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/
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