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 <568691@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 <568691@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@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?



2015-06-23 20:40 GMT+03:00 Alexandru Covalschi <568691@gmail.com>:

2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla <miconda@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@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@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@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 list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla
http://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 list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla
http://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/