[VoiceOps] Question about Packet Jitter

[Jim Dalton]

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