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

[Alexandru Covalschi]

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>:

>  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 at 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
>
>
> _______________________________________________
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> sr-users at lists.sip-router.org
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>
>


-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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