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/
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