On Fri, Apr 23, 2010 at 4:34 PM, russian qwerty <russian.qwe...@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. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
