2015-06-23 14:08 GMT+03:00 Alexandru Covalschi <568691@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@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@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/
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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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--

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/