If that is the case what is the best way to handle ACK from kamailio? I am using it as proxy server only and for caching registrations.
On Thu, Apr 28, 2022 at 8:27 PM Henning Westerholt <[email protected]> wrote: > Hello, > > > > the ACK to a 200 OK is not part of the (previous) SIP INVITE transaction. > Please refer to the standard for more details: > https://datatracker.ietf.org/doc/html/rfc3261#section-17 > > > > Cheers, > > > > Henning > > > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com > > > > *From:* sr-users <[email protected]> *On Behalf Of *Linux > Vince > *Sent:* Thursday, April 28, 2022 12:36 PM > *To:* Kamailio (SER) - Users Mailing List <[email protected]> > *Subject:* [SR-Users] t_check_trans failed on ACK > > > > I am using Kamailio as registrar and proxy for underlying asterisk server. > > > > I am able to call but ACK on 200 OK respond from client phones are not > validating transaction and thus not relaying correctly. > > > > Below are my Invites and 200 OK responses. > > > > Invite from Asterisk to Kamailio > > > > INVITE sip:[email protected]:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 65.20.69.72:5060 > ;rport;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 > > From: "R31313005 N31313005" <sip:[email protected]:5060;user=phone > >;tag=GR52RWG346-34 > > To: "[email protected]" <sip:[email protected]:5060> > > Call-ID: [email protected] > > CSeq: 1 INVITE > > Contact: <sip:65.20.69.72:5060> > > max-forwards: 70 > > Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, > BYE, CANCEL, INVITE > > Content-Type: application/sdp > > Content-Length: 228 > > > > v=0 > > o=Clarent 155866 155867 IN IP4 65.20.69.72 > > s=Clarent CallManager > > c=IN IP4 65.20.69.72 > > t=0 0 > > m=audio 45168 RTP/AVP 0 8 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > > > Invite from Kamailio to Phone > > > > INVITE sip:[email protected];lhst=180.211.119.50:25759;lm=midreg > SIP/2.0 > > Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> > > Via: SIP/2.0/UDP > 65.20.67.118;branch=z9hG4bK6cc3.dad7580d3e28f55cbb749dbb82f0c9eb.0 > > Via: SIP/2.0/UDP 65.20.69.72:5060 > ;received=65.20.69.72;rport=5060;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 > > From: "R31313005 N31313005" <sip:[email protected]:5060;user=phone > >;tag=GR52RWG346-34 > > To: "[email protected]" <sip:[email protected]:5060> > > Call-ID: [email protected] > > CSeq: 1 INVITE > > Contact: <sip:65.20.69.72:5060> > > max-forwards: 69 > > Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, > BYE, CANCEL, INVITE > > Content-Type: application/sdp > > Content-Length: 228 > > > > v=0 > > o=Clarent 155866 155867 IN IP4 65.20.69.72 > > s=Clarent CallManager > > c=IN IP4 65.20.69.72 > > t=0 0 > > m=audio 45168 RTP/AVP 0 8 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > > > > > 200 OK from Phone to Kamailio > > > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP > 65.20.67.118;received=65.20.67.118;branch=z9hG4bK6cc3.dad7580d3e28f55cbb749dbb82f0c9eb.0 > > Via: SIP/2.0/UDP 65.20.69.72:5060 > ;rport=5060;received=65.20.69.72;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 > > Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> > > Call-ID: [email protected] > > From: "R31313005 N31313005" <sip:[email protected] > ;user=phone>;tag=GR52RWG346-34 > > To: "[email protected]" <sip:[email protected] > >;tag=4ecc1f21f1544fd783965f27ff6d95d2 > > CSeq: 1 INVITE > > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > > Contact: <sip:[email protected]:57935;ob> > > Supported: replaces, 100rel, timer, norefersub > > Content-Type: application/sdp > > Content-Length: 322 > > > > v=0 > > o=- 3860150297 3860150298 IN IP4 180.211.119.50 > > s=pjmedia > > b=AS:84 > > t=0 0 > > a=X-nat:0 > > m=audio 4016 RTP/AVP 0 101 > > c=IN IP4 180.211.119.50 > > b=TIAS:64000 > > a=rtcp:4017 IN IP4 180.211.119.50 > > a=sendrecv > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ssrc:1461220511 cname:3b657c275d2a43f6 > > > > > > > > 200 OK from kamailio to asterisk > > > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP 65.20.69.72:5060 > ;rport=5060;received=65.20.69.72;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 > > Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> > > Call-ID: [email protected] > > From: "R31313005 N31313005" <sip:[email protected] > ;user=phone>;tag=GR52RWG346-34 > > To: "[email protected]" <sip:[email protected] > >;tag=4ecc1f21f1544fd783965f27ff6d95d2 > > CSeq: 1 INVITE > > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > > Contact: <sip:[email protected]:57935;ob> > > Supported: replaces, 100rel, timer, norefersub > > Content-Type: application/sdp > > Content-Length: 322 > > > > v=0 > > o=- 3860150297 3860150298 IN IP4 180.211.119.50 > > s=pjmedia > > b=AS:84 > > t=0 0 > > a=X-nat:0 > > m=audio 4016 RTP/AVP 0 101 > > c=IN IP4 180.211.119.50 > > b=TIAS:64000 > > a=rtcp:4017 IN IP4 180.211.119.50 > > a=sendrecv > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ssrc:1461220511 cname:3b657c275d2a43f6 > > > > ACK from phone to Kamailio > > > > ACK sip:[email protected]:57935;ob SIP/2.0 > > Via: SIP/2.0/UDP 65.20.69.72:5060 > ;rport;branch=z9hG4bK111270de2729ade9e048afa0f9fbb8df > > Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> > > From: "R31313005 N31313005" <sip:[email protected]:5060;user=phone > >;tag=GR52RWG346-34 > > To: "[email protected]" <sip:[email protected] > >;tag=4ecc1f21f1544fd783965f27ff6d95d2 > > Call-ID: [email protected] > > CSeq: 1 ACK > > Contact: <sip:65.20.69.72:5060> > > max-forwards: 70 > > Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, > BYE, CANCEL, INVITE > > > > > > This ACK is failing on transaction check and not routing correctly to > phone. > > > > Any help? > > > > Thanks >
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