Subject: [Ieprep] Voice quality metrics

From: Henning Schulzrinne <[email-masked]>
Date: Wed, 13 Nov 2002 10:45:07 -0500

We have looked at the QOS issue for audio in depth. See 
http://www.cs.columbia.edu/~wenyu for a set of papers. Summary: it's not 3%.

The ITU e-model (cited in the papers) combines delay and codec 
distortions into a single metric and provides bounds on acceptable 
quality. Many of the metrics are relevant for analog transmission 
distortions (such as various frequency or noise distortions).

The delay bound is also described in ITU Rec. G.114, based on a fair 
number of human-subject measurements.

I think the jitter bound is largely immaterial; playout buffers 
translate jitter into absolute delay, so it's not an independent 
quantity (assuming well-functioning playout algorithms).

My first-order approximation is that a network that has usable telnet 
performance works well for voice. With high losses, you're not going to 
be enjoying your telnet session due to TCP retransmission delay. (The 
only difference is that temporary telnet delay spikes are probably more 
tolerable than for audio.)

Henning

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