Subject: [ippm] metric requirements

From: "H.E.Holland" <[email-masked]>
Date: Sat, 07 Sep 2002 14:13:38 +0200

Dear all,

Where might I find quantitative metric requirements for services such as
VoIP? E.g. what are recommended values for one-way delay and packet loss
for a VoIP service?

Regard,
Henrik.

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From: stanislav shalunov <[email-masked]>
Date: 30 Sep 2002 12:10:04 -0400

"H.E.Holland" <[email-masked]> writes:

> Where might I find quantitative metric requirements for services
> such as VoIP? E.g. what are recommended values for one-way delay and
> packet loss for a VoIP service?

While others pointed out the ITU documents for one-way delay, I don't
think we answered the `loss' part of the question.

There can possibly be no answer, because codecs have various degrees
of tolerance and redundancy.  Relevant gedanken Experiment: Imagine
that your favorite codec were to send all its packets in triplicate;
that would change the threshold of tolerance for loss, wouldn't it?

-- 
Stanislav Shalunov		http://www.internet2.edu/~shalunov/

"I didn't attend the funeral, but I sent a nice letter saying that I
approved of it."				-- Mark Twain
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From: "Alan Clark" <[email-masked]>
Date: Mon, 30 Sep 2002 15:18:38 -0400

Fortunately many codecs behave in a consistent way and don't generate triple
the amount of traffic in order to improve resilience!!

Work done within ITU SG12, ETSI TIPHON and a few other groups has
characterized the behavior of a wide range of codecs under packet loss
conditions up to 20-30%.  It can however be a little tricky deducing if
packet loss concealment is in use from the info available in RTP headers!

One key point when considering the effects of packet loss on VoIP is the
distribution of packet loss.  This does not behave, as many would suggest,
as a simple burst channel exhibiting only consecutive loss.  Bursts of loss
are often sparse (e.g. 20%) and extend for periods of seconds - largely
caused by periods of congestion.  You may also see the extreme case of
bursts of consecutive loss extending for hundreds of packets, due to link
failures.
The effect of this is that call quality will be fine for some period of
time, and then experience several seconds of badly degraded quality before
improving again.

Alan Clark

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From: [email-masked] [mailto:[email-masked]]On Behalf
Of stanislav shalunov
Sent: Monday, September 30, 2002 12:10 PM
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Subject: Re: [ippm] metric requirements