BHARAT SANCHAR NIGAM LIMITED
(A Govt. of India Enterprise)
SDE/AGM
LICE MATERIAL
AGM/DGM LICE
INDEX
1 SIGNALING IN PSTN2
2 NGN ARCHITECTURE
3 NGN PROTOCOLS (SIP/H.248/RTP/SIGTRAN)
4 CDOT MAX NG
5 C5 NGN LMG ARCHITECTURE
6 SSTP IMPLEMENTATION IN BSNL
7 IMS NETWORK ARCHITECTURE
1 SIGNALING IN PSTN
1.1 LEARNING OBJECTIVES
Types of signaling
Channel associated Signaling
Common Channel Signaling
Types of Signal Units
1.2 INTRODUCTION
A telecommunication network establishes and realizes temporary connections, in
accordance with the instructions and information received from subscriber lines and
interexchange 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.
Signalling has diversified area like land line telephone communication, mobile
communication or data communication. In this chapter, signaling in PSTN will be
discussed in brief.
1.3 TYPES OF SIGNALLING
a) Subscriber Line signaling
b) Inter-Exchange Signalling
Subscriber Line signaling
I. 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.
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.
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.
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.
Ring-back tone to the calling subscriber and ringing current to the called
subscriber, if the called line is free.
Busy-tone to the calling subscriber, if the called line is busy or otherwise
inaccessible.
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.
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. 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.
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.
II Called Subscriber Line Signaling.
1. Ringing 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.
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.
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.
4. 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.
1.4 INTER-EXCHANGE SIGNALING
The two type of digital inter-exchange signaling are channel associated signaling
and common channel signaling.
1.4.1 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 the Table.
As the dialling pulses are also conveyed by these conditions, the line state
recognition time is therefore, above a threshold value. The bit bf is normally kept at 0,
and the value 1 indicates a fault. However, the utilisation of such a dedicated channel for
signalling for each speech channel is highly inefficient as it remains idle during the
speech phase. Hence, another form of signalling known as common-channel signalling
evolved.
State Bit Value
Forward Backward.
af bf ab bb
Idle 1 0 1 0
Seizure 0 0 1 0
Seizure acknowledge 0 0 1 1
Answer 0 0 0 1
Clear Forward 1 0 0/1 1
Clear Back 0 0 1 1
1.4.2 COMMON CHANNEL SIGNALING SYSTEM No. 7 (CCS#7)
Signaling System No. 7 (SS7) is a signaling protocol that has become a worldwide
standard for modern telecommunications networks. SS7 is a layered protocol following
the OSI reference model .It enables network elements to share more than just basic call-
control information through the many services provided by the SS7's Integrated Services
Digital Network-User Part (ISUP), and the Transaction Capabilities Application Part
(TCAP). The functions of the TCAP and ISUP layers correspond to the Application Layer
of the OSI reference model, and allow for new services such as User-to-User signaling,
Closed-User Group, Calling Line Identification, various options on Call Forwarding and
the rendering of services based on a centralized database (e.g., 800 and 900 service). All
of these services may be offered between any two network subscribers. 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.
1.5 CC#7 SIGNALLING TERMINOLOGY
In CCS7 the signalling messages are sent via separate signalling links. 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.
Figure 1: Signaling via Common Channel Signaling link
1.5.1 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.
1.5.2 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.
1.5.3 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.
Figure 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. This signalling mode is favourable for traffic relations with low capacity
utilization, as the same signalling link can be used for several destinations.
Figure 3: Quasi-associated mode
1.5.4 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.
1.5.5 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.
The signalling functions in CCS7 are distributed among the following parts :
- message transfer part (MTP)
- function – specific user parts (UP)
The MTP represents a user-neutral means of transport for messages between the
users. The term user is applied here for all functional units which use the transport
capability of the MTP.
Each user part encompasses the functions, protocols and coding for the signalling
via CCS7 for a specific user type (e.g. telephone service, data service, ISDN). In this way,
the user parts control the set-up and release of circuit connections, the processing of
facilities as well as administration and maintenance functions for the circuits.
The functions of the MTP and the UP of CCS7 are divided into 4 levels. Levels to
3 are allotted to the MTP while the UPs form level 4 .
Figure 4: Functional Levels of CCS7
The message transfer part (MTP) is used in CCS7 by all user parts (UPs) as a transport
system for message exchange. Messages to be transferred from one UP to another are
given to the MTP (See Fig.5). The MTP ensures that the messages reach the addressed UP
in the correct order without information loss, duplication or sequence alteration and
without any bit errors.
Figure 5: Message exchange between two Signalling Points with CCS7
1.6 FUNCTIONAL LEVELS OF CCS 7
Level I (Signalling Data Link) defines the physical, electrical and functional
characteristics of a signalling data link and the access units. Level 1 represents the bearer
for a signalling link. In a digital network, 64-kbit/s channels are generally used as
signalling data links. In addition, analog channels (preferablywith a bit rate of 4.8 kbit/s)
can also be used via modems as a signalling data link.
Level 2 (Signalling Link) defines the functions and procedures for a correct exchange of
user messages via a signalling link. The following functions must be carried out at level 2
- delimitation of the signal units by flags.
- elimination of superfluous flags.
- error detection using check bits.
- error correction by re-transmitting signal units.
- error rate monitoring on the signalling data link.
- restoration of fault-free operation, for example, after disruption of the signalling data
link.
Level 3 (Signalling Network) defines the inter-working of the individual signalling
links. A distinction is made between the two following functional areas :
- message handling, i.e. directing the messages to the desired signalling line, or to the
correct UP.
- signalling network management, i.e. control of the message traffic, for example, by
means of changeover of signalling links if a fault is detected and changeback to normal
operation after the fault is corrected.
The various functions of level 3 operate with one another, with functions of other levels
and with corresponding functions of other signalling of other SPs.
1.7 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.
1.8 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.
Figure 6: Format of Various Signal Units
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.
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 Subservice field (SSF, 4 bits) whose last 2 bits are Network Indicator (NI).
An SI is assigned to each user of the MTP. It informs the MTP which UP has sent the
message and which UP is to receive it. Four SI bits can define 16 UPs (3-SCCP, 4-TUP,
5-ISUP, 6-DATAUP, 8-MTP test, etc.). The NI indicates whether the traffic is
international (00,01) or national (10,11). In CCS7 a SP can belong to both national and
international network at the same time. So SSF field indicate where the SP belongs.
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).
Figure 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).
Figure 8: Distribution of Functions in MTP
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. s
The check bit and flag generator (level 2) generates CKs for safeguarding against
transmission errors for the MUS and sets the flag for separating the SUs. In order that any
section of code identical to the flag (01111110) occurring by chance is not mistaken for
the flag, the user messages are monitored before the flag is added to see if five
consecutive ones (1) appear in the message. A zero (0) is automatically inserted after five
consecutive 1s. On the receive side, the zero following the five 1s is then automatically
removed and the user message thereby regains its original coding.
The check-bit and flag generator transfers a complete MSU to level 1. In level 1, the MUS
is sent on the signalling data link.
The bit stream along a signalling data link is received in level 1 and transferred to
level 2. Flag detection (level 2) examines the received bit stream for flags. The bit
sequence between two flags corresponds to one SU. The alignment detection (level 2)
monitors the synchronism of transmit and receive sides with the bit pattern of the flags.
Using the CKs transmitted, error detection (level 2) checks whether the SU was
correctly received. A fault-free SU is transferred to the receive control, while a faulty SU
is discarded. The reception of a faulty SU is reported to error rate monitoring, in order to
keep a continuous check on the error rate on the receive side of the signalling link. If a
specified error rate is exceeded, this is reported to the signalling link status control by
error rate monitoring. The signalling link status control then takes the signalling link out
of service and sends a report to level 3.
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").
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.
(a) Signalling Network Management
The signalling network management is a function of level 3. It controls the operation
and the inter working of the individual signalling links in the signalling network. To
this end, the signalling network management exchanges messages and control
instructions with the signalling links of level 2, sends message to the UPs and works
together with the signalling network management in adjacent SPs. For the inter
working with other SPs the signalling network management uses the transport
function of the MTP.
Management messages are transferred in MSUs like user messages. For
discrimination, the management messages have their own SI. The signalling network
management contains 3 function blocks :
(b) 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.
(c) The signalling route mangement controls and monitors the operability of
signalling routes. It exchanges messages with the signalling route management in the
adjacent STPs for this purpose. The signalling route management receives, for
example, messages concerning the failure or re-availability of signalling routes or the
overloading of STPs. In cooperation with the signalling traffic management, it
initiates the appropriate actions in order to maintain the signalling operation to the
signalling destinations involved.
(c) The signalling traffic management controls the diversion of the signalling traffic
from faulty signalling links or routes to fault-free signalling links or routes. It also
controls the load distribution on the signalling links and routes. To achieve this, it can
initiate the following actions :
- changeover; on failure of a signalling link the signalling traffic management
switches the signalling traffic from the failed signalling link to a fault-free signalling
link.
- changeback; when signalling link becomes available again after a fault has been
corrected, the signalling traffic management reverse the effect of the changeover.
- rerouting; when SP can no longer be reached on a normal route, the signalling traffic
management diverts the signalling traffic to a predefined alternative route.
When overloading occurs, the signalling traffic management sends messages to the users
in its own SP in order that they reduce the load. The management also informs the
adjacent SPs of the overloading in its own SP and requests them to also reduce the load.
The signalling traffic management accomplishes its functions by
- receiving messages from the signalling link and signalling route management.
- sending control instructions to signalling link and signalling route management.
- directly accessing the signalling links, e.g. during emergency alignment.
- modifying the message routing on failure of signalling routes.
- 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.
Figure 9: Message Transfer Part Users
1.9 MTP USERS
1.9.1 Telephone User Part (TUP)
Use of CCS7 for telephone call control signalling requires (i) application of TUP
functions, in combination with (ii) application of an appropriate set of MTP functions.
The TUP is one of level 4 users in CCS7. It is specified with the aim of providing the
same features for telephone signalling as other telephone signalling systems. It exchanges
signalling messages through MTP. Signalling messages contain information relating to
call set up and conditions of speech path. The TUP message consists of SIF and a SIO.
These signalling information are generated by the TUP of the originating exchange. The
label is 40 bits long, comprises DPC, OPC and CIC. CIC indicates one of the speech
circuit connecting the destination and originating points. Level 3 identifies the user to
which a message belongs by SIO, which comprises a SI and SSF. For TUP SI value is 4.
The SSF distinguishes the signalling message is for national or international network.
1.9.2 Integrated Services Digital Network User Part
The ISDN-UP covers the signalling functions for the control of calls, for the processing of
services and facilities and for the administration of circuits in ISDN. The ISDN-UP has
interface to the MTP and the SCCP for the transport of MSUs. The ISDN-UP can use
SCCP functions for end-to-end signalling.
1.9.3 Integrated Services Digital Network User Part
The integrated services digital network user part (ISUP) is the protocol which provides
the signalling functions required by CCITT No. 7 signalling to support basic bearer
services and supplementary services for voice and non-voice applications in an Integrated
Services Digital Network (ISDN).
The ISUP is suited for application in dedicated telephone and circuit-switched data
networks and in analogue and moved analogue/digital networks. In particular, the ISUP
meets the requirements defined by the CCITT for world-wide International semiautomatic
and automatic telephone and circuit-switched data traffic.
The ISUP can be used for national and international applications. The signalling
procedures, information elements and message type specified are for both applications.
Coding space has been reserved to allow national administrations and recognized private
operating agencies to introduce network specific signalling messages and elements of
information within the protocol structure.
The ISUP makes use of the services provided by the messages transfer part (MTP) (1)
and, in some cases, by the signalling connection control part (SCCP0 of CCITT No.7
signalling for the transfer of information between ISDN user parts.
Services Supported by the ISDN User Part
The ISUP protocol supports the basic bearer service; that is the establishment, supervision
and release of 64 kbit/s circuit-switched network connections between customer line
exchange terminations.
In addition to the basic bearer service the ISUP is expected to support (in the 1988
Recommendations) the following supplementary services :
Calling line identification (presentation and restriction).
Call forwarding,
Closed user group,
Direct dialling-in, and
User-to-user signalling.
1.9.4 Signaling Connection Control Part
The SCCP function is covered in ITU-T recommendations Q.711 to Q.714 and Q.716.
The signalling connection control part provides additional functions to message transfer
part for transfer of circuit related and non-circuit related signalling information and other
type of information between exchanges and other specialized centrals in
telecommunications network via SS#7 networks. The overall objective of SSCP is to
provide means for:
A transfer capability for signalling data units with or without the use of logical signalling
connections.
A logical signalling connection between two SCCP users with the SS#7 network.
Enhanced addressing capabilities.
The following figure illustrates the SCCP position in the SS#7 hierarchy:
Figure 10: SCCP
The functions of SCCP are used for handling transactions required by TCAP and also for
transfer of circuit related and call related signalling information for ISDN UP with or
without set up of end-to-end logical signalling connections.
The SCCP relies on the MTP to route the signalling information from one node to another
node. For this, it interacts with the user parts and with the MTP. Primitives are used to
convey information between the levels. Primitives are nothing but set of commands and
their respective responses associated with the services requested of the SCCP.
SCCP and OSI model
The SCCP enhances the services of MTP to provide the functional equivalent of Network
layer (i.e. layer #3 of OSI model). The MTP and the SCCP together is also referred to as
Network Service Part (NSP).
SCCP Addressing
The addressing capability of MTP is limited to delivering the message to a node
(identified by Network indicator and DPC) and to distribute it to a user using four bit
service indicator (octet SIO ). SCCP supplements this capability by providing an
addressing capability that uses DPC + SSN .The SSN is a local addressing information
used by SCCP to identify each of the SCCP users at a node.
SCCP provides enhanced addressing capability to MTP to enable it to address messages
with Global Title (GT). A Global Title is an address that does not explicitly contain
information usable for routing by MTP.
Figure 11: SCCP Addressing
SCCP provides enhanced addressing capability to MTP to enable it to address messages
with Global Title (GT). A Global Title is an address that does not explicitly contain
information usable for routing by MTP.
Global Title
The SS#7 signalling method identifies the destination and origination using signalling
point codes. Since a signalling point code has only fourteen bits. It is too small to be
uniquely addressed on a global scale. For this reason, signalling point codes are always
combined with a Network indicator- which means that a code is only valid in one
particular network.
To facilitate unambiguous global addressing, a unique international address or sender
information is necessary. The global address is known as The Global title and is sent in
the SCCP message in SS#7 messages. Since, SPCs are only ever valid in individual
networks, so called Global title translation must be performed at each relevant network
gateway. The SCCP performs the Global title translation, whereby an internationally
unique address (Global Title) is translated to an SPC and Network indicator in order to be
transferred to the network border or to a destination if it is located in the same network.
Global title translation is always used if no SPC for the destination is available at all. For
example, the HLR is to be identified on the basis of the IMSI.
SCCP Functional Units
The services supported by
SCCP are divided into two groups viz. Connection-oriented services and Connectionless
services. The protocols used for providing these services are divided in four classes; two
for connectionless services and two for connection-oriented services. Each protocol class
defines which level of services SCCP to provide. The four protocol classes are described
below:
Connectionless services
Class 0 – Basic connectionless
Class 1 – Sequenced connectionless
Connection oriented services
Class 2 – Basic connection oriented
Class 3 – Flow control connection oriented
The first two classes 0 and 1, support the connectionless environment, for example, for
use by TCAP. These are particularly suitable for frequent transmission of short messages.
As an example, to check validity of the credit card, an interrogation message can sent to a
data centre and reply received on the same route. The connectionless services are all that
is used in today’s networks. Classes 2 and 3 are used for connection-oriented services, for
example by ISDN–UP and, even though well-defined, are not used in today’s network.
The SCCP is divided into four functional units:
SCCP routing control (SCRC).
SCCP connectionless control (SCLC).
SCCP connection-oriented control.
SCCP management control (SCMG).
Figure 12: SCCP Functional Units
1.9.5 Transaction Capabilities Application Part (TCAP)
The TCAP recommendations are covered in ITU-T Q.771 to Q.775. The TCAP
portion of the CCS#7 protocol is used to transfer non-circuit related information between
two signalling points in the network. It is used to communicate between the SSP, SCP or
other SSPs through an exchange of TCAP messages. There is no setup of speech/data
channel connections.
Non-circuit related information would be such things as data queries for services
(1600) where there is not a physical end-to-end connection between the signalling points.
TCAP supports the exchange of non-circuit related data between applications across the
SS7 network using the SCCP connectionless service. Queries and responses sent between
SSPs and SCPs are carried in TCAP messages. For example, an SSP sends a TCAP query
to determine the routing number associated with a dialed 1600 number and to check the
personal identification number (PIN) of a calling card user. In mobile networks (IS-41
and GSM), TCAP carries Mobile Application Part (MAP) messages sent between
mobile switches and databases to support user authentication, equipment identification,
and roaming.
Applications for the TCAP
In mobile networks to report the location of a mobile network subscriber to the
home exchanges. In credit card service to check the validity and to execute account
transactions.
Functions of TCAP
TCAP supports real-time remote operations and is structured in two sub-layers:
i) Component sub-layer, dealing with individual actions called components.
ii) ii)Transaction sub-layer,dealing with the exchange of messages containing
components
The component sub-layer is above the transaction sub-layer.
The TCAP layer interfaces directly with SCCP layer.
A component consists of a request to invoke an operation.
An invocation of the operation is identified by a Component ID.
Components are passed individually between TCAP users.
Component
ASE1 ASE2
TC-primitives TC primitives
TCAP messages
TCAP-A TCAP-B
N-primitives N-primitive
SCCP-A SCCP-B
MTP-primitives MTP-primitives
MTP-A MTP-B
MSUs
Figure 13: Messages and message paths
The originating TC user may send several components to the component sub-layer before
they are transmitted in a single message to the remote end. At the remote end each one is
delivered individually to the destinating TC-user. Successive component exchanged
between TC-users in order to perform an application constitute a dialogue. The
component sub-layer allows several dialogues to run concurrently between two TC-users
each being identified by a particular ID.
TCAP serves all ( application specific0 ASEs in a node.To send a message an
ASE passes a series of TC requests to TCAP and TCAP passes the message to SCCP.
When a TCAP receives a message from its SCCP it passes the contents to the destination
ASE in its node.
1.10 CONCLUSION
CCS 7 messages are exchanged over out-of-band (OOB) signaling links, so the
voice carrier channel is isolated from the signaling channel. Hence out-of-band signaling
channel is inherently more secure and reliable than a channel that contains the voice (or
data) and the signal and improves control over fraudulent network use. It facilitates user
services like number portability, toll-free calling, and enhanced features like Caller ID.
2 NGN ARCHITECTURE
2.1 LEARNING OBJECTIVES
NGN – Vision and Definition
NGN Architecture
Protocols used in NGN
NGN Implementation in BSNL
2.2 INTRODUCTION
Telecommunication industry is changing at a rapid pace. This change in the
industry is basically driven by demand of new services from subscriber's side and urge to
reduce capex (capital expenditure) and opex (operational expenditure) from carrier side.
Today all most all telecommunication giants are installing and maintaining at least three
kinds of basic network.
PSTN: public switch telephone network was basically developed and engineered for
giving voice connectivity to the wire line subscribers. The network consists of
local exchange/rsu as a part of access network and taxs as a part of core network.
Already huge amount of money has been invested in pstn setup. Because of tough
competition from mobile & voice over ip, it is becoming white elephant day by
day for the operators. Another fact about pstn is that most of its equipment are
going to exhaust their lives in coming years.
PLMN: (public land mobile network): PLMN has been developed to provide voice
services for wireless subscribers. Though in recent times sms has emerged as
killer application for mobile. Plmn includes bts/bsc as access network and msc as
a core network.
DATA network: this network was basically designed for accessing remote files and
servers for defense people and universities but now a days nobody can think of
living with data network services. The basic and most popular application of data
networks is internet. Other applications include e-commerce, online banking,
online gaming, e-shopping, iptv video on demand and many more. Data network
is an assembly of routers, which are responsible for forwarding information from
one end to other.
The interesting fact about the current generation is that these networks
have been developed during different time zones. That’s why they are separate
network infrastructure. There is no sharing of infrastructure among them.
However some gateways are available for inter network communication.
Another disadvantage of the current scenario is that all the three networks
are having their own service platforms in other words services are tightly coupled
with their networks because of that carrier or operators have to introduce service
separately for separate networks.
Because all the three networks are having separate access transport and switching
network service provider has to invest in all the three networks separately. Hence CAPEX
increases on the other hand for maintenance of three different networks operational cost
also increases. Manpower of the company has to have knowledge of multiple
technologies.
2.3 NGN VISION
Next Generation Network is the framework where operator will have a common
transport network based on Internet Protocol for providing all kinds of telecommunication
services. Hence operators will have to install and maintain only a single network which
will reduce its CAPEX and OPEX significantly. Moreover service provisioning will
become easier because of the introduction of new and intelligent servers. NGN is able to
provide Vendor independence because of the standard protocols it uses for interaction
with network elements.
2.4 NGN DEFINITION
A Next Generation Network (NGN) is a packet-based network able to provide
Telecommunication Services to users and able to make use of multiple broadband, QoS-
enabled transport technologies and in which service-related functions are independent of
the underlying transport-related technologies. It enables unfettered access for users to
networks and to competing service providers and services of their choice. It supports
generalised mobility which will allow consistent and ubiquitous provision of services to
users.
2.5 GENERALISED MOBILITY
At present subscribers are enjoying terminal mobility where network
identification system is available in the form of SIM and the same is inserted in the
terminal. If user is having that terminal he will be mobile with the identity of the SIM.
In NGN subscriber can have Generalised mobility. Here each individual will have its
own network identity in the form of "SIPURL: xyz @ domain name.com". Users have to
make registration from his devices against the given URL. Registrar servers of the
company will maintain bindings with URL and physical location of registered devices.
Users can register for more than one device at a time. With this subscribers need not to
depend upon terminal they can login with any device enabled with required protocols
(SIP) if they want and call will come to that device.
2.6 NGN ARCHITECTURE
NGN is a layered architecture consisting of transport, access, control and
application layer. It is important to note that all the layers are independent from each
other. Change in one layer should not affect other layers.
2.6.1 Access Layer
Access Layers is responsible for direct subscriber attachment function. NGN can
support all kind of existing access as well as upcoming access. In fact NGN does not
matter about type of access. NGN is capable of processing traffic originated from PSTN,
GSM, CDMA, xDSL, WiMAX or any other access system. Depending upon the type of
access, protocol conversion and/or media conversion may be required at the NGN
Gateways.Access Layer consists of Gateways. Example of getaways are media Gateway,
Access gateway. Signalling gateway.
Media gateway terminates media, coming from PSTN/PLMN in E1 / STM. Here
it is responsible for packetisation of media under the instruction of control layer. After
packetisation of information it throws packets to the transport Network. Access gateway
is nearer to subscriber. Subscriber can directly be terminated in Access Gateway. All the
required configuration of such subscribers should be done at control layer. Access
Gateway and Media Gateways are responsible for carriage of Media whereas Signalling
gateway is carrying signalling generated by PSTN and informs Control Layer about the
signalling in required format.
2.6.2 Transport layer
Transport Layer of NGN is based on IP. It can utilize the advantage of MPLS.
Transport Layer forms the core of the Network. It basically consists of Routers, which are
responsible for carrying traffic originated by access layer. As the same core network is
going to be used for all kinds of subscribers enjoying different kind of real time and non
real time services, it should be able to make use of band width policies and Qos policies.
Operator has to think of managed Network for its subscribers. It is basically an assembly
of routers connected with optical network. Traffic coming from gat ways is properly
routed by those routers.
2.6.3 Control layer
It is responsible of call setup, routing and charging policies and other controls in
NGN environment. It consists of call servers where all information of the network
resides. These call servers are responsible for setting up, modifying, charging and tear
down of the calls. NGN may work on soft switch principle. It consists of MGC (Media
Gateway Controller) as an overall controller and MGs (Media Gateway) for termination
of traffic. MGC is basically a server and it is having all the necessary information of
network MGC instructs MGs for establishing the call. Under the control of MGC, MG
performs different call related tasks such as connection, modification and termination of
media streams, packetisation of media etc.
2.6.4 Application layer
It is responsible for OSS/BSS. Enhanced services to the subscribers will be
provided with the help of application servers. It may include prepaid servers,
announcement servers, Service servers etc. Hence NGN is making service separation
from Network. Any service can be introduced with the help of server at any time without
any modifications in the control, transport or access.
Figure 14: NGN Architecture
2.7 PSTN VERSUS NGN:
Figure 15: PSTN Versus NGN
As shown in above figure PSTN switch consists of interface, switching and call
control. All the functional entities are shown in one box that means they are interacting
with each other using proprietary protocol. Where as in NGN model entities are
interacting using standard protocols.
In PSTN each node should have call control separately whereas NGN may have
centralised call control.
PSTN is dedicated network for providing voice services to the subscribers
whereas NGN is developing with the idea of carrying all kind of traffic over it.
PSTN is working on circuit switched principle whereas NGN is working on
packet switching.
PSTN provides excellent quality of voice and it is tested in all conditions whereas
NGN will provide good quality of voice and it is to be tested in adverse network
conditions.
In PSTN service integration is very difficult and because of vendor dependent
technologies it is difficult to introduce services easily. Whereas NGN shall be able to
provide separate service platform for introduction of services without depending upon
underlying network related technologies.
2.8 NGN IMPLEMENTATION IN BSNL
After the introduction of NGN it was a huge task to migrate all the TDM
exchanges into IP based NGN Network with seamless running of services. Also with the
increased traffic handling capacity of NGN network the whole network had to be
migrated and redeployed in a manner to decrease CAPEX.
In this exercise BSNL has adopted phase wise methodology to migrate all its
exchanges to NGN Network and had given tenders to various equipment vendors.
2.9 MIGRATION STRATEGY
As far as migration of PSTN towards NGN is concerned, two types of migration
scenarios could be foreseen based either on a replacement or on an overlay strategy:
(i) Replacement strategy
It consists of replacing traditional PSTN equipment by next generation equipment.
This can be done in the core of the network in order to provide further capacity and to
enable a better utilization of the network resources, or/and at the edges of the network, in
order to provide advanced services to the customer.
(ii) Overlay Strategy
With an overlay strategy, the NGN network will integrate current circuit-switched
and packet-switched technologies. The modern packet-based overlay network will
provide advanced services whilst the circuit switched PSTN network will continue to
provide basic telephony services. Both networks will remain interconnected via gateways
as required by specific types of services (e.g. VoIP call originated from an IP phone and
terminated on the PSTN, or Internet data traffic originated from the PSTN). When the
overlay network eventually becomes capable of providing sufficient QoS, all traffic
could be diverted from the circuit switched PSTN to the packet-switched overlay
network.
The strategy adopted by BSNL would be the overlay one as it has a huge base of
circuit switched network that will coexist with packet switched network for a
considerable period of time. The migration steps would be as follows :
• Introduce IP in Transit network at Level-1 TAX locations (IP TAX Project) -
Class 4 NGN
• Extend IP network to Level-2 TAXs and large scale implementation in Access
Network. – Class 5 NGN
• Develop MPLS core at Circle and LDCA Level.
• Offer Voice and Multimedia services to Broadband Subscribers using DSL,
Optical Ethernet technologies.
IP Tax Project is the 1st step towards NGN. The name given to this project has been IP
TAX Project and is a class4 NGN implementation. The equipment for IP Tax is provided
by M/s ZTE.
2.9.1 Components of IP TAX Network:
1. Media Gateway:
The media gateway (MG) supports packetized voice and the interface to whatever
medium the voice is to be transported on. The MG performs the task of packetizing voice
and providing connections from switched circuits (TDM) to packetized circuits (IP,
Frame Relay, or ATM). The MG does not necessarily reside at a subscriber premise, but
interfaces to equipment located at the subscriber premise. The MG is responsible for
media conversion, resource allocation and resource management, and event notifications.
It is responsible for reporting events to the media gateway controller (MGC) within its
zone. Following functions are performed by Media Gateway:
• Media Conversion
The MG must be able to provide conversion from TDM circuit-switched
connections to ATM, IP, or Frame Relay connections. This includes the packetization of
the voice itself, media processing including transcoding, conferencing, interactive voice
recognition, and other audio resource functions.
• Resource Allocation
Resource allocation includes the reservation and release of all resources. It is
important to understand that although the MG is responsible for resource allocation and
management, it does so under the direction of the MGC. The MGC holds the ultimate
responsibility of defining what resources are to be allocated for a particular call. The MG
is capable of providing either point-to-point connections or point-to-multipoint
connections (such as in a conference call). The MG must support all types of media viz.
voice, data, video, facsimile etc.
• Event Notification
The MG must also maintain the state of all resources and report the state to the
MGC. If a particular resource fails, the MG reports the failure to the MGC. The MGC
maintains a state table for all resources within the MGs in its zone.
2 Softswitch (Media gateway controller) :
The MGC is responsible for controlling all the MGs within its zone. As previously
mentioned in the description of the MG, the MGC performs a number of call control
functions that are typically found in the call control portion of a legacy switch. This
includes determining and controlling signals used on specific circuits, such as ringing and
busy tones. The MGC also determines the QoS required for any one connection at the
MG. In short, the call handling is performed at the MGC, rather than at the MG itself.
This enables carriers to deploy less expensive equipment at the edge of their networks.
3 Signaling gateways :
The IETF defines the signaling gateway (SG) as being the bridge to the PSTN. It
supports STP functions to the network. The SG should be capable of providing
conversions between SS7 addresses (point codes) and IP addresses. One method of
address resolution utilizes the point code of an incoming message from the PSTN, along
with the circuit identification code (CIC), to determine the appropriate IP address of the
receiving MG. If a message is TCAP, the address resolution is provided using the point
code and subsystem number.
Using IP in signaling networks rather than TDM point-to-point connections has a
number of advantages. One of the most significant advantages is a more efficient (and
less expensive) use of bandwidth. With TDM circuits, a lot of ancillary equipment is
required to maintain the circuit. With IP networks, the equipment is much less expensive.
The addressing in IP signaling networks provides far more flexibility than in conventional
SS7 networks. All entities in the SS7 environment are addressed through the use of point
codes. The point code administrator in each country issues point codes. When a carrier
uses an IP network, the entities in the IP network are addressed by IP addresses rather
than point codes. This requires the use of an SG to resolve the addresses from the SS7
network to the addresses in the IP network. The routing is based usually on a destination
point code and, in the case of ISUP messages. When a database requires additional
servers to increase capacity, an additional point code is not required (as is the case in
conventional SS7 networks). The new server can be added using IP addressing, and the
routing assignments can be made in the SG to ensure load sharing and subsystem
management. One of the biggest advantages to launching services in IP networks is the
flexibility provided to the carrier. Carriers can develop their own applications deployed
on carrier-grade servers. Already many carriers have begun developing creative new
services themselves, rather than depending on vendors. This factor alone will greatly
enhance the concept of IP networking and provide great benefits to new carriers looking
to differentiate themselves from the competition. Of course, as is the case with traditional
signaling networks, the signaling network should be deployed first, providing an
infrastructure that will enable the carrier to expand into new markets quickly and launch
exciting new services to address the needs of their customers.
4 Announcement server:
Announcement server is within the IP domain itself and contains all the
announcements required for various purposes.
2.9.2 Scope of IP TAX project
This project was allocated to ZTE and as per the Solution provided by them, 3
types of sites are built according to different requirements of TMG capacity, and
corresponding application scenarios, etc.
• Primary NOC Site
• Primary Site
• Secondary Site
BSNL’s 2052 KC MPLS Based IP TAX Network, which has been deployed
throughout the country consist of the following products.
• ZTE Soft switch ZXSS10 SS1b,
• Trunk Media Gateways ZXSS10 MSG 9000(TMG)
• Announcement Server ZXSS10 MSG 9000
2.9.3 Soft switch Control: ZXSS10 SS1b
The Soft switch control device ZXSS10 SS1b mainly carries out the functions of
call control, signaling process, resource management, accounting management, user
management and protocol adaptation within its own domain, and uses 100M Ethernet
interface to connect to the data network. Soft switch can support maximum load of traffic
2M BHCA/shelf without extension. When extension shelves (max.8) are present it can
support Maximum traffic load of 16M BHCA . Billing records are stored at 3 levels.
Maximum capacity of trunk is 200,000 DS0/shelf.
2.9.4 Trunk Media Gateway: ZXMSG 9000
The ZXMSG 9000(Trunk Media Gateway) is located on the core layer for
connecting No.7 trunk users and PRI users. It connects the PSTN subscriber to the
NGN to implement the conversion between voice/fax on the PSTN/ISDN trunk side and
voice/fax on the IP network side. The ZXMSG 9000 can provide the functions of TG, SG
and AG through different board and software configurations. When serving as TG, the
ZXMSG 9000 is responsible to access PSTN to IP core network through trunk line and
convert the voice/fax between PSTN/ISDN trunk side and IP network. It supports 5,600
E1 as Trunk Gateway.
2.9.5 Announcement Server: ZXMSG 9000
ZXMSG 9000 can be configured as announcement server, it is capable to provide
sufficient announcement resources for all TMG under its control by interlocking with
control device via ZXSS1b.
2.9.6 Network Management: ZXNMS
The softswitch integrated network management is developed independently by
ZTE, which implement unified network management for softswitch product and relevant
devices of ZTE. It can provide centralized management of facilities (Softswitch, TMG,
SG, Data devices etc) with unified customer's interface, and can provide management
interface for devices of other manufacturers.
Every Site connects to MPLS/IP core packet network via a LAN Switch. Different types
of sites consisting of the Soft switch control device, the service platform and network
management system will be constructed that is responsible for the call and service control
and network management of the whole network.
2.10 NOC SOLUTION
The architecture of network management system is divided into typical 3 layers.
From bottom to top, they are Network Element Layer (NEL), Element Management
Layer (EML) and Network Management Layer (NML). Different layers focus on
different perspectives of network management. The expansion in each layer will not have
impact on others and provides the flexibility.
Figure 16: Primary NOC Site
The main NOC location is in Chennai for managing the whole NGN
Network. Its Disaster Recovery NMS (DR Site) has been established at Delhi.In
NOC location, Raid technology, server redundancy are deployed for high stability
and expandability. Also, dual Ethernet switch, router, and firewall are utilized for
networking reliability. Between the NOC and Disaster Recovery Site, the Disaster
Backup Plan is used.
Figure 17: Primary Site
Figure 18: Secondary Site
2.11 CLASS 5 NGN IMPLEMENTATION:
For Class 5 implementation of Access equipment BSNL has adopted two different
approaches:
• Soft Switch based
• IMS based
In Soft Switch based approach BSNL has given tender to M/s CDOT and for IMS based
approach tender is given to M/s Huawei with additional capacity tender given to M/s
UTSTARCOM.
2.11.1 SoftSwitch Based Class 5 Implementation.
The project is executed by M/s CDOT which includes equipment planning for
CORE,ACCESS and NOC.
Zone Name Primary DR Subscriber NEBS
Softswitch Softswitch Ultimate Compliant
Site Site Capacity Chassis/
(Main + DR) Server
North Zone Gurgaon Chandigarh 33Lacs + 5 / 48
33Lacs
East Zone Kolkata Cuttack 16Lacs + 5 / 48
16Lacs
South Zone Bangalore Hyderabad 44Lacs + 5 / 48
44Lacs
West Zone Pune Bhopal 29Lacs + 5 / 48
29Lacs
Table 1. CDOT MAX NG Core Location
The table above summarizes the CORE locations divided zone wise with capacity of each
zone. The zones were divided in Primary site and DR site for redundancy purpose.
Each of the core site is having following configuration:
• Racks (+1 Spare Rack) will be installed at every core site.
• 1 Chassis (12 HP servers) on each rack will be installed at every core site. (Total
48 servers)
• 1 Spare Chassis at each Core site.
• All servers will be loaded with Linux Operating system and the C-DOT
Application Software as per the requirement.
The 12 HP servers in each chassis is loaded with the software with specific functionality
namely:
• SBC: Session Border Controller. It is the entry point for the core. All the access
equipments will be aggregated and connected to SBC of the core location. It acts as a
firewall for the CDOT NGN.
• C5 Soft switch: It is having different elements inside it namely Database server,
Feature server, Media server, billing server and call processing engine.
• C4 Soft switch: It is responsible for routing calls from/to CDOT NGN to
PLMN/PSTN. It supports MEGACO,SIP protocols for routing signalling messages.
• RATER Engine: For processing of the CDR received from C5 and to push it
towards billing server.
• SG Server: For equipping SS7 and SIGTRAN protocol stacks. It helps in
converting SS7 to SIGTRAN and vice versa.
• NGEMS: For NMS functionality to support O&M of core elements.
• LAGEMS: It provides NMS support for all the LAGU access units. Backup and
restoration of LAGU can be done through LAGEMS.
• LIS Server: For monitoring of calls.
• INSSP: For routing calls which requires trigger towards SCP database.
For Equipment planning at the Access site the inputs taken are:
• Equipped Capacity
• Working Capacity
• No. of co-located BM
• No. of Trunk BM
• No. of Remote BM
• Total No. of AN-RAX and WLL parented on V5.2 Protocol
• Total No. of PRI Connection
There are namely two access components which are part of CDOT MAX NG:
1. LAGU: Line access gateway unit
Table below shows equipage for a single LAGU unit:
Table 2. LAGU Configuration
Fully Equipped Line Access Gateway Unit (LAGU) System
Hardware Modules / Items Qty.
C-DOT Assembled Board
NGTK Card Line Access Gway Appl. With Intg. EEB 02
module (NGTJ-A00)
Mother Board/Chassis
MAX-NG LAG Frame Assy. (2-Slot) Per Unit Qty.
MAX-NG LAG Frame Assy. (2-Slot) 1
Frame Support Assy. 2
Mounting Angle(LAG 6 Slot) 2
Support Bracket (LAG 2-Slot) 1
MB Stiffener Assy. (2 Slot) 1
MAX-NG Motherboard (2-Slot) 1
Accessories/Cables Items/Tools etc.
External Cables Set for 2-Slot Chassis Per Unit Qty.
RS232 Monitor Cable Assy. 1
Internal Cables Set for 2-Slot Chassis Per Unit Qty.
TUB-TSI Power Cable Assy. 2
Internal Chassis Ground Cable Assy. 2
Sys. Internal Chassis-Gnd Cable Assy. 1
I/P Power-MMT Cable Assy. 2
A single LAGU unit can replace a BM along with two LM working in concentration
mode. The NGTK card is responsible for processing and converting analogue voice
traffic from ATU’s of BM to RTP packets. The LAGU unit is connected to CACU via
E1’s or over Ethernet.
2. CACU: Central Access Control Unit:
Table below shows the equipage of a CACU unit.
Table 3. CACU Configuration
Fully Equipped Line Access Gateway Unit (LAGU) System
Hardware Modules / Items Qty.
C-DOT Assembled Board
02
NGTK Card Line Access Gway Appl. With Intg.
EEB module (NGTJ-A00)
Mother Board/Chassis
MAX-NG LAG Frame Assy. (2-Slot) Per Unit Qty.
MAX-NG LAG Frame Assy. (2-Slot) 1
Frame Support Assy. 2
Mounting Angle(LAG 6 Slot) 2
Support Bracket (LAG 2-Slot) 1
MB Stiffener Assy. (2 Slot) 1
MAX-NG Motherboard (2-Slot) 1
Accessories/Cables Items/Tools etc.
External Cables Set for 2-Slot Chassis Per Unit Qty.
RS232 Monitor Cable Assy. 1
Internal Cables Set for 2-Slot Chassis Per Unit Qty.
TUB-TSI Power Cable Assy. 2
Internal Chassis Ground Cable Assy. 2
Sys. Internal Chassis-Gnd Cable Assy. 1
I/P Power-MMT Cable Assy. 2
Following cards are present in CACU unit:
• NGTK: It is equipped with gateway functionality namely PRIAG, V5AG, MG,
SG.
• EBM: For converting E1 traffic to Ethernet and vice versa.
• MLS: It acts as a L3 switch and is used for aggregating all the LAGU to CACU.
• SLM: It is a shelf monitoring card for the CACU unit.
Presently only PRI is supported by CDOT MAX NGN. All the existing ANRAX
are terminated to NGTK card over V5.2 protocol.MG & SG gateways are used for
interworking with PLMN/PSTN and works under the control of C4 soft switch located at
core.
For every Co-located Line BM and Remote BM there is requirement of LAGU
system. For every MBM system there is requirement of one CACU system.
Equipment planning of NOC is also done.
Location of NOC
i. Gurgaon
ii. Bangalore
ONE Rack with 12 HP servers containing C-DOT proprietary data provisioning
software will be loaded.
2.12 IMS BASED APPROACH.
New Technology switches by IMS Class 5 NGN. These are provided by M/s
Huawei and M/s UTStarcom. The IMS based project is implemented in two parts
consisting of Package-1 IMS core elements and Package-2 Access Elements. As is done
in case of CDOT MAX NGN, IMS is also deployed in core and access parts with core
located in four zones.
The Table below summarizes the location of CORE equipments and its capacity.
ZONE PR Site DR Site Subscriber
Capacity
NORTH Chandigarh Lucknow 800000
SOUTH Hyderabad Bengaluru 1500000
EAST Bhubhaneshwar Kolkatta 500000
WEST Ahmedabad Pune 1200000
Table 4. IMS Core Sites
Figure 19: Core location of IMS
Figure 20: Huawei IMS solution to BSNL
An additional tender for 2.4 Mn lines is given to M/s UTStarcom, it has deployed
core equipment with PR at Chandigarh and DR at Hyderabad site.
Package 2 Access Equipments are supplied by M/s Huawei, M/s ZTE and M/s
UTStarcom.
The Access equipment called as LMG or Line Media Gateway is a replacement to
all the TDM based local exchanges. These LMG’s are equipped with both voice and
ADSL functionality thus eliminating the use of DSLAM’s. The subscriber can avail voice
as well as internet services from same equipment.
For providing PRI/BRI facility separate IAG equipment is being provided.
Three Different types of LMG’s in BSNL:
1. UTStarcom LMG: B1205 and B 1205E (E for extended )
B1205 LMG is having capacity of 256 lines and B1205E LMG is having capacity
of 512 lines. Both the LMG’s are having common control and network cards the only
difference being in the number of line cards it supports. The number of voice line cards
and ADSL cards are in the ratio of 4:1 means for every four voice cards one ADSL cards
will be equipped. The slots for voice/ADSL cards are universal.
2. Huawei LMG:
The UA5000 Universal Access Unit (referred to as the UA5000) is a device for
accessing both the narrowband and broadband services.
With the increasing demands for telecom services, such as data, video, voice, and
multimedia services, the requirement for an access network that provides large-capacity,
high-rate, and high-quality integrated services has become imperative.
Besides quality traditional voice and broadband access services, the UA5000 provides
subscribers with perfect IP-based voice services and multimedia services
3. ZTE (Vayam Tech ) LMG:
ZTE LMGs come in different capacities viz. LMG 128 ports, LMG 256 ports,
LMG 512 ports, LMG 1024 ports, LMG 2048 ports. For LMG capacity of 256 and less
ZTE ZXA10 350M architecture is used.
In all cases LMG’s can be cascaded to increase the capacity.
2.13 CONCLUSION
With the implementation of NGN in BSNL network, BSNL has been able to
reduce expenditure on O&M of the Legacy systems. Also NGN allows us to offer newer
services fast as compare to the legacy systems. More over with the introduction of all IP
based systems in NGN architecture the providers are able to offer quality services to its
customer. The NGN technology is an access agnostic frame work which reduces the
limitation of terminal type and vendor dependability.
3 NGN PROTOCOLS (SIP/H.248/RTP/SIGTRAN)
3.1 LEARNING OBJECTIVE
In this chapter the reader will get acquainted with various protocols that are used in
the NGN Architecture. The reader will have the overview of SIP, H.248 , RTP and
SIGTRAN protocols.
3.2 INTRODUCTION
The NGN devices exchange messages among themselves with the help of various
protocols. These protocols are responsible for call control, media control and event
related data which are generated and are necessary to be captured during the different
phases of telephony services. These protocols are responsible to delivers services to the
end customers based on the service type offered by these protocols. Some of the protocols
which are widely used are SIP, H.248, SIGTRAN and RTP/RTCP.
As NGN Architecture is a layered architecture, there are different entities working in
different layers. At each layers these entities perform specific functions and support
specific set of protocols. These protocols are divided according to the functionality which
they support and as summarized below:
1. Call Control protocols: SIP (IETF), H.323 (ITU-T), BICC (ITU-T)
2. Media Gateway Control Protocols: H.248(ITU-T) /Megaco (IETF)
3. Signalling transport protocols: UDP(IETF), TCP(IETF) and SCTP(IETF)
4. Media Transport Protocols: RTP/RTCP(IETF) over UDP(IETF)
The figure below shows the NGN protocol family
Figure 21: NGN Protocols
3.3 MEGACO/H.248 PROTOCOL
The Megaco/H.248 protocol is used between the SoftSwitch control equipment
and various media gateways. Two abstractions are defined in the protocol, i.e.
Termination and Context.
A Termination sources and/or sinks one or more streams. In a multimedia
conference, a Termination can be multimedia and sources or sinks multiple media
streams. The media stream parameters, as well as modem, and bearer parameters are
encapsulated within the Termination.
A Context is an association between a collection of Terminations. There is a
special type of Context, the null Context, which contains all Terminations that are not
associated to any other Termination. For instance, in a decomposed access gateway, all
idle lines are represented by Terminations in the null Context.
There are eight commands defined in H.248. They are Add, Modify, Subtract, Move,
Audit Value, Audit Capability, Notify and Service Change.
A group of commands between the MG and the SoftSwtich control equipment
implement the transaction interaction. One event interworking can be identified by a
TransactionID. Transactions consist of one or more Actions. An Action consists of a
series of Commands that are limited to operating within a single Context. Consequently
each Action typically must specify a ContextID. Figure below shows a graphic
representation of the Transaction, Action and Command relationships.
Figure 22: Transaction, Command and Action relationship
Transactions guarantee ordered Command processing. It means: the commands in
an event interworking are executed in order. Ordering of Transactions is NOT guaranteed
- transactions may be executed in any order, or simultaneously.
At the first failure of a Command in a Transaction, the processing of the remaining
Commands in that Transaction will stop.
Applications SHOULD implement an application level timer per transaction. Expiration
of the timer should cause a retransmission of the request. Receipt of a Reply should
cancel the timer. Receipt of Pending should restart the timer.
3.4 SIGTRAN PROTOCOL
The SIGRAN protocol stack transmits the SCN Signaling in the IP network,
supports the interface with the standard primary language of the SCN Signaling
application without any modification, thus to guarantee that the existent SCN Signaling
application can be put into use without modification. Meanwhile, it utilizes standard IP
transmission protocol as the lower layer transmission Signaling, and satisfies the
requirement by SCN Signaling transmission through adding the function of the protocol.
Figure below shows the model of a universal Signaling transmission function
Figure 23: SIGTRAN Functional Model
In the model, the Signaling gateway terminates the Signaling in the circuit
switching network, and then transmits the Signaling content to the Media Gateway
Controller (MGC) for processing through SIGTRAN. MG terminates the inter-office
trunk and controls the trunk according to the control instruction from MGC.
SIGTRAN protocol is composed of three parts: Signaling adaptation layer, Signaling
transmission layer and IP transmission layer.
The Signaling adapting layer provides the standard primary language interface
with SCN Signaling, the Signaling transmission layer provides the real-time reliable
transmission service required by SCN Signaling, and the bottom layer is the standard IP
protocol layer. Figure below shows the model of SIGTRAN protocol.
Figure 24: SIGTRAN Protocol Stack Model
3.5 SIP-SESSION INITIATION PROTOCOL
SIP has been chosen as the session control protocol for the Next Generation
Network This protocol was developed by IETF. The prime function performed by this
protocol is to establish and manage multimedia sessions over IP networks. SIP is a
complete end to end signaling protocol. It is a text based protocol and so is easier to
extend, debug and can be used to develop services. The main functionalities of SIP are
Session establishment, Modification and Termination.
The users in SIP are identified by the URLs. The format used by SIP to address
SIP entities is identical to the one used on the web and by e-mail systems.
Mohit Sharma
Telephone : +91 761 2604100
Mobile : +91 9425001232
Fax : +91 761 2601458
Email :
[email protected] Mohit Sharma
SIP : Mohit.sharma @ bsnl.com
SIP reduces the amount of contact information
Figure 25: SIP URL examples
Having a SIP URL that can be redirected to any other means of communication
considerably reduces the amount of different contact information needed by an
individual, currently, an average business card contains at least a fixed number, a mobile
no, , a fax no. and an e-mail address. SIP distills all this to a single URL.
SIP protocol defines several entities and it is necessary to understand their role
inside SIP architecture.
1. User Agents: A user agent is an end system that acts on behalf of someone
who wants to participate in calls. In general, a user agent contains both a
protocol client – called a user agent client (UAC) and a protocol server - called
a user agent server (UAS). The UAC is used to initiate a call and the UAS is
used to answer a call.
All interaction between users and SIP protocol are mediated by UAs. SIP UAs
are implemented on top of many systems. They can run, for instance in a
computer as one among many applications, or they can be implemented in
dedicated device, such as SIP phone.
2. Network Servers: There are three types of network servers
a. Proxy Server: Proxy servers make subsequent attempts on behalf of the
user, rather than sending new contact information to the user. It is the first
point of contact in the network.
b. Redirect Server: Redirect servers do not forward the request on behalf of
UA’s but sends back the location information to the UA. Thus they help
locate SIP UAs by providing alternative locations where the user can be
reachable.
c. Registrar Server: Registrar refers to a SIP server accepting registrations.
The contact information is then made available to other SIP servers within
the same administrative domain. A registrar is usually co-located with a
redirect server or a proxy server.
3. Location Server: They are not SIP entities. A location server stores and returns
the possible locations for users. Most registrars upload location updates to a
location server.
3.6 RTP/RTCP
RTP, the Real Time Transport Protocol, has gained widespread acceptance as the
transport protocol for voice and video on the Internet. Its companion control protocol, the
Real Time Control Protocol (RTCP), is used for loose session control, QoS reporting, and
media synchronization, among other functions.
The real-time transport protocol (RTP), provides end-to-end delivery services for
data with real-time characteristics, such as interactive audio and video. Those services
include payload type identification, sequence numbering, time stamping and delivery
monitoring. .Applications typically run RTP on top of UDP to make use of its
multiplexing and checksum services; both protocols contribute parts of the transport
protocol functionality.
Note that RTP itself does not provide any mechanism to ensure timely delivery or
provide other quality-of-service guarantees, but relies on lower-layer services to do so. It
does not guarantee delivery or prevent out-of-order delivery, nor does it assume that the
underlying network is reliable and delivers packets in sequence. The sequence numbers
included in RTP allow the receiver to reconstruct the sender's packet sequence, but
sequence numbers might also be used to determine the proper location of a packet, for
example in video decoding, without necessarily decoding packets in sequence.
RTP, consisting of two closely-linked parts:
The real-time transport protocol (RTP), to carry data that has real-time properties.
The RTP control protocol (RTCP), to monitor the quality of service and to convey
information about the participants in an on-going session.
3.7 CONCLUSION
By the virtue of these protocols which are based on open interfaces, it is very easy
to deploy new services onto the existing network without much capex involved.
4 CDOT MAX NG
4.1 LEARNING OBJECTIVE
This chapter lays down the foundation for implementation of CDOT MAX NG.
The chapter covers the parent DSS Family with its introduction and the need for MAX
NG and changes in CDOT for migrating to MAX-NG network.
4.2 INTRODUCTION
C-DOT DSS MAX is a universal digital switch which can be configured for
different applications as local, transit, or integrated local and transit switch. High
traffic/load handling capacity up to 8,00,000 BHCA with termination capacity of 40,000
Lines as Local Exchange or 15,000 trunks as Trunk Automatic Exchange, the C-DOT
DSS family is ideally placed to meet the different requirements of any integrated digital
network.
The design of C-DOT DSS MAX has envisaged a family concept. The advantages
of family concept are standardized components, commonality in hardware,
documentation, training, installation and field support for all products and minimization
of inventory of spares. In fact this modular design has been consciously achieved by
employing appropriate hardware, software, and equipment practices.
The equipment practices provide modular packaging. Common cards and advanced
components have been used in the system hardware in order to reduce the number and
type of cards. Standard cards, racks, frames, cabinets and distribution frames are used
which facilitate flexible system growth. Interconnection technology has been standardized
at all levels of equipment packaging. All these features, together with ruggedised design,
make C-DOT DSS MAX easy to maintain and highly reliable.
4.3 BASIC GROWTH/BUILDING MODULES
C-DOT DSS MAX exchanges can be configured using four basic modules
Base Module
Central Module
Administrative Module
Input Output Module
i) BASE MODULE
The Base Module (BM) is the basic growth unit of the system. It interfaces the
external world to the switch. The interfaces may be subscriber lines, analog and digital
trunks, CCM and PBX lines. Each Base Module can interface upto 2024 terminations.
The number of Base Modules directly corresponds to the exchange size. It carries out
majority of call processing functions and, in a small-exchange application, it also carries
out operation and maintenance functions with the help of the Input Output Module.
In Single Base Module (SBM) exchange configuration, the Base Module acts as an
independent switching system and provides connections to 1500 lines and 128 trunks. In
such a configuration, the Base Module directly interfaces with the Input Output Module
for bulk data storage, operations and maintenance functions. Clock and synchronization is
provided by a source within the Base Module. It is a very useful application for small
urban and rural environments.
With minimum modifications in hardware through only one type of card, a Base Module
can be remotely located as a Remote Switch Unit (RSU), parented to the main exchange
using PCM links.
ii) CENTRAL MODULE
Central Module (CM) consists of a message switch and a space switch to provide
inter-module communication and perform voice and data switching between Base
Modules. It provides control message communication between any two Base Modules,
and between Base Modules and Administrative Module for operation and maintenance
functions. It also provides clock and synchronization on a centralized basis.
iii) ADMINISTRATIVE MODULE
Administrative Module (AM) performs system-level resource allocation and
processing function on a centralized basis. It performs all the memory and time intensive
call processing support functions and also administration and maintenance functions. It
communicates with the Base Module via the Central Module. It supports the Input Output
Module for providing man- machine interface. It also supports the Alarm Display Panel
for the audio-visual indication of faults in the system.
iv) INPUT OUTPUT MODULE (I0M)
Input, Output Module (IOM) consists of duplicated Input Output Processor
(IOP). The Input Output Processor (IOP) is a general-purpose computer with UNIX
Operating System. It is used as the front-end processor in C-DOT DSS. It handles all the
input and output functions in C-DOT DSS. The IOP is connected to AP/BP via HDLC
links. During normal operation, two IOP’s interconnected by a HDLC link, operate in a
duplex configuration. Working as front-end processor, it provides initial code down load
to the subsystems, man machine interface and data storage for billing and other
administrative information.
IOP interfaces various secondary storage devices like' disk drives, cartridge tape
drive and floppy drive. It supports printers and upto 8 serial ports for video display units
which are used for man- machine communication interface. All the bulk data processing
and storage is done in this module
Thus, a C-DOT DSS exchange, depending upon its size and application, consists of
Base Modules (maximum 32), Central Module, Administrative Module, Input/Output
Module and Alarm Display Panel. The Base Modules can be remotely located or co-
located depending on the requirement.
4.4 REMOTE SWITCH UNIT
Remote Switch Unit (RSU) is an integral part of C-DOT DSS architecture. In
order to realise a RSU, the normal BM can be modified for remoting with the host
exchange via 2 Mbps digital links. The number of 2 Mbps links between the Main
Exchange and RSU is primarily determined by the traffic. A maximum 16 PCMs can be
provided between a RSU & Main exchange. Analog and Digital trunk interfaces are also
implemented in RSU to support direct parenting of small exchanges from RSU itself
instead of parenting it to the main exchange which will ultimately save the media required
from main exchange. As far as call processing is concerned, RSU is an autonomous
exchange capable of local-call completion. Operation and maintenance functions are
handled by the host exchange. In the event of failure of PCM links, RSU goes into
standalone mode of operation. In case it is not possible to process a call request due to
unavailability of links to the host, the subscriber is connected to appropriate tone or
announcement.
During standalone mode of operation, the local and incoming terminating calls in
RSU are switched and the metering information of all the RSU subscribers is stored in the
RSU. It is sent to the host whenever the PCM links are available again.
Only the even numbered BMs can be configured as RSU i.e. a maximum 16 RSUs
are possible in C-DOT DSS MAX-XL and 8 RSUs in MAX-L.
4.5 SYSTEM FEATURES
MAX
Main Automatic Exchange MAX is expandable to large capacities of order of
2000 lines or beyond. The MAX may have Remote Modules (RM) and Remote Line
Concentrators (RLC) connected to it.
RAX
Rural Automatic Exchange (RAX) is a small exchange and is expandable upto
2000 lines capacity. Single Base Module configuration (i.e. CDOT SBM RAX with or
without concentration) comes under the RAX category.
4.5.1 Types Of Application
The system can be put to the following applications:
Replacements
The exchange can serve as replacement of an existing switching system due to be
phased out from the network.
New Exchanges
Wherever new exchanges are opened, the CDOT DSS MAX can provide the
switching network within the existing telecom network.
Extensions
The capacity of an existing CDOT switching system can be increased. For example
if the capacity of an existing CDOT exchange is 512 points, it can be increased, to say,
4000 lines.
TYPE OF SYSTEM
The system is Stored Programme Controlled (SPC) which makes it possible to work
in attended/non-attended type of working environment.
TYPE OF NETWORK
The switching network within the system is 4-wire digital.
TYPE OF COMPONENTS
The different type of components used include integrated circuits, miniature
relays, PCB, etc. The connecting scheme between various modules emphasis
connectorised hardware.
4.6 WHY MAX-NG?
Consolidation of Data and Voice-Converged Services & Networks
Less Infrastructure cost in installation & expansion
Better enhanced services such as Video calling, Centrex, Multilingual
announcements etc.
Distributed architecture & Centralized Control
More calls with less bandwidth
Lower costs per call, especially for long-distance calls
Involves more of software, simplified hardware infrastructure and changing
upgrade economics, hence ease in upgradation
Remote support
Taking care of Component obsolescence
Figure 26: Options for Upgrading to All-IP
4.7 DSS MAX UPGRADE PROCEDURE
Retain and convert line termination units to VoIP Access Gateways(LAG)
o Convert trunk interfaces to VoIP Trunk Media Gateways(MG)
o Convert the SS7 unit to a Signalling Gateway with STP functionality(SG)
o Retain the Integrated Local and Tandem (ILT) functionality by moving
Class 4 and Class 5 switching functions to an external softswitch (C4/C5)
o Replace internal media and control paths with IP-enabled routing
o Add on broadband interfaces for copper and wireless access.
4.8 MAX-NG COMPONENT
CORE COMPONENT
ACCESS COMPONENT
The MAX-NG network is structurally divided into the following four sub-network
segments
a. NGN Core Network for delivery of Services
b. NGN Access Network consisting of MAX exchanges upgraded to MAXNG
exchanges
c. Operations Support Network consisting of
• i. MAX-NG Billing System Interface
• ii. MAX-NG Service Provisioning Interface
• iii. MAX-NG Network Management
d. The existing IP/MPLS based NIB/NIB-II Transport Network
The NGN Core Network handles
All Session Establishment, Call Processing and Service
Delivery functions centrally.
Figure 27: MAX NG Architecture
4.8.1 Core Component
The core network consists mainly of following servers.
• C5 Softswitch
• C4 Softswitch
• Session Border controller (SBC)
• Rating Engine
• SG Server
• EMS Server
• LIS Server
4.8.1.1 C5 Softswitch
A central device in a telecommunications network which connects telephone
calls from one phone line to another, entirely by means of software running on a server.
SIP protocol is used to establish calls.Currently in MAX N/W switching is carried out by
hardware, with physical switchboards to route the calls.
• Soft switch
– Class 5 SS – Call Agent, Caller Server.
– Class 4 SS – Media Gateway Controller
4.8.1.2 C4 Softswitch
Class 4 soft switch is used to deliver calls of MAX-NG subscribers to
PLMN/PSTN network via Signaling and Media Gateway and vice-versa MGC perform
call control and signaling routing for PSTN and Signaling System 7 (SS7) voice traffic.
MGC uses a Signaling Gateway (SG) to get SS7 signaling over IP, and provides ISUP to
SIP and back translation. MGC controls Media Gateway (MG) using MEGACO/H.248 to
translate the voice/media call component from TDM to RTP and back.
4.8.1.3 Session Border controller (SBC)
A Session Border Controller (SBC) is a network function which secures voice
over IP (VoIP) infrastructures while providing interworking between incompatible
signaling messages and media flows (sessions) from end devices or application servers.
Along with processing signaling messages, Session Border Controllers also handle all
media traffic, typically in the form of RTP.
4.8.1.4 Rating Engine
The Rating engine is a system used to determine the customer chargeable units
based on:
Type of the day
Time of day
Call duration
4.8.1.5 SG Server
A Signaling Gateway is a network component solely responsible for translating
signaling messages between one medium (usually IP) and another (PSTN) .
4.8.1.6 EMS Server
EMS (Element Management System) is a GUI based solution for configuring and
monitoring the different systems installed as MAX-NG components.
4.8.1.7 LIS Server (Lawful Interception System)
Monitoring the calls of a particular subscriber.
The calls are diverted to another number as configured
4.8.2 ACCESS COMPONNET
Figure 28: MAX to MAX NG
4.8.2.1 LINE ACCESS GATEWAY UNIT(LAGU)
C-DOT LAGU is designed to migrate fixed line TDM based technology of DSS-
MAX to IP based technology.
It converts basic voice into RTP and Signaling information to SIP and connects to
the IP network.
LAGU provides gateway functionality through a pair of network gateway cards
working in active-standby mode.
LAGU systems are installed at all co-located and remote Base Modules of the
MAX exchanges.
A BM system is converted to LAGU system replacing BPU and TSU with NGTJ
cards and TSI cards.
All TUs with PSUs, TCs (LCC), TIC, SPC and TUI cards are retained.
2-slot chassis consists of NGTJ card and TSI card interfacing with TUI cards in
ATU frame.
Supports maximum 32 E1s from RBM site to co-located BM site.
LAGU Chassis
Figure 29: LAGU Chassis
LAGU chassis contains two NGTJ & two TSI cards.
TSI cards interface with TUI cards in ATUs through TT cables.
NGTK variants are NGTK-A00 (EEB integrated) and NGTK-B00 (EEB not
integrated).
LAGU Chassis operates at -48V DC.
5VDC & 12 VDC for LAGU is fed from ATU.
4.8.2.2 Central Access Control Unit(CACU)
KEY COMPONENTS IN CACU CHASSIS
The 6-slot MAX-NG Card Frame (CACU) variant consists of the
following components:
◦ Four Universal slots
◦ Pair of redundant aggregation slots
◦ Pair of redundant Shelf Management and Controller slots
◦ Pair of redundant -48V power feeds
◦ A FAN tray at the bottom
◦ Temperature sensor card with digital temp sensing for each slot
4.8.2.3 CARDS INSTALLED IN CACU CHASSIS
Shelf Manager (SLM) cards: A pair of SLM cards will be installed in the two
slots at the lower left side of CACU Chassis.
NGTKcards: NGTK cards can be installed in any of the first four universal slots.
These cards work in 1+1 configuration.
EBM cards: EBM cards can be installed/jacked in into any of the first four
universal slots. These cards work in 1+1 configuration.
MLS cards: A pair of MLS cards can be installed in the last two slots.
TSN cards : Temperature sensor cards
Figure 30: CACU unit
4.9 CONCLUSION
CDOT MAX NG is a cost effective and easy to migrate solution developed by
CDOT. It uses the existing infrastructure at the subscriber termination point thereby
reducing the time for migration to an IP based NGN Network.
5 CLASS 5 LMS ARCHITECTURE
5.1 LEARNING OBJECTIVES
● Hardware components of UTStarcom LMG.
● Functionality of different modules in LMG.
● Initial configuration of LMG.
5.2 INTRODUCTION
B1205 is a cost effective & versatile Digital Loop Carrier (DLC) and Multi
Service Access Node (MSAN) optimal solution from UTStarcom that provides a unique
set of capabilities enabling service providers to deliver the most competitive triple play
service offerings. It enables service providers to smoothly migrate to IP based next-
generation applications while continuing to offer traditional TDM based services to
customers.
The B1205 supports a range of technologies such as POTS,
ADSL/ADSL2/ADSL2+, VDSL2, SHDSL(EFM) which allows the service providers to
offer highly interactive and bandwidth intensive applications. B1205 Facts as a traditional
TDM based DLC, IPDSLAM, Media Gateway platform rolled into one.
5.2.1 B1205/ B1205E Shelf
The iAN B1205E shelf contains modules, adapters, backplanes, fan trays and air-filter.
Figure below shows the iAN B1205E shelf front view.
Figure 31: iAN B1205E Front View
B1205E Backplane
The backplane is vertically installed on back of the shelf. It provides interconnection for
the modules.
Slots:
There are 18 slots in the shelf. They are slot P1, P2, U1-U3, A, B, 1-10 and slot fan. The
slots are used to install modules or fan tray and guide them to the backplane. Module hot-
pluggable. All modules can be replaced without affecting service.
Slot P1 and P1: used for POW1A only.
Slot U1, U2 and U3:
- Slot U2 and U3 can be used to install VPM1A
- Slot U1 can be used to install PCU1A
Slot A and B: used for CSM1A only.
Slot 1-10: used for user interface module, including FXS1A and ASL1A
Slot FAN: used for fan tray only.
Figure 32: iANB1205E Shelf Slots
FAN TRAY
The iAN B1205E shelf is equipped with four plug-in fan trays located at right
side of the shelf.
Figure 33: Fan Tray Faceplate
1 Each fan tray contains four axial fans which are connected in parallel
with 48 VDC power input.
2 The fan trays are hot-swappable and can be replaced without interrupting
system operation.
3 Each fan tray transfers two fan alarms to PCU1A.
4 There are two LED indicators on the fan tray faceplate ALIVE and DIAG
5 When fuse protection fails, the DIAG LED will be turned ON, while the ALIVE
LED will keep ON.
5.3 AN B1205E/ AN B1205 MODULES AND THEIR FUNCTIONS
1. Common Modules:
a) CSM1A:
The system control module controls the whole system operation which supports VoIP/AG
services, AN services and DSLAM services. 4xGE interface with configurable SFP in
front panel as WAN ports. Can be used as shelf cascade
The CSM1A (Control & Switching Module) is the central processing module of iAN
B1205E that controls the entire system operation. The CSM1A module is installed in slots
A or B.
The CSM1A module supports VoIP/AG services, AN services and DSLAM services.
It has the following features:
L2 Switch with 20 GE ports (40 GE ports with Redundant CSM1A):
● 4xGE uplink interfaces with configurable SFP in front panel
● Uplink interfaces can be used for shelf cascading also.
● 2xGE interface to each of Slots 1 - 10
● 1xGE interface to each of Slots U1 and U3.
● 2xGE interface to slot U2.
1+1 Hot Redundancy
● Manual switch
● Auto switch
DSLAM service processing
● System OAM features
● Download system and module firmware
● Stores system and module configuration data
● Detects, initialize and configure modules
● Communicates with PCU1A for external alarms
Management and Maintenance:
● GUI (SNMP) network management interface,
● CLI (Debug) interface
● Module self-test and diagnostic
● Support IPv6
The CSM1A module has two buttons on the faceplate
1. SWITCH:
To switchover the CSM1A module.
When there is only one CSM1A module in the shelf, pressing the SWITCH
button on the active CSM1A module will have no effect.
When there are two (active and standby) CSM1A modules in the shelf, pressing
the SWITCH button on the standby CSM1A module will have no effect.
When there are two (active and standby) CSM1A modules in the shelf and
the CSM1A modules are in redundant operation mode, pressing SWITCH
button on the active CSM1A module will switch the active CSM1A to
standby CSM1A.
2. RESET:
To restart the CSM1A module, the CSM1A module will reset all hardware and
firmware.
When there is only one CSM1A module in the shelf, pressing the RESET
button on the active CSM1A module will cause it to reset itself, and no
switchover will occur.
When there are two (active and standby) CSM1A modules in the shelf,
pressing RESET button on standby CSM1A module will reset itself, and no
switchover will occur
When there are two (active and standby) CSM1A modules in the shelf and
the CSM1A modules are in redundant operation mode, pressing the RESET
button on the active CSM1A module will cause a protected switchover to
occur, and the standby CSM1A will switch to active CSM1A.
b) POW1A:
The power supply module with ring generator, converts -48V DC to +3.3V DC and +75V
DC ringing voltage and distributes power to modules via the backplane. The POW1A
module is installed in slots P1 or P2.
The POW1A module has the following features:
1 Supports DC/DC converter:
- -48V DC current input, up to 20A
- -48V to +3.3V DC/DC converter, up to 30A
- -48VDC to +75VDC DC/DC converter, up to 30W (POW1A only)
2 Supports 1+1 hot redundancy
3 Management and Maintenance
- Surge protection
- EMC/EMI arrestor
- Provides un-present alarm
- Provides -48V power supply alarm
- Provides +3.3V power supply alarm
- Provides ringing module alarm (POW1A only)
- Integrated management by the CSM via I2C bus
2. TDM user interface module:
a) FXS1A:
The remote user module provides 64 POTS subscriber connections and supports built- in
line test (1.8Kohm loop impedance). The FXS1A module has the following features:
Provides 64 PSTN subscriber line interface ports
BORSCHT function
● Battery feeding
● Over-voltage and over-current protection
● Ringing control
● Supervision
● Codec
● Hybrid
● Test
Management and Maintenance
● Supports line test, self-test and diagnostics
● Debug interface
● Integrated management by the CSM1A
Special features:
● Configurable A-law and μ-law codec both
● Standard 1800 ohm loop length
● Balanced ringing signal for maximum 5 REN per port.
● Dial pulse with both 10 and 20 PPS.
● On-hook transmission for Calling Line Identity.
● Polarity reversal
● 16Khz and 12Khz tone generation
● Anti-lightning protection
3. IPxDSL user interface module
a) ASL1A:
ASL1A is high-performance ADSL concentration modules which provide 64 ADSL ports
in asymmetric data mode. The ASL1A module is installed in slots 1-10. The ASL1A
module has the following features:
Supports ADSL and ADSL2/2+ standards
● ANSI T1.413, G.992.2(G.lite),G.992.1(G.dmt) Annex-A, G.992.3(ADSL2),
G.992.5(ADSL2+), G.992.3(ADSL2) Annex M, G.992.5(ADSL2+),
G.992.3(ADSL2) (READSL) Annex L
● Up to 28 Mbps downstream.
● Up to 1 Mbps upstream for ADSL2+/Annex A,
● 3 Mbps upstream for Annex M standard.
● 8 PVCs per ADSL port
Supports Layer-2 switching
● 4K MAC address table
● MAC address limit per bridge port
Supports QoS
● 8 queues in DSL port
● 8 queues in WAN port
Supports multicast
● IGMP snooping v1/v2/v3
● Up to 256 multicast groups per module
Supports security
● MAC Flooding Control, MAC address spoofing, MAC filtering
● DHCP filter, DHCP Option82
● Gateway ARP spoofing, IP spoofing prevention
● Access Control List (ACL)
Management and Maintenance
● Supports SELT/DELT and diagnostics
● Debug interface
● Integrated management by the CSM
4. VOIP process module
a) VPM1A:
The voice process module provides up to 384 channels. VPM1A. The VPM1A
(Voice Processing Module) enables VoIP services on iAN B1200 systems. It provides
media and packet processing functions on the system. The VPM1A module is installed in
slots U1, U2 or U3. The VPM1A module has the following features:
DSP Channel Capacity
● G.711: 256 channels ,
● G.729: 168 channels
● G.723.1: 128 channels
● G.726ADPCM : 256 channels
Voice Codec En-Coding & De-Coding
● G.711A/U Law, G.729AB ,G.723.1,G.726 ADPCM
Tone Detector and Tone Generator
● Voice Activity Detection (VAD)
● Comfortable Noise Generation (CNG)
Line Echo Canceller
DTMF/MF Relay, RFC2833
Packet Loss Concealment & Recovery
Fax Relay
● CED/CNG Tone Detection in Fax Mode
● V.21, V.27ter, V.29, V.17, V.33, V.34 HDX, V.34 Fax Forced Fallback
● T.38 over UDPTL, T.38 over RTP, T.38 Version 0/1/2/3
CallerID
● Bellcore , ETSI, NTT, Chinese and UK CID Generation & Detection
Performance Statistics
● RTCP (RFC3550/3551), RTCP-XR (RFC3611)
Management and Maintenance
● Debug interface on the faceplate
● Integrated management by the CSM1A
5. MISC:
a) PCU1A:
The Peripheral Control Unit provides the external signals access for the iAN B1205E
system PCU1A (Peripheral Control Unit) provides the external signals access for the iAN
B1205E system. PCU1A is installed in slot U1.
The PCU1A module has the following features:
1 Collects 12 external alarms
2 Buffer and distribute 2.048 MHz clock from central office to CSM
3 Provides one GE port for port mirroring function of CSM
4 Integrated management by the CSM via SCB bus
5.4 IANB1205:
The iAN B1205 shelf contains modules, adapters, backplanes, fan trays and air-
filter. Figure shows the iAN B1205 shelf front view.
Figure 34: iAN B1205 Front View
B1205 BACKPLANE:
The backplane is vertically installed on back of the shelf. It provides interconnection for
the modules.
SLOTS:
There are 13 slots in the shelf. They are slot P1, P2, U1-U3, A, B, 1-5 and slot
fan. The slots are used to install modules or fan tray and guide them to the
backplane. Figure shows shelf slots arrangements.
Figure 35: iANB1205 slot arrangement
Slot P1 and P1: used for POW1A only.
Slot U1, U2 and U3:
- Slot U2 and U3 can be used to install VPM1A
- Slot U1 can be used to install PCU1A
Slot A and B: used for CSM1A only.
Slot 1-5: used for user interface module, including FXS1A, ASL1A
Slot FAN: used for fan tray only.
The iAN B1205 shelf is equipped with four plug-in fan trays located at right
side of the shelf. Figure shows the fan tray faceplate.
Figure 36: Fan Tray Faceplate
1 Each fan tray contains four axial fans which are connected in parallel
with 48 VDC power input.
2 The fan trays are hot-swappable and can be replaced without interrupting
system operation.
3 Each fan tray transfers two fan alarms to PCU1A.
4 There are two LED indicators on the fan tray faceplate ALIVE and DIAG
5 When fuse protection fails, the DIAG LED will be turned ON, while the ALIVE
LED will keep ON.
5.5 AN B1205 MODULES AND THEIR FUNCTIONS:
1. Common Modules:
CSM1A :
The system control module controls the whole system operation which supports VoIP/AG
services, AN services and DSLAM services. 4xGE interface with configurable SFP in
front panel as WAN ports. Can be used as shelf cascade
POW1A
The power supply module with ring generator, converts -48V DC to +3.3V DC and +75V
DC ring.
2. TDM user interface module
FXS1A:
The remote user module provides 64 POTS subscriber connections and supports built- in
line test (1.8Kohm loop impedance).
3. IPxDSL user interface module
ASL1A:
The ADSL user module provides 64 ADSL user connections.
4. VOIP process module
VPM1A:
The voice process module provides up to 384 channels.
5. MISC:
PCU1A:
The Peripheral Control Unit provides the external signals access for the iAN B1205E
system
5.6 POWER DISTRIBUTION PANEL :
The PDP (Power Distribution Panel) provides DC power supply and overload
protection for the iAN B1205E / the iAN B1205 system. It is installed at top of the rack,
and occupies 3U of rack space.
The type of PDP which is used in the iAN B1205E system is PDP-B. It is
installed at the top of the rack.
5.7 VOICE APPLICATION OF LMG:
Netman 4000
Phone Lines
iAN B1205E
IP
Network
Phone Lines
iAN B1205
IMS
Figure 37: Voice Application
5.8 DATA APPLICATION OF LMGS:
Data Application
Netman 4000
ADSL Lines
iAN B1205E
IP
Network
ADSL Lines
iAN B1205
BNG
Figure 38: Data Application
5.9 LMG UPLINK CONNECTIVITY VIA LAN SWITCH
Figure 39: LMG Uplink COnnectivity
5.10 INITIAL CONFIGURATION OF LMGS
Console Port Connectivity
Connect the Debug Cable between the “COM” connector on the CSM faceplate and serial
RS 232 connector on PC
Figure 40: CONSOLE PORT Connectivity
Hyper Terminal Configuration
Figure 41: Hyper Terminal Configuration
Bits per second: 115200
Data bits: 8
Parity: None
Stop bits: 1
Flow control: None
5.11 NODE MANAGEMENT IP CONFIGURATION OF 256P & 500P
LMG
The default username/password is admin/admin.
Log on to the CSM1A
[email protected]:
● ~#admin
● You are now in the Privileged Mode
Configure the IP address
● B1205#ip
● B1205(IP)#management address 10.230.7.5 netmask 255.255.255.0
● This command will change the IP address to 10.230.7.5.
● Execute anyway? (yes or no)
● Yes
Configure default route
● B1205(IP)#route 0.0.0.0 netmask 0.0.0.0 gateway 10.230.7.254
Check the IP address and default route setting
● B1205(IP)#show ip
● B1205(IP)#show route
Save Configuration
● B1205#save config bin
Now out band management port can be used to access CSM1A.
5.12 CONCLUSION
LMG is an access equipment provides a simple solution to migrate from TDM
based local exchanges to IP based equipment with less power and minimum
maintenance. It also provides facilities for DSL services thereby eliminating the use of
additional DSLAM. The compact build of LMG’s provides an excellent choice for
operators ready to migrate to an all IP based NGN network with reduced CAPEX and
OPEX.
6 SSTP ARCHITECTURE AND IMPLEMENTATION
6.1 LEARNING OBJECTIVE
This chapter deals about the UTStarcom SSTP and its implementation in BSNL network.
6.2 INTRODUCTION
The efficiency of SS7 had made a numbers of applications possible with e.g. fast
connection setup in PSTN, "short message service" and "location update” messages in
GSM world. As the size of the network grew, it became more and more difficult to
manage the direct SS7 links between the switches and from switches to other network
elements like HLR, SMSC, SCPs etc. The introduction of Standalone Signal Transfer
Point (SSTP) was a historic step from that perspective. It immediately solved issues
related to the complexity by converting the mesh networks into the star networks. It is
now able to handle the signaling very efficiently. This capability also offloaded some of
the processing power required in the L-1 TAX and all switches could breathe easier.
SSTP also handle the non call related messages efficiently. These messages exist
in all the technology and may not be needed to be handled by a switch. Many times these
will actually involve multiple technologies and in absence of the uniform signaling layer,
the complexity increase many fold. For example a simple service like SMS, a CDMA
subscriber may send the message to a GSM subscriber that could go through the long
distance network. This involves three different networks and three technologies. However
one thing is common, which is signaling as the transport to carry this.
The SSTP suddenly became the vantage point in the network because of the
simple reason that the signaling protocol was common i.e. SS7, independent of the
technology and the access. Be it GSM or CDMA, the connectivity was based on SS7 with
different application parts (MAP, INAP etc.). Thus the SSTPs are actually a centralized
routing database and not a transmission system for SS7 packet.
As the subscribers are growing, the number of nodes involved in the routing is
also increasing exponentially. If the management of the routing is to be done on these
individual nodes, the complexity increases accordingly, which also mean the increased
chances of error and hence loosing revenue? The SSTP, enabling the uniform signaling in
SS7 domain, provided a single routing database, which is managed centrally. This routing
database is able to make the routing decision based on the destination point codes (DPC),
global title translation (GTT), routing keys etc.
Moreover when any mobile subscriber of a private operator roams into service
area of other operator (say BSNL), the signaling traffic or SMS are being handled by
signaling channel of BSNL taken by private operator against the POI. BSNL was not able
to measure the traffic and it was also not known about the type of signaling traffic. To
measure and know type of signaling traffic, SSTP was planned and installed so that
billing to private operators can be done accordingly.
6.3 NEED OF SSTP
Due to the reasons explained above, BSNL decided to have a separate signaling network
by installing a number SSTPs at various locations.
Following were the main objectives:
Regulate, measure, and account for inter-network traffic including SMS
messages from mobile networks including GSM and CDMA
Achieve a flexibility and transparency in management of signaling for
BSNL's wired and wireless networks.
Optimal expansion of Mobile network of BSNL
Introduction of new services.
Offer CCS7 & IP Signaling Services to other Wire line & Wireless
Network Operators.
To enable migration of TDM based network to IP based network.
6.4 ADVANTAGES OF SSTP
• Works as central command for network growth and maintenance
• Optimizes traffic capacity
– Full mated pair redundancy
• Dedicated signaling processors, resources
• Upgrade path divorced from MSC / SSP functions, growth
• Most effective method to manage network level resources, features
• Frees up processing capacity from the switches
• Can host most of the applications, centrally
• Enhanced MNP processing capacity
• Support for newer protocols SIP/Diameter.
6.5 BSNL IMPLEMENTATION
BSNL existing SSTP network comprising of 16 SSTP nodes installed in mated
pair configuration
The proposed SSTP solution is for complete new Signaling Solution
System(Option 1) and it comprises of 16 No's of FNMP functionality compliant
SSTP nodes, 200Mn MNP DB, EMS/DR EMS system and migration of all
network elements to new SSTP system of BSNL network.
The SSTPs at Delhi, Chennai, Pune, & Ernakulum shall be with International
Signaling Gateway functionality
Each of the TAXs/IP TAXs & MSCs in BSNL Network shall be connected to at
least two SSTPs through IP and/or E1 link per SSTP on load balancing and
failover manner
The MSCs in the Indian Telecom Network connected to TAXs/IP TAXs of BSNL
Network shall be routed through one of the sixteen SSTPs installed as part of this
tender.
SSTPs shall be connected with the BSNL’s IP MPLS network through two L3
LAN switch with minimum two GE interfaces The Layer-3 switches shall be
deployed in high availability mode (Active-Active) across different arms of each
site.
SSTPs shall be interconnected with mated SSTP node with FE links /HSL links
through the SDH network of BSNL for redundancy purposes in addition to
interconnecting the SSTPs amongst themselves and to the EMS locations on the
IP MPLS networks. Some network elements are also connected with HSL/FE
links.
NOC/ DR NOC at Bangalore and Mumbai
6.5.1 Locations of SSTP in BSNL Network
All the SSTP nodes are installed at the location show in figure below
Table 5. SSTP Node Location
16 SSTP Node at SSTP Field site and
NOC and DR NOC for EMS
2 SSTP nodes @ Training site
Figure 42: SSTP Architecture in BSNL Network
6.5.2 BSNL Connectivity
The Figure below shows the SSTP connectivity
Figure 43: SSTP Connectivity
Figure 44: Mated SSTP Connectivity
6.5.3 Capacity Requirement of SSTP
The Table below depicts the Capacity and performance requirement of the SSTP nodes
Table 6. SSTP Capacity
S. No. Parameter Unit Value
a. MSU (Message Signaling MSU/sec 5,00,000
Unit) throughput
b. GTT ( Global Title GTT/sec 2,00,000
Translation) throughput
c IN Transactions throughput Per sec 20,000
d. GTT entries Numbers 400,000
e. GT length Digits 24
f. SCTP associations over IP Numbers 4000
g. Link-sets Numbers 1,024
h. Route-sets Numbers 10,000
i. Routes/link-set Numbers 1,024
j. Routes/route-set Numbers 4
k. Routes Numbers 16,000
l. Screening Matrix 512 x 512
m. Number portability Transactions per 7000
Transactions per second second
Total capacity of SSTP at each node is 500 TPS(Transaction per seconds)
6.6 UTSTARCOM SSTP
The UTStarcom SSTP comes with the product name of iSG6400. The chassis
accommodate the same types of function boards that fall into the following six categories:
MPU Card (MPU1A):
This is the system main control board. It manages all hardware resources;
provides a common, manageable and HA platform for the system. The MPU board in the
chassis functions as a communication agent for the SNMS.
Line Card (PEM-S8):
It supports TDM-based MTP2 protocol and IP-based M2PA protocol; and
implements reliable signaling link transmission between iSG6400 and SS7 devices. It
also provides translation between SS7 common channel signaling and MTP3 messages.
CLOCK I/O board (PCU1A):
This provides the synchronization clock signal for E1/T1 trunks.
SPU ( Signaling Processing Unit):
This is the signal processing equipment that provides SIGTRAN and SS7
signaling messages translation and distribution functionality. It supports MTP3, SCCP,
and M3UA protocols.
6.6.1 Hardware Platform: X86 based Server
The figure below shows the chassis view of iSG6400
Figure 45: Chassis view of iSG6400
6.6.2 Reliability Configuration of one SSTP site
Both ISG6400 Chassis are active and active load sharing mode. Failure of any
hardware module in one chassis does not impact any services
The LSL and HSL links from SSP are duplicated and to be connected to E1 ports
on both chassis
LAN SW Pair works in Active –Active mode
SPU (Signaling Processing Unit):
This software application runs on IBM severs . This is the signal processing equipment
that provides SIGTRAN and SS7 signaling messages translation and distribution
functionality. It supports MTP3, SCCP, and M3UA protocols.
Two SPUs run on active – Active load sharing mode
The Figure below shows the reliability configuration of SSTP node
Figure 46: SSTP Reliability configuration
6.6.3 Key parameters of SSTP node
Some of the key parameters of UTStarcom SSTP are shown in figure below
Figure 47: Key Parameters
6.6.4 Dimensioning of links
Assuming
The worst case, the load of every LSL/HLS is 0.9 ERLANG. In this case, SSTP
shall still be able to handle the traffic.
The average length of SS7 is 40 bytes
1. TPS of LSL
Worst case , one LSL can bear the traffic up to: 64Kbps/8 * 0.9ERLANG/40 =
180 SS7 messages/sec,
One TPS is defined as receiving one request message and sending one
response message, so it is 180/2 = 90TPS per LSL
One line card has 256 LSL, so TPS of one line card = 90*256 = 23 K TPS.
With 4 Line cards : 23* 4= 92 K TPS is offered against tender 80K TPS
2. TPS of HSL
Worst case for HSL: (30*64Kbps/8) * 0.9ERLANG/40 = 5400
messages/sec, one TPS is defined as receiving one request message and
sending one response message, so it is 5400/2= 2700 TPS/HSL
The normal case, the TPS of one HSL: 30* 64Kbps/8 *
0.3ERLANG/40/2 = 900 TPS
TPS of 24 HSL = 2700 TPS*24 = 64.4 K while tender requirement is 60K TPS
approx in worst possible case.
3. TPS of SIGTRAN/M3UA
One SPU(SIGTRAN and M3UA module of SSTP node) can serve up to 15
line cards. TPS of one line card is 23 K TPS.
The worst case is that, all traffic from line card go to SIGTRAN/M3UA
module , so total capacity of SIGTRAN/M3UA of one SSTP node is:
15*23K = 345K TPS
TPS of UT SSTP is well above the tender reuirement which is 160K TPS.
6.7 IMPLEMENTATION FOR MNP
Number portability functionality is required for GSM mobile and CDMA
networks and fixed networks also as per tender ,FMNP Capacity should be 200 Mn DB
as per SOR and expandable by additional 25 m .
For the MNP, iSG6400 is the mixer of SSTP and SRF, it will play role of NP
HLR and reply the LOCREQ, SRI Req etc with new Routing Number, and originating
network will route the call to ported in network basing on the new routing number.
SLR is the MemDB, it is also named NP AS
All processing occurs in RAM
Flexible design allows data for multiple applications to be searched in a single
SLR
SLR, its capacity increased by adding more servers, one SLR server with 28G
RAM caches FMNP data of 280M subs (assuming one FMNP record consume
100 bytes)
Oracle DB Enterprise 11g at NOC/DR-NOC is used for keeping NP data
permanently, principally it has no limitation to size of data with external
storage
The Figure below shows the MNP architecture implemented in BSNL
Figure 48: MNP Implementation
A virtual IP address (VIP or VIPA) is an IP address that doesn't correspond to
an actual physical network interface (port) ;
VIP addresses are also used for connection redundancy by providing alternative
fail-over options for one machine.
6.8 FEATURES OF UT SSTP
UT SSTP Other vendor
eMS, DB and
All the h/w of SSTP are
Hardware Lock- SIGTRAN/SIP/DIAMTER routing use
proprietary hardware,
In Off-the-Shelf X86 Server , no hardware
hardware lock-in is there
lock in
So far MNP capacity is
One Line card support 256 LSL, MNP
120M entries, capacity of
Capacity capacity is 250M NP entries and can be
one line card is far less than
further expanded
256 LSL
• UT SSTP use Oracle DB for
eMS and NP DB. Oracle
database is a truly carrier-class
DB, with high reliability,
No centralized DB, every
centralized data management
SSTP node has its own DB,
• UT SSTP network is composed
so it need a lot of DB
by distributed SSTP nodes and
Centralized license, which is more
Centralized eMS/NP SRV /DB
DataBase costly, and more difficult to
SRV, it is more flexible and a
do DB sync and
better cost structure. All SSTP
management. It is higher
node share the centralized
CAPEX and OPEX
DB/eMS/NP SRV
• Centralized DB means low
CAPEX and OPEX
DB Veritas used to synchronize the Oracle
Use proprietary software to
synchronization DB between different NOC/DR-NOC to
synchronize DB, it is not
between implement DB Geographic Redundancy.
stable and reliable
Redundancy sites Veritas is most reliable tools to do this
• No GUI, all
management are
• Centralized eMS manage all the
done by MML.
SSTP node which is deployed
• No centralized
OA&M around PAN India.
management point ,
• eMS is GUI based, easy to
• it means higher
operate and use, more friendly
OPEX
• Support SS7 and SIGTRAN
Emerging • Support the emerging Support only SS7 and
Protocol Support DIAMETER AND SIP SIGTRAN
6.9 CONCLUSION
The Document introduces the requirements and implementation of SSTP in BSNL. The
new SSTPs will be capable of supporting new signaling technologies like SIP and
diameter, in addition to existing SS7/SIGTRAN and planned to cater to the signaling
needs of BSNL network for future.
7 IMS ARCHITECTURE
7.1 LEARNING OBJECTIVES
Learn about the Architecture of IMS
Different working elements of IMS Core
IMS interfaces
Application Servers to provide functionality.
Basic Call Flow in IMS
7.2 INTRODUCTION
Many successful services are available today on the Internet, including e-mail, web
browsing, chat, and audio and video downloading/streaming, Internet telephony and Multimedia
Communications Services.
Both fixed and mobile operators face problem of subscriber churn, and the issue is getting
worse as new service providers offers cheap, or free, calls over the Internet continue to arrive on
the scene and gain market share.
One key way to attract and retain subscribers is to offer differentiation in areas like
personalization, service bundling, co-branding, business-to-business relations, tariffs,
single sign-on and quality of service.
Another key way to retain subscribers is to build on and strengthen the customer
relationship so that subscribers are far more reluctant to switch suppliers, even if
switching means lower call charges in the short term.
In this case, they will have to rapidly push IMS before proprietary solutions become largely
adopted. IMS is the only standardized solution in the telecommunications world.
7.3 WHAT IS IMS ?
IMS – IP Multimedia Subsystem standardized by the telecommunications world is a new
architecture based on new concepts, new technologies, new partners and ecosystem.
IMS provides real-time multimedia sessions (voice session, video session , conference session,
etc) and non real-time multimedia sessions (Push to talk, Presence, instant messaging) over an all-
IP network.
IMS targets convergence of services supplied indifferently by different types of networks : fixed,
mobile, Internet. IMS is also called Multimedia NGN (Next Generation Network).
IMS deployment is a strategic decision, not a network technology decision. It can be taken either
by a traditional service provider in the context of repositioning its business on IP services or by
any entity that would decide to start an activity in IP services even without owning an access or
transport network.
IMS offers standardized service enablers and network interfaces that will make interoperability of new
MM services easier to achieve.
IMS is a tool for operators that enable the creation and delivery of PS based person-to-person MM
services in a way that protects the operator business model and generates new revenue.
Service scalability is solved by the IMS architecture. It offers support to compose services and
expand existing services.
The core of IMS is combining the best of two worlds datacom industry & telecom industry.
7.4 WHY IMS?
Operator perspective End-user perspective General
Quality Of Service New, exciting services and Faster time to market with
enhancements of existing new services
services
Service Integration Same services available Grow and protect subscriber
regardless of terminal and base, increase ARPU
access type
Keeps charging relation Ease of use & Security Controlling CAPEX and
with user OPEX
7.5 IMS STANDARDIZATION
The IMS was initially standardized by the 3rd Generation Partnership Projects (3GPP) as
part of its Release 5 specifications & is practically speaking targeted at supporting non – real
time services .The second release is 3GPP Release 6 & is targeted at supporting real time
services .3GPP release added inter-working with WLAN.
With the increasing penetration of Wireless Local Area Networks (WLANs) and
emerging Wireless Metropolitan Area Networks (WiMax) as access network technologies, the
IMS scope is now extended within the ongoing Release 7 standardization for any IP access
network, including fixed access networks, i.e. DSL.
7.6 IMS ARCHITECTURE AS DEFINED BY 3 GPP
The IMS provides all the network entities and procedures to support real-time voice and
multimedia IP applications. It uses SIP to support signaling and session control for real-time
services Depending on the specific tasks performed by a CSCF, CSCFs can be divided into three
different types.
Serving CSCF (S-CSCF).
Proxy CSCF (P-CSCF).
Interrogating CSCF (I-CSCF).
Figure 49: GPP IP Multimedia Subsystems
7.6.1 S-CSCF (Serving Call State Control Function)
An S-CSCF provides session control services for a user. It maintains session states for a
registered user’s on-going sessions and performs the following main tasks.
a. Registration: An S-CSCF can act as a SIP Registrar to accept users’ SIP
registration requests and make users’ registration and location information
available to location servers such as the HSS (Home Subscriber Server).
b. Session Control: An S-CSCF can perform SIP session control functions for a
registered user. Relay SIP requests and responses between calling and called
parties.
c. Proxy Server: An S-CSCF may act as a SIP Proxy Server that relays SIP
messages between users and other CSCFs or SIP servers.
d. Interactions with Application Servers: An S-CSCF acts as the interface to
application servers and other IP or legacy service platforms.
e. Other functions: An S-CSCF performs a range of other functions not mentioned
above. For example, it provides service-related event notifications to users and
generates Call Detail Records (CDRs) neededfor accounting and billing
7.6.2 P-CSCF
A P-CSCF is a mobile’s first contact point inside a local (or visited) IMS. It acts as a SIP
Proxy Server. In other words, the P-CSCF accepts SIP requests from the mobiles and then either
serves these requests internally or forwards them to other servers. The P-CSCF includes a Policy
Control Function (PCF) that controls the policy regarding how bearers in the packet-switched
network should be used. The P-CSCF performs the following specific functions:
o Forward SIP REGISTER request from a mobile to the mobile’s home network. If an I-
CSCF is used in the mobile’s home network, the P-CSCF will forward the SIP
REGISTER request to the I-CSCF. Otherwise, the P- CSCF will forward the SIP
REGISTER request to an S-CSCF in the mobile’s home network. The P-CSCF
determines where a SIP REGISTER request should be forwarded based on the home
domain name in the SIP REGISTER Request received from the mobile.
o Forward other SIP messages from a mobile to a SIP server (e.g. the mobile’s S-CSCF in
the mobile’s home network). The P-CSCF determines to which SIP server the messages
should be forwarded based on the result of the SIP registration process.
o Forward SIP messages from the network to a mobile.
o Compression and decompression of SIP messages. Compression is required to minimize
the air-interface time.
o Perform necessary modifications to the SIP requests before forwarding them to other
network entities.
o Maintain a security association with the mobile.
o Detect emergency session.
o Create CDRs.
7.6.3 I-CSCF
An I-CSCF is an optional function that can be used to hide an operator networks internal
structure from an external network when an I-CSCF is used. It serves as a central contact point
within an operator’s network for all sessions destined to a subscriber of that network or a
roaming user currently visiting that network. Its main function is to select an S-CSCF for a
user’s session, route SIP requests to the selected S-CSCF. The I-CSCF selects an S-CSCF based
primarily on the following information:
o Capabilities required by the user.
o Capabilities and availability of the S-CSCF and
o Topological information, such as the location of an S-CSCF and the location of
o the users P-CSCFs if they are in the same operators network as the S-CSCF.
7.6.4 The Databases: (HSS And SLF)
HSS (Home Subs Server):
• It is just like HLR & Authentication Centre (AuC).
• All the database of users are stored in HSS ie, authentication data , service profile ,charging etc
will be in HSS.
• No VLR concept in IMS.
• HSS is mandatory. Whereas SLF is optional.
• HSS is master user database that supports IMS N/W entities that actually handle call.
• It contain subscriber profile , perform authentication & authorisation of the user & can provide
information about subscriber location & IP information.
SLF ( Subs Location Function)
•Whenever n/w size is so big that if one HSS cannot store data then SLF is required ,this is an addl.
Component.
•Suppose S-CSCF has done some authorisation then it has to contact HSS for downloading
,authentication etc .
–If one HSS is there then no ambiguity.
–But if more than one HSS then SLF will check which HSS.
•Both HSS & SLF communicate through Diameter protocol.
•This diameter is called as AAA protocol.
•SLF will have -(User-ID/ HSS-ID ).
Both the HSS and the SLF implement the Diameter protocol (RFC 3588) with an IMS-specific
Diameter application.
The Media Gateway Control Function (MGCF) and the IM Media Gateway(IM-MGW) are
responsible for signaling and media inter-working, respectively, between the PS domain and
circuit-switched networks (e.g. PSTN).
Multimedia Resource Function Processor (MRFP) –Provides resources to be controlled by
the MRFC
Sources media streams (for multimedia announcements)
Processes media streams (e.g. audio transcoding, media analysis)
Tones and announcements –Applied on receipt of ACK, self-timed with BYE or
stopped on BYE
Support DTMF within the bearer path.
The Multimedia Resource Function Controller (MRFC) interprets signaling information
from an S-CSCF or a SIP-based Application Server and controls the media streams resources in the
MRFP accordingly.
The Breakout Gateway Control Function (BGCF) selects to which PSTN network a session
should be forwarded. IT will then be responsible for forwarding the session signaling to the
appropriate MGCF and BGCF in the destination PSTN network
Figure 50: Service architecture
With both service architectures, the initial SIP request from a mobile travels from the
originating mobile to the visited P-CSCF first, which then forwards the request to the I-CSCF (if
used) in the originating mobiles home network. This I-CSCF selects an S- CSCF in the home
network for this user session and forwards the SIP request to session will travel directly between
the visited P-CSCF and the S-CSCF in the mobiles home network.
The S-CSCF is responsible for interfacing with internal and external service platforms
as illustrated in Fig. 3. There are three types of standardized platforms:
(1) SIP application server
(2) Open Service Access (OSA) Service Capability Server (SCS) and
(3) IP Multimedia Service Switching Function (IM-SSF).
OSA
Application Servers
MAP
CAMEL Service
OSA API HSS Environment
Sh Sh SI
CAP
OSA SCS SIP Cx IM-SSF
(for Providing Application Servers (for Interface withLegacy
Open Access to Service Platform)
3rd Party Service
ISC
Providers) ISC
ISC
S-CSCF
IP Multimedia
Subsystems
Figure 51: Interaction between SCSCF and service platforms
The services offered by them are value-added services (VAS or operator-specific services. The
S-CSCF uses the same interface, IMS Service Control (ISC) interface, to interface with all service
platforms. The signaling protocol over the ISC interface is SIP.
The OSA SCS and IM-SSF by themselves are not application servers. Instead, they are
gateways to other service environments. As depicted in Fig. 3, the OSA SCS and IM- SSF
interface to the OSA application server and CAMEL Service Environment (CSE), respectively.
From the perspective of the S-CSCF, however, they all exhibit the same ISC interface behavior.
The services are briefly described:
7.7 SIP APPLICATION SERVER:
In addition to session control, a SIP server can also provide various value-added services. A
lightweight SIP-based server enables the CSCF to utilize the SIP-based services and interact with
the ISP application servers without additional components
7.7.1 Camel Service Environment (CSE):
The CSE provides legacy Intelligent Network (IN) services. It allows operators leverage
existing infrastructure for IMS services. As specified earlier, the CSCF interacts with CSE through
IM-SSF. The IM-SSF hosts the CAMEL features and interfaces with CSE by CAP (CAMEL
Application Part).
7.7.2 OSA Application Server:
Applications may be developed by a third party that is not the owner of the network
infrastructure. The OSA application server framework provides a standardized way for a third
party to secure access to the IMS. The OSA reference architecture defines an OSA Application
Server as the service execution environment for third-party applications. The OSA application
server then interfaces with the CSCF through the OSA SCS by OSA API (Application
Programming Interface).
Figure 52: Simplified 3GPP IMS Architecture
7.8 EXAMPLES OF FLOW OF INFORMATION:
Registration and Re-Registration
Figure 53: Basic Message Flow
Figure 54: Routing of Mobile-To-Mobile Calls - Session Initiation
7.9 CONCLUSION:
The IP Multimedia Subsystem (IMS) seems to be the technology that will prevail in Next
Generation Networks (NGNs) and its main goal to make convergence between any IP networks
and a vertical handoff may happening depend on the user requirements (services,QoS,.etc). In this
chapter it was presented an IMS based interworking architecture for NGN networking through
which it prevail that how any two user from any two different IP based network can be involved in
a session under the umbrella of IMS management. By presenting a complete signaling flow for
concerning the authorization, registration, session set up and vertical handoff processes between
two networks.
ISDN (PRI/BRI), CENTREX, EPBAX, SIP SERVICES
8.1 LEARNING OBJECTIVES
At the end of session, the trainees will be able to
Explain the concept of Centrex
Understand the salient features of Centrex
Understand the new NGN based IP Centrex
Understand the features of IP Centrex
8.2 INTRODUCTION:
A group of users may form themselves into a CUG, creating a Pseudo "Private" sub-
networking, utilizing the facilities of the network. This service enables the subscribers
belonging to the same CUG to make communications with each other preventing access
from or to the world outside the group.
The CUG has opened up newer channels of communication for the upwardly mobile,
heavy duty user. An organization with CUG would be like a giant EPABX Exchange over the
whole country. CUG is a facility, wherein a group of subscribers within an organization can
talk (or send messages) to each other at a differential cost or for free, thereby significantly
reducing the overall cost of communication for the company.
The service can also be customized as per the requirement of the client. In this case
subscribers are tied in a pre-specified loop whereby they can send or receive message to all
users in the group, at one turn. The configurations of the user-group numbers are
captured in thesystem at the service center to facilitate the benefits of CUG.
Service providers these days are vying with each other to woo more customers for
CUG by providing more STD operating procedures ex caller identifications, call forwarding
are some of the options that come free with CUG package. Corporate plans are cheaper,
convenient and come with more utilities.
8.3 CENTREX
Centrex is a communication service which is available at centrally in an organization,
integrates many landline telephone numbers into a single functional group with many features
having free calling at any time in that landline group without any extra equipment with new
tariff at lowest rates. Centrex is the facility under which landline customers can talk free
of cost with each other,within a group.
BSNL Landline Group Plan, connects all the telephone numbers of old and new into
one group, connecting each number with free calling at any time without any extra
registration amount, initial deposit, and installation charges.
For Centrex facility, there is no limit for members to join into the same centrex group,
and the customers who avail this Centrex facility will be allowed to get free calls as usually
according with normal plan along with free calling between the group numbers. According
with technical feasibility, BSNL offers consolidated billing, payment and discount, under the
“Corporate Account Holders Scheme” where applicable, may be provided to the groups or
organizations covered under the Centrex.
8.3.1 THE SALIENT FEATURES OF CENTREX FACILITIES:
The existing customer can opt for this facility and new customers can avail this facility
atthe time of booking of telephone connection.
Minimum limit of members in a group of Centrex would be five.
Calls within the Group are free.
The registration amount, initial Deposit, monthly rental and installation charges etc.
for Telephone Connection under Centrex shall be same as normal Telephone
connection.
The free monthly calls as per tariff plan shall be allowed on the telephone
connection covered under the Centrex.
Anybody can join/leave the existing Group at any time.
The numbers belonging to public Services like, call Centers, enquiry numbers, ISP
Dial in numbers, paging service numbers, franchises, PCO holders and only incoming
number shall not be part of the Centrex Group.
Centrex Facility will be available in all Tariff Plans.
The Free Centrex facility shall be for the Cooperative Group Housing Society/Resident
Welfare Associations and DDA having clearly defined boundaries with maximum 500
dwelling units. Above 500 dwelling units the case will be considered by the concerned
CGMs on case to case basis.
This scheme is restricted to the Members of Society residing within the same premises.
No additional rent will be charged from the Members for availing Centrex facility.
There will not be any security deposit, Registration/Installation charges. However
usage charges shall be made by the Society. This free Centrex facility will also be
extended to Shopping malls with well defined boundaries with minimum 10 users.
7.10 SALIENT FEATURES OF CENTREX SUBSCRIBER
a. CLIP for Intercom and network calls
b. Conditional call forward On busy call, On no reply, Only PSTN calls, Only Intercom
calls
c. Call pick up Within Group, Directed, Night Service
d. Call Transfer Features All Calls, Incoming calls only, Internal only
e. Voice/Data protection
In CENTREX, no physical PABX needs to be installed. Facility can be extended to
any ordinary telephone. A special dial tone which is different from the normal dial tone is
provided.A number of such lines can form a group called CENTREX group.
Like a PABX, following dialing procedure is adopted:
a. For internal call (call within the group), dial a 2 or 3 digit number.
b. For outgoing call (to call outside the group), dial '9' - get normal dial tone-
dialLocal/STD/ISD digits.
c. For incoming call - a CENTREX number can be reached from outside
the group bydialing the 7 or 8 digit directory number.
8.3.2 CENTREX FACILITY FOR CIC/CORPORATE CUSTOMERS
A. CENTREX FACILITY IN THE BSNL
NETWORK FOR CIC/CORPORATE CUSTOMERS:
(i) The charges for Centrex facility are available on the site bsnl.co.in. The Centrex charges
will not be levied for CIC/Corporate customer whose annual revenue is more than Rs. 4
lakhs with combined billing wherever technical feasible.
(ii) Centrex facility may also be allowed to the trade related groups, which are
identified/authorized by respective associations like medicine, hardware, sanitary, doctors,
lawyers and general store etc. within SDCA after charging the Centrex charges.
B. RENT FREE INCOMING CONNECTION:
Rent free connection for corporate customers, housing societies multistory complex
etc. may also be provided for the use of incoming calls and making calls to other Centrex
numbers subject to the fulfillment of following conditions:
TABLE -1 CENTREX SIZE AND ITS RENT FREE CONNECTION
Size of Centrex Group Rent free incoming connection
15-25 01
26-50 02
51-100 03
For every additional 100 1 additional connection in additionabove
connections
Note:
i. The security deposit, fixed monthly charges, installation charges, free calls etc shall
be same as applicable to normal DEL.
ii. All intra centrex calls shall be free.
iii. The free monthly call shall be allowed on the DELs covered under the centrex.
iv. There is no upper limit on the number of members in the centrex group.
v. The Dels belonging to public services like, call centers, enquiry numbers, ISP Dial in
numbers, paging service numbers, PCO holders shall not be part of the Centrex group.
(C) CENTREX FACILITY TO TRADE RELATED GROUP, WHICH
ARE IDENTIFIED/AUTHORISED BY RESPECTIVE ASSOCIATIONS
LIKE MEDICINE,HARDWARE,SANITARY,DOCTORS,LAWYERS
AND GENERAL STORE ETC.
TABLE-2 CENTREX GROUP PLANS
Size of the Centrex Monthly plan Monthly plan Monthly plan
Group charges for Intra charges for Inter charges for
circle circle (in Rs.) SDCA
(in Rs.) (in Rs.)
Less than 25 75 150 50
More than 25 90 180 60
Note:
Rental for Centrex will be based on number of connections within the group and
category of Centrex.
There will be possibility of change of Category.
IP Centrex will be offered on fixed rental basis to the customers
Subscriber will not be charged for per call (both local &STD) within Centrex
IP Centex services will be provided to both Post and Prepaid subscribers
Plan/rental would be applied on individual subscriber
Even if some of subscriber moves out of group , the slab will not change.
8.4 WINGS CLOSED USER GROUP(ALL INDIA IP CENTREX)
A user can setup a WINGS closed user group across the country with both fixed line
and mobile users of BSNL as members of the group. The users include both NGN and Non
NGN customers of BSNL. This facility can be availed after paying a nominal charge per
month per user of IP Centrex facility. It allows small and medium size enterprises to deploy
PABX/ CUG service without really installing a system physically in their premises. You can
have a PAN India IP centrex i.e. numbers anywhere in the country can be brought into a
single Centrex group.
7.11 WINGS CLOSED USED GROUP (IP CENTREX):-
Features of IP Centrex
i. Master number can be any BSNL landline no.
ii. IP Centrex group may have 10000 members in a group. For creation of IP Centrex group
for more than 10000 NCNGN Circle may be contacted through EB Cell/Circle.
iii. Intra SDCA is applicable to landline number only as part of group. (Useful for
apartment, group housing society, colony etc.)
iv. Call made by full number dialing instead of short dialing will be charges as per existing
landline or mobile plan.
v. GSM mobile customers can make Centrex calling as Access code 1286(1261-Old
Code) + Short code.
vi. For GSM Mobile customers while roaming no additional charge for Centrex calling if
possible.
vii. Sulabh connection can be part of group (on request) but cannot be master of the IP
Centrex group.
viii. Billing/deduction of rental is to be from respective prepaid and postpaid account of
individual group number but in case of default billing to be borne by master number.
ix. It will be responsibility of applicant to ensure group numbers provided their consent to
be part of IP Centrex Group.
8.4.1 IP CENTREX FEATURES
IP Centrex allows small and medium size enterprises to deploy PABX/ CUG service
without really installing system physically in their premises.
IP Centrex have no limitation of Zone . It is PAN India Service.
A customer can have a PAN India IP centrex i.e. numbers anywhere in the country can
be brought into a single Centrex group. Each member of the group can access the other
by dialing a short code.
The IP centrex supports different types of terminals of the enterprise in the same group,
such as mobile terminals, landline phones.
Number Presentation/CLIP of intra-CENTREX call is the short number.
Dialing with the long number is recognized as off-net call, even if the caller and callee
are Centrex users and in the same group.
It works between NGN, PLMN and 5ESS and EWSD PSTN Subscribers.
A CENTREX Group can have max 10,000 subscribers.
Customer can request for any short code range except starting with 0 ,1 and 9.
If a BSNL mobile subscriber roams to other operator’s network, he can’t use the Centrex
service.
8.4.2 IP CENTREX GROUP CATEGORIES
Within STD Code/City/Apartment
Within SSA
Within Circle
Within Zone
Across the Zones
Number Within Group may be from Different Zones and from different Services like
LL,GSM.
8.4.3 THE CENTREX DIALING METHOD
Centrex Short Code length is 3 to 5 digits.
Landline Customer will dial short code to access the LL/Mobile number of the same
group. Mobile Customer will dial 1261 +Short code to access the LL /Mobile number
of the same Group.
7.12 HOW TO GET IP CENTREX?
Customers can Get the IP Centrex service through SMS/E-mail/IVRS/BSNL
Portal/BSNL CSC/Call Centre.
Any customer requiring this service has to submit the list of numbers (LL/Mobile )
to be added to IP Centrex Group. Fill up the CAF, selecting IP Centrex category
and has to submit it to CSR/EB with required documents
Sending SMS request for IPCENTREX Service requests Format is : IPCENTREX
followed by STD code i.e IPCENTREX <space><std code> to 54141. SMS is
pushed to Call Centre. Call Centre agent speaks to customer and raises request on
CRM.
Fault handling mechanism: Subscribers can book service request / complaint
through 1500/198.
8.5 CONCLUSION :
It is central office based communication service, which integrates all your multi
located telephone lines (Existing and New) into a single highly functional communication
groupwith more distinctive features without any additional equipment (like EPABX) at
your premises. Highly cost effective, Free Intra Centrex calls, No worry to select
innovative (PBXs) equipment,No risk of obsolescence of technology, No requirement of
power supply, No need to waste valuable floor space, No annual maintenance charges,
totally flexible are some of the salient features of Centrex facility.
PARTNER MANAGEMENT (TIP AND SYSTEM INTEGRATOR)
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9.1 LEARNING OBJECTIVE
After reading this unit, you should be able to understand services managed by Franchise/ TIP:
Enterprise Wi-Fi service
Establishment and Maintenance of Customized Wide Area Network (WAN)
PABX
Network System Solution
Bulk SMS solution etc.
9.2 INTRODUCTION
With the objective of increasing the customer base and enterprise business of BSNL and to
utilize the channels already associated with BSNL for sales, marketing and delivery in various
segments TIP (Telecom Infrastructure Provider) and Franchisee is introduced.
The channel partners would require to identify the prospective Business & Customers and to
convert such prospects (Leads) into a business order by regular visits, presentations,
submissions and explaining the various services and plans to them and to convince them to
bring them on board for taking BSNL services. The channel partners succeeding in garnering
business for BSNL would get suitable remuneration in the form of payment of commission
depending upon the category of customer, type of product(s) and volume of business booked.
Some Enterprise Business which is managed by Franchisee/TIP is listed below:
a. Enterprise Wi-Fi service
b. Establishment and Maintenance of Customized Wide Area Network (WAN)
c. PABX
d. Network System Solution
e. Bulk SMS solution etc.
Details scheme of above mentioned Enterprise solution is discussed one by one.
9.3 ENTERPRISE WI-FI SERVICE
BSNL is providing Telecommunications services to Enterprise customers such as Government
customers, PSUs, MNCs, Educational Institutions & other customers. There is large scope of
business opportunity to provide Wi-Fi hotspot services for these customers.
BSNL has deployed Wi-Fi core system with main at Bangalore and DR at Pune BSNL.
Corporate Office has instructed all the telecom circles to engage public Wi-Fi providers for
providing Wi-Fi services under Bulk user plan to Enterprise customers on revenue share basis.
9.3.1 ENTERPRISE WI-FI MODEL
a. Public Wi-Fi Providers/ Partners (PWPs) shall be permitted on first come first
served basis as per the requirement of SSAs/ Circles.
b. The PWP can deploy the Access Points APs both Indoor and Outdoor using
the available models (including RAPs with mesh APs).
c. Public Wi-Fi Provider (PWP) shall supply, install commission, operate and
maintain the PoEs/APs/UPS in customer premises.
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d. BSNL Reserves the right to broadcast its own SSID Other than captive traffic
in further with addition additional bandwidth provision.
e. The required bandwidth for provisioning of bulk user plans shall be bundled
with the various plans. Beyond this bandwidth, incremental BW charges shall
be applicable for which BSNL will raise separate invoice to the bulk
customers.
f. Internet shall be unlimited.
g. Captive portal for customer authentication shall be customized as per client
requirement. SSID will be configured as per the Clients/Partner’s choice.
9.3.2 RESPONSIBILITY OF PUBLIC WI-FI PARTNERS (PWPS)
9.3.2.1 The Public Wi-Fi Partners engaged through this open policy shall supply, install,
commission, operate and maintain Wi-Fi Hotspot location equipment such as
Access Points (which may be indoor or outdoor), UPS, POE, racks and associated
installation materials at hotspot locations.
9.3.2.2 BSNL shall extend bandwidth as per tariff plans purchased by customer. The
Public Wi-Fi Partner shall be responsible for extending the Bandwidth connectivity
from this single point to the remaining access points at that hotspot location
including electrical wiring staring from UPS, Optical, Data, LAN cabling etc.
9.3.2.3 Access Points (APs) shall be integrated with OEM’s WLC and create CAP/WAP
tunnel as per standard CAP/WAP protocol.
9.3.2.4 The PWP shall plan the hotspot locations by site survey, plan logistics etc. and get
the acceptance testing etc completed.
9.3.2.5 The Wi-Fi partner will be fully responsible for the operation and maintenance of
HS/APs/PoEs/WLCs/EMSs being deployed under this partnership and shall have
back to back arrangement with OEM of AP/WLC.
9.3.2.6 Details scope of work regarding installation of Wi-Fi Hotspot, Integration details of
WLCs with BSNL’s existing WAG/DHCP and integration of EMSs/CMSs shall be
discussed and finalized with the Wi-Fi NoC team at the time of validation/testing.
9.3.2.7 Promotion and Marketing of service – Public Wi-Fi Partners shall proactively
market Wi – Fi services by educating the customer on the usage /benefits of the
service and BSNL tariff plans etc.
9.3.2.8 The Public Wi-Fi partner shall carry – out formalities for Customer acquisition,
assisting customers in filling up of CAF (customer acquisition form).
9.3.3 RESPONSIBILITY OF BSNL.
BSNL shall provide space and power at Central location for Installation of WLC/EMS free of
cost. No co-location charges shall be taken from the OEM/Authorized partner of OEM.
9.3.3.1 The B/W or backhaul shall be bundled by BSNL as per the bulk user plan.
9.3.3.2 BSBL shall be responsible for integration of WLC/EMS with BSNL Core,
9.3.3.3 Provisioning and Billing shall be BSNL’s responsibility.
9.3.3.4 BSNL shall provide necessary support to the PWP/OEM/authorized OEM for
integration with BSNL Core network, marketing strategies and use of prepaid
Wallet system etc.
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9.3.4 ELIGIBILITY REQUIREMENT FOR PUBLIC WI-FI PARTNERS
(PWPS)
9.3.4.1 Any registered/partnership/proprietorship firm/society including existing Telecom
Infra Provider, having, minimum turnover of Rs. 5 Lakhs per year during the last
three consecutive years shall be eligible.
9.3.4.2 The registered/partnership/proprietorship firm/society shall have worked with
Telecom Service provider(s) /ISP(s) for minimum 1 year.
9.3.4.3 Existing FTTH Franchisee with good record of providing FTTH connections/BSNL
service shall be eligible to become Public Wi-Fi Partners.
9.3.4.4 One time Registration charges of Rs. 10,000/- (non-refundable) and applicable.
Taxes shall be taken at the time of registration.
9.3.5 REVENUE SHARING STRUCTURE
a) For feasible areas where BSNL bandwidth is available within 250 meters of the
customer premises, Public Wi-Fi partner (PWP) shall be responsible for
extending the Bandwidth connectivity from this single point to the remaining
access points:
b) Revenue share to the PWP shall be at 25% of the net revenue for 1- 4 APs
Plans.
c) Revenue share to the PWP shall be at 28% of the net revenue for 5- 10 APs
Plans.
d) Share of revenue to the PWP shall be @ 30% of the net revenue for Plans
beyond 4 APs.
e) Plan charges and applicable revenue to the PWP for 1 – 4 APs, 5 – 10 APs and
11 – 20 APs Plans is provided in the table below:
Table 1: Revenue sharing
Min. bandwidth in
Min. bandwidth in
Min. bandwidth in
Annual plan charges
Annual plan charges
Annual plan charges
for 11-20 AP plans
Revenue share to
Revenue share to
Revenue share to
for 5-10 AP plans
for 1-4 AP plans
Mbps
Mbps
Mbps
PWP
PWP
PWP
10 300000 NO NO NO NO NO NO
the
net
revenue
the net
25% of
realized
20 350000 NO NO NO NO NO NO
30 400000 YES 525000 NO NO NO
25
%
of
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40 450000 YES 575000 NO NO NO
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50 500000 YES 625000 YES 900000 30% of
the
60 550000 YES 675000 YES 950000 ne
t revenue
100 Revenue share to the PWP shall be as per YES 1050000
the no. of AP plans, however, incremental
BW charges applicable which shall be
payable to BSNL only.
9.4 ESTABLISHMENT AND MAINTENANCE OF
CUSTOMIZED WIDE AREA NETWORK (WAN)
BSNL is a major telecommunication service provider in India. It offers all kinds of
telecommunication services like Basic (both fixed and wireless), Broadband, Internet Leased
Line, Point to Point Leased Line, MPLS etc.
Keeping pace with the technological trend to provide single window enterprise
solution to its customers, BSNL, Calcutta Telephones is providing complete network system
solutions in building Wide Area Network on turnkey basis including supply, installation,
integration and maintenance of networking equipment.
9.4.1OBJECTIVES TO BE ACHIEVED BY FRANCHISEE
9.4.1.1 Establishment of LAN/WAN on turnkey basis including supply,
installation, integration and maintenance of networking equipment.
9.4.1.2 Complete system integration, networking, design,
implementation andmanagement.
C) Establishment of LAN/WAN on turnkey basis including supply, installation, integration
and maintenance of Video Conferencing Equipment, Public addressing System and other
upcoming Corporate Office Solutions.
9.4.2 METHOD OF OPERATION
9.4.2.1 For providing complete end-to-end solution to customer, BSNL
intends to empanel System Integrators who will not only do all the operations
and maintenance activities of customer end networking equipment but will also
do the marketing.
9.4.2.2 The empanelled SIs will submit the best possible rate so as to
enable BSNL to bid successfully in any tender. Based on the quotes received
from various SIs, the price for each model and each make will be finalized on
the lowest of the quote received. This will form the base price. On which BSNL
will submit its own proposal to the customer. For providing last mile
connectivity to the customer, BSNL will co-ordinate internally with all
agencies
/Departments to enable the same.
9.4.3 ELIGIBILITY CRITERIA
9.4.3.1 Network/System Integrator should be an IT/Networking sector
company in case of National and Circle SIs. And for Circle-Silver SI it may be
an individual or proprietorship/partnership concern.
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9.4.3.2 SI or its parent company should be a public limited or private
limited company registered in India in case of National and Circle SIs. Circle-
Silver SI may be an Individual/
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Proprietor/ partnership Concern /LLP / Company and registered as per commercial laws to
undertake the activities mentioned in scope of empanelment.
9.4.3.3 Network/System Integrator shall be a direct owner of technology
or have a direct teaming agreement with each of technology companies that
form the core building block for WAN or related project implementation.
9.4.3.4 The Network/System Integrator should provide letters of support
from Original Equipment Manufacturer (OEM) stating that their solution will be
supported on the platform proposed by Network/System Integrator for the next
five years.
9.4.3.5 The Network/ System Integrator should be ISO 9001:2008 or
higher certified for services.
9.4.3.6 The technical team of Network/System Integrators will assist
BSNL in coming out with the cost effective solution for the customers and will
be required to give joint presentation with BSNL to customers.
9.4.3.7 Each empanelled Network/System Integrator will need to submit
refundable security deposit in the form of Bank Guarantee from any scheduled
bank valid for one year.
9.4.3.8 Network/System Integrator shall also submit additional PBG of at least
5% of the
P.O. value, whenever a work is awarded to Network/System Integrator valid for the duration
of the project.
i) Network/System Integrator shall provide 24X7 help centre either web-based or
IVRS based. Network/System Integrator shall ensure consultation, assistance and advice
within 4(four) hours or as defined in SLA entered with customer.
j) Network/System Integrator shall support SLA requirements of BSNL customers
and ensure its compliance. In case SLA commitments are not met, Network/System
Integrator shall be responsible for payment of penalties, if any, imposed by the customer.
k) The PBG valid for five and half years shall be collected from the empanelled SIs
subject to minimum of Rs. 15 Lakhs, 3 Lakhs and Rs. 50,000/- for National level, Circle level
and Circle-Silver SIs respectively for abiding by the general rules of empanelment
agreement. The refundable SD of Rs. 1.0 Lakh, Rs.50,000 and Rs.10,000 respectively
submitted at the time of application for empanelment would stand released thereafter.
l) Once a SI is empanelled in National Category in any of the Circle, it can get
empanelled in any other Circle with a consent letter
m) NTR Circle would be treated as a Circle with its geographical territory as NCT,
Delhi for the purpose of SI empanelment and delivery of service.
n) Validity of the agreement to be signed between circle and empanelled SI (in case of
National, Circle and Circle-Silver level) shall be 5 (Five) years i.e. the agreement shall be
signed between circle and empanelled SI for a period of 5 years.
9.5 PABX
BSNL is providing Telecommunication Services to Enterprise customer such as
Government Organizations, Public Sector Undertakings, MNCs, Corporate, Hotels, Housing
Societies and all other Commercially Important Customers (CIC’s) through the provision of
Voice & Data PABX at their premises on a rental cum usage charge basis and accordingly
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seeks to empanel eligible firms / companies at National / Circle level for providing such
Voice & Data PABX service on Revenue Share Basis as PABX Franchisees of BSNL.
9.5.1SALIENT POINTS OF THE FREE PABX SCHEME
BSNL contemplates to provide the following services to its customers under this
scheme
a) Intra & Inter Office Voice Connectivity .................... Essential
b) Data Connectivity & Internet Access… .................... Optional
c) Value Added Services (VAS)… ................................ Optional
d) The VAS services may include services like Voice mail, audio/video conferencing,
Predictive Dialers, Automatic Call Distribution, call forwarding, encrypted communication,
Information Pop-Up Screens, VoIP, Voice Mail, Inter Office Video-Conferencing etc.
e) The above Telecommunication Services portfolio may be modified / supplemented
to include latest services in line with the customers’ preferences, market demands and in
accordance with BSNL requirements, from time to time.
9.5.2 KEY FEATURES OF THE SCHEME
9.5.2.1 The proposed agreement with the prospective firms /
companies shall be entered for an initial period of five years. Any customer
enrolled under the scheme during the agreement period would have to be served
by the concerned franchisee for a minimum periodof 3 (Three) Years or for such
extended periods as agreed upon by the Customer and BSNL from the
commencement of such services to the specific Customer.
9.5.2.2 PABX franchisee shall supply, install, operate and maintain the
Voice & Data PABX on BSNL’s advice.
9.5.2.3 BSNL shall provide all back haul voice and data connectivity to
the said Voice & Data PABX.
9.5.2.4 The customers under the scheme shall not be charged the rentals
for the junction connectivity and shall only be charged a rental on the extensions
provided. Call usage and data connectivity shall be charged on applicable tariff
of BSNL.
9.5.2.5 BSNL shall share both the rentals and the call/data usage
revenue with the PABX franchisee at the applicable rates prescribed. The tariff
may be modified by BSNL as per the business requirements.
9.5.2.6 Revenue share to the PABX Franchisee shall be made only on
the revenue realized/collected.
9.5.2.7 The services shall be provided under the brand names of BSNL only.
9.5.2.8 Both BSNL and PABX Franchisee shall jointly market the scheme.
9.5.3 ELIGIBILITY CRITERION
9.5.3.1 For National Level empanelment
9.5.3.1.1 Indian registered Firm / Company.
9.5.3.1.2 Shall be manufacturer / Distributer of the Voice and Data PABX
Switch. In case ofDistributor, must have 10 Years support agreement from OEM.
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9.5.3.1.3 Shall have a minimum annual turnover of Rs.1.50 Crores during each of
the last three financial years.
9.5.3.1.4 Experience of Installation / Sales & Service / Operation & Maintenance
of minimum 20,000 PABX ports spread in different parts of country (Minimum 3 States)
duringlast financial year.
9.5.3.1.5 Shall not be a Licensed Telecom Service Provider for any of the
Telecom Services in India.
9.5.3.2 FOR CIRCLE LEVEL EMPANELMENT
9.5.3.2.1 Indian registered Firm / Company.
9.5.3.2.2 Shall be manufacturer / Distributer of the Voice and Data PABX
Switch. In case ofDistributor, must have 10 Years support agreement from OEM.
9.5.3.2.3 Shall have a minimum annual turnover of Rs.0.50 Crores during each
of the lastthree financial years.
9.5.3.2.4 Experience of Installation / Sales & Service / Operation &
Maintenance ofminimum 2000 PABX ports during last financial year.
9.5.3.2.5 Shall not be a Licensed Telecom Service Provider for any of the
Telecom Servicesin India.
9.5.4 PABX FRANCHISEE’S ROLE & RESPONSIBILITIES
9.5.4.1 PABX Franchisee shall market/promote the PABX Services at
its own cost under the brand name of BSNL.
9.5.4.2 PABX Franchisee will provide following equipment at the
customer premises intimated in writing by BSNL in order to deliver the
Telecommunication Services to the BSNL Customers in the Areas of Operation.
9.5.4.3 Hardware & Software for Voice & Data PABX as per the demand of
customer.
9.5.4.4 FCBC, Batteries, MDF, Inter-Connect Cables (from PABX to MDF),
DDF etc
9.5.4.5 Modems, Media Convertors etc. required for interface to PABX.
9.5.4.6 Customer Premise Equipment for Broadband access service viz.
Router, xDSL Modems& Splitters.
9.5.4.7 Cabling/ Wiring at Customer Premises, if required by the
customer separately chargeable.
9.5.4.8 PABX Franchisee shall comply with all applicable laws,
byelaws, rules, regulations, orders, directions, notifications etc as per Law of the
land and of Government/ court/ tribunals.
9.5.4.9 PABX Franchisee shall comply and enable BSNL in complying
all laws related to Lawful Interception of all voice and data communication
provided through the PABX system and may be asked to supply CDR/IPDR
records.
9.5.5 BSNL’S ROLE & RESPONSIBILITIES
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9.5.5.1 BSNL shall provide the required bandwidth and connectivity to
its network for thePABX, subject to technical feasibility as per the applicable
policies of BSNL
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9.5.5.2 The customers garnered by the PABX Franchisees, shall be
directly allocated to the respective PABX Franchisee. However, customer
registering directly with BSNL, shall be allocated between the different
franchisees in a round-robin manner.
9.5.5.3 BSNL shall raise the bills for these services utilizing its own
resources and the collection of the same shall be done.
9.5.5.4 BSNL shall fix the tariffs applicable to customers for
Telecommunication Services in different circles taking into consideration the
local factors, prevailing competition, market scenario, etc. BSNL shall be free to
modify these tariffs from time to time.
9.5.5.5 BSNL shall share the revenue earned as per the agreement
between BSNL and PABX Franchisee.
9.5.5.6 Special business cases may however be signed by BSNL with
any customer for special periods, special needs and with special conditions in
consultation with the empanelled Franchisee(s) with the same or different
revenue share arrangement.
9.6 NETWORK SYSTEM SOLUTION
Network System Solution is required to supply, configure, Install, Commission the networking
equipments/hardware/Software at third party premise, along with integration with the existing
LAN wherever applicable & Acceptance Testing of system installed.Maintenance & supply of
spare parts of the equipments supplied & installed will be the integral part of the work.
Required functionality of the proposed network is as follows:
a) BSNL has provided 1 Gbps links from third party premises to NKN PoP in CTD
through optical fibre. The link is terminated in the NKN router at third party premises.
b) Equipments shall be supplied at the sites as specified at the time of placement of
P.O.
c) Installation, Configuration, Commissioning of LAN & wi-fi system and Integration
with existing Local Area Network (LAN) at third party location as per the third party
requirement shall be done by the bidder. The bidder shall be fully responsible for any damage
caused by the installation of LAN & wi-fi equipment to the third party buildings and will have
to repair and rectify any such damage in at his own cost before completion of the network.
Restoration of faults related to equipments has to be done by the bidder.
d) Change of configuration if required due to conversion/shifting of links to be done
by bidder.
9.6.1BROAD TECHNICAL
REQUIREMENTS:
NETWORKING
DEVICES:
9.6.1.1 Firewall with necessary accessories at third party location
- capable of supporting 1 Gbps Internet Bandwidth.
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9.6.1.2 Core Switch with necessary accessories – capable of handling traffic to
and from above firewall and interfacing with access switches.
9.6.1.3 Access switches with necessary accessories at third party location -
capable of handling traffic between core switch & end users at third party location.
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9.6.1.4 Wi-fi controller & access points for wireless network connectivity at
third partylocation.
9.6.1.5 I/O points for LAN connectivity of end users at third party location.
9.6.1.6 Integration with existing LAN at third party location.
9.7 BULK SMS SOLUTION
Bulk SMS solution business is for providing Bulk SMS Services to The Financial
organization. The proposal from bidder must integrate with The Financial organization’s
existing infrastructure seamlessly. The Current SMS volume is about 1 crore per month (which
includes Domestic & International SMS) which is projected to reach 4-5 Crores SMS per
month with due time. The Solution includes Real-time Push SMS Solution, Two-way SMS
solution, Bulk Campaign Messaging. In addition to the above, add-on services such as Missed
Call Alert Solution, Interactive SMS solution etc, needs to be provided as desired by The
Financial organization.
9.7.1 MINIMUM REQUIREMENTS FROM SMS SYSTEM
INTEGRATOR FOR PROVIDING BULK SMS SOLUTION
9.7.1.1 SMS solution for current & future requirement
9.7.1.2 Experience in SMS solution
9.7.1.3 Hardware, Software, Middleware & connectivity requirement
9.7.1.4 The System Integrator Solution should be able to establish connections
simultaneously to multiple SMSC ports provided by BSNL and the solution should be able to
support upto 500 TPS (Transactions per second) per port and a combined support of 2000
TPS.
9.7.1.5 BSNL shall provide necessary Domestic & International SMS
accounts/ports at single/multiple SMSC. System Integrator should integrate the same as per
the requirement of The Financial organization.
9.7.1.6 The System Integrator solution should have load balancing & auto-
failover features in the SMS solution and should divert the SMS to other SMSC port/account
in case of any failure at BSNL SMSC.
9.7.1.7 The System Integrators Solution should be able to identify Priority
Messages received with Priority flags (eg: P1 for High Priority, P2 for Other Priority SMS,
P3 for Batch SMS, P4 for Promotional SMS etc.,). Based upon the priority of the SMS, such
SMSes are to be pushed to appropriate BSNL’s SMSC Ports/Accounts.
9.7.1.8 It will be the responsibility of the System Integrator to pull the SMS
from customers’ servers and to push them to BSNL SMSC with an average delay of only 2 to
3 seconds in his server. The maximum delay should not exceed 5 Seconds failing which may
attract penalty to the System Integrator.
9.7.1.9 It will be the responsibility of the System Integrator to pull the SMS
from customers’ servers and to push them to BSNL SMSC with an average delay of only 2 to
3 seconds in his server. The maximum delay should not exceed 5 Seconds failing which may
attract penalty to the System Integrator.
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9.7.1.10 The System Integrator will be responsible to take care of the
SMPP connectivity from his Server to SMSC as well as Customer’s Server. An
Overall uptime of 99.95% is to bemaintained failing which will attract penalty.
9.7.1.11 As per TCCCPR regulations, for Promotional SMS, the System
Integrator should be connected to dedicated SMSC Port allotted for Promotional
SMS for DND Scrubbing. First level DND Scrubbing should be done by the System
Integrator before pushing the messages to Promotional SMS Port at BSNL SMSC.
9.7.1.12 System Integrator shall provide necessary Two-Way SMS
solution for The Financial organization’s requirement such as Mobile Banking,
Interactive SMS etc., using the existing/new long code provided by BSNL/The
Financial organization. If desired, BSNL may request the selected System Integrator to
arrange long code from Other Operator at extra cost (if any).
9.7.1.13 Any add-on services such as Missed Call Alert, Interactive SMS
solution etc., are to be provided by the System Integrator at free of cost (if required) as
and when requested by The Financial organization during the contract period.
9.7.1.14 The System Integrator should provide 24*7*365 Off-site
support. System Integrator should submit escalation matrix of the support team in their
technical Proposal. As and when desired by The Financial organization/BSNL,
qualified on-site engineers/managers shall be deputed at The Financial
organization/BSNL premises for on-site support on temporary/regular basis.
9.7.1.15 System Integrator should have backend Software, Hardware and
other technical team at their office to incorporate new features as requested by the
customer from time to time and for bug fixing and any kind of support.
9.7.1.16 Necessary Back up devices are to be provided at The Financial
organization’s DC and minimum 1 year Back up is to be maintained. Old backup is to
be handed over to the Customer.
9.7.1.17 System Integrator should provide integration support with all
user departments of The Financial organization with respect to SMS Solution not
limited to https, SMPP, file upload (various formats) and other interfaces as required
by The Financial organization from time to time.
9.8 CONCLUSION
During the discussion of the chapter we have discussed the following points of
Enterprise Business which is managed by Franchisee/TIP:
a. Enterprise Wi-Fi service
b. Establishment and Maintenance of Customized Wide Area Network (WAN)
c. PABX
d. Network System Solution
e. Bulk SMS solution etc.
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