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Problems with audio delay using VoIP #524

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago 
What steps will reproduce the problem?
1.Install BBB on Ubuntu 9.04 and 10.04, on XEN virtual machine and hardware 
server. 
2.Go to conference, enable audio
3.in first time others hears my voice a little delay 1-2sec. But a delay of 
up to 5 minutes for 10 seconds, 10 minutes - 20 .. this is very bad.

What version of the product are you using? On what operating system?
BigBlueButton Version 0.64, Ubuntu 9.04, Ubuntu 10.04

Original issue reported on code.google.com by polum...@gmail.com on 11 May 2010 at 12:09

GoogleCodeExporter commented 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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 

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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago

Original comment by ffdixon@gmail.com on 14 Jul 2010 at 11:24

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
Issue 587 has been merged into this issue.

Original comment by Me.S...@gmail.com on 26 Jul 2010 at 3:11

GoogleCodeExporter commented 9 years ago

Original comment by ffdixon@gmail.com on 9 Aug 2010 at 11:31

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
Is it possible to have trunck repository to test it ?

Original comment by qdqdsq...@gmail.com on 8 Sep 2010 at 11:20

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
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

GoogleCodeExporter commented 9 years ago 
Fixed....on going testing in demo.bigbluebutton.org

Original comment by ritza...@gmail.com on 17 Oct 2010 at 4:52

GoogleCodeExporter commented 9 years ago 
Issue 676 has been merged into this issue.

Original comment by ffdixon@gmail.com on 24 Oct 2010 at 12:21

GoogleCodeExporter commented 9 years ago 
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