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

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/

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 &lt;568691(a)gmail.com&gt;om>: > Here is it > http://pastebin.com/JkkM4M5m > > 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla &lt;miconda(a)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 &lt;568691(a)gmail.com&gt;om>: >> >>> 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 < >>>> miconda(a)gmail.com&gt;gt;: >>>> >>>>> 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/ >>> >> >> >> >> -- >> 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/