Olle E. Johansson wrote:
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
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Is the May 4 version the newest?
<http://www.astertest.com/downloads/sip jitter buffer/latest/>
Best regards
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