[asterisk-users] JITTERBUFFER function

[Torbjorn Abrahamsson]

Hello!

 

I am going to use the JITTERBUFFER function in a SIP (and local channels)
only setup, but have some questions of how to use it:

 

1.       Do I need to activate jbenable in sip.conf? Or is it enough to call
the JITTERBUFFER function?



2.       What is the preferred way to invoke this function? Say I have
channel A which is not in need of buffering, while channel B do need it. If
A calls B and I do Set(JITTERBUFFER(fixed)=default), my guess is that it
will be attached to channel A:s read side. This is not the desired outcome,
as I would like to have it on B:s read side. How should I invoke this to
make the buffer belong to channel B? Maybe using b option to Dial? So that
when a JB-enabled device (B) calls out one just calls JITTERBUFFER from the
normal dialplan flow, and if there is a call to the device (B) one need to
use b option? Sound correct?

 

BR,

Torbjörn Abrahamsson

 

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