[asterisk-users] Jitter Buffer and MeetMe.

[russian qwerty]

Hello, David.

Thank you for reply. But my problem is certainly in the size of JitterBuffer
of chan_local. I realy need to know how to change the size of JB (reduce).

BTW:
1. The file /etc/asterisk/dsp.conf doesn't exist in my Asterisk 1.6.0.6
(something wrong?).
2. VAD is already disable for all trunks.
3. And 'talker optimization' future is already disable by patch for
app_meetme.c.

2010/4/25 David Backeberg <dbackeberg at gmail.com>

> On Fri, Apr 23, 2010 at 4:34 PM, russian qwerty
> <russian.qwerty at gmail.com> wrote:
> > Hello.
> >
> > As I see, there are a lot of threads about jitter buffer... Maybe anybody
> > knows something about my case? Any help will be appreciate.
> >
> > So, the problem with voice quality was completely solved, BUT some
> customers
> > have informed me about big latency. It's really hard to make dialogue
> with
> > current latency.
>
> You're on the right track here, but I don't think your problem is
> jitter. I think your problem is VoIP and voice activity detection, and
> depending on your version of asterisk, MeetMe conference 'talker
> optimization'.
>
> I've posted all of this before. Here goes again...
>
> * 'talker optimization' should be disabled on MeetMe() conferences.
> * /etc/asterisk/dsp.conf set silencethreshold=1024
> * /etc/asterisk/codecs.conf set vad=>false
>
> Give those a try, restart or reload asterisk to apply changes, and
> tell us if it fixes it.
>
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