24 Jun
2015
24 Jun
'15
9:18 a.m.
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(a)gmail.com
<mailto: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(a)gmail.com
<mailto: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(a)gmail.com <mailto:568691@gmail.com>>:
Here is it
http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto: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:
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
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(a)gmail.com <mailto: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(a)gmail.com <mailto: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(a)gmail.com
<mailto:miconda@gmail.com>>:
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
--
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(a)lists.sip-router.org
<mailto: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 <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
web: http://abs-telecom.com/ 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(a)lists.sip-router.org
<mailto: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://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(a)lists.sip-router.org
<mailto: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 <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(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users