[SR-Users] 404 Not here as response to BYE requests

[Marko Seidenglanz]

Hello,

We have a setup where Kamailio 4.2 is used in front of Asterisk as WebRTC
Proxy doing the encryption and NAT Traversal.

Everything works as expected, except that BYE Requests sent by the WebRTC
Client are not forwarded by Kamailio to Asterisk. We use record routing.
Instead Kamailio responds with a "404 Not here".


 INVITE Headers: Kamailio --> Browser:

sip:e3W5LffMkg0PZjP3SIGf6 at wh2.24dial.com SIP/2.0
Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on>
Via: SIP/2.0/UDP 104.155.14.169:5064
;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0
Via: SIP/2.0/UDP 10.240.177.13:5060
;received=127.0.0.1;branch=z9hG4bK2d4f70db
Max-Forwards: 70
From: "Anonymous" <sip:anonymous at anonymous.invalid>;tag=as32dd9e24
To: <sip:e3W5LffMkg0PZjP3SIGf6 at wh2.24dial.com>
Contact: <sip:anonymous at 10.240.177.13:5060>
Call-ID: 21cdb2366ccaab492fd1e69f590620b4 at 10.240.177.13:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1
Date: Wed, 29 Oct 2014 10:04:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 676




 BYE Headers Browser --> Kamailio:

BYE sip:anonymous at anonymous.invalid SIP/2.0
Via: SIP/2.0/UDP 104.155.14.169:5064
;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0
 , SIP/2.0/UDP 10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK2d4f70db
From:  <sip:e3W5LffMkg0PZjP3SIGf6 at wh2.24dial.com
>;tag=2TQ878KMAVLA43TXVZHNAWCWVKU6BLPBURF3
To: "Anonymous" <sip:anonymous at anonymous.invalid>;tag=as32dd9e24
Call-ID: 21cdb2366ccaab492fd1e69f590620b4 at 10.240.177.13:5060
CSeq: 0 BYE
Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on>
Reason: Q.850;cause=16




BYE Response Kamailio --> Browser:

SIP/2.0 404 Not here
Via: SIP/2.0/UDP 104.155.14.169:5064
;branch=z9hG4bK65df.6fd637d055286a45aa6f3e12c5ac873c.0
 , SIP/2.0/UDP 10.240.177.13:5060
;received=127.0.0.1;branch=z9hG4bK51c69f33;received=80.255.2.37
From:  <sip:e3W5LffMkg0PZjP3SIGf6 at wh2.24dial.com
>;tag=47J6E76F5EVB583A682FQR799J6XDSU46MW8
To: "Anonymous" <sip:anonymous at anonymous.invalid>;tag=as0b12a022
Call-ID: 683def08655393622c55299c1a7c92d5 at 10.240.177.13:5060
CSeq: 0 BYE
Server: kamailio (4.2.0 (x86_64/linux))
Content-Length: 0



In the logs I can see the following messages:
[loose.c:113]: find_first_route(): No Route headers found
[loose.c:929]: loose_route(): There is no Route HF


Does anybody know, why Kamailio may respond with 404 Not here? Do I have to
send the BYE request directly to Asterisk?

Kind regards,
Marko
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20141029/64d1a7e4/attachment.html>