[asterisk-users] Jitter Buffer and MeetMe.

[David Backeberg]

On Fri, Apr 23, 2010 at 4:34 PM, russian qwerty
<russian.qwerty at gmail.com> wrote:
> Hello.
>
> As I see, there is 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.