[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).
Seems like it would be much simpler and more effective to force sip-sip
bridge jitter buffering with jbforce=yes (1.4)
I cannot comment on 1.4 as we are still not even close to implementing
it. In the case of out-of-order packets, you are correct. Our solution
does not fix that. But it does fix jitter better than any other
solution up to 1.2. Out-of-order packets are much harder to come by
than regular 30-60ms jitter which we do find on at least 30% of
international calls.
At any rate, thanks for the information on the new sequence number in
the asterisk sip-sip bridge in 1.0.x. have you done any testing in 1.2
or 1.4 to confirm this is still the case?
I cannot remember doing testing in 1.2, but since there wasn't a readily
available jitter buffer for SIP in Asterisk 1.2 we continued using our
solution. When we get ready for 1.4 we will start all over again with
our testing to see if the new jitter buffer is as good as what we can
get with the ZAP timers.
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Andres
Technical Support
http://www.telesip.net
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