[asterisk-users] jitterbuffer in pure voip (sip/iax) - what is best practice

[Pavel Jezek]

I know, that jitterbuffer should be set at receiving side and on 
outgoing call leg,
ie. if sipphone calls to asterisk and outgoing to zap chanel, I should 
set jitterbuffer on zap channel (to dejjitter audio stream from sipphone)

but what about pure voip situation (i.e. iax-iax, sip-iax, skinny-iax etc.)?
I have following setup (homeworkers using sip phone connected to home 
asterisk via SIP and registered via iax to central asterisk in main office)

sipphone1->LAN->(sip) asterisk-home1 (iax)----jittery link in upload 
direction----> (iax) _asterisk-office_ (iax) <------jittery link in 
upload direction---(iax) asterisk-home2 (sip)<-LAN<-sipphone2

on asterisk-office I have jitterbuffer enabled in both users&peers 
iax.conf definitions to home asterisk (and forced because central 
asterisk bridge voip-voip call legs, for what is jitterbuffer normaly 
not activated),
on asterisk-home I have jitterbuffer also enabled (but _not_ forced) on 
client side of iax connection
probably jitterbuffer on home asterisk never be activated, because it 
bridges voip call legs with two jitterbuffer implementation - generic jb 
for sip/rtp and iax jitterbuffer - am I right?

in this setup I have still many problems with jerky audio, what I'm 
doing wrong, _please_ help.
btw, sometimes call is even one-way audio after several minutes, I 
reported this:
http://bugs.digium.com/view.php?id=8325

many thanks.
PJ