[asterisk-users] Jitter buffer question

[Ondrej Valousek]

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?

Many thanks,
Ondrej