AW: Use of RTCP statistics for estimating QoS of VoIP
ernst.horvath at SIEMENS.COM
Thu Nov 7 09:17:32 EST 2002
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.
> -----Ursprüngliche Nachricht-----
> Von: Robert R. Gilman [mailto:rrg at AVAYA.COM]
> Gesendet am: Donnerstag, 07. November 2002 02:13
> An: ITU-SG16 at echo.jf.INTEL.COM
> Betreff: Re: Use of RTCP statistics for estimating QoS of VoIP
> 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
> or trial basis. It seemed better to set the framework now, so that
> 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 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
> > 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
> > (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
> > For help on this mail list, send "HELP ITU-SG16" in a message to
> > listserv at lists.intel.com
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