On Sun, Apr 25, 2010 at 7:13 AM, russian qwerty
wrote:
> Hello, David.
> 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).
> BTW:
> 1. The file /etc/asterisk/dsp.conf doesn't exist in my Asteris
Hello, David.
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).
BTW:
1. The file /etc/asterisk/dsp.conf doesn't exist in my Asterisk 1.6.0.6
(something wrong?).
2. VAD is already disable for all
On Fri, Apr 23, 2010 at 4:34 PM, russian qwerty
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 informe
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.
Thanks in advance.
-- Original message --
From: russian qwerty
Date: 2010/3/31
Subject: Jitter Buffer and MeetMe.
To: asterisk-users@lists.
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.ne
