[asterisk-users] Measuring Jitter in Asterisk

[John Todd]

At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
>
>How can I objectively measure jitter in Asterisk on a SIP channel?
>
>I don't just want to turn the new 1.4 jitter buffer on. I want to 
>measure jitter.
>
>Thanks,
>Doug.

You could look at the txjitter and rxjitter values (and other values) 
stored in the CHANNEL() function, and those values you're looking for 
were previously known as RTPAUDIOQOS.  Or is this not sufficient?

I opened a request ticket to allow viewing of arbitrary CHANNEL() 
data on any active channel, but to my knowledge it has not been 
implemented.  The RTP source of media has however been impelemented 
in the CHANNEL() structure.  It may be possible to use chan_local to 
ascertain media data on the "other" leg of a call, but I have not 
experimented with that.

http://bugs.digium.com/view.php?id=9620

JT