Thanks Philipp That's the same dialog, 100rel present there in subsequent SIP/183.
Let me show the call flow. Substituted calling, called, IPs etc. Can the last SIP/183 be considered as incorrect from https://tools.ietf.org/html/rfc3264#section-8 perspective, because there is "c=" added to media level and "o=" origin must be the same (as previous) except the version must increment by one. INVITE sip:+called_num@req_uri_host:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP calling_VIA:5060;branch=z9hG4bK04B3fb4750baf5bf89b From: "calling_name" <sip:+calling_num@calling_VIA;user=phone>;tag=gK046df8ed To: <sip:+called_num@req_uri_host;user=phone> Call-ID: 52701823_134068140@calling_VIA CSeq: 533420 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: "calling_name" <sip:+calling_num@calling_VIA:5060> Supported: timer,100rel,precondition,replaces Session-Expires: 1800 Min-SE: 90 Content-Length: 182 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=username_1 252686 420906 IN IP4 calling_VIA s=session_1 c=IN IP4 calling_media_IP t=0 0 m=audio 19250 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=maxptime:20 SIP/2.0 100 Trying Via: SIP/2.0/UDP calling_VIA:5060;branch=z9hG4bK04B3fb4750baf5bf89b From: "calling_name" <sip:+calling_num@calling_VIA;user=phone>;tag=gK046df8ed To: <sip:+called_num@req_uri_host;user=phone> Call-ID: 52701823_134068140@calling_VIA CSeq: 533420 INVITE Content-Length: 0 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP calling_VIA:5060;branch=z9hG4bK04B3fb4750baf5bf89b To: <sip:+called_num@req_uri_host;user=phone>;tag=bc6ce47f4s From: "calling_name" <sip:+calling_num@calling_VIA;user=phone>;tag=gK046df8ed Call-ID: 52701823_134068140@calling_VIA CSeq: 533420 INVITE Contact: <sip:called_num@req_uri_host:5060> P-Asserted-Identity: <sip:called_num@P_IP_1;user=phone> Remote-Party-ID: <sip:called_num@P_IP_1;user=phone> Content-Type: application/sdp Content-Length: 149 v=0 o=username_2 188 1 IN IP4 req_uri_host s=session_2 c=IN IP4 called_media_IP_1 t=0 0 m=audio 20158 RTP/AVP 0 a=rtpmap:0 PCMU/8000 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP calling_VIA:5060;branch=z9hG4bK04B3fb4750baf5bf89b To: <sip:+called_num@req_uri_host;user=phone>;tag=bc6ce47f4s From: "calling_name" <sip:+calling_num@calling_VIA;user=phone>;tag=gK046df8ed Call-ID: 52701823_134068140@calling_VIA CSeq: 533420 INVITE Contact: <sip:called_num@req_uri_host:5060> P-Asserted-Identity: <sip:called_num@P_IP_1;user=phone> Remote-Party-ID: <sip:called_num@P_IP_1;user=phone> Content-Type: application/sdp Content-Length: 149 v=0 o=username_2 188 1 IN IP4 req_uri_host s=session_2 c=IN IP4 called_media_IP_1 t=0 0 m=audio 20158 RTP/AVP 0 a=rtpmap:0 PCMU/8000 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP calling_VIA:5060;branch=z9hG4bK04B3fb4750baf5bf89b To: <sip:+called_num@req_uri_host;user=phone>;tag=bb4ffe310s From: "calling_name" <sip:+calling_num@calling_VIA;user=phone>;tag=gK046df8ed Call-ID: 52701823_134068140@calling_VIA CSeq: 533420 INVITE Contact: <sip:called_num@req_uri_host:5060> P-Asserted-Identity: <sip:called_num@P_IP_2;user=phone> Remote-Party-ID: <sip:called_num@P_IP_2;user=phone> Content-Type: application/sdp Content-Length: 150 v=0 o=username_2 188 1 IN IP4 req_uri_host s=session_2 c=IN IP4 called_media_IP_2 t=0 0 m=audio 17724 RTP/AVP 0 a=rtpmap:0 PCMU/8000 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP calling_VIA:5060;branch=z9hG4bK04B3fb4750baf5bf89b From: "calling_name" <sip:+calling_num@calling_VIA;user=phone>;tag=gK046df8ed To: <sip:+called_num@req_uri_host;user=phone>;tag=bae4a15d7s Call-ID: 52701823_134068140@calling_VIA CSeq: 533420 INVITE Require: 100rel RSeq: 8617 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: <sip:called_num@called_media_IP_3>;party=called;screen=no;privacy=off Contact: <sip:called_num@req_uri_host:5060> Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 182 v=0 o=username_3 3814 9111 IN IP4 req_uri_host s=session_3 c=IN IP4 called_media_IP_3 t=0 0 m=audio 19346 RTP/AVP 0 c=IN IP4 called_media_IP_3 a=rtpmap:0 PCMU/8000 CANCEL sip:+called_num@req_uri_host:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP calling_VIA:5060;branch=z9hG4bK04B3fb4750baf5bf89b From: "calling_name" <sip:+calling_num@calling_VIA;user=phone>;tag=gK046df8ed To: <sip:+called_num@req_uri_host;user=phone> Call-ID: 52701823_134068140@calling_VIA CSeq: 533420 CANCEL Max-Forwards: 70 Reason: SIP;cause=503;text="Service Unavailable" Supported: 100rel Content-Length: 0 24.06.2019, 23:06, "Philipp Schöning" <schoenin...@gmail.com>: > Are the different SIP183 in the same dialog or are this different dialogs? > > Is the SIP183 sent reliable (Require: 100rel with PRACK-procedure)? > > Does this c-line apply on the session or media level? > See RFC4566, 5.7. Connection Data ("c=") for more details... > _______________________________________________ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
