On Fri, 2007-08-03 at 12:31 -0700, 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. You can use Wireshark (formerly Ethereal) to analyze the RTP stream after it's been captured. You can either use Wireshark itself to do the network capture, or you can capture the traffic with tcpdump and then pull the file into Wireshark at a later time. Inside Wireshark, go to Statistics, RTP, Show All Streams, and then select a stream and hit the "Analyze" button. -- Jared Smith Community Relations Manager Digium, Inc. _______________________________________________ --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
