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From: Mirja =?iso-8859-1?q?K=FChlewind?= <mirja.kuehlewind@ikr.uni-stuttgart.de>
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Subject: [rmcat] First meeting
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Dear working group!

We are about to plan our first meeting in Atlanta. We will have a 2 hour time 
slot. Time and date are not announced yet but will be this week.

Our first milestones include the following items:

 * Draft of requirements

 * Draft of evaluation criteria

* Draft of RTCP extensions and interaction with applications/RTP flow (if 
needed)

Please let us know if you are working on a draft that will address one of 
these items and if you are planing to present this work at the next meeting.

Especially everyone who is working on a proposal for a congestion control 
algorithm, we would like to encourage to evaluate your own proposal (against 
other proposals) and contribute to the evaluation criteria draft. We would 
like to see sufficient discussion on each proposed algorithm on the mailing 
list before discussing the proposal at a meeting or even adopting a proposal 
as working group item. That also means that drafts describing a proposal in 
detail should be submitted early enough in advance of a meeting. 

See you in Atlanta!
Mirja

From michawe@ifi.uio.no  Wed Oct  3 04:14:50 2012
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From: Michael Welzl <michawe@ifi.uio.no>
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Cc: rmcat WG <rmcat@ietf.org>, "Eggert, Lars" <lars@netapp.com>
Subject: Re: [rmcat] First meeting
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Hi!

A comment / clarification below...


On 2. okt. 2012, at 18:09, Mirja K=FChlewind wrote:

> Dear working group!
>=20
> We are about to plan our first meeting in Atlanta. We will have a 2 =
hour time=20
> slot. Time and date are not announced yet but will be this week.
>=20
> Our first milestones include the following items:
>=20
> * Draft of requirements
>=20
> * Draft of evaluation criteria
>=20
> * Draft of RTCP extensions and interaction with applications/RTP flow =
(if=20
> needed)

About the "interaction with applications" bit: there was, I think, =
general agreement pre-BOF / at the BOF that we need to some more =
information than now to be exchanged between applications (i.e. the =
codec, I think) and the transport (i.e. RTP/UDP). Some interesting =
thoughts on that are in the cc workshop's paper 19, for example: =
http://www.iab.org/activities/workshops/cc-workshop/papers/

One particular point that there seemed to be agreement on is that =
packets will have different value to an application, and these =
priorities should be specified to the transport to enable better =
mechanisms at that level.

So this is, essentially about the API between applications and =
transport. Unsurprisingly, the word "API" caused some trouble, which I =
resolved by rewording it into "interactions". Indeed we don't want to =
specify an API, but still, what type of information is exchanged at =
that, well, interface. I guess getting this right requires quite a bit =
of discussion...

I'm interested in this and volunteer to participate in document writing. =
I'm probably not enough of an RTP expert to be the main author, though.


> Please let us know if you are working on a draft that will address one =
of=20
> these items and if you are planing to present this work at the next =
meeting.
>=20
> Especially everyone who is working on a proposal for a congestion =
control=20
> algorithm, we would like to encourage to evaluate your own proposal =
(against=20
> other proposals) and contribute to the evaluation criteria draft. We =
would=20
> like to see sufficient discussion on each proposed algorithm on the =
mailing=20
> list before discussing the proposal at a meeting or even adopting a =
proposal=20
> as working group item. That also means that drafts describing a =
proposal in=20
> detail should be submitted early enough in advance of a meeting.=20
>=20
> See you in Atlanta!
> Mirja
> _______________________________________________
> rmcat mailing list
> rmcat@ietf.org
> https://www.ietf.org/mailman/listinfo/rmcat

Cheers,
Michael


--Apple-Mail=_C019F444-BD94-4256-9F4A-0CC1E9CDF955
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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Hi!<div><br></div><div>A comment / clarification =
below...</div><div><br></div><div><br><div><div>On 2. okt. 2012, at =
18:09, Mirja K=FChlewind wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div>Dear =
working group!<br><br>We are about to plan our first meeting in Atlanta. =
We will have a 2 hour time <br>slot. Time and date are not announced yet =
but will be this week.<br><br>Our first milestones include the following =
items:<br><br> * Draft of requirements<br><br> * Draft of evaluation =
criteria<br><br>* Draft of RTCP extensions and interaction with =
applications/RTP flow (if =
<br>needed)<br></div></blockquote><div><br></div><div>About the =
"interaction with applications" bit: there was, I think, general =
agreement pre-BOF / at the BOF that we need to some more information =
than now to be exchanged between applications (i.e. the codec, I think) =
and the transport (i.e. RTP/UDP). Some interesting thoughts on that are =
in the cc workshop's paper 19, for example:&nbsp;<a =
href=3D"http://www.iab.org/activities/workshops/cc-workshop/papers/">http:=
//www.iab.org/activities/workshops/cc-workshop/papers/</a></div><div><br><=
/div><div>One particular point that there seemed to be agreement on is =
that packets will have different value to an application, and these =
priorities should be specified to the transport to enable better =
mechanisms at that level.</div><div><br></div>So this is, essentially =
about the API between applications and transport. Unsurprisingly, the =
word "API" caused some trouble, which I resolved by rewording it into =
"interactions". Indeed we don't want to specify an API, but still, what =
type of information is exchanged at that, well, interface.&nbsp;I guess =
getting this right requires quite a bit of =
discussion...</div><div><br></div><div>I'm interested in this and =
volunteer to participate in document writing. I'm probably not enough of =
an RTP expert to be the main author, =
though.</div><div><br></div><div><br><blockquote type=3D"cite"><div>Please=
 let us know if you are working on a draft that will address one of =
<br>these items and if you are planing to present this work at the next =
meeting.<br><br>Especially everyone who is working on a proposal for a =
congestion control <br>algorithm, we would like to encourage to evaluate =
your own proposal (against <br>other proposals) and contribute to the =
evaluation criteria draft. We would <br>like to see sufficient =
discussion on each proposed algorithm on the mailing <br>list before =
discussing the proposal at a meeting or even adopting a proposal <br>as =
working group item. That also means that drafts describing a proposal in =
<br>detail should be submitted early enough in advance of a meeting. =
<br><br>See you in =
Atlanta!<br>Mirja<br>_______________________________________________<br>rm=
cat mailing list<br><a =
href=3D"mailto:rmcat@ietf.org">rmcat@ietf.org</a><br>https://www.ietf.org/=
mailman/listinfo/rmcat<br></div></blockquote></div><br></div><div>Cheers,<=
/div><div>Michael</div><div><br></div></body></html>=

--Apple-Mail=_C019F444-BD94-4256-9F4A-0CC1E9CDF955--

From john@jlc.net  Wed Oct  3 08:00:39 2012
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From: John Leslie <john@jlc.net>
To: rmcat WG <rmcat@ietf.org>
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References: <9DB4BEBD-8C49-443D-8C7D-4787EF314453@ifi.uio.no> <5060725F.4090709@alvestrand.no> <50608290.5020601@jesup.org>
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Subject: [rmcat] Piggybacking
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Randell Jesup <randell-ietf@jesup.org> wrote:
> On 9/24/2012 10:46 AM, Harald Alvestrand wrote:
>>
>> In the TCP case, the feedback mechanism is the ack, which is needed 
>> for reasons beyond congestion control. In the RTP case, there isn't 
>> such a single, simple mechanism needed for other reasons.
> 
> Agreed, though Michael indicated that we can piggyback congestion info 
> on reverse traffic.
> 
> This may be possible and I've thought about it.  Part of the problem is 
> that if you want to signal data back quickly, you have to wait up to 
> 20ms and in some cases longer to piggyback (and on 
> low-upstream-bandwidth links, the uplink transit time if you're 
> piggybacking on a 1500-byte video packet might be 20+ms).

   This is missing the point, IMHO.

   We are dealing in IP packets -- "unreliable" packets: meaning that
we aren't entitled to expect any particular packet to be delivered
at all.

   The signal we wish to send is tiny -- probably on the order of one
byte. If we allocate one byte per packet for feedback, it's almost
lost in the noise.

   But if we expect that byte and it doesn't arrive, that's a clear
congestion signal.

   (Granted its loss signals congestion in the path _to_ us...)

   Trying to optimize 20 msec of reaction time for real-time traffic
isn't the right paradigm. We want real-time traffic to adapt more
slowly than web-surfing traffic, but stay low for longer (to avoid
self-congestion).

   Thus, we can be pretty sure that the absence of an expected feedback
justifies a delay in any ramp-up of outgoing bit-rate currently planned.

   Beyond that, it gets more complicated. Nonetheless, we can easily
signal in our next scheduled outgoing packet that we missed an
expected feedback; and that signal (or the failure to receive our
next outgoing packeti) can trigger a retransmission -- catching up
typically faster than TCP can do.

   I'd like to suggest that we plan RMCAT around bidirectional
(though possibly very small in one direction) media packets. There
aren't many cases where the traffic is _so_ unidirectional that
an "I'm still interested in this stream" isn't appropriate.
(I don't mean to imply that the packet-rate needs to be the same
in both directions, least of all the bit-rate.)

> If you do piggyback, it has to be optional on a instance-by-instance
> basis 

   This is precisely what I _don't_ want to see. Perhaps the size of
the feedback might vary (though I doubt that's needed); but I want the
_absence_ of feedback to be a primary signal.

> (perhaps even estimating when the next chance will be, and the priority 
> of the data - but that speaks against a dumb receiver).

   (I don't think it makes sense to even think about a "dumb receiver.)

> TCP can "get away" with high packet count feedback because flows are 
> typically largely unidirectional or alternating flows, or on 
> well-connected devices, not edge nodes.  The PacketCable/etc issue with 
> shared upstreams and slot allocation is an example of reacting to this - 
> there even with the typically largely downstream flows, the upstream 
> acks become a limiting factor.

   Yes!

   This issue is very real in cable-upstream, and also significant in
many wireless setups.

   But the right approach is to measure "upstream" congestion and adapt
our packet rate to it. Really-tiny upstream packet aren't much "cheaper"
than medium-sized upstream packets.

   Granted, it's easy to confuse upstream-congestion with downstream-
congestion; but it's not _that_ hard to separate them.

>> Stickler: Amount of feedback has little correlation with packet drop. 
>> If feedback is lost due to congestion, more feedback will lead to more 
>> packet drops. What is true is that when feedback occurs more rarely, 
>> losing one feedback packet will lead to a greater gap between feedback 
>> packets.

   This is true, but some possible inferences are not true...

> I'd say less feedback means the odds that a drop will hit a feedback 
> packet are *lower* (as routers typically either tail-drop or 
> random-drop, but in either case in a static evaluation a 
> lower-bandwidth/packet flow (i.e. the feedback packets) will get hit 
> less often), but as Harald says the impact is higher.  And in a dynamic 
> evaluation this may shift some.

   In particular, inferring that we must do slavish AIMD is not true.

>>> - receiver-side-congestion-control-based feedback reduction as in 
>>> RRTCC means to reduce the feedback whenever feedback isn't urgent for 
>>> the sender. I'd argue that, in fact, we should reduce feedback only 
>>> when the channel carrying it (the backwards channel) is congested, 
>>> and that is a generic function, not bound to the sender-side 
>>> congestion control behavior.

   I agree that reducing feedback when the feedback path is not
congested really doesn't help.

   (When we _do_ need to reduce feedback, I suggest we need to lower
the packet-rate, letting the chips fall where they may; and be slow
to increase it again.)

> Personally, I assume both channels are always congested, or will be soon 
> (if we're doing our job right and aren't on a huge link),

   The assumption isn't particularly bad; but I envision many cases
where it isn't true.

> so bandwidth reduction for feedback always allows more bandwidth for
> useful data.

   While technically true, it's a bad tradeoff to discard information
about congestion -- this leads to growing your bit-rate too fast, and
more packet-loss than necessary.

>... 
> And my position is that any *good* two-way communication will run 
> near-but-not-quite-over the bandwidth limit as much of the time as 
> possible, so ACKs always have the potential to cause problems (and 
> always will force you a little further down the quality curve).

   I don't agree with this conclusion.

   With "ideal" feedback, we can be sure we don't initiate any
congestion problems, while "minimal" feedback means some of our
guesses will be wrong.

   And I don't think we have a good understanding of "the quality
curve" yet.

   Some real-time stacks will run very close to the maximum data-rate,
and hopefully will have good redundancy algorithms to reconstruct
lost packets. Others will choose to stay below the maximum. We shouldn't
prejudge that tradeoff, but limit ourselves to timely delivery of the
best congestion information, and algorithms for rate-adjustment.

--
John Leslie <john@jlc.net>

From michawe@ifi.uio.no  Thu Oct  4 03:44:20 2012
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Subject: Re: [rmcat] Piggybacking
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Hi,


On 3. okt. 2012, at 17:00, John Leslie wrote:

> Randell Jesup <randell-ietf@jesup.org> wrote:
>> On 9/24/2012 10:46 AM, Harald Alvestrand wrote:
>>>=20
>>> In the TCP case, the feedback mechanism is the ack, which is needed=20=

>>> for reasons beyond congestion control. In the RTP case, there isn't=20=

>>> such a single, simple mechanism needed for other reasons.
>>=20
>> Agreed, though Michael indicated that we can piggyback congestion =
info=20
>> on reverse traffic.
>>=20
>> This may be possible and I've thought about it.  Part of the problem =
is=20
>> that if you want to signal data back quickly, you have to wait up to=20=

>> 20ms and in some cases longer to piggyback (and on=20
>> low-upstream-bandwidth links, the uplink transit time if you're=20
>> piggybacking on a 1500-byte video packet might be 20+ms).
>=20
>   This is missing the point, IMHO.
>=20
>   We are dealing in IP packets -- "unreliable" packets: meaning that
> we aren't entitled to expect any particular packet to be delivered
> at all.
>=20
>   The signal we wish to send is tiny -- probably on the order of one
> byte. If we allocate one byte per packet for feedback, it's almost
> lost in the noise.
>=20
>   But if we expect that byte and it doesn't arrive, that's a clear
> congestion signal.
>=20
>   (Granted its loss signals congestion in the path _to_ us...)
>=20
>   Trying to optimize 20 msec of reaction time for real-time traffic
> isn't the right paradigm. We want real-time traffic to adapt more
> slowly than web-surfing traffic, but stay low for longer (to avoid
> self-congestion).
>=20
>   Thus, we can be pretty sure that the absence of an expected feedback
> justifies a delay in any ramp-up of outgoing bit-rate currently =
planned.

Indeed, I'm not sure a quick signal is always needed. Either way, as =
John says, a packet may be lost and the signal is tiny. We could agree =
to always piggyback - if the signal really can't wait, we can send it =
immediately but include it piggybacked onto the next datapacket too, for =
redundancy, making the whole thing more stable in case a feedback packet =
gets lost.


>   Beyond that, it gets more complicated. Nonetheless, we can easily
> signal in our next scheduled outgoing packet that we missed an
> expected feedback; and that signal (or the failure to receive our
> next outgoing packeti) can trigger a retransmission -- catching up
> typically faster than TCP can do.

I don't get that, and it sounds complicated and inefficient - you're =
adding round-trips here?


>   I'd like to suggest that we plan RMCAT around bidirectional
> (though possibly very small in one direction) media packets. There

> aren't many cases where the traffic is _so_ unidirectional that
> an "I'm still interested in this stream" isn't appropriate.
> (I don't mean to imply that the packet-rate needs to be the same
> in both directions, least of all the bit-rate.)
>=20
>> If you do piggyback, it has to be optional on a instance-by-instance
>> basis=20
>=20
>   This is precisely what I _don't_ want to see. Perhaps the size of
> the feedback might vary (though I doubt that's needed); but I want the
> _absence_ of feedback to be a primary signal.
>=20
>> (perhaps even estimating when the next chance will be, and the =
priority=20
>> of the data - but that speaks against a dumb receiver).
>=20
>   (I don't think it makes sense to even think about a "dumb receiver.)

Call it a *generic* receiver. That's what was meant in this discussion.

The problem is really that it's a (major, I think) benefit for =
deployment to have one side generic, and a generic sender is a more =
complicated setting than a generic receiver... in my opinion, none of =
the two choices are really ideal. To have things combined in the right =
place, I'll address this in a separate email in which I extend my =
previous analysis of the sender/receiver situation.


>> TCP can "get away" with high packet count feedback because flows are=20=

>> typically largely unidirectional or alternating flows, or on=20
>> well-connected devices, not edge nodes.  The PacketCable/etc issue =
with=20
>> shared upstreams and slot allocation is an example of reacting to =
this -=20
>> there even with the typically largely downstream flows, the upstream=20=

>> acks become a limiting factor.
>=20
>   Yes!
>=20
>   This issue is very real in cable-upstream, and also significant in
> many wireless setups.
>=20
>   But the right approach is to measure "upstream" congestion and adapt
> our packet rate to it. Really-tiny upstream packet aren't much =
"cheaper"
> than medium-sized upstream packets.

That was my argument too; it does add some complexity though. See my =
next email on sender vs receiver...

As for John's comments below, I agree with many of them.


>   Granted, it's easy to confuse upstream-congestion with downstream-
> congestion; but it's not _that_ hard to separate them.
>=20
>>> Stickler: Amount of feedback has little correlation with packet =
drop.=20
>>> If feedback is lost due to congestion, more feedback will lead to =
more=20
>>> packet drops. What is true is that when feedback occurs more rarely,=20=

>>> losing one feedback packet will lead to a greater gap between =
feedback=20
>>> packets.
>=20
>   This is true, but some possible inferences are not true...
>=20
>> I'd say less feedback means the odds that a drop will hit a feedback=20=

>> packet are *lower* (as routers typically either tail-drop or=20
>> random-drop, but in either case in a static evaluation a=20
>> lower-bandwidth/packet flow (i.e. the feedback packets) will get hit=20=

>> less often), but as Harald says the impact is higher.  And in a =
dynamic=20
>> evaluation this may shift some.
>=20
>   In particular, inferring that we must do slavish AIMD is not true.
>=20
>>>> - receiver-side-congestion-control-based feedback reduction as in=20=

>>>> RRTCC means to reduce the feedback whenever feedback isn't urgent =
for=20
>>>> the sender. I'd argue that, in fact, we should reduce feedback only=20=

>>>> when the channel carrying it (the backwards channel) is congested,=20=

>>>> and that is a generic function, not bound to the sender-side=20
>>>> congestion control behavior.
>=20
>   I agree that reducing feedback when the feedback path is not
> congested really doesn't help.
>=20
>   (When we _do_ need to reduce feedback, I suggest we need to lower
> the packet-rate, letting the chips fall where they may; and be slow
> to increase it again.)
>=20
>> Personally, I assume both channels are always congested, or will be =
soon=20
>> (if we're doing our job right and aren't on a huge link),
>=20
>   The assumption isn't particularly bad; but I envision many cases
> where it isn't true.
>=20
>> so bandwidth reduction for feedback always allows more bandwidth for
>> useful data.
>=20
>   While technically true, it's a bad tradeoff to discard information
> about congestion -- this leads to growing your bit-rate too fast, and
> more packet-loss than necessary.
>=20
>> ...=20
>> And my position is that any *good* two-way communication will run=20
>> near-but-not-quite-over the bandwidth limit as much of the time as=20
>> possible, so ACKs always have the potential to cause problems (and=20
>> always will force you a little further down the quality curve).
>=20
>   I don't agree with this conclusion.
>=20
>   With "ideal" feedback, we can be sure we don't initiate any
> congestion problems, while "minimal" feedback means some of our
> guesses will be wrong.
>=20
>   And I don't think we have a good understanding of "the quality
> curve" yet.
>=20
>   Some real-time stacks will run very close to the maximum data-rate,
> and hopefully will have good redundancy algorithms to reconstruct
> lost packets. Others will choose to stay below the maximum. We =
shouldn't
> prejudge that tradeoff, but limit ourselves to timely delivery of the
> best congestion information, and algorithms for rate-adjustment.
>=20
> --
> John Leslie <john@jlc.net>
> _______________________________________________
> rmcat mailing list
> rmcat@ietf.org
> https://www.ietf.org/mailman/listinfo/rmcat


From michawe@ifi.uio.no  Thu Oct  4 03:58:06 2012
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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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I herewith extend my own analysis of the sender- vs. receiver-based =
congestion control question, with one (in my opinion) significant plus =
in each camp:

> Ok, I think you got me convinced. The way I see it, there's a choice =
between:
> 1) sending a lot of feedback, and only reducing it when there is =
backwards congestion, by doing congestion control on the backwards =
channel like ACK-CC
> 2) simply always sending as little feedback as possible, based on the =
(forward) congestion control mechanism's requirements
>=20
> If we think of e.g. a videoconference, indeed congestion from all the =
feedback on the uplink is quite realistic. 1) can then become rather =
complicated... you may get congestion on your uplink, and it might be =
necessary to reduce the rate of your data stream, or it might suffice to =
reduce it on some of the feedback streams, but which? And then we have =
to apply priorities between feedback packets and your payload...
>=20

> 2) seems simpler, but it has the problem of making the mechanism more =
fragile, and perhaps unnecessarily so (in a situation where it might be =
fine to send much more feedback). I say "unnecessarily" because I'm not =
convinced about "both channels are always congested, or will be soon" =
because codecs may not be greedy, and streams such as VoIP may simply =
not generate enough traffic for that.


2), i.e. receiver-side congestion control, also has the ADVANTAGE that =
it doesn't misinterpret the loss of feedback as an indication of forward =
congestion. A receiver-side controller has a clear notion of all packets =
that have arrived in an interval and knows exactly what was dropped on =
the way from the sender to the receiver and what wasn't.

2), i.e. receiver-side congestion control, also has the DISADVANTAGE =
that tight interaction with the sending application gets harder - e.g., =
to consider per-packet priorities in the congestion control mechanism. =
With a receiver-side scheme, this requires richer feedback (in the =
style: "for high-priority packets: this rate; for low-priority and =
high-priority packets combined: that rate"). Then, the receiver would =
have to know how many priorities are there on the sender side to give =
the right amount of feedback...  this will make things unnecessarily =
complicated or, as I rather suspect, not happen, and then greatly limit =
us.



Cheers,
Michael


From randell-ietf@jesup.org  Thu Oct  4 07:08:34 2012
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Subject: Re: [rmcat] Piggybacking
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On 10/4/2012 6:44 AM, Michael Welzl wrote:
> Hi,
>
>
> On 3. okt. 2012, at 17:00, John Leslie wrote:
>
>> Randell Jesup <randell-ietf@jesup.org> wrote:
>>> On 9/24/2012 10:46 AM, Harald Alvestrand wrote:
>>>> In the TCP case, the feedback mechanism is the ack, which is needed
>>>> for reasons beyond congestion control. In the RTP case, there isn't
>>>> such a single, simple mechanism needed for other reasons.
>>> Agreed, though Michael indicated that we can piggyback congestion info
>>> on reverse traffic.
>>>
>>> This may be possible and I've thought about it.  Part of the problem is
>>> that if you want to signal data back quickly, you have to wait up to
>>> 20ms and in some cases longer to piggyback (and on
>>> low-upstream-bandwidth links, the uplink transit time if you're
>>> piggybacking on a 1500-byte video packet might be 20+ms).
>>    This is missing the point, IMHO.
>>
>>    We are dealing in IP packets -- "unreliable" packets: meaning that
>> we aren't entitled to expect any particular packet to be delivered
>> at all.
>>
>>    The signal we wish to send is tiny -- probably on the order of one
>> byte. If we allocate one byte per packet for feedback, it's almost
>> lost in the noise.
>>
>>    But if we expect that byte and it doesn't arrive, that's a clear
>> congestion signal.
>>
>>    (Granted its loss signals congestion in the path _to_ us...)
>>
>>    Trying to optimize 20 msec of reaction time for real-time traffic
>> isn't the right paradigm. We want real-time traffic to adapt more
>> slowly than web-surfing traffic, but stay low for longer (to avoid
>> self-congestion).
>>
>>    Thus, we can be pretty sure that the absence of an expected feedback
>> justifies a delay in any ramp-up of outgoing bit-rate currently planned.
> Indeed, I'm not sure a quick signal is always needed. Either way, as John says, a packet may be lost and the signal is tiny. We could agree to always piggyback - if the signal really can't wait, we can send it immediately but include it piggybacked onto the next datapacket too, for redundancy, making the whole thing more stable in case a feedback packet gets lost.

That's closer to what I was thinking - use independent packets 
*sometimes*, but mostly use piggybacked (which reduced the overall 
impact of feedback).  I'd use independent packets when there's an 
expected (or measured) long delay waiting to piggyback (for example if 
muted or using VAD), and when we have new (unexpected) information to send.

There is an issue with 'how' piggybacking is done on RTP.  In RTCP, it's 
fairly easy (though more than one byte!)  In RTP, you either are 
changing the payload formats or you're using a header extension.  (Or 
we're designing RTP v3....  That would be 'interesting'.)

WebRTC 'bundle' helps us here (more packets to piggyback on), and in 
general if we have multiple flows sharing 
congestion-measurement/reporting domains.  (Also more data (faster 
smoothing of noise), etc).  In my previous implementation one common 
case was that on packet loss (video), the system would send an immediate 
RTCP NACK/rtcpfb pli/etc to ask for a refresh, and congestion info would 
hitchhike along with any RTCP packet we sent.

>>    Beyond that, it gets more complicated. Nonetheless, we can easily
>> signal in our next scheduled outgoing packet that we missed an
>> expected feedback; and that signal (or the failure to receive our
>> next outgoing packeti) can trigger a retransmission -- catching up
>> typically faster than TCP can do.
> I don't get that, and it sounds complicated and inefficient - you're adding round-trips here?

Basically PLI for congestion control.  However, you need a jitter/delay 
buffer to handle transit jitter and reordering, so it's not so simple.  
Or you decide fairly frequent false positives are ok, which might be 
possible.

>>    I'd like to suggest that we plan RMCAT around bidirectional
>> (though possibly very small in one direction) media packets. There

I think it has to handle *well* both one-directional and bidirectional 
flows.  But opportunistic piggybacking would do that (if you solve the 
'how' question above for RTP).  You definitely can piggyback with other 
RTCP traffic.

>> aren't many cases where the traffic is _so_ unidirectional that
>> an "I'm still interested in this stream" isn't appropriate.
>> (I don't mean to imply that the packet-rate needs to be the same
>> in both directions, least of all the bit-rate.)

Right, though that also means we're not talking about ack-per-packet in 
unidirectional.
While I'm not suggesting this, when there was no other RTCP traffic to 
piggyback on (and no loss) I ran ~1 second congestion reports.

>>
>>> If you do piggyback, it has to be optional on a instance-by-instance
>>> basis
>>    This is precisely what I _don't_ want to see. Perhaps the size of
>> the feedback might vary (though I doubt that's needed); but I want the
>> _absence_ of feedback to be a primary signal.

That's not at odds with what I was saying - I meant 
packet-by-packet/signal-by-signal you decide to piggyback or not - and 
if you're waiting to piggyback you can react to a timer expiry and send 
it standalone.  So you *could* design it to have "feedback every Nms" 
(with adjustable N), and have the receiver send feedback every <=N, 
using piggyback or not as available.

>>> TCP can "get away" with high packet count feedback because flows are
>>> typically largely unidirectional or alternating flows, or on
>>> well-connected devices, not edge nodes.  The PacketCable/etc issue with
>>> shared upstreams and slot allocation is an example of reacting to this -
>>> there even with the typically largely downstream flows, the upstream
>>> acks become a limiting factor.
>>    Yes!
>>
>>    This issue is very real in cable-upstream, and also significant in
>> many wireless setups.
>>
>>    But the right approach is to measure "upstream" congestion and adapt
>> our packet rate to it. Really-tiny upstream packet aren't much "cheaper"
>> than medium-sized upstream packets.
> That was my argument too; it does add some complexity though. See my next email on sender vs receiver...

Agreed.  Many/most routers buffer by packet not by packet-size. Each 
WiFi packet is much heavierweight than the payload size would indicate, 
etc.  Which is why I like reducing ACK *packet* flows.

>
> As for John's comments below, I agree with many of them.
>
>
>>    Granted, it's easy to confuse upstream-congestion with downstream-
>> congestion; but it's not _that_ hard to separate them.
>>
>>>> Stickler: Amount of feedback has little correlation with packet drop.
>>>> If feedback is lost due to congestion, more feedback will lead to more
>>>> packet drops. What is true is that when feedback occurs more rarely,
>>>> losing one feedback packet will lead to a greater gap between feedback
>>>> packets.
>>    This is true, but some possible inferences are not true...
>>
>>> I'd say less feedback means the odds that a drop will hit a feedback
>>> packet are *lower* (as routers typically either tail-drop or
>>> random-drop, but in either case in a static evaluation a
>>> lower-bandwidth/packet flow (i.e. the feedback packets) will get hit
>>> less often), but as Harald says the impact is higher.  And in a dynamic
>>> evaluation this may shift some.
>>    In particular, inferring that we must do slavish AIMD is not true.
>>
>>>>> - receiver-side-congestion-control-based feedback reduction as in
>>>>> RRTCC means to reduce the feedback whenever feedback isn't urgent for
>>>>> the sender. I'd argue that, in fact, we should reduce feedback only
>>>>> when the channel carrying it (the backwards channel) is congested,
>>>>> and that is a generic function, not bound to the sender-side
>>>>> congestion control behavior.
>>    I agree that reducing feedback when the feedback path is not
>> congested really doesn't help.
>>
>>    (When we _do_ need to reduce feedback, I suggest we need to lower
>> the packet-rate, letting the chips fall where they may; and be slow
>> to increase it again.)
>>
>>> Personally, I assume both channels are always congested, or will be soon
>>> (if we're doing our job right and aren't on a huge link),
>>    The assumption isn't particularly bad; but I envision many cases
>> where it isn't true.
>>
>>> so bandwidth reduction for feedback always allows more bandwidth for
>>> useful data.
>>    While technically true, it's a bad tradeoff to discard information
>> about congestion -- this leads to growing your bit-rate too fast, and
>> more packet-loss than necessary.
>>
>>> ...
>>> And my position is that any *good* two-way communication will run
>>> near-but-not-quite-over the bandwidth limit as much of the time as
>>> possible, so ACKs always have the potential to cause problems (and
>>> always will force you a little further down the quality curve).
>>    I don't agree with this conclusion.
>>
>>    With "ideal" feedback, we can be sure we don't initiate any
>> congestion problems, while "minimal" feedback means some of our
>> guesses will be wrong.
>>
>>    And I don't think we have a good understanding of "the quality
>> curve" yet.
>>
>>    Some real-time stacks will run very close to the maximum data-rate,
>> and hopefully will have good redundancy algorithms to reconstruct
>> lost packets. Others will choose to stay below the maximum. We shouldn't
>> prejudge that tradeoff, but limit ourselves to timely delivery of the
>> best congestion information, and algorithms for rate-adjustment.
>>
>> --
>> John Leslie <john@jlc.net>
>> _______________________________________________
>> rmcat mailing list
>> rmcat@ietf.org
>> https://www.ietf.org/mailman/listinfo/rmcat


-- 
Randell Jesup
randell-ietf@jesup.org


From john@jlc.net  Thu Oct  4 07:46:40 2012
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Michael Welzl <michawe@ifi.uio.no> wrote:
> 
> I herewith extend my own analysis of the sender- vs. receiver-based
> congestion control question, with one (in my opinion) significant plus
> in each camp:
> 
>>... The way I see it, there's a choice between:
>> 1) sending a lot of feedback, and only reducing it when there is
>>    backwards congestion, by doing congestion control on the backwards
>>    channel like ACK-CC
>> 2) simply always sending as little feedback as possible, based on the
>>    (forward) congestion control mechanism's requirements
> 
> 2), i.e. receiver-side congestion control, also has the ADVANTAGE that
>     it doesn't misinterpret the loss of feedback as an indication of
>     forward congestion.

   We should try pretty hard to avoid the need to interpret the lack
of feedback as indicating what happened on the forward path.

>     A receiver-side controller has a clear notion of all packets that
>     have arrived in an interval and knows exactly what was dropped on
>     the way from the sender to the receiver and what wasn't.

   (Assuming, of course, it knows the schedule of what the sender sent.)

> 2), i.e. receiver-side congestion control, also has the DISADVANTAGE
>     that tight interaction with the sending application gets harder -
>     e.g., to consider per-packet priorities in the congestion control
>     mechanism.

   I don't see any necessary coupling between priorities of what the
sender would like to send and what the receiver would like to receive.

>     With a receiver-side scheme, this requires richer feedback (in
>     the style: "for high-priority packets: this rate; for low-priority
>     and high-priority packets combined: that rate"). Then, the receiver
>     would have to know how many priorities are there on the sender
>     side to give the right amount of feedback...  this will make
>     things unnecessarily complicated or, as I rather suspect, not
>     happen, and then greatly limit us.

   Congestion is somewhat orthogonal to priorities: we're interested
in avoiding congestion far more than we're worried about which packets
have the highest priority.

   The receiver may or may not wish to express an opinion of which
streams should have highest priority. If it does, this preference
should be stable over any particular period of congestion.

   The sender may well believe that certain events should have high
priority regardless of the receiver's priorities. It will no doubt
find a way to send these with higher priority regardless of what we
may specify.

   We should concentrate our efforts on getting both endpoints a
measure of congestion along the path they're using to send, and
one or more algorithms to back off the _overall_ sending rate. The
network doesn't care one whit for our application-layer priorities.

--
John Leslie <john@jlc.net>

From michawe@ifi.uio.no  Thu Oct  4 15:09:43 2012
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On Oct 4, 2012, at 4:46 PM, John Leslie wrote:

> Michael Welzl <michawe@ifi.uio.no> wrote:
>>
>> I herewith extend my own analysis of the sender- vs. receiver-based
>> congestion control question, with one (in my opinion) significant  
>> plus
>> in each camp:
>>
>>> ... The way I see it, there's a choice between:
>>> 1) sending a lot of feedback, and only reducing it when there is
>>>   backwards congestion, by doing congestion control on the backwards
>>>   channel like ACK-CC
>>> 2) simply always sending as little feedback as possible, based on  
>>> the
>>>   (forward) congestion control mechanism's requirements
>>
>> 2), i.e. receiver-side congestion control, also has the ADVANTAGE  
>> that
>>    it doesn't misinterpret the loss of feedback as an indication of
>>    forward congestion.
>
>   We should try pretty hard to avoid the need to interpret the lack
> of feedback as indicating what happened on the forward path.
>
>>    A receiver-side controller has a clear notion of all packets that
>>    have arrived in an interval and knows exactly what was dropped on
>>    the way from the sender to the receiver and what wasn't.
>
>   (Assuming, of course, it knows the schedule of what the sender  
> sent.)
>
>> 2), i.e. receiver-side congestion control, also has the DISADVANTAGE
>>    that tight interaction with the sending application gets harder -
>>    e.g., to consider per-packet priorities in the congestion control
>>    mechanism.
>
>   I don't see any necessary coupling between priorities of what the
> sender would like to send and what the receiver would like to receive.

Not "what the receiver would like to receive", but between the  
sender's priorities and congestion control.


>>    With a receiver-side scheme, this requires richer feedback (in
>>    the style: "for high-priority packets: this rate; for low-priority
>>    and high-priority packets combined: that rate"). Then, the  
>> receiver
>>    would have to know how many priorities are there on the sender
>>    side to give the right amount of feedback...  this will make
>>    things unnecessarily complicated or, as I rather suspect, not
>>    happen, and then greatly limit us.
>
>   Congestion is somewhat orthogonal to priorities: we're interested
> in avoiding congestion far more than we're worried about which packets
> have the highest priority.
>
>   The receiver may or may not wish to express an opinion of which
> streams should have highest priority. If it does, this preference
> should be stable over any particular period of congestion.
>
>   The sender may well believe that certain events should have high
> priority regardless of the receiver's priorities. It will no doubt
> find a way to send these with higher priority regardless of what we
> may specify.
>
>   We should concentrate our efforts on getting both endpoints a
> measure of congestion along the path they're using to send, and
> one or more algorithms to back off the _overall_ sending rate. The
> network doesn't care one whit for our application-layer priorities.

First, to clarify, I'm talking about per-packet priorities within a  
stream. Second, my point is that congestion control could be done  
better if packet priorities would be known, and could then be taken  
into account.
Right now, we are essentially facing greedy or non-greedy traffic, and  
we have the LEDBAT notion of a generally lower priority. In fact a  
mechanism could opt for a more LEDBAT'ish behavior when its buffer is  
full of low-priority packets and be more aggressive when it has high- 
priority packets.

To the best of my knowledge, these things don't really exist yet, but  
I believe that this is largely because this knowledge is not typically  
available to a congestion control mechanism. Note that conforming to  
such priorities is not just some obscure idea, as there are hard facts  
for a significant qualitative impact, e.g. in the case of video.

But if a mechanism does such things, and it's on the receiver side,  
whereas the send buffer is on the sender side, things get problematic.  
Either we fix this, or we move the mechanism to the sender, or we  
ignore the chance we have, here, to do things better. I guess these  
are our choices.

Cheers,
Michael


From john@jlc.net  Thu Oct  4 15:31:16 2012
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Michael Welzl <michawe@ifi.uio.no> wrote:
> On Oct 4, 2012, at 4:46 PM, John Leslie wrote:
>... 
>> I don't see any necessary coupling between priorities of what the
>> sender would like to send and what the receiver would like to receive.
> 
> Not "what the receiver would like to receive", but between the  
> sender's priorities and congestion control.

   Two different things...

>... 
> First, to clarify, I'm talking about per-packet priorities within a  
> stream. Second, my point is that congestion control could be done  
> better if packet priorities would be known, and could then be taken  
> into account.

   I guess I'm missing why packet priorities would ever _not_ be the
domain of the sender -- even if advised by receiver preferences.

> Right now, we are essentially facing greedy or non-greedy traffic,

   Do you mean we're "contending against" those? If so, I agree.

> and we have the LEDBAT notion of a generally lower priority. In fact a  
> mechanism could opt for a more LEDBAT'ish behavior when its buffer is  
> full of low-priority packets and be more aggressive when it has high- 
> priority packets.

   If you're saying that sending rate may deserve to increase faster
when there are higher priority packets to send, I could agree but it
seems unlikely such a situation will happen. If OTOH, the sender has
_only_ higher priority packets, a more aggressive sending rate _may_
be justified. Either way, it seems unlikely to be a good idea to
increase sending rate too aggressively for low-priority packets...

> To the best of my knowledge, these things don't really exist yet, but  
> I believe that this is largely because this knowledge is not typically  
> available to a congestion control mechanism. Note that conforming to  
> such priorities is not just some obscure idea, as there are hard facts  
> for a significant qualitative impact, e.g. in the case of video.

   This sounds worth discussing. (I'm seriously hopeful that the
"video codec" BoF will lead to a family of video codecs with smoother
bit-rate than is typical today.)

> But if a mechanism does such things, and it's on the receiver side,  
> whereas the send buffer is on the sender side, things get problematic.  
> Either we fix this, or we move the mechanism to the sender, or we  
> ignore the chance we have, here, to do things better. I guess these  
> are our choices.

   To tell truth, I think it's premature to specify what will be
receiver-side vs. sender-side.

   Especially, I think we need intelligence at both ends in order to
communicate useful congestion information in the two directions.
Once we have designed that we will be better able to specify the
control loops.

   (Assuming ConEx ever finishes, we are likely to see some kind of
congestion-based pricing both for senders and receivers -- and the
control loops ideally would serve the price-based needs of both
senders and receivers.)

--
John Leslie <john@jlc.net>

From michawe@ifi.uio.no  Fri Oct  5 02:46:29 2012
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To: John Leslie <john@jlc.net>
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Cc: rmcat WG <rmcat@ietf.org>
Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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On 5. okt. 2012, at 00:31, John Leslie wrote:

> Michael Welzl <michawe@ifi.uio.no> wrote:
>> On Oct 4, 2012, at 4:46 PM, John Leslie wrote:
>> ...=20
>>> I don't see any necessary coupling between priorities of what the
>>> sender would like to send and what the receiver would like to =
receive.
>>=20
>> Not "what the receiver would like to receive", but between the =20
>> sender's priorities and congestion control.
>=20
>   Two different things...
>=20
>> ...=20
>> First, to clarify, I'm talking about per-packet priorities within a =20=

>> stream. Second, my point is that congestion control could be done =20
>> better if packet priorities would be known, and could then be taken =20=

>> into account.
>=20
>   I guess I'm missing why packet priorities would ever _not_ be the
> domain of the sender -- even if advised by receiver preferences.

It's just not what I meant. I meant priorities influencing the =
congestion control behavior (I should say the "sender's" behavior, but =
this discussion is about having the congestion control logic on the =
sender or receiver side... so...).


>> Right now, we are essentially facing greedy or non-greedy traffic,
>=20
>   Do you mean we're "contending against" those? If so, I agree.

I would agree with that too, but I meant: congestion control mechanism =
design is considering only these two classes of traffic. Sorry for my =
imprecise wording here!


>> and we have the LEDBAT notion of a generally lower priority. In fact =
a =20
>> mechanism could opt for a more LEDBAT'ish behavior when its buffer is =
=20
>> full of low-priority packets and be more aggressive when it has high-=20=

>> priority packets.
>=20
>   If you're saying that sending rate may deserve to increase faster
> when there are higher priority packets to send, I could agree but it
> seems unlikely such a situation will happen. If OTOH, the sender has

Yes that's what I'm saying. I don't understand why that would be =
unlikely?


> _only_ higher priority packets, a more aggressive sending rate _may_
> be justified. Either way, it seems unlikely to be a good idea to
> increase sending rate too aggressively for low-priority packets...

Yes, exactly. We can be less aggressive when we have low-priority =
packets to send.


>> To the best of my knowledge, these things don't really exist yet, but =
=20
>> I believe that this is largely because this knowledge is not =
typically =20
>> available to a congestion control mechanism. Note that conforming to =20=

>> such priorities is not just some obscure idea, as there are hard =
facts =20
>> for a significant qualitative impact, e.g. in the case of video.
>=20
>   This sounds worth discussing. (I'm seriously hopeful that the
> "video codec" BoF will lead to a family of video codecs with smoother
> bit-rate than is typical today.)
>=20
>> But if a mechanism does such things, and it's on the receiver side, =20=

>> whereas the send buffer is on the sender side, things get =
problematic. =20
>> Either we fix this, or we move the mechanism to the sender, or we =20
>> ignore the chance we have, here, to do things better. I guess these =20=

>> are our choices.
>=20
>   To tell truth, I think it's premature to specify what will be
> receiver-side vs. sender-side.
>=20
>   Especially, I think we need intelligence at both ends in order to
> communicate useful congestion information in the two directions.
> Once we have designed that we will be better able to specify the
> control loops.

I hear you, but have a hard time imagining this without envisioning =
whether the control decision is made on the sender or receiver side, as =
this influences what needs to be signalled.

So I think it's a key architectural decision that we'd better get right =
from the outset. This is because having one side generic ("dumb") is =
probably very important for deployment. Look at TCP, with its many =
congestion control flavors in Linux' or FreeBSD's pluggable congestion =
control. If I run a TCP based (e.g. web) server, I can just pick a =
mechanism with a simple command. No need to hope for clients to support =
it; no need to signal to the client what mechanism I use. No need to =
even standardize it, as we had to learn the hard way...

In case of rmcat in rtcweb, this would mean that browser vendor X could =
simply start playing with a new congestion control mechanism (offer it =
to its users or enable it by default), and that would work with all =
other rtcweb/rmcat browsers, without ever having support for this =
mechanism from browser vendor Y or Z. If we want that to be possible, we =
need to decide which side should be the generic one at some point, and =
then agree on the exact functionality of this generic side.


>   (Assuming ConEx ever finishes, we are likely to see some kind of
> congestion-based pricing both for senders and receivers -- and the
> control loops ideally would serve the price-based needs of both
> senders and receivers.)

I would prefer to leave this consideration aside, because 1) rmcat works =
on a much tighter schedule than ConEx, as it will probably be deployed =
in the rtcweb context fast, while I don't know how long we'd have to =
wait for reasonably wide-spread ConEx deployment...   2) I don't think =
that this matters much for our decision - in the worst case, we would =
find that ConEx related information needs to be signalled from one end =
to the other, and then we could consider extending to whatever =
signalling scheme we define.

Cheers,
Michael


From csp@csperkins.org  Fri Oct  5 02:53:37 2012
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Subject: Re: [rmcat] Piggybacking
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On 4 Oct 2012, at 15:06, Randell Jesup wrote:
> On 10/4/2012 6:44 AM, Michael Welzl wrote:
>> On 3. okt. 2012, at 17:00, John Leslie wrote:
...
> There is an issue with 'how' piggybacking is done on RTP.  In RTCP, =
it's fairly easy (though more than one byte!)  In RTP, you either are =
changing the payload formats or you're using a header extension.  (Or =
we're designing RTP v3....  That would be 'interesting'.)

Or defining a new RTP Profile (see section 5.3 of RFC 3550).


--=20
Colin Perkins
http://csperkins.org/




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From: "Eggert, Lars" <lars@netapp.com>
To: =?iso-8859-1?Q?Mirja_K=FChlewind?= <mirja.kuehlewind@ikr.uni-stuttgart.de>
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Hi,

On Oct 2, 2012, at 18:09, Mirja K=FChlewind =
<mirja.kuehlewind@ikr.uni-stuttgart.de> wrote:
> We are about to plan our first meeting in Atlanta. We will have a 2 =
hour time=20
> slot. Time and date are not announced yet but will be this week.

our TENTATIVE slot is THURSDAY, November 8, 2012, 1300-1500 Afternoon =
Session I. This may still change.

The ID cutoff for new -00 drafts (which most of ours will be) is Oct 15. =
Please make sure your draft-yourname-rmcat-* is submitted by then.

Begin discussing drafts on the list as soon as they become available. =
Mirja and me will gauge the need for facetime in Atlanta based on =
whether list discussion indicated interest in a draft or topic.

Lars=

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From john@jlc.net  Fri Oct  5 07:58:40 2012
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From: John Leslie <john@jlc.net>
To: Michael Welzl <michawe@ifi.uio.no>
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References: <9DB4BEBD-8C49-443D-8C7D-4787EF314453@ifi.uio.no> <5060725F.4090709@alvestrand.no> <50608290.5020601@jesup.org> <9BE23689-DFA9-4CA9-87CB-F82C7A2DC565@ifi.uio.no> <9E9AE97E-2E9E-406B-8E55-2FAA966DF9D3@ifi.uio.no> <20121004144639.GC97796@verdi> <CB17D93B-96C0-4A6C-937F-B111C6D5DFD7@ifi.uio.no> <20121004223115.GE97796@verdi> <9CADC40E-6432-4DFA-9512-8C780A6A0E4C@ifi.uio.no>
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Cc: rmcat WG <rmcat@ietf.org>
Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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Michael Welzl <michawe@ifi.uio.no> wrote:
> On 5. okt. 2012, at 00:31, John Leslie wrote:
>> Michael Welzl <michawe@ifi.uio.no> wrote:
>> 
>>> First, to clarify, I'm talking about per-packet priorities within a  
>>> stream. Second, my point is that congestion control could be done  
>>> better if packet priorities would be known, and could then be taken  
>>> into account.
>> 
>> I guess I'm missing why packet priorities would ever _not_ be the
>> domain of the sender -- even if advised by receiver preferences.
> 
> It's just not what I meant. I meant priorities influencing the
> congestion control behavior (I should say the "sender's" behavior,
> but this discussion is about having the congestion control logic
> on the sender or receiver side... so...).

   "Priorities influencing" is actually quite separable from packet
scheduling.

   I think we're hopelessly talking past each other here. It might
help if you outlined some "likely" mixes of priorities for packets.

>>> Right now, we are essentially facing greedy or non-greedy traffic,
>> 
>> Do you mean we're "contending against" those? If so, I agree.
> 
> I would agree with that too, but I meant: congestion control
> mechanism design is considering only these two classes of traffic.

   Alas, I don't understand that statement either (and it sounds as if
I would disagree if I did understand it).

>>> In fact a mechanism could opt for a more LEDBAT'ish behavior when
>>> its buffer is full of low-priority packets and be more aggressive
>>> when it has high-priority packets.
>> 
>> If you're saying that sending rate may deserve to increase faster
>> when there are higher priority packets to send, I could agree but
>> it seems unlikely such a situation will happen.
> 
> Yes that's what I'm saying. I don't understand why that would be
> unlikely?

   I expressed myself poorly, but I think we're suffering from the
lack of a shared vision of "priority" again...

>> If OTOH, the sender has _only_ higher priority packets, a more
>> aggressive sending rate _may_ be justified. Either way, it seems
>> unlikely to be a good idea to increase sending rate too aggressively
>> for low-priority packets...
> 
> Yes, exactly. We can be less aggressive when we have low-priority
> packets to send.

   In essence, I was trying to say that an increased aggressiveness
for sending high-priority packets shouldn't spill over into the rate
for low-priority packets. (I still think we're talking past each
other, though.)

>> To tell truth, I think it's premature to specify what will be
>> receiver-side vs. sender-side.
>> 
>> Especially, I think we need intelligence at both ends in order to
>> communicate useful congestion information in the two directions.
>> Once we have designed that we will be better able to specify the
>> control loops.
> 
> I hear you, but have a hard time imagining this without envisioning
> whether the control decision is made on the sender or receiver side,
> as this influences what needs to be signalled.

   I think it's way to early to talk bits on the wire.

> So I think it's a key architectural decision that we'd better get
> right from the outset. This is because having one side generic ("dumb")
> is probably very important for deployment.

   To the extent that _is_ important for deployment (IMHO) we'd need
to design for both sending _and_ receiving to be "dumb" on one side
of a two-way session.

   I don't think of optimizing for "one side dumb" as important for
deployment, though I would certainly agree that fallback when one side
turns out to be "dumb" is critical.

> Look at TCP, with its many congestion control flavors in Linux' or
> FreeBSD's pluggable congestion control. If I run a TCP based (e.g. web)
> server, I can just pick a mechanism with a simple command. No need to
> hope for clients to support it; no need to signal to the client what
> mechanism I use. No need to even standardize it, as we had to learn
> the hard way...

   Those are all sender-based-control. Are you arguing we should give
up considering receiver-based-control?

> In case of rmcat in rtcweb, this would mean that browser vendor X
> could simply start playing with a new congestion control mechanism
> (offer it to its users or enable it by default), and that would work
> with all other rtcweb/rmcat browsers, without ever having support for
> this mechanism from browser vendor Y or Z.

   That indeed is a worthy goal. But do understand that X, Y, and Z
all will be both sender and receiver.

> If we want that to be possible, we need to decide which side should
> be the generic one at some point, and then agree on the exact
> functionality of this generic side.

   I don't agree that's necessary or sufficient. OTOH, I _do_ believe
we need to establish a feedback mechanism with a clear definition
of what the feedback means (in both directions, of course).

>> (Assuming ConEx ever finishes, we are likely to see some kind of
>> congestion-based pricing both for senders and receivers -- and the
>> control loops ideally would serve the price-based needs of both
>> senders and receivers.)

   I expressed myself poorly again. We will (IMHO) see congestion-
based pricing for both senders and receivers anyway -- it's happening
already, though the measure of "congestion" is very poor.

   I meant to indicate that, even if absent from the first version,
our control loops should allow receivers to request "lower congestion"
and senders to moderate their sending rate to avoid "congestion"
charges.

> I would prefer to leave this consideration aside, because
> 1) rmcat works on a much tighter schedule than ConEx, as it will
>    probably be deployed in the rtcweb context fast, while I don't
>    know how long we'd have to wait for reasonably wide-spread ConEx
>    deployment...

   Agreed: we may not see widespread ConEx deployment in our lifetime.

> 2) I don't think that this matters much for our decision - in the
>    worst case, we would find that ConEx related information needs
>    to be signalled from one end to the other, and then we could
>    consider extending to whatever signalling scheme we define.

   Although there may (someday) be value to selective ConEx marking
by the sender, it's way premature to talk about such bits on the wire.

--
John Leslie <john@jlc.net>

From michawe@ifi.uio.no  Sun Oct  7 06:18:08 2012
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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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Hi,

Very sorry for any disadvantages that might have come from having  
expressed myself unclear. I'll try to clean things up in this email now:


On Oct 5, 2012, at 4:58 PM, John Leslie wrote:

> Michael Welzl <michawe@ifi.uio.no> wrote:
>> On 5. okt. 2012, at 00:31, John Leslie wrote:
>>> Michael Welzl <michawe@ifi.uio.no> wrote:
>>>
>>>> First, to clarify, I'm talking about per-packet priorities within a
>>>> stream. Second, my point is that congestion control could be done
>>>> better if packet priorities would be known, and could then be taken
>>>> into account.
>>>
>>> I guess I'm missing why packet priorities would ever _not_ be the
>>> domain of the sender -- even if advised by receiver preferences.
>>
>> It's just not what I meant. I meant priorities influencing the
>> congestion control behavior (I should say the "sender's" behavior,
>> but this discussion is about having the congestion control logic
>> on the sender or receiver side... so...).
>
>   "Priorities influencing" is actually quite separable from packet
> scheduling.
>
>   I think we're hopelessly talking past each other here. It might
> help if you outlined some "likely" mixes of priorities for packets.

Maybe you're considering one type of priority and me another.

Indeed, in rmcat we're facing two kinds of priorities. One, priorities  
between streams. I think we *must* be able to support these priorities  
because they are required by rtcweb (A23 in draft-ietf-rtcweb-use- 
cases-and-requirements). In case of rtcweb, these priorities are under  
control of the user or at least the web site programmer via the  
javascript API, and should therefore probably be expected to be  
reasonably long-lived.

Second (and this is what I meant in this current discussion),  
individual packets in a multimedia stream differ in importance. One  
easy way to say this is that a video consists of e.g. I-frames, P- 
frames and B-frames, where the priority is I > P > B. In reality,  
things are more complex: a frame typically spans multiple packets, but  
there are blocks in frames of different type (e.g. a P-frame may  
contain I- P- and B-blocks), and then there is also a dependency  
chain: some blocks may have many other blocks depend upon them, making  
them more important, and others may have less, making them less  
important. Honoring these priorities is important for the receiver's  
subjective quality - here's one out of many, many existing  
publications documenting this in one way or another:
Michael Schier, Michael Welzl: "Optimizing Selective ARQ for H.264  
Live Streaming: A Novel Method for Predicting Loss-Impact in  
Realtime", IEEE Transactions on Multimedia 14(2), April 2012.
http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=6095636

Suffice to say, we can have different priorities per packet as a  
function of the multimedia encoding, and these priorities could  
quickly change (this is a way to represent the dependence - if the  
sending application is in control of its data until the last moment,  
and if it is informed about the loss of packet X, it could quickly  
assign a very low priority to packets that depend on X.

Supporting these priorities in rmcat doesn't seem to be a MUST in  
accordance with the charter and rtcweb requirements. I am arguing that  
it would be great if we could support them.

What I mean with supporting them via congestion control is: going back  
to the overly simplistic example of I- P- and B-"packets", one may  
decide to be less aggressive when the buffer is full of "B-packets"  
and more aggressive when it is full of "I-packets". One out of many  
possibilities... my point is, such functionality should be enabled by  
handing over the right information to the congestion controller, and  
this requires the controller to be close to the send buffer, or the  
information to be signalled to the other side. Both can be okay, but  
we should be clear about this.


>>>> Right now, we are essentially facing greedy or non-greedy traffic,
>>>
>>> Do you mean we're "contending against" those? If so, I agree.
>>
>> I would agree with that too, but I meant: congestion control
>> mechanism design is considering only these two classes of traffic.
>
>   Alas, I don't understand that statement either (and it sounds as if
> I would disagree if I did understand it).

Let's leave it at that, this bit doesn't really matter for the  
discussion anyway.


>>>> In fact a mechanism could opt for a more LEDBAT'ish behavior when
>>>> its buffer is full of low-priority packets and be more aggressive
>>>> when it has high-priority packets.
>>>
>>> If you're saying that sending rate may deserve to increase faster
>>> when there are higher priority packets to send, I could agree but
>>> it seems unlikely such a situation will happen.
>>
>> Yes that's what I'm saying. I don't understand why that would be
>> unlikely?
>
>   I expressed myself poorly, but I think we're suffering from the
> lack of a shared vision of "priority" again...

Perhaps; I hope my explanation above helps.


>>> If OTOH, the sender has _only_ higher priority packets, a more
>>> aggressive sending rate _may_ be justified. Either way, it seems
>>> unlikely to be a good idea to increase sending rate too aggressively
>>> for low-priority packets...
>>
>> Yes, exactly. We can be less aggressive when we have low-priority
>> packets to send.
>
>   In essence, I was trying to say that an increased aggressiveness
> for sending high-priority packets shouldn't spill over into the rate
> for low-priority packets. (I still think we're talking past each
> other, though.)
>
>>> To tell truth, I think it's premature to specify what will be
>>> receiver-side vs. sender-side.
>>>
>>> Especially, I think we need intelligence at both ends in order to
>>> communicate useful congestion information in the two directions.
>>> Once we have designed that we will be better able to specify the
>>> control loops.
>>
>> I hear you, but have a hard time imagining this without envisioning
>> whether the control decision is made on the sender or receiver side,
>> as this influences what needs to be signalled.
>
>   I think it's way to early to talk bits on the wire.
>
>> So I think it's a key architectural decision that we'd better get
>> right from the outset. This is because having one side generic  
>> ("dumb")
>> is probably very important for deployment.
>
>   To the extent that _is_ important for deployment (IMHO) we'd need
> to design for both sending _and_ receiving to be "dumb" on one side
> of a two-way session.
>
>   I don't think of optimizing for "one side dumb" as important for
> deployment, though I would certainly agree that fallback when one side
> turns out to be "dumb" is critical.

See immediately below, I think this is clearer:


>> Look at TCP, with its many congestion control flavors in Linux' or
>> FreeBSD's pluggable congestion control. If I run a TCP based (e.g.  
>> web)
>> server, I can just pick a mechanism with a simple command. No need to
>> hope for clients to support it; no need to signal to the client what
>> mechanism I use. No need to even standardize it, as we had to learn
>> the hard way...
>
>   Those are all sender-based-control. Are you arguing we should give
> up considering receiver-based-control?

I'm arguing that we should have either this, or strictly only-receiver- 
based control. Then we could get the same functionality: you could  
just change the receiver to use a new mechanism X, without waiting for  
the sender to support the new mechanism X or even signalling to the  
sender what mechanism to use. Sender- or receiver- based, but not both.

Note that this doesn't exclude two-way behavior at all. To get back to  
the example of TCP and sender-based control, it's absolutely possible  
to have two way communication on a connection there, where one side  
uses say CUBIC congestion control and the other Compound TCP. Each  
side doesn't require any specific behavior from the other, because it  
simply expects the other side to show the simple generic receiver  
behavior. This is what I'd like to see in rmcat.

(again, the situation is the same for the case of receiver-based  
control with a generic, generally agreed upon sender behavior - but  
spreading each congestion controller over both sides ruins this ability)


>> In case of rmcat in rtcweb, this would mean that browser vendor X
>> could simply start playing with a new congestion control mechanism
>> (offer it to its users or enable it by default), and that would work
>> with all other rtcweb/rmcat browsers, without ever having support for
>> this mechanism from browser vendor Y or Z.
>
>   That indeed is a worthy goal. But do understand that X, Y, and Z
> all will be both sender and receiver.

See just above, I hope this was clear enough.


>> If we want that to be possible, we need to decide which side should
>> be the generic one at some point, and then agree on the exact
>> functionality of this generic side.
>
>   I don't agree that's necessary or sufficient. OTOH, I _do_ believe
> we need to establish a feedback mechanism with a clear definition
> of what the feedback means (in both directions, of course).

I do believe it's necessary (sufficient, no - indeed we'd need a  
clearly defined feedback mechanism). For example, with strict receiver- 
based control, only having a clearly defined feedback mechanism will  
not make flexible use of controls as with TCP possible. Or, in other  
words: having e.g. RRTCC's sender behavior but multiply defined  
receiver behaviors would make this flexibility possible, but having  
multiple sender- AND receiver-behaviors will not.


>>> (Assuming ConEx ever finishes, we are likely to see some kind of
>>> congestion-based pricing both for senders and receivers -- and the
>>> control loops ideally would serve the price-based needs of both
>>> senders and receivers.)
>
>   I expressed myself poorly again. We will (IMHO) see congestion-
> based pricing for both senders and receivers anyway -- it's happening
> already, though the measure of "congestion" is very poor.
>
>   I meant to indicate that, even if absent from the first version,
> our control loops should allow receivers to request "lower congestion"
> and senders to moderate their sending rate to avoid "congestion"
> charges.

Agreed, but I don't see a problem in doing this with either sender- or  
receiver-based control.


>> I would prefer to leave this consideration aside, because
>> 1) rmcat works on a much tighter schedule than ConEx, as it will
>>   probably be deployed in the rtcweb context fast, while I don't
>>   know how long we'd have to wait for reasonably wide-spread ConEx
>>   deployment...
>
>   Agreed: we may not see widespread ConEx deployment in our lifetime.
>
>> 2) I don't think that this matters much for our decision - in the
>>   worst case, we would find that ConEx related information needs
>>   to be signalled from one end to the other, and then we could
>>   consider extending to whatever signalling scheme we define.
>
>   Although there may (someday) be value to selective ConEx marking
> by the sender, it's way premature to talk about such bits on the wire.
>
> --
> John Leslie <john@jlc.net>


Cheers,
Michael


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To: Michael Welzl <michawe@ifi.uio.no>, rmcat WG <rmcat@ietf.org>
Thread-Topic: [rmcat] Piggybacking
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Subject: Re: [rmcat] Piggybacking
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Hi

This brings up old memories (OK, not that old)
I proposed something very 3GPP AMR specific ~6 years ago.=20
http://tools.ietf.org/html/draft-johansson-avt-rtp-shim-01=20
The message in the AVT WG was then to define a new RTP profile or use RTCP,=
 we ended up with reduced size RTCP (RFC5506)

/Ingemar

-----Original Message-----
From: Michael Welzl [mailto:michawe@ifi.uio.no]=20
Sent: den 4 oktober 2012 12:44
To: rmcat WG
Subject: Re: [rmcat] Piggybacking

Hi,


On 3. okt. 2012, at 17:00, John Leslie wrote:

> Randell Jesup <randell-ietf@jesup.org> wrote:
>> On 9/24/2012 10:46 AM, Harald Alvestrand wrote:
>>>=20
>>> In the TCP case, the feedback mechanism is the ack, which is needed=20
>>> for reasons beyond congestion control. In the RTP case, there isn't=20
>>> such a single, simple mechanism needed for other reasons.
>>=20
>> Agreed, though Michael indicated that we can piggyback congestion=20
>> info on reverse traffic.
>>=20
>> This may be possible and I've thought about it.  Part of the problem=20
>> is that if you want to signal data back quickly, you have to wait up=20
>> to 20ms and in some cases longer to piggyback (and on=20
>> low-upstream-bandwidth links, the uplink transit time if you're=20
>> piggybacking on a 1500-byte video packet might be 20+ms).
>=20
>   This is missing the point, IMHO.
>=20
>   We are dealing in IP packets -- "unreliable" packets: meaning that=20
> we aren't entitled to expect any particular packet to be delivered at=20
> all.
>=20
>   The signal we wish to send is tiny -- probably on the order of one=20
> byte. If we allocate one byte per packet for feedback, it's almost=20
> lost in the noise.
>=20
>   But if we expect that byte and it doesn't arrive, that's a clear=20
> congestion signal.
>=20
>   (Granted its loss signals congestion in the path _to_ us...)
>=20
>   Trying to optimize 20 msec of reaction time for real-time traffic=20
> isn't the right paradigm. We want real-time traffic to adapt more=20
> slowly than web-surfing traffic, but stay low for longer (to avoid=20
> self-congestion).
>=20
>   Thus, we can be pretty sure that the absence of an expected feedback=20
> justifies a delay in any ramp-up of outgoing bit-rate currently planned.

Indeed, I'm not sure a quick signal is always needed. Either way, as John s=
ays, a packet may be lost and the signal is tiny. We could agree to always =
piggyback - if the signal really can't wait, we can send it immediately but=
 include it piggybacked onto the next datapacket too, for redundancy, makin=
g the whole thing more stable in case a feedback packet gets lost.


>   Beyond that, it gets more complicated. Nonetheless, we can easily=20
> signal in our next scheduled outgoing packet that we missed an=20
> expected feedback; and that signal (or the failure to receive our next=20
> outgoing packeti) can trigger a retransmission -- catching up=20
> typically faster than TCP can do.

I don't get that, and it sounds complicated and inefficient - you're adding=
 round-trips here?


>   I'd like to suggest that we plan RMCAT around bidirectional (though=20
> possibly very small in one direction) media packets. There

> aren't many cases where the traffic is _so_ unidirectional that an=20
> "I'm still interested in this stream" isn't appropriate.
> (I don't mean to imply that the packet-rate needs to be the same in=20
> both directions, least of all the bit-rate.)
>=20
>> If you do piggyback, it has to be optional on a instance-by-instance=20
>> basis
>=20
>   This is precisely what I _don't_ want to see. Perhaps the size of=20
> the feedback might vary (though I doubt that's needed); but I want the=20
> _absence_ of feedback to be a primary signal.
>=20
>> (perhaps even estimating when the next chance will be, and the=20
>> priority of the data - but that speaks against a dumb receiver).
>=20
>   (I don't think it makes sense to even think about a "dumb receiver.)

Call it a *generic* receiver. That's what was meant in this discussion.

The problem is really that it's a (major, I think) benefit for deployment t=
o have one side generic, and a generic sender is a more complicated setting=
 than a generic receiver... in my opinion, none of the two choices are real=
ly ideal. To have things combined in the right place, I'll address this in =
a separate email in which I extend my previous analysis of the sender/recei=
ver situation.


>> TCP can "get away" with high packet count feedback because flows are=20
>> typically largely unidirectional or alternating flows, or on=20
>> well-connected devices, not edge nodes.  The PacketCable/etc issue=20
>> with shared upstreams and slot allocation is an example of reacting=20
>> to this - there even with the typically largely downstream flows, the=20
>> upstream acks become a limiting factor.
>=20
>   Yes!
>=20
>   This issue is very real in cable-upstream, and also significant in=20
> many wireless setups.
>=20
>   But the right approach is to measure "upstream" congestion and adapt=20
> our packet rate to it. Really-tiny upstream packet aren't much "cheaper"
> than medium-sized upstream packets.

That was my argument too; it does add some complexity though. See my next e=
mail on sender vs receiver...

As for John's comments below, I agree with many of them.


>   Granted, it's easy to confuse upstream-congestion with downstream-=20
> congestion; but it's not _that_ hard to separate them.
>=20
>>> Stickler: Amount of feedback has little correlation with packet drop.=20
>>> If feedback is lost due to congestion, more feedback will lead to=20
>>> more packet drops. What is true is that when feedback occurs more=20
>>> rarely, losing one feedback packet will lead to a greater gap=20
>>> between feedback packets.
>=20
>   This is true, but some possible inferences are not true...
>=20
>> I'd say less feedback means the odds that a drop will hit a feedback=20
>> packet are *lower* (as routers typically either tail-drop or=20
>> random-drop, but in either case in a static evaluation a=20
>> lower-bandwidth/packet flow (i.e. the feedback packets) will get hit=20
>> less often), but as Harald says the impact is higher.  And in a=20
>> dynamic evaluation this may shift some.
>=20
>   In particular, inferring that we must do slavish AIMD is not true.
>=20
>>>> - receiver-side-congestion-control-based feedback reduction as in=20
>>>> RRTCC means to reduce the feedback whenever feedback isn't urgent=20
>>>> for the sender. I'd argue that, in fact, we should reduce feedback=20
>>>> only when the channel carrying it (the backwards channel) is=20
>>>> congested, and that is a generic function, not bound to the=20
>>>> sender-side congestion control behavior.
>=20
>   I agree that reducing feedback when the feedback path is not=20
> congested really doesn't help.
>=20
>   (When we _do_ need to reduce feedback, I suggest we need to lower=20
> the packet-rate, letting the chips fall where they may; and be slow to=20
> increase it again.)
>=20
>> Personally, I assume both channels are always congested, or will be=20
>> soon (if we're doing our job right and aren't on a huge link),
>=20
>   The assumption isn't particularly bad; but I envision many cases=20
> where it isn't true.
>=20
>> so bandwidth reduction for feedback always allows more bandwidth for=20
>> useful data.
>=20
>   While technically true, it's a bad tradeoff to discard information=20
> about congestion -- this leads to growing your bit-rate too fast, and=20
> more packet-loss than necessary.
>=20
>> ...=20
>> And my position is that any *good* two-way communication will run=20
>> near-but-not-quite-over the bandwidth limit as much of the time as=20
>> possible, so ACKs always have the potential to cause problems (and=20
>> always will force you a little further down the quality curve).
>=20
>   I don't agree with this conclusion.
>=20
>   With "ideal" feedback, we can be sure we don't initiate any=20
> congestion problems, while "minimal" feedback means some of our=20
> guesses will be wrong.
>=20
>   And I don't think we have a good understanding of "the quality=20
> curve" yet.
>=20
>   Some real-time stacks will run very close to the maximum data-rate,=20
> and hopefully will have good redundancy algorithms to reconstruct lost=20
> packets. Others will choose to stay below the maximum. We shouldn't=20
> prejudge that tradeoff, but limit ourselves to timely delivery of the=20
> best congestion information, and algorithms for rate-adjustment.
>=20
> --
> John Leslie <john@jlc.net>
> _______________________________________________
> rmcat mailing list
> rmcat@ietf.org
> https://www.ietf.org/mailman/listinfo/rmcat



From michawe@ifi.uio.no  Mon Oct  8 00:47:46 2012
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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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This gave me an extra thought:

> Second (and this is what I meant in this current discussion), =
individual packets in a multimedia stream differ in importance. One easy =
way to say this is that a video consists of e.g. I-frames, P-frames and =
B-frames, where the priority is I > P > B. In reality, things are more =
complex: a frame typically spans multiple packets, but there are blocks =
in frames of different type (e.g. a P-frame may contain I- P- and =
B-blocks), and then there is also a dependency chain: some blocks may =
have many other blocks depend upon them, making them more important, and =
others may have less, making them less important. Honoring these =
priorities is important for the receiver's subjective quality - here's =
one out of many, many existing publications documenting this in one way =
or another:
> Michael Schier, Michael Welzl: "Optimizing Selective ARQ for H.264 =
Live Streaming: A Novel Method for Predicting Loss-Impact in Realtime", =
IEEE Transactions on Multimedia 14(2), April 2012.
> http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=3D6095636

I don't know how many systems do these sort of thing in practice, but I =
know that there are lots and lots of papers on such content-aware =
mechanisms. What they tend to do is either ARQ or FEC, and yes, these =
things can operate in real-time (also the mechanism in our paper above =
was designed for that), i.e. they work for interactive media.

For this stuff to work, what is needed at the sender is precise =
knowledge about which packets have been lost, as quickly as possible. =
This means that, if we go for a =
receiver-side-congestion-control-sending-as-little-feedback-as-possible =
scheme, we should have "special" feedback whenever loss happens: provide =
fine-grain information about exactly which packets were lost, and =
probably increase the feedback frequency.

Given that this is about loss on the sender=3D>receiver path which may =
not be affected by increased traffic on the receiver=3D>sender path, I =
don't see a big problem in doing this. It just complicates the =
minimum-feedback-due-to-receiver-side-cc. case somewhat, I suppose.


Cheers,
Michael


From holmer@google.com  Mon Oct  8 04:43:12 2012
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From: Stefan Holmer <stefan@webrtc.org>
To: Michael Welzl <michawe@ifi.uio.no>
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Cc: rmcat WG <rmcat@ietf.org>, John Leslie <john@jlc.net>
Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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On Sun, Oct 7, 2012 at 3:17 PM, Michael Welzl <michawe@ifi.uio.no> wrote:

> Hi,
>
> Very sorry for any disadvantages that might have come from having
> expressed myself unclear. I'll try to clean things up in this email now:
>
>
>
> On Oct 5, 2012, at 4:58 PM, John Leslie wrote:
>
>  Michael Welzl <michawe@ifi.uio.no> wrote:
>>
>>> On 5. okt. 2012, at 00:31, John Leslie wrote:
>>>
>>>> Michael Welzl <michawe@ifi.uio.no> wrote:
>>>>
>>>>  First, to clarify, I'm talking about per-packet priorities within a
>>>>> stream. Second, my point is that congestion control could be done
>>>>> better if packet priorities would be known, and could then be taken
>>>>> into account.
>>>>>
>>>>
>>>> I guess I'm missing why packet priorities would ever _not_ be the
>>>> domain of the sender -- even if advised by receiver preferences.
>>>>
>>>
>>> It's just not what I meant. I meant priorities influencing the
>>> congestion control behavior (I should say the "sender's" behavior,
>>> but this discussion is about having the congestion control logic
>>> on the sender or receiver side... so...).
>>>
>>
>>   "Priorities influencing" is actually quite separable from packet
>> scheduling.
>>
>>   I think we're hopelessly talking past each other here. It might
>> help if you outlined some "likely" mixes of priorities for packets.
>>
>
> Maybe you're considering one type of priority and me another.
>
> Indeed, in rmcat we're facing two kinds of priorities. One, priorities
> between streams. I think we *must* be able to support these priorities
> because they are required by rtcweb (A23 in draft-ietf-rtcweb-use-cases-**and-requirements).
> In case of rtcweb, these priorities are under control of the user or at
> least the web site programmer via the javascript API, and should therefore
> probably be expected to be reasonably long-lived.
>
> Second (and this is what I meant in this current discussion), individual
> packets in a multimedia stream differ in importance. One easy way to say
> this is that a video consists of e.g. I-frames, P-frames and B-frames,
> where the priority is I > P > B. In reality, things are more complex: a
> frame typically spans multiple packets, but there are blocks in frames of
> different type (e.g. a P-frame may contain I- P- and B-blocks), and then
> there is also a dependency chain: some blocks may have many other blocks
> depend upon them, making them more important, and others may have less,
> making them less important. Honoring these priorities is important for the
> receiver's subjective quality - here's one out of many, many existing
> publications documenting this in one way or another:
> Michael Schier, Michael Welzl: "Optimizing Selective ARQ for H.264 Live
> Streaming: A Novel Method for Predicting Loss-Impact in Realtime", IEEE
> Transactions on Multimedia 14(2), April 2012.
> http://ieeexplore.ieee.org/**xpls/abs_all.jsp?arnumber=**6095636<http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=6095636>
>
> Suffice to say, we can have different priorities per packet as a function
> of the multimedia encoding, and these priorities could quickly change (this
> is a way to represent the dependence - if the sending application is in
> control of its data until the last moment, and if it is informed about the
> loss of packet X, it could quickly assign a very low priority to packets
> that depend on X.
>
> Supporting these priorities in rmcat doesn't seem to be a MUST in
> accordance with the charter and rtcweb requirements. I am arguing that it
> would be great if we could support them.
>
> What I mean with supporting them via congestion control is: going back to
> the overly simplistic example of I- P- and B-"packets", one may decide to
> be less aggressive when the buffer is full of "B-packets" and more
> aggressive when it is full of "I-packets". One out of many possibilities...
> my point is, such functionality should be enabled by handing over the right
> information to the congestion controller, and this requires the controller
> to be close to the send buffer, or the information to be signalled to the
> other side. Both can be okay, but we should be clear about this.


You want to allow the sender to, for a short time, override the bandwidth
estimate by sending at a higher rate because it has important packets to
send? Isn't this possible even though the estimator is at the receiver? I'm
also not sure that is something you want to do. Sending important packets
at a higher rate increases the risk of losing those important packets,
which is the opposite of what you want, right?


>
>
>
>  Right now, we are essentially facing greedy or non-greedy traffic,
>>>>>
>>>>
>>>> Do you mean we're "contending against" those? If so, I agree.
>>>>
>>>
>>> I would agree with that too, but I meant: congestion control
>>> mechanism design is considering only these two classes of traffic.
>>>
>>
>>   Alas, I don't understand that statement either (and it sounds as if
>> I would disagree if I did understand it).
>>
>
> Let's leave it at that, this bit doesn't really matter for the discussion
> anyway.
>
>
>
>  In fact a mechanism could opt for a more LEDBAT'ish behavior when
>>>>> its buffer is full of low-priority packets and be more aggressive
>>>>> when it has high-priority packets.
>>>>>
>>>>
>>>> If you're saying that sending rate may deserve to increase faster
>>>> when there are higher priority packets to send, I could agree but
>>>> it seems unlikely such a situation will happen.
>>>>
>>>
>>> Yes that's what I'm saying. I don't understand why that would be
>>> unlikely?
>>>
>>
>>   I expressed myself poorly, but I think we're suffering from the
>> lack of a shared vision of "priority" again...
>>
>
> Perhaps; I hope my explanation above helps.
>
>
>
>  If OTOH, the sender has _only_ higher priority packets, a more
>>>> aggressive sending rate _may_ be justified. Either way, it seems
>>>> unlikely to be a good idea to increase sending rate too aggressively
>>>> for low-priority packets...
>>>>
>>>
>>> Yes, exactly. We can be less aggressive when we have low-priority
>>> packets to send.
>>>
>>
>>   In essence, I was trying to say that an increased aggressiveness
>> for sending high-priority packets shouldn't spill over into the rate
>> for low-priority packets. (I still think we're talking past each
>> other, though.)
>>
>>  To tell truth, I think it's premature to specify what will be
>>>> receiver-side vs. sender-side.
>>>>
>>>> Especially, I think we need intelligence at both ends in order to
>>>> communicate useful congestion information in the two directions.
>>>> Once we have designed that we will be better able to specify the
>>>> control loops.
>>>>
>>>
>>> I hear you, but have a hard time imagining this without envisioning
>>> whether the control decision is made on the sender or receiver side,
>>> as this influences what needs to be signalled.
>>>
>>
>>   I think it's way to early to talk bits on the wire.
>>
>>  So I think it's a key architectural decision that we'd better get
>>> right from the outset. This is because having one side generic ("dumb")
>>> is probably very important for deployment.
>>>
>>
>>   To the extent that _is_ important for deployment (IMHO) we'd need
>> to design for both sending _and_ receiving to be "dumb" on one side
>> of a two-way session.
>>
>>   I don't think of optimizing for "one side dumb" as important for
>> deployment, though I would certainly agree that fallback when one side
>> turns out to be "dumb" is critical.
>>
>
> See immediately below, I think this is clearer:
>
>
>
>  Look at TCP, with its many congestion control flavors in Linux' or
>>> FreeBSD's pluggable congestion control. If I run a TCP based (e.g. web)
>>> server, I can just pick a mechanism with a simple command. No need to
>>> hope for clients to support it; no need to signal to the client what
>>> mechanism I use. No need to even standardize it, as we had to learn
>>> the hard way...
>>>
>>
>>   Those are all sender-based-control. Are you arguing we should give
>> up considering receiver-based-control?
>>
>
> I'm arguing that we should have either this, or strictly
> only-receiver-based control. Then we could get the same functionality: you
> could just change the receiver to use a new mechanism X, without waiting
> for the sender to support the new mechanism X or even signalling to the
> sender what mechanism to use. Sender- or receiver- based, but not both.
>
> Note that this doesn't exclude two-way behavior at all. To get back to the
> example of TCP and sender-based control, it's absolutely possible to have
> two way communication on a connection there, where one side uses say CUBIC
> congestion control and the other Compound TCP. Each side doesn't require
> any specific behavior from the other, because it simply expects the other
> side to show the simple generic receiver behavior. This is what I'd like to
> see in rmcat.
>
> (again, the situation is the same for the case of receiver-based control
> with a generic, generally agreed upon sender behavior - but spreading each
> congestion controller over both sides ruins this ability)
>
>
>
>  In case of rmcat in rtcweb, this would mean that browser vendor X
>>> could simply start playing with a new congestion control mechanism
>>> (offer it to its users or enable it by default), and that would work
>>> with all other rtcweb/rmcat browsers, without ever having support for
>>> this mechanism from browser vendor Y or Z.
>>>
>>
>>   That indeed is a worthy goal. But do understand that X, Y, and Z
>> all will be both sender and receiver.
>>
>
> See just above, I hope this was clear enough.
>
>
>
>  If we want that to be possible, we need to decide which side should
>>> be the generic one at some point, and then agree on the exact
>>> functionality of this generic side.
>>>
>>
>>   I don't agree that's necessary or sufficient. OTOH, I _do_ believe
>> we need to establish a feedback mechanism with a clear definition
>> of what the feedback means (in both directions, of course).
>>
>
> I do believe it's necessary (sufficient, no - indeed we'd need a clearly
> defined feedback mechanism). For example, with strict receiver-based
> control, only having a clearly defined feedback mechanism will not make
> flexible use of controls as with TCP possible. Or, in other words: having
> e.g. RRTCC's sender behavior but multiply defined receiver behaviors would
> make this flexibility possible, but having multiple sender- AND
> receiver-behaviors will not.
>
>
>
>  (Assuming ConEx ever finishes, we are likely to see some kind of
>>>> congestion-based pricing both for senders and receivers -- and the
>>>> control loops ideally would serve the price-based needs of both
>>>> senders and receivers.)
>>>>
>>>
>>   I expressed myself poorly again. We will (IMHO) see congestion-
>> based pricing for both senders and receivers anyway -- it's happening
>> already, though the measure of "congestion" is very poor.
>>
>>   I meant to indicate that, even if absent from the first version,
>> our control loops should allow receivers to request "lower congestion"
>> and senders to moderate their sending rate to avoid "congestion"
>> charges.
>>
>
> Agreed, but I don't see a problem in doing this with either sender- or
> receiver-based control.
>
>
>
>  I would prefer to leave this consideration aside, because
>>> 1) rmcat works on a much tighter schedule than ConEx, as it will
>>>   probably be deployed in the rtcweb context fast, while I don't
>>>   know how long we'd have to wait for reasonably wide-spread ConEx
>>>   deployment...
>>>
>>
>>   Agreed: we may not see widespread ConEx deployment in our lifetime.
>>
>>  2) I don't think that this matters much for our decision - in the
>>>   worst case, we would find that ConEx related information needs
>>>   to be signalled from one end to the other, and then we could
>>>   consider extending to whatever signalling scheme we define.
>>>
>>
>>   Although there may (someday) be value to selective ConEx marking
>> by the sender, it's way premature to talk about such bits on the wire.
>>
>> --
>> John Leslie <john@jlc.net>
>>
>
>
> Cheers,
> Michael
>
>

--14dae9340dc9c43a7804cb8ab9bf
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Sun, O=
ct 7, 2012 at 3:17 PM, Michael Welzl <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:michawe@ifi.uio.no" target=3D"_blank" class=3D"cremed">michawe@ifi.uio.no=
</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<br>
Very sorry for any disadvantages that might have come from having expressed=
 myself unclear. I&#39;ll try to clean things up in this email now:<div cla=
ss=3D"im"><br>
<br>
<br>
On Oct 5, 2012, at 4:58 PM, John Leslie wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Michael Welzl &lt;<a href=3D"mailto:michawe@ifi.uio.no" target=3D"_blank" c=
lass=3D"cremed">michawe@ifi.uio.no</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 5. okt. 2012, at 00:31, John Leslie wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Michael Welzl &lt;<a href=3D"mailto:michawe@ifi.uio.no" target=3D"_blank" c=
lass=3D"cremed">michawe@ifi.uio.no</a>&gt; wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
First, to clarify, I&#39;m talking about per-packet priorities within a<br>
stream. Second, my point is that congestion control could be done<br>
better if packet priorities would be known, and could then be taken<br>
into account.<br>
</blockquote>
<br>
I guess I&#39;m missing why packet priorities would ever _not_ be the<br>
domain of the sender -- even if advised by receiver preferences.<br>
</blockquote>
<br>
It&#39;s just not what I meant. I meant priorities influencing the<br>
congestion control behavior (I should say the &quot;sender&#39;s&quot; beha=
vior,<br>
but this discussion is about having the congestion control logic<br>
on the sender or receiver side... so...).<br>
</blockquote>
<br>
=A0 &quot;Priorities influencing&quot; is actually quite separable from pac=
ket<br>
scheduling.<br>
<br>
=A0 I think we&#39;re hopelessly talking past each other here. It might<br>
help if you outlined some &quot;likely&quot; mixes of priorities for packet=
s.<br>
</blockquote>
<br></div>
Maybe you&#39;re considering one type of priority and me another.<br>
<br>
Indeed, in rmcat we&#39;re facing two kinds of priorities. One, priorities =
between streams. I think we *must* be able to support these priorities beca=
use they are required by rtcweb (A23 in draft-ietf-rtcweb-use-cases-<u></u>=
and-requirements). In case of rtcweb, these priorities are under control of=
 the user or at least the web site programmer via the javascript API, and s=
hould therefore probably be expected to be reasonably long-lived.<br>

<br>
Second (and this is what I meant in this current discussion), individual pa=
ckets in a multimedia stream differ in importance. One easy way to say this=
 is that a video consists of e.g. I-frames, P-frames and B-frames, where th=
e priority is I &gt; P &gt; B. In reality, things are more complex: a frame=
 typically spans multiple packets, but there are blocks in frames of differ=
ent type (e.g. a P-frame may contain I- P- and B-blocks), and then there is=
 also a dependency chain: some blocks may have many other blocks depend upo=
n them, making them more important, and others may have less, making them l=
ess important. Honoring these priorities is important for the receiver&#39;=
s subjective quality - here&#39;s one out of many, many existing publicatio=
ns documenting this in one way or another:<br>

Michael Schier, Michael Welzl: &quot;Optimizing Selective ARQ for H.264 Liv=
e Streaming: A Novel Method for Predicting Loss-Impact in Realtime&quot;, I=
EEE Transactions on Multimedia 14(2), April 2012.<br>
<a href=3D"http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=3D6095636" =
target=3D"_blank" class=3D"cremed">http://ieeexplore.ieee.org/<u></u>xpls/a=
bs_all.jsp?arnumber=3D<u></u>6095636</a><br>
<br>
Suffice to say, we can have different priorities per packet as a function o=
f the multimedia encoding, and these priorities could quickly change (this =
is a way to represent the dependence - if the sending application is in con=
trol of its data until the last moment, and if it is informed about the los=
s of packet X, it could quickly assign a very low priority to packets that =
depend on X.<br>

<br>
Supporting these priorities in rmcat doesn&#39;t seem to be a MUST in accor=
dance with the charter and rtcweb requirements. I am arguing that it would =
be great if we could support them.<br>
<br>
What I mean with supporting them via congestion control is: going back to t=
he overly simplistic example of I- P- and B-&quot;packets&quot;, one may de=
cide to be less aggressive when the buffer is full of &quot;B-packets&quot;=
 and more aggressive when it is full of &quot;I-packets&quot;. One out of m=
any possibilities... my point is, such functionality should be enabled by h=
anding over the right information to the congestion controller, and this re=
quires the controller to be close to the send buffer, or the information to=
 be signalled to the other side. Both can be okay, but we should be clear a=
bout this.</blockquote>
<div><br></div><div>You want to allow the sender to, for a short time, over=
ride the bandwidth estimate by sending at a higher rate because it has impo=
rtant packets to send? Isn&#39;t this possible even though the estimator is=
 at the receiver? I&#39;m also not sure that is something you want to do. S=
ending important packets at a higher rate increases the risk of losing thos=
e important packets, which is the opposite of what you want, right?</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div class=3D"im"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Right now, we are essentially facing greedy or non-greedy traffic,<br>
</blockquote>
<br>
Do you mean we&#39;re &quot;contending against&quot; those? If so, I agree.=
<br>
</blockquote>
<br>
I would agree with that too, but I meant: congestion control<br>
mechanism design is considering only these two classes of traffic.<br>
</blockquote>
<br>
=A0 Alas, I don&#39;t understand that statement either (and it sounds as if=
<br>
I would disagree if I did understand it).<br>
</blockquote>
<br></div>
Let&#39;s leave it at that, this bit doesn&#39;t really matter for the disc=
ussion anyway.<div class=3D"im"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
In fact a mechanism could opt for a more LEDBAT&#39;ish behavior when<br>
its buffer is full of low-priority packets and be more aggressive<br>
when it has high-priority packets.<br>
</blockquote>
<br>
If you&#39;re saying that sending rate may deserve to increase faster<br>
when there are higher priority packets to send, I could agree but<br>
it seems unlikely such a situation will happen.<br>
</blockquote>
<br>
Yes that&#39;s what I&#39;m saying. I don&#39;t understand why that would b=
e<br>
unlikely?<br>
</blockquote>
<br>
=A0 I expressed myself poorly, but I think we&#39;re suffering from the<br>
lack of a shared vision of &quot;priority&quot; again...<br>
</blockquote>
<br></div>
Perhaps; I hope my explanation above helps.<div><div class=3D"h5"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">

If OTOH, the sender has _only_ higher priority packets, a more<br>
aggressive sending rate _may_ be justified. Either way, it seems<br>
unlikely to be a good idea to increase sending rate too aggressively<br>
for low-priority packets...<br>
</blockquote>
<br>
Yes, exactly. We can be less aggressive when we have low-priority<br>
packets to send.<br>
</blockquote>
<br>
=A0 In essence, I was trying to say that an increased aggressiveness<br>
for sending high-priority packets shouldn&#39;t spill over into the rate<br=
>
for low-priority packets. (I still think we&#39;re talking past each<br>
other, though.)<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
To tell truth, I think it&#39;s premature to specify what will be<br>
receiver-side vs. sender-side.<br>
<br>
Especially, I think we need intelligence at both ends in order to<br>
communicate useful congestion information in the two directions.<br>
Once we have designed that we will be better able to specify the<br>
control loops.<br>
</blockquote>
<br>
I hear you, but have a hard time imagining this without envisioning<br>
whether the control decision is made on the sender or receiver side,<br>
as this influences what needs to be signalled.<br>
</blockquote>
<br>
=A0 I think it&#39;s way to early to talk bits on the wire.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
So I think it&#39;s a key architectural decision that we&#39;d better get<b=
r>
right from the outset. This is because having one side generic (&quot;dumb&=
quot;)<br>
is probably very important for deployment.<br>
</blockquote>
<br>
=A0 To the extent that _is_ important for deployment (IMHO) we&#39;d need<b=
r>
to design for both sending _and_ receiving to be &quot;dumb&quot; on one si=
de<br>
of a two-way session.<br>
<br>
=A0 I don&#39;t think of optimizing for &quot;one side dumb&quot; as import=
ant for<br>
deployment, though I would certainly agree that fallback when one side<br>
turns out to be &quot;dumb&quot; is critical.<br>
</blockquote>
<br></div></div>
See immediately below, I think this is clearer:<div class=3D"im"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Look at TCP, with its many congestion control flavors in Linux&#39; or<br>
FreeBSD&#39;s pluggable congestion control. If I run a TCP based (e.g. web)=
<br>
server, I can just pick a mechanism with a simple command. No need to<br>
hope for clients to support it; no need to signal to the client what<br>
mechanism I use. No need to even standardize it, as we had to learn<br>
the hard way...<br>
</blockquote>
<br>
=A0 Those are all sender-based-control. Are you arguing we should give<br>
up considering receiver-based-control?<br>
</blockquote>
<br></div>
I&#39;m arguing that we should have either this, or strictly only-receiver-=
based control. Then we could get the same functionality: you could just cha=
nge the receiver to use a new mechanism X, without waiting for the sender t=
o support the new mechanism X or even signalling to the sender what mechani=
sm to use. Sender- or receiver- based, but not both.<br>

<br>
Note that this doesn&#39;t exclude two-way behavior at all. To get back to =
the example of TCP and sender-based control, it&#39;s absolutely possible t=
o have two way communication on a connection there, where one side uses say=
 CUBIC congestion control and the other Compound TCP. Each side doesn&#39;t=
 require any specific behavior from the other, because it simply expects th=
e other side to show the simple generic receiver behavior. This is what I&#=
39;d like to see in rmcat.<br>

<br>
(again, the situation is the same for the case of receiver-based control wi=
th a generic, generally agreed upon sender behavior - but spreading each co=
ngestion controller over both sides ruins this ability)<div class=3D"im">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
In case of rmcat in rtcweb, this would mean that browser vendor X<br>
could simply start playing with a new congestion control mechanism<br>
(offer it to its users or enable it by default), and that would work<br>
with all other rtcweb/rmcat browsers, without ever having support for<br>
this mechanism from browser vendor Y or Z.<br>
</blockquote>
<br>
=A0 That indeed is a worthy goal. But do understand that X, Y, and Z<br>
all will be both sender and receiver.<br>
</blockquote>
<br></div>
See just above, I hope this was clear enough.<div class=3D"im"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
If we want that to be possible, we need to decide which side should<br>
be the generic one at some point, and then agree on the exact<br>
functionality of this generic side.<br>
</blockquote>
<br>
=A0 I don&#39;t agree that&#39;s necessary or sufficient. OTOH, I _do_ beli=
eve<br>
we need to establish a feedback mechanism with a clear definition<br>
of what the feedback means (in both directions, of course).<br>
</blockquote>
<br></div>
I do believe it&#39;s necessary (sufficient, no - indeed we&#39;d need a cl=
early defined feedback mechanism). For example, with strict receiver-based =
control, only having a clearly defined feedback mechanism will not make fle=
xible use of controls as with TCP possible. Or, in other words: having e.g.=
 RRTCC&#39;s sender behavior but multiply defined receiver behaviors would =
make this flexibility possible, but having multiple sender- AND receiver-be=
haviors will not.<div class=3D"im">
<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">

(Assuming ConEx ever finishes, we are likely to see some kind of<br>
congestion-based pricing both for senders and receivers -- and the<br>
control loops ideally would serve the price-based needs of both<br>
senders and receivers.)<br>
</blockquote></blockquote>
<br>
=A0 I expressed myself poorly again. We will (IMHO) see congestion-<br>
based pricing for both senders and receivers anyway -- it&#39;s happening<b=
r>
already, though the measure of &quot;congestion&quot; is very poor.<br>
<br>
=A0 I meant to indicate that, even if absent from the first version,<br>
our control loops should allow receivers to request &quot;lower congestion&=
quot;<br>
and senders to moderate their sending rate to avoid &quot;congestion&quot;<=
br>
charges.<br>
</blockquote>
<br></div>
Agreed, but I don&#39;t see a problem in doing this with either sender- or =
receiver-based control.<div class=3D"im"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I would prefer to leave this consideration aside, because<br>
1) rmcat works on a much tighter schedule than ConEx, as it will<br>
=A0 probably be deployed in the rtcweb context fast, while I don&#39;t<br>
=A0 know how long we&#39;d have to wait for reasonably wide-spread ConEx<br=
>
=A0 deployment...<br>
</blockquote>
<br>
=A0 Agreed: we may not see widespread ConEx deployment in our lifetime.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
2) I don&#39;t think that this matters much for our decision - in the<br>
=A0 worst case, we would find that ConEx related information needs<br>
=A0 to be signalled from one end to the other, and then we could<br>
=A0 consider extending to whatever signalling scheme we define.<br>
</blockquote>
<br>
=A0 Although there may (someday) be value to selective ConEx marking<br>
by the sender, it&#39;s way premature to talk about such bits on the wire.<=
br>
<br>
--<br>
John Leslie &lt;<a href=3D"mailto:john@jlc.net" target=3D"_blank" class=3D"=
cremed">john@jlc.net</a>&gt;<br>
</blockquote>
<br>
<br></div>
Cheers,<br>
Michael<br>
<br>
</blockquote></div><br></div>

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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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On 8. okt. 2012, at 13:43, Stefan Holmer wrote:

>=20
>=20
>=20
> On Sun, Oct 7, 2012 at 3:17 PM, Michael Welzl <michawe@ifi.uio.no> =
wrote:
> Hi,
>=20
> Very sorry for any disadvantages that might have come from having =
expressed myself unclear. I'll try to clean things up in this email now:
>=20
>=20
>=20
> On Oct 5, 2012, at 4:58 PM, John Leslie wrote:
>=20
> Michael Welzl <michawe@ifi.uio.no> wrote:
> On 5. okt. 2012, at 00:31, John Leslie wrote:
> Michael Welzl <michawe@ifi.uio.no> wrote:
>=20
> First, to clarify, I'm talking about per-packet priorities within a
> stream. Second, my point is that congestion control could be done
> better if packet priorities would be known, and could then be taken
> into account.
>=20
> I guess I'm missing why packet priorities would ever _not_ be the
> domain of the sender -- even if advised by receiver preferences.
>=20
> It's just not what I meant. I meant priorities influencing the
> congestion control behavior (I should say the "sender's" behavior,
> but this discussion is about having the congestion control logic
> on the sender or receiver side... so...).
>=20
>   "Priorities influencing" is actually quite separable from packet
> scheduling.
>=20
>   I think we're hopelessly talking past each other here. It might
> help if you outlined some "likely" mixes of priorities for packets.
>=20
> Maybe you're considering one type of priority and me another.
>=20
> Indeed, in rmcat we're facing two kinds of priorities. One, priorities =
between streams. I think we *must* be able to support these priorities =
because they are required by rtcweb (A23 in =
draft-ietf-rtcweb-use-cases-and-requirements). In case of rtcweb, these =
priorities are under control of the user or at least the web site =
programmer via the javascript API, and should therefore probably be =
expected to be reasonably long-lived.
>=20
> Second (and this is what I meant in this current discussion), =
individual packets in a multimedia stream differ in importance. One easy =
way to say this is that a video consists of e.g. I-frames, P-frames and =
B-frames, where the priority is I > P > B. In reality, things are more =
complex: a frame typically spans multiple packets, but there are blocks =
in frames of different type (e.g. a P-frame may contain I- P- and =
B-blocks), and then there is also a dependency chain: some blocks may =
have many other blocks depend upon them, making them more important, and =
others may have less, making them less important. Honoring these =
priorities is important for the receiver's subjective quality - here's =
one out of many, many existing publications documenting this in one way =
or another:
> Michael Schier, Michael Welzl: "Optimizing Selective ARQ for H.264 =
Live Streaming: A Novel Method for Predicting Loss-Impact in Realtime", =
IEEE Transactions on Multimedia 14(2), April 2012.
> http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=3D6095636
>=20
> Suffice to say, we can have different priorities per packet as a =
function of the multimedia encoding, and these priorities could quickly =
change (this is a way to represent the dependence - if the sending =
application is in control of its data until the last moment, and if it =
is informed about the loss of packet X, it could quickly assign a very =
low priority to packets that depend on X.
>=20
> Supporting these priorities in rmcat doesn't seem to be a MUST in =
accordance with the charter and rtcweb requirements. I am arguing that =
it would be great if we could support them.
>=20
> What I mean with supporting them via congestion control is: going back =
to the overly simplistic example of I- P- and B-"packets", one may =
decide to be less aggressive when the buffer is full of "B-packets" and =
more aggressive when it is full of "I-packets". One out of many =
possibilities... my point is, such functionality should be enabled by =
handing over the right information to the congestion controller, and =
this requires the controller to be close to the send buffer, or the =
information to be signalled to the other side. Both can be okay, but we =
should be clear about this.
>=20
> You want to allow the sender to, for a short time, override the =
bandwidth estimate by sending at a higher rate because it has important =
packets to send? Isn't this
> possible even though the estimator is at the receiver? I'm also not =
sure that is something you want to do. Sending important packets at a =
higher rate increases the risk of losing those important packets, which =
is the opposite of what you want, right?

That's a possibility; I was actually thinking of the exact opposite - to =
be overly cautious when you only have low-priority packets to send. Tune =
your filter to say "yes there's congestion, don't increase" when it =
might actually be noise in the measured delay signal in these =
conditions.=20

Yes this can also be done when the controller is at the receiver, but to =
do such a thing you need to inform the controller about the current =
state of the sender buffer (how many low-, medium- and high-priority =
packets are waiting there) =3D> more signalling. I'm not saying this is =
a big problem, but it's probably another weight in the "minus" part of =
the scale. Either way, it's just something to keep in mind, I think.

Cheers,
Michael


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On 8. okt. 2012, at 13:43, Stefan Holmer wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><br><div =
class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Sun, Oct 7, =
2012 at 3:17 PM, Michael Welzl <span dir=3D"ltr">&lt;<a =
href=3D"mailto:michawe@ifi.uio.no" target=3D"_blank" =
class=3D"cremed">michawe@ifi.uio.no</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
Very sorry for any disadvantages that might have come from having =
expressed myself unclear. I'll try to clean things up in this email =
now:<div class=3D"im"><br>
<br>
<br>
On Oct 5, 2012, at 4:58 PM, John Leslie wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">
Michael Welzl &lt;<a href=3D"mailto:michawe@ifi.uio.no" target=3D"_blank" =
class=3D"cremed">michawe@ifi.uio.no</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">
On 5. okt. 2012, at 00:31, John Leslie wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">
Michael Welzl &lt;<a href=3D"mailto:michawe@ifi.uio.no" target=3D"_blank" =
class=3D"cremed">michawe@ifi.uio.no</a>&gt; wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">
First, to clarify, I'm talking about per-packet priorities within a<br>
stream. Second, my point is that congestion control could be done<br>
better if packet priorities would be known, and could then be taken<br>
into account.<br>
</blockquote>
<br>
I guess I'm missing why packet priorities would ever _not_ be the<br>
domain of the sender -- even if advised by receiver preferences.<br>
</blockquote>
<br>
It's just not what I meant. I meant priorities influencing the<br>
congestion control behavior (I should say the "sender's" behavior,<br>
but this discussion is about having the congestion control logic<br>
on the sender or receiver side... so...).<br>
</blockquote>
<br>
&nbsp; "Priorities influencing" is actually quite separable from =
packet<br>
scheduling.<br>
<br>
&nbsp; I think we're hopelessly talking past each other here. It =
might<br>
help if you outlined some "likely" mixes of priorities for packets.<br>
</blockquote>
<br></div>
Maybe you're considering one type of priority and me another.<br>
<br>
Indeed, in rmcat we're facing two kinds of priorities. One, priorities =
between streams. I think we *must* be able to support these priorities =
because they are required by rtcweb (A23 in =
draft-ietf-rtcweb-use-cases-<u></u>and-requirements). In case of rtcweb, =
these priorities are under control of the user or at least the web site =
programmer via the javascript API, and should therefore probably be =
expected to be reasonably long-lived.<br>

<br>
Second (and this is what I meant in this current discussion), individual =
packets in a multimedia stream differ in importance. One easy way to say =
this is that a video consists of e.g. I-frames, P-frames and B-frames, =
where the priority is I &gt; P &gt; B. In reality, things are more =
complex: a frame typically spans multiple packets, but there are blocks =
in frames of different type (e.g. a P-frame may contain I- P- and =
B-blocks), and then there is also a dependency chain: some blocks may =
have many other blocks depend upon them, making them more important, and =
others may have less, making them less important. Honoring these =
priorities is important for the receiver's subjective quality - here's =
one out of many, many existing publications documenting this in one way =
or another:<br>

Michael Schier, Michael Welzl: "Optimizing Selective ARQ for H.264 Live =
Streaming: A Novel Method for Predicting Loss-Impact in Realtime", IEEE =
Transactions on Multimedia 14(2), April 2012.<br>
<a href=3D"http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=3D6095636"=
 target=3D"_blank" =
class=3D"cremed">http://ieeexplore.ieee.org/<u></u>xpls/abs_all.jsp?arnumb=
er=3D<u></u>6095636</a><br>
<br>
Suffice to say, we can have different priorities per packet as a =
function of the multimedia encoding, and these priorities could quickly =
change (this is a way to represent the dependence - if the sending =
application is in control of its data until the last moment, and if it =
is informed about the loss of packet X, it could quickly assign a very =
low priority to packets that depend on X.<br>

<br>
Supporting these priorities in rmcat doesn't seem to be a MUST in =
accordance with the charter and rtcweb requirements. I am arguing that =
it would be great if we could support them.<br>
<br>
What I mean with supporting them via congestion control is: going back =
to the overly simplistic example of I- P- and B-"packets", one may =
decide to be less aggressive when the buffer is full of "B-packets" and =
more aggressive when it is full of "I-packets". One out of many =
possibilities... my point is, such functionality should be enabled by =
handing over the right information to the congestion controller, and =
this requires the controller to be close to the send buffer, or the =
information to be signalled to the other side. Both can be okay, but we =
should be clear about this.</blockquote>
<div><br></div><div>You want to allow the sender to, for a short time, =
override the bandwidth estimate by sending at a higher rate because it =
has important packets to send? Isn't =
this</div></div></div></blockquote><blockquote type=3D"cite"><div =
class=3D"gmail_extra"><div class=3D"gmail_quote"><div> possible even =
though the estimator is at the receiver? I'm also not sure that is =
something you want to do. Sending important packets at a higher rate =
increases the risk of losing those important packets, which is the =
opposite of what you want, =
right?</div></div></div></blockquote><div><br></div><div><div>That's a =
possibility; I was actually thinking of the exact opposite - to be =
overly cautious when you&nbsp;only&nbsp;have low-priority packets to =
send. Tune your filter to say "yes there's congestion, don't increase" =
when it might actually be noise in the measured delay signal in these =
conditions.&nbsp;</div><div><br></div><div>Yes this can also be done =
when the controller is at the receiver, but to do such a thing you need =
to inform the controller about the current state of the sender buffer =
(how many low-, medium- and high-priority packets are waiting there) =
=3D&gt; more signalling. I'm not saying this is a big problem, but it's =
probably another weight in the "minus" part of the scale. Either way, =
it's just something to keep in mind, I =
think.</div><div><br></div><div>Cheers,</div><div>Michael</div><div><br></=
div></div></div></body></html>=

--Apple-Mail=_0FF2E28D-0849-4079-84D1-6DC3D17B50CF--

From john@jlc.net  Mon Oct  8 06:19:42 2012
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From: John Leslie <john@jlc.net>
To: Michael Welzl <michawe@ifi.uio.no>
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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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Michael Welzl <michawe@ifi.uio.no> wrote:
> 
> Maybe you're considering one type of priority and me another.

   Indeed!

> Indeed, in rmcat we're facing two kinds of priorities. One, priorities  
> between streams. I think we *must* be able to support these priorities  
> because they are required by rtcweb (A23 in
> draft-ietf-rtcweb-use-cases-and-requirements).
] 
] A23             The Web API MUST provide means for the
]                 application to specify the priority to
]                 apply for outgoing streams and data.

   Agreed.

> In case of rtcweb, these priorities are under control of the user or
> at least the web site programmer via the javascript API, and should
> therefore probably be expected to be reasonably long-lived.

   Yes.

> Second (and this is what I meant in this current discussion),  
> individual packets in a multimedia stream differ in importance. One  
> easy way to say this is that a video consists of e.g. I-frames, P- 
> frames and B-frames, where the priority is I > P > B.

   Hmmm...

   One should tread carefully here. Most video codecs are designed to
expect in-order delivery; and priorities can mess that up.

   (Or are you actually talking about trying to jiggle probability-of-
loss, rather than what goes first?)

> In reality, things are more complex: a frame typically spans multiple
> packets, but there are blocks in frames of different type (e.g. a
> P-frame may contain I- P- and B-blocks),

   You're starting to lose me... :^(

   There are a heck of a lot of video codec standards. Are you referring
to one in particular?

   I think of an I-frame as a picture complete in itself, and both
P-frames and B-frames as trying to describe motion between frames of
the original video signal -- the difference being that P-frames refer
to previous I-frames while B-frames refer to following I-frames as well.
(This in itself is enough to make my head ache!)

   I'm reduced to guesswork to try to attach a meaning to "P-block" and
"B-block"... I'd rather not go too far into guesswork...

> and then there is also a dependency chain: some blocks may have many
> other blocks depend upon them, making them more important, and others
> may have less, making them less important. Honoring these priorities
> is important for the receiver's subjective quality

   That sounds like probability-of-loss...

> - here's one out of many, many existing publications documenting this
> in one way or another:
> Michael Schier, Michael Welzl: "Optimizing Selective ARQ for H.264  
> Live Streaming: A Novel Method for Predicting Loss-Impact in  
> Realtime", IEEE Transactions on Multimedia 14(2), April 2012.
> http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=6095636

   Many of us won't have time to access and read this...

> Suffice to say, we can have different priorities per packet as a  
> function of the multimedia encoding, and these priorities could  
> quickly change (this is a way to represent the dependence - if the  
> sending application is in control of its data until the last moment,  
> and if it is informed about the loss of packet X, it could quickly  
> assign a very low priority to packets that depend on X.

   I have been assuming the sender _will_ be in control of what goes
into each packet sent (in other words, the sender will compose
packets, not streams).

   But you're losing me at "assign a very low priority"... There are
many things that could mean.

   The most obvious is QoS bits. I dislike that because it _asks_ to
mess with order of delivery.

   I won't list any other meanings -- this email is complicated enough
already...

> Supporting these priorities in rmcat doesn't seem to be a MUST in  
> accordance with the charter and rtcweb requirements. I am arguing that  
> it would be great if we could support them.

   To the extent you mean in what order the sender packs the pieces
into outgoing packets, I very much want to support that. If you mean
some "priority" mechanism other than QoS bits attached to packets, I'm
all ears to learn of it...

> What I mean with supporting them via congestion control is: going back  
> to the overly simplistic example of I- P- and B-"packets", one may  
> decide to be less aggressive when the buffer is full of "B-packets"  
> and more aggressive when it is full of "I-packets". One out of many  
> possibilities... my point is, such functionality should be enabled by  
> handing over the right information to the congestion controller, and  
> this requires the controller to be close to the send buffer, or the  
> information to be signalled to the other side. Both can be okay, but  
> we should be clear about this.

   Philosophically, I'm more friendly to congestion-control-at-sender;
but I still don't think we need to settle that yet.

>> Are you arguing we should give up considering receiver-based-control?
> 
> I'm arguing that we should have either this, or strictly only-receiver- 
> based control. Then we could get the same functionality: you could  
> just change the receiver to use a new mechanism X, without waiting for  
> the sender to support the new mechanism X or even signalling to the  
> sender what mechanism to use. Sender- or receiver- based, but not both.

   This is a bit convoluted... I agree we could accomplish what we need
with either receiver-based or sender-based congestion control; thus I
don't see the need to settle it yet.

   I disagree that we can't do it with a mix (though at first blush
that would seem a very poor choice).

>> I _do_ believe we need to establish a feedback mechanism with a clear
>> definition of what the feedback means (in both directions, of course).
 
> I do believe it's necessary (sufficient, no - indeed we'd need a  
> clearly defined feedback mechanism).

   :^)

> For example, with strict receiver-based control, only having a
> clearly defined feedback mechanism will not make flexible use of
> controls as with TCP possible. Or, in other words: having e.g.
> RRTCC's sender behavior but multiply defined receiver behaviors
> would make this flexibility possible, but having multiple sender-
> AND receiver-behaviors will not.

   I think you've lost most of our readers here. :^(

   Presumably you mean:

http://tools.ietf.org/id/draft-alvestrand-rtcweb-congestion-02.txt

I _could_ follow the math there, had I more time to spare... I'm
guessing the critical part comes in section 3.4:
] 
] 3.4.  Rate control
] 
]  The rate control at the receiving side is designed to increase the
]  receive-side estimate of the available bandwidth A_hat as long as the
]  detected state is normal.  Doing that assures that we, sooner or
]  later, will reach the available bandwidth of the channel and detect
]  an over-use.
]...

any you're saying A_hat isn't sufficient to the purpose. (I don't even
see why that's necessarily true, unless you mean that any intelligence
whatsoever at sender-side is forbidden.)

--
John Leslie <john@jlc.net>

From michawe@ifi.uio.no  Mon Oct  8 12:25:24 2012
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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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On Oct 8, 2012, at 3:19 PM, John Leslie wrote:

> Michael Welzl <michawe@ifi.uio.no> wrote:
>>
>> Maybe you're considering one type of priority and me another.
>
>   Indeed!
>
>> Indeed, in rmcat we're facing two kinds of priorities. One,  
>> priorities
>> between streams. I think we *must* be able to support these  
>> priorities
>> because they are required by rtcweb (A23 in
>> draft-ietf-rtcweb-use-cases-and-requirements).
> ]
> ] A23             The Web API MUST provide means for the
> ]                 application to specify the priority to
> ]                 apply for outgoing streams and data.
>
>   Agreed.
>
>> In case of rtcweb, these priorities are under control of the user or
>> at least the web site programmer via the javascript API, and should
>> therefore probably be expected to be reasonably long-lived.
>
>   Yes.
>
>> Second (and this is what I meant in this current discussion),
>> individual packets in a multimedia stream differ in importance. One
>> easy way to say this is that a video consists of e.g. I-frames, P-
>> frames and B-frames, where the priority is I > P > B.
>
>   Hmmm...
>
>   One should tread carefully here. Most video codecs are designed to
> expect in-order delivery; and priorities can mess that up.
>
>   (Or are you actually talking about trying to jiggle probability-of-
> loss, rather than what goes first?)

Well, there is only so much you can squeeze through a pipe. When what  
you'd like to send is more than the network allows, a decision has to  
be taken, and that should be done on priorities.
In a way that's similar to a loss - it's a decision on the sender  
side, to not send certain packets in favor of others.


>> In reality, things are more complex: a frame typically spans multiple
>> packets, but there are blocks in frames of different type (e.g. a
>> P-frame may contain I- P- and B-blocks),
>
>   You're starting to lose me... :^(
>
>   There are a heck of a lot of video codec standards. Are you  
> referring
> to one in particular?

AFAIK many codecs are based on MPEG in one way or another. This basis  
is what I'm referring to.


>   I think of an I-frame as a picture complete in itself, and both
> P-frames and B-frames as trying to describe motion between frames of
> the original video signal -- the difference being that P-frames refer
> to previous I-frames while B-frames refer to following I-frames as  
> well.
> (This in itself is enough to make my head ache!)
>
>   I'm reduced to guesswork to try to attach a meaning to "P-block" and
> "B-block"... I'd rather not go too far into guesswork...

A block is just a part of a frame. Anyway, as I said in my previous  
email, for simplicity it's enough to think about I- vs. P- vs. B- 
frames, exactly as you describe them here.


>> and then there is also a dependency chain: some blocks may have many
>> other blocks depend upon them, making them more important, and others
>> may have less, making them less important. Honoring these priorities
>> is important for the receiver's subjective quality
>
>   That sounds like probability-of-loss...

... in the sender. See above.


>> - here's one out of many, many existing publications documenting this
>> in one way or another:
>> Michael Schier, Michael Welzl: "Optimizing Selective ARQ for H.264
>> Live Streaming: A Novel Method for Predicting Loss-Impact in
>> Realtime", IEEE Transactions on Multimedia 14(2), April 2012.
>> http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=6095636
>
>   Many of us won't have time to access and read this...

Not needed to get my point. It's just one out of many examples, to  
clarify that honoring such priorities can make a pronounced quality  
difference.


>> Suffice to say, we can have different priorities per packet as a
>> function of the multimedia encoding, and these priorities could
>> quickly change (this is a way to represent the dependence - if the
>> sending application is in control of its data until the last moment,
>> and if it is informed about the loss of packet X, it could quickly
>> assign a very low priority to packets that depend on X.
>
>   I have been assuming the sender _will_ be in control of what goes
> into each packet sent (in other words, the sender will compose
> packets, not streams).

That's good. I hope the sender will be in control until the very last  
moment in actual implementations coming out of rmcat. This is not as  
simple and obvious as it may sound, I think.


>   But you're losing me at "assign a very low priority"... There are
> many things that could mean.

>   The most obvious is QoS bits. I dislike that because it _asks_ to
> mess with order of delivery.
>
>   I won't list any other meanings -- this email is complicated enough
> already...


Let's stay with the I-, P-, B-frame model. That makes priority 1, 2,  
3. I am then saying that priorities could be changed to be 3. My  
context is the send buffer, influencing the congestion control  
behavior but nothing else. Nothing on the wire. I'm not trying to make  
things THAT complicated!  :-)


>> Supporting these priorities in rmcat doesn't seem to be a MUST in
>> accordance with the charter and rtcweb requirements. I am arguing  
>> that
>> it would be great if we could support them.
>
>   To the extent you mean in what order the sender packs the pieces
> into outgoing packets, I very much want to support that. If you mean
> some "priority" mechanism other than QoS bits attached to packets, I'm
> all ears to learn of it...

I mean: if the sender's buffer is full of 1-priority packets, be more  
aggressive. If the sender's buffer is full of 3-priority packets, be  
less aggressive.
This can be implemented in more or less reasonable ways. One, to me,  
reasonable method is what I described in the email answering Stefan:  
only implement low priorities, by being overly cautious in increasing  
the rate when your buffer is full of low-priority packets. ("Overly  
cautious" has a bad sound to it, but in fact any delay-based mechanism  
must be based on some parameters which can never be ideal - always  
trade-offs there, and I'm merely suggesting to tune these trade-offs  
based on what is in the sender buffer). Here's another example:
RRTCC: http://tools.ietf.org/id/draft-alvestrand-rtcweb-congestion-02.txt
has the sender send at least as much as one standard TCP would, using  
the TCP equation. With the MulTCP equation:
http://heim.ifi.uio.no/~michawe/research/projects/multfrc/index.html
one could easily tune this to send at least as much as N standard  
TCP's, where N could for example be 1.5 or 0.8. N could depend on the  
number of priority-1, priority-2 or priority-3 packets waiting to be  
transmitted.


>> What I mean with supporting them via congestion control is: going  
>> back
>> to the overly simplistic example of I- P- and B-"packets", one may
>> decide to be less aggressive when the buffer is full of "B-packets"
>> and more aggressive when it is full of "I-packets". One out of many
>> possibilities... my point is, such functionality should be enabled by
>> handing over the right information to the congestion controller, and
>> this requires the controller to be close to the send buffer, or the
>> information to be signalled to the other side. Both can be okay, but
>> we should be clear about this.
>
>   Philosophically, I'm more friendly to congestion-control-at-sender;
> but I still don't think we need to settle that yet.
>
>>> Are you arguing we should give up considering receiver-based- 
>>> control?
>>
>> I'm arguing that we should have either this, or strictly only- 
>> receiver-
>> based control. Then we could get the same functionality: you could
>> just change the receiver to use a new mechanism X, without waiting  
>> for
>> the sender to support the new mechanism X or even signalling to the
>> sender what mechanism to use. Sender- or receiver- based, but not  
>> both.
>
>   This is a bit convoluted... I agree we could accomplish what we need
> with either receiver-based or sender-based congestion control; thus I
> don't see the need to settle it yet.
>
>   I disagree that we can't do it with a mix (though at first blush
> that would seem a very poor choice).

I don't understand how a mix would be as easily deployable as a one- 
sided approach.


>>> I _do_ believe we need to establish a feedback mechanism with a  
>>> clear
>>> definition of what the feedback means (in both directions, of  
>>> course).
>
>> I do believe it's necessary (sufficient, no - indeed we'd need a
>> clearly defined feedback mechanism).
>
>   :^)
>
>> For example, with strict receiver-based control, only having a
>> clearly defined feedback mechanism will not make flexible use of
>> controls as with TCP possible. Or, in other words: having e.g.
>> RRTCC's sender behavior but multiply defined receiver behaviors
>> would make this flexibility possible, but having multiple sender-
>> AND receiver-behaviors will not.
>
>   I think you've lost most of our readers here. :^(
>
>   Presumably you mean:
>
> http://tools.ietf.org/id/draft-alvestrand-rtcweb-congestion-02.txt
>
> I _could_ follow the math there, had I more time to spare... I'm
> guessing the critical part comes in section 3.4:
> ]
> ] 3.4.  Rate control
> ]
> ]  The rate control at the receiving side is designed to increase the
> ]  receive-side estimate of the available bandwidth A_hat as long as  
> the
> ]  detected state is normal.  Doing that assures that we, sooner or
> ]  later, will reach the available bandwidth of the channel and detect
> ]  an over-use.
> ]...
>
> any you're saying A_hat isn't sufficient to the purpose. (I don't even

Err... now you're losing me. I think your interpretation of what I'm  
saying is way more complicated than what I mean  :-)
Sorry if I'm not clear enough, I promise I'm doing my best here...


> see why that's necessarily true, unless you mean that any intelligence
> whatsoever at sender-side is forbidden.)

I mean that any mechanism-specific intelligence at sender-side should  
be forbidden, if we settle for receiver-based congestion control.  
That's what I mean with "generic": not mechanism-specific.
RMCAT will perhaps standardize 3 mechanisms. A, B and C. Say each is  
split into a sender and receiver behavior: A_send and A_recv, B_send  
and B_recv, C_send and C_recv. Say they are not compatible, i.e.  
A_send cannot talk to B_recv, and C_send cannot talk to A_recv and so  
on. Then, for any mechanism X to work, you need both X_send and X_recv  
to be in place.

Say, we decide to do receiver-based congestion control. Then I argue  
that we should agree on G_send, which is a generic sender behavior.  
Then, A, B and C only consist of A_recv, B_recv and C_recv, and you  
always only need to deploy this receiving end to use the new mechanism.

Symmetrically, the same is true for pure sender-based congestion  
control. It's a mix of the two that's hard to deploy.

Cheers,
Michael


From paalh@ifi.uio.no  Tue Oct  9 01:30:32 2012
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Subject: [rmcat] PhD position available in the area of in Time-Dependent Networking
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AVAILABLE PHD POSITION

At Simula Research Laboratory (Norway) there is an available PhD =
position in the area of in Time-Dependent Networking. The PhD Student =
will work in the area of networking support for time-dependent =
applications, focusing on thin streams. Thin streams are generated by =
applications that send data with high interarrival time and also often =
small packet sizes. These traffic patterns are in stark contrast to =
greedy streams where the application always has data to send and the =
sending rate is limited by congestion control. Typical for thin streams =
is that they are generated by interactive applications with a need for =
low latency communication.

The following topics are central for the work in this position:

=95 Develop mechanisms to improve thin-stream latency when transmitted =
via different buffer types with varying drop schemes.
=95 Investigate the congestion control support of conditionally thin =
streams (i.e. not everything must be sent at all times) via per-packet =
priorities.
=95 Work towards a formal definition of the concept of thin streams.
=95 Work towards defining what TCP fairness must mean in thin-stream =
scenarios.



The position is connected to the "Traffic behaviour of interactive =
time-dependent thin streams on the modern Internet" (TimeIn) research =
project cooperating with partners like UNINETT, University of Oslo, =
Karlstad University, University of Kaiserslautern, CAIDA San Diego, =
Cisco and Funcom.

It is planned to align the research with, and possibly contribute to, =
ongoing standardization efforts in the RTP Media Congestion Avoidance =
Techniques (RMCAT) working group in the IETF.


For more information about the project, the requirements, Simula, =
salary, etc.,=20
please see
  http://simula.no/jobs/PhDStudent-TimeIn

or contact:
  Postdoc Andreas Petlund, e-mail: apetlund@simula.no or
  Professor P=E5l Halvorsen, e-mail: paalh@simula.no
(where you include the following phrase in the subject field: =93TimeIn =
PhD position=94)

The application deadline is as soon as possible, but no later than 20 =
October 2012.

To apply for this position, click here:=20
=
https://emea3.recruitmentplatform.com/appproc/index.cfm?event=3DcreateSess=
ionAfterSessionClear&ID=3DQL5FK026203F3VBQBV779QWAD&jobboard=3D0&nPTID=3D2=
8&bSessionClear=3Dtrue=

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From: "Eggert, Lars" <lars@netapp.com>
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Hi,

On Oct 5, 2012, at 15:50, "Eggert, Lars" <lars@netapp.com> wrote:
> The ID cutoff for new -00 drafts (which most of ours will be) is Oct =
15. Please make sure your draft-yourname-rmcat-* is submitted by then.

and even if your draft is not submitted yet, please already send Mirja =
and me requests for agenda time, so we can start the planning process.

Thanks,
Lars=

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From john@jlc.net  Tue Oct  9 06:06:46 2012
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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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Michael Welzl <michawe@ifi.uio.no> wrote:
> On Oct 8, 2012, at 3:19 PM, John Leslie wrote:
>> Michael Welzl <michawe@ifi.uio.no> wrote:
> 
> Well, there is only so much you can squeeze through a pipe. When what  
> you'd like to send is more than the network allows, a decision has to  
> be taken, and that should be done on priorities.

   I'm not sure "priorities" is a good name for this -- it's allowed me
to get confused with packet-forwarding priorities.

>> There are a heck of a lot of video codec standards. Are you  
>> referring to one in particular?
> 
> AFAIK many codecs are based on MPEG in one way or another. This basis  
> is what I'm referring to.

   That's probably a good family of codecs to think in terms of; but
we might find ourselves working with something different. Not
surprisingly, "Moving Pictures Experts Group" concentrated on motion,
while we will have uses where motion is limited...

>> I have been assuming the sender _will_ be in control of what goes
>> into each packet sent (in other words, the sender will compose
>> packets, not streams).
> 
> That's good. I hope the sender will be in control until the very last  
> moment in actual implementations coming out of rmcat. This is not as  
> simple and obvious as it may sound, I think.

   It may indeed not be "simple" for web-browsers in general. Our
charter calls for:
" 
" - Find or develop candidate congestion control algorithms, verify that
" these can be tested on the Internet without significant risk, and
" publish one or more of these as Experimental RFCs.

which I think gives us sufficient wiggle-room for demonstrating a
congestion control algorithm that not all web-browsers can initially
support.

   Nonetheless, I see sender control of what gets packed into packets
as essential to our eventual success; and I think that browsers can
evolve to escape the TCP-stream paradigm.

>> But you're losing me at "assign a very low priority"... There are
>> many things that could mean...
> 
> Let's stay with the I-, P-, B-frame model. That makes priority 1, 2,  
> 3. I am then saying that priorities could be changed to be 3. My  
> context is the send buffer, influencing the congestion control  
> behavior but nothing else. Nothing on the wire. I'm not trying to make  
> things THAT complicated!  :-)

   :^)

   In some sense, you seem to be talking of pre-determining x-frame
loss, rather than waiting for the network to drop something. "Priority"
IMHO isn't a good word for that; though it may turn out to be a very
good paradigm.

   Fundamentally, under "congestion control" we want to back off
sending bit-rate when congestion is detected, and ramp it back up
when congestion appears to have cleared. Packing less into the packets
we send is certainly an appropriate mechanism.

   "Priority" may be a good term for academic types to describe
policies for which frames don't get sent, but I'm afraid that to
operations types it brings too much baggage. Also, "priority" suggests
we're talking about a scalar, when IMHO the problem space is multi-
dimensional.

>> To the extent you mean in what order the sender packs the pieces
>> into outgoing packets, I very much want to support that. If you mean
>> some "priority" mechanism other than QoS bits attached to packets, I'm
>> all ears to learn of it...
> 
> I mean: if the sender's buffer is full of 1-priority packets, be more  
> aggressive. If the sender's buffer is full of 3-priority packets, be  
> less aggressive.

   This deserves its own thread. (I agree it's a promising path.)

> This can be implemented in more or less reasonable ways. One, to me,  
> reasonable method is what I described in the email answering Stefan:  
> only implement low priorities, by being overly cautious in increasing  
> the rate when your buffer is full of low-priority packets.

   (I have gone beyond this, to suggest maintaining two rates: a
slower one for less important packets and a faster one for the most
important packets; both to be reduced when congestion is detected.
This may deserve discussion in some other thread...)

> ("Overly cautious" has a bad sound to it, but in fact any delay-based
> mechanism must be based on some parameters which can never be ideal -
> always trade-offs there, and I'm merely suggesting to tune these
> trade-offs based on what is in the sender buffer).

   This may not be the easiest thing to model... :^(

   The way I think about it is that we want to avoid a situation where
_our_ packets are growing the bit-rate at the bottleneck too quickly.
There are plenty of uses out there which deserve to be called greedy:
where every second of delay in delivering a fixed-size web-page
reduces the user experience. For these, multiple TCP streams are "the"
way to go; but this leads to overshoot and congestion more severe
than "should" be needed.

   Thus, by the "first, do no harm" principle, we should avoid driving
a bottleneck into severe congestion. In trade, we should be able to
back off a bit more slowly when congestion _is_ encountered. Our true
aim is to deliver a _consistent_ user experience -- not an "optimized"
delivery of a fixed-size message.

> Here's another example:
> RRTCC: http://tools.ietf.org/id/draft-alvestrand-rtcweb-congestion-02.txt
> has the sender send at least as much as one standard TCP would, using  
> the TCP equation. With the MulTCP equation:
> http://heim.ifi.uio.no/~michawe/research/projects/multfrc/index.html
> one could easily tune this to send at least as much as N standard  
> TCP's, where N could for example be 1.5 or 0.8. N could depend on the  
> number of priority-1, priority-2 or priority-3 packets waiting to be  
> transmitted.

   At first blush, that seems a reasonable way to model the situation;
but I suspect we can do better. N>1 TCP is a big part of the problem...

>> This is a bit convoluted... I agree we could accomplish what we need
>> with either receiver-based or sender-based congestion control; thus I
>> don't see the need to settle it yet.
>>
>> I disagree that we can't do it with a mix (though at first blush
>>that would seem a very poor choice).
> 
> I don't understand how a mix would be as easily deployable as a one- 
> sided approach.

   I'm not talking about "ease of deployment". We need first to
establish we have something worth deploying.

>>... Presumably you mean:
>>
>> http://tools.ietf.org/id/draft-alvestrand-rtcweb-congestion-02.txt
>>
>> I _could_ follow the math there, had I more time to spare... I'm
>> guessing the critical part comes in section 3.4:
>>]
>>] 3.4.  Rate control
>>]
>>]  The rate control at the receiving side is designed to increase the
>>]  receive-side estimate of the available bandwidth A_hat as long as the
>>]  detected state is normal.  Doing that assures that we, sooner or
>>]  later, will reach the available bandwidth of the channel and detect
>>]  an over-use.
>>]...
>>
>> and you're saying A_hat isn't sufficient to the purpose. (I don't
>> even see why that's necessarily true, unless you mean that any
>> intelligence whatsoever at sender-side is forbidden.)
> 
> I mean that any mechanism-specific intelligence at sender-side should  
> be forbidden, if we settle for receiver-based congestion control.  

   I don't agree that would be helpful.

> That's what I mean with "generic": not mechanism-specific.
> RMCAT will perhaps standardize 3 mechanisms. A, B and C. Say each is  
> split into a sender and receiver behavior: A_send and A_recv, B_send  
> and B_recv, C_send and C_recv. Say they are not compatible, i.e.  
> A_send cannot talk to B_recv, and C_send cannot talk to A_recv and so  
> on. Then, for any mechanism X to work, you need both X_send and X_recv  
> to be in place.

   Logically true; but not particularly interesting...

   We're chartered to explore Experimental protocols. There's every
reason to believe when the time comes for Proposed Standard, we will
have learned from the experiments. IMHO, at this stage it's more
important to design good experiments than worry about deployment of
something not yet designed.

--
John Leslie <john@jlc.net>

From michawe@ifi.uio.no  Tue Oct  9 06:38:49 2012
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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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On Oct 9, 2012, at 3:06 PM, John Leslie wrote:

> Michael Welzl <michawe@ifi.uio.no> wrote:
>> On Oct 8, 2012, at 3:19 PM, John Leslie wrote:
>>> Michael Welzl <michawe@ifi.uio.no> wrote:
>>
>> Well, there is only so much you can squeeze through a pipe. When what
>> you'd like to send is more than the network allows, a decision has to
>> be taken, and that should be done on priorities.
>
>   I'm not sure "priorities" is a good name for this -- it's allowed me
> to get confused with packet-forwarding priorities.

I understand; sorry. I herewith coin the term "Packet  
Importance" (PI)   :-)


>>> There are a heck of a lot of video codec standards. Are you
>>> referring to one in particular?
>>
>> AFAIK many codecs are based on MPEG in one way or another. This basis
>> is what I'm referring to.
>
>   That's probably a good family of codecs to think in terms of; but
> we might find ourselves working with something different. Not
> surprisingly, "Moving Pictures Experts Group" concentrated on motion,
> while we will have uses where motion is limited...

Different importance of different parts of media is a natural  
occurrence if you want to compress, really. And compression is a  
natural occurrence if you want to squeeze, say, video through the  
network.
We might not only consider video, yes, but for cases where you have  
similar-importance packets, well, you have no problem. It's just that  
WHEN different PIs (defined as per above) occur, it can be useful to  
let the congestion controller know about them.


>>> I have been assuming the sender _will_ be in control of what goes
>>> into each packet sent (in other words, the sender will compose
>>> packets, not streams).
>>
>> That's good. I hope the sender will be in control until the very last
>> moment in actual implementations coming out of rmcat. This is not as
>> simple and obvious as it may sound, I think.
>
>   It may indeed not be "simple" for web-browsers in general. Our
> charter calls for:
> "
> " - Find or develop candidate congestion control algorithms, verify  
> that
> " these can be tested on the Internet without significant risk, and
> " publish one or more of these as Experimental RFCs.
>
> which I think gives us sufficient wiggle-room for demonstrating a
> congestion control algorithm that not all web-browsers can initially
> support.
>
>   Nonetheless, I see sender control of what gets packed into packets
> as essential to our eventual success; and I think that browsers can
> evolve to escape the TCP-stream paradigm.

I'm completely with you here.


>>> But you're losing me at "assign a very low priority"... There are
>>> many things that could mean...
>>
>> Let's stay with the I-, P-, B-frame model. That makes priority 1, 2,
>> 3. I am then saying that priorities could be changed to be 3. My
>> context is the send buffer, influencing the congestion control
>> behavior but nothing else. Nothing on the wire. I'm not trying to  
>> make
>> things THAT complicated!  :-)
>
>   :^)
>
>   In some sense, you seem to be talking of pre-determining x-frame
> loss, rather than waiting for the network to drop something.  
> "Priority"
> IMHO isn't a good word for that; though it may turn out to be a very
> good paradigm.

PI hereinafter  :-)   and "pre-determining x-frame loss" is a rather  
complicated phrase for what really is: "reducing the send rate".  
Adaptive multimedia.


>   Fundamentally, under "congestion control" we want to back off
> sending bit-rate when congestion is detected, and ramp it back up
> when congestion appears to have cleared. Packing less into the packets
> we send is certainly an appropriate mechanism.
>
>   "Priority" may be a good term for academic types to describe
> policies for which frames don't get sent, but I'm afraid that to
> operations types it brings too much baggage. Also, "priority" suggests
> we're talking about a scalar, when IMHO the problem space is multi-
> dimensional.

Nononononono please! A scalar please!!!
I herewith very very quickly define that "Packet Importance" is a  
scalar with values 1, 2 and 3. Not multiple dimensions!!!


>>> To the extent you mean in what order the sender packs the pieces
>>> into outgoing packets, I very much want to support that. If you mean
>>> some "priority" mechanism other than QoS bits attached to packets,  
>>> I'm
>>> all ears to learn of it...
>>
>> I mean: if the sender's buffer is full of 1-priority packets, be more
>> aggressive. If the sender's buffer is full of 3-priority packets, be
>> less aggressive.
>
>   This deserves its own thread. (I agree it's a promising path.)
>
>> This can be implemented in more or less reasonable ways. One, to me,
>> reasonable method is what I described in the email answering Stefan:
>> only implement low priorities, by being overly cautious in increasing
>> the rate when your buffer is full of low-priority packets.
>
>   (I have gone beyond this, to suggest maintaining two rates: a
> slower one for less important packets and a faster one for the most
> important packets; both to be reduced when congestion is detected.
> This may deserve discussion in some other thread...)

Ok about another thread. About your suggestion: another possibility...  
there are many. All I'm saying is that such stuff could be done, and  
would probably be beneficial, and for that to be possible the  
congestion controller must be informed about the PIs of packets in the  
buffer. Which, in case of sender-side control, is straightforward, and  
in case of receiver-side control, needs signalling.


>> ("Overly cautious" has a bad sound to it, but in fact any delay-based
>> mechanism must be based on some parameters which can never be ideal -
>> always trade-offs there, and I'm merely suggesting to tune these
>> trade-offs based on what is in the sender buffer).
>
>   This may not be the easiest thing to model... :^(
>
>   The way I think about it is that we want to avoid a situation where
> _our_ packets are growing the bit-rate at the bottleneck too quickly.
> There are plenty of uses out there which deserve to be called greedy:
> where every second of delay in delivering a fixed-size web-page
> reduces the user experience. For these, multiple TCP streams are "the"
> way to go; but this leads to overshoot and congestion more severe
> than "should" be needed.
>
>   Thus, by the "first, do no harm" principle, we should avoid driving
> a bottleneck into severe congestion. In trade, we should be able to
> back off a bit more slowly when congestion _is_ encountered. Our true
> aim is to deliver a _consistent_ user experience -- not an "optimized"
> delivery of a fixed-size message.
>
>> Here's another example:
>> RRTCC: http://tools.ietf.org/id/draft-alvestrand-rtcweb-congestion-02.txt
>> has the sender send at least as much as one standard TCP would, using
>> the TCP equation. With the MulTCP equation:
>> http://heim.ifi.uio.no/~michawe/research/projects/multfrc/index.html
>> one could easily tune this to send at least as much as N standard
>> TCP's, where N could for example be 1.5 or 0.8. N could depend on the
>> number of priority-1, priority-2 or priority-3 packets waiting to be
>> transmitted.
>
>   At first blush, that seems a reasonable way to model the situation;
> but I suspect we can do better. N>1 TCP is a big part of the  
> problem...

Our equation also allows for N<1 TCP  :-)    Like, 0.5  :-)


>>> This is a bit convoluted... I agree we could accomplish what we need
>>> with either receiver-based or sender-based congestion control;  
>>> thus I
>>> don't see the need to settle it yet.
>>>
>>> I disagree that we can't do it with a mix (though at first blush
>>> that would seem a very poor choice).
>>
>> I don't understand how a mix would be as easily deployable as a one-
>> sided approach.
>
>   I'm not talking about "ease of deployment". We need first to
> establish we have something worth deploying.
>
>>> ... Presumably you mean:
>>>
>>> http://tools.ietf.org/id/draft-alvestrand-rtcweb-congestion-02.txt
>>>
>>> I _could_ follow the math there, had I more time to spare... I'm
>>> guessing the critical part comes in section 3.4:
>>> ]
>>> ] 3.4.  Rate control
>>> ]
>>> ]  The rate control at the receiving side is designed to increase  
>>> the
>>> ]  receive-side estimate of the available bandwidth A_hat as long  
>>> as the
>>> ]  detected state is normal.  Doing that assures that we, sooner or
>>> ]  later, will reach the available bandwidth of the channel and  
>>> detect
>>> ]  an over-use.
>>> ]...
>>>
>>> and you're saying A_hat isn't sufficient to the purpose. (I don't
>>> even see why that's necessarily true, unless you mean that any
>>> intelligence whatsoever at sender-side is forbidden.)
>>
>> I mean that any mechanism-specific intelligence at sender-side should
>> be forbidden, if we settle for receiver-based congestion control.
>
>   I don't agree that would be helpful.
>
>> That's what I mean with "generic": not mechanism-specific.
>> RMCAT will perhaps standardize 3 mechanisms. A, B and C. Say each is
>> split into a sender and receiver behavior: A_send and A_recv, B_send
>> and B_recv, C_send and C_recv. Say they are not compatible, i.e.
>> A_send cannot talk to B_recv, and C_send cannot talk to A_recv and so
>> on. Then, for any mechanism X to work, you need both X_send and  
>> X_recv
>> to be in place.
>
>   Logically true; but not particularly interesting...
>
>   We're chartered to explore Experimental protocols. There's every
> reason to believe when the time comes for Proposed Standard, we will
> have learned from the experiments. IMHO, at this stage it's more
> important to design good experiments than worry about deployment of
> something not yet designed.

I disagree on both accounts.
1) About the importance: the IETF has standardized a wealth of  
undeployed / undeployable stuff, e.g. DCCP.
2) About the design aspect: I don't see any notable benefit from  
having a per-mechanism sender *and* receiver behavior, but, as stated,  
a significant deployment problem with two-sided approaches. And I  
believe it's important to get these major architectural decisions  
right from the outset.

But I don't believe we will come to an agreement here, and have yet  
converged to a point where we can probably just leave it at that, for  
now  :)

Cheers,
Michael


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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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On 10/8/2012 3:24 PM, Michael Welzl wrote:
>
> On Oct 8, 2012, at 3:19 PM, John Leslie wrote:
>
>> Michael Welzl <michawe@ifi.uio.no> wrote:
>>>
>>> Maybe you're considering one type of priority and me another.
>>
>>   Indeed!
>>
>>> Indeed, in rmcat we're facing two kinds of priorities. One, priorities
>>> between streams. I think we *must* be able to support these priorities
>>> because they are required by rtcweb (A23 in
>>> draft-ietf-rtcweb-use-cases-and-requirements).
>> ]
>> ] A23             The Web API MUST provide means for the
>> ]                 application to specify the priority to
>> ]                 apply for outgoing streams and data.
>>
>>   Agreed.
>>
>>> In case of rtcweb, these priorities are under control of the user or
>>> at least the web site programmer via the javascript API, and should
>>> therefore probably be expected to be reasonably long-lived.
>>
>>   Yes.
>>
>>> Second (and this is what I meant in this current discussion),
>>> individual packets in a multimedia stream differ in importance. One
>>> easy way to say this is that a video consists of e.g. I-frames, P-
>>> frames and B-frames, where the priority is I > P > B.
>>
>>   Hmmm...
>>
>>   One should tread carefully here. Most video codecs are designed to
>> expect in-order delivery; and priorities can mess that up.
>>
>>   (Or are you actually talking about trying to jiggle probability-of-
>> loss, rather than what goes first?)
>
> Well, there is only so much you can squeeze through a pipe. When what 
> you'd like to send is more than the network allows, a decision has to 
> be taken, and that should be done on priorities.
> In a way that's similar to a loss - it's a decision on the sender 
> side, to not send certain packets in favor of others.

Except in most cases we're not sitting with a pile of frames/packets and 
deciding which ones get dropped (liek a router); we're encoding data to 
meet the current-moment bandwidth requirements.  There is a separate 
component of feeding data out onto the wire, which would be applied to 
non-encode-controlled streams like a SCTP/DTLS DataChannel sharing the 
pipe.  For encoder-based streams, we generally want to control via the 
encoder not by inducing possible internal delay in how we select packets 
for transmission.

If the data doesn't fit in the current pipe estimate, we shouldn't be 
generating it in the first place.  And intermediate nodes don't know our 
priorities to adjust probability of drop except maybe indirectly via 
markings, which may instead mess with order of delivery (not a good thing).

>>   I'm reduced to guesswork to try to attach a meaning to "P-block" and
>> "B-block"... I'd rather not go too far into guesswork...
>
>
> A block is just a part of a frame. Anyway, as I said in my previous 
> email, for simplicity it's enough to think about I- vs. P- vs. 
> B-frames, exactly as you describe them here.

Most packets are at a much higher level than blocks, so barring 
considerable swizzling of internal order it's likely that all packets 
for a frame should have similar priority and likelyhood of drop.

>
>>   That sounds like probability-of-loss...
>
>
> ... in the sender. See above.

We should never drop in the sender. :-)

>> see why that's necessarily true, unless you mean that any intelligence
>> whatsoever at sender-side is forbidden.)
>
> I mean that any mechanism-specific intelligence at sender-side should 
> be forbidden, if we settle for receiver-based congestion control. 
> That's what I mean with "generic": not mechanism-specific.
> RMCAT will perhaps standardize 3 mechanisms. A, B and C. Say each is 
> split into a sender and receiver behavior: A_send and A_recv, B_send 
> and B_recv, C_send and C_recv. Say they are not compatible, i.e. 
> A_send cannot talk to B_recv, and C_send cannot talk to A_recv and so 
> on. Then, for any mechanism X to work, you need both X_send and X_recv 
> to be in place.
>
> Say, we decide to do receiver-based congestion control. Then I argue 
> that we should agree on G_send, which is a generic sender behavior. 
> Then, A, B and C only consist of A_recv, B_recv and C_recv, and you 
> always only need to deploy this receiving end to use the new mechanism.
>
> Symmetrically, the same is true for pure sender-based congestion 
> control. It's a mix of the two that's hard to deploy.

Not that I'm arguing for it at this moment: but you could design things 
so that 'generic'/dumb mechanisms are available to support either sender 
or receiver-based control, with the decision on which done by 
negotiation or even in-band decision.  It simply requires defining both, 
and defining a way to select the one that's non-default.

This would allow innovation at either end in the future.

As to whether this is useful... I haven't given it the thought it 
deserves yet.  But it's quite doable.

(And on a separate note, a 'generic'/dumb mechanism doesn't mean there 
can't be feedback from the other side affecting how it operates, or that 
it can't be a more complex state machine than "send one packet per 
packet received").  Just that it be relatively simple and well-defined.  
(A bit like the whole video-codec "we'll define the quite-complex 
decoder and let the encoder innovate")

-- 
Randell Jesup
randell-ietf@jesup.org


From michawe@ifi.uio.no  Wed Oct 10 09:10:42 2012
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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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On Oct 9, 2012, at 10:04 PM, Randell Jesup wrote:

> On 10/8/2012 3:24 PM, Michael Welzl wrote:
>>
>> On Oct 8, 2012, at 3:19 PM, John Leslie wrote:
>>
>>> Michael Welzl <michawe@ifi.uio.no> wrote:
>>>>
>>>> Maybe you're considering one type of priority and me another.
>>>
>>>  Indeed!
>>>
>>>> Indeed, in rmcat we're facing two kinds of priorities. One,  
>>>> priorities
>>>> between streams. I think we *must* be able to support these  
>>>> priorities
>>>> because they are required by rtcweb (A23 in
>>>> draft-ietf-rtcweb-use-cases-and-requirements).
>>> ]
>>> ] A23             The Web API MUST provide means for the
>>> ]                 application to specify the priority to
>>> ]                 apply for outgoing streams and data.
>>>
>>>  Agreed.
>>>
>>>> In case of rtcweb, these priorities are under control of the user  
>>>> or
>>>> at least the web site programmer via the javascript API, and should
>>>> therefore probably be expected to be reasonably long-lived.
>>>
>>>  Yes.
>>>
>>>> Second (and this is what I meant in this current discussion),
>>>> individual packets in a multimedia stream differ in importance. One
>>>> easy way to say this is that a video consists of e.g. I-frames, P-
>>>> frames and B-frames, where the priority is I > P > B.
>>>
>>>  Hmmm...
>>>
>>>  One should tread carefully here. Most video codecs are designed to
>>> expect in-order delivery; and priorities can mess that up.
>>>
>>>  (Or are you actually talking about trying to jiggle probability-of-
>>> loss, rather than what goes first?)
>>
>> Well, there is only so much you can squeeze through a pipe. When  
>> what you'd like to send is more than the network allows, a decision  
>> has to be taken, and that should be done on priorities.
>> In a way that's similar to a loss - it's a decision on the sender  
>> side, to not send certain packets in favor of others.
>
> Except in most cases we're not sitting with a pile of frames/packets  
> and deciding which ones get dropped (liek a router); we're encoding  
> data to meet the current-moment bandwidth requirements.  There is a  
> separate component of feeding data out onto the wire, which would be  
> applied to non-encode-controlled streams like a SCTP/DTLS  
> DataChannel sharing the pipe.  For encoder-based streams, we  
> generally want to control via the encoder not by inducing possible  
> internal delay in how we select packets for transmission.
>
> If the data doesn't fit in the current pipe estimate, we shouldn't  
> be generating it in the first place.  And intermediate nodes don't  
> know our priorities to adjust probability of drop except maybe  
> indirectly via markings, which may instead mess with order of  
> delivery (not a good thing).

I agree about marking, or at least I'd like to keep this a separate  
issue, so I ignore your last sentence above.

About the rest of what you say: I guess that may be case-dependent? I  
would think that always generating exactly the right amount, as you  
describe, is an ideal case - but sometimes you may not be able to get  
exactly that amount, and then you will have to have a (of course, very  
small) send buffer, and then honoring the importance of packets in  
this buffer is useful.

BUT I am waving my hands here. Note that the case I previusly dealt  
with, this Ph.D. thesis:
http://heim.ifi.uio.no/~michawe/research/students/michael_schier-diss_final.pdf
... is about real-time communication all right, but for real-time one- 
way TV transmission with interactive controls, having timing  
constraints which are probably not as strict as e.g. in WebRTC. Maybe  
we're only interested in codecs that would give you just the right  
amount of data at the right moment. I don't know (after all, rmcat is  
not exclusively about WebRTC too). I thought that we could probably  
have such situations with a short send buffer, and then honoring the  
importance of packets in this buffer would be useful, that's all.


>>>  I'm reduced to guesswork to try to attach a meaning to "P-block"  
>>> and
>>> "B-block"... I'd rather not go too far into guesswork...
>>
>>
>> A block is just a part of a frame. Anyway, as I said in my previous  
>> email, for simplicity it's enough to think about I- vs. P- vs. B- 
>> frames, exactly as you describe them here.
>
> Most packets are at a much higher level than blocks, so barring  
> considerable swizzling of internal order it's likely that all  
> packets for a frame should have similar priority and likelyhood of  
> drop.

That depends on various factors and is not always true. Going back to  
the Ph.D. thesis I linked to above, and papers that have come out of  
it, in this case we had a clearly distinguishable importance for  
packets within the same frame, depending on block and macroblock  
types, their dependencies and such. I guess this will greatly depend  
on video configuration and whatnot.


>>>  That sounds like probability-of-loss...
>>
>>
>> ... in the sender. See above.
>
> We should never drop in the sender. :-)

That might have to depend on the granularity of send rates the codec  
produces. But again, I'm waving my hands.


>>> see why that's necessarily true, unless you mean that any  
>>> intelligence
>>> whatsoever at sender-side is forbidden.)
>>
>> I mean that any mechanism-specific intelligence at sender-side  
>> should be forbidden, if we settle for receiver-based congestion  
>> control. That's what I mean with "generic": not mechanism-specific.
>> RMCAT will perhaps standardize 3 mechanisms. A, B and C. Say each  
>> is split into a sender and receiver behavior: A_send and A_recv,  
>> B_send and B_recv, C_send and C_recv. Say they are not compatible,  
>> i.e. A_send cannot talk to B_recv, and C_send cannot talk to A_recv  
>> and so on. Then, for any mechanism X to work, you need both X_send  
>> and X_recv to be in place.
>>
>> Say, we decide to do receiver-based congestion control. Then I  
>> argue that we should agree on G_send, which is a generic sender  
>> behavior. Then, A, B and C only consist of A_recv, B_recv and  
>> C_recv, and you always only need to deploy this receiving end to  
>> use the new mechanism.
>>
>> Symmetrically, the same is true for pure sender-based congestion  
>> control. It's a mix of the two that's hard to deploy.
>
> Not that I'm arguing for it at this moment: but you could design  
> things so that 'generic'/dumb mechanisms are available to support  
> either sender or receiver-based control, with the decision on which  
> done by negotiation or even in-band decision.  It simply requires  
> defining both, and defining a way to select the one that's non- 
> default.
>
> This would allow innovation at either end in the future.
>
> As to whether this is useful... I haven't given it the thought it  
> deserves yet.  But it's quite doable.

This is an interesting idea! Not sure if it's useful either. Perhaps?


> (And on a separate note, a 'generic'/dumb mechanism doesn't mean  
> there can't be feedback from the other side affecting how it  
> operates, or that it can't be a more complex state machine than  
> "send one packet per packet received").  Just that it be relatively  
> simple and well-defined.  (A bit like the whole video-codec "we'll  
> define the quite-complex decoder and let the encoder innovate")

Yes, agree 100%

Cheers,
Michael


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From: "Eggert, Lars" <lars@netapp.com>
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Thread-Topic: interest in streaming the session?
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Hi,

is there an interest in having our Atlanta session streamed online? =
Please reply off-list if so.

Lars=

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From mirja.kuehlewind@ikr.uni-stuttgart.de  Thu Oct 11 04:56:13 2012
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From: Mirja =?iso-8859-1?q?K=FChlewind?= <mirja.kuehlewind@ikr.uni-stuttgart.de>
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References: <5060725F.4090709@alvestrand.no> <ED4A9962-6071-493D-B9E5-05E5F0A09D07@ifi.uio.no> <50748338.7050900@jesup.org>
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Cc: Randell Jesup <randell-ietf@jesup.org>, Michael Welzl <michawe@ifi.uio.no>
Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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Hi,

skipped the part on (intra-flow packet) priorities because that a topic on its 
own. 

> > I mean that any mechanism-specific intelligence at sender-side should
> > be forbidden, if we settle for receiver-based congestion control.
> > That's what I mean with "generic": not mechanism-specific.
> > RMCAT will perhaps standardize 3 mechanisms. A, B and C. Say each is
> > split into a sender and receiver behavior: A_send and A_recv, B_send
> > and B_recv, C_send and C_recv. Say they are not compatible, i.e.
> > A_send cannot talk to B_recv, and C_send cannot talk to A_recv and so
> > on. Then, for any mechanism X to work, you need both X_send and X_recv
> > to be in place.
> >
> > Say, we decide to do receiver-based congestion control. Then I argue
> > that we should agree on G_send, which is a generic sender behavior.
> > Then, A, B and C only consist of A_recv, B_recv and C_recv, and you
> > always only need to deploy this receiving end to use the new mechanism.
> >
> > Symmetrically, the same is true for pure sender-based congestion
> > control. It's a mix of the two that's hard to deploy.
>
> Not that I'm arguing for it at this moment: but you could design things
> so that 'generic'/dumb mechanisms are available to support either sender
> or receiver-based control, with the decision on which done by
> negotiation or even in-band decision.  It simply requires defining both,
> and defining a way to select the one that's non-default.
>
> This would allow innovation at either end in the future.
>
> As to whether this is useful... I haven't given it the thought it
> deserves yet.  But it's quite doable.
>
> (And on a separate note, a 'generic'/dumb mechanism doesn't mean there
> can't be feedback from the other side affecting how it operates, or that
> it can't be a more complex state machine than "send one packet per
> packet received").  Just that it be relatively simple and well-defined.
> (A bit like the whole video-codec "we'll define the quite-complex
> decoder and let the encoder innovate")

I want to note that the algorithm design is mostly independent of this 
question who sends feedback to whom. A certain algorithm should be 
implementable at sender or receiver side. But as soon as you have one 
specific algorithm you might be able to analyze which implementation causes 
how much feedback to send. 

I personally agree that it would be nice to have on one side a very simple and 
well defined state maschine (dumb side) such that only one side needs to be 
changed to deploy new algorithms. But actually having this simple state 
maschine for both sides and then negotiate at the beginning who will do the 
actual control would be an option.

For my understand in TCP the reason why sometmes a receiver control would be 
more useful, is because the receiver might be close to be bottleneck and thus 
might be able to use additional information on the bottleneck (without 
additional signaling and signaling delay to the sender). In case of 
interactive audio/video up- and downstream traffic between to home costumers 
it might be not that clear where the bottleneck is...

Mirja

From michawe@ifi.uio.no  Thu Oct 11 05:05:21 2012
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Subject: Re: [rmcat] [R-C] Sender- vs. receiver-based, yet again...
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Hi,

Just saying, I agree 100% with every word here.

Cheers,
Michael


On Oct 11, 2012, at 1:56 PM, Mirja K=FChlewind wrote:

> Hi,
>
> skipped the part on (intra-flow packet) priorities because that a =20
> topic on its
> own.
>
>>> I mean that any mechanism-specific intelligence at sender-side =20
>>> should
>>> be forbidden, if we settle for receiver-based congestion control.
>>> That's what I mean with "generic": not mechanism-specific.
>>> RMCAT will perhaps standardize 3 mechanisms. A, B and C. Say each is
>>> split into a sender and receiver behavior: A_send and A_recv, B_send
>>> and B_recv, C_send and C_recv. Say they are not compatible, i.e.
>>> A_send cannot talk to B_recv, and C_send cannot talk to A_recv and =20=

>>> so
>>> on. Then, for any mechanism X to work, you need both X_send and =20
>>> X_recv
>>> to be in place.
>>>
>>> Say, we decide to do receiver-based congestion control. Then I argue
>>> that we should agree on G_send, which is a generic sender behavior.
>>> Then, A, B and C only consist of A_recv, B_recv and C_recv, and you
>>> always only need to deploy this receiving end to use the new =20
>>> mechanism.
>>>
>>> Symmetrically, the same is true for pure sender-based congestion
>>> control. It's a mix of the two that's hard to deploy.
>>
>> Not that I'm arguing for it at this moment: but you could design =20
>> things
>> so that 'generic'/dumb mechanisms are available to support either =20
>> sender
>> or receiver-based control, with the decision on which done by
>> negotiation or even in-band decision.  It simply requires defining =20=

>> both,
>> and defining a way to select the one that's non-default.
>>
>> This would allow innovation at either end in the future.
>>
>> As to whether this is useful... I haven't given it the thought it
>> deserves yet.  But it's quite doable.
>>
>> (And on a separate note, a 'generic'/dumb mechanism doesn't mean =20
>> there
>> can't be feedback from the other side affecting how it operates, or =20=

>> that
>> it can't be a more complex state machine than "send one packet per
>> packet received").  Just that it be relatively simple and well-=20
>> defined.
>> (A bit like the whole video-codec "we'll define the quite-complex
>> decoder and let the encoder innovate")
>
> I want to note that the algorithm design is mostly independent of this
> question who sends feedback to whom. A certain algorithm should be
> implementable at sender or receiver side. But as soon as you have one
> specific algorithm you might be able to analyze which implementation =20=

> causes
> how much feedback to send.
>
> I personally agree that it would be nice to have on one side a very =20=

> simple and
> well defined state maschine (dumb side) such that only one side =20
> needs to be
> changed to deploy new algorithms. But actually having this simple =20
> state
> maschine for both sides and then negotiate at the beginning who will =20=

> do the
> actual control would be an option.
>
> For my understand in TCP the reason why sometmes a receiver control =20=

> would be
> more useful, is because the receiver might be close to be bottleneck =20=

> and thus
> might be able to use additional information on the bottleneck (without
> additional signaling and signaling delay to the sender). In case of
> interactive audio/video up- and downstream traffic between to home =20
> costumers
> it might be not that clear where the bottleneck is...
>
> Mirja


From michawe@ifi.uio.no  Thu Oct 11 05:19:19 2012
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Subject: [rmcat] Reliable feedback
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Hi all,

Just a quick thought, below, quoting my own old email again (sorry):

On Oct 4, 2012, at 12:58 PM, Michael Welzl wrote:

> I herewith extend my own analysis of the sender- vs. receiver-based  
> congestion control question, with one (in my opinion) significant  
> plus in each camp:
>
>> Ok, I think you got me convinced. The way I see it, there's a  
>> choice between:
>> 1) sending a lot of feedback, and only reducing it when there is  
>> backwards congestion, by doing congestion control on the backwards  
>> channel like ACK-CC
>> 2) simply always sending as little feedback as possible, based on  
>> the (forward) congestion control mechanism's requirements
>>
>> If we think of e.g. a videoconference, indeed congestion from all  
>> the feedback on the uplink is quite realistic. 1) can then become  
>> rather complicated... you may get congestion on your uplink, and it  
>> might be necessary to reduce the rate of your data stream, or it  
>> might suffice to reduce it on some of the feedback streams, but  
>> which? And then we have to apply priorities between feedback  
>> packets and your payload...
>>
>
>> 2) seems simpler, but it has the problem of making the mechanism  
>> more fragile, and perhaps unnecessarily so (in a situation where it  
>> might be fine to send much more feedback). I say "unnecessarily"  
>> because I'm not convinced about "both channels are always  
>> congested, or will be soon" because codecs may not be greedy, and  
>> streams such as VoIP may simply not generate enough traffic for that.

This increased fragility is, to me, the biggest problem of 2) above  
(feedback minimization via receiver-based control).

But: I think this could largely be compensated for if we make that  
feedback reliable (a la ACKs-of-ACKs).
What I mean is: a sender could piggyback a short message in its data  
packets to the receiver, to acknowledge the reception of feedback. If  
a receiver only sends feedback very rarely, it could use a quite  
aggressive timeout to send the feedback again if it doesn't get an  
acknowledgement fast.

Yes, this message would be piggybacked onto an unreliable data stream,  
but such problems have been fixed before (e.g. TCP's ECN-ECE-CWR  
sequence).

What does everyone think? Good idea, bad idea?
(yes, I know, such stuff would ideally be discussed in the context of  
the RTCP extensions draft, but I clearly don't know enough about RTP/ 
RTCP to lead this effort... happy to contribute though!)


> 2), i.e. receiver-side congestion control, also has the ADVANTAGE  
> that it doesn't misinterpret the loss of feedback as an indication  
> of forward congestion. A receiver-side controller has a clear notion  
> of all packets that have arrived in an interval and knows exactly  
> what was dropped on the way from the sender to the receiver and what  
> wasn't.
>
> 2), i.e. receiver-side congestion control, also has the DISADVANTAGE  
> that tight interaction with the sending application gets harder -  
> e.g., to consider per-packet priorities in the congestion control  
> mechanism. With a receiver-side scheme, this requires richer  
> feedback (in the style: "for high-priority packets: this rate; for  
> low-priority and high-priority packets combined: that rate"). Then,  
> the receiver would have to know how many priorities are there on the  
> sender side to give the right amount of feedback...  this will make  
> things unnecessarily complicated or, as I rather suspect, not  
> happen, and then greatly limit us.
>
>
>
> Cheers,
> Michael
>


From lars@netapp.com  Fri Oct 12 01:29:22 2012
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From: "Eggert, Lars" <lars@netapp.com>
To: Xiaoqing Zhu <zhuxq@alumni.stanford.edu>
Thread-Topic: [rmcat] send agenda requests (was Re: First meeting)
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Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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Hi,

On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu> =
wrote:
> My colleague and I are working on a congestion control scheme for
> real-time conferencing applications at Cisco. A brief version of that
> algorithm was presented at the IAB Workshop in Vancouver (paper #12,
> "Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
> to time constraints, however, we did not get a chance to mention the
> delay-based variant of that scheme, which may be a better fit for
> rmcat.
>=20
> We would like to request for a slot to present in Atlanta. We will
> also submit the ID draft before Oct 15, for further discussions on
> this mailing list.

I look forward to seeing the draft being discussed on the list!

Also, I'm noting your request for a slot. Because the actual CC =
mechanisms are not our most pressing milestones initially - the =
requirements and eval criteria are - we may decide to focus our meeting =
time on those other drafts, if demand for agenda time is high.=20

Lars=

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From john@jlc.net  Fri Oct 12 03:41:04 2012
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Date: Fri, 12 Oct 2012 06:41:02 -0400
From: John Leslie <john@jlc.net>
To: Michael Welzl <michawe@ifi.uio.no>
Message-ID: <20121012104102.GE22359@verdi>
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Cc: rmcat WG <rmcat@ietf.org>
Subject: Re: [rmcat] Reliable feedback
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Michael Welzl <michawe@ifi.uio.no> wrote:
>... 
>>> The way I see it, there's a choice between:
>>> 1) sending a lot of feedback, and only reducing it when there is  
>>>    backwards congestion, by doing congestion control on the
>>>    backwards channel like ACK-CC
>>> 2) simply always sending as little feedback as possible, based on  
>>>    the (forward) congestion control mechanism's requirements

   I'm not commenting on this choice!

> This increased fragility is, to me, the biggest problem of 2) above  
> (feedback minimization via receiver-based control).

   (I'm not commenting on that either.)

> But: I think this could largely be compensated for if we make that  
> feedback reliable (a la ACKs-of-ACKs).

   We should be careful about using the term "reliable".

> What I mean is: a sender could piggyback a short message in its data  
> packets to the receiver, to acknowledge the reception of feedback.
> If a receiver only sends feedback very rarely, it could use a quite  
> aggressive timeout to send the feedback again if it doesn't get an  
> acknowledgement fast.

   This is, indeed, possible.

   But since we're inherently talking "media" here, there will inherently
be a fairly-regular succession of packets from "sender" to "receiver".
Keeping an event-counter and always sending a few low-order bits of that
counter gives us a better approximation of what we want than any attempt
at "reliable" by re-transmission.

   (I also claim that many of our use-cases will involve media in both
directions, and that the remainder deserve periodic "I'm still listening"
messages; so there is always a packet stream to attach these low-order
bits to, although the packet rates may be quite different in the two
directions.)

> Yes, this message would be piggybacked onto an unreliable data stream,  
> but such problems have been fixed before (e.g. TCP's ECN-ECE-CWR  
> sequence).

   ECN-ECE-CWR is doing essentially the event-counter sending one
low-order bit. I prefer to think of the general case, and recommend
two or three low-order bits at least. (Bits just aren't as scarce today
as they were ten years ago -- heck, RFC3168 was trying to squeeze into
a 1981 straitjacket!)

> What does everyone think? Good idea, bad idea?

   I prefer the always-send-counter approach to reliable-by-retransmit.

   I have no objection to _also_ sending an immediate event-report,
but I'm not sure it's worth the trouble. We care about avoiding
self-congestion; but that's mostly a matter of avoiding the TCP habit
of ramping-up too quickly. Depending on an unreliable event report to
tune the ramp-up is a suboptimal design.

> (yes, I know, such stuff would ideally be discussed in the context of  
> the RTCP extensions draft, but I clearly don't know enough about RTP/ 
> RTCP to lead this effort... happy to contribute though!)

   Hint to chairs: I think Michael would be a good Document Editor for
this...

--
John Leslie <john@jlc.net>

From xiaoqzhu@cisco.com  Fri Oct 12 11:42:50 2012
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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: "Eggert, Lars" <lars@netapp.com>
Thread-Topic: [rmcat] send agenda requests (was Re: First meeting)
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Cc: rmcat WG <rmcat@ietf.org>, Xiaoqing Zhu <zhuxq@alumni.stanford.edu>, "Rong Pan \(ropan\)" <ropan@cisco.com>
Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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Hi Lars,=20

Thanks a lot for your prompt response.  Fully agree with the chosen priorit=
ies for the topics=20
to be discussed in the meeting.  I'll keep you posted once the draft is sub=
mitted.=20

Best,
Xiaoqing
=20
On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:

> Hi,
>=20
> On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu> wrote=
:
>> My colleague and I are working on a congestion control scheme for
>> real-time conferencing applications at Cisco. A brief version of that
>> algorithm was presented at the IAB Workshop in Vancouver (paper #12,
>> "Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
>> to time constraints, however, we did not get a chance to mention the
>> delay-based variant of that scheme, which may be a better fit for
>> rmcat.
>>=20
>> We would like to request for a slot to present in Atlanta. We will
>> also submit the ID draft before Oct 15, for further discussions on
>> this mailing list.
>=20
> I look forward to seeing the draft being discussed on the list!
>=20
> Also, I'm noting your request for a slot. Because the actual CC mechanism=
s are not our most pressing milestones initially - the requirements and eva=
l criteria are - we may decide to focus our meeting time on those other dra=
fts, if demand for agenda time is high.=20
>=20
> Lars


From vsingh.ietf@gmail.com  Mon Oct 15 13:33:59 2012
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Greetings,

We have submitted an initial draft outlining the criteria for evaluating
congestion control algorithms. The metrics section is a bit light
on details and will to expand on that in the next revision.

We hope that the draft is a good starting point to get the discussion
started about metrics and scenarios for simulation and testbed experiments.

Cheers,
Varun

Begin forwarded message:


A new version of I-D, draft-singh-rmcat-cc-eval-00.txt
has been successfully submitted by Varun Singh and posted to the
IETF repository.

Filename:	 draft-singh-rmcat-cc-eval
Revision:	 00
Title:		 Evaluating Congestion Control for Interactive Real-time Media.
Creation date:	 2012-10-15
WG ID:		 Individual Submission
Number of pages: 9
URL:
http://www.ietf.org/internet-drafts/draft-singh-rmcat-cc-eval-00.txt
Status:          http://datatracker.ietf.org/doc/draft-singh-rmcat-cc-eval
Htmlized:        http://tools.ietf.org/html/draft-singh-rmcat-cc-eval-00


Abstract:
  The Real-time Transport Protocol (RTP) is used to transmit media in
  telephony and video conferencing applications.  This document
  describes the guidelines to evaluate new congestion control
  algorithms for interactive point-to-point real-time media.




The IETF Secretariat

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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: "Eggert, Lars" <lars@netapp.com>
Thread-Topic: [rmcat] send agenda requests (was Re: First meeting)
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And below is the notification of our submission:

Thanks,
Xiaoqing

--------------------------------------
A new version of I-D, draft-zhu-rmcat-nada-00.txt
has been successfully submitted by Xiaoqing Zhu and posted to the
IETF repository.

Filename:  draft-zhu-rmcat-nada
Revision:  00
Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
Creation date:  2012-10-14
WG ID:  Individual Submission
Number of pages: 10
URL:             http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-0=
0.txt
Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00


Abstract:
  This document describes a scheme named network-assisted dynamic
  adaptation (NADA), a novel congestion control approach for
  interactive real-time media applications, such as video conferencing.
  In the proposed scheme, the sender regulates its sending rate based
  on either implicit or explicit congestion signaling, in a unified
  approach. The scheme can reap the benefits of explicit congestion
  notification markings from network nodes. It also maintains
  consistent sender behavior in the absence of such markings, by
  reacting to queuing delays instead.

  We present here the overall system architecture, recommended
  behaviors at the sender and the receiver, as well as expected network
  nodes operations. Results from extensive simulation studies of the
  proposed scheme are available upon request.




The IETF Secretariat
--------------------------------------
On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:

Hi,

On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu<mailto:z=
huxq@alumni.stanford.edu>> wrote:
My colleague and I are working on a congestion control scheme for
real-time conferencing applications at Cisco. A brief version of that
algorithm was presented at the IAB Workshop in Vancouver (paper #12,
"Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
to time constraints, however, we did not get a chance to mention the
delay-based variant of that scheme, which may be a better fit for
rmcat.

We would like to request for a slot to present in Atlanta. We will
also submit the ID draft before Oct 15, for further discussions on
this mailing list.

I look forward to seeing the draft being discussed on the list!

Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is high.

Lars


--_000_E7175A8E3DC14048A7D3020E0338C1FA0B3EE9xmbalnx13ciscocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
And below is the notification of our submission:&nbsp;
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </spa=
n>&nbsp;draft-zhu-rmcat-nada<br>
Revision:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </spa=
n>&nbsp;00<br>
Title:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </span><=
span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp;NAD=
A: A Unified Congestion Control Scheme for Real-Time Media<br>
Creation date:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span>&nbsp;2012-10-14<br>
WG ID:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </span><=
span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp;Ind=
ividual Submission<br>
Number of pages: 10<br>
URL: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<a href=3D"http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.tx=
t">http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt</a><br>
Status: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"ht=
tp://datatracker.ietf.org/doc/draft-zhu-rmcat-nada">http://datatracker.ietf=
.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools=
.ietf.org/html/draft-zhu-rmcat-nada-00">http://tools.ietf.org/html/draft-zh=
u-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp;&nbsp;This document describes a scheme named network-assisted dynamic=
<br>
&nbsp;&nbsp;adaptation (NADA), a novel congestion control approach for<br>
&nbsp;&nbsp;interactive real-time media applications, such as video confere=
ncing.<br>
&nbsp;&nbsp;In the proposed scheme, the sender regulates its sending rate b=
ased<br>
&nbsp;&nbsp;on either implicit or explicit congestion signaling, in a unifi=
ed<br>
&nbsp;&nbsp;approach. The scheme can reap the benefits of explicit congesti=
on<br>
&nbsp;&nbsp;notification markings from network nodes. It also maintains<br>
&nbsp;&nbsp;consistent sender behavior in the absence of such markings, by<=
br>
&nbsp;&nbsp;reacting to queuing delays instead.<br>
<br>
&nbsp;&nbsp;We present here the overall system architecture, recommended<br=
>
&nbsp;&nbsp;behaviors at the sender and the receiver, as well as expected n=
etwork<br>
&nbsp;&nbsp;nodes operations. Results from extensive simulation studies of =
the<br>
&nbsp;&nbsp;proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a href=3D"mailto:zhuxq@alumni.=
stanford.edu">zhuxq@alumni.stanford.edu</a>&gt; wrote:<br>
<blockquote type=3D"cite">My colleague and I are working on a congestion co=
ntrol scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. A b=
rief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in Va=
ncouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">&quot;Network-Assisted Dynamic Adaptation (NADA):=
 A Design Summary&quot;). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a ch=
ance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may be =
a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present in=
 Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for furth=
er discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
<br>
</div>
</body>
</html>

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From michawe@ifi.uio.no  Tue Oct 16 01:11:42 2012
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Date: Tue, 16 Oct 2012 10:11:32 +0200
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References: <201210021809.19685.mkuehle@ikr.uni-stuttgart.de> <D4D47BCFFE5A004F95D707546AC0D7E9068D4895@SACEXCMBX01-PRD.hq.netapp.com> <D4D47BCFFE5A004F95D707546AC0D7E9068DEB13@SACEXCMBX01-PRD.hq.netapp.com> <CAFT3WbWX3PPXWsry5LTnDT7uWOjEhUoTDV2q0+NH+oKgUzSdKg@mail.gmail.com> <D4D47BCFFE5A004F95D707546AC0D7E918563B83@SACEXCMBX01-PRD.hq.netapp.com> <E7175A8E3DC14048A7D3020E0338C1FA0B3EE9@xmb-aln-x13.cisco.com>
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Cc: rmcat WG <rmcat@ietf.org>, Xiaoqing Zhu <zhuxq@alumni.stanford.edu>, "Eggert, Lars" <lars@netapp.com>, "Rong Pan \(ropan\)" <ropan@cisco.com>
Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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Hi,

I took a quick look at this:

I like the general idea of using as much as possible of existing network =
support - but I'm critical about the actual control you propose.

In section 5.3.2, you suggest to replace eqn. 1 from section 5.3 with =
eqn. 2 when ECN feedback is available. This means to ignore a delay =
signal from now on, and interpret ECN marks in the exact same way as you =
interpreted delay before - this is too simplistic, I think. After all, =
ECN marks require a queue to grow, and this group intends to minimize =
delay... why would you stop using delay once you have ECN in addition?

In practice, using your scheme would probably mean that, as soon as I =
enable RED with ECN on the bottleneck between the two end systems, the =
users' observed delay grows. Not exactly an incentive?

Cheers,
Michael


On 15. okt. 2012, at 23:43, Xiaoqing Zhu (xiaoqzhu) wrote:

> And below is the notification of our submission:=20
>=20
> Thanks,
> Xiaoqing
>=20
> --------------------------------------
> A new version of I-D, draft-zhu-rmcat-nada-00.txt
> has been successfully submitted by Xiaoqing Zhu and posted to the
> IETF repository.
>=20
> Filename:  draft-zhu-rmcat-nada
> Revision:  00
> Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
> Creation date:  2012-10-14
> WG ID:  Individual Submission
> Number of pages: 10
> URL:             =
http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
> Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00
>=20
>=20
> Abstract:
>   This document describes a scheme named network-assisted dynamic
>   adaptation (NADA), a novel congestion control approach for
>   interactive real-time media applications, such as video =
conferencing.
>   In the proposed scheme, the sender regulates its sending rate based
>   on either implicit or explicit congestion signaling, in a unified
>   approach. The scheme can reap the benefits of explicit congestion
>   notification markings from network nodes. It also maintains
>   consistent sender behavior in the absence of such markings, by
>   reacting to queuing delays instead.
>=20
>   We present here the overall system architecture, recommended
>   behaviors at the sender and the receiver, as well as expected =
network
>   nodes operations. Results from extensive simulation studies of the
>   proposed scheme are available upon request.
>=20
>=20
>=20
>=20
> The IETF Secretariat
> --------------------------------------
> On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:
>=20
>> Hi,
>>=20
>> On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu> =
wrote:
>>> My colleague and I are working on a congestion control scheme for
>>> real-time conferencing applications at Cisco. A brief version of =
that
>>> algorithm was presented at the IAB Workshop in Vancouver (paper #12,
>>> "Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
>>> to time constraints, however, we did not get a chance to mention the
>>> delay-based variant of that scheme, which may be a better fit for
>>> rmcat.
>>>=20
>>> We would like to request for a slot to present in Atlanta. We will
>>> also submit the ID draft before Oct 15, for further discussions on
>>> this mailing list.
>>=20
>> I look forward to seeing the draft being discussed on the list!
>>=20
>> Also, I'm noting your request for a slot. Because the actual CC =
mechanisms are not our most pressing milestones initially - the =
requirements and eval criteria are - we may decide to focus our meeting =
time on those other drafts, if demand for agenda time is high.=20
>>=20
>> Lars
>=20


--Apple-Mail=_47AD916F-E81B-4512-8265-7D9492391FB7
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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Hi,<div><br></div><div>I took a quick look at =
this:</div><div><br></div><div>I like the general idea of using as much =
as possible of existing network support - but I'm critical about the =
actual control you propose.</div><div><br></div><div>In section 5.3.2, =
you suggest to replace eqn. 1 from section 5.3 with eqn. 2 when ECN =
feedback is available. This means to ignore a delay signal from now on, =
and interpret ECN marks in the exact same way as you interpreted delay =
before - this is too simplistic, I think. After all, ECN marks require a =
queue to grow, and this group intends to minimize delay... why would you =
stop using delay once you have ECN in =
addition?</div><div><br></div><div>In practice, using your scheme would =
probably mean that, as soon as I enable RED with ECN on the bottleneck =
between the two end systems, the users' observed delay grows. Not =
exactly an =
incentive?</div><div><br></div><div>Cheers,</div><div>Michael</div><div><b=
r></div><div><br><div><div>On 15. okt. 2012, at 23:43, Xiaoqing Zhu =
(xiaoqzhu) wrote:</div><br class=3D"Apple-interchange-newline"><blockquote=
 type=3D"cite">

<meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dus-ascii">

<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">
And below is the notification of our submission:&nbsp;
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span>&nbsp;draft-zhu-rmcat-nada<br>
Revision:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span>&nbsp;00<br>
Title:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span><span class=3D"Apple-tab-span" style=3D"white-space: pre; =
"></span>&nbsp;NADA: A Unified Congestion Control Scheme for Real-Time =
Media<br>
Creation date:<span class=3D"Apple-tab-span" style=3D"white-space: pre; =
"> </span>&nbsp;2012-10-14<br>
WG ID:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span><span class=3D"Apple-tab-span" style=3D"white-space: pre; =
"></span>&nbsp;Individual Submission<br>
Number of pages: 10<br>
URL: =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a=
 =
href=3D"http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt">h=
ttp://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt</a><br>
Status: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a =
href=3D"http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada">http://datat=
racker.ietf.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a =
href=3D"http://tools.ietf.org/html/draft-zhu-rmcat-nada-00">http://tools.i=
etf.org/html/draft-zhu-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp;&nbsp;This document describes a scheme named network-assisted =
dynamic<br>
&nbsp;&nbsp;adaptation (NADA), a novel congestion control approach =
for<br>
&nbsp;&nbsp;interactive real-time media applications, such as video =
conferencing.<br>
&nbsp;&nbsp;In the proposed scheme, the sender regulates its sending =
rate based<br>
&nbsp;&nbsp;on either implicit or explicit congestion signaling, in a =
unified<br>
&nbsp;&nbsp;approach. The scheme can reap the benefits of explicit =
congestion<br>
&nbsp;&nbsp;notification markings from network nodes. It also =
maintains<br>
&nbsp;&nbsp;consistent sender behavior in the absence of such markings, =
by<br>
&nbsp;&nbsp;reacting to queuing delays instead.<br>
<br>
&nbsp;&nbsp;We present here the overall system architecture, =
recommended<br>
&nbsp;&nbsp;behaviors at the sender and the receiver, as well as =
expected network<br>
&nbsp;&nbsp;nodes operations. Results from extensive simulation studies =
of the<br>
&nbsp;&nbsp;proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a =
href=3D"mailto:zhuxq@alumni.stanford.edu">zhuxq@alumni.stanford.edu</a>&gt=
; wrote:<br>
<blockquote type=3D"cite">My colleague and I are working on a congestion =
control scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. =
A brief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in =
Vancouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">"Network-Assisted Dynamic Adaptation (NADA): A =
Design Summary"). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a =
chance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may =
be a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present =
in Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for =
further discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I'm noting your request for a slot. Because the actual CC =
mechanisms are not our most pressing milestones initially - the =
requirements and eval criteria are - we may decide to focus our meeting =
time on those other drafts, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
<br>
</div>
</div>

</blockquote></div><br></div></body></html>=

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From: Piers O'Hanlon <p.ohanlon@gmail.com>
In-Reply-To: <E7175A8E3DC14048A7D3020E0338C1FA0B3EE9@xmb-aln-x13.cisco.com>
Date: Tue, 16 Oct 2012 09:54:39 +0100
Message-Id: <DE34BE14-280A-4A63-848E-DD45EA383CEC@gmail.com>
References: <201210021809.19685.mkuehle@ikr.uni-stuttgart.de> <D4D47BCFFE5A004F95D707546AC0D7E9068D4895@SACEXCMBX01-PRD.hq.netapp.com> <D4D47BCFFE5A004F95D707546AC0D7E9068DEB13@SACEXCMBX01-PRD.hq.netapp.com> <CAFT3WbWX3PPXWsry5LTnDT7uWOjEhUoTDV2q0+NH+oKgUzSdKg@mail.gmail.com> <D4D47BCFFE5A004F95D707546AC0D7E918563B83@SACEXCMBX01-PRD.hq.netapp.com> <E7175A8E3DC14048A7D3020E0338C1FA0B3EE9@xmb-aln-x13.cisco.com>
To: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
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Cc: rmcat WG <rmcat@ietf.org>, Xiaoqing Zhu <zhuxq@alumni.stanford.edu>, "Eggert, Lars" <lars@netapp.com>, "Rong Pan \(ropan\)" <ropan@cisco.com>
Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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Hi,

I too had a quick look at the proposal.

I was interested as to how kappa is calculated in equation (1) or =
whether it is a fixed value?

Also your slow start algorithm only seems to be time dependent - =
literally it appears to start slowly from t_0 to some preset time (T) - =
is there any mechanism to terminate slow start on loss/marked packets? =
Does slow start ever occur mid stream? (e.g. after a silence period?)

Out of interest it was mentioned that there were extensive simulation =
studies - would it be possible to post a simulation of multiple NADA =
flows competing. And also a competition of NADA with a TCP flow?

Thanks,

Piers.

On 15 Oct 2012, at 22:43, Xiaoqing Zhu (xiaoqzhu) wrote:

> And below is the notification of our submission:=20
>=20
> Thanks,
> Xiaoqing
>=20
> --------------------------------------
> A new version of I-D, draft-zhu-rmcat-nada-00.txt
> has been successfully submitted by Xiaoqing Zhu and posted to the
> IETF repository.
>=20
> Filename:  draft-zhu-rmcat-nada
> Revision:  00
> Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
> Creation date:  2012-10-14
> WG ID:  Individual Submission
> Number of pages: 10
> URL:             =
http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
> Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00
>=20
>=20
> Abstract:
>   This document describes a scheme named network-assisted dynamic
>   adaptation (NADA), a novel congestion control approach for
>   interactive real-time media applications, such as video =
conferencing.
>   In the proposed scheme, the sender regulates its sending rate based
>   on either implicit or explicit congestion signaling, in a unified
>   approach. The scheme can reap the benefits of explicit congestion
>   notification markings from network nodes. It also maintains
>   consistent sender behavior in the absence of such markings, by
>   reacting to queuing delays instead.
>=20
>   We present here the overall system architecture, recommended
>   behaviors at the sender and the receiver, as well as expected =
network
>   nodes operations. Results from extensive simulation studies of the
>   proposed scheme are available upon request.
>=20
>=20
>=20
>=20
> The IETF Secretariat
> --------------------------------------
> On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:
>=20
>> Hi,
>>=20
>> On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu> =
wrote:
>>> My colleague and I are working on a congestion control scheme for
>>> real-time conferencing applications at Cisco. A brief version of =
that
>>> algorithm was presented at the IAB Workshop in Vancouver (paper #12,
>>> "Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
>>> to time constraints, however, we did not get a chance to mention the
>>> delay-based variant of that scheme, which may be a better fit for
>>> rmcat.
>>>=20
>>> We would like to request for a slot to present in Atlanta. We will
>>> also submit the ID draft before Oct 15, for further discussions on
>>> this mailing list.
>>=20
>> I look forward to seeing the draft being discussed on the list!
>>=20
>> Also, I'm noting your request for a slot. Because the actual CC =
mechanisms are not our most pressing milestones initially - the =
requirements and eval criteria are - we may decide to focus our meeting =
time on those other drafts, if demand for agenda time is high.=20
>>=20
>> Lars
>=20


--Apple-Mail=_83DB50FE-0A30-47A5-8CE3-3F5233EB7349
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Hi,<div><br></div><div>I too had a quick look at the =
proposal.</div><div><br></div><div>I was interested as to how kappa is =
calculated in equation (1) or whether it is a fixed =
value?</div><div><br></div><div>Also your slow start algorithm only =
seems to be time dependent - literally it appears to start slowly from =
t_0 to some preset time (T) - is there any mechanism to terminate slow =
start on loss/marked packets? Does slow start ever occur mid stream? =
(e.g. after a silence period?)</div><div><br></div><div>Out of interest =
it was mentioned that there were extensive simulation studies - would it =
be possible to post a simulation of multiple NADA flows competing. And =
also a competition of NADA with a TCP =
flow?</div><div><br></div><div>Thanks,</div><div><br></div><div>Piers.</di=
v><div><br><div><div>On 15 Oct 2012, at 22:43, Xiaoqing Zhu (xiaoqzhu) =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite">

<meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dus-ascii">

<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">
And below is the notification of our submission:&nbsp;
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span>&nbsp;draft-zhu-rmcat-nada<br>
Revision:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span>&nbsp;00<br>
Title:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span><span class=3D"Apple-tab-span" style=3D"white-space: pre; =
"></span>&nbsp;NADA: A Unified Congestion Control Scheme for Real-Time =
Media<br>
Creation date:<span class=3D"Apple-tab-span" style=3D"white-space: pre; =
"> </span>&nbsp;2012-10-14<br>
WG ID:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span><span class=3D"Apple-tab-span" style=3D"white-space: pre; =
"></span>&nbsp;Individual Submission<br>
Number of pages: 10<br>
URL: =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a=
 =
href=3D"http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt">h=
ttp://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt</a><br>
Status: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a =
href=3D"http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada">http://datat=
racker.ietf.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a =
href=3D"http://tools.ietf.org/html/draft-zhu-rmcat-nada-00">http://tools.i=
etf.org/html/draft-zhu-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp;&nbsp;This document describes a scheme named network-assisted =
dynamic<br>
&nbsp;&nbsp;adaptation (NADA), a novel congestion control approach =
for<br>
&nbsp;&nbsp;interactive real-time media applications, such as video =
conferencing.<br>
&nbsp;&nbsp;In the proposed scheme, the sender regulates its sending =
rate based<br>
&nbsp;&nbsp;on either implicit or explicit congestion signaling, in a =
unified<br>
&nbsp;&nbsp;approach. The scheme can reap the benefits of explicit =
congestion<br>
&nbsp;&nbsp;notification markings from network nodes. It also =
maintains<br>
&nbsp;&nbsp;consistent sender behavior in the absence of such markings, =
by<br>
&nbsp;&nbsp;reacting to queuing delays instead.<br>
<br>
&nbsp;&nbsp;We present here the overall system architecture, =
recommended<br>
&nbsp;&nbsp;behaviors at the sender and the receiver, as well as =
expected network<br>
&nbsp;&nbsp;nodes operations. Results from extensive simulation studies =
of the<br>
&nbsp;&nbsp;proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a =
href=3D"mailto:zhuxq@alumni.stanford.edu">zhuxq@alumni.stanford.edu</a>&gt=
; wrote:<br>
<blockquote type=3D"cite">My colleague and I are working on a congestion =
control scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. =
A brief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in =
Vancouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">"Network-Assisted Dynamic Adaptation (NADA): A =
Design Summary"). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a =
chance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may =
be a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present =
in Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for =
further discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I'm noting your request for a slot. Because the actual CC =
mechanisms are not our most pressing milestones initially - the =
requirements and eval criteria are - we may decide to focus our meeting =
time on those other drafts, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
<br>
</div>
</div>

</blockquote></div><br></div></body></html>=

--Apple-Mail=_83DB50FE-0A30-47A5-8CE3-3F5233EB7349--

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Date: Tue, 16 Oct 2012 11:01:21 +0200
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From: Luca De Cicco <ldecicco@gmail.com>
To: Varun Singh <vsingh.ietf@gmail.com>
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Dear Varun,

I've read the draft and I have a couple of comments.

1) regarding the definition of stability which is given in the draft:

Stability: is the period of time when the endpoint's encoding
         rate is relatively stable, i.e., the bandwidth utilization is
         constant.

I would use "steady-state" instead of "stability" since the latter has a
formal definition which has a precise meaning. Moreover, it has to be
noticed that "steady-state" can be reached if and only if the
available bandwidth is constant.

2) for what concerns the video quality indices, it is well known
that PSNR is not well related to the QoE. (See for instance
http://www.pscr.gov/about_pscr/press/video/video_quality_experts_interview_sigmm-122009.pdf)
Are we sure we want to cite PSNR?

Cheers,
--
Luca De Cicco, PhD, Eng.
Politecnico di Bari
Dipartimento di Elettrotecnica ed Elettronica
Via Re David, 200 - Bari - ITALY
Office: +39 080 596 3851


On Mon, Oct 15, 2012 at 10:33 PM, Varun Singh <vsingh.ietf@gmail.com> wrote:
>
> Greetings,
>
> We have submitted an initial draft outlining the criteria for evaluating
> congestion control algorithms. The metrics section is a bit light
> on details and will to expand on that in the next revision.
>
> We hope that the draft is a good starting point to get the discussion
> started about metrics and scenarios for simulation and testbed experiments.
>
> Cheers,
> Varun
>
> Begin forwarded message:
>
>
> A new version of I-D, draft-singh-rmcat-cc-eval-00.txt
> has been successfully submitted by Varun Singh and posted to the
> IETF repository.
>
> Filename:        draft-singh-rmcat-cc-eval
> Revision:        00
> Title:           Evaluating Congestion Control for Interactive Real-time Media.
> Creation date:   2012-10-15
> WG ID:           Individual Submission
> Number of pages: 9
> URL:
> http://www.ietf.org/internet-drafts/draft-singh-rmcat-cc-eval-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-singh-rmcat-cc-eval
> Htmlized:        http://tools.ietf.org/html/draft-singh-rmcat-cc-eval-00
>
>
> Abstract:
>   The Real-time Transport Protocol (RTP) is used to transmit media in
>   telephony and video conferencing applications.  This document
>   describes the guidelines to evaluate new congestion control
>   algorithms for interactive point-to-point real-time media.
>
>
>
>
> The IETF Secretariat

From vsingh.ietf@gmail.com  Tue Oct 16 03:57:34 2012
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Tue, 16 Oct 2012 13:57:12 +0300
Message-ID: <CAEbPqrwcfOXaUB+sG1vWt_3Cis3bZx+hvshvtPtVhA3ge2rxOw@mail.gmail.com>
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Hi Luca,

Thanks for your comments, responses inline.

On Tue, Oct 16, 2012 at 12:01 PM, Luca De Cicco <ldecicco@gmail.com> wrote:
[snip]
> 1) regarding the definition of stability which is given in the draft:
>
> Stability: is the period of time when the endpoint's encoding
>          rate is relatively stable, i.e., the bandwidth utilization is
>          constant.
>
> I would use "steady-state" instead of "stability" since the latter has a
> formal definition which has a precise meaning. Moreover, it has to be

True, will change stability to steady-state.

> noticed that "steady-state" can be reached if and only if the
> available bandwidth is constant.


I defined steady-state as a period when bandwidth utilization is
constant (not close to 1).
Apart from the situation when the congestion control is operating
close to the bottleneck capacity, it can also have constant utilization
if it is under-utilizing the link or detects cross-traffic and
attempts to be fair.

The metrics section has intentionally been left a bit open for discussion
because I want to tune the definitions to match the exact requirements
of the congestion control.


>
> 2) for what concerns the video quality indices, it is well known
> that PSNR is not well related to the QoE. (See for instance
> http://www.pscr.gov/about_pscr/press/video/video_quality_experts_interview_sigmm-122009.pdf)
> Are we sure we want to cite PSNR?
>

I believe we need some objective metric for video either
PSNR or something else. I am happy to include or
update the document based on that discussion.
(Note: I have not read the above document but
will go through it.)

However, when choosing these quality metrics we should be aware
of their defined operating range (some models are limited to specific
range of video resolutions or bit rates, etc.).

Cheers,
Varun


> Cheers,
> --
> Luca De Cicco, PhD, Eng.
> Politecnico di Bari
> Dipartimento di Elettrotecnica ed Elettronica
> Via Re David, 200 - Bari - ITALY
> Office: +39 080 596 3851
>
>
> On Mon, Oct 15, 2012 at 10:33 PM, Varun Singh <vsingh.ietf@gmail.com> wrote:
>>
>> Greetings,
>>
>> We have submitted an initial draft outlining the criteria for evaluating
>> congestion control algorithms. The metrics section is a bit light
>> on details and will to expand on that in the next revision.
>>
>> We hope that the draft is a good starting point to get the discussion
>> started about metrics and scenarios for simulation and testbed experiments.
>>
>> Cheers,
>> Varun
>>
>> Begin forwarded message:
>>
>>
>> A new version of I-D, draft-singh-rmcat-cc-eval-00.txt
>> has been successfully submitted by Varun Singh and posted to the
>> IETF repository.
>>
>> Filename:        draft-singh-rmcat-cc-eval
>> Revision:        00
>> Title:           Evaluating Congestion Control for Interactive Real-time Media.
>> Creation date:   2012-10-15
>> WG ID:           Individual Submission
>> Number of pages: 9
>> URL:
>> http://www.ietf.org/internet-drafts/draft-singh-rmcat-cc-eval-00.txt
>> Status:          http://datatracker.ietf.org/doc/draft-singh-rmcat-cc-eval
>> Htmlized:        http://tools.ietf.org/html/draft-singh-rmcat-cc-eval-00
>>
>>
>> Abstract:
>>   The Real-time Transport Protocol (RTP) is used to transmit media in
>>   telephony and video conferencing applications.  This document
>>   describes the guidelines to evaluate new congestion control
>>   algorithms for interactive point-to-point real-time media.
>>
>>
>>
>>
>> The IETF Secretariat



-- 
http://www.netlab.tkk.fi/~varun/

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On Tue, Oct 16, 2012 at 12:57 PM, Varun Singh <vsingh.ietf@gmail.com> wrote:

> Hi Luca,
>
> Thanks for your comments, responses inline.
>
> On Tue, Oct 16, 2012 at 12:01 PM, Luca De Cicco <ldecicco@gmail.com>
> wrote:
> [snip]
> > 1) regarding the definition of stability which is given in the draft:
> >
> > Stability: is the period of time when the endpoint's encoding
> >          rate is relatively stable, i.e., the bandwidth utilization is
> >          constant.
> >
> > I would use "steady-state" instead of "stability" since the latter has a
> > formal definition which has a precise meaning. Moreover, it has to be
>
> True, will change stability to steady-state.
>
> > noticed that "steady-state" can be reached if and only if the
> > available bandwidth is constant.
>
>
> I defined steady-state as a period when bandwidth utilization is
> constant (not close to 1).
> Apart from the situation when the congestion control is operating
> close to the bottleneck capacity, it can also have constant utilization
> if it is under-utilizing the link or detects cross-traffic and
> attempts to be fair.
>
> The metrics section has intentionally been left a bit open for discussion
> because I want to tune the definitions to match the exact requirements
> of the congestion control.
>
>
> >
> > 2) for what concerns the video quality indices, it is well known
> > that PSNR is not well related to the QoE. (See for instance
> >
> http://www.pscr.gov/about_pscr/press/video/video_quality_experts_interview_sigmm-122009.pdf
> )
> > Are we sure we want to cite PSNR?
> >
>
> I believe we need some objective metric for video either
> PSNR or something else. I am happy to include or
> update the document based on that discussion.
> (Note: I have not read the above document but
> will go through it.)
>
> However, when choosing these quality metrics we should be aware
> of their defined operating range (some models are limited to specific
> range of video resolutions or bit rates, etc.).
>
>
I agree that PSNR isn't likely to be a good choice, I would assume that we
at least need a metric which takes both the video quality and the delay
into account? The experienced video quality depends on several aspects
other than the congestion control, such as encoder settings, transmitted
spatial and temporal resolution, video jitter buffer algorithm at the
receiver, etc. Therefore I'm not sure it's a good idea to evaluate our
congestion control algorithms based on video quality. If possible I would
rather see us concentrate on simpler metrics as those you described earlier
in your draft, which are more related to how the algorithm actually adapts
to changes in bandwidth.


> Cheers,
> Varun
>
>
> > Cheers,
> > --
> > Luca De Cicco, PhD, Eng.
> > Politecnico di Bari
> > Dipartimento di Elettrotecnica ed Elettronica
> > Via Re David, 200 - Bari - ITALY
> > Office: +39 080 596 3851
> >
> >
> > On Mon, Oct 15, 2012 at 10:33 PM, Varun Singh <vsingh.ietf@gmail.com>
> wrote:
> >>
> >> Greetings,
> >>
> >> We have submitted an initial draft outlining the criteria for evaluating
> >> congestion control algorithms. The metrics section is a bit light
> >> on details and will to expand on that in the next revision.
> >>
> >> We hope that the draft is a good starting point to get the discussion
> >> started about metrics and scenarios for simulation and testbed
> experiments.
> >>
> >> Cheers,
> >> Varun
> >>
> >> Begin forwarded message:
> >>
> >>
> >> A new version of I-D, draft-singh-rmcat-cc-eval-00.txt
> >> has been successfully submitted by Varun Singh and posted to the
> >> IETF repository.
> >>
> >> Filename:        draft-singh-rmcat-cc-eval
> >> Revision:        00
> >> Title:           Evaluating Congestion Control for Interactive
> Real-time Media.
> >> Creation date:   2012-10-15
> >> WG ID:           Individual Submission
> >> Number of pages: 9
> >> URL:
> >> http://www.ietf.org/internet-drafts/draft-singh-rmcat-cc-eval-00.txt
> >> Status:
> http://datatracker.ietf.org/doc/draft-singh-rmcat-cc-eval
> >> Htmlized:
> http://tools.ietf.org/html/draft-singh-rmcat-cc-eval-00
> >>
> >>
> >> Abstract:
> >>   The Real-time Transport Protocol (RTP) is used to transmit media in
> >>   telephony and video conferencing applications.  This document
> >>   describes the guidelines to evaluate new congestion control
> >>   algorithms for interactive point-to-point real-time media.
> >>
> >>
> >>
> >>
> >> The IETF Secretariat
>
>
>
> --
> http://www.netlab.tkk.fi/~varun/
>

--485b397dd1c9c2973104cc2b7622
Content-Type: text/html; charset=ISO-8859-1
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<br><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Tue, O=
ct 16, 2012 at 12:57 PM, Varun Singh <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:vsingh.ietf@gmail.com" target=3D"_blank" class=3D"cremed">vsingh.ietf@gma=
il.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi Luca,<br>
<br>
Thanks for your comments, responses inline.<br>
<br>
On Tue, Oct 16, 2012 at 12:01 PM, Luca De Cicco &lt;<a href=3D"mailto:ldeci=
cco@gmail.com" class=3D"cremed">ldecicco@gmail.com</a>&gt; wrote:<br>
[snip]<br>
<div class=3D"im">&gt; 1) regarding the definition of stability which is gi=
ven in the draft:<br>
&gt;<br>
&gt; Stability: is the period of time when the endpoint&#39;s encoding<br>
&gt; =A0 =A0 =A0 =A0 =A0rate is relatively stable, i.e., the bandwidth util=
ization is<br>
&gt; =A0 =A0 =A0 =A0 =A0constant.<br>
&gt;<br>
&gt; I would use &quot;steady-state&quot; instead of &quot;stability&quot; =
since the latter has a<br>
&gt; formal definition which has a precise meaning. Moreover, it has to be<=
br>
<br>
</div>True, will change stability to steady-state.<br>
<div class=3D"im"><br>
&gt; noticed that &quot;steady-state&quot; can be reached if and only if th=
e<br>
&gt; available bandwidth is constant.<br>
<br>
<br>
</div>I defined steady-state as a period when bandwidth utilization is<br>
constant (not close to 1).<br>
Apart from the situation when the congestion control is operating<br>
close to the bottleneck capacity, it can also have constant utilization<br>
if it is under-utilizing the link or detects cross-traffic and<br>
attempts to be fair.<br>
<br>
The metrics section has intentionally been left a bit open for discussion<b=
r>
because I want to tune the definitions to match the exact requirements<br>
of the congestion control.<br>
<div class=3D"im"><br>
<br>
&gt;<br>
&gt; 2) for what concerns the video quality indices, it is well known<br>
&gt; that PSNR is not well related to the QoE. (See for instance<br>
&gt; <a href=3D"http://www.pscr.gov/about_pscr/press/video/video_quality_ex=
perts_interview_sigmm-122009.pdf" target=3D"_blank" class=3D"cremed">http:/=
/www.pscr.gov/about_pscr/press/video/video_quality_experts_interview_sigmm-=
122009.pdf</a>)<br>

&gt; Are we sure we want to cite PSNR?<br>
&gt;<br>
<br>
</div>I believe we need some objective metric for video either<br>
PSNR or something else. I am happy to include or<br>
update the document based on that discussion.<br>
(Note: I have not read the above document but<br>
will go through it.)<br>
<br>
However, when choosing these quality metrics we should be aware<br>
of their defined operating range (some models are limited to specific<br>
range of video resolutions or bit rates, etc.).<br>
<br></blockquote><div><br></div><div>I agree that PSNR isn&#39;t likely to =
be a good choice, I would assume that we at least need a metric which takes=
 both the video quality and the delay into account? The experienced video q=
uality depends on several aspects other than the congestion control, such a=
s encoder settings, transmitted spatial and temporal resolution, video jitt=
er buffer algorithm at the receiver, etc. Therefore I&#39;m not sure it&#39=
;s a good idea to evaluate our congestion control algorithms based on video=
 quality. If possible I would rather see us concentrate on simpler metrics =
as those you described earlier in your draft, which are more related to how=
 the algorithm actually adapts to changes in bandwidth.</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
Cheers,<br>
Varun<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
&gt; Cheers,<br>
&gt; --<br>
&gt; Luca De Cicco, PhD, Eng.<br>
&gt; Politecnico di Bari<br>
&gt; Dipartimento di Elettrotecnica ed Elettronica<br>
&gt; Via Re David, 200 - Bari - ITALY<br>
&gt; Office: <a href=3D"tel:%2B39%20080%20596%203851" value=3D"+39080596385=
1" class=3D"cremed">+39 080 596 3851</a><br>
&gt;<br>
&gt;<br>
&gt; On Mon, Oct 15, 2012 at 10:33 PM, Varun Singh &lt;<a href=3D"mailto:vs=
ingh.ietf@gmail.com" class=3D"cremed">vsingh.ietf@gmail.com</a>&gt; wrote:<=
br>
&gt;&gt;<br>
&gt;&gt; Greetings,<br>
&gt;&gt;<br>
&gt;&gt; We have submitted an initial draft outlining the criteria for eval=
uating<br>
&gt;&gt; congestion control algorithms. The metrics section is a bit light<=
br>
&gt;&gt; on details and will to expand on that in the next revision.<br>
&gt;&gt;<br>
&gt;&gt; We hope that the draft is a good starting point to get the discuss=
ion<br>
&gt;&gt; started about metrics and scenarios for simulation and testbed exp=
eriments.<br>
&gt;&gt;<br>
&gt;&gt; Cheers,<br>
&gt;&gt; Varun<br>
&gt;&gt;<br>
&gt;&gt; Begin forwarded message:<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; A new version of I-D, draft-singh-rmcat-cc-eval-00.txt<br>
&gt;&gt; has been successfully submitted by Varun Singh and posted to the<b=
r>
&gt;&gt; IETF repository.<br>
&gt;&gt;<br>
&gt;&gt; Filename: =A0 =A0 =A0 =A0draft-singh-rmcat-cc-eval<br>
&gt;&gt; Revision: =A0 =A0 =A0 =A000<br>
&gt;&gt; Title: =A0 =A0 =A0 =A0 =A0 Evaluating Congestion Control for Inter=
active Real-time Media.<br>
&gt;&gt; Creation date: =A0 2012-10-15<br>
&gt;&gt; WG ID: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
&gt;&gt; Number of pages: 9<br>
&gt;&gt; URL:<br>
&gt;&gt; <a href=3D"http://www.ietf.org/internet-drafts/draft-singh-rmcat-c=
c-eval-00.txt" target=3D"_blank" class=3D"cremed">http://www.ietf.org/inter=
net-drafts/draft-singh-rmcat-cc-eval-00.txt</a><br>
&gt;&gt; Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/=
doc/draft-singh-rmcat-cc-eval" target=3D"_blank" class=3D"cremed">http://da=
tatracker.ietf.org/doc/draft-singh-rmcat-cc-eval</a><br>
&gt;&gt; Htmlized: =A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/dra=
ft-singh-rmcat-cc-eval-00" target=3D"_blank" class=3D"cremed">http://tools.=
ietf.org/html/draft-singh-rmcat-cc-eval-00</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; Abstract:<br>
&gt;&gt; =A0 The Real-time Transport Protocol (RTP) is used to transmit med=
ia in<br>
&gt;&gt; =A0 telephony and video conferencing applications. =A0This documen=
t<br>
&gt;&gt; =A0 describes the guidelines to evaluate new congestion control<br=
>
&gt;&gt; =A0 algorithms for interactive point-to-point real-time media.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; The IETF Secretariat<br>
<br>
<br>
<br>
</div></div><span class=3D"HOEnZb"><font color=3D"#888888">--<br>
<a href=3D"http://www.netlab.tkk.fi/~varun/" target=3D"_blank" class=3D"cre=
med">http://www.netlab.tkk.fi/~varun/</a><br>
</font></span></blockquote></div><br></div>

--485b397dd1c9c2973104cc2b7622--

From vsingh.ietf@gmail.com  Tue Oct 16 05:35:04 2012
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Tue, 16 Oct 2012 15:34:40 +0300
Message-ID: <CAEbPqryxVzPsV0EXw5JuJqE8fci+qx_TxeGt+=ftg1JeEUvTXg@mail.gmail.com>
To: Stefan Holmer <stefan@webrtc.org>
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Cc: rmcat WG <rmcat@ietf.org>, Luca De Cicco <ldecicco@gmail.com>
Subject: Re: [rmcat] Video Quality for RMCAT
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Hi Stefan,

changing the email subject and snipping away the rest of the message.

Comments inline.

>> >
>> > 2) for what concerns the video quality indices, it is well known
>> > that PSNR is not well related to the QoE. (See for instance
>> >
>> > http://www.pscr.gov/about_pscr/press/video/video_quality_experts_interview_sigmm-122009.pdf)
>> > Are we sure we want to cite PSNR?
>> >
>>
>> I believe we need some objective metric for video either
>> PSNR or something else. I am happy to include or
>> update the document based on that discussion.
>> (Note: I have not read the above document but
>> will go through it.)
>>
>> However, when choosing these quality metrics we should be aware
>> of their defined operating range (some models are limited to specific
>> range of video resolutions or bit rates, etc.).
>>
>
> I agree that PSNR isn't likely to be a good choice, I would assume that we
> at least need a metric which takes both the video quality and the delay into
> account? The experienced video quality depends on several aspects other than

Apart from delay there is also balancing throughput and loss.
In the scenario of a single RTP flow we could come up with a
simple metric but with competing traffic it may be more difficult.
By introducing a video quality metric in those scenarios,
I was intending to quantify the balance of throughput, loss and delay.


> the congestion control, such as encoder settings, transmitted spatial and
> temporal resolution, video jitter buffer algorithm at the receiver, etc.

Since we are talking about interactive real-time media, we could *agree* on
running a baseline setup with preset parameters and reference video
sequences. :)

> Therefore I'm not sure it's a good idea to evaluate our congestion control
> algorithms based on video quality. If possible I would rather see us
> concentrate on simpler metrics as those you described earlier in your draft,
> which are more related to how the algorithm actually adapts to changes in
> bandwidth.

In my opinion, if we can come up with a good QoE metric then it
would be nice to have in addition to the throughput, loss and delay metrics.
OTOH if we come up with a quality metric that is based on the relationship
of the three then we may have a single metric to compare each algorithm.
I am not sure which of these two methods would be better, but I am happy
to hear more arguments for and against QoE, also other QoE methods
before we decide on a quality metric.


Regards,
Varun

From michawe@ifi.uio.no  Tue Oct 16 06:01:36 2012
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Subject: Re: [rmcat] Video Quality for RMCAT
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Sorry if this is stupid, perhaps I'm missing something - but:

even if we could agree on a baseline test setup, standard movies etc.: =
isn't the very idea of a QoE metric inevitably bound to the codec, which =
then binds evaluating congestion control mechanisms to a debate that =
this really shouldn't be a part of?

Cheers,
Michael


On 16. okt. 2012, at 14:34, Varun Singh wrote:

> Hi Stefan,
>=20
> changing the email subject and snipping away the rest of the message.
>=20
> Comments inline.
>=20
>>>>=20
>>>> 2) for what concerns the video quality indices, it is well known
>>>> that PSNR is not well related to the QoE. (See for instance
>>>>=20
>>>> =
http://www.pscr.gov/about_pscr/press/video/video_quality_experts_interview=
_sigmm-122009.pdf)
>>>> Are we sure we want to cite PSNR?
>>>>=20
>>>=20
>>> I believe we need some objective metric for video either
>>> PSNR or something else. I am happy to include or
>>> update the document based on that discussion.
>>> (Note: I have not read the above document but
>>> will go through it.)
>>>=20
>>> However, when choosing these quality metrics we should be aware
>>> of their defined operating range (some models are limited to =
specific
>>> range of video resolutions or bit rates, etc.).
>>>=20
>>=20
>> I agree that PSNR isn't likely to be a good choice, I would assume =
that we
>> at least need a metric which takes both the video quality and the =
delay into
>> account? The experienced video quality depends on several aspects =
other than
>=20
> Apart from delay there is also balancing throughput and loss.
> In the scenario of a single RTP flow we could come up with a
> simple metric but with competing traffic it may be more difficult.
> By introducing a video quality metric in those scenarios,
> I was intending to quantify the balance of throughput, loss and delay.
>=20
>=20
>> the congestion control, such as encoder settings, transmitted spatial =
and
>> temporal resolution, video jitter buffer algorithm at the receiver, =
etc.
>=20
> Since we are talking about interactive real-time media, we could =
*agree* on
> running a baseline setup with preset parameters and reference video
> sequences. :)
>=20
>> Therefore I'm not sure it's a good idea to evaluate our congestion =
control
>> algorithms based on video quality. If possible I would rather see us
>> concentrate on simpler metrics as those you described earlier in your =
draft,
>> which are more related to how the algorithm actually adapts to =
changes in
>> bandwidth.
>=20
> In my opinion, if we can come up with a good QoE metric then it
> would be nice to have in addition to the throughput, loss and delay =
metrics.
> OTOH if we come up with a quality metric that is based on the =
relationship
> of the three then we may have a single metric to compare each =
algorithm.
> I am not sure which of these two methods would be better, but I am =
happy
> to hear more arguments for and against QoE, also other QoE methods
> before we decide on a quality metric.
>=20
>=20
> Regards,
> Varun


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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: Michael Welzl <michawe@ifi.uio.no>
Thread-Topic: [rmcat] send agenda requests (was Re: First meeting)
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Hi Michael,

Thanks for your comments.  Please find my replies inline below.

Best,
Xiaoqing


On Oct 16, 2012, at 1:11 AM, Michael Welzl wrote:

Hi,

I took a quick look at this:

I like the general idea of using as much as possible of existing network su=
pport - but I'm critical about the actual control you propose.

In section 5.3.2, you suggest to replace eqn. 1 from section 5.3 with eqn. =
2 when ECN feedback is available. This means to ignore a delay signal from =
now on, and interpret ECN marks in the exact same way as you interpreted de=
lay before - this is too simplistic, I think. After all, ECN marks require =
a queue to grow, and this group intends to minimize delay... why would you =
stop using delay once you have ECN in addition?


In our design, we considered the two cases, with or without ECN markings, s=
eparately.  Therefore, the dynamic scenario when ECN becomes available in t=
he middle of a video conferencing session has not yet been fully studied.  =
Just wondering, are cases where ECN marking are turned on and off dynamical=
ly common in reality?



In practice, using your scheme would probably mean that, as soon as I enabl=
e RED with ECN on the bottleneck between the two end systems, the users' ob=
served delay grows. Not exactly an incentive?

What you mentioned (uers observing higher delay in ECN-enabled systems) wil=
l not happen.

But rather, our simulation results have indicated that ECN with RED-based m=
arking yields fairly similar results as the default delay-based scheme. The=
y certainly do not incur higher delay, since the endpoint will automaticall=
y back off in face of higher ECN markings, which are indications of higher =
delay. The two systems (delay-based and ECN-based) can be tuned to stabiliz=
e that the same queuing delay at the bottleneck.

Cheers,
Michael


On 15. okt. 2012, at 23:43, Xiaoqing Zhu (xiaoqzhu) wrote:

And below is the notification of our submission:

Thanks,
Xiaoqing

--------------------------------------
A new version of I-D, draft-zhu-rmcat-nada-00.txt
has been successfully submitted by Xiaoqing Zhu and posted to the
IETF repository.

Filename:  draft-zhu-rmcat-nada
Revision:  00
Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
Creation date:  2012-10-14
WG ID:  Individual Submission
Number of pages: 10
URL:             http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-0=
0.txt
Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00


Abstract:
  This document describes a scheme named network-assisted dynamic
  adaptation (NADA), a novel congestion control approach for
  interactive real-time media applications, such as video conferencing.
  In the proposed scheme, the sender regulates its sending rate based
  on either implicit or explicit congestion signaling, in a unified
  approach. The scheme can reap the benefits of explicit congestion
  notification markings from network nodes. It also maintains
  consistent sender behavior in the absence of such markings, by
  reacting to queuing delays instead.

  We present here the overall system architecture, recommended
  behaviors at the sender and the receiver, as well as expected network
  nodes operations. Results from extensive simulation studies of the
  proposed scheme are available upon request.




The IETF Secretariat
--------------------------------------
On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:

Hi,

On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu<mailto:z=
huxq@alumni.stanford.edu>> wrote:
My colleague and I are working on a congestion control scheme for
real-time conferencing applications at Cisco. A brief version of that
algorithm was presented at the IAB Workshop in Vancouver (paper #12,
"Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
to time constraints, however, we did not get a chance to mention the
delay-based variant of that scheme, which may be a better fit for
rmcat.

We would like to request for a slot to present in Atlanta. We will
also submit the ID draft before Oct 15, for further discussions on
this mailing list.

I look forward to seeing the draft being discussed on the list!

Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is high.

Lars




--_000_E7175A8E3DC14048A7D3020E0338C1FA0B4C5Exmbalnx13ciscocom_
Content-Type: text/html; charset="us-ascii"
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
Hi Michael,&nbsp;
<div><br>
</div>
<div>Thanks for your comments. &nbsp;Please find my replies inline below.&n=
bsp;</div>
<div><br>
</div>
<div>Best,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>&nbsp;<br>
<div>
<div>On Oct 16, 2012, at 1:11 AM, Michael Welzl wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
Hi,
<div><br>
</div>
<div>I took a quick look at this:</div>
<div><br>
</div>
<div>I like the general idea of using as much as possible of existing netwo=
rk support - but I'm critical about the actual control you propose.</div>
<div><br>
</div>
<div>In section 5.3.2, you suggest to replace eqn. 1 from section 5.3 with =
eqn. 2 when ECN feedback is available. This means to ignore a delay signal =
from now on, and interpret ECN marks in the exact same way as you interpret=
ed delay before - this is too simplistic,
 I think. After all, ECN marks require a queue to grow, and this group inte=
nds to minimize delay... why would you stop using delay once you have ECN i=
n addition?</div>
</div>
</blockquote>
<div><br>
</div>
<div><br>
</div>
<div>In our design, we considered the two cases, with or without ECN markin=
gs, separately. &nbsp;Therefore, the dynamic scenario when ECN becomes avai=
lable in the middle of a video conferencing session has not yet been fully =
studied.&nbsp;&nbsp;Just wondering, are cases where
 ECN marking are turned on and off dynamically common in reality? &nbsp;</d=
iv>
<div><br>
</div>
<br>
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
<div><br>
</div>
<div>In practice, using your scheme would probably mean that, as soon as I =
enable RED with ECN on the bottleneck between the two end systems, the user=
s' observed delay grows. Not exactly an incentive?</div>
<div><br>
</div>
</div>
</blockquote>
<div>What you mentioned (uers observing higher delay in ECN-enabled systems=
) will not happen.&nbsp;</div>
<div>
<div><br>
</div>
<div>But rather, our simulation results have indicated that ECN with RED-ba=
sed marking yields fairly similar results as the default delay-based scheme=
. They certainly do not incur higher delay, since the endpoint will automat=
ically back off in face of higher
 ECN markings, which are indications of higher delay. The two systems (dela=
y-based and ECN-based) can be tuned to stabilize that the same queuing dela=
y at the bottleneck.&nbsp;</div>
</div>
<br>
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
<div>Cheers,</div>
<div>Michael</div>
<div><br>
</div>
<div><br>
<div>
<div>On 15. okt. 2012, at 23:43, Xiaoqing Zhu (xiaoqzhu) wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
And below is the notification of our submission:&nbsp;
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </spa=
n>&nbsp;draft-zhu-rmcat-nada<br>
Revision:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </spa=
n>&nbsp;00<br>
Title:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </span><=
span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp;NAD=
A: A Unified Congestion Control Scheme for Real-Time Media<br>
Creation date:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span>&nbsp;2012-10-14<br>
WG ID:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </span><=
span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp;Ind=
ividual Submission<br>
Number of pages: 10<br>
URL: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<a href=3D"http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.tx=
t">http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt</a><br>
Status: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"ht=
tp://datatracker.ietf.org/doc/draft-zhu-rmcat-nada">http://datatracker.ietf=
.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools=
.ietf.org/html/draft-zhu-rmcat-nada-00">http://tools.ietf.org/html/draft-zh=
u-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp;&nbsp;This document describes a scheme named network-assisted dynamic=
<br>
&nbsp;&nbsp;adaptation (NADA), a novel congestion control approach for<br>
&nbsp;&nbsp;interactive real-time media applications, such as video confere=
ncing.<br>
&nbsp;&nbsp;In the proposed scheme, the sender regulates its sending rate b=
ased<br>
&nbsp;&nbsp;on either implicit or explicit congestion signaling, in a unifi=
ed<br>
&nbsp;&nbsp;approach. The scheme can reap the benefits of explicit congesti=
on<br>
&nbsp;&nbsp;notification markings from network nodes. It also maintains<br>
&nbsp;&nbsp;consistent sender behavior in the absence of such markings, by<=
br>
&nbsp;&nbsp;reacting to queuing delays instead.<br>
<br>
&nbsp;&nbsp;We present here the overall system architecture, recommended<br=
>
&nbsp;&nbsp;behaviors at the sender and the receiver, as well as expected n=
etwork<br>
&nbsp;&nbsp;nodes operations. Results from extensive simulation studies of =
the<br>
&nbsp;&nbsp;proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a href=3D"mailto:zhuxq@alumni.=
stanford.edu">zhuxq@alumni.stanford.edu</a>&gt; wrote:<br>
<blockquote type=3D"cite">My colleague and I are working on a congestion co=
ntrol scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. A b=
rief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in Va=
ncouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">&quot;Network-Assisted Dynamic Adaptation (NADA):=
 A Design Summary&quot;). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a ch=
ance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may be =
a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present in=
 Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for furth=
er discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</body>
</html>

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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: "Piers O'Hanlon" <p.ohanlon@gmail.com>
Thread-Topic: [rmcat] send agenda requests (was Re: First meeting)
Thread-Index: AQHNp9grf2quh6JMt0CYpuBNP/SHCpe1q3yAgAWVAgCAALtjgIABUiQA
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References: <201210021809.19685.mkuehle@ikr.uni-stuttgart.de> <D4D47BCFFE5A004F95D707546AC0D7E9068D4895@SACEXCMBX01-PRD.hq.netapp.com> <D4D47BCFFE5A004F95D707546AC0D7E9068DEB13@SACEXCMBX01-PRD.hq.netapp.com> <CAFT3WbWX3PPXWsry5LTnDT7uWOjEhUoTDV2q0+NH+oKgUzSdKg@mail.gmail.com> <D4D47BCFFE5A004F95D707546AC0D7E918563B83@SACEXCMBX01-PRD.hq.netapp.com> <E7175A8E3DC14048A7D3020E0338C1FA0B3EE9@xmb-aln-x13.cisco.com> <DE34BE14-280A-4A63-848E-DD45EA383CEC@gmail.com>
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Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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Hi,

On Oct 16, 2012, at 1:54 AM, Piers O'Hanlon wrote:

Hi,

I too had a quick look at the proposal.

I was interested as to how kappa is calculated in equation (1) or whether i=
t is a fixed value?

The scaling values, kappa for (1) and eta for (2) are fixed values at the v=
ideo conferencing endpoint during any session.  The system typically works =
with a range of values for kappa and eta, leading to the same steady-state =
behavior, but with different dynamics during transients.


Also your slow start algorithm only seems to be time dependent - literally =
it appears to start slowly from t_0 to some preset time (T) - is there any =
mechanism to terminate slow start on loss/marked packets? Does slow start e=
ver occur mid stream? (e.g. after a silence period?)

Good point.  Right now slow start only works at the beginning of the stream=
. The rationale being that when a stream has not yet gathered enough conges=
tion information (implicit or explicit) from the network, it rather stay on=
 the cautious side.  Note that the sending rate is chosen to be the minimum=
 of slow-start rate R_ss and congestion-based rate R_o, so if excessive con=
gestion has been observed initially, the sender will still back off.

We haven't yet looked into the case whether slow start is needed mid-stream=
.  Open for comments/suggestions in this aspect.


Out of interest it was mentioned that there were extensive simulation studi=
es - would it be possible to post a simulation of multiple NADA flows compe=
ting. And also a competition of NADA with a TCP flow?

Thanks for the suggestions.  I'll try to compile a set of representative re=
sults on multiple self-competing NADA streams to share over this mailing li=
st.  Some modifications are need to make NADA compete fairly with loss-base=
d TCP, we are still working on that.

Thanks,

Piers.

On 15 Oct 2012, at 22:43, Xiaoqing Zhu (xiaoqzhu) wrote:

And below is the notification of our submission:

Thanks,
Xiaoqing

--------------------------------------
A new version of I-D, draft-zhu-rmcat-nada-00.txt
has been successfully submitted by Xiaoqing Zhu and posted to the
IETF repository.

Filename:  draft-zhu-rmcat-nada
Revision:  00
Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
Creation date:  2012-10-14
WG ID:  Individual Submission
Number of pages: 10
URL:             http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-0=
0.txt
Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00


Abstract:
  This document describes a scheme named network-assisted dynamic
  adaptation (NADA), a novel congestion control approach for
  interactive real-time media applications, such as video conferencing.
  In the proposed scheme, the sender regulates its sending rate based
  on either implicit or explicit congestion signaling, in a unified
  approach. The scheme can reap the benefits of explicit congestion
  notification markings from network nodes. It also maintains
  consistent sender behavior in the absence of such markings, by
  reacting to queuing delays instead.

  We present here the overall system architecture, recommended
  behaviors at the sender and the receiver, as well as expected network
  nodes operations. Results from extensive simulation studies of the
  proposed scheme are available upon request.




The IETF Secretariat
--------------------------------------
On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:

Hi,

On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu<mailto:z=
huxq@alumni.stanford.edu>> wrote:
My colleague and I are working on a congestion control scheme for
real-time conferencing applications at Cisco. A brief version of that
algorithm was presented at the IAB Workshop in Vancouver (paper #12,
"Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
to time constraints, however, we did not get a chance to mention the
delay-based variant of that scheme, which may be a better fit for
rmcat.

We would like to request for a slot to present in Atlanta. We will
also submit the ID draft before Oct 15, for further discussions on
this mailing list.

I look forward to seeing the draft being discussed on the list!

Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is high.

Lars




--_000_E7175A8E3DC14048A7D3020E0338C1FA0B4CC9xmbalnx13ciscocom_
Content-Type: text/html; charset="us-ascii"
Content-ID: <A03F4F3063D5CE4DB5FBE1AA4073C6BF@cisco.com>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
Hi,&nbsp;
<div><br>
<div>
<div>On Oct 16, 2012, at 1:54 AM, Piers O'Hanlon wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
Hi,
<div><br>
</div>
<div>I too had a quick look at the proposal.</div>
<div><br>
</div>
<div>I was interested as to how kappa is calculated in equation (1) or whet=
her it is a fixed value?</div>
</div>
</blockquote>
<div><br>
</div>
The scaling values, kappa for (1) and eta for (2) are fixed values at the v=
ideo conferencing endpoint during any session. &nbsp;The system typically w=
orks with a range of values for kappa and eta, leading to the same steady-s=
tate behavior, but with different dynamics
 during transients.&nbsp;</div>
<div><br>
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
<div><br>
</div>
<div>Also your slow start algorithm only seems to be time dependent - liter=
ally it appears to start slowly from t_0 to some preset time (T) - is there=
 any mechanism to terminate slow start on loss/marked packets? Does slow st=
art ever occur mid stream? (e.g.
 after a silence period?)</div>
</div>
</blockquote>
<div><br>
</div>
<div>Good point. &nbsp;Right now slow start only works at the beginning of =
the stream. The rationale being that when a stream has not yet gathered eno=
ugh congestion information (implicit or explicit) from the network, it rath=
er stay on the cautious side. &nbsp;Note that
 the sending rate is chosen to be the minimum of slow-start rate R_ss and c=
ongestion-based rate R_o, so if excessive congestion has been observed init=
ially, the sender will still back off.&nbsp;</div>
<div><br>
</div>
<div>We haven't yet looked into the case whether slow start is needed mid-s=
tream. &nbsp;Open for comments/suggestions in this aspect. &nbsp;</div>
<br>
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
<div><br>
</div>
<div>Out of interest it was mentioned that there were extensive simulation =
studies - would it be possible to post a simulation of multiple NADA flows =
competing. And also a competition of NADA with a TCP flow?</div>
</div>
</blockquote>
<div><br>
</div>
Thanks for the suggestions. &nbsp;I'll try to compile a set of representati=
ve results on multiple self-competing NADA streams to share over this maili=
ng list. &nbsp;Some modifications are need to make NADA compete fairly with=
 loss-based TCP, we are still working on that.&nbsp;<br>
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
<div><br>
</div>
<div>Thanks,</div>
<div><br>
</div>
<div>Piers.</div>
<div><br>
<div>
<div>On 15 Oct 2012, at 22:43, Xiaoqing Zhu (xiaoqzhu) wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
And below is the notification of our submission:&nbsp;
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </spa=
n>&nbsp;draft-zhu-rmcat-nada<br>
Revision:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </spa=
n>&nbsp;00<br>
Title:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </span><=
span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp;NAD=
A: A Unified Congestion Control Scheme for Real-Time Media<br>
Creation date:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> =
</span>&nbsp;2012-10-14<br>
WG ID:<span class=3D"Apple-tab-span" style=3D"white-space: pre; "> </span><=
span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp;Ind=
ividual Submission<br>
Number of pages: 10<br>
URL: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<a href=3D"http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.tx=
t">http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt</a><br>
Status: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"ht=
tp://datatracker.ietf.org/doc/draft-zhu-rmcat-nada">http://datatracker.ietf=
.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools=
.ietf.org/html/draft-zhu-rmcat-nada-00">http://tools.ietf.org/html/draft-zh=
u-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp;&nbsp;This document describes a scheme named network-assisted dynamic=
<br>
&nbsp;&nbsp;adaptation (NADA), a novel congestion control approach for<br>
&nbsp;&nbsp;interactive real-time media applications, such as video confere=
ncing.<br>
&nbsp;&nbsp;In the proposed scheme, the sender regulates its sending rate b=
ased<br>
&nbsp;&nbsp;on either implicit or explicit congestion signaling, in a unifi=
ed<br>
&nbsp;&nbsp;approach. The scheme can reap the benefits of explicit congesti=
on<br>
&nbsp;&nbsp;notification markings from network nodes. It also maintains<br>
&nbsp;&nbsp;consistent sender behavior in the absence of such markings, by<=
br>
&nbsp;&nbsp;reacting to queuing delays instead.<br>
<br>
&nbsp;&nbsp;We present here the overall system architecture, recommended<br=
>
&nbsp;&nbsp;behaviors at the sender and the receiver, as well as expected n=
etwork<br>
&nbsp;&nbsp;nodes operations. Results from extensive simulation studies of =
the<br>
&nbsp;&nbsp;proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a href=3D"mailto:zhuxq@alumni.=
stanford.edu">zhuxq@alumni.stanford.edu</a>&gt; wrote:<br>
<blockquote type=3D"cite">My colleague and I are working on a congestion co=
ntrol scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. A b=
rief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in Va=
ncouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">&quot;Network-Assisted Dynamic Adaptation (NADA):=
 A Design Summary&quot;). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a ch=
ance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may be =
a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present in=
 Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for furth=
er discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</body>
</html>

--_000_E7175A8E3DC14048A7D3020E0338C1FA0B4CC9xmbalnx13ciscocom_--

From michawe@ifi.uio.no  Wed Oct 17 00:32:10 2012
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References: <201210021809.19685.mkuehle@ikr.uni-stuttgart.de> <D4D47BCFFE5A004F95D707546AC0D7E9068D4895@SACEXCMBX01-PRD.hq.netapp.com> <D4D47BCFFE5A004F95D707546AC0D7E9068DEB13@SACEXCMBX01-PRD.hq.netapp.com> <CAFT3WbWX3PPXWsry5LTnDT7uWOjEhUoTDV2q0+NH+oKgUzSdKg@mail.gmail.com> <D4D47BCFFE5A004F95D707546AC0D7E918563B83@SACEXCMBX01-PRD.hq.netapp.com> <E7175A8E3DC14048A7D3020E0338C1FA0B3EE9@xmb-aln-x13.cisco.com> <C12C3E50-3435-4E49-A6DA-E5CD4A401A3F@ifi.uio.no> <E7175A8E3DC14048A7D3020E0338C1FA0B4C5E@xmb-aln-x13.cisco.com>
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On 17. okt. 2012, at 06:58, Xiaoqing Zhu (xiaoqzhu) wrote:

> Hi Michael,=20
>=20
> Thanks for your comments.  Please find my replies inline below.=20
>=20
> Best,
> Xiaoqing
>=20
> =20
> On Oct 16, 2012, at 1:11 AM, Michael Welzl wrote:
>=20
>> Hi,
>>=20
>> I took a quick look at this:
>>=20
>> I like the general idea of using as much as possible of existing =
network support - but I'm critical about the actual control you propose.
>>=20
>> In section 5.3.2, you suggest to replace eqn. 1 from section 5.3 with =
eqn. 2 when ECN feedback is available. This means to ignore a delay =
signal from now on, and interpret ECN marks in the exact same way as you =
interpreted delay before - this is too simplistic, I think. After all, =
ECN marks require a queue to grow, and this group intends to minimize =
delay... why would you stop using delay once you have ECN in addition?
>=20
>=20
> In our design, we considered the two cases, with or without ECN =
markings, separately.  Therefore, the dynamic scenario when ECN becomes =
available in the middle of a video conferencing session has not yet been =
fully studied.  Just wondering, are cases where ECN marking are turned =
on and off dynamically common in reality? =20

First, anything can happen when you have a routing change.
Second, I'm really sorry, but this wasn't what I meant. I shouldn't have =
written "when" but "if". I wasn't referring to a case where we go from =
no-ECN-support to ECN-support. So this is about the ECN-always-supported =
case.


>> In practice, using your scheme would probably mean that, as soon as I =
enable RED with ECN on the bottleneck between the two end systems, the =
users' observed delay grows. Not exactly an incentive?
>>=20
> What you mentioned (uers observing higher delay in ECN-enabled =
systems) will not happen.=20
>=20
> But rather, our simulation results have indicated that ECN with =
RED-based marking yields fairly similar results as the default =
delay-based scheme. They certainly do not incur higher delay, since the =
endpoint will automatically back off in face of higher ECN markings, =
which are indications of higher delay. The two systems (delay-based and =
ECN-based) can be tuned to stabilize that the same queuing delay at the =
bottleneck.=20

Whether you see significant delay before you see a large enough amount =
of ECN markings depends on the queue length and RED parameters, none of =
which are under the end system's control. ECN is also a probabilistic =
signal, so sometimes you may get delay growth but just be lucky (or =
unlucky) enough to not get an ECN mark.

So, I'd have to see results that involve various RED settings and queue =
lengths before I believe that.

Cheers,
Michael


From p.ohanlon@gmail.com  Wed Oct 17 04:25:36 2012
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Hi Varun,

The draft looks like a good start.

I think it would good to add evaluation of idle and data-limited =
behaviours - as these are important as regards media transport - they =
were the reasons the original TFRC (RFC3448) was updated to RFC5348. =
See: http://dx.doi.org/10.1016%2fj.comcom.2011.05.003
"TCP-Friendly Rate Control (TFRC) for bursty media flows", Arjuna =
Sathiaseelan, Gorry Fairhurst, Comcom 2011

Furthermore it would also be useful to evaluate the algorithms with =
respect to their operation in terms of competing against a small number =
of flows vs a large number - see:=20
On the Effectiveness of Delay-Based Congestion Avoidance, Ravi S. =
Prasad, Manish Jain Georgia, Constantinos Dovrolis, PFLDnet 2004

Cheers,

Piers




On 15 Oct 2012, at 21:33, Varun Singh wrote:

> Greetings,
>=20
> We have submitted an initial draft outlining the criteria for =
evaluating
> congestion control algorithms. The metrics section is a bit light
> on details and will to expand on that in the next revision.
>=20
> We hope that the draft is a good starting point to get the discussion
> started about metrics and scenarios for simulation and testbed =
experiments.
>=20
> Cheers,
> Varun
>=20
> Begin forwarded message:
>=20
>=20
> A new version of I-D, draft-singh-rmcat-cc-eval-00.txt
> has been successfully submitted by Varun Singh and posted to the
> IETF repository.
>=20
> Filename:	 draft-singh-rmcat-cc-eval
> Revision:	 00
> Title:		 Evaluating Congestion Control for Interactive =
Real-time Media.
> Creation date:	 2012-10-15
> WG ID:		 Individual Submission
> Number of pages: 9
> URL:
> http://www.ietf.org/internet-drafts/draft-singh-rmcat-cc-eval-00.txt
> Status:          =
http://datatracker.ietf.org/doc/draft-singh-rmcat-cc-eval
> Htmlized:        =
http://tools.ietf.org/html/draft-singh-rmcat-cc-eval-00
>=20
>=20
> Abstract:
>  The Real-time Transport Protocol (RTP) is used to transmit media in
>  telephony and video conferencing applications.  This document
>  describes the guidelines to evaluate new congestion control
>  algorithms for interactive point-to-point real-time media.
>=20
>=20
>=20
>=20
> The IETF Secretariat


From tterriberry@mozilla.com  Wed Oct 17 17:31:29 2012
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Subject: Re: [rmcat] Video Quality for RMCAT
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Varun Singh wrote:
> Apart from delay there is also balancing throughput and loss.
> In the scenario of a single RTP flow we could come up with a
> simple metric but with competing traffic it may be more difficult.
> By introducing a video quality metric in those scenarios,
> I was intending to quantify the balance of throughput, loss and delay.

If you want to measure video quality in the presence of changing 
bitrates, loss, and delay, you have a difficult problem on your hands. 
Not the least of which is that PSNR (or similar metrics) require a 
reference to compare against, which starts to get meaningless once you 
change resolution or framerate in the encoder (with dropping frames 
being a very crude form of changing framerate), all of which are things 
the encoder can do to adapt to changes in available bandwidth. I 
discussed some of this in more detail in my submission to the workshop: 
<http://www.tschofenig.priv.at/cc-workshop/irtf_iab-ccirtcpaper26.pdf>

From tterriberry@mozilla.com  Wed Oct 17 17:38:43 2012
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Subject: Re: [rmcat] Video Quality for RMCAT
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Michael Welzl wrote:
> Sorry if this is stupid, perhaps I'm missing something - but:
>
> even if we could agree on a baseline test setup, standard movies etc.: isn't the very idea of a QoE metric inevitably bound to the codec, which then binds evaluating congestion control mechanisms to a debate that this really shouldn't be a part of?

Yes. I think there are two separable problems here:

1) Estimating the channel capacity
2) Allocating that capacity among multiple flows constrained by the same 
bottleneck (or multiple, multiplexed streams on the same flow, if you like).

The only interaction between problem 1 and the codec(s) is how quickly 
they can adapt to the changing estimates that are produced. I'm not sure 
that requires looking at the quality of the resulting content at all, 
however.

For problem 2, actual video/audio quality becomes much more interesting, 
but I think is something this is a problem that has to be solved by the 
application itself, not the congestion control layer. Only the 
application knows things like "I'm using G.711: the only way I can adapt 
its rate is to drop the packets, so all reductions must come from my 
video stream," or "I'm transmitting two video streams, but one of them 
is much easier to encode than the other, so when the rate drops, I 
should steal more bits from that one."

From wes@mti-systems.com  Wed Oct 17 20:14:35 2012
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Subject: Re: [rmcat] Video Quality for RMCAT
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On 10/17/2012 8:38 PM, Timothy B. Terriberry wrote:
> Michael Welzl wrote:
>> Sorry if this is stupid, perhaps I'm missing something - but:
>>
>> even if we could agree on a baseline test setup, standard movies etc.:
>> isn't the very idea of a QoE metric inevitably bound to the codec,
>> which then binds evaluating congestion control mechanisms to a debate
>> that this really shouldn't be a part of?
> 
> Yes. I think there are two separable problems here:
> 
> 1) Estimating the channel capacity
> 2) Allocating that capacity among multiple flows constrained by the same
> bottleneck (or multiple, multiplexed streams on the same flow, if you
> like).
> 
> The only interaction between problem 1 and the codec(s) is how quickly
> they can adapt to the changing estimates that are produced. I'm not sure
> that requires looking at the quality of the resulting content at all,
> however.
> 
> For problem 2, actual video/audio quality becomes much more interesting,
> but I think is something this is a problem that has to be solved by the
> application itself, not the congestion control layer. Only the
> application knows things like "I'm using G.711: the only way I can adapt
> its rate is to drop the packets, so all reductions must come from my
> video stream," or "I'm transmitting two video streams, but one of them
> is much easier to encode than the other, so when the rate drops, I
> should steal more bits from that one."
> 
> 


I totally agree with this.  The elasticity of the codecs matters, and
the congestion controller will benefit from an understanding of the
envelopes they can operate within, but it's the job of the codecs to
ensure quality within their envelopes of elasticity, and not at all the
congestion controller's job.

The CC may say "you now have N kbps to work within" to each codec, and
this will be based on what it estimates is safe for the network given
the current ensemble of codecs and their stated operating envelopes,
*not* on any explicit attempt by the CC to maximize user experience.

-- 
Wes Eddy
MTI Systems

From xiaoqzhu@cisco.com  Wed Oct 17 20:49:47 2012
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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: Michael Welzl <michawe@ifi.uio.no>
Thread-Topic: [rmcat] send agenda requests (was Re: First meeting)
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Your points are well taken, Michael. =20


Agree that ultimately, it would be nice to have the scheme consider a merge=
 of delay and ECN signals, instead of switching naively between the two. We=
 considered that as a stretch goal though, according to the priorities disc=
ussed in the last BOF. We are still in the process of improving the algorit=
hm in that aspect.=20

I will put in some results from ECN-based marking in the compiled file to s=
hare. But admitted, we have not yet explored the impact of all variations o=
f ECN configurations.  I will need to run more tests for that.=20

-Xiaoqing

On Oct 17, 2012, at 12:31 AM, Michael Welzl wrote:

>=20
> On 17. okt. 2012, at 06:58, Xiaoqing Zhu (xiaoqzhu) wrote:
>=20
>> Hi Michael,=20
>>=20
>> Thanks for your comments.  Please find my replies inline below.=20
>>=20
>> Best,
>> Xiaoqing
>>=20
>>=20
>> On Oct 16, 2012, at 1:11 AM, Michael Welzl wrote:
>>=20
>>> Hi,
>>>=20
>>> I took a quick look at this:
>>>=20
>>> I like the general idea of using as much as possible of existing networ=
k support - but I'm critical about the actual control you propose.
>>>=20
>>> In section 5.3.2, you suggest to replace eqn. 1 from section 5.3 with e=
qn. 2 when ECN feedback is available. This means to ignore a delay signal f=
rom now on, and interpret ECN marks in the exact same way as you interprete=
d delay before - this is too simplistic, I think. After all, ECN marks requ=
ire a queue to grow, and this group intends to minimize delay... why would =
you stop using delay once you have ECN in addition?
>>=20
>>=20
>> In our design, we considered the two cases, with or without ECN markings=
, separately.  Therefore, the dynamic scenario when ECN becomes available i=
n the middle of a video conferencing session has not yet been fully studied=
.  Just wondering, are cases where ECN marking are turned on and off dynami=
cally common in reality? =20
>=20
> First, anything can happen when you have a routing change.
> Second, I'm really sorry, but this wasn't what I meant. I shouldn't have =
written "when" but "if". I wasn't referring to a case where we go from no-E=
CN-support to ECN-support. So this is about the ECN-always-supported case.
>=20
>=20
>>> In practice, using your scheme would probably mean that, as soon as I e=
nable RED with ECN on the bottleneck between the two end systems, the users=
' observed delay grows. Not exactly an incentive?
>>>=20
>> What you mentioned (uers observing higher delay in ECN-enabled systems) =
will not happen.=20
>>=20
>> But rather, our simulation results have indicated that ECN with RED-base=
d marking yields fairly similar results as the default delay-based scheme. =
They certainly do not incur higher delay, since the endpoint will automatic=
ally back off in face of higher ECN markings, which are indications of high=
er delay. The two systems (delay-based and ECN-based) can be tuned to stabi=
lize that the same queuing delay at the bottleneck.=20
>=20
> Whether you see significant delay before you see a large enough amount of=
 ECN markings depends on the queue length and RED parameters, none of which=
 are under the end system's control. ECN is also a probabilistic signal, so=
 sometimes you may get delay growth but just be lucky (or unlucky) enough t=
o not get an ECN mark.
>=20
> So, I'd have to see results that involve various RED settings and queue l=
engths before I believe that.
>=20
> Cheers,
> Michael
>=20


From michawe@ifi.uio.no  Thu Oct 18 00:45:33 2012
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Cc: "Timothy B. Terriberry" <tterriberry@mozilla.com>, rmcat WG <rmcat@ietf.org>
Subject: Re: [rmcat] Video Quality for RMCAT
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On 18. okt. 2012, at 05:14, Wesley Eddy wrote:

> On 10/17/2012 8:38 PM, Timothy B. Terriberry wrote:
>> Michael Welzl wrote:
>>> Sorry if this is stupid, perhaps I'm missing something - but:
>>>=20
>>> even if we could agree on a baseline test setup, standard movies =
etc.:
>>> isn't the very idea of a QoE metric inevitably bound to the codec,
>>> which then binds evaluating congestion control mechanisms to a =
debate
>>> that this really shouldn't be a part of?
>>=20
>> Yes. I think there are two separable problems here:
>>=20
>> 1) Estimating the channel capacity
>> 2) Allocating that capacity among multiple flows constrained by the =
same
>> bottleneck (or multiple, multiplexed streams on the same flow, if you
>> like).
>>=20
>> The only interaction between problem 1 and the codec(s) is how =
quickly
>> they can adapt to the changing estimates that are produced. I'm not =
sure
>> that requires looking at the quality of the resulting content at all,
>> however.
>>=20
>> For problem 2, actual video/audio quality becomes much more =
interesting,
>> but I think is something this is a problem that has to be solved by =
the
>> application itself, not the congestion control layer. Only the
>> application knows things like "I'm using G.711: the only way I can =
adapt
>> its rate is to drop the packets, so all reductions must come from my
>> video stream," or "I'm transmitting two video streams, but one of =
them
>> is much easier to encode than the other, so when the rate drops, I
>> should steal more bits from that one."
>>=20
>>=20
>=20
>=20
> I totally agree with this.  The elasticity of the codecs matters, and
> the congestion controller will benefit from an understanding of the
> envelopes they can operate within, but it's the job of the codecs to
> ensure quality within their envelopes of elasticity, and not at all =
the
> congestion controller's job.

Exactly, I also totally agree. Do we then have a consensus to kick out =
the QoE metric?   :-)
I think that would be a good thing.


> The CC may say "you now have N kbps to work within" to each codec, and
> this will be based on what it estimates is safe for the network given
> the current ensemble of codecs and their stated operating envelopes,
> *not* on any explicit attempt by the CC to maximize user experience.

Agreed. Note, what you describe here is an "interaction between =
applications and RTP flows", which is a quote from the charter, a part =
of a milestone due in December... is anyone working on a draft for this?

This is where "packet importance" to be handed over to the congestion =
control should also fit... (that was discussed as a part of the sender- =
vs. receiver-based thread) IF we decide to have that.

Cheers,
Michael


From harald@alvestrand.no  Thu Oct 18 06:26:00 2012
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Subject: [rmcat] Multihop congestion control - two words of caution
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I got into a discussion in Vancouver with someone about multihop 
congestion control, why it was hard, and why it should be ruled out of 
scope - and promised to say something about it later.
But at this point, I no longer remember who it was with, so I'm posting 
it here instead.

One interesting thing about congestion control is a tendency to 
oscillation. If there is a feedback signal, that's always based on past 
history; the decisions based on that signal will affect traffic in the 
future.

So there's always a delay between the events we take action on and the 
action we take.
If, for instance, the delay is 1 second (the argument's the same for 1 
millisecond, but a second is easier to think about), the feedback signal 
is "congestion", and the decision is "reduce traffic" .. the signal an 
instant after the decision is made is based on the *previous* second's 
worth of traffic, and will still say "congestion"; if we take action on 
that, we'll reduce traffic even more - perhaps too much.

Once we detect "no congestion", the same thing happens in reverse. If 
the delays and the size of action is just right, we can get into 
oscillation: jumping between inefficient use of bandwidth (sending too 
little) and serious congestion (sending too much).

The mitigation against this kind of behaviour is that one designs the 
strength of the action to be commensurate with the delay so as to avoid 
damping; in TCP, this is obtained by scaling reaction time by the RTT; 
we don't react faster than 1 unit of reaction per RTT.

Now consider what happens if we have two hops after each other, possibly 
with different delays, each running their own congestion control 
algorithm - and a linkage that tries to feed congestion control 
information from the second hop back to the congestion control algorithm 
running on the first hop.

The overall system will have a delay - but that delay won't be visible 
to the congestion control algorithm on the first hop, so the reaction 
time and size can't be scaled appropriately.

The result is that the overall system is very likely to exhibit 
oscillatory behaviour - simply because the time constants we need to 
observe for a proper design are invisible to the observer.

We can imagine ways of working around this limitation - but for the 
purposes of RMCAT, I think we should just stick with two principles:

- We don't design for the two-hop case.
- We advise designers of two-hop systems (like relay nodes) that if they 
want to feed capacity information from a second hop back to the first 
hop, they should reflect those changes SLOWLY.

Experts, did this sound reasonable?

                      Harald

From harald@alvestrand.no  Thu Oct 18 06:32:09 2012
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Subject: Re: [rmcat] Video Quality for RMCAT
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On 10/18/2012 02:31 AM, Timothy B. Terriberry wrote:
> Varun Singh wrote:
>> Apart from delay there is also balancing throughput and loss.
>> In the scenario of a single RTP flow we could come up with a
>> simple metric but with competing traffic it may be more difficult.
>> By introducing a video quality metric in those scenarios,
>> I was intending to quantify the balance of throughput, loss and delay.
>
> If you want to measure video quality in the presence of changing 
> bitrates, loss, and delay, you have a difficult problem on your hands. 
> Not the least of which is that PSNR (or similar metrics) require a 
> reference to compare against, which starts to get meaningless once you 
> change resolution or framerate in the encoder (with dropping frames 
> being a very crude form of changing framerate), all of which are 
> things the encoder can do to adapt to changes in available bandwidth. 
> I discussed some of this in more detail in my submission to the 
> workshop: 
> <http://www.tschofenig.priv.at/cc-workshop/irtf_iab-ccirtcpaper26.pdf>
Wouldn't a simple (simplistic?) way of measuring quality in the presence 
of packet drops be to always use a test rig where one end looped back 
the encoded video signal to the other end?

Thus, the one that injects the signal could compare the pre-encoding 
video source to the post-decoding video sink, which I suppose is what 
PSNR measurements need.

This is inappropriate for measurements on a live service, but might be 
appropriate for generating benchmarks.



From radhika.r.roy.civ@mail.mil  Thu Oct 18 07:36:48 2012
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From: "Roy, Radhika R CIV (US)" <radhika.r.roy.civ@mail.mil>
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Thread-Topic: [rmcat] Multihop congestion control - two words of caution
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Hi, Herald:

In general, your observation is OK. There are a couple of things are as
follows:

1. We have to look into aggregate statistical model of traffic congestions.
2. Contributions to congestions are different for different types of
traffic.
3. Behavior of each kind of traffic is also different. 
4. Performance requirements of each kind of traffic are also different.
4. High bandwidth-intensive video vs. low bandwidth audio vs. low-high
bandwidth with very busty data with different performance requirements over
the same IP network will need to have very different requirements for taking
actions when congestions are detected.

The point that I am making is this: IETF is being the technical organization
needs to try to solve, if not complete solutions for congestions, but at
least to look into in addressing the problems especially when video is
becoming (or has become) the most dominant traffic. As we gain more and more
insights of this complex problem, we will be developing the technical
standards accordingly as time goes by.

BR/Radhika

-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of
Harald Alvestrand
Sent: Thursday, October 18, 2012 9:26 AM
To: rmcat@ietf.org
Subject: [rmcat] Multihop congestion control - two words of caution

I got into a discussion in Vancouver with someone about multihop congestion
control, why it was hard, and why it should be ruled out of scope - and
promised to say something about it later.
But at this point, I no longer remember who it was with, so I'm posting it
here instead.

One interesting thing about congestion control is a tendency to oscillation.
If there is a feedback signal, that's always based on past history; the
decisions based on that signal will affect traffic in the future.

So there's always a delay between the events we take action on and the
action we take.
If, for instance, the delay is 1 second (the argument's the same for 1
millisecond, but a second is easier to think about), the feedback signal is
"congestion", and the decision is "reduce traffic" .. the signal an instant
after the decision is made is based on the *previous* second's worth of
traffic, and will still say "congestion"; if we take action on that, we'll
reduce traffic even more - perhaps too much.

Once we detect "no congestion", the same thing happens in reverse. If the
delays and the size of action is just right, we can get into
oscillation: jumping between inefficient use of bandwidth (sending too
little) and serious congestion (sending too much).

The mitigation against this kind of behaviour is that one designs the
strength of the action to be commensurate with the delay so as to avoid
damping; in TCP, this is obtained by scaling reaction time by the RTT; we
don't react faster than 1 unit of reaction per RTT.

Now consider what happens if we have two hops after each other, possibly
with different delays, each running their own congestion control algorithm -
and a linkage that tries to feed congestion control information from the
second hop back to the congestion control algorithm running on the first
hop.

The overall system will have a delay - but that delay won't be visible to
the congestion control algorithm on the first hop, so the reaction time and
size can't be scaled appropriately.

The result is that the overall system is very likely to exhibit oscillatory
behaviour - simply because the time constants we need to observe for a
proper design are invisible to the observer.

We can imagine ways of working around this limitation - but for the purposes
of RMCAT, I think we should just stick with two principles:

- We don't design for the two-hop case.
- We advise designers of two-hop systems (like relay nodes) that if they
want to feed capacity information from a second hop back to the first hop,
they should reflect those changes SLOWLY.

Experts, did this sound reasonable?

                      Harald

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From ford@isoc.org  Thu Oct 18 07:56:09 2012
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Subject: Re: [rmcat] Multihop congestion control - two words of caution
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On 18 Oct 2012, at 14:25, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> - We don't design for the two-hop case.

For me at least, visualisation helps: =
http://www.youtube.com/watch?v=3DQXf95_EKS6E


From mirja.kuehlewind@ikr.uni-stuttgart.de  Thu Oct 18 08:58:47 2012
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From: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>
Organization: University of Stuttgart (Germany), IKR
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Cc: Harald Alvestrand <harald@alvestrand.no>
Subject: Re: [rmcat] Multihop congestion control - two words of caution
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Hi Harald,

i believe do not understand your scenario:

> Now consider what happens if we have two hops after each other, possibly
> with different delays, each running their own congestion control
> algorithm - and a linkage that tries to feed congestion control
> information from the second hop back to the congestion control algorithm
> running on the first hop.

Are we talking of end-to-end congestion control here? Then if we have=20
congestion feedback from different nodes on one path, this feedback signal=
=20
will have the same delay. And usually there is only one bottleneck at one=20
point of time as there is always on link with the smallest available=20
bandwidth. Is this the scenario your are talking about or something else?

Mirja



>
> The overall system will have a delay - but that delay won't be visible
> to the congestion control algorithm on the first hop, so the reaction
> time and size can't be scaled appropriately.
>
> The result is that the overall system is very likely to exhibit
> oscillatory behaviour - simply because the time constants we need to
> observe for a proper design are invisible to the observer.
>
> We can imagine ways of working around this limitation - but for the
> purposes of RMCAT, I think we should just stick with two principles:
>
> - We don't design for the two-hop case.
> - We advise designers of two-hop systems (like relay nodes) that if they
> want to feed capacity information from a second hop back to the first
> hop, they should reflect those changes SLOWLY.
>
> Experts, did this sound reasonable?
>
>                       Harald



=2D-=20
=2D------------------------------------------------------------------
Dipl.-Ing. Mirja K=FChlewind
Institute of Communication Networks and Computer Engineering (IKR)
University of Stuttgart, Germany
Pfaffenwaldring 47, D-70569 Stuttgart

tel: +49(0)711/685-67973
email: mirja.kuehlewind@ikr.uni-stuttgart.de
web: www.ikr.uni-stuttgart.de
=2D------------------------------------------------------------------

From randell-ietf@jesup.org  Thu Oct 18 21:50:47 2012
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I won't bother quoting people, but I'll say that trying to evaluate a CC 
algorithm (even one meant for media) via a video QoE test is a really 
bad idea.  It's way, way too subjective - the problem is far *harder* 
than it is for pre-recorded sequences, and there's no great agreement 
there on how to weight all the factors. (From what I hear).

It may be *interesting* to look at, but I'd never try to use it as any 
sort of definitive test.

-- 
Randell Jesup
randell-ietf@jesup.org


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On 10/18/2012 3:45 AM, Michael Welzl wrote:
>
>> The CC may say "you now have N kbps to work within" to each codec, and
>> this will be based on what it estimates is safe for the network given
>> the current ensemble of codecs and their stated operating envelopes,
>> *not* on any explicit attempt by the CC to maximize user experience.
> Agreed. Note, what you describe here is an "interaction between applications and RTP flows", which is a quote from the charter, a part of a milestone due in December... is anyone working on a draft for this?

I have made a proposal in the W3 bugzilla and on the list.  There was a 
small amount of discussion; none since.  I'd suggest starting from 
there.  It may be worthwhile to re-examine the API in light of the stats 
proposals to avoid conflicting/overlapping data.

> This is where "packet importance" to be handed over to the congestion control should also fit... (that was discussed as a part of the sender- vs. receiver-based thread) IF we decide to have that.

Well, packet importance would (generally) be in the internals, not at 
the application/JS layer, except maybe to provide stream-level priorities.

-- 
Randell Jesup
randell-ietf@jesup.org


From michawe@ifi.uio.no  Fri Oct 19 00:52:18 2012
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Cc: rmcat@ietf.org, Harald Alvestrand <harald@alvestrand.no>
Subject: Re: [rmcat] Multihop congestion control - two words of caution
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Hi all,

I feel in a bit of a dilemma answering Harald's email, because this =
requires that I call myself an "expert"... yet I'm tempted to answer... =
tsk!   :-)     Anyway, I'll try to reply the emails from Harald, Radhika =
and Mirja in one go here, in the hope that I can clarify things a =
little:

Harald: all of what you have written sounds 100% reasonable to me.

Radhika: your answer appears (to me) to miss the point, and maybe (sorry =
if I'm wrong!) it's because you misunderstood the scenario? So maybe my =
answer to Mirja below helps.

Mirja: RTP/RTCP can let end systems take the role of an intermediate =
node, to take care of data stream specific things before traffic reaches =
the receiver (synchronizing, mixing, ...). Essentially, this is an =
overlay network. I think Harald refers to such a case with multiple RTP =
end systems, i.e. a path like this:

Node1 [ RTP1 sender ] =3D> Node 2 [RTP1 receiver + RTP2 sender] =3D> =
Node 3 [RTP2 receiver]

...and his description of problems goes into the behavior of node 2 =
(which is, in principle, quite similar to the behavior of some PEPs =
(RFC3135), I guess, e.g. TCP splitters. The decision at the BOF was not =
*yet* to define what is needed to make Node 2 work correctly, which is =
what we referred to as multi-hop congestion control. The exact phrasing =
in the charter is: "implications on multi-hop connections will be =
considered at a later stage". The reason for writing this is that Magnus =
Westerlund (quite correctly, I think) stated that defining this will at =
some point be needed.

I hope my interpretation of Harald's scenario is correct, and I hope =
that this was helpful!

Cheers,
Michael


On 18. okt. 2012, at 17:58, Mirja Kuehlewind wrote:

> Hi Harald,
>=20
> i believe do not understand your scenario:
>=20
>> Now consider what happens if we have two hops after each other, =
possibly
>> with different delays, each running their own congestion control
>> algorithm - and a linkage that tries to feed congestion control
>> information from the second hop back to the congestion control =
algorithm
>> running on the first hop.
>=20
> Are we talking of end-to-end congestion control here? Then if we have=20=

> congestion feedback from different nodes on one path, this feedback =
signal=20
> will have the same delay. And usually there is only one bottleneck at =
one=20
> point of time as there is always on link with the smallest available=20=

> bandwidth. Is this the scenario your are talking about or something =
else?
>=20
> Mirja
>=20
>=20
>=20
>>=20
>> The overall system will have a delay - but that delay won't be =
visible
>> to the congestion control algorithm on the first hop, so the reaction
>> time and size can't be scaled appropriately.
>>=20
>> The result is that the overall system is very likely to exhibit
>> oscillatory behaviour - simply because the time constants we need to
>> observe for a proper design are invisible to the observer.
>>=20
>> We can imagine ways of working around this limitation - but for the
>> purposes of RMCAT, I think we should just stick with two principles:
>>=20
>> - We don't design for the two-hop case.
>> - We advise designers of two-hop systems (like relay nodes) that if =
they
>> want to feed capacity information from a second hop back to the first
>> hop, they should reflect those changes SLOWLY.
>>=20
>> Experts, did this sound reasonable?
>>=20
>>                      Harald
>=20
>=20
>=20
> --=20
> -------------------------------------------------------------------
> Dipl.-Ing. Mirja K=FChlewind
> Institute of Communication Networks and Computer Engineering (IKR)
> University of Stuttgart, Germany
> Pfaffenwaldring 47, D-70569 Stuttgart
>=20
> tel: +49(0)711/685-67973
> email: mirja.kuehlewind@ikr.uni-stuttgart.de
> web: www.ikr.uni-stuttgart.de
> -------------------------------------------------------------------


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On 19. okt. 2012, at 06:52, Randell Jesup wrote:

> On 10/18/2012 3:45 AM, Michael Welzl wrote:
>>=20
>>> The CC may say "you now have N kbps to work within" to each codec, =
and
>>> this will be based on what it estimates is safe for the network =
given
>>> the current ensemble of codecs and their stated operating envelopes,
>>> *not* on any explicit attempt by the CC to maximize user experience.
>> Agreed. Note, what you describe here is an "interaction between =
applications and RTP flows", which is a quote from the charter, a part =
of a milestone due in December... is anyone working on a draft for this?
>=20
> I have made a proposal in the W3 bugzilla and on the list.  There was =
a small amount of discussion; none since.  I'd suggest starting from =
there.  It may be worthwhile to re-examine the API in light of the stats =
proposals to avoid conflicting/overlapping data.

I think I've asked you before, and I think you probably answered =3D> so =
I'm sorry for being too chaotic, but could you maybe give us that link =
again?


>> This is where "packet importance" to be handed over to the congestion =
control should also fit... (that was discussed as a part of the sender- =
vs. receiver-based thread) IF we decide to have that.
>=20
> Well, packet importance would (generally) be in the internals, not at =
the application/JS layer, except maybe to provide stream-level =
priorities.

Yes, but to me, "application" in the rmcat context is not the JS =
application but the codec. This is at least where that "interactions" =
deliverable originated - to address comments like this one:

from: http://www.ietf.org/proceedings/84/minutes/minutes-84-rmcat
" - Stephan Wenger suggested that the discussions have been behind the
   state of the art in joint source-channel coding. The group should
   consider that not all packets are equal in importance when designing
   congestion control algorithms. "


Cheers,
Michael


--Apple-Mail=_82949183-54B8-4833-8CFD-6446313134DD
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=iso-8859-1

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On 19. okt. 2012, at 06:52, Randell Jesup =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div>On 10/18/2012 3:45 AM, Michael Welzl =
wrote:<br><blockquote type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">The CC may say "you now have N =
kbps to work within" to each codec, =
and<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">this will be based on what it estimates is safe for the =
network given<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">the current ensemble of codecs =
and their stated operating =
envelopes,<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">*not* on any explicit attempt by =
the CC to maximize user =
experience.<br></blockquote></blockquote><blockquote type=3D"cite">Agreed.=
 Note, what you describe here is an "interaction between applications =
and RTP flows", which is a quote from the charter, a part of a milestone =
due in December... is anyone working on a draft for =
this?<br></blockquote><br>I have made a proposal in the W3 bugzilla and =
on the list. &nbsp;There was a small amount of discussion; none since. =
&nbsp;I'd suggest starting from there. &nbsp;It may be worthwhile to =
re-examine the API in light of the stats proposals to avoid =
conflicting/overlapping data.<br></div></blockquote><div><br></div>I =
think I've asked you before, and I think you probably answered =3D&gt; =
so I'm sorry for being too chaotic, but could you maybe give us that =
link again?</div><div><br></div><div><br><blockquote =
type=3D"cite"><div><blockquote type=3D"cite">This is where "packet =
importance" to be handed over to the congestion control should also =
fit... (that was discussed as a part of the sender- vs. receiver-based =
thread) IF we decide to have that.<br></blockquote><br>Well, packet =
importance would (generally) be in the internals, not at the =
application/JS layer, except maybe to provide stream-level =
priorities.<br></div></blockquote><div><br></div><div>Yes, but to me, =
"application" in the rmcat context is not the JS application but the =
codec. This is at least where that "interactions" deliverable originated =
- to address comments like this =
one:</div><div><br></div><div>from:&nbsp;<a =
href=3D"http://www.ietf.org/proceedings/84/minutes/minutes-84-rmcat">http:=
//www.ietf.org/proceedings/84/minutes/minutes-84-rmcat</a></div><div>"&nbs=
p;- Stephan Wenger suggested that the discussions have been behind =
the</div><div>&nbsp; &nbsp;state of the art in joint source-channel =
coding. The group should</div><div>&nbsp; &nbsp;consider that not all =
packets are equal in importance when designing</div><div>&nbsp; =
&nbsp;congestion control algorithms. =
"</div><div><br></div><div><br></div><div>Cheers,</div><div>Michael</div><=
div><br></div></div></body></html>=

--Apple-Mail=_82949183-54B8-4833-8CFD-6446313134DD--

From vsingh.ietf@gmail.com  Fri Oct 19 03:12:11 2012
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Fri, 19 Oct 2012 13:11:49 +0300
Message-ID: <CAEbPqryQpOreaAcyPuf5YDMjyd0daqNORBeW_N33znr2ubgPZg@mail.gmail.com>
To: "Piers O'Hanlon" <p.ohanlon@gmail.com>
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Subject: Re: [rmcat] New Version Notification for draft-singh-rmcat-cc-eval-00.txt
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Hi Piers,

Comments inline,

On Wed, Oct 17, 2012 at 2:25 PM, Piers O'Hanlon <p.ohanlon@gmail.com> wrote=
:
> Hi Varun,
>
> The draft looks like a good start.
>
> I think it would good to add evaluation of idle and data-limited behaviou=
rs - as these are important as regards media transport - they were the reas=
ons the original TFRC (RFC3448) was updated to RFC5348. See: http://dx.doi.=
org/10.1016%2fj.comcom.2011.05.003
> "TCP-Friendly Rate Control (TFRC) for bursty media flows", Arjuna Sathias=
eelan, Gorry Fairhurst, Comcom 2011
>

Do you mean the audio media traffic should be modelled as having idle
periods? and video because of I and P frames will have bursty
behaviour?


> Furthermore it would also be useful to evaluate the algorithms with respe=
ct to their operation in terms of competing against a small number of flows=
 vs a large number - see:
> On the Effectiveness of Delay-Based Congestion Avoidance, Ravi S. Prasad,=
 Manish Jain Georgia, Constantinos Dovrolis, PFLDnet 2004
>
In [S5], I have an editors note:
[many and few short TCP flows may depend on the bottleneck link capacity.]

I can add a line to both cross traffic and self fairness guideline
(Section 4) to evaluate the performance with small number and large
number of competing flows.

Cheers,
Varun

> Cheers,
>
> Piers
>
>
>
>
> On 15 Oct 2012, at 21:33, Varun Singh wrote:
>
>> Greetings,
>>
>> We have submitted an initial draft outlining the criteria for evaluating
>> congestion control algorithms. The metrics section is a bit light
>> on details and will to expand on that in the next revision.
>>
>> We hope that the draft is a good starting point to get the discussion
>> started about metrics and scenarios for simulation and testbed experimen=
ts.
>>
>> Cheers,
>> Varun
>>
>> Begin forwarded message:
>>
>>
>> A new version of I-D, draft-singh-rmcat-cc-eval-00.txt
>> has been successfully submitted by Varun Singh and posted to the
>> IETF repository.
>>
>> Filename:      draft-singh-rmcat-cc-eval
>> Revision:      00
>> Title:                 Evaluating Congestion Control for Interactive Rea=
l-time Media.
>> Creation date:         2012-10-15
>> WG ID:                 Individual Submission
>> Number of pages: 9
>> URL:
>> http://www.ietf.org/internet-drafts/draft-singh-rmcat-cc-eval-00.txt
>> Status:          http://datatracker.ietf.org/doc/draft-singh-rmcat-cc-ev=
al
>> Htmlized:        http://tools.ietf.org/html/draft-singh-rmcat-cc-eval-00
>>
>>
>> Abstract:
>>  The Real-time Transport Protocol (RTP) is used to transmit media in
>>  telephony and video conferencing applications.  This document
>>  describes the guidelines to evaluate new congestion control
>>  algorithms for interactive point-to-point real-time media.
>>
>>
>>
>>
>> The IETF Secretariat
>



--=20
http://www.netlab.tkk.fi/~varun/

From vsingh.ietf@gmail.com  Fri Oct 19 03:26:36 2012
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Fri, 19 Oct 2012 13:26:15 +0300
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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rmcat] Video Quality for RMCAT
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On Thu, Oct 18, 2012 at 4:32 PM, Harald Alvestrand <harald@alvestrand.no> wrote:
> On 10/18/2012 02:31 AM, Timothy B. Terriberry wrote:
>>
>> Varun Singh wrote:
>>>
>>> Apart from delay there is also balancing throughput and loss.
>>> In the scenario of a single RTP flow we could come up with a
>>> simple metric but with competing traffic it may be more difficult.
>>> By introducing a video quality metric in those scenarios,
>>> I was intending to quantify the balance of throughput, loss and delay.
>>
>>
>> If you want to measure video quality in the presence of changing bitrates,
>> loss, and delay, you have a difficult problem on your hands. Not the least
>> of which is that PSNR (or similar metrics) require a reference to compare
>> against, which starts to get meaningless once you change resolution or
>> framerate in the encoder (with dropping frames being a very crude form of
>> changing framerate), all of which are things the encoder can do to adapt to
>> changes in available bandwidth. I discussed some of this in more detail in
>> my submission to the workshop:
>> <http://www.tschofenig.priv.at/cc-workshop/irtf_iab-ccirtcpaper26.pdf>
>
> Wouldn't a simple (simplistic?) way of measuring quality in the presence of
> packet drops be to always use a test rig where one end looped back the
> encoded video signal to the other end?
>
> Thus, the one that injects the signal could compare the pre-encoding video
> source to the post-decoding video sink, which I suppose is what PSNR
> measurements need.
>

I was thinking about a similar setup when writing the document.
Perhaps the Quality Metrics should not be used to compare
between congestion control algorithms but compare the
behaviour of the congestion control (all metrics not only quality)
in different evaluation scenarios.


> This is inappropriate for measurements on a live service, but might be
> appropriate for generating benchmarks.




-- 
http://www.netlab.tkk.fi/~varun/

From michawe@ifi.uio.no  Fri Oct 19 03:31:28 2012
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Subject: Re: [rmcat] Video Quality for RMCAT
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On 19. okt. 2012, at 12:26, Varun Singh wrote:

> On Thu, Oct 18, 2012 at 4:32 PM, Harald Alvestrand =
<harald@alvestrand.no> wrote:
>> On 10/18/2012 02:31 AM, Timothy B. Terriberry wrote:
>>>=20
>>> Varun Singh wrote:
>>>>=20
>>>> Apart from delay there is also balancing throughput and loss.
>>>> In the scenario of a single RTP flow we could come up with a
>>>> simple metric but with competing traffic it may be more difficult.
>>>> By introducing a video quality metric in those scenarios,
>>>> I was intending to quantify the balance of throughput, loss and =
delay.
>>>=20
>>>=20
>>> If you want to measure video quality in the presence of changing =
bitrates,
>>> loss, and delay, you have a difficult problem on your hands. Not the =
least
>>> of which is that PSNR (or similar metrics) require a reference to =
compare
>>> against, which starts to get meaningless once you change resolution =
or
>>> framerate in the encoder (with dropping frames being a very crude =
form of
>>> changing framerate), all of which are things the encoder can do to =
adapt to
>>> changes in available bandwidth. I discussed some of this in more =
detail in
>>> my submission to the workshop:
>>> =
<http://www.tschofenig.priv.at/cc-workshop/irtf_iab-ccirtcpaper26.pdf>
>>=20
>> Wouldn't a simple (simplistic?) way of measuring quality in the =
presence of
>> packet drops be to always use a test rig where one end looped back =
the
>> encoded video signal to the other end?
>>=20
>> Thus, the one that injects the signal could compare the pre-encoding =
video
>> source to the post-decoding video sink, which I suppose is what PSNR
>> measurements need.
>>=20
>=20
> I was thinking about a similar setup when writing the document.
> Perhaps the Quality Metrics should not be used to compare
> between congestion control algorithms but compare the
> behaviour of the congestion control (all metrics not only quality)
> in different evaluation scenarios.

I can't parse that. Not compare between but compare the behaviour?

Anyway, I still think that a quality metric makes no sense here...  you =
can't get rid of the fact that it's bound to the codec?

Cheers,
Michael


From p.ohanlon@gmail.com  Fri Oct 19 04:59:01 2012
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From: Piers O'Hanlon <p.ohanlon@gmail.com>
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To: Varun Singh <vsingh.ietf@gmail.com>
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Subject: Re: [rmcat] New Version Notification for draft-singh-rmcat-cc-eval-00.txt
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Hi Varun,

On 19 Oct 2012, at 11:11, Varun Singh wrote:

> Hi Piers,
>=20
> Comments inline,
>=20
> On Wed, Oct 17, 2012 at 2:25 PM, Piers O'Hanlon <p.ohanlon@gmail.com> =
wrote:
>> Hi Varun,
>>=20
>> The draft looks like a good start.
>>=20
>> I think it would good to add evaluation of idle and data-limited =
behaviours - as these are important as regards media transport - they =
were the reasons the original TFRC (RFC3448) was updated to RFC5348. =
See: http://dx.doi.org/10.1016%2fj.comcom.2011.05.003
>> "TCP-Friendly Rate Control (TFRC) for bursty media flows", Arjuna =
Sathiaseelan, Gorry Fairhurst, Comcom 2011
>>=20
>=20
> Do you mean the audio media traffic should be modelled as having idle
> periods? and video because of I and P frames will have bursty
> behaviour?
>=20
Yes both media can have idle and data-limited periods due to silence or =
low motion (depending upon codec type etc) - If we're trying stay away =
from adding full QoE evaluation these terms describe more generic =
behaviours that should be addressed when testing a congestion control =
scheme. These were some of the original complaints about TFRC operation =
that lead to the formation of this group.

These behaviours are also linked to the corresponding TCP Congestion =
Window Validation techniques (e.g. RFC2861)

>=20
>> Furthermore it would also be useful to evaluate the algorithms with =
respect to their operation in terms of competing against a small number =
of flows vs a large number - see:
>> On the Effectiveness of Delay-Based Congestion Avoidance, Ravi S. =
Prasad, Manish Jain Georgia, Constantinos Dovrolis, PFLDnet 2004
>>=20
> In [S5], I have an editors note:
> [many and few short TCP flows may depend on the bottleneck link =
capacity.]
>=20
> I can add a line to both cross traffic and self fairness guideline
> (Section 4) to evaluate the performance with small number and large
> number of competing flows.
>=20
That sounds good. This issue is more of a concern as most of the =
proposed schemes are likely to be primarily based upon some sort of =
delay measurement which can become more problematic in certain scenarios =
as mentioned in the short paper (and elsewhere).

Cheers,

Piers


> Cheers,
> Varun
>=20
>> Cheers,
>>=20
>> Piers
>>=20
>>=20
>>=20
>>=20
>> On 15 Oct 2012, at 21:33, Varun Singh wrote:
>>=20
>>> Greetings,
>>>=20
>>> We have submitted an initial draft outlining the criteria for =
evaluating
>>> congestion control algorithms. The metrics section is a bit light
>>> on details and will to expand on that in the next revision.
>>>=20
>>> We hope that the draft is a good starting point to get the =
discussion
>>> started about metrics and scenarios for simulation and testbed =
experiments.
>>>=20
>>> Cheers,
>>> Varun
>>>=20
>>> Begin forwarded message:
>>>=20
>>>=20
>>> A new version of I-D, draft-singh-rmcat-cc-eval-00.txt
>>> has been successfully submitted by Varun Singh and posted to the
>>> IETF repository.
>>>=20
>>> Filename:      draft-singh-rmcat-cc-eval
>>> Revision:      00
>>> Title:                 Evaluating Congestion Control for Interactive =
Real-time Media.
>>> Creation date:         2012-10-15
>>> WG ID:                 Individual Submission
>>> Number of pages: 9
>>> URL:
>>> http://www.ietf.org/internet-drafts/draft-singh-rmcat-cc-eval-00.txt
>>> Status:          =
http://datatracker.ietf.org/doc/draft-singh-rmcat-cc-eval
>>> Htmlized:        =
http://tools.ietf.org/html/draft-singh-rmcat-cc-eval-00
>>>=20
>>>=20
>>> Abstract:
>>> The Real-time Transport Protocol (RTP) is used to transmit media in
>>> telephony and video conferencing applications.  This document
>>> describes the guidelines to evaluate new congestion control
>>> algorithms for interactive point-to-point real-time media.
>>>=20
>>>=20
>>>=20
>>>=20
>>> The IETF Secretariat
>>=20
>=20
>=20
>=20
> --=20
> http://www.netlab.tkk.fi/~varun/


From harald@alvestrand.no  Fri Oct 19 06:09:36 2012
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Cc: rmcat@ietf.org, Varun Singh <vsingh.ietf@gmail.com>
Subject: Re: [rmcat] Video Quality for RMCAT
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On 10/19/2012 12:31 PM, Michael Welzl wrote:
> On 19. okt. 2012, at 12:26, Varun Singh wrote:
>
>> On Thu, Oct 18, 2012 at 4:32 PM, Harald Alvestrand <harald@alvestrand.no> wrote:
>>> On 10/18/2012 02:31 AM, Timothy B. Terriberry wrote:
>>>> Varun Singh wrote:
>>>>> Apart from delay there is also balancing throughput and loss.
>>>>> In the scenario of a single RTP flow we could come up with a
>>>>> simple metric but with competing traffic it may be more difficult.
>>>>> By introducing a video quality metric in those scenarios,
>>>>> I was intending to quantify the balance of throughput, loss and delay.
>>>>
>>>> If you want to measure video quality in the presence of changing bitrates,
>>>> loss, and delay, you have a difficult problem on your hands. Not the least
>>>> of which is that PSNR (or similar metrics) require a reference to compare
>>>> against, which starts to get meaningless once you change resolution or
>>>> framerate in the encoder (with dropping frames being a very crude form of
>>>> changing framerate), all of which are things the encoder can do to adapt to
>>>> changes in available bandwidth. I discussed some of this in more detail in
>>>> my submission to the workshop:
>>>> <http://www.tschofenig.priv.at/cc-workshop/irtf_iab-ccirtcpaper26.pdf>
>>> Wouldn't a simple (simplistic?) way of measuring quality in the presence of
>>> packet drops be to always use a test rig where one end looped back the
>>> encoded video signal to the other end?
>>>
>>> Thus, the one that injects the signal could compare the pre-encoding video
>>> source to the post-decoding video sink, which I suppose is what PSNR
>>> measurements need.
>>>
>> I was thinking about a similar setup when writing the document.
>> Perhaps the Quality Metrics should not be used to compare
>> between congestion control algorithms but compare the
>> behaviour of the congestion control (all metrics not only quality)
>> in different evaluation scenarios.
> I can't parse that. Not compare between but compare the behaviour?
>
> Anyway, I still think that a quality metric makes no sense here...  you can't get rid of the fact that it's bound to the codec?
And that makes some sense.... algorithms need to be defined without 
reference to codecs, and some of their evaluation criteria 
(slef-fairness, lack of congestion collapse) need to be evaluated 
independent of the codec, but what impact they have on media streams 
encoded with existing codecs matters for real life deployments.



From radhika.r.roy.civ@mail.mil  Fri Oct 19 07:17:18 2012
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Hi, Michael:

Yes, in one way, I did misunderstood Harald's scenarios. The key point that
I have been pointing out that we have not got any mathematical models to
quantify the situations of congestions for multimedia traffic with competing
goals of bandwidth and performance requirements over UDP transport.

In absence of the above, we are trying something that is based on some
heuristic things hoping to work. Well, I am not against this approach. All I
am trying to point to the long-term approach in addressing the traffic
congestions problems.

BR/Radhika

-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of
Michael Welzl
Sent: Friday, October 19, 2012 3:52 AM
To: Mirja Kuehlewind
Cc: rmcat@ietf.org; Harald Alvestrand
Subject: Re: [rmcat] Multihop congestion control - two words of caution

...

Radhika: your answer appears (to me) to miss the point, and maybe (sorry if
I'm wrong!) it's because you misunderstood the scenario? So maybe my answer
to Mirja below helps.

...

I hope my interpretation of Harald's scenario is correct, and I hope that
this was helpful!

Cheers,
Michael



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From randell-ietf@jesup.org  Fri Oct 19 09:19:03 2012
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Subject: Re: [rmcat] Video Quality for RMCAT
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On 10/19/2012 3:56 AM, Michael Welzl wrote:
>
> On 19. okt. 2012, at 06:52, Randell Jesup wrote:
>
>> On 10/18/2012 3:45 AM, Michael Welzl wrote:
>>>
>>>> The CC may say "you now have N kbps to work within" to each codec, and
>>>> this will be based on what it estimates is safe for the network given
>>>> the current ensemble of codecs and their stated operating envelopes,
>>>> *not* on any explicit attempt by the CC to maximize user experience.
>>> Agreed. Note, what you describe here is an "interaction between 
>>> applications and RTP flows", which is a quote from the charter, a 
>>> part of a milestone due in December... is anyone working on a draft 
>>> for this?
>>
>> I have made a proposal in the W3 bugzilla and on the list.  There was 
>> a small amount of discussion; none since.  I'd suggest starting from 
>> there.  It may be worthwhile to re-examine the API in light of the 
>> stats proposals to avoid conflicting/overlapping data.
>
> I think I've asked you before, and I think you probably answered => so 
> I'm sorry for being too chaotic, but could you maybe give us that link 
> again?

https://www.w3.org/Bugs/Public/show_bug.cgi?id=15861

It was also discussed on the lists (rtcweb or public-webrtc more likely)

-- 
Randell Jesup
randell-ietf@jesup.org


From randell-ietf@jesup.org  Fri Oct 19 09:27:20 2012
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Just FYI, I'm still waiting for draft-jesup-rmcat-reqs-00 to be posted; 
if it doesn't get posted in time I'll update 
draft-jesup-rtp-congestion-reqs to -01 and put everything there. (My 
rmcat draft is a derivative of the rtp-congestion-reqs document.)  Once 
-00 is posted I still plan to revise it before the deadline.

-- 
Randell Jesup
randell-ietf@jesup.org	


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Cc: rmcat@ietf.org
Subject: Re: [rmcat] Video Quality for RMCAT
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On Oct 19, 2012, at 6:16 PM, Randell Jesup wrote:

> On 10/19/2012 3:56 AM, Michael Welzl wrote:
>>
>> On 19. okt. 2012, at 06:52, Randell Jesup wrote:
>>
>>> On 10/18/2012 3:45 AM, Michael Welzl wrote:
>>>>
>>>>> The CC may say "you now have N kbps to work within" to each  
>>>>> codec, and
>>>>> this will be based on what it estimates is safe for the network  
>>>>> given
>>>>> the current ensemble of codecs and their stated operating  
>>>>> envelopes,
>>>>> *not* on any explicit attempt by the CC to maximize user  
>>>>> experience.
>>>> Agreed. Note, what you describe here is an "interaction between  
>>>> applications and RTP flows", which is a quote from the charter, a  
>>>> part of a milestone due in December... is anyone working on a  
>>>> draft for this?
>>>
>>> I have made a proposal in the W3 bugzilla and on the list.  There  
>>> was a small amount of discussion; none since.  I'd suggest  
>>> starting from there.  It may be worthwhile to re-examine the API  
>>> in light of the stats proposals to avoid conflicting/overlapping  
>>> data.
>>
>> I think I've asked you before, and I think you probably answered =>  
>> so I'm sorry for being too chaotic, but could you maybe give us  
>> that link again?
>
> https://www.w3.org/Bugs/Public/show_bug.cgi?id=15861
>
> It was also discussed on the lists (rtcweb or public-webrtc more  
> likely)

Thanks!!

And: well, I took a look - that seems really unrelated to the type of  
communication needed between a congestion controller and a codec??

Cheers,
Michael


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From: "Eggert, Lars" <lars@netapp.com>
To: Varun Singh <vsingh.ietf@gmail.com>
Thread-Topic: [rmcat] New Version Notification for draft-singh-rmcat-cc-eval-00.txt
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Hi,

you may also want to take a look at the following paper, which attempted =
to define some community-agreed evaluation scenarios for proposed TCP =
extensions. Maybe there are some aspects in there that are applicable to =
your draft.

Towards a Common TCP Evaluation Suite. Lachlan Andrew, Cesar Marcondes, =
Sally Floyd, Lawrence Dunn, Wang Gang, Lars Eggert, Sangtae Ha and =
Injong Rhee. Proc. International Workshop on Protocols for Fast =
Long-Distance Networks (PFLDnet),Manchester, UK, March 5-7, 2008. =
http://eggert.org/papers/2008-pfldnet.pdf

Lars=

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There isn't much change to this document, but some new discussion on 
combining information across multiple streams.

Given the time with respect to the draft deadline, I didn't change the name.

                 Harald


-------- Original Message --------
Subject: 	New Version Notification for 
draft-alvestrand-rtcweb-congestion-03.txt
Date: 	Mon, 22 Oct 2012 05:44:03 -0700
From: 	internet-drafts@ietf.org
To: 	harald@alvestrand.no
CC: 	holmer@google.com



A new version of I-D, draft-alvestrand-rtcweb-congestion-03.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Filename:	 draft-alvestrand-rtcweb-congestion
Revision:	 03
Title:		 A Google Congestion Control Algorithm for Real-Time Communication on the World Wide Web
Creation date:	 2012-10-22
WG ID:		 Individual Submission
Number of pages: 18
URL:             http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-congestion-03.txt
Status:          http://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-congestion
Htmlized:        http://tools.ietf.org/html/draft-alvestrand-rtcweb-congestion-03
Diff:            http://www.ietf.org/rfcdiff?url2=draft-alvestrand-rtcweb-congestion-03

Abstract:
    This document describes two methods of congestion control when using
    real-time communications on the World Wide Web (RTCWEB); one sender-
    based and one receiver-based.

    It is published as an input document to the RMCAT working group on
    congestion control for media streams.  The mailing list of that WG is
    rmcat@ietf.org.


                                                                                   


The IETF Secretariat





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    There isn't much change to this document, but some new discussion on
    combining information across multiple streams.<br>
    <br>
    Given the time with respect to the draft deadline, I didn't change
    the name.<br>
    <br>
    Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â  Harald<br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>New Version Notification for
              draft-alvestrand-rtcweb-congestion-03.txt</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Mon, 22 Oct 2012 05:44:03 -0700</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:holmer@google.com">holmer@google.com</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>A new version of I-D, draft-alvestrand-rtcweb-congestion-03.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Filename:	 draft-alvestrand-rtcweb-congestion
Revision:	 03
Title:		 A Google Congestion Control Algorithm for Real-Time Communication on the World Wide Web
Creation date:	 2012-10-22
WG ID:		 Individual Submission
Number of pages: 18
URL:             <a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-congestion-03.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-congestion-03.txt</a>
Status:          <a class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-congestion">http://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-congestion</a>
Htmlized:        <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-alvestrand-rtcweb-congestion-03">http://tools.ietf.org/html/draft-alvestrand-rtcweb-congestion-03</a>
Diff:            <a class="moz-txt-link-freetext" href="http://www.ietf.org/rfcdiff?url2=draft-alvestrand-rtcweb-congestion-03">http://www.ietf.org/rfcdiff?url2=draft-alvestrand-rtcweb-congestion-03</a>

Abstract:
   This document describes two methods of congestion control when using
   real-time communications on the World Wide Web (RTCWEB); one sender-
   based and one receiver-based.

   It is published as an input document to the RMCAT working group on
   congestion control for media streams.  The mailing list of that WG is
   <a class="moz-txt-link-abbreviated" href="mailto:rmcat@ietf.org">rmcat@ietf.org</a>.


                                                                                  


The IETF Secretariat

</pre>
      <br>
      <br>
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    <br>
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</html>

--------------000508010305000202010704--

From mzanaty@cisco.com  Mon Oct 22 11:55:44 2012
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Michael Welzl <michawe@ifi.uio.no>, Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>
Thread-Topic: [rmcat] Multihop congestion control - two words of caution
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References: <50800364.1070908@alvestrand.no> <201210181758.44757.mirja.kuehlewind@ikr.uni-stuttgart.de> <0AC4ED0D-88AB-466F-B4EA-EC75A5EC53FB@ifi.uio.no>
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Cc: "rmcat@ietf.org" <rmcat@ietf.org>, Harald Alvestrand <harald@alvestrand.no>
Subject: Re: [rmcat] Multihop congestion control - two words of caution
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Do we consider a TURN server or other UDP relay to be multi-hop? By relay, =
I mean something which never alters or processes the RTP/RTCP at all. It is=
 certainly another hop from the network and congestion perspectives, but no=
t the RTP perspective.

The multi-hop case is certainly more complex, but complexity alone should n=
ot determine whether we tackle it or not. The importance of the problem and=
 the need for a solution should be considered along with complexity. I susp=
ect many major deployments of RMCAT will have multi-hop cases, and they may=
 perhaps even be the dominant use cases. If we punt based on complexity, th=
ere will likely be wildly different behavior across these cases. (Even wild=
er than double pendulums...)

Mo

-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of M=
ichael Welzl
Sent: Friday, October 19, 2012 3:52 AM
To: Mirja Kuehlewind
Cc: rmcat@ietf.org; Harald Alvestrand
Subject: Re: [rmcat] Multihop congestion control - two words of caution

Hi all,

I feel in a bit of a dilemma answering Harald's email, because this require=
s that I call myself an "expert"... yet I'm tempted to answer... tsk!   :-)=
     Anyway, I'll try to reply the emails from Harald, Radhika and Mirja in=
 one go here, in the hope that I can clarify things a little:

Harald: all of what you have written sounds 100% reasonable to me.

Radhika: your answer appears (to me) to miss the point, and maybe (sorry if=
 I'm wrong!) it's because you misunderstood the scenario? So maybe my answe=
r to Mirja below helps.

Mirja: RTP/RTCP can let end systems take the role of an intermediate node, =
to take care of data stream specific things before traffic reaches the rece=
iver (synchronizing, mixing, ...). Essentially, this is an overlay network.=
 I think Harald refers to such a case with multiple RTP end systems, i.e. a=
 path like this:

Node1 [ RTP1 sender ] =3D> Node 2 [RTP1 receiver + RTP2 sender] =3D> Node 3=
 [RTP2 receiver]

...and his description of problems goes into the behavior of node 2 (which =
is, in principle, quite similar to the behavior of some PEPs (RFC3135), I g=
uess, e.g. TCP splitters. The decision at the BOF was not *yet* to define w=
hat is needed to make Node 2 work correctly, which is what we referred to a=
s multi-hop congestion control. The exact phrasing in the charter is: "impl=
ications on multi-hop connections will be considered at a later stage". The=
 reason for writing this is that Magnus Westerlund (quite correctly, I thin=
k) stated that defining this will at some point be needed.

I hope my interpretation of Harald's scenario is correct, and I hope that t=
his was helpful!

Cheers,
Michael


On 18. okt. 2012, at 17:58, Mirja Kuehlewind wrote:

> Hi Harald,
>=20
> i believe do not understand your scenario:
>=20
>> Now consider what happens if we have two hops after each other, possibly
>> with different delays, each running their own congestion control
>> algorithm - and a linkage that tries to feed congestion control
>> information from the second hop back to the congestion control algorithm
>> running on the first hop.
>=20
> Are we talking of end-to-end congestion control here? Then if we have=20
> congestion feedback from different nodes on one path, this feedback signa=
l=20
> will have the same delay. And usually there is only one bottleneck at one=
=20
> point of time as there is always on link with the smallest available=20
> bandwidth. Is this the scenario your are talking about or something else?
>=20
> Mirja
>=20
>=20
>=20
>>=20
>> The overall system will have a delay - but that delay won't be visible
>> to the congestion control algorithm on the first hop, so the reaction
>> time and size can't be scaled appropriately.
>>=20
>> The result is that the overall system is very likely to exhibit
>> oscillatory behaviour - simply because the time constants we need to
>> observe for a proper design are invisible to the observer.
>>=20
>> We can imagine ways of working around this limitation - but for the
>> purposes of RMCAT, I think we should just stick with two principles:
>>=20
>> - We don't design for the two-hop case.
>> - We advise designers of two-hop systems (like relay nodes) that if they
>> want to feed capacity information from a second hop back to the first
>> hop, they should reflect those changes SLOWLY.
>>=20
>> Experts, did this sound reasonable?
>>=20
>>                      Harald
>=20
>=20
>=20
> --=20
> -------------------------------------------------------------------
> Dipl.-Ing. Mirja K=FChlewind
> Institute of Communication Networks and Computer Engineering (IKR)
> University of Stuttgart, Germany
> Pfaffenwaldring 47, D-70569 Stuttgart
>=20
> tel: +49(0)711/685-67973
> email: mirja.kuehlewind@ikr.uni-stuttgart.de
> web: www.ikr.uni-stuttgart.de
> -------------------------------------------------------------------


From p.ohanlon@gmail.com  Mon Oct 22 14:14:06 2012
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From: "Piers O'Hanlon" <p.ohanlon@gmail.com>
Date: Mon, 22 Oct 2012 22:13:44 +0100
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Subject: [rmcat] Fwd: New Version Notification for draft-ohanlon-rmcat-dflow-01.txt
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Here's an updated version of the draft on our DFlow congestion control scheme.

It is still under development and testing so comments welcome.

Cheers,

Piers.

Begin forwarded message:

From: <internet-drafts@ietf.org>
Subject: New Version Notification for draft-ohanlon-rmcat-dflow-01.txt
Date: 22 October 2012 21:53:49 GMT+01:00
To: <piers.ohanlon@oii.ox.ac.uk>
Cc: <carlberg@g11.org.uk>


A new version of I-D, draft-ohanlon-rmcat-dflow-01.txt
has been successfully submitted by Piers O'Hanlon and posted to the
IETF repository.

Filename: draft-ohanlon-rmcat-dflow
Revision: 01
Title: Congestion control algorithm for lower latency and lower loss
media transport
Creation date: 2012-10-22
WG ID: Individual Submission
Number of pages: 10
URL:
http://www.ietf.org/internet-drafts/draft-ohanlon-rmcat-dflow-01.txt
Status:          http://datatracker.ietf.org/doc/draft-ohanlon-rmcat-dflow
Htmlized:        http://tools.ietf.org/html/draft-ohanlon-rmcat-dflow-01
Diff:            http://www.ietf.org/rfcdiff?url2=draft-ohanlon-rmcat-dflow-01

Abstract:
  This memo provides an initial design for a congestion control
  algorithm, for media transport, which aims to provide for lower delay
  and lower loss communications.




The IETF Secretariat

From harald@alvestrand.no  Tue Oct 23 04:16:35 2012
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Cc: "rmcat@ietf.org" <rmcat@ietf.org>, Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>, Michael Welzl <michawe@ifi.uio.no>
Subject: Re: [rmcat] Multihop congestion control - two words of caution
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On 10/22/2012 08:55 PM, Mo Zanaty (mzanaty) wrote:
> Do we consider a TURN server or other UDP relay to be multi-hop? By relay, I mean something which never alters or processes the RTP/RTCP at all. It is certainly another hop from the network and congestion perspectives, but not the RTP perspective.
>
> The multi-hop case is certainly more complex, but complexity alone should not determine whether we tackle it or not. The importance of the problem and the need for a solution should be considered along with complexity. I suspect many major deployments of RMCAT will have multi-hop cases, and they may perhaps even be the dominant use cases. If we punt based on complexity, there will likely be wildly different behavior across these cases. (Even wilder than double pendulums...)

I don't think of a link via a TURN server as multihop; the TURN server 
should have the same characteristics as a switch or router in the path 
(lose packets if congested, but don't attempt to shape the stream).

I'd like to see guidelines (such as "go slower than X when you're a 
destination changing the desirable bandwidth based on something other 
than source/destination congestion") that avoid the double-pendulum 
style chaotic behaviour. I don't want to start off trying to write a 
controller for efficient management of double pendulums.

>
> Mo
>
> -----Original Message-----
> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of Michael Welzl
> Sent: Friday, October 19, 2012 3:52 AM
> To: Mirja Kuehlewind
> Cc: rmcat@ietf.org; Harald Alvestrand
> Subject: Re: [rmcat] Multihop congestion control - two words of caution
>
> Hi all,
>
> I feel in a bit of a dilemma answering Harald's email, because this requires that I call myself an "expert"... yet I'm tempted to answer... tsk!   :-)     Anyway, I'll try to reply the emails from Harald, Radhika and Mirja in one go here, in the hope that I can clarify things a little:
>
> Harald: all of what you have written sounds 100% reasonable to me.
>
> Radhika: your answer appears (to me) to miss the point, and maybe (sorry if I'm wrong!) it's because you misunderstood the scenario? So maybe my answer to Mirja below helps.
>
> Mirja: RTP/RTCP can let end systems take the role of an intermediate node, to take care of data stream specific things before traffic reaches the receiver (synchronizing, mixing, ...). Essentially, this is an overlay network. I think Harald refers to such a case with multiple RTP end systems, i.e. a path like this:
>
> Node1 [ RTP1 sender ] => Node 2 [RTP1 receiver + RTP2 sender] => Node 3 [RTP2 receiver]
>
> ...and his description of problems goes into the behavior of node 2 (which is, in principle, quite similar to the behavior of some PEPs (RFC3135), I guess, e.g. TCP splitters. The decision at the BOF was not *yet* to define what is needed to make Node 2 work correctly, which is what we referred to as multi-hop congestion control. The exact phrasing in the charter is: "implications on multi-hop connections will be considered at a later stage". The reason for writing this is that Magnus Westerlund (quite correctly, I think) stated that defining this will at some point be needed.
>
> I hope my interpretation of Harald's scenario is correct, and I hope that this was helpful!
>
> Cheers,
> Michael
>
>
> On 18. okt. 2012, at 17:58, Mirja Kuehlewind wrote:
>
>> Hi Harald,
>>
>> i believe do not understand your scenario:
>>
>>> Now consider what happens if we have two hops after each other, possibly
>>> with different delays, each running their own congestion control
>>> algorithm - and a linkage that tries to feed congestion control
>>> information from the second hop back to the congestion control algorithm
>>> running on the first hop.
>> Are we talking of end-to-end congestion control here? Then if we have
>> congestion feedback from different nodes on one path, this feedback signal
>> will have the same delay. And usually there is only one bottleneck at one
>> point of time as there is always on link with the smallest available
>> bandwidth. Is this the scenario your are talking about or something else?
>>
>> Mirja
>>
>>
>>
>>> The overall system will have a delay - but that delay won't be visible
>>> to the congestion control algorithm on the first hop, so the reaction
>>> time and size can't be scaled appropriately.
>>>
>>> The result is that the overall system is very likely to exhibit
>>> oscillatory behaviour - simply because the time constants we need to
>>> observe for a proper design are invisible to the observer.
>>>
>>> We can imagine ways of working around this limitation - but for the
>>> purposes of RMCAT, I think we should just stick with two principles:
>>>
>>> - We don't design for the two-hop case.
>>> - We advise designers of two-hop systems (like relay nodes) that if they
>>> want to feed capacity information from a second hop back to the first
>>> hop, they should reflect those changes SLOWLY.
>>>
>>> Experts, did this sound reasonable?
>>>
>>>                       Harald
>>
>>
>> -- 
>> -------------------------------------------------------------------
>> Dipl.-Ing. Mirja Kühlewind
>> Institute of Communication Networks and Computer Engineering (IKR)
>> University of Stuttgart, Germany
>> Pfaffenwaldring 47, D-70569 Stuttgart
>>
>> tel: +49(0)711/685-67973
>> email: mirja.kuehlewind@ikr.uni-stuttgart.de
>> web: www.ikr.uni-stuttgart.de
>> -------------------------------------------------------------------


From michawe@ifi.uio.no  Tue Oct 23 04:23:41 2012
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Cc: "rmcat@ietf.org" <rmcat@ietf.org>, Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>, "Mo Zanaty \(mzanaty\)" <mzanaty@cisco.com>
Subject: Re: [rmcat] Multihop congestion control - two words of caution
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On 23. okt. 2012, at 13:16, Harald Alvestrand wrote:

> On 10/22/2012 08:55 PM, Mo Zanaty (mzanaty) wrote:
>> Do we consider a TURN server or other UDP relay to be multi-hop? By =
relay, I mean something which never alters or processes the RTP/RTCP at =
all. It is certainly another hop from the network and congestion =
perspectives, but not the RTP perspective.
>>=20
>> The multi-hop case is certainly more complex, but complexity alone =
should not determine whether we tackle it or not. The importance of the =
problem and the need for a solution should be considered along with =
complexity. I suspect many major deployments of RMCAT will have =
multi-hop cases, and they may perhaps even be the dominant use cases. If =
we punt based on complexity, there will likely be wildly different =
behavior across these cases. (Even wilder than double pendulums...)
>=20
> I don't think of a link via a TURN server as multihop; the TURN server =
should have the same characteristics as a switch or router in the path =
(lose packets if congested, but don't attempt to shape the stream).
>=20
> I'd like to see guidelines (such as "go slower than X when you're a =
destination changing the desirable bandwidth based on something other =
than source/destination congestion") that avoid the double-pendulum =
style chaotic behaviour. I don't want to start off trying to write a =
controller for efficient management of double pendulums.

Things don't have to get THAT bad... after all, TCP splitters do pretty =
much the same job that an intermediate hop in our setup would be doing, =
probably without getting into the fun of a double pendulum. Now, that's =
always easy to say, with little proof out there... but we could dig =
through the early literature on PEPs, I suppose, for a start...

Cheers,
Michael


From mirja.kuehlewind@ikr.uni-stuttgart.de  Tue Oct 23 04:51:35 2012
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Subject: Re: [rmcat] Multihop congestion control - two words of caution
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Hi Micheal,

thanks for the explanation. I believe I understood the scenario now but I'm=
=20
still not sure about the problem. In TCP you have separate congestion contr=
ol=20
on each hop. If the second hop is slower than the first one, you also have=
=20
flow control to not overload the receiver (by announcing a certain receive=
=20
window). Is there a benefit in having a common congestion control over both=
=20
hops?

Mirja


On Friday 19 October 2012 09:52:03 Michael Welzl wrote:
> Hi all,
>
> I feel in a bit of a dilemma answering Harald's email, because this
> requires that I call myself an "expert"... yet I'm tempted to answer...
> tsk!   :-)     Anyway, I'll try to reply the emails from Harald, Radhika
> and Mirja in one go here, in the hope that I can clarify things a little:
>
> Harald: all of what you have written sounds 100% reasonable to me.
>
> Radhika: your answer appears (to me) to miss the point, and maybe (sorry =
if
> I'm wrong!) it's because you misunderstood the scenario? So maybe my answ=
er
> to Mirja below helps.
>
> Mirja: RTP/RTCP can let end systems take the role of an intermediate node,
> to take care of data stream specific things before traffic reaches the
> receiver (synchronizing, mixing, ...). Essentially, this is an overlay
> network. I think Harald refers to such a case with multiple RTP end
> systems, i.e. a path like this:
>
> Node1 [ RTP1 sender ] =3D> Node 2 [RTP1 receiver + RTP2 sender] =3D> Node=
 3
> [RTP2 receiver]
>
> ...and his description of problems goes into the behavior of node 2 (which
> is, in principle, quite similar to the behavior of some PEPs (RFC3135), I
> guess, e.g. TCP splitters. The decision at the BOF was not *yet* to define
> what is needed to make Node 2 work correctly, which is what we referred to
> as multi-hop congestion control. The exact phrasing in the charter is:
> "implications on multi-hop connections will be considered at a later
> stage". The reason for writing this is that Magnus Westerlund (quite
> correctly, I think) stated that defining this will at some point be neede=
d.
>
> I hope my interpretation of Harald's scenario is correct, and I hope that
> this was helpful!
>
> Cheers,
> Michael
>
> On 18. okt. 2012, at 17:58, Mirja Kuehlewind wrote:
> > Hi Harald,
> >
> > i believe do not understand your scenario:
> >> Now consider what happens if we have two hops after each other, possib=
ly
> >> with different delays, each running their own congestion control
> >> algorithm - and a linkage that tries to feed congestion control
> >> information from the second hop back to the congestion control algorit=
hm
> >> running on the first hop.
> >
> > Are we talking of end-to-end congestion control here? Then if we have
> > congestion feedback from different nodes on one path, this feedback
> > signal will have the same delay. And usually there is only one bottlene=
ck
> > at one point of time as there is always on link with the smallest
> > available bandwidth. Is this the scenario your are talking about or
> > something else?
> >
> > Mirja
> >
> >> The overall system will have a delay - but that delay won't be visible
> >> to the congestion control algorithm on the first hop, so the reaction
> >> time and size can't be scaled appropriately.
> >>
> >> The result is that the overall system is very likely to exhibit
> >> oscillatory behaviour - simply because the time constants we need to
> >> observe for a proper design are invisible to the observer.
> >>
> >> We can imagine ways of working around this limitation - but for the
> >> purposes of RMCAT, I think we should just stick with two principles:
> >>
> >> - We don't design for the two-hop case.
> >> - We advise designers of two-hop systems (like relay nodes) that if th=
ey
> >> want to feed capacity information from a second hop back to the first
> >> hop, they should reflect those changes SLOWLY.
> >>
> >> Experts, did this sound reasonable?
> >>
> >>                      Harald
> >
> > --
> > -------------------------------------------------------------------
> > Dipl.-Ing. Mirja K=FChlewind
> > Institute of Communication Networks and Computer Engineering (IKR)
> > University of Stuttgart, Germany
> > Pfaffenwaldring 47, D-70569 Stuttgart
> >
> > tel: +49(0)711/685-67973
> > email: mirja.kuehlewind@ikr.uni-stuttgart.de
> > web: www.ikr.uni-stuttgart.de
> > -------------------------------------------------------------------



=2D-=20
=2D------------------------------------------------------------------
Dipl.-Ing. Mirja K=FChlewind
Institute of Communication Networks and Computer Engineering (IKR)
University of Stuttgart, Germany
Pfaffenwaldring 47, D-70569 Stuttgart

tel: +49(0)711/685-67973
email: mirja.kuehlewind@ikr.uni-stuttgart.de
web: www.ikr.uni-stuttgart.de
=2D------------------------------------------------------------------

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Hi,


On 23. okt. 2012, at 13:51, Mirja Kuehlewind wrote:

> Hi Micheal,
>=20
> thanks for the explanation. I believe I understood the scenario now =
but I'm=20
> still not sure about the problem. In TCP you have separate congestion =
control=20
> on each hop. If the second hop is slower than the first one, you also =
have=20
> flow control to not overload the receiver (by announcing a certain =
receive=20

Right;


> window). Is there a benefit in having a common congestion control over =
both=20
> hops?

I really don't know. Maybe? Not sure if someone suggested that?
Anyway, all I was trying to explain is that this is the scenario that, =
according to the charter, we're not looking at (and we keep it for the =
future).

This being said, personally, I'm interested in it and don't have a =
problem with discussions  :)   I just wanted to explain/clarify.

Cheers,
Michael


>=20
> Mirja
>=20
>=20
> On Friday 19 October 2012 09:52:03 Michael Welzl wrote:
>> Hi all,
>>=20
>> I feel in a bit of a dilemma answering Harald's email, because this
>> requires that I call myself an "expert"... yet I'm tempted to =
answer...
>> tsk!   :-)     Anyway, I'll try to reply the emails from Harald, =
Radhika
>> and Mirja in one go here, in the hope that I can clarify things a =
little:
>>=20
>> Harald: all of what you have written sounds 100% reasonable to me.
>>=20
>> Radhika: your answer appears (to me) to miss the point, and maybe =
(sorry if
>> I'm wrong!) it's because you misunderstood the scenario? So maybe my =
answer
>> to Mirja below helps.
>>=20
>> Mirja: RTP/RTCP can let end systems take the role of an intermediate =
node,
>> to take care of data stream specific things before traffic reaches =
the
>> receiver (synchronizing, mixing, ...). Essentially, this is an =
overlay
>> network. I think Harald refers to such a case with multiple RTP end
>> systems, i.e. a path like this:
>>=20
>> Node1 [ RTP1 sender ] =3D> Node 2 [RTP1 receiver + RTP2 sender] =3D> =
Node 3
>> [RTP2 receiver]
>>=20
>> ...and his description of problems goes into the behavior of node 2 =
(which
>> is, in principle, quite similar to the behavior of some PEPs =
(RFC3135), I
>> guess, e.g. TCP splitters. The decision at the BOF was not *yet* to =
define
>> what is needed to make Node 2 work correctly, which is what we =
referred to
>> as multi-hop congestion control. The exact phrasing in the charter =
is:
>> "implications on multi-hop connections will be considered at a later
>> stage". The reason for writing this is that Magnus Westerlund (quite
>> correctly, I think) stated that defining this will at some point be =
needed.
>>=20
>> I hope my interpretation of Harald's scenario is correct, and I hope =
that
>> this was helpful!
>>=20
>> Cheers,
>> Michael
>>=20
>> On 18. okt. 2012, at 17:58, Mirja Kuehlewind wrote:
>>> Hi Harald,
>>>=20
>>> i believe do not understand your scenario:
>>>> Now consider what happens if we have two hops after each other, =
possibly
>>>> with different delays, each running their own congestion control
>>>> algorithm - and a linkage that tries to feed congestion control
>>>> information from the second hop back to the congestion control =
algorithm
>>>> running on the first hop.
>>>=20
>>> Are we talking of end-to-end congestion control here? Then if we =
have
>>> congestion feedback from different nodes on one path, this feedback
>>> signal will have the same delay. And usually there is only one =
bottleneck
>>> at one point of time as there is always on link with the smallest
>>> available bandwidth. Is this the scenario your are talking about or
>>> something else?
>>>=20
>>> Mirja
>>>=20
>>>> The overall system will have a delay - but that delay won't be =
visible
>>>> to the congestion control algorithm on the first hop, so the =
reaction
>>>> time and size can't be scaled appropriately.
>>>>=20
>>>> The result is that the overall system is very likely to exhibit
>>>> oscillatory behaviour - simply because the time constants we need =
to
>>>> observe for a proper design are invisible to the observer.
>>>>=20
>>>> We can imagine ways of working around this limitation - but for the
>>>> purposes of RMCAT, I think we should just stick with two =
principles:
>>>>=20
>>>> - We don't design for the two-hop case.
>>>> - We advise designers of two-hop systems (like relay nodes) that if =
they
>>>> want to feed capacity information from a second hop back to the =
first
>>>> hop, they should reflect those changes SLOWLY.
>>>>=20
>>>> Experts, did this sound reasonable?
>>>>=20
>>>>                     Harald
>>>=20
>>> --
>>> -------------------------------------------------------------------
>>> Dipl.-Ing. Mirja K=FChlewind
>>> Institute of Communication Networks and Computer Engineering (IKR)
>>> University of Stuttgart, Germany
>>> Pfaffenwaldring 47, D-70569 Stuttgart
>>>=20
>>> tel: +49(0)711/685-67973
>>> email: mirja.kuehlewind@ikr.uni-stuttgart.de
>>> web: www.ikr.uni-stuttgart.de
>>> -------------------------------------------------------------------
>=20
>=20
>=20
> --=20
> -------------------------------------------------------------------
> Dipl.-Ing. Mirja K=FChlewind
> Institute of Communication Networks and Computer Engineering (IKR)
> University of Stuttgart, Germany
> Pfaffenwaldring 47, D-70569 Stuttgart
>=20
> tel: +49(0)711/685-67973
> email: mirja.kuehlewind@ikr.uni-stuttgart.de
> web: www.ikr.uni-stuttgart.de
> -------------------------------------------------------------------


From lars@netapp.com  Wed Oct 24 06:27:30 2012
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From: "Eggert, Lars" <lars@netapp.com>
To: rmcat WG <rmcat@ietf.org>
Thread-Topic: preliminary agenda online
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Date: Wed, 24 Oct 2012 13:26:35 +0000
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Hi,

a preliminary agenda for our Atlanta meeting is online at =
http://www.ietf.org/proceedings/85/agenda/agenda-85-rmcat

Send comments.

Lars=

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From: Michael Welzl <michawe@ifi.uio.no>
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References: <201210021809.19685.mkuehle@ikr.uni-stuttgart.de> <D4D47BCFFE5A004F95D707546AC0D7E9068D4895@SACEXCMBX01-PRD.hq.netapp.com> <D4D47BCFFE5A004F95D707546AC0D7E9068DEB13@SACEXCMBX01-PRD.hq.netapp.com> <CAFT3WbWX3PPXWsry5LTnDT7uWOjEhUoTDV2q0+NH+oKgUzSdKg@mail.gmail.com> <D4D47BCFFE5A004F95D707546AC0D7E918563B83@SACEXCMBX01-PRD.hq.netapp.com> <E7175A8E3DC14048A7D3020E0338C1FA0B3EE9@xmb-aln-x13.cisco.com> <C12C3E50-3435-4E49-A6DA-E5CD4A401A3F@ifi.uio.no> <E7175A8E3DC14048A7D3020E0338C1FA0B4C5E@xmb-aln-x13.cisco.com> <7829361A-29CE-429B-A40B-BB63B76C56C9@ifi.uio.no> <E7175A8E3DC14048A7D3020E0338C1FA0B5CD5@xmb-aln-x13.cisco.com>
To: Xiaoqing Zhu (xiaoqzhu) <xiaoqzhu@cisco.com>
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Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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Actually...

It just occurred to me that the dynamic scenario (going from =
ECN-supported to non-ECN-supported) is also quite realistic.
Consider a situation where you traverse two routers: router 1, followed =
by router 2. Router 1 isn't usually your bottleneck, but it supports =
ECN. Router 2 doesn't support ECN, but is your bottleneck most of the =
time.

Now, if a sudden traffic burst causes the queue of router 1 to grow, you =
will start to believe that there is ECN support, and stop using delay. =
But then, this queue might never grow again, all the congestion could =
happen in router 2...

My point is, *relying* on ECN to be there no matter how much feedback =
you get is probably always the wrong thing to do. We can always only =
take it as a nice extra if its available, but I think we should never =
"switch to ECN mode".

Cheers,
Michael


On 18. okt. 2012, at 05:49, Xiaoqing Zhu (xiaoqzhu) wrote:

> Your points are well taken, Michael. =20
>=20
>=20
> Agree that ultimately, it would be nice to have the scheme consider a =
merge of delay and ECN signals, instead of switching naively between the =
two. We considered that as a stretch goal though, according to the =
priorities discussed in the last BOF. We are still in the process of =
improving the algorithm in that aspect.=20
>=20
> I will put in some results from ECN-based marking in the compiled file =
to share. But admitted, we have not yet explored the impact of all =
variations of ECN configurations.  I will need to run more tests for =
that.=20
>=20
> -Xiaoqing
>=20
> On Oct 17, 2012, at 12:31 AM, Michael Welzl wrote:
>=20
>>=20
>> On 17. okt. 2012, at 06:58, Xiaoqing Zhu (xiaoqzhu) wrote:
>>=20
>>> Hi Michael,=20
>>>=20
>>> Thanks for your comments.  Please find my replies inline below.=20
>>>=20
>>> Best,
>>> Xiaoqing
>>>=20
>>>=20
>>> On Oct 16, 2012, at 1:11 AM, Michael Welzl wrote:
>>>=20
>>>> Hi,
>>>>=20
>>>> I took a quick look at this:
>>>>=20
>>>> I like the general idea of using as much as possible of existing =
network support - but I'm critical about the actual control you propose.
>>>>=20
>>>> In section 5.3.2, you suggest to replace eqn. 1 from section 5.3 =
with eqn. 2 when ECN feedback is available. This means to ignore a delay =
signal from now on, and interpret ECN marks in the exact same way as you =
interpreted delay before - this is too simplistic, I think. After all, =
ECN marks require a queue to grow, and this group intends to minimize =
delay... why would you stop using delay once you have ECN in addition?
>>>=20
>>>=20
>>> In our design, we considered the two cases, with or without ECN =
markings, separately.  Therefore, the dynamic scenario when ECN becomes =
available in the middle of a video conferencing session has not yet been =
fully studied.  Just wondering, are cases where ECN marking are turned =
on and off dynamically common in reality? =20
>>=20
>> First, anything can happen when you have a routing change.
>> Second, I'm really sorry, but this wasn't what I meant. I shouldn't =
have written "when" but "if". I wasn't referring to a case where we go =
from no-ECN-support to ECN-support. So this is about the =
ECN-always-supported case.
>>=20
>>=20
>>>> In practice, using your scheme would probably mean that, as soon as =
I enable RED with ECN on the bottleneck between the two end systems, the =
users' observed delay grows. Not exactly an incentive?
>>>>=20
>>> What you mentioned (uers observing higher delay in ECN-enabled =
systems) will not happen.=20
>>>=20
>>> But rather, our simulation results have indicated that ECN with =
RED-based marking yields fairly similar results as the default =
delay-based scheme. They certainly do not incur higher delay, since the =
endpoint will automatically back off in face of higher ECN markings, =
which are indications of higher delay. The two systems (delay-based and =
ECN-based) can be tuned to stabilize that the same queuing delay at the =
bottleneck.=20
>>=20
>> Whether you see significant delay before you see a large enough =
amount of ECN markings depends on the queue length and RED parameters, =
none of which are under the end system's control. ECN is also a =
probabilistic signal, so sometimes you may get delay growth but just be =
lucky (or unlucky) enough to not get an ECN mark.
>>=20
>> So, I'd have to see results that involve various RED settings and =
queue lengths before I believe that.
>>=20
>> Cheers,
>> Michael
>>=20
>=20


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Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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Hi,


On 25. okt. 2012, at 02:20, Xiaoqing Zhu wrote:

> Michael,=20
>=20
> In general, I agree with you that we need a more graceful scheme to =
accommodate both implicit and explicit congestion signals. Our current =
-00 draft obviously does not solve that yet.  I'll add this important =
point to the "open issue" section of the draft, and also strive to find =
a solution for that. =20

Sure, I think we had already agreed, I just wanted to share this =
scenario that came to my mind.


> However, I am not sure the dynamic scenario you describe will happen. =
If a packet passes along both ECN-supported and non-ECN-supported =
routers, wouldn't the ECN field bits in its header be marked as '00' =
(not ECN-capable), as specified in RFC 3168?  In that case, even if =
congestion rises on the ECN-enabled route, the sender will still be =
reacting to the rise in observed delay.=20

RFC 3168 is long and it's been a while since the last time I read it, =
but: I can't recall reading what you describe here.
My understanding of ECN (in a nutshell) is: ECN-capable end hosts sent =
01 or 10; ECN-incapable routers might not react to ECN at all, so anyway =
you can't trust them to do something based on RFC 3168 if they don't =
implement RFC 3168  :-)    ; if an ECN-capable router sees a 01 or 10 =
packet, it may mark it to 11; if an ECN-capable router sees a 00 packet, =
it drops it instead of marking it.

So, no way to reliably determine if there are some routers that do *not* =
support ECN.


> Also, my understanding from the previous email threads on this mailing =
list seems to say that multi-hop congested links are hard scenarios to =
tackle anyway.  Do you think we should focus on that now, or defer it =
after we obtain some satisfactory results from the simpler case of =
single-congested link?=20

As I tried to explain in my earlier email about that:
http://www.ietf.org/mail-archive/web/rmcat/current/msg00069.html
this was about multiple hops in the overlay - multiple RTP end systems =
attached to each other, to form a longer path, if you will. The underlay =
can always have more than just one hop.

Cheers,
Michael


>=20
> Thanks,
> Xiaoqing
>=20
>=20
> On Wed, Oct 24, 2012 at 7:22 AM, Michael Welzl <michawe@ifi.uio.no> =
wrote:
> Actually...
>=20
> It just occurred to me that the dynamic scenario (going from =
ECN-supported to non-ECN-supported) is also quite realistic.
> Consider a situation where you traverse two routers: router 1, =
followed by router 2. Router 1 isn't usually your bottleneck, but it =
supports ECN. Router 2 doesn't support ECN, but is your bottleneck most =
of the time.
>=20
> Now, if a sudden traffic burst causes the queue of router 1 to grow, =
you will start to believe that there is ECN support, and stop using =
delay. But then, this queue might never grow again, all the congestion =
could happen in router 2...
>=20
> My point is, *relying* on ECN to be there no matter how much feedback =
you get is probably always the wrong thing to do. We can always only =
take it as a nice extra if its available, but I think we should never =
"switch to ECN mode".
>=20
> Cheers,
> Michael
>=20
>=20
> On 18. okt. 2012, at 05:49, Xiaoqing Zhu (xiaoqzhu) wrote:
>=20
> > Your points are well taken, Michael.
> >
> >
> > Agree that ultimately, it would be nice to have the scheme consider =
a merge of delay and ECN signals, instead of switching naively between =
the two. We considered that as a stretch goal though, according to the =
priorities discussed in the last BOF. We are still in the process of =
improving the algorithm in that aspect.
> >
> > I will put in some results from ECN-based marking in the compiled =
file to share. But admitted, we have not yet explored the impact of all =
variations of ECN configurations.  I will need to run more tests for =
that.
> >
> > -Xiaoqing
> >
> > On Oct 17, 2012, at 12:31 AM, Michael Welzl wrote:
> >
> >>
> >> On 17. okt. 2012, at 06:58, Xiaoqing Zhu (xiaoqzhu) wrote:
> >>
> >>> Hi Michael,
> >>>
> >>> Thanks for your comments.  Please find my replies inline below.
> >>>
> >>> Best,
> >>> Xiaoqing
> >>>
> >>>
> >>> On Oct 16, 2012, at 1:11 AM, Michael Welzl wrote:
> >>>
> >>>> Hi,
> >>>>
> >>>> I took a quick look at this:
> >>>>
> >>>> I like the general idea of using as much as possible of existing =
network support - but I'm critical about the actual control you propose.
> >>>>
> >>>> In section 5.3.2, you suggest to replace eqn. 1 from section 5.3 =
with eqn. 2 when ECN feedback is available. This means to ignore a delay =
signal from now on, and interpret ECN marks in the exact same way as you =
interpreted delay before - this is too simplistic, I think. After all, =
ECN marks require a queue to grow, and this group intends to minimize =
delay... why would you stop using delay once you have ECN in addition?
> >>>
> >>>
> >>> In our design, we considered the two cases, with or without ECN =
markings, separately.  Therefore, the dynamic scenario when ECN becomes =
available in the middle of a video conferencing session has not yet been =
fully studied.  Just wondering, are cases where ECN marking are turned =
on and off dynamically common in reality?
> >>
> >> First, anything can happen when you have a routing change.
> >> Second, I'm really sorry, but this wasn't what I meant. I shouldn't =
have written "when" but "if". I wasn't referring to a case where we go =
from no-ECN-support to ECN-support. So this is about the =
ECN-always-supported case.
> >>
> >>
> >>>> In practice, using your scheme would probably mean that, as soon =
as I enable RED with ECN on the bottleneck between the two end systems, =
the users' observed delay grows. Not exactly an incentive?
> >>>>
> >>> What you mentioned (uers observing higher delay in ECN-enabled =
systems) will not happen.
> >>>
> >>> But rather, our simulation results have indicated that ECN with =
RED-based marking yields fairly similar results as the default =
delay-based scheme. They certainly do not incur higher delay, since the =
endpoint will automatically back off in face of higher ECN markings, =
which are indications of higher delay. The two systems (delay-based and =
ECN-based) can be tuned to stabilize that the same queuing delay at the =
bottleneck.
> >>
> >> Whether you see significant delay before you see a large enough =
amount of ECN markings depends on the queue length and RED parameters, =
none of which are under the end system's control. ECN is also a =
probabilistic signal, so sometimes you may get delay growth but just be =
lucky (or unlucky) enough to not get an ECN mark.
> >>
> >> So, I'd have to see results that involve various RED settings and =
queue lengths before I believe that.
> >>
> >> Cheers,
> >> Michael
> >>
> >
>=20
>=20


--Apple-Mail=_BDEC0611-DBCE-43C5-933E-B84EBE11DC78
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=iso-8859-1

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Hi,<div><br></div><div><br><div><div>On 25. okt. 2012, at 02:20, =
Xiaoqing Zhu wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div>Michael,&nbsp;</div><div><br></div><div>In general, I =
agree with you that we need a more graceful scheme to accommodate both =
implicit and explicit congestion signals. Our current -00 draft =
obviously does not solve that yet. &nbsp;I'll add this important point =
to the "open issue" section of the draft, and also strive to find a =
solution for that. &nbsp;</div></blockquote><div><br></div><div>Sure, I =
think we had already agreed, I just wanted to share this scenario that =
came to my mind.</div><div><br></div><br><blockquote type=3D"cite">
<div>However, I am not sure the dynamic scenario you describe will =
happen.&nbsp;If a packet passes along both ECN-supported and =
non-ECN-supported routers, wouldn't the ECN field bits in its header be =
marked as '00' (not ECN-capable), as specified in RFC 3168? &nbsp;In =
that case, even if congestion rises on the ECN-enabled route, the sender =
will still be reacting to the rise in observed =
delay.&nbsp;</div></blockquote><div><br></div><div>RFC 3168 is long and =
it's been a while since the last time I read it, but: I can't recall =
reading what you describe here.</div>My understanding of ECN (in a =
nutshell) is: ECN-capable end hosts sent 01 or 10; ECN-incapable routers =
might not react to ECN at all, so anyway you can't trust them to do =
something based on RFC 3168 if they don't implement RFC 3168 &nbsp;:-) =
&nbsp; &nbsp;; if an ECN-capable router sees a 01 or 10 packet, it may =
mark it to 11; if an ECN-capable router sees a 00 packet, it drops it =
instead of marking it.</div><div><br></div><div>So, no way to reliably =
determine if there are some routers that do *not* support =
ECN.</div><div><br></div><div><br><blockquote type=3D"cite">
<div>Also, my understanding from the previous email threads on this =
mailing list seems to say that multi-hop congested links are hard =
scenarios to tackle anyway. &nbsp;Do you think we should focus on that =
now, or defer it after we obtain some satisfactory results from the =
simpler case of single-congested =
link?&nbsp;</div></blockquote><div><br></div><div>As I tried to explain =
in my earlier email about that:</div><div><a =
href=3D"http://www.ietf.org/mail-archive/web/rmcat/current/msg00069.html">=
http://www.ietf.org/mail-archive/web/rmcat/current/msg00069.html</a></div>=
<div>this was about multiple hops in the overlay - multiple RTP end =
systems attached to each other, to form a longer path, if you will. The =
underlay can always have more than just one =
hop.</div><div><br></div>Cheers,</div><div>Michael</div><div><br></div><di=
v><br><blockquote type=3D"cite">
=
<div><br></div><div>Thanks,</div><div>Xiaoqing</div><div><div><div><br></d=
iv><div><br><div class=3D"gmail_quote">On Wed, Oct 24, 2012 at 7:22 AM, =
Michael Welzl <span dir=3D"ltr">&lt;<a href=3D"mailto:michawe@ifi.uio.no" =
target=3D"_blank">michawe@ifi.uio.no</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">Actually...<br>
<br>
It just occurred to me that the dynamic scenario (going from =
ECN-supported to non-ECN-supported) is also quite realistic.<br>
Consider a situation where you traverse two routers: router 1, followed =
by router 2. Router 1 isn't usually your bottleneck, but it supports =
ECN. Router 2 doesn't support ECN, but is your bottleneck most of the =
time.<br>

<br>
Now, if a sudden traffic burst causes the queue of router 1 to grow, you =
will start to believe that there is ECN support, and stop using delay. =
But then, this queue might never grow again, all the congestion could =
happen in router 2...<br>

<br>
My point is, *relying* on ECN to be there no matter how much feedback =
you get is probably always the wrong thing to do. We can always only =
take it as a nice extra if its available, but I think we should never =
"switch to ECN mode".<br>

<br>
Cheers,<br>
Michael<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
On 18. okt. 2012, at 05:49, Xiaoqing Zhu (xiaoqzhu) wrote:<br>
<br>
&gt; Your points are well taken, Michael.<br>
&gt;<br>
&gt;<br>
&gt; Agree that ultimately, it would be nice to have the scheme consider =
a merge of delay and ECN signals, instead of switching naively between =
the two. We considered that as a stretch goal though, according to the =
priorities discussed in the last BOF. We are still in the process of =
improving the algorithm in that aspect.<br>

&gt;<br>
&gt; I will put in some results from ECN-based marking in the compiled =
file to share. But admitted, we have not yet explored the impact of all =
variations of ECN configurations. &nbsp;I will need to run more tests =
for that.<br>

&gt;<br>
&gt; -Xiaoqing<br>
&gt;<br>
&gt; On Oct 17, 2012, at 12:31 AM, Michael Welzl wrote:<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; On 17. okt. 2012, at 06:58, Xiaoqing Zhu (xiaoqzhu) wrote:<br>
&gt;&gt;<br>
&gt;&gt;&gt; Hi Michael,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Thanks for your comments. &nbsp;Please find my replies =
inline below.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Best,<br>
&gt;&gt;&gt; Xiaoqing<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On Oct 16, 2012, at 1:11 AM, Michael Welzl wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Hi,<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I took a quick look at this:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I like the general idea of using as much as possible of =
existing network support - but I'm critical about the actual control you =
propose.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; In section 5.3.2, you suggest to replace eqn. 1 from =
section 5.3 with eqn. 2 when ECN feedback is available. This means to =
ignore a delay signal from now on, and interpret ECN marks in the exact =
same way as you interpreted delay before - this is too simplistic, I =
think. After all, ECN marks require a queue to grow, and this group =
intends to minimize delay... why would you stop using delay once you =
have ECN in addition?<br>

&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; In our design, we considered the two cases, with or without =
ECN markings, separately. &nbsp;Therefore, the dynamic scenario when ECN =
becomes available in the middle of a video conferencing session has not =
yet been fully studied. &nbsp;Just wondering, are cases where ECN =
marking are turned on and off dynamically common in reality?<br>

&gt;&gt;<br>
&gt;&gt; First, anything can happen when you have a routing change.<br>
&gt;&gt; Second, I'm really sorry, but this wasn't what I meant. I =
shouldn't have written "when" but "if". I wasn't referring to a case =
where we go from no-ECN-support to ECN-support. So this is about the =
ECN-always-supported case.<br>

&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;&gt;&gt; In practice, using your scheme would probably mean =
that, as soon as I enable RED with ECN on the bottleneck between the two =
end systems, the users' observed delay grows. Not exactly an =
incentive?<br>

&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt; What you mentioned (uers observing higher delay in =
ECN-enabled systems) will not happen.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; But rather, our simulation results have indicated that ECN =
with RED-based marking yields fairly similar results as the default =
delay-based scheme. They certainly do not incur higher delay, since the =
endpoint will automatically back off in face of higher ECN markings, =
which are indications of higher delay. The two systems (delay-based and =
ECN-based) can be tuned to stabilize that the same queuing delay at the =
bottleneck.<br>

&gt;&gt;<br>
&gt;&gt; Whether you see significant delay before you see a large enough =
amount of ECN markings depends on the queue length and RED parameters, =
none of which are under the end system's control. ECN is also a =
probabilistic signal, so sometimes you may get delay growth but just be =
lucky (or unlucky) enough to not get an ECN mark.<br>

&gt;&gt;<br>
&gt;&gt; So, I'd have to see results that involve various RED settings =
and queue lengths before I believe that.<br>
&gt;&gt;<br>
&gt;&gt; Cheers,<br>
&gt;&gt; Michael<br>
&gt;&gt;<br>
&gt;<br>
<br>
</div></div></blockquote></div><br></div></div></div>
</blockquote></div><br></div></body></html>=

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--14dae934061fd1000d04cd332add
Content-Type: text/plain; charset=ISO-8859-1

Hi Xiaoqing,

Interesting draft, I have a few questions related to the queuing delay part
of it.

"The role of the receiver is fairly straightforward. It observes and
estimates end-to-end queuing delay d and ECN marking ratio p of the stream.
The former can be obtained from the RTP timestamp provided by the sender."

You don't provide much detail on how the end-to-end queuing delay is
estimated from the timestamps. Would it be possible to be more specific?
Are you employing a filter? Related to that, you seem to be running into a
divide by zero in equation 1 if your queuing delay turns out to be zero. It
also seems a bit bold to go directly from R_min to R_max if the queuing
delay is low enough.

/Stefan




On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (xiaoqzhu) <
xiaoqzhu@cisco.com> wrote:

>  And below is the notification of our submission:
>
>  Thanks,
> Xiaoqing
>
>  --------------------------------------
> A new version of I-D, draft-zhu-rmcat-nada-00.txt
> has been successfully submitted by Xiaoqing Zhu and posted to the
> IETF repository.
>
> Filename:  draft-zhu-rmcat-nada
> Revision:  00
> Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
> Creation date:  2012-10-14
> WG ID:  Individual Submission
> Number of pages: 10
> URL:
> http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
> Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00
>
>
> Abstract:
>   This document describes a scheme named network-assisted dynamic
>   adaptation (NADA), a novel congestion control approach for
>   interactive real-time media applications, such as video conferencing.
>   In the proposed scheme, the sender regulates its sending rate based
>   on either implicit or explicit congestion signaling, in a unified
>   approach. The scheme can reap the benefits of explicit congestion
>   notification markings from network nodes. It also maintains
>   consistent sender behavior in the absence of such markings, by
>   reacting to queuing delays instead.
>
>   We present here the overall system architecture, recommended
>   behaviors at the sender and the receiver, as well as expected network
>   nodes operations. Results from extensive simulation studies of the
>   proposed scheme are available upon request.
>
>
>
>
> The IETF Secretariat
> --------------------------------------
>  On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:
>
>  Hi,
>
> On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu> wrote:
>
> My colleague and I are working on a congestion control scheme for
>
> real-time conferencing applications at Cisco. A brief version of that
>
> algorithm was presented at the IAB Workshop in Vancouver (paper #12,
>
> "Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
>
> to time constraints, however, we did not get a chance to mention the
>
> delay-based variant of that scheme, which may be a better fit for
>
> rmcat.
>
>
>  We would like to request for a slot to present in Atlanta. We will
>
> also submit the ID draft before Oct 15, for further discussions on
>
> this mailing list.
>
>
> I look forward to seeing the draft being discussed on the list!
>
> Also, I'm noting your request for a slot. Because the actual CC mechanisms
> are not our most pressing milestones initially - the requirements and eval
> criteria are - we may decide to focus our meeting time on those other
> drafts, if demand for agenda time is high.
>
> Lars
>
>
>

--14dae934061fd1000d04cd332add
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Xiaoqing,<div><br></div><div>Interesting draft, I have a few questions r=
elated to the queuing delay part of it.</div><div><br></div><div>&quot;The =
role of the receiver is fairly straightforward. It observes and estimates e=
nd-to-end queuing delay d and ECN marking ratio p of the stream. The former=
 can be obtained from the RTP timestamp provided by the sender.&quot;</div>
<div><br></div><div>You don&#39;t provide much detail on how the end-to-end=
 queuing delay is estimated from the timestamps. Would it be possible to be=
 more specific? Are you employing a filter? Related to that, you seem to be=
 running into a divide by zero in equation 1 if your queuing delay turns ou=
t to be zero. It also seems a bit bold to go directly from R_min to R_max i=
f the queuing delay is low enough.</div>
<div><br></div><div>/Stefan</div><div><br></div><div><br></div><div class=
=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Oct 15, 2012 at=
 11:43 PM, Xiaoqing Zhu (xiaoqzhu) <span dir=3D"ltr">&lt;<a href=3D"mailto:=
xiaoqzhu@cisco.com" target=3D"_blank">xiaoqzhu@cisco.com</a>&gt;</span> wro=
te:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">



<div style=3D"word-wrap:break-word">
And below is the notification of our submission:=A0
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span style=3D"white-space:pre-wrap"> </span>=A0draft-zhu-rmcat-na=
da<br>
Revision:<span style=3D"white-space:pre-wrap"> </span>=A000<br>
Title:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>=A0NADA: A Unified Congestion Control Scheme for Real-T=
ime Media<br>
Creation date:<span style=3D"white-space:pre-wrap"> </span>=A02012-10-14<br=
>
WG ID:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>=A0Individual Submission<br>
Number of pages: 10<br>
URL: =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0<a href=3D"http://www.ietf.org/int=
ernet-drafts/draft-zhu-rmcat-nada-00.txt" target=3D"_blank">http://www.ietf=
.org/internet-drafts/draft-zhu-rmcat-nada-00.txt</a><br>
Status: =A0=A0=A0=A0=A0=A0=A0=A0=A0<a href=3D"http://datatracker.ietf.org/d=
oc/draft-zhu-rmcat-nada" target=3D"_blank">http://datatracker.ietf.org/doc/=
draft-zhu-rmcat-nada</a><br>
Htmlized: =A0=A0=A0=A0=A0=A0=A0<a href=3D"http://tools.ietf.org/html/draft-=
zhu-rmcat-nada-00" target=3D"_blank">http://tools.ietf.org/html/draft-zhu-r=
mcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
=A0=A0This document describes a scheme named network-assisted dynamic<br>
=A0=A0adaptation (NADA), a novel congestion control approach for<br>
=A0=A0interactive real-time media applications, such as video conferencing.=
<br>
=A0=A0In the proposed scheme, the sender regulates its sending rate based<b=
r>
=A0=A0on either implicit or explicit congestion signaling, in a unified<br>
=A0=A0approach. The scheme can reap the benefits of explicit congestion<br>
=A0=A0notification markings from network nodes. It also maintains<br>
=A0=A0consistent sender behavior in the absence of such markings, by<br>
=A0=A0reacting to queuing delays instead.<br>
<br>
=A0=A0We present here the overall system architecture, recommended<br>
=A0=A0behaviors at the sender and the receiver, as well as expected network=
<br>
=A0=A0nodes operations. Results from extensive simulation studies of the<br=
>
=A0=A0proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div><div class=3D"im">
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br>
</div><div><div class=3D"h5"><blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a href=3D"mailto:zhuxq@alumni.=
stanford.edu" target=3D"_blank">zhuxq@alumni.stanford.edu</a>&gt; wrote:<br=
>
<blockquote type=3D"cite">My colleague and I are working on a congestion co=
ntrol scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. A b=
rief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in Va=
ncouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">&quot;Network-Assisted Dynamic Adaptation (NADA):=
 A Design Summary&quot;). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a ch=
ance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may be =
a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present in=
 Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for furth=
er discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I&#39;m noting your request for a slot. Because the actual CC mechani=
sms are not our most pressing milestones initially - the requirements and e=
val criteria are - we may decide to focus our meeting time on those other d=
rafts, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div></div></div>
<br>
</div>
</div>

</blockquote></div><br></div>

--14dae934061fd1000d04cd332add--

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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: Stefan Holmer <stefan@webrtc.org>
Thread-Topic: [rmcat] send agenda requests (was Re: First meeting)
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Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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Hi Stefan,

Thanks a lot for your comments. I'll add in more details on the description=
 for the receiver actions.

You are right, we did employ some filtering of the observed per-packet queu=
ing delay, to avoid erratic reactions in the sending rate. The queuing dela=
y employed in the rate calculation in Eq. (1) is actually the time-smoothed=
 averaging d_avg:

d_avg =3D (1-alpha)*d_avg + d_n, where d_n represents the measured queuing =
delay  (end-to-end delay minus minimum observed value) for the n-th packet.=
 The parameter alpha adjusts the level of smoothness in our average process=
. In our experiments, we are using alpha =3D 10e-4.

Also, it's obvious that Eq.(1) wouldn't hold for for d=3D0, in which case t=
he rate is chosen to be R_max.  As mentioned in the paragraph below Eq. (1)=
, any calculated target rate R_o greater than R_max will also be clipped to=
 the R_max.

Initially, I omitted these details in the draft proposal for brevity. Given=
 your comments and concerns, looks like I'd better put them back in for cla=
rity.

Best,
Xiaoqing


On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:

Hi Xiaoqing,

Interesting draft, I have a few questions related to the queuing delay part=
 of it.

"The role of the receiver is fairly straightforward. It observes and estima=
tes end-to-end queuing delay d and ECN marking ratio p of the stream. The f=
ormer can be obtained from the RTP timestamp provided by the sender."

You don't provide much detail on how the end-to-end queuing delay is estima=
ted from the timestamps. Would it be possible to be more specific? Are you =
employing a filter? Related to that, you seem to be running into a divide b=
y zero in equation 1 if your queuing delay turns out to be zero. It also se=
ems a bit bold to go directly from R_min to R_max if the queuing delay is l=
ow enough.

/Stefan




On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (xiaoqzhu) <xiaoqzhu@cisco.c=
om<mailto:xiaoqzhu@cisco.com>> wrote:
And below is the notification of our submission:

Thanks,
Xiaoqing

--------------------------------------
A new version of I-D, draft-zhu-rmcat-nada-00.txt
has been successfully submitted by Xiaoqing Zhu and posted to the
IETF repository.

Filename:  draft-zhu-rmcat-nada
Revision:  00
Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
Creation date:  2012-10-14
WG ID:  Individual Submission
Number of pages: 10
URL:             http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-0=
0.txt
Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00


Abstract:
  This document describes a scheme named network-assisted dynamic
  adaptation (NADA), a novel congestion control approach for
  interactive real-time media applications, such as video conferencing.
  In the proposed scheme, the sender regulates its sending rate based
  on either implicit or explicit congestion signaling, in a unified
  approach. The scheme can reap the benefits of explicit congestion
  notification markings from network nodes. It also maintains
  consistent sender behavior in the absence of such markings, by
  reacting to queuing delays instead.

  We present here the overall system architecture, recommended
  behaviors at the sender and the receiver, as well as expected network
  nodes operations. Results from extensive simulation studies of the
  proposed scheme are available upon request.




The IETF Secretariat
--------------------------------------
On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:

Hi,

On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu<mailto:z=
huxq@alumni.stanford.edu>> wrote:
My colleague and I are working on a congestion control scheme for
real-time conferencing applications at Cisco. A brief version of that
algorithm was presented at the IAB Workshop in Vancouver (paper #12,
"Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
to time constraints, however, we did not get a chance to mention the
delay-based variant of that scheme, which may be a better fit for
rmcat.

We would like to request for a slot to present in Atlanta. We will
also submit the ID draft before Oct 15, for further discussions on
this mailing list.

I look forward to seeing the draft being discussed on the list!

Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is high.

Lars




--_000_E7175A8E3DC14048A7D3020E0338C1FA0D7B82AExmbalnx13ciscoc_
Content-Type: text/html; charset="us-ascii"
Content-ID: <C62DCB7ADF8DC84EBFA54910D6628CA5@cisco.com>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
Hi Stefan,&nbsp;
<div><br>
</div>
<div>Thanks a lot for your comments. I'll add in more details on the descri=
ption for the receiver actions.&nbsp;</div>
<div><br>
</div>
<div>You are right, we did employ some filtering of the observed per-packet=
 queuing delay, to avoid erratic reactions in the sending rate. The queuing=
 delay employed in the rate calculation in Eq. (1) is actually the time-smo=
othed averaging d_avg:&nbsp;</div>
<div><br>
</div>
<div>d_avg =3D (1-alpha)*d_avg &#43; d_n, where d_n represents the measured=
 queuing delay &nbsp;(end-to-end delay minus minimum observed value) for th=
e n-th packet. The parameter alpha adjusts the level of smoothness in our a=
verage process. In our experiments, we are
 using alpha =3D 10e-4.&nbsp;</div>
<div><br>
</div>
<div>Also, it's obvious that Eq.(1) wouldn't hold for for d=3D0, in which c=
ase the rate is chosen to be R_max. &nbsp;As mentioned in the paragraph bel=
ow Eq. (1), any calculated target rate R_o greater than R_max will also be =
clipped to the R_max.&nbsp;</div>
<div><br>
</div>
<div>Initially, I omitted these details in the draft proposal for brevity. =
Given your comments and concerns, looks like I'd better put them back in fo=
r clarity.&nbsp;</div>
<div><br>
</div>
<div>Best,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div><br>
</div>
<div>
<div>
<div>On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">Hi Xiaoqing,
<div><br>
</div>
<div>Interesting draft, I have a few questions related to the queuing delay=
 part of it.</div>
<div><br>
</div>
<div>&quot;The role of the receiver is fairly straightforward. It observes =
and estimates end-to-end queuing delay d and ECN marking ratio p of the str=
eam. The former can be obtained from the RTP timestamp provided by the send=
er.&quot;</div>
<div><br>
</div>
<div>You don't provide much detail on how the end-to-end queuing delay is e=
stimated from the timestamps. Would it be possible to be more specific? Are=
 you employing a filter? Related to that, you seem to be running into a div=
ide by zero in equation 1 if your
 queuing delay turns out to be zero. It also seems a bit bold to go directl=
y from R_min to R_max if the queuing delay is low enough.</div>
<div><br>
</div>
<div>/Stefan</div>
<div><br>
</div>
<div><br>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (=
xiaoqzhu)
<span dir=3D"ltr">&lt;<a href=3D"mailto:xiaoqzhu@cisco.com" target=3D"_blan=
k">xiaoqzhu@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">And below is the notification of our su=
bmission:&nbsp;
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span style=3D"white-space:pre-wrap"> </span>&nbsp;draft-zhu-rmcat=
-nada<br>
Revision:<span style=3D"white-space:pre-wrap"> </span>&nbsp;00<br>
Title:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>&nbsp;NADA: A Unified Congestion Control Scheme for Rea=
l-Time Media<br>
Creation date:<span style=3D"white-space:pre-wrap"> </span>&nbsp;2012-10-14=
<br>
WG ID:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>&nbsp;Individual Submission<br>
Number of pages: 10<br>
URL: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<a href=3D"http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.tx=
t" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-zhu-rmcat-na=
da-00.txt</a><br>
Status: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"ht=
tp://datatracker.ietf.org/doc/draft-zhu-rmcat-nada" target=3D"_blank">http:=
//datatracker.ietf.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools=
.ietf.org/html/draft-zhu-rmcat-nada-00" target=3D"_blank">http://tools.ietf=
.org/html/draft-zhu-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp;&nbsp;This document describes a scheme named network-assisted dynamic=
<br>
&nbsp;&nbsp;adaptation (NADA), a novel congestion control approach for<br>
&nbsp;&nbsp;interactive real-time media applications, such as video confere=
ncing.<br>
&nbsp;&nbsp;In the proposed scheme, the sender regulates its sending rate b=
ased<br>
&nbsp;&nbsp;on either implicit or explicit congestion signaling, in a unifi=
ed<br>
&nbsp;&nbsp;approach. The scheme can reap the benefits of explicit congesti=
on<br>
&nbsp;&nbsp;notification markings from network nodes. It also maintains<br>
&nbsp;&nbsp;consistent sender behavior in the absence of such markings, by<=
br>
&nbsp;&nbsp;reacting to queuing delays instead.<br>
<br>
&nbsp;&nbsp;We present here the overall system architecture, recommended<br=
>
&nbsp;&nbsp;behaviors at the sender and the receiver, as well as expected n=
etwork<br>
&nbsp;&nbsp;nodes operations. Results from extensive simulation studies of =
the<br>
&nbsp;&nbsp;proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div class=3D"im">
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br>
</div>
<div>
<div class=3D"h5">
<blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a href=3D"mailto:zhuxq@alumni.=
stanford.edu" target=3D"_blank">zhuxq@alumni.stanford.edu</a>&gt; wrote:<br=
>
<blockquote type=3D"cite">My colleague and I are working on a congestion co=
ntrol scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. A b=
rief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in Va=
ncouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">&quot;Network-Assisted Dynamic Adaptation (NADA):=
 A Design Summary&quot;). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a ch=
ance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may be =
a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present in=
 Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for furth=
er discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
</div>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</body>
</html>

--_000_E7175A8E3DC14048A7D3020E0338C1FA0D7B82AExmbalnx13ciscoc_--

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From: Stefan Holmer <stefan@webrtc.org>
To: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
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Cc: rmcat WG <rmcat@ietf.org>, Xiaoqing Zhu <zhuxq@alumni.stanford.edu>, "Eggert, Lars" <lars@netapp.com>, "Rong Pan \(ropan\)" <ropan@cisco.com>
Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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On Mon, Oct 29, 2012 at 3:53 PM, Xiaoqing Zhu (xiaoqzhu) <xiaoqzhu@cisco.com
> wrote:

>  Hi Stefan,
>
>  Thanks a lot for your comments. I'll add in more details on the
> description for the receiver actions.
>
>  You are right, we did employ some filtering of the observed per-packet
> queuing delay, to avoid erratic reactions in the sending rate. The queuing
> delay employed in the rate calculation in Eq. (1) is actually the
> time-smoothed averaging d_avg:
>
>  d_avg = (1-alpha)*d_avg + d_n, where d_n represents the measured queuing
> delay  (end-to-end delay minus minimum observed value) for the n-th packet.
> The parameter alpha adjusts the level of smoothness in our average process.
> In our experiments, we are using alpha = 10e-4.
>

Thanks. Just to be clear,

d_n = receiver_time_n - timestamp_n - min_observed

and min_observed is the smallest d_n seen in some window?


>
>  Also, it's obvious that Eq.(1) wouldn't hold for for d=0, in which case
> the rate is chosen to be R_max.  As mentioned in the paragraph below Eq.
> (1), any calculated target rate R_o greater than R_max will also be clipped
> to the R_max.
>
>  Initially, I omitted these details in the draft proposal for brevity.
> Given your comments and concerns, looks like I'd better put them back in
> for clarity.
>
>  Best,
> Xiaoqing
>
>
>   On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:
>
> Hi Xiaoqing,
>
>  Interesting draft, I have a few questions related to the queuing delay
> part of it.
>
>  "The role of the receiver is fairly straightforward. It observes and
> estimates end-to-end queuing delay d and ECN marking ratio p of the stream.
> The former can be obtained from the RTP timestamp provided by the sender."
>
>  You don't provide much detail on how the end-to-end queuing delay is
> estimated from the timestamps. Would it be possible to be more specific?
> Are you employing a filter? Related to that, you seem to be running into a
> divide by zero in equation 1 if your queuing delay turns out to be zero. It
> also seems a bit bold to go directly from R_min to R_max if the queuing
> delay is low enough.
>
>  /Stefan
>
>
>
>
> On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (xiaoqzhu) <
> xiaoqzhu@cisco.com> wrote:
>
>> And below is the notification of our submission:
>>
>>  Thanks,
>> Xiaoqing
>>
>>  --------------------------------------
>> A new version of I-D, draft-zhu-rmcat-nada-00.txt
>> has been successfully submitted by Xiaoqing Zhu and posted to the
>> IETF repository.
>>
>> Filename:  draft-zhu-rmcat-nada
>> Revision:  00
>> Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
>> Creation date:  2012-10-14
>> WG ID:  Individual Submission
>> Number of pages: 10
>> URL:
>> http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt
>> Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
>> Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00
>>
>>
>> Abstract:
>>   This document describes a scheme named network-assisted dynamic
>>   adaptation (NADA), a novel congestion control approach for
>>   interactive real-time media applications, such as video conferencing.
>>   In the proposed scheme, the sender regulates its sending rate based
>>   on either implicit or explicit congestion signaling, in a unified
>>   approach. The scheme can reap the benefits of explicit congestion
>>   notification markings from network nodes. It also maintains
>>   consistent sender behavior in the absence of such markings, by
>>   reacting to queuing delays instead.
>>
>>   We present here the overall system architecture, recommended
>>   behaviors at the sender and the receiver, as well as expected network
>>   nodes operations. Results from extensive simulation studies of the
>>   proposed scheme are available upon request.
>>
>>
>>
>>
>> The IETF Secretariat
>> --------------------------------------
>>  On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:
>>
>>   Hi,
>>
>> On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu>
>> wrote:
>>
>> My colleague and I are working on a congestion control scheme for
>>
>> real-time conferencing applications at Cisco. A brief version of that
>>
>> algorithm was presented at the IAB Workshop in Vancouver (paper #12,
>>
>> "Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
>>
>> to time constraints, however, we did not get a chance to mention the
>>
>> delay-based variant of that scheme, which may be a better fit for
>>
>> rmcat.
>>
>>
>>  We would like to request for a slot to present in Atlanta. We will
>>
>> also submit the ID draft before Oct 15, for further discussions on
>>
>> this mailing list.
>>
>>
>> I look forward to seeing the draft being discussed on the list!
>>
>> Also, I'm noting your request for a slot. Because the actual CC
>> mechanisms are not our most pressing milestones initially - the
>> requirements and eval criteria are - we may decide to focus our meeting
>> time on those other drafts, if demand for agenda time is high.
>>
>> Lars
>>
>>
>>
>
>

--e89a8ff253d2a8769404cd3440b3
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, O=
ct 29, 2012 at 3:53 PM, Xiaoqing Zhu (xiaoqzhu) <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:xiaoqzhu@cisco.com" target=3D"_blank" class=3D"cremed">xiaoqzh=
u@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">



<div style=3D"word-wrap:break-word">
Hi Stefan,=A0
<div><br>
</div>
<div>Thanks a lot for your comments. I&#39;ll add in more details on the de=
scription for the receiver actions.=A0</div>
<div><br>
</div>
<div>You are right, we did employ some filtering of the observed per-packet=
 queuing delay, to avoid erratic reactions in the sending rate. The queuing=
 delay employed in the rate calculation in Eq. (1) is actually the time-smo=
othed averaging d_avg:=A0</div>

<div><br>
</div>
<div>d_avg =3D (1-alpha)*d_avg + d_n, where d_n represents the measured que=
uing delay =A0(end-to-end delay minus minimum observed value) for the n-th =
packet. The parameter alpha adjusts the level of smoothness in our average =
process. In our experiments, we are
 using alpha =3D 10e-4.=A0</div></div></blockquote><div><br></div><div>Than=
ks. Just to be clear,=A0</div><div><br></div><div>d_n =3D receiver_time_n -=
 timestamp_n - min_observed</div><div><br></div><div>and min_observed is th=
e smallest d_n seen in some window?</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div style=3D"word-wrap:break-=
word">
<div><br>
</div>
<div>Also, it&#39;s obvious that Eq.(1) wouldn&#39;t hold for for d=3D0, in=
 which case the rate is chosen to be R_max. =A0As mentioned in the paragrap=
h below Eq. (1), any calculated target rate R_o greater than R_max will als=
o be clipped to the R_max.=A0</div>

<div><br>
</div>
<div>Initially, I omitted these details in the draft proposal for brevity. =
Given your comments and concerns, looks like I&#39;d better put them back i=
n for clarity.=A0</div>
<div><br>
</div>
<div>Best,</div>
<div>Xiaoqing</div><div><div class=3D"h5">
<div><br>
</div>
<div><br>
</div>
<div>
<div>
<div>On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:</div>
<br>
<blockquote type=3D"cite">Hi Xiaoqing,
<div><br>
</div>
<div>Interesting draft, I have a few questions related to the queuing delay=
 part of it.</div>
<div><br>
</div>
<div>&quot;The role of the receiver is fairly straightforward. It observes =
and estimates end-to-end queuing delay d and ECN marking ratio p of the str=
eam. The former can be obtained from the RTP timestamp provided by the send=
er.&quot;</div>

<div><br>
</div>
<div>You don&#39;t provide much detail on how the end-to-end queuing delay =
is estimated from the timestamps. Would it be possible to be more specific?=
 Are you employing a filter? Related to that, you seem to be running into a=
 divide by zero in equation 1 if your
 queuing delay turns out to be zero. It also seems a bit bold to go directl=
y from R_min to R_max if the queuing delay is low enough.</div>
<div><br>
</div>
<div>/Stefan</div>
<div><br>
</div>
<div><br>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (=
xiaoqzhu)
<span dir=3D"ltr">&lt;<a href=3D"mailto:xiaoqzhu@cisco.com" target=3D"_blan=
k" class=3D"cremed">xiaoqzhu@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">And below is the notification of our su=
bmission:=A0
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span style=3D"white-space:pre-wrap"> </span>=A0draft-zhu-rmcat-na=
da<br>
Revision:<span style=3D"white-space:pre-wrap"> </span>=A000<br>
Title:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>=A0NADA: A Unified Congestion Control Scheme for Real-T=
ime Media<br>
Creation date:<span style=3D"white-space:pre-wrap"> </span>=A02012-10-14<br=
>
WG ID:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>=A0Individual Submission<br>
Number of pages: 10<br>
URL: =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0<a href=3D"http://www.ietf.org/int=
ernet-drafts/draft-zhu-rmcat-nada-00.txt" target=3D"_blank" class=3D"cremed=
">http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt</a><br>
Status: =A0=A0=A0=A0=A0=A0=A0=A0=A0<a href=3D"http://datatracker.ietf.org/d=
oc/draft-zhu-rmcat-nada" target=3D"_blank" class=3D"cremed">http://datatrac=
ker.ietf.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: =A0=A0=A0=A0=A0=A0=A0<a href=3D"http://tools.ietf.org/html/draft-=
zhu-rmcat-nada-00" target=3D"_blank" class=3D"cremed">http://tools.ietf.org=
/html/draft-zhu-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
=A0=A0This document describes a scheme named network-assisted dynamic<br>
=A0=A0adaptation (NADA), a novel congestion control approach for<br>
=A0=A0interactive real-time media applications, such as video conferencing.=
<br>
=A0=A0In the proposed scheme, the sender regulates its sending rate based<b=
r>
=A0=A0on either implicit or explicit congestion signaling, in a unified<br>
=A0=A0approach. The scheme can reap the benefits of explicit congestion<br>
=A0=A0notification markings from network nodes. It also maintains<br>
=A0=A0consistent sender behavior in the absence of such markings, by<br>
=A0=A0reacting to queuing delays instead.<br>
<br>
=A0=A0We present here the overall system architecture, recommended<br>
=A0=A0behaviors at the sender and the receiver, as well as expected network=
<br>
=A0=A0nodes operations. Results from extensive simulation studies of the<br=
>
=A0=A0proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div>
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br>
</div>
<div>
<div>
<blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a href=3D"mailto:zhuxq@alumni.=
stanford.edu" target=3D"_blank" class=3D"cremed">zhuxq@alumni.stanford.edu<=
/a>&gt; wrote:<br>
<blockquote type=3D"cite">My colleague and I are working on a congestion co=
ntrol scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. A b=
rief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in Va=
ncouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">&quot;Network-Assisted Dynamic Adaptation (NADA):=
 A Design Summary&quot;). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a ch=
ance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may be =
a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present in=
 Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for furth=
er discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I&#39;m noting your request for a slot. Because the actual CC mechani=
sms are not our most pressing milestones initially - the requirements and e=
val criteria are - we may decide to focus our meeting time on those other d=
rafts, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
</div>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div></div></div>

</blockquote></div><br></div>

--e89a8ff253d2a8769404cd3440b3--

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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: Stefan Holmer <stefan@webrtc.org>
Thread-Topic: [rmcat] send agenda requests (was Re: First meeting)
Thread-Index: AQHNp9grf2quh6JMt0CYpuBNP/SHCpe1q3yAgAWVAgCAFYClAIAADUEAgAAIdICAAAFVgA==
Date: Mon, 29 Oct 2012 15:28:20 +0000
Message-ID: <E7175A8E3DC14048A7D3020E0338C1FA0D7B838F@xmb-aln-x13.cisco.com>
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Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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On Oct 29, 2012, at 10:23 AM, Stefan Holmer wrote:




On Mon, Oct 29, 2012 at 3:53 PM, Xiaoqing Zhu (xiaoqzhu) <xiaoqzhu@cisco.co=
m<mailto:xiaoqzhu@cisco.com>> wrote:
Hi Stefan,

Thanks a lot for your comments. I'll add in more details on the description=
 for the receiver actions.

You are right, we did employ some filtering of the observed per-packet queu=
ing delay, to avoid erratic reactions in the sending rate. The queuing dela=
y employed in the rate calculation in Eq. (1) is actually the time-smoothed=
 averaging d_avg:

d_avg =3D (1-alpha)*d_avg + d_n, where d_n represents the measured queuing =
delay  (end-to-end delay minus minimum observed value) for the n-th packet.=
 The parameter alpha adjusts the level of smoothness in our average process=
. In our experiments, we are using alpha =3D 10e-4.

Thanks. Just to be clear,

d_n =3D receiver_time_n - timestamp_n - min_observed

and min_observed is the smallest d_n seen in some window?

That's right.  BTW, while the time-averaged filtering works well for us, fo=
r now, we are also considering building in more robustness against individu=
al bad samples. The Kalman filtering approach presented in your proposal se=
ems to be a good candidate for that. Maybe there is a way to incorporate th=
at approach here for the delay estimation as well.  What do you think?




Also, it's obvious that Eq.(1) wouldn't hold for for d=3D0, in which case t=
he rate is chosen to be R_max.  As mentioned in the paragraph below Eq. (1)=
, any calculated target rate R_o greater than R_max will also be clipped to=
 the R_max.

Initially, I omitted these details in the draft proposal for brevity. Given=
 your comments and concerns, looks like I'd better put them back in for cla=
rity.

Best,
Xiaoqing


On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:

Hi Xiaoqing,

Interesting draft, I have a few questions related to the queuing delay part=
 of it.

"The role of the receiver is fairly straightforward. It observes and estima=
tes end-to-end queuing delay d and ECN marking ratio p of the stream. The f=
ormer can be obtained from the RTP timestamp provided by the sender."

You don't provide much detail on how the end-to-end queuing delay is estima=
ted from the timestamps. Would it be possible to be more specific? Are you =
employing a filter? Related to that, you seem to be running into a divide b=
y zero in equation 1 if your queuing delay turns out to be zero. It also se=
ems a bit bold to go directly from R_min to R_max if the queuing delay is l=
ow enough.

/Stefan




On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (xiaoqzhu) <xiaoqzhu@cisco.c=
om<mailto:xiaoqzhu@cisco.com>> wrote:
And below is the notification of our submission:

Thanks,
Xiaoqing

--------------------------------------
A new version of I-D, draft-zhu-rmcat-nada-00.txt
has been successfully submitted by Xiaoqing Zhu and posted to the
IETF repository.

Filename:  draft-zhu-rmcat-nada
Revision:  00
Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
Creation date:  2012-10-14
WG ID:  Individual Submission
Number of pages: 10
URL:             http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-0=
0.txt
Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00


Abstract:
  This document describes a scheme named network-assisted dynamic
  adaptation (NADA), a novel congestion control approach for
  interactive real-time media applications, such as video conferencing.
  In the proposed scheme, the sender regulates its sending rate based
  on either implicit or explicit congestion signaling, in a unified
  approach. The scheme can reap the benefits of explicit congestion
  notification markings from network nodes. It also maintains
  consistent sender behavior in the absence of such markings, by
  reacting to queuing delays instead.

  We present here the overall system architecture, recommended
  behaviors at the sender and the receiver, as well as expected network
  nodes operations. Results from extensive simulation studies of the
  proposed scheme are available upon request.




The IETF Secretariat
--------------------------------------
On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:

Hi,

On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu<mailto:z=
huxq@alumni.stanford.edu>> wrote:
My colleague and I are working on a congestion control scheme for
real-time conferencing applications at Cisco. A brief version of that
algorithm was presented at the IAB Workshop in Vancouver (paper #12,
"Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
to time constraints, however, we did not get a chance to mention the
delay-based variant of that scheme, which may be a better fit for
rmcat.

We would like to request for a slot to present in Atlanta. We will
also submit the ID draft before Oct 15, for further discussions on
this mailing list.

I look forward to seeing the draft being discussed on the list!

Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is high.

Lars






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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<br>
<div>
<div>On Oct 29, 2012, at 10:23 AM, Stefan Holmer wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite"><br>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Mon, Oct 29, 2012 at 3:53 PM, Xiaoqing Zhu (x=
iaoqzhu)
<span dir=3D"ltr">&lt;<a href=3D"mailto:xiaoqzhu@cisco.com" target=3D"_blan=
k" class=3D"cremed">xiaoqzhu@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">Hi Stefan,&nbsp;
<div><br>
</div>
<div>Thanks a lot for your comments. I'll add in more details on the descri=
ption for the receiver actions.&nbsp;</div>
<div><br>
</div>
<div>You are right, we did employ some filtering of the observed per-packet=
 queuing delay, to avoid erratic reactions in the sending rate. The queuing=
 delay employed in the rate calculation in Eq. (1) is actually the time-smo=
othed averaging d_avg:&nbsp;</div>
<div><br>
</div>
<div>d_avg =3D (1-alpha)*d_avg &#43; d_n, where d_n represents the measured=
 queuing delay &nbsp;(end-to-end delay minus minimum observed value) for th=
e n-th packet. The parameter alpha adjusts the level of smoothness in our a=
verage process. In our experiments, we are
 using alpha =3D 10e-4.&nbsp;</div>
</div>
</blockquote>
<div><br>
</div>
<div>Thanks. Just to be clear,&nbsp;</div>
<div><br>
</div>
<div>d_n =3D receiver_time_n - timestamp_n - min_observed</div>
<div><br>
</div>
<div>and min_observed is the smallest d_n seen in some window?</div>
</div>
</div>
</blockquote>
<div><br>
</div>
That's right. &nbsp;BTW, while the time-averaged filtering works well for u=
s, for now, we are also considering building in more robustness against ind=
ividual bad samples. The Kalman filtering approach presented in your propos=
al seems to be a good candidate for that.
 Maybe there is a way to incorporate that approach here for the delay estim=
ation as well. &nbsp;What do you think? &nbsp;&nbsp;</div>
<div><br>
</div>
<div><br>
<blockquote type=3D"cite">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div><br>
</div>
<div>Also, it's obvious that Eq.(1) wouldn't hold for for d=3D0, in which c=
ase the rate is chosen to be R_max. &nbsp;As mentioned in the paragraph bel=
ow Eq. (1), any calculated target rate R_o greater than R_max will also be =
clipped to the R_max.&nbsp;</div>
<div><br>
</div>
<div>Initially, I omitted these details in the draft proposal for brevity. =
Given your comments and concerns, looks like I'd better put them back in fo=
r clarity.&nbsp;</div>
<div><br>
</div>
<div>Best,</div>
<div>Xiaoqing</div>
<div>
<div class=3D"h5">
<div><br>
</div>
<div><br>
</div>
<div>
<div>
<div>On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:</div>
<br>
<blockquote type=3D"cite">Hi Xiaoqing,
<div><br>
</div>
<div>Interesting draft, I have a few questions related to the queuing delay=
 part of it.</div>
<div><br>
</div>
<div>&quot;The role of the receiver is fairly straightforward. It observes =
and estimates end-to-end queuing delay d and ECN marking ratio p of the str=
eam. The former can be obtained from the RTP timestamp provided by the send=
er.&quot;</div>
<div><br>
</div>
<div>You don't provide much detail on how the end-to-end queuing delay is e=
stimated from the timestamps. Would it be possible to be more specific? Are=
 you employing a filter? Related to that, you seem to be running into a div=
ide by zero in equation 1 if your
 queuing delay turns out to be zero. It also seems a bit bold to go directl=
y from R_min to R_max if the queuing delay is low enough.</div>
<div><br>
</div>
<div>/Stefan</div>
<div><br>
</div>
<div><br>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (=
xiaoqzhu)
<span dir=3D"ltr">&lt;<a href=3D"mailto:xiaoqzhu@cisco.com" target=3D"_blan=
k" class=3D"cremed">xiaoqzhu@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">And below is the notification of our su=
bmission:&nbsp;
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span style=3D"white-space:pre-wrap"> </span>&nbsp;draft-zhu-rmcat=
-nada<br>
Revision:<span style=3D"white-space:pre-wrap"> </span>&nbsp;00<br>
Title:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>&nbsp;NADA: A Unified Congestion Control Scheme for Rea=
l-Time Media<br>
Creation date:<span style=3D"white-space:pre-wrap"> </span>&nbsp;2012-10-14=
<br>
WG ID:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>&nbsp;Individual Submission<br>
Number of pages: 10<br>
URL: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<a href=3D"http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.tx=
t" target=3D"_blank" class=3D"cremed">http://www.ietf.org/internet-drafts/d=
raft-zhu-rmcat-nada-00.txt</a><br>
Status: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"ht=
tp://datatracker.ietf.org/doc/draft-zhu-rmcat-nada" target=3D"_blank" class=
=3D"cremed">http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools=
.ietf.org/html/draft-zhu-rmcat-nada-00" target=3D"_blank" class=3D"cremed">=
http://tools.ietf.org/html/draft-zhu-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp;&nbsp;This document describes a scheme named network-assisted dynamic=
<br>
&nbsp;&nbsp;adaptation (NADA), a novel congestion control approach for<br>
&nbsp;&nbsp;interactive real-time media applications, such as video confere=
ncing.<br>
&nbsp;&nbsp;In the proposed scheme, the sender regulates its sending rate b=
ased<br>
&nbsp;&nbsp;on either implicit or explicit congestion signaling, in a unifi=
ed<br>
&nbsp;&nbsp;approach. The scheme can reap the benefits of explicit congesti=
on<br>
&nbsp;&nbsp;notification markings from network nodes. It also maintains<br>
&nbsp;&nbsp;consistent sender behavior in the absence of such markings, by<=
br>
&nbsp;&nbsp;reacting to queuing delays instead.<br>
<br>
&nbsp;&nbsp;We present here the overall system architecture, recommended<br=
>
&nbsp;&nbsp;behaviors at the sender and the receiver, as well as expected n=
etwork<br>
&nbsp;&nbsp;nodes operations. Results from extensive simulation studies of =
the<br>
&nbsp;&nbsp;proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div>
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br>
</div>
<div>
<div>
<blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a href=3D"mailto:zhuxq@alumni.=
stanford.edu" target=3D"_blank" class=3D"cremed">zhuxq@alumni.stanford.edu<=
/a>&gt; wrote:<br>
<blockquote type=3D"cite">My colleague and I are working on a congestion co=
ntrol scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. A b=
rief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in Va=
ncouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">&quot;Network-Assisted Dynamic Adaptation (NADA):=
 A Design Summary&quot;). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a ch=
ance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may be =
a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present in=
 Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for furth=
er discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
</div>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</body>
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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: Michael Welzl <michawe@ifi.uio.no>
Thread-Topic: [rmcat] send agenda requests (was Re: First meeting)
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Cc: rmcat WG <rmcat@ietf.org>, Xiaoqing Zhu <zhuxq@alumni.stanford.edu>, "Eggert, Lars" <lars@netapp.com>, "Rong Pan \(ropan\)" <ropan@cisco.com>
Subject: Re: [rmcat] send agenda requests (was Re: First meeting)
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Ah, okay.  Thanks a lot for the clarification for the multi-hop scenario.

In that case I will only worry about the scenario where ECN-capable and ECN=
-incapable routers co-exist on the same path.

-Xiaoqing


On Oct 25, 2012, at 1:45 AM, Michael Welzl wrote:

Hi,


On 25. okt. 2012, at 02:20, Xiaoqing Zhu wrote:

Michael,

In general, I agree with you that we need a more graceful scheme to accommo=
date both implicit and explicit congestion signals. Our current -00 draft o=
bviously does not solve that yet.  I'll add this important point to the "op=
en issue" section of the draft, and also strive to find a solution for that=
.

Sure, I think we had already agreed, I just wanted to share this scenario t=
hat came to my mind.


However, I am not sure the dynamic scenario you describe will happen. If a =
packet passes along both ECN-supported and non-ECN-supported routers, would=
n't the ECN field bits in its header be marked as '00' (not ECN-capable), a=
s specified in RFC 3168?  In that case, even if congestion rises on the ECN=
-enabled route, the sender will still be reacting to the rise in observed d=
elay.

RFC 3168 is long and it's been a while since the last time I read it, but: =
I can't recall reading what you describe here.
My understanding of ECN (in a nutshell) is: ECN-capable end hosts sent 01 o=
r 10; ECN-incapable routers might not react to ECN at all, so anyway you ca=
n't trust them to do something based on RFC 3168 if they don't implement RF=
C 3168  :-)    ; if an ECN-capable router sees a 01 or 10 packet, it may ma=
rk it to 11; if an ECN-capable router sees a 00 packet, it drops it instead=
 of marking it.

So, no way to reliably determine if there are some routers that do *not* su=
pport ECN.


Also, my understanding from the previous email threads on this mailing list=
 seems to say that multi-hop congested links are hard scenarios to tackle a=
nyway.  Do you think we should focus on that now, or defer it after we obta=
in some satisfactory results from the simpler case of single-congested link=
?

As I tried to explain in my earlier email about that:
http://www.ietf.org/mail-archive/web/rmcat/current/msg00069.html
this was about multiple hops in the overlay - multiple RTP end systems atta=
ched to each other, to form a longer path, if you will. The underlay can al=
ways have more than just one hop.

Cheers,
Michael



Thanks,
Xiaoqing


On Wed, Oct 24, 2012 at 7:22 AM, Michael Welzl <michawe@ifi.uio.no<mailto:m=
ichawe@ifi.uio.no>> wrote:
Actually...

It just occurred to me that the dynamic scenario (going from ECN-supported =
to non-ECN-supported) is also quite realistic.
Consider a situation where you traverse two routers: router 1, followed by =
router 2. Router 1 isn't usually your bottleneck, but it supports ECN. Rout=
er 2 doesn't support ECN, but is your bottleneck most of the time.

Now, if a sudden traffic burst causes the queue of router 1 to grow, you wi=
ll start to believe that there is ECN support, and stop using delay. But th=
en, this queue might never grow again, all the congestion could happen in r=
outer 2...

My point is, *relying* on ECN to be there no matter how much feedback you g=
et is probably always the wrong thing to do. We can always only take it as =
a nice extra if its available, but I think we should never "switch to ECN m=
ode".

Cheers,
Michael


On 18. okt. 2012, at 05:49, Xiaoqing Zhu (xiaoqzhu) wrote:

> Your points are well taken, Michael.
>
>
> Agree that ultimately, it would be nice to have the scheme consider a mer=
ge of delay and ECN signals, instead of switching naively between the two. =
We considered that as a stretch goal though, according to the priorities di=
scussed in the last BOF. We are still in the process of improving the algor=
ithm in that aspect.
>
> I will put in some results from ECN-based marking in the compiled file to=
 share. But admitted, we have not yet explored the impact of all variations=
 of ECN configurations.  I will need to run more tests for that.
>
> -Xiaoqing
>
> On Oct 17, 2012, at 12:31 AM, Michael Welzl wrote:
>
>>
>> On 17. okt. 2012, at 06:58, Xiaoqing Zhu (xiaoqzhu) wrote:
>>
>>> Hi Michael,
>>>
>>> Thanks for your comments.  Please find my replies inline below.
>>>
>>> Best,
>>> Xiaoqing
>>>
>>>
>>> On Oct 16, 2012, at 1:11 AM, Michael Welzl wrote:
>>>
>>>> Hi,
>>>>
>>>> I took a quick look at this:
>>>>
>>>> I like the general idea of using as much as possible of existing netwo=
rk support - but I'm critical about the actual control you propose.
>>>>
>>>> In section 5.3.2, you suggest to replace eqn. 1 from section 5.3 with =
eqn. 2 when ECN feedback is available. This means to ignore a delay signal =
from now on, and interpret ECN marks in the exact same way as you interpret=
ed delay before - this is too simplistic, I think. After all, ECN marks req=
uire a queue to grow, and this group intends to minimize delay... why would=
 you stop using delay once you have ECN in addition?
>>>
>>>
>>> In our design, we considered the two cases, with or without ECN marking=
s, separately.  Therefore, the dynamic scenario when ECN becomes available =
in the middle of a video conferencing session has not yet been fully studie=
d.  Just wondering, are cases where ECN marking are turned on and off dynam=
ically common in reality?
>>
>> First, anything can happen when you have a routing change.
>> Second, I'm really sorry, but this wasn't what I meant. I shouldn't have=
 written "when" but "if". I wasn't referring to a case where we go from no-=
ECN-support to ECN-support. So this is about the ECN-always-supported case.
>>
>>
>>>> In practice, using your scheme would probably mean that, as soon as I =
enable RED with ECN on the bottleneck between the two end systems, the user=
s' observed delay grows. Not exactly an incentive?
>>>>
>>> What you mentioned (uers observing higher delay in ECN-enabled systems)=
 will not happen.
>>>
>>> But rather, our simulation results have indicated that ECN with RED-bas=
ed marking yields fairly similar results as the default delay-based scheme.=
 They certainly do not incur higher delay, since the endpoint will automati=
cally back off in face of higher ECN markings, which are indications of hig=
her delay. The two systems (delay-based and ECN-based) can be tuned to stab=
ilize that the same queuing delay at the bottleneck.
>>
>> Whether you see significant delay before you see a large enough amount o=
f ECN markings depends on the queue length and RED parameters, none of whic=
h are under the end system's control. ECN is also a probabilistic signal, s=
o sometimes you may get delay growth but just be lucky (or unlucky) enough =
to not get an ECN mark.
>>
>> So, I'd have to see results that involve various RED settings and queue =
lengths before I believe that.
>>
>> Cheers,
>> Michael
>>
>





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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
Ah, okay. &nbsp;Thanks a lot for the clarification for the multi-hop scenar=
io.&nbsp;
<div><br>
</div>
<div>In that case I will only worry about the scenario where ECN-capable an=
d ECN-incapable routers co-exist on the same path.&nbsp;</div>
<div><br>
</div>
<div>-Xiaoqing</div>
<div><br>
</div>
<div><br>
<div>
<div>
<div>On Oct 25, 2012, at 1:45 AM, Michael Welzl wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
Hi,
<div><br>
</div>
<div><br>
<div>
<div>On 25. okt. 2012, at 02:20, Xiaoqing Zhu wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div>Michael,&nbsp;</div>
<div><br>
</div>
<div>In general, I agree with you that we need a more graceful scheme to ac=
commodate both implicit and explicit congestion signals. Our current -00 dr=
aft obviously does not solve that yet. &nbsp;I'll add this important point =
to the &quot;open issue&quot; section of the draft,
 and also strive to find a solution for that. &nbsp;</div>
</blockquote>
<div><br>
</div>
<div>Sure, I think we had already agreed, I just wanted to share this scena=
rio that came to my mind.</div>
<div><br>
</div>
<br>
<blockquote type=3D"cite">
<div>However, I am not sure the dynamic scenario you describe will happen.&=
nbsp;If a packet passes along both ECN-supported and non-ECN-supported rout=
ers, wouldn't the ECN field bits in its header be marked as '00' (not ECN-c=
apable), as specified in RFC 3168? &nbsp;In
 that case, even if congestion rises on the ECN-enabled route, the sender w=
ill still be reacting to the rise in observed delay.&nbsp;</div>
</blockquote>
<div><br>
</div>
<div>RFC 3168 is long and it's been a while since the last time I read it, =
but: I can't recall reading what you describe here.</div>
My understanding of ECN (in a nutshell) is: ECN-capable end hosts sent 01 o=
r 10; ECN-incapable routers might not react to ECN at all, so anyway you ca=
n't trust them to do something based on RFC 3168 if they don't implement RF=
C 3168 &nbsp;:-) &nbsp; &nbsp;; if an ECN-capable
 router sees a 01 or 10 packet, it may mark it to 11; if an ECN-capable rou=
ter sees a 00 packet, it drops it instead of marking it.</div>
<div><br>
</div>
<div>So, no way to reliably determine if there are some routers that do *no=
t* support ECN.</div>
<div><br>
</div>
<div><br>
<blockquote type=3D"cite">
<div>Also, my understanding from the previous email threads on this mailing=
 list seems to say that multi-hop congested links are hard scenarios to tac=
kle anyway. &nbsp;Do you think we should focus on that now, or defer it aft=
er we obtain some satisfactory results
 from the simpler case of single-congested link?&nbsp;</div>
</blockquote>
<div><br>
</div>
<div>As I tried to explain in my earlier email about that:</div>
<div><a href=3D"http://www.ietf.org/mail-archive/web/rmcat/current/msg00069=
.html">http://www.ietf.org/mail-archive/web/rmcat/current/msg00069.html</a>=
</div>
<div>this was about multiple hops in the overlay - multiple RTP end systems=
 attached to each other, to form a longer path, if you will. The underlay c=
an always have more than just one hop.</div>
<div><br>
</div>
Cheers,</div>
<div>Michael</div>
<div><br>
</div>
<div><br>
<blockquote type=3D"cite">
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div>
<div>
<div><br>
</div>
<div><br>
<div class=3D"gmail_quote">On Wed, Oct 24, 2012 at 7:22 AM, Michael Welzl <=
span dir=3D"ltr">
&lt;<a href=3D"mailto:michawe@ifi.uio.no" target=3D"_blank">michawe@ifi.uio=
.no</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Actually...<br>
<br>
It just occurred to me that the dynamic scenario (going from ECN-supported =
to non-ECN-supported) is also quite realistic.<br>
Consider a situation where you traverse two routers: router 1, followed by =
router 2. Router 1 isn't usually your bottleneck, but it supports ECN. Rout=
er 2 doesn't support ECN, but is your bottleneck most of the time.<br>
<br>
Now, if a sudden traffic burst causes the queue of router 1 to grow, you wi=
ll start to believe that there is ECN support, and stop using delay. But th=
en, this queue might never grow again, all the congestion could happen in r=
outer 2...<br>
<br>
My point is, *relying* on ECN to be there no matter how much feedback you g=
et is probably always the wrong thing to do. We can always only take it as =
a nice extra if its available, but I think we should never &quot;switch to =
ECN mode&quot;.<br>
<br>
Cheers,<br>
Michael<br>
<div class=3D"HOEnZb">
<div class=3D"h5"><br>
<br>
On 18. okt. 2012, at 05:49, Xiaoqing Zhu (xiaoqzhu) wrote:<br>
<br>
&gt; Your points are well taken, Michael.<br>
&gt;<br>
&gt;<br>
&gt; Agree that ultimately, it would be nice to have the scheme consider a =
merge of delay and ECN signals, instead of switching naively between the tw=
o. We considered that as a stretch goal though, according to the priorities=
 discussed in the last BOF. We are
 still in the process of improving the algorithm in that aspect.<br>
&gt;<br>
&gt; I will put in some results from ECN-based marking in the compiled file=
 to share. But admitted, we have not yet explored the impact of all variati=
ons of ECN configurations. &nbsp;I will need to run more tests for that.<br=
>
&gt;<br>
&gt; -Xiaoqing<br>
&gt;<br>
&gt; On Oct 17, 2012, at 12:31 AM, Michael Welzl wrote:<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; On 17. okt. 2012, at 06:58, Xiaoqing Zhu (xiaoqzhu) wrote:<br>
&gt;&gt;<br>
&gt;&gt;&gt; Hi Michael,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Thanks for your comments. &nbsp;Please find my replies inline =
below.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Best,<br>
&gt;&gt;&gt; Xiaoqing<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On Oct 16, 2012, at 1:11 AM, Michael Welzl wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Hi,<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I took a quick look at this:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I like the general idea of using as much as possible of ex=
isting network support - but I'm critical about the actual control you prop=
ose.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; In section 5.3.2, you suggest to replace eqn. 1 from secti=
on 5.3 with eqn. 2 when ECN feedback is available. This means to ignore a d=
elay signal from now on, and interpret ECN marks in the exact same way as y=
ou interpreted delay before - this is too simplistic,
 I think. After all, ECN marks require a queue to grow, and this group inte=
nds to minimize delay... why would you stop using delay once you have ECN i=
n addition?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; In our design, we considered the two cases, with or without EC=
N markings, separately. &nbsp;Therefore, the dynamic scenario when ECN beco=
mes available in the middle of a video conferencing session has not yet bee=
n fully studied. &nbsp;Just wondering, are cases where
 ECN marking are turned on and off dynamically common in reality?<br>
&gt;&gt;<br>
&gt;&gt; First, anything can happen when you have a routing change.<br>
&gt;&gt; Second, I'm really sorry, but this wasn't what I meant. I shouldn'=
t have written &quot;when&quot; but &quot;if&quot;. I wasn't referring to a=
 case where we go from no-ECN-support to ECN-support. So this is about the =
ECN-always-supported case.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;&gt;&gt; In practice, using your scheme would probably mean that, a=
s soon as I enable RED with ECN on the bottleneck between the two end syste=
ms, the users' observed delay grows. Not exactly an incentive?<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt; What you mentioned (uers observing higher delay in ECN-enabled=
 systems) will not happen.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; But rather, our simulation results have indicated that ECN wit=
h RED-based marking yields fairly similar results as the default delay-base=
d scheme. They certainly do not incur higher delay, since the endpoint will=
 automatically back off in face of higher
 ECN markings, which are indications of higher delay. The two systems (dela=
y-based and ECN-based) can be tuned to stabilize that the same queuing dela=
y at the bottleneck.<br>
&gt;&gt;<br>
&gt;&gt; Whether you see significant delay before you see a large enough am=
ount of ECN markings depends on the queue length and RED parameters, none o=
f which are under the end system's control. ECN is also a probabilistic sig=
nal, so sometimes you may get delay growth
 but just be lucky (or unlucky) enough to not get an ECN mark.<br>
&gt;&gt;<br>
&gt;&gt; So, I'd have to see results that involve various RED settings and =
queue lengths before I believe that.<br>
&gt;&gt;<br>
&gt;&gt; Cheers,<br>
&gt;&gt; Michael<br>
&gt;&gt;<br>
&gt;<br>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</body>
</html>

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From p.ohanlon@gmail.com  Tue Oct 30 03:09:01 2012
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To: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
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Hi,

Presumably by time-smooth average you mean EWMA - so you're missing an =
alpha?

> d_avg =3D (1-alpha)*d_avg + d_n

Should be:  d_avg =3D (1-alpha)*d_avg + alpha*d_n

Piers.

On 29 Oct 2012, at 15:28, Xiaoqing Zhu (xiaoqzhu) wrote:

>=20
> On Oct 29, 2012, at 10:23 AM, Stefan Holmer wrote:
>=20
>>=20
>>=20
>>=20
>> On Mon, Oct 29, 2012 at 3:53 PM, Xiaoqing Zhu (xiaoqzhu) =
<xiaoqzhu@cisco.com> wrote:
>> Hi Stefan,=20
>>=20
>> Thanks a lot for your comments. I'll add in more details on the =
description for the receiver actions.=20
>>=20
>> You are right, we did employ some filtering of the observed =
per-packet queuing delay, to avoid erratic reactions in the sending =
rate. The queuing delay employed in the rate calculation in Eq. (1) is =
actually the time-smoothed averaging d_avg:=20
>>=20
>> d_avg =3D (1-alpha)*d_avg + d_n, where d_n represents the measured =
queuing delay  (end-to-end delay minus minimum observed value) for the =
n-th packet. The parameter alpha adjusts the level of smoothness in our =
average process. In our experiments, we are using alpha =3D 10e-4.=20
>>=20
>> Thanks. Just to be clear,=20
>>=20
>> d_n =3D receiver_time_n - timestamp_n - min_observed
>>=20
>> and min_observed is the smallest d_n seen in some window?
>=20
> That's right.  BTW, while the time-averaged filtering works well for =
us, for now, we are also considering building in more robustness against =
individual bad samples. The Kalman filtering approach presented in your =
proposal seems to be a good candidate for that. Maybe there is a way to =
incorporate that approach here for the delay estimation as well.  What =
do you think?  =20
>=20
>=20
>> =20
>>=20
>> Also, it's obvious that Eq.(1) wouldn't hold for for d=3D0, in which =
case the rate is chosen to be R_max.  As mentioned in the paragraph =
below Eq. (1), any calculated target rate R_o greater than R_max will =
also be clipped to the R_max.=20
>>=20
>> Initially, I omitted these details in the draft proposal for brevity. =
Given your comments and concerns, looks like I'd better put them back in =
for clarity.=20
>>=20
>> Best,
>> Xiaoqing
>>=20
>>=20
>> On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:
>>=20
>>> Hi Xiaoqing,
>>>=20
>>> Interesting draft, I have a few questions related to the queuing =
delay part of it.
>>>=20
>>> "The role of the receiver is fairly straightforward. It observes and =
estimates end-to-end queuing delay d and ECN marking ratio p of the =
stream. The former can be obtained from the RTP timestamp provided by =
the sender."
>>>=20
>>> You don't provide much detail on how the end-to-end queuing delay is =
estimated from the timestamps. Would it be possible to be more specific? =
Are you employing a filter? Related to that, you seem to be running into =
a divide by zero in equation 1 if your queuing delay turns out to be =
zero. It also seems a bit bold to go directly from R_min to R_max if the =
queuing delay is low enough.
>>>=20
>>> /Stefan
>>>=20
>>>=20
>>>=20
>>>=20
>>> On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (xiaoqzhu) =
<xiaoqzhu@cisco.com> wrote:
>>> And below is the notification of our submission:=20
>>>=20
>>> Thanks,
>>> Xiaoqing
>>>=20
>>> --------------------------------------
>>> A new version of I-D, draft-zhu-rmcat-nada-00.txt
>>> has been successfully submitted by Xiaoqing Zhu and posted to the
>>> IETF repository.
>>>=20
>>> Filename:  draft-zhu-rmcat-nada
>>> Revision:  00
>>> Title:  NADA: A Unified Congestion Control Scheme for Real-Time =
Media
>>> Creation date:  2012-10-14
>>> WG ID:  Individual Submission
>>> Number of pages: 10
>>> URL:             =
http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt
>>> Status:          =
http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
>>> Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00
>>>=20
>>>=20
>>> Abstract:
>>>   This document describes a scheme named network-assisted dynamic
>>>   adaptation (NADA), a novel congestion control approach for
>>>   interactive real-time media applications, such as video =
conferencing.
>>>   In the proposed scheme, the sender regulates its sending rate =
based
>>>   on either implicit or explicit congestion signaling, in a unified
>>>   approach. The scheme can reap the benefits of explicit congestion
>>>   notification markings from network nodes. It also maintains
>>>   consistent sender behavior in the absence of such markings, by
>>>   reacting to queuing delays instead.
>>>=20
>>>   We present here the overall system architecture, recommended
>>>   behaviors at the sender and the receiver, as well as expected =
network
>>>   nodes operations. Results from extensive simulation studies of the
>>>   proposed scheme are available upon request.
>>>=20
>>>=20
>>>=20
>>>=20
>>> The IETF Secretariat
>>> --------------------------------------
>>> On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:
>>>=20
>>>> Hi,
>>>>=20
>>>> On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu> =
wrote:
>>>>> My colleague and I are working on a congestion control scheme for
>>>>> real-time conferencing applications at Cisco. A brief version of =
that
>>>>> algorithm was presented at the IAB Workshop in Vancouver (paper =
#12,
>>>>> "Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). =
Due
>>>>> to time constraints, however, we did not get a chance to mention =
the
>>>>> delay-based variant of that scheme, which may be a better fit for
>>>>> rmcat.
>>>>>=20
>>>>> We would like to request for a slot to present in Atlanta. We will
>>>>> also submit the ID draft before Oct 15, for further discussions on
>>>>> this mailing list.
>>>>=20
>>>> I look forward to seeing the draft being discussed on the list!
>>>>=20
>>>> Also, I'm noting your request for a slot. Because the actual CC =
mechanisms are not our most pressing milestones initially - the =
requirements and eval criteria are - we may decide to focus our meeting =
time on those other drafts, if demand for agenda time is high.=20
>>>>=20
>>>> Lars
>>>=20
>>>=20
>>=20
>>=20
>=20


--Apple-Mail=_BE385D87-819D-4576-847F-D7D75130CF3A
Content-Transfer-Encoding: 7bit
Content-Type: text/html;
	charset=us-ascii

<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi,<div><br></div><div>Presumably by time-smooth average you mean EWMA - so you're missing an alpha?</div><div><br></div><div><blockquote type="cite"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><div class="gmail_extra"><div class="gmail_quote"><div style="word-wrap: break-word; "><div>d_avg = (1-alpha)*d_avg + d_n</div></div></div></div></div></div></blockquote><br>Should be: &nbsp;d_avg = (1-alpha)*d_avg + alpha*d_n</div><div><br></div><div>Piers.</div><div><br><div><div>On 29 Oct 2012, at 15:28, Xiaoqing Zhu (xiaoqzhu) wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">

<meta http-equiv="Content-Type" content="text/html; charset=us-ascii">

<div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">
<br>
<div>
<div>On Oct 29, 2012, at 10:23 AM, Stefan Holmer wrote:</div>
<br class="Apple-interchange-newline">
<blockquote type="cite"><br>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Mon, Oct 29, 2012 at 3:53 PM, Xiaoqing Zhu (xiaoqzhu)
<span dir="ltr">&lt;<a href="mailto:xiaoqzhu@cisco.com" target="_blank" class="cremed">xiaoqzhu@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0.8ex; border-left-width: 1px; border-left-color: rgb(204, 204, 204); border-left-style: solid; padding-left: 1ex; position: static; z-index: auto; ">
<div style="word-wrap:break-word">Hi Stefan,&nbsp;
<div><br>
</div>
<div>Thanks a lot for your comments. I'll add in more details on the description for the receiver actions.&nbsp;</div>
<div><br>
</div>
<div>You are right, we did employ some filtering of the observed per-packet queuing delay, to avoid erratic reactions in the sending rate. The queuing delay employed in the rate calculation in Eq. (1) is actually the time-smoothed averaging d_avg:&nbsp;</div>
<div><br>
</div>
<div>d_avg = (1-alpha)*d_avg + d_n, where d_n represents the measured queuing delay &nbsp;(end-to-end delay minus minimum observed value) for the n-th packet. The parameter alpha adjusts the level of smoothness in our average process. In our experiments, we are
 using alpha = 10e-4.&nbsp;</div>
</div>
</blockquote>
<div><br>
</div>
<div>Thanks. Just to be clear,&nbsp;</div>
<div><br>
</div>
<div>d_n = receiver_time_n - timestamp_n - min_observed</div>
<div><br>
</div>
<div>and min_observed is the smallest d_n seen in some window?</div>
</div>
</div>
</blockquote>
<div><br>
</div>
That's right. &nbsp;BTW, while the time-averaged filtering works well for us, for now, we are also considering building in more robustness against individual bad samples. The Kalman filtering approach presented in your proposal seems to be a good candidate for that.
 Maybe there is a way to incorporate that approach here for the delay estimation as well. &nbsp;What do you think? &nbsp;&nbsp;</div>
<div><br>
</div>
<div><br>
<blockquote type="cite">
<div class="gmail_extra">
<div class="gmail_quote">
<div>&nbsp;</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">
<div><br>
</div>
<div>Also, it's obvious that Eq.(1) wouldn't hold for for d=0, in which case the rate is chosen to be R_max. &nbsp;As mentioned in the paragraph below Eq. (1), any calculated target rate R_o greater than R_max will also be clipped to the R_max.&nbsp;</div>
<div><br>
</div>
<div>Initially, I omitted these details in the draft proposal for brevity. Given your comments and concerns, looks like I'd better put them back in for clarity.&nbsp;</div>
<div><br>
</div>
<div>Best,</div>
<div>Xiaoqing</div>
<div>
<div class="h5">
<div><br>
</div>
<div><br>
</div>
<div>
<div>
<div>On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:</div>
<br>
<blockquote type="cite">Hi Xiaoqing,
<div><br>
</div>
<div>Interesting draft, I have a few questions related to the queuing delay part of it.</div>
<div><br>
</div>
<div>"The role of the receiver is fairly straightforward. It observes and estimates end-to-end queuing delay d and ECN marking ratio p of the stream. The former can be obtained from the RTP timestamp provided by the sender."</div>
<div><br>
</div>
<div>You don't provide much detail on how the end-to-end queuing delay is estimated from the timestamps. Would it be possible to be more specific? Are you employing a filter? Related to that, you seem to be running into a divide by zero in equation 1 if your
 queuing delay turns out to be zero. It also seems a bit bold to go directly from R_min to R_max if the queuing delay is low enough.</div>
<div><br>
</div>
<div>/Stefan</div>
<div><br>
</div>
<div><br>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (xiaoqzhu)
<span dir="ltr">&lt;<a href="mailto:xiaoqzhu@cisco.com" target="_blank" class="cremed">xiaoqzhu@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">And below is the notification of our submission:&nbsp;
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span style="white-space:pre-wrap"> </span>&nbsp;draft-zhu-rmcat-nada<br>
Revision:<span style="white-space:pre-wrap"> </span>&nbsp;00<br>
Title:<span style="white-space:pre-wrap"> </span><span style="white-space:pre-wrap"></span>&nbsp;NADA: A Unified Congestion Control Scheme for Real-Time Media<br>
Creation date:<span style="white-space:pre-wrap"> </span>&nbsp;2012-10-14<br>
WG ID:<span style="white-space:pre-wrap"> </span><span style="white-space:pre-wrap"></span>&nbsp;Individual Submission<br>
Number of pages: 10<br>
URL: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href="http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt" target="_blank" class="cremed">http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt</a><br>
Status: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href="http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada" target="_blank" class="cremed">http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href="http://tools.ietf.org/html/draft-zhu-rmcat-nada-00" target="_blank" class="cremed">http://tools.ietf.org/html/draft-zhu-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp;&nbsp;This document describes a scheme named network-assisted dynamic<br>
&nbsp;&nbsp;adaptation (NADA), a novel congestion control approach for<br>
&nbsp;&nbsp;interactive real-time media applications, such as video conferencing.<br>
&nbsp;&nbsp;In the proposed scheme, the sender regulates its sending rate based<br>
&nbsp;&nbsp;on either implicit or explicit congestion signaling, in a unified<br>
&nbsp;&nbsp;approach. The scheme can reap the benefits of explicit congestion<br>
&nbsp;&nbsp;notification markings from network nodes. It also maintains<br>
&nbsp;&nbsp;consistent sender behavior in the absence of such markings, by<br>
&nbsp;&nbsp;reacting to queuing delays instead.<br>
<br>
&nbsp;&nbsp;We present here the overall system architecture, recommended<br>
&nbsp;&nbsp;behaviors at the sender and the receiver, as well as expected network<br>
&nbsp;&nbsp;nodes operations. Results from extensive simulation studies of the<br>
&nbsp;&nbsp;proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div>
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br>
</div>
<div>
<div>
<blockquote type="cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a href="mailto:zhuxq@alumni.stanford.edu" target="_blank" class="cremed">zhuxq@alumni.stanford.edu</a>&gt; wrote:<br>
<blockquote type="cite">My colleague and I are working on a congestion control scheme for<br>
</blockquote>
<blockquote type="cite">real-time conferencing applications at Cisco. A brief version of that<br>
</blockquote>
<blockquote type="cite">algorithm was presented at the IAB Workshop in Vancouver (paper #12,<br>
</blockquote>
<blockquote type="cite">"Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due<br>
</blockquote>
<blockquote type="cite">to time constraints, however, we did not get a chance to mention the<br>
</blockquote>
<blockquote type="cite">delay-based variant of that scheme, which may be a better fit for<br>
</blockquote>
<blockquote type="cite">rmcat.<br>
</blockquote>
<blockquote type="cite"><br>
</blockquote>
<blockquote type="cite">We would like to request for a slot to present in Atlanta. We will<br>
</blockquote>
<blockquote type="cite">also submit the ID draft before Oct 15, for further discussions on<br>
</blockquote>
<blockquote type="cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I'm noting your request for a slot. Because the actual CC mechanisms are not our most pressing milestones initially - the requirements and eval criteria are - we may decide to focus our meeting time on those other drafts, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
</div>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>

</blockquote></div><br></div></body></html>
--Apple-Mail=_BE385D87-819D-4576-847F-D7D75130CF3A--

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On Mon, Oct 29, 2012 at 4:28 PM, Xiaoqing Zhu (xiaoqzhu) <xiaoqzhu@cisco.com
> wrote:

>
>  On Oct 29, 2012, at 10:23 AM, Stefan Holmer wrote:
>
>
>
>
> On Mon, Oct 29, 2012 at 3:53 PM, Xiaoqing Zhu (xiaoqzhu) <
> xiaoqzhu@cisco.com> wrote:
>
>> Hi Stefan,
>>
>>  Thanks a lot for your comments. I'll add in more details on the
>> description for the receiver actions.
>>
>>  You are right, we did employ some filtering of the observed per-packet
>> queuing delay, to avoid erratic reactions in the sending rate. The queuing
>> delay employed in the rate calculation in Eq. (1) is actually the
>> time-smoothed averaging d_avg:
>>
>>  d_avg = (1-alpha)*d_avg + d_n, where d_n represents the measured
>> queuing delay  (end-to-end delay minus minimum observed value) for the n-th
>> packet. The parameter alpha adjusts the level of smoothness in our average
>> process. In our experiments, we are using alpha = 10e-4.
>>
>
>  Thanks. Just to be clear,
>
>  d_n = receiver_time_n - timestamp_n - min_observed
>
>  and min_observed is the smallest d_n seen in some window?
>
>
>  That's right.  BTW, while the time-averaged filtering works well for us,
> for now, we are also considering building in more robustness against
> individual bad samples. The Kalman filtering approach presented in your
> proposal seems to be a good candidate for that. Maybe there is a way to
> incorporate that approach here for the delay estimation as well.  What do
> you think?
>
>
That would be an option. Note that RRTCC isn't based on one-way delay
estimates but on inter-arrival time, which is changes in one-way delay.


>
>
>
>>
>>  Also, it's obvious that Eq.(1) wouldn't hold for for d=0, in which case
>> the rate is chosen to be R_max.  As mentioned in the paragraph below Eq.
>> (1), any calculated target rate R_o greater than R_max will also be clipped
>> to the R_max.
>>
>>  Initially, I omitted these details in the draft proposal for brevity.
>> Given your comments and concerns, looks like I'd better put them back in
>> for clarity.
>>
>>  Best,
>> Xiaoqing
>>
>>
>>   On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:
>>
>> Hi Xiaoqing,
>>
>>  Interesting draft, I have a few questions related to the queuing delay
>> part of it.
>>
>>  "The role of the receiver is fairly straightforward. It observes and
>> estimates end-to-end queuing delay d and ECN marking ratio p of the stream.
>> The former can be obtained from the RTP timestamp provided by the sender."
>>
>>  You don't provide much detail on how the end-to-end queuing delay is
>> estimated from the timestamps. Would it be possible to be more specific?
>> Are you employing a filter? Related to that, you seem to be running into a
>> divide by zero in equation 1 if your queuing delay turns out to be zero. It
>> also seems a bit bold to go directly from R_min to R_max if the queuing
>> delay is low enough.
>>
>>  /Stefan
>>
>>
>>
>>
>> On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (xiaoqzhu) <
>> xiaoqzhu@cisco.com> wrote:
>>
>>> And below is the notification of our submission:
>>>
>>>  Thanks,
>>> Xiaoqing
>>>
>>>  --------------------------------------
>>> A new version of I-D, draft-zhu-rmcat-nada-00.txt
>>> has been successfully submitted by Xiaoqing Zhu and posted to the
>>> IETF repository.
>>>
>>> Filename:  draft-zhu-rmcat-nada
>>> Revision:  00
>>> Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
>>> Creation date:  2012-10-14
>>> WG ID:  Individual Submission
>>> Number of pages: 10
>>> URL:
>>> http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt
>>> Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
>>> Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00
>>>
>>>
>>> Abstract:
>>>   This document describes a scheme named network-assisted dynamic
>>>   adaptation (NADA), a novel congestion control approach for
>>>   interactive real-time media applications, such as video conferencing.
>>>   In the proposed scheme, the sender regulates its sending rate based
>>>   on either implicit or explicit congestion signaling, in a unified
>>>   approach. The scheme can reap the benefits of explicit congestion
>>>   notification markings from network nodes. It also maintains
>>>   consistent sender behavior in the absence of such markings, by
>>>   reacting to queuing delays instead.
>>>
>>>   We present here the overall system architecture, recommended
>>>   behaviors at the sender and the receiver, as well as expected network
>>>   nodes operations. Results from extensive simulation studies of the
>>>   proposed scheme are available upon request.
>>>
>>>
>>>
>>>
>>> The IETF Secretariat
>>> --------------------------------------
>>>  On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:
>>>
>>>   Hi,
>>>
>>> On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu>
>>> wrote:
>>>
>>> My colleague and I are working on a congestion control scheme for
>>>
>>> real-time conferencing applications at Cisco. A brief version of that
>>>
>>> algorithm was presented at the IAB Workshop in Vancouver (paper #12,
>>>
>>> "Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
>>>
>>> to time constraints, however, we did not get a chance to mention the
>>>
>>> delay-based variant of that scheme, which may be a better fit for
>>>
>>> rmcat.
>>>
>>>
>>>  We would like to request for a slot to present in Atlanta. We will
>>>
>>> also submit the ID draft before Oct 15, for further discussions on
>>>
>>> this mailing list.
>>>
>>>
>>> I look forward to seeing the draft being discussed on the list!
>>>
>>> Also, I'm noting your request for a slot. Because the actual CC
>>> mechanisms are not our most pressing milestones initially - the
>>> requirements and eval criteria are - we may decide to focus our meeting
>>> time on those other drafts, if demand for agenda time is high.
>>>
>>> Lars
>>>
>>>
>>>
>>
>>
>
>

--14dae934061fa4783804cd44fe3e
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, O=
ct 29, 2012 at 4:28 PM, Xiaoqing Zhu (xiaoqzhu) <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:xiaoqzhu@cisco.com" target=3D"_blank" class=3D"cremed">xiaoqzh=
u@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">



<div style=3D"word-wrap:break-word">
<br>
<div><div class=3D"im">
<div>On Oct 29, 2012, at 10:23 AM, Stefan Holmer wrote:</div>
<br>
<blockquote type=3D"cite"><br>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Mon, Oct 29, 2012 at 3:53 PM, Xiaoqing Zhu (x=
iaoqzhu)
<span dir=3D"ltr">&lt;<a href=3D"mailto:xiaoqzhu@cisco.com" target=3D"_blan=
k" class=3D"cremed">xiaoqzhu@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">Hi Stefan,=A0
<div><br>
</div>
<div>Thanks a lot for your comments. I&#39;ll add in more details on the de=
scription for the receiver actions.=A0</div>
<div><br>
</div>
<div>You are right, we did employ some filtering of the observed per-packet=
 queuing delay, to avoid erratic reactions in the sending rate. The queuing=
 delay employed in the rate calculation in Eq. (1) is actually the time-smo=
othed averaging d_avg:=A0</div>

<div><br>
</div>
<div>d_avg =3D (1-alpha)*d_avg + d_n, where d_n represents the measured que=
uing delay =A0(end-to-end delay minus minimum observed value) for the n-th =
packet. The parameter alpha adjusts the level of smoothness in our average =
process. In our experiments, we are
 using alpha =3D 10e-4.=A0</div>
</div>
</blockquote>
<div><br>
</div>
<div>Thanks. Just to be clear,=A0</div>
<div><br>
</div>
<div>d_n =3D receiver_time_n - timestamp_n - min_observed</div>
<div><br>
</div>
<div>and min_observed is the smallest d_n seen in some window?</div>
</div>
</div>
</blockquote>
<div><br>
</div></div>
That&#39;s right. =A0BTW, while the time-averaged filtering works well for =
us, for now, we are also considering building in more robustness against in=
dividual bad samples. The Kalman filtering approach presented in your propo=
sal seems to be a good candidate for that.
 Maybe there is a way to incorporate that approach here for the delay estim=
ation as well. =A0What do you think? =A0=A0</div><div><div class=3D"h5">
<div><br></div></div></div></div></blockquote><div><br></div><div>That woul=
d be an option. Note that RRTCC isn&#39;t based on one-way delay estimates =
but on inter-arrival time, which is changes in one-way delay.</div><div>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div style=3D"word-wrap:break-word"=
><div><div class=3D"h5"><div>
</div>
<div><br>
<blockquote type=3D"cite">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div><br>
</div>
<div>Also, it&#39;s obvious that Eq.(1) wouldn&#39;t hold for for d=3D0, in=
 which case the rate is chosen to be R_max. =A0As mentioned in the paragrap=
h below Eq. (1), any calculated target rate R_o greater than R_max will als=
o be clipped to the R_max.=A0</div>

<div><br>
</div>
<div>Initially, I omitted these details in the draft proposal for brevity. =
Given your comments and concerns, looks like I&#39;d better put them back i=
n for clarity.=A0</div>
<div><br>
</div>
<div>Best,</div>
<div>Xiaoqing</div>
<div>
<div>
<div><br>
</div>
<div><br>
</div>
<div>
<div>
<div>On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:</div>
<br>
<blockquote type=3D"cite">Hi Xiaoqing,
<div><br>
</div>
<div>Interesting draft, I have a few questions related to the queuing delay=
 part of it.</div>
<div><br>
</div>
<div>&quot;The role of the receiver is fairly straightforward. It observes =
and estimates end-to-end queuing delay d and ECN marking ratio p of the str=
eam. The former can be obtained from the RTP timestamp provided by the send=
er.&quot;</div>

<div><br>
</div>
<div>You don&#39;t provide much detail on how the end-to-end queuing delay =
is estimated from the timestamps. Would it be possible to be more specific?=
 Are you employing a filter? Related to that, you seem to be running into a=
 divide by zero in equation 1 if your
 queuing delay turns out to be zero. It also seems a bit bold to go directl=
y from R_min to R_max if the queuing delay is low enough.</div>
<div><br>
</div>
<div>/Stefan</div>
<div><br>
</div>
<div><br>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (=
xiaoqzhu)
<span dir=3D"ltr">&lt;<a href=3D"mailto:xiaoqzhu@cisco.com" target=3D"_blan=
k" class=3D"cremed">xiaoqzhu@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">And below is the notification of our su=
bmission:=A0
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span style=3D"white-space:pre-wrap"> </span>=A0draft-zhu-rmcat-na=
da<br>
Revision:<span style=3D"white-space:pre-wrap"> </span>=A000<br>
Title:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>=A0NADA: A Unified Congestion Control Scheme for Real-T=
ime Media<br>
Creation date:<span style=3D"white-space:pre-wrap"> </span>=A02012-10-14<br=
>
WG ID:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>=A0Individual Submission<br>
Number of pages: 10<br>
URL: =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0<a href=3D"http://www.ietf.org/int=
ernet-drafts/draft-zhu-rmcat-nada-00.txt" target=3D"_blank" class=3D"cremed=
">http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.txt</a><br>
Status: =A0=A0=A0=A0=A0=A0=A0=A0=A0<a href=3D"http://datatracker.ietf.org/d=
oc/draft-zhu-rmcat-nada" target=3D"_blank" class=3D"cremed">http://datatrac=
ker.ietf.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: =A0=A0=A0=A0=A0=A0=A0<a href=3D"http://tools.ietf.org/html/draft-=
zhu-rmcat-nada-00" target=3D"_blank" class=3D"cremed">http://tools.ietf.org=
/html/draft-zhu-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
=A0=A0This document describes a scheme named network-assisted dynamic<br>
=A0=A0adaptation (NADA), a novel congestion control approach for<br>
=A0=A0interactive real-time media applications, such as video conferencing.=
<br>
=A0=A0In the proposed scheme, the sender regulates its sending rate based<b=
r>
=A0=A0on either implicit or explicit congestion signaling, in a unified<br>
=A0=A0approach. The scheme can reap the benefits of explicit congestion<br>
=A0=A0notification markings from network nodes. It also maintains<br>
=A0=A0consistent sender behavior in the absence of such markings, by<br>
=A0=A0reacting to queuing delays instead.<br>
<br>
=A0=A0We present here the overall system architecture, recommended<br>
=A0=A0behaviors at the sender and the receiver, as well as expected network=
<br>
=A0=A0nodes operations. Results from extensive simulation studies of the<br=
>
=A0=A0proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div>
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br>
</div>
<div>
<div>
<blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a href=3D"mailto:zhuxq@alumni.=
stanford.edu" target=3D"_blank" class=3D"cremed">zhuxq@alumni.stanford.edu<=
/a>&gt; wrote:<br>
<blockquote type=3D"cite">My colleague and I are working on a congestion co=
ntrol scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. A b=
rief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in Va=
ncouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">&quot;Network-Assisted Dynamic Adaptation (NADA):=
 A Design Summary&quot;). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a ch=
ance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may be =
a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present in=
 Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for furth=
er discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I&#39;m noting your request for a slot. Because the actual CC mechani=
sms are not our most pressing milestones initially - the requirements and e=
val criteria are - we may decide to focus our meeting time on those other d=
rafts, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
</div>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div></div></div>

</blockquote></div><br></div>

--14dae934061fa4783804cd44fe3e--

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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: "Piers O'Hanlon" <p.ohanlon@gmail.com>
Thread-Topic: [rmcat] send agenda requests (was Re: First meeting)
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Oops. Sorry about the typo.  Yes, there is the alpha for the second term.

Best,
Xiaoqing

On Oct 30, 2012, at 5:08 AM, Piers O'Hanlon wrote:

Hi,

Presumably by time-smooth average you mean EWMA - so you're missing an alph=
a?

d_avg =3D (1-alpha)*d_avg + d_n

Should be:  d_avg =3D (1-alpha)*d_avg + alpha*d_n

Piers.

On 29 Oct 2012, at 15:28, Xiaoqing Zhu (xiaoqzhu) wrote:


On Oct 29, 2012, at 10:23 AM, Stefan Holmer wrote:




On Mon, Oct 29, 2012 at 3:53 PM, Xiaoqing Zhu (xiaoqzhu) <xiaoqzhu@cisco.co=
m<mailto:xiaoqzhu@cisco.com>> wrote:
Hi Stefan,

Thanks a lot for your comments. I'll add in more details on the description=
 for the receiver actions.

You are right, we did employ some filtering of the observed per-packet queu=
ing delay, to avoid erratic reactions in the sending rate. The queuing dela=
y employed in the rate calculation in Eq. (1) is actually the time-smoothed=
 averaging d_avg:

d_avg =3D (1-alpha)*d_avg + d_n, where d_n represents the measured queuing =
delay  (end-to-end delay minus minimum observed value) for the n-th packet.=
 The parameter alpha adjusts the level of smoothness in our average process=
. In our experiments, we are using alpha =3D 10e-4.

Thanks. Just to be clear,

d_n =3D receiver_time_n - timestamp_n - min_observed

and min_observed is the smallest d_n seen in some window?

That's right.  BTW, while the time-averaged filtering works well for us, fo=
r now, we are also considering building in more robustness against individu=
al bad samples. The Kalman filtering approach presented in your proposal se=
ems to be a good candidate for that. Maybe there is a way to incorporate th=
at approach here for the delay estimation as well.  What do you think?




Also, it's obvious that Eq.(1) wouldn't hold for for d=3D0, in which case t=
he rate is chosen to be R_max.  As mentioned in the paragraph below Eq. (1)=
, any calculated target rate R_o greater than R_max will also be clipped to=
 the R_max.

Initially, I omitted these details in the draft proposal for brevity. Given=
 your comments and concerns, looks like I'd better put them back in for cla=
rity.

Best,
Xiaoqing


On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:

Hi Xiaoqing,

Interesting draft, I have a few questions related to the queuing delay part=
 of it.

"The role of the receiver is fairly straightforward. It observes and estima=
tes end-to-end queuing delay d and ECN marking ratio p of the stream. The f=
ormer can be obtained from the RTP timestamp provided by the sender."

You don't provide much detail on how the end-to-end queuing delay is estima=
ted from the timestamps. Would it be possible to be more specific? Are you =
employing a filter? Related to that, you seem to be running into a divide b=
y zero in equation 1 if your queuing delay turns out to be zero. It also se=
ems a bit bold to go directly from R_min to R_max if the queuing delay is l=
ow enough.

/Stefan




On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (xiaoqzhu) <xiaoqzhu@cisco.c=
om<mailto:xiaoqzhu@cisco.com>> wrote:
And below is the notification of our submission:

Thanks,
Xiaoqing

--------------------------------------
A new version of I-D, draft-zhu-rmcat-nada-00.txt
has been successfully submitted by Xiaoqing Zhu and posted to the
IETF repository.

Filename:  draft-zhu-rmcat-nada
Revision:  00
Title:  NADA: A Unified Congestion Control Scheme for Real-Time Media
Creation date:  2012-10-14
WG ID:  Individual Submission
Number of pages: 10
URL:             http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-0=
0.txt
Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-00


Abstract:
  This document describes a scheme named network-assisted dynamic
  adaptation (NADA), a novel congestion control approach for
  interactive real-time media applications, such as video conferencing.
  In the proposed scheme, the sender regulates its sending rate based
  on either implicit or explicit congestion signaling, in a unified
  approach. The scheme can reap the benefits of explicit congestion
  notification markings from network nodes. It also maintains
  consistent sender behavior in the absence of such markings, by
  reacting to queuing delays instead.

  We present here the overall system architecture, recommended
  behaviors at the sender and the receiver, as well as expected network
  nodes operations. Results from extensive simulation studies of the
  proposed scheme are available upon request.




The IETF Secretariat
--------------------------------------
On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:

Hi,

On Oct 11, 2012, at 18:45, Xiaoqing Zhu <zhuxq@alumni.stanford.edu<mailto:z=
huxq@alumni.stanford.edu>> wrote:
My colleague and I are working on a congestion control scheme for
real-time conferencing applications at Cisco. A brief version of that
algorithm was presented at the IAB Workshop in Vancouver (paper #12,
"Network-Assisted Dynamic Adaptation (NADA): A Design Summary"). Due
to time constraints, however, we did not get a chance to mention the
delay-based variant of that scheme, which may be a better fit for
rmcat.

We would like to request for a slot to present in Atlanta. We will
also submit the ID draft before Oct 15, for further discussions on
this mailing list.

I look forward to seeing the draft being discussed on the list!

Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is high.

Lars








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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
Oops. Sorry about the typo. &nbsp;Yes, there is the alpha for the second te=
rm.
<div><br>
</div>
<div>Best,</div>
<div>Xiaoqing</div>
<div>&nbsp;&nbsp;<br>
<div>
<div>On Oct 30, 2012, at 5:08 AM, Piers O'Hanlon wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
Hi,
<div><br>
</div>
<div>Presumably by time-smooth average you mean EWMA - so you're missing an=
 alpha?</div>
<div><br>
</div>
<div>
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div style=3D"word-wrap: break-word; ">
<div>d_avg =3D (1-alpha)*d_avg &#43; d_n</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<br>
Should be: &nbsp;d_avg =3D (1-alpha)*d_avg &#43; alpha*d_n</div>
<div><br>
</div>
<div>Piers.</div>
<div><br>
<div>
<div>On 29 Oct 2012, at 15:28, Xiaoqing Zhu (xiaoqzhu) wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
<br>
<div>
<div>On Oct 29, 2012, at 10:23 AM, Stefan Holmer wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite"><br>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Mon, Oct 29, 2012 at 3:53 PM, Xiaoqing Zhu (x=
iaoqzhu)
<span dir=3D"ltr">&lt;<a href=3D"mailto:xiaoqzhu@cisco.com" target=3D"_blan=
k" class=3D"cremed">xiaoqzhu@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin-top: 0px; margin-right: 0=
px; margin-bottom: 0px; margin-left: 0.8ex; border-left-width: 1px; border-=
left-color: rgb(204, 204, 204); border-left-style: solid; padding-left: 1ex=
; position: static; z-index: auto; ">
<div style=3D"word-wrap:break-word">Hi Stefan,&nbsp;
<div><br>
</div>
<div>Thanks a lot for your comments. I'll add in more details on the descri=
ption for the receiver actions.&nbsp;</div>
<div><br>
</div>
<div>You are right, we did employ some filtering of the observed per-packet=
 queuing delay, to avoid erratic reactions in the sending rate. The queuing=
 delay employed in the rate calculation in Eq. (1) is actually the time-smo=
othed averaging d_avg:&nbsp;</div>
<div><br>
</div>
<div>d_avg =3D (1-alpha)*d_avg &#43; d_n, where d_n represents the measured=
 queuing delay &nbsp;(end-to-end delay minus minimum observed value) for th=
e n-th packet. The parameter alpha adjusts the level of smoothness in our a=
verage process. In our experiments, we are
 using alpha =3D 10e-4.&nbsp;</div>
</div>
</blockquote>
<div><br>
</div>
<div>Thanks. Just to be clear,&nbsp;</div>
<div><br>
</div>
<div>d_n =3D receiver_time_n - timestamp_n - min_observed</div>
<div><br>
</div>
<div>and min_observed is the smallest d_n seen in some window?</div>
</div>
</div>
</blockquote>
<div><br>
</div>
That's right. &nbsp;BTW, while the time-averaged filtering works well for u=
s, for now, we are also considering building in more robustness against ind=
ividual bad samples. The Kalman filtering approach presented in your propos=
al seems to be a good candidate for that.
 Maybe there is a way to incorporate that approach here for the delay estim=
ation as well. &nbsp;What do you think? &nbsp;&nbsp;</div>
<div><br>
</div>
<div><br>
<blockquote type=3D"cite">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div><br>
</div>
<div>Also, it's obvious that Eq.(1) wouldn't hold for for d=3D0, in which c=
ase the rate is chosen to be R_max. &nbsp;As mentioned in the paragraph bel=
ow Eq. (1), any calculated target rate R_o greater than R_max will also be =
clipped to the R_max.&nbsp;</div>
<div><br>
</div>
<div>Initially, I omitted these details in the draft proposal for brevity. =
Given your comments and concerns, looks like I'd better put them back in fo=
r clarity.&nbsp;</div>
<div><br>
</div>
<div>Best,</div>
<div>Xiaoqing</div>
<div>
<div class=3D"h5">
<div><br>
</div>
<div><br>
</div>
<div>
<div>
<div>On Oct 29, 2012, at 9:05 AM, Stefan Holmer wrote:</div>
<br>
<blockquote type=3D"cite">Hi Xiaoqing,
<div><br>
</div>
<div>Interesting draft, I have a few questions related to the queuing delay=
 part of it.</div>
<div><br>
</div>
<div>&quot;The role of the receiver is fairly straightforward. It observes =
and estimates end-to-end queuing delay d and ECN marking ratio p of the str=
eam. The former can be obtained from the RTP timestamp provided by the send=
er.&quot;</div>
<div><br>
</div>
<div>You don't provide much detail on how the end-to-end queuing delay is e=
stimated from the timestamps. Would it be possible to be more specific? Are=
 you employing a filter? Related to that, you seem to be running into a div=
ide by zero in equation 1 if your
 queuing delay turns out to be zero. It also seems a bit bold to go directl=
y from R_min to R_max if the queuing delay is low enough.</div>
<div><br>
</div>
<div>/Stefan</div>
<div><br>
</div>
<div><br>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Mon, Oct 15, 2012 at 11:43 PM, Xiaoqing Zhu (=
xiaoqzhu)
<span dir=3D"ltr">&lt;<a href=3D"mailto:xiaoqzhu@cisco.com" target=3D"_blan=
k" class=3D"cremed">xiaoqzhu@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">And below is the notification of our su=
bmission:&nbsp;
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>--------------------------------------<br>
A new version of I-D, draft-zhu-rmcat-nada-00.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span style=3D"white-space:pre-wrap"> </span>&nbsp;draft-zhu-rmcat=
-nada<br>
Revision:<span style=3D"white-space:pre-wrap"> </span>&nbsp;00<br>
Title:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>&nbsp;NADA: A Unified Congestion Control Scheme for Rea=
l-Time Media<br>
Creation date:<span style=3D"white-space:pre-wrap"> </span>&nbsp;2012-10-14=
<br>
WG ID:<span style=3D"white-space:pre-wrap"> </span><span style=3D"white-spa=
ce:pre-wrap"></span>&nbsp;Individual Submission<br>
Number of pages: 10<br>
URL: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<a href=3D"http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-00.tx=
t" target=3D"_blank" class=3D"cremed">http://www.ietf.org/internet-drafts/d=
raft-zhu-rmcat-nada-00.txt</a><br>
Status: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"ht=
tp://datatracker.ietf.org/doc/draft-zhu-rmcat-nada" target=3D"_blank" class=
=3D"cremed">http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools=
.ietf.org/html/draft-zhu-rmcat-nada-00" target=3D"_blank" class=3D"cremed">=
http://tools.ietf.org/html/draft-zhu-rmcat-nada-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp;&nbsp;This document describes a scheme named network-assisted dynamic=
<br>
&nbsp;&nbsp;adaptation (NADA), a novel congestion control approach for<br>
&nbsp;&nbsp;interactive real-time media applications, such as video confere=
ncing.<br>
&nbsp;&nbsp;In the proposed scheme, the sender regulates its sending rate b=
ased<br>
&nbsp;&nbsp;on either implicit or explicit congestion signaling, in a unifi=
ed<br>
&nbsp;&nbsp;approach. The scheme can reap the benefits of explicit congesti=
on<br>
&nbsp;&nbsp;notification markings from network nodes. It also maintains<br>
&nbsp;&nbsp;consistent sender behavior in the absence of such markings, by<=
br>
&nbsp;&nbsp;reacting to queuing delays instead.<br>
<br>
&nbsp;&nbsp;We present here the overall system architecture, recommended<br=
>
&nbsp;&nbsp;behaviors at the sender and the receiver, as well as expected n=
etwork<br>
&nbsp;&nbsp;nodes operations. Results from extensive simulation studies of =
the<br>
&nbsp;&nbsp;proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat</div>
<div>--------------------------------------<br>
<div>
<div>
<div>On Oct 12, 2012, at 1:29 AM, Eggert, Lars wrote:</div>
<br>
</div>
<div>
<div>
<blockquote type=3D"cite">
<div>Hi,<br>
<br>
On Oct 11, 2012, at 18:45, Xiaoqing Zhu &lt;<a href=3D"mailto:zhuxq@alumni.=
stanford.edu" target=3D"_blank" class=3D"cremed">zhuxq@alumni.stanford.edu<=
/a>&gt; wrote:<br>
<blockquote type=3D"cite">My colleague and I are working on a congestion co=
ntrol scheme for<br>
</blockquote>
<blockquote type=3D"cite">real-time conferencing applications at Cisco. A b=
rief version of that<br>
</blockquote>
<blockquote type=3D"cite">algorithm was presented at the IAB Workshop in Va=
ncouver (paper #12,<br>
</blockquote>
<blockquote type=3D"cite">&quot;Network-Assisted Dynamic Adaptation (NADA):=
 A Design Summary&quot;). Due<br>
</blockquote>
<blockquote type=3D"cite">to time constraints, however, we did not get a ch=
ance to mention the<br>
</blockquote>
<blockquote type=3D"cite">delay-based variant of that scheme, which may be =
a better fit for<br>
</blockquote>
<blockquote type=3D"cite">rmcat.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">We would like to request for a slot to present in=
 Atlanta. We will<br>
</blockquote>
<blockquote type=3D"cite">also submit the ID draft before Oct 15, for furth=
er discussions on<br>
</blockquote>
<blockquote type=3D"cite">this mailing list.<br>
</blockquote>
<br>
I look forward to seeing the draft being discussed on the list!<br>
<br>
Also, I'm noting your request for a slot. Because the actual CC mechanisms =
are not our most pressing milestones initially - the requirements and eval =
criteria are - we may decide to focus our meeting time on those other draft=
s, if demand for agenda time is
 high. <br>
<br>
Lars</div>
</blockquote>
</div>
</div>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</body>
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