You are on page 1of 18

3G Channels

Here is the list of different channels used in 3G network for transporting information at
much faster speed.
These are divided into Logical, Transport & Physical.
UTRA Channels
UTRA FDD radio interface has logical channels, which are mapped to
transport channels, which are again mapped to physical channels. Logical to
transport channel conversion happens in Medium Access Control (MAC) layer, which is
a lower sublayer in Data Link Layer (Layer 2).
Logical Channels:
Broadcast Control Channel (BCCH), Downlink (DL).
Paging Control Channel (PCCH), DL
Dedicated Control Channel (DCCH), UL/DL
Common Control Channel (CCCH), UL/DL
Dedicated Traffic Channel (DTCH), UL/DL
Common Traffic Channel (CTCH), Unidirectional (one to many)
Transport Channels:
Dedicated Transport Channel (DCH), UL/DL, mapped to DCCH and DTCH
Broadcast Channel (BCH), DL, mapped to BCCH
Forward Access Channel (FACH), DL, mapped to BCCH, CCCH, CTCH,
Paging Channel (PCH), DL, mapped to PCCH
Random Access Channel (RACH), UL, mapped to CCCH, DCCH and DTCH
Uplink Common Packet Channel (CPCH), UL, mapped to DCCH and DTCH
Downlink Shared Channel (DSCH), DL, mapped to DCCH and DTCH
Physical Channels:
Primary Common Control Physical Channel (PCCPCH), mapped to BCH
Secondary Common Control Physical Channel (SCCPCH), mapped to FACH,
Physical Random Access Channel (PRACH), mapped to RACH
Dedicated Physical Data Channel (DPDCH), mapped to DCH
Dedicated Physical Control Channel (DPCCH), mapped to DCH
Physical Downlink Shared Channel (PDSCH), mapped to DSCH
Physical Common Packet Channel (PCPCH), mapped to CPCH
Synchronization Channel (SCH)
Common Pilot Channel (CPICH)
Acquisition Indicator Channel (AICH)
Paging Indication Channel (PICH)
CPCH Status Indication Channel (CSICH)
Collision Detection/Channel Assignment Indication Channel (CD/CA-ICH)
Here we have tried to explain MNP call flow (Prepaid).
Terms which are used in explanation are already discussed earlier posts.
So let us begin with action.
Assumptions: A and B both are Vodafone Delhi Subscriber in different MSC/MSS coverage area.
Let us discuss call flow step by step.
1) Subscriber A (prepaid) call B.
2) Since A is prepaid first query(IDP) has to go IN/SCP with "calling party number" Subscriber A MSISDN and
"called party Subscriber" Subscriber B MSISDN. Here is change from normal prepaid call flow, in normal
case IDP would have gone straight to serving SCP but in case of MNP IDP will sent to MNP server.
3) MNP server will check its database for B MSISDN and add LRN/RN according to operator to which B
subscriber is registered, in above case it is Vodafone Delhi.
After addition of LRN/RN IDP is forwarded to SCP.
4) IDP received by SCP contains LRN/RN + B MSISDN in "called party number" field and "calling party field"
contains A MSISDN. Charging is done on the basis of LRN/RN. Here LRN is of Vodafone Delhi so local call
rates apply to this call. In normal scenario charging would have be done on the basis on B party MSISDN.
In response to IDP SCP revert with Connect/Continue message to MSC which contains "called party number"
5) MSC check called party number and removes LRN (as its own LRN) and forward SRI to MNP server.
Hereafter normal MNP call flow is followed which is already discussed in detail in earlier post.
6) MNP server checks B MSISDN and forward SRI to HLR.
7) HLR queries with MSC B and provide MSRN to MSC A
8) IAM is send out to MSC B with called number at B party MSRN.
Thereafter normal terminating call flow taken place.
In this post we have given emphasis on changes in prepaid leg for MNP implementation.
Hope it had been informative for you. More call flows to come in case of MNP till then happy reading.
For further queries feel free to revert to undersigned.
Posted by Shobhit at 4:20 PM 4 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google Buzz
Links to this post
Labels: MNP - Mobile Number Portability
Saturday, October 30, 2010
GPRS Call Flow
GPRS (General Packet Radio Service) is a packet based communication service for mobile devices
that allows data to be sent and received across a mobile telephone network. It's a step towards 3G and is
often referred to as 2.5G.
Its an upgrade to the existing network that sits along side the GSM network. Many of the devices
such as the BTS and BSC are still used. Often devices need to be upgraded be it software, hardware or
both. When deploying GPRS many of the software changes can be made remotely.
There are however 2 New Functional Elements which play a major role in how GPRS works - SGSN
& GGSN. In simple terms there are in practice two different networks working in parallel, GSM and GPRS.
In any GSM network there will be several BSCs. When implementing GPRS a software and hardware
upgrade of this unit is required. The hardware upgrade consists of adding a PCU (Packet Control Unit). This
extra piece of hardware differentiates data destined for the standard GSM network or Circuit Switched
Data and data destined for the GPRS network or Packet Switched Data.
PCU can be a separate entity.
SGSN (Serving GPRS Support Node) - It takes care of some important tasks, including Routing, Handover
and IP address assignment. Its a logical connection to the GPRS device. One job of the SGSN is to make
sure the connection is not interrupted as you make your journey passingfrom cell to cell. It works out
which BSC to route your connection through. If the user moves into a segment of the network that is
managed by a different SGSN it will perform a handoff to the new SGSN, this is done extremely quickly and
generally the user will not notice this has happened. Any packets that are lost during this process are
retransmitted. The SGSN converts mobile data into IP and is connected to the GGSN via a tunneling
GGSN (Gateway GPRS support node) - It is the last port of call in the GPRS network before a connection
between an ISP (Internet Service Provider) or corporate networks router occurs. The GGSN is basically
agateway, router and firewall. It also confirms user details with RADIUS servers for security, which are
usually situated in the IP network and outside of the GPRS network.
The connection between the two GPRS Support Nodes is made with a protocol called GPRS
Tunneling Protocol (GTP). GTP sits on top of TCP/IP and is also responsible for the collection of mediation
and billing information. GPRS is billed on per megabyte basis.
GPRS Call Scenario :-
A subscriber accesses the Internet with GPRS mobile phone to set the APN (Access Point Names)
& gateway IP address defined on subscription. In fact, APN is a logical name indicating the
external data network in GGSN. A subscriber can select different GGSNs via different APNs.
Currently, however, only one APN can be activated at a time. The purpose of selecting different
APNs is to access the external network via different GGSNs, because without GGSN, a subscriber
cannot access the PDN (Public Data Network). An APN consists of a fully qualified DNS (Domain
Name Server) name e.g., which should be parsed by DNS to get the real IP
address of GGSN.
The call reaches the SGSN of the GPRS network. The SGSN triggers the service in the
corresponding SCP (Service Control Point) according to subscriber's authentication information on
the HLR interconnected to the corresponding home SCP for processing.
The DNS parses the APN and get the IP address of the GGSN.
The call is routed to the GGSN according to the IP address.
The GGSN assigns the IP address to the subscriber.
After SCP verifies the subscriber, the subscriber begins to transmit data and log in to the
external web sites via the gateway whose IP address is set in the mobile phone.
The subscriber may select the service from the portal web site to connect the SP/CP web site
that provides the service, or enter the IP address of the SP/CP in the mobile phone to access the
SP/CP web site.
More Information from Readers are Expected !!!
Posted by Rajeev Bhatia at 11:16 PM 1 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google Buzz
Links to this post
Labels: Call Flow
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
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
7. CIC matching should be there with other operator.
8. Network Equipment like MSC, etc. should not be congested n Many
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).
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
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 !!!
Posted by Rajeev Bhatia at 11:09 PM 3 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
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
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
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
65 Bearer capability not implemented
69 Requested facility not implemented
Only restricted digital
bearer capability is available
79 Service or option not implemented, unspecified
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
96 Mandatory information element is missing
97 Message type non-existing or not implemented
Message not compatible with call state
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
127 Interworking, unspecified
Posted by Rajeev 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
Tuesday, August 17, 2010
MNP Call Flow to Ported Out or Other Operator
Hi All,
Want to share MNP call flow in case of Ported Out subscriber or to Other Operator subscriber (Local/NLD).
Here i would like to explain new terms.
1) Ported In/Out--> Ported In/Out subscriber are those which have changed there Service provider
Exp: A was originally Airtel Subscriber and availed MNP and changed to
Vodafone so A will be Ported in number for Vodafone and Ported out Number for Airtel.
2) RN/LRN--> Called as Routing Number(RN) for NLD calls or Local Routing number(LRN) in case of Local
call. Is four digit Unique Number allocated by TRAI to each operator.
Now let us go through call flow.
Here again i have taken example of Vodafone Delhi, call is originated by A which is Vodafone Delhi
subscriber to B number which is either other operator number or Ported out number.
Let us go through call flow in steps:
1) A dials B number
2) After receiving B number MSC send SRI to MNPDB. This is major difference between normal/MNP call
flow. In normal call flow there would have been IAM to other operators MSC.
3) MNP check its database and returns with RN/LRN+ B number in SRI response as its not Vodafone Delhi
4) After receiving RN/LRN+ B Number MSC agin checks its B number routing table and sends IAM to that
Call flow remains same in case NLD also, IAM (LRN + B number) is sent to NLD in that case.
After this calls flow remains same as in case of normal call which already posted in this blog.
Hope it had been informative for you. More call flows to come in case of MNP till then happy reading.
+~@w ] g~#
Posted by Shobhit at 8:30 PM 0 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google Buzz
Links to this post
Labels: MNP - Mobile Number Portability
Sunday, June 13, 2010
MNP Call Flow Normal Mobile to Mobile Call
Hi All,
Writing this to throw some light on call flows when MNP(Mobile Number Portability) will be implemented in
There is one very interesting a full form of MNP floating in industry "Mujhe Nahi Pata", hope after reading
this we will be able to say goodbye to that :)
Before coming to the point i would like to elaborate some terms which will used later in explaination.
1) MNPDB--> Mobile Number Portability Database, where in all details of subscriber is stored.
2) Ported In/Out--> Ported In/Out subscriber are those which have changed there Service provider
Exp: A was originally Airtel Subscriber and availed MNP and changed to
Vodafone so A will be Ported in number for Vodafone and Ported out Number for Airtel.
These terms are enough to get a feel of normal mobile to mobile in MNP.
Here i have taken example to Vodafone Delhi where Subscriber A in MSC1 and Subscriber B in Coverage of
Let us now go through signaling flow in steps:
1) Subscriber A dials Subscriber B
2) MSC1 receives B number and SRI is sent to MNP server instead of HLR as in case of traditional call flow.
3) MNP checks its database and founds that this is own subscriber and relays that SRI to HLR.
4) HLR on receiving SRI check the VLR address of subscriber and send PRN to MSC2.
5) In response to PRN MSC2 returns with MSRN.
6) HLR forward that MSRN number to MSC1 in SRI Response Message
7) MSRN is now dialled out from MSC1 to MSC2 to establish the voice path between two MSCs
After this call flow remains same as in case of traditional call flow which is already posted in this blog.
**There may be some changes in call flow as Implementing MNP is operator specific, they implement the
way which is best feasible for them.
But Implementing Concept remains the same.
Hope it had been informative for you.
More call flows to come in case of MNP as this was normal scenario, till then happy reading.
Remember sharing knowledge is best way to gain knowledge, expecting inputs/comments from readers as
well :)
Thanks & Regards
Posted by Shobhit at 4:02 PM 14 comments
Email This BlogThis! Share to Twitter Share to Facebook Share to Google Buzz
Links to this post
Labels: MNP - Mobile Number Portability
Saturday, March 27, 2010
KPI - Key Performance Indicators
KPI tells the performance of a network on a daily/weekly/monthlybasis, 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
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
Duration of calls
Billed amount on each call
% 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 no.of times network successfully find the mobiles to the no.of
times network tries to locate the mobiles within its area.
PSR = (No.of Network Paging Response / No.of Network Paging Attempts)*100.
Standard Value >= 92%.
More Information from Readers are Expected !!!
Posted by Rajeev Bhatia at 7:12 PM 11 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 -- theStream Control Transmission
Protocol (SCTP) & a set of User Adaption (UA) layers, which mimic the services of lower layers of SS7 &
Ordered, Reliable Transfer.
Redundancy in case of Link Failure.
Low Loss & Delay.
The key components in the SIGTRAN architecture are :-
MGCMedia 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.
SGSignaling Gateway, responsible for interfacing to the SS7 network and passing signaling
messages to the IP nodes.
MGMedia 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.
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.