[asterisk-users] Jitter buffer question

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

> Hi List,
> 
> I have a question regarding jitterbuffer in Asterisk 1.4.24. I see
> that 
> jitterbuffer is only effective on the receiving channels.
> My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch
> 
> office.
> Questions:
> 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.

> 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.


> 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].

> 
> Many thanks for all answers, I have tried hard to google out them, but
> 
> no success so far.
> Ondrej
> 
> 
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