Also, an interesting thing - if you can see in Kamailio log, a check of the
proto of user "300" is being made. But 300 is $tU, and $tU proto is being
checked only if source IP is asterisks IP.
Here's the part of config where rtpengine is engaged (in NATmanage route)
if((src_ip==10.0.0.87))
{
xlog("L_NOTICE","====== select proto from sipusers where
name=$tU");
sql_xquery("ca_asterisk", "select proto from sipusers where
name=$tU", "ra");
xlog("L_NOTICE","===== $tU has proto $xavp(ra=>proto)");
if ($xavp(ra=>proto)=="ws")
{
xlog("L_NOTICE","===== $tU has WEBSOCKETS");
rtpengine_manage("trust-address replace-origin
replace-session-connection ICE=force RTP/SAVPF");
}
else
{
xlog("L_NOTICE","===== $tU has NO fucken WEBSOCKETS");
rtpengine_manage("trust-address replace-origin
replace-session-connection");
}
} else {
xlog("L_NOTICE","====== select proto from sipusers where
name=$fU");
sql_xquery("ca_asterisk", "select proto from sipusers where
name=$fU", "ra");
if ($xavp(ra=>proto)=="ws")
{
xlog("L_NOTICE","===== $fU has WEBSOCKETS");
rtpengine_manage("trust-address replace-origin
replace-session-connection ICE=force RTP/AVP");
}
else
{
xlog("L_NOTICE","===== $fU has NO WEBSOCKETS");
rtpengine_manage("replace-origin
replace-session-connection RTP/AVP");
}
}
2015-06-24 16:14 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> 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 <568...@gmail.com>:
>
>> Well.. Guys, sorry, it was totally my fault. I just used VPN.
>>
>> 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568...@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 <568...@gmail.com>:
>>>
>>>> Here is it
>>>> http://pastebin.com/JkkM4M5m
>>>>
>>>> 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla <mico...@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 <568...@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 <568...@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 <
>>>>>>> mico...@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-us...@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@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-us...@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@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/
>
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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