There's always the RTP/RTCP RFC 3550. See section 6.4.1, interarrival jitter subsection for generating RTCP jitter stats. There's even sample code to implement this in Appendix A.8 Howard Hart Ooma Operations On 10/29/2010 05:15 AM, Jim Dalton wrote:
Jitter is an average measure of what's essentially noise (random variations in the arrival times of packets from a mean), and so I'd expect - for two jitter sources in series producing average jitter of j1 and j2 respectively - the overall jitter to be sqrt(j1^2 + j2^2).
Intuitively, a packet which is late as a result of the first jitter source might then be further delayed or delivered early by the second - so just summing the average jitter values isn't appropriate.
--Dave
[JD wrote:] Very good point. The relationship of overall jitter to be sqrt(j1^2 + j2^2) makes sense. However, I am having trouble working through a proof of the math. Any suggestions of a reference on jitter that might address this?
Many thanks for your insight,
Jim D.
-----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops- bounces at voiceops.org] On Behalf Of David Knell Sent: Friday, October 29, 2010 1:37 AM To: voiceops at voiceops.org Subject: Re: [VoiceOps] Question about Packet Jitter
On Thu, 2010-10-28 at 16:26 -0400, Scott Berkman wrote:
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.
I think I'd disagree with this. Jitter is an average measure of what's essentially noise (random variations in the arrival times of packets from a mean), and so I'd expect - for two jitter sources in series producing average jitter of j1 and j2 respectively - the overall jitter to be sqrt(j1^2 + j2^2).
Intuitively, a packet which is late as a result of the first jitter source might then be further delayed or delivered early by the second - so just summing the average jitter values isn't appropriate.
--Dave
-----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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