[Asterisk-Users] SIP Jitter buffer

[Rich Adamson]

> 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? 

My understanding is the new jitterbuffer code (in cvs-head) has 
been applied to iax connections, and the objective is to make it 
available for sip/rtp (and possibly other channel types) after 
things are cool in iax.

A jitterbuffer is only required when the delivery of rtp 
packets is inconsistent (eg, jerky).

Its my understanding that sip phones have at least some sort
of jitterbuffer built into firmware. Don't know how effective
they are for large variations in packet delivery though.