I am writing a sip stack in C++. I need to know the all possible responses of SIP requests(REGISTER, INVITE, OPTION etc) separately. So far i have got the following...
:::::::::: All possible responses of REGISTER request ::::::::::
The UAC may receive 2xx, 3xx(except 300, 380) or 4xx(except 402, 409, 480, 481, 486, 487, 488, 489, 491) responses excluding.
:::::::::: All possible responses of INVITE request ::::::::::
The UAC may receive 1xx, 2xx, 4xx(Except 409, 410, 433, 481, 487) responses.
:::::::::: All possible responses of OPTION request ::::::::::
A proxy never generates an OPTION request. A UAC or UAS responds to the OPTION request as it would to an INVITE. It would respond with 4xx(Except 409, 410, 433, 481, 487) or 6xx responses.
:::::::::: All possible responses of ACK request ::::::::::
ACKs are only used for INVITE. The UAC or UAS may send or receive, and ACK in response to the 2xx responses and it is End-to-End. So, for 2xx, ACK is end-to-end. And for non-2xx, ACK is hop-by-hop.
:::::::::: All possible responses of CANCEL request ::::::::::
CANCELs are hop-by-hop request. A UA confirms the call CANCEL request with 2xx response and replies to the invite with 487 response.
:::::::::: All possible responses of BYE request ::::::::::
A BYE is only used to terminate a established media sessions and is sent only by UAs participation in a sessions, never by proxies or other 3rd parties. For a BYE request we can get 481, 2xx responses.
I am sure that those are not all. Please indicate what else need to change/add/delete. And also please indicate if i am wrong. Any help would be highly appreciated...
All possible responses of various SIP requests
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- Posts: 1
- Joined: Wed Nov 06, 2013 7:48 pm
Re: All possible responses of various SIP requests
The SIP RFC contains all that information...
Re: All possible responses of various SIP requests
Never got around to testing this....
Code: Select all
when /^341/ then sys.Dial("hangup$____@conference.freeswitch.org") # hangup$HANGUP_CAUSE@conference.freeswitch.org ## https://wiki.freeswitch.org/wiki/Hangup_Causes OR http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
=begin =====================================
400 Bad Request 41 Temporary Failure
401 Unauthorized 21 Call rejected (*)
402 Payment required 21 Call rejected
403 Forbidden 21 Call rejected
404 Not found 1 Unallocated number
405 Method not allowed 63 Service or option unavailable
406 Not acceptable 79 Service/option not implemented (+)
407 Proxy authentication required 21 Call rejected (*)
408 Request timeout 102 Recovery on timer expiry
410 Gone 22 Number changed (w/o diagnostic)
413 Request Entity too long 127 Interworking (+)
414 Request-URI too long 127 Interworking (+)
415 Unsupported media type 79 Service/option not implemented (+)
416 Unsupported URI Scheme 127 Interworking (+)
420 Bad extension 127 Interworking (+)
421 Extension Required 127 Interworking (+)
423 Interval Too Brief 127 Interworking (+)
480 Temporarily unavailable 18 No user responding
481 Call/Transaction Does not Exist 41 Temporary Failure
482 Loop Detected 25 Exchange - routing error
483 Too many hops 25 Exchange - routing error
484 Address incomplete 28 Invalid Number Format (+)
485 Ambiguous 1 Unallocated number
486 Busy here 17 User busy
487 Request Terminated --- (no mapping)
488 Not Acceptable here --- by Warning header
500 Server internal error 41 Temporary failure
501 Not implemented 79 Not implemented, unspecified
502 Bad gateway 38 Network out of order
503 Service unavailable 41 Temporary failure
504 Server time-out 102 Recovery on timer expiry
504 Version Not Supported 127 Interworking (+)
513 Message Too Large 127 Interworking (+)
600 Busy everywhere 17 User busy
603 Decline 21 Call rejected
604 Does not exist anywhere 1 Unallocated number
606 Not acceptable --- by Warning header
=end =====================================
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VSP: SIPTalk¹, Symbio² & MyNetFone³ via SIPSorcery Cloud PBX. [Warning! Avoid DIDLogic!!! Beware!!]
H/W: Asus RT-AC86U; Cisco SPA232D, 2 x SPA901's, SPA3102; Yealink SIP-T46G.
VSP: SIPTalk¹, Symbio² & MyNetFone³ via SIPSorcery Cloud PBX. [Warning! Avoid DIDLogic!!! Beware!!]
H/W: Asus RT-AC86U; Cisco SPA232D, 2 x SPA901's, SPA3102; Yealink SIP-T46G.