Você está na página 1de 54

SIGNALLING IN TELECOMMUNICATION

5.1 Introduction
A telecommunication network establishes and realizes temporary connections,
in accordance with the instructions and information received from subscriber lines
and inter- exchange trunks, in form of various signals. Therefore, it is necessary to
interchange information between an exchange and it external environment i.e.
between subscriber lines and exchange, and between different exchanges. Though
these signals may differ widely in their implementation they are collectively known
as telephone signals.

A signalling system uses a language which enables two switching equipments


to converse for the purpose of setting up calls. Like any other language. it possesses a
vocabulary of varying size and varying precision, ie. a list of signals which may also
vary in size and a syntax in the form of a complex set of rules governing the assembly
of these signals. This handout discusses the growth of signalling and various type of
signalling codes used in Indian Telecommunication.

Telephony started with the invention of magneto telephone which used a


magneto to generate the ringing current, the only signal, sent over a dedicated line
between two subscribers. The need for more signals was felt with the advent of manual
switching. Two additional signals were, therefore, introduced to indicate call request
and call release. The range of signals increased further with the invention of electro-
mechanical automatic exchanges and is still growing further at a very fast pace, after
the advent of SPC electronic exchanges.

The interchange of signaling information can be illustrated with the help of a


typical call connection sequence. The circled number in Fig. 1 correspond to the steps
listed below

i. A request for originating a call is initiated when the calling subscriber


lifts the handset.
ii. The exchange sends dial-tone to the calling subscriber to indicate to him to start
dialing.
iii. The called number is transmitted to the exchange, when the calling subscriber dials the
number.
iv. If the number is free, the exchange sends ringing current to him.
v. Feed-back is provided to the calling subscriber by the exchange by sending,

a) Ring-back tone, if the called subscriber is free(shown in fig.1)


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

vi. The called subscriber indicates acceptance of the incoming call by lifting the handset
vii. The exchange recognizing the acceptance terminates the ringing current and the ring-
back tone, and establishes a connection between the calling and called subscribers.
viii. The connection is released when either subscriber replaces the handset. When the
called subscriber is in a different exchange, the following inter-exchange trunk.
signal functions are also involved, before the call can be set up.
ix The originating exchange seizes an idle inter exchange trunk, connected to a digit
register at the terminating exchange.
x. The originating exchange sends the digit. The steps iv to viii are then performed to set
up the call.

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 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.

5.2.1.2 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 1477Hz

ABC DEF
097 Hz 1 2 3

GHI JKL MNO


170 Hz 4 5 6

PRS TUV WXY


7 8 9
662 Hz

OPER
0 #
41 Hz *
FIGURE 2. TONE-DIALLING FREQUENCY GROUPS.

5.2.1.3 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.

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) 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.

5.2.2.1 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.

5.2.2.2 Answer Signal


When the called subscriber, lifts the hand-set on receipt of ring, the line impedance
goes low. This is detected by the exchange which cuts off the ringing current and ring-
back tone.

5.2.2.3 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.

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. 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

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 frequency (SF) may be used to carry the on/off
hook information. The dialing pulses can also be transmitted by pulsing of the
states. The number of signals is small and they can be transmitted in-band or
out-of band. The states involved are shown in Table 1.

TABLE 1. SINGLE FREQUENCY SIGNALING STATES TONE SIGNAL


CONDITION

State Forward Backward


Idle (On hook) On On
FORWARD
Seizure(off hook) off on
Release (on hook) on off/on
BACKWARD
Answer(off hook) off off
Clear Back (on hook) off on
Blocking (off hook) on off

For in band signaling the tone frequency is chosen to be 2600Hz. or 2400 Hz. As the
frequency lies within the speech band, simulation of tone-on condition indicating end-
of call signal by the speech, has to be guarded against, for pre-mature disconnection.
Out-of- Band signaling overcomes the problem of tone on condition imitation by the
speech by selecting a tone frequency of 3825 Hz which is beyond the speech band.
However, this adds up to the hard-ware costs.

5.3.1.3 E & M Signals


E & M lead signaling may be used for signaling on per-trunk basis. An
additional pair of circuit, reserved for signaling is employed. One wire is
dedicated to the forward signals ((M-Wire for transmit or mouth) which
corresponds to receive or R-lead of the destination exchange, and the other
wire dedicated to the backward signals (E-wire for receive or ear) which
corresponds transmit or send wire or S-Lead of the destination exchange. The
signaling states are shown in table2.

TABLE 2. E & M SIGNALING STATES

State Outgoing Exchange Incoming Exchange


M- lead E-lead M- lead Elead
Idle Earth Open Earth Open
(On hook)

FORWARD Battery Open Earth Earth


seizure
(off hook)
Release Earth Earth/open Battery/Earth Open
(On hook)
BACKWARD
Answer battery Earth Battery Earth
(off hook)
Clear Back battery Open Earth earth
(On hook)
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.
forward
signal outgoing register
incomming register 2-and-2only
time signal recognition

signal cessation acknowledgement backward


time
recognition signal and request for next
signal
signal cessation
recognition
compelled signal sequence

next forward
signal
acknowledgement backward
signal

receiving

Sending

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 1380 1500 1620 1740 1860

Backward 1140 1020 900 780 660

Index f0 f1 f2 f3 f4

Weight Code 0 1 2 4 7

TABLE 4-FORWARD SIGNALS


Signal Weight Group I Group II
1 0+1 Digit 1 Ordinary subscriber
2 0+2 Digit2 Subscriber with
priority Test / Mtce,
equipment
3 1+2 Digit3 Spare
4 0+4 Digit4 STD Barred
5 1+4 Digit5 Spare
6 2+4 Digit6 CCB
7 0+7 Digit7 Changed Number to
Operator
8 1+7 Digit8 Closed Number
9 2+7 Digit9 Closed Number
10 4+7 Digit0 Spare

TABLE 5 -BACKWARD SIGNALS


Signal No. Weight Code Group A Group B
1 0+1 Send next digit Called line free with
out metering
2 0+2 Restart Changed number
3 1+2 Address complete, Called line busy
Changeover to
reception of group B
signals
4 0+4 Calling line Local congestion
identification for
malicious calls
5 1+4 send calling Number unobtainable
subscribers category
6 2+4 Set up speech called line fee, with
connection metering
7 0+7 Send last but 1 digit Route congestion
8 1+7 Send last but 2 digit Spare
9 2+7 Send last but 3 digit Route Breakdown
10 4+7 Spare Malicious call
blocking

Note : Signals A2, and A7 to A9 are used in Tandem working only.

It can be seen from the tables that


1. Forward signals are used for sending the address information of the called
subscriber, and category and address, information of the calling subscriber.
2. Backward signals are used for demanding address information and caller’s
category and for sending condition and category of called line.

R2 signaling is fully compelled and the backward signal is transmitted as an


acknowledgement to the forward signal. This speeds up the interchange of
information, reducing the call set up time. However, the satellite circuits are an
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
exchange
4. Terminating exchange acknowledges this by A1, demanding for next
digit.
5. Originating exchange, acknowledges A1 by sending any of 1-1to 1-10
sending the digit.
6. The digits are sent in succession by interchange of steps v and vi.
7. On receipt of last digit, the terminating exchange carries out group and
line selection and then sends A3, indicating switching over to group B
signals.
8. This is acknowledgement by the originating exchange by sending the
caller’s category again.
9. The terminating exchange acknowledgements by sending the called line
condition by sending any of B2 to B6.
10. In response to B6, the originating exchanges switches through the
speech path and the registers are released. Alternatively, in response to
B2 to B5, the registers are released and appropriate tone is fed to the
calling subscriber by the originating exchange.

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 signalling
In the PCM systems the signalling information is conveyed on a separate channel
which is rigidly associated with the speech channel. Hence, this method is known as
channel associated signalling (CAS). Though the speech sampling rate is 8 Khz, the
signals do not change as rapidly as speech and hence, a lower sampling rate of 500 Hz,
for digitisation of signals can suffice. Based on this concept, TS 16 of each frame of
125 microseconds is used to carry signals of 2 speech channels, each using 4 bits.
Hence, for a 30 channel PCM system, 15 frames are required to carry all the signals.
To constitute a 2 millisecond multiframe of 16 frames. F 0 to F 15 TS 16 of the frame
F 0 is used for multiframe synchronisation. TS 16 of F1 contains signal for speech
channels 1 and 16 being carried in TS 1 and TS 17, respectively, TS16 of F2
contains signals of speech channels 2 and 17 being carried in TS2 and TS 18,
respectively and so on, Both line signals and address information can be conveyed by
this method.

Although four bits per channel are available for signalling only two bits are used. As
the transmission is separate in the forward and backward direction, the bits in the
forward link are called af and bf, and those in the backward link are called ab and bb.
Values for these bits are assigned as shown in Table 6.

As the dialling pulses are also conveyed by these conditions, the line state recognition
time is therefore, above a threshold value. The bit bf is normally kept at 0, and the
value 1 indicates a fault.
However, the utilisation of such a dedicated channel for signalling for each speech
channel is highly inefficient as it remains idle during the speech phase. Hence, another
form of signalling known as common-channel signalling evolved.

State Bit Value


Forward Backward.
af bf ab bb
Idle 1 0 1 0

Seizure 0 0 1 0

Seizure 0 0 1 1
acknowledge

Answer 0 0 0 1

Clear Forward 1 0 0/1 1

Clear Back 0 0 1 1

5.4.2 COMMON CHANNEL SIGNALLING SYSTEM No. 7


(CCS#7)

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 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 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.

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
- 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 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, 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.
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-of-
service (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 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.

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 BILLINGÆPULSE_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 & Exchange
Acccounting
System
TCP/IP
RS232 Interface

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. 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

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) 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

(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.

(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.

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.

(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


Fig. 10
Home 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.
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.

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


the subscriber line.

Fig. 12
The 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).

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) :

• 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.

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. 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.

(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.

• 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.

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 • To carry signaling information for


circuit switching
64 kbps

H H0 : 384 kbps • To carry high-speed packet data such


as facsimile and video
H11 : 1536 kbps
• An H channel does not carry
H12 : 1920 kbps signaling information for circuit
switching by the ISDN

Note : K
• H0 : 64 X 6 = 384 kbps
K
• H11 : 64 X 24 = 1536 kbps
K
• H12 : 64 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.

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.

Fig. 16
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.

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.
• 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:

• Voice response system


• Announcements
• Voice mail boxes
• Speech recognition system
• Text-to-speech converters

The IP is switch based or is a specialized computer. It interfaces to the SSPs via ISDN
Primary Rate Interface or G.703 interface at which ISUP, INAP, TCAP, SCCP and MTP
protocols of the CCS7 protocol stack are defined.
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

SSF SSF

CCAF CCF CCF CCF CCAF

Management interface
In real time interface
Signaling circuit interface
Fig. 2 Distributed Functional Entities
The distribution of functional entities over the physical entities and their inter-
connection is summarized in Table 1 and 2 below. It may be noted that all the physical entities
may not be present in all INs as the choice of functional entities to be provisioned is entirely
up to the service provider.

Table 1
Distribution of FE's over PE's
Physical Entity Possible Functional Entities
SSP CCF, SSF, CCAF
SCP SCF, SDF
SMP SCEF, SMF, SMAF
IP SRF

Table 2
FE-FE Relationship to PE-PE Relationship
FE-FE PE-PE Protocol
SSF-SCF SSP-SCP INAP, TCAP, SCCP and MTP
SCF-SDF SCP-SDP X.25 or Proprietary
SCF-SRF SCP-IP INAP, TCAP, SCCP and MTP
SCP-SSP-IP ISUP, 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.

(1) ABD – Abbreviated dialing


The subscriber can register a short dialing code and use the same for access to any PSTN
Number.

(2) ACC – Account Card Calling


• A special telephone instrument is required.
• User dials an access code and gets acceptance tone.
• Then he dials a PIN (personal identification no.) code and dials the called no.
The Exchange reads the account number from card.
• The Billing is debited to an account number (Telephone no.) as defined by the
card.
• In another variation of the service, the account number can be given through
DTMF telephone instrument.
• The follow-on feature facilitates the subscriber to dial another number without
disconnecting the call and without need to dial PIN and account number again.
(3) AAB – Automatic Alternative Billing
• Call can be initiated by any user and any instrument.
• The call charges are billed in user’s account and that account need not be a
calling or a called party.
• The user first dials access code.
• Receives an announcement to dial account code and PIN (which is given by
management).
• The account code and PIN are validated to check its correctness and expired
credit limit.
• On getting acceptance tone the user dials the called number.
• In another variation of the service, the called party may be billed based on his
concurrence.

(4) CD – Call Distribution


• This service allows subscribers to have I/C calls routed to different destinations
according to allocation law specified by management (The Subscriber has
multiple installations).
• Three types of laws exist :
- Uniform load distribution
- % Load distribution
- Priority list distribution
• In case of congestion or fault the alternative over flow is specified.

(5) CFU – Call Forwarding Unconditional


The subscriber can forward all incoming calls to a specified destination number. Optionally an
alerting ring/reminder ring can be given to the forwarding subscriber whenever there is an
incoming call.

(6) CRD – Call Rerouting Distribution


• Calls are rerouted as per conditions encountered, e.g. busy or no reply (time
specified) or overload or call limiter.
• Then as per selected condition the call is rerouted to predefined choice, e.g.
paper, vocal box, announcement or queue.

(7) Completion of calls to busy subscriber


The service cannot be fully implemented with CSI capability since the status of called party
need to be known.
• The calls are completed when subscriber who is busy becomes free.
• On getting busy tone – user dials a code.
• The user disconnects.
• On called party becoming free, call is made by the exchange first to originating
then to terminating subscriber (without any call attempt by the user).

(8) CON – Conference Calling


The service cannot be fully implemented with CSI capability. In adding or dropping the
parties concerned it is not possible to check the authenticity of the parties. This service
requires a special transmission bridge to allow conversation among multiple subscribers.
CON-Add-ON-Conference Calling
• User reserves the CON resources in advance indicating date, time of
conference and duration.
• Controlled by user.
• In active phase of conference parties can be added, deleted, isolated again
reattached or split the group of parties.
• CON-Meet-ME – Conference calling meet me
• User reserve the resource same as 8A.
• Each participant dials a special number at specific time (specified at the time of
booking of conference) and reach the conference bridge.

(9) CCC – Credit Card Calling


• The Credit Card Calling service allows subscribers to place calls from any
normal access interface to any destination number and have the cost of these
calls charged to account specified by the CCC number.
• A special instrument is not required. The caller has to dial card number and
PIN using DTMF instrument.
• Follow-on feature may be provided optionally.

(10) DCR – Destination Call Routing


The call is routed to destination pertaining to following conditions :
• Time of day, day of week
• Area of call originating
• Calling identity of customer
• Services attributes (non payment charges against subscriber)
• Priority
• Charge rates applicable for destination
• Proportional routing of traffic
• Optionally the subscribers can be provided with traffic details

(11) FMD – Follow me Diversion


• A subscriber can remotely control the call forwarding capabilities.
• It can be done from any point in the network using a password.
• It is required if subscriber moves from place to place in a day.
• The service subscriber will pay for diverted portion of the call.

(12) FPH – Free Phone


• The called subscriber is charged for active phase of a call.
• For the calling user, no charging is done.
• The called subscriber can have multiple destinations and have DCR facility.

(13) MCI – Malicious Call Indication


• The subscriber requests the Administration to register his number for MCI.
• Administration registers the subscriber for MCI.
• The called subscriber (who has registered this service) invokes the service
during the active phase of the call if he feels that the call is malicious.
• The call is logged in the network with calling and called party number and
Date and time of invoking the service.
• Optionally, the network can log unanswered calls also.
• Optionally, the facility to HOLD the connection may be provided.
(14) MAS – Mass Calling
• It involves high volume of traffic.
• Calls can be routed to one or multiple destinations depending on geographical
location or time of day.
• Mainly used in Televoting.
• The network operator allots a service number.
• The user dials this number to register his vote.
• The user is played an announcement and asked to give his choice.
• At the end of the service, the network operator provides the call details and the
count on various preferences.
• After the service the same number can be reallocated to another subscriber.
• Calls made to this MAS number may be charged differently.

(15) OCS – Originating Call Screening


• This helps subscriber to screen outgoing call as per day and time.
• The screening list may be managed by subscriber.
• The restriction of screening list may be override by PIN or password. Three
call cases are possible :
- Call screened and allowed
- Call screened and rejected
- Call passed by using override option

(16) PRM – Premium Rate


• The local call is charged at a higher (premium) rate.
• This service is used by service providers for value added information services,
e.g. jobs, fortune, forecast, etc.
• The revenue is shared between network operator and service provider.
• 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.
• 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 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

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

Você também pode gostar