> -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of John Todd > Sent: Friday, August 03, 2007 2:12 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Measuring Jitter in Asterisk > > 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. _______________________________________________ --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
