits per second (kbps).

Meaning of Broadband services

Broadband services are defined in various terms by different organization. Few of these

are given below: -

Original Bell System Definition

“A broadband channel is a communications channel having a Bandwidth greater than a

voice-grade channel, and therefore capable of higher-speed data transmission.”

1996 Telecom Reform Act

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Broadband services are capable of carrying “high-quality” voice, data, graphics, & video.

CCITT definition

A service requiring transmission channels capable of supporting rates greater than 1.5

Mbps or primary rate in ISDN or T1/E1 in digital terminology.

Definition of broadband

Recognizing the potential of ubiquitous Broadband service in growth of GDP and

enhancement in quality of life through societal applications including tele-education,

tele-medicine, e-governance, entertainment as well as employment generation by way of

high speed access to information and web-based communication, Government have

finalised a policy to accelerate the growth of Broadband services. In India, DoT has

issued a Broadband policy in 2004. Keeping in view the present status, Broadband

connectivity is defined at present as: -

“An ‘always-on’ data connection that is able to support interactive services

including Internet access and has the capability of the minimum download speed of 256

kilo bits per second (kbps) to an individual subscriber from the Point Of Presence (POP)

of the service provider intending to provide Broadband service where multiple such

individual Broadband connections are aggregated and the subscriber is able to access

these interactive services including the Internet through this POP. The interactive

services will exclude any services for which a separate licence is specifically required, for

example, real-time voice transmission, except to the extent that it is presently permitted

under ISP licence with Internet Telephony.”

Implementation of Broadband To Strengthen Broadband Penetration, the Government of India has formulated the

Broadband Policy whose main objectives are to:-

• Establish a regulatory framework for the carriage and the content of information

in the scenario of convergence.

• Facilitate development of national infrastructure for an information based society.

• Make available broadband interactive multimedia services to users in the public

network.

• Provide high speed data and multimedia capability using new technologies to all

towns with a population greater than 2 lakhs.

• Make available Internet services at panchayat (village) level for access to

information to provide product consultancy and marketing advice.

• Deploy state of art and proven technologies to facilitate introduction of new

services.

• Strengthen research and development efforts in the telecom technologies.

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Need of Broadband

The Internet, e-mail, web sites, software downloads, file transfers: they are all now part

of the fabric of doing business. But until now, it has not been possible for businesses to

fully take advantage of the benefits that technology can truly deliver. The reason for this

is a simple one - a lack of bandwidth. Even for small businesses, narrowband dial-up

access is no longer sufficient. It simply takes too long to do basic tasks, like downloading

a large file, and is increasingly being recognized as insufficient and inconvenient.

Kim Maxwell in his book-"Residential Broadband: An Insider's Guide to the Battle for

the Last Mile" has grouped potential residential broadband applications into three general

categories: "professional activities " (activities related to users' employment),

"entertainment activities " (from game playing to movie watching), and "consumer

activities " (all other non-employment and non-entertainment activities). as follows:

Professional Activities:

• Telecommuting (access to corporate networks and systems to support working at

home on a regular basis)

• Video conferencing (one-to-one or multi-person video telephone calls)

• Home-based business (including web serving, e-commerce with customers, and

other financial functions)

• Home office (access to corporate networks and e-mail to supplement work at a

primary office location)

Entertainment Activities:

• Web surfing (as today, but at higher speeds with more video content)

• Video-on-demand (movies and rerun or delayed television shows)

• Video games (interactive multi-player games)

Consumer Activities:

• Shopping (as today, but at higher speeds with more video content)

• Telemedicine (including remote doctor visits and remote medical analyses by

medical specialists)

• Distance learning (including live and pre-recorded educational presentations)

• Public services (including voting and electronic town hall meetings)

• Information gathering (using the Web for non-entertainment purposes)

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• Photography (editing, distributing, and displaying of digital photographs)

• Video conferencing among friends and family

These applications have different bandwidth requirements, and some of them are still out

of reach today. For example, all of the "professional" activities will likely be supported

with less than 1.0 Mbps of bandwidth. Similarly, web surfing and home shopping will be

supported with less than 1.0 Mbps of bandwidth.

Movies and video, however, demand more bandwidth. Feature length movies can

probably be delivered with 1.5 Mbps of bandwidth, but broadcast quality video will

probably require more— perhaps as much as 6.0 Mbps. Moreover, if high definition

television ("HDTV") is widely accepted as a new broadcast standard, that quality of

video would require almost 20.0 Mbps of bandwidth — much higher than the current

broadband technologies will support. Thus, although the technology is moving toward

flexible, high-quality video-on-demand, the necessary speed is probably still more than a

few years away from becoming a reality.

The Internet is poised to spin off thousands of specialized broadband services. The access

network needs to provide the platform for delivery of these services. Following are the

various applications or services, which are very popular in society and needs broadband

connectivity: -

Virtual Networks

The private virtual networks (LAN/WAN) can be used in an ample variety of multimedia

services, like bank accounts and central offices.

Education by distance

Education will not have any limits to reach from source to destination. Along with the

traditional school a concept of remote leaning center is emerged out and popular for

various courses. There is no limit of distance, area or location in such distance learning.

The student situated in the remote station can intervene directly to his class with a double

system via videoconference, whilst this happens, simultaneously, the file ex change

Telework

Organization firm workers that incorporate communication systems via satellite, can

work remotely connecting directly to their head offices Internet by a high speed

connection that permits users to work efficiently and comfortable.

Telemedicine

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Doctors situated in different clinics can stay in contact and consult themselves directly to

other regional medical centers, using videoconference and the exchange of high quality

images, giving out test results and any type of information. Also rural zone can have the

opinion of specialists situated in remote hospitals quickly and efficiently.

Electronic commerce

Electronic commerce is a system that permits users to pay goods and services by Internet.

Thanks to this service, any person connected to the network can aquire such services with

independence from the place that he is situated and during the 24 hours, simply using a

portable computer.

Services on BB-Multiplay (Triple Play i.e. Voice, Data and Video)

TVOIP

1. TVOIP (also called as IPTV) delivers television programming to households via

broadband connection using Internet protocols.

2. Internet Protocol Television (IPTV) is expected to change the way people watch

TV. As the name suggests, IPTV is television programs delivered to subscribers

through the Internet

3. It requires a subscription and IPTV set-top box (STB).

4. IPTV is typically bundled with other services like Video on Demand (VOD),

Voice Over IP (VOIP) or digital Phone, and Web access.

5. IPTV viewers will have full control over functionality such as rewind, fast-

forward, pause, and so on.

6. IPTV (Internet Protocol Television) is a system where a digital television service

is delivered by using Internet Protocol over a network.

7. If you've ever watched a video clip on your computer, you've used an IPTV

system in its broadest sense.

8. For residential users, IPTV is provided with Video On Demand and may be

bundled with Internet services such as Web access and VoIP.

9. Microsoft is one of the many companies developing solutions to support the

Internet Protocol TV (IPTV) market.

10. IPTV is an emerging technology and will evolve into a completely interactive

experience in the future!

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11. First things first: the Set-Top Box (STB), on its way out in the cable world, will

make resurgence in IPTV systems.

12. The box will connect to the home DSL line and is responsible for reassembling

the packets into a video stream and then decoding the contents.

13. The video stream is broken up into IP packets and dumped into the core network,

which is a massive IP network that handles all sorts of other traffic (data, voice,

etc.)

VOIP

1. The technology used to transmit voice conversations over a data network using

the Internet Protocol.

2. A category of hardware and software that enables people to use the Internet as the

transmission medium for telephone calls.

3. VoIP works through sending voice information in digital form in packets,

4. VoIP also is referred to as Internet telephony, IP telephony, or Voice over the

Internet (VOI)

Benefits of VoIP

1. Cost reduction

a. Toll by-pass

b. WAN Cost Reduction

2. Operational Improvement

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a. Common network infrastructure

b. Simplification of Routing Administration Business Tool

3. Integration

c. Voice mail, email and fax mail integration

d. Web + Call

e. Mobility using IP

The concept of socio economy has an important role in the field of

communication of data, audio, video, speech or any other kind of application. It is an era

of CAPEX and OPEX. Service providers and customers both are interested in economy

with fastest tool of communication with more throughput. Traditional circuit switching

network are not supporting the effective fast communication for new applications.

Broadband has emerged out with the evolution of packet switching network.

Communication of data for various applications is feasible to carry with different

throughput. These applications have different bandwidth requirements and most of them

need more bandwidth.

Various technologies are available to service providers by which they can extend

the Broadband services to customers. These technologies are mainly classified under two

categories i.e. Wire line and Wireless technologies. Existing infrastructure used to access

telecom services is exploited for broadband for economical aspects and faster

development. DSL on copper loops, Optical fiber, cable TV are the popular technologies

for Broadband. World has also entered in the field of wireless to provide the broadband

through GSM, CDMA, LMDS, MMDS, WiMax and Wi-Fi.

The public sector will be one of the key drivers of broadband demand. Pooling

requirements from hospitals, schools etc. could permit more cost effective procurement

and stimulate broadband rollout.

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TECHNOLOGY OPTIONS FOR BROADBAND SERVICES

Broadband Access Technology

Broadband access technology is broadly classified into two categories. They are

Wired Line & Wireless which is further classified as given below. Wireline technologies

include traditional telephone lines, coaxial cable lines, and fiber optic lines. Wireless

communications involve cellular and fixed wireless technology, high speed short range

communications and satellite transmission.

Wireless Wireline

3G Mobile DSL (Digital Sub’s Line)

Wi-Fi (Wireless Fidelity) Cable Modem

WiMAX Optical Fibre Technologies

LMDS & MMDS PLC(Power Line Communication)

FSO (Free Space Optics)

Satellite

Because physical infrastructure and geography are vastly different from country to

country, technology that works well in one geographic area may not work as well in

another. Therefore, it is up to each individual locality to determine the technologies that

best meet its needs. To handle the increasing bandwidth demand, localities are

considering upgrading their current telecommunications infrastructure or installing new

infrastructure. It is essential that communities and operators consider the present and

future needs of their citizens when examining the most appropriate systems. Broadband

should be considered an accelerator of economic development .

Factors affecting Broadband Access Choices

o Population density

o Existing infrastructure (e.g., twisted pair, cable, fiber)

o Government policies

o Competitive and regulatory dynamics

o Technology evolution

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o

The broadband services reached to customer from the three providers. Basically these are

Service Provider, Network Provider and Access Provider. The role of Network Provider

is to provide the services offered to customer through the access extended by Access

Provider. There are various types of networks which are capable of transmitting and

managing the broadband traffic to desired nodes or locations.

Wireline access technology through DSL, Fiber, Cable etc., generally adopts:

• IP based Network

• ATM Network

Wireless access technology through Wi-Fi, Wi-MAX, 3G mobile etc provides wireless

access to ingress point of any core network and migrates to Internet world.

Broadband technologies used in Asian countries

Broadband technologies go through two stages of development in Asian countries. In the

early stage, sharp technological divisions exist among players due to regulatory

constraints. There are various mode of access used by service providers in this field.

Following was the beginning scenario in various countries like Hong Kong, Malaysia,

Indonesia, India and Singapore: -

• Basic Telecom service providers adopted the use of ISDN/DSL

• CATV operators use cable modems

• Competitive players use wireless technologies.

In the later stage of development, technological divisions are shaped by geography and

infrastructure. The broadband started establishing and due to a progressive regulatory

framework it has matured in the market. In the countries like Korea and Philippines

service providers employ several technologies for the broadband in their networks.

• DSL and cable modems are used where the PSTN and CATV are in place.

Farm

Homes

MDU GigE Fiber

Cable

modem Twisted pair

DSL

Satellite

City: High-rise multi-family units

Suburbs: Individual single-family units

Rural: isolated single family unit

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• Where rainfall is light, LMDS is used to serve densely populated areas with little

infrastructure and unwired business districts.

• Satellite is used to service rural areas where population densities are low

Once newer technologies are available in the market, ISDN becomes relatively less

important. Established telephone companies are calculating the economics of converting

the Last Mile of existing networks to all-digital systems. Hong Kong and Singapore

citizens already have broadband access, such as movies on demand, through their local

telecom network. Cable-TV operators, too, are venturing into high-speed Internet access

through modified networks and end-user "cable modems." Advances in wireless

communications means that people starts surfing the net with cell phones at speeds

comparable to or greater than current home access.

The service provider converged voice and data network promises to be

implemented as nodes in a neighborhood or remote switches in regional locations.

NNeettwwoorrkk AArrcchhiitteeccttuurree ooff BBrrooaaddbbaanndd

Network architecture can be broadly classified into four categories:

1. Core or Backbone Network supporting multiple services of different QoS

implemented as packet switching nodes.

2. Aggregation or Distribution Network for extending the reachability to remote

locations and able to provide a cost effective solution to access the backbone

node.

3. Last mile connecting the subscriber – Access Network

4. Home Network or Subscriber’s network.

Summary

There are tremendous changes in the telecommunication technologies. With the evolution

of Internet telecom world has merged rapidly in computer network.

Broadband Internet connections allow users to download web pages and data many times

faster than conventional 'narrowband' Internet access. Broadband services are 'always-on'

- the computer is connected to the Internet continuously. Users pay a flat rate independent

of how long they spend on the Internet or the amount of data downloaded. Broadband

users typically spend four times as long online as narrowband customers and broadband

take-up has been faster than many comparable technologies, competitiveness. Broadband

is needed in the present scenario due to new technologies and emerging out various types

of Data communication applications.

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Digital Subscriber Line Technology

DSL is the family of technology, which transforms the narrowband Copper access

network into broadband. DSL does not refer to a physical line but the equipment, which

transforms the already existing media into a digital line. DSL has various family

members, which can be broadly classified as

• Symmetric DSL

Provide identical data rates upstream & downstream

• Asymmetric DSL

Provide relatively lower rates upstream but higher rates downstream.

DSL exploits the copper wires which have a much greater bandwidth or range of

frequencies than that demanded for voice without disturbing the line's ability to carry

phone conversations. The wires themselves can carry frequencies up to several million

Hertz.

There are several forms of digital subscriber lines, or xDSL with the x depending on the

particular variety of DSL. All xDSL connections use the same ordinary pair of twisted

copper wires that already carry phones calls among homes and businesses. Unlike cable

modem connections, which broadcast everyone’s cable signals to everyone on a cable

hub, xDSL is a point-to-point connection, unshared with others using the service.

The most common form of xDSL is ADSL. The A stands for asymmetric, meaning that

data transmission rate is not the same in both directions ie.,more bandwidth, or data-

carrying capacity, is devoted to data traveling downstream-from the Internet to your PC-

than to upstream data traveling from your PC to the Internet. The reason for the

imbalance is that, generally upstream traffic is very limited to a few words at a time, like

for example –an URL request and downstream traffic, carrying graphics, multimedia, and

shareware program downloads needs the extra capacity. Downstream data moves at about

8Mbps for the most common forms of DSL.

Transmission rates depend on the quality of the phone line, the type of equipment it uses,

the distance from the PC to a phone company switching office, and the type of xDSL

being used.

Common types of DSL:

HDSL (High Data-rate Digital Subscriber Line)

HDSL is a better way of transmitting T1/E1(Primary rate as per American standard

(1.544Mb/s) & European standard (2.048Mb/s)) over copper wires using less

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bandwidth without repeaters. Can be viewed as equivalent of PCM stream. It offers

the same bandwidth, both upstream and downstream. It can work up to a distance of

3.66 to 4.57 kms depending upon the speed required. When delivering 2048 kbps –On

2 phone lines, each line carries 1168 kbps and On 3 phone lines, each line carries 784

kbps. No provision exists for voice because it uses the voice band. HDSL over single

phone line requires more aggressive modulation, works only shorter distance and

requires better phone line

SDSL (Single-Line Digital Subscriber Line)

SDSL is a single line version of HDSL that transmits T1/E1 signals over a single

twisted pair line. It can work up to 3.7 kms on 0.5 mm dia cable. However, SDSL

will not reach much beyond 10,000 feet while ADSL reaches rates above 6Mbps at

the same distance. SDSL is mainly used by small businesses. It does not allow to use

the phone at the same time but the speed of downloading and uploading is the same

RADSL (Rate-Adaptive Digital Subscriber Line)

RADSL is a variation of ADSL where the modem adjusts the speed of connection

depending on the length and quality of the line.

VDSL •Very-high Data-rate DSL

Originally named VADSL (A –Asymmetric) but was later extended to support both

symmetric & asymmetric. Requires one phone line and supports both voice & data. It

works between 0.3-1.37 kms depending on speed. Upstream data rate can be 1.6-2.3

Mbps. Downstream data rate is 13-52 Mbps

ADSL Asymmetric Digital Subscriber Line

ADSL is one of the number of access technologies that can be used to convert the

access line into a high speed digital link and to avoid overloading the circuit switched

PSTN. You can use the same phone line for Internet service at the same time it’s

carrying a voice call because the two signals use widely separated areas of the

frequency spectrum. A splitter next to your xDSL modem combines the low-

frequency voice signals and the higher-frequency data signals. A splitter on the other

end of the line breaks the voice and data signals apart again, sending voice calls into

the plain old telephone system (POTS) and computer data through high-speed lines to

the Internet.

An ADSL circuit connects an ADSL modem on each end of a twisted pair telephone

line, creating three information channels

1. A high speed downstream channel

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2. A medium speed duplex channel

3. A basic telephone service channel

The basic telephone service channel is split off from the digital modem by filters, thus

guaranteeing uninterrupted basic telephone service, even if ADSL fails. To create

multiple channels, ADSL modems divide the available bandwidth of a telephone line

in one of two ways: frequency-division multiplexing (FDM) or echo cancellation, as

shown in Figure. FDM assigns one band for upstream data and another band for

downstream data. The downstream path is then divided by time-division multiplexing

into one or more high-speed channels and one or more low-speed channels. The

upstream path is also multiplexed into corresponding low-speed channels. Echo

cancellation assigns the upstream band to overlap the downstream, and separates the

two by means of local echo cancellation, a technique. With either technique, ADSL

splits off a 4-kHz region for basic telephone service using a splitter.

ADSL uses a different set of frequencies for data and does not interfere with

telephone conversation. Conversely, making a phone call while accessing the Internet

does not affect the speed of the ADSL connection.

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There are two different standards for ADSL:

• CAP (Carrier less Amplitude/Phase) which was used on the early installations of

ADSL.

• DMT (Discrete MultiTone) which is the official ANSI standard.

CAP is well-understood and relatively inexpensive because it is a single-carrier

modulation technique that uses a wide passband and is susceptible to narrowband

interference. DMT uses multiple carriers. that uses many narrowband channels.

Therefore, DMT is capable of more speed than CAP. This is one reason that the ANSI

committee T1E1.4 accorded it standards status in document T1.413. This standard calls

for 256 sub bands of 4 KHz each, thereby occupying 1.024 GHz.

CAP divides the signals of the telephone line into three bands: voice, upstream channel

and downstream channel. Voice conversations are carried in the 0 to 4 KHz band as they

are in all POTS circuits. The upstream channel that carries data from the user to the

server is between 25 and 160 KHz. The downstream channel begins at 240KHz with a

maximum of 1.5 MHz which depends on a number of conditions such as distance, line

noise and number of users. CAP by keeping the three channels widely separated,

minimises the possibility of interference both between channels on one line and signals

on different lines.

DMT also operates by dividing signals into separate channels without using two quite

broad channels for upstream and downstream. The modulation technique that has become

standard for ADSL is called the Discrete Multitone Technique, which combines QAM

and FDM. In ADSL, the available bandwidth of 1.104 MHz is divided into 256 channels.

Each channel uses a bandwidth of 4.312 KHz. Each channel is 4KHz wide with a guard

band of .312KHz.. Hence the name Discrete Multitone.

Each sub carrier can support maximum15 number of bits. Depending on signal to noise

ratio for that sub carrier, a decision is taken as to how many bits that particular sub carrier

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can support. Every channel is monitored and if the quality is low, the signal is shifted to

another channel. DMT constantly shifts signals between different channels, looking for

the best channels for transmission and reception. Moreover, some of the lower frequency

channels, are used as bi-directional channels for upstream and downstream. Keeping up

with the quality of all channels, monitoring and sorting the information on the bi-

directional channels, makes the implementation of DMT more complex than CAP.

However, it provides more flexibility on lines of different quality.

� Voice Channel 0 is reserved for voice communication.

� Idle Channels 1 to 5 are not used, to allow a gap between voice and data

communication.

� Upstream data and control. Channels 6 to 30 (25 channels) are used for upstream

data transfer and control. One channel is for control and 24 channels are for data

transfer. If there are 24 channels, each using 4KHz (out of 4.312KHz available)

with QAM modulation, we have 24x4000x15, or a 1.44Mbps bandwidth, in the

upstream direction.

� Downstream data and control Channels 31 to 255 (225 channels) are used for

downstream data transfer and control. One channel is for control, and 224

channels are for data. If there are 224 channels, we can achieve upto 224 channels

are for data. If there are 224 channels, we can achieve upto 224 x 4000 x 15 or

13.4 Mbps.

This is the equivalent of 250 modems connected to your PC at once

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LOW-PASS FILTER

CAP and DMT are similar in one way as they both use frequencies above 4KHz. That is

why every ADSL user needs small filters to attach to the outlets that do not provide

signal to the ADSL modem. The low pass filter blocks all signals above 4KHz. The

reason is that all voice conversations take place below 4KHz and the low-pass filter

prevents the data signals from interfering with standard telephone calls. The basic

telephone service channel is split off from the digital modem by splitter at client site

ADSL characteristics:

1. Asymmetric ? The data can flow faster in one direction than the other. More

precisely, Data transmission is faster downstream (to the user) to the subscriber

than upstream (from the user). Customers do not need a high bi-directional

transmission speed. They actually connect to the internet in a relatively passive

mode because the amount of data they download is enormously higher than the

amount of data they transmitting.

2. Digital ? No type of communication is transferred in an analog method. All data is

purely digital, and only at the end, modulated to be carried over the line.

3. Subscriber L ine ? The data is carried over a single twisted pair copper loop to the

subscriber premises

ADSL Architecture

Delivery of ADSL services requires a single copper pair configuration of a standard voice

circuit with an ADSL modem at each end of the line, splits the telephone line into three

information channels: a high speed downstream channel, a medium speed upstream

channel, and a Plain Old Telephone Service (POTS) channel for voice or an ISDN

channel.

The ADSL network components are:

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1. The ADSL modem at the customer premises, that is called an ADSL transceiver

unit-remote (ATU-R). It provides local loop termination on the customer side.

2. The modem of the central office that is called an ADSL Transmission Unit-

central office (ATU-C). It terminates the ADSL local loop at the central office

premises.

3. DSLAM - DSL access multiplexer. Many ATU-C units are inserted into the

DSLAM. This unit can connect through an ATM or an ETHERNET access

network to the internet.

4. Splitter :- An electronic low pass filter that separates the analogue voice or ISDN

signal from ADSL data frequencies when they get to the subscriber premises. For

outgoing traffic, when they are transmitted from the subscriber premises, it

combines the voice and the data frequencies onto one line. This allows a POTS

phone connection to operate at the same time as ADSL digital data is transmitted

or received on the same line. One splitter is located at the central office and

another at the subscriber premises. The splitter at the central office can be

separate device or may be incorporated into the DSLAM.

Figure 1. ADSL Loop Architechture

The ADSL modem will combine the data stream from the computer, divide it into the

available aggregate channels, interleave, attach an ECC and then finally modulate the

data on the copper wires to be received by the demodulator at the other end. The

receiving end will perform error checking, correct the errors or request for re-transmit if

not possible. Interleaved blocks will be kept in the buffers until the entire blocks can be

reconstructed from subsequent frames.

Performance

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On the outside, ADSL looks simple and transparent, but on the inside there is a miracle of

modern technology.

In ADSL, downstream data rates can be between 1.5 Mbps and 8 Mbps, while upstream

data rates are between 16 Kbps and 832 Kbps. The minimum configuration provides 1.5

or 2.0 Mbps downstream and a 16 kbps duplex channel; others provide rates of 6.1 Mbps

and 64 kbps duplex. Products with downstream rates up to 8 Mbps and duplex rates up to

640 kbps are also available.

Known Problems & Solutions

The factors that can influence the downstream data rates are the length of the line, the

gauge of the line, presence of bridged taps and crosstalk from other wires in the same

time that cause noise. Line attenuation increases with line length and frequency, and

decreases as wire diameter increases. Also, some copper loops use different gauge wires

at different points and this can cause reflections in the signal, effectively attenuating

some frequencies. Ignoring bridged taps, ADSL will perform as follows:

Data Rate Wire gauge Wire size Distance

1.5 or 2 Mbps 24 AWG 0.5 mm 5.5 km

1.5 or 2 Mbps 26 AWG 0.4 mm 4.6 km

6.1 Mbps 24 AWG 0.5 mm 3.7 km

1.5 or 2 Mbps 26 AWG 0.4 mm 2.7

Table 1. ADSL performance over different loop conditions

Some solutions to the problems that influence the data rate in ADSL:

1. Placing an optical network unit closer to the neighborhoods which are located too

far from the central office to obtain useful ADSL connection speeds. Data is

transmitted over fiber to the optical network unit, which then distributes the

signals to ADSL modems for connection to the residences.

2. Most ADSL providers select a series of speeds that are smaller than the maximum

so that more customers can obtain higher speeds, because the speed varies with

the length and the gauge of the line.

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3. ADSL modems incorporate forward error correction that dramatically reduces

errors caused by impulse noise. Error correction on a symbol-by-symbol basis

also reduces errors caused by continuous noise coupled into a line.

Framing

The data of the ADSL transport is organized into frames and superframes. One

superframe is built of 68 ADSL frames and one sync frame. The sync frame is used to

provide superframe synchronization, identifying the start of the superframe.

Because the ADSL lines can vary and are asymmetrical, the frame sizes may vary also.

Therefore, there are no absolute frame sizes for the superframe.

Each superframe must be sent every 17 ms and each frame must be sent every

250microseconds.

Each frame contains:

1. Fast Byte - used for special superframe-related processing functions.

2. Fast Data - transmits time sensitive information and it can vary in length.

3. Forward Error Correction (FEC) - is used to ensure the accuracy of the fast

data.

4. Interleaved Data - is the user data that the ADSL interface transmits, typically

internet data, and it can be vary in length.

ADSL DMT frame rate synchronized to the 4 kHz, and multiplexed into two data buffers

fast and interleaved. FEC is applied to each of the buffers.

Figure1. ADSL superframe

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Features of ADSL

• Allows simultaneous access to the line by the telephone and the computer

• In case of power/ADSL failure, data transmission is lost but basic

telephone service will be operational

• ADSL Provides 16-640 kbps upstream and 1.5-9 Mbps downstream. It can

work up to a distance of 3.7 to 5.5 kms depending upon the speed required

Advantages of ADSL

• You can leave your Internet connection open and still use the phone line

for voice calls.

• The speed is much higher than a regular modem

• DSL doesn't necessarily require new wiring; it can use the phone line you

already have.

• The company that offers DSL will usually provide the modem as part of

the installation.

Variants of ADSL

THE ADSL2 FAMILY

As ADSL popularity grew, it begun spreading world-wide and clearly became the

carriers, service providers and subscribers choice of broadband media. Their feedback,

along with rising demand for improvements, contributed to the completion of a new

family of standards named ADSL2. These, beside offering higher rates, are more user-

friendly for subscribers on one hand, and more profitable to carriers on the other hand. In

addition, most ADSL2's modems are backward compatible and support the simple

ADSL, making it easier to step into the next level when upgrading.

How can we choose the best configuration for a connection between a certain consumer

and the CO? Well, one of ADSL2's characteristics is being almost entirely automated,

and here is no different. A new feature called "Automode" enables service providers to

give their customers the optimal level of service, by analyzing the line condition and

gathering various information during connection between the CO and the customer

premise, and then choosing (automatically) the best line configuration.

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Figure 1. Automode in action.

The CO detects on initialization that the line to the CPE is too long for conditional

ADSL2/2+. Also, the consumer doesn't require extremely high speed services such as

HD video, and so automode decide to connect via RE-ADSL2 for optimum results.

ADSL2

The ADSL2 standard G.992.3 (and G.992.4 - the splitterless ADSL2 'lite' version) was

approved in October 02 by the ITU, and is looking like a promising next-generation

ADSL technology. The basic form of ADSL2 over POTS is also referred to as ADSL2

annex A, or simply ADSL2, and was one of its earliest member of the family. We now

discuss the new characteristics and main benefits of ADSL2 over its predecessor - the

ADSL:

Rate and Reach improvement –

• ADSL2 provides greater download stream (12Mbps compared to 8Mbps in

ADSL) due to improved modulation efficiency, and greater upstream (1Mbps).

Higher data rates are also gained due to improved modulation technique.

• ADSL2 also presents several improvements in the initialization state machine,

which also helps avoiding line interference. This has also enabled to extend the

reach of ADSL2 in regard to ADSL by approx 600 feet (~185m), and to increase

data rates. Some of these improvements are listed in the next paragraph.

• In addition, ADSL2 reduce framing overhead by allowing to determine the

number of overhead bits in range 4-32 kbps (compared to ADSL frames which

had a fixed 32kbps overhead per frame).

• Also, ADSL2 achieves higher coding gain due to new framing algorithms (the

reed-solomon forward error correction code).

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Improved initialization --

• ADSL2 improves the the line initialization process, which is more automated than

ADSL, thus reducing error rate and increasing throughput. Improvements

includes:

• Reducing near-end echo and crosstalk interferences at the binder.

• Enabling RFI (radio frequency interference) cancellation techniques by disabling

tones during initialization.

• Receiver allocated pilot tone helps avoiding bridged taps and RFI.

• Receiver and Transmitter reach optimum in signal processing functions by

controlling the length of certain initializations states.

• Better equalization with spectrum shaped init signals to improve channel

discovery.

Diagnostics –

ADSL2's transceivers have been enhanced with Real-time performance-monitoring

capabilities to provide measurements of line noise, loop attenuation, and signal-to-noise

ratio (SNR) at both ends of the line, even when connection can not be achieved due to

poor quality of the line. This is of great value to both service providers which can use it

to track and prevent failures, and to carriers to determine if a customer qualifies for

higher data-rate services.

Power consumption –

ADSL2 introduces two power management modes to help reduce power consumption:

the first is called "L2 low power mode" which enables power saving at the transceivers in

the central office (the ATU-C) by going into low power state whenever internet traffic is

decreased. The moment traffic has increased its detected by the transceiver which goes

back into full power mode (called L0).

The other power saving technique is called "L3 low power mode", and is used on both

CO's transceivers and the remote transceivers (the ATU-R), and basically enters the

transceiver into sleep mode when no traffic is detected on the connection. Returning into

full power mode requires re-initialization, but as we see shortly it is quite fast.

Fast startup -

A fast start-up mode reduces initialization time from about 10 seconds (in ADSL) to

approx 3 seconds.

ADSL2 Plus

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ADSL2Plus (also known as ADSL2+), has joined the ADSL2 family In January 2003 as

standard g992.5 after been approved by the ITU.

ADSL2+ doubles the downstream frequency band from 1.1Mhz in basic ADSL2, up to

2.2Mhz in ADSL2+, therefore increases the downstream data rate on shorter phone lines,

reaching 20Mbps on lines of max length of 5000ft (~1.5km). ADSL2+ upstream remains

1Mbps, depending of course on loop conditions.

Figure1: ADSL2+ compared to ADSL/ADSL2 in terms of bandwidth

ADSL2+ can also be used to reduce crosstalk. It has the ability to use only tones between

1.1Mhz-2.2Mhz, it can mask all downstream frequencies below 1.1Mhz. This comes in

handy when ADSL2+ services from a remote terminal co-exist with other ADSL services

(from the CO) and are sharing the same binder. Crosstalk interferences from the remote

terminal could be performance hazard for the ADSL services coming from the CO. The

masking of frequencies below 1.1 from the remote terminal eliminates the crosstalk

interferences almost completely.

Figure2: Reducing Crosstalk with ADSL2+

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RADSL

The Rate-adaptive ADSL is a non standard version of ADSL, in which the DSL modem

have the additional capability of adjusting bandwidth to the quality of the phone line not

only at the start of the connection but also at any time during the data transmission.

RADSL increases the maximum distance supported from 3.5 to around 5.5km (18000ft)

which makes it ideal for suburban neighborhoods. However, once again we pay the cost

of reduced data rate.

CABLE MODEM

Cable companies are now competing with telephone companies for the customer who

wants high-speed access to the Internet. DSL technology provides high-data-rate

connections for residential subscribers over the local loop. This imposes an upper limit on

the data rate. Cable TV operators use the cable TV network . In this section, we briefly

discuss this technology.

Traditional cable network: -

Cable TV network started to distribute broadcast video signal to locations with poor or no

reception in the late 1940s .It was called Community Antenna TV (CATV) because an

antenna at the top of a hill or building received the signals from the TV stations and

distributed them, via coaxial cables, to the community. Figure 1 shows a schematic

diagram of a traditional cable TV network.

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Traditional cable TV network

The cable TV office called the head end, receives video signals from broadcasting

stations and feed the signals into coaxial cables. The signals become weaker and weaker,

so amplifiers were installed through the network to amplify the signals. There could be

up to 35 amplifiers between the head end and the subscriber premises. At the other end,

splitters split the cable, and drop cables makes the connections to the subscriber premises.

The traditional cable TV system used coaxial cable end to end. Due to attenuation of the

signals and the use of a large number of amplifiers, communication in the traditional

network was unidirectional (one way). Video signals were transmitted downstream, from

the head end to the subscriber premises.

HFC Network

The second generation of cable network is called a hybrid fiber-coaxial (HFC) network.

The network uses a combination of fiber-optic and coaxial cable. The transmission

medium from the cable TV office to a box, called the fiber node, is optical fiber. From

the node through the neighborhood and into the subscriber premises is still coaxial cable.

The transmission medium from the cable TV office to a box, called the fiber node, is

Figure 1

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optical fiber; from the fiber node through the neighborhood and into the house is still

coaxial cable, figure shows a schematic diagram of an HFC network.

The regional cable head (RCH) normally serves up to 400,000 subscribers. The RCHs

feed the distribution hubs, each of which serves up to 40,000subscribers . The distribution

hubs plays an important role in the new infrastructure. Modulation and distribution of

signals are done here; the signals are then fed to the fiber node through fiber-optic cable.

The fiber node split into the analog signals so that the same signal is sent to the each

coaxial cable. Each coaxial cable serves up to 1000subscribers .the uses of fiber optic

cable reduce the need for amplifiers down eight or less.

One reason for moving from traditional to hybrid infrastructure is to make the cable

network bi-directional (two way).

Hybrid fiber-coaxial (HFC) network

Even in an HFC system, the last part of the network, from the fiber node to the subscriber

premises, is still coaxial cable has a bandwidth that ranges from 5 to 750 MHz

(approximately). The cable company has divided this bandwidth into three bands,: video,

downstream data, and upstream data, as shown in figure 3.

Figure 2

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Video Band

The downstream only video band occupies frequencies from 54 to 550 MHz. Since each

cable TV channels occupies 6MHz, this can accommodate more than 80 channels.

Division of coaxial cable band by CATV

Downstream Data Band

The downstream data (from the internet to the subscriber premises) occupies the upper

band , from 550 to 750 MHz .this band is also divided into 6-MHz channels .

Modulation Downstream data are modulated using the 64 - QAM (or possibly 256-

QAM) modulation technique.

Data rate There are 6bits for each baud in 64 –QAM. One bit is used for forward

error correction; this leaves 5 bits of data per baud. The standard

specifies 1Hz for each baud; this means that theoretically, downstream

data can be received at 30 Mbps (5 bits /Hz x 6 MHz). The standard

specifies only 27 Mbps. However, since the cable modem is connected

Figure 3

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to the computer through a 10 base –T cable, this limits the data rate to 10

Mbps.

Upstream Data Band

The up stream data (from the subscriber premises to the internet) occupies the lower

band, from 5 to 42MHz. This band is also divided into 6-MHz channels.

Modulation The upper stream data used lower frequencies that are more susceptible

to noise and interference. For this reason, the QAM technique is not

suitable for this band. a better solution is QPSK.

Data rate There are 2 bit for each baud in QPSK. The standard specifies 1 Hz for

each baud; this means that, theoretically, up stream data can be sent at

12 Mbps (2 bit/ Hz ҳ 6 Mbps). However, the data rate is usually less

than 12 Mbps.

Sharing

Both upstream and down band are shared by the subscribers.

Upstream sharing

The up stream data bandwidth is only 37 MHz. This means that there are only six 6-MHz

channels available in the upstream direction. A subscriber needs to use channels to send

data in the upstream direction. The question is, how can six channels be shared in an area

with 1000, 2000 or even 100,000 subscribers? The solution is timesharing. The

bandwidth is divided into channels using FDM; these channels must be shared between

subscribers in the same neighborhood. The cable provider allocates one channel,

statically for a group of subscribers. If one subscriber wanted to send data, she or he

contends for the channel with other who wants access; the subscriber must wait until the

channel is available.

Downstream sharing

We have a similar situation in the down stream .the downstream band has 33 channels of

6 MHz. a cable provider probably has more than 33 subscriber; there –fore, each channel

must be shared between a group of subscribers. However the situations is different for the

downstream directions; here we have a multicasting situation .if there are data for any of

the subscriber in the group, the data are sent to that channels. Each subscriber is sent the

data. But since each subscriber also has an address registered with the provider, the cable

modem for the group matches the address carried with the data to the address assigned by

the provider. If the address matches, the data are kept; otherwise, they are discarded.

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CM AND CMTS

To use a cable network for data transmission, we need two key devices; a CM and a

CMTS.

CM

The cable modem (CM) is installed on the subscriber premises. it is similar to an ADSL

modem .figure 5 shows its location

CMTS

The cable modem transmission system (CMTS) is installed inside the distribution hub by

the cable company. It receives data from the Internet and passes them to the combiner,

which sends them to the subscriber. The CMTS also receives data from the subscriber

and passes them to the Internet.

Cable modem transmission system (CMTS)

Figure 4

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Cable modem (CM)

Data Transmission Schemes:

During the last few decades, several schemes have been designed to create a standard for

data transmission over an HFC network. Prevalent is one devised by Multimedia Cable

Network System (MCNS), called Data Over Cable System Interface Specification

(DOCSIS). DOCSIS defines all the protocols necessary to transport data from a CMTS to

a CM.

Up stream communication

The following is a very simplified version of the protocol defined by DOCSIS for

upstream communication. It describes the steps that must be followed by a CM.

1. The CM checks the downstream channels for a specific packet periodically sent

by the CMTS. The packet asks any new CM to announce itself on a specific

upstream channels.

2. The CMTS send a packet to the CM, defining its allocated downstream and

upstream channels.

3. The CM then starts a process, called ranging, which determine the distance

between the CM and CMTS. This process is required for synchronization between

Figure 5

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all CM s and CMTS s for the minislots used for timesharing of the upstream

channels.

4. The CM sends a packet to the ISP, asking for the Internet address.

5. The CM and CMTS then exchange some packets to establish security parameters,

which are need for public network such as cable TV.

6. The CM sends its unique identifier to the CMTS.

7. Upstream communication can start in the allocated upstream channel; the CM can

contend for the mini slots to sent data.

Downstream communication

In the downstream direction, the communication is much simpler. There is no contention because there is only one sender. The CMTS sends the packet with the address of the receiving CM, using the allocated downstream channel.

DIGITAL SUBSCRIBER LINE ACCESS MULTIPLEXER(DSLAM)

Introduction

To enable DSL technology, service providers must have a DSLAM located in their

networks to interact with the customer premises equipment (CPE) at the end user

location.

DSLAM is an integrated hardware and software system that allows the user to access

Broadband services as well as originate and terminate telephone calls over the same

single pair of copper wires

A Digital Subscriber Line Access Multiplexer (DSLAM) delivers exceptionally high-

speed data transmission over existing copper telephone lines

A DSLAM separates the voice-frequency signals from the high-speed data traffic and

controls and routes digital subscriber line (xDSL) traffic between the subscriber's end-

user equipment (router, modem, or network interface card [NIC]) and the network service

provider's network. A DSLAM takes connections from many customers and aggregates

them onto a single, high-capacity connection to the Internet. DSLAMs are generally

flexible and able to support multiple types of DSL in a single central office, and different

varieties of protocol and modulation, both CAP and DMT, in the same type of DSL. The

DSLAM may provide additional functions including routing or dynamic IP address

assignment for the customers.

The DSLAMs is in general be collocated with existing PSTN exchanges which provide

last mile access to customers over copper wire up to average span lengths of 3 kms.

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Features of DSLAM

A digital subscriber line access multiplexer (DSLAM) delivers exceptionally high-speed

data transmission over existing copper telephone lines. A multiservice DSLAM is a

broadband-access network element (NE) that combines support for multiple DSL

transmission types. When coupled with high-capacity asynchronous transfer mode

(ATM) switching, multiservice DSLAMs deliver scalability, port density, and a

redundant architecture for reliability. Multiservice DSLAMs, together with various CPE

elements, can enable the relatively efficient deployment of broadband networks for high-

speed Internet access as well as voice and video applications. The basic features of

Digital Subscriber Line Access Multiplexure (DSLAM) are describes below: -

• DSLAM aggregates the subscriber lines

• A Digital Subscriber Access Multiplexer delivers exceptionally high speed data

transmission over existing copper telephone lines

• DSLAM separates Voice and Data of the Subscriber i.e. it separates the voice

frequency signal from High Speed data traffic

• Routes and Controls Digital Subscriber Line (xDSL) traffic between the

subscriber’s end-user equipment (Router, Modem, or Network Interface Card

(NIC) and the Network Service Provider’s network.

• Voice is given to the exchange switch

• Data is fed to the IP Network through the LAN Switch

• DSLAMs have been categorized in to 6 types based on no. of ports (480, 240,120,

64, 48 & 24) provided and planned for deployment based on the expected demand

• DSLAM provides Access from 128Kbps to 8Mbps

• DSLAM supports for QOS features such as Committed Access Rate between CPE

and DSLAM, Traffic Policing per port

• DSLAM works Satisfactory without any degradation in performance and without

using any repeater/regenerator over a distance for various access speeds for

0.5mm copper pair.

Distance wise downstream bit rate in DSLAM

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Downstream bit rate Distance

6 Mbps 1.5 Kms

2 Mbps 3.5 Kms

1 Mbps 4.0 Kms

Implementation Of DSLAM

Broadband connectivity is extended to these DSLAM through the core network via the

LAN switch. Commonly it is available with 480, 240, 120, 64, 48 and 24 ports. DSLAMs

are generally aggregated through a Fast Ethernet or Gigabit Ethernet Interface. DSLAMs

are available with different types of access modules and capacities. The FX or GBIC

module in DSLAM and LAN switch should be capable of driving up to 10km on a single

mode fibre. The SX or GBIC module will support

Connectivity of DSLAM

DSLAM is connected to ATM or IP based core network through the networking

elements. It aggregates the data traffic of all the users provided to it and extends to core

network. The telephone traffic of each user is separated by splitter available in it and

transmits to PSTN network. DSLAM provides user access through user access layer and

Connectivity to IP backbone is provided through IP convergence layer.

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CHAPTER - 13

INTELLIGENT NETWORK

Overview of Intelligent Network Architecture Over the last thirty years, one of the major changes in the implementation of

Public Switched Telephone Networks (PSTNs) has been the migration from analogue to

digital switches. Coupled with this change has been the growth of intelligence in the

switching nodes. From a customer's and network provider's point of view this has meant

that new features could be offered and used.

Since the feature handling functionality was resident in the switches, the way in

which new features were introduced into the network was by introducing changes in all

the switches. This was time consuming and fraught with risk of malfunction because of

proprietary feature handling in the individual switches.

To overcome these constraints the Intelligent Network architecture was evolved

both as a network and service architecture.

In the IN architecture, the service logic and service control functions are taken out

of the individual switches and centralized in a special purpose computer. The interface

between the switches and the central computer is standardised. The switches utilize the

services of the specialized computer whenever a call involving a service feature is to be

handled. The call is switched according to the advice received by the requesting switch

from the computer. For normal call handling, the switches do not have to communicate

with the central computer.

Objectives of the Intelligent Network

The main objectives of the IN are the introduction and modification of new services in a manner which leads to substantial reduction in lead times and hence development costs, and to introduce more complex network functions.

An objective of IN is also to allow the inclusion of the additional capabilities and flexibility to facilitate the provisioning of services independent of the underlying network's details. Service independence allows the service providers to define their own services independent of the basic call handling implementation of the network owner.

The key needs that are driving the implementation of IN are :

• Rapid Service Deployment

Most business today require faster response from their suppliers, including

telecommunication operators. By separating the service logic from the underlying switch

call processing software, IN enables operator to provide new services much more rapidly.

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• Reduced Deployment Risk

Prior to IN, the risk associated with the deployment of new services was

substantial. Major investments had to be made in developing the software for the services

and then deploying them in all of the switches.

With the service creation environment available, the IN services can be

prototyped, tested and accessed by multiple switches simultaneously. The validated

services can then be rolled out to other networks as well.

• Cost Reduction

Because the IN services are designed from the beginning to be reusable, many

new services can be implemented by building on or modifying an existing service.

Reusability reduces the overall cost of developing services. Also, IN is an architecture

independent concept, i.e. it allows a network operator to choose suitable development

hardware without having to redevelop a service in the event that the network

configuration changes.

• Customization

Prior to IN, due to complexity of switch based feature handling software, the

considerable time frame required for service development prevented the provider from

easily going back to redefine the service after the customer started to use it. With IN, the

process of modifying the service or customization of service for a specific customer is

much less expensive and time consuming.

The customization of services is further facilitated by the integration of advanced

peripherals in the IN through standard interfaces. Facilities such as voice response

system, customized announcements and text to speech converters lead to better call

completion rate and user-friendliness of the services.

IN Architecture

Building upon the discussion in the previous section, one can envisage that an IN

would consist of the following nodes :

• Specialized computer system for – holding service logic, feature control, service

creation, customer data, and service management.

• Switching nodes for basic call handling.

• Specialized resources node.

The physical realization of the various nodes and the functions inherent in them is flexible. This accrues form the "open" nature of IN interfaces.

Let us now look at the nodes that are actually to be found in an IN implementation.

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The service logic is concentrated in a central node called the Service Control Point (SCP0.

The switch with basic call handling capability and modified call processing model for querying the SCP is referred to as the Service Switching Point (SSP).

Intelligent Peripheral (IP) is also a central node and contains specialized resources required for IN service call handling. It connects the requested resource towards a SSP upon the advice of the SCP.

Service Management Point (SMP0 is the management node which manages services logic, customers data and traffic and billing data. The concept of SMP was introduced in order to prevent possible SCP malfunction due to on-the-fly service logic or customer data modification. These are first validated at the SMP and then updated at the SCP during lean traffic hours. The user interface to the SCP is thus via the SMP.

All the nodes communicate via standard interfaces at which protocols have been defined by international standardization bodies. The distributed functional architecture, which is evident from the above discussion, and the underlying physical entities are best described in terms of layers or planes. The following sections are dedicated to the discussion of the physical and functional planes.

Physical Plane

Service Switching Point (SSP)

The SSP serves as an access point for IN services. All IN services calls must first

be routed through the PSTN to the "nearest" SSP. The SSP identifies the incoming call as

an IN service call by analysing the initial digits (comprising the "Service Key") dialled by

the calling subscriber and launches a Transaction Capabilities Application Part (TCAP)

query to the SCP after suspending further call processing. When a TCAP response is

obtained from the SCP containing advice for further call processing, SSP resumes call

processing.

The interface between the SCP and the SSP is G.703 digital trunk. The MTR,

SCCP, TCAP and INAP protocols of the CCS7 protocol stack are defined in this

interface.

Service Control Point (SCP)

The SCP is a fault-tolerant online computer system. It communicates with the SSPs

and the IP for providing guidelines on handling IN service calls. The physical interface to

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the SSPs is G.703 digital trunk. It communicates with the IP via the requesting SSP for

connecting specialized resources.

SCP stores large amounts of data concerning the network, service logic, and the IN customers. For this, secondary storage and I/O devices are supported. For more details refer to the chapter on the "SCP Architecture".

As has been commented before, the service programs and the data at the SCP are

updated from the SMP.

Service Management Point (SMP)

The SMP, which is a computer system, is the front-end to the SCP and provides the user interface. It is sometimes referred to as the Service Management System (SMS). It updates the SCP with new data and programs (service logic) and collects statistics from it. The SMP also enables the service subscriber to control his own service parameters via a remote terminal connected through dial-up connection or X.25 PSPDN. This modification is filtered or validated by the network operator before replicating it on the SCP.

The SMP may contain the service creation environment as well. In that case the new services are created and validated first on the SMP before downloading to the SCP.

One SMP may be used to manage more than one SCPs. Intelligent Peripheral (IP)

The IP provides enhanced services to all the SSPs in an IN under the control of the SCP. It is centralized since it is more economical for several users to share the specialized resources available in the IP which may be too expensive to replicate in all the SSPs. The following are examples of resources that may be provided by an IP:

• Voice response system • Announcements • Voice mail boxes • Speech recognition system • Text-to-speech converters

The IP is switch based or is a specialized computer. It interfaces to the SSPs via ISDN Primary Rate Interface or G.703 interface at which ISUP, INAP, TCAP, SCCP and MTP protocols of the CCS7 protocol stack are defined.

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The IN architecture is depicted in Fig.1

Fig. 1 IN Architecture

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Distributed Functional Plane

Functional model of IN contains nine functional entities (FE's) which are

distributed over various physical entities (PE's) described in the previous section. A

functional entity is a set of unique functions. Brief description of the FE's is given below :

CCAF

Call Control Agent Function, gives users access to the network.

CCF

Call Control Function provides the basic facility for connecting the transport (e.g.

speech). It involves the basic switching function and trigger function for handling the

criteria relating to the use of IN.

SSF

Service Switching Function is used to switch calls based on the advice of the SCF

at the SCP. This function provides a service independent interface.

SCF

It contains the service logic components and advises the SSF at SSP on further call

handling.

SDF

Service Data Function contains the user related data and data internal to the network.

SRF

Specialized Resources Function covers all types of specialized resources other

than the connection resources that are in the exchange (e.g. recorded announcements,

tones, conference bridges, etc.).

SCEF

Service Creation Environment Function specifies, develops, tests and deploys the

services on the network.

SMAF

Service Management Access Function provides an interface between service

management function and the service manager who may be an operator.

SMF

Service Management Function enables a service to be deployed and used on IN.

Fig. 2 depicts the distribution and interconnection of the various functional entities.

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Management interface In real time interface Signaling circuit interface

Fig. 2 Distributed Functional Entities

The distribution of functional entities over the physical entities and their inter-connection is summarized in Table 1 and 2 below. It may be noted that all the physical entities may not be present in all INs as the choice of functional entities to be provisioned is entirely up to the service provider.

Table 1 Distribution of FE's over PE's

Physical Entity Possible Functional Entities SSP CCF, SSF, CCAF SCP SCF, SDF SMP SCEF, SMF, SMAF

IP SRF

Table 2 FE-FE Relationship to PE-PE Relationship

FE-FE PE-PE Protocol SSF-SCF SSP-SCP INAP, TCAP, SCCP and MTP SCF-SDF SCP-SDP X.25 or Proprietary SCF-SRF SCP-IP

SCP-SSP-IP INAP, TCAP, SCCP and MTP ISUP, INAP, TCAP, SCCP and MTP

SRF-SSF SSP-IP ISUP and MTP

8.5 IN Services

SMF

SMAF

SCEF

SDF

SRF

SCF

SSF SSF

CCAF CCF CCF CCF CCAF

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The IN services proposed to be introduced in Indian network have been derived from ITU-T recommendations. Q.1211 (April ’92). This document briefly gives the description of 25 services mentioned in Capability set no. 1 (CS1) of above mentioned ITU-T recommendations. CS1 basically deals with single ended services (which ITU-T calls as Type-A services). Single needed services apply to only one party in the call.

(1) ABD – Abbreviated dialing The subscriber can register a short dialing code and use the same for access to any PSTN Number.

(2) ACC – Account Card Calling • A special telephone instrument is required.

• User dials an access code and gets acceptance tone.

• Then he dials a PIN (personal identification no.) code and dials the called no. The Exchange reads the account number from card.

• The Billing is debited to an account number (Telephone no.) as defined by the card.

• In another variation of the service, the account number can be given through DTMF telephone instrument.

• The follow-on feature facilitates the subscriber to dial another number without disconnecting the call and without need to dial PIN and account number again.

(3) AAB – Automatic Alternative Billing • Call can be initiated by any user and any instrument.

• The call charges are billed in user’s account and that account need not be a calling or a called party.

• The user first dials access code.

• Receives an announcement to dial account code and PIN (which is given by management).

• The account code and PIN are validated to check its correctness and expired credit limit.

• On getting acceptance tone the user dials the called number.

• In another variation of the service, the called party may be billed based on his concurrence.

(4) CD – Call Distribution • This service allows subscribers to have I/C calls routed to different

destinations according to allocation law specified by management (The Subscriber has multiple installations).

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• Three types of laws exist :

- Uniform load distribution

- % Load distribution

- Priority list distribution

• In case of congestion or fault the alternative over flow is specified.

(5) CFU – Call Forwarding Unconditional The subscriber can forward all incoming calls to a specified destination number. Optionally an alerting ring/reminder ring can be given to the forwarding subscriber whenever there is an incoming call.

(6) CRD – Call Rerouting Distribution • Calls are rerouted as per conditions encountered, e.g. busy or no reply

(time specified) or overload or call limiter.

• Then as per selected condition the call is rerouted to predefined choice, e.g. paper, vocal box, announcement or queue.

(7) Completion of calls to busy subscriber The service cannot be fully implemented with CSI capability since the status of called party need to be known.

• The calls are completed when subscriber who is busy becomes free.

• On getting busy tone – user dials a code.

• The user disconnects.

• On called party becoming free, call is made by the exchange first to originating then to terminating subscriber (without any call attempt by the user).

(8) CON – Conference Calling The service cannot be fully implemented with CSI capability. In adding or dropping the parties concerned it is not possible to check the authenticity of the parties. This service requires a special transmission bridge to allow conversation among multiple subscribers.

CON-Add-ON-Conference Calling

• User reserves the CON resources in advance indicating date, time of conference and duration.

• Controlled by user.

• In active phase of conference parties can be added, deleted, isolated again reattached or split the group of parties.

• CON-Meet-ME – Conference calling meet me

• User reserve the resource same as 8A.

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• Each participant dials a special number at specific time (specified at the time of booking of conference) and reach the conference bridge.

(9) CCC – Credit Card Calling • The Credit Card Calling service allows subscribers to place calls from any

normal access interface to any destination number and have the cost of these calls charged to account specified by the CCC number.

• A special instrument is not required. The caller has to dial card number and PIN using DTMF instrument.

• Follow-on feature may be provided optionally.

(10) DCR – Destination Call Routing The call is routed to destination pertaining to following conditions :

• Time of day, day of week

• Area of call originating

• Calling identity of customer

• Services attributes (non payment charges against subscriber)

• Priority

• Charge rates applicable for destination

• Proportional routing of traffic

• Optionally the subscribers can be provided with traffic details

(11) FMD – Follow me Diversion • A subscriber can remotely control the call forwarding capabilities.

• It can be done from any point in the network using a password.

• It is required if subscriber moves from place to place in a day.

• The service subscriber will pay for diverted portion of the call.

(12) FPH – Free Phone • The called subscriber is charged for active phase of a call.

• For the calling user, no charging is done.

• The called subscriber can have multiple destinations and have DCR facility.

(13) MCI – Malicious Call Indication • The subscriber requests the Administration to register his number for

MCI.

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• Administration registers the subscriber for MCI.

• The called subscriber (who has registered this service) invokes the service during the active phase of the call if he feels that the call is malicious.

• The call is logged in the network with calling and called party number and Date and time of invoking the service.

• Optionally, the network can log unanswered calls also.

• Optionally, the facility to HOLD the connection may be provided.

(14) MAS – Mass Calling • It involves high volume of traffic.

• Calls can be routed to one or multiple destinations depending on geographical location or time of day.

• Mainly used in Televoting.

• The network operator allots a service number.

• The user dials this number to register his vote.

• The user is played an announcement and asked to give his choice.

• At the end of the service, the network operator provides the call details and the count on various preferences.

• After the service the same number can be reallocated to another subscriber.

• Calls made to this MAS number may be charged differently.

(15) OCS – Originating Call Screening • This helps subscriber to screen outgoing call as per day and time.

• The screening list may be managed by subscriber.

• The restriction of screening list may be override by PIN or password. Three call cases are possible :

- Call screened and allowed

- Call screened and rejected

- Call passed by using override option

(16) PRM – Premium Rate • The local call is charged at a higher (premium) rate.

• This service is used by service providers for value added information services, e.g. jobs, fortune, forecast, etc.

• The revenue is shared between network operator and service provider.

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• The network operator allots a specific number to service provider, which can be reached from any point in the network.

• The provision exists for multiple site provider, in order to achieve minimum expenditure on actual call.

(17) SEC – Security Screening • This capability allows security screening to be performed in the network

before an end user gains access to subscriber’s network, systems or application.

• It detects the invalid access attempts : how many, over what time period, by whom and from where.

• It provides an added layer of security.

(18) SCF – Selected Call Forwarding (Busy/Don’t answer) • This facility is used for a group of 5 to 10 subscribers.

• A list of SCF is prepared by a subscriber.

• The list contains the choices as per conditions and calling subscribers of the group.

• A call from outside the group is forwarded to default telephone number.

• The variation in SCF list can be done as per time of the day.

(19) SPL – Split Charging • It allows service subscriber to share the call charges with calling party on

per call basis.

(20) VOT - Televoting • It is used to survey the public opinion by different agencies.

• The network operator allocates a single telephone number to surveyor.

• Each time user makes a call he can get access to televoting.

• An announcement asks him to input further choice digits as per preference.

• As the user presses the digits the choice counter is incremented.

• After voting is ceased the service subscriber is supplied the results.

(21) TCS – Terminating Call Screening • The incoming calls are screened as per screening list.

• Calls are allowed as per list and time of the day.

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(22) UAN – Universal Access Number • National number is published by the subscriber.

• The subscriber may specify the incoming calls to be routed to number of different destinations based on geographical locations of caller.

(23) UPT – Universal Personal Telecommunications • A universal number is defined.

• Whenever subscriber changes the destination, he inputs that number from telephone.

• When a call comes, UPT number is translated to actual number.

• This number can be accessed across various multiple networks, e.g. mobile and fixed.

• It can be accessed from any user network access.

(24) UDR – User Defined Routing • The user is allowed to define the routing of outgoing calls through

different network such as private, public, virtual or mixed network.

• As per time of the day, for example the call is routed to either public or private network whichever is cheaper.

• For example, outstation calls can have different routes at different times of the day.

(25) VPN – Virtual Private Network • A private network is built using public network resources.

• A virtual PABX is created using different switches.

• A PNP (private numbering plan) can be incorporated on those numbers.

• Facilities such as CT, CH, dialed restrictions and other supplementary services can be provided within the network.

• Each line or user is assigned a class of service and specific rights in the network.

• To access the VPN from outside by one of VPN user, he is required to dial a password.

• Screening feature can be used to put restriction on outgoing and incoming calls.

• Call charges are assigned to VPN service subscriber.

• Additional Account Codes are assigned to service subscriber to analyse the cost line wise.

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8.6 Charging The IN services can be broadly divided into three categories for charging purposes :

- No charging for calling user

- Charging of calling user as per local call

- Charging of calling user at higher rates

No charging for calling user : FPH, VCC and VPN services fall under this category. Level ‘160’ is free at present and is proposed to be allotted to such services. Local exchanges need to analyse only ‘160’ and route the call to SSP. This level has to be created as charge free. New services of this type can be introduced in future without any requirement of further modification in local exchanges

Charging of calling user as per local call : UN (local) falls under this category. Level ‘190’ is free at present and is proposed to be allotted to such services. Local exchanges need to analyse only ‘190’ and route the call to SSP. This level has to be created as local charge. New services of this type can be introduced in future without any requirement of further modification in local exchanges.

Charging of calling user at higher rates : PRM and UN (long distance) falls under this category. Since the charging is at higher rate it is proposed that prefix ‘0’ may be used to have barring facility. Level ‘090’ may be used for such purpose. Local exchange will analyse ‘090’ and route the call to SSP. This level has to be created as ‘charge on junction pulses’. New services of this type can be introduced in future without any requirement of further modification in local exchanges.

The access code of various IN services as proposed is as follows :

No charging for calling user :

FPH 1600

VCC 1601

Password change for VCC 1602

VPN 1603

Charging of calling user as per local call :

UN (local) 1901

Televoting 1902

Charging of calling user at higher rates :

PRM 0900

UN (Long distance) 0901

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