Hello,
SIP Transport Does Not come in room. I am able to register from
linphone.but when i go from pc. SIP Transport doesn't come into room.
Please Help :(
--
*Gaganjot Singh*
*(+91-9899025098)*
*B.Tech - Electronics Communication*
*Guru Tegh Bahadur Institute of Technology / GGSIPU*
it activated SIP Room No. too i.e. 40013
On Wed, Aug 21, 2013 at 3:19 PM, Gagan sethi gaganjotsse...@gmail.comwrote:
yes it's enabled.
On Wed, Aug 21, 2013 at 3:18 PM, Vasiliy Degtyarev va...@unipro.ruwrote:
Do you check the Enable SIP transport in the room checkbox in the
http://openmeetings.apache.org/red5sip-integration_2.1.html
I Use this for installation.
On Wed, Aug 21, 2013 at 3:32 PM, Vasiliy Degtyarev va...@unipro.ru wrote:
Do you install asterisk integration like this
https://cwiki.apache.org/**confluence/display/**OPENMEETINGS/OpenMeetings+
12 Aug 18:48:04 - [TRACE] o.r.s.a.RTPStreamMultiplexingSender: Sleep pause:
14 decrementTime: 0
12 Aug 18:48:04 - [TRACE] o.r.s.a.RTPStreamReceiver: Sleep pause: 14
12 Aug 18:48:04 - [TRACE] o.r.s.a.RTPStreamReceiver: Sleep pause: 14
12 Aug 18:48:04 - [TRACE] o.r.s.a.RTPStreamMultiplexingSender:
There is no log for current date.
On Wed, Aug 21, 2013 at 4:37 PM, Gagan sethi gaganjotsse...@gmail.comwrote:
12 Aug 18:48:04 - [TRACE] o.r.s.a.RTPStreamMultiplexingSender: Sleep
pause: 14 decrementTime: 0
12 Aug 18:48:04 - [TRACE] o.r.s.a.RTPStreamReceiver: Sleep pause: 14
12 Aug 18:48:04
Please check that red5sip service is started.
On 21.08.2013 18:15, Gagan sethi wrote:
There is no log for current date.
On Wed, Aug 21, 2013 at 4:37 PM, Gagan sethi gaganjotsse...@gmail.comwrote:
12 Aug 18:48:04 - [TRACE] o.r.s.a.RTPStreamMultiplexingSender: Sleep
pause: 14 decrementTime:
sudo service --status-all
[ + ] red5sip
it means red5sip is started
On Wed, Aug 21, 2013 at 5:25 PM, Vasiliy Degtyarev va...@unipro.ru wrote:
Please check that red5sip service is started.
On 21.08.2013 18:15, Gagan sethi wrote:
There is no log for current date.
On Wed, Aug 21, 2013
Hi,
I have seen there has been a similar problem on july so I investigated some
time in my configs but still did not found a solution.
*Background:*
I have installed OM + Asterisk in a Amazon EC2 machine for some tests.
After some time I now got a connection to asterisk and OM. So I am able to
Hello folks,
how difficult (if technically possible) would be the following:
1) understand that video / sound of some participant experiences delays;
2) stop video for that participant;
in other words, switch off the video for those who does not have enough channel?
Maybe something like
At the risk of creating an unnecessary diversion, have you considered
using an adaptive streaming protocol. I'm involved with dash.js a
reference implementation of MPEG-DASH. I'd love to see it used here in
open meetings. However that is a client, not a server. I'm not sure if
there are any open
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