[Asterisk-Users] sip jitter buffer

[Derek Beaumont]

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.