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

Wed Jun 24 00:27:50 CEST 2015

Well.. Guys, sorry, it was totally my fault. I just used VPN.

2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568691 at gmail.com>:

> 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/
>

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