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
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