[SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)
Daniel-Constantin Mierla
miconda at gmail.com
Wed Jun 24 15:30:52 CEST 2015
Do you have proper routing rules between the local ips of kamailio and
asterisk? Why aren't you use only external IPs if they are on different
servers? Asterisk has also the option to set external ip. It can reduce
the complexity of doing bridging of signaling and rtp. Once you get that
working you can start adding bridging step by step.
Cheers,
Daniel
On 24/06/15 15:25, Alexandru Covalschi wrote:
> Asterisk localip=10.0.0.87, sorry
>
> 2015-06-24 16:24 GMT+03:00 Alexandru Covalschi <568691 at gmail.com
> <mailto: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 <mailto: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 <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 <tel:%2B37367398493>
>> web: http://abs-telecom.com/
>
> --
> 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
>
>
>
>
> --
> 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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