Hello.<br><br>As I see, there is a lot of threads about jitter buffer... Maybe anybody knows something about my case? Any help will be appreciate.<br><br>Thanks in advance.<br><br><div class="gmail_quote">---------- Original message ----------<br>
From: <b class="gmail_sendername">russian qwerty</b> <span dir="ltr"><<a href="mailto:russian.qwerty@gmail.com">russian.qwerty@gmail.com</a>></span><br>Date: 2010/3/31<br>Subject: Jitter Buffer and MeetMe.<br>To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
<br>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/" target="_blank">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>
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