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
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
