Andrew Kohlsmith wrote: > On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote: > >>As I understand it, adding VAD/Silence would require redesigning the >>entire RTP stack of Asterisk. > > > My understanding is that with the new jitter buffer both of these things are > completely doable now since nothing's timed off the incoming stream... > ...when the new jitterbuffer is included and if it's enabled...
Please help us test the SIP/RTP jitterbuffer! It's available in the bug tracker! /Olle _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
