23 Jun
2015
23 Jun
'15
11:53 a.m.
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(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://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.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/
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--
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
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<mailto:sr-users@lists.sip-router.org>
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--
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/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users