Daily Archives: January 3, 2014

SIP Messages Explanied

Traditional SIP headers for Call setup are INVITE, ACK and teardown are CANCEL or BYE
However with more adoption newer methods specific to services were added such as
MESSAGE Methods for Instant Message based services
SUBSCRIBE, NOTIFY standardised by Event notification extension RFC 3856
PUBLISH to push presence information to the network

Outlining the SIP Requests and Responses in tables below,

1. Request Message

Request Message

Description

REGISTERA Client use this message to register an address with a SIP server
INVITEA User or Service use this message to let another user/service participate in a session. The body of this message would include a description of the session to which the callee is being invited.
ACKThis is used only for INVITE indicating that the client has received a final response to an INVITE request
CANCELThis is used to cancel a pending request
BYEA User Agent Client use this message to terminate the call
OPTIONSThis is used to query a server about its capabilities

2. Response Message

Code

Category

Description

1xxProvisionalThe request has been received and processing is continuing
2xxSuccessAn ACK, to indicate that the action was successfully received, understood, and accepted.
3xxRedirectionFurther action is required to process this request
4xxClient ErrorThe request contains bad syntax and cannot be fulfilled at this server
5xxServer ErrorThe server failed to fulfill an apparently valid request
6xxGlobal FailureThe request cannot be fulfilled at any server

, based on RFC 3261


SIP headers :

Display names are described in RFC 2822
From also contains a display name and a SIP URI that indicate the originator of the request.  The From also contains a tag parameter which is used for identification purposes.
Call-ID contains a globally unique identifier for this call. Mandatory
CSeq or Command Sequence contains an integer and a method name. The CSeq number is incremented for each new request within a dialog and is a traditional sequence number.
Contact contains a SIP URI that represents a direct route to the originator usually composed of a username at a fully qualified domain name (FQDN). While an FQDN is preferred, many end systems do not have registered domain names, so IP addresses are permitted.  The Contact header field tells other elements where to send future requests.
Max-Forwards serves to limit the number of hops a request can make on the way to its destination. It consists of an integer that is decremented by one at each hop.
Content-Type contains a description of the message body.
Content-Length contains an octet (byte) count of the message body.
 
sip headers 1
sip headers 2
sip headers 3

Mandatory SIP headers

  • INVITE sip:altanai@domain.comSIP/2.0
  • Via: SIP/2.0/UDP host.domain.com:5060
  • From: Bob <sip:bob@domain.com>
  • To: Altanai <sip:domain@wcom.com>
  • Call-ID: 163784@host.domain.com
  • CSeq: 1 INVITE

session description in SDP

sdp

  • v=  (protocol version)  Mandatory
  • o=  (owner/creator and session identifier).   Mandatory
  • s=  (session name)   Mandatory
  • t=  (time the session is active)   Mandatory
  • i=* (session information)
  • u=* (URI of description)
  • e=* (email address)
  • p=* (phone number)
  • c=* (connection information – not required if included in all media)
  • b=* (bandwidth information)
  • z=* (time zone adjustments)
  • k=* (encryption key)
  • a=* (zero or more session attribute lines)
  • r=* (zero or more repeat times)Media description
  • m=  (media name and transport address)  Mandatory
  • i=* (media title)

TYPICAL SIP INVITE :


INVITE&amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;nbsp;sip:01150259917040@67.135.76.4 SIP/2.0
Via: SIP/2.0/UDP 69.7.163.154:5060;branch=z9hG4bK400fc6e6
From: "8069664170" &amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;lt;sip:8069664170@69.7.163.154&amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;gt;;tag=as42e2ecf6
To: &amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;lt;sip:01150259917040@67.135.76.4&amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;gt;
Contact: &amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;lt;sip:8069664170@69.7.163.154&amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;gt;
Call-ID: 2485823e63b290b47c042f20764d990a@69.7.163.154
CSeq: 102 INVITE
User-Agent:&amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;nbsp;MatrixSwitch
Date: Thu, 22 Dec 2005 18:38:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 14040 14040 IN IP4 69.7.163.154
s=session
c=IN IP4 69.7.163.154
t=0 0
m=audio 26784 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=fmtp:18&amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;amp;nbsp;annexb=no - - - -
c=* (connection information - optional if included at session-level)
b=* (bandwidth information)
a=* (zero or more media attribute lines)

SIP Responses

sip resp

1xx—Provisional Responses

response that tells to its recipient that the associated request was received but result of the processing is not known yet which could be if the processing hasnt finished immediately. The sender must stop retransmitting the request upon reception of a provisional response.

100 Trying
180 Ringing
181 Call is Being Forwarded
182 Queued
183 Session in Progress199 Early Dialog Terminated

2xx—Successful Responses

final responses express result of the processing of the associated request and they terminate the transactions.

200 OK
202 Accepted
204 No Notification

3xx—Redirection Responses

redirection response gives information about the user’s new location or an alternative service that the caller should try for the call. Used for cases when the server cant satisfy the call and wants the caller to try elsewhere . After this the caller is suppose to resend the request to the new location.

300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
305 Use Proxy
380 Alternative Service

4xx—Client Failure Responses

negative final responses indicating that the request couldn’t be processed  due to callers fault , for reasons such as t contains bad syntax or cannot be fulfilled at that server.

400 Bad Request
401 Unauthorized
402 Payment Required
403 Forbidden
404 Not Found
405 Method Not Allowed
406 Not Acceptable
407 Proxy Authentication Required
408 Request Timeout
409 Conflict
410 Gone
411 Length Required
412 Conditional Request Failed
413 Request Entity Too Large
414 Request-URI Too Long
415 Unsupported Media Type
416 Unsupported URI Scheme
417 Unknown Resource-Priority
420 Bad Extension
421 Extension Required
422 Session Interval Too Small
423 Interval Too Brief
424 Bad Location Information
428 Use Identity Header
429 Provide Referrer Identity
430 Flow Failed
433 Anonymity Disallowed
436 Bad Identity-Info
437 Unsupported Certificate
438 Invalid Identity Header
439 First Hop Lacks Outbound Support
470 Consent Needed
480 Temporarily Unavailable
481 Call/Transaction Does Not Exist
482 Loop Detected.
483 Too Many Hops
484 Address Incomplete
485 Ambiguous
486 Busy Here
487 Request Terminated
488 Not Acceptable Here
489 Bad Event
491 Request Pending
493 Undecipherable
494 Security Agreement Required

5xx—Server Failure Responses

negative responses but indicating that fault is at server’s side for cases such as server cant or doesnt want to respond the the request.

500 Server Internal Error
501 Not Implemented
502 Bad Gateway
503 Service Unavailable
504 Server Time-out
505 Version Not Supported
513 Message Too Large
580 Precondition Failure

6xx—Global Failure Responses

request cannot be fulfilled at any server with definitive information

600 Busy Everywhere
603 Decline
604 Does Not Exist Anywhere
606 Not Acceptable

Mandatory Headers in SIP Response 
  • SIP/2.0 200 OK
  • Via: SIP/2.0/UDP host.domain.com:5060
  • From: Bob<sip:bob@domain.com>
  • To: Altanai<sip:altanai@domain.com>
  • Call-ID: 163784@host.domain.com
  • CSeq: 1 INVITE
Note : – 

Via, From, To, Call-ID , and  CSeq  

are copied exactly from Request. 

You can read more about SIP based Architecture here :SIP based architecture

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