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 <568...@gmail.com > <mailto:568...@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 <568...@gmail.com > <mailto:568...@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 > <mico...@gmail.com <mailto:mico...@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 <tel:%2B37367398493> >> web: http://abs-telecom.com/ >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> sr-users@lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla > http://twitter.com/#!/miconda <http://twitter.com/#%21/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 > <mailto: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 <tel:%2B37367398493> > web: http://abs-telecom.com/ > > > > > -- > Alexandru Covalschi > ABRISS-Solutions > VoIP engineer and system administrator > phone: +37367398493 <tel:%2B37367398493> > 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 list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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