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

[Alexandru Covalschi]

Here is it
http://pastebin.com/JkkM4M5m

2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla <miconda at gmail.com>:

>  There are no major changes in 4.3 comparing with 4.2 in regards to
> websocket -- the implementation is quite mature for a long time.
>
> Looks like websocket connection is not available. Can you look at
> javascript debug console in the browser to see what is printing?
>
> Daniel
>
>
> On 23/06/15 17:23, Alexandru Covalschi wrote:
>
>  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/
>
>
> _______________________________________________
> 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/
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