status on jitter buffer for SIP/RTP? (OT?)
- From: Adam Moffett <adam (at) plexicomm.net>
- Date: Wed, 08 Mar 2006 10:20:49 -0500
This might be a better question for the dev list, but does anyone know
the status of a jitter buffer for SIP channels?
I know they created a generic jitter buffer and implemented it for IAX
channels. Does it work yet for SIP? Like is it there and disabled or
not there at all?
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