<div dir="ltr"><div>Also, an interesting thing - if you can see in Kamailio log, a check of the proto of user "300" is being made. But 300 is $tU, and $tU proto is being checked only if source IP is asterisks IP. <br><br></div>Here's the part of config where rtpengine is engaged (in NATmanage route)<br><br> if((src_ip==10.0.0.87))<br> {<br> xlog("L_NOTICE","====== select proto from sipusers where name=$tU");<br> sql_xquery("ca_asterisk", "select proto from sipusers where name=$tU", "ra");<br> xlog("L_NOTICE","===== $tU has proto $xavp(ra=>proto)");<br> if ($xavp(ra=>proto)=="ws")<br> {<br> xlog("L_NOTICE","===== $tU has WEBSOCKETS");<br><br> rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF");<br> }<br> else<br> {<br> xlog("L_NOTICE","===== $tU has NO fucken WEBSOCKETS");<br> rtpengine_manage("trust-address replace-origin replace-session-connection");<br> }<br> } else {<br> xlog("L_NOTICE","====== select proto from sipusers where name=$fU");<br> sql_xquery("ca_asterisk", "select proto from sipusers where name=$fU", "ra");<br> if ($xavp(ra=>proto)=="ws")<br> {<br><br> xlog("L_NOTICE","===== $fU has WEBSOCKETS");<br> rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/AVP");<br> }<br> else<br> {<br> xlog("L_NOTICE","===== $fU has NO WEBSOCKETS");<br> rtpengine_manage("replace-origin replace-session-connection RTP/AVP");<br> }<br><br> }<br><br></div><div class="gmail_extra"><br><div class="gmail_quote">2015-06-24 16:14 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"><div><div>Heh... <br></div>Well, I still have troubles with my configuration. And in SDP media adress is Amazon public interface - but rtpengine has replace-origin replace-session-connection session, so it must be local address. <br></div><div>Any ideas?<br></div><div>Asterisk log <a href="http://pastebin.com/MFt9V9qK" target="_blank">http://pastebin.com/MFt9V9qK</a><br></div><div>Kamailio log <a href="http://pastebin.com/jZceP2Rn" target="_blank">http://pastebin.com/jZceP2Rn</a><br></div><div>Javascript log <a href="http://pastebin.com/4ZLePyKz" target="_blank">http://pastebin.com/4ZLePyKz</a><br></div><div><br></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">2015-06-24 1:27 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">Well.. Guys, sorry, it was totally my fault. I just used VPN. </div><div><div><div class="gmail_extra"><br><div class="gmail_quote">2015-06-24 0:59 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"><div>I used <a href="https://github.com/caruizdiaz/kamailio-ws" target="_blank">https://github.com/caruizdiaz/kamailio-ws</a> configuration that 100% works on other then Amazon EC2 environment and I still get this error. Maybe it is somehow related to NAT traversal?<br></div><div class="gmail_extra"><br></div><div class="gmail_extra">Kamailio log: <a href="http://pastebin.com/jZceP2Rn" target="_blank">http://pastebin.com/jZceP2Rn</a><br></div><div class="gmail_extra">javascript log: <a href="http://pastebin.com/9Y4Pv43W" target="_blank">http://pastebin.com/9Y4Pv43W</a><br></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">2015-06-23 20:40 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:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">Here is it<br><a href="http://pastebin.com/JkkM4M5m" target="_blank">http://pastebin.com/JkkM4M5m</a><br></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">2015-06-23 18:53 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:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
There are no major changes in 4.3 comparing with 4.2 in regards to
websocket -- the implementation is quite mature for a long time.<br>
<br>
Looks like websocket connection is not available. Can you look at
javascript debug console in the browser to see what is printing?<span><font color="#888888"><br>
<br>
Daniel</font></span><div><div><br>
<br>
<div>On 23/06/15 17:23, Alexandru Covalschi
wrote:<br>
</div>
<blockquote type="cite">
<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:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);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>
<div>
<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:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);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:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);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>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><span><font color="#888888">
</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/#%21/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>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio
(OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<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>
<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>
<br>
<fieldset></fieldset>
<br>
<pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
</blockquote>
<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>
</div></div></div>
<br>_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<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></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><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>