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:
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@anonymous.invalid>;tag=as32dd9e24
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@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
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as32dd9e24
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
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as0b12a022
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