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 $tU proto is being checked only if source IP is asterisks IP.
Here's the part of config where rtpengine is engaged (in NATmanage route)
if((src_ip==10.0.0.87)) { xlog("L_NOTICE","====== select proto from sipusers where name=$tU"); sql_xquery("ca_asterisk", "select proto from sipusers where name=$tU", "ra"); xlog("L_NOTICE","===== $tU has proto $xavp(ra=>proto)"); if ($xavp(ra=>proto)=="ws") { xlog("L_NOTICE","===== $tU has WEBSOCKETS");
rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); } else { xlog("L_NOTICE","===== $tU has NO fucken WEBSOCKETS"); rtpengine_manage("trust-address replace-origin replace-session-connection"); } } else { xlog("L_NOTICE","====== select proto from sipusers where name=$fU"); sql_xquery("ca_asterisk", "select proto from sipusers where name=$fU", "ra"); if ($xavp(ra=>proto)=="ws") {
xlog("L_NOTICE","===== $fU has WEBSOCKETS"); rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); } else { xlog("L_NOTICE","===== $fU has NO WEBSOCKETS"); rtpengine_manage("replace-origin replace-session-connection RTP/AVP"); }
}
2015-06-24 16:14 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Heh... 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. Any ideas? Asterisk log http://pastebin.com/MFt9V9qK Kamailio log http://pastebin.com/jZceP2Rn Javascript log http://pastebin.com/4ZLePyKz
2015-06-24 1:27 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Well.. Guys, sorry, it was totally my fault. I just used VPN.
2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
I used https://github.com/caruizdiaz/kamailio-ws configuration that 100% works on other then Amazon EC2 environment and I still get this error. Maybe it is somehow related to NAT traversal?
Kamailio log: http://pastebin.com/jZceP2Rn javascript log: http://pastebin.com/9Y4Pv43W
2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Here is it http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla miconda@gmail.com :
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.
Looks like websocket connection is not available. Can you look at javascript debug console in the browser to see what is printing?
Daniel
On 23/06/15 17:23, Alexandru Covalschi wrote:
without fix_nated_contact error behaviour is the same maybe I should upgrade to 4.3 ?
2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
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 http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
> I solved the SIP voice trouble, but WebRTC problem still exists. > What kind of trace I must do to make my post more informative? > > 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla < > miconda@gmail.com>: > >> Hello, >> >> On 23/06/15 04:10, Alexandru Covalschi wrote: >> >> Hello. I'm trying to set up this (v 4.2 stable): >> peer <--> ec2 <--kamailio+rtpengine--> asterisk >> scheme >> >> I use advertised adress for SIP and WS connections. >> 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. >> >> >> 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. >> >> Cheers, >> Daniel >> >> In case of WebRTC I get lot's of erros: >> >> 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 >> 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 >> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> >> [forward.c:584]: forward_request(): building failed >> 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) >> >> The call reaches Asterisk, but not vice-versa. No media is being >> transferred. >> >> Rtpengine flags I use: >> For SIP: rtpengine_manage("trust-adress replace-origin >> replace-session-connection RTP/AVP"); >> For WS: rtpengine_manage("trust-address replace-origin >> replace-session-connection ICE=force RTP/AVP"); >> >> Do you have any ideas how ti fix that? I also make REGFWD's to >> Asterisk >> -- >> Alexandru Covalschi >> ABRISS-Solutions >> VoIP engineer and system administrator >> phone: +37367398493 >> web: http://abs-telecom.com/ >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Book: SIP Routing With Kamailio - http://www.asipto.com >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > > -- > Alexandru Covalschi > ABRISS-Solutions > VoIP engineer and system administrator > phone: +37367398493 > web: http://abs-telecom.com/ >
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/