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 <568...@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 <mico...@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-us...@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
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@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
> web: http://abs-telecom.com/
>
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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