[asterisk-users] Jitter buffer question

[Vinícius Fontes]

----- "Ondrej Valousek" <webserv at s3group.cz> escreveu:

> Hi Vinicius.
> 
> >>/ 1. To enable jitter buffer on SIP channels it seems I have to
> enable
> />>/ and 
> />>/ force it, right?
> /
> > Not sure about the forcing part (don't know exacly how it works),
> but I always set jbforce=yes to be sure.
> Ok, thanks!
> 
> >>/ 2. If I enable and force jitter buffer, Asterisk would always have
> to
> />>/ 
> />>/ stay in media path to make it function, right? If I am right,
> this 
> />>/ effectively disables native RTP bridging.
> /
> > Yes, there's no way Asterisk can create buffers if it's not on the
> media path.
> Yes, that makes a sense. I was just wondering if it is possible to
> configure it the 
> way that the jitterbuffer is enabled only if the asterisk server can
> not do native RTP bridging...
> 
> 
> >>/ 3. Is it possible to only enable jitter buffer on calls where the
> SIP
> />>/ 
> />>/ trunk is involved? It is no use for me to enable the jitter
> buffer 
> />>/ between SIP phones on the same LAN.
> /
> > Sure, just put the jbenable and other options on the SIP section of
> that trunk, instead of putting it on [general].
> 
> Well, I think that would not work since the jitterbuffer is only
> effective on the outgoing channels.
> If I receive a call from the SIP trunk, I hear jitter. To suppress it,
> I would have to enable jbforce/jbenable on my 
> local SIP channel as this is the outgoing one - the SIP trunk is the
> incoming one, right?

You're right. I just found a webpage that explains in detail the way the jitter buffer works: http://www.asterisk.org/node/48317.