Asterisk localip=10.0.0.87, sorry 2015-06-24 16:24 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> Ok, so my scheme. > Kamailio and Asterisk are in Amazon EC2 > Kamailio externip=54.197.230.121 localip=10.145.45.103 > Asterisk localip=10.145.45.103, externip doesn't matter > > Call should flow like that: > webrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localip > but now it's webrtc --> kamailio-externip --> kamailio--localip --> > asterisk-localip --> kamailio-externip --> peer > > I have the voice, but it's wrong scheme, and Asterisk drops call because > of retransmissions failure > > > 2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla <mico...@gmail.com>: > >> 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>: >> >>> 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>: >>> >>>> 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>: >>>> >>>>> Here is it >>>>> http://pastebin.com/JkkM4M5m >>>>> >>>>> 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla <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>: >>>>>> >>>>>>> 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/ >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> 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/ >>> >> >> >> >> -- >> Alexandru Covalschi >> ABRISS-Solutions >> VoIP engineer and system administrator >> phone: +37367398493 >> web: http://abs-telecom.com/ >> >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - >> http://www.linkedin.com/in/miconda >> Book: SIP Routing With Kamailio - http://www.asipto.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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