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

Wed Jun 24 15:14:18 CEST 2015

Heh...
Well, I still have troubles with my configuration. And in SDP media adress
is Amazon public interface - but rtpengine has replace-origin
replace-session-connection session, so it must be local address.
Any ideas?
Asterisk log http://pastebin.com/MFt9V9qK
Kamailio log http://pastebin.com/jZceP2Rn
Javascript log http://pastebin.com/4ZLePyKz

2015-06-24 1:27 GMT+03:00 Alexandru Covalschi <568691 at gmail.com>:

> 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/
>

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