Hello, David.<div><br></div><div>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).</div><div><br></div><div>BTW:</div><div>
1. The file /etc/asterisk/dsp.conf doesn't exist in my Asterisk 1.6.0.6 (something wrong?).</div><div>2. VAD is already disable for all trunks.</div><div>3. And 'talker optimization' future is already disable by patch for app_meetme.c.</div>
<div><br></div><div>2010/4/25 David Backeberg <span dir="ltr"><<a href="mailto:dbackeberg@gmail.com">dbackeberg@gmail.com</a>></span></div><div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class="im">On Fri, Apr 23, 2010 at 4:34 PM, russian qwerty<br>
<<a href="mailto:russian.qwerty@gmail.com">russian.qwerty@gmail.com</a>> wrote:<br>
> Hello.<br>
><br>
> As I see, there are a lot of threads about jitter buffer... Maybe anybody<br>
> knows something about my case? Any help will be appreciate.<br>
><br>
</div><div class="im">> So, the problem with voice quality was completely solved, BUT some customers<br>
> have informed me about big latency. It's really hard to make dialogue with<br>
> current latency.<br>
<br>
</div>You're on the right track here, but I don't think your problem is<br>
jitter. I think your problem is VoIP and voice activity detection, and<br>
depending on your version of asterisk, MeetMe conference 'talker<br>
optimization'.<br>
<br>
I've posted all of this before. Here goes again...<br>
<br>
* 'talker optimization' should be disabled on MeetMe() conferences.<br>
* /etc/asterisk/dsp.conf set silencethreshold=1024<br>
* /etc/asterisk/codecs.conf set vad=>false<br>
<br>
Give those a try, restart or reload asterisk to apply changes, and<br>
tell us if it fixes it.<br>
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