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

[Russell Bryant]

Damon Estep wrote:
> Anyone know the answer? Has it been validated with packet captures, or 
> code review?

All of the timing information should be passed across the bridge in all of the 
frames that come in over RTP.  I can't say I verified this with packet captures, 
but I did look for this in the code review for the jitterbuffer code in 1.4.  I 
know there is explicit code to ensure this is the case.

-- 
Russell Bryant
Software Engineer
Digium, Inc.