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Subject: Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02
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On 2014-02-17 14:53, Eggert, Lars wrote:
> Hi,
> 
> we'd like to start the working group last call for 
> draft-ietf-rmcat-cc-requirements-02. Please send your feedback to
> the list.
> 

I have reviewed the draft and here are my feedback on the draft.

1. Abstract:

   This document is derived from draft-jesup-rtp-congestion-reqs
   [I-D.jesup-rtp-congestion-reqs].


I think this can be removed from the abstract. It doesn't appear
relevant at this stage to be in the abstract.

2. Section 1

   ... and since it's consumed in real
   time, data delivered late is useless.

I agree that is commonly useless, but not always. I don't know if we
want to be more precise here and write this as:

and since it's consumed in real
   time, data delivered late is commonly useless.

3. Section 1

   Given that this use case is the focus of this document, use cases
   involving noninteractive media such as YouTube-like video streaming,
   and use cases using multicast/broadcast-type technologies, are out of
   scope.

I wonder if it is appropriate to use the trademark "YouTube" in this
text. I would suggest to remove it.

4. Section 2, Req 1.D:

   Similarly periodic bursty flows such as DASH or proprietary

I would suggest adding a reference to DASH and possibly expand it.

5. Section 2, Req 1.D
            or
            may cause a shift in the location of the bottleneck fir the
            duration of the burst.

Spelling fir/for

5. Section 2, Req 3.A

        A.  If possible, it should also share information and adaptation
            with other non-RTP flows between the same endpoints, such as
            a WebRTC data channel

I would recommend a reference after "WebRTC data channel"

6. Section 2, Req 5.A
        A.  In order to react sufficiently quickly when using RTCP for a
            backchannel, an RTP profile such as AVPF/SAVPF that allows
            sufficiently frequent feedback [RFC4585] MUST be used.

The references and the RTP Profiles don't align well, I would suggest
that this is modified to:

        A.  In order to react sufficiently quickly when using RTCP for a
            backchannel, an RTP profile such as RTP/AVPF [RFC4585] or
            RTP/SAVPF [RFC5124] that allows
            sufficiently frequent feedback MUST be used.


7. Section 2, Req 5

A back channel requirement I don't know if it has been discussed is
the need for back channel congestion control.

8. Section 2, Req 6

        Flows managed by this algorithm and flows competed against at a
        bottleneck may have different DSCP markings depending on the
        type of traffic.  A particular bottleneck or section of the
        network path may or may not honor these markings.

A. This talks about DSCP only. What about flow based QoS? Is the not
included here as differnet network behaviors would require differnet
5-tuples. The RTP streams could still be under common application
control so I wonder if this is relevant?

B. I would ad a reference here to DSCP, and maybe you need to include
the word Diffserv here.

9. Section 2, Req 6.A

        A.  In WebRTC, a division of packets into 4 classes is
            envisioned in order of priority: faster-than-audio, audio,
            video, best-effort, and bulk-transfer.  Typically the flows
            managed by this algorithm would be audio or video in that
            heirarchy, and feedback flows would be faster-than-audio.

This doesn't match what is currently in
draft-dhesikan-tsvwg-rtcweb-qos-05. I wonder if this needs to be
either generalized or reference this document?

10. Section 2.

Another requirement that might be missing and is related to 6.A, but
not the same. It is the possibility to indicate the relative priority
between multiple streams in the same session context. This doesn't
require any QoS, just that the sending application indicates relative
priority and that affects how bit-rate gets distributed. Or has the
discussion been that it will be the higher layers that do this
re-distribution between streams for which one have indication if they
can increase or reduce the stream.


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
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Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From nobody Tue Mar  4 07:40:33 2014
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On 3/3/2014 2:28 PM, Magnus Westerlund wrote:
> On 2014-02-17 14:53, Eggert, Lars wrote:
>> Hi,
>>
>> we'd like to start the working group last call for
>> draft-ietf-rmcat-cc-requirements-02. Please send your feedback to
>> the list.
>>
> I have reviewed the draft and here are my feedback on the draft.

Thanks!  Sorry I can't be there in person (I'd planned to until a couple 
of weeks ago).

Consider my response to be "ok, will do" to everything I don't mention 
below.

>
>
> 7. Section 2, Req 5
>
> A back channel requirement I don't know if it has been discussed is
> the need for back channel congestion control.

This has been discussed, and the draft mentions that RTCP backchannel 
messages may be delayed due to RTCP bandwidth rules, and that bandwidth 
should be limited.  It also mentions that this is a flexible area for 
design and different approaches to backchannels are expected to be 
considered.  I think that's enough to cover it.

>
> 8. Section 2, Req 6
>
>          Flows managed by this algorithm and flows competed against at a
>          bottleneck may have different DSCP markings depending on the
>          type of traffic.  A particular bottleneck or section of the
>          network path may or may not honor these markings.
>
> A. This talks about DSCP only. What about flow based QoS? Is the not
> included here as differnet network behaviors would require differnet
> 5-tuples. The RTP streams could still be under common application
> control so I wonder if this is relevant?

If you want to propose something specific, please do.  "Is the not 
included here" is confusing...  Mostly this is just a warning that we 
can't rely on markings to be honored, even if they're honored on one 
part of the path.


Also s/competed/competing/


>
> 9. Section 2, Req 6.A
>
>          A.  In WebRTC, a division of packets into 4 classes is
>              envisioned in order of priority: faster-than-audio, audio,
>              video, best-effort, and bulk-transfer.  Typically the flows
>              managed by this algorithm would be audio or video in that
>              heirarchy, and feedback flows would be faster-than-audio.
>
> This doesn't match what is currently in
> draft-dhesikan-tsvwg-rtcweb-qos-05. I wonder if this needs to be
> either generalized or reference this document?

We can reference the document and make it align.

>
> 10. Section 2.
>
> Another requirement that might be missing and is related to 6.A, but
> not the same. It is the possibility to indicate the relative priority
> between multiple streams in the same session context. This doesn't
> require any QoS, just that the sending application indicates relative
> priority and that affects how bit-rate gets distributed. Or has the
> discussion been that it will be the higher layers that do this
> re-distribution between streams for which one have indication if they
> can increase or reduce the stream.

See section 3.

Perhaps something there along the lines of "When there are multiple 
streams across the same 5-tuple coordinating their bandwidth use and 
congestion control, it should be possible for the application to control 
the relative split of available bandwidth."

-- 
Randell Jesup -- rjesup a t mozilla d o t com


From nobody Tue Mar  4 07:46:55 2014
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Subject: Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02
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On 2/19/2014 5:25 PM, Michael Welzl wrote:
> Hi,
>
> Sorry if I missed this in Randell's responses to the reviews - but this text is still there and should be fixed IMO:
>
> ***
>     2.   The algorithm must be fair to other flows, both realtime flows
>          (such as other instances of itself), and TCP flows, both long-
>          lived and bursts such as the traffic generated by a typical web
>          browsing session.  Note that 'fair' is a rather hard-to-define
>          term.
>
>          A.  Existing flows at a bottleneck must also be fair to new
>              flows to that bottleneck, and must allow new flows to ramp
>              up to a useful share of the bottleneck bandwidth quickly.
> ***
>
> Being fair to other instances of itself is one issue, being "fair" to TCP is another one. The latter is covered by the text about trying to avoid starvation of TCP, so I suggest that this item here should not bring TCP into the mix and only talk about other instances of itself.
>
> Regarding fairness, I don't think that saying "must be fair" and "Note that 'fair' is hard to define" isn't really useful - we need to be clear about what we mean. (Sub-item A doesn't make the paragraph above it clearer, it just adds a restriction. I have no problem with sub-item A although I think it's unnecessary.)
>
> I once proposed: "Send at roughly the same rate under similar conditions."
> (with more details in a previous email with subject "Fairness", see the list archives).

This is why it's hard.  Fairness is actually larger than what your 
statement covers - it implies how it reacts when one side has (say) a 
10ms RTT and 10Mbps access links and another flow at the bottleneck has 
150ms RTT and 1Mbps.  (Exactly *what* happens in that case is tough to 
define in detail, but it falls under "fairness" and isn't covered by 
your statement.)

And while we have most control over self-fairness, a workable solution 
to the problem here must coexist "fairly" with other traffic which 
includes TCP (and DNS, etc).

> It has sldo been said that we should mandate RTT-independence too. I'm for it.

Mandate as a requirement?  Not sure; I wouldn't throw out an otherwise 
excellent solution that fell down some there in favor of a so-so 
RTT-independent solution.  Goal or a positive point in evaluation?  yes.


-- 
Randell Jesup -- rjesup a t mozilla d o t com


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Subject: Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02
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On 4. mars 2014, at 15:45, Randell Jesup <randell-ietf@jesup.org> wrote:

> On 2/19/2014 5:25 PM, Michael Welzl wrote:
>> Hi,
>>=20
>> Sorry if I missed this in Randell's responses to the reviews - but =
this text is still there and should be fixed IMO:
>>=20
>> ***
>>    2.   The algorithm must be fair to other flows, both realtime =
flows
>>         (such as other instances of itself), and TCP flows, both =
long-
>>         lived and bursts such as the traffic generated by a typical =
web
>>         browsing session.  Note that 'fair' is a rather =
hard-to-define
>>         term.
>>=20
>>         A.  Existing flows at a bottleneck must also be fair to new
>>             flows to that bottleneck, and must allow new flows to =
ramp
>>             up to a useful share of the bottleneck bandwidth quickly.
>> ***
>>=20
>> Being fair to other instances of itself is one issue, being "fair" to =
TCP is another one. The latter is covered by the text about trying to =
avoid starvation of TCP, so I suggest that this item here should not =
bring TCP into the mix and only talk about other instances of itself.
>>=20
>> Regarding fairness, I don't think that saying "must be fair" and =
"Note that 'fair' is hard to define" isn't really useful - we need to be =
clear about what we mean. (Sub-item A doesn't make the paragraph above =
it clearer, it just adds a restriction. I have no problem with sub-item =
A although I think it's unnecessary.)
>>=20
>> I once proposed: "Send at roughly the same rate under similar =
conditions."
>> (with more details in a previous email with subject "Fairness", see =
the list archives).
>=20
> This is why it's hard.  Fairness is actually larger than what your =
statement covers - it implies how it reacts when one side has (say) a =
10ms RTT and 10Mbps access links and another flow at the bottleneck has =
150ms RTT and 1Mbps.  (Exactly *what* happens in that case is tough to =
define in detail, but it falls under "fairness" and isn't covered by =
your statement.)

I'm aware of that  :-)   My point is that a mechanism must *at least* do =
what I request here. If it can't achieve a roughly equal rate under =
completely similar conditions, we deal with a beast that exhibits a =
behavior that we can't do much with.


> And while we have most control over self-fairness, a workable solution =
to the problem here must coexist "fairly" with other traffic which =
includes TCP (and DNS, etc).

Yes, for some definition of that... I think trying to nail the fairness =
with TCP down into a very specific definition is bound to fail. The =
current text with "avoid starving and avoid being starved" seems ok to =
me


>> It has sldo been said that we should mandate RTT-independence too. =
I'm for it.
>=20
> Mandate as a requirement?  Not sure; I wouldn't throw out an otherwise =
excellent solution that fell down some there in favor of a so-so =
RTT-independent solution.  Goal or a positive point in evaluation?  yes.

I think the group should have a discussion about this later today.

Cheers,
Michael


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Subject: Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02
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On 3/4/2014 10:53 AM, Michael Welzl wrote:
> On 4. mars 2014, at 15:45, Randell Jesup <randell-ietf@jesup.org> wrote:
>
>> On 2/19/2014 5:25 PM, Michael Welzl wrote:
>>> I once proposed: "Send at roughly the same rate under similar conditions."
>>> (with more details in a previous email with subject "Fairness", see the list archives).
>> This is why it's hard.  Fairness is actually larger than what your statement covers - it implies how it reacts when one side has (say) a 10ms RTT and 10Mbps access links and another flow at the bottleneck has 150ms RTT and 1Mbps.  (Exactly *what* happens in that case is tough to define in detail, but it falls under "fairness" and isn't covered by your statement.)
> I'm aware of that  :-)   My point is that a mechanism must *at least* do what I request here. If it can't achieve a roughly equal rate under completely similar conditions, we deal with a beast that exhibits a behavior that we can't do much with.

Ok, as a subset of fairness, I can state something along that way (it's 
implied by other statements, but this adds a bit more teeth to it).

>> And while we have most control over self-fairness, a workable solution to the problem here must coexist "fairly" with other traffic which includes TCP (and DNS, etc).
> Yes, for some definition of that... I think trying to nail the fairness with TCP down into a very specific definition is bound to fail. The current text with "avoid starving and avoid being starved" seems ok to me

ok

>>> It has sldo been said that we should mandate RTT-independence too. I'm for it.
>> Mandate as a requirement?  Not sure; I wouldn't throw out an otherwise excellent solution that fell down some there in favor of a so-so RTT-independent solution.  Goal or a positive point in evaluation?  yes.
> I think the group should have a discussion about this later today.

Ok.  I still don't see it as a hard MUST requirement here, but a 
goal/part of evaluation.  Let's see where the discussion goes.


-- 
Randell Jesup -- rjesup a t mozilla d o t com


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Here are my comments on the document.
   1.   The congestion control algorithm must attempt to provide as-low-=0A=
        as-possible-delay transit for real-time traffic while still=0A=
        providing a useful amount of bandwidth=2C even when faced with=0A=
        intermediate bottlenecks and competing flows.
[BA] I am not sure that this requirement is meaningful because of the poten=
tial influence of the competing flows.  For example=2C when TCP is competin=
g withrealtime traffic=2C it will tend to fill the queue to the point of in=
ducingloss=2C and there is not much that the RMCAT algorithm can do about t=
his - the "attempts" will be in vain.  So I would separate the requirement =
into multiple parts=2C one stating goals in the absence of competing traffi=
c=2C and another part talking about goals in the presence of competing flow=
s.=20
        There may be=0A=
        lower limits on the amount of bandwidth that is useful=2C but this=
=0A=
        is largely application-specific and the application may be able=0A=
        to modify or remove flows in order allow some useful flows to=0A=
        get enough bandwidth.=20
[BA] There are situations such as scalable video coding where the "lowerlim=
it" is not application-specific=2C because it relates to the characteristic=
s of the base layer.  So in these situations=2C once you have stopped sendi=
ngextension layers=2C you only have the base layer=2C whose characteristics=
 maynot be flexible because of the relationship between the base and extens=
ionlayers.=20
        A.  It should also handle routing changes and interface changes=0A=
            (WiFi to 3G data=2C etc) which may radically change the=0A=
            bandwidth available=2C and react quickly=2C especially if there=
=0A=
            is a reduction in available bandwidth.
[BA] We need to separate routing and interface changes=2C because routingch=
anges are typically invisible to the application=2C whereas interfacechange=
s will not be.  I would argue that reacting "quickly" to routingtransients =
is a non-goal because this is a regular feature of the Internet=2Cand the m=
ajor requirement is to be resilient against them.   This is oneof the reaso=
ns that Circuit Breakers is likely to be ineffective in practice- shutting =
down communications and requiring manual intervention inresponse to a routi=
ng transient is going to be annoying at best.=20
Interface changes are a distinct issue=2C because information could beprovi=
ded (either through use of bw estimates or information relatingto the inter=
face itself) that could help reset the parameters of thevideo encoder short=
ly after the interface change. =20
        C.  The algorithm must not overreact to short-term bursts (such=0A=
            as web-browsing) which can quickly saturate a local-=0A=
            bottleneck router or link=2C but also clear quickly=2C and=0A=
            should recover quickly when the burst ends.  This is=0A=
            inherently at odds with the need to react quickly-enough to=0A=
            avoid queue buildup.
[BA] It is also at odds with the ability to react quickly to rate adaptatio=
n on wireless networks.=20
        D.  Similarly periodic bursty flows such as DASH or proprietary=0A=
            media streaming algorithms may compete in bursts with the=0A=
            algorithm=2C and may not be adaptive within a burst.  They are=
=0A=
            often are layered on top of TCP.  The algorithm must avoid=0A=
            too much delay buildup during those bursts=2C and quickly=0A=
            recover.  Note that this traffic may on an access link=2C or=0A=
            may cause a shift in the location of the bottleneck fir the=0A=
            duration of the burst.
[BA] I would argue that it is not within the power of the algorithm to avoi=
ddelay buildup due to competing flows.  They are what they are.  The onlyth=
ing RMCAT can do is to decide how it wishes to behave in response.
        A.  If possible=2C it should also share information and adaptation=
=0A=
            with other non-RTP flows between the same endpoints=2C such as=
=0A=
                              a WebRTC data channel=20
[BA] Cooperative adaptation between RTP and SCTP is a tall order=2C particu=
larly since the RMCAT WG is not chartered to work on SCTP congestion contro=
l.=20
   11.  It should be evaluated in how it works both with backbone-router=0A=
        bottlenecks=2C (asymmetric) local-loop bottlenecks=2C and local-lan=
=0A=
        (WiFi/etc) bottlenecks=2C and in competition with varying numbers=
=0A=
        and types of streams (TCP=2C TCP variants in use=2C LEDBAT=0A=
        [I-D.ietf-ledbat-congestion]=2C inflexible VoIP UDP flows).
[BA] WiFi=2C with its (often broken) rate adaptation is going to be a speci=
alchallenge for evaluation=2C because the loss could be due to interference=
 or poor rate adaptation=2C rather than dueto congestion.  I would probably=
 delete this.=20



 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'><div>Here are my comments on the=
 document.</div><div><br></div><div><pre class=3D"newpage" style=3D"font-si=
ze: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: alwa=
ys=3B">   1.   The congestion control algorithm must attempt to provide as-=
low-=0A=
        as-possible-delay transit for real-time traffic while still=0A=
        providing a useful amount of bandwidth=2C even when faced with=0A=
        intermediate bottlenecks and competing flows.</pre><pre class=3D"ne=
wpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B =
page-break-before: always=3B"><br></pre><pre class=3D"newpage" style=3D"fon=
t-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: =
always=3B">[BA] I am not sure that this requirement is meaningful because o=
f the potential influence of the competing flows.  For example=2C when TCP =
is competing with</pre><pre class=3D"newpage" style=3D"font-size: 1em=3B ma=
rgin-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B">realti=
me traffic=2C it will tend to fill the queue to the point of inducing</pre>=
<pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin=
-bottom: 0px=3B page-break-before: always=3B">loss=2C and there is not much=
 that the RMCAT algorithm can do about this - the "attempts" will be in vai=
n.  So <span style=3D"font-size: 12pt=3B font-family: Calibri=2C sans-serif=
=3B">I would separate the requirement into multiple parts=2C one stating go=
als in </span><span style=3D"font-size: 12pt=3B font-family: Calibri=2C san=
s-serif=3B">the absence of competing traffic=2C and another part talking a<=
/span><span style=3D"font-size: 12pt=3B font-family: Calibri=2C sans-serif=
=3B">bout goals in the presence of competing flows. </span></pre><pre class=
=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0=
px=3B page-break-before: always=3B"><span style=3D"font-size: 12pt=3B font-=
family: Calibri=2C sans-serif=3B"><br></span></pre><pre class=3D"newpage" s=
tyle=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-bre=
ak-before: always=3B">        There may be=0A=
        lower limits on the amount of bandwidth that is useful=2C but this=
=0A=
        is largely application-specific and the application may be able=0A=
        to modify or remove flows in order allow some useful flows to=0A=
        get enough bandwidth. </pre><pre class=3D"newpage" style=3D"font-si=
ze: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: alwa=
ys=3B"><br></pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-t=
op: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B">[BA] There a=
re situations such as scalable video coding where the "lower</pre><pre clas=
s=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: =
0px=3B page-break-before: always=3B">limit" is not application-specific=2C =
because it relates to the characteristics of the base layer.  So in these s=
ituations=2C once you have stopped sending</pre><pre class=3D"newpage" styl=
e=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-=
before: always=3B">extension layers=2C you only have the base layer=2C whos=
e characteristics may</pre><pre class=3D"newpage" style=3D"font-size: 1em=
=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B">=
not be flexible because of the relationship between the base and extension<=
/pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B m=
argin-bottom: 0px=3B page-break-before: always=3B">layers. </pre><pre class=
=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0=
px=3B page-break-before: always=3B"><br></pre><pre class=3D"newpage" style=
=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-b=
efore: always=3B">        A.  It should also handle routing changes and int=
erface changes=0A=
            (WiFi to 3G data=2C etc) which may radically change the=0A=
            bandwidth available=2C and react quickly=2C especially if there=
=0A=
            is a reduction in available bandwidth.</pre><pre class=3D"newpa=
ge" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B pag=
e-break-before: always=3B"><br></pre><pre class=3D"newpage" style=3D"font-s=
ize: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: alw=
ays=3B">[BA] We need to separate routing and interface changes=2C because r=
outing</pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0=
px=3B margin-bottom: 0px=3B page-break-before: always=3B">changes are typic=
ally invisible to the application=2C whereas interface</pre><pre class=3D"n=
ewpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B=
 page-break-before: always=3B">changes will not be.  I would argue that rea=
cting "quickly" to routing</pre><pre class=3D"newpage" style=3D"font-size: =
1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=
=3B">transients is a non-goal because this is a regular feature of the Inte=
rnet=2C</pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: =
0px=3B margin-bottom: 0px=3B page-break-before: always=3B">and the major re=
quirement is to be resilient against them.   This is one</pre><pre class=3D=
"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=
=3B page-break-before: always=3B">of the reasons that Circuit Breakers is l=
ikely to be ineffective in practice- shutting down communications and requi=
ring manual intervention in</pre><pre class=3D"newpage" style=3D"font-size:=
 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=
=3B">response to a routing transient is going to be annoying at best. </pre=
><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margi=
n-bottom: 0px=3B page-break-before: always=3B"><br></pre><pre class=3D"newp=
age" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B pa=
ge-break-before: always=3B">Interface changes are a distinct issue=2C becau=
se information could be</pre><pre class=3D"newpage" style=3D"font-size: 1em=
=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B">=
provided (either through use of bw estimates or information relating</pre><=
pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-=
bottom: 0px=3B page-break-before: always=3B">to the interface itself) that =
could help reset the parameters of the</pre><pre class=3D"newpage" style=3D=
"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-befo=
re: always=3B">video encoder shortly after the interface change.  </pre><pr=
e class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bo=
ttom: 0px=3B page-break-before: always=3B"><br></pre><pre class=3D"newpage"=
 style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-b=
reak-before: always=3B">        C.  The algorithm must not overreact to sho=
rt-term bursts (such=0A=
            as web-browsing) which can quickly saturate a local-=0A=
            bottleneck router or link=2C but also clear quickly=2C and=0A=
            should recover quickly when the burst ends.  This is=0A=
            inherently at odds with the need to react quickly-enough to=0A=
            avoid queue buildup.</pre><pre class=3D"newpage" style=3D"font-=
size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: al=
ways=3B"><br></pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margin=
-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B">[BA] It is=
 also at odds with the ability to react quickly to rate adaptation on wirel=
ess networks. </pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margi=
n-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B"><br></pre=
><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margi=
n-bottom: 0px=3B page-break-before: always=3B">        D.  Similarly period=
ic bursty flows such as DASH or proprietary=0A=
            media streaming algorithms may compete in bursts with the=0A=
            algorithm=2C and may not be adaptive within a burst.  They are=
=0A=
            often are layered on top of TCP.  The algorithm must avoid=0A=
            too much delay buildup during those bursts=2C and quickly=0A=
            recover.  Note that this traffic may on an access link=2C or=0A=
            may cause a shift in the location of the bottleneck fir the=0A=
            duration of the burst.</pre><pre class=3D"newpage" style=3D"fon=
t-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: =
always=3B"><br></pre><pre class=3D"newpage" style=3D"font-size: 1em=3B marg=
in-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B">[BA] I w=
ould argue that it is not within the power of the algorithm to avoid</pre><=
pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-=
bottom: 0px=3B page-break-before: always=3B">delay buildup due to competing=
 flows.  They are what they are.  The only</pre><pre class=3D"newpage" styl=
e=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-=
before: always=3B">thing RMCAT can do is to decide how it wishes to behave =
in response.</pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-=
top: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B"><br></pre><=
pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-=
bottom: 0px=3B page-break-before: always=3B">        A.  If possible=2C it =
should also share information and adaptation=0A=
            with other non-RTP flows between the same endpoints=2C such as=
=0A=
</pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B =
margin-bottom: 0px=3B page-break-before: always=3B"><span style=3D"font-siz=
e: 12pt=3B font-family: Calibri=2C sans-serif=3B">                         =
     a WebRTC data channel</span> </pre><pre class=3D"newpage" style=3D"fon=
t-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: =
always=3B"><br></pre><pre class=3D"newpage" style=3D"font-size: 1em=3B marg=
in-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B">[BA] Coo=
perative adaptation between RTP and SCTP is a tall order=2C particularly si=
nce the RMCAT WG is not chartered to work on SCTP congestion control. </pre=
><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margi=
n-bottom: 0px=3B page-break-before: always=3B"><br></pre><pre class=3D"newp=
age" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B pa=
ge-break-before: always=3B">   11.  It should be evaluated in how it works =
both with backbone-router=0A=
        bottlenecks=2C (asymmetric) local-loop bottlenecks=2C and local-lan=
=0A=
        (WiFi/etc) bottlenecks=2C and in competition with varying numbers=
=0A=
        and types of streams (TCP=2C TCP variants in use=2C LEDBAT=0A=
        [<a href=3D"http://tools.ietf.org/html/draft-ietf-rmcat-cc-requirem=
ents-02#ref-I-D.ietf-ledbat-congestion">I-D.ietf-ledbat-congestion</a>]=2C =
inflexible VoIP UDP flows).</pre><pre class=3D"newpage" style=3D"font-size:=
 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=
=3B"><br></pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top=
: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B">[BA] WiFi=2C w=
ith its (often broken) rate adaptation is going to be a specialchallenge <s=
pan style=3D"font-size: 12pt=3B font-family: Calibri=2C sans-serif=3B">for =
evaluation=2C because the loss could be due to interference or poor rate ad=
aptation=2C rather than due</span></pre><pre class=3D"newpage" style=3D"fon=
t-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: =
always=3B"><span style=3D"font-size: 12pt=3B font-family: Calibri=2C sans-s=
erif=3B">to congestion.  I would probably delete this. </span></pre><pre cl=
ass=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom=
: 0px=3B page-break-before: always=3B"><span style=3D"font-size: 12pt=3B fo=
nt-family: Calibri=2C sans-serif=3B"><br></span></pre><pre class=3D"newpage=
" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-=
break-before: always=3B"><span style=3D"font-size: 12pt=3B font-family: Cal=
ibri=2C sans-serif=3B"><br></span></pre><pre class=3D"newpage" style=3D"fon=
t-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=3B page-break-before: =
always=3B"><br></pre><pre class=3D"newpage" style=3D"font-size: 1em=3B marg=
in-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B"><br></pr=
e></div> 		 	   		  </div></body>
</html>=

--_24c0795b-8849-4951-8c7c-170cab7a2d2b_--


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>>>> It has sldo been said that we should mandate RTT-independence too. =
I'm for it.
>>> Mandate as a requirement?  Not sure; I wouldn't throw out an =
otherwise excellent solution that fell down some there in favor of a =
so-so RTT-independent solution.  Goal or a positive point in evaluation? =
 yes.
>> I think the group should have a discussion about this later today.
>=20
> Ok.  I still don't see it as a hard MUST requirement here, but a =
goal/part of evaluation.  Let's see where the discussion goes.

So I'm not so concerned about the "throw out an otherwise excellent =
solution" thing because you can have an excellent solution and *force* =
it to be RTT-independent... it's about changing the controller, e.g. via =
using the RTT itself to take it out of the equation - a pretty simple =
way to do this was described in this paper which is the oldest =
occurrence of this kind of thing I know of:

http://www.icir.org/floyd/papers/phase.pdf
Check out the beginning of page 37 of the pdf.

Cheers,
Michael


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--001a11c219ea50960d04f3d860e9
Content-Type: text/plain; charset=UTF-8

For delay-based CC I have considered the built-in RTT dependency of ack
clocked algorithms a feature. The longer the RTT, the longer it will take
for you to have a chance to do something about the queues you are building
up. Isn't it right that you should be a bit more careful pumping out your
packets in that case?

I guess it boils down to what we mean with fairness... Equal rates or equal
right to build up queues before backing off. Or maybe I've just gotten this
wrong. :)

/Stefan


On Tue, Mar 4, 2014 at 6:04 PM, Michael Welzl <michawe@ifi.uio.no> wrote:

> >>>> It has sldo been said that we should mandate RTT-independence too.
> I'm for it.
> >>> Mandate as a requirement?  Not sure; I wouldn't throw out an otherwise
> excellent solution that fell down some there in favor of a so-so
> RTT-independent solution.  Goal or a positive point in evaluation?  yes.
> >> I think the group should have a discussion about this later today.
> >
> > Ok.  I still don't see it as a hard MUST requirement here, but a
> goal/part of evaluation.  Let's see where the discussion goes.
>
> So I'm not so concerned about the "throw out an otherwise excellent
> solution" thing because you can have an excellent solution and *force* it
> to be RTT-independent... it's about changing the controller, e.g. via using
> the RTT itself to take it out of the equation - a pretty simple way to do
> this was described in this paper which is the oldest occurrence of this
> kind of thing I know of:
>
> http://www.icir.org/floyd/papers/phase.pdf
> Check out the beginning of page 37 of the pdf.
>
> Cheers,
> Michael
>
>

--001a11c219ea50960d04f3d860e9
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">For delay-based CC I have considered the built-in RTT depe=
ndency of ack clocked algorithms a feature. The longer the RTT, the longer =
it will take for you to have a chance to do something about the queues you =
are building up. Isn&#39;t it right that you should be a bit more careful p=
umping out your packets in that case?<div>
<br></div><div>I guess it boils down to what we mean with fairness... Equal=
 rates or equal right to build up queues before backing off. Or maybe I&#39=
;ve just gotten this wrong. :)</div><div><br></div><div><div>/Stefan</div>
</div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">O=
n Tue, Mar 4, 2014 at 6:04 PM, 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"><div class=3D"">&gt;&gt;&gt;&gt; It has sldo=
 been said that we should mandate RTT-independence too. I&#39;m for it.<br>
&gt;&gt;&gt; Mandate as a requirement? =C2=A0Not sure; I wouldn&#39;t throw=
 out an otherwise excellent solution that fell down some there in favor of =
a so-so RTT-independent solution. =C2=A0Goal or a positive point in evaluat=
ion? =C2=A0yes.<br>

&gt;&gt; I think the group should have a discussion about this later today.=
<br>
&gt;<br>
&gt; Ok. =C2=A0I still don&#39;t see it as a hard MUST requirement here, bu=
t a goal/part of evaluation. =C2=A0Let&#39;s see where the discussion goes.=
<br>
<br>
</div>So I&#39;m not so concerned about the &quot;throw out an otherwise ex=
cellent solution&quot; thing because you can have an excellent solution and=
 *force* it to be RTT-independent... it&#39;s about changing the controller=
, e.g. via using the RTT itself to take it out of the equation - a pretty s=
imple way to do this was described in this paper which is the oldest occurr=
ence of this kind of thing I know of:<br>

<br>
<a href=3D"http://www.icir.org/floyd/papers/phase.pdf" target=3D"_blank">ht=
tp://www.icir.org/floyd/papers/phase.pdf</a><br>
Check out the beginning of page 37 of the pdf.<br>
<br>
Cheers,<br>
Michael<br>
<br>
</blockquote></div><br></div>

--001a11c219ea50960d04f3d860e9--


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From: "Eggert, Lars" <lars@netapp.com>
To: "rmcat@ietf.org" <rmcat@ietf.org>
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Hi,

I wanted to put my proposal from the session into writing, so we can =
discuss it here.

I think we've made some good progress on the evaluation criteria and =
scenarios, to where we can begin to discuss the properties of some of =
the proposed solution candidate algorithms (on the list and at the next =
meeting.)

My proposal would be that the proponents of said candidates discuss =
among themselves and with Varun to pick a small number of them for =
initial evaluation. Varun & co-authors will focus their efforts on =
fleshing those out in more detail, and meanwhile, the solution =
proponents begin evaluation of the candidates in (at least) those =
scenarios.

In the next session, we'd begin by having Varun walk the group through =
the details of those scenarios, and then see evaluation results for the =
various candidates.

Because I want to encourage folks to implement and evaluate each others =
work* (instead of just their own proposal), I'd like to "reward" those =
teams that can show results for at least one competitor - in addition to =
their own candidate - with more session time in Toronto.

Thoughts?

Lars

* The reason being that this will result in more directly comparable =
numbers, while at the same time improving the quality of the =
specifications of the various candidates, because we will verify that =
they actually can be implemented based on the textual description.

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On 2014-03-05 15:35, Eggert, Lars wrote:

>
> My proposal would be that the proponents of said candidates discuss
> among themselves and with Varun to pick a small number of them for
> initial evaluation. Varun & co-authors will focus their efforts on
> fleshing those out in more detail, and meanwhile, the solution
> proponents begin evaluation of the candidates in (at least) those
> scenarios.

The proponents may pick small number of the testcases but they must 
select the same test cases.

But I would like to get input and comments on all the test cases on the 
list. And we(=the authors) will try to address the comments and concerns 
in all those test cases in the next version. Otherwise, we run a risk of 
loosing the non-picked testcases as for some reason those were down 
prioritized. I personally didn't have any preference when those were 
written and the indexing of the test case does not show any priority 
either. I would be interesting to know what the proponents are 
considering when selecting the test cases.

 From my experience I can say, if somebody setup the testbed and agree 
on the attribute values specified in the respective test cases, then 
running all the test cases and as small number of them should not make 
much of a difference.

>
> In the next session, we'd begin by having Varun walk the group
> through the details of those scenarios, and then see evaluation
> results for the various candidates.
In this meeting the focus was on the structure and the list of test 
cases but in future we definitely have to discuss the details and 
attribute values in each of the test cases.

BR

Zahed

==================================================
ANM ZAHEDUZZAMAN SARKER


Ericsson AB
Services, Media and Network Features
Laboratoriegränd 11
97128 Luleå, Sweden
Phone +46 10 717 37 43
Fax +46 920 996 21
SMS/MMS +46 76 115 37 43
zaheduzzaman.sarker@ericsson.com
www.ericsson.com

==================================================


From nobody Wed Mar  5 09:41:05 2014
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Wed, 5 Mar 2014 17:40:34 +0000
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Subject: Re: [rmcat] for discussion: candidate solution evaluation
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Hi all,

comments inline.

On Wed, Mar 5, 2014 at 4:21 PM, Zaheduzzaman Sarker
<zaheduzzaman.sarker@ericsson.com> wrote:
>
> On 2014-03-05 15:35, Eggert, Lars wrote:
>> My proposal would be that the proponents of said candidates discuss
>> among themselves and with Varun to pick a small number of them for

Those interested in this should drop me a note at varun@comnet.tkk.fi.

>> initial evaluation. Varun & co-authors will focus their efforts on
>> fleshing those out in more detail, and meanwhile, the solution
>> proponents begin evaluation of the candidates in (at least) those
>> scenarios.
>
>
> The proponents may pick small number of the testcases but they must selec=
t
> the same test cases.

I concur, whichever tests are picked we should make sure that they are
the same.
However, I would expect the proponents to run test with different set
of attributes.
For example, different values of RTT

>
> But I would like to get input and comments on all the test cases on the
> list. And we(=3Dthe authors) will try to address the comments and concern=
s in
> all those test cases in the next version. Otherwise, we run a risk of
> loosing the non-picked testcases as for some reason those were down
> prioritized. I personally didn't have any preference when those were writ=
ten
> and the indexing of the test case does not show any priority either. I wo=
uld
> be interesting to know what the proponents are considering when selecting
> the test cases.
>
> From my experience I can say, if somebody setup the testbed and agree on =
the
> attribute values specified in the respective test cases, then running all
> the test cases and as small number of them should not make much of a
> difference.
>
>
>>
>> In the next session, we'd begin by having Varun walk the group
>> through the details of those scenarios, and then see evaluation
>> results for the various candidates.
>
> In this meeting the focus was on the structure and the list of test cases
> but in future we definitely have to discuss the details and attribute val=
ues
> in each of the test cases.
>

The attribute values are not set in stone, they can be changed. We
just have to make sure that all the proponents use roughly the same
values.

Cheers,
Varun

> BR
>
> Zahed
>
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D
> ANM ZAHEDUZZAMAN SARKER
>
>
> Ericsson AB
> Services, Media and Network Features
> Laboratoriegr=E4nd 11
> 97128 Lule=E5, Sweden
> Phone +46 10 717 37 43
> Fax +46 920 996 21
> SMS/MMS +46 76 115 37 43
> zaheduzzaman.sarker@ericsson.com
> www.ericsson.com
>
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D
>



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


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To: Zaheduzzaman Sarker <zaheduzzaman.sarker@ericsson.com>
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Hi,

On 2014-3-5, at 16:21, Zaheduzzaman Sarker =
<zaheduzzaman.sarker@ericsson.com> wrote:
> The proponents may pick small number of the testcases but they must =
select the same test cases.

yep.

> But I would like to get input and comments on all the test cases on =
the list. And we(=3Dthe authors) will try to address the comments and =
concerns in all those test cases in the next version. Otherwise, we run =
a risk of loosing the non-picked testcases as for some reason those were =
down prioritized. I personally didn't have any preference when those =
were written and the indexing of the test case does not show any =
priority either. I would be interesting to know what the proponents are =
considering when selecting the test cases.

Also agreed. I didn't mean to say that all other activity should stop, =
and we will obviously give session time to our other milestones and =
individual drafts as needed.

Lars

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From: Michael Welzl <michawe@ifi.uio.no>
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Date: Wed, 5 Mar 2014 20:11:38 +0000
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Subject: Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02
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No, you have it right - it's a trade-off. What you say here is indeed =
the (only?) counter-argument for RTT-independence.
We have to weigh the two against each other.

I'm not 100% sure - I mean, your argument *is* valid. I'd still lean =
towards RTT-independence but can live with both decisions.

Cheers,
Michael


On 5. mars 2014, at 09:09, Stefan Holmer <stefan@webrtc.org> wrote:

> For delay-based CC I have considered the built-in RTT dependency of =
ack clocked algorithms a feature. The longer the RTT, the longer it will =
take for you to have a chance to do something about the queues you are =
building up. Isn't it right that you should be a bit more careful =
pumping out your packets in that case?
>=20
> I guess it boils down to what we mean with fairness... Equal rates or =
equal right to build up queues before backing off. Or maybe I've just =
gotten this wrong. :)
>=20
> /Stefan
>=20
>=20
> On Tue, Mar 4, 2014 at 6:04 PM, Michael Welzl <michawe@ifi.uio.no> =
wrote:
> >>>> It has sldo been said that we should mandate RTT-independence =
too. I'm for it.
> >>> Mandate as a requirement?  Not sure; I wouldn't throw out an =
otherwise excellent solution that fell down some there in favor of a =
so-so RTT-independent solution.  Goal or a positive point in evaluation? =
 yes.
> >> I think the group should have a discussion about this later today.
> >
> > Ok.  I still don't see it as a hard MUST requirement here, but a =
goal/part of evaluation.  Let's see where the discussion goes.
>=20
> So I'm not so concerned about the "throw out an otherwise excellent =
solution" thing because you can have an excellent solution and *force* =
it to be RTT-independent... it's about changing the controller, e.g. via =
using the RTT itself to take it out of the equation - a pretty simple =
way to do this was described in this paper which is the oldest =
occurrence of this kind of thing I know of:
>=20
> http://www.icir.org/floyd/papers/phase.pdf
> Check out the beginning of page 37 of the pdf.
>=20
> Cheers,
> Michael
>=20
>=20


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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-line-break: after-white-space;">No, =
you have it right - it's a trade-off. What you say here is indeed the =
(only?) counter-argument for RTT-independence.<div>We have to weigh the =
two against each other.</div><div><br></div><div>I'm not 100% sure - I =
mean, your argument *is* valid. I'd still lean towards RTT-independence =
but can live with both =
decisions.</div><div><br></div><div>Cheers,</div><div>Michael</div><div><b=
r></div><div><br><div><div>On 5. mars 2014, at 09:09, Stefan Holmer =
&lt;<a href=3D"mailto:stefan@webrtc.org">stefan@webrtc.org</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div dir=3D"ltr">For delay-based CC I have considered the =
built-in RTT dependency of ack clocked algorithms a feature. The longer =
the RTT, the longer it will take for you to have a chance to do =
something about the queues you are building up. Isn't it right that you =
should be a bit more careful pumping out your packets in that case?<div>
<br></div><div>I guess it boils down to what we mean with fairness... =
Equal rates or equal right to build up queues before backing off. Or =
maybe I've just gotten this wrong. =
:)</div><div><br></div><div><div>/Stefan</div>
</div></div><div class=3D"gmail_extra"><br><br><div =
class=3D"gmail_quote">On Tue, Mar 4, 2014 at 6:04 PM, 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"><div =
class=3D"">&gt;&gt;&gt;&gt; It has sldo been said that we should mandate =
RTT-independence too. I'm for it.<br>
&gt;&gt;&gt; Mandate as a requirement? &nbsp;Not sure; I wouldn't throw =
out an otherwise excellent solution that fell down some there in favor =
of a so-so RTT-independent solution. &nbsp;Goal or a positive point in =
evaluation? &nbsp;yes.<br>

&gt;&gt; I think the group should have a discussion about this later =
today.<br>
&gt;<br>
&gt; Ok. &nbsp;I still don't see it as a hard MUST requirement here, but =
a goal/part of evaluation. &nbsp;Let's see where the discussion =
goes.<br>
<br>
</div>So I'm not so concerned about the "throw out an otherwise =
excellent solution" thing because you can have an excellent solution and =
*force* it to be RTT-independent... it's about changing the controller, =
e.g. via using the RTT itself to take it out of the equation - a pretty =
simple way to do this was described in this paper which is the oldest =
occurrence of this kind of thing I know of:<br>

<br>
<a href=3D"http://www.icir.org/floyd/papers/phase.pdf" =
target=3D"_blank">http://www.icir.org/floyd/papers/phase.pdf</a><br>
Check out the beginning of page 37 of the pdf.<br>
<br>
Cheers,<br>
Michael<br>
<br>
</blockquote></div><br></div>
</blockquote></div><br></div></body></html>=

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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: Michael Welzl <michawe@ifi.uio.no>, Stefan Holmer <stefan@webrtc.org>
Thread-Topic: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02
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Stefan:

I agree with your argument that the longer the RTT, more careful the flow n=
eeds to be in pumping out the packets. But, why would it necessarily transl=
ate to a lower portion of bandwidth sharing *at steady state*?  Or maybe I =
misunderstood you somewhere?

Thanks,
Xiaoqing



________________________________
From: rmcat [rmcat-bounces@ietf.org] on behalf of Michael Welzl [michawe@if=
i.uio.no]
Sent: Wednesday, March 05, 2014 2:11 PM
To: Stefan Holmer
Cc: Randell Jesup; rmcat WG
Subject: Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02

No, you have it right - it's a trade-off. What you say here is indeed the (=
only?) counter-argument for RTT-independence.
We have to weigh the two against each other.

I'm not 100% sure - I mean, your argument *is* valid. I'd still lean toward=
s RTT-independence but can live with both decisions.

Cheers,
Michael


On 5. mars 2014, at 09:09, Stefan Holmer <stefan@webrtc.org<mailto:stefan@w=
ebrtc.org>> wrote:

For delay-based CC I have considered the built-in RTT dependency of ack clo=
cked algorithms a feature. The longer the RTT, the longer it will take for =
you to have a chance to do something about the queues you are building up. =
Isn't it right that you should be a bit more careful pumping out your packe=
ts in that case?

I guess it boils down to what we mean with fairness... Equal rates or equal=
 right to build up queues before backing off. Or maybe I've just gotten thi=
s wrong. :)

/Stefan


On Tue, Mar 4, 2014 at 6:04 PM, Michael Welzl <michawe@ifi.uio.no<mailto:mi=
chawe@ifi.uio.no>> wrote:
>>>> It has sldo been said that we should mandate RTT-independence too. I'm=
 for it.
>>> Mandate as a requirement?  Not sure; I wouldn't throw out an otherwise =
excellent solution that fell down some there in favor of a so-so RTT-indepe=
ndent solution.  Goal or a positive point in evaluation?  yes.
>> I think the group should have a discussion about this later today.
>
> Ok.  I still don't see it as a hard MUST requirement here, but a goal/par=
t of evaluation.  Let's see where the discussion goes.

So I'm not so concerned about the "throw out an otherwise excellent solutio=
n" thing because you can have an excellent solution and *force* it to be RT=
T-independent... it's about changing the controller, e.g. via using the RTT=
 itself to take it out of the equation - a pretty simple way to do this was=
 described in this paper which is the oldest occurrence of this kind of thi=
ng I know of:

http://www.icir.org/floyd/papers/phase.pdf
Check out the beginning of page 37 of the pdf.

Cheers,
Michael




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<html dir=3D"ltr">
<head>
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1">
<style type=3D"text/css" id=3D"owaParaStyle"></style>
</head>
<body style=3D"word-wrap:break-word" fpstyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">Stefan:&nbsp;
<div><br>
</div>
<div>I agree with your argument that the longer the RTT, more careful the f=
low needs to be in pumping out the packets. But, why would it necessarily t=
ranslate to a lower portion of bandwidth sharing *at steady state*? &nbsp;O=
r maybe I misunderstood you somewhere?</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
<div style=3D"font-family: Times New Roman; color: #000000; font-size: 16px=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF240157" style=3D"direction: ltr;"><font face=3D"Tahoma" si=
ze=3D"2" color=3D"#000000"><b>From:</b> rmcat [rmcat-bounces@ietf.org] on b=
ehalf of Michael Welzl [michawe@ifi.uio.no]<br>
<b>Sent:</b> Wednesday, March 05, 2014 2:11 PM<br>
<b>To:</b> Stefan Holmer<br>
<b>Cc:</b> Randell Jesup; rmcat WG<br>
<b>Subject:</b> Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02<br=
>
</font><br>
</div>
<div></div>
<div>No, you have it right - it's a trade-off. What you say here is indeed =
the (only?) counter-argument for RTT-independence.
<div>We have to weigh the two against each other.</div>
<div><br>
</div>
<div>I'm not 100% sure - I mean, your argument *is* valid. I'd still lean t=
owards RTT-independence but can live with both decisions.</div>
<div><br>
</div>
<div>Cheers,</div>
<div>Michael</div>
<div><br>
</div>
<div><br>
<div>
<div>On 5. mars 2014, at 09:09, Stefan Holmer &lt;<a href=3D"mailto:stefan@=
webrtc.org" target=3D"_blank">stefan@webrtc.org</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div dir=3D"ltr">For delay-based CC I have considered the built-in RTT depe=
ndency of ack clocked algorithms a feature. The longer the RTT, the longer =
it will take for you to have a chance to do something about the queues you =
are building up. Isn't it right that
 you should be a bit more careful pumping out your packets in that case?
<div><br>
</div>
<div>I guess it boils down to what we mean with fairness... Equal rates or =
equal right to build up queues before backing off. Or maybe I've just gotte=
n this wrong. :)</div>
<div><br>
</div>
<div>
<div>/Stefan</div>
</div>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Tue, Mar 4, 2014 at 6:04 PM, Michael Welzl <s=
pan 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:1=
px #ccc solid; padding-left:1ex">
<div class=3D"">&gt;&gt;&gt;&gt; It has sldo been said that we should manda=
te RTT-independence too. I'm for it.<br>
&gt;&gt;&gt; Mandate as a requirement? &nbsp;Not sure; I wouldn't throw out=
 an otherwise excellent solution that fell down some there in favor of a so=
-so RTT-independent solution. &nbsp;Goal or a positive point in evaluation?=
 &nbsp;yes.<br>
&gt;&gt; I think the group should have a discussion about this later today.=
<br>
&gt;<br>
&gt; Ok. &nbsp;I still don't see it as a hard MUST requirement here, but a =
goal/part of evaluation. &nbsp;Let's see where the discussion goes.<br>
<br>
</div>
So I'm not so concerned about the &quot;throw out an otherwise excellent so=
lution&quot; thing because you can have an excellent solution and *force* i=
t to be RTT-independent... it's about changing the controller, e.g. via usi=
ng the RTT itself to take it out of the equation
 - a pretty simple way to do this was described in this paper which is the =
oldest occurrence of this kind of thing I know of:<br>
<br>
<a href=3D"http://www.icir.org/floyd/papers/phase.pdf" target=3D"_blank">ht=
tp://www.icir.org/floyd/papers/phase.pdf</a><br>
Check out the beginning of page 37 of the pdf.<br>
<br>
Cheers,<br>
Michael<br>
<br>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_E7175A8E3DC14048A7D3020E0338C1FA217526B2xmbalnx13ciscoc_--


From nobody Thu Mar  6 01:26:51 2014
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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: "rmcat@ietf.org" <rmcat@ietf.org>, "holmer@google.com" <holmer@google.com>, "l.decicco@poliba.it" <l.decicco@poliba.it>, "mascolo@poliba.it" <mascolo@poliba.it>
Thread-Topic: Comments on draft-alvestrand-rmcat-congestion-02
Thread-Index: Ac85E2PU6TOD6ZoqTRmcFuWj0MVbhg==
Date: Thu, 6 Mar 2014 09:26:38 +0000
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Cc: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, Zaheduzzaman Sarker <zaheduzzaman.sarker@ericsson.com>
Subject: [rmcat] Comments on draft-alvestrand-rmcat-congestion-02
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Hi

Had a look at http://www.ietf.org/proceedings/89/slides/slides-89-rmcat-5.p=
df . I have done some studies around the problem that FTP starves out the m=
edia flow.

The algorithm proposed on page 3 seems to do the job. There is however a ri=
sk that the algo in itself causes self-inflicted congestion in that it incr=
eases a threshold against a measured value(and in the process causes the me=
asured value to increase).=20

I have experienced this behavior with another algorithm that I have worked =
on for a while, the details are not the same as in your algo but I suspect =
that you may get the same problem that I describe. In any case, the solutio=
n that I have tried out which seems to work is to blank the video every onc=
e in a while and sense the one way delay during these periods. If the one w=
ay delay (or RTT) drops consideraby during the blank period  then the thres=
hold (or gamma in your case) should be reduced. I have tried out a one RTT =
blank every 10seconds and it seems to do the job. With that said, it should=
 be possible to solve the problem I outline above.

Another, possibly more complicating matter, which I have not managed to sol=
ve is that the media (video) is eventually starved out by FTP if I stick to=
 reasonable settings in the congestion/flow control. The only way to avoid =
this is to make the algorithm very insensitive to packet loss.=20
I believe (hypothesis) that the root cause to this problem is that while th=
e FTP has an unlimited source bitrate (there are always bytes available for=
 the flow control to send if possible), the media has a limited source bitr=
ate as it is not possible to queue up too many bytes (delay constraint), ad=
d the fact that the media source bitrate can change due to content... The e=
ffect of this is that the media sender must become even more opportunistic =
when it has data to send than FTP.=20
In my implementation I adjust a beta value (similar to Cubic's beta value) =
to control the backoff in case of congestion events. If I set the beta valu=
e for very little backoff then I can avoid that media is starved out by FTP=
* and even get a fair split of the bandwidth. The problem is also that I in=
crease the packet loss rate when the FTP load is removed, this is particula=
rly obvious when I use a CoDel AQM in the bottleneck. I believe that I can =
see that the packet loss rate increases quite considerably on page 5 in you=
r presentation when you switch on AT (right figure), this indicates an aggr=
essive behavior.=20

All this is not a rant against your algorithm, rather it makes me wonder ho=
w reasonable it is with test cases that involves parallel FTP and media ove=
r the same bottleneck. Guess it is Ok to try it out but one can question wh=
at weight it should have when a figure of merit or whatever is computed.=20

Comments welcome, even if you punch a hole in my hypothesis above.=20

* I can also allow more video frames to be queued up in the sender to alway=
s have data to send but that that causes a higher  e2e delay.

/Ingemar


 =20


=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D
Ingemar Johansson  M.Sc.=20
Senior Researcher

Ericsson AB
Wireless Access Networks
Labratoriegr=E4nd 11
971 28, Lule=E5, Sweden
Phone +46-1071 43042
SMS/MMS +46-73 078 3289
ingemar.s.johansson@ericsson.com
www.ericsson.com

"I was gratified to be able to answer promptly, and I did.=20
 I said I didn't know."
Samuel Clemens
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D


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Archived-At: http://mailarchive.ietf.org/arch/msg/rmcat/7E6HtHbtWBKiWKAwp7Xs7gM9tKM
Subject: [rmcat] RTT-independence (Re: WGLC for draft-ietf-rmcat-cc-requirements-02)
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Changing the subject to allow concern separation...

On 03/05/2014 09:30 PM, Xiaoqing Zhu (xiaoqzhu) wrote:
> Stefan:
>
> I agree with your argument that the longer the RTT, more careful the 
> flow needs to be in pumping out the packets. But, why would it 
> necessarily translate to a lower portion of bandwidth sharing *at 
> steady state*?  Or maybe I misunderstood you somewhere?

A longer time to achieve the steady state for long RTT flows is probably 
a more relevant formulation - I think that's inevitable, or close to it. 
When the stabilization time is a significant fraction of the total 
lifetime of the flow, or when it's long compared to the interval between 
changes in the equilibrium, a long-RTT flow will probably experience 
worse performance than a short-RTT flow, even if they are 
RTT-independent at equilibrium.

> Thanks,
> Xiaoqing
>
>
>
> ------------------------------------------------------------------------
> *From:* rmcat [rmcat-bounces@ietf.org] on behalf of Michael Welzl 
> [michawe@ifi.uio.no]
> *Sent:* Wednesday, March 05, 2014 2:11 PM
> *To:* Stefan Holmer
> *Cc:* Randell Jesup; rmcat WG
> *Subject:* Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02
>
> No, you have it right - it's a trade-off. What you say here is indeed 
> the (only?) counter-argument for RTT-independence.
> We have to weigh the two against each other.
>
> I'm not 100% sure - I mean, your argument *is* valid. I'd still lean 
> towards RTT-independence but can live with both decisions.
>
> Cheers,
> Michael
>
>
> On 5. mars 2014, at 09:09, Stefan Holmer <stefan@webrtc.org 
> <mailto:stefan@webrtc.org>> wrote:
>
>> For delay-based CC I have considered the built-in RTT dependency of 
>> ack clocked algorithms a feature. The longer the RTT, the longer it 
>> will take for you to have a chance to do something about the queues 
>> you are building up. Isn't it right that you should be a bit more 
>> careful pumping out your packets in that case?
>>
>> I guess it boils down to what we mean with fairness... Equal rates or 
>> equal right to build up queues before backing off. Or maybe I've just 
>> gotten this wrong. :)
>>
>> /Stefan
>>
>>
>> On Tue, Mar 4, 2014 at 6:04 PM, Michael Welzl <michawe@ifi.uio.no 
>> <mailto:michawe@ifi.uio.no>> wrote:
>>
>>     >>>> It has sldo been said that we should mandate
>>     RTT-independence too. I'm for it.
>>     >>> Mandate as a requirement?  Not sure; I wouldn't throw out an
>>     otherwise excellent solution that fell down some there in favor
>>     of a so-so RTT-independent solution.  Goal or a positive point in
>>     evaluation?  yes.
>>     >> I think the group should have a discussion about this later today.
>>     >
>>     > Ok.  I still don't see it as a hard MUST requirement here, but
>>     a goal/part of evaluation.  Let's see where the discussion goes.
>>
>>     So I'm not so concerned about the "throw out an otherwise
>>     excellent solution" thing because you can have an excellent
>>     solution and *force* it to be RTT-independent... it's about
>>     changing the controller, e.g. via using the RTT itself to take it
>>     out of the equation - a pretty simple way to do this was
>>     described in this paper which is the oldest occurrence of this
>>     kind of thing I know of:
>>
>>     http://www.icir.org/floyd/papers/phase.pdf
>>     Check out the beginning of page 37 of the pdf.
>>
>>     Cheers,
>>     Michael
>>
>>
>


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    <div class="moz-cite-prefix">Changing the subject to allow concern
      separation...<br>
      <br>
      On 03/05/2014 09:30 PM, Xiaoqing Zhu (xiaoqzhu) wrote:<br>
    </div>
    <blockquote
cite="mid:E7175A8E3DC14048A7D3020E0338C1FA217526B2@xmb-aln-x13.cisco.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <style type="text/css" id="owaParaStyle"></style>
      <div style="direction: ltr;font-family: Tahoma;color:
        #000000;font-size: 10pt;">Stefan:&nbsp;
        <div><br>
        </div>
        <div>I agree with your argument that the longer the RTT, more
          careful the flow needs to be in pumping out the packets. But,
          why would it necessarily translate to a lower portion of
          bandwidth sharing *at steady state*? &nbsp;Or maybe I misunderstood
          you somewhere?</div>
      </div>
    </blockquote>
    <br>
    A longer time to achieve the steady state for long RTT flows is
    probably a more relevant formulation - I think that's inevitable, or
    close to it. When the stabilization time is a significant fraction
    of the total lifetime of the flow, or when it's long compared to the
    interval between changes in the equilibrium, a long-RTT flow will
    probably experience worse performance than a short-RTT flow, even if
    they are RTT-independent at equilibrium.<br>
    <br>
    <blockquote
cite="mid:E7175A8E3DC14048A7D3020E0338C1FA217526B2@xmb-aln-x13.cisco.com"
      type="cite">
      <div style="direction: ltr;font-family: Tahoma;color:
        #000000;font-size: 10pt;">
        <div>Thanks,</div>
        <div>Xiaoqing</div>
        <div><br>
        </div>
        <div><br>
        </div>
        <div><br>
          <div style="font-family: Times New Roman; color: #000000;
            font-size: 16px">
            <hr tabindex="-1">
            <div id="divRpF240157" style="direction: ltr;"><font
                face="Tahoma" size="2" color="#000000"><b>From:</b>
                rmcat [<a class="moz-txt-link-abbreviated" href="mailto:rmcat-bounces@ietf.org">rmcat-bounces@ietf.org</a>] on behalf of Michael
                Welzl [<a class="moz-txt-link-abbreviated" href="mailto:michawe@ifi.uio.no">michawe@ifi.uio.no</a>]<br>
                <b>Sent:</b> Wednesday, March 05, 2014 2:11 PM<br>
                <b>To:</b> Stefan Holmer<br>
                <b>Cc:</b> Randell Jesup; rmcat WG<br>
                <b>Subject:</b> Re: [rmcat] WGLC for
                draft-ietf-rmcat-cc-requirements-02<br>
              </font><br>
            </div>
            <div>No, you have it right - it's a trade-off. What you say
              here is indeed the (only?) counter-argument for
              RTT-independence.
              <div>We have to weigh the two against each other.</div>
              <div><br>
              </div>
              <div>I'm not 100% sure - I mean, your argument *is* valid.
                I'd still lean towards RTT-independence but can live
                with both decisions.</div>
              <div><br>
              </div>
              <div>Cheers,</div>
              <div>Michael</div>
              <div><br>
              </div>
              <div><br>
                <div>
                  <div>On 5. mars 2014, at 09:09, Stefan Holmer &lt;<a
                      moz-do-not-send="true"
                      href="mailto:stefan@webrtc.org" target="_blank">stefan@webrtc.org</a>&gt;
                    wrote:</div>
                  <br class="Apple-interchange-newline">
                  <blockquote type="cite">
                    <div dir="ltr">For delay-based CC I have considered
                      the built-in RTT dependency of ack clocked
                      algorithms a feature. The longer the RTT, the
                      longer it will take for you to have a chance to do
                      something about the queues you are building up.
                      Isn't it right that you should be a bit more
                      careful pumping out your packets in that case?
                      <div><br>
                      </div>
                      <div>I guess it boils down to what we mean with
                        fairness... Equal rates or equal right to build
                        up queues before backing off. Or maybe I've just
                        gotten this wrong. :)</div>
                      <div><br>
                      </div>
                      <div>
                        <div>/Stefan</div>
                      </div>
                    </div>
                    <div class="gmail_extra"><br>
                      <br>
                      <div class="gmail_quote">On Tue, Mar 4, 2014 at
                        6:04 PM, Michael Welzl <span dir="ltr">
                          &lt;<a moz-do-not-send="true"
                            href="mailto:michawe@ifi.uio.no"
                            target="_blank">michawe@ifi.uio.no</a>&gt;</span>
                        wrote:<br>
                        <blockquote class="gmail_quote" style="margin:0
                          0 0 .8ex; border-left:1px #ccc solid;
                          padding-left:1ex">
                          <div class="">&gt;&gt;&gt;&gt; It has sldo
                            been said that we should mandate
                            RTT-independence too. I'm for it.<br>
                            &gt;&gt;&gt; Mandate as a requirement? &nbsp;Not
                            sure; I wouldn't throw out an otherwise
                            excellent solution that fell down some there
                            in favor of a so-so RTT-independent
                            solution. &nbsp;Goal or a positive point in
                            evaluation? &nbsp;yes.<br>
                            &gt;&gt; I think the group should have a
                            discussion about this later today.<br>
                            &gt;<br>
                            &gt; Ok. &nbsp;I still don't see it as a hard
                            MUST requirement here, but a goal/part of
                            evaluation. &nbsp;Let's see where the discussion
                            goes.<br>
                            <br>
                          </div>
                          So I'm not so concerned about the "throw out
                          an otherwise excellent solution" thing because
                          you can have an excellent solution and *force*
                          it to be RTT-independent... it's about
                          changing the controller, e.g. via using the
                          RTT itself to take it out of the equation - a
                          pretty simple way to do this was described in
                          this paper which is the oldest occurrence of
                          this kind of thing I know of:<br>
                          <br>
                          <a moz-do-not-send="true"
                            href="http://www.icir.org/floyd/papers/phase.pdf"
                            target="_blank">http://www.icir.org/floyd/papers/phase.pdf</a><br>
                          Check out the beginning of page 37 of the pdf.<br>
                          <br>
                          Cheers,<br>
                          Michael<br>
                          <br>
                        </blockquote>
                      </div>
                      <br>
                    </div>
                  </blockquote>
                </div>
                <br>
              </div>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------030604000207000405030302--


From nobody Sun Mar  9 12:32:39 2014
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Here's my review of draft-ietf-rmcat-cc-requirements-02. I hope it's useful.

Summary:

Content seems just about ready for IETF Last Call (modulo a few nits).
Form needs some work.

Structural:

I feel two sections are missing:

- Term definitions. There’s a pointer to -overview, but many terms are 
not defined there. Nor is it good to have a normative dependency on 
-overview outside of WebRTC, since this is going to be hanging around in 
dependency hell “forever”.

Especially the terms “flow” and “fair” seem to be works of art for this 
memo, but also “starvation”.

- Evaluation considerations. This is mainly to have somewhere to move 
requirement 11 to. Now that we have draft-singh-rmcat-cc-eval adopted as 
a WG document, this may be deleted as “overtaken by events” - but an 
informative reference to that document may be a Good Thing.

The role of bullets and sub-bullets needs to be managed. Most of the 
time, bullets give requirements (except when they don’t), and the 
sub-bullets give additional information to expand on understanding the 
requirements, or further detail about implementation techniques that can 
be used to achieve them. But sometimes not.

Nits:


Section 2

1.C Too many hyphens - web-browsing, local-bottleneck, quickly-enough 
all need to lose their hyphens.

1.D “are often are layered” -> “are often layered”, “may on an access 
link” -> “may be on an access link”, “fir” -> “for”

2. Don’t use the “fair” term. Use “behave reasonably”.

3. Commas - “The algorithm should, where possible, merge”.
Technical: Flows with different Diffserv PHBs may experience different 
congestion, so amount of merge needs to be conservative. DSCP markings 
occur in section 6, but need to be menthined here too.

5. Suggest reformulating the point as “When the streams to be managed 
are all RTP streams, backchannel information needs to be sent via RTCP 
or via header extensions in an RTP channel going in the opposite 
direction.” We then need a sub-bullet noting that there are cases when 
there is no RTP channel in the opposite direction.

6. A WebRTC doesn’t seem to have settled on these 4 classes; 
draft-dhesikan doesn’t. General requirement should be that feedback flow 
has DSCP markings that cause it to travel faster than audio, so that the 
contribution of jitter in the feedback path to estimate errors is as 
small as possible.

8. “As best as possible” is not English. “As far as possible”?
Define “starvation” somewhere.

10. “both if the initial bandwidth is above or below the bottleneck 
bandwidth” looks meaningful, but doesn’t parse grammatically.

10.A. “other problems for user experience” is too abstract. Say “Can 
cause packet loss, which gives other problems for the user experience”.

11 does not belong in requirements. It belongs in evaluation.

12 should relate to bullet 10 - starting a flow and resuming a flow 
should have some of the same characteristics, including how much you 
trust previous estimates of capacity to the same destination.





From nobody Mon Mar 10 03:28:03 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the RTP Media Congestion Avoidance Techniques Working Group of the IETF.

        Title           : Evaluating Congestion Control for Interactive Real-time Media
        Authors         : Varun Singh
                          Joerg Ott
	Filename        : draft-ietf-rmcat-eval-criteria-01.txt
	Pages           : 12
	Date            : 2014-03-10

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 datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rmcat-eval-criteria/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rmcat-eval-criteria-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rmcat-eval-criteria-01


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Mon Mar 10 03:32:59 2014
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From: Varun Singh <vsingh.ietf@gmail.com>
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Hi all,

As per the feedback from the meeting in London, I have removed the
section on test parameters and scenarios. The test parameters section
will be incorporated in the next version of
draft-sarker-rmcat-eval-test.

Cheers,
Varun

On Mon, Mar 10, 2014 at 12:27 PM,  <internet-drafts@ietf.org> wrote:
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the RTP Media Congestion Avoidance Techniques Working Group of the IETF.
>
>         Title           : Evaluating Congestion Control for Interactive Real-time Media
>         Authors         : Varun Singh
>                           Joerg Ott
>         Filename        : draft-ietf-rmcat-eval-criteria-01.txt
>         Pages           : 12
>         Date            : 2014-03-10
>
> 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 datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rmcat-eval-criteria/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rmcat-eval-criteria-01
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rmcat-eval-criteria-01
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>



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


From nobody Mon Mar 10 10:28:22 2014
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References: <3B7006B2-C2A5-472B-BC74-15BD2F4D0208@netapp.com> <6EDFD6CF-D260-4D15-9844-F50299F593B5@ifi.uio.no> <5315F511.1060505@jesup.org> <42D6008D-C8ED-4EB1-81D6-A9FECDD2C368@ifi.uio.no> <5315F93B.2060508@jesup.org> <BC0F15C5-5FEC-461F-8A17-C0E4BB799058@ifi.uio.no> <CAEdus3LS2TeVecSsRq5a0SC3y5T_XQAiq66xXnbwpiHLGDsd9g@mail.gmail.com>, <876FA176-FA9C-4867-BC5A-EDAE58E19A6F@ifi.uio.no> <E7175A8E3DC14048A7D3020E0338C1FA217526B2@xmb-aln-x13.cisco.com>, <531CC124.3000209@alvestrand.no>
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Subject: Re: [rmcat] RTT-independence (Re: WGLC for draft-ietf-rmcat-cc-requirements-02)
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Thanks for the clarification.

Then, looks like we are on the same page?  we want RTT-independence at equi=
librium, and may tolerate RTT-dependence during transit.

Best,
Xiaoqing

________________________________
From: rmcat [rmcat-bounces@ietf.org] on behalf of Harald Alvestrand [harald=
@alvestrand.no]
Sent: Sunday, March 09, 2014 1:29 PM
To: rmcat@ietf.org
Subject: [rmcat] RTT-independence (Re: WGLC for draft-ietf-rmcat-cc-require=
ments-02)

Changing the subject to allow concern separation...

On 03/05/2014 09:30 PM, Xiaoqing Zhu (xiaoqzhu) wrote:
Stefan:

I agree with your argument that the longer the RTT, more careful the flow n=
eeds to be in pumping out the packets. But, why would it necessarily transl=
ate to a lower portion of bandwidth sharing *at steady state*?  Or maybe I =
misunderstood you somewhere?

A longer time to achieve the steady state for long RTT flows is probably a =
more relevant formulation - I think that's inevitable, or close to it. When=
 the stabilization time is a significant fraction of the total lifetime of =
the flow, or when it's long compared to the interval between changes in the=
 equilibrium, a long-RTT flow will probably experience worse performance th=
an a short-RTT flow, even if they are RTT-independent at equilibrium.

Thanks,
Xiaoqing



________________________________
From: rmcat [rmcat-bounces@ietf.org<mailto:rmcat-bounces@ietf.org>] on beha=
lf of Michael Welzl [michawe@ifi.uio.no<mailto:michawe@ifi.uio.no>]
Sent: Wednesday, March 05, 2014 2:11 PM
To: Stefan Holmer
Cc: Randell Jesup; rmcat WG
Subject: Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02

No, you have it right - it's a trade-off. What you say here is indeed the (=
only?) counter-argument for RTT-independence.
We have to weigh the two against each other.

I'm not 100% sure - I mean, your argument *is* valid. I'd still lean toward=
s RTT-independence but can live with both decisions.

Cheers,
Michael


On 5. mars 2014, at 09:09, Stefan Holmer <stefan@webrtc.org<mailto:stefan@w=
ebrtc.org>> wrote:

For delay-based CC I have considered the built-in RTT dependency of ack clo=
cked algorithms a feature. The longer the RTT, the longer it will take for =
you to have a chance to do something about the queues you are building up. =
Isn't it right that you should be a bit more careful pumping out your packe=
ts in that case?

I guess it boils down to what we mean with fairness... Equal rates or equal=
 right to build up queues before backing off. Or maybe I've just gotten thi=
s wrong. :)

/Stefan


On Tue, Mar 4, 2014 at 6:04 PM, Michael Welzl <michawe@ifi.uio.no<mailto:mi=
chawe@ifi.uio.no>> wrote:
>>>> It has sldo been said that we should mandate RTT-independence too. I'm=
 for it.
>>> Mandate as a requirement?  Not sure; I wouldn't throw out an otherwise =
excellent solution that fell down some there in favor of a so-so RTT-indepe=
ndent solution.  Goal or a positive point in evaluation?  yes.
>> I think the group should have a discussion about this later today.
>
> Ok.  I still don't see it as a hard MUST requirement here, but a goal/par=
t of evaluation.  Let's see where the discussion goes.

So I'm not so concerned about the "throw out an otherwise excellent solutio=
n" thing because you can have an excellent solution and *force* it to be RT=
T-independent... it's about changing the controller, e.g. via using the RTT=
 itself to take it out of the equation - a pretty simple way to do this was=
 described in this paper which is the oldest occurrence of this kind of thi=
ng I know of:

http://www.icir.org/floyd/papers/phase.pdf
Check out the beginning of page 37 of the pdf.

Cheers,
Michael





--_000_E7175A8E3DC14048A7D3020E0338C1FA217535F4xmbalnx13ciscoc_
Content-Type: text/html; charset="iso-8859-1"
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<html dir=3D"ltr">
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body bgcolor=3D"#FFFFFF" fpstyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">Thanks for the clarification.&nbsp;
<div><br>
</div>
<div>Then, looks like we are on the same page? &nbsp;we want RTT-independen=
ce at equilibrium, and may tolerate RTT-dependence during transit. &nbsp;</=
div>
<div><br>
</div>
<div>Best,</div>
<div>Xiaoqing</div>
<div><br>
<div style=3D"font-family: Times New Roman; color: #000000; font-size: 16px=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF376273" style=3D"direction: ltr;"><font face=3D"Tahoma" si=
ze=3D"2" color=3D"#000000"><b>From:</b> rmcat [rmcat-bounces@ietf.org] on b=
ehalf of Harald Alvestrand [harald@alvestrand.no]<br>
<b>Sent:</b> Sunday, March 09, 2014 1:29 PM<br>
<b>To:</b> rmcat@ietf.org<br>
<b>Subject:</b> [rmcat] RTT-independence (Re: WGLC for draft-ietf-rmcat-cc-=
requirements-02)<br>
</font><br>
</div>
<div></div>
<div>
<div class=3D"moz-cite-prefix">Changing the subject to allow concern separa=
tion...<br>
<br>
On 03/05/2014 09:30 PM, Xiaoqing Zhu (xiaoqzhu) wrote:<br>
</div>
<blockquote type=3D"cite"><style type=3D"text/css" id=3D"owaParaStyle"></st=
yle>
<div style=3D"direction:ltr; font-family:Tahoma; color:#000000; font-size:1=
0pt">Stefan:&nbsp;
<div><br>
</div>
<div>I agree with your argument that the longer the RTT, more careful the f=
low needs to be in pumping out the packets. But, why would it necessarily t=
ranslate to a lower portion of bandwidth sharing *at steady state*? &nbsp;O=
r maybe I misunderstood you somewhere?</div>
</div>
</blockquote>
<br>
A longer time to achieve the steady state for long RTT flows is probably a =
more relevant formulation - I think that's inevitable, or close to it. When=
 the stabilization time is a significant fraction of the total lifetime of =
the flow, or when it's long compared
 to the interval between changes in the equilibrium, a long-RTT flow will p=
robably experience worse performance than a short-RTT flow, even if they ar=
e RTT-independent at equilibrium.<br>
<br>
<blockquote type=3D"cite">
<div style=3D"direction:ltr; font-family:Tahoma; color:#000000; font-size:1=
0pt">
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
<div style=3D"font-family:Times New Roman; color:#000000; font-size:16px">
<hr tabindex=3D"-1">
<div id=3D"divRpF240157" style=3D"direction:ltr"><font face=3D"Tahoma" size=
=3D"2" color=3D"#000000"><b>From:</b> rmcat [<a class=3D"moz-txt-link-abbre=
viated" href=3D"mailto:rmcat-bounces@ietf.org" target=3D"_blank">rmcat-boun=
ces@ietf.org</a>] on behalf of Michael Welzl [<a class=3D"moz-txt-link-abbr=
eviated" href=3D"mailto:michawe@ifi.uio.no" target=3D"_blank">michawe@ifi.u=
io.no</a>]<br>
<b>Sent:</b> Wednesday, March 05, 2014 2:11 PM<br>
<b>To:</b> Stefan Holmer<br>
<b>Cc:</b> Randell Jesup; rmcat WG<br>
<b>Subject:</b> Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02<br=
>
</font><br>
</div>
<div>No, you have it right - it's a trade-off. What you say here is indeed =
the (only?) counter-argument for RTT-independence.
<div>We have to weigh the two against each other.</div>
<div><br>
</div>
<div>I'm not 100% sure - I mean, your argument *is* valid. I'd still lean t=
owards RTT-independence but can live with both decisions.</div>
<div><br>
</div>
<div>Cheers,</div>
<div>Michael</div>
<div><br>
</div>
<div><br>
<div>
<div>On 5. mars 2014, at 09:09, Stefan Holmer &lt;<a href=3D"mailto:stefan@=
webrtc.org" target=3D"_blank">stefan@webrtc.org</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div dir=3D"ltr">For delay-based CC I have considered the built-in RTT depe=
ndency of ack clocked algorithms a feature. The longer the RTT, the longer =
it will take for you to have a chance to do something about the queues you =
are building up. Isn't it right that
 you should be a bit more careful pumping out your packets in that case?
<div><br>
</div>
<div>I guess it boils down to what we mean with fairness... Equal rates or =
equal right to build up queues before backing off. Or maybe I've just gotte=
n this wrong. :)</div>
<div><br>
</div>
<div>
<div>/Stefan</div>
</div>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Tue, Mar 4, 2014 at 6:04 PM, Michael Welzl <s=
pan 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=0A=
                          0 0 .8ex; border-left:1px #ccc solid; padding-lef=
t:1ex">
<div class=3D"">&gt;&gt;&gt;&gt; It has sldo been said that we should manda=
te RTT-independence too. I'm for it.<br>
&gt;&gt;&gt; Mandate as a requirement? &nbsp;Not sure; I wouldn't throw out=
 an otherwise excellent solution that fell down some there in favor of a so=
-so RTT-independent solution. &nbsp;Goal or a positive point in evaluation?=
 &nbsp;yes.<br>
&gt;&gt; I think the group should have a discussion about this later today.=
<br>
&gt;<br>
&gt; Ok. &nbsp;I still don't see it as a hard MUST requirement here, but a =
goal/part of evaluation. &nbsp;Let's see where the discussion goes.<br>
<br>
</div>
So I'm not so concerned about the &quot;throw out an otherwise excellent so=
lution&quot; thing because you can have an excellent solution and *force* i=
t to be RTT-independent... it's about changing the controller, e.g. via usi=
ng the RTT itself to take it out of the equation
 - a pretty simple way to do this was described in this paper which is the =
oldest occurrence of this kind of thing I know of:<br>
<br>
<a href=3D"http://www.icir.org/floyd/papers/phase.pdf" target=3D"_blank">ht=
tp://www.icir.org/floyd/papers/phase.pdf</a><br>
Check out the beginning of page 37 of the pdf.<br>
<br>
Cheers,<br>
Michael<br>
<br>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<br>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_E7175A8E3DC14048A7D3020E0338C1FA217535F4xmbalnx13ciscoc_--


From nobody Wed Mar 12 06:58:05 2014
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From: Mirja =?iso-8859-1?q?K=FChlewind?= <mirja.kuehlewind@ikr.uni-stuttgart.de>
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Cc: Randell Jesup <randell-ietf@jesup.org>
Subject: Re: [rmcat] RTT-independence (Re: WGLC for draft-ietf-rmcat-cc-requirements-02)
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Hi (as individual contributor),

I'd think RTT-fairness is in general a good thing. The question would be, is 
there in rmcat an requirement for RTT-fairness that is more important than 
for other kinds of traffic? I'd actually say yes, because the quality of my 
call should not depend on the fact that my communication partner sits next 
door or on a different continent. (I know it still might, but hopefully not 
because of a RTT-fairness issue...). 

That's for me the reason why this could/should be listed explicitly in the 
requirement document. Like actually all requirements, RTT-fairness should 
only be a 'should', thus it will not be the doorkeeper to not except a 
candidate. But that's similar for all requirements. We might not be able to 
find a solution that can always fulfill all requirement, thus at the end we 
need to have a discussion which requirements are okay to break.

Mirja



On Monday 10 March 2014 18:28:04 Xiaoqing Zhu (xiaoqzhu) wrote:
> Thanks for the clarification.
>
> Then, looks like we are on the same page?  we want RTT-independence at
> equilibrium, and may tolerate RTT-dependence during transit.
>
> Best,
> Xiaoqing
>
> ________________________________
> From: rmcat [rmcat-bounces@ietf.org] on behalf of Harald Alvestrand
> [harald@alvestrand.no] Sent: Sunday, March 09, 2014 1:29 PM
> To: rmcat@ietf.org
> Subject: [rmcat] RTT-independence (Re: WGLC for
> draft-ietf-rmcat-cc-requirements-02)
>
> Changing the subject to allow concern separation...
>
> On 03/05/2014 09:30 PM, Xiaoqing Zhu (xiaoqzhu) wrote:
> Stefan:
>
> I agree with your argument that the longer the RTT, more careful the flow
> needs to be in pumping out the packets. But, why would it necessarily
> translate to a lower portion of bandwidth sharing *at steady state*?  Or
> maybe I misunderstood you somewhere?
>
> A longer time to achieve the steady state for long RTT flows is probably a
> more relevant formulation - I think that's inevitable, or close to it. When
> the stabilization time is a significant fraction of the total lifetime of
> the flow, or when it's long compared to the interval between changes in the
> equilibrium, a long-RTT flow will probably experience worse performance
> than a short-RTT flow, even if they are RTT-independent at equilibrium.
>
> Thanks,
> Xiaoqing
>
>
>
> ________________________________
> From: rmcat [rmcat-bounces@ietf.org<mailto:rmcat-bounces@ietf.org>] on
> behalf of Michael Welzl [michawe@ifi.uio.no<mailto:michawe@ifi.uio.no>]
> Sent: Wednesday, March 05, 2014 2:11 PM
> To: Stefan Holmer
> Cc: Randell Jesup; rmcat WG
> Subject: Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02
>
> No, you have it right - it's a trade-off. What you say here is indeed the
> (only?) counter-argument for RTT-independence. We have to weigh the two
> against each other.
>
> I'm not 100% sure - I mean, your argument *is* valid. I'd still lean
> towards RTT-independence but can live with both decisions.
>
> Cheers,
> Michael
>
>
> On 5. mars 2014, at 09:09, Stefan Holmer
> <stefan@webrtc.org<mailto:stefan@webrtc.org>> wrote:
>
> For delay-based CC I have considered the built-in RTT dependency of ack
> clocked algorithms a feature. The longer the RTT, the longer it will take
> for you to have a chance to do something about the queues you are building
> up. Isn't it right that you should be a bit more careful pumping out your
> packets in that case?
>
> I guess it boils down to what we mean with fairness... Equal rates or equal
> right to build up queues before backing off. Or maybe I've just gotten this
> wrong. :)
>
> /Stefan
>
> On Tue, Mar 4, 2014 at 6:04 PM, Michael Welzl 
<michawe@ifi.uio.no<mailto:michawe@ifi.uio.no>> wrote:
> >>>> It has sldo been said that we should mandate RTT-independence too. I'm
> >>>> for it.
> >>>
> >>> Mandate as a requirement?  Not sure; I wouldn't throw out an otherwise
> >>> excellent solution that fell down some there in favor of a so-so
> >>> RTT-independent solution.  Goal or a positive point in evaluation? 
> >>> yes.
> >>
> >> I think the group should have a discussion about this later today.
> >
> > Ok.  I still don't see it as a hard MUST requirement here, but a
> > goal/part of evaluation.  Let's see where the discussion goes.
>
> So I'm not so concerned about the "throw out an otherwise excellent
> solution" thing because you can have an excellent solution and *force* it
> to be RTT-independent... it's about changing the controller, e.g. via using
> the RTT itself to take it out of the equation - a pretty simple way to do
> this was described in this paper which is the oldest occurrence of this
> kind of thing I know of:
>
> http://www.icir.org/floyd/papers/phase.pdf
> Check out the beginning of page 37 of the pdf.
>
> Cheers,
> Michael



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Hi (as individual contributor),

find below some comments. I might repeat some of the comment that have already 
be made, but I also think those points are important. 

A few general comments:
1) Would be helpful to spell out most of the acronyms and give the respective 
references as Magnus already mentioned.
2) I know you told me that the order does not have a specific meaning, but it 
might still be useful the rethink the order to e.g. group more related points 
together. Some more concrete remarks below.

More detailed comments:

Abstract:
I think the scope of the document is just right but the abstract does not 
really reflect the content very well. Let me just try to propose a different 
abstract:

"Due to an increasing amount of (RTP-based) the real-time media traffic in 
today's Internet (e.g. with the introduction of RTCweb [ref]), it is 
especially important to ensure that this kind of traffic is congestion 
controlled to provide a proper capacity sharing between different flows and 
avoid a congestion collapse. As the requirements for interactive, 
point-to-point real time multimedia are quite different from the requirements 
for traditional congestion control as developed for bulk transfers like FTP 
or bursty transfers like web, this document describes a set of requirements 
that can be used to design and evaluate such new RTP Media Congestion 
Avoidance Techniques (rmcat)."

1. Intro
s/The traditional TCP congestion control requirements were developed in
   order to promote efficient use of the Internet for reliable bulk
   transfer of non-time-critical data/
   Most of today's TCP congestion control schemes were developed with a focus 		 
   on an use of the Internet for reliable bulk transfer of non-time-critical 
   data/

100s -> ITU-T names 150ms in G.114

As already said by a couple of reviewers, please don't only point to 
rtcweb-overview but also list the relevant terminology (because this is rmcat 
and not rtcweb wg).

2. Requirements

Req 1) 
- This seems to be two requirements in one: as-low-as-possible delay and 
useful amount of bandwidth. While the useful amount of bandwidth req seems 
also be part of Req 2 and 8... maybe split up or have separate subbullets on 
these two things
- Don't really get the different between 1C and 1D...?
- s/They are often are layered/They are often layered/

Req 2)
- I agree with other reviewers that fair is not a good word here. I would 
instead say 'must be able to get a proper share of the bandwidth'.
- Then one could merge Req 8 as one point into a new requirement 2B ( or at 
least req 2 and req 8 should be listed right next to each other).

Req 3)
s/between the same endpoints/that share a common bottleneck/

Req 4) 
please spell out ECN and give a ref

Req 5)
Not sure what is actually a requirement in 5 or what are just some explanation 
on RTP. Maybe you add the assumption that we talk about RTP traffic and stuff 
that needs to be know about RTP/RTCP in the Introduction (incl. respective 
references) and make the requirements a little more clear?

Req 6)
Do not really understand what the actually requirement is -> there should be 
a 'should' somewhere....

Req 7)
no comments :-)

Req 8)
said this already: merge with 2 or move below 2

Req 9)
could potentially be merged with 4 or moved below 4

Req 10)
This is an important point which I'm, coming from CC design, didn't think 
about at the beginning. I would list this requirement right after req 2 as 
there are also some similarities.

Req 11)
Please remove, there shouldn't be a requirement for CC design that the design 
should be evaluated based on this requirement... You could add this text in 
the intro but as we have an own doc on evaluation I don't think this point is 
needed here.

Req 12)
Isn't this maybe a subbullet of 10?

4. Security
Not sure if we actually need this text here, because there is nothing specific 
for rmcat here (that's all true for cc in general) and if we really need a 
text like this, it probably should be part of the candidate document. There 
doesn't really need to be security considerations in an requirements 
document. If you really, really want to say something, than just say, that 
requiring cc for rt media is a very important for the stability of the 
Internet.

Mirja


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Hi,

I've uploaded the minutes:

http://www.ietf.org/proceedings/89/minutes/minutes-89-rmcat

Please check and send me an email if you think there are changes needed (as 
something that were said was not reflected correctly)!

There are also some names missing in the minutes, so if you have said 
something on the mic, go and have a look at the minutes, please!

Thanks,
Mirja



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Please join me in welcoming Karen Nielsen as new co-chair for RMCAT, 
joining Lars and Mirja.

Thank you, Karen, for being willing to serve the community in this way.

Spencer, as AD


From nobody Wed Mar 12 11:21:48 2014
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Welcome Karen, and thanks to Mirja for her services!


On 12. mars 2014, at 16:42, Spencer Dawkins =
<spencerdawkins.ietf@gmail.com> wrote:

> Please join me in welcoming Karen Nielsen as new co-chair for RMCAT, =
joining Lars and Mirja.
>=20
> Thank you, Karen, for being willing to serve the community in this =
way.
>=20
> Spencer, as AD
>=20


From nobody Wed Mar 12 11:31:31 2014
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ARGH - SORRY!!!!!

That's the result of reading real quick - my apologies to Mirja, very =
sorry for that!

I didn't know groups can have 3 chairs and I just read too fast and =
thought one of you must be stepping down when a chair is filled, I'm =
really really sorry for that.


On 12. mars 2014, at 19:21, Michael Welzl <michawe@ifi.uio.no> wrote:

> Welcome Karen, and thanks to Mirja for her services!
>=20
>=20
> On 12. mars 2014, at 16:42, Spencer Dawkins =
<spencerdawkins.ietf@gmail.com> wrote:
>=20
>> Please join me in welcoming Karen Nielsen as new co-chair for RMCAT, =
joining Lars and Mirja.
>>=20
>> Thank you, Karen, for being willing to serve the community in this =
way.
>>=20
>> Spencer, as AD
>>=20
>=20


From nobody Thu Mar 13 01:50:43 2014
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From: Erlendur Karlsson <erlendur.karlsson@ericsson.com>
To: "rmcat@ietf.org" <rmcat@ietf.org>
Thread-Topic: Pdf document formatting problems
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Date: Thu, 13 Mar 2014 08:49:59 +0000
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Subject: [rmcat] Pdf document formatting problems
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Hi,

The pdf versions of the documents
draft-sarker-rmcat-eval-test-00<http://datatracker.ietf.org/doc/draft-sarke=
r-rmcat-eval-test/> and draft-ietf-rmcat-cc-requirements-02<http://datatrac=
ker.ietf.org/doc/draft-ietf-rmcat-cc-requirements/>,
seem to be formatted in such a way that a document page is split onto two p=
ages.

It would be greatly appreciated if the pdf formatting of these documents co=
uld be fixed.

Br,
Erlendur


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osoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" xmlns=3D"http:=
//www.w3.org/TR/REC-html40">
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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The pdf versions of the documents<o:p></o:p></p>
<p class=3D"MsoNormal"><a href=3D"http://datatracker.ietf.org/doc/draft-sar=
ker-rmcat-eval-test/"><span style=3D"font-size:10.0pt;font-family:&quot;Ari=
al&quot;,&quot;sans-serif&quot;;background:white">draft-sarker-rmcat-eval-t=
est-00</span></a> and
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rmcat-cc-requirements=
/"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans=
-serif&quot;;background:#EDF5FF">draft-ietf-rmcat-cc-requirements-02</span>=
</a>,
<o:p></o:p></p>
<p class=3D"MsoNormal">seem to be formatted in such a way that a document p=
age is split onto two pages.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">It would be greatly appreciated if the pdf formattin=
g of these documents could be fixed. &nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Br,<o:p></o:p></p>
<p class=3D"MsoNormal">Erlendur<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

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From: "Eggert, Lars" <lars@netapp.com>
To: Erlendur Karlsson <erlendur.karlsson@ericsson.com>
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On 2014-3-13, at 9:49, Erlendur Karlsson =
<erlendur.karlsson@ericsson.com> wrote:
> The pdf versions of the documents
> draft-sarker-rmcat-eval-test-00 and =
draft-ietf-rmcat-cc-requirements-02,
> seem to be formatted in such a way that a document page is split onto =
two pages.

http://tools.ietf.org/pdf/draft-sarker-rmcat-eval-test-00.pdf and
http://tools.ietf.org/pdf/draft-ietf-rmcat-cc-requirements-02.pdf look =
fine here?

Lars


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From: Erlendur Karlsson <erlendur.karlsson@ericsson.com>
To: "Eggert, Lars" <lars@netapp.com>
Thread-Topic: [rmcat] Pdf document formatting problems
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In both of them I get the foot-line for the first page on top of the second=
 page, which is empty except for that line. This line offset evolves for th=
e following pages. While for other documents such as http://datatracker.iet=
f.org/doc/draft-ietf-rmcat-eval-criteria/ I have no problems with the pdf f=
ile, the foot-lines for all the pages are exactly where they should. It see=
ms they are formatted differently.

Br,
Erlendur


-----Original Message-----
From: Eggert, Lars [mailto:lars@netapp.com]=20
Sent: den 13 mars 2014 10:37
To: Erlendur Karlsson
Cc: rmcat@ietf.org
Subject: Re: [rmcat] Pdf document formatting problems

On 2014-3-13, at 9:49, Erlendur Karlsson <erlendur.karlsson@ericsson.com> w=
rote:
> The pdf versions of the documents
> draft-sarker-rmcat-eval-test-00 and=20
> draft-ietf-rmcat-cc-requirements-02,
> seem to be formatted in such a way that a document page is split onto two=
 pages.

http://tools.ietf.org/pdf/draft-sarker-rmcat-eval-test-00.pdf and http://to=
ols.ietf.org/pdf/draft-ietf-rmcat-cc-requirements-02.pdf look fine here?

Lars


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Are you printing them or viewing them on screen?

If the former, I suspect a page size issue.

Lars

On 2014-3-13, at 13:12, Erlendur Karlsson =
<erlendur.karlsson@ericsson.com> wrote:

> In both of them I get the foot-line for the first page on top of the =
second page, which is empty except for that line. This line offset =
evolves for the following pages. While for other documents such as =
http://datatracker.ietf.org/doc/draft-ietf-rmcat-eval-criteria/ I have =
no problems with the pdf file, the foot-lines for all the pages are =
exactly where they should. It seems they are formatted differently.
>=20
> Br,
> Erlendur
>=20
>=20
> -----Original Message-----
> From: Eggert, Lars [mailto:lars@netapp.com]=20
> Sent: den 13 mars 2014 10:37
> To: Erlendur Karlsson
> Cc: rmcat@ietf.org
> Subject: Re: [rmcat] Pdf document formatting problems
>=20
> On 2014-3-13, at 9:49, Erlendur Karlsson =
<erlendur.karlsson@ericsson.com> wrote:
>> The pdf versions of the documents
>> draft-sarker-rmcat-eval-test-00 and=20
>> draft-ietf-rmcat-cc-requirements-02,
>> seem to be formatted in such a way that a document page is split onto =
two pages.
>=20
> http://tools.ietf.org/pdf/draft-sarker-rmcat-eval-test-00.pdf and =
http://tools.ietf.org/pdf/draft-ietf-rmcat-cc-requirements-02.pdf look =
fine here?
>=20
> Lars
>=20


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From: Erlendur Karlsson <erlendur.karlsson@ericsson.com>
To: "Eggert, Lars" <lars@netapp.com>
Thread-Topic: [rmcat] Pdf document formatting problems
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I am viewing them on the screen and it does not matter if it is in a Chrome=
 browser or Acrobat Reader.=20

Br,
Erlendur

-----Original Message-----
From: rmcat [mailto:rmcat-bounces@ietf.org] On Behalf Of Eggert, Lars
Sent: den 13 mars 2014 14:02
To: Erlendur Karlsson
Cc: rmcat@ietf.org
Subject: Re: [rmcat] Pdf document formatting problems

Are you printing them or viewing them on screen?

If the former, I suspect a page size issue.

Lars

On 2014-3-13, at 13:12, Erlendur Karlsson <erlendur.karlsson@ericsson.com> =
wrote:

> In both of them I get the foot-line for the first page on top of the seco=
nd page, which is empty except for that line. This line offset evolves for =
the following pages. While for other documents such as http://datatracker.i=
etf.org/doc/draft-ietf-rmcat-eval-criteria/ I have no problems with the pdf=
 file, the foot-lines for all the pages are exactly where they should. It s=
eems they are formatted differently.
>=20
> Br,
> Erlendur
>=20
>=20
> -----Original Message-----
> From: Eggert, Lars [mailto:lars@netapp.com]=20
> Sent: den 13 mars 2014 10:37
> To: Erlendur Karlsson
> Cc: rmcat@ietf.org
> Subject: Re: [rmcat] Pdf document formatting problems
>=20
> On 2014-3-13, at 9:49, Erlendur Karlsson <erlendur.karlsson@ericsson.com>=
 wrote:
>> The pdf versions of the documents
>> draft-sarker-rmcat-eval-test-00 and=20
>> draft-ietf-rmcat-cc-requirements-02,
>> seem to be formatted in such a way that a document page is split onto tw=
o pages.
>=20
> http://tools.ietf.org/pdf/draft-sarker-rmcat-eval-test-00.pdf and http://=
tools.ietf.org/pdf/draft-ietf-rmcat-cc-requirements-02.pdf look fine here?
>=20
> Lars
>=20


From nobody Thu Mar 13 09:52:41 2014
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Date: Thu, 13 Mar 2014 17:52:28 +0100
From: Stefan Alfredsson <Stefan.Alfredsson@kau.se>
To: "Eggert, Lars" <lars@netapp.com>
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Subject: Re: [rmcat] Pdf document formatting problems
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* Quoting Eggert, Lars <lars@netapp.com> [13 Mar-14 13:01]:
> Are you printing them or viewing them on screen?
> 
> If the former, I suspect a page size issue.

I became very curious of a PDF document that would *reformat*
in different viewers (xpdf showed blank pages, but chrome was OK).

However, the discrepancy is caused by a stale web cache:

$ wget 'http://tools.ietf.org/pdf/draft-sarker-rmcat-eval-test-00.pdf'
Connecting to tools.ietf.org (tools.ietf.org)|77.72.230.30|:80...
Length: 76566 (75K) [application/pdf]

### wait for it...

$ wget 'http://tools.ietf.org/pdf/draft-sarker-rmcat-eval-test-00.pdf'
 Connecting to tools.ietf.org (tools.ietf.org)|64.170.98.42|:80...
Length: 91706 (90K) [application/pdf]


(or you can do a couple of shift-reload in Chrome)


The PDF served from 77.72.230.30 has bad formatting,
but the PDF from 64.170.98.42 is fine.
As indicated above, they have different sizes and are
also created on different days:

$ pdfinfo draft-sarker-rmcat-eval-test-00.pdf | grep Creation
CreationDate:   Thu Feb 20 02:21:32 2014

$ pdfinfo draft-sarker-rmcat-eval-test-00.pdf.1 | grep Creation
CreationDate:   Wed Feb 19 13:38:50 2014

I'm CCing the tools-webmaster so they can investigate.

/Stefan

-- 
Stefan Alfredsson, PhD         Tel: +46 (0) 54 700 1668
Datavetenskap                  Kontor: 21F-425 (Hus Vanern)
Karlstads universitet          PGP 0xB19B4B16
SE-651 88 Karlstad             http://www.cs.kau.se/stefalfr/


From nobody Thu Mar 13 21:00:06 2014
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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: "rmcat@ietf.org" <rmcat@ietf.org>
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Hi:=0A=
=0A=
The main changes in this version are as follows:=0A=
=0A=
a)  Added Sec 5.4 to discuss some of the issue related to using different f=
orms of delay as congestion indication. Current implementation of NADA reli=
es on absolute one-way delay (OWD), but that's only reliable in controlled =
environments between senders and receivers with well-synchronized clocks. I=
n this section, we also discuss alternative delay signals NADA can use (rel=
ative OWD or RTT), and the pros and cons of each approach.  Thanks to earli=
er discussions with Pal and Michael W on the mailing list for raising this =
issue.=0A=
=0A=
b) Fixed various typos in notations in Sec 3 (System Model) so that they no=
w match to the system diagrams and later descriptions in Sec 6 (Sender Beha=
vior). =0A=
=0A=
c) Slightly revise the text in Sec 8 (Implementation Status) to reflect the=
 fact that we've obtained first set of testbed evaluation results (not too =
interesting yet, stay tuned ...)=0A=
=0A=
As always, comments are welcome. =0A=
=0A=
Thanks,=0A=
Xiaoqing=0A=
=0A=
________________________________________=0A=
From: internet-drafts@ietf.org [internet-drafts@ietf.org]=0A=
Sent: Thursday, March 13, 2014 10:47 PM=0A=
To: Rong Pan (ropan); Xiaoqing Zhu (xiaoqzhu); Rong Pan (ropan); Xiaoqing Z=
hu (xiaoqzhu)=0A=
Subject: New Version Notification for draft-zhu-rmcat-nada-03.txt=0A=
=0A=
A new version of I-D, draft-zhu-rmcat-nada-03.txt=0A=
has been successfully submitted by Xiaoqing Zhu and posted to the=0A=
IETF repository.=0A=
=0A=
Name:           draft-zhu-rmcat-nada=0A=
Revision:       03=0A=
Title:          NADA: A Unified Congestion Control Scheme for Real-Time Med=
ia=0A=
Document date:  2014-03-13=0A=
Group:          Individual Submission=0A=
Pages:          13=0A=
URL:            http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-03=
.txt=0A=
Status:         https://datatracker.ietf.org/doc/draft-zhu-rmcat-nada/=0A=
Htmlized:       http://tools.ietf.org/html/draft-zhu-rmcat-nada-03=0A=
Diff:           http://www.ietf.org/rfcdiff?url2=3Ddraft-zhu-rmcat-nada-03=
=0A=
=0A=
Abstract:=0A=
   This document describes a scheme named network-assisted dynamic=0A=
   adaptation (NADA), a novel congestion control approach for=0A=
   interactive real-time media applications, such as video conferencing.=0A=
   In the proposed scheme, the sender regulates its sending rate based=0A=
   on either implicit or explicit congestion signaling, in a unified=0A=
   approach. The scheme can benefit from explicit congestion=0A=
   notification (ECN) markings from network nodes. It also maintains=0A=
   consistent sender behavior in the absence of such markings, by=0A=
   reacting to queuing delays and packet losses instead.=0A=
=0A=
   We present here the overall system architecture, recommended=0A=
   behaviors at the sender and the receiver, as well as expected network=0A=
   node operations. Results from extensive simulation studies of the=0A=
   proposed scheme are available upon request.=0A=
=0A=
=0A=
=0A=
=0A=
Please note that it may take a couple of minutes from the time of submissio=
n=0A=
until the htmlized version and diff are available at tools.ietf.org.=0A=
=0A=
The IETF Secretariat=0A=
=0A=


From nobody Fri Mar 14 02:30:42 2014
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Subject: Re: [rmcat] Pdf document formatting problems
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On 2014-03-13 17:52 Stefan Alfredsson said the following:
> * Quoting Eggert, Lars <lars@netapp.com> [13 Mar-14 13:01]:
>> Are you printing them or viewing them on screen?
>> 
>> If the former, I suspect a page size issue.
> 
> I became very curious of a PDF document that would *reformat*
> in different viewers (xpdf showed blank pages, but chrome was OK).
> 
> However, the discrepancy is caused by a stale web cache:
> 
> $ wget 'http://tools.ietf.org/pdf/draft-sarker-rmcat-eval-test-00.pdf'
> Connecting to tools.ietf.org (tools.ietf.org)|77.72.230.30|:80...
> Length: 76566 (75K) [application/pdf]
> 
> ### wait for it...
> 
> $ wget 'http://tools.ietf.org/pdf/draft-sarker-rmcat-eval-test-00.pdf'
>  Connecting to tools.ietf.org (tools.ietf.org)|64.170.98.42|:80...
> Length: 91706 (90K) [application/pdf]
> 
> 
> (or you can do a couple of shift-reload in Chrome)
> 
> 
> The PDF served from 77.72.230.30 has bad formatting,
> but the PDF from 64.170.98.42 is fine.
> As indicated above, they have different sizes and are
> also created on different days:
> 
> $ pdfinfo draft-sarker-rmcat-eval-test-00.pdf | grep Creation
> CreationDate:   Thu Feb 20 02:21:32 2014
> 
> $ pdfinfo draft-sarker-rmcat-eval-test-00.pdf.1 | grep Creation
> CreationDate:   Wed Feb 19 13:38:50 2014
> 
> I'm CCing the tools-webmaster so they can investigate.

The issue should be fixed now.  The underlying issue with the flawed generation
was fixed already about a month ago, but I'd neglected to clean out the cached
files on at least one of the tools servers.

Thanks for the analysis and alert!


Best regards,

	Henrik


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Subject: [rmcat] ECN marking for webrtc and stun packets?
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I am curious if anyone has outlined an approach to CC for
webrtc (rtp, stun, etc) based on more fully utilizing ECN for
it?

The only thing that I see that sort of touches upon it was this:

http://www.tschofenig.priv.at/cc-workshop/draft-arkko-wscc-deployment-00.tx=
t



--=20
Dave T=E4ht


From nobody Fri Mar 14 05:56:20 2014
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From: "Eggert, Lars" <lars@netapp.com>
To: Dave Taht <dave.taht@gmail.com>
Thread-Topic: [rmcat] ECN marking for webrtc and stun packets?
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Subject: Re: [rmcat] ECN marking for webrtc and stun packets?
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Hi,

there is also RFC6679, which deals with RTP but not (necessarily) in the =
context of webrtc.

Lars

On 2014-3-14, at 13:39, Dave Taht <dave.taht@gmail.com> wrote:

> I am curious if anyone has outlined an approach to CC for
> webrtc (rtp, stun, etc) based on more fully utilizing ECN for
> it?
>=20
> The only thing that I see that sort of touches upon it was this:
>=20
> =
http://www.tschofenig.priv.at/cc-workshop/draft-arkko-wscc-deployment-00.t=
xt
>=20
>=20
>=20
> --=20
> Dave T=E4ht
>=20


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From nobody Fri Mar 14 06:02:52 2014
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Subject: Re: [rmcat] ECN marking for webrtc and stun packets?
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... and NADA quite explicitly incorporates ECN:
http://tools.ietf.org/html/draft-zhu-rmcat-nada-03

Cheers,
Michael


On 14. mars 2014, at 13:56, Eggert, Lars wrote:

> Hi,
>=20
> there is also RFC6679, which deals with RTP but not (necessarily) in =
the context of webrtc.
>=20
> Lars
>=20
> On 2014-3-14, at 13:39, Dave Taht <dave.taht@gmail.com> wrote:
>=20
>> I am curious if anyone has outlined an approach to CC for
>> webrtc (rtp, stun, etc) based on more fully utilizing ECN for
>> it?
>>=20
>> The only thing that I see that sort of touches upon it was this:
>>=20
>> =
http://www.tschofenig.priv.at/cc-workshop/draft-arkko-wscc-deployment-00.t=
xt
>>=20
>>=20
>>=20
>> --=20
>> Dave T=E4ht
>>=20
>=20


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">... =
and NADA quite explicitly incorporates ECN:<div><a =
href=3D"http://tools.ietf.org/html/draft-zhu-rmcat-nada-03">http://tools.i=
etf.org/html/draft-zhu-rmcat-nada-03</a></div><div><br></div><div>Cheers,<=
/div><div>Michael</div><div><br></div><div><br><div><div>On 14. mars =
2014, at 13:56, Eggert, Lars wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div>Hi,<br><br>there is also RFC6679, which deals with =
RTP but not (necessarily) in the context of =
webrtc.<br><br>Lars<br><br>On 2014-3-14, at 13:39, Dave Taht &lt;<a =
href=3D"mailto:dave.taht@gmail.com">dave.taht@gmail.com</a>&gt; =
wrote:<br><br><blockquote type=3D"cite">I am curious if anyone has =
outlined an approach to CC for<br></blockquote><blockquote =
type=3D"cite">webrtc (rtp, stun, etc) based on more fully utilizing ECN =
for<br></blockquote><blockquote =
type=3D"cite">it?<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The only thing =
that I see that sort of touches upon it was =
this:<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite"><a =
href=3D"http://www.tschofenig.priv.at/cc-workshop/draft-arkko-wscc-deploym=
ent-00.txt">http://www.tschofenig.priv.at/cc-workshop/draft-arkko-wscc-dep=
loyment-00.txt</a><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">-- =
<br></blockquote><blockquote type=3D"cite">Dave =
T=E4ht<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><br></div></blockquote></div><br></div></bo=
dy></html>=

--Apple-Mail=_425907C2-1515-4557-B525-BD8B8B8657C8--


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Subject: Re: [rmcat] ECN marking for webrtc and stun packets?
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On Fri, Mar 14, 2014 at 6:02 AM, Michael Welzl <michawe@ifi.uio.no> wrote:
> ... and NADA quite explicitly incorporates ECN:
> http://tools.ietf.org/html/draft-zhu-rmcat-nada-03
>
> Cheers,
> Michael

I obviously did not google hard enough. Thank you both.

(This is in reaction to the ongoing thread on using extensive QoS
markings in stun with the implied assumption that using AFxx
would actually accomplish anything on common media types like
wifi)

It does not appear if nada has made it as far as the webrtc sources.

>
> On 14. mars 2014, at 13:56, Eggert, Lars wrote:
>
> Hi,
>
> there is also RFC6679, which deals with RTP but not (necessarily) in the
> context of webrtc.
>
> Lars
>
> On 2014-3-14, at 13:39, Dave Taht <dave.taht@gmail.com> wrote:
>
> I am curious if anyone has outlined an approach to CC for
>
> webrtc (rtp, stun, etc) based on more fully utilizing ECN for
>
> it?
>
>
> The only thing that I see that sort of touches upon it was this:
>
>
> http://www.tschofenig.priv.at/cc-workshop/draft-arkko-wscc-deployment-00.=
txt
>
>
>
>
> --
>
> Dave T=E4ht
>
>
>
>



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html


From nobody Thu Mar 20 07:58:57 2014
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Subject: Re: [rmcat] RTT-independence (Re: WGLC for draft-ietf-rmcat-cc-requirements-02)
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On 3/12/2014 9:57 AM, Mirja Kühlewind wrote:
> Hi (as individual contributor),
>
> I'd think RTT-fairness is in general a good thing. The question would be, is
> there in rmcat an requirement for RTT-fairness that is more important than
> for other kinds of traffic? I'd actually say yes, because the quality of my
> call should not depend on the fact that my communication partner sits next
> door or on a different continent. (I know it still might, but hopefully not
> because of a RTT-fairness issue...).
>
> That's for me the reason why this could/should be listed explicitly in the
> requirement document. Like actually all requirements, RTT-fairness should
> only be a 'should', thus it will not be the doorkeeper to not except a
> candidate. But that's similar for all requirements. We might not be able to
> find a solution that can always fulfill all requirement, thus at the end we
> need to have a discussion which requirements are okay to break.
>

100% agree: it can be a goal we include in evaluation, but not a hard 
blocker

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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Subject: Re: [rmcat] RTT-independence (Re: WGLC for draft-ietf-rmcat-cc-requirements-02)
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On 20. mars 2014, at 15:58, Randell Jesup wrote:

> On 3/12/2014 9:57 AM, Mirja K=FChlewind wrote:
>> Hi (as individual contributor),
>>=20
>> I'd think RTT-fairness is in general a good thing. The question would =
be, is
>> there in rmcat an requirement for RTT-fairness that is more important =
than
>> for other kinds of traffic? I'd actually say yes, because the quality =
of my
>> call should not depend on the fact that my communication partner sits =
next
>> door or on a different continent. (I know it still might, but =
hopefully not
>> because of a RTT-fairness issue...).
>>=20
>> That's for me the reason why this could/should be listed explicitly =
in the
>> requirement document. Like actually all requirements, RTT-fairness =
should
>> only be a 'should', thus it will not be the doorkeeper to not except =
a
>> candidate. But that's similar for all requirements. We might not be =
able to
>> find a solution that can always fulfill all requirement, thus at the =
end we
>> need to have a discussion which requirements are okay to break.
>>=20
>=20
> 100% agree: it can be a goal we include in evaluation, but not a hard =
blocker

Fine by me.

(and for those who totally want it, coupled cc. should be able to force =
it onto the controller anyway, at least for the flows that originate =
from one host)


From nobody Mon Mar 31 06:12:24 2014
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From: "Eggert, Lars" <lars@netapp.com>
To: rmcat WG <rmcat@ietf.org>
Thread-Topic: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02
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Hi,

On 2014-2-17, at 15:53, Eggert, Lars <lars@netapp.com> wrote:
> we'd like to start the working group last call for =
draft-ietf-rmcat-cc-requirements-02. Please send your feedback to the =
list.
>=20
> We're letting this LC run until March 12 (after the London meeting), =
so there will also be a chance to discuss any feedback there.

the WGLC has concluded. Thank you all for commenting; we are expecting =
the editor to submit a version reflecting them shortly. Afterwards, we =
will forward the document to the IESG.

Lars

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From: Mirja =?iso-8859-1?q?K=FChlewind?= <mirja.kuehlewind@ikr.uni-stuttgart.de>
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Subject: [rmcat] Test Cases for Evaluation
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Hi group,

we would like to know who has read the document(s) that are specifying 
evaluation test cases, especially draft-sarker-rmcat-eval-test-00, and if 
those people think the content of the document(s) is interesting for the 
working group and that the group should continue to work on this (which means 
potentially adding a milestone)?

Thanks,
Mirja



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Subject: [rmcat] Please review draft-ietf-rmcat-eval-criteria-01
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Hi group,

draft-ietf-rmcat-eval-criteria-01 was updated after the last meeting 
addressing all comments from the meeting. We would really need people to 
review this document now! Please do so!

https://datatracker.ietf.org/doc/draft-ietf-rmcat-eval-criteria/

Mirja



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Subject: Re: [rmcat] WGLC for draft-ietf-rmcat-cc-requirements-02
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Planning an update incorporating feedback later today.

Incorporated feedback from Magnus.  Three notes:

On 3/3/2014 2:28 PM, Magnus Westerlund wrote:
> 4. Section 2, Req 1.D:
>
>     Similarly periodic bursty flows such as DASH or proprietary
>
> I would suggest adding a reference to DASH and possibly expand it.

Done, though I had to guess at the correct way to reference an ISO spec.

> 9. Section 2, Req 6.A
>
>          A.  In WebRTC, a division of packets into 4 classes is
>              envisioned in order of priority: faster-than-audio, audio,
>              video, best-effort, and bulk-transfer.  Typically the flows
>              managed by this algorithm would be audio or video in that
>              heirarchy, and feedback flows would be faster-than-audio.
>
> This doesn't match what is currently in
> draft-dhesikan-tsvwg-rtcweb-qos-05. I wonder if this needs to be
> either generalized or reference this document?

The division I mention was spoken to (can't remember if it was on-list, 
or in a meeting) in the terms I gave above.  To be honest, I'd want 
something equivalent to that, and if the draft given isn't providing it 
yet we need to address that.  I'd want feedback data to have higher 
priority than audio if at all possible; if not higher then equal to 
audio.  This is complicated by there being no standard ordering of DSCP 
classes.  Thus far I'm not referencing that personal draft.  And 
"envisioned" is a very soft term.

>
> 10. Section 2.
>
> Another requirement that might be missing and is related to 6.A, but
> not the same. It is the possibility to indicate the relative priority
> between multiple streams in the same session context. This doesn't
> require any QoS, just that the sending application indicates relative
> priority and that affects how bit-rate gets distributed. Or has the
> discussion been that it will be the higher layers that do this
> re-distribution between streams for which one have indication if they
> can increase or reduce the stream.
>

Yes, if there's a shared congestion context with multiple streams, then 
something is deciding how many bits of the shared resource each stream 
gets, and that's not part of this requirement.  I'd consider how the 
bits are apportioned between flows in a coupled-cc case as not being 
part of this requirement.

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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On 3/4/2014 11:26 AM, Bernard Aboba wrote:
> Here are my comments on the document.
>
>     1.   The congestion control algorithm must attempt to provide as-low-
>          as-possible-delay transit for real-time traffic while still
>          providing a useful amount of bandwidth, even when faced with
>          intermediate bottlenecks and competing flows.
> [BA] I am not sure that this requirement is meaningful because of the potential influence of the competing flows.  For example, when TCP is competing with
> realtime traffic, it will tend to fill the queue to the point of inducing
> loss, and there is not much that the RMCAT algorithm can do about this - the "attempts" will be in vain.  SoI would separate the requirement into multiple parts, one stating goals inthe absence of competing traffic, and another part talking about goals in the presence of competing flows.

I separated that out into a bullet point, and mentioned there's a limit 
on what's possible to achieve.

>
>          There may be
>          lower limits on the amount of bandwidth that is useful, but this
>          is largely application-specific and the application may be able
>          to modify or remove flows in order allow some useful flows to
>          get enough bandwidth.
> [BA] There are situations such as scalable video coding where the "lower
> limit" is not application-specific, because it relates to the characteristics of the base layer.  So in these situations, once you have stopped sending
> extension layers, you only have the base layer, whose characteristics may
> not be flexible because of the relationship between the base and extension
> layers.

I don't see this as affecting the requirement; we state there may be 
lower limits, and that this is "largely" application-specific (and 
"application-specific" would include a layered codec and the 
configuration of that codec).

>          A.  It should also handle routing changes and interface changes
>              (WiFi to 3G data, etc) which may radically change the
>              bandwidth available, and react quickly, especially if there
>              is a reduction in available bandwidth.
> [BA] We need to separate routing and interface changes, because routing
> changes are typically invisible to the application, whereas interface
> changes will not be.  I would argue that reacting "quickly" to routing
> transients is a non-goal because this is a regular feature of the Internet,
> and the major requirement is to be resilient against them.   This is one
> of the reasons that Circuit Breakers is likely to be ineffective in practice- shutting down communications and requiring manual intervention in
> response to a routing transient is going to be annoying at best.
> Interface changes are a distinct issue, because information could be
> provided (either through use of bw estimates or information relating
> to the interface itself) that could help reset the parameters of the
> video encoder shortly after the interface change.

Separated them, and also mentioned that on an interface change we assume 
the algorithm can be notified.

>          C.  The algorithm must not overreact to short-term bursts (such
>              as web-browsing) which can quickly saturate a local-
>              bottleneck router or link, but also clear quickly, and
>              should recover quickly when the burst ends.  This is
>              inherently at odds with the need to react quickly-enough to
>              avoid queue buildup.
> [BA] It is also at odds with the ability to react quickly to rate adaptation on wireless networks.

Among other things.  We like things that are inherently at odds in 
low-delay congestion control ;-) :-/

>          D.  Similarly periodic bursty flows such as DASH or proprietary
>              media streaming algorithms may compete in bursts with the
>              algorithm, and may not be adaptive within a burst.  They are
>              often are layered on top of TCP.  The algorithm must avoid
>              too much delay buildup during those bursts, and quickly
>              recover.  Note that this traffic may on an access link, or
>              may cause a shift in the location of the bottleneck fir the
>              duration of the burst.
> [BA] I would argue that it is not within the power of the algorithm to avoid
> delay buildup due to competing flows.  They are what they are.  The only
> thing RMCAT can do is to decide how it wishes to behave in response.

I disagree in detail, though I agree in general.  Our algorithm will 
affect other protocols, if in no other way than how quickly they ramp up 
and exactly where they stabilize.  Since these are time-limited, we may 
have some luck in allowing moderate delay buildup during a burst, or 
perhaps simply giving in for the burst and immediately popping back up 
after it ends will work better. (Especially if the algorithm can detect 
a regular bursty flow or simply the end of a competing flow.)  For 
example, we could notice a sudden very-greedy flow, back way off, and as 
soon as we see the queue start to drain immediately try restoring our 
previous flow level instead of ramping back up.  If we immediately start 
building queues, we know it wasn't a simple burst and can switch behaviors.

So I'll keep the current text and let's see what the proposals can manage.

>          A.  If possible, it should also share information and adaptation
>              with other non-RTP flows between the same endpoints, such as
>                                a WebRTC data channel  
> [BA] Cooperative adaptation between RTP and SCTP is a tall order, particularly since the RMCAT WG is not chartered to work on SCTP congestion control.

"If possible."  This might occur at the application or 
congestion-controller level.

>     11.  It should be evaluated in how it works both with backbone-router
>          bottlenecks, (asymmetric) local-loop bottlenecks, and local-lan
>          (WiFi/etc) bottlenecks, and in competition with varying numbers
>          and types of streams (TCP, TCP variants in use, LEDBAT
>          [I-D.ietf-ledbat-congestion  <http://tools.ietf.org/html/draft-ietf-rmcat-cc-requirements-02#ref-I-D.ietf-ledbat-congestion>], inflexible VoIP UDP flows).
> [BA] WiFi, with its (often broken) rate adaptation is going to be a specialchallengefor evaluation, because the loss could be due to interference or poor rate adaptation, rather than due
> to congestion.  I would probably delete this.
>

it should be evaluated with regards to those.  This is for the 
evaluation spec.  And some of those evaluations may be more useful than 
others, or easier than others.  WiFi may be problematic, but it's a HUGE 
usecase, and larger going forward.  In terms of importance, I'd likely 
put it on a par with wired and perhaps even higher (thinking to the 
future).  Of course, it may work better with newer/better/less-buggy 
wifi routing too.


-- 
Randell Jesup -- rjesup a t mozilla d o t com


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    <div class="moz-cite-prefix">On 3/4/2014 11:26 AM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU181-W6213C0A0D8FAF0EF3F82B2938E0@phx.gbl"
      type="cite">
      <style><!--
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      <div dir="ltr">
        <div>Here are my comments on the document.</div>
        <div><br>
        </div>
        <div>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">   1.   The congestion control algorithm must attempt to provide as-low-
        as-possible-delay transit for real-time traffic while still
        providing a useful amount of bandwidth, even when faced with
        intermediate bottlenecks and competing flows.</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">[BA] I am not sure that this requirement is meaningful because of the potential influence of the competing flows.  For example, when TCP is competing with</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">realtime traffic, it will tend to fill the queue to the point of inducing</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">loss, and there is not much that the RMCAT algorithm can do about this - the "attempts" will be in vain.  So <span style="font-size: 12pt; font-family: Calibri, sans-serif;">I would separate the requirement into multiple parts, one stating goals in </span><span style="font-size: 12pt; font-family: Calibri, sans-serif;">the absence of competing traffic, and another part talking a</span><span style="font-size: 12pt; font-family: Calibri, sans-serif;">bout goals in the presence of competing flows. </span></pre>
        </div>
      </div>
    </blockquote>
    <br>
    I separated that out into a bullet point, and mentioned there's a
    limit on what's possible to achieve.<br>
    <br>
    <blockquote cite="mid:BLU181-W6213C0A0D8FAF0EF3F82B2938E0@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="font-size: 12pt; font-family: Calibri, sans-serif;">
</span></pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">        There may be
        lower limits on the amount of bandwidth that is useful, but this
        is largely application-specific and the application may be able
        to modify or remove flows in order allow some useful flows to
        get enough bandwidth. </pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">[BA] There are situations such as scalable video coding where the "lower</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">limit" is not application-specific, because it relates to the characteristics of the base layer.  So in these situations, once you have stopped sending</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">extension layers, you only have the base layer, whose characteristics may</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">not be flexible because of the relationship between the base and extension</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">layers. </pre>
        </div>
      </div>
    </blockquote>
    <br>
    I don't see this as affecting the requirement; we state there may be
    lower limits, and that this is "largely" application-specific (and
    "application-specific" would include a layered codec and the
    configuration of that codec).<br>
    <br>
    <blockquote cite="mid:BLU181-W6213C0A0D8FAF0EF3F82B2938E0@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">        A.  It should also handle routing changes and interface changes
            (WiFi to 3G data, etc) which may radically change the
            bandwidth available, and react quickly, especially if there
            is a reduction in available bandwidth.</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">[BA] We need to separate routing and interface changes, because routing</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">changes are typically invisible to the application, whereas interface</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">changes will not be.  I would argue that reacting "quickly" to routing</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">transients is a non-goal because this is a regular feature of the Internet,</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">and the major requirement is to be resilient against them.   This is one</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">of the reasons that Circuit Breakers is likely to be ineffective in practice- shutting down communications and requiring manual intervention in</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">response to a routing transient is going to be annoying at best. </pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">Interface changes are a distinct issue, because information could be</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">provided (either through use of bw estimates or information relating</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">to the interface itself) that could help reset the parameters of the</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">video encoder shortly after the interface change.  </pre>
        </div>
      </div>
    </blockquote>
    <br>
    Separated them, and also mentioned that on an interface change we
    assume the algorithm can be notified.<br>
    <br>
    <blockquote cite="mid:BLU181-W6213C0A0D8FAF0EF3F82B2938E0@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">        C.  The algorithm must not overreact to short-term bursts (such
            as web-browsing) which can quickly saturate a local-
            bottleneck router or link, but also clear quickly, and
            should recover quickly when the burst ends.  This is
            inherently at odds with the need to react quickly-enough to
            avoid queue buildup.</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">[BA] It is also at odds with the ability to react quickly to rate adaptation on wireless networks. </pre>
        </div>
      </div>
    </blockquote>
    <br>
    Among other things.&nbsp; We like things that are inherently at odds in
    low-delay congestion control ;-) :-/<br>
    <br>
    <blockquote cite="mid:BLU181-W6213C0A0D8FAF0EF3F82B2938E0@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">        D.  Similarly periodic bursty flows such as DASH or proprietary
            media streaming algorithms may compete in bursts with the
            algorithm, and may not be adaptive within a burst.  They are
            often are layered on top of TCP.  The algorithm must avoid
            too much delay buildup during those bursts, and quickly
            recover.  Note that this traffic may on an access link, or
            may cause a shift in the location of the bottleneck fir the
            duration of the burst.</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">[BA] I would argue that it is not within the power of the algorithm to avoid</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">delay buildup due to competing flows.  They are what they are.  The only</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">thing RMCAT can do is to decide how it wishes to behave in response.</pre>
        </div>
      </div>
    </blockquote>
    <br>
    I disagree in detail, though I agree in general.&nbsp; Our algorithm will
    affect other protocols, if in no other way than how quickly they
    ramp up and exactly where they stabilize.&nbsp; Since these are
    time-limited, we may have some luck in allowing moderate delay
    buildup during a burst, or perhaps simply giving in for the burst
    and immediately popping back up after it ends will work better.&nbsp;
    (Especially if the algorithm can detect a regular bursty flow or
    simply the end of a competing flow.)&nbsp; For example, we could notice a
    sudden very-greedy flow, back way off, and as soon as we see the
    queue start to drain immediately try restoring our previous flow
    level instead of ramping back up.&nbsp; If we immediately start building
    queues, we know it wasn't a simple burst and can switch behaviors.<br>
    <br>
    So I'll keep the current text and let's see what the proposals can
    manage.<br>
    <br>
    <blockquote cite="mid:BLU181-W6213C0A0D8FAF0EF3F82B2938E0@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">        A.  If possible, it should also share information and adaptation
            with other non-RTP flows between the same endpoints, such as
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="font-size: 12pt; font-family: Calibri, sans-serif;">                              a WebRTC data channel</span> </pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">[BA] Cooperative adaptation between RTP and SCTP is a tall order, particularly since the RMCAT WG is not chartered to work on SCTP congestion control. </pre>
        </div>
      </div>
    </blockquote>
    <br>
    "If possible."&nbsp; This might occur at the application or
    congestion-controller level.<br>
    <br>
    <blockquote cite="mid:BLU181-W6213C0A0D8FAF0EF3F82B2938E0@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">   11.  It should be evaluated in how it works both with backbone-router
        bottlenecks, (asymmetric) local-loop bottlenecks, and local-lan
        (WiFi/etc) bottlenecks, and in competition with varying numbers
        and types of streams (TCP, TCP variants in use, LEDBAT
        [<a moz-do-not-send="true" href="http://tools.ietf.org/html/draft-ietf-rmcat-cc-requirements-02#ref-I-D.ietf-ledbat-congestion">I-D.ietf-ledbat-congestion</a>], inflexible VoIP UDP flows).</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">[BA] WiFi, with its (often broken) rate adaptation is going to be a specialchallenge <span style="font-size: 12pt; font-family: Calibri, sans-serif;">for evaluation, because the loss could be due to interference or poor rate adaptation, rather than due</span></pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><span style="font-size: 12pt; font-family: Calibri, sans-serif;">to congestion.  I would probably delete this. </span></pre>
          <br>
        </div>
      </div>
    </blockquote>
    <br>
    it should be evaluated with regards to those.&nbsp; This is for the
    evaluation spec.&nbsp; And some of those evaluations may be more useful
    than others, or easier than others.&nbsp; WiFi may be problematic, but
    it's a HUGE usecase, and larger going forward.&nbsp; In terms of
    importance, I'd likely put it on a par with wired and perhaps even
    higher (thinking to the future).&nbsp; Of course, it may work better with
    newer/better/less-buggy wifi routing too.<br>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Randell Jesup -- rjesup a t mozilla d o t com
</pre>
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the RTP Media Congestion Avoidance Techniques Working Group of the IETF.

        Title           : Congestion Control Requirements For RMCAT
        Author          : Randell Jesup
	Filename        : draft-ietf-rmcat-cc-requirements-03.txt
	Pages           : 9
	Date            : 2014-03-31

Abstract:
   Congestion control is needed for all data transported across the
   Internet, in order to promote fair usage and prevent congestion
   collapse.  The requirements for interactive, point-to-point real time
   multimedia, which needs low-delay, semi-reliable data delivery, are
   different from the requirements for bulk transfer like FTP or bursty
   transfers like Web pages.

   This document attempts to describe a set of requirements that can be
   used to evaluate other congestion control mechanisms in order to
   figure out their fitness for this purpose, and in particular to
   provide a set of possible requirements for proposals coming out of
   the RMCAT Working Group.



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