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

[Marko Seidenglanz]

The question is, how I can forward the BYE message back to asterisk?

2014-10-29 15:24 GMT+01:00 Marko Seidenglanz <marko.seidenglanz at gmail.com>:

> 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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>>
>>
>
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