[asterisk-dev] Jitter Buffer

[Andrew Kohlsmith]

On Tuesday 24 October 2006 13:30, Pavel Jezek wrote:
> Zoa, I think, that we should dejjitter when possible, always _before_
> any processing within asterisk (sending to some app, bridging to
> channel, etc.),
> is there any reason, why (in your example) dejjitter only sip-zap and
> not sip-iax?

yes -- you do NOT want to dejitter VOIP connections, as you will invariably 
increase latency for no reason.  The other voip link will introduce jitter of 
its own. 

You should ONLY dejitter if the other side requires precisely-timed packets, 
and 99% of VOIP systems do NOT require this, as the end hardware will 
dejitter.

-A.