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

Alexandru Covalschi 568691 at gmail.com
Wed Jun 24 15:25:39 CEST 2015


Asterisk localip=10.0.0.87, sorry

2015-06-24 16:24 GMT+03:00 Alexandru Covalschi <568691 at gmail.com>:

> Ok, so my scheme.
> Kamailio and Asterisk are in Amazon EC2
> Kamailio externip=54.197.230.121 localip=10.145.45.103
> Asterisk localip=10.145.45.103, externip doesn't matter
>
> Call should flow like that:
> webrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localip
> but now it's webrtc --> kamailio-externip --> kamailio--localip -->
> asterisk-localip --> kamailio-externip --> peer
>
> I have the voice, but it's wrong scheme, and Asterisk drops call because
> of retransmissions failure
>
>
> 2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla <miconda at gmail.com>:
>
>>  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>:
>>
>>> 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/
>>
>>
>> --
>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Book: SIP Routing With Kamailio - http://www.asipto.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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