<div dir="ltr"><div>without fix_nated_contact error behaviour is the same<br></div>maybe I should upgrade to 4.3 ?<br></div><div class="gmail_extra"><br><div class="gmail_quote">2015-06-23 14:08 GMT+03:00 Alexandru Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@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">Here's the trace on port which I use for ws server. Don't look at fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't establish a ws connection properly. Client is SIPML5 demo phone<br><a href="http://pastebin.com/LvAk2HkP" target="_blank">http://pastebin.com/LvAk2HkP</a><br></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">2015-06-23 14:03 GMT+03:00 Alexandru Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@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">I solved the SIP voice trouble, but WebRTC problem still exists. What kind of trace I must do to make my post more informative?<br></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">2015-06-23 10:46 GMT+03:00 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,<span><br>
<br>
<div>On 23/06/15 04:10, Alexandru Covalschi
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>Hello. I'm trying to set up this (v 4.2 stable):<br>
</div>
peer <--> ec2 <--kamailio+rtpengine--> asterisk<br clear="all">
<div>
<div>
<div>scheme<br>
<br>
</div>
<div>I use advertised adress for SIP and WS connections.<br>
</div>
<div>The problem is that on SIP I get one way audio - I can
receive audio from asterisk, but I can't transmit audio
there - my SIP UA tries to send data to Kamailio-s local
EC2 IP.</div>
</div>
</div>
</div>
</blockquote>
<br></span>
you should grab a ngrep trace on server to see what happens in the
signaling in order to be able to provide some hints on solving it.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<blockquote type="cite"><span>
<div dir="ltr">
<div>
<div>
<div> In case of WebRTC I get lot's of erros:<br>
<br>
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]:
WARNING: <core> [msg_translator.c:2778]:
via_builder(): TCP/TLS connection (id: 0) for WebSocket
could not be found<br>
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR:
<core> [msg_translator.c:1996]:
build_req_buf_from_sip_req(): could not create Via header<br>
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR:
<core> [forward.c:584]: forward_request(): building
failed<br>
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR:
sl [sl_funcs.c:387]: sl_reply_error(): ERROR:
sl_reply_error used: I'm terribly sorry, server error
occurred (1/SL)<br>
<br>
</div>
<div>The call reaches Asterisk, but not vice-versa. No media
is being transferred.<br>
<br>
</div>
<div>Rtpengine flags I use:<br>
</div>
<div>For SIP: rtpengine_manage("trust-adress replace-origin
replace-session-connection RTP/AVP");<br>
</div>
<div>For WS: rtpengine_manage("trust-address replace-origin
replace-session-connection ICE=force RTP/AVP");<br>
<br>
</div>
<div>Do you have any ideas how ti fix that? I also make
REGFWD's to Asterisk<br>
</div>
<div>-- <br>
<div>
<div dir="ltr">Alexandru Covalschi<br>
ABRISS-Solutions
<div>VoIP engineer and system administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
</div>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
</span><pre>_______________________________________________
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</font></span></pre><span><font color="#888888">
</font></span></blockquote><span><font color="#888888">
<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>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
</font></span></div>
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<br></blockquote></div><br><br clear="all"><br>-- <br><div><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
</div>
</div></div></blockquote></div><br><br clear="all"><br>-- <br><div><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
</div>
</div></div></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature"><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: +37367398493<br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
</div>