Here is it http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla <mico...@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 <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>: >> >>> 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>: >>> >>>> 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-us...@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-us...@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/
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