[Asterisk-Users] SIP Jitter buffer(control?)

[Matt Riddell]

Matt Schulte wrote:
> I'm assuming asterisk does not have a SIP jitter buffer in place? Any
> ideas on how to help with this going over a data T1 where VoIP is shared
> with regular traffic? Problem is when people are downloading the voice
> is jittery, even lossy.

I think what you are looking for is QOS (quality of service).  There is 
a good wiki page (www.voip-info.org) on it.  I personally use the 
wondershaper script.

-- 
Cheers,

Matt Riddell
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