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 568691@gmail.com:
Well.. Guys, sorry, it was totally my fault. I just used VPN.
2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568691@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 568691@gmail.com:
Here is it http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla miconda@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 568691@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 568691@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 < miconda@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@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-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@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/