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

Tue Jun 23 23:59:51 CEST 2015

I used https://github.com/caruizdiaz/kamailio-ws configuration that 100%
works on other then Amazon EC2 environment and I still get this error.
Maybe it is somehow related to NAT traversal?

Kamailio log: http://pastebin.com/jZceP2Rn
javascript log: http://pastebin.com/9Y4Pv43W

2015-06-23 20:40 GMT+03:00 Alexandru Covalschi <568691 at gmail.com>:

> 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/
>

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