Closed GoogleCodeExporter closed 9 years ago
Hi polumish,
The mixing of audio is sensitive to CPU usage. Even though we run
demo.bigbluebutton.org on a virtualized machine (KVM), you'll get better
performance
when installing BigBlueButton natively on a host.
How much memory are you giving the XEN instance?
Original comment by ffdixon@gmail.com
on 11 May 2010 at 1:35
On XEN virtual host I have 1.5 Gb memory, and 2 core 2.2 CPU.
On native host I have 1 Gb memory and 2 core 2.6 CPU.
On both servers BBB the same problem.
Original comment by polum...@gmail.com
on 11 May 2010 at 3:43
Several people have complained of this in 0.64, and it doesn't seem it was a
problem in
0.63. It would be best to try to reproduce and investigate so we don't release
0.7 with
the same bug
Original comment by Me.S...@gmail.com
on 20 May 2010 at 6:59
I can confirm this.
Native install (non virtual)
2gb ram
CPU0: Intel(R) Pentium(R) 4 CPU 2.80GHz stepping 09
ubuntu 9.04 fresh install,
add repo and install via apt-get
Original comment by 808blog...@gmail.com
on 7 Jun 2010 at 9:13
Does any one working on this issue. Because I think the problem become from the
Red5 0.91. I try to upgrade with the RED5 trunck 1.0.0, but a can connnect to
asterisk because I have an RTMP handshake problem. I will work on this problem.
Original comment by qdqdsq...@gmail.com
on 3 Jul 2010 at 7:37
It's the same on the demo server.
It's impossible to use conference with this issue.
It the same with RED5 trunck 1.0.0. May be it came from red5 phone or JAVA !
Original comment by qdqdsq...@gmail.com
on 7 Jul 2010 at 4:23
Hi jproussandies,
High-quality VoIP conferencing is a tough nut to crack. TCP/IP does not
guarantee delivery of packets within a certain time. As well, the network
latency (i.e. a remote user's upload bandwidth and distance from the server)
plays a big factor.
We know this problem well. Here's some more background information along with
some of our suggestions/plans to improve it
http://code.google.com/p/bigbluebutton/wiki/FAQ#Why_is_there_a_delay_in_the_audio_when_I_use_VoIP?
Regards,... Fred
We have a pretty good idea where the
Original comment by ffdixon@gmail.com
on 7 Jul 2010 at 7:12
Thank, I read the FAQ, and the problem came from the transcode latency in red5
phone.
Flash Player 10 support SPEECH and ADPCM codec, who are natively supported in
Asterisk.
All the sip rtp to and from asterisk can be in SPEECH (I check and the
bbb-voice manage the SPEECH), also the FP 10 will talk in SPEECH.
The ubuntu 10.4, comme with asterisk 1.6.2 and a new function confbridge who
can mix no-linear codec like SPEECH.
If we need to do some voice transcode, it can be perform by Asterisk.
What do you think of my idea ?
Original comment by qdqdsq...@gmail.com
on 8 Jul 2010 at 10:43
Hi jproussanides,
Thanks for your suggestions. We are currently using a modified version of
app_konference for mixing voice. We're working on packages for Ubuntu 10.04,
which give us Asterisk 1.6.2.
We're aware of the support for speex in FP 10, but at the moment asterisk only
supports 8 khz speex. See
http://lists.digium.com/pipermail/asterisk-dev/2010-May/044147.html
We'll definitely be exploring using speex in a future iteration of
BigBlueButton.
Original comment by ffdixon@gmail.com
on 8 Jul 2010 at 11:13
We've been looking into this and we think the problem is related to
timestamping of incoming RTMP packets.
For the audio packages, if the BigBlueButton server gets loaded down (such as
with a slide conversion), the transcoding of some of the audio packages may
lag, causing the queue of incoming audio packets to build-up for a user.
When the red5 server does transcode the packets, they are now lagging behind
and the user's voice comes in as delayed.
One solution is to drop packets that are too old to catch up, rather than now
leave the user 2, 3, 4, or 5 seconds behind in the audio.
More investigation needed.
Original comment by ffdixon@gmail.com
on 14 Jul 2010 at 11:23
Original comment by ffdixon@gmail.com
on 14 Jul 2010 at 11:24
We are still investigating trying to narrow down where the problem is. We
notice that not all experience the delay at the same time.
A workaround is to the user who is experiencing the delay to leave and join the
voice conference (i.e. clicking on the headset icon).
Original comment by ritza...@gmail.com
on 15 Jul 2010 at 3:42
Issue 587 has been merged into this issue.
Original comment by Me.S...@gmail.com
on 26 Jul 2010 at 3:11
Original comment by ffdixon@gmail.com
on 9 Aug 2010 at 11:31
Experiencing similar issues myself. Audio is delayed by about 1 second (1 way
only, inbound Audio).
How can you get the incorrectly stamped packets to be dropped? I would either
like them to be dropped or un-queued.
Original comment by colinjam...@gmail.com
on 31 Aug 2010 at 1:44
We're looking at the time stamping of the audio packets as one of the ways to
improve the audio. See
http://groups.google.com/group/bigbluebutton-dev/browse_thread/thread/6ffa48133ea7d2e6#
Original comment by ffdixon@gmail.com
on 31 Aug 2010 at 9:41
Some here on Ubuntu 10.04 LTS/BBB 0.7 on Xen and on physical
Given the current implementation of the VoIP in BBB, do you think is possible
to just drop the late-come-packets and resync? What if I want to help implement
such a workaround?
Original comment by davide.v...@gmail.com
on 8 Sep 2010 at 9:23
I checked out BBB on github and saw a 14 hours old "throw away delayed rtp
packets" commit on bbb-voice! Great! You guys rock! I'll test right now.
Original comment by davide.v...@gmail.com
on 8 Sep 2010 at 9:32
Thanks! Let us know what you find. We've been working hard on this and we
really want to solve it (as much as possible). The goal of BigBlueButton is to
offer remote students a high-quality learning experience, and that includes
high-quality audio.
Davide, if you are experienced with this type of programming, please contact us
directly.
Original comment by ffdixon@gmail.com
on 8 Sep 2010 at 11:15
Is it possible to have trunck repository to test it ?
Original comment by qdqdsq...@gmail.com
on 8 Sep 2010 at 11:20
At the moment, you would need to have setup a BigBlueButton development
environment
http://code.google.com/p/bigbluebutton/wiki/DevelopingBBB
And be working on the "bleeding edge"
http://code.google.com/p/bigbluebutton/wiki/BleedingEdge
Once we into our testing phase, we'll make an announcement to bigbluebutton-dev
and provide a test server everyone can try out. If your not a developer, best
wait until you see that announcement.
Original comment by ffdixon@gmail.com
on 8 Sep 2010 at 11:31
Asterisk now supports 16kHz Speex: https://issues.asterisk.org/view.php?id=17501
(The contradictory comment above was posted to the Asterisk-dev mailing list in
May; the patch was posted June 12.)
Original comment by rod.mont...@gmail.com
on 10 Sep 2010 at 8:55
rod.montgomery,
Yes, that issue was reported by a developer that we (GenericConf.com) paid to
add Speex support, specifically for BigBlueButton. Richard is aware of it and
using it to support WB Speex already for the 0.71 release. We were able to get
our WB Speex support into trunk for the upcoming Asterisk 1.8 release, but it
was not possible to add it to the core in 1.6 (because all of the available
codec slots were already full, meaning you had to remove support for other
codecs to add this one). Instead, we created a patch so that we could build a
custom version of Asterisk 1.6 with WB Speex. See
http://github.com/jthomerson/AsteriskAudioKonf/tree/master/asterisk-patches/
for more information
Original comment by jeremyth...@gmail.com
on 11 Sep 2010 at 4:00
Fixed....on going testing in demo.bigbluebutton.org
Original comment by ritza...@gmail.com
on 17 Oct 2010 at 4:52
Issue 676 has been merged into this issue.
Original comment by ffdixon@gmail.com
on 24 Oct 2010 at 12:21
Please can you just explain what the actual delay is. Because of the 2-3
seconds delay in the past. Is this also better / faster now
Original comment by rb.p...@gmail.com
on 2 Nov 2010 at 6:36
Original issue reported on code.google.com by
polum...@gmail.com
on 11 May 2010 at 12:09