Thank you for the fast reply,

I have enabled NAT Traversal like in the default config. The problem seems to be that the cannot be assigned to any transaction.

2014-10-29 13:08 GMT+01:00 Daniel-Constantin Mierla <miconda@gmail.com>:

Hello,

the BYE is coming with a rather strange R-URI. That should be taken from INVITE contact. Also, apparently the INVITE comes from behind NAT, you should use nat traversal logic to update the contact (e.g., add/set alias parameter).

See default configuration file for nat traversal, same should be applied here.

Cheers,
Daniel

On 29/10/14 11:55, Marko Seidenglanz wrote:

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@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@anonymous.invalid>;tag=as32dd9e24

To: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com>

Contact: <sip:anonymous@10.240.177.13:5060>

Call-ID: 21cdb2366ccaab492fd1e69f590620b4@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@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@wh2.24dial.com>;tag=2TQ878KMAVLA43TXVZHNAWCWVKU6BLPBURF3

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

Call-ID: 21cdb2366ccaab492fd1e69f590620b4@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@wh2.24dial.com>;tag=47J6E76F5EVB583A682FQR799J6XDSU46MW8

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

Call-ID: 683def08655393622c55299c1a7c92d5@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

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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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