[SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Daniel-Constantin Mierla]

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 at gmail.com
> <mailto:568691 at 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 at gmail.com
>     <mailto:568691 at 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 at gmail.com <mailto:miconda at 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 <tel:%2B37367398493>
>>             web: http://abs-telecom.com/
>>
>>
>>             _______________________________________________
>>             SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>             sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>>             http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>             -- 
>             Daniel-Constantin Mierla
>             http://twitter.com/#!/miconda <http://twitter.com/#%21/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 at lists.sip-router.org
>             <mailto:sr-users at 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 <tel:%2B37367398493>
>         web: http://abs-telecom.com/
>
>
>
>
>     -- 
>     Alexandru Covalschi
>     ABRISS-Solutions
>     VoIP engineer and system administrator
>     phone: +37367398493 <tel:%2B37367398493>
>     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 list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com

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