[asterisk-users] Measuring Jitter in Asterisk

[Jared Smith]

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.