[Asterisk-Users] SIP Jitter buffer

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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

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