Escolar Documentos
Profissional Documentos
Cultura Documentos
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.
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.
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.
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.
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.
ABC DEF
097 Hz 1 2 3
OPER
0 #
41 Hz *
FIGURE 2. TONE-DIALLING FREQUENCY GROUPS.
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.
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.
5.3.1.1 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.,
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.
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.
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
next forward
signal
acknowledgement backward
signal
receiving
Sending
In other words, register signals are interchanged between registers during a phase
between receipt of trunk seizure signal and the exchange switching to the speech
phase. These signals are proceed-to-send (PTS) signals, address, signals, and signals
indicating the result of the call attempt.
The register signals may be transmitted in band or out of band. however, in the latter
case, the signaling is relatively slow and only limited range of signals may be used.
For example, a single out-of-band frequency may be selected and information sent as
pulses.
In-band transmission can be used easily as there can be no possible interference with
the speech signals. To reduce transmission time and to increase reliability, a number of
frequencies are used in groups. Normally 2 out of 6 frequencies are used. To make the
system more reliable compelled sequence is used. Hence, this system is normally
called compelled sequence Multi-frequency (CSMF) signaling as shown in Fig.3. In
CCITT terminology it is termed as R2 system. As the frequencies need be transmitted
only for a short duration to convey the entire information, the post dialling delay is
reduced.
When more than two exchanges are involved in setting up the connections the
signaling may be done in either of the two modes
i. End-to-end 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.
5.3.3 R2 Signalling
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.
Index f0 f1 f2 f3 f4
Weight Code 0 1 2 4 7
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.
Seizure 0 0 1 0
Seizure 0 0 1 1
acknowledge
Answer 0 0 0 1
Clear Back 0 0 1 1
Introduction
Communication networks generally connect two subscriber terminating equipment units
together via several line sections and switches for message exchange (e.g. speech, data, text or
images). Control information has to be transferred between the exchanges for call control and
for the use of facilities. In analog communication networks, channel-associated 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 :
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 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
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.
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.
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.
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.
6.2 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.
• Call Request
• Call Disconnect
• Calling number
• Called number
• Incoming circuit or Trunk identifier
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:
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:
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
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
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.
(4) A network that utilizes Signaling System No. 7 (SS7) for signaling between switching
systems. Fig. 5 shows the signaling connection between Switching Systems.
(5) A network offers standard user network interface. Fig.6 shows the standard user
network interface.
ISDN Services
(1) 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.
(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
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.
• 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).
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).
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
• 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.
• 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.
(d) S-Interface :
(e) T-Interface :
(f) R-Interface :
• A physical interface used for single customer terminator between TE2 and
TA.
(g) U-Interface :
•
ISDN User Network Interface Points
(1) Requirements 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 :
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.
(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.
(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.
Note : K
• H0 : 64 X 6 = 384 kbps
K
• H11 : 64 X 24 = 1536 kbps
K
• H12 : 64 X 30 = 1920 kbps
Fig. 15
Basic Interface Structure
• 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.
Fig. 16
Primary Group Interface Structure
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 :
• 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.
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.
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.
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:
Fig. 1
IN Architecture
SMF
SCEF
SCF SDF
SRF
SSF SSF
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.
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.
************