[VoiceOps] Question about Packet Jitter

[Scott Berkman]

The only portion missing is the jitter through the SBC itself, which really
should be negligible.  Assuming that is 0, your equation below would be
correct.  However, unless you are trying to place blame, the only jitter
that really matters is the end to end since if the jitter buffer at the far
end overflows, you have problem regardless of the origin of the jitter (OK,
I am assuming you are not trying to do something with the RTP at any
intermediate point).

If you need to check for jitter through the SBC you should be able to take
traces on each side and compare the inter-arrival timings.

	-Scott

-----Original Message-----
From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org]
On Behalf Of Jim Dalton
Sent: Thursday, October 28, 2010 8:34 AM
To: voiceops at voiceops.org
Subject: [VoiceOps] Question about Packet Jitter

I have a question about calculating jitter.  Consider the call diagram
below.
A SIP call flows from the source device to the Session Border Controller
(SBC)
to the destination device.  All RTP packets are proxied through the SBC.

+========+                    +=====+                   +=============+
| Source |  Ingress Call Leg  | SBC |  Egress Call Leg  | Destination |
| Device |------------------->|     |------------------>|   Device    |
+========+                    +=====+                   +=============+
            Jitter Src to SBC
          ------------------->
 
                    Jitter Source to Destination
          --------------------------------------------->

If the packet jitter is known for the Ingress Call Leg from the source to
SBC and for end to end packet flow from the source to the destination, is it
possible to calculate jitter for the Egress Call Leg from the SBC to the
destination device? 

I do not think the following relationship is accurate.
(jitter Source to Destination) less (jitter Src to SBC) = (jitter SBC to
Destination)

Can anyone provide some guidance on this question?

Thank you,

Jim Dalton
VoIP Least Cost Routing, Analysis, Billing
1.404.526.6053
www.TransNexus.com



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