Can you specify exactly which side received what IP and what you would expect there? It is not easy to digests lots of logs and also guess what would you expect to happen...
Cheers, Daniel On 24/06/15 15:14, Alexandru Covalschi wrote: > 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 <568...@gmail.com > <mailto:568...@gmail.com>>: > > Well.. Guys, sorry, it was totally my fault. I just used VPN. > > 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568...@gmail.com > <mailto:568...@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 > <568...@gmail.com <mailto:568...@gmail.com>>: > > Here is it > http://pastebin.com/JkkM4M5m > > 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla > <mico...@gmail.com <mailto: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 <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 <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 <tel:%2B37367398493> > web: http://abs-telecom.com/ > > > > > -- > Alexandru Covalschi > ABRISS-Solutions > VoIP engineer and system administrator > phone: +37367398493 > web: http://abs-telecom.com/ -- 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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