<div dir="ltr"><div>The question is, how I can forward the BYE message back to asterisk?</div></div><div class="gmail_extra"><br><div class="gmail_quote">2014-10-29 15:24 GMT+01:00 Marko Seidenglanz <span dir="ltr"><<a href="mailto:marko.seidenglanz@gmail.com" target="_blank">marko.seidenglanz@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Thank you for the fast reply,<div><br></div><div>I have enabled NAT Traversal like in the default config. The problem seems to be that the cannot be assigned to any transaction.</div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">2014-10-29 13:08 GMT+01:00 Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
Hello,<br>
<br>
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).<br>
<br>
See default configuration file for nat traversal, same should be
applied here.<br>
<br>
Cheers,<br>
Daniel<div><div><br>
<br>
<div>On 29/10/14 11:55, Marko Seidenglanz
wrote:<br>
</div>
</div></div><blockquote type="cite"><div><div>
<div dir="ltr">Hello,
<div><br>
</div>
<div>We have a setup where Kamailio 4.2 is used in front of
Asterisk as WebRTC Proxy doing the encryption and NAT
Traversal.</div>
<div><br>
</div>
<div>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".</div>
<div><br>
</div>
<div><br>
</div>
<div> INVITE Headers: Kamailio --> Browser:</div>
<div><br>
</div>
<div>
<div><a href="mailto:sip%3Ae3W5LffMkg0PZjP3SIGf6@wh2.24dial.com" target="_blank">sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com</a>
SIP/2.0</div>
<div>Record-Route:
<a><sip:104.155.11.255:5060;nat=yes;lr=on></a></div>
<div>Via: SIP/2.0/UDP
104.155.14.169:5064;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0</div>
<div>Via: SIP/2.0/UDP
10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK2d4f70db</div>
<div>Max-Forwards: 70</div>
<div>From: "Anonymous"
<a><sip:anonymous@anonymous.invalid></a>;tag=as32dd9e24</div>
<div>To: <<a href="mailto:sip%3Ae3W5LffMkg0PZjP3SIGf6@wh2.24dial.com" target="_blank">sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com</a>></div>
<div>Contact: <<a href="http://sip:anonymous@10.240.177.13:5060" target="_blank">sip:anonymous@10.240.177.13:5060</a>></div>
<div>Call-ID: <a href="http://21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060" target="_blank">21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060</a></div>
<div>CSeq: 102 INVITE</div>
<div>User-Agent: Asterisk PBX 11.13.1</div>
<div>Date: Wed, 29 Oct 2014 10:04:22 GMT</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE</div>
<div>Supported: replaces, timer</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 676</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div> BYE Headers Browser --> Kamailio:<br>
</div>
<div><br>
</div>
<div>
<div>BYE <a>sip:anonymous@anonymous.invalid</a> SIP/2.0</div>
<div>Via: SIP/2.0/UDP
104.155.14.169:5064;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0</div>
<div> , SIP/2.0/UDP
10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK2d4f70db</div>
<div>From: <<a href="mailto:sip%3Ae3W5LffMkg0PZjP3SIGf6@wh2.24dial.com" target="_blank">sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com</a>>;tag=2TQ878KMAVLA43TXVZHNAWCWVKU6BLPBURF3</div>
<div>To: "Anonymous"
<a><sip:anonymous@anonymous.invalid></a>;tag=as32dd9e24</div>
<div>Call-ID: <a href="http://21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060" target="_blank">21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060</a></div>
<div>CSeq: 0 BYE</div>
<div>Record-Route:
<a><sip:104.155.11.255:5060;nat=yes;lr=on></a></div>
<div>Reason: Q.850;cause=16</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>BYE Response Kamailio --> Browser:<br>
</div>
<div><br>
</div>
<div>
<div>SIP/2.0 404 Not here</div>
<div>Via: SIP/2.0/UDP
104.155.14.169:5064;branch=z9hG4bK65df.6fd637d055286a45aa6f3e12c5ac873c.0</div>
<div> , SIP/2.0/UDP
10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK51c69f33;received=80.255.2.37</div>
<div>From: <<a href="mailto:sip%3Ae3W5LffMkg0PZjP3SIGf6@wh2.24dial.com" target="_blank">sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com</a>>;tag=47J6E76F5EVB583A682FQR799J6XDSU46MW8</div>
<div>To: "Anonymous"
<a><sip:anonymous@anonymous.invalid></a>;tag=as0b12a022</div>
<div>Call-ID: <a href="http://683def08655393622c55299c1a7c92d5@10.240.177.13:5060" target="_blank">683def08655393622c55299c1a7c92d5@10.240.177.13:5060</a></div>
<div>CSeq: 0 BYE</div>
<div>Server: kamailio (4.2.0 (x86_64/linux))</div>
<div>Content-Length: 0</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>In the logs I can see the following messages:</div>
<div>[loose.c:113]: find_first_route(): No Route headers found<br>
</div>
<div>[loose.c:929]: loose_route(): There is no Route HF<br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>Does anybody know, why Kamailio may respond with 404 Not
here? Do I have to send the BYE request directly to Asterisk?</div>
<div><br>
</div>
<div>Kind regards,</div>
<div>Marko</div>
</div>
<br>
<fieldset></fieldset>
<br>
</div></div><pre>_______________________________________________
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<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a></pre>
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