AW: Use of RTCP statistics for estimating QoS of VoIP
Alan,
I fully agree with Bob's reply below. In my view H.460.9 is now sufficiently open to be used for transporting also advanced statistical data, even if the initial choice reflects the content of today's RTCP reports.
The information content of QoS reports is only one part of the standard. Another - maybe the major - part is the mechanism for conveying these reports from endpoints to gatekeepers and between gatekeepers. And this part does not change if improved QoS statistics become available.
It was the unanimous opinion of the experts at the SG16 meeting that H.460.9 will be used for extended QoS reports as soon as the relevant work in IETF has finished. How this is best done will be decided when the RFC is available. Any input from your side is certainly welcome.
Regards, Ernst Horvath Siemens AG
-----Ursprüngliche Nachricht----- Von: Robert R. Gilman [mailto:rrg@AVAYA.COM] Gesendet am: Donnerstag, 07. November 2002 02:13 An: ITU-SG16@echo.jf.INTEL.COM Betreff: 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@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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participants (1)
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Horvath Ernst