This is kind of a repost of one part of a previous question I have had. Peer Username Call ID Seq (Tx/Rx) Lag Jitter Format 213.137.73.178 xxxxxxxxxx 3705df0a5f7 00103/00000 00000ms 0000ms 4 1 active SIP channel(s) I see that there is 0ms Jitter set. How can I set a Jitter buffer for use with sip channels? I can't seem to find any documentation about this. Any help is always appreciated.
What is the status of a jitter buffer implemenation for SIP ? Implemented / planed / total void ? chris. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041025/274ba4af/attachment.htm
Hi I am using CVS latest Is it correct there is no jitter buffer for SIP (RTP) Are there any plans for this? prob a stupid question: Is it required / do the endpoints handle this - if the src and destination are both SIP and there is no transcoding but asterisk is still in the media path? Thanks Jack __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com