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

[Alexandru Covalschi]

without fix_nated_contact error behaviour is the same
maybe I should upgrade to 4.3 ?

2015-06-23 14:08 GMT+03:00 Alexandru Covalschi <568691 at 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 at 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 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
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at 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/
>



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