29 Oct
2014
29 Oct
'14
8:16 p.m.
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(a)gmail.com>om>:
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(a)gmail.com>om>:
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 Mierlahttp://twitter.com/#!/miconda -
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