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A telecommunication network establishes and realizes temporary connections, in accordance with the instructions and information received from subscriber lines and inter- exchange trunks, in form of various signals. Therefore, it is necessary to interchange information between an exchange and it external environment i.e. between subscriber lines and exchange, and between different exchanges. Though these signals may differ widely in their implementation they are collectively known as telephone signals. A signalling system uses a language which enables two switching equipments to converse for the purpose of setting up calls. Like any other language. it possesses a vocabulary of varying size and varying precision, ie. a list of signals which may also vary in size and a syntax in the form of a complex set of rules governing the assembly of these signals. This handout discusses the growth of signalling and various type of signalling codes used in Indian Telecommunication. Telephony started with the invention of magneto telephone which used a magneto to generate the ringing current, the only signal, sent over a dedicated line between two subscribers. The need for more signals was felt with the advent of manual switching. Two additional signals were, therefore, introduced to indicate call request and call release. The range of signals increased further with the invention of electromechanical 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. ii. iii. iv. v.
A request for originating a call is initiated when the calling subscriber lifts the handset. The exchange sends dial-tone to the calling subscriber to indicate to him to start dialing. The called number is transmitted to the exchange, when the calling subscriber dials the number. If the number is free, the exchange sends ringing current to him. Feed-back is provided to the calling subscriber by the exchange by sending, a) Ring-back tone, if the called subscriber is free(shown in fig.1) b) Busy tone if the called subscriber is busy ( not shown in the figure), or c)Recorded message, if provision exists, for non completion of call due to some other constraint ( not shown in figure).
vi. vii. viii.
The called subscriber indicates acceptance of the incoming call by lifting the handset The exchange recognizing the acceptance terminates the ringing current and the ringback tone, and establishes a connection between the calling and called subscribers. 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.
The originating exchange seizes an idle inter exchange trunk, connected to a digit register at the terminating exchange. The originating exchange sends the digit. The steps iv to viii are then performed to set up the call.
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. 220.127.116.11 Call request When the subscriber is idle, the line impedance is high. The line impedance falls, as soon as, the subscriber lifts the hand-set, resulting in increase of line current. This is detected as a new call signal and the exchange after connecting an appropriate equipment to receive the address information sends back dial-tone signal to the subscriber. 18.104.22.168 Address signal After the receipt of the dial tone signal, the subscriber proceeds to send the address digits. The digits may be transmitted either by decade dialing or by multifrequency pushbutton dialling. 1. Decadic Dialling The address digits may be transmitted as a sequence of interruption of the DC loop by a rotary dial or a decadic push-button key pad. The number of interruption (breaks) indicate the digit, exept0, for which there are 10 interruptions. The rate of such interruptions is 10 per second and the make/break ration is 1:2. There has to be a inter-digital pause of a few hundred milliseconds to enable the exchange to distinguish between consecutive digits. This method is, therefore, relatively slow and signals cannot be transmitted during the speech phase. 2. Multifrequency Push-button Dialling This method overcomes the constraints of the decadic dialling. It uses two sets of four voice frequencies. pressing a button (key), generates a signal comprising of two frequencies. one from each group. Hence, it is also called Dual-Tone Multi-frequency (DTMF) dialling. The signal is transmitted as long as the key is kept pressed. This provides 16 different combinations. As there are only 10 digits, at present the highest frequency, viz., 1633 Hz, is not used and only 7 frequencies are used, as shown in Fig.2. By this method, the dialling time is reduced and almost 10 digits can be transmitted per second. As frequencies used lie in the speech band, information may be transmitted during the speech phase also, and hence, DTMF telephones can be used as access terminals 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.
HIGH FREQUENCY GROUP 1209 Hz 1336 Hz ABC 2 1477Hz DEF 3
FIGURE 2. TONE-DIALLING FREQUENCY GROUPS.
22.214.171.124 End of selection signal The address receiver is disconnected after the receipt of complete address. After the connection is established or if the attempt has failed the exchange sends any one of the following signals. 1. Ring-back tone to the calling subscriber and ringing current to the called subscriber, if the called line is free. 2. Busy-tone to the calling subscriber, if the called line is busy or otherwise inaccessible. 3. Recorded announcement to the calling subscriber, if the provision exists, to indicate reasons for call failure, other than called line busy. Ring back, tone and ringing current are always transmitted from the called subscriber local exchange and busy tone and recorded announcements, if any, by the equipment as close to the calling subscriber as possible to avoid unnecessary busying of equipment and trunks. 126.96.36.199 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.
188.8.131.52 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. 184.108.40.206 Permanent Line (PG) Signal Permanent line or permanent glow (PG) signal is sent to the calling subscriber if he fails to release the call even after the called subscriber has gone on-hook and the call is released after a time delay. The PG signal may also be sent, in case the subscriber takes too long to dial. It is normally busy tone. 5.2.2 Called subscriber line signals. 220.127.116.11 Ring Signal On receipt of a call to the subscriber whose line is free, the terminating exchange sends the ringing current to the called telephone. This is typically 25 or 50Hz with suitable interruptions. Ring-back tone is also fed back to the calling subscriber by the terminating exchange. 18.104.22.168 Answer Signal When the called subscriber, lifts the hand-set on receipt of ring, the line impedance goes low. This is detected by the exchange which cuts off the ringing current and ringback tone. 22.214.171.124 Release Signal If after the speech phase, the called subscriber goes on hook before the calling subscriber, the state of line impedance going high from a low value, is detected. The exchange sends a permanent line signal to the calling subscriber and releases the call after a time delay, if the calling subscriber fails to clear in the meantime. 5.2.3 Register Recall Signal With the use of DTMF telephones, it is possible to enhance the services, e.g., by dialing another number while holding on to the call in progress, to set up a call to a third subscriber. The signal to recall the dialling phase during the talking phase, is called Register Recall Signal. It consists of interruption of the calling subscriber’s loop for duration less than the release signal. it may be of 200 to 320 milliseconds duration.
Inter-exchange signaling 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-ofband signaling). The signaling may be i. Pulsed
The signal is transmitted in pulses. Change from idle condition to one of active states for a particular duration characterizes the signal, e.g., address information ii. Continuous The signal consists of transition from one condition to another, a steady state condition does not characterizes any signal. iii. Compelled It is similar to the pulsed mode but the transmission is not of fixed duration but condones till acknowledgement of the receiving unit is received back at the sending unit. It is a highly reliable mode of signal transmission of complex signals. 5.3.1 Line signals 126.96.36.199 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. ii. 188.8.131.52 Circuit seizure/release corresponding to off/on-hook signal of the subscriber. Address information in the from of decade pulses.
In-Band and Out-of-Band Signals Exchanges separated by long distance cannot use any form of DC line signaling. Suitable interfaces have to be interposed between them, for conversion of the signals into certain frequencies, to enable them to be carried over long distance. A signal frequency (SF) may be used to carry the on/off hook information. The dialing pulses can also be transmitted by pulsing of the states. The number of signals is small and they can be transmitted in-band or out-of band. The states involved are shown in Table 1. TABLE 1. SINGLE FREQUENCY SIGNALING STATES TONE SIGNAL CONDITION State Idle (On hook) FORWARD Seizure(off hook) Release (on hook) BACKWARD Answer(off hook) Clear Back (on hook) Blocking (off hook) Forward On off on off off on On on off/on off on off Backward
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 endof 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. 184.108.40.206 E & M Signals E & M lead signaling may be used for signaling on per-trunk basis. An additional pair of circuit, reserved for signaling is employed. One wire is dedicated to the forward signals ((M-Wire for transmit or mouth) which corresponds to receive or R-lead of the destination exchange, and the other wire dedicated to the backward 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 Idle (On hook) FORWARD seizure (off hook) Release (On hook) BACKWARD Answer (off hook) Clear Back (On hook) Blocking Outgoing Exchange M- lead E-lead Earth Open Battery Earth battery battery Earth Open Earth/open Earth Open Earth Incoming Exchange M- lead Elead Earth Open Earth Battery/Earth Battery Earth Battery Earth Open Earth earth 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.
forward signal time signal cessation time recognition
signal cessation recognition
outgoing register incomming register
2-and-2only signal recognition acknowledgement backward signal and request for next signal compelled signal sequence
next forward signal acknowledgement backward signal
Fig.3. Compelled signalling procedure In other words, register signals are interchanged between registers during a phase between receipt of trunk seizure signal and the exchange switching to the speech phase. These signals are proceed-to-send (PTS) signals, address, signals, and signals indicating the result of the call attempt. The register signals may be transmitted in band or out of band. however, in the latter case, the signaling is relatively slow and only limited range of signals may be used. For example, a single out-of-band frequency may be selected and information sent as pulses. In-band transmission can be used easily as there can be no possible interference with the speech signals. To reduce transmission time and to increase reliability, a number of frequencies are used in groups. Normally 2 out of 6 frequencies are used. To make the system more reliable compelled sequence is used. Hence, this system is normally called compelled sequence Multi-frequency (CSMF) signaling as shown in Fig.3. In CCITT terminology it is termed as R2 system. As the frequencies need be transmitted only for a short duration to convey the entire information, the post dialling delay is reduced. When more than two exchanges are involved in setting up the connections the signaling may be done in either of the two modes i. End-to-end signaling The signaling is always between the ends of the connection, as the call progresses. Considering a three exchanges, A-B-C, connection, initially
the signaling is between A-B, then between A-C after the B-C connection is established. ii. Link-By-Link signaling The signaling is always confined to individual links. Hence, initially the signaling is between A-B, then between B-C after the B-C connection is established.
Generally supervisory (or line) and subscriber signaling is necessarily on link-by-link basis. Address component may be signalled either by end-to-end or link-by-link depending upon the network configuration. 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 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 Backward Index Weight Code 1380 1140 f0 0 1500 1020 f1 1 1620 900 f2 2 1740 780 f3 4 1860 660 f4 7
Signal 1 2 3 4 5 6 7
TABLE 4-FORWARD SIGNALS Weight Group I 0+1 Digit 1 0+2 Digit2 1+2 0+4 1+4 2+4 0+7 Digit3 Digit4 Digit5 Digit6 Digit7
Group II Ordinary subscriber Subscriber with priority Test / Mtce, equipment Spare STD Barred Spare CCB Changed Number to
8 9 10
1+7 2+7 4+7
Digit8 Digit9 Digit0
Operator Closed Number Closed Number Spare
Signal No. 1 2 3
TABLE 5 -BACKWARD SIGNALS Weight Code Group A 0+1 Send next digit 0+2 1+2 Restart Address complete, Changeover to reception of group B signals Calling line identification for malicious calls send calling subscribers category Set up speech connection Send last but 1 digit Send last but 2 digit Send last but 3 digit Spare
Group B Called line free with out metering Changed number Called line busy
4 5 6 7 8 9 10
0+4 1+4 2+4 0+7 1+7 2+7 4+7
Local congestion Number unobtainable called line fee, with metering Route congestion Spare Route Breakdown Malicious call blocking
Note : Signals A2, and A7 to A9 are used in Tandem working only. It can be seen from the tables that 1. Forward signals are used for sending the address information of the called subscriber, and category and address, information of the calling subscriber. 2. Backward signals are used for demanding address information and caller’s category and for sending condition and category of called line. R2 signaling is fully compelled and the backward signal is transmitted as an acknowledgement to the forward signal. This speeds up the interchange of information, reducing the call set up time. However, the satellite circuits are an 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. 2. 3. 4. 5. 6. 7. 8. 9. 10.
Originating exchange sends first digit Receipt of the digit is acknowledged by the terminating exchanges by sending A5 (demanding the caller’s category). A5 is acknowledgement by sending any11-1 to 11-5 by the originating exchange Terminating exchange acknowledges this by A1, demanding for next digit. Originating exchange, acknowledges A1 by sending any of 1-1to 1-10 sending the digit. The digits are sent in succession by interchange of steps v and vi. 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. This is acknowledgement by the originating exchange by sending the caller’s category again. The terminating exchange acknowledgements by sending the called line condition by sending any of B2 to B6. 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.
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.
Channel- Associated signalling
In the PCM systems the signalling information is conveyed on a separate channel which is rigidly associated with the speech channel. Hence, this method is known as channel associated signalling (CAS). Though the speech sampling rate is 8 Khz, the signals do not change as rapidly as speech and hence, a lower sampling rate of 500 Hz, for digitisation of signals can suffice. Based on this concept, TS 16 of each frame of 125 microseconds is used to carry signals of 2 speech channels, each using 4 bits. Hence, for a 30 channel PCM system, 15 frames are required to carry all the signals. To constitute a 2 millisecond multiframe of 16 frames. F 0 to F 15 TS 16 of the frame F 0 is used for multiframe synchronisation. TS 16 of F1 contains signal for speech channels 1 and 16 being carried in TS 1 and TS 17, respectively, TS16 of F2 contains signals of speech channels 2 and 17 being carried in TS2 and TS 18,
respectively and so on, Both line signals and address information can be conveyed by this method. Although four bits per channel are available for signalling only two bits are used. As the transmission is separate in the forward and backward direction, the bits in the forward link are called af and bf, and those in the backward link are called ab and bb. Values for these bits are assigned as shown in Table 6. As the dialling pulses are also conveyed by these conditions, the line state recognition time is therefore, above a threshold value. The bit bf is normally kept at 0, and the value 1 indicates a fault. However, the utilisation of such a dedicated channel for signalling for each speech channel is highly inefficient as it remains idle during the speech phase. Hence, another form of signalling known as common-channel signalling evolved. State Forward af Idle Seizure Seizure acknowledge Answer Clear Forward Clear Back 1 0 0 0 1 0 0 0 0 0 0 0 bf 1 1 1 0 0/1 1 ab 0 0 1 1 1 1 Bit Value Backward. bb
COMMON CHANNEL SIGNALLING SYSTEM No. 7 (CCS#7)
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 signalling systems have so far been used to carry the control information. Fault free operation is guaranteed with the channel-associated signalling 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 signalling systems. For this reason, a new, efficient signalling system is required in digital, processor-controlled communication networks. The CCITT has, therefore, specified the common channel signalling system no.7 (CCS-7). CCS-7 is optimised for application in digital networks. It is characterised by the following main features : • internationally standardized (national variations possible).
• suitable for the national, international and intercontinental network level. • suitable for various communication services such as telephony, text services, data services digital network (ISDN). • high performance and flexibility along with a future-oriented concept which well meet new requirements. • high reliability for message transfer.
• processor-friendly structure of messages (signal units of multiples of 8 bits). • signalling on separate signalling links; the bit rate of the circuits is, therefore, exclusively for communication. • • • signalling links always available, even during existing calls. use of the signalling links for transferring user data also. used on various transmission media • • • cable (copper, optical fiber) 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.
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 Signalling Points (SP) A distinction is made between signalling points (SP) and signalling transfer points (STP). The SPs are the sources (originating points) and the sinks (destination points) of signalling traffic. In a communication network these are primarily the exchanges. The STPs switch signalling messages received to another STP or to a SP on the basis of the destination address. No call processing of the signalling messages 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. 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.
Signalling Modes Two different signalling modes can be used in the signalling networks for CCS7, viz. associated mode and quasi-associated mode. In the associated mode of signalling, the signalling link is routed together with the circuit group belonging to the link. In other words, the signalling link is directly connected to SPs which are also the terminal points of the circuit group (See Fig.2). This mode of signalling is recommended when the capacity of the traffic relation between the SPs A and B is heavily utilized.
Fig. 2 Associated Mode of Signalling In the quasi-associated mode of signalling, the signalling link and the speech circuit group run along different routes, the circuit group connecting the SP A directly with the SP B. For this mode, the signalling for the circuit group is carried out via one or more defined STPs (See Fig. 3.3). This signalling mode is favourable for traffic relations with low capacity utilization, as the same signalling link can be used for several destinations.
Fig. 3 Quasi-associated Mode of Signalling 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.
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 quasiassociated) 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. 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 (b) (c) mode of signalling selection of the STPs signalling type (en block or overlap) assignment of the addresses to SPs.
signalling data links, e.g. 64 kbit/s digital or 4.8 kbit/s analog safety requirements load sharing between signalling links
diverting the signalling traffic to alternative routes in event of faults. (d) error correction
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 1 to 3 are allotted to the MTP while the UPs form level 4 (See Fig.3.4).
Fig. 4 Functional Levels of CCS7 Message Transfer Part (CCITT Blue Book Recommendations Q.701 to Q.707) 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. Fig. 5
Message exchange between two Signalling Points with CCS7 Functional Levels 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 (preferably with a bit rate of 4.8 kbit/s) can also be used via modems as a signalling data link. 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 retransmitting signal units. error rate monitoring on the signalling data link.
restoration of fault-free operation, for example, after disruption of the signalling data link. 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 change back to normal operation after the fault is corrected. The various functions of level 3 operate with one another, with functions of other levels and with corresponding functions of other signalling of other SPs. 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.
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. 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. 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. 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. 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. 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.
Fig. 6 Format of Various Signal Units Check bits (CK) : (16 bits) The CKs are formed on the transmission side from the contents of the SU and are added to the SUs as redundancy. On the receive side, the MTP can determine with the CKs whether the SU was transferred without any errors. The SUs acknowledged as either positive or faulty on the basis of the check. Fields specific to MSUs : Service Information Octet (SIO) : (8 bits) It contains the Service Indicator (SI, 4 bits) and Sub service 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, 5ISUP, 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. 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. Fields Specific to LSSUs 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-ofservice (011), Local processor outage (100) status, etc. Addressing of the SUs (in SIF) A code is assigned to each SP in the signalling network according to a numbering plan. The MTP uses the code for message routing. The destination of a SU is specified in a routing label. The routing label is a component of every user message and is transported in the SIF. The routing label in a MSU consists of the following (See Fig. 7).
Fig. 7 Routing Label of a Message Signal Unit Destination Point Code (DPC) : (14 bits) identifies the SP to which this message is to be transferred.
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. 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. 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). 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. 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 the transmit and receive sides with the bit pattern of the flags. Using the CKs also transmitted, error detection (level 2) checks whether the SU was correctly received. A fault-free SU is transferred to the receive control, while a faulty SU is discarded. The reception of a faulty SU is reported to error rate monitoring, in order to keep a continuous check on the error rate on the receive side of the signalling link. If a specified error rate is exceeded, this is reported to the signalling link status control by error rate monitoring. The signalling link status control then takes the signalling link out of service and sends a report to level 3. 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").
Fig. 8 Distribution of Functions in Message Transfer Part
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 preselection 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. Signalling Network Management The signalling network management is a function of level 3. It controls the operation and the interworking 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 interworking 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 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 management 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 non 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. change back; 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.
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
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 identiies 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.
6.THE 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 BILLING PULSE_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 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
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 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 all locations would occur in real time and provide users with instant access to all data. Implementation in BSNL
Centralised Billing & Acccounting System TCP/IP LAN | WAN or dialup CCT
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. Countrywide 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
7. ISDN 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. 1 The Network Configuration Without ISDN Fig.2 shows individual networks that will be fully integrated in the future.
Fig. 2 The Network Configuration With ISDN ISDN Definition
The CCITT defines the ISDN as follows : (1) A complete, terminal-to-terminal digital network. Fig.3 shows the end-to-end digital connectivity.
Fig. 3 End-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.
Fig. 4 Voice and Non-Voice Service in the Same Network (Example)
(3) (4) A network based on a digital telephone network. 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
(1) A wide range of services (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. The ISDN concept can be summarized by two statements : • • (2) ISDN offers a variety of services, such as telephone, data and image transmission through one network. ISDN handles all information digitally.
Standard user-network interface. Fig.8 shows the user-terminal/network interface.
Fig. 8 User-Terminal/Network Interface
The subscriber line is connected with an NT (Network Termination) installed at the customer premises. 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.
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. 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.
Fig. 9 shows operation of various terminals in the home.
Fig. 9 Operation 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. 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. 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.
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
Fig. 10 Home Shopping and Home Banking Service
• • (4) 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.
Home Medical System • • Fig.11 shows home medical system. Fig.11 shows another service provided by ISDN : the receiving of medical care at home.
Fig. 11 Home 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. An NT (Network Termination) is installed at the subscriber’s home and connected to the subscriber line.
Fig. 12 The Interface between the User
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 onpremises 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. 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. CCITT recommends the use of AMI (Alternative Mark Inversion) code at the S/T point. AMI code is a bipolar waveform. 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).
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.
User-Network Interface Reference Points
The following describes the user-access points and the function of each for basic usernetwork 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) : • 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.
• (b) (c) (2)
Data transmission speed can be selected.
Plural number of terminals can be concurrently connected. The portability of terminals can be ensured.
Basic Structure of User-Network Interface. 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.
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. 14 Multi 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.
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. 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. The D-channel also permits multiple logical channels to be established for use in packet communications.
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. Table 1 outlines these three channel types and characteristics.
Table 1 : Channel Types and Characteristics
Channel Type B Bit Rate 64 kbps • • D 16 kbps 64 kbps H H0 : 384 kbps H11 : 1536 kbps H12 : 1920 kbps
Function To carry user information Circuit switchingmode and packet switching mode To carry signaling information for circuit switching 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
• • •
H0 : 64 X 6 = 384 kbps H11 : 64 X 24 = 1536 kbps H12 : 64 X 30 = 1920 kbps
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.
Fig. 15 Basic 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.
Primary Group Interface Structure
(c) Table 2 shows the typical user network interface structure.
8. INTELLIGENT NETWORK
8.0 Overview of Intelligent Network Architecture Over the last thirty years, one of the major changes in the implementation of Public Switched Telephone Networks (PSTNs) has been the migration from analogue to digital switches. Coupled with this change has been the growth of intelligence in the switching nodes. From a customer's and network provider's point of view this has meant that new features could be offered and used. Since the feature handling functionality was resident in the switches, the way in which new features were introduced into the network was by introducing changes in all the switches. This was time consuming and fraught with risk of malfunction because of proprietary feature handling in the individual switches. To overcome these constraints the Intelligent Network architecture was evolved both as a network and service architecture. In the IN architecture, the service logic and service control functions are taken out of the individual switches and centralized in a special purpose computer. The interface between the switches and the central computer is standardised. The switches utilize the services of the specialized computer whenever a call involving a service feature is to be handled. The call is switched according to the advice received by the requesting switch from the computer. For normal call handling, the switches do not have to communicate with the central computer. 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 : • Rapid Service Deployment Most business today require faster response from their suppliers, including telecommunication operators. By separating the service logic from the underlying switch call processing software, IN enables operator to provide new services much more rapidly.
Reduced Deployment Risk Prior to IN, the risk associated with the deployment of new services was substantial. Major investments had to be made in developing the software for the services and then deploying them in all of the switches. With the service creation environment available, the IN services can be prototyped, tested and accessed by multiple switches simultaneously. The validated services can then be rolled out to other networks as well.
Cost Reduction Because the IN services are designed from the beginning to be reusable, many new services can be implemented by building on or modifying an existing service. Reusability reduces the overall cost of developing services. Also, IN is an architecture independent concept, i.e. it allows a network operator to choose suitable development hardware without having to redevelop a service in the event that the network configuration changes. Customization Prior to IN, due to complexity of switch based feature handling software, the considerable time frame required for service development prevented the provider from easily going back to redefine the service after the customer started to use it. With IN, the process of modifying the service or customization of service for a specific customer is much less expensive and time consuming. The customization of services is further facilitated by the integration of advanced peripherals in the IN through standard interfaces. Facilities such as voice response system, customized announcements and text to speech converters lead to better call completion rate and user-friendliness of the services.
Building upon the discussion in the previous section, one can envisage that an IN would consist of the following nodes :
• • •
Specialized computer system for – holding service logic, feature control, service creation, customer data, and service management. Switching nodes for basic call handling. Specialized resources node. The physical realization of the various nodes and the functions inherent in them is flexible. This accrues form the "open" nature of IN interfaces. Let us now look at the nodes that are actually to be found in an IN implementation.
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. 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: • • • • • • 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. Voice response system Announcements Voice mail boxes Speech recognition system Text-to-speech converters
The IN architecture is depicted in Fig.1
Fig. 1 IN Architecture 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.
CCF Call Control Function provides the basic facility for connecting the transport (e.g. speech). It involves the basic switching function and trigger function for handling the criteria relating to the use of IN. SSF Service Switching Function is used to switch calls based on the advice of the SCF at the SCP. This function provides a service independent interface. SCF It contains the service logic components and advises the SSF at SSP on further call handling. SDF Service Data Function contains the user related data and data internal to the network. SRF Specialized Resources Function covers all types of specialized resources other than the connection resources that are in the exchange (e.g. recorded announcements, tones, conference bridges, etc.). SCEF Service Creation Environment Function specifies, develops, tests and deploys the services on the network. SMAF Service Management Access Function provides an interface between service management function and the service manager who may be an operator. SMF Service Management Function enables a service to be deployed and used on IN. Fig. 2 depicts the distribution and interconnection of the various functional entities.
SMAF SMF SCEF SCF SDF SRF
Management interface In real time interface Signaling circuit interface
Fig. 2 Distributed Functional Entities
The distribution of functional entities over the physical entities and their interconnection is summarized in Table 1 and 2 below. It may be noted that all the physical entities may not be present in all INs as the choice of functional entities to be provisioned is entirely up to the service provider. Table 1 Distribution of FE's over PE's Physical Entity SSP SCP SMP IP Possible Functional Entities CCF, SSF, CCAF SCF, SDF SCEF, SMF, SMAF SRF
Table 2 FE-FE Relationship to PE-PE Relationship FE-FE SSF-SCF SCF-SDF SCF-SRF SRF-SSF PE-PE SSP-SCP SCP-SDP SCP-IP SCP-SSP-IP SSP-IP Protocol INAP, TCAP, SCCP and MTP X.25 or Proprietary INAP, TCAP, SCCP and MTP ISUP, INAP, TCAP, SCCP and MTP 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. (1) ABD – Abbreviated dialing The subscriber can register a short dialing code and use the same for access to any PSTN Number. (2) ACC – Account Card Calling • A special telephone instrument is required. • User dials an access code and gets acceptance tone. • Then he dials a PIN (personal identification no.) code and dials the called no. The Exchange reads the account number from card. • The Billing is debited to an account number (Telephone no.) as defined by the card. • In another variation of the service, the account number can be given through DTMF telephone instrument. • The follow-on feature facilitates the subscriber to dial another number without disconnecting the call and without need to dial PIN and account number again.
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. 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 Unconditional The subscriber can forward all incoming calls to a specified destination number. Optionally an alerting ring/reminder ring can be given to the forwarding subscriber whenever there is an incoming call. (6) CRD – Call Rerouting Distribution • Calls are rerouted as per conditions encountered, e.g. busy or no reply (time specified) or overload or call limiter. • Then as per selected condition the call is rerouted to predefined choice, e.g. paper, vocal box, announcement or queue.
(7) Completion of calls to busy subscriber The service cannot be fully implemented with CSI capability since the status of called party need to be known. • The calls are completed when subscriber who is busy becomes free. • On getting busy tone – user dials a code. • The user disconnects. • On called party becoming free, call is made by the exchange first to originating then to terminating subscriber (without any call attempt by the user). (8) CON – Conference Calling The service cannot be fully implemented with CSI capability. In adding or dropping the parties concerned it is not possible to check the authenticity of the parties. This service requires a special transmission bridge to allow conversation among multiple subscribers. CON-Add-ON-Conference Calling
• • • • • • (9)
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.
CCC – Credit Card Calling • The Credit Card Calling service allows subscribers to place calls from any normal access interface to any destination number and have the cost of these calls charged to account specified by the CCC number. • A special instrument is not required. The caller has to dial card number and PIN using DTMF instrument. • Follow-on feature may be provided optionally.
(10) DCR – Destination Call Routing The call is routed to destination pertaining to following conditions : • Time of day, day of week • Area of call originating • Calling identity of customer • Services attributes (non payment charges against subscriber) • Priority • Charge rates applicable for destination • Proportional routing of traffic • Optionally the subscribers can be provided with traffic details (11) FMD – Follow me Diversion • A subscriber can remotely control the call forwarding capabilities. • It can be done from any point in the network using a password. • It is required if subscriber moves from place to place in a day. • The service subscriber will pay for diverted portion of the call. 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. 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.
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. OCS – Originating Call Screening • This helps subscriber to screen outgoing call as per day and time. • The screening list may be managed by subscriber. • The restriction of screening list may be override by PIN or password. Three call cases are possible : Call screened and allowed Call screened and rejected Call passed by using override option 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. 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. 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. SPL – Split Charging • It allows service subscriber to share the call charges with calling party on per call basis.
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. TCS – Terminating Call Screening • The incoming calls are screened as per screening list. • Calls are allowed as per list and time of the day. UAN – Universal Access Number • National number is published by the subscriber. • The subscriber may specify the incoming calls to be routed to number of different destinations based on geographical locations of caller. 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. 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. 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.
The IN services can be broadly divided into three categories for charging purposes : No charging for calling user Charging of calling user as per local call Charging of calling user at higher rates No charging for calling user : FPH, VCC and VPN services fall under this category. Level ‘160’ is free at present and is proposed to be allotted to such services. Local exchanges need to analyse only ‘160’ and route the call to SSP. This level has to be created as charge free. New services of this type can be introduced in future without any requirement of further modification in local exchanges Charging of calling user as per local call : UN (local) falls under this category. Level ‘190’ is free at present and is proposed to be allotted to such services. Local exchanges need to analyse only ‘190’ and route the call to SSP. This level has to be created as local charge. New services of this type can be introduced in future without any requirement of further modification in local exchanges. Charging of calling user at higher rates : PRM and UN (long distance) falls under this category. Since the charging is at higher rate it is proposed that prefix ‘0’ may be used to have barring facility. Level ‘090’ may be used for such purpose. Local exchange will analyse ‘090’ and route the call to SSP. This level has to be created as ‘charge on junction pulses’. New services of this type can be introduced in future without any requirement of further modification in local exchanges. The access code of various IN services as proposed is as follows : No charging for calling user : FPH 1600 VCC 1601 Password change for VCC 1602 VPN 1603 Charging of calling user as per local call : UN (local) 1901 Televoting 1902 Charging of calling user at higher rates : PRM 0900 UN (Long distance) 0901
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