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 <sr-users-bounces@lists.kamailio.org>
On Behalf Of Linux Vince
Sent: Thursday, April 28, 2022 12:36 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
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:31313004@rigel.com:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 65.20.69.72:5060;rport;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87
From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060;user=phone>;tag=GR52RWG346-34
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:31313004@65.20.67.118;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:31313005@65.20.69.72:5060;user=phone>;tag=GR52RWG346-34
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>
From: "R31313005 N31313005" <sip:31313005@65.20.69.72;user=phone>;tag=GR52RWG346-34
To: "31313004@rigel.com" <sip:31313004@rigel.com>;tag=4ecc1f21f1544fd783965f27ff6d95d2
CSeq: 1 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:31313004@180.211.119.50:57935;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>
From: "R31313005 N31313005" <sip:31313005@65.20.69.72;user=phone>;tag=GR52RWG346-34
To: "31313004@rigel.com" <sip:31313004@rigel.com>;tag=4ecc1f21f1544fd783965f27ff6d95d2
CSeq: 1 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:31313004@180.211.119.50:57935;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:31313004@180.211.119.50:57935;ob SIP/2.0
Via: SIP/2.0/UDP 65.20.69.72:5060;rport;branch=z9hG4bK111270de2729ade9e048afa0f9fbb8df
From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060;user=phone>;tag=GR52RWG346-34
To: "31313004@rigel.com" <sip:31313004@rigel.com>;tag=4ecc1f21f1544fd783965f27ff6d95d2
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