the telecom protocols_call flows

21st March

SIP responses generated by the UAS or SIP servers.

SIP Response Classes:

1xx - Informational - Indicates the status of call prior to the completion.2xx - Success - Request has succeeded.3xx - Redirection - The client should try to complete the request at another server.4xx - Client Error - The request has failed due to error due to the client. The client may retry.5xx - Server failure - The request has failed due to server error. request may retry to another server.6xx - Global failure - the request has failed and can not retry to any server.

1xx --Informational :

100 Trying : This response is used to indicate the next node receives the request and stop theretransmission. This response is sent if there is delay in sending the final response more the 200ms.180 Ringing: The response is generated if UA receives the INVITE and started the ringing. It may usedto initiate local ring back.181 Call is being Forwarded: This response is indication of call is being forwarded to differentdestination.182 Call Queued: The called server is overloaded or temporary unavailable. the server sends this statuscode to queue the call. When server ready to take the call, it initiates appropriate final response.183 Call Progress: This response may be used to send extra information for a call which is still being set up.

199 Early Dialog Terminated: Can be used by User Agent Server to indicate to upstream SIP entities(including the User Agent Client (UAC)) that an early dialog has been terminated.

2xx—Successful Responses

200 OK: Indicates the request was successful.202 Accepted: Indicates that the request has been accepted for processing, but the processing has notbeen completed.204 No Notification : Indicates the request was successful, but the corresponding response will not bereceived.

3xx—Redirection Response

300 Multiple Choices: The address resolved to one of several options for the user or client to choosebetween, which are listed in the message body or the message's Contact fields.301 Moved Permanently: The original Request-URI is no longer valid, the new address is given in theContact header field, and the client should update any records of the original Request-URI with the newvalue.302 Moved Temporarily: The client should try at the address in the Contact field. If an Expires field ispresent, the client may cache the result for that period of time.305 Use Proxy: The Contact field details a proxy that must be used to access the requested destination.380 Alternative Service : The call failed, but alternatives are detailed in the message body.

SIP Responses

The Telecom Protocols http://telecomprotocols.blogspot.in/2012/10/location-update-message-de...

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401 Unauthorized: The request requires user authentication. This response is issued by UASs andregistrars.402 Payment Required: Reserved for future use.403 Forbidden : The server understood the request, but is refusing to fulfill it404 Not Found : The server has definitive information that the user does not exist at the domain specifiedin the Request-URI. This status is also returned if the domain in the Request-URI does not match any ofthe domains handled by the recipient of the request.405 Method Not Allowed: The method specified in the Request-Line is understood, but not allowed forthe address identified by the Request-URI.406 Not Acceptable: The resource identified by the request is only capable of generating responseentities that have content characteristics but not acceptable according to the Accept header field sent inthe request.407 Proxy Authentication Required: The request requires user authentication. This response is issuedby proxys408 Request Timed Out: Couldn't find the user in time.409 Conflict : User already registered.410 Gone : The user existed once, but is not available here any more.411 Length Required : The server will not accept the request without a valid Content-Length.412 Conditional Request Failed : The given precondition has not been met.413 Request Entity Too Large : Request body too large.414 Request-URI Too Long : The server is refusing to service the request because the Request-URI islonger than the server is willing to interpret.415 Unsupported Media Type : Request body in a format not supported.416 Unsupported URI Scheme : Request-URI is unknown to the server.417 Unknown Resource-Priority : There was a resource-priority option tag, but no Resource-Priorityheader.420 Bad Extension: Bad SIP Protocol Extension used, not understood by the server.421 Extension Required: The server needs a specific extension not listed in the Supported header.422 Session Interval Too Small: The received request contains a Session-Expires header field with aduration below the minimum timer.423 Interval Too Brief: Expiration time of the resource is too short.424 Bad Location Information : The request's location content was malformed or otherwiseunsatisfactory.428 Use Identity Header : The server policy requires an Identity header, and one has not been provided.429 Provide Referrer Identity : The server did not receive a valid Referred-By token on the request.430 Flow Failed : A specific flow to a user agent has failed, although other flows may succeed. Thisresponse is intended for use between proxy devices, and should not be seen by an endpoint (and if it isseen by one, should be treated as a 400 Bad Request response).433 Anonymity Disallowed: The request has been rejected because it was anonymous.436 Bad Identity-Info: The request has an Identity-Info header, and the URI scheme in that headercannot be dereferenced.437 Unsupported Certificate : The server was unable to validate a certificate for the domain that signedthe request.438 Invalid Identity Header: The server obtained a valid certificate that the request claimed was used tosign the request, but was unable to verify that signature.439 First Hop Lacks Outbound Support : The first outbound proxy the user is attempting to registerthrough does not support the "outbound" feature of RFC 5626, although the registrar does.470 Consent Needed: The source of the request did not have the permission of the recipient to makesuch a request.480 Temporarily Unavailable : Callee currently unavailable.481 Call/Transaction Does Not Exist : Server received a request that does not match any dialog ortransaction.

…The Telecom Protocols All about the telecom protoc


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485 Ambiguous : Request-URI is ambiguous.486 Busy Here : Callee is busy.487 Request Terminated : Request has terminated by bye or cancel.488 Not Acceptable Here : Some aspects of the session description of the Request-URI is notacceptable.489 Bad Event : The server did not understand an event package specified in an Event header field.491 Request Pending : Server has some pending request from the same dialog.493 Undecipherable : Request contains an encrypted MIME body, which recipient can not decrypt.494 Security Agreement Required : The server has received a request that requires a negotiatedsecurity mechanism, and the response contains a list of suitable security mechanisms for the requesterto choose between or a digest authentication challenge.

5xx—Server Failure Responses

500 Server Internal Error : The server could not fulfill the request due to some unexpected condition.501 Not Implemented : The server does not have the ability to fulfill the request, such as because it doesnot recognize the request method. (Compare with 405 Method Not Allowed, where the server recognizesthe method but does not allow or support it.)502 Bad Gateway : The server is acting as a gateway or proxy, and received an invalid response from adownstream server while attempting to fulfill the request.503 Service Unavailable : The server is undergoing maintenance or is temporarily overloaded and socannot process the request. A "Retry-After" header field may specify when the client may re attempt itsrequest.504 Server Time-out : The server attempted to access another server in attempting to process therequest, and did not receive a prompt response.505 Version Not Supported : The SIP protocol version in the request is not supported by the server513 Message Too Large : The request message length is longer than the server can process.580 Precondition Failure : The server is unable or unwilling to meet some constraints specified in theoffer.

6xx—Global Failure Responses

600 Busy Everywhere : All possible destinations are busy. Unlike the 486 response, this responseindicates the destination knows there are no alternative destinations (such as a voicemail server) able toaccept the call.603 Decline : The destination does not wish to participate in the call, or cannot do so, and additionally theclient knows there are no alternative destinations (such as a voicemail server) willing to accept the call.604 Does Not Exist Anywhere : The server has authoritative information that the requested user doesnot exist anywhere.606 Not Acceptable : The user's agent was contacted successfully but some aspects of the sessiondescription such as the requested media, bandwidth, or addressing style were not acceptable.

Posted 21st March by Pramod Kumar

Labels: SIP

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3 of 54 02-10-2013 11:56

20th March

SIP Servers:

Proxy Servers:

- A stateless proxy server processes each SIP request or response based solely on the message contents. Once the

message has been parsed, processed, and forwarded or responded to,no information about the message is stored—no

dialog information is stored. A stateless proxy never re-transmits a message, and does not use any SIP timers.

- A stateful proxy server keeps track of requests and responses received in the past and uses that information in

processing future requests and responses. For example, a stateful proxy server starts a timer when a request is

forwarded. If no response to the request is received within the timer period, the proxy will

re-transmit the request, relieving the user agent of this task. Also, a stateful proxy can require user agent

authentication.

Back-to-Back User Agents (B2BUA):An B2BUA is a type of SIP device that receives the SIP request, that reformulates the request and send it out as new

request. Response to the request are reformulated and sent back to the UA in opposite direction.

SIP Methods (Request):

1.) INVITE: The INVITE is used to establish the media session between the users. An Invite usually has a message

body containing the media session information as SDP. it also contains other information like QoS and security

information. If INVITE does not contain the media information, the ACK message contains the media information

of the UAC.

A media session is considered established when the INVITE, 200 OK, and ACK messages have been exchanged

between the UAC and the UAS. If the media information contained in the ACK is not acceptable, then the called

party must send a BYE to cancel the session, a CANCEL cannot be sent because the session is already established.

A UAC that originates an INVITE to establish a dialog creates a globally unique Call-ID that is used for the

duration of the call. A CSeq count is initialized (which need not be set to 1, but must be an integer) and incremented

for each new request for the same Call-ID. The To and From headers are populated with the remote and local

addresses. A From tag is included in the INVITE, and the UAS includes a To tag in any responses. A To tag in a

200 OK response to an INVITE is used in the To header field of the ACK and all future requests within the dialog.

The combination of the To tag, From tag, and Call-ID is the unique identifier for the dialog.

Re-Invite: An INVITE sent for an existing dialog references the same Call-ID as the original INVITE and contains

the same To and From tags. Sometimes called a re-INVITE, the request is used to change the session characteristics

or refresh the state of the dialog. The CSeq command sequence number is incremented so that a UAS can

distinguish the re-INVITE from a re-transmission of the original INVITE.

UPDATE: A re-INVITE must not be sent by a UAC until a final response to the initial INVITE has been received

instead, an UPDATE request can be sent.

An Expires header in an INVITE indicates to the UAS how long the call request is valid. For example, the UAS

SIP Methods


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session

2.) REGISTER: REGISTER message used to register the Address of record to Registrar server. The REGISTER

method is used by a user agent to notify a SIP network of its current Contact URI (IP address) and the URI that

should have requests routed to this Contact.The registrar binds the SIP URI of marconi and the IP address of the

device in a database that can be used.

----------------------------------------------------------

REGISTER sip:registrar.text.com SIP/2.0

Via: SIP/2.0/UDP 200.201.202.203:5060;branch=z9hG4bKus1812

Max-Forwards: 70

To: Marconi <sip:[email protected]>

From: Marconi <sip:[email protected]>

;tag=3431

Call-ID: [email protected]

CSeq: 1 REGISTER

Contact: sip:[email protected]

Content-Length: 0

--------------------------------------------------------

SIP/2.0 200 OK

Via: SIP/2.0/UDP 200.201.202.203:5060;branch=z9hG4bKus19

To: Marconi <sip:[email protected]>;tag=8771

From: Marconi <sip:[email protected]>

;tag=3431

Call-ID: [email protected]

CSeq: 1 REGISTER

Contact:Marconi <sip:[email protected]>;expires=3600Content-Length: 0

--------------------------------------------------------The Contact URI is returned along with an expires parameter, which indicates how long the registration is valid,

which in this case is 1 hour (3,600 seconds).

Marconi Registrar Server

=================================

--------------REGISTER--------------------->

<---------------200 OK-----------------------

3.) BYE: The BYE method is used to terminate an established media session. BYE is sent only by useragents participating in the session, never by proxies or other third parties. It is an end-to-end method, soresponses are only generated by the other user agent.

BYE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 100.120.100.100:5060;branch=z9hG4bK3145rMax-Forwards:70To: Marconi <sip:[email protected]>;tag=63104From: Tela <sip:[email protected]>;tag=9341123Call-ID: [email protected]: 12 BYEContent-Length: 0



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to all other requests are never acknowledged. Final responses are defined as 2xx, 3xx, 4xx, 5xx, or 6xxclass responses. The CSeq number is never incremented for an ACK, but the CSeq method is changedto ACK. This is so that a UAS can match the CSeq number of the ACK with the number of thecorresponding INVITE.

An ACK may contain an application/sdp message body. This is permitted if the initial INVITE did notcontain a SDP message body. If the INVITE contained a message body, the ACK may not contain amessage body. The ACK may not be used to modify a media description that has already been sent inthe initial INVITE; a re-INVITE must be used for this purpose.

For 2xx responses, the ACK is end-to-end, but for all other final responses it is done on a hop-by-hopbasis when stateful proxies are involved. The end-toend nature of ACKs to 2xx responses allows amessage body to be transported. A hop-by-hop ACK reuses the same branch ID as the INVITE since it isconsidered part of the same transaction. An end-to-end ACK uses a different branch ID as it isconsidered a new transaction.

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/TCP 100.200.102.100:5060;branch=z9hG4bK1234Max-Forwards:70To: Marconi <sip:[email protected]>;tag=902332From: Tesla <sip:[email protected]>;tag=887823Call-ID: [email protected]: 3 ACKContent-Type: application/sdpContent-Length: 100

(SDP not shown)

5.) CANCEL: The CANCEL method is used to terminate pending call attempts. It can be generated byeither user agents or proxy servers provided that a 1xx response containing a tag has been received, butno final response has been received. CANCEL is a hop-by-hop request and receives a responsegenerated by the next stateful element. The CSeq is not incremented for this method so that proxies anduser agents can match the CSeq of the CANCEL with the CSeq of the pending INVITE to which itcorresponds. The branch ID for a CANCEL matches the INVITE that it is canceling.

CANCEL sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 100.100.122.122:5060;branch=z9hG4bK3134324Max-Forwards:70To: Marconi <sip:[email protected]>From: Tesla <sip:[email protected]>;tag=034324Call-ID: [email protected]: 1 CANCELContent-Length: 0

6.) OPTIONS: The OPTIONS method is used to query a user agent or server about its capabilities anddiscover its current availability. The response to the request lists thecapabilities of the user agent or server. A proxy never generates an OPTIONS request.

OPTIONS sip:[email protected] SIP/2.0



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To: Marconi <sip:[email protected]>From: Tesla <[email protected]>;tag=34Call-ID: [email protected]: 1 OPTIONSContent-Length: 0

7.) REFER: The REFER method is used by a user agent to request another user agent to access a URIor URL resource. The resource is identified by a URI or URL in the required Refer-To header field. Whenthe URI is a sip or sips URI, the REFER is probably being used to implement a call transfer service.REFER can also used to implement peer-to-peer call control.

REFER sip:[email protected] SIP/2.0Via SIP/2.0/UDP test.com:5060;branch=z9hG4bK9323249Max-Forwards: 69To: <sip:[email protected]>;tag=324234From: Tesla <sip:[email protected]>;tag=44432Call-ID: 3419fak32343243s1A9dklCSeq: 5412 REFERRefer-To: <sip:[email protected]>Content-Length: 0

8.) SUBSCRIBE: The SUBSCRIBE method is used by a user agent to establish a subscription for thepurpose of receiving notifications (via the NOTIFY method) about a particular event. The subscriptionrequest contains an Expires header field, which indicates the desired duration of the existence of thesubscription. After this time period passes, the subscription is automatically terminated. The subscriptioncan be refreshed by sending another SUBSCRIBE within the dialog before the expiration time. A serveraccepting a subscription returns a 200 OK response also obtaining an Expires header field. There is no“UNSUBSCRIBE” method used in SIP—instead a SUBSCRIBE with Expires:0 requests the terminationof a subscription and hence the dialog. A terminated subscription (either due to timeout out or atermination request) will result in a final NOTIFY indicating that the subscription has been terminated.

UA Proxy-Server Presence Agent====================================------------Subscribe-----------> -----------Subscribe--------------> <-----202 Accepted---------------<-----------------202 Accepted---<------------------------------------NOTIFY-----------------------------------------------------200 OK--------------------------------------------->----------------------------Subscribe--------------------------------------><-----------------------------------200 OK-------------------------------

SUBSCRIBE sip:[email protected] SIP/2.0Via SIP/2.0/UDP 200:200:200:201:5060;branch=z9hG4bKABDA ;received=192.0.3.4Max-Forwards: 69To: <sip:[email protected]>From: Tesla <sip:[email protected]>;tag=1814Call-ID: 452k59252058234924lk34



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Event: dialogContent-Length: 0

9.) NOTIFY: The NOTIFY method is used by a user agent to convey information about the occurrence ofa particular event. A NOTIFY is always sent within a dialog, when a subscription exists between thesubscriber and the notifier. A NOTIFY request normally receives a 200 OK response to indicate that ithas been received.A NOTIFY requests contain an Event header field indicating the package and a Subscription-Stateheader field indicating the current state of the subscription. The Event header field will contain thepackage name used in the subscription.A NOTIFY is always sent at the start of a subscription and at the termination of a subscription.

Posted 20th March by Pramod Kumar

Labels: SIP

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10th January

Four variants of BICC IP bearer set-up procedures are defined:- Fast Forward

- Delayed Forward- Fast Backward- Delayed Backward

Those procedures differ on the way the bearer control information are exchanged, and on whether anAPM (Connected) message shall be sent by the originating BICC node once the bearer is ready for use.

- Bearer control information exchanges :- In Fast bearer setup (forward or backward) and in delayed forward bearer establishmentprocedures, the IP bearer establishment is done in the forward direction, i.e. the IPBCP requestis sent from the originating towards the terminating MGWs ; the bearer establishment request issent in the IAM message in fast (forward or backward) procedures, while it is sent insubsequent APM message, after a first IAM/APM exchange, in case of delayed forward bearerestablishment.- In reverse, in the Delayed backward bearer establishment procedure, the IP bearerestablishment is done in the direction reverse to the call establishment direction, i.e. the IPBCPrequest is sent from the terminating towards the originating MGWs, through a backward APMmessage.

- APM (Connected) exchange :- In Forward bearer setup procedures, the terminating BICC node decides by its ownwhether it requires or not the originating BICC node to send an APM (Connected) message

Type of BICC calls

BICC Bearer establishment procedures:



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- In Fast backward bearer setup procedure, an APM(Connected) message shall always besent (BICC protocol).

Posted 10th January by Pramod Kumar

Labels: Codec Management, VOIP (Voice over IP), Call Flows

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3rd December 2012

The interfaces and protocols supported towards the networks are mentioned below:

[http://4.bp.blogspot.com/-J59OmJgtxwM/ULxV9a5i0aI/AAAAAAAABPg/1DYT1mqCKWs/s1600/interfaces.JPG]Network Interfaces and Protocols

Posted 3rd December 2012 by Pramod

Labels: Codec Management

Core Network Architecture - Interfaces



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23rd October 2012

Codec Management has following goals:1.) Optimize the voice quality.2.) Optimize the bandwidth efficiency3.) Optimize the transcoder resource in MGW.4.) Optimize the delay

Voice quality is measured in terms of the level, attenuation, delay, echo, and so on and may be used as abasis for the iQoS assessments for conventional VoIP. Voice Quality is measured in terms of R-value andbelow 50 are not recommended.

Standards Intri nsic quality (R)------------------------------------------------------

ITU G.711 (64kbps) 94.3ITU G.728 (12.kbps) 74.3ETSI GSM-FR (13kbps) 74.3ETSI GSM-HR (5.6kbps) 71.3ETSI GSM-EFR (12.2kbps) 89.3

-----------------------------------------------------

Transcoding can be harmful for voice quality of a call and should be avoided if possible. In 2G, BSC is incharge of transcoding while MSC is the incharge of transcoding in 3G. We have intention to minimise theTranscoder to enhance the voice quality.

[http://1.bp.blogspot.com/-MC4wZ4nx8ZI/UIY4bi3uelI/AAAAAAAAAKE/MrBbw85JEuY/s1600/codec1.JPG]

Legacy networks use a consistent 64 kbps per channel. Use G.711, packet networks easily surpass 64kbps. Therefore, compress codecs must be used on core. Compression ratio of 8:1can be achieved withcompressed codecs along with the following techniques : silence suppression, AAL2/RTP multiplexing, IPheader compression.

The third goal of the codec management is to optimize the transcoder resources in theMGW. So, TrFO must be used whenever possible.

Codec Management

Codec Management Objectives

Voice Quality:

Bandwidth efficiency:

Transcoder Resources optimization:


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Delay can be minimized if reduce the number of transcoders. We can reduce the transcoding time in call.

Posted 23rd October 2012 by Pramod Kumar


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19th October 2012

In Early legacy telephone network, A call between the two mobiles involved with two transcoding functionat both the end which decrease the voice quality. The Transcoding Function done by the TRAU(Transcoding and Rate Adaptation Unit) to compress and decompress the speech.

TFO (Tandem Free Operation) enables to avoid the traditional double speech encoding / decoding inMS to MS call configurations. TFO uses in-band signalling and procedures for transcoders to enablecompressed speech to be maintained between a pair of transcoders. So the main objective of TFO is the improvement of the voice quality for calls between 2 mobilesubscribers, but no resource optimisation is introduced as transcoder functions are always present in thepath.

If TFO is activated between two end nodes, TFO Frames with compressed speech (e.g. AMR in LSB) aspayload are carried over 8 or 16 kbit/s channels mapped onto the least or two least significant bits of the64 kbit/s PCM speech samples. It is also carried the G.711 codec in MSB if it is not possible to achievethe TFO. So we can say that TFO is used to achieve the voice quality if possible.

[http://2.bp.blogspot.com/-cdr2NJlVsYM/UIEVctA-oiI/AAAAAAAAAJw/A2Whhqdc0sU/s1600/tfo.jpg]

Tandem Free Operation is activated and controlled by the Transcoder Units after the completion of thecall set-up phase at both ends of an MS-MS, MS-UE, or UE-UE call configuration. The TFO protocol isfully handled and terminated in the Transcoder Units. For this reason, the Transcoder Units cannot bebypassed in Tandem Free Operation. This is the key difference with the feature called Transcoder FreeOperation (TrFO).

TFO - Tandem Free Operation



11 of 54 02-10-2013 11:56


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17th October 2012

In Early legacy telephone network, A call between the two mobiles involved with two transcoding functionat both the end which decrease the voice quality. The Transcoding Function done by the TRAU(Transcoding and Rate Adaptation Unit) to compress and decompress the speech.

Transcoder Free Operation (TrFO) is transport of compressed speech, whicheliminates unnecessary coding and decoding of the signals when both end uses the same codecs. TrFOutilizes out of band signaling capabilities that include the ability to determine the negotiated codec type tobe used at the two end nodes. If the two end nodes are capable of the same codec operations, it may bepossible to traverse the entire packet network using only the compressed speech (of thepreferred codec). TrFO is basic function of codec Management.

Following are the benefits of achieving the TrFO:

1.) Improve the voice quality because of elimination of coding and decoding of suppressed codecs.2.) We can optimize the resources by skipping the transcodes at both ends.3.) We can increase the bandwidth in the Core Network.4.) Increase the transmission delay by skipping the compression and decompression to G.711 codec.

It can improve the voice quality and above features using the TrFo and TFO simultaneously.

[http://4.bp.blogspot.com/-gETJBRJA6do

/UH5hAFlaI4I/AAAAAAAAAJc/moj04H3ZQuI/s1600/Trfo.jpg]TrFO Operation

Posted 17th October 2012 by Pramod Kumar


TrFO - Transcoder Free Operation



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11th October 2012

HEX VALUE CAUSE----------------------------------------------------------------

03 No route to destination

06 Channel unacceptable

08 Preemption

0E Query on Release QoR

10 Normal clearing

11 User busy

12 No user responding

13 No answer from user (user alerted)

14 Subscriber absent

15 Call rejected

16 Number changed

17 Redirection to new destination

18 Call rejected because of a feature at the destination

19 Exchange routing error

1A Non selected user clearing

1B Destination out of Order

1C Invalid number format (address incomplete)

1D Facility rejected

1E Response to Status Enquiry

1F Normal, unspecified

22 No circuit/channel available

26 Network out of order

29 Temporary failure

39 Bearer capability not authorized

51 Invalid call reference value

52 Identified channel does not exist

54 Call identity in use

57 User not member of Closed User Group

58 Incompatible destination

5F Invalid message, unspecified

60 Mandatory information element is missing

6F Protocol error, unspecified

7F Interworking, unspecified


Labels: SS7 Protocol Stack

ISUP Release Cause Values

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13 of 54 02-10-2013 11:56

11th October 2012

Service Name HEX BINARY==================================================

CLIP 11 00010001CLIR 12 00010010 COLP 13 00010011 COLR 14 00010100 All Call Forward 20 00100000 CFU 21 00100001

All Conditional call forward 28 00101000 CFB 29 00101001CFNRY 2A 00101010CFNRc 2B 00101011 CD 24 00100100 ECT 31 00110001 CW 41 01000001 CH 42 01000010 CCBs-A 43 01000011CCBs-B 44 01000011 MPTY 51 01010001 AOCI (information) 71 01110001 AOCC (Charging) 72 01110010 All call Barring 90 10010000 Outgoing call Barring 91 10010001 baoc 92 10010010 boic 93 10010011 boicExHC 94 10010100

Barring of Incoming calls 99 10011001 baic 9A 10011010 bicRoam 9B 10011011

Abbreviation :

CLIP : Calling Line Identification PresentationCLIR: Calling Line Identification RestrictionCOLP: Connected Line Identification PresentationCOLR: Connected Line Identification RestrictionCFU: Call Forwarding UnconditionalCFB: Call Forwarding BusyCFNRy: Call Forwarding No ReplyCFNRc: Call Forwarding Not ReachableECT: Explicit Call TransferCW: Call WaitingMPTY: Multi PartyCH: Call HoldAOCI: Advice of Charge IndicatorBAOC : Barring of All Outgoing Calls

Supplementary Service Codes



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Labels: GSM

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9th October 2012

Here is UMTS subscriber call flow with the core network.

MS-A MSC MS-B****************************************************************************************************-------Initial_UE(CM_SERVICE_REQ) -->|------------Dir_Trnx(AUTH_REQ) <--|-------------Dir_Trnx(AUTH_RSP) -->|---------------SecurityModeCmd <--|--------------SecurityModeComp -->|-----Dir_Trnx(TMSI_RELOC_CMD) <--|----Dir_Trnx(TMSI_RELOC_COMP) -->| -------------Dir_Trnx(SETUP) -->|---------Dir_Trnx(CALL_PROC) <--|----------------RabAssignReq <--|----------------RabAssignRsp -->|

|--> PAGING --------------------------- |<-- Initial_UE(PAGE_RSP) ------------- |--> Dir_Trnx(AUTH_REQ)-------------- |<-- Dir_Trnx(AUTH_RSP)--------------- |--> SecurityModeCmd ------------------- |<-- SecurityModeComp------------------ |--> Dir_Trnx(TMSI_RELOC_CMD)------ |<-- Dir_Trnx(TMSI_RELOC_COMP) --- |--> Dir_Trnx(SETUP) ------------------- |<-- Dir_Trnx(CALL_CONF) -------------- |--> RabAssignReq ------------------------ |<-- RabAssignRsp ------------------------- |<-- Dir_Trnx(CALL_ALERT) ------------- -----------Dir_Trnx(CALL_ALERT) <--| |<-- Dir_Trnx(CALL_CONNECT)---------- |--> Dir_Trnx(CALL_CONNECTACK)---- -------Dir_Trnx(CALL_CONNECT) <--|---Dir_Trnx(CALL_CONNECTACK) -->|

<--------------------------------------Call Connected at this stage----------------------------> |<-- Dir_Trnx(CALL_DISCONNECT) ------ |--> Dir_Trnx(CALL_REL) ------------------ |<-- Dir_Trnx(CALL_RELCOMP) -----------

3G Call Flow



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|<-- IuReleaseComp -------------------------- --------Dir_Trnx(CALL_RELCOMP) <--|------------------IuReleaseCmd <--|------------------IuReleaseComp -->|*************************************************************************************************


Labels: Call Flows

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6th October 2012

*********************************************************************** Mobile Station MSC

*************************************************************----------------------Initial_UE(CM_SERVICE_REQ) -->|---------------------------Dir_Trnx(AUTH_REQ) <--|---------------------------Dir_Trnx(AUTH_RSP) -->|-----------------------------Cipher Mode Command <--|-----------------------------Cipher Mode Complete -->|---------------------Dir_Trnx(TMSI_RELOC_CMD) <--|----------------------Dir_Trnx(LOCUPD_ACCEPT) <--|------------------------TMSI_RELOC_COMP <--|--------------------------------Clear Cmd <--|--------------------------------Clear Comp -->|

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

*******************************************************************00 21 57 05 08 00 13 f0 79 00 01 00 01 17 12 05 08 70 13 f0 79 ff fe 30 08 89 88 88 12 45 12 00 00 21 01 00

00 Message discriminator

21 length

Location Update Message Decoding

Location Update Flow :

CM Service Request/Location Update Request



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05 IE

08 Length

00 Cell Identifier discriminator

13 F0 MCC1-> 3, MCC2 ->1, MCC3-> 0 MCC: 31079 MNC1->9, MNC2->7 MNC: 9700 01 LAC: 1.Represent LAC in hex and put in two bytes00 01 CI: 1 represents CI in hex and put in two bytes

17 12 message type and length of layer3

05 TI flag, value and protocol discriminator (Mobility management).08 message type of LU

70 Chiperkey sequence No.

13 f0 79 ff fe Location Area Identification

30 08 89 88 88 12 45 12 00 00 IMSI

21 01 00 Mobility Management.

********************************************01 00 13 05 12 02 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00

01 Message Discriminator

00 Ctrl Channel not further specified (00), spare (000), sapi (000)13 Length (19)

05 Mobility Management

12 Message Type DTAP

01/02 Possible value for ciphering key sequence number

00 00 00 00 00 00 00 00 00 Authentication RAND

00 00 00 00 00 00 00

**********************************************01 00 06 05 54 00 00 00 00


00 Ctrl Channel not further specified (00), spare (000), sapi (000)

06 Length (06)


54 Message Type

00 00 00 00 Authentication SRES

Authentication Request

Authentication Response

Class Mark Request



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01 Length (1)

58 Message Type

***********************************00 0b 54 12 03 30 59 81 13 02 60 14 00


0b Length (10)

54 Message Type

12 Classmark Information Type 2

03 Length (03)

30 59 81 Class Mark

13 Classmark Information Type 3

02 Length (02)

60 14 Class Mark

00 end of Optional parameters.

********************************************00 0e 53 0a 09 07 00 00 00 00 00 00 00 00 23 01


0e Length (15)

53 Message Type

0a Encryption Information

09 Length (09)

07 00 00 00 00 00 00 00 00 Encryption Information

23 Cipher Response Mode (Phase 2)

01 Spare/ IMEISV must be Included By The Mobile Station

00 10 55 20 0b 17 09 33 05 70 87 70 35 17 01 f9 2c 01


10 Length (17)

55 Message Type

20 Layer 3 Message contents (Phase 2)

0b Length (11)

Class Mark Update

Cipher Mode Command

Cipher Mode Complete



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********************************************01 00 0e 05 02 13 f0 69 00 03 17 05 f4 0f 2f 20 04 00 02 01


00 Ctrl Channel not further specified (00), spare (000), sapi (000)

0e Length (15)


02 Message Type

13 F0 MCC1-> 3, MCC2 ->1, MCC3-> 0 MCC: 310

69 MNC1->9, MNC2->6 MNC: 96

00 03 LAC

17 Mobility Identity

05 Parameter Length

f4 ST 0/Even Number of Address Signals / TMSI

0f 2f 20 04 00 02 01 Identity Digits

******************************00 04 20 04 01 09


04 Length (04)

20 Message Type

04 Cause

01 Length

09 Extension bit (last octet) / Call Control Normal Event

*****************************00 01 21


01 Length (1)

21 Message Type


Labels: Call Flows

Location Update Accept

Clear Command

Clear Complete



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27th September 2012

[http://1.bp.blogspot.com/-L7w1xQzk7wQ/UGP2Fhb-

vLI/AAAAAAAAAI4/rBUclMCwTRI/s1600/sccp.JPG]

The Signalling Connection Control Part (SCCP) message are used by the peer to peer protocol.Following are the SCCP message used by the peer to connection oriented and connection less services.

Application that uses the service of SCCP are called Subsystems. Refer the SCCP structure[http://telecomprotocols.blogspot.com/2012/09/ss7-protocol-stack-sccp.html] for detail SCCP structure.

Classes of service :

• Class 0 —Basic connectionless class - it has no sequencing control. i does not impose any condition onSLS, therefore SCCP message can be delivered in out-of-sequece.

• Class 1 —In-sequence delivery connectionless class - it adds the sequence control to class 0 service byrequiring to insert the same SLS to all NSDU.

• Class 2 —Basic connection-oriented class - Assign the local reference numbers (SLR,DLR) to createlogical connection. it does not provide the flow control, loss, and mis-sequence detection.• Class 3 —Flow control connection-oriented class - Class 3 is an enhanced connection-oriented servicethat offers detection of both message loss and mis-sequencing

1. Connection Request (CR): Connection Request message is initiated by a calling SCCP to a called SCCP to

request the setting up of a signalling connection between two entities. On Reception of CR message, the called

SCCP initiates the setup signalling connection. CR message have the Source Local Reference (SLR) as an address

of originating entity. It is used during connection establishment phase by connection-oriented protocol class 2 or 3.

2. Connection Confirm (CC) : Connection Confirm message is initiated by the called SCCP to indicate

to the calling SCCP that it has performed the setup of the signaling connection. On reception of

SCCP Messages



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3. Connection Refused (CREF) : Connection Refused message is initiated by the calledSCCP or an intermediate node SCCP to indicate to the calling SCCP that the setup of thesignalling connection has been refused. It is used during connection establishment phase by

connection-oriented protocol class 2 or 3.

4. Data Acknowledgement (AK) : Data Acknowledgement message is used to control the window flow control

mechanism, which has been selected for the data transfer phase. It is used during the data transfer phase inprotocol class 3.

5. Data Form 1 (DT1) : A Data Form 1 message is sent by either end of a signalling connection to passtransparently SCCP user data between two SCCP nodes. It is used during the data transfer phase inprotocol class 2 only.

6. Data Form 2 (DT2) : A Data Form 2 message is sent by either end of a signalling connection to passtransparently SCCP user data between two SCCP nodes and to acknowledge messages flowing in theother direction. It is used during the data transfer phase in protocol class 3 only.

7. Expedited Data (ED) : An Expedited Data message functions as a Data Form 2 message but includesthe ability to bypass the flow control mechanism which has been selected for the data transfer phase. Itmay be sent by either end of the signalling connection. It is used during the data transfer phase inprotocol class 3 only.

8. Expedited Data Acknowledgement (EA) : An Expedited Data Acknowledgement message is used toacknowledge an Expedited Data message. Every ED message has to be acknowledged by an EAmessage before another ED message may be sent. It is used during the data transfer phase in protocolclass 3 only.

9. Inactivity Test (IT) : An Inactivity Test message may be sent periodically by either end of a signallingconnection section to check if this signalling connection is active at both ends, and to audit the

consistency of connection data at both ends. It is used in protocol classes 2 and 3.

10. Protocol Data Unit Error (ERR) : A Protocol Data Unit Error message is sent on detection of anyprotocol errors. It is used during the data transfer phase in protocol classes 2 and 3.

11. Released (RLSD) : A Released message is sent, in the forward or backward direction, to indicate thatthe sending SCCP wants to release a signalling connection and the associated resources at the sendingSCCP have been brought into the disconnect pending condition. It also indicates that the receiving nodeshould release the connection and any other associated resources as well. It is used during connectionrelease phase in protocol classes 2 and 3.

12. Release Complete (RLC) : A Release Complete message is sent in response to the Releasedmessage indicating that the Released message has been received, and the appropriate procedures havebeen completed. It is used during connection release phase in protocol classes 2 and 3.

13. Reset Request (RSR) : A Reset Request message is sent to indicate that the sending SCCP wantsto initiate a reset procedure (re-initialization of sequence numbers) with the receiving SCCP. It is usedduring the data transfer phase in protocol class 3.

14. Reset Confirm (RSC) : A Reset Confirm message is sent in response to a Reset Request message



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15. Subsystem-Prohibited (SSP) : A Subsystem-Prohibited message is sent to concerned destinationsto inform SCCP Management (SCMG) at those destinations of the failure of a subsystem. It is used forSCCP subsystem management.

16. Subsystem-Allowed (SSA) : A Subsystem-Allowed message is sent to concerned destinations toinform those destinations that a subsystem which was formerly prohibited is now allowed or that a SCCPwhich was formerly unavailable is now available. It is used for SCCP management.

17. Subsystem-Status-Test (SST) : A Subsystem-Status-Test message is sent to verify thestatus of a subsystem marked prohibited or the status of an SCCP marked unavailable. It isused for SCCP management.

18. UnitData (UDT) : A Unitdata message can be used by an SCCP wanting to send data in a

connectionless mode. It is used in connectionless protocol classes 0 and 1.

19. Unitdata Service (UDTS) : A Unitdata Service message is used to indicate to the originating SCCPthat a UDT sent cannot be delivered to its destination. Exceptionally and subject to protocol interworkingconsiderations, a UDTS might equally be used in response to an XUDT or LUDT message. A UDTSmessage is sent only when the option field in that UDT is set to "return on error". It is used inconnectionless protocol classes 0 and 1.

20. Extended Unitdata (XUDT) : An Extended Unitdata message is used by the SCCP wanting to senddata (along with optional parameters) in a connectionless mode. It is used for the segmentation of largemessage into more XUDT messages. It is used in connectionless protocol classes 0 and 1.

21. Extended Unitdata Service (XUDTS) : An Extended Unitdata Service message is usedto indicate to the originating SCCP that an XUDT cannot be delivered to its destination. It isused in connectionless protocol classes 0 and 1.

22. Subsystem Congested (SSC) : A Subsystem Congested message is sent by an SCCP node when itexperiences congestion. It is used for SCCP subsystem management.

23. Long Unitdata (LUDT) : Long Unitdata message is used by the SCCP to send data (along withoptional parameters) in a connectionless mode. When MTP capabilities according toRecommendation Q.2210 are present, it allows sending of NSDU sizes up to 3952 octets withoutsegmentation. It is used in Connectionless protocol classes 0 and 1.

24. Long Unitdata Service (LUDTS) : A Long Unitdata Service message is used to indicateto the originating SCCP that an LUDT cannot be delivered to its destination. It is used in

connectionless protocol classes 0 and 1.

25. Subsystem-Out-of-Service-Request (SOR) : A Subsystem-Out-of-Service-Request message isused to allow subsystems to go out-of-service without degrading performance of the network. When asubsystem wishes to go out-of-service, the request is transferred by means of a Subsystem-Out-of-Service-Request message between the SCCP at the subsystem's node and the SCCP at the duplicatesubsystems, node.It is used for SCCP subsystem management.

26. Subsystem-Out-of-Service-Grant (SOG) : A Subsystem-Out-of-Service-Grant message is sent, in



22 of 54 02-10-2013 11:56

Posted 27th September 2012 by Pramod Kumar

Labels: SS7 Protocol Stack

0 Add a comment

11th September 2012

[http://4.bp.blogspot.com/-rS3es3A3NQI/UEmDWcT8ipI/AAAAAAAAAH8

/y2eJXo7pHAI/s1600/3gpp.JPG]

Why LTE? The first question came into mind is why LTE? we are having 2G and 3G well established inmarket, then what is the requirement of LTE or so called 4G.

But before proceeding let me clear LTE is not considered as 4G but the 3.9G due to some limitation.

To answer the question, the subscribers and business users are discovered the power of wirelessbroadband through the advanced phones. Today internet are used for video streaming, live video, YouTube, Maps, Social Sites and web search.Because of so much to do on internet, consumers wants the high speed in data transfer on the go andthe solution is LTE. The standards of LTE developed by 3GPP.

LTE Provides following features and application for users and business:

Improve QoS by decreasing the latency time.provide the connectivity for non-traditional device like carsprovide the communication, entertainments, personal assistance.Improving the servicesReducing the transport network cost.All IP Network (AIPN).

2G/3G vs LTE:

- 2G and 3G supports voice traffic on CS (Circuit Switched) Network and Data service on packet netwok- LTE provides voice and data over IP (packet network); single channel End to End and all-IP. howevercurrent release of LTE (3GPP Release 8) does not support voice over LTE.

LTE Architecture :

LTE - Long Term Evolution



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[http://3.bp.blogspot.com/-VOOHx-XOGpc/UEbrF1tT2lI/AAAAAAAAAHU/gUrMhig9kJ0/s1600/lte_archi.JPG]

Entity Summery:MME (Mobile Management Entity) : MME provides mobility and session control management.SGW (Serving Gateway) : Routes and forwards the user data packets.PGW (PDN Gateway): Provides UE session connectivity to external packet data network (PDN).PCRF : Supports service data flow detection, policy enforcement, and flow based charging.eNodeB : Receive and sends radio signals to and from the antennas. Schedules uplink and downlink data to/from the UE. Provides Ethernet links to the EPC elements and other eNodeB.

LTE eUTRAN architecture elements:

MIMO (Multiple-Input and Multiple-Output): Input output refers to the channels. It requires multipleantennas at transmitter and receivers. It increase throughput.

[http://4.bp.blogspot.com/-gbZy-Kqvn-Q/UEcaBFvzKcI/AAAAAAAAAHo/AR9UNRVc1uA/s1600/lte_mimo.JPG]

LTE Modulation Techniques:

OFDMA (Orthogonal Frequency Division Multiple acce ss): OFDMA on the downlink,



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SC-FDMA (Single-Carrier Frequency Division Multiple Access): SC-FDMA on the uplink, low sensitivity to carrier frequency offset. Chosen over OFDMA for uplink because OFDMA uses a lot ofpower. Lower throughput than OFDMA because no overlap and it require less power. It is used UL toconvey UE battery.

MME Functionality:

Communicates with the HSS for the user authentication and subscriber profile downloads. Communicates with the eNodeB and SGW for the session control and bearer setup.

MME Interfaces:

S10: To other MMEsS1-MME: MME to eNodeB S6a: MME to HSSS11: MME to HSSS13: MME to EIRX1_1 and X2: to IRI for Lawful interception



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[http://3.bp.blogspot.com/-pymIzX984YY/UEmFVPeGUjI/AAAAAAAAAIM/9tLleQia-8U/s1600/lte_interfaces.JPG]

SGW Functionality:

Serves as local mobility anchor for the UE - terminates the packet data network interface towards theeUTRAN.Support user-plan mobility - performs IP routing and forwarding functions and maintains data pathsbetween the eNodeBs and the PGW.

SGW Interfaces:

S5 from the PGW (User and Control traffic)S8 from visiting SGW to Home PGW S11 to the MME (For Control Traffic)S1-U to the eNodeB (User Traffic)

PGW Functionality:

Provide the UE with the IP address.



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to Non-3GPP/3GPP access.Per-User based packet filtering.

PGW interfaces:

S5 from the SGW ( for the user and control traffic)S8 from the visiting SGW to Home PGW (Roaming, user and control traffic)SGi to the packet data network (User traffic).Gx to the PCRF (for the policy control)

PCRF Functionality:

Policy management entity that provides dynamic control of QoS and charging policies for the servicedata flows (SDFs)Decide how the SDFs will be treated by the PGWOn the UE attachments, the PCRF:

Receive the request for the policies for the default bearers.1.Retrieve the user profiles from SPR and executes the rule-sets for the decision for the policy andcharging.

2.

Responds the PGW with the PCC rule.3.

PCRF Interfaces:

Gx to the PCRF (policy contol)Sp to the SPR (For the subscriber repository).


Labels: LTE, 4G

0 Add a comment

10th September 2012

For call related message, there are two type of solutions defined for portability Domain:

A. Mobile Number Portability-Signaling Relay Functi on (MNP- SRF): it is based solution acts onSCCP addressing and also makes use of NP database.

B. IN- Related Solution : IN based solution allows the MSCs to retrieve routing information from NPDB.

A. Mobile Number Portability-Signaling Relay Functi on (MNP- SRF) Scenarios:

MNP Call Flows



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Call A.1 : Call to a non-ported number :

[http://1.bp.blogspot.com/-ymwWPwa2w2I/UESJARDo5SI/AAAAAAAAADQ/pU7N7iI4zIY/s1600/non-ported.JPG]

Fig 1: Call to a non-ported number

From an Originating Exchange a call is set up to MSISDN. The call is routed to the subscriptionnetwork being the number range holder network, if the number is non-ported.

1.

When GMSCa receives the ISUP IAM, , it requests routing information by submitting a MAP SRI tothe MNP_SRF/MATF.

2.

When the MNP_SRF/MATF receives the message, the MNP_SRF/MATF analyses the MSISDN inthe CdPA and identifies the MSISDN as being non-ported. The MNP_SRF/MATF function thenreplaces the CdPA by an HLRB address. After modifying the CdPA, the message is routed to HLRB.

3.

When HLRB receives the SRI, it responds to the GMSCb by sending an SRI ack with an MSRN thatidentifies the MSB in the VMSCb.

4.

GMSCb uses the MSRN to route the call to VMSCb.5.IAM requires special NOA 6.

Call A.2 : Call to the Ported Number – Originating Network = Subscription Network – Direct Routing:

[http://4.bp.blogspot.com/-Kij9u85sfCs/UESK4avA2rI/AAAAAAAAADY/XE0Ivwcd-mE/s1600/ported-direct.JPG]

Fig 2: Call to the Ported Number – Originating Network = Subscription Network

MSA originates a call to MSISDN.1.VMSCa routes the call to the network’s GMSCa.2.When GMSCa receives the ISUP IAM, it requests routing information by submitting a MAP SRI tothe MNP_SRF/MATF.

3.

When the MNP_SRF/MATF receives the message, it analyses the MSISDN in the CdPA andidentifies the MSISDN as being ported into the network. The MNP_SRF/MATF function then replacesthe CdPA by an HLRA address. After modifying the CdPA, the message is routed to HLRA.

4.

When HLRA receives the SRI, it responds to the GMSCa by sending an SRI ack with an MSRN thatidentifies the MSB in the VMSCb.

5.

GMSCa uses the MSRN to route the call to VMSCb.6.

[http://3.bp.blogspot.com/-Ulcgcb-y8Ps/UESOg7RSy6I/AAAAAAAAADo/mcYnuAk3Dps/s1600/ported-out.JPG]Fig3: Mobile Originated Call to a Ported or not known to be Ported Number – Direct Routing

MSA originates a call to MSISDN.1.

VMSCA routes the call to the network’s GMSCA.2.

Call A.3: Mobile Originated Call to a Ported or no t known to be Ported Number –Originating Network=Subscription Network – Direct R outing



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and identifies the MSISDN as not known to be ported or being ported to anothernetwork. As the message is a SRI message, the MNP_SRF/MATF responds to theGMSCa by sending an SRI ack with a RN + MSISDN; For the case the number is notknown to be ported the routing number may be omitted. GMSCa uses the (RN +) MSISDN to route the call to GMSCb in the subscriptionnetwork. Depending on the interconnect agreement, the RN will be added in the IAM ornot.

5.

Call 4: Call to a Ported Number – Indirect Routing

[http://3.bp.blogspot.com/-ayYVUEYKKm4/UESQAgN-doI/AAAAAAAAADw/silf4RmuMIQ/s1600/ported-

indirect.JPG]

Fig4: Call to a Ported Number – Indirect Routing

From an Originating Exchange a call is set up to MSISDN. The call is routed to the number rangeholder network.

1.

When GMSCa in the number range holder network receives the ISUP IAM, it requests routinginformation by submitting a MAP SRI to MNP_SRF/MATF.

2.

When the MNP_SRF/MATF receives the message, it analyses the MSISDN in the CdPA andidentifies the MSISDN as being ported to another network. As the message is an SRI message, theMNP_SRF/MATF responds to the GMSCa by sending an SRI ack with a RN + MSISDN

3.

GMSCa uses the RN + MSISDN to route the call to GMSCb in the subscription network. Dependingon the interconnect agreement, the RN will be added in the IAM or not.

4.

Call A.5: Call to a Ported Number Indirect Routein g with Reference to Subscription Network

[http://3.bp.blogspot.com/-FtDddkt5tlY/UESUN3vxI2I/AAAAAAAAAEI/k_FEH4LO-rM/s1600

/portedout_indirectrouting.JPG]

Fig5: Call to a Ported Number Indirect Routeing with Reference to Subscription Network

From an Originating Exchange a call is set up to MSISDN. The call is routed to the number rangeholder network.

1.

When GMSCA in the number range holder network receives the ISUP IAM, it requests routeinginformation by submitting a MAP SRI to the MNP_SRF/MATF. The TT on SCCP may be set to SRI.

2.

When MNP_SRF/MATF receives the message, MNP_SRF/MATF operation is triggered. TheMNP_SRF/MATF functionality analyses the MSISDN in the CdPA and identifies the MSISDN asbeing ported to another network. As the message is a SRI message, the MNP_SRF/MATF functionrelays the message to the subscription network by adding a routeing number to the CdPA whichinformation may be retrieved from a database. After modifying the CdPA, the message is routed tothe subscription network.

3.

When MNP_SRF/MATF in the subscription network receives the SRI, it responds to the GMSCA inthe number range holder network by sending a SRI ack with a RN + MSISDN.

4.

GMSCA uses the (RN +) MSISDN to route the call to GMSCB in the subscription network;Depending on the interconnect agreement, the RN will be added in the IAM or not.

5.



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MNP_SRF/MATF functionality analyses the MSISDN in the CdPA and identifies the MSISDN asbeing ported into the network. The MNP_SRF/MATF function then replaces the CdPA by an HLRBaddress which information may be retrieved from a database. After modifying the CdPA, themessage is routed to HLRB.When HLRB receives the SRI, it responds to the GMSCB by sending an SRI ack with an MSRN thatidentifies the MSB in the VMSCB.

8.

GMSCB uses the MSRN to route the call to VMSCB.9.

B. IN- Related Solution :

The following network operator options are defined for the MT calls in the GMSC: - Terminating call Query on Digit Analysis (TQoD)- Query on HLR Release (QoHR).

The following network operator option is defined for MO calls in VMSCA and for forwarded calls in theGMSC and VMSCB:- Originating call Query on Digit Analysis (OQoD).

Call B.1 Call to a non-ported number, no NP query required:

[http://4.bp.blogspot.com/-ho398TcUftU/UESWqM3-VVI/AAAAAAAAAEY/NaeROM5v7tA/s1600

/non-ported_b.1.JPG]Fig 6: Call to non-ported Number, no query required

From an Originating Exchange a call is set up to MSISDN. The call is routed to the Number rangeholder network being the Subscription network.

1.

When GMSCB receives the ISUP IAM, it requests routeing information by submitting a MAP SRI tothe HLRB including the MSISDN in the request.

2.

The HLRB requests an MSRN from the MSC/VLRB where the mobile subscriber currently isregistered.

3.

The MSC/VLRB returns an MSRN back to the HLRB.4.The HLRB responds to the GMSCB by sending an SRI ack with an MSRN.5.GMSCB uses the MSRN to route the call to VMSCB.6.

Call B.2: TQoD Number is not ported

[http://4.bp.blogspot.com/-bh3pnZ6qySI/UESlPOKup8I/AAAAAAAAAEo/7EtdE5THQ8M/s1600/Tqod-

Notported.JPG]Fig 7: TQoD Number is not ported

From an Originating Exchange a call is set up to MSISDN. The call is routed to the Number rangeholder network being the Subscription network.

1.

When GMSCB receives the ISUP IAM, it will send a database query to the NPDB as a result ofanalysis of the received MSISDN. The MSISDN is included in the query to the NPDB.

2.

The NPDB detects that the MSISDN is not ported and responds back to the GMSCB to continue thenormal call setup procedure for MT calls.

3.

The GMSCB requests routeing information by submitting a MAP SRI to the HLRB, including theMSISDN in the request.

4.



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The HLRB responds to the GMSCB by sending an SRI ack with an MSRN. 7.GMSCB uses the MSRN to route the call to VMSCB.8.

Call B.3: TQoD Number is ported

[http://2.bp.blogspot.com/-grscysZP02M/UESl5CPCI-I/AAAAAAAAAEw/-JRbA8ane88/s1600/Tqod-ported.JPG]Fig 8: TQoD Number is ported

From an Originating Exchange a call is set up to MSISDN. The call is routed to the Number rangeholder network.

1.

When GMSCA receives the ISUP IAM, it will send a database query, including the MSISDN, to theNPDB as a result of analysis of the received MSISDN.

2.

The NPDB detects that the MSISDN is ported and responds back to the GMSCA with a RouteingNumber pointing out the Subscription network.

3.

The call is routed to the Subscription network based on the Routeing Number carried in ISUP IAMmessage; also the MSISDN is included in IAM.

4.

The GMSCB requests routeing information by submitting a MAP SRI to the HLRB, including theMSISDN in the request. The capability to route messages to the correct HLR is required.

5.


6.

The MSC/VLRB returns an MSRN back to the HLRB. 7.The HLRB responds to the GMSCB by sending an SRI ack with an MSRN. 8.GMSCB uses the MSRN to route the call to VMSCB.9.



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Call B.4: QoHR Number is ported

[http://2.bp.blogspot.com/-plVAprSq4LE/UESm2iHYwVI/AAAAAAAAAE4/GFYIDDj14kc/s1600/HLR-ported.JPG]Fig 9: QoHR Number is ported

From an Originating Exchange a call is set up to MSISDN. The call is routed to the Number rangeholder network.

1.

When GMSCA receives the ISUP IAM, it requests routeing information by submitting a MAP SRI tothe HLRA including the MSISDN in the request.

2.

The HLRA returns a MAP SRI ack with an Unknown Subscriber error since no record was found forthe subscriber in the HLRA.

3.

When GMSCA receives the error indication form the HLRA, this will trigger the sending of adatabase query to the NPDB, including the MSISDN in the query.

4.

The NPDB detects that the MSISDN is ported and responds back to the GMSCA with a RouteingNumber pointing out the Subscription network.

5.


6.


7.


8.


Call B.5: OQoD Number is not ported



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[http://3.bp.blogspot.com/-bP5zgf-YFE8/UESnttyMlrI/AAAAAAAAAFA/9Ot94QcRh2c/s1600/Oqod-notported.JPG]Fig 10: OQoD Number is not ported

A call is initiated by Mobile Subscriber A towards Mobile Subscriber B, using the MSISDN of thecalled subscriber.

1.

When VMSCA receives the call setup indication, it will send a database query to the NPDB as aresult of analysis of the received MSISDN, including the MSISDN in the query.

2.

The NPDB detects that the MSISDN is not ported and responds back to the VMSCA to continue thenormal call setup procedure for MO calls. Depending on database configuration option, the NPDBcould either return a Routeing Number on not ported calls, as done for ported calls, or the call isfurther routed using the MSISDN number only towards the Number range holder network.

3.

The call is routed to the Number range holder/Subscription network based on the MSISDN orRouteing Number carried in ISUP IAM message.

4.

The GMSCB requests routeing information by submitting a MAP SRI to the HLRB, including theMSISDN in the request.

5.


6.

The MSC/VLRB returns an MSRN back to the HLRB.7. The HLRB responds to the GMSCB by sending an SRI ack with an MSRN.8.GMSCB uses the MSRN to route the call to VMSCB.9.

Call B.6: OQoD- Number is ported

[http://3.bp.blogspot.com/-GzIWOpIYcxY/UESoqbnEO8I/AAAAAAAAAFI/Jg14ftyLyDY/s1600/Oqod-ported.JPG]Fig 11: OQoD- Number is ported

A call is initiated by Mobile Subscriber A towards Mobile Subscriber B, using the MSISDN of the1.



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The NPDB detects that the MSISDN is ported and responds back to the VMSCA with a RouteingNumber pointing out the Subscription network.

3.


4.


5.


6.



Labels: MNP, Mobile Number Portability

0 Add a comment

9th September 2012

H.248 is protocol used between the MGC and MG in Master-Slave fashion. MEGACO is similar to MGCP.MGC uses this protocol to control the MG.

MEGACO provide the following enhancement over the MGCP.

- Support multimedia and multi point conference enhanced service.- Improve syntax for more efficient semantic message processing.- TCP and UDP transport support- Support either binary or text encoding.

Message Structure:Message { Transaction { Action { Command { Descriptor { Package { Property { }}}}}}

MTACDPP…

Message : Multiple Transactions can be concatenated into a message, which contains header, theversion, and one or more Transactions. Syntax: MEGECA /version [senderIPAddress]:portNumer

H.248/MEGACO Protocol



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The message header contains the identity of the sender. The message identity is set to a provisionedname of the entity transmitting the message. The version number is two digit numbers, beginning with theversion 1 for the present version of the protocol.

Transaction : Command between the MGC and MG are grouped into the transaction. Each of which identified by Transaction ID. Transaction ID assigned by the sender. Transaction reply isinvoked by receiver. There is one reply invocation per transaction.

Transaction ( transactionID { context ID {{{{ }}}})where Transaction ID = 1 to 4294967295 (a 32bit value) Context ID= null to 65535 ( 16bit value)

Reply ( TID {CID})The TID parameter must be the same as the corresponding transaction request. The CID must bespecifying a value to pertain to all responses for the actions.

Transaction Pending is invoked by the receiver indicates that the transaction is actively being processed,but has not been completed. It is used to prevent the sender from assuming that the TransactionRequest was lost if the transaction takes some time to complete. The syntax for command is :

Pending ( TID {})The TID must be same as the corresponding Transaction request.

The Root property ‘normalMGExecutionTime’ is used to specify the interval within which the MGCexpects a response to any transaction from the MG. Another Root property normalMGCExecutionTime’ isused to indicate the interval within which the MG should expect a response to any transaction from theMGC. Both of these properties are configurable by the MGC and have the following value ranges

NormalMGExecutionTime = 100 to 5000 millisecondsNormalMGCExecutionTime = 100 to 5000 milliseconds

Action : Action is a group of command to be executed in the same context. it does not have an ownidentifier.

Context is identified by a Context_ID, which is assigned by the MG when the first termination is Added toa context. The context is deleted when the last termination is subtracted from a termination.

Command : Command are used to manipulate the logical entity of the protocol connection model, contextand termination. Commands provide the complete control of properties of the context and theterminations.Commands are:

- ADD: The Add command adds a Termination to a Context. The first Termination add to a contextcreates a new Context.

Request: Add = Termination_ID { [MediaDescriptor] [,EventDescriptor] [,SignalsDescritor][,AuditDescriptor]}

Reply: Add = Termination_ID { [MediaDescriptor] [,EventDescriptor] [,SignalsDescritor][,ObservedEventsDescriptor] [,StatisticsDescriptor] [,PackagesDescriptor] [,ErrorDescriptor][,AuditDescriptor]}



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Request: Modify = Termination_ID { [ MediaDescriptor] [,EventDescriptor] [,SignalsDescritor][,AuditDescriptor]}

Reply: Modify = Termination_ID { [MediaDescriptor] [,EventDescriptor] [,SignalsDescritor][,ObservedEventsDescriptor] [,StatisticsDescriptor] [,PackagesDescriptor] [,ErrorDescriptor][,AuditDescriptor]}

- SUBTRACT: The Subtract command disconnects a Termination from its Context and returns statisticson the Termination's participation in the Context. The Subtract command on the last Termination in aContext deletes the Context.

Request: Subtract = Termination_ID { [AuditDescriptor]}

Reply: Subtract = Termination_ID { [ MediaDescriptor] [,EventDescriptor] [,SignalsDescritor][,ObservedEventsDes[,StatisticsDescriptor[,PackagesDescriptor[,ErrorDescriptor] [,AuditDescriptor]}

- MOVE: The Move command atomically moves a Termination to another Context.

Request: Move = Termination_ID { [ MediaDescriptor] [,EventDescriptor] [,SignalsDescritor][,AuditDescriptor]}

Reply: Move = Termination_ID { [ MediaDescriptor] [,EventDescriptor] [,SignalsDescritor][,ObservedEventsDescriptor] [,StatisticsDescriptor] [,PackagesDescriptor] [,ErrorDescriptor][,AuditDescriptor]}

- Audit-value: The AuditValue command returns the current state of properties, events, signals and statistics of Terminations.

Request: AuditValue = Termination_ID { AuditDescriptor}

Reply: AuditValue = Termination_ID { [MediaDescriptor] [,EventDescriptor] [,SignalsDescritor][,ObservedEventsDescriptor] [,StatisticsDescriptor] [,PackagesDescriptor]}

- Audit-Capability: Audit Capability commands returns the all possible proprties, events, signals andstatistics of the Termination.

- Notify : The Notify command allows the MG to inform the MGC of the occurrence of events in the MG.

Request: Notify = Termination_ID { [,ObservedEventsDescriptor] [,ErrorDescriptor]}

Reply: Notify = Termination_ID { [ErrorDescriptor]}

- Service Change: The ServiceChange command allows the MG to notify the MGC that a Termination orgroup of Terminations is about to be taken out of service or has just been returned to service.ServiceChange is also used by the MG to announce its availability to a MGC (registration), and to notifythe MGC of impending or completed restart of the MG. The MGC may announce a handover to the MGby sending it a ServiceChange command. The MGC may also use ServiceChange to instruct the MG totake a Termination or group of Terminations in or out of service.



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Terminations: A Termination is a logical entity on a MG that sources and/or sinks media and/or controlstreams. A Termination is described by a number of characterizing Properties, which are grouped in aset of Descriptors that are included in commands. Terminations have unique identities (TerminationIDs),assigned by the MG at the time of their creation.

There are two type of terminations:Ephemeral Terminations are created by means of an Add command. They are destroyed by means ofa Subtract command. These exist only for the duration of their use. These are created and destroyed.Physical Termination: The physical terminations have the semi-permanent existence on the MGW. Forexample the TDM channels that are exist as long as its provisioned on the MGW.

Context : Context is an association between collections of termination. There is special type of contextcalled null context, which contains the terminations that are not associated with any other termination.

Descriptors: Properties of the termination are organized syntactically into the descriptors. Parametersto the commands are called Descriptors. Many commands share same descriptors.

- Modem Descriptors: Specifies the modem type and parameters.

- Multiplex Descriptors: Associates with the media and bearer in multimedia calls.

- Media Descriptors: Specifies the parameters of all media streams.These parameters are structured intotwo descriptors: a TerminationState descriptor, which specifies the properties of a termination that are notstream dependent and one or more Stream descriptors each of which describes a single media stream.

- Event Descriptors: Specifies the list of events that MG is requested to detect and report.

- EventBuffer Descriptors: Specifies the list of events and their parameters that MG is requested to detectand buffer when EventBufferControl equals to lockStep.

- Signal Descriptors: Specifies the set of signals that media gateway is asked to apply to terminations.

- Audit Descriptors: Specifies the information is to be audited.

- Service Change Descriptors: Specifies the parameters between the MGC and MG when MG power up,termination state change, MG or MGC failure happens, or MGC handoff.

- Digit Map Descriptors: A DigitMap is a dialing plan resident in the MG used for detecting and reportingdigit events received on a termination.

- Statistics Descriptors: The Statistics descriptor provides information describing the status and usage ofa termination during its existence within a specific Context.

- Package Descriptors: returns the list of package realized by the termination and it is used with theAuditValue command.

- ObservedEvent Descriptors: notify event to MGC when detected used with the Notify Command.

- Topology Descriptors: A Topology descriptor is used to specify flow directions between terminations in acontext. The topology descriptor applies to a context instead of a termination.



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Package:All properties, Events, Signals, and statistics used in the H.248 protocols are defined in packages. it isuniquely identified by the packageID. Following are the Package defined:

- Generic (g): Signal Completion Event

- Base Root (root): Defines the generic properties of MG i.e. max number of contexts, system times valueand terminations

- Tone Generator (tg): define signal to generates the Tones.

- Tone Detection (tonedet): Defines events for audio tone detection. Needed for detection the DTMFtones.

- Basic DTMF Generators (dg): Defines signals for DTMF generation

- DTMF Detection (dd): Defines events for DTMF detection.

- Call Progress tones generators (cg): defines signals for call progress tones generation.

- Basic Continuity (ct): Defines events & signals for continuity tests.

- Network (nt): : Defines termination properties independent of the network type .

- Real Time Transport protocol (rtp): RTP transfer of the multimedia stream.

- TDM Circuits (tdmc): use for termination to support gain and echo control.

- Generic Announcements: Allows to support announcement functionality at a MG. Announcement will beplayed endlessly by the MG, until the MGC stops the announcement.

- Media gateway resource congestion handling (chp): Allows the MG to control its load.

- 3GUP (threegup): Configures the User Plane functions in the MG

- TFO (threegtfoc): Defines events and properties for Tandem Free Operation

- Bearer characteristics (bcp) : Identify which bearer services are to be supported by a MG

Posted 9th September 2012 by Pramod

Labels: VOIP (Voice over IP), MEGACO, H.248

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8th September 2012 Mobile Number Portability (MNP)



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subscription network within a portability domain while retaining her original MSISDN.

[http://3.bp.blogspot.com/-veVG8GphPhQ

/UER4tAYyW1I/AAAAAAAAACw/Fu9cOkFxiVo/s1600/port1.JPG]

Fig 1: General Architecture of Portability for Calls

Few Definitions to understand the MNP feature:

Number Range Holder Network (NRHN): The Network to which number range containing the portednumber has been allocated. For Example if a number which is in process of porting is belongs to Vodafone network, then theVodafone treated as Number Range Holder Network (NRHN).

Donor Network : A subscription network from which a number is ported-out in porting process. This mayor may not be the Number range holder network. For example if a subscriber is ported-out from Vodafone to Airtel, then Vodafone network if called asDonor network. If this number range belongs to Vodafone then it also called NRHN so in this NRHN andDonor network would be same i.e. Vodafone. but if this number is already ported in to Vodafone fromIdea and again it is ported out to Airtel, then the Vodafone network is called as only Donor network andIdea would be NRHN.

Number Portability Database (NPDB ): A Database which provides portability information like Number isported out or not (portability Status) and if ported out then provides the Routeing number (RN).

Routeing Number (RN) : A Routeing number route the call to recipient network or subscription network.This is provided by NPDB or MNP-SRF.

Number Portability Status: Information indicating the status of portability for Mobile subscriber.Portability can be:



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- Foreign number ported out to foreign network.- Foreign number not know to ported

Originating Network: Network where the calling party located.

Subscription Network or Recipient Network : An recipient network for the Mobile subscriber in portingprocess. For Example if a number which is in process of porting is belongs to Vodafone network and ported out toAirtel, then Airtel would be Recipient Network in this case.

Following routing methods are mentioned: these are the method implemented in portability domain basedon the operator agreements.

- Direct Routing: Direct Routing of calls is PLMN options that allows to route calls directly from thePLMNs supporting this option to the ported subscriber's subscription network. For example: In the Fig 1, if a message(7) is originated inside the portability domain, in a PLMN supportsthe direct routing, this IAM (7) directly routed to the subscription network.

- Indirect Routing : Indirect Routeing of calls is a PLMN option which allows to route calls from the PLMNsupporting this option via the number range holder network to the ported subscriber’s subscriptionnetwork. For example In Fig 1, If a message(2) originated inside the portability domain, in a PLMN support indirectrouting routes this IAM(2) to Number Range holder network. The Number range holder network route thecalls to subscription network.

- If a call originated outside the Portability domain, then the IAM(1) routes to NRHN and then NRHNroutes message(1) to Subscription network.

What changes needed to perform the portability:case 1: if the number range holder network is identical with the donor network (Example if number is ofAirtel and ported to Vodafone):

Airtel(NRHN, Donor N/W) Vodafone (Recipient N/w) Idea ( Other N/w,Direct Routing) ================================================================================HLR REMOVE ADD NPDB rn=ADD to voda ADD ADDfor vodafonecase 2: if the number range holder network is identical with the recipient network:(Example if number isof Airtel and ported in to airtel from vodafone) – Normal call.

Airtel(NRHN, receipint) Vodafone (DONOR N/w) Idea ( Other N/w, DirectRouting) ================================================================================HLR ADD REMOVENPDB rn=remove remove removeCase 3: if the number range holder network is different from both the recipient and the donor network:(Example if number is of Airtel and ported in to idea from vodafone) Airtel (NRHN) Vodafone(Donor N/W) Idea (Recipient N/w) BSNL(



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HLR delete addNPDB rn=update delete add update


Labels: MNP, Mobile Number Portability

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7th September 2012

SIP stack handled over the following layer and data transfer based on following Internet Media Protocolstack:

Application Layer: H.232, SIP, RTP, DNS, DHCPTransport Layer: TCP, UDPInternet Layer: IPPhysical Layer: ATM, V90, Ethernet, Wireless 802.11

1. Physical Layer : it can be following: - Ethernet LAN, - DSL, - ATM - Wireless 802.11 network.

2. Internet Layer : used to route the packet across the network using the destination IP address. IP offersfollowing functionality and drawbacks with simplicity: -- Connection less -- Best-effort packet delivery protocol. -- IP packets can be lost -- IP packets can be delayed -- IP packets cab be received out of sequence. -- IP packet are not Acknowledged.

IP Address used over the public internet are assigned in blocks by the IANA (Internet AssignedNumber Association and IP address are globally unique.

3. Transport Layer: TCP/UDP/SCTP

Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP)



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- Connection oriented transport over IP.- TCP uses the sequence numbers- TCP uses ACK for reliability of transfer of message.- Lost segments retransmitted until they are successfully received- TCP works on well know port number.

The TCP cleint sends SYN (synchronization) message to open the connection, which (SYN)contains initial sequence number, the client will use during the connection. The server respond with SYNmessage containing own initial sequence number, and an acknowledgement number, indicating that itreceived the SYN from client. The client completes the three-way handshake with an ACK or a DATApacket with the AK flag set to the server acknowledging the server's sequence number. Now that theconnection is open, either client or server can send data in DATA packets called segments.

After sending the the segment, sender starts the timer,and if it expires, sender resend the segment.FINmessage use to close the TCP connection. Window size is use representing the initial maximum numberof unacknowledged segments is sent.

TCP-Client TCP-Server ------------------------SYN(SN_client)-----------------------> <--------------SYN/AK(SN_server, SN_client)---------------- ----------------------------ACK-------------------------------> --------------------------------DATA--------------------------> .................... <--------------------------------FIN---------------------------- ---------------------------------ACK--------------------------->

3.2 UDP:

- UDP provides the unreliable transport across the Internet. No Ack of sent datagram. - It does not have complexity like TCP. - Best effort delivery service. 3.3 TLS:

TLS is based on SSL (secure sockets layer) and uses TCP for transport. it is used in https URIschemes. TLS protocol have two layers:

3.3.1 TLS Trasport protocol : - Used to provide the reliable and private transport, - It is encrypted so that third party can not intercept the data.

3.3.2 TLS Handshake protocol : - Used to establish the connection - Negotiate the encryption keys used by TLS transport protocol and provide the authentication.

3.4 SCTP: SCTP is steam-based protocol. it is similar like TCP but have some advantage over theTCP. Advantage of SCTP: - Segmentation - multi hoaming



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The SIP is an application layer protocol develop by IETF to setup, modify, and tear down multimediasession such as Internet telephony calls over IP.

DNS ( Domain Name Service): - DNS is used in the internet to map a symbolic name (like thomas.com) to an IP address (like100.100.100.1).- Domain is used to give the internet a human friendly feel.- Domain names are organised in hierarchy. Each level of name is separated by the dot, with the highestlevel domain on the right hand side.

Address Record (A Record):- CNAME (Canonical Name)- MX (Mail Exchange)- TXT (Free Form text record)- SRV (Service Record)- NAPTR (Naming Authority Pointer)

SIP Request Message:- INVITE- ACK- BYE- REGISTER- OPTIONS- CANCEL

SIP Responses: Response code are generated with Numerical Numbers- 1xx (Informational Class) like 100 Trying, 180 Ringing, 183 Session Progress.- 2xx (Final Response Class) like 200 ok.- 3xx (Forwarding Class)- 4xx- 5xx

SIP Call Flow:

UACa UACb

INVITE -------------------------------------->100 Trying ------------------------------------->

<----------------------------------- 180 Ringing <----------------------------------- 200 OK

ACK ------------------------------------>

<-------------Media Session Established--------------->

<------------------------------------ BYE -------------------------------------> 200 OK

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP lab.high-voltage.org:5060;branch=z9hG4bKfw19bMax-Forwards: 70To: G. Marconi <sip:[email protected]>



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Subject: About That Power Outage...Contact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 158v=0o=Tesla 2890844526 2890844526 IN IP4 lab.high-voltage.orgs=Phone Callc=IN IP4 100.101.102.103t= 0 0m= audio 1201 RTP/AVP 0 98a=rtpmap:0 PCMU/8000a=rtpmap:98 AMR/8000a=fmtp:98 mode-set=0,2,4,7

SIP is text-encoded protocol. Description of header information:- Via contains the address at which sender is expecting to receive responses to this request. usuallywritten as a host name that can be resolved into an IP address using a DNS query. It also contains abranch parameter that identifies this transaction.

- Max-Forwards header field, which is initialized to some large integer and decremented by each SIPserver, which receives and forwards the request, providing simple loop detection.

- From header fields, which show the originator of the SIP request.

- TO header fields, which show the destination of the SIP request.

- Call-ID contains a globally unique identifier for this call, generated by the combination of a randomstring and the softphone's host name or IP address.

The combination of the local tag (contained in the From header field), remote tag (contained in the Toheader field), and the Call-ID uniquely identifies the established session, known as a “dialog.”

- CSeq or Command Sequence contains an integer and a method name. The CSeq number isincremented for each new request within a dialog and is a traditional sequence number.

The Via header fields plus the Max-Forwards, To, From, Call-ID, and CSeq header fields represent theminimum required header field set in any SIP request message.

- Contact contains a SIP or SIPS URI that represents a direct route to contact to sender. Itis mandatory in invite request.

- Content-Type contains a description of the message body(SDP).

- Content-Length contains an octet (byte) count of the message body.

The 180 Ringing response has the following structure:SIP/2.0 180 RingingVia: SIP/2.0/UDP lab.high-voltage.org:5060;branch=z9hG4bKfw19b;received=100.101.102.103To: G. Marconi <sip:[email protected]>t;;tag=a53e42From: Nikola Tesla <sip:[email protected]>;tag=76341Call-ID:[email protected]: 1 INVITE



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Via header field contains original branch parameter and additional received parameter, this parametercontains the literal IP address that the request was received from (IP address of requester from DNS).

To and From header field are not reversed but same as INVITE, which show SIP indicates the directionof request, not the direction of message. To header now contains the tag. Response also contains thecontact which have direct address of recipient.

SIP/2.0 200 OKVia: SIP/2.0/UDP lab.high-voltage.org:5060;branch=z9hG4bKfw19b;received=100.101.102.103To: G. Marconi <sip:[email protected]>;tag=a53e42From: Nikola Tesla <sip:[email protected]>;tag=76341Call-ID:[email protected]: 1 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 155v=0o=Marconi 2890844528 2890844528 IN IP4 tower.radio.orgs=Phone Callc=IN IP4 200.201.202.203t=0 0m=audio 60000 RTP/AVP 0a=rtpmap:0 PCMU/8000

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP lab.high-voltage.org:5060;branch=z9hG4bK321gMax-Forwards: 70To: G. Marconi <sip:[email protected]>;tag=a53e42From: Nikola Tesla <sip:[email protected]>;tag=76341Call-ID: <[email protected]: 1 ACKContent-Length: 0

ACK have the same Cseq number but method is set to ACK. Branch parameter in the Via header fieldcontains a new transaction identifiers than the invite, since an ACK sent to acknowledge a 200 OK isconsidered a separate transaction.This message exchange shows the SIP as an end-to-end signalingprotocol.

a BYE request is sent by Marconi to terminate the media session:

BYE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP tower.radio.org:5060;branch=z9hG4bK392kfMax-Forwards: 70To: Nikola Tesla <sip:[email protected]>;tag=76341From: G. Marconi <sip:[email protected]>;tag=a53e42Call-ID: [email protected]: 1 BYEContent-Length: 0

The Via header field in this example is populated with Marconi’s host address and contains a newtransaction identifier since the BYE is considered a separate transaction from the INVITE or ACK



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tear down the correct media session.

SIP/2.0 200 OKVia: SIP/2.0/UDP tower.radio.org:5060;branch=z9hG4bK392kf;received=200.201.202.203To: Nikola Tesla <sip:[email protected]>;tag=76341From: G. Marconi <sip:[email protected]>;tag=a53e42Call-ID: <a href="mailto:[email protected]">[email protected]</a>CSeq: 1 BYEContent-Length: 0

User Agent Client (UAC) initiates the request and User Agent Server (UAS) generates the responses.During the call, a user agent will usally operate as both UAC and a UAS. A UA must understand anyextensions listed in a Require header field in a request.

A UA should advertise its capabilities and features in any request it sends. This allows other UAs to learnof them without having to make an explicit capabilities query.For example, the methods that a UA supports should be listed in an Allow header field. SIP extensionsshould be listed in a Supported header field.Message body types that are supported should be listed in an Accept header field.

SIP has two broad categories of URIs:- Address of Record (AoR): corresponds to the user, it requires the database look-up and can be sent tomore than one device. Usually it is populated in To, From.- Contact: it is device URI, and typically not required the database lookup.

SIP call with the Proxy:

Marconi Proxy Server Tesla==================================================-------INVITE-------------> ------------INVITE---------------> <---------------180 RINGING----<------180 Ringing---------- <--------------200 OK-----------<------200 OK-------------

----------------------------------ACK--------------------------><=================Media Session Established=========><----------------------------BYE--------------------------------------------------------------200 OK------------------------------>

Proxy is not really in the call. It facilitates the two end points locating and contacting each other. Proxycan be used in further required message using the Record-Route header field.Media is always end-to-end and not through the proxy.

REQUEST: INVITE, REGISTER, BYE, ACK, CANCEL and OPTIONS original six methods in SIP 3261.The REFER, SUBSCRIBE, NOTIFY, MESSAGE, UPDATE, INFO, and PRACK methods are describedin separate RFCs.

- INVITE: used to establish the media session b/w UAs. INVITE is always acknowledge by ACK method.If the media information contained in the ACK is not acceptable, then the called party must send a BYE



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between the UAC and the UAS. A successful INVITE.

An INVITE sent for an existing dialog references the same Call-ID as the original INVITE and containsthe same To and From tags. called a re-INVITE, the request is used to change the session characteristicsor refresh the state of the dialog. The CSeq command sequence number is incremented so that a UAScan distinguish the re-INVITE from a retransmission of the original INVITE. UPDATE is sent to refersh thediagloue if media session did not established.

- BYE: The BYE method is used to terminate an established media session. Cancel is used to terminatethe call which did not establish the media session. BYE can only initiated by user agents. Proxies caninitiates it.

- ACK : The ACK method is used to acknowledge final responses to INVITE requests.An ACK maycontain an application/sdp message body. This is permitted if the initial INVITE did not contain a SDPmessage body. If the INVITE contained a message body, the ACK may not contain a message body. TheACK may not be used to modify a media description that has already been sent in the initial INVITE; are-INVITE must be used for this purpose.

- CANCEL : The CANCEL method is used to terminate pending searches or call attempts. User agentand or Proxy can generate it.

- OPTIONS: The OPTIONS method is used to query a user agent or server about its capabilities anddiscover its current availability. A success class (2xx) response can contain Allow, Accept, Accept-Encoding, Accept-Language, and Supported headers indicating its capabilities.

- SUBSCRIBE : The SUBSCRIBE method [5] is used by a user agent to establish a subscription for thepurpose of receiving notifications (via the NOTIFY method) about a particular event.

- NOTIFY: The NOTIFY method [5] is used by a user agent to convey information about the occurrenceof a particular event.

- REFER: The REFER method is used by a user agent to request another user agent to access a URI orURL resource. The resource is identified by a URI or URL in the required Refer-To header field. theREFER is probably being used to implement a call transfer service.

- MESSAGE: The MESSAGE method is used to transport instant messages (IM) using SIP.- INFO: The INFO method is used by a user agent to send call signaling information to another useragent with which it has an established media session. This is different from a re-INVITE since it does notchange the media characteristics of the call. The request is end-to-end, and is never initiated by proxies.

- PRACK : The PRACK method is used to acknowledge receipt of reliably transported provisionalresponses. A PRACK is generated by a UAC when a provisional response has been received containinga RSeq reliable sequence number and a Supported: 100rel header. The PRACK echoes the number inthe RSeq and the CSeq of the response in a RAck header

SIP/2.0 180 RingingVia: SIP/2.0/UDP lucasian.trinity.cambridge.edu.uk;branch=z9hG4bK452352;received=1.2.3.4To: Descartes <sip:[email protected]>;tag=12323From: Newton <sip:[email protected]>;tag=981Call-ID: <a href="mailto:[email protected]">[email protected]</a>RSeq: 314CSeq: 1 INVITE



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Via: SIP/2.0/UDP lucasian.trinity.cambridge.edu.uk;branch=z9hG4bKdtywMax-Forwards: 70To: Descartes <sip:[email protected]>;tag=12323From: Newton <sip:[email protected]>;tag=981Call-ID: <a href="mailto:[email protected]">[email protected]</a>CSeq: 2 PRACKRAck: 314 1 INVITEContent-Length: 0

SIP/2.0 200 OKVia: SIP/2.0/UDP lucasian.trinity.cambridge.edu.uk;branch=z9hG4bKdtyw ;received=1.2.3.4To: Descartes <sip:[email protected]>;tag=12323From: Newton <sip:[email protected]>;tag=981Call-ID: <a href="mailto:[email protected]">[email protected]</a>CSeq: 2 PRACKContent-Length: 0

-UPDATE: The UPDATE method is used to modify the state of a session without changing the state ofthe dialog, used when offer -answer model (media session) is not completed.

Responses:- 180 Ringing : This response is used to indicate that the INVITE has been received by the user agentand that alerting is taken place.- 181 Call is Being Forwarded: This response is used to indicate that the call has been handed off toanother end-point.- 182 Call Queued : This response is used to indicate that the INVITE has been received, and will beprocessed in a queue.- 183 Session Progress : A typical use of this response is to allow a UAC to hear ring tone, busy tone, ora recorded announcement in calls through a gateway into the PSTN.

Headers :

CSeq: The command sequence CSeq header field is a required header field in every request. The CSeqheader field contains a decimal number that increases foreach request. Usually, it increases by 1 for each new request, with the exception of CANCEL and ACKrequests, which use the CSeq number of the INVITE request to which it refers.

From:The From header field is a required header field that indicates the originator of the request.A Fromheader field may contain a tag, used to identify a particular call.

Subject :The contents of the header field can also be displayed during alerting to aid the user in deciding whetherto accept the call.

supported :The contents of the header field can also be displayed during alerting to aid the user in deciding whetherto accept the call.




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6th September 2012

M3UA is a protocol for supporting the transport of any SS7 MTP3-User signaling (e.g., ISUP andSCCP messages) over IP using the services of the Stream Control Transmission Protocol (SCTP). Inaddition, provision is made for protocol elements that enable a seamless operation of theMTP3-User peers in the SS7 and IP domains. This protocol would be used between a Signaling Gateway(SG) and a Media Gateway Controller (MGC) or IP-resident Database.

SG: SG is a Signaling Agent, which interface between the SS7 and IP network. It appears the SS7 SP tothe SS7 . SG is logical entity with contains the one of more SGP.

SGP: SGP is logical instance of SG. SGP is a physical entity, which has the association between to ASP.All SGP within the SG have the view towards the SS7 Network.

AS: AS is a logical entity, which serves as a specific Routing key. It is present in IP domain. An AScontains set of one or more ASP. There is 1:1 mapping of AS to routing key i.e. there is one routing perAS.

ASP: ASP is process instance AS. it is physical entity which have association with the SGP. Unlike SGP,An ASP can be part of one or more AS.

IPSP: IPSP is a process instance of IP-based application. it is same as ASP but not connected to SS7network via SGP. When two users are present in IP domain we only need IPSP.

Routing Key: Routing key is set of SS7 parameters and parameter values that uniquely define thesignaling traffic to be handled by particular AS.

Routing Context: RC is a unique identifier of a Routing key. RC can user configurable. The scope of RCis global on SGP side.

ASP/IPSP SGP/IPSP =================================== ——————- ASP UP————–> INACTIVE State <—————–ASP UP ACK——— ——————-ASP ACTIVE——–> ACTIVE state <————ASP ACTIVE ACK——- ——————BEAT——————-> <—————-BEAT ACK————-

To make down the M3UA association:

ASP/IPSP SGP/IPSP ======================================= ——————-ASP DOWN————–> <——————ASP DOWN ACK——-

MTP 3 User Adaptation (M3UA) - SIGTRANStack


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Labels: SIGTRAN Stack - M3UA

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4th September 2012

SCTP is designed to transport PSTN signaling messages over IP networks, but is capable of broader

applications. SCTP is a reliable transport protocol operating on top of a connectionless packet network such as

IP.

It offers the following services to its users:

- Error Free, Non-duplicated transfer of user data.

- Data fragmentation.

- Optional bundling of multiple user message in one single SCTP packet.

- Fault tolerance using the multi-homing.

SCTP provide following features:- it is transport layer protocol, like TCP and UDP.

- It is unicast protocol communication between 2 endpoint.

- It is session oriented protocol. it creates association between the endpoint. Endpoints are identified by the IP

address and logical port number.

- It provide the multihoming - more than one IP address of one endpoint to provide the multi-path, endpoints are

identified by the port number. Only one path (association) can be active at a given time. multi homing is provided

for path failure (redundancy) not for load sharing.

- Provide the reliable transmission using SACK method. Retransmission take place time out in ACK has the gap

in TSN.

- provide the path failure detection using the heartbeat mechanism.

- provide the security consideration using the verification tag and cookies.

- It is message oriented protocol.

SCTP Association initialization EndPoint-A EndPoint-B

closed state ———INIT(veri tag, init tag, IP)————–> Cloesed State

cookie wait state <—–INIT ACK(init tag,IP,verification tag)—-

cookie echoed ————–COOKIE_ECHO (cookie)————>

Established state <————-COOKIE ACK————————– Established

<—————–DATA——————————-

——————–SACK—————————->

init and init ack must not be bundled with other chunk. if an error received at init/initAck, ABORT is sent.

Handle Stream: Endpoints sends (in init and initACK) the number of outbound stream (OS), and maximum

inbound stream. if peer’s MIS is less than the endpoints OS, than the endpoint either use the MIS outbound

stream, or abort the association.

Shutdown the association :

ENDPoint-A ENDPoint-B

———-SHUTDOWN—————–>

<——–SHUTDOWN ACK————

———SHUTDOWN COMPLETE–>

Chunk : A unit of information within an SCTP packet, consisting of a chunk header and chunk-specific content.

Congestion Window (cwnd): An SCTP variable that limits the data, in number of bytes, a sender can send to a

Stream Control Transmission Protocol (SCTP)



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Receiver Window (rwnd): An SCTP variable a data sender uses to store the most recently calculated receiver

window of its peer, in number of bytes. This gives the sender an indication of the space available in

the receiver’s inbound buffer.

SCTP association: A protocol relationship between SCTP endpoints, composed of the two SCTP endpoints and

protocol state information including Verification Tags and the currently active set of Transmission Sequence

Numbers (TSNs), etc. An association can be uniquely identified by the transport addresses used by the

endpoints in the association. Two SCTP endpoints MUST NOT have more than one SCTP association between

them at any given time.

SCTP endpoint: The logical sender/receiver of SCTP packets. On a multi-homed host, an SCTP endpoint is

represented to its peers as a combination of a set of eligible destination transport addresses to which

SCTP packets can be sent and a set of eligible source transport addresses from which SCTP packets can be

received. All transport addresses used by an SCTP endpoint must use the same port number, but can use

multiple IP addresses. A transport address used by an SCTP endpoint must not be used by another

SCTP endpoint.

Stream Sequence Number: A 16-bit sequence number used internally by SCTP to assure sequenced delivery

of the user messages within a given stream. One stream sequence number is attached to each user message.

Transmission Sequence Number (TSN): A 32-bit sequence number used internally by SCTP. One TSN is

attached to each chunk containing user data to permit the receiving SCTP endpoint toacknowledge its receipt

and detect duplicate deliveries.

Transport Address: In the case of SCTP running over IP, a transport address is defined by the combination of

an IP address and an SCTP port number (where SCTP is the Transport protocol).

Verification Tag: A 32 bit unsigned integer that is randomly generated. The Verification Tag provides a key that

allows a receiver to verify that the SCTP packet belongs to the current association and is not an old or stale

packet from a previous association.

SCTP Packet Format: SCTP provide the bundling of more than on chunk in one SCTP packet except for the

INIT, INIT ACK, and SHUTDOWN COMPLETE chunks and segmentation if size if giver.

— Common Header —

|Checksum|Verification Tag| Destination Port Address| Source Port Address|

– Source Port Number (16bit, Sender Port Number)

– Destination Port Number (16bit, Receiver Port

– Verification Tag (32bit, to validate the sender, it should same as initiate tag received

in INIT during the starting the association. in INIT, it should be zero and in SHUTDOWN COMPLETE, it should

same as SHUTDOWN-ACK.

- Checksum (CRC32bit, to check the error in packets)

— CHUNK header —

|value|Length|Type| …………… |value|length|Type| SCTP Common Header|

– Chunk Type (8bit, it can be init, initack, shutdown, heartbeat, etc…)

- Chunk Flags (8bit, depend on chunk type, otherwise zero)

- Chunk Length (16bit, provide the length of chunk including the headers)

– Chunk Value (varaible length, actual data Payload)

INIT Chunk: |Type=1|Chunk Flags|Chunk Length|Initiate tag|a_rwnd|Number of OS|Number of IS|Initial

TSN|optianal Param|

SCTP Features:

- Transport Layer Protocol - Alternative to TCP and UDP.

- Uni-cast Protocol - Communication between the 2 end points.

- Session Oriented - "associated" between 2 endpoints.

End points are identified by the near and far end IP address and logical Address.



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are transferred in single stream so that if error occurs, TCP holds up delivery of all data. While SCTP supports

message oriented data transfer in multi stream fashion which insures if errors occurs at one stream there would

be no impact on transmission of other streams data.

[http://1.bp.blogspot.com/-0O56igaFhwg

/UEWd1M7_zUI/AAAAAAAAAGI/ovcLvZQkp6M/s1600/multistreamed.JPG]

Define structured frames of dataAllow to encapsulate upper layer within the SCTP message.

- Reliable Delivery : undelivered messages are re-transmitted.

Using Sequenced acknowledges (SACK)TSN (Transmission sequence numbers) are used to provide reliable delivery.Retransmission takes places if: 1.Timeouts 2. Ack has gap in TSN.

[http://4.bp.blogspot.com/-1z55tffCkG4/UEWfB2pvtkI/AAAAAAAAAGQ/7F7hOFsE2fE/s1600/sctp_features.JPG]


Labels: SIGTRAN

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52 of 54 02-10-2013 11:56

4th September 2012

SIGTRAN (Signalling Transport) is a set of protocols defined to transport SS7 messages overIP networks.

[http://1.bp.blogspot.com/-GjPvWnulWwE

/UEWkth1UhUI/AAAAAAAAAGo/T3fLdFrEXbQ/s1600/sig.JPG]

SIGTRAN allows IP networks to inter-worked with the Public switched Telephone Network (PSTN) andvice-versa.

[http://3.bp.blogspot.com/-ZOEmaQpWYMQ/UEWTnRouBXI/AAAAAAAAAFY/JiRUDCBNrUs/s1600/sigtran.JPG]

The Sigrtan protocol stack based on following components:

[http://4.bp.blogspot.com/-m3veyev1Ros/UEWUzlJSPPI/AAAAAAAAAFg/jHkKW7zb-u8/s1600/sigtran-

component.JPG]

- A Standard IP stack.- A common signalling transport protocol, that we called as SCTP (stream transmission Protocol)- Adaptation Layer: These are the Adaptation layers for MTP2, MTP3 and ISUP, called M2PA, M2UA,

SIGTRAN


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SIGTRAN Stack :

[http://3.bp.blogspot.com/-5ONMD2JDl3k/UEWXicXCeXI/AAAAAAAAAF4/cH8X7eKxaUM/s1600/sigtran_stack.JPG]


Labels: SIGTRAN

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54 of 54 02-10-2013 11:56

the telecom protocols_call flows

Documents

success request

original requesturi

server error

server failure

user agent server

client error

server ready

response classes