[SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)
Daniel-Constantin Mierla
miconda at gmail.com
Wed Jun 24 15:18:56 CEST 2015
Can you specify exactly which side received what IP and what you would
expect there? It is not easy to digests lots of logs and also guess what
would you expect to happen...
Cheers,
Daniel
On 24/06/15 15:14, Alexandru Covalschi wrote:
> 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
> <mailto: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
> <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
>> <mailto: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 <tel:%2B37367398493>
>>> web: http://abs-telecom.com/
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>> --
>> Daniel-Constantin Mierla
>> http://twitter.com/#!/miconda <http://twitter.com/#%21/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
>> <mailto: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 <tel:%2B37367398493>
>> web: http://abs-telecom.com/
>>
>>
>>
>>
>> --
>> Alexandru Covalschi
>> ABRISS-Solutions
>> VoIP engineer and system administrator
>> phone: +37367398493 <tel:%2B37367398493>
>> web: http://abs-telecom.com/
>>
>>
>>
>>
>> --
>> Alexandru Covalschi
>> ABRISS-Solutions
>> VoIP engineer and system administrator
>> phone: +37367398493 <tel:%2B37367398493>
>> web: http://abs-telecom.com/
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda <http://twitter.com/#%21/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
> <mailto: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 <tel:%2B37367398493>
> web: http://abs-telecom.com/
>
>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493 <tel:%2B37367398493>
> web: http://abs-telecom.com/
>
>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493 <tel:%2B37367398493>
> web: http://abs-telecom.com/
>
>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
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