[VoiceOps] Question about Packet Jitter

[Hiers, David]

Before you get down into the weeds with the math...

All the ITU docs that I've read portray the "network" as set of "non-rtp-aware, forward-as-fast-as-you-can router-type-things that probably have some qos capability".  The "edge" is implicitly defined as the first occurrence of a jitter buffer.  

As near as I can tell, there are no rtp jitter buffers in an ITU "network" for which all this fancy math applies.  If you transcode, scrub media with a ditech pvp, or otherwise implement anything that knows it should generate an rtp stream by sending a packet every X milliseconds, the ITU math is bounded by that device.

If the SBC in your example transcodes or has a jitter/loss-concealment buffer, it *is* the edge, and you cannot average the link jitters as described in the ITU docs.



David Hiers

CCIE (R/S, V), CISSP
ADP Dealer Services
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-----Original Message-----
From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Jim Dalton
Sent: Friday, October 29, 2010 6:43 AM
To: 'Schwarz, Albrecht (Albrecht)'; voiceops at voiceops.org
Subject: Re: [VoiceOps] Question about Packet Jitter

> VoIP performance experts should be familiar with
> 
> - ITU-T Y.1541 concerning IPDV (IP packet delay variation = jitter);
> there are many sections on IPDV, particularily also on "aggregated
> PDV";
> 

[JD wrote:] ITU-T Y.1541 "Amendment 1: New Appendix X - An example showing
how to calculate IPDV across multiple sections" addresses my question.

Thank you,

Jim D.


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