FW: [AVT] Resolution of G.726 payload format endian-issue

[Campos, Simao]

Alan,

I have just had a look at the final version sof H.460.9 and I could not
find any text that explains the real value of the QoS information that
can be transported (not including future extensions of course). This is
a real pity, because I do agree that many implementors will not be well
informed enough to determine the value of the information from the
standard RTCP packets.

Regards,

Frank





Alan Clark <alan at TELCHEMY.COM>
Sent by: Mailing list for parties associated with ITU-T Study Group 16
<ITU-SG16 at echo.jf.INTEL.COM>
2002-11-08 18:29
Please respond to alan


        To:     ITU-SG16 at echo.jf.INTEL.COM
        cc:     (bcc: Frank Derks/HVS/BE/PHILIPS)
        Subject:        Re: Use of RTCP statistics for estimating QoS of VoIP
        Classification:



Frank

I did suggest (via Mike Buckley) that the following be added to H.460.9
however don't know if it was actually included.

" It should be noted that RTCP reports were not originally intended to
provide accurate QoS metrics but for informative exchanges between
endpoints
to support possible dynamic configuration.  According to RFC1889 (a)
received packet count may include duplicate packets and hence it is
possible
to report negative packet loss (b) reported mean packet-to-packet delay
variation (jitter) is averaged over the last 16 received packets and hence
provides information only on the 160-480mS period immediately preceding
transmission of the RTCP report."

I do agree with you that potential implementors may well feel
uncomfortable
that the Recommendation in its initial form contains information that has
little or no value.  I am even more concerned that less well informed
implementors may accept what is in the Recommendation as being
"sufficient"
and hence attempt to manage VoIP services with metrics that can be quite
misleading.

Regards

Alan


-----Original Message-----
From: frank.derks at philips.com [mailto:frank.derks at philips.com]
Sent: Friday, November 08, 2002 2:46 AM
To: alan at TELCHEMY.COM
Cc: ITU-SG16 at echo.jf.INTEL.COM
Subject: Re: Use of RTCP statistics for estimating QoS of VoIP


Alan,

I certainly agree that, if the main body of the Recommendation covers
metrics
that are of little practical use of the recipient of these metrics, the
Recommendation can be misleading. Only with the proper background
information
will the reader be able to distinguish the real value in the particular
metrics.

I feel somewhat uncomfortable when the main body of a Recommendation
provides
me with something that is of little value, but which can be extended to
include
more useful information.

A possible way "out of this", is to include some text that clearly
explains the
situation and that reflects the concerns that were raised and agreed upon.

Regards,

Frank






Bob

I appreciate that a mechanism was provided to permit extension of H.460.9
and that it was agreed to incorporate the RTCP Extensions, once
standardized.  This does make a lot of sense and would definitely
accomodate
the new metrics when available.

The reason behind my email yesterday was that we often find that equipment
manufacturers, service providers and network managers don't understand
this
area as well as those who are actively conducting R&D in this area, and
hence in this case they may assume that RFC1889 metrics are "good" for
monitoring VoIP QoS since they are contained in H.460.9.  This is already
a
continual re-education process with the metrics in RFC1889 .. their
inclusion in H.460.9 will make the task even more difficult.

People within the industry have great respect for the ITU, understand that
there is an extensive process of technical discussion and review that is
part of the process of developing ITU standards, and hence place
considerable faith in the technical content of ITU Recommendations.  In
this
specific case H.460.9 contains metrics which are known to be inadequate
for
the purpose of measuring VoIP QoS and hence there is a risk of deployment
of
an incorrect or inappropriate element due to its incorporation in an ITU
Recommendation.

I am not trying to be "difficult" but tried to input essentially the same
viewpoints prior to the SG16 meeting and am genuinely concerned over the
outcome.

Regards

Alan Clark




-----Original Message-----
From: Mailing list for parties associated with ITU-T Study Group 16
[mailto:ITU-SG16 at echo.jf.INTEL.COM]On Behalf Of Robert R. Gilman
Sent: Wednesday, November 06, 2002 8:13 PM
To: ITU-SG16 at echo.jf.INTEL.COM
Subject: Re: Use of RTCP statistics for estimating QoS of VoIP


Alan-
Mike Buckley brought up the same questions you raise at the last SG16
meeting.
I don't believe anyone objects to your arguments, and, in fact, the
resulting
work, H.460.9, was modified to permit extension to other measures and the
exclusion of the current, IETF-standardized, measures.  Once the IETF
draft
you reference becomes a standard, it will be easy to refer to the
resulting
RFC and to incorporate the resulting measures as appropriate.  In the
meantime,
you have the option to use the existing measures and, through the
definition
of non-standard elements, any other measures you wish to implement on a
permanent
or trial basis.  It seemed better to set the framework now, so that
implementation
and use can begin, and to do so in such a way that future improvements may
be
incorporated quickly and easily.  Does this make sense?
-Bob
----------------------------------------------------
Bob Gilman       rrg at avaya.com      +1 303 538 3868


Alan Clark wrote:
>
> Paul
>
> I am responding to the recent SG16 email that encouraged direct input
from
> those not participating directly in the work of the Study Group.
>
> One particular recent issue relates to the recent addition of QoS
reporting
> based on RTCP statistics.  We have spent the last three years analyzing
and
> modeling the effects of IP network impairments on VoIP and would like to
> share our experience on this issue.
>
> RTCP statistics were intended to provide a "rough" real time feedback
> between endpoints to give some sense of network quality and facilitate
> potential adaptation.  IP network impairments are transient in nature
and
> the "averaged" metrics that RFC1889 produces are as a result very
> misleading.  The addition of QoS reporting based on RFC1889 metrics is,
in
> our opinion, a backward step as it infers that these statistics are
useful
> for VoIP quality estimation whereas they are in fact almost meaningless.
>
> (i) Packet Loss - counts
> RFC1889 counts of loss are not compensated for duplicate packets, hence
it
> is possible to have incorrect and even negative (!!) loss counts.
>
> (ii) Packet Loss- distribution
> Voice quality is very sensitive to the distribution of packet loss - if
2%
> loss occurs but the losses are spread out evenly then with PLC quality
would
> be affected only marginally, if 1% loss occurs but losses occur during
short
> intervals (typically the case) then quality can be affected quite
> significantly.
> RFC1889 does not consider the distribution of lost packets and hence
packet
> loss numbers can only grossly be equated to call quality (i.e. loss rate
of
> 20% is bad but you couldn't say for sure if 2% was bad or not)
>
> (iii) Jitter - metric
> The average packet-to-packet delay variation used in RFC1889 does not
> properly reflect the effects of congestion related delay changes.  For
> example
> (a) if congesion occurs and a queue starts to fill then delay increases
> however the packet-to-packet delay change may be small however the
increase
> in delay may cause the jitter buffer to resynchronize at some stage.
> (b) LAN congestion can cause short term spikes in delay, and hence may
lead
> to packet discards.  The average packet-to-packet delay variation will
not
> reflect this due to the smoothing effect of the averaging algorithm.
>
> (iv) Jitter - reporting interval
> As jitter is reported as a running average with a scaling factor of 16,
an
> RTCP report only describes the jitter level during the 200mS or so prior
to
> the report being generated and says nothing about the jitter level
during
> the much larger period between reports.
>
> We have spoken to, and worked with, people in the industry that have
been
> looking at this issue in detail and have found general agreement with
the
> comments above.  Unfortunately, there are also a fair number of people
that
> blindly accept that something must be "ok" because it is "in a standard"
..
> this obviously raised the concern that incorporating these metrics into
the
> H series of standards further proliferates this viewpoint.
>
> Within the IETF AVT group this issue was discussed almost a year ago,
and
> the group agreed to develop a set of metrics that were more accurate and
> appropriate for VoIP QoS reporting purposes.  These include measures of
> packet loss, packets discarded due to jitter, length and density of
bursts
> and gaps (of combined loss and discard), delay, call quality metrics...
> This is contained in the following:-
>
> draft-ietf-avt-rtcp-report-extns-01.txt
>
> I would very much appreciate understanding the technical rationale
behind
> incorporating RFC1889 based metrics into VoIP QoS reports, given the
> comments above.
>
> Regards
>
> Alan Clark
> Telchemy
>
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