9 Feb
2012
9 Feb
'12
8:52 p.m.
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