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
