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@echo.jf.INTEL.COM]On Behalf Of Robert R. Gilman Sent: Wednesday, November 06, 2002 8:13 PM To: ITU-SG16@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@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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