[asterisk-users] Measuring Jitter in Asterisk

[Douglas Garstang]

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Jared Smith
> Sent: Friday, August 03, 2007 1:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
> 
> 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.

Jared, that won't do. I don't want to run the wireshark GUI, and I don't
wan't to run it on every single Asterisk box, connecting back to a local
X server running on my desktop. I also don't want to capture the RTP
data, and store it somewhere for later analysis. I'm looking at a
situation here with millions of subscribers and dozens of ITSP's.

What I do want to do is record QoS data to every single ITSP in real
time. I can then lease cost route based not just on route cost, but also
on historical QoS data. Whatever tool is used to collect the QoS data
has to stick it somewhere, such as MySQL, and then when I route a call,
I will have to query that data from MySQL.

> 
> Inside Wireshark, go to Statistics, RTP, Show All Streams, and then
> select a stream and hit the "Analyze" button.

I'm trying to avoid post-eyeballing the data.