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2006 Nov 16
0
jitterbuffer in pure voip (sip/iax) - what is best practice
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...