[asterisk-users] Jitter buffer question

[Ondrej Valousek]

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?
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.
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.

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