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/
>
>
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--
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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