The jitter buffer is not enabled by default. On Feb 9, 2012, at 15:52, Holger Freyther <holger@freyther.de> wrote:
Hi all,
I have a setup with a FreeTDM/wanpipe trunk and most of the time the audio skips on the path to this trunk (e.g. after a period of silence, RTP marker bit set by the sender). It is not an issue with FreeTDM/wanpipe but more the way the RTP input is handled by FreeSWITCH. My theory is the audio skips because the RTP 'queue' has an under-run (while it shouldn't, there is no packet loss).
So how can I see if the jitterbuffer is enabled for a given call (or globally)? Can I somehow see under-runs? The recovered clock/ clock skew? I couldn't find any of this in the wiki[1].
thanks holger
[1] http://wiki.freeswitch.org/wiki/Jitterbuffer
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