[asterisk-users] SIP jitter buffer and asterisk native bridge

[Russell Bryant]

Damon Estep wrote:
> Thanks a bunch. So in theory the media gateway at the far end should be
> able to properly jitter buffer the entire RTP path from the ATA via
> asterisk, correct?
> 
> Would this be the same in 1.2 and it 1.4?

Yes, that is correct, but only for 1.4.  In the case of Asterisk 1.2, if the 
media is flowing through Asterisk, then I believe the timestamp information is lost.

> The best practice in the example given would be to rely on adaptive
> jitter buffers at the ATA and the media gateway, and not force jitter
> buffers in the SIP<>SIP asterisk bridge (1.4)

Correct.  That would be unless the endpoint has no jitterbuffer.  If that was 
the case, then forcing it to happen in the middle at Asterisk may help, but 
obviously can't fix anything that happens between Asterisk and the endpoint.

-- 
Russell Bryant
Software Engineer
Digium, Inc.