Damon Estep wrote:
http://www.asterisk.org/node/48317 does a nice job of explaining the
1.4 jitter buffer, however it raised a question in my mind.
In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the
UDP RTP packets renumbered on transmit, or is the original
[Damon Estep]
I can see how bridging sip to sip via a zap channel would fix minor
jitter issues, since the zap timers are very accurate, however I cannot
see how this would correct out of order packets like a true jitter
buffer does (without the use of a jitter buffer on the sip-zap bridge).
