[asterisk-dev] Jitter Buffer

[John Lange]

On Fri, 2006-10-20 at 18:52 +0300, Zoa wrote:

> Thats correct because the voip side can accept jitter and should not 
> have a jitter buffer.

I assume you mean the Voip Device side? Yes the device can accept jitter
and dejitter the audio from Asterisk on its own. I'm not concerned about
that.

But if you mean Asterisk can accept jitter I'm not clear what you mean
by that? Asterisk needs to dejitter the Audio it receives from the SIP
device _before_ it routes it on to the Zap device.

> The thing that should be dejittered is the audio coming from asterisk 
> going to the PSTN.

I suppose thats another way of looking at it. It makes the logic very
complex though.

If I understand what you are saying; putting jbenable=yes in sip.conf
causes Asterisk to dejitter outgoing audio on the sip channel if the
source of the audio can cause jitter?

Therefore, in order to dejitter audio coming from a SIP device before it
goes out the Zap channel I would need to put "jbenable=yes" in
zapata.conf?

John