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

[Steve Kann]

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