Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

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 &lt;568691(a)gmail.com&gt;om>: > 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 &lt;miconda(a)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(a)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/