I have used this freeware tool in the past: http://sineapps.com/sinestatiax.php maybe you can have a look at it as well.... l.
In data Thu, 09 Aug 2007 02:07:49 +0200, John Todd <[EMAIL PROTECTED]> ha scritto: > 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 > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Home of QueueMetrics - http://queuemetrics.com _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users