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You have to do nat traversal logic for signaling -- see default
config file for set_contact_alias() and handle_uri_alias().<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div class="moz-cite-prefix">On 07/08/14 21:16, Manuel Camarg wrote:<br>
</div>
<blockquote
cite="mid:CAMjEs7T-8T3+4U2WFJT0RNd6NOL7DUbQLsRw1=YDse_+Bs6ZHA@mail.gmail.com"
type="cite">
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<div>I have this config:</div>
<div>WebRTC sipml5 -> Kamailio -> Asterisk -> Kamailio
-> WebRTC sipml5</div>
<div><br>
</div>
<div>When placing a communication between two sipml5 points I
get this errors:</div>
<div><br>
</div>
<div>ERROR: <core> [resolve.c:1733]: sip_hostport2su():
ERROR: sip_hostport2su: could not resolve hostname:
"df7jal23ls0d.invalid"</div>
<div>ERROR: <core> [forward.c:532]: forward_request():
ERROR: forward_request: bad host name df7jal23ls0d.invalid,
dropping packet</div>
<div>ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR:
sl_reply_error used: Unresolvable destination (478/SL)<br>
</div>
<div><br>
</div>
<div>If I place a comm between WebRTC sipml5 -> Kamailio
-> Asterisk -> Softphone (connected to Asterisk)</div>
<div>Everything works fine</div>
<div><br>
</div>
<div>I'm running RTPEngine as RTP Proxy for DTLS/SRTP issues</div>
<div><br>
</div>
<div>SIP INVITE signalling Ringing 180 message and 200 OK gets
done fine</div>
<div><br>
</div>
<div>Any ideas where this issue might come from? Looks something
related with media traffic (RTP).</div>
<div><br>
</div>
<div>Kind regards!</div>
</div>
<br>
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</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
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Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA</pre>
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