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@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
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