[asterisk-users] grandstream and jitter buffer

[Frank Bulk]

If the users' understanding of jitter is technically correct, and they're
complaining about quality issues (due to jitter or packet loss), then
lowering the jitter buffer isn't going to help.

An ADSL link, depending on the sync rate, can have 40+ msec of latency
between the DSL modem and DSLAM.  If the link quality is good, ask your DSL
provider to change the mode from INTERLEAVED to FAST; that should drop the
latency at least 10 msec.  

But before all that, you may want to ascertain how much of the issue is
packet loss versus jitter.

Regards,

Frank

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vinícius
Fontes
Sent: Wednesday, July 22, 2009 3:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] grandstream and jitter buffer

jbmaxsize=80 is way overkill. If your jitter is really close to 80ms then
you have some serious issues on your link and it's not suitable for VoIP at
all. Try jbmaxsize=40.


Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e
telefonia IP

----- "Kelvin Chan" <kelvinc at positronics.com> escreveu:

> Hi guys,
> 
> I have a bunch grandstream phones using ulaw and my users are
> complaining they are jittery when I use "canreinvite=yes". The data
> connection is an ADSL link dedicated for phone traffic. At any given
> time, I have at most 2 calls in parallel.
> 
> I'm not a huge fan of asterisk being in media path doing buffering
> because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and
> sometimes my users complain that "are you on a sat phone?"
> 
> Any suggestions?
> 
> -  
> Kelvin Chan
> 
> 
> 
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