[SR-Users] t_check_trans failed on ACK

Linux Vince linuxv at gmail.com
Thu Apr 28 12:36:24 CEST 2022


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 at 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 at 65.20.69.72:5060
;user=phone>;tag=GR52RWG346-34
To: "31313004 at rigel.com" <sip:31313004 at rigel.com:5060>
Call-ID: 1558172022-144148450000000029280449 at 65.20.69.72
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 at 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 at 65.20.69.72:5060
;user=phone>;tag=GR52RWG346-34
To: "31313004 at rigel.com" <sip:31313004 at rigel.com:5060>
Call-ID: 1558172022-144148450000000029280449 at 65.20.69.72
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: 1558172022-144148450000000029280449 at 65.20.69.72
From: "R31313005 N31313005" <sip:31313005 at 65.20.69.72
;user=phone>;tag=GR52RWG346-34
To: "31313004 at rigel.com" <sip:31313004 at rigel.com
>;tag=4ecc1f21f1544fd783965f27ff6d95d2
CSeq: 1 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Contact: <sip:31313004 at 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>
Call-ID: 1558172022-144148450000000029280449 at 65.20.69.72
From: "R31313005 N31313005" <sip:31313005 at 65.20.69.72
;user=phone>;tag=GR52RWG346-34
To: "31313004 at rigel.com" <sip:31313004 at rigel.com
>;tag=4ecc1f21f1544fd783965f27ff6d95d2
CSeq: 1 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Contact: <sip:31313004 at 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 at 180.211.119.50:57935;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:31313005 at 65.20.69.72:5060
;user=phone>;tag=GR52RWG346-34
To: "31313004 at rigel.com" <sip:31313004 at rigel.com
>;tag=4ecc1f21f1544fd783965f27ff6d95d2
Call-ID: 1558172022-144148450000000029280449 at 65.20.69.72
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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