how can I tune SIP jitter? is it possible today in asterisk?
I assume you are asking for how to alleviate the effects of jitter on the
RTP audio streams initated by SIP? Asterisk currently only has a jitter
buffer for IAX, not for RTP streams. There are pland for the next
generation jitter buffer code to hook into RTP as well.
There is an entry on the bug tracker that touches on this topic.
thanks
is this in HEAD yet?
roy
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