[asterisk-users] Jitter Buffer and MeetMe.

russian qwerty russian.qwerty at gmail.com
Wed Mar 31 07:06:19 CDT 2010


Hello.

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 (
http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/)
I added following scheme in dialplan:

[some-context]
exten => 123,1,Dial(Local/124 at some-context/nj)
exten => 124,1,MeetMe(some-room,dM)

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.

And there are some questions:

1. Where can I find "the best practice" to solve the issue with JB and
applications (MeetMe)?
2. Is it possible to adjust (reduce) "generic JB" in chan_local and for
Local/.../nj construction?

BR, Alexey
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