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

[Marko Seidenglanz]

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 at 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 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
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
>
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