asterisk users - Jan 2005 - SIP Jitter buffer(control?)

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.

	Matt

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.
>  
>
Where do the calls go?

If it goes <sip> <*> <VoIP> <endpoint>, the timestamps
from the other
VoIP leg get bridged through, and you shouldn't need a jitterbuffer for 
SIP. If your calls go to zap, the local audio device, or a meetme 
conference, though, you don't have many options other than trying to 
clean up your network..


However, I have a generic jitterbuffer implementation that's about ready 
now to be integrated into asterisk, to replace the present IAX2 
jitterbuffer, and be used also for RTP media streams.. It's presently 
being used in iaxclient. So, no timeline, but it's coming..

-SteveK

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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