Hello.<br><br>I'm having Asterisk 1.6.0.x and trying to solve the issue concerning with a bad quality of voice for incoming SIP calls into the app_meetme. As I know, in my case of calls, jitter buffer is NOT executed on anyone channel. So, after reading Russell Bryant's post (<a href="http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/">http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/</a>) I added following scheme in dialplan:<br>
<br>[some-context]<br>exten => 123,1,Dial(Local/124 at some-context/nj)<br>exten => 124,1,MeetMe(some-room,dM)<br><br>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.<br>
<br>And there are some questions:<br><br>1. Where can I find "the best practice" to solve the issue with JB and applications (MeetMe)?<br>2. Is it possible to adjust (reduce) "generic JB" in chan_local and for Local/.../nj construction?<br>
<br>BR, Alexey<br>