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/
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