[asterisk-users] Measuring Jitter in Asterisk

[John Todd]

At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:
>  > 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?
>
>Are txjitter and rxjitter working reliably? These calls are going to be
>placed from AMI and bridged together. Do you think the variables would
>be correctly set for each leg of the call?
>
>Doug.

I think the best way to determine this would be to compare the 
numbers provided by CHANNEL() versus the numbers provided by 
something with a little more reliability, such as wireshark, in a 
controlled set of circumstances.

Please post your results here - it would be an interesting test.

JT