I have the voice, but it's wrong scheme, and Asterisk drops call because of retransmissions failurebut now it's webrtc --> kamailio-externip --> kamailio--localip --> asterisk-localip --> kamailio-externip --> peerwebrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localipCall should flow like that:Asterisk localip=10.145.45.103, externip doesn't matterKamailio externip=54.197.230.121 localip=10.145.45.103Ok, so my scheme.Kamailio and Asterisk are in Amazon EC22015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla <miconda@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@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?
Kamailio log: http://pastebin.com/jZceP2Rn
javascript log: http://pastebin.com/9Y4Pv43W
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 samemaybe 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
_______________________________________________ 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
--
Alexandru Covalschi
ABRISS-Solutions
--
Alexandru Covalschi
ABRISS-Solutions
_______________________________________________ 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
--
Alexandru Covalschi
ABRISS-Solutions
--
Alexandru Covalschi
ABRISS-Solutions
--
Alexandru Covalschi
ABRISS-Solutions
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
--Alexandru Covalschi
ABRISS-Solutions