BHARAT SANCHAR NIGAM LIMITED(A Government of India Enterprise)

FUNDAMENTALS OF

ELECTRONIC EXCHANGES

FUNDAMENTALS OF ELECTRONIC EXCHANGES

INDEX

Chapter Page

1. Introduction to Electronic Exchanges 2-11

1. Basic Concept of Telephone Traffic 12-16

2a. Quantitative Indicators For Quality Of Service 17-18

2. Basic Principles of Electronic Exchanges 19-28

3. Digital Switching 29-41

4. Signalling in Telecommunication 42-86

5. Billing Process & CDR Based Billing 87-91

6. ISDN Introduction 92-108

7. Intelligent Network 109-122

8. V5.2 Interface 123-125

9. NGN 126-142

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CHAPTER 1

INTRODUCTION TO ELECTRONIC EXCHANGES

1.0 Introduction

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 Sundsvall, 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.

1.1 The first electronic exchange employing Space-Division switching (Analog switching) was commissioned in 1965 at Succasunna, New Jersey. This exchange used one physical path for one call and, hence, full availability could still not be achieved. Further research resulted in development of Time-Division switching (Digital Switching) which enabled sharing a single path by several calls, thus providing full availability. The first digital exchange was commissioned in 1970 in Brittany, France.

This handout reviews the evolution of the electronic exchanges, lists the chronological developments in this field and briefly describes the facilities provided to subscribers, administration and maintenance personnel.

Table 1 Chronological Development of Electronic Exchanges.

ANALOG

1965 No.1 ESS Local Bell Labs, USA1972 D 10 Local and Transit NEC. Japan.1973 Metaconta Local LMT. France1974 No. 1 ESS Centrex Local and Transit Bell Labs. USA1975 Proteo Local & Transit Proteo, Italy1976 AXE Local PTT & LM Ericsson, Sweden1976 No.4 ESS Transit Bell Labs, USA

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1978 AXE Local LM Eiricsson, Sweden.

Table 2: Development of Electronic Exchanges

MODELAnalog

Capacity (in thousands) Traffic

Lines Trunks Erlangs Call Attempts per second

No. 1 ESS 10-65 - 6,000 30No. 1 ESS 20-128 32 10,000 65NO. 4 A XB ETS - 22.4 6,200 35No. 4 ESS - 107 47,500 150D 10 98 14.3 4,400 30XE 1 - 13 2,500 3.6EWSD 30 - 2,000 11-16EWSP - 13 5,000 -TXE-4 40 - 5,000 50Proteo 30 15 - -AXE 64 - 6,000 35PRX-205 10 - 1,000 10-15Digital Exchange - - - -E-10B 30 4 2,400 25Mentaconta 10-60 - 10,000 28-60MT 20 - 64 20,000 83-110E 12 - 65 15,000 86System X 100 60 25,000 800000AXE -10 64 - 26,000 800000FETEX-150L 290 60 24,000 1800000OCB-283 200 60 25,000 800000EWSD 250 60 25,200 1000000No.5ESS - - - -NEAX-61E 100 60 27,000 1000000

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1.2 ADVANTAGES OF ELECTRONIC EXCHANGE OVER ELECTROMECHANICAL EXCHANGES

Electromechanical Exchanges - Electronic Exchanges

Category, Analysis, Routing, translation, etc;, done by relays.

Any changes in facilities require addition of hardware and/or large amount of wiring change. Flexibility limited.

Testing is done manually externally and is time consuming. No logic analysis carried out.

Partial full-availability, hence blocking.limited facilities to the subscribers.

Slow in speed. Dialing speed is max. 11 Ips and switching speed is in l milliseconds.

Switch room occupies large volume.

Lot of switching noise.

Long installation and testing time.Large maintenance effort and preventive maintenance necessary.

Translation, speech path Sub’s Facilities, etc., managed by MAP and other DATA.

Changes can be carried out by simple commands. A few changes can be made by Subs himself. Hence, highly flexible.

Testing carried out periodically automatically and analysis printed out.

Full availability, hence no blocking. A large number of different types of services possible very easily.Very fast. Dialing speed up to 11 digits /sec possible. Switching is achieved in a few microseconds.Much lesser volume required floor space of switch room reduced to about one-sixth.

Almost noiseless.

Short installation and testing period.Remedial maintenance is very easy due to plug-in type circuit boards. Preventive maintenance not required.

1.3 Influence of Electronics in Exchange Design.

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, Relays, the logic elements in the electromechanical systems, have operate and release times which are roughly equal to the duration of telephone signals to maintain required accuracy. However, to achieve the requisite simultaneous call processing capacity, it became essential for such system to have number of such electrical control units (Called registers in a Cross-bar Exchange), in parallel, each handling one call at a time. In other words, it was necessary to have an individual control system to process each call.

Electronic logic components on the other hand, can operate a thousand or ten thousand times during a telephone signal. This led to a concept of using a single electronic control device to simultaneously process a number of calls on time-sharing basis. Though

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such centralisation of control is definitely more economical it has the disadvantage of making the switching system more vulnerable to total system failure. This can, however be overcome by having a standby control device.

Another major consequence of using electronics in control subsystems of a telephone exchange was to make it technically and economically feasible to realize powerful processing units employing complex sequence of instructions. Part of the control equipment capacity could then be employed for functions other than call processing, viz., exchange operation and maintenance. It resulted in greatly improved system reliability without excessively increasing system cost. This development led to a form of centralized control in which the same processor handled all the functions, i.e., call processing, operation and maintenance functions of the entire exchange.

In the earlier versions of electronic control equipment, the control system was of a very large size, fixed cost unit. It lacked modularity. It was economically competitive for very large capacity exchanges. Initially, small capacity processors were costlier due to high cost per bit of memory and logic gates. Therefore, for small exchanges, processor cost per line was too high. However, with the progressive development of the small size low cost processor using microprocessor, it became possible to employ electronic controls for all capacities. In addition control equipment could also be made modular aiding the future expansion.

The impact of electronics on exchanges is not static and it is still changing as a function of advances in electronic technology.

1.4 Phased Developments

Many electronic switching systems, including the recent ones, had an electromechanical switching network and used miniature electromagnetic relays in junctors and subscriber line equipments None-the-less the trend is towards all electronic equipments for both public and private switching and the switching network has already been made fully electronic with the advent of digital switching.

However, very recently, several countries have developed or specified stored program equipment for upgrading electromechanical exchanges. This typically involves replacing the registers and translators of crossbar exchanges by processor-based facilities. These allow the exchange subscribers to benefit from new services like abbreviated dialing call forwarding automatic alarm call, and detailed billing. They, very significantly, enhance exchange administration and maintenance capabilities for day-to-day operations, such as, modifying a subscriber’s class of service, changing the way traffic is routed, collecting traffic and load data, call charging, etc.

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1.5 Facilities provided by Electronic Exchanges.

Facilities offered by electronic exchanges can be categorised in three arts.(i) Facilities to the Subscribers.(ii) Facilities to the Administration.(iii) Facilities to the Maintenance Personnel.

1. Facilities to the Subscribers.

MFC Push-button Dialing. All subscribers in an electronic exchange can use push-button telephones, which use Dual Tone Multi- frequency, for sending the dialed digits. Sending of eleven digits per second is possible, thus increasing the dialing speed.

Priority Subscriber LinesPriority Subscribers lines may be provided in electronic exchanges. These subscribers are attended to, according to their priority level, by the central processor, even during heavy congestion or emergency.

Toll (Outgoing Call) RestrictionThe facility of toll restriction or blocking of subscriber line for specific types of outgoing traffic, viz., long distance STD calls, can be availed of by all subscribers. This can be easily achieved by keying-in certain service codes.

Service InterceptionIncoming calls to a subscriber can be automatically forwarded during his absence, to a customer service position or a recorded announcement. The customer service position answers the calls and forwards any message meant for the subscriber.

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.

Call ForwardingThe subscriber having the call forwarding facility can keep his telephone in the transfer condition in case he wishes his incoming calls to be transferred to another telephone number during his absence.

Do Not DisturbThis service enables the subscriber to free himself from attending to his incoming calls. In such a case, the incoming calls are routed to an operator position or a talking machine. This position or machine informs the caller that called subscriber is temporarily inaccessible.

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Conference Calls Subscribers can set up connections to more than one subscriber and conduct telephone conferences under the provision of this facility.

Camp On Busy Incoming call to a busy subscriber can be “Camped on” until the called subscriber gets free. This avoids wastage of time in redialing a busy telephone number.

Call Waiting The ‘Call Waiting’ service notifies the already busy subscriber of a third party calling him. He is fed with a special tone during his conversation. It is purely his choice either to ignore the third party or to interrupt the existing connection and have a conversation with the third party while holding the first party on the line.

Call RepetitionInstead of camp on busy a call can automatically be repeated. The calling party can replace his hand set after receiving the busy tone. A Periodic check is carried out on the called party’s status. When idle status is ascertained, the connection is set up and ringing current fed to both the parties.

Third party InquiryThis system permits consultation and the transfer of call to other subscribers. Consultation can be initiated by means of a special signal from the subscriber telephone and by dialing the directory number of the desired subscriber without disconnecting the previous connection.

Priority of calls to Emergency Positions Emergency calls such as ambulance, fire, etc., are processed in priority to other calls.

Subscriber charge Indicator By placing a charge indicator at the subscriber’s premises the charges of each call made can be ascertained by him.

Call Charge printout or immediate BillingThe subscriber can request automatic post call charge notification in the printout form for individual calls or for all calls. The information containing called number, date and time, and the charges can be had on a Tele-type-write.

Malicious Call Identification Malicious Call Identification is done immediately and the information is Obtained in the printout form either automatically or by dialing an identification code.

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Interception or Announcement.In the following conditions, an announcement is automatically conveyed to calling subscribers.1. Change of a particular number of transferred subscriber.2. Dialing of an unallocated cods.3. Dialing of an unobtainable number.4. Route congested or out of order.5. Subscriber’s line temporarily out of order.6. Suspension of service due to non-payment.

Connection Without Dialing.This allows the subscribers to have a specific connection set up, after lifting the handset, Without dialing. If the subscriber wishes to dial another number, then he has to start dialing within a specified time period, say 10 seconds, after lifting the handset.

Automatic Wake Up.Automatic wake up service or morning alarm is possible, without any human intervention.

Hot Line or Private Wire.Hot line service enables the subscriber to talk to a specific subscriber by only lifting the handset. This service cannot be used. along with normal dialing facility. The switching starts as soon as the receiver is lifted.

Denied Incoming CallA Subscriber may desire that no incoming call should come on a particular line. He can ask for such a facility so that he can use the line for making only outgoing calls.

Instrument Locking A few subscribers may like to have their telephone sets locked up against any misuse. Dialing of a secret code will extend such a facility to them.

Free of charge Calls Calls free of charge are possible on certain special services such as booking of complaints, booking of telegrams, etc.

Collect callIf so desired, the incoming subscriber is billed for all the calls made to him, instead of the calling subscriber.

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2. Facilities to the Administration

Reduced Switch Room AccommodationReduction in switch room accommodation to about 1/6th to 1/4th as compared to Cross-bar system is possible.

Faster installation and Easy ExtensionThe reduced volume of equipment, plug-in assemblies for interconnecting cables, printed cards and automatic testing of exchange equipment result in faster installation (about six months for a 10,000 line exchange) Due to modular structure, the expansion is also easier and quicker

Economic Consideration The switching speed being much faster as compared to Cross-bar system, the use of principle of full availability of trunk circuits and other equipment makes the system economically superior to electromechanical systems.

Automatic test of Subscriber lineRoutine testing of subscriber lines for Insulation, capacitance, foreign potential, etc., are automatically carried out during night. The results of the testing can be obtained in the printout form, the next day.

3. Maintenance Facilities

Fault ProcessingAutomatic fault processing facility is available for checking all hardware components and complete internal working of the exchange. Changeover from a faulty sub-system to stand-by sub-system is automatically affected without any human intervention. Only information is given out so that the maintenance staff is able to attend to the faulty sub-system.

DiagnosticsOnce a fault is reported by the system, ‘on demand’ programs are available which help the maintenance staff to localise the fault, who can replace the defective printed card and restore the faulty sub-system. The faulty card is attended at a centralised maintenance centre specifically equipped for this purpose.

Statistical programs Statistical programs are available to gather information about the traffic conditions and trunks occupancy rate to assess and plan the solutions in cases of anticipated problems. This facility helps the maintenance and administration personnel to maintain a specified level of grade of service.

BlockingIn case of congestion or breakdown of a specific route, facility of blocking such routes is available in modes, such as

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(i) Blocking of a specified percentage of calls in such a route either automatically or manually.

(ii) Blocking a specific category of subscribers.

Overloading Security Overloading of central processor in an electronic exchange can lead to disastrous results. To prevent this, central processor occupancy is measured automatically periodically, when it exceeds a specified percentage, audio-visual alarms are activated, in addition to printing out the message. Maintenance personnel have the following options.

(i) Block some of the facilities temporarily, or(ii) Reduce the load by blocking some of the congested routes.

1.6 Constraints of Electronic Exchanges

Though there are a number of definite advantages of Electronic exchanges, over the electromechanical exchanges, there are certain constraints, which should be considered, at the planning stage for deciding between the two systems.

Traffic Handling Capacity

Apparently, the traffic handling capacity of an exchange is limited by the number of subscriber lines and trunks connected to the switching network, and the number of simultaneous paths available through the switching network. However, in electronic exchanges, the prime limitation is the number of simultaneous calls, which can be handled by the control equipment, as it has to execute a number of instructions depending on the type of the call. Therefore the extent of loading of the exchange will be guided solely by the amount of processor loading. Moreover, the facilities to the subscribers will also have to be limited accordingly.

Power Supply

The power supply should be highly stable for trouble free operation as the components are sensitive to variations beyond +10%. It is almost essential to have a stand-by power supply arrangement.

Total Protection from Dust

All possible precautions should be observed for ensuring dust-free environment.

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Temperature and Humidity Control

Due to the presence of quiescent current in the components and because of their compactness, heat generated per unit volume is highest in electronic exchanges. Moreover, as the component characteristics drift substantially with the temperature and humidity, the air-conditioning load is higher. Obviously, the air-conditioning system should be highly reliable and preferably there should be a stand-by arrangement. The installation is also carried out in air-conditioned environment.

Static Electricity and Electromagnetic interference.

Due to the presence of static electricity on the body of persons handling the equipment, the stored data may get vitiated. Handling of PCB’s therefore, should be done with utmost care and should be minimised care should also be taken to protect the cards from exposure to stray electromagnetic fields.

PCB Repair

The repair of PCB’s is extremely complicated and sophisticated equipments are required for diagnosing the faults. This results in having costly inventory and a costly repair centre. With the frequent improvement and changes in the cards, proper documentation of cards becomes essential.

Faster Obsolescence

The changes in the field of electronics are almost revolutionary with the very fast improvements. Hence, the current technology becomes obsolete at a very fast rate. The equipment becomes obsolete before it can possibly complete one third of its life and it might be impossible to get spare parts for the entire currency of the life of the system.

1.7 Conclusion

After 1950, the development in the field of electronic devices induced the telephone system designers to make use of innumerable advantages offered by their inventions. Therefore, telephone switching system with both electronic and electromechanical components was evolved. Later on, Stored Program Control concept was evolved and adapted to the electromechanical exchanges. This developmental step opened a new era of innumerable additional facilities to the subscribers, administration and maintenance personnel.

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CHAPTER 2 BASIC CONCEPT OF TELEPHONE TRAFFIC

2.0 IntroductionTelephone traffic is originated by the individual needs of different subscribers

and so it is beyond the control of telephone administration. Any and every subscriber can originate a call at any and every moment without giving any previous information and the duration of calls is also not previously known. Although the individual telephone traffic originates at random, the average telephone traffic for a particular exchange follows the general pattern of activity in the exchange area. Normally there is a peak in morning, a dip during lunch period followed by a afternoon peak. In some localities the traffic has seasonal characteristic, for example at a holiday resort. A typical 24 hours variations in calling rate is shown below.

2.1 Whatever be the nature of variation of traffic, a telephone engineer is interested in maximum traffic that occurs in an exchange.The hour in which maximum traffic usually occurs in an exchange is known as Busy Hour.Busy Hour Traffic is the average value of maximum traffic in the busy hour. In computing Busy Hour Traffic the seasonal effects are also taken into account. Sometimes it is convenient to refer to Busy hour calling rate (BHCR). Busy hour calling rate is the number of calls originated per subscriber in the busy hour. This provides a simple means for designing the exchange with respect to the number of subscribers. It also provides probable growth of traffic to the estimated growth in number of subscribers. The busy hour calling rate may vary about 0.3 for a small country exchange and 1.5 or more for a busy exchange in business area in a city.

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When the volume of traffic is quoted in terms of number of calls originated in a given time, this is insufficient to determine the consequent occupancy of lines and equipment. Therefore, measurement of traffic should not only consider number of calls but also their duration. The duration during which equipments and circuits are held when a call is made is called HOLDING TIME. Normally, it is average holding time per call for the particular item of equipment that is taken into account, so far as the caller is concerned the useful time is during the conversation only. However, the total time during which equipments and circuits are held when a call is made also includes, the period during which call is being established and time taken to release the equipment after the call has concluded.

2.2 Measurement of Telephone Traffic.

The total cost of providing telephone service can be roughly divided into those charge which are constant and independent of volume of traffic and those, which are determined by the amount of traffic. The cost of subscriber’s line and instrument and certain individual equipment in the exchange is totally independent of the volume of traffic. The quantity of common switching equipment required is almost entirely dependent by volume of traffic. The quantity of such equipment is dependent not only on number of calls but also on duration of calls. Therefore to determine the quantity of switching equipment in automatic exchange or staffing in manual exchange telephone traffic may be measured in terms of both the number of calls and the duration of calls.

For certain purpose it is sufficient to specify a Traffic Volume which is product of number of calls occurred during the time concerned by their average duration. however for the purposes of automatic exchange a more precise unit of traffic flow is required. this is called Traffic Intensity. Traffic intensity is the average number of calls simultaneously in progress. The unit of traffic intensity is Erlang.

A traffic intensity of one erlang is obtained in any specified period when the average number of calls simultaneously in progress during that period in unity. The specified period is always one hour and is taken as being the busy hour unless some other period is indicated.

There is a more precise way to define traffic intensity. The average Traffic Intensity during a specified period T, carried by a group of circuits or equipments, is given by the sum of the holding times divided by T. The holding times and period T all being expressed in the same unit.

Sometime it is stated that the average traffic intensity is equal to the average number of calls, which originate during the average holding time.All the above three definitions give the same numerical result.

The foregoing relationships may be expressed symbolically as follows.Let S be sum of holding times during a given period T , both expressed in hours. Then by definition.

A = S/T

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Where A is the average traffic intensity. Let C be the total number of calls during the period T then the average holding time ‘t’ hours per call, is given by

t=S/C

Then A = S/T Can also be written as

A = Ct/T

It also follows that when the average call duration is known, the average call intensity can be obtained by determining the number of calls occurring during the period T. Also because A is equal to average number of calls simultaneously in progress, an approximate value of A can be obtained by counting the number of occupied circuits or equipments at uniform interval during the time T and finding the average value.

2.3 Grade of service.

Owing to the fact that calls originated in a pure chance manner, it is likely that during the busy hour some calls may fail to mature due to insufficiency of switching equipment. To ensure that the number of calls so lost is reasonably small, it is the standard practice switching equipment such that on the average not more than one call out of every 500 in the busy hour is lost at each switching stage, with the provision that loss does not fall below 1 in 100 with a 10 percent increase of traffic.

This allowable loss is termed the grade of service and is usually represented by the symbol ‘B’ with one lost call in 500 the grade of service is written as

B= 1/500 or B= 0.002The Grade of service is a factor employed for dimensions of the exchange equipment.

A few typical problems are Worked out below to illustrate how the terms and definitions of telephone traffic are actually applied in practice.

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Example 1

If the calling rate per line per day in an exchange of 5000 lines is 6.0 and proportion of the traffic that occurs in the busy hours is 12 percent, what is the busy hours traffic in Erlangs, assuming an average holding time of 2.5 minutes per call?

Calling rate per line per day = 6.0Capacity of the exchange = 50000 linesTotal number of calls made in a day = 5000 x 6 = 30,000Number of calls originated in the hours = 30,000 x 12/100Holding time of a call = 2.5 minutesBusy hour traffic = C x t/60 = 3600 x 2.5/60 = 150 Erlangs or T.u.s.

Example 2

A group of selectors observed for ten busy hours carried an average of twenty Erlangs and the total number of calls lost was twelve. The calls had an average duration of two minutes. What grade of service was given?

Traffic carried by the selectors in one busy hour = 20ErlangsAverage holding time = 2 minutes Total number of calls carried in one busy hour = 20 x 60/2 = 600Number of calls lost in ten busy hours = 12Average number of calls lost in one busy hour = 12/10 = 1.2Total number of calls offered in busy hour = 600 + 1.2 = 601.2Grade of service = Number of calls lost number of calls offered = 1.2/601.2 = 0.001996 Say, = 0.002

2.4 Scanning MethodThis is the practical method for measuring traffic in SPC switches.Here the observation of traffic is not continuous. The group of equipments are scanned at regular intervals and the traffic flow is calculated. s

A=1/S ∑ Fv v=1

or A= I/S [f1+f2+f3+……..+fs]

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where A=Tele traffic intensity in Erlangs S=Number of scans made on the group.Fv=The number of occupied devices found in the vth scan

Example A group of equipments were scanned for ascertaining the traffic flow. The scanning was done once in 5 seconds for one minute. The number of occupied devices in each scan is as follows

1st scan=4,2nd scan=3,3rd scan=24th scan=3,5th scan=1,6th scan=37th scan=2,8th scan=4,9th scan=310th scan=5,11th scan=4,12th scan=2 Calculate the intensity of traffic.

Duration of observation = 60 sFrequency of scanning = 5 sNumber of scans = 12 1A = ── [ f1+f2+f3+…….+f12] S

1 = ──*36 = 3 Erlangs 12

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2A.QUANTITATIVE INDICATORS FOR QUALITY OF SERVICE

The quality of service of a telecommunications network is characterized by the level of satisfaction of the customers connected to it. There are a number of technical and customer services indicators that determine the quality of service. Technical performance indicators encompass reliability (fault rate and time to clear faults), connectivity (dial tone delay and call completion rates) and operator response time for booking calls (manual operations). Specific technical performance indicators are:

(a) fault rate, that is number of faults per main line per year; (b) average number of lines faulty any day as percent (%) of total main lines; (c) percent (%) of faults cleared by next working day; (d) dial tone delay, that is time (in seconds) before dial tone received after call is originated; (e) call completion rates, that is percent (%) of originated calls successfully completed; and (f) time to answer for operator service.

Fault Rate

The number of faults per main line per year defines the frequency of breakdown of the telephone lines. For a well constructed and well maintained network, the average number of faults per main line per year should be 0.2 or less; that is the telephone line should not be out of order more than once in five years. Because the figure is normally small in industrialized countries, this indicator is often expressed in faults per 100 main lines. The actual situation in developing countries is much worse, with the average number of faults in some countries exceeding three faults per main line per year.

The number of lines faulty on any day as percent (%) of total lines in service is an important performance indicator for the company because it actually represents the percentage of the network that is not generating revenues at any particular time. This indicator is closely related to the fault rate and the time to clear .

Fault Clearance

The time to clear faults is normally expressed in terms of the percentage of reported faults cleared within a given time. The significant time frame normally applied is "by next working day".

Call Completion Rate

The Call Completion Rate (CCR) measures the percentage of originated calls successfully completed. The CCR, which is normally measured during the peak traffic hour, is an indication of the probability of establishing a connection at the end of dialing. In practice, dialing can commence only after the dial tone is received; hence, connectivity also depends on the availability of a dial tone, the ability of the network to establish a transmission path between the calling and the called party and to switch the call to the called party. The network components involved for a local call are:

(a) the customer premises equipment (terminal equipment such as a telephone and indoor wiring);

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(b) the local cable network; and (c) the local switching equipment.

For domestic long distance calls, in addition to the above equipment, long-distance switching equipment and transmission media and equipment are required while for international calls, international switching equipment and transmission media and equipment are required. Hence, the CCR for the international calls depends on the quality of the total network - local, domestic long-distance and international.

A successful call could be defined in two ways. First, the call could be considered as successfully completed only if the called party answers and communication (voice, data, fax, etc.) is established. Another interpretation of a successful call could be establishing a connection successfully to the called number although the called party may not answer. In respect of telephone calls, the called party may not answer because of a number of reasons including:

(a) called party is not available near the phone and hence the phone keeps on ringing without an answer. In the age of answering machines, the probability of not receiving an answer is low; and (b) called line is busy and therefore the telephone at the called number does not actually ring. The probability of this happening is also being reduced through use of "Call Waiting" facility by many users.

The CCR reflects directly the degree of congestion in the network and indirectly the fault rate. The CCR depends on the equipment available to switch and transmit the signaling messages. The equipment may not be available either because of under dimensioning in which case the available equipment is not adequate to handle the traffic, or faulty equipment which would cause the same effect. In many developing countries, the poor CCR is mainly due to faulty switching equipment; however, because of poor maintenance, the outside plant network could also contribute to the poor CCR.

In the international network, the CCR has been further categorized into:

(a) Answer Bid Ratio (ABR); (b) Answer Seizure Ratio (ASR); and (c) Congestion (CONG).

The ABR is the ratio of successful calls to total originating international calls. The ratio is the measure of effective international calls, reflects the performance of the total international network between the calling and called country and hence is the CCR for the entire international network or the probability of a call being successful. The ASR is the ratio of successful calls to total incoming international calls. It is a measure of the performance of the called country's telephone network and hence reflects its CCR. The CONG is the percentage of calls lost due to congestion in the international network. It is a measure of the inadequacy in the number of international circuits between the two countries.

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CHAPTER 3BASIC PRINCIPLES OF ELECTRONIC EXCHANGES

3.0 Introduction

The prime purpose of an exchange is to provide a temporary path for simultaneous. bi- directional transmission of speech between

(i) Subscriber lines connected to same exchange (local switching) (ii) Subscriber lines and trunks to other exchange(outgoing trunk call)(iii) Subscriber lines and trunks from other exchanges(incoming trunk calls) and(iv) Pairs of trunks towards different exchanges (transit switching)

These are also called the switching functions of an exchange and are implemented through the equipment called the switching network. An exchange, which can setup just the first three types of connections., is called a Subscriber or Local Exchange. If an exchange can setup only the fourth type of connections, it is called a Transit or Tandem Exchange. The other distinguished functions of an exchange are

i) Exchange of information with the external environment (Subscriber lines or other exchanges) i.e. signaling.

ii) Processing the signaling information and controlling the operation of signaling network, i.e. control, and

iii) Charging and billing

All these functions can be provided more efficiently using computer controlled electronic exchange, than by the conventional electromechanical exchanges.

This handout describes the basic principals of SPC exchanges and explains how the exchange functions are achieved.

3.1 Stored Programme Controlled Exchange:

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

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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 cased, 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 performed from a single centralised 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 centralised 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 decentralising the

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control. Some routine functions, such as scanning, signal distributing, marking, which are independent of call processing, can be delegated to auxiliary or peripheral processors. These peripheral units, each with specialised function, are often themselves controlled by a small stored programmes processors, thus reducing the size and complexity at central control level. Since, they have to handle only one function, their programmes are less voluminous and far less subjected to change than those at central. Therefore, the associated programme memory need not be modifiable (generally, semiconductors ROM's are used).

3.2 Block Schematic of SPC Exchange

Despite the many difference between the electronic switching systems, and all over the world there is a general similarity between most of the systems in terms of their functional subdivisions. In it’s simplest from. an SPC exchange consists of five main sub-systems, as shown in fig.

i. Terminal equipment, provides on individual basis for each subscriber line and for interexchange trunk.

ii. Switching network, may be space- division or time-division, uni-directional or bi-directional.

iii. Switching processor, consisting mainly of processors and memories.

iv. Switching peripherals ( Scanner, Distributor and Marker ), are Interface Circuits between control system terminal equipment and switching network.

v. Signaling interfaces depending on type of signaling used, and

vi. Data Processing Peripherals ( Tele - typewriters, Printers, etc. ) for man- machine dialogue for operation and maintenance of the exchange.

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(6)Man

MachineDialogue

peripherals

Fig. FUNCTIONAL SUBDIVISIONS OF AN SPC EXCHANGE 1.Terminal Equipment.

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(3)Central control

CCMemories

(1)

(2)

Switching

network

Common channel

Signaling links

Common channel

signaling terminal

(5)

Channel associated signaling terminal

(5)

(4)Distributor

(4)Scanner

(4)Marker

Terminal Equipment Line & Trunks

(6)Man

Machine dialogue

peripherals

SP

In this equipment, line, trunk, and service circuits are terminated, for detection, signaling, speech transmission, and supervision of calls. The Line Circuits carry out the traditional functions of supervising and providing battery feed to each subscriber line. The Trunk Circuits are used on outgoing, incoming and transit calls for battery feed and supervision. Service Circuits perform specific functions, like, transmission and reception of decadic dial pulses or MF signals, which may be economically handled by a specialised common pool of circuits. In contrast to electromechanical circuits, the Trunk and Service circuits in SPC exchanges, are considerably simpler because functions, like counting, pulsing, timing charging, etc. are delegated to stored programme.

2. 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 ,i. Switching, i.e., setting up temporary connection between two or more exchange

terminations, and ii. 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.

2.1 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, viz., reed relay, mini-cross bar derivative switches, etc.were favored. They have the advantage of compatibility with the existing line and trunk signaling conditions in the network.

2.2 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 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

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limitations of analog and PAM signals. PCM signals are suitable for both transmission and switching. The PCM switching is popularly called Digital Switching.

Compatibility with Existing Network

In this area, the application of electronic techniques has encountered the greatest difficulty. To appreciate the reasons, let us consider the basic requirements of a conventional switching network.

High OFF resistance and low ON resistance. Sufficient power handling capacity for transmitting ringing

current, battery feed etc..., on subscriber lines. Good frequency response (300-3400 Khz ) Bi-directional path (preferable) D.C. signaling path to work with existing junction equipment (preferable) Economy Easy to control. Low power consumption, and Immunity to extraneous noise, voltage surges.

The present day electronic devices cannot meet all these requirements adequately. It is seen that requirement iii,v, vi and vii only, can easily be met by electronic devices. These considerations show that substitutions of the analog mode of electromechanical switching network by fully electronic equipment is not, straight way practical. The main virtue of the existing electromechanical devices is their immunity to extraneous noise voltage surge, etc., which are frequently experienced in a telephone network. Moreover, metal contact switches offer little restriction on the voltages and currents to be carried. In the existing network and subscriber handsets, typically, 80 volt peak to peak ringing current is required to be transmitted on the line. This is difficult, if not impractical, for electronic switches to handle. Therefore, to avail of the advantages of the electronic exchanges, either of the two following alternatives may be adopted.i. Deploy a new range of peripherals/ equipments, suited to the

characteristics of the electronic switching devices, on one hand, and requirements of telephone network on the other hand. i.e. employ Time Division Switching systems, or

ii. Continue to use metal contact switches, while other sub-systems may be changed to electronic. i.e. semi-electronic type of exchanges rather than fully electronic exchanges, to employ Space Division Switching Systems.

3. Switching Processor

The switching processor is a special purpose real time computer, designed and optimised for dedicated applications of processing telephone calls. It has to perform certain real time functions (which have to be performed at the time of occurrence and cannot be deferred), such as, reception of dialed digits, and sending of digits in case of transit exchange. The block schematic of a switching processor, consisting of central control programme store is shown in fig.2.

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Fig.2 Switching Processor

Central Control (CC) is a high speed data processing unit, which controls the operation of the switching network. In Programme store, sets of instructions. called programmes, are stored. The programmes are interpreted and executed by the central control. Data Store provides for the temporary storage of transient data, required in processing telephone calls, such as digits dialed by the subscriber, busy / idle states of lines and trunks etc. Translation Store contains information regarding lines. e.g. category of calling and called line. routing code, charging information, etc. Data Stores is temporary memory, whereas Translation and Programme Stores are of semi-permanent type. The information in the Semi-permanent memories does not change during the processing of the call, but the information in Data Store changes continuously with origination and termination of each call.

4 Switching Peripheral Equipment

The time intervals, in which the processor operates, is in the order of microseconds, while the components in the telephone switching section operate in milliseconds ( if the switching network is of the analog type). The equipment, known as the switching peripheral, is the interface between these two equipments working at different speeds. The interface equipment acts as speed buffer, as well as, enables conversion of digital logic signals from the processor to the appropriate electrical signals to operate relays and cross-points, etc. Scanners, Signal distributors and Marker fall under this category of devices.

4.1 ScannerIts purpose is to detect and inform CC of all significant events / signals on subscriber lines and trunks. connected to the exchange. These signals may either be continuous or discrete. The equipments at which the events / signals must be detected are equally diverse.

i. Terminal equipment for subscriber lines and inter-exchange trunks and. ii. Common equipment such as DTMF (Dual - Tone Multi Frequency) or MFC digit

receivers and inter-exchange signaling senders / receivers connected to the lines and trunks.

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Central control Processor

ProgrammeStore

Translation Store

Data Store

To Switching Network

In view of this wide diversity in the types of lines. trunks and signaling, the scanning rate, i.e. the frequency at which scan points are read, depends upon the maximum rate at which events / signals may occur. For example, on a subscriber line, with decadic pules signaling with 1:2 make -break ratio, the necessary precision, required for pulse detection, is of the order of ten milliseconds, while other continuos signals ( clear, off hook, etc.) on the same line are usually several hundred mili-seconds long and the same high precision is not required. To detect new calls, while complying with the dial tone connection specifications, each line must be scanned about every 300 milliseconds. It means that in a 40,000 lines exchange ( normal size electronic exchange ) 5000 orders are to be issued every 300 milliseconds, assuming that eight lines are scanned simultaneously.

4.2 MarkerMarker performs physical setup and release of paths through the switching network, under the control of CC. A path is physically operated only when it has been reserved in the central control memory. Similarly, paths are physically released before being cleared in memory, to keep the memory information updated vis-a-vis switching network, Depending upon whether is switching is Time division or Space division, marker either writes information in the control memory of time and space stages. (Time Division Switching), or physical operates the cross - points (Space Division Switching)

4.3 DistributorIt is a buffer between high - speed - low - power CC and relatively slow-speed-high-power signaling terminal circuits. A signal distributor operates or releases electrically latching relays in trunks and service circuits, under the direction of central control.

4.4 Bus SystemVarious switching peripherals are connected to the central processor by means of a common system. A bus is a group of wires on which data and commands pulses are transmitted between the various sub- units of a switching processor or between switching processor and switching peripherals. The device to be activated is addressed by sending its address on the address bus. The common bus system avoids the costly mesh type of interconnection among various devices.

4.5 Line Interface CircuitsTo enable an electronic exchange to function with the existing outdoor telephone network, certain interfaces are required between the network and the electronic exchange.

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4.5.1 Analogue Subscriber Line InterfaceThe functions of a Subscriber Line Interface, for each two wire line, are often known by the acronym : BORSHT

B : Battery feedO : Overload protectionR : RingingS : Supervision of loop statusH : HybridT : Connection to test equipment

All these functions cannot be performed directly by the electronic circuits and, therefore, suitable interfaces are required.

4.5.2 Transmission InterfaceTransmission interface between analogue trunks and digital trunks (individual or multiplexed) such as, A/D and D/A converters, are known as CODEC, These may be provided on a per-line and per-trunk basis or on the basis of one per 30 speech channels.

4.5.3 Signaling Interfaces

A typical telephone network may have various exchange systems (Manual,Strowger, Cross bar, electronic) each having different signaling schemes. In such an environment, an exchange must accommodate several different signaling codes.

SignalingInitially, all signaling between automatic exchanges was decadic i.e. telephone numbers were transmitted as trains of 1to 10 pulses, each train representing one digit. To increase the speed at which the calls could be set up, and to improve the reliability of signaling, compelled sequence multi frequency signaling system was then introduced. In this system, each signal is transmitted as a combination of 2 out of a group of say 5 or 6 frequencies. In both decadic and multi frequency methods, the signals for each call are sent over a channel directly associated with the inter-exchange speech transmission circuit used for that call. This is termed as channel associated signaling. Recently, a different technique has been developed, known as common channel signaling. In this technique, all the signaling information for a number of calls is sent over a signaling link independent of the inter-exchange speech circuits. Higher transmission rate can be utilised to enable exchange of much larger amount of information. This results in faster call setup, introduction of new services, e.g.., abbreviated dialing, and more retrials ultimately accomplishing higher call completion rate, Moreover, it can provided an efficient means of collecting information and transmitting orders for network management and traffic engineering.

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4.5.4 Data Processing Peripherals.Following basic categories of Data Processing Peripherals are used in operation and maintenance of exchange.i. Man - machine dialogue terminals, like Tele-typewriter (TTY) and

Visual Display Units (VDU), are used to enter operator commands and to give out low-volume date concerning the operation of the switching system. These terminals may be local i.e. within a few tense of meters of the exchange, or remotely located. These peripherals have been adopted in the switching Systems for their ease and flexibility of operation.

ii. Special purpose peripheral equipment is, sometimes employed for carrying out repeated functions, such as, subscriber line testing, where speed is more important than flexibility.

iii. High speed large capacity data storage peripherals (Magnetic Tape Drives, magnetic Disc Unit) are used for loading software in the processor memory.

iv. Maintenance peripherals, such as, Alarm Annunciators and Special Consoles, are used primarily to indicate that automatic maintenance procedure have failed and manual attention is necessary.

3.3 Conclusion

The electronic exchanges work on the principle of Stored Programme Control. All the call processing functions are performed on the basis ofpre-designed programme which is stored in the memory of the Central Processor. Though the initially designed Electronic Exchanges had single centralised processor. the control is being decentralised, providing dedicated micro - processor controlled sub- systems for improved efficiency and security of the system. This modular architecture also aids future expansions.

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CHAPTER 4DIGITAL SWITCHING

4.0 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.

4.1 Time and Space Switching

Generally, a digital switching system several time divisionmultiplexed (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.

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FIG 1 DIGITAL SWITCH

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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 Switchingii. 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.

4.2 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 .

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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

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to the address, is operated 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 carried on

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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.

4.3 Digital Time Switch

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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.

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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.

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

4.3.1 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.

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FIG 4 OUTPUT ASSOCIATED CONTROL SWITCH

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.

4.3.2 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.

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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

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4.4 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.

4.5 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.

4.6 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.

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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.

4.7 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

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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 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 HWY0Step 2 Space stage (S)Tsx HWY0 to Tsx HWY3Step 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

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designed as output-controlled switch, whereas the second time stage is input-controlled.

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.

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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 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.

4.8 Switching Network Configuration of some Modern Switches

1. E10B - T-S-T

2. EWSD - T-S-S-S-T

3. AXE10 - T-S-T

4. CDOT(MBM) - T-S-T

5. 5ESS - T-S-T

6. OCB 283 - T

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CHAPTER 5SIGNALLING IN TELECOMMUNICATION

5.1 IntroductionA 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)

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b) Busy tone if the called subscriber is busy ( not shown in the figure), orc)Recorded message, if provision exists, for non completion of call due to some other constraint ( not shown in figure).

vi. The called subscriber indicates acceptance of the incoming call by lifting the handsetvii. 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.

5.2 Types of Signalling

Subscriber Line signalling

5.2.1 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.

5.2.1.1 Call requestWhen 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.

5.2.1.2 Address signalAfter 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 DiallingThe 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 distinguish between consecutive digits. This method is, therefore, relatively slow and signals cannot be transmitted during the speech phase.

2. Multifrequency Push-button Dialling

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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.

5.2.1.3 End of selection signal

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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.

5.2.1.4 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.

5.2.1.5 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.

5.2.1.6 Permanent Line (PG) SignalPermanent 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.

5.2.2 Called subscriber line signals.

5.2.2.1 Ring SignalOn 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.

5.2.2.2 Answer SignalWhen 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.

5.2.2.3 Release Signal

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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.

5.2.3 Register Recall SignalWith 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.

5.3 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. PulsedThe 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. ContinuousThe signal consists of transition from one condition to another, a steady state condition does not characterizes any signal.

iii. CompelledIt 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.

5.3.1 Line signals

5.3.1.1 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.,

i. Circuit seizure/release corresponding to off/on-hook signal of the subscriber.ii. Address information in the from of decade pulses.

5.3.1.2 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

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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 BackwardIdle (On hook) FORWARDSeizure(off hook)Release (on hook)BACKWARDAnswer(off hook)Clear Back (on hook)Blocking (off hook)

On

offon

offoffon

On

onoff/on

offonoff

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 SignalsE & 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 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 ExchangeM- lead E-lead

Incoming ExchangeM- lead Elead

Idle(On hook)

Earth Open Earth Open

FORWARDseizure(off hook)

Battery Open Earth Earth

Release(On hook)

Earth Earth/open Battery/Earth Open

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BACKWARDAnswer(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.

5.3.2 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.

Fig.3. Compelled signalling procedure

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outgoing register

incomming register

forwardsignal

time

signal cessation recognition

next forward signal

2-and-2onlysignal recognition

acknowledgement backwardsignal and request for next signal

compelled signal sequence

acknowledgement backward signal

receiving

time

signal cessationrecognition

Sending

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 signalingThe 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 signalingThe 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.

5.3.3 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

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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

TABLE 4-FORWARD SIGNALSSignal Weight Group I Group II

1 0+1 Digit 1 Ordinary subscriber2 0+2 Digit2 Subscriber with

priority Test / Mtce, equipment

3 1+2 Digit3 Spare4 0+4 Digit4 STD Barred5 1+4 Digit5 Spare6 2+4 Digit6 CCB7 0+7 Digit7 Changed Number to

Operator8 1+7 Digit8 Closed Number9 2+7 Digit9 Closed Number10 4+7 Digit0 Spare

TABLE 5 -BACKWARD SIGNALSSignal 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 signalsCalled line busy

4 0+4 Calling line identification for malicious calls Local congestion5 1+4 send calling subscribers category Number unobtainable

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6 2+4 Set up speech connection called line fee, with metering7 0+7 Send last but 1 digit Route congestion8 1+7 Send last but 2 digit Spare9 2+7 Send last but 3 digit Route Breakdown10 4+7 Spare Malicious call blockingNote : Signals A2, and A7 to A9 are used in Tandem working only.

It can be seen from the tables that1. 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 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.

5.3.4 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

exchange4. 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.

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5.4 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 common channel signalling, the exchange is connected to only a limited number of signalling links through a special terminal.

5.4.1 Channel- Associated signallingIn 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 bbIdle 1 0 1 0

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Seizure 0 0 1 0Seizure acknowledge 0 0 1 1Answer 0 0 0 1Clear Forward 1 0 0/1 1Clear Back 0 0 1 1

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

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- cable (copper, optical fiber)

- 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 link

2.2. Signalling Points (SP)

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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 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.

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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.

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.

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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.

(b) Signalling data links, e.g. 64 kbit/s digital or 4.8 kbit/s analog

(c) Safety requirements

- load sharing between signalling links

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- 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 .

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Fig. 4 Functional Levels of CCS7

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

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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.

- 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.

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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 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.

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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.

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

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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.

5.10Fields 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).

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Fig. 7 Routing Label of a Message Signal Unit

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).

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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 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

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Fig. 8Distribution of Functions in Message Transfer Part

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.

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

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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.

- 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.

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- 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

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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.

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.

9. Signaling Connection Control Part

9.1 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.

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Enhanced addressing capabilities.

The following figure illustrates the SCCP position in the SS#7 hierarchy:

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.

9.2 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).

9.3 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

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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

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.

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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 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).

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SCCP Functional Units

10. 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.

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.

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.

ASE1

SCCP-A

MTP-A MTP-B

SCCP-B

TCAP-B

ASE2

TCAP-A

Component

TCAP messages

MSUs

TC-primitives TC primitives

N-primitiveN-primitives

MTP-primitives MTP-primitives

Messages and message paths

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SSTPCommon Channel Signaling NetworksSignaling System No. 7 (SS7) is a signaling protocol that has become a worldwide standard for modern telecommunications networks. SS7 is a layered protocol following the OSI reference model .It enables network elements to share more than just basic call-control information through the many services provided by the SS7's Integrated Services Digital Network-User Part (ISUP), and the Transaction Capabilities Application Part (TCAP).The functions of the TCAP and ISUP layers correspond to the Application Layer of the OSI reference model, and allow for new services such as User-to-User signaling, Closed-User Group, Calling Line Identification, various options on Call Forwarding and the rendering of services based on a centralized database (e.g., 800 and 900 service). All of these services may be offered between any two network subscribers.

CCS Network Architecture

The CCS Network is comprised of Four Major Components;

Service Switching Points [SSP]

Signaling Transfer Points [STP]

Service Control Points [SCP]

Data Signaling Links (SLK)

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SSP

STP

SCP AINSCP

SLK

Service Switching Point (SSP)

SSP is typically found in tandem or Class 5 offices and is the interface to the networks outside of SS7.

A SSP can be any of the following: Customer switch End office Access tandem Tandem

Usually, a switch is used to interface to the customer premise, The CO switch then interfaces to the SS7 network via the SSP

The SSP is the interface between the subscriber and the telecom network, and provide the following functions:

Call Processing function provides dial tone routes calls between links and trunks provides tones, and announcements maintenance and revenue collection and generation Query Processing When necessary, generates queries toward another signaling node or database to receive

information necessary for certain calls. SS7 Response Processing Upon receiving queried information, carries out the connection function for proper handling

of calls. Resource Interface For AIN services, establishes and maintains connections to

The SSPs are the legacy switches of the telecommunications network. SSPs are referred to as an “End Office switch”, “Central Office switch”, “Toll Tandem switch”, etc. The central offices that house the SSP are identified by classes of ranging from a class 5-lowet, to a class 1 – highest office. The lowest class office in a network will be the one providing dial tone to subscribers.

SSP

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Intelligent Peripherals (IPs)Service Control Point (SCP)

The SCPs and AIN SCPs are centralized database that provide real-time access to call completion and

information services such as: Toll-Free (800/888) Database Service Alternate Billing Service (ABS) Custom Local Area Signaling Services (CLASS) Advanced Intelligent Network Services (AIN) Local Number Portability (LNP) Home Location Register (HLR) Visitor Location Register (VLR)

Common SCP TypesCall Management Services Database (CMSDB) – Call processing information

Routing instructions for 800, 888, 900 special service numbers Billing information such as billing address and third party billing

Line Information Database (LIDB) – Alternate Billing Services (ABS) are performed such as: Calling card validation – PIN information Third party billing Collect call handling instructions Originating line number screening – Call forwarding and Speed dialing

Custom Local Area Signaling Services (CLASS) Calling name delivery (CNAM) Caller ID with calls waiting Customer originated trace Automatic callback / recall Selective call acceptance / forwarding / rejection

Toll-Free AINABS LNPCLASS HLR

VLR

SSP STP SCPAINSCP

or

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Local Number Portability Database (LNP) Stores subscriber ported number information Stores the Location Routing Number (LRN) of Servicing Carriers Places LRN information in the called party parameter of LNP query responses.

Home Location Register (HLR) – Used in wireless networks and used to store wireless subscriber information such as:

- Billing information Services allowed Current location information for retrieval by Mobile Switching Centers

Visitor Location Register (VLR) When a wireless telephone is not recognized by the local Mobile Switching Center (MSC)

the MSC originates a query into the network requesting validation information from the subscriber’s HLR

When information is received from the subscriber’s HLR, it is stored in the VLR where the subscriber is currently located

Signaling Transfer Point (STP)

STPs are routers that are placed within the heart of the CCS Networks. STPs are packet switches that provide common channel message routing and transport. STPs are stored programmed control switches that use information contained in messages in

conjunction with information stored in memory to route message to the appropriate destination signaling point.

STP Deployment

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SSP

STP

SCP

SCP

STPs are generally deployed in pairs with mirrored databases. If one of the STPs are removed from service or signaling links fail, the mate can process all of the traffic that is typically shared by the mated pair.

STP mated pairs are geographically separated , This helps ensure protection for message routing they perform if a natural disaster occur, etc.

STP two-level Architecture in CCS Network

In large CCS networks, STPs are deployed in a hierarchical arrangement, and typically identified as Regional STPs, and Local STPs.

There are no functional differences in the two STPs.

The LSTP handles call set-up and network management traffic within the network.

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SSP STP SSP

STP

SSP

LSTP

SSP LSTP

SSP RSTP

SSP RSTP

SCP

The RSTP only handles query traffic within the network requiring access to SCP databases.

STP Functions SS7 Message routing Global Title Translation SS7 Network Management Network Interconnection Gateway Screening

STP Function – Message Routing

Message Routing: By using outgoing DPC contained in MTP’s routing label in a datagram environment (where a separate route may be chosen for each message packet) Routing tables which are prepared to allow message transport between any given pair of STPs are stored and maintained within STPs. The STP’s SNM (signaling network management) functions control message routing during periods of link congestion or failure.

Routing is performed using Destination Point Codes (DPCs) similar to street address for the Postal Service. STPs have the ability to route messages to all types of signaling points.

All nodes in the network are identified by a unique point code. This point code is used by CCSS #7 as the Origination Point Code (OPC) and the Destination Point Code (DPC) in the routing label of all Message Signaling Units (MSUs).

STP Function – Global Title Translation

Global Title translation : By using SCCP to translate addresses (Global titles) from signaling messages that do not contain explicit information allowing the MTP to route the message. For (e.g. STP translates dialed 1+ 800 number into an SCP’s DPC for MTP routing and gives sub system number SSN for delivery of the good data base application at the SCP

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When more information is needed to process a call, such as an 800 number, queries are processed for SSPs. STPs contain a GTT table with routing information for the type of query and address of SCP.

STP Function – network ManagementActs as traffic cop to route traffic around failures in a network, and to control link congestion.

TFP tells the connecting nodes not to send anything that is destined for the affected node.TFR tells the connecting nodes – if all possible, not to send anything that is destined for the affected node.

STP function – Gateway Screening

Screening is the capability to examine Incoming and Outgoing packets and allow those which are authorized. This is done by going through a series of Gateway screening tables that must be configured by the service provider. For example out of the messages which are coming via a link set only ISUP messages can be allowed whereas on another link only SCCP messages can be allowed by utilizing two basic function allow and block..

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1-1-1

2-2-2 4-4-4254

STP

STP

STP

SSP

STP

SCP

TFR

TFRTFR

Software in STPs with inter-network connection is used to control who has access into a Telco’s network.

Why SSTP is required in BSNL

• When any mobile subscriber roams to other Service Area then the signalling traffic or SMS are being handled by the signalling channel of BSNL taken by private operator against POI. BSNL is not able to measure the traffic and kind of traffic. Now by putting SSTP in system we can measure the traffic and bill to private operators.

• It will enable migration from TDM based network to IP based networks. Traditional Mesh Network Analysis

SSP

MSC

SSP

MSC

MSC

MSC

SS7 associated / mesh signaling

HLRHLRHLRHLR

HLRHLRHLR

HLR

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Advantages:

• Cost effective at start-up• Signaling is added as bearer capacity added• STP function not mandatory

Disadvantages:• Difficult to “grow”• Traffic capacity not optimized

– Fixed relationship between sites– Fixed relationship between bearer channels and signaling links (30:1)

• No STP functionality• Verification on signaling traffic not possible• Difficult to add network-level services, features

Integrated STP Network Analysis

• Advantages:– Cost effective at start-up– Uses SSP processor, call control

• Disadvantages:– Difficult to “grow”– Trade-off between SSP functions and STP functions– Upgrades driven by SSP, not STP– Traffic capacity not optimized– Difficult to add network-level services, features

SSP

SSP

SSP

SSP

SSP

SSP

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Stand-alone STP Network Analysis

–––––

•• Advantages:• Dedicated signaling processors, resources• Upgrade path divorced from MSC / SSP functions, growth• Most effective method to manage network level resources, features• Frees up processing capacity from the switches• Can host most of the applications, centrally

• Full mated pair redundancy• 0.4 Erlang normal load

Disadvantages:• Requires additional investment (However compensated by freeing up extra resources of the

switches)• Requires traffic study, SS7 management.

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SSP SSP SSP SSPSSP

MSCMSCMSC

MSCHLRHLR VMSVMSSMSCSMSC

CHAPTER 6THE BILLING PROCESS & CDR-BASED BILLING

In a digital exchange during the course of performing the

switching functions, a number of events are significant from the billing or charging point of

view. These events include the dialed digits, the moment the customer answers and the

moment of disconnection. The first step in the billing process is the recognition of these

events and recording of data. This data collection is done by the call processing software

and a Call Detail Record (CDR) is prepared .A CDR is a data record that contains

information related to a telephone call such as the origination and destination addresses of

the call, the exact time the call started and ended, the duration of the call, the time of the day

the call was made and charges for operator services among other details of the call. The

CDRs can be used for billing and administrative purposes. By compiling CDRs, it is also

possible to keep track of successful and failed call events.

6.1 CONVENTIONAL METHOD OF BILLINGPULSE_BASED BILLING

In the conventional method of billing, the charging for a call is done at the originating local

exchange. This is known as local automatic message accounting. Each subscriber line has

an individual charge meter defined in the exchange memory to accumulate the charges

payable by the subscriber. The charge for a call is computed based on metering pulses

(Periodic pulses). The metering pulse rate (the interval between successive incrementing of

subscriber’s meter) depends not only on the distance between calling and called party but

also on other parameters such as the time of the day and type of the day (Normal Working

day or Holiday) etc which are predefined. The pulses for metering may be locally generated

or may come from the leading TAX exchange

.The meter reading contents of the subscribers or the CDRs present in the buffer of the

switch are periodically copied on to a portable secondary storage device such as a magnetic

tape or cartridge and are then manually transported to the Telecom Revenue Accounting

Centre. A copy of the magnetic tape or cartridge is preserved in the exchange for future

verification. At TRA billing centre, these tapes are processed for billing. The billing

computer calculates the bills for individual lines based on difference between the current

and previous meter readings. For STD/ISD calls made by the subscribers, detailed bills or

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itemized bills are also generated which contain details about the call such as a) Number

dialled by the subscriber b) Date and time of call c) Chargeable duration of call d) Number

of Chargeable units etc.

6.2 CDR : CDR is a text record of call related data. The CDR’s are collected in files so

that they can be uploaded to a CDR Buffer. CDR files are continually updated to a

centralized billing and accounting server to prevent file-overwrites and disk capacity

problems.

A typical CDR may contain the following fields:

Time : The date and time of call origination or disconnection

Qualifier : Qualifies the type of event. There are 4 qualifiers

Call Request

Call Disconnect

Setup Fail : An incoming call was denied or failed

Disc Fail : A disconnect request was denied or failed

Calling number

Called number

Incoming circuit or Trunk identifier

The bearer channel Timeslot identifier

For eg: 1 through 31 for E1

A description of the cause for call disconnect

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All incoming call requests are recorded, time-stamped and identified by the call request

qualifier to help trace network events triggered by call request. Call failures may occur

during call setup or tear-down and the failures will be recorded in CDR files which will

include all available information identifying the call as well as failure codes. Some

examples of failure codes in mnemonics are:

1) Normal call clearing

2) No user response

3) Call rejected etc.

Call detail records, both local and long distance, can be used for usage verification, billing

reconciliation, network management and to monitor telephone usage to determine volume of

the phone usage as well as misuse of the company’s telephone system.CDR analysis gives

the following advantages:

Review all CDRs for accuracy

Verify costs and usage

Resolves discrepancies with vendors

Disconnect unused service

Terminate leases on unused equipments

Deter or detect toll fraud of long distant services

Negotiate the most cost-effective call routing

6.3 CDR based billing

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Regardless of their size, most telephone exchanges output CDRs. Generally, these get

created at the end of a call but on some phone systems the data is available during the call.

This data is output from the phone system through a serial link to a CDR buffer where they

are temporarily stored until retrieved by a call accounting software. Since they provide a

reliable method of safely transferring information to a centralized call accounting or Tele-

management system, call record buffers have long been broadly accepted as the preferred

storage device as a safe-guard against cases of delayed call collection or communication

failure. A CDR buffer would be placed at each exchange for collection of call data. A PC

with sufficient memory and installed with a suitable software may serve as the CDR buffer.

The software has the capability of scheduling the CDR downloading without manual

intervention. The CDRs will be sent to the centralized billing centre over LAN/WAN

arrangements or over dial-up circuits.

The centralized billing and accounting centre to which all CDR buffers will be connected is

a powerful, real-time, PC based system capable of processing call records to the tune of tens

of thousands per second and generating reports. CDRs are immediately available for

viewing and reporting – allowing users to monitor and address business, legal and security

issues those need immediate attention such as emergency calls, internal phone abuse

(sexual harassment, bomb threats etc.) , potential toll fraud and others.

The CDR based call accounting & billing system will be fully web-enabled and any

authorized user can access the centralized billing system over the company intranet and run

reports right from their desktops using the web browser. The centralized call accounting

application can be administered for any number of sites, from one single location.

Regardless of number of sites and number of stations, data for multiple sites is maintained

in a single database.

The CDR buffers at the exchanges connect to a TCP/IP Ethernet network and send

data continuously over LAN/WAN to the centralized server. The CDR retrieval for

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LAN | WAN or dialup CCT

RS232 Interface

CDR Buffer BufferBuffer

CentralisedBilling &

AcccountingSystem

Exchange

TCP/IP Interface

all locations would occur in real time and provide users with instant access to all

data.

Implementation in BSNL

BSNL is proposing to implement CDR based customer care and convergent billing

system. Since all the switches do not support generation of 100% CDRs, it is proposed

that the billing system should also support the conventional meter reading based billing

in addition to CDR based billing. A centralized integrated billing system with suitable

communication infrastructure will be deployed. This will require a countrywide BSNL

Intranet. There will be 6 Zonal billing centers, in 3 pairs. In each pair, one will act as

disaster recovery centre for the other. Country-wide exclusive TCP/IP based intranet

required for collection of CDRs and meter readings will cover most of the major

exchanges having more than thousand lines. Remaining exchanges will be connected

through dial-up circuits.

There will be centralized data base servers and application servers in each billing

centre with provision of client connectivity upto SSA/SDCA level. All processed

customer-care related data has to flow from centralized billing and customer care center

to designated centers of SSA. Interconnect billing system is proposed at the respective

zonal billing center

6.4 Conclusion

After the Interconnect Usage Charge (IUC) regime has been introduced, it has become

necessary to evolve suitable method of generating CDRs for all the calls of private

operators handled by BSNL switches and collecting CDRs at the Telecom Revenue

Accounting Centre for raising bills. The Telecom Regulatory Authority of India (TRAI)

has also stipulated that BSNL should migrate to CDR based billing from the

conventional meter reading based billing. Accuracy, speed and customer satisfaction

through viewing the reports are the important advantages of CDR based billing

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CHAPTER 7ISDN INTRODUCTION

What is ISDN ?The ISDN is an abbreviation of Integrated Services Digital Network. The current communications

networks vary with the type of service, such as telephone network, telex network, and digital data

transmission network. On the other hand, the ISDN is an integrated network for various types of

communications services handling digitized voice (telephone) and non voice (data) information.

Fig.1 shows the current network configuration with individual networks, such as telephone network

and a data network existing independently, and telephone sets, data terminals, etc. connected

individually to each network

(Current Telephone : Individual access to multiplex networks)

Fig. 1The Network Configuration Without ISDN

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Fig.2 shows individual networks that will be fully integrated in the future.

Fig. 2 The Network Configuration With ISDN

ISDN DefinitionThe CCITT defines the ISDN as follows :

(1) A complete, terminal-to-terminal digital network. Fig.3 shows the end-to-end digital

connectivity.

Fig. 3End-to-End Digital Connectivity

(2) A network that provides both telephone and non-telephone services in the same network.

Fig.4 shows the voice and non-voice services in the same network.

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Fig. 4 Voice and Non-Voice Service in the Same Network (Example)

(3) A network based on a digital telephone network.

(4) A network that utilizes Signaling System No. 7 (SS7) for signaling between switching

systems. Fig. 5 shows the signaling connection between Switching Systems.

Fig. 5 The Signaling Connection between Switching Systems

(5) A network offers standard user network interface. Fig.6 shows the standard user network

interface.

Fig. 6 Standard User Network Interface

ISDN Services(1) A wide range of services

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(a) The ISDN provides the following functions, as shown in Fig.7.

Packet switching service

Circuit switching service

Leased circuit service

Fig. 7 A Wide Range of Services

Circuit switching service includes both telephone and data circuit switching.

(b) As shown in the figure, ISDN can interface with various terminals, such as a

telephone set, FAX, Video terminal or personal computer to provide a wide range of

services.

(c) The ISDN concept can be summarized by two statements :

ISDN offers a variety of services, such as telephone, data and image

transmission through one network.

ISDN handles all information digitally.

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(2) Standard user-network interface. Fig.8 shows the user-terminal/network interface.

Fig. 8 User-Terminal/Network Interface

(a) The subscriber line is connected with an NT (Network Termination) installed at the

customer premises.

(b) Various terminals are connected to the NT. These terminals can include digital

telephones, multi media terminal, digital facsimile machines, personal computers,

etc. as shown in the figure.

(c) The NT and terminals are connected by S or T interface (S/T interface), as

recommended by the CCITT. Up to 8 terminals are connected to one S/T interface.

The NT and terminals are connected using an 8-pin connector, which is also

recommended by the CCITT.

(d) As shown in this figure, the personal computer uses the RS232C interface that is

different from the ISDN S/T interfaces, so a TA (Terminal Adapter) is provided to

adapt the RS232C interface for use with the ISDN interfaces.

c

Fig. 9 shows operation of various terminals in the home.

Fig. 9Operation of Various Terminals in the Home

(a) Each terminal is connected to the NT through S/T interface which, in turn, is

connected to the switching system through the subscriber line.

(b) At the upper left of the figure a person is using a television telephone called a Video

Phone, at the lower left, a person is watching a picture on a Videotext terminal.

(c) At the upper right of the figure, a person is operating a personal computer, which

requires the use of a TA to convert the computer’s RS232C interface to the S/T

interfaces used by ISDN. At the lower right, a person is doing catalog shopping

using a Videotex terminal.

(3) Home Shopping and Home Banking

Fig.10 shows home shopping and home banking services.

Fig.10 shows a typical service made possible by ISDN. It shows something is being

ordered to a department store, and then delivered

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Fig. 10Home Shopping and Home Banking Service

The goods are ordered using the Videotex terminal, and an instruction is output to

the bank to transfer the amount of the bill from your account.

The department store delivers the ordered goods.

(4) Home Medical System

Fig.11 shows home medical system.

Fig.11 shows another service provided by ISDN : the receiving of medical care at

home.

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Fig. 11Home Medical System

The upper left shows the measuring of blood pressure, with the result shown on the

videotex screen both at home and at a medical facility (show at the bottom right of

the figure).

The lower left shows a consultation for medication using a TV telephone.

User Network Interface

ISDN User Network Interface Configuration(1) Fig.12 shows the interface between the user and the network. Telephone service makes use

of two wires for the subscriber line between the switching system and customer’s premises.

These same two wires can be ued by ISDN to receive ISDN services.

(2) An NT (Network Termination) is installed at the subscriber’s home and connected to the

subscriber line.

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Fig. 12The Interface between the User

(3) The Interface between the NT and the ISDN exchange (switching system) is called U

interface. This interface has not been defined in the CCITT Recommendations because

circumstances are different in each country. The point between the NT and the on-premises

terminals is called the S or T reference point. The ISDN user/network interface refers to

these S/T points, and is defined in the CCITT Recommendations.

(4) The S/T interface uses four wires, two for sending and two for receiving. Since U interface

uses two wires, the NT provides a two-wire/four-wire conversion function.

(5) CCITT recommends the use of AMI (Alternative Mark Inversion) code at the S/T point.

AMI code is a bipolar waveform.

(6) As shown in the figure, the ISDN Terminal provides S/T interface that follows the CCITT

Recommendations, and can be connected directly to the NT. Since the personal computer

and the analog FAX utilize a different interface from S/T interface, they require protocol

conversion by a TA (Terminal Adapter).

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Service Access Points (Reference Points)(1) In the existing telephone network, a point at which a service is provided for a user, that is, a

service access point is located at a rossete between the user’s telephone set and the

subscriber line.

Since the ISDN provides various types of service other than telephone service through a

plural number of terminals, various service access points are provided. Thus, service access

points would have to be defined corresponding to the ISDN Services.

(2) Fig. 13 shows the user-network interface reference points which is based on the CCITT

reference model and identifies the important reference points of the model.

Fig. 13

User-Network Interface Reference Points

(3) The following describes the user-access points and the function of each for basic user-

network interface.

(a) Network Termination (NT) :

The NT can be split into NT1 and NT2. NT1 and NT2 are terminating equipment for

the network.

In this case, NT1 provides the Layer 1 functions, such as circuit termination, timing

and supply of electricity, while NT2 provides the layer 2 functions, such as protocol,

control and concentration functions.

(b) Terminal Equipment (TE) :

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The TE can be split into TE1 and TE2. TE1 is an ISDN terminal which is connected

to ISDN via the S/T interface. TE2 is a non-ISDN terminal which is connected to

ISDN via a Terminal Adapter (TA) such as personal computer or analog FAX as

described in Fig. 12.

(c) Terminal Adapter (TA) :

A TA is a physical device which is connected to a non-ISDN terminal (TE2) to

permit access to ISDN.

(d) S-Interface :

A 4-wire physical interface used for a single customer termination between a TA

and NT2 or between TE1 and NT2.

(e) T-Interface :

A 4-wire physical interface between NT1 and NT2.

(f) R-Interface :

A physical interface used for single customer terminator between TE2 and TA.

(g) U-Interface :

The subscriber line is called U-Interface and utilizes 2-wires.

ISDN User Network Interface Points(1) Requirements of User-Network Interface

For us to utilize “integrated services” including voice and non-voice communications and

the use of some new media, such as facsimile in offices and home, the following features

must be provided for user-network interfaces :

(a) Different services for each call

A switching mode (packet switched/circuit switched function) can be

selected.

Data transmission speed can be selected.

(b) Plural number of terminals can be concurrently connected.

(c) The portability of terminals can be ensured.

(2) Basic Structure of User-Network Interface.

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The basic conditions for structuring the user-network interface that satisfy the preceding

requirements can be summarized into the following three points :

(a) Multi services

Common use of various services telephone/non telephone and existing/new

services. As shown in Fig.12, ISDN termianls, personal computers, FAX

machines, etc. are connected to S/T points to offer various services.

(b) Multi points

Up to eight (8) terminals can be connected to one (1) NT as well as point to

point connection.

Fig.14 shows the multi points connection.

Fig. 14Multi Points Connection

(c) Portability

Terminals can be carried from place to place and connected to different sockets for use, just

as home electrical appliances can be carried around and plugged into AC outlets.

(3) Channel Classification

Various channels can be used to transmit information between a terminal and the switching

system. These include B, D and H channels. Each channel has a different bit rate and

information carrying attributes.

(a) B-channel

The B-channel carries user information such as voice and packet data at a

rate of 64 kbps. However, the B-channel does not carry signaling

information.

(b) D-channel

The D-channel interface carries mainly signaling information such as

originating or terminating subscriber number, call origination and

disconnect signals for circuit switching and packet switched user data at 16

kbps or 64 kbps.

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The D-channel also permits multiple logical channels to be established for

use in packet communications.

(c) H-channel

The H-channel carries high-speed user information such as high-speed

facsimile, video, high-speed data, etc. H channels do not carry signaling

information for circuit switching by the ISDN.

(d) Table 1 outlines these three channel types and characteristics.

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Table 1 : Channel Types and Characteristics

Channel Type Bit Rate Function

B 64 kbps To carry user information

Circuit switchingmode and packet switching mode

D 16 kbps

64 kbps

To carry signaling information for circuit switching

H H0 : 384 kbps

H11 : 1536 kbps

H12 : 1920 kbps

To carry high-speed packet data such as facsimile and video

An H channel does not carry signaling information for circuit switching by the ISDN

Note : H0 : 64K X 6 = 384 kbps

H11 : 64K X 24 = 1536 kbps

H12 : 64K X 30 = 1920 kbps

(3) Typical Interface Structures

(a) Basic Interface

This interface is primarily for home use.

The basic interface is set at a transmission speed of 144 kbps. This provides

two (2) 64 kbps B-channels for user information exchange and a 16 kbps D-

channel for signaling and control.

The interface is thus referred to as 2B+D.

Fig.15 shows the basic interface structure.

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Fig. 15Basic Interface Structure

(b) Primary Group Interface

These interface are primarily for business use. The primary group interface

for ATT system consists of a 1.544 Mbps line. This line can thus provide up

to 23 B-channels at 64 kbps and a single D-channel at 64 kbps.

In Europe and other countries using CEPT system standards, the primary

group is 2.048 Mbps and the interface is 30B-channels and single 64 kbps

D-channel.

This line is used for PABX etc.

Fig.16 shows the primary group interface structure.

Fig. 16Primary Group Interface Structure

(c) Table 2 shows the typical user network interface structure.

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CHAPTER 8.

INTELLIGENT NETWORK

8.0 Overview of Intelligent Network ArchitectureOver 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.

8.1 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 :

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Rapid Service DeploymentMost 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.

Reduced Deployment RiskPrior 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 ReductionBecause 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.

CustomizationPrior 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.

8.2 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.

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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.

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.

8.3 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.

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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 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

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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.

The IN architecture is depicted in Fig.1

Fig. 1IN Architecture

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Management interfaceIn real time interfaceSignaling circuit interface

8.4 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.

CCFCall 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.

SSFService Switching Function is used to switch calls based on the advice of the SCF at the

SCP. This function provides a service independent interface.

SCFIt contains the service logic components and advises the SSF at SSP on further call

handling.

SDFService Data Function contains the user related data and data internal to the network.

SRFSpecialized 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.).

SCEFService Creation Environment Function specifies, develops, tests and deploys the services

on the network.

SMAFService Management Access Function provides an interface between service management

function and the service manager who may be an operator.

SMFService 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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SMF

SMAF

SCEF

SDF

SRF

SCF

SSFSSFCCAF CCF CCF CCF CCAF

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 1Distribution of FE's over PE's

Physical Entity Possible Functional EntitiesSSP CCF, SSF, CCAFSCP SCF, SDFSMP SCEF, SMF, SMAF

IP SRF

Table 2FE-FE Relationship to PE-PE Relationship

FE-FE PE-PE ProtocolSSF-SCF SSP-SCP INAP, TCAP, SCCP and MTPSCF-SDF SCP-SDP X.25 or ProprietarySCF-SRF SCP-IP

SCP-SSP-IPINAP, TCAP, SCCP and MTPISUP, INAP, TCAP, SCCP and MTP

SRF-SSF SSP-IP ISUP and MTP

8.5 IN Services

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.

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(1) ABD – Abbreviated dialingThe 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).

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 UnconditionalThe 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.

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(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 CallingThe 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. 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

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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. 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

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- 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. 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.

(22) UAN – Universal Access Number National number is published by the subscriber.

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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.

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

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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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************

CHAPTER 9

INTERFACE V5.2

1.0 Introduction

The V reference points are ITU-T Recommendations, describing configurations of the Access Digital Section. They are all reference points in an architecture where ISDN traffic is carried through an Access Network to a Local Exchange. V5.2 is one of the full-fledged interface which can carry both ISDN and PSTN traffic. All the telecom switches available in the market today (EWSD, OCB-283, C-DOT, FETEX, 5ESS, AXE etc) are V5.2 compatible, and therefore, access network based on FILL (Fibre in Local Loop )and WLL (Wireless in Local Loop) without having complex switching and advance features can be parented to new technology switches, extending all the features to users transparently.

InterfaceV5.2 specifies the physical procedural and protocol requirements that control communication between Local Exchange (LE) and Access Network (AN). Different from protocols that are used uniquely at each proprietary, Interface V5.2 is an open interface for all LE vendors and also all types of access networks. User ports (PSTN/ISDN) interface terminates on the AN instead of directly terminating on the LE. Access network provides the narrow-band services to the users connected to it. Call control responsibility still resides at the local exchange.Interface V5.2 being an open interface interoperability with other devices that have the same or different allocation but with different specifications is possible. It is based on G 703/G704 interface at 2048 kbps ( E1). One V5.2 interface can have maximum 16 E1 links. Access Network (AN) forms a system that connects telecommunication customers to local Exchange(LE). This system can be in the form of physical, functional or interface connection. AN method and technique are developed based on certain access technology. Basically there are three types of applicable technologies.

(i) The conventional copper cable technology(ii) The Fibre optic technology and(iii) The Radio access technology.

A typical AN- LE interconnection, various services, links and maintenance terminals are shown in figure-

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V5.2 InterfaceNew Technology

Switch(OCB, 5ESS, EWSD, CDoT MBM)

Sub Access Node (SAN)

1. DLC2. AN RAX3. WLL BSC

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1.1 V 5.2 philosophy

Interface V 5.2 is 2 Mbps interface ( G 703) LE-AN with following advantages.

Interface V5.2 is used for connecting LE to various types of access systems as fibre optic, coaxial and radio.

Interface V5.2 is multi vendor interface or open interface which means by using interface V 5.2, AN from any vendor can be connected to LE from any vendor.

Interface V5.2 simplifies signal handling process especially for ISDN service, since the signal does not need to be brought to analog level but just to 2 Mbps level. Only control protocol is defined in the V5.2 for the exchange of the individual function and messages required for the co-ordination with the call control procedure in the LE. The PSTN user part is converted into a functional part of the V 5.2 protocol for signaling to the AN side.By using interface V5.2, there is no need of Main Distribution Frame (MDF) any more over Fibre Access Network. The absence of MDF will make more simple configuration and the problems caused due to cabling system and jumpering can be reduced. This will make operation and maintenance easier and more economical.

Network development only requires AN implementation and network can be expanded as it uses interface V5.2 as multi vendor interface. All the features of Main exchange(LE) are transparently extended to all AN connected users.

Allows network management (Both Administration and Maintenance of Resources ) of Access Network via Q3 interface.

Allows user selection of the Local Exchange. Effective usage of bandwidth available Types of services that can be passed through interface V5.2 are customers of regulars

telephone( POTS) ISDN BRA/ ISDN PRAPABX DID and other analog or digital accesses for semi-permanent connections with out associated out band signaling information.

By AN and interface V5.2, all line signaling from customers are interpreted and handled by AN to be passed through to LE. Data signaling format in the digital signals, structure between AN and LE is standardized in interface V 5.2. Interface V5.2 is dynamic multiplus, means that there is no traffic channel allocation for permanent customers. Each customer can occupy any unoccupied channel. One interface V 5.2 can consist of maximum 16 links E1( 16 links 2 Mbps)which means maximum 480 traffic channels.

1.2 Conclusion

In todays telecom scenario where switches from various vendors exists in network transparent inter networking of all such switches has become very necessary so that uniform numbering scheme can be adopted and all the features of the switches are available to all customers and V 5.2 can solve all these requirements. It can extend the telecom services to non feasible and rural areas using WLL technology as last mile solution. V 5.2 based DLC system can solve cable pair

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problems in high concentration areas. Remote Switching Unit (RSU) of different vendors can be parented to one exchange for uniformity in numbering scheme, billing and other subs management functions. Therefore, V 5.2 based telecom equipments can bring revolutionary changes in Telecom Net works by transparently extending services

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CHAPTER -10

NEXT GENERATION NETWORKS

Why NGN?

NGN is about the network infrastructure that will enable the provision of the existing telecommunications services and innovative applications of the next generation. The term NGN refers to a converged network capable of carrying both voice and data over the same physical network, with all traffic (voice and data) carried as IP.“IP networks are likely to be simpler and easier to run and maintain as compared to the existing

legacy networks and provide the operators with sufficient flexibility in their cost base to reduce both opex and capex. In addition, all IP networks allow for innovation in terms of new services and applications, with a truly converged product offering that bridges the current distinction between fixed and mobile networks.”The process of realization of NGN will lead to a revolution in the design and build-up of

telecommunications network architecture, which will result in a change in the way service providers offer their services and the way people communicate. Ultimately, NGN would phase out the existing legacy networks at a point of time in the future.

There are some practical factors that have collectively formed the key drivers for NGN migration. Firstly, the existing network operators are facing fierce competition in the market and they have to remain competitive to survive. In order to achieve this, operators are trying to build cost-effective businesses on the one hand and create new business models and generate new revenue streams on the other hand. The convergence of fixed and mobile networks and integration of voice and non-voice services are becoming their targets because such approachwould lower operational cost and allow greater flexibilities for service innovation and shorter time-to-market.

Secondly, the increasing service requirements from end users call for innovative applications / multimedia services, high flexibility of service access, high bandwidth, high quality of service and etc. Apparently, the operators’ need for remaining competitive and the end-users’ demand for increased service requirements are together forming a strong driving force pushing the development of NGN all over the world with characteristics and features that would fulfil the needs of network operators, service providers and end-users.

NGN is as much about easier provision of advanced services such as VoIP, Broadband, multimedia applications etc. as it is about cost saving through simplification of network.

A migration to NGN will bring about a complete change in the existing business models which is a source of concern for both operators and regulators world over.

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Introduction to the NGN concept

Historically, incumbent operators typically ran one network — the Public Switched Telephone Network (PSTN). The PSTN was designed to carry voice when voice was the only communication carried. As demand for data communications developed the incumbents adapted their networks to also carry data traffic. However, typically, rather than replacing the PSTN operators typically built new networks that they ran in parallel – which is called the overlay network. These new overlay networks were designed specifically to carry data traffic.

Fig. 1 NGN IS ABOUT SIMPLIFYING NETWORKSToday : Many Networks Tomorrow : Single IP Network

As network technology continued to develop, the number of networks multiplied in step. As a result, today, many operators run typically 5-10 different network platforms (ATM, IP, Frame

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All IP

network

ing

DSL SDH Frame Relay X.25 Gig E Dial-UP

Accessnetwor

k

Opportumity

Communication(Voice,Data)

Lifestyle / Workstyle(Wireless, Home etc.)

(TV, DSL,

Games )

Infotainment

Frame RelayX.25IPPSTN

CoreNetwork

ATMPSTN class4/5SDHIP

Limited,separate services

Edge Network

Relay, ISDN, PSTN, X.25 etc.). The problem with this multi-network approach is that it has created a web of complexity resulting in management complexity, operational inefficiencies, maintenance issues and duplicating capex.

Next Generation Networks aim to reverse the clock and go back to the simplicity of one single network. NGN is all about deploying one network platform capable of supporting all traffic types while facilitating service innovation (Fig. 1).NGN Definition

ITU defines Next Generation Network (NGN) as “a packet-based network able to provide telecommunication services and able to make use of multiple broadband, QoS-enabled transport technologies and in which service-related functions are independent from underlying transport related technologies.

It enables unfettered access for users to networks and to competing service providers and/or services of their choice. It supports generalized mobility which will allow consistent and ubiquitous provision of services to users”. Convergence between Telecom and Internet

It is believed that the rapid and widespread growth in the use of the Internet has become the catalyst to the fostering of such a concept of NGN. With Broadband access service becoming increasingly popular, easily accessible and more affordable to any corporate entity and individuals, more and more applications and services have been developed and evolved based on the IP technology of Internet, varying from narrowband voice telephony services (i.e. VoIP) to broadband applications such as high-speed Internet access, video conferencing, multi-casting of TV programmes and etc. The increasing proliferation of IP-based services has in turn driven the rapid development of packet-based networks in the access, transport and core layers of the telecommunications infrastructure in order to cater for the drastic increase in the volume of IP traffic. Such a change in telecommunications services brought about by the Internet has paved the path and laid a foundation for the development of IP-based NGN.NGN Principles

From a high-level perspective, Next Generation Networks rely on three main principles. First of all, NGNs are implemented in such a way that the functions performed by the network are separated into functional planes. These functional planes include access, transport, control & intelligence, and service layers. Layers are independent in the sense that they can be modified or upgraded regardless of other functional layers. This layered architecture provides a flexible and scalable network, reducing time to market for the implementation of new services. Moreover, the functional planes are separated by open interfaces in order to facilitate the interconnection to other operators’ networks but also the integration of third-parties’ services and applications. Provided that commercial agreement is reached between the different parties, such a principle can widen the operator’s coverage and service scope and can also provide end users with an access to a greater number of services.

Last but not least, NGN is a multi-service network meaning that an NGN can be used to provide many services, as opposed to legacy networks that are only used for specific services. This multi-service network enables operators to implement converged and new services in addition to POTS. From the users’ perspective, the convergence of services will enable the emergence of the seamless service concepts, where users can access their “desired” services from any type of access network.

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The NGN architecture

Along with a new architecture, Next Generation Networks will bring an additional level of complexity over existing networks. The addition of support for multiple access technologies and for mobility results in the need to support a wide variety of network configurations.

The NGN architecture supports different services including multimedia services and content delivery services such as video streaming and broadcasting. NGN provides support for PSTN/ISDN replacement (i.e. PSTN/ISDN emulation) as well as PSTN/ISDN simulation. In addition, the NGN provides infrastructure for Value Added Service addition by 3rd party. The NGN reference architecture comprises three distinct levels:

The transfer network carries out the transport in the form of packets, of information flows interchanged between peripheral units, user terminals and service provider servers:

The network control includes the functions necessary to establish the links needed to transfer information in conformity with request from the applications, whether these be implemented by users, operators or service providers.

The Service control not an integral part of network is related to the final service provided for the user.

Fig 2 The NGN Architecture

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Transfer network

The NGN transfer process can be divided into five main parts User facility: depends upon what type of equipments user is using. Connecting user facility: connection is hereby defined as the part of the network through

which user facilities are connected to the first equipment unit from which user flows are multiplexed or concentrated onto shared transmission media. Customers may process their own connection resource (dedicated wire access) but the connection may also be shared at least partially as in the case of radio access or cable networks. The connection network will be considered to include the first network unit shared by several customers to be defined by the generic term connection unit (Access Node, Access Muxer, Access Gateway) and classified in terms of base stations and base station controller for radio access, DSLAM for ADSL access, distribution centre for cable network, etc.

Aggregation: The aggregation functions in linking connection units to the peripheral units to the peripheral routing nodes at which level communication between users are setup. It is independent of connection technologies and is implemented in networks that each agglomerate flows originating from customer connections from one area and convey them to peripheral routing node managing several areas.

Peripheral routing: It terminates customers virtual access and handles the elementary information flows they carry by sorting, classifying and finally routing them either individually to local customers (connected upto same routing node) or in groups to the network core. As well as handling information flows, routing extracts the control flows sent by customers and directs them to the network control.

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Transit : The functions of transit is carried out by the core network, a very high speed meshed network that handles the aggregated flows that are transported through various channels, giving priority to routing speed and economy rather than optimization of resources, which are a source of processing complexity and high costs. The NGN core network combines, then high transmission rates, long distance and low costs.

Network control

Controlling the NGN involves four main functions : mediation for access to the services, user mobility and presence management and control of resources.

Mediation: Accessing services in a competitive environment: The mediation function represents the interface between customers and service providers. It takes the form of a portal which will enable interchanges with the customer by calling on different technologies depending upon the terminal used : HTML pages on PC, WML on a mobile phone, audio on a fixed telephone, etc. through this portal customers can navigate among the components of their service package, either to use them or manage them.LIKDPGKPFDGPD

Managing mobilityThe aim of mobility management is to allow users to enjoy their services wherever they are.

This involves maintaining a constant relationship ( necessary for the routing of information) between the address of the terminal used and the address of its active point of connection to the network, whether this connection is by wire or radio. The characteristics of the terminal and of the access to which it is connected must also be constantly available in order to be able to adapt services to their performance ( access speed, user dialogue mode, terminal operating system, etc.).

Managing presence: in the network, in the services, or both?In traditional telephone networks, the status of the user was simply ‘free’ or ‘busy’ and was

supplied only after a call was made. Presence management, however, consists in permanently posting the status that the user of the network wishes to make known to others: ‘I can be contacted by members of my sports club’, etc. The concept ha the potential to transform the way people communicate, and theoretically could be applied to all forms of communications. It could also be shared by many services and thus become, at least in part, one of the generic control functions associated with the network.

Controlling resourcesResource control consists essentially in the process of receiving a request of a given

application, deciding on a route between two peripheral points on the network ( where the relevant terminals or servers are located ) and configuring or reserving resources along this route. Controlling I performed at two levels, that of the global network and that of the equipment, and corresponds to several entities: physical link, virtual circuit, address, transmission rate, memory capacity, etc.)Request from services are expressed in terms of parameters that are independent of techniques and mechanisms implemented by the network. This makes it necessary to translate, for examples, the identifiers (URL, directory number, etc.) used by the service or the expression of the quality of the service expected into a language adapted to the equipment and to the resources themselves. Although it is an important characteristic of NGN, this ‘de-correlation’ is nevertheless difficult to apply, and this is one of the reasons for the continuing variations in vocabulary used to describe this function: ‘ Gatekeeper’, ‘Call Server’, ‘ Policy Server’, or ‘Session Server’ are all used for sets of functions that are not always identical.Even though the above definition is evocative of information transfer in particular, the control of resources also concerns other types of equipment that may be used in applications ( voice servers,

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cache servers, transcoders, etc.)

CONTROLLING SERVICES

Each service is controlled in a specific way. Algorithms and data are combined to enable the implementation of end services between two customers or between a customer and a server. For example , for a VoD service, the user I given a choice of commands foe selecting, then navigating within a movie supplied by a provider , using such command as fast forward, rewind, pause etc. which are commonly used on a VCR or DVD player.Service control is not part of the network, but uses it so that the media component of the service transferred between the terminals and the servers with the expected quality of service and with a minimum of constraints regarding location of customers and providers.Control of resources and the transfer network

As described in previous sections control of the network includes a resource control function which

has the objective of controlling, when requested by applications, the reservation and the

configuration of all the resources needed for these applications to run smoothly, in accordance with

the parameters that express the required QoS. These resources may be associated with information

transfer: either directly (to attribute an address, establish a channel, reserve a transmission rate)or

indirectly (via the intervention of a server, etc). These resources may also be linked, for example,

with the dialogue between user and application (via the intervention of a voice server, etc.) or with

security functions (implementation of a firewall, etc). It is noticeable that the control of resources

comprises a “global” part, located in the network control, and a “decentralized” part, at the level of

the network elements themselves.

As regards the control of transfer resources in the NGN, three main aspects should be mentioned.

-The consideration of differentiated QoS, which implies that the network elements know, flow

by flow, which level of QoS to observe, in such a way as to be able to implement the

appropriate mechanisms

-The possibility of modifications during a single session, which might mean a change of the bit

rate allocated to a flow that has already been established, the processing of a complementary

flow or the routing of a given flow via an intermediate entity such as a transcoding device.

-The convergence of transfer resources onto a single control interface whatever the applications

involved. In conventional networks, the control of transfer resources uses one of two quite

distinct channels according to the origin of the request : network operator for a management

application or user for a communication service. This has led to the need to manage two

different interfaces each with specific protocols: the “equipment-management” interface

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(considered a weak or average real-time constraint) and the equipment-call processing’ interface

(a severe real-time constraint). However, in the NGN convergence between the fields of

management and control of services(illustrated, from the end user’s point of view, by the fact

that he will himself increasingly have direct access to management facilities) should also lead to

the convergence of the control of resources towards a single interface at the equipment level.

NGN Control

From Service Control to resource control

With the aim of ensuring the connection between users and the terminals and servers

involved in a telecommunications service, it is necessary to decide which paths the media flows

between these terminals will be routed along. This may or may not mean building real new

paths by manipulating addresses and indications of directions, and deciding whether or not to

set aside network capacity, or establishing priorities between flows sharing the same path,

according to the characteristics of the media flows that support the service requested.

Establishing this ‘connectivity’ and managing its evolution comes under the title of “network

control”, whose prime role is to give directives that will allow flows to follow the right paths

under satisfactory conditions. This role is known more precisely by the term “resource control”.

It must, of course, take into account the techniques used on the path being established. It is a

different task to establish a path through a 64kbps switched digital network than to do the same

through an end-to-end IP network, where it is enough to indicate to an SIP terminal, for

example, which IP address should be used to reach the requested party, and perhaps to

implement the appropriate mechanisms to guarantee the required level of QoS.

One of the features of NGN principles in network control is to distinguish between resource

control and “call control” or “session control”. The aim here is to separate.

- The implementation of actual resources, specific to transfer techniques, that are necessary for establishing physical connectivity, with the required QoS, between network inputs and outputs, these being represented by physical addresses. This is the role of resource control.

- The actions, on a more generic level, ideally independent of transfer techniques, that manipulate the various possible representations of these addresses, and that determine the general characteristics of the connection to be made, particularly in terms of bit rate and QoS. This is the role of call control or session control.

The first distinction, proposed within the framework of NGN, has never really been

implemented before. In telephone networks, for example, resource control and call control are

integrated in the “call-processing” of the switches.,

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A second distinction, also included in the NGN, separates everything to do with network

control, i.e. the more or less simple connection (in a session) of terminals, from everything to do

with the control of the services provided during these sessions. Control of services here can be

extended to include even the consideration of complex information necessary for the complete

establishment of the session. This may concern the identification and authentication of the users,

their presence and availability, network and even geographic localization, their means of

payment (especially in the case of prepayment)etc. This distinction may be considered as an

extension of the IN (Intelligent Network) concepts but experience shows it to be difficult to

define and even more difficult to stabilize, In fact, certain services can be processed only at the

level of call control. This is the case, for example, for a simple telephone or videophone

communication. For other services, the definition may appear to be quite a simple matter, as in

the case of a pre-paid multimedia messaging service. However, it is easy to imagine that this

distinction between network control and service control would be less easy to define in complex

applications involving several actors, network operators and service providers. The availability

of open interfaces, which has not been achieved in the IN context applied to telephony, can be

seen to be even more necessary in multi-service networks like the NGN.

It will also be noted that, according to this principle of separation, the network becomes, in a

sense, unaware of the application that is actually being implemented once the session is

established, which is different from the traditional context of telephone networks. For example

for a terminal that connects to a web server in normal (best efforts) mode via an ADSL link, the

NGN network control will set up a link to the internet access provider (identification of access,

allocation of an IP address, session initiation, routing), then will supervise the connection, but it

will not “see” which services the user is actually using during the session. Depending on how

the application proceeds, the network control may be solicited to set up a new telephone or

videophone call.

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NGN Control Functions

The Control functions presented here are not new and are already being used in today’s

telecommunications networks. However, evolving from a context of dedicated networks to a

potentially multi-service one implies a rethinking of these control functions in terms of their

application, no longer to a determined set of services, but rather to the whole range of services

accessible through the NGN. This analysis does not take into account any particular

implementations, which might be very heterogeneous, as indeed has been proved by the variety

of initial commercial products ( ‘ softswitch,’ call server, ‘ application server, etc.) These

functions should be seen as being interdependent, and their implementation can use common

resources, notably databases.

IDENTIFICATION AND AUTHENTICATIONThe identification function consists in establishing the link between a terminal that accesses a

network, the user of such a terminal and a customer who has a contract with a network or

service operator. This customer may be represented by an anonymous account. Such is the case,

for example, with anonymous pre-paid communications, where the contract is implicit.

This identification may be authenticated in order to reduce the risk of forgery and to ensure the

identification of a terminal matches its associated physical address and its user. In conventional

telephony, for call set-up purposes, identification is deduced from the address of the

subscriber’s copper line and no authentication is performed. On GSM and UMTS mobiles,

identification is carried out through the SIM card and authenticated through the PIN code. For

Internet applications, including access, the still prevalent, simple pair of operations ‘

login/password’ gives an authenticated identification. During a single access session that

supports different services, several identifications might have to be made, and it could be useful

to try and join them in order to avoid duplication or , conversely, to distinguish between them.

An identification function is usually one of the first functions to be used, at least when it comes

to finding an address or giving one to a terminal.

SERVICE MEDIATION

A characteristic function of the NGN is the mediation of access to services. From a single

terminal or from several terminals, a user can launch successively or simultaneously different

access and communication session. This may also occur during an active service session. For

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example, on a GSM terminal it is possible to choose a mobile access network from among those

available at a given place. In a wider sense, it is possible to imagine that a wide range of

interpersonal communication services ( voice or videophone combining several media), mixing

direct interaction with the distribution of shared information, might be accessible via an

advanced control interface that would prompt for call and resource controls in order to set up

new sessions (or sub-sessions). This control interface would already be part of another third

party communication session. The important joint here is that communication services can be

started from an active service session. For example, during a best effort IP session, it could be

requested to set up a videophone session from a web page between two terminals on the

network Such access to services through mediation is characterized in the Internet sphere, at

least in part, by the concept of ‘ portals’, with the added possibility of managing one’s own

session parameters or personal preferences.

PRESENCE MANAGEMENT AND INDICATION OF AVAILABILITY

Presence management originated on the Internet. As the presence of a person on the network

could not be detected, because he was not associated with a fixed network address, and as

machines could also be connected in the absence of their users, it became useful to know if a

potential communication partner could be ‘ reached’ on the Internet. Presence management was

initially used in instant messaging, which is different from conventional asynchronous

messaging in that an ‘ instant’ reply is expected to a short message. It is also now seen as a

potential delayed trigger for interpersonal relations, where information is sought regarding the

manner in which parties can and wish to be contacted before they are actually contacted. It is

even possible for net surfers to cause other people to contact them by prior posting of their

availability. The combination of presence management and mediation in portals will be one of

the major developments of the future in interpersonal relations. It will lead to the registration

and management of the information provided by users on the conditions under which they

would wish to set up a communication with ( or be reached by ) specific people or groups by

means of such and such a session type. This is related to the discussion on ‘self-management’.

Furthermore, this same information will be accessible to the network control in order to set up

( or not) the sessions requested , taking into account conditions named by the users.

MOBILITY AND NOMADISM

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One characteristic of NGN networks is the consideration given to the management of all forms

of mobility and nomadism. Under the concept of mobility, the most obvious aspect is radio

access networks allowing users to communicate from wherever they happen to be through a

terminal they carry with them, and to move about during the communication. However, another

form of mobility, established at about the same time, developed on fixed networks, first with

post-paid cards linked to a contract, allowing the user to adopt provisionally a fixed access in

order to make available from there some of his personal services, such as a directory. This is

referred to as nomadism, mobility being a term reserved for the ability to move around during a

single session.

Nomadism means a user being able to get through to his communication and information

retrieval services from different physical accesses, whether they be different terminals with

different identifications and capabilities or different network access interfaces and local loop

operators. Yet nomadism, unlike mobility, is not really concerned with the continuity of an

access or communication session. By combining mobility and nomadism it would become

possible to suspend and resume communication sessions ( i.e. with the same context of

communication and the same initial identification) on different access sessions. To this extent,

mobile networks already offer the user a combination of mobility and nomadism through the

possibility of establishing an access on the networks of different operators ( usually those

having roaming agreements) and of accessing his own services. In the standardization of

UMTS, there is a definition of the concept of ‘VHE’, or Virtual Home Environment, that has

the objective of giving the user a consistent impression of the execution, the presentation and

the management of his services, no matter where he may be located, no matter what type of

access and terminal he is using and no matter which network is being visited. Implementing

mobility and nomadism requires a combination of control of services, networks and resources

according to the type of session that needs to be established and maintained, and will usually

require the co-operation of several different actors. Firstly, the relationships between different

addresses will have to be established and developed, implying the designation of the user,

connection to a terminal and connection to a network access. Any matches between addresses

that may be made inside each of these domains will have to be ignored, and the fact that these

addresses that may may be made inside each of these domains will have to be ignored, and the

fact that these addresses may be under the control of different operators must be taken into

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account Next, the information must be found and exchanged between these domains and

between operators, making sure that it is not attached to a physical location, as was the case in

the PSTN.

METERING AND MONITORING

Making resources available is a part of the contract between the actors, and it is therefore

necessary to check that these contracts are being respected. Providing means to control and

count the resources used in an access or service session is the business of service and network

control, and the existence of this metering and monitoring has strong consequences for the

service and network architecture. The distribution of control and the possibility of establishing

different qualities of service at the same interfaces seriously complicate the implementation of

this monitoring and metering as well as the possibility of forwarding the related data without

any distortion up to contractual points of reference. In the conventional circuit or packet

networks, local contractual metering was considered sufficient, as long as what passed for a

contract could be deduced from what actually happened on that access. In the simplest cases, the

local exchange of a PSTN measures the duration of the communications, and billing then

depends on this duration and on the number dialed. The circuit-mode structure of the network

automatically ensures that the service contract is respected and no other verification is

necessary. The situation became more complicated once rerouting mechanisms were introduced

and the metering was effectively distributed. Signalling was used to trace the metering

information back to the origin of the call, but disruptions can upset this process, and can lead to

the need to abandon the service. In routed packet (IP) networks, it is also at access that either

the time or the quantity of packets transmitted can be measured , but it is very difficult to check

that an end-to-end quality of service contract has been strictly respected, and it comes back to

ensuring beforehand, through the resource control, that the architecture will guarantee that the

contract is respected In fact, it is easier today to establish contracts making available large

capacities of resources between network or service operators than to verify the dynamic and

distant consumption of such resources by individual users. This situation may be improved by

distributed metering and post hoc verification or payment, but the major difficulty, between

integrating control and metering, remains the generalization of pre-payment possibilities to all

types of services.

TERMINALS IN THE CONTROL

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A further feature of NGN will be the very great variety of terminals with widely differing

characteristics ( type of codec, window size, storage and display capacities, etc.) that must be

known to applications servers, but also that must be accessible to the network control so that the

service required can be adapted to the terminal used. This may require the use of intermediate

transcoding devices , for example, between communicating terminals. These terminals will be in

part, manageable by or via the network , Software could be downloaded onto them and updated,

data could be synchronized etc. This will not be developed in the short-term, as it has as yet

rarely been put into practice, but it seems sure that terminals equipped with everincreasing

capacities ( for processing and storage) will be integrated totally into the handling of network

and service control functions.

SECURITY

The traditional fixed telecommunications networks operate with a very high level of quality,

and this value is noticed by users. Their handling of emergency calls is viewed with a high

degree of confidence and they offer relatively good protection against intrusions. The is not the

case with the Internet, where users do not always find their services are available and are

sometimes victims of unwelcome, or even malicious intrusions. It is, therefore, of particular

importance to bring the same performance levels of these conventional networks, in terms of

availability and security, to packet-mode networks like the NGN. Network control through its

many links with a very diversified environment ( of users, other networks, serevice providers,

etc.) will obviously be an active participant in this process.

User Profile and Customer profile

The implementation of all the functions mentioned above requires the management of data

related to terminals, users and contracts, if the required communication sessions are to be set up.

Instrumental to this is the concept of user profile, linked to the activation of services, and that of

customer profile linked to contracts and payment. The customer profile can be defined in

abstract terms as the set of customer-related data that correspond to the ‘User Agent’ as defined

in TINA and the user profile as the set of user-related data, whether it be for access to

communication services or for the use of these services, Several user profiles can fit a single

customer. For example, in a company or in a family there are usually several service users for a

single customer. Moreover, several user profiles can fit a single person. These are often related

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to different customer profiles, often concerning the contract or to different types of access

concerning usage ( e.g. mobile and ADSL access for the same person).

Conventional telecommunications developed without making a distinction between these

two types of profile , Normally, an access is associated with an identifiable customer and

anonymous user at a given moment, and the data are distributed all over the network and the

Information System. However, it could be considered that this type of approach already exists,

either in the Information System (a single invoice containing the billing information of several

users for example) or in the network (by using the intelligent network to distinguish between

users in, for example, virtual private networks).

The GSM mobile networks already have a real profile base too, thanks to the use, in the

terminal, of the SIM card, and in the network, of a data base holding the authorization and usage

profile of a mobile number ( the HLR). The network also has access to information concerning

location, which is therefore related to the temporary situation of a terminal and to its actual

access. But all the service data needed by the person who uses the terminal carrying the

contract-related SIM card are not present in the HLR and there remains a certain association

between customer and user. In a way, the user is equated with his SIM card, and is authenticated

thanks to a PIN code. The UMTS networks will build their architecture around an extension of

HLR, the HSS ( ‘Home Subscriber Server’), which includes more service data, particularly

related to the packet network. These are databases that are used as a model for profile aspects,

but are extended by user information and preferences such as those stored in the portals of the

Internet.

A lot of different types of information are needed to implement these control functions. Here are

just a few examples.

— access- related information: identification of the access address, of the address linked to a service ( which may be different from the former, e.g. the ATMVC address for an ADSL access, the IP address for services arriving at the terminal). Features of the terminal used. Characteristics of the type of access;

— user-related information ( at the network or service level) : identification of the user, the associated method and means of authentication, services that can be activated by the user according to the access selected, preferences, user’s rights, location data, data related to presence and to the ability to be reached, credit remaining, etc.;

— customer-related information delegation of rights to other users, contract validity, billing information or pre-payment accounts, etc.

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These data may be static, changing very slowly, and be modified by explicit acts of

management; they are above all contractual data. They may also be temporary and dependent on

active sessions. But their static, manageable and temporary nature does not always remain stable

in the evolution of networks and services. A typical example is that of a terminal’s IP address

that has been attributed in a fixed manner through contract, then attributed by an address server

at the set-up time of an access session, and which may be allocated in a fixed way again, e.g. by

using Ipv6.

The data will be accessible to and modifiable by a large number of people (network operators,

service operators, customers, users), by network servers and by terminals during access or

service sessions. Conceptually speaking, it would be easier to store all data needed for

communication in a single database, indexed with a single identifier related, for example, to the

user. However either for reasons of time of access to the data or for reasons of ownership and

confidentiality, this is not possible, and the data will have to be distributed over bases which

might need to be dedicated. The profile of a user will, in fact, be made up of a series of profile

fragments, located in the various domains, and featuring a certain degree of redundancy or

replication. The bases of profile fragments will be an important part of the NGN economy. For

the moment, in the definition of the profile bases, two approaches can be identified. Whose

principles are, for the most part, agreed upon; one, the previously mentioned HSS within the

framework of 3G mobile networks; the other having the objective of forming profile bases of

identified and authenticated users on the Internet with, as the first type of application,

simplification and increased security for e-commerce applications.

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