Difference between NGN & TDM Network

Hi,

In today's scenario, NGN is gradually replacing traditional TDM network,

So below are the some differences / benefits of NGN system w.r.t. traditional TDM

Keep checking for new posts till then HaPPy Reading.

ChEEeeerS
Telecom Tigers Team
telecomtigers@[Link]
[Link]

Posted by Telecom Tigers Blog at 9:21 PM 0 comments

Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Friday, October 15, 2010

Major Steps to Improve KPI !!!
Here are some parameters which highly effects network performance like -

 Network Overall ASR (Answer Seizure Ration).

 Location Update Success Rate.
 Paging Success Rate.
 Handover Success Rate.

& these parameters needs to be monitored continuously for smooth

functioning of mobile network.

Network Overall ASR - This is very-very important parameter in telecom

industry because it directly relates to Revenue (Money), So we need to keep
close monitoring & take certain precautions to keep it higher-n-higher.
Its standard value lies between 35% - 45% & rest % is left considering
Subscriber Behaviour i.e. miss call, no answers after long ring, etc.

Points to be closely monitoring for improvement of ASR :-

1. POI Utilization (whether more E1's are required or not).

2. Routing of Levels.
3. Selection of Routes.
4. CIC (Circuit) Hunting e.g. Odd-Even Selection or Sequential Routing.
5. Unallocated Numbers, e.g. subscriber which are churn (De-active), delete those
numbers on regular basis.
6. Proper Announcements, so that Subscriber won't re-attempts again n again.
7. CIC matching should be there with other operator.
8. Network Equipment like MSC, etc. should not be congested n Many more......
9. 1). CIC Selection method with other exchange should be different.
2). Length of B-number.
3). RBT (RING BACK TONE) timer should not be less than 45 seconds.
4). Cranck-back routing (ALTERNATE PATH ROUTING) should be enable for
Congested codes/routes.
5). Utilization should be kept below 70%.
6). Regular audits for B-number/Routing case.
10. Top 15 cause codes – check release code then you can find abnormal release which can
affect ASR

Location Update Success Rate - It is Number of Successful Location

Updates w.r.t. Total Number of Location Updating Attempts. This parameter
is calculated for 24 Hrs. Its standard value >= 95%.

LUSR = 100*(Number of successful location updates) / (Total number of

location updating attempts)
Where above both paramteres are considered for Non-Registered Mobile
Subscribers & Already Registered Mobile Subscribers.

Major contributor for decreasing LUSR -

1. Congestion in C7 Signaling.
2. Incorrect IMSI definition of IMSI analysis in Switch.
3. Incorrect roaming subscriber definition in Switch.
4. SDDCH Congestion.
5. LU timers setting.
6. Network Synchronization problem.

Improvement Plan -

1. Continuous Monitor of C7 Signaling utilization and it should be

optimize as much as possible.
2. Correct definition of IMSI and Roaming Subscriber.
3. For Narrow Band Signaling the utilization should not go above 0.3 Erl.
& for High Speed Signaling the utilization must be kept below 0.4
Erl(Per time slot).

Paging Success Rate - It is rate of successful page responses to First and

Repeated Page Attempts to a location area w.r.t. Number of Initial and
Repeated Page Attempts to a location area. This parameter is calculated for
24 Hours. Its standard value >= 92%.

LSR = (Number of Page responses to first page to an LA + Number of Page

responses to repeated page to an LA) / Number of Page Attempts to an LA
(Location Area).
OR
LSR = (first paging response+ repeated paging response)*100/first paging
request).

Major contributor for decreasing PSR -

1. Improper Paging / LU (Location Update) related parameter setting.

2. O&M issue i.e Outages.
3. Lower RACH success rate.
4. Air Interface Issues like Interference, SDCCH Congestion, etc.
5. Footprints.
6. Paging overload on BSC i.e. paging capacity of BSC compared with
the actual paging.
7. Congestion on A-bis interface i.e. Paging command from BSC is
delivered to BTS via A-bis.
Improvement Plan -

1. Paging / LU timers setting, like Paging Timers in MSC must longer

than Paging Timer in BSC (prolonging 1st and repeated page) and
also paging strategy (local vs global), or repeated page on/off.
2. LAC optimization.
3. Paging / LU related parameter setting like increasing paging capacity
through uncombined BCCH, changing Access grant and MFRMS
(multiframe) parameters.
4. Address Coverage issues.
5. Check Discard/Paging queue on cell level.

HandOver Success Rate - It is the mechanism that transfers an ongoing

call from one cell to another as a user moves through the coverage area of a
cellular system.
The handover success rate shows the percentage of successful
handovers of all handover attempts. A handover attempt is when a handover
command is sent to the mobile.

HOSR = (Successful Incoming Inter-Cell Handover + Successful Outgoing

Inter-Cell Handover) / (Incoming Inter-Cell Handover + Outgoing Inter-Cell
Handover)

Major contributor for decreasing HOSR -

1. C/I Ratio (Carrier-to-Interference ratio), Lower value gives Worst

Connection Quality.
2. High Interference, Co-Channel or adjacent i.e., High Bit-Error Ratio.
3. Bad Antenna Installation.
4. Bad Radio Coverage.
5. Incorrect Locating Parameter Settings.
6. Insufficient Planning in Certain Areas.
7. Repeated Handover between two base stations, caused by rapid
fluctuations in the received signal strengths from both base stations.
8. Un-Necessary Handover often leads to Increased Signaling Traffic.

Improvement Plan -

1. Updating & Optimising Neighbours List.

2. Removing Neighbours which have fewer no of HOs and cells having
poor HOSR,
 3. Avoid same BCCH+BSIC Combination.

More Information from Readers are Expected !!!

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 11:09 PM 2 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Saturday, August 28, 2010

Standard ISUP Release Causes

ISUP Release Cause Values and their meanings

REL
Cause Meaning
Value

NORMAL EVENT CLASS

1 Unallocated number
2 No route to specified transit network
3 No route to destination
4 Send special information tone
5 Misdialled trunk prefix
6 Channel unacceptable
7 Call awarded and being delivered in an established channel
16 Normal call clearing
17 User busy
18 No user responding
19 No answer from user (user alerted)
20 Subscriber absent
21 Call rejected
22 Number changed
26 Non selected user clearing
27 Destination out of order
28 Invalid format (address incomplete)
29 Facility rejected
30 Response to status enquiry
31 Normal, unspecified

RESOURCE UNAVAILABLE CLASS

34 No circuit/channel available
38 Network out of order
41 Temporary failure
42 Switching equipment congestion
43 Access information discarded
44 Request circuit/channel not available
47 Resource unavailable, unspecified

SERVICE/OPTION NOT AVAILABLE CLASS

49 Quality of service unavailable
50 Requested facility not subscribed
55 Incoming calls barred within CUG
57 Bearer capability not authorized
58 Bearer capability not presently available
63 Service or option not available, unspecified

SERVICE/OPTION NOT IMPLEMENTED CLASS

65 Bearer capability not implemented
69 Requested facility not implemented
Only restricted digital
70 information
bearer capability is available
79 Service or option not implemented, unspecified

INVALID MESSAGE CLASS

81 Invalid call reference value
82 Identified channel does not exist
83 A suspend call existing but this call identity does not
84 Call identity in use
85 No call suspended
86 Call having the requested call identity has been cleared
87 User not member of CUG
88 Incompatible destination
91 Invalid transit network selection
95 Invalid message, unspecified

PROTOCOL ERROR CLASS

96 Mandatory information element is missing
 97 Message type non-existing or not implemented
Message not compatible with call state
98 or message type non-
existing or not implemented
99 Information element non-existing or not implemented
100 Invalid information element contents
101 Message not compatible call state
102 Recovery on timer expiry
103 Parameter non-existent or not implemented
109 Unrecognized message has been passed on
110 Message with unrecognized parameter discarded
111 Protocol error, unspecified

INTERWORK CLASS
127 Interworking, unspecified

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 9:51 PM 0 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Saturday, March 27, 2010

KPI - Key Performance Indicators
Hi,

KPI tells the performance of a network on a daily/weekly/monthly

basis, which helps to improve the network, so that operator & customer both
enjoys the service at its most.

Key Performance Indicators for Telecom Industry are :-

Systems and Network Performance Analysis / Capacity Planning

 Availability
 Grade of service
 Service life of equipment
 Bit error ratio (data, bits & elements transfer)
 Bit rate (data, bits and elements transfer)
 Downtime / Time out of service
  Call completion ratio
 Cost of support systems
 Cost of operational systems
 Average call length
 Analysis of ASR routes
 Network traffic, congestion
 Idle time on network
 Dropped calls

Quality / Usage (Airtime): Analysis of the volume of successful calls

 Mean Opinion Score

 Service
 Duration of calls
 Billed amount on each call

Coverage

 % of land covered with services

 % of population covered with services
 Average land unavailable to services
 Average population unavailable to services
 Access to customer service

Faults and complains (Trouble tickets analysis)

 % of open and level of escalation priority required

 % closed
 Mean time to resolved
 Work in progress
 Customer service level statistics

Customer Analysis

 Customer segmentation
 Analysis of subscriptions
 Top N customers
 Churn (No. of Subscriber who stopped using Services or left particular
network)

ASR (Answer Seizure Ratio) - Number of successfully answered calls

divided by the total number of calls attempted (seizures) multiplied by 100.
(Answer / Seizure) * 100 = Answer Seizure Ratio.
Standard Value = 40% - 45%.

MOU (Minutes of Usage) per Subscriber – It calculates the Total Minutes

used in a Network divided by the number of subscribers.

CCR (Call Completion Ratio) - Total no of calls completed / Total no of

calls attempted * 100%
Higher the ratio is better.
Standard Value > 98%.

LUSR (Location Update Success Rate) - Its a ratio of no. of times mobiles
update its location successfully to the no. of times mobiles request network
for Location update.
LUSR = (Location Update Success / Location Update Request)*100.
Standard Value >= 98%.

PSR (Paging Success Rate) - Its a ratio of [Link] times network successfully
find the mobiles to the [Link] times network tries to locate the mobiles within
its area.
PSR = ([Link] Network Paging Response / [Link] Network Paging
Attempts)*100.
Standard Value >= 92%.

More Information from Readers are Expected !!!

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 7:12 PM 6 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Monday, January 25, 2010

What is Sigtran
SIGTRAN (SIGnaling TRANsport) :-
It is a set of protocols defined by IETF to transport SS7 messages over IP
networks. It allows IP networks to inter-work with the Public Switched
Telephone Network (PSTN) and vice versa.
The telco switch sends SS7 signals to a signaling gateway (SG) that
converts them into SIGTRAN packets, which travel over IP to the next
signaling gateway or to a softswitch if the destination is not another PSTN.

Sigtran Protocol Suite is made up of a new Transport Layer -- the

Stream Control Transmission Protocol (SCTP) & a set of User
Adaption (UA) layers, which mimic the services of lower layers of SS7 &
ISDN.

Why SIGTRAN :

 Ordered, Reliable Transfer.

 Redundancy in case of Link Failure.
 Low Loss & Delay.

The key components in the SIGTRAN architecture are :-

 MGC–Media Gateway Controller, responsible for mediating call

control (between the SG (Signaling Gateway) and MG (Media
Gateway)) and controlling access from the IP world to/from the PSTN.
 SG–Signaling Gateway, responsible for interfacing to the SS7
network and passing signaling messages to the IP nodes.
 MG–Media Gateway, responsible for packetization of voice traffic and
transmitting the traffic towards the destination.
 IP SCP – an IP-enabled Service Control Point (SCP). This exists wholly
within the IP network, but is addressable from the SS7 network.
 IP Phone – generically referred to as a “terminal.”

SIGTRAN protocol stack consists of 3 components :

 A standard IP layer.
 A common signaling transport protocol, Stream Control Transmission
Protocol (SCTP)
 An Adaptation layer, like - M2PA, M2UA, M3UA, and SUA.

SCTP :- Stream Control Transmission Protocol

SCTP is designed to transport SS7 signaling messages over IP
networks. It operates directly on top of IP at the same level as TCP. SCTP's
basic service is connection oriented reliable transfer of messages between
peer SCTP users. Its aim of designing, is to address the Shortcomings of TCP.
SCTP is a general purpose protocol, a replacement for TCP.

SCTP has the following set of features :-

 It is a Unicast Protocol - data exchange is between two known

endpoints.
 It defines timers of much shorter duration than TCP.
 SCTP uses periodic heart-beat messages to confirm the status of each
end point.
 It provide reliable transport of data - detecting when data is corrupt or
out of sequence, and performing repair as necessary.
 It is Rate-Adaptive, responding to network congestion
 It support Multi-Homing - Each SCTP endpoint may be known by
multiple IP addresses. Routing to one address is independent of all
others, & if one route fails, another will be used.
 It uses an initialization process, based on cookies, to prevent denial-of-
service attacks.
 It supports Bundling, where a single SCTP message may carries
multiple "Chunks" of data, each of which contains a whole signaling
message.
 It support fragmentation, where a single signaling message may be
split into multiple SCTP messages in order to be accommodated
within a underlying PDU.
 It is message-oriented, defining structured frame of data, on the other
hand, TCP imposes no structured on the transmitted stream of bytes.
 It has a multi-streaming capacity, data is split into multiple streams,
each with independent sequenced delivery, TCP has no such feature.

Sigtran Adaptation Layers serves common purposes like :-

 To carry upper layer Signaling Protocol over a reliable IP-based

transport.
 To provide same class of services offered at the interface of PSTN
equivalent.
 Transparent.
 To remove as much need for the lower SS7 layers as possible.

Sigtran currently defines SIX adaption layers :-

1. M2UA :- It provides the services of MTP2 in a Client-Server Situation,

such as SG to MGC. Its user would be MTP3.
 2. M2PA :- It provides the services of MTP2 in a Peer-to-Peer Situation,
such as SG to SG Connections. Its user would be MTP3.
3. M3UA :- It provides the services of MTP3 in both a Client-Server
Situation (SG to MGC) & Peer-to-Peer Architecture, Its user would be
SCCP and/or ISUP.
4. SUA :- It provides the services of SCCP in a Peer-to-Peer Situation,
such as SG to IP SCP Connections. Its user would be TCAP.
5. IUA :- It provides the services of the ISDN Data Link Layer (LAPD), Its
user would be an ISDN Layer 3 (Q.931) entity.
6. V5UA :- It provides the services of the V.5.2 Protocol.

More Information from Readers are Expected !!!

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 10:46 PM 0 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Thursday, December 24, 2009

NGN - Next Generation Network
The general idea behind NGN is that one network transports all
information and services (voice, data, and all sorts of media such as video)
by encapsulating these into packets, like it is on the Internet. NGNs start
making sense when voice, data, video are all in IP format.

Basically, the core network should have a common service delivery

architecture, with any access network hanging off the core.
Some important service characteristics for NGN would be real-time, multi-
media communications, more personal intelligence, more network
intelligence, more simplicity for users, personal service customization and
management.

It will benefit from its advanced control, management, and

signaling capabilities, enabling a much broader array of service types,
such as

 Specialized resource services (provision and management of

transcoders, multimedia multipoint-conferencing bridges, processing
and storage services),
 Middleware services (brokering, security, licensing, transactions),
  Application-specific services (business applications, e-commerce
applications, supply-chain management applications, interactive
video games),
 Content-provision services (electronic training, information push
services).

Next generation IP networks or NGN IP will be the key enabler of

mobility and convergence. This would mean that convergence would not just
be limited to wired networks. WLAN too can run voice, data, & video.
NGN IP would help organizations achieve new levels of enhanced
productivity, reduced operational costs, increased operational efficiency and
better overall profitability.
NGN IP would also support new levels of personal mobility, allowing
for seamless integration of fixed and mobile networks. And, for enterprises it
helps to scale their network architecture and prioritize bandwidth usage, and
reduce network management complexities.
In an NGN environment, almost 60–70% bandwidth would be
reserved for data, and as voice would be on the same pipe, it would
come almost free. With no additional costs for using voice, the usage of
voice services would increase. And it is voice over IP (VoIP) that would be the
killer application for NGNs.

Fundamental to Next Generation Networking :-

 Packet-Based Data Transfer.

 Separate control functions for bearer capabilities, calls/sessions and
applications/services.
 De-coupling of service provision from the network, and provision of
open interfaces.
 Support for a wide range of service applications and mechanisms
based on service building blocks (including real-time/streaming/non-
real-time services and multi-media).
 Broadband capabilities with end-to-end QoS and transparency.
 Interworking with legacy networks via open interfaces.
 Generalized mobility.
 Converged services between Fixed and Mobile networks.

Issues to be kept in mind while planning for NGN deployment :-

1. Latency (Delay)
2. Jitter
3. Bandwidth
4. Packet Loss
5. Reliability
6. Security
7. Inter-operability
NGN involves three main architectural changes that need to be looked at
separately :-

 In the Core Network, NGN implies a consolidation of several (dedicated

or overlay) transport networks each historically built for a different
service into one core transport network (often based on IP and
Ethernet). It implies amongst others the migration of voice from a
circuit-switched architecture (PSTN) to VoIP, and also migration of
legacy services such as X.25, Frame Relay (either commercial
migration of the customer to a new service like IP VPN, or technical
emigration by emulation of the "legacy service" on the NGN).
 In the Wired Access Network, NGN implies the migration from the dual
system of legacy voice next to xDSL setup in the local exchanges to
a converged setup in which the DSLAMs integrate voice ports or VoIP,
making it possible to remove the voice switching infrastructure from
the exchange.
 In Cable Access Network, NGN convergence implies migration of
constant bit rate voice to PacketCable (CableLabs standards that
provide VoIP and SIP services).

PacketCable Networks use the Internet Protocol (IP) to enable a wide range
of multimedia services, such as Voice over IP (IP telephony), multimedia
conferencing, interactive gaming, and general multimedia applications.

NGN Technology Components :-

 NGNs are based on Internet technologies including Internet Protocol

(IP) and Multiprotocol Label Switching (MPLS). At the application
level, Session Initiation Protocol (SIP) seems to be taking over from
ITU-T H.323.
 For voice applications, one of the most important devices in NGN is a
Softswitch - a programmable device that controls Voice over IP
(VoIP) calls. It enables correct integration of different protocols within
NGN. The most important function of the Softswitch is creating the
interface to the existing telephone network, PSTN, through Signalling
Gateways and Media Gateways.
 Gatekeeper - This was originally a VoIP device, which converted
(using gateways) voice and data from their analog or digital
switched-circuit form (PSTN, SS7) to the packet-based one (IP). It
controlled one or more gateways. As soon as this kind of device
started using the Media Gateway Control Protocol, the name was
changed to Media Gateway Controller (MGC).
 IP Multimedia Subsystem (IMS) is a standardized NGN architecture
for an Internet media-services capability.
SoftSwitch :- It's a central device in a telecommunications network which
connects calls from one phone line to another, entirely by means of software
running on a computer system. This work was formerly carried out by
hardware, with physical switchboards to route the calls.
It is typically used to control connections at the junction point
between circuit and packet networks. A single device containing both the
switching logic and the switching fabric can be used for this purpose.
however, modern technology has led to a preference for decomposing this
device into a Call Agent and a Media Gateway.
Call Agent takes care of functions like billing, call routing,
signalling, call services and so on and is the 'brains' of the outfit. A Call
Agent may control several different Media Gateways in geographically
dispersed areas over a TCP/IP link.
Media Gateway connects different types of digital media stream
together to create an end-to-end path for the media (voice and data) in the
call. It may have interfaces to connect to traditional PSTN networks like DS1
or DS3 ports (E1 or STM1), it may have interfaces to connect to ATM and IP
networks and in the modern system will have Ethernet interfaces to connect
VoIP calls. The call agent will instruct the media gateway to connect media
streams between these interfaces to connect the call.
In more recent times (i.e., in IP Multimedia Subsystem or IMS), the
Softswitch element is represented by the Media Gateway Controller (MGC)
element, and the term "Softswitch" is rarely used in the IMS context, but
another word of AGCF(Access Gateway Control Function).
Feature Server, often built into a call agent/softswitch, is the
functional component that provides call-related features. Capabilities such as
call forwarding, call waiting, and last call return, if implemented in the
network, are implemented in the feature server. The feature server works
closely with the call agent, and may call upon the media server to provide
these services. These features do not require the subscriber to explicitly
request them but tend to be triggered within the call handling logic.

More Information from Readers are Expected !!!

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 6:01 PM 1 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Wednesday, December 16, 2009

Location Update Procedure
In order to make a mobile terminated call, The GSM network
should know the location of the MS (Mobile Station), despite of its movement.
For this purpose the MS periodically reports its location to the network using
the Location Update procedure.

Location Area (LA)

A GSM network is divided into cells. A group of cells is considered
a location area. A mobile phone in motion keeps the network informed
about changes in the location area. If the mobile moves from a cell in one
location area to a cell in another location area, the mobile phone should
perform a location area update to inform the network about the exact
location of the mobile phone.

The Location Update procedure is performed:

 When the MS has been switched off and wants to become active, or
 When it is active but not involved in a call, and it moves from one
location area to another, or
 After a regular time interval.

Location registration takes place when a mobile station is turned on. This
is also known as IMSI Attach because as soon as the mobile station is
switched on it informs the Visitor Location Register (VLR) that it is now back
in service and is able to receive calls. As a result of a successful registration,
the network sends the mobile station two numbers that are stored in the SIM
(Subscriber Identity Module) card of the mobile station.

These two numbers are :-

1. Location Area Identity (LAI)

 2. Temporary Mobile Subscriber Identity (TMSI).

The network, via the control channels of the air interface, sends the LAI. The
TMSI is used for security purposes, so that the IMSI of a subscriber does not
have to be transmitted over the air interface. The TMSI is a temporary
identity, which regularly gets changed.

 A Location Area Identity (LAI) is a globally unique number.

 A Location Area Code (LAC) is only unique in a particular network.

Every time the mobile receives data through the control channels, it reads
the LAI and compares it with the LAI stored in its SIM card. A generic location
update is performed if they are different. The mobile starts a Location
Update process by accessing the MSC/VLR that sent the location data.
A channel request message is sent that contains the subscriber identity (i.e.
IMSI/TMSI) and the LAI stored in the SIM card. When the target MSC/VLR
receives the request, it reads the old LAI which identifies
the MSC/VLR that has served the mobile phone up to this point. A signalling
connection is established between the two MSC/VLRs and the subscriber’s
IMSI is transferred from the old MSC to the new MSC. Using this IMSI, the new
MSC requests the subscriber data from the HLR and then updates the VLR
and HLR after successful authentication.

Periodic location update is carried out when the network does not receive
any location update request from the mobile in a specified time. Such a
situation is created when a mobile is switched on but no traffic is carried, in
which case the mobile is only reading and measuring the information sent by
the network. If the subscriber is moving within a single location area, there is
no need to send a location update request.
A timer controls the periodic updates and the operator of the VLR sets the
timer value. The network broadcasts this timer value so that a mobile station
knows the periodic location update timer values.
Therefore, when the set time is up, the mobile station initiates a registration
process by sending a location update request signal. The VLR receives the
request and confirms the registration of the mobile in
the same location area. If the mobile station does not follow this procedure,
it could be that the batteries of the mobile are exhausted or the subscriber is
in an area where there is no network coverage. In such
a case, the VLR changes the location data of the mobile station to
“unknown”.
The Location Update process consists of the following phases

 Request for service; the MS detects that it has entered a new Location
Area and requests to update its location. The new MSC/VLR identifies
the MS.
 Authentication - The new MSC/VLR requests to the AUC for
authentication parameters (SRES, Kc, RAND). Using these parameters
the MS is authenticated.
 Ciphering - Using the parameters which were made available earlier
during the authentication the uplink and the downlink are ciphered.
 Update HLR/VLR - The new MSC/VLR requests to update the MS
location in the HLR. The MS is de-registered in the old VLR.
 TMSI re-allocation - The MS is assigned a new TMSI.

1. The MS detects that it has entered a new Location Area and

transmits a Channel Request message over the Random Access
Channel (RACH).
2. Once the BSS receives the Channel Request message, it allocates a
Stand-alone Dedicated Control Channel (SDCCH) and forwards this
channel assignment information to the MS over the Access Grant
Channel (AGCH). It is over the SDCCH that the MS will communicate
with the BSS and MSC.
3. The MS transmits a location update request message to the BSS over
the SDCCH. Included in this message are the MS Temporary Mobile
Subscriber Identity (TMSI) and the old Location Area Identification
(oldLAI). The MS can identify itself either with its IMSI or TMSI. The
BSS forwards the location update request message to the MSC.
4. The VLR analyzes the LAI supplied in the message and determines
that the TMSI received is associated with a different VLR (old VLR). In
order to proceed with the registration, the IMSI of the MS must be
determined. The new VLR derives the identity of the old VLR by using
the received LAI, supplied in the location update request message. It
also requests the old VLR to supply the IMSI for a particular TMSI.
5. The new VLR sends a request to the HLR/AUC (Authentication Center)
requesting the “authentication triplets” (RAND, SRES, and Kc)
available for the specified IMSI.
6. The AUC, using the IMSI, extracts the subscriber's authentication key
(Ki). The AUC then generates a random number (RAND), applies the
Ki and RAND to both the authentication algorithm (A3) and the cipher
key generation algorithm (A8) to produce an authentication Signed
Response (SRES) and a Cipher Key (Kc). The AUC then returns to the
new VLR an authentication triplet: RAND, SRES, and Kc.
7. The MSC/VLR keeps the two parameters Kc and SRES for later use
and then sends a message to the MS. The MS reads its
Authentication key (Ki) from the SIM, applies the received random
number (RAND) and Ki to both its Authentication Algorithm (A3) and
Cipher key generation Algorithm (A8) to produce an authentication
Signed Response (SRES) and Cipher Key (Kc). The MS saves Kc for
later, and will use Kc when it receives command to cipher the
channel.
8. The MS returns the generated SRES to the MSC/VLR. The VLR
compares the SRES returned from the MS with the expected SRES
received earlier from the AUC. If equal, the mobile passes
authentication. If unequal, all signaling activities will be aborted.
9. The new MSC/VLR requests the BSS to cipher the radio channel.
Included in this message is the Cipher Key (Kc), which was made
available earlier during the authentication.
10. The BSS retrieves the cipher key, Kc, from the message and then
transmits a request to the MS requesting it to begin ciphering the
uplink channel.
11. The MS uses the cipher key generated previously when it was
authenticated to cipher the uplink channel, and transmits a
confirmation over the ciphered channel to the BSS.
 12. The BSS upon ciphering the downlink channel sends a cipher
complete message to the MSC. At this point, we are ready to inform
the HLR that the MS is under control of a new VLR and that the MS
can be de-registered from the old VLR.
13. The new VLR sends a message to the HLR informing it that the
given IMSI has changed locations and can be reached by routing all
incoming calls to the VLR address included in the message.
14. The HLR requests the old VLR to remove the subscriber record
associated with the given IMSI. The request is acknowledged.
15. The HLR updates the new VLR with subscriber data (mobiles
subscriber’s customer profile).
16. The MSC forwards the location update accept message to the
MS. This message includes the new TMSI.
17. The MS retrieves the new TMSI value from the message and
updates its SIM with this new value. The mobile sends then an
update complete message back to the MSC.
18. The MSC requests from the BSS that the signaling connection be
released between the MSC and the MS.
19. The MSC releases its portion of the signaling connection when it
receives the clear complete message from the BSS.
20. The BSS sends a "radio resource" channel release message to
the MS and then frees up the Stand-alone Dedicated Control Channel
(SDCCH) that was allocated previously. The BSS then informs the
MSC that the signaling connections has been cleared.

More Information from Readers are Expected !!!

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 6:12 PM 11 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Sunday, November 8, 2009

STP (Signaling Transfer Point)
Hi,

There are different types of signaling points in SS7 Standards, like

Service Switching Point (SSP) - like MSC, Tandem Switch, etc.
Service Control Points (SCP) - like HLR, etc.
Signaling Transfer Point (STP).

These Signaling points provide access to the SS7 network, databases, &
transfer messages to other signaling points.

STP :- It is a vital element in SS7 network serving as a Signaling hub for the
transfer of digital data packets between network nodes.

It routes messages throughout the network using, call information & network
addressing structured within SS7 data packets.

It serves as dynamic router, controlling traffic flow & access to variety of SS7
nodes & network.

Functions of STP :-

1. Receives the MSU's & direct them to appropriate destination.

2. Network Management.
3. (ANSI to ITU) or (ITU to ANSI) protocol conversion.
4. Global Title Translation (GTT).
5. Measurement of Data.
6. Gateway Function.
7. Gateway Screening (GWS)
8. Local Number Portability (LNP).

Different Links used with STP :-

 A-Link - It provide STP/SSP or STP/SCP connectivity. Maximum of 16

links can be there in a link set or maximum of 32 links in combined
set.

 B-Link - It connects one STP to other STP of same hierarchical level.

Maximum of 8 links in quad configuration of link set.
 C-Link - Maximum of 16 links in a link set.
 D-Link - Maximumof 8 links in a link set.

 E Links - It connects a STP to other STP other then its Home STP &
provides an alternate route for SS7 messages if congestion occurs at
home STP.
  F Links - It provides SSP to SSP connectivity. It provides only Call
Setup/TearDown capability & it should be adjacent.

Why STP ???

 It provides Centralized Network Management, facilitating the delivery

of Intelligent Services throughout the network.
 Reduction in Signaling Terminal Hardware in SSPs / SCPs. (like MSC, IN,
HLR, etc.)
 Central Database for GTT at STP, Minimizes Errors.
 Efficient Routing of Messages.
 Flexible SS7 Network Management.
 Fast Integeration of New Nodes in the network.
 Easy Migration to Next Generation Networks like VOIP, Soft Switch, etc.

Expecting More Information from Readers.

Thanks & Regards
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 6:15 PM 0 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Thursday, November 5, 2009

What is E1 / T1
The PDH (plesiochronous Digital Hierarchy) has 2 primary communication
systems as its foundation.

These are,
T1 system based on 1544kbit/s that is recommended by ANSI &
E1 system based on 2048kbit/s that is recommended by ITU-T.

Common Characteristics :-

1. Both are having Same Sampling Frequency i.e. 8kHz.

2. In both (E1 & T1) Number of samples/telephone signal =
8000/sec.
3. In both (E1 & T1) Length of PCM Frame = 1/8000s = 125µs.
4. In both (E1 & T1) Number of Bits in each code word = 8.
5. In both (E1 & T1) Telephone Channel Bit Rate = 8000/s x 8 Bit =
64 kbit/s.

Differing Characteristics :-

1. In E1 Encoding/Decoding is followed by A-Law while in T1

Encoding/Decoding is followed by µ-Law.
2. In E1 - 13 Number of Segments in Characteristics while in T1 -
15 Number of Segments in Characteristics.
3. In E1 - 32 Number of Timeslots / PCM Frame while in T1 - 24
Number of Timeslots / PCM Frame.
4. In E1 - 8 x 32 = 256 number of bits / PCM Frame while in T1 - 8
x 24 + 1* = 193 number of bits / PCM Frame. (* Signifies an
additional bit).
5. In E1 - (125µs x 8)/256 = approx 3.9µs is the length of an 8-bit
Timeslot while in T1 - (125µs x 8)/193 = approx 5.2µs is the
length of an 8-bit Timeslot.
 6. In E1 - 8000/s x 256 bits = 2048kbit/s is the Bit Rate of Time-
Division Multiplexed Signal while in T1 - 8000/s x 193 bits =
1544kbit/s is the Bit Rate of Time-Division Multiplexed Signal.

Thanks & Regards

telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 11:36 PM 1 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Wednesday, October 14, 2009

What is GPRS
GPRS (General Packet Radio Service)

It is a non-voice service added to existing TDMA networks, It is an

enhancement to GSM or TDMA (IS-95) network. It uses existing cellular
network infrastructure with software upgrade at base stations and the
addition of a GPRS Gateway that connects the GPRS network to the Internet.
TDMA is the underlying transport mechanism used by GSM networks.
GPRS provides the transmission of IP packets over existing cellular networks.

GPRS Architecture -
Components :-

1. SGSN (Serving GPRS Support Node) - It monitors the state of the

mobile station and tracks its movements within a given geographical
area. It is also responsible for establishing and managing the data
connections between the mobile user and the destination network.
2. GGSN (Gateway GPRS Support node) - It provides the point of
attachment between the GPRS domain and external data networks
such as the internet and Corporate Intranets. Each external network
is given a unique Access Point Name (APN) which is used by the
mobile user to establish the connection to the required destination
network.
3. WAP Servers - Its used for General Information Services like Train
Timetables, etc.
4. RADIUS Server - Remote Access Dial-in User Server.

How GPRS Works?

Because it is a packet switched network, a GPRS user station doesn't

occupy a dedicated path during an Internet connection. However, each end
user station (e.g. mobile phone) is allocated several time slots out of 8
GSM/TDMA available time-slots for GPRS service. Each time slot has a
maximum capacity of 14.4 kbps, depending on how many time slots are
allocated for the downlink (from a base station to a user station) and the
uplink (from a user station to a base station), GPRS devices are divided into
multi-slot classes.
A multi-slot class is often represented by the number of downlink and uplink
slots. For example, Class 10 is also known as Class 4+2. While active slots
indicate the maximum number of slots that can be allocated for both
downlink and uplink in a specific class.

Class Types :-

 2+1 (two slots for download + 1 for upload)

 3+1 (three slots for download + 1 for upload)
 4+1 (four slots for download + 1 for upload)

GPRS is a mobile data upgrade to a GSM mobile phone network. This

provides users with packet data services (similar to the Internet) using the
GSM digital radio network. Each voice circuit in GSM transmits the speech on
a secure 14kbps digital radio link between the mobile phone and a nearby
GSM transceiver station.

The GPRS service joins together multiple speech channels to provide higher
bandwidth data connections for GPRS data users. The radio bandwidth
remains the same, it is just shared between the voice users and the data
users. The network operator has the choice of prioritizing one or the other.
GPRS users will also benefit from being able to use GPRS while traveling as
the GSM system should transparently hand over the GPRS connection from
one base station to another.

GPRS Roaming - There are two type of GPRS Roaming

Home Network Roaming - Here all data is transmitted from wherever you
connect to a GPRS network to your home GPRS network where it is
connected to the Internet or your company LAN as if you were indeed in your
home country.

Local Network Roaming - Data is just connected to a local Internet

connection point and will be subject to local conditions for security and
performance.

Radio Interface
Each GSM radio transceiver uses Time Division Multiplexing to deliver
eight voice circuits on one radio channel. Each radio site may have one or
more transceivers to provide sufficient channels to end users (maximum
numbers are limited by many factors including - operators radio license,
interference with other nearby GSM cells, cost of equipment, capacity of
radio site infrastructure, etc.)

Each 14kbps channel may be shared by multiple 'connected' GPRS users

(many users will be connected to the network but transmitting very little
data). As a user's data requirements grow, they will use more of the
available capacity within that timeslot, and then more available timeslots up
to the maximum available or the maximum supported by their device.

In general the higher the data rate, the more power the mobile device will
use and the shorter the battery life and the higher the transmitted RF power.

GPRS Mobile devices

The key use for GPRS is to send and receive data to a computer application
such as Email, web browsing or even telemetry (telemetry refers to devices
not being controlled by humans such as cash point machines or traffic
monitoring cameras etc.). To use GPRS the service is 'dialed' in a similar
manner to a standard data call (though there is no phone no.) at which point
the user is 'attached' and an IP address is allocated. From then on data can
flow to and from the Internet until either the network unattaches you (maybe
because of a time-out, fault or congestion) or you manually unattach.

Some of the key issues are:

 Using GPRS will not stop you making or receiving voice [Link]
phones will usually suspend the data session while a voice call takes
place.
 Battery life will be reduced when using GPRS.
 Do not keep it near your ear for long periods while data transfers are
taking place.

As a business GPRS user you will have a choice of methods to connect to the
GPRS network - by far the most common method will be via the Internet. For
larger users you may connect your company LAN to the GPRS networks using
leased lines or Frame Relay virtual circuits.

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 5:20 PM 0 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms
Tuesday, October 13, 2009
When Mobile is turned ON
Initial Process When Mobile is turned ON :-

Network Attachment
Network attachment is the process of selecting an appropriate cell (radio
frequency) by the mobile station to provide the available services, and
making its location known to the network. The process starts when the
mobile station is switched on, and ends when the mobile station enters the
idle mode. In idle mode the mobile station does not have a traffic channel
allocated to make or receive a call, but the Public Land Mobile Network
(PLMN) is aware of the existence of the mobile station within the chosen cell.

The network attachment process consists of the following tasks:

Cell identification - When a mobile station is switched on, it attempts to

make contact with a GSM PLMN by performing the following actions:

• Measure the BCCH (Broadcast Control Channel) channels.

• Search for a suitable cell.

The mobile station measures the signal strength of the BCCH channels
received. It stores in list information about 30 of these BCCH channels, such
as the signal strength and the frequency corresponding to these BCCH
channels.

PLMN selection - A suitable PLMN is chosen.

PLMN Area - This area is the geographical area in which a particular PLMN
operator provides land mobile communication services to the public. From
any position within a PLMN area, the mobile user can set up calls to another
user of the same network, or to a user of another network. The other
network may be a fixed network, another GSM PLMN, or another type of
PLMN. Other network users, and users of other networks, can also call a
mobile user who is active in the PLMN area.
When there are several PLMN operators, the geographical areas covered by
their networks may overlap. National borders normally limit the extent of a
PLMN area.

Cell selection - Cell selection is the process of selecting an appropriate cell

(radio frequency) by the mobile station to provide the available services.

Location update - In order to initiate a call or to receive a call, the mobile

station tunes to the control channel (BCCH plus CCCH – Common Control
Channel) of the chosen cell. Then, it registers its presence in this cell
(registration process) by means of a location updating procedure.

No suitable cell found - If the mobile station is unable to find a suitable

cell to access, it attempts to access a cell irrespective of the PLMN identity,
and enters a "limited service" state in which it can only attempt to make
emergency calls.

PLMN selection mechanism - The particular PLMN to be contacted can be

selected either in one of the following modes:
• Automatic mode - In automatic mode, the mobile station will choose which
PLMNs to try all by it. The automatic mode is based on the existence of the
preferred list, which is stored in a non-volatile memory in the SIM. This list
includes a number of PLMN identities in order of preference and is under
control of the user. The most preferred is usually the home PLMN. The list is
filled in by the user through a mechanism to be specified by the mobile
station manufacturer.
• Manual mode - In manual mode, the user is presented a list containing all
found PLMNs. The user chooses one of the PLMNs from the list.

Cell selection criteria - The mobile station attempts to find a suitable cell
by passing through the list in descending order of received signal strength;
the first BCCH channel which satisfies a set of requirements is selected. The
requirements that a cell must satisfy before a mobile station can provide
service from it, are:
• It should be a cell of the selected PLMN - The mobile station checks
whether the cell is part of the selected PLMN.
• It should not be "barred" - The PLMN operator may decide not to allow
mobile stations to access certain cells. These cells may, for example, only be
used for handover traffic. Barred cell information is broadcast on the BCCH to
instruct mobile stations not to access these cells.
• The radio path loss - This loss between the mobile station and the selected
BTS must be below a threshold set by the PLMN operator.

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 2:50 PM 0 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Thursday, October 8, 2009

GSM IDENTIFIERS
IMEI - International Mobile Equipment Identity
Its an internationally-unique serial number allocated to Mobile Station (MS)
hardware at the time of manufacture. It is registered by the network
operator & stored in Aunthetication Center (AuC) for Validation purpose. This
number consists of type approval code, final assembly code and serial
number of the mobile station. The network stores the IMEI numbers in the
Equipment Identity Register (EIR).

IMSI - International Mobile Subscriber Number

When a subscriber registered with a network operator, a unique subscriber
IMSI identifier is issued & stored in the SIM (Subscriber Identity Module) of
the MS. A MS can only functional fully, if it is operated with a valid SIM
inserted into a MS with a valid IMEI.
The total length of the IMSI is 15 digits and it consists of the following
elements:

MCC = Mobile Country Code (three digits)

MNC = Mobile Network Code (two digits)
MSIN = Mobile Subscriber Identification Number (ten digits)

TMSI - Temporary Mobile Subscriber Identity

Its used to protect the true identity (IMSI) of a subscriber. It is issued by &
stored within a VLR (not in the HLR) when an IMSI attach takes place or a
Location Area (LA) updates takes place. The issues TMSI only has validity
within a specific LA. The TMSI is used for security purposes, so that the IMSI
of a subscriber does not have to be transmitted over the air interface. Its a
temporary identity, which regularly gets changed.

MSISDN - Mobile Subscriber International ISDN number

It represents the True or dailed number associated with the subscriber. It is
assigned to the subscriber by the network operator at registration & is stored
in SIM. Its possible for a MS to hold multiple MSISDNs, each associated with
different services. It contains the following elements,

CC= Country code (33=France, 358=Finland, etc.)

NDC= National Destination Code
SN= Subscriber Number

MSRN - Mobile Subscriber Roaming Number

It is temporary, location-dependent ISDN number issued by the parent VLR to
all MSs within its area of responsibility. It is stored in VLR & associated HLR
but not in the MS. The MSRN is used by VLR associated MSC for call routing
within MSC/VLR service area.

CC = Country Code (of the visited country)

NDC = National Destination Code (of the serving network)
SN = Subscriber Number

LAI - Location Area Identity

Each location area within PLMN (Public Landline Mobile Network) has an
associated Intenationally unique identifier (LAI). The LAI is broadcasted
regularly by the BTSs on the Broadcast Control Channel (BCCH), thus
uniquely identify each cell within an associated location area (LA). It
structure is as follows,

MCC= Mobile Country Code (of the visited country)

MNC= Mobile Network Code (of the serving PLMN)
LAC= Location Area Code

CI - Cell Identifier
Its an identifier assigned to each cell within a network. however, CI is only
unique within a specific LA. When combined with the internationally unique
LAI for its associated LA, the Global Cell Identity (GCI) is produced which is
also internationally unique.

BSIC - Base Station Identity Code

Each BTS is issued with a unique identity, the BSIC & is used to distinguish
neighbouring BTSs. It is needed to identify that the frequency strength being
measured by the mobile station is coming from a particular base station.

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 5:55 PM 0 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Monday, October 5, 2009

What is GTT
GTT - Global Title Translation

First of all what is GT (Global Title)

*Global Title* (GT) is an *address* used in the *SCCP* protocol for routing
signaling messages on telecommunications networks. In theory, a global title
is a unique address which refers to only one destination, though in practice
destinations can change over time.

OR

The routing address within an ISDN network is termed the point code. Global
titles identify end terminals that may be beyond the ISDN network. To route
information the Global Title is translated into a point code, this is typically
conducted at a STP (Signalling Transfer Point).

GTT can also be defined in terms of routing as SCCP routing,

I mean for SCCP routing we usually requires GTT not only to STP but also to
Switch that supports SS7 protocol, e.g.

In any MSU (a Signal Unit or data in SS7 n/w) if we have an OPC as well as
DPC then we don't need to find out SSN for the destine node (that mean the
STP will not manipulate the data its receiving from the Originating node it
will simply route the data in the format in which it was receiving (this is
called MTP routing that we usually don't perform) while in case of SCCP
routing we do have an idea about only OPC not about DPC, to find out DPC
we extract the some other parameters such as SSN, NAI, NP, and TT from the
MSU (this all parameters comes under SCCP part of SS7 architecture (This is
called SCCP routing).

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 4:10 PM 1 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Wednesday, September 30, 2009

What is Handover
HANDOVER :-

In a mobile communications network, the subscriber can move around freely,

& to maintain the constant connection with subscriber, so that he can use all
his services without any disturbance is done with the help of Hand-Over.

The basic concept is simple - when the subscriber moves from the coverage
area of one cell to another, a new connection with the target cell has to be
set up and the connection with the old cell has to be released.

There are two reasons for performing a handover:

1. Handover due to measurements - It occurs when the quality or the

strength of the radio signal falls below certain parameters specified
in the BSC. The deterioration of the signal is detected by the constant
signal measurements carried out by both the mobile station and the
BTS. As a consequence, the connection is handed over to a cell with
a stronger signal.
2. Handover due to traffic reasons - It occurs when the traffic capacity
of a cell has reached its maximum or is approaching it. In such a
case, the mobile stations near the edges of the cell may be handed
over to neighbouring cells with less traffic load.

The decision to perform a handover is always made by the BSC that is

currently serving the subscriber, except for the handover for traffic reasons.
In the latter case the MSC makes the decision.
There are four different types of handover

 Intra cell - Intra BSC handover - The smallest of the handovers is the
intra cell handover where the Subscriber is handed over to another
traffic channel (generally in another frequency) within the same cell.
In this case the BSC controlling the cell makes the decision to
perform handover.

Inter cell - Intra BSC handover - The subscriber moves from cell 1 to
cell 2. In this case the handover process is controlled by BSC. The
traffic connection with cell 1 is released when the connection with
cell 2 is set up successfully.


Inter cell - Inter BSC handover - The subscriber moves from cell 2 to
cell 3, which is served by another BSC. In this case the handover
process is carried out by the MSC, but, the decision to make the
handover is still done by the first BSC. The connection with the first
BSC (and BTS) is released when the connection with the new BSC
(and BTS) is set up successfully.

 Inter MSC handover - The subscriber moves from a cell controlled by

one MSC/VLR to a cell in the domain of another MSC/VLR. This case is
a bit more complicated. Considering that the first MSC/VLR is
connected to the GMSC via a link that passes through PSTN lines, it is
 evident that the second MSC/VLR can not take over the first one just
like that. The MSC/VLR currently serving the subscriber (also known
as the anchor MSC), contacts the target MSC/VLR and the traffic
connection is transferred to the target MSC/VLR. As both MSCs are
part of the same network, the connection is established smoothly. It
is important to notice, however, that the target MSC and the source
MSC are two telephone exchanges. The call can be transferred
between two exchanges only if there is a telephone number
identifying the target MSC. Such a situation makes it necessary to
generate a new number, the Handover Number (HON).

Thanks
telecomtigers@[Link]
[Link]
Posted by Ashish Bhatia at 6:22 PM 0 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google
Buzz
Links to this post
Labels: Telecom Terms

Tuesday, September 29, 2009

Common Terms in Signaling Network
 Signaling Point (SP) - The switching or processing node in a
signaling network, where the function of SS7 are implemented. like
MSC in GSM or Transmit Exchange in PSTN. Every SP is identified by
a number determined by Network Identifier (NI) & Signaling Point
Code (SPC).
 Network Identifier (NI) - It provides discrimination between
International & National messages or between 2 national signaling
switches. SPC uniquely idetifies a SP within the signaling network.
 Originating Point (OP) - The SP at which the signaling message is
generated. It is identified by Originating Point Code (OPC).
 Destination Point (DP) - The SP to which the signaling message is
destined. It is identified by Destination Point Code (DPC).
 Signaling Transfer Point (STP) - This is a SP, that is able to route
signaling messages. In GSM, every SP is STP as soon as it routes the
signaling messages that must be delivered to different destination
point, in this case, only MTP is used, upper layers are not involved.
 Signaling Link (SL) - The packet data link that connects 2 SPs is a
signaling link. It is not neccessary to have a SL in each PCM line,
according to maximum load, there may be more than one SL
between 2 SPs depending on network structure.
 Link Set (LS) - A number of parallel SL connecting the same SP is
Signaling LinkSet.
 Signaling Route (SR) - The predetermined path a message takes
through the signaling network between OP & DP is called SR. A
linkset may carry several SR and hence convey traffic to several
destination.
 Signaling Route Set (SRS) - The signaling network groups all SR
that may be used for message transferring between OP & DP is SRS.
 Circuit Intetity Code (CIC) - The message requires for call setup
carries an identity. The identity carries a label with source &
destination of the message & also the identity of speech trunk the
message is refering to, this identity is termed as CIC.
Interview Questions ?

a. Explain the Basic Unit of Traffic ?

ANS: An Erlang is a unit of telecommunications traffic measurement. Strictly speaking,

an Erlang represents the continuous use of one voice path. In practice, it is used to
describe the total traffic volume of one hour.
For example, if a group of user made 30 calls in one hour, and each call had an average
call duration of 5 minutes, then the number of Erlangs this represents is worked out as
follows:

Minutes of traffic in the hour = number of calls x duration

Minutes of traffic in the hour = 30 x 5

Minutes of traffic in the hour = 150

Hours of traffic in the hour = 150 / 60
Hours of traffic in the hour = 2.5
Traffic figure = 2.5 Erlangs

2. Define term "End of Selection" ?

ANS: End-of-selection code used when the call set-up is regarded as

unsuccessful. It contains the reason for failure. It can be received and used for
analysis purpose until an answer is Received

3. Role of LRN/RN in MNP ?

ANS: It is a four digit code and describe the name and circle of specific
[Link] a exchange can identify where I should terminate the call.

4. why TMSI used for Paging ?

i. ANS: The TMSI is a temporary identifier with a length of four octets,
which is assigned to the mobile when it registers in an MSC. The TMSI
is used to increase subscriber confidentiality by avoiding sending the
IMSI on the air interface. Once a TMSI has been assigned to an MS, the
 MS shall use the TMSI to identify itself in the network.
Also it is used to reduced the processer load at the time of paging
beacuse its length is shorter than IMSI

5. Role of MSRN ?

Regarding your query.

Now in most of telecom Equipments atleast 5-6 hunting modes are available
1) Min
2) Max
3) Master/Slave
4) Random
5) Cylic

Now we come on congestion part which can happen due to wrong selection of
hunting mode.
Congestion is non availability of circuit in Trunk Group.
Trunk can be of three types
1) Incoming trunk
2) Outgoing Trunk
3) Bi Directional trunk

For first kind of trunk groups hunting selection will be done by peer end as you are
only receiving calls on that.
For second kind choose any hunting mode it doesn't matter there wont be any issue
of congestion.
Third kind is most common and here hunting mode really plays a major role.
Exp:
At your end you selected MIN and peer end also selected MIN while defining Trunk at
their end
In MIN MSS/MSC selects the minimum CIC for outgoing call.
If you have 1-30 CICs defined in trunk MSS will select first CIC i.e 1 if 1 is busy MSS
will select 2
now suppose third call arrives and CIC 1 in now free so MSS will again select CIC 1
even though CIC 3-30 were free.

Same will happening at peer end also as hunting mechanism selected in same.
 So there will many instants where both MSS/MSC select same CIC at same time,
which leads to call drop and pegged in congestion counter. This particular
phenomenon is call "DUAL SEIZURE" in telecom term.

Recommended hunting mode for Bi-directional trunk group is Master/Slave

In this mode two MSS/MSCs connected take control of EVEN and ODD CICs according
to there point code.
Higher Point code controls EVEN CICs and lower Point Code controls ODD CICs
EVEN controlling MSC wont select ODD CIC unless all EVEN CICs are busy same is
with ODD controlling MSC.
So in this case there is very less chance of Dual Seizure unless and until there is real
congestion on trunk.

IMP

Regarding your query - why 1 E1 contain only 32 slots?

•An E1 link is of 2.048M (Standard), with PCM coding.

•Human Voice ranges from 300 - 3300 Hz, - Maximum 4000 Hz.

•Nyquist Sampling Theorem/Principle says, sampling a signal (e.g., converting

from an analog signal to digital), the sampling frequency must be greater than
twice(2) the bandwidth of the input signal in order to be able to reconstruct the
original perfectly from the sampled version.

•So 8000 (4000 * 2) samples are taken per second for each voice signal.

•So 8000 samples/second for each voice signal. =1 sample/every 125 micro sec. =
8 bit.

•Data capacity required per individual = 8bits/sec. x 8000 samples = 64,000

bits/sec = 64 kbps i.e. Data Speed.

•Total data transferred per second = 32 channels x (64,000 bits/sec)

•Bandwidth = 2,048,000 bits / second = 2.048 Mbps = E1.

•We need to take atleast 8000 samples to faithfully recreate human voice.
So, as per above calculation, 32 slots are required in an E1.