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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: "rmcat WG (rmcat@ietf.org)" <rmcat@ietf.org>
Thread-Topic: SCReAM Questions, comments and answers
Thread-Index: Ac+v67toGzBoGVtLRPKlIIR7Ckl9og==
Date: Mon, 4 Aug 2014 16:17:28 +0000
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Subject: [rmcat] SCReAM Questions, comments and answers
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Hi
First of all, thanks for the comments to the SCReAM proposal during the RMC=
AT session. Below I add some more info based on Piers's comments as well as=
 the comments that I see in the meeting minutes.

+ Intraprotocol fairness (Piers's comment)  : A method very similar multipl=
icative decrease is implemented in the SCReAM code that we use. In a few wo=
rds it reduces CWND if the smoothed OWD exceeds half the target delay AND m=
ore than 10 RTTs have elapsed since the last CWND reduction. This works so-=
so but I needed to add another extra check that inhibits this function if t=
he bitrate is low.

+ Media on hold or frame blanking (also related to Piers's comment): The id=
ea is to avoid sending high bitrate media (video) for a short duration (~1 =
RTT) and measure the OWD during this time. To aviod poor video performance =
it is recommended to inhibit the video encoder during these periods as it m=
akes it possible to maintain a correct codec state.
This serves two purposes:
1) Make possible for the flow (as well as other intraprotocol flows) to mor=
e correctly update their base delays, the rationale is that  the queue occu=
pancy in the bottleneck should drop to zero and then all intraprotocol flow=
s should get a  more correct base delay estimate. This solves the intraprot=
ocol fairness issue quite well.
2) Detection of competing (TCP) flows and adjustment of delay target. If th=
ere is only one flow then the OWD will drop to zero during the media on hol=
d period otherwise OWD will have a higher value, the delay target is then s=
et equal to the measured OWD. This is complemented by function which period=
ically checks if the delay target can be decreased in the case that the com=
peting flow is removed.

Comment to what is written in the preliminary minutes.

+ Frame blanking and bursts: I am not sure that I understand the comment. T=
he idea is to inhibit the video encoder for a short duration, the duration =
is so short that the codec state and the congestion window is maintained as=
 is, in other words, no slow start is taking place. Now, because the predic=
tion gain is reduced, there is a possibility that the PSNR may drop tempora=
rly after a frame blanking period because of reduced prediction gain, curre=
ntly I don't know exactly what the effect of frame blanking is.
+ 32 bits ACK vector: This was a design choise in the simulator. Currently =
it is more than enough. It is however possible that RLE encoding over a lar=
ger span is needed if the bitrates are higher and/or min RTT is larger.
+ Bytes in flight or packets in flight : I would recommend bytes in flight,=
 packets in flight can become messy as not all RTP packets use up the entir=
e MTU, one example is if one IR frame is split into several RTP packets due=
 to MTU restrictions.
+ Dropping RTP packets : This should be avoided and used only as a last res=
ort to remove RTP packets that are anyway too old to be rendered. It is ins=
tead better to avoid encoding new RTP packets (frame skipping) as long as t=
he sender queue is too larger, this helps to keep the video codec state in =
a healthy condition. There may be other methods than frame skipping as Mo m=
ention, the main point is that it should be possible to reduce the temporal=
 sender queue size growth in cases where the link thoughput drops.
+ Prediction in order to reduce jitter: I have tried out a method where the=
 video bitrate increase is slowed down around the last known congestion (in=
flection) point. This is very similar to how TCP BIC or Cubic behaves, the =
method increases the elapsed time between delay spikes but does not reduce =
the size that much. In addition I implemented a 1st order prediction on the=
 OWD samples and use that as a trend indicator. This manages to remove almo=
st all jitter spikes and for some unknown reason it improves the intra prot=
ocol fairness greatly. There is more experimentation needed to make sure th=
at it does work well with the LTE test cases.

Probably missed a few comments/questions, please feel free to ask.

Will add more details around the above in the next update of the draft.

/Ingemar

=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.
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<mailto:ingemar.s.johansson@ericsson.com>
www.ericsson.com

"The difference between stupidity and
genius is that genius has its limits."
Albert Einstein
=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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<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal">First of all, thanks for the comments to the SCReAM =
proposal during the RMCAT session. Below I add some more info based on Pier=
s&#8217;s comments as well as the comments that I see in the meeting minute=
s.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&#43; Intraprotocol fairness (Piers&#8217;s comment)=
&nbsp; : A method very similar multiplicative decrease is implemented in th=
e SCReAM code that we use. In a few words it reduces CWND if the smoothed O=
WD exceeds half the target delay AND more than 10
 RTTs have elapsed since the last CWND reduction. This works so-so but I ne=
eded to add another extra check that inhibits this function if the bitrate =
is low.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&#43; Media on hold or frame blanking (also related =
to Piers&#8217;s comment): The idea is to avoid sending high bitrate media =
(video) for a short duration (~1 RTT) and measure the OWD during this time.=
 To aviod poor video performance it is recommended
 to inhibit the video encoder during these periods as it makes it possible =
to maintain a correct codec state.<o:p></o:p></p>
<p class=3D"MsoNormal">This serves two purposes:<o:p></o:p></p>
<p class=3D"MsoNormal">1) Make possible for the flow (as well as other intr=
aprotocol flows) to more correctly update their base delays, the rationale =
is that &nbsp;the queue occupancy in the bottleneck should drop to zero and=
 then all intraprotocol flows should get
 a&nbsp; more correct base delay estimate. This solves the intraprotocol fa=
irness issue quite well. &nbsp;&nbsp;&nbsp;&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal">2) Detection of competing (TCP) flows and adjustment=
 of delay target. If there is only one flow then the OWD will drop to zero =
during the media on hold period otherwise OWD will have a higher value, the=
 delay target is then set equal to
 the measured OWD. This is complemented by function which periodically chec=
ks if the delay target can be decreased in the case that the competing flow=
 is removed.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Comment to what is written in the preliminary minute=
s.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&#43; Frame blanking and bursts: I am not sure that =
I understand the comment. The idea is to inhibit the video encoder for a sh=
ort duration, the duration is so short that the codec state and the congest=
ion window is maintained as is, in other
 words, no slow start is taking place. Now, because the prediction gain is =
reduced, there is a possibility that the PSNR may drop temporarly after a f=
rame blanking period because of reduced prediction gain, currently I don&#8=
217;t know exactly what the effect of
 frame blanking is.<o:p></o:p></p>
<p class=3D"MsoNormal">&#43; 32 bits ACK vector: This was a design choise i=
n the simulator. Currently it is more than enough. It is however possible t=
hat RLE encoding over a larger span is needed if the bitrates are higher an=
d/or min RTT is larger.<o:p></o:p></p>
<p class=3D"MsoNormal">&#43; Bytes in flight or packets in flight : I would=
 recommend bytes in flight, packets in flight can become messy as not all R=
TP packets use up the entire MTU, one example is if one IR frame is split i=
nto several RTP packets due to MTU restrictions.<o:p></o:p></p>
<p class=3D"MsoNormal">&#43; Dropping RTP packets : This should be avoided =
and used only as a last resort to remove RTP packets that are anyway too ol=
d to be rendered. It is instead better to avoid encoding new RTP packets (f=
rame skipping) as long as the sender queue
 is too larger, this helps to keep the video codec state in a healthy condi=
tion. There may be other methods than frame skipping as Mo mention, the mai=
n point is that it should be possible to reduce the temporal sender queue s=
ize growth in cases where the link
 thoughput drops.<o:p></o:p></p>
<p class=3D"MsoNormal">&#43; Prediction in order to reduce jitter: I have t=
ried out a method where the video bitrate increase is slowed down around th=
e last known congestion (inflection) point. This is very similar to how TCP=
 BIC or Cubic behaves, the method increases
 the elapsed time between delay spikes but does not reduce the size that mu=
ch. In addition I implemented a 1<sup>st</sup> order prediction on the OWD =
samples and use that as a trend indicator. This manages to remove almost al=
l jitter spikes and for some unknown
 reason it improves the intra protocol fairness greatly. There is more expe=
rimentation needed to make sure that it does work well with the LTE test ca=
ses.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Probably missed a few comments/questions, please fee=
l free to ask.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Will add more details around the above in the next u=
pdate of the draft.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">/Ingemar<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">=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<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ingemar Joh=
ansson&nbsp; M.Sc.
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Senior Rese=
archer<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;=
</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ericsson AB=
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Wireless Ac=
cess Networks<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Labratorieg=
r=E4nd 11<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">971 28, Lul=
e=E5, Sweden<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Phone &#43;=
46-1071 43042<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">SMS/MMS &#4=
3;46-73 078 3289<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"SV" styl=
e=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"=
><a href=3D"mailto:ingemar.s.johansson@ericsson.com"><span lang=3D"EN-US" s=
tyle=3D"color:windowtext">ingemar.s.johansson@ericsson.com</span></a></span=
><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-s=
erif&quot;"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"SV" styl=
e=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"=
><a href=3D"www.ericsson.com"><span lang=3D"EN-US" style=3D"color:windowtex=
t">www.ericsson.com</span></a></span><span style=3D"font-size:10.0pt;font-f=
amily:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;=
</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:black"=
>&#8220;</span><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot=
;,&quot;sans-serif&quot;;color:black;background:white">The difference betwe=
en stupidity
 and <o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:black;=
background:white">genius is that genius has its limits.</span>&quot;<span s=
tyle=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quo=
t;;color:black"><br>
</span><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;=
sans-serif&quot;">Albert Einstein<br>
</span><span lang=3D"SV" style=3D"font-size:10.0pt;font-family:&quot;Arial&=
quot;,&quot;sans-serif&quot;">=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<o:p></o:p></span></p=
>
<p class=3D"MsoNormal" style=3D"margin-bottom:7.5pt;line-height:15.0pt"><sp=
an style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif=
&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Piers O'Hanlon <p.ohanlon@gmail.com>, Zaheduzzaman Sarker <zaheduzzaman.sarker@ericsson.com>
Thread-Topic: [iccrg] draft-johansson-rmcat-scream-cc
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Date: Mon, 4 Aug 2014 16:17:26 +0000
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Hi Piers

Thanks for the pointer, for some reason I have missed this paper.
I will address this as well as other comments in a separate email.

/Ingemar


From: Piers O'Hanlon [mailto:p.ohanlon@gmail.com]
Sent: den 24 juli 2014 11:03
To: Zaheduzzaman Sarker
Cc: Ingemar Johansson S; iccrg@irtf.org; rmcat WG
Subject: Re: [iccrg] draft-johansson-rmcat-scream-cc

Hi Zahed,

On 23 Jul 2014, at 19:31, Zaheduzzaman Sarker wrote:


Hi Piers,
Thanks for the comments.
Could you please send us pointers to the research papers you mentioned in y=
our reply regarding performance issue with "media on hold" in the wireline =
network?.

I was talking about the suggestions of related approaches such as the use o=
f gaps (RFC6817 Sec 4.4) or 'Random window dropping' (section V.B) in "The =
quest for LEDBAT fairness, Giovanna Carofiglio, Luca Muscariello, Dario Ros=
si and Silvio Valenti, Globcom 2010. As mentioned in the paper the utility =
of media on hold type of approaches is inversely related to the number of f=
lows.

Piers



BR

Zahed


From: Piers O'Hanlon [mailto:p.ohanlon@gmail.com]
Sent: den 23 juli 2014 08:26
To: Ingemar Johansson S
Cc: Zaheduzzaman Sarker; iccrg@irtf.org<mailto:iccrg@irtf.org>; rmcat WG
Subject: Re: [iccrg] draft-johansson-rmcat-scream-cc

Hi Ingemar,

Thanks for your replies - more inline.

Regards,

Piers.


On 17 Jul 2014, at 19:53, Ingemar Johansson S wrote:



Hi

Thanks for the interest in this contribution. Answers/comments inline.

Regards
Ingemar

From: Piers O'Hanlon [mailto:p.ohanlon@gmail.com]
Sent: den 17 juli 2014 13:24
To: Ingemar Johansson S; Zaheduzzaman Sarker
Cc: iccrg@irtf.org<mailto:iccrg@irtf.org>; rmcat WG
Subject: Re: [iccrg] draft-johansson-rmcat-scream-cc

Hi Ingemar and Zahed,

Great to see this work come to draft stage - I like the name. The approach =
looks interesting - I had a couple of questions:

- Since Scream is based upon LEDBAT I wondered if you had looked into some =
of the issues it suffers from like; intra-procotol fairness, late-comer adv=
antage, delay-impact on other flows, and yielding behaviour to short of lon=
ger lived TCP flows etc.
[IJ] Zahed will present a few test cases at the RMCAT WG sessions. A few ad=
ditional comments. Yes, I have see similar issues that are reported for LED=
BAT in terms of late comers advantage etc. also in SCReAM, the current solu=
tion to this is to put media on hold for about one RTT at regular intervals=
 if the one way delay exceeds a threshold and measure the OWD during that p=
eriod.
This makes it possible to updated the base delay, and also competing LEDBAT=
 based flows will update the base delay more accurately even though they do=
n't use this solution. The behavior in the precense of competing TCP flows =
depend much on the type of AQM in the bottleneck, a few examples will be gi=
ven in the presentation.

I guess this sort of technique may be more likely to work in LTE type of en=
vironments where each bearer channel only carries a few flows. But on wired=
 links with more flows some research papers seem to show it may not be as e=
ffective? I wonder if there's an AQM that more suited to LEDBAT/Vegas style=
 OWD measurement...? I suppose the 'low delay' AQMs would be beneficial but=
 if you're running one of those then most protocols will benefit in terms o=
f low delay.

And also I assume that it's behaviour would be similar to LEDBAT when compe=
ting with TCP type flows?




- It is mentioned that there's an 'Adaptive delay target' - but in the code=
 owdTarget does not change - Did you have strategies for choosing the TARGE=
T delay on different links - or is it fixed? Or did you mean the adaptation=
 based on off_target?
[IJ] Correct, there is not yet any adaptive target delay functionality in t=
he code, the adaptive delay target function use the same technique to put t=
he media on hold and measure the OWD during that time and update the delay =
target. If there are no competing flows then the OWD will drop considerably=
, if there are competing flows then the OWD will not drop that much, and th=
is is an indication that the delay target can be increased.
Ah ok - Do you see any strange behaviour where different flows end up with =
different targets and suffer or do they correct themselves quickly enough?

- What period do use to measure the Base Delay (or do use LEDBAT's 1 minute=
 value) - the default seems quite high for mobile deployments...?
[IJ] Don't remember right now but it is definitely shorter than default LED=
BAT, will look it up when I am back from the vacation for good in mid Augus=
t
It would be interesting to know how the actual RTT of on LTE can vary as a =
user moves, and I imagine handovers would lead to some complications.



- What sort of filtering do you propose for queuing delay beyond MIN - Mobi=
le provides for very noisy environments I wondered if you had any alternati=
ves?
[IJ] Yes it is noisy but sofar I don't use any filtering at all, seems like=
 it is sufficient with the filtering that is inherent in the CWND adjustmen=
t.
I guess it depends on the variation of the noise - what sort of variation d=
o you typically see?



- How does it compete against a LEDBAT flow with a similar target?
[IJ] We don't have any example of it, I tried it out in the earlier stages =
but the results are lost and it is probably better to re run and see how it=
 behaves.
Yes it would be interesting to see how it compares to LEDBAT and to aid in =
understanding the advantages Scream brings.





Cheers,

Piers



On 17 Jun 2014, at 18:35, Ingemar Johansson S wrote:




Hi

This draft can be of interest for those of you interested in congestion con=
trol for realtime media.
http://www.ietf.org/internet-drafts/draft-johansson-rmcat-scream-cc-00.txt

Abstract : "This memo describes a rate adaptation framework for conversatio=
nal
   video services.  The solution conforms to the packet conservation
   principle and uses a hybrid loss and delay based congestion control
   algorithm.  The framework is evaluated over both simulated bottleneck
   scenarios as well as in a LTE (Long Term Evolution) system simulator
   and is shown to achieve both low latency and high video throughput in
   these scenarios"

/Ingemar


=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.
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<mailto:ingemar.s.johansson@ericsson.com>
www.ericsson.com<x-msg://1596/www.ericsson.com>

"The difference between stupidity and
genius is that genius has its limits."
Albert Einstein
=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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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Piers<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks for the pointer, f=
or some reason I have missed this paper.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I will address this as we=
ll as other comments in a separate email.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">/Ingemar<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Piers O'=
Hanlon [mailto:p.ohanlon@gmail.com]
<br>
<b>Sent:</b> den 24 juli 2014 11:03<br>
<b>To:</b> Zaheduzzaman Sarker<br>
<b>Cc:</b> Ingemar Johansson S; iccrg@irtf.org; rmcat WG<br>
<b>Subject:</b> Re: [iccrg] draft-johansson-rmcat-scream-cc<o:p></o:p></spa=
n></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi Zahed,<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On 23 Jul 2014, at 19:31, Zaheduzzaman Sarker wrote:=
<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Piers,</span><o:p></o:=
p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks for the comments.<=
/span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Could you please send us =
pointers to the research papers you mentioned in your reply regarding perfo=
rmance issue with &#8220;media on hold&#8221; in the wireline network?.</sp=
an><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">I was talking about the suggestions of related appro=
aches such as the use of gaps (RFC6817 Sec 4.4) or 'Random window dropping'=
 (section V.B) in &quot;The quest for LEDBAT fairness,&nbsp;Giovanna Carofi=
glio, Luca Muscariello, Dario Rossi and Silvio
 Valenti, Globcom 2010. As mentioned in the paper the utility of media on h=
old type of approaches is inversely related to the number of flows.<o:p></o=
:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Piers<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Zahed</span><o:p></o:p></=
p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</div>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt;border-width:initial;border-color:initial">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm;border-width:initial;border-color:initial">
<div>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span class=3D"apple-=
converted-space"><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&q=
uot;,&quot;sans-serif&quot;">&nbsp;</span></span><span style=3D"font-size:1=
0.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">Piers
 O'Hanlon [<a href=3D"mailto:p.ohanlon@gmail.com">mailto:p.ohanlon@gmail.co=
m</a>]<span class=3D"apple-converted-space">&nbsp;</span><br>
<b>Sent:</b><span class=3D"apple-converted-space">&nbsp;</span>den 23 juli =
2014 08:26<br>
<b>To:</b><span class=3D"apple-converted-space">&nbsp;</span>Ingemar Johans=
son S<br>
<b>Cc:</b><span class=3D"apple-converted-space">&nbsp;</span>Zaheduzzaman S=
arker; <a href=3D"mailto:iccrg@irtf.org">
iccrg@irtf.org</a>; rmcat WG<br>
<b>Subject:</b><span class=3D"apple-converted-space">&nbsp;</span>Re: [iccr=
g] draft-johansson-rmcat-scream-cc</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Hi Ingemar,<o:p></o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Thanks for your replies - more inline.<o:p></o:p></p=
>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Piers.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">On 17 Jul 2014, at 19:53, Ingemar Johansson S wrote:=
<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi</span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks for the interest i=
n this contribution. Answers/comments inline.</span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards</span><o:p></o:p>=
</p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Ingemar</span><o:p></o:p>=
</p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</div>
</div>
<div style=3D"border:none;border-left:solid windowtext 3.0pt;padding:0cm 0c=
m 0cm 4.0pt;border-width:initial;border-color:initial;border-width:initial;=
border-color:initial;z-index:auto">
<div>
<div style=3D"border:none;border-top:solid windowtext 3.0pt;padding:3.0pt 0=
cm 0cm 0cm;border-width:initial;border-color:initial;border-width:initial;b=
order-color:initial">
<div>
<div>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span class=3D"apple-=
converted-space"><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&q=
uot;,&quot;sans-serif&quot;">&nbsp;</span></span><span style=3D"font-size:1=
0.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">Piers
 O'Hanlon [<a href=3D"mailto:p.ohanlon@gmail.com">mailto:p.ohanlon@gmail.co=
m</a>]<span class=3D"apple-converted-space">&nbsp;</span><br>
<b>Sent:</b><span class=3D"apple-converted-space">&nbsp;</span>den 17 juli =
2014 13:24<br>
<b>To:</b><span class=3D"apple-converted-space">&nbsp;</span>Ingemar Johans=
son S; Zaheduzzaman Sarker<br>
<b>Cc:</b><span class=3D"apple-converted-space">&nbsp;</span><a href=3D"mai=
lto:iccrg@irtf.org">iccrg@irtf.org</a>; rmcat WG<br>
<b>Subject:</b><span class=3D"apple-converted-space">&nbsp;</span>Re: [iccr=
g] draft-johansson-rmcat-scream-cc</span><o:p></o:p></p>
</div>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Hi I<span style=3D"color:#1F497D">ng</span>emar and&=
nbsp;Zahed,<o:p></o:p></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">Great to see this work come to draft stage - I like =
the name. The approach looks interesting - I had a couple of questions:<o:p=
></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">- Since Scream is based upon LEDBAT I wondered if yo=
u had looked into some of the issues it suffers from like; intra-procotol f=
airness, late-comer advantage, delay-impact on other flows, and yielding be=
haviour to short of longer lived TCP
 flows etc.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">[IJ] Zahed will present a=
 few test cases at the RMCAT WG sessions. A few additional comments. Yes, I=
 have see similar issues that are reported for LEDBAT in
 terms of late comers advantage etc. also in SCReAM, the current solution t=
o this is to put media on hold for about one RTT at regular intervals if th=
e one way delay exceeds a threshold and measure the OWD during that period.=
</span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">This makes it possible to=
 updated the base delay, and also competing LEDBAT based flows will update =
the base delay more accurately even though they don&#8217;t use
 this solution. The behavior in the precense of competing TCP flows depend =
much on the type of AQM in the bottleneck, a few examples will be given in =
the presentation.</span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</div>
</div>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">I guess this sort of technique may be more likely to=
 work in LTE type of environments where each bearer channel only carries a =
few flows. But on wired links with more flows some research papers seem to =
show it may not be as effective? I
 wonder if there's an AQM that more suited to LEDBAT/Vegas style OWD measur=
ement...? I suppose the 'low delay' AQMs would be beneficial but if you're =
running one of those then most protocols will benefit in terms of low delay=
.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">And also I assume that it's behaviour would be simil=
ar to LEDBAT when competing with TCP type flows?<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
</div>
<div>
<div style=3D"border:none;border-left:solid windowtext 3.0pt;padding:0cm 0c=
m 0cm 4.0pt;border-width:initial;border-color:initial;border-width:initial;=
border-color:initial;z-index:auto">
<div>
<div>
<div>
<p class=3D"MsoNormal">- It is mentioned that there's an 'Adaptive delay ta=
rget' - but in the code owdTarget does not change -&nbsp;Did you have strat=
egies for choosing the TARGET delay on different links - or is it fixed? Or=
 did you mean the adaptation based on off_target?<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">[IJ] Correct, there is no=
t yet any adaptive target delay functionality in the code, the adaptive del=
ay target function use the same technique to put the media
 on hold and measure the OWD during that time and update the delay target. =
If there are no competing flows then the OWD will drop considerably, if the=
re are competing flows then the OWD will not drop that much, and this is an=
 indication that the delay target
 can be increased.</span><o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Ah ok - Do you see any strange behaviour where diffe=
rent flows end up with different targets and suffer or do they correct them=
selves quickly enough?<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div style=3D"border:none;border-left:solid windowtext 3.0pt;padding:0cm 0c=
m 0cm 4.0pt;border-width:initial;border-color:initial;border-width:initial;=
border-color:initial;z-index:auto">
<div>
<div>
<div>
<p class=3D"MsoNormal">- What period do use to measure the Base Delay (or d=
o use LEDBAT's 1 minute value) - the default seems quite high for mobile de=
ployments...?<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">[IJ] Don&#8217;t remember=
 right now but it is definitely shorter than default LEDBAT, will look it u=
p when I am back from the vacation for good in mid August</span><o:p></o:p>=
</p>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<div>
<p class=3D"MsoNormal">It would be interesting to know how the actual RTT o=
f on LTE can vary as a user moves, and I imagine handovers would lead to so=
me complications.<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
</div>
<div>
<div style=3D"border:none;border-left:solid windowtext 3.0pt;padding:0cm 0c=
m 0cm 4.0pt;border-width:initial;border-color:initial;border-width:initial;=
border-color:initial;z-index:auto">
<div>
<div>
<div>
<p class=3D"MsoNormal">- What sort of filtering do you propose for queuing =
delay beyond MIN - Mobile provides for very noisy environments I wondered i=
f you had any alternatives?<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">[IJ] Yes it is noisy but =
sofar I don&#8217;t use any filtering at all, seems like it is sufficient w=
ith the filtering that is inherent in the CWND adjustment.</span><o:p></o:p=
></p>
</div>
</div>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">I guess it depends on the variation of the noise - w=
hat sort of variation do you typically see?<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
</div>
<div>
<div style=3D"border:none;border-left:solid windowtext 3.0pt;padding:0cm 0c=
m 0cm 4.0pt;border-width:initial;border-color:initial;border-width:initial;=
border-color:initial;z-index:auto">
<div>
<div>
<div>
<p class=3D"MsoNormal">- How does it compete against a LEDBAT flow with a s=
imilar target?<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">[IJ] We don&#8217;t have =
any example of it, I tried it out in the earlier stages but the results are=
 lost and it is probably better to re run and see how it behaves.</span><o:=
p></o:p></p>
</div>
</div>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Yes it would be interesting to see how it compares t=
o LEDBAT and to aid in understanding the advantages Scream brings.<o:p></o:=
p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
</div>
<div>
<div style=3D"border:none;border-left:solid windowtext 3.0pt;padding:0cm 0c=
m 0cm 4.0pt;border-width:initial;border-color:initial;border-width:initial;=
border-color:initial;z-index:auto">
<div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">Cheers,<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">Piers<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal">On 17 Jun 2014, at 18:35, Ingemar Johansson S wrote:=
<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<br>
<o:p></o:p></p>
</div>
</div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Hi</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">This draft can be of interest for those=
 of you interested in congestion control for realtime media.</span><o:p></o=
:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><a href=3D"http://www.ietf.org/internet=
-drafts/draft-johansson-rmcat-scream-cc-00.txt">http://www.ietf.org/interne=
t-drafts/draft-johansson-rmcat-scream-cc-00.txt</a></span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Abstract : &#8220;This memo describes a=
 rate adaptation framework for conversational</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;&nbsp; video services.&nbsp; The =
solution conforms to the packet conservation</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;&nbsp; principle and uses a hybri=
d loss and delay based congestion control</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;&nbsp; algorithm.&nbsp; The frame=
work is evaluated over both simulated bottleneck</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;&nbsp; scenarios as well as in a =
LTE (Long Term Evolution) system simulator</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;&nbsp; and is shown to achieve bo=
th low latency and high video throughput in</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;&nbsp;<span class=3D"apple-conver=
ted-space">&nbsp;</span></span><span lang=3D"SV" style=3D"font-size:11.0pt;=
font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">these scenarios&#82=
21;</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p=
>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;">/Ingemar</span><o:p></o:p><=
/p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p=
>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p=
>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:10.0pt;font-fam=
ily:&quot;Arial&quot;,&quot;sans-serif&quot;">=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</s=
pan><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:10.0pt;font-fam=
ily:&quot;Arial&quot;,&quot;sans-serif&quot;">Ingemar Johansson&nbsp; M.Sc.=
</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Senior Researcher</span><o:p></o:p></p>
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</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Ericsson AB</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Wireless Access Networks</span><o:p></o:p=
></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Labratoriegr=E4nd 11</span><o:p></o:p></p=
>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">971 28, Lule=E5, Sweden</span><o:p></o:p>=
</p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Phone &#43;46-1071 43042</span><o:p></o:p=
></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">SMS/MMS &#43;46-73 078 3289</span><o:p></=
o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:10.0pt;font-fam=
ily:&quot;Arial&quot;,&quot;sans-serif&quot;"><a href=3D"mailto:ingemar.s.j=
ohansson@ericsson.com"><span lang=3D"EN-US" style=3D"color:windowtext">inge=
mar.s.johansson@ericsson.com</span></a></span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:10.0pt;font-fam=
ily:&quot;Arial&quot;,&quot;sans-serif&quot;"><a href=3D"x-msg://1596/www.e=
ricsson.com"><span lang=3D"EN-US" style=3D"color:windowtext">www.ericsson.c=
om</span></a></span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:black">&#8220;<span style=3D"backgro=
und:white">The difference between stupidity and</span></span><o:p></o:p></p=
>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;;color:black;background:white">genius is th=
at genius has its limits.</span><span style=3D"font-size:11.0pt;font-family=
:&quot;Calibri&quot;,&quot;sans-serif&quot;">&quot;</span><span style=3D"fo=
nt-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:b=
lack"><br>
</span><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;=
sans-serif&quot;">Albert Einstein<br>
</span><span lang=3D"SV" style=3D"font-size:10.0pt;font-family:&quot;Arial&=
quot;,&quot;sans-serif&quot;">=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</span><o:p></o:p></p=
>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:7.5pt;line-height:15.0pt"><sp=
an style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif=
&quot;;color:black">&nbsp;</span><o:p></o:p></p>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">_______________________________________________<br>
iccrg mailing list<br>
<a href=3D"mailto:iccrg@irtf.org">iccrg@irtf.org</a><br>
<a href=3D"https://www.irtf.org/mailman/listinfo/iccrg">https://www.irtf.or=
g/mailman/listinfo/iccrg</a><o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
</html>

--_000_81564C0D7D4D2A4B9A86C8C7404A13DA31FAF764ESESSMB205erics_--


From nobody Tue Aug  5 06:52:04 2014
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From: "Eggert, Lars" <lars@netapp.com>
To: "rmcat@ietf.org" <rmcat@ietf.org>
Thread-Topic: IPR Disclosure: Polycom, Inc.'s Statement about IPR related to draft-singh-rmcat-adaptive-fec-00
Thread-Index: AQHPrbLE7yL0LpvGXE2yncWn9NHVNA==
Date: Tue, 5 Aug 2014 13:51:36 +0000
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Begin forwarded message:

> From: IETF Secretariat <ietf-ipr@ietf.org>
> Subject: IPR Disclosure: Polycom, Inc.'s Statement about IPR related =
to draft-singh-rmcat-adaptive-fec-00
> Date: August 1, 2014 at 20:02:28 GMT+2
> To: <varun@comnet.tkk.fi>, <marcin.nagy@aalto.fi>, <jo@comnet.tkk.fi>, =
<lars@netapp.com>
> Cc: <jari.arkko@piuha.net>, <ipr-announce@ietf.org>
>=20
>=20
> Dear Varun Singh, Marcin Nagy, Joerg Ott, Lars Eggert:
>=20
> An IPR disclosure that pertains to your Internet-Draft entitled =
"Congestion
> Control Using FEC for Conversational Media" =
(draft-singh-rmcat-adaptive-fec) was
> submitted to the IETF Secretariat on 2014-08-01 and has been posted on =
the "IETF
> Page of Intellectual Property Rights Disclosures"
> (https://datatracker.ietf.org/ipr/2400/). The title of the IPR =
disclosure is
> "Polycom, Inc.'s Statement about IPR related to =
draft-singh-rmcat-adaptive-
> fec-00."");
>=20
> The IETF Secretariat
>=20


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From mirja.kuehlewind@tik.ee.ethz.ch  Wed Aug  6 05:06:50 2014
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Date: Wed, 06 Aug 2014 14:06:42 +0200
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Hi everybody,

I changed jobs this week and therefore have a new email address.
Please use the following address in future:

     mirja.kuehlewind@tik.ee.ethz.ch

The old address is still active but I will check it less frequently over 
time.

Thanks,
Mirja


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From: "Eggert, Lars" <lars@netapp.com>
To: "rmcat@ietf.org" <rmcat@ietf.org>
Thread-Topic: [ipr-announce] IPR Disclosure: Cisco's Statement of IPR Related to draft-zhu-rmcat-nada-03
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Begin forwarded message:

> From: IETF Secretariat <ietf-ipr@ietf.org>
> Subject: [ipr-announce] IPR Disclosure: Cisco's Statement of IPR =
Related to draft-zhu-rmcat-nada-03
> Date: August 6, 2014 at 23:56:27 GMT+2
> To: <xiaoqzhu@cisco.com>, <ropan@cisco.com>
> Cc: <jari.arkko@piuha.net>, <ipr-announce@ietf.org>
>=20
>=20
> Dear Xiaoqing Zhu, Rong Pan:
>=20
> An IPR disclosure that pertains to your Internet-Draft entitled "NADA: =
A
> Unified Congestion Control Scheme for Real-Time Media" =
(draft-zhu-rmcat-nada)
> was submitted to the IETF Secretariat on 2014-08-06 and has been =
posted on the
> "IETF Page of Intellectual Property Rights Disclosures"
> (https://datatracker.ietf.org/ipr/2409/). The title of the IPR =
disclosure is
> "Cisco's Statement of IPR Related to draft-zhu-rmcat-nada-03."");
>=20
> The IETF Secretariat
>=20
> _______________________________________________
> ipr-announce mailing list
> ipr-announce@ietf.org
> https://www.ietf.org/mailman/listinfo/ipr-announce


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From nobody Thu Aug  7 01:20:46 2014
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From: "Eggert, Lars" <lars@netapp.com>
To: rmcat WG <rmcat@ietf.org>
Thread-Topic: [rmcat] Summary of Toronto decisions
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Hi,

On 2014-7-29, at 11:19, Eggert, Lars <lars@netapp.com> wrote:
>> In the meantime, I wanted to summarize the decisions we took and =
confirm them on the list:
>>=20
>> * draft-perkins-rmcat-rtp-cc-feedback adopted as WG item
>> * draft-sarker-rmcat-eval-test adopted as WG item
>> * draft-zanaty-rmcat-app-interaction adopted as WG item
>>=20
>> Please speak up if you disagree with these decisions.
>=20
> Please comment by Aug 8.

we've gotten additional support for these decisions by mail, with nobody =
disagreeing, so these three drafts are now adopted as WG items.

Editors, whenever you resubmit the next versions please change the names =
to draft-rmcat-xyz-00. (You may wait with these resubmissions until you =
have other edits to submit.)

Lars

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From nobody Thu Aug  7 01:56:20 2014
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Subject: [rmcat] I-D Action: draft-ietf-rmcat-app-interaction-00.txt
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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           : RTP Application Interaction with Congestion Control
        Authors         : Mo Zanaty
                          Varun Singh
                          Suhas Nandakumar
                          Zaheduzzaman Sarker
	Filename        : draft-ietf-rmcat-app-interaction-00.txt
	Pages           : 14
	Date            : 2014-08-07

Abstract:
   Interactive real-time media applications that use the Real-time
   Transport Protocol (RTP) over the User Datagram Protocol (UDP) must
   use congestion control techniques above the UDP layer since it
   provides none.  This memo describes the interactions and conceptual
   interfaces necessary between the application components that relate
   to congestion control, including the RTP layer, the higher-level
   media codec control layer, and the lower-level transport interface,
   as well as components dedicated to congestion control functions.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rmcat-app-interaction/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rmcat-app-interaction-00


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 Thu Aug  7 02:29:21 2014
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Date: Thu, 7 Aug 2014 12:29:14 +0300
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--90e6ba6e8488dfce38050006b82b
Content-Type: text/plain; charset=UTF-8

Hi,

No changes were made, just submitted the last version of the document as
the WG draft.

-Varun
On 7 Aug 2014 11:56, <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           : RTP Application Interaction with Congestion
> Control
>         Authors         : Mo Zanaty
>                           Varun Singh
>                           Suhas Nandakumar
>                           Zaheduzzaman Sarker
>         Filename        : draft-ietf-rmcat-app-interaction-00.txt
>         Pages           : 14
>         Date            : 2014-08-07
>
> Abstract:
>    Interactive real-time media applications that use the Real-time
>    Transport Protocol (RTP) over the User Datagram Protocol (UDP) must
>    use congestion control techniques above the UDP layer since it
>    provides none.  This memo describes the interactions and conceptual
>    interfaces necessary between the application components that relate
>    to congestion control, including the RTP layer, the higher-level
>    media codec control layer, and the lower-level transport interface,
>    as well as components dedicated to congestion control functions.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rmcat-app-interaction/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rmcat-app-interaction-00
>
>
> 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/
>
>

--90e6ba6e8488dfce38050006b82b
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<p dir=3D"ltr">Hi,</p>
<p dir=3D"ltr">No changes were made, just submitted the last version of the=
 document as the WG draft. </p>
<p dir=3D"ltr">-Varun</p>
<div class=3D"gmail_quote">On 7 Aug 2014 11:56,  &lt;<a href=3D"mailto:inte=
rnet-drafts@ietf.org">internet-drafts@ietf.org</a>&gt; wrote:<br type=3D"at=
tribution"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bor=
der-left:1px #ccc solid;padding-left:1ex">
<br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the RTP Media Congestion Avoidance Techn=
iques Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : RTP =
Application Interaction with Congestion Control<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Mo Zanaty=
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Varun Singh<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Suhas Nandakumar<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Zaheduzzaman Sarker<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename =C2=A0 =C2=A0 =C2=A0 =C2=A0: draft-iet=
f-rmcat-app-interaction-00.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : 14<b=
r>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 2014-08-07<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0Interactive real-time media applications that use the Real-tim=
e<br>
=C2=A0 =C2=A0Transport Protocol (RTP) over the User Datagram Protocol (UDP)=
 must<br>
=C2=A0 =C2=A0use congestion control techniques above the UDP layer since it=
<br>
=C2=A0 =C2=A0provides none. =C2=A0This memo describes the interactions and =
conceptual<br>
=C2=A0 =C2=A0interfaces necessary between the application components that r=
elate<br>
=C2=A0 =C2=A0to congestion control, including the RTP layer, the higher-lev=
el<br>
=C2=A0 =C2=A0media codec control layer, and the lower-level transport inter=
face,<br>
=C2=A0 =C2=A0as well as components dedicated to congestion control function=
s.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rmcat-app-interactio=
n/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rmcat-app=
-interaction/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rmcat-app-interaction-00" =
target=3D"_blank">http://tools.ietf.org/html/draft-ietf-rmcat-app-interacti=
on-00</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
</blockquote></div>

--90e6ba6e8488dfce38050006b82b--


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Subject: [rmcat] Fwd: WGLC: draft-ietf-dart-dscp-rtp-02
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Hi,

The DART working group has started a last call for =
draft-ietf-dart-dscp-rtp-02, ending on August 22. This informational =
draft describes considerations for the use of DiffServ code points in =
situations where one multiplexes multiple RTP packet streams, and =
potentially other protocol streams, that share the same 5-tuple.=20

Since this draft is likely of interest to several working groups, I =
would like to solicit additional reviews from participants of TSVWG, =
AVTCORE, MMUSIC, CLUE, RTCWEB, and RMCAT. Please send any feedback =
(including feedback to the effect of "this is ready to progress") to the =
DART working group mailing list.

Thanks!

Ben.

Begin forwarded message:

> Resent-From: draft-alias-bounces@tools.ietf.org
> From: Ben Campbell <ben@nostrum.com>
> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
> Date: August 9, 2014 at 12:50:16 PM CDT
> Resent-To: ben@nostrum.com, david.black@emc.com, paulej@packetizer.com
> To: dart@ietf.org
> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, mls.ietf@gmail.com, =
Richard Barnes <rlb@ipv.sx>
>=20
> This is a DART Working Group Last Call of draft-ietf-dart-dscp-rtp-02. =
It's available on the data tracker at the following URL:
>=20
> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
>=20
> The WGLC will conclude on 22 August, 2014. Please send your comments =
to the authors and the DART mailing list. If you've reviewed it and =
think it's good to go, please say so.
>=20
> Thanks!
>=20
> Ben.
>=20


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To: Ben Campbell <ben@nostrum.com>
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Cc: mmusic@ietf.org, draft-ietf-dart-dscp-rtp.all@tools.ietf.org, "clue@ietf.org" <clue@ietf.org>, avt@ietf.org, "rmcat@ietf.org" <rmcat@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Subject: Re: [rmcat] [rtcweb] Fwd: WGLC: draft-ietf-dart-dscp-rtp-02
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This is also of some interest to me at least, and perhaps some members
of the AQM working group who are working out weighted fair queuing and
wred like approaches to these needs.

Anyway, I do try to track thinking in this area in the ongoing
development of cerowrt's sqm system, which has pluggable modules for
all the known aqm and fq schedulers, and a three tier system modeled
on wondershaper for managing traffic,
which uses what diffserv markings that I have observed make sense...

Some relevant links:

http://www.bufferbloat.net/projects/cerowrt/wiki/Wondershaper_Must_Die

http://www.bufferbloat.net/projects/cerowrt/wiki/Setting_up_SQM_for_CeroWrt=
_310

http://www.bufferbloat.net/projects/cerowrt/wiki/Smart_Queue_Management


Missing from this discussion so far is what is actually deployed out
there... I am only familiar with stuff like the above as an
egress/ingress queue management system...

AND: in the case of linux based wireless routers, none of the AF
codepoints are respected explicitly, and how the codepoints are mapped
are described by 802.11e - and that is far more about scheduling
priority for airtime rather than drop probability.

It is kind of hard to wrap your head around how 802.11e actually
works, but, this is a start:

http://en.wikipedia.org/wiki/IEEE_802.11e-2005#Enhanced_distributed_channel=
_access_.28EDCA.29

In linux wifi at least:

CS0, CS3 are mapped into the BE hw queue
CS1, CS2 are mapped into the BK hw queue
CS4, CS5 are mapped into the VI hw queue
CS6, CS7 are mapped into the VO  hw queue

Bit patterns for the other classes to whatever extent they map into
CSX are mapped into the underlying hardware queue.

I am not saying this is an optimal state of affairs, (as notably VO
should be as completely disabled as possible in the 802.11n and ac
age), and am very interested in how other OSes (ios, osx, windows),
and router/switch OSes do it, but the wireless scheduling aspect of
all these codepoints does not seem to be well touched upon.

On a personal note I tend to think that having 17+ ways to slice up
traffic is about 13 or 14 too many, particularly those targetting a
drop probability. When you have drop rates well below a few percentage
points, adding another percentage point of granularity seems futile.

Lastly, there is also an implementation issue in linux presently in
that doing extensive diffserv classification basically requires
one iptables or tc rule per class, which is inefficient.  Someone (not
me!) doing some code for doing diffserv saner there would go a long
way...




On Sat, Aug 9, 2014 at 11:48 AM, Ben Campbell <ben@nostrum.com> wrote:
> Hi,
>
> The DART working group has started a last call for draft-ietf-dart-dscp-r=
tp-02, ending on August 22. This informational draft describes consideratio=
ns for the use of DiffServ code points in situations where one multiplexes =
multiple RTP packet streams, and potentially other protocol streams, that s=
hare the same 5-tuple.
>
> Since this draft is likely of interest to several working groups, I would=
 like to solicit additional reviews from participants of TSVWG, AVTCORE, MM=
USIC, CLUE, RTCWEB, and RMCAT. Please send any feedback (including feedback=
 to the effect of "this is ready to progress") to the DART working group ma=
iling list.
>
> Thanks!
>
> Ben.
>
> Begin forwarded message:
>
>> Resent-From: draft-alias-bounces@tools.ietf.org
>> From: Ben Campbell <ben@nostrum.com>
>> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
>> Date: August 9, 2014 at 12:50:16 PM CDT
>> Resent-To: ben@nostrum.com, david.black@emc.com, paulej@packetizer.com
>> To: dart@ietf.org
>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, mls.ietf@gmail.com, Ric=
hard Barnes <rlb@ipv.sx>
>>
>> This is a DART Working Group Last Call of draft-ietf-dart-dscp-rtp-02. I=
t's available on the data tracker at the following URL:
>>
>> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
>>
>> The WGLC will conclude on 22 August, 2014. Please send your comments to =
the authors and the DART mailing list. If you've reviewed it and think it's=
 good to go, please say so.
>>
>> Thanks!
>>
>> Ben.
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Dave T=C3=A4ht

NSFW: https://w2.eff.org/Censorship/Internet_censorship_bills/russell_0296_=
indecent.article


From nobody Thu Aug 14 13:14:11 2014
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Subject: [rmcat] I-D Action: draft-ietf-rmcat-eval-test-00.txt
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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           : Test Cases for Evaluating RMCAT Proposals
        Authors         : Zaheduzzaman Sarker
                          Varun Singh
                          Xiaoqing Zhu
                          Michael A. Ramalho
	Filename        : draft-ietf-rmcat-eval-test-00.txt
	Pages           : 36
	Date            : 2014-08-14

Abstract:
   The Real-time Transport Protocol (RTP) is used to transmit media in
   multimedia telephony applications, these applications are typically
   required to implement congestion control.  The RMCAT working group is
   currently working on candidate algorithms for such interactive real-
   time multimedia applications.  This document describes the test cases
   to be used in the performance evaluation of those candidate
   algorithms.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rmcat-eval-test-00


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 Thu Aug 14 13:28:14 2014
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Thu, 14 Aug 2014 23:27:49 +0300
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Subject: Re: [rmcat] I-D Action: draft-ietf-rmcat-eval-test-00.txt
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Hi all,

No changes were made, this is a resubmission of the latest version of
the individual document as a WG document.

-Varun

On Thu, Aug 14, 2014 at 11:14 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           : Test Cases for Evaluating RMCAT Proposals
>         Authors         : Zaheduzzaman Sarker
>                           Varun Singh
>                           Xiaoqing Zhu
>                           Michael A. Ramalho
>         Filename        : draft-ietf-rmcat-eval-test-00.txt
>         Pages           : 36
>         Date            : 2014-08-14
>
> Abstract:
>    The Real-time Transport Protocol (RTP) is used to transmit media in
>    multimedia telephony applications, these applications are typically
>    required to implement congestion control.  The RMCAT working group is
>    currently working on candidate algorithms for such interactive real-
>    time multimedia applications.  This document describes the test cases
>    to be used in the performance evaluation of those candidate
>    algorithms.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rmcat-eval-test/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rmcat-eval-test-00
>
>
> 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/


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On 30 Jul 2014, at 21:47, The IESG <iesg-secretary@ietf.org> wrote:
> The IESG has received a request from the RTP Media Congestion =
Avoidance
> Techniques WG (rmcat) to consider the following document:
> - 'Congestion Control Requirements For RMCAT'
>  <draft-ietf-rmcat-cc-requirements-05.txt> as Informational RFC
>=20
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action. Please send substantive comments to the
> ietf@ietf.org mailing lists by 2014-08-13.=20

Apologies for the late response, but I have some extensive comments on =
this
draft:

- Requirement 1A states that the congestion control algorithm should try =
to
  keep delays to a minimum. I agree that this is a primary goal. =
However,
  as I noted at IETF 90, I think the draft should also include a =
secondary
  requirement to keep delay variation (jitter) down, where possible, =
since
  larger delay variation needs larger receiver-side buffers to =
compensate,
  increasing overall latency.=20

- Requirement 1B states that the algorithm should react "quickly" to any
  reduction in bandwidth or increase in delay. The draft needs to define
  what quickly means in this context, since the expected timing could be
  different to other congestion control algorithms. Is the expectation=20=

  that the response occurs with an RTT, or with next video frame, GoP, =
or
  talkspurt? This comment also applies to requirements 1C, 1E, 1F, and =
4.

- Requirement 1B uses an "increase in bottleneck delay" as a congestion
  signal, but it's not clear how the congestion control can distinguish =
an
  increase in delay at the bottleneck from an increase in delay =
elsewhere.
  Perhaps this should be =93an increase in observed delay=94?

- Requirement 1C requires the congestion controller to react to =
interface
  changes. Does this apply to changes in the local interface only, or to=20=

  changes in either the local or remote interface? One is a local =
matter,
  while the other requires signalling, so the distinction may be =
important.

- Requirement 1D seems to describe possible behaviour of the system, but
  the requirement on the algorithm is not clear. Can you either clarify,
  or remove if there is no concrete requirement?

- Requirement 1E states that "the algorithm must not overreact". Can you
  define (even approximately) what overreacting means in this context?
  Without this definition, it's not clear how this requirement could be
  translated into protocol behaviour. Similarly, what is "short-term"?

- Requirement 1F states that the algorithm "must avoid too much delay=20
  buildup during those bursts". This is, at least partly, outside the=20
  control of the algorithm, since it depends on the behaviour of the
  cross traffic. Perhaps the requirement ought to be that the algorithm=20=

  does not significantly increase the delay buildup during the bursts?

- Requirement 2 states that flows should get "roughly equal" shares of
  bandwidth, but gives no definition of =93roughly equal". I realise =
that=20
  this is a difficult issue, but some guidance, however approximate, =
would
  be helpful.=20

- Requirement 2A is written in terms of "a useful share" of bandwidth.
  Again, defining "useful" would help: perhaps for the flow to ramp up =
to=20
  a bandwidth that allows it to support a minimal quality media stream?

- Requirement 3A is unclear. Is the intent that there is no requirement
  that the algorithm compete equally with TCP flows that have different
  RTT?

- Requirement 3B is unclear. Is the requirement that "the congestion
  control should prioritise achieving a useful (as defined in =
Requirement
  2A) share of the bandwidth over achieving as low as possible transit
  delay, when these two requirements are in conflict"?
=20
- Requirements 4A, 4B, and 4C seem to describe possible behaviours of =
the
  system, but the actual requirements are unclear. Requirment 4A could =
be
  that the algorithm must reach a useful sending rate for audio calls =
fast
  enough that the initial "Hello" is not clipped; it's not clear how =
this
  applies to other media.=20

- Requirement 5 is for "stable" behaviour, but it is not clear what
  stability means with discontinuous transmission, since the media will
  have an on-off profile that is not stable.=20

- Requirement 5A lists possible behaviour for the algorithm, but it is =
not
  clear what requirement is being placed on an implementation. It "may
  adapt more quickly", but is it required to do so? Previous bandwidth
  estimates "may need to be aged or thrown away", but are they required
  to be?

- Requirement 6 could be interpreted as requiring general multipath
  congestion control. Is that the intent, or is the goal only to share
  information between flows between two endpoint when those flows share=20=

  a common bottleneck?

- Requirement 6C gives a possible use for information, but it's not =
clear
  what requirement is being placed on implementations.

- Requirement 8 talks about mechanisms, using RTCP feedback and/or =
header
  extensions. It's appropriate that this draft gives give general =
feedback
  requirements, but I don't think it should be making explicit =
requirements
  on how RTCP or RTP header extensions are used without justification,=20=

  especially since the RMCAT working group has a milestone to produce a
  draft giving such recommendations. I suggest keeping requirement 8E,
  but removing the rest of requirement 8.


I also noted some editorial issues:

- The RMCAT working group should have a limited lifetime. Accordingly,=20=

  the title might be better written as "Congestion Control Requirements
  for Interactive Real-time Media Transport=94?

- The 4th paragraph of Section 1 talks about "requirements for RMCAT" =
and
  "The RMCAT congestion control algorithm". I suggest these be changed =
to
  "requirements for congestion control of interactive real-time media" =
and
  "A congestion control algorithm for interactive real-time media".

- The last paragraph of Section 1 states that terminology from the =
WebRTC
  overview is used. This is appropriate. It might also be appropriate to
  align with draft-ietf-avtext-rtp-grouping-taxonomy and cite that =
draft.

- Requirement 1E talks about "short-term bursts (such as web-browsing)".
  This would be clearer if written "short-term bursts of competing =
traffic
  (such as web browsing)", since the bursts are not due to the =
congestion
  control algorithm.


Colin



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




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Subject: Re: [rmcat] Last Call: <draft-ietf-rmcat-cc-requirements-05.txt> (Congestion Control Requirements For RMCAT) to Informational RFC
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I'll preface my remarks with a paraphrase of the US Supreme Court:  In 
many cases, I know something is X when I see it, but it's hard to put a 
written "X is <foo>" down, or the parameters of <foo> are fuzzy and 
multidimensional.  In the end, the proposals will be evaluated by humans 
looking at all the data people can provide, turning it over in their 
heads, arguing, and saying "X is best" or "Y is best for XYZ usage, and 
Z is best for ABC usage".   Kinda like "fairness"....

That said, I'll dive in:

On 8/18/2014 10:49 AM, Colin Perkins wrote:
> On 30 Jul 2014, at 21:47, The IESG <iesg-secretary@ietf.org> wrote:
>> The IESG has received a request from the RTP Media Congestion Avoidance
>> Techniques WG (rmcat) to consider the following document:
>> - 'Congestion Control Requirements For RMCAT'
>>   <draft-ietf-rmcat-cc-requirements-05.txt> as Informational RFC
>>
>> The IESG plans to make a decision in the next few weeks, and solicits
>> final comments on this action. Please send substantive comments to the
>> ietf@ietf.org mailing lists by 2014-08-13.
> Apologies for the late response, but I have some extensive comments on this
> draft:

Thanks for looking at it closely.

> - Requirement 1A states that the congestion control algorithm should try to
>    keep delays to a minimum. I agree that this is a primary goal. However,
>    as I noted at IETF 90, I think the draft should also include a secondary
>    requirement to keep delay variation (jitter) down, where possible, since
>    larger delay variation needs larger receiver-side buffers to compensate,
>    increasing overall latency.

How about:
           <t>Jitter (variation in the bitrate over short timescales)
           also is relevant, though moderate amounts of jitter will be
           absorbed by receiver jitter buffers.  Transit delay should be 
considered
           to include the short-term maximums of delay including jitter.</t>


> - Requirement 1B states that the algorithm should react "quickly" to any
>    reduction in bandwidth or increase in delay. The draft needs to define
>    what quickly means in this context, since the expected timing could be
>    different to other congestion control algorithms. Is the expectation
>    that the response occurs with an RTT, or with next video frame, GoP, or
>    talkspurt? This comment also applies to requirements 1C, 1E, 1F, and 4.

I'm loathe to give a hard number here, and instead leave that to the 
evaluation, since as with all things like this it's a tradeoff.  I could 
state that "quickly" means a timescale on the order of RTT times (i.e. 
O(RTT)), and certainly well sub-second.  (IMHO, a congestion mechanism 
needs to adapt down ASAP to avoid delay buildup it will then have to 
drain out).

> - Requirement 1B uses an "increase in bottleneck delay" as a congestion
>    signal, but it's not clear how the congestion control can distinguish an
>    increase in delay at the bottleneck from an increase in delay elsewhere.
>    Perhaps this should be “an increase in observed delay”?

Sure.

> - Requirement 1C requires the congestion controller to react to interface
>    changes. Does this apply to changes in the local interface only, or to
>    changes in either the local or remote interface? One is a local matter,
>    while the other requires signalling, so the distinction may be important.

I would say local interface changes should be noticed and both 
directions should take this into account if possible (for example as a 
sender by restarting adaptation from a known startpoint, or increase 
sensitivity/adaptation-rate temporarily in combination with an immediate 
drop in send rate as a precaution).  On the receive side, it can 
indicate an interface change via RTCP, or just let the other side adapt 
if the bottleneck has changed.  (Indicating a change may be preferable 
in order to more-intelligently adapt, but may not be required, and might 
not even help or help much.)

Really I'm interested to see what the proposals find are the best way to 
handle this problem, without constricting the solution space.

Suggested wording changes?

> - Requirement 1D seems to describe possible behaviour of the system, but
>    the requirement on the algorithm is not clear. Can you either clarify,
>    or remove if there is no concrete requirement?
Old:
           <t>The offered load may be less than the available bandwidth
           at any given moment, and may vary dramatically over time,
           including dropping to no load and then resuming a high load,
           such as in a mute operation.  The reaction time between a
           change in the bandwidth available from the algorithm and a
           change in the offered load is variable, and may be different
           when increasing versus decreasing.</t>

Proposal:
           <t>The algorithm must deal with offered loads that are less 
than the available bandwidth
           at any given moment, and may vary dramatically over time,
           including dropping to no load and then resuming a high load,
           such as in a mute/unmute operation.  Note that the reaction 
time between a
           change in the bandwidth available from the algorithm and a
           change in the offered load is variable, and may be different
           when increasing versus decreasing.</t>

i.e. make it into a requirement for the algorithm to deal with Mute 
operations and that offered load will not change immediately in response 
to the algorithm's idea of bandwidth (which was what was intended here).

> - Requirement 1E states that "the algorithm must not overreact". Can you
>    define (even approximately) what overreacting means in this context?
>    Without this definition, it's not clear how this requirement could be
>    translated into protocol behaviour. Similarly, what is "short-term"?

"the algorithm must deal with traffic bursts caused by actions such as 
webpage loading [by an aggressive browser/website combination]"?   (with 
the part in [] being perhaps removable). Again, the definition of 
"quickly" comes into play.

> - Requirement 1F states that the algorithm "must avoid too much delay
>    buildup during those bursts". This is, at least partly, outside the
>    control of the algorithm, since it depends on the behaviour of the
>    cross traffic. Perhaps the requirement ought to be that the algorithm
>    does not significantly increase the delay buildup during the bursts?

Perhaps that's a good rewording; that's a tough/painful case since they 
may smoke your router's buffers regardless of what you do, so you may 
want to keep sending at a "normal" rate or close once the algorithm 
realizes it's lost the battle (temporarily).

> - Requirement 2 states that flows should get "roughly equal" shares of
>    bandwidth, but gives no definition of “roughly equal". I realise that
>    this is a difficult issue, but some guidance, however approximate, would
>    be helpful.

This is "fair" of course.   I'm not sure how I can say much more without 
constraining the solution-space artificially.  And it's worse because 
it's not a single-point measurement; defining it in detail would require 
looking at startup ramp, steady-state, reaction to BW changes and 
cross-traffic (does it remain "roughly equal"), etc.  Down this road 
lives insanity...

I can try alternate wiggle-words, or some additional somewhat-vague phrases.

> - Requirement 2A is written in terms of "a useful share" of bandwidth.
>    Again, defining "useful" would help: perhaps for the flow to ramp up to
>    a bandwidth that allows it to support a minimal quality media stream?

A "minimal quality media stream" may be 8Kbps (or less!) or it might be 
8Mbps, so that makes it hard to define here....

> - Requirement 3A is unclear. Is the intent that there is no requirement
>    that the algorithm compete equally with TCP flows that have different
>    RTT?

3A says "don't be a hog" (likely not a problem anyways, but since once 
an algorithm has "given up" on low delay due to TCP competition you 
don't want it to starve the TCP streams by going CBR for example.)  And 
for long-lived flows, it can't compete equally even with flows with the 
same RTT.

>
> - Requirement 3B is unclear. Is the requirement that "the congestion
>    control should prioritise achieving a useful (as defined in Requirement
>    2A) share of the bandwidth over achieving as low as possible transit
>    delay, when these two requirements are in conflict"?

That might be a nice way to state that, thanks.

> - Requirements 4A, 4B, and 4C seem to describe possible behaviours of the
>    system, but the actual requirements are unclear. Requirment 4A could be
>    that the algorithm must reach a useful sending rate for audio calls fast
>    enough that the initial "Hello" is not clipped; it's not clear how this
>    applies to other media.

Those were to push the solutions in the direction of having a reasonable 
start-of-call/stream behavior, which otherwise might get ignored and not 
looked at or optimized for.  I agree they're somewhat editorial.

> - Requirement 5 is for "stable" behaviour, but it is not clear what
>    stability means with discontinuous transmission, since the media will
>    have an on-off profile that is not stable.

"must deal with"?

>
> - Requirement 5A lists possible behaviour for the algorithm, but it is not
>    clear what requirement is being placed on an implementation. It "may
>    adapt more quickly", but is it required to do so? Previous bandwidth
>    estimates "may need to be aged or thrown away", but are they required
>    to be?

<t>After stream resumption, the algorithm should attempt to regain a 
share of bandwidth as quickly as possible.</t>
("useful share"?)  *ducks*

> - Requirement 6 could be interpreted as requiring general multipath
>    congestion control. Is that the intent, or is the goal only to share
>    information between flows between two endpoint when those flows share
>    a common bottleneck?

The latter.  I thought it was clear; wording suggestions?

> - Requirement 6C gives a possible use for information, but it's not clear
>    what requirement is being placed on implementations.

Right, it's editorial reminding them they can do that.

> - Requirement 8 talks about mechanisms, using RTCP feedback and/or header
>    extensions. It's appropriate that this draft gives give general feedback
>    requirements, but I don't think it should be making explicit requirements
>    on how RTCP or RTP header extensions are used without justification,
>    especially since the RMCAT working group has a milestone to produce a
>    draft giving such recommendations. I suggest keeping requirement 8E,
>    but removing the rest of requirement 8.

Ok.

>
>
> I also noted some editorial issues:
>
> - The RMCAT working group should have a limited lifetime. Accordingly,
>    the title might be better written as "Congestion Control Requirements
>    for Interactive Real-time Media Transport”?

Sure

>
> - The 4th paragraph of Section 1 talks about "requirements for RMCAT" and
>    "The RMCAT congestion control algorithm". I suggest these be changed to
>    "requirements for congestion control of interactive real-time media" and
>    "A congestion control algorithm for interactive real-time media".

Sure

>
> - The last paragraph of Section 1 states that terminology from the WebRTC
>    overview is used. This is appropriate. It might also be appropriate to
>    align with draft-ietf-avtext-rtp-grouping-taxonomy and cite that draft.

I think we can do that...  (I'll see how much editing it needs to do so)

> - Requirement 1E talks about "short-term bursts (such as web-browsing)".
>    This would be clearer if written "short-term bursts of competing traffic
>    (such as web browsing)", since the bursts are not due to the congestion
>    control algorithm.

Sure.

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


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Thanks - responses inline.

Cheers,
Colin




On 20 Aug 2014, at 00:16, Randell Jesup <randell-ietf@jesup.org> wrote:
> I'll preface my remarks with a paraphrase of the US Supreme Court:  In =
many cases, I know something is X when I see it, but it's hard to put a =
written "X is <foo>" down, or the parameters of <foo> are fuzzy and =
multidimensional.  In the end, the proposals will be evaluated by humans =
looking at all the data people can provide, turning it over in their =
heads, arguing, and saying "X is best" or "Y is best for XYZ usage, and =
Z is best for ABC usage".   Kinda like =93fairness"....

A lot of these are difficult to be precise about, I agree.

> That said, I'll dive in:
>=20
> On 8/18/2014 10:49 AM, Colin Perkins wrote:
>> On 30 Jul 2014, at 21:47, The IESG <iesg-secretary@ietf.org> wrote:
>>> The IESG has received a request from the RTP Media Congestion =
Avoidance
>>> Techniques WG (rmcat) to consider the following document:
>>> - 'Congestion Control Requirements For RMCAT'
>>>  <draft-ietf-rmcat-cc-requirements-05.txt> as Informational RFC
>>>=20
>>> The IESG plans to make a decision in the next few weeks, and =
solicits
>>> final comments on this action. Please send substantive comments to =
the
>>> ietf@ietf.org mailing lists by 2014-08-13.
>> Apologies for the late response, but I have some extensive comments =
on this
>> draft:
>=20
> Thanks for looking at it closely.
>=20
>> - Requirement 1A states that the congestion control algorithm should =
try to
>>   keep delays to a minimum. I agree that this is a primary goal. =
However,
>>   as I noted at IETF 90, I think the draft should also include a =
secondary
>>   requirement to keep delay variation (jitter) down, where possible, =
since
>>   larger delay variation needs larger receiver-side buffers to =
compensate,
>>   increasing overall latency.
>=20
> How about:
>          <t>Jitter (variation in the bitrate over short timescales)
>          also is relevant, though moderate amounts of jitter will be
>          absorbed by receiver jitter buffers.  Transit delay should be =
considered
>          to include the short-term maximums of delay including =
jitter.</t>

Sure.

>> - Requirement 1B states that the algorithm should react "quickly" to =
any
>>   reduction in bandwidth or increase in delay. The draft needs to =
define
>>   what quickly means in this context, since the expected timing could =
be
>>   different to other congestion control algorithms. Is the =
expectation
>>   that the response occurs with an RTT, or with next video frame, =
GoP, or
>>   talkspurt? This comment also applies to requirements 1C, 1E, 1F, =
and 4.
>=20
> I'm loathe to give a hard number here, and instead leave that to the =
evaluation, since as with all things like this it's a tradeoff.  I could =
state that "quickly" means a timescale on the order of RTT times (i.e. =
O(RTT)), and certainly well sub-second.  (IMHO, a congestion mechanism =
needs to adapt down ASAP to avoid delay buildup it will then have to =
drain out).

I agree that we don=92t want to give a hard number, but I think some =
guidelines to set expectations would be useful. React within some small =
multiple of the RTT would seem appropriate, possibly with some wording =
about possible codec constraints also (e.g., react on the next video =
frame, or maybe at the start of the next talk spurt)? Agree that =
sub-second reaction is desirable.=20

Saying =93In this memo, a =91quick=92 reaction to congestion means=85=94 =
would then cover the later uses of the term.

>> - Requirement 1B uses an "increase in bottleneck delay" as a =
congestion
>>   signal, but it's not clear how the congestion control can =
distinguish an
>>   increase in delay at the bottleneck from an increase in delay =
elsewhere.
>>   Perhaps this should be =93an increase in observed delay=94?
>=20
> Sure.
>=20
>> - Requirement 1C requires the congestion controller to react to =
interface
>>   changes. Does this apply to changes in the local interface only, or =
to
>>   changes in either the local or remote interface? One is a local =
matter,
>>   while the other requires signalling, so the distinction may be =
important.
>=20
> I would say local interface changes should be noticed and both =
directions should take this into account if possible (for example as a =
sender by restarting adaptation from a known startpoint, or increase =
sensitivity/adaptation-rate temporarily in combination with an immediate =
drop in send rate as a precaution).  On the receive side, it can =
indicate an interface change via RTCP, or just let the other side adapt =
if the bottleneck has changed.  (Indicating a change may be preferable =
in order to more-intelligently adapt, but may not be required, and might =
not even help or help much.)
>=20
> Really I=92m interested to see what the proposals find are the best =
way to handle this problem, without constricting the solution space.

Agree, and agree that those are likely mechanisms for doing this.=20

> Suggested wording changes?

Maybe something like =93It should detect handover between different =
interfaces (e.g., Wi-Fi to 3G) at sender or receiver, and signal these =
to the congestion control algorithm where possible, to help the =
algorithm respond quickly to such path changes=94?=20

>> - Requirement 1D seems to describe possible behaviour of the system, =
but
>>   the requirement on the algorithm is not clear. Can you either =
clarify,
>>   or remove if there is no concrete requirement?
> Old:
>          <t>The offered load may be less than the available bandwidth
>          at any given moment, and may vary dramatically over time,
>          including dropping to no load and then resuming a high load,
>          such as in a mute operation.  The reaction time between a
>          change in the bandwidth available from the algorithm and a
>          change in the offered load is variable, and may be different
>          when increasing versus decreasing.</t>
>=20
> Proposal:
>          <t>The algorithm must deal with offered loads that are less =
than the available bandwidth
>          at any given moment, and may vary dramatically over time,
>          including dropping to no load and then resuming a high load,
>          such as in a mute/unmute operation.  Note that the reaction =
time between a
>          change in the bandwidth available from the algorithm and a
>          change in the offered load is variable, and may be different
>          when increasing versus decreasing.</t>
>=20
> i.e. make it into a requirement for the algorithm to deal with Mute =
operations and that offered load will not change immediately in response =
to the algorithm=92s idea of bandwidth (which was what was intended =
here).

Sure, makes sense.

>> - Requirement 1E states that "the algorithm must not overreact". Can =
you
>>   define (even approximately) what overreacting means in this =
context?
>>   Without this definition, it's not clear how this requirement could =
be
>>   translated into protocol behaviour. Similarly, what is =
"short-term"?
>=20
> "the algorithm must deal with traffic bursts caused by actions such as =
webpage loading [by an aggressive browser/website combination]"?   (with =
the part in [] being perhaps removable). Again, the definition of =
=93quickly" comes into play.

Perhaps =93Short-term bursts of competing traffic (e.g., due to local =
web browser use) can saturate a local bottleneck link, but also clear =
quickly. The algorithm needs to be able respond quickly to such events =
to help prevent queue build-up; it is desirable if it can also quickly =
recover its sending rate when they conclude=94?=20

>> - Requirement 1F states that the algorithm "must avoid too much delay
>>   buildup during those bursts". This is, at least partly, outside the
>>   control of the algorithm, since it depends on the behaviour of the
>>   cross traffic. Perhaps the requirement ought to be that the =
algorithm
>>   does not significantly increase the delay buildup during the =
bursts?
>=20
> Perhaps that=92s a good rewording; that's a tough/painful case since =
they may smoke your router's buffers regardless of what you do, so you =
may want to keep sending at a "normal" rate or close once the algorithm =
realizes it's lost the battle (temporarily).

Sure, it=92s difficult. If it=92s a transient burst that fills the queue =
for a short time then disappears, then you want to back off to avoid =
making the situation worse, then continue, but if the competing traffic =
is determined to keep the queue full, you=92re better keeping sending =
and accepting the latency, and let the user hang-up if it gets too bad.

>> - Requirement 2 states that flows should get "roughly equal" shares =
of
>>   bandwidth, but gives no definition of =93roughly equal". I realise =
that
>>   this is a difficult issue, but some guidance, however approximate, =
would
>>   be helpful.
>=20
> This is "fair" of course.   I'm not sure how I can say much more =
without constraining the solution-space artificially.  And it's worse =
because it's not a single-point measurement; defining it in detail would =
require looking at startup ramp, steady-state, reaction to BW changes =
and cross-traffic (does it remain "roughly equal"), etc.  Down this road =
lives insanity=85

Of course.

> I can try alternate wiggle-words, or some additional somewhat-vague =
phrases.

Add a =93It is difficult to give a definition of =91roughly equal=92 =
because=85=94 to make it clear why the definition is imprecise? I=92m =
less concerned that the definitions are vague, than that they=92re vague =
without explanation of the factors that affect them.

>> - Requirement 2A is written in terms of "a useful share" of =
bandwidth.
>>   Again, defining "useful" would help: perhaps for the flow to ramp =
up to
>>   a bandwidth that allows it to support a minimal quality media =
stream?
>=20
> A "minimal quality media stream" may be 8Kbps (or less!) or it might =
be 8Mbps, so that makes it hard to define here=85.

True, but the issue ought to be discussed. Quickly ramping up to get a =
useful VoIP flow is straight-forward (as you say, 8kbps), but it=92s a =
much harder problem for higher rate media. This draft should be setting =
expectations that the ramp-up will take longer the higher your useful =
rate, and that doesn=92t come across now.

>> - Requirement 3A is unclear. Is the intent that there is no =
requirement
>>   that the algorithm compete equally with TCP flows that have =
different
>>   RTT?
>=20
> 3A says "don't be a hog" (likely not a problem anyways, but since once =
an algorithm has "given up" on low delay due to TCP competition you =
don't want it to starve the TCP streams by going CBR for example.)  And =
for long-lived flows, it can=92t compete equally even with flows with =
the same RTT.

I got that from 3, but don=92t see how 3A adds to it. I=92d possibly =
suggest removing 3A, since I think it=92s covered elsewhere in 2.

>> - Requirement 3B is unclear. Is the requirement that "the congestion
>>   control should prioritise achieving a useful (as defined in =
Requirement
>>   2A) share of the bandwidth over achieving as low as possible =
transit
>>   delay, when these two requirements are in conflict"?
>=20
> That might be a nice way to state that, thanks.
>=20
>> - Requirements 4A, 4B, and 4C seem to describe possible behaviours of =
the
>>   system, but the actual requirements are unclear. Requirment 4A =
could be
>>   that the algorithm must reach a useful sending rate for audio calls =
fast
>>   enough that the initial "Hello" is not clipped; it's not clear how =
this
>>   applies to other media.
>=20
> Those were to push the solutions in the direction of having a =
reasonable start-of-call/stream behavior, which otherwise might get =
ignored and not looked at or optimized for.  I agree they=92re somewhat =
editorial.

Pushing solutions in that direction makes sense, but to fit the way this =
draft is structured as a set of requirements, I think these need to be =
rephrased.

>> - Requirement 5 is for "stable" behaviour, but it is not clear what
>>   stability means with discontinuous transmission, since the media =
will
>>   have an on-off profile that is not stable.
>=20
> =93must deal with"?

Maybe, although might be better to just merge with 5A.

>> - Requirement 5A lists possible behaviour for the algorithm, but it =
is not
>>   clear what requirement is being placed on an implementation. It =
"may
>>   adapt more quickly", but is it required to do so? Previous =
bandwidth
>>   estimates "may need to be aged or thrown away", but are they =
required
>>   to be?
>=20
> <t>After stream resumption, the algorithm should attempt to regain a =
share of bandwidth as quickly as possible.</t>
> ("useful share"?)  *ducks*

=93After stream resumption, the algorithm should attempt to rapidly =
regain its previous share of the bandwidth; the aggressiveness with =
which this is done will decay with the length of the pause=94??

>> - Requirement 6 could be interpreted as requiring general multipath
>>   congestion control. Is that the intent, or is the goal only to =
share
>>   information between flows between two endpoint when those flows =
share
>>   a common bottleneck?
>=20
> The latter.  I thought it was clear; wording suggestions?

=93across multiple RTP streams sent between two endpoints, when those =
RTP streams share a common bottleneck, whether or not those streams are =
multiplexed onto the same ports=94?

>> - Requirement 6C gives a possible use for information, but it's not =
clear
>>   what requirement is being placed on implementations.
>=20
> Right, it=92s editorial reminding them they can do that.

Sure, but it=92s in the requirements section, so needs to be phrased as =
such. Suggest changing =93it should be possible for the application to =
control=94 to =93the algorithm should allow the application to control=94

>> - Requirement 8 talks about mechanisms, using RTCP feedback and/or =
header
>>   extensions. It's appropriate that this draft gives give general =
feedback
>>   requirements, but I don't think it should be making explicit =
requirements
>>   on how RTCP or RTP header extensions are used without =
justification,
>>   especially since the RMCAT working group has a milestone to produce =
a
>>   draft giving such recommendations. I suggest keeping requirement =
8E,
>>   but removing the rest of requirement 8.
>=20
> Ok.
>=20
>>=20
>>=20
>> I also noted some editorial issues:
>>=20
>> - The RMCAT working group should have a limited lifetime. =
Accordingly,
>>   the title might be better written as "Congestion Control =
Requirements
>>   for Interactive Real-time Media Transport=94?
>=20
> Sure
>=20
>>=20
>> - The 4th paragraph of Section 1 talks about "requirements for RMCAT" =
and
>>   "The RMCAT congestion control algorithm". I suggest these be =
changed to
>>   "requirements for congestion control of interactive real-time =
media" and
>>   "A congestion control algorithm for interactive real-time media".
>=20
> Sure
>=20
>>=20
>> - The last paragraph of Section 1 states that terminology from the =
WebRTC
>>   overview is used. This is appropriate. It might also be appropriate =
to
>>   align with draft-ietf-avtext-rtp-grouping-taxonomy and cite that =
draft.
>=20
> I think we can do that...  (I'll see how much editing it needs to do =
so)
>=20
>> - Requirement 1E talks about "short-term bursts (such as =
web-browsing)".
>>   This would be clearer if written "short-term bursts of competing =
traffic
>>   (such as web browsing)", since the bursts are not due to the =
congestion
>>   control algorithm.
>=20
> Sure.
>=20
> --=20
> Randell Jesup -- rjesup a t mozilla d o t com
>=20



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




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On 8/20/2014 8:14 AM, Colin Perkins wrote:
>
>>> - Requirement 1B states that the algorithm should react "quickly" to any
>>>    reduction in bandwidth or increase in delay. The draft needs to define
>>>    what quickly means in this context, since the expected timing could be
>>>    different to other congestion control algorithms. Is the expectation
>>>    that the response occurs with an RTT, or with next video frame, GoP, or
>>>    talkspurt? This comment also applies to requirements 1C, 1E, 1F, and 4.
>> I'm loathe to give a hard number here, and instead leave that to the evaluation, since as with all things like this it's a tradeoff.  I could state that "quickly" means a timescale on the order of RTT times (i.e. O(RTT)), and certainly well sub-second.  (IMHO, a congestion mechanism needs to adapt down ASAP to avoid delay buildup it will then have to drain out).
> I agree that we don’t want to give a hard number, but I think some guidelines to set expectations would be useful. React within some small multiple of the RTT would seem appropriate, possibly with some wording about possible codec constraints also (e.g., react on the next video frame, or maybe at the start of the next talk spurt)? Agree that sub-second reaction is desirable.
>
> Saying “In this memo, a ‘quick’ reaction to congestion means…” would then cover the later uses of the term.

Ok, I'll do that.

>
>>> - Requirement 1B uses an "increase in bottleneck delay" as a congestion
>>>    signal, but it's not clear how the congestion control can distinguish an
>>>    increase in delay at the bottleneck from an increase in delay elsewhere.
>>>    Perhaps this should be “an increase in observed delay”?
>> Sure.
>>
>>> - Requirement 1C requires the congestion controller to react to interface
>>>    changes. Does this apply to changes in the local interface only, or to
>>>    changes in either the local or remote interface? One is a local matter,
>>>    while the other requires signalling, so the distinction may be important.
>> I would say local interface changes should be noticed and both directions should take this into account if possible (for example as a sender by restarting adaptation from a known startpoint, or increase sensitivity/adaptation-rate temporarily in combination with an immediate drop in send rate as a precaution).  On the receive side, it can indicate an interface change via RTCP, or just let the other side adapt if the bottleneck has changed.  (Indicating a change may be preferable in order to more-intelligently adapt, but may not be required, and might not even help or help much.)
>>
>> Really I’m interested to see what the proposals find are the best way to handle this problem, without constricting the solution space.
> Agree, and agree that those are likely mechanisms for doing this.
>
>> Suggested wording changes?
> Maybe something like “It should detect handover between different interfaces (e.g., Wi-Fi to 3G) at sender or receiver, and signal these to the congestion control algorithm where possible, to help the algorithm respond quickly to such path changes”?

Ok, I'll come up with something like that.

>
>>> - Requirement 1E states that "the algorithm must not overreact". Can you
>>>    define (even approximately) what overreacting means in this context?
>>>    Without this definition, it's not clear how this requirement could be
>>>    translated into protocol behaviour. Similarly, what is "short-term"?
>> "the algorithm must deal with traffic bursts caused by actions such as webpage loading [by an aggressive browser/website combination]"?   (with the part in [] being perhaps removable). Again, the definition of “quickly" comes into play.
> Perhaps “Short-term bursts of competing traffic (e.g., due to local web browser use) can saturate a local bottleneck link, but also clear quickly. The algorithm needs to be able respond quickly to such events to help prevent queue build-up; it is desirable if it can also quickly recover its sending rate when they conclude”?

That sounds pretty good, thanks

>
>>> - Requirement 1F states that the algorithm "must avoid too much delay
>>>    buildup during those bursts". This is, at least partly, outside the
>>>    control of the algorithm, since it depends on the behaviour of the
>>>    cross traffic. Perhaps the requirement ought to be that the algorithm
>>>    does not significantly increase the delay buildup during the bursts?
>> Perhaps that’s a good rewording; that's a tough/painful case since they may smoke your router's buffers regardless of what you do, so you may want to keep sending at a "normal" rate or close once the algorithm realizes it's lost the battle (temporarily).
> Sure, it’s difficult. If it’s a transient burst that fills the queue for a short time then disappears, then you want to back off to avoid making the situation worse, then continue, but if the competing traffic is determined to keep the queue full, you’re better keeping sending and accepting the latency, and let the user hang-up if it gets too bad.

Right.  Maybe I'll combine your original and some of that.

>
>>> - Requirement 2 states that flows should get "roughly equal" shares of
>>>    bandwidth, but gives no definition of “roughly equal". I realise that
>>>    this is a difficult issue, but some guidance, however approximate, would
>>>    be helpful.
>> This is "fair" of course.   I'm not sure how I can say much more without constraining the solution-space artificially.  And it's worse because it's not a single-point measurement; defining it in detail would require looking at startup ramp, steady-state, reaction to BW changes and cross-traffic (does it remain "roughly equal"), etc.  Down this road lives insanity…
> Of course.
>
>> I can try alternate wiggle-words, or some additional somewhat-vague phrases.
> Add a “It is difficult to give a definition of ‘roughly equal’ because…” to make it clear why the definition is imprecise? I’m less concerned that the definitions are vague, than that they’re vague without explanation of the factors that affect them.

Sure, sounds good.

>
>>> - Requirement 2A is written in terms of "a useful share" of bandwidth.
>>>    Again, defining "useful" would help: perhaps for the flow to ramp up to
>>>    a bandwidth that allows it to support a minimal quality media stream?
>> A "minimal quality media stream" may be 8Kbps (or less!) or it might be 8Mbps, so that makes it hard to define here….
> True, but the issue ought to be discussed. Quickly ramping up to get a useful VoIP flow is straight-forward (as you say, 8kbps), but it’s a much harder problem for higher rate media. This draft should be setting expectations that the ramp-up will take longer the higher your useful rate, and that doesn’t come across now.

Ok, how about "a useful share of the bandwidth" and "A useful share will 
depend on the media types involved and total bandwidth available, but 
should allow reasonable communication quickly assuming similar media 
types and required bandwidths".  Also, it's more about the existing 
flows ramping down fast enough so that the new flow (which will likely 
start at a "reasonable minimum" bandwidth) doesn't cause all the flows 
to start seeing delay and/or loss.  Likely applications using this will 
lower-limit the bandwidth and when forced down there will live with 
extra delay (i.e. the compete with greedy TCP case), and if delay is 
egregious or if loss builds up badly as well then it may decide to drop 
that stream or indicate connection failure.

>>> - Requirement 3A is unclear. Is the intent that there is no requirement
>>>    that the algorithm compete equally with TCP flows that have different
>>>    RTT?
>> 3A says "don't be a hog" (likely not a problem anyways, but since once an algorithm has "given up" on low delay due to TCP competition you don't want it to starve the TCP streams by going CBR for example.)  And for long-lived flows, it can’t compete equally even with flows with the same RTT.
> I got that from 3, but don’t see how 3A adds to it. I’d possibly suggest removing 3A, since I think it’s covered elsewhere in 2.

ok

>
>>> - Requirements 4A, 4B, and 4C seem to describe possible behaviours of the
>>>    system, but the actual requirements are unclear. Requirment 4A could be
>>>    that the algorithm must reach a useful sending rate for audio calls fast
>>>    enough that the initial "Hello" is not clipped; it's not clear how this
>>>    applies to other media.
>> Those were to push the solutions in the direction of having a reasonable start-of-call/stream behavior, which otherwise might get ignored and not looked at or optimized for.  I agree they’re somewhat editorial.
> Pushing solutions in that direction makes sense, but to fit the way this draft is structured as a set of requirements, I think these need to be rephrased.

I'll take a shot at that.

>
>>> - Requirement 5 is for "stable" behaviour, but it is not clear what
>>>    stability means with discontinuous transmission, since the media will
>>>    have an on-off profile that is not stable.
>> “must deal with"?
> Maybe, although might be better to just merge with 5A.
>
>>> - Requirement 5A lists possible behaviour for the algorithm, but it is not
>>>    clear what requirement is being placed on an implementation. It "may
>>>    adapt more quickly", but is it required to do so? Previous bandwidth
>>>    estimates "may need to be aged or thrown away", but are they required
>>>    to be?
>> <t>After stream resumption, the algorithm should attempt to regain a share of bandwidth as quickly as possible.</t>
>> ("useful share"?)  *ducks*
> “After stream resumption, the algorithm should attempt to rapidly regain its previous share of the bandwidth; the aggressiveness with which this is done will decay with the length of the pause”??

That's better.  (I'd say "may decay" or "should decay" to avoid 
hard-restricting the solutions).

>
>>> - Requirement 6 could be interpreted as requiring general multipath
>>>    congestion control. Is that the intent, or is the goal only to share
>>>    information between flows between two endpoint when those flows share
>>>    a common bottleneck?
>> The latter.  I thought it was clear; wording suggestions?
> “across multiple RTP streams sent between two endpoints, when those RTP streams share a common bottleneck, whether or not those streams are multiplexed onto the same ports”?

ok

>
>>> - Requirement 6C gives a possible use for information, but it's not clear
>>>    what requirement is being placed on implementations.
>> Right, it’s editorial reminding them they can do that.
> Sure, but it’s in the requirements section, so needs to be phrased as such. Suggest changing “it should be possible for the application to control” to “the algorithm should allow the application to control”

ok


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


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Hi Randell,

That all sounds good - thanks!

Colin


On 26 Aug 2014, at 04:40, Randell Jesup <randell-ietf@jesup.org> wrote:
> On 8/20/2014 8:14 AM, Colin Perkins wrote:
>>=20
>>>> - Requirement 1B states that the algorithm should react "quickly" =
to any
>>>>   reduction in bandwidth or increase in delay. The draft needs to =
define
>>>>   what quickly means in this context, since the expected timing =
could be
>>>>   different to other congestion control algorithms. Is the =
expectation
>>>>   that the response occurs with an RTT, or with next video frame, =
GoP, or
>>>>   talkspurt? This comment also applies to requirements 1C, 1E, 1F, =
and 4.
>>> I'm loathe to give a hard number here, and instead leave that to the =
evaluation, since as with all things like this it's a tradeoff.  I could =
state that "quickly" means a timescale on the order of RTT times (i.e. =
O(RTT)), and certainly well sub-second.  (IMHO, a congestion mechanism =
needs to adapt down ASAP to avoid delay buildup it will then have to =
drain out).
>> I agree that we don=92t want to give a hard number, but I think some =
guidelines to set expectations would be useful. React within some small =
multiple of the RTT would seem appropriate, possibly with some wording =
about possible codec constraints also (e.g., react on the next video =
frame, or maybe at the start of the next talk spurt)? Agree that =
sub-second reaction is desirable.
>>=20
>> Saying =93In this memo, a =91quick=92 reaction to congestion means=85=94=
 would then cover the later uses of the term.
>=20
> Ok, I'll do that.
>=20
>>=20
>>>> - Requirement 1B uses an "increase in bottleneck delay" as a =
congestion
>>>>   signal, but it's not clear how the congestion control can =
distinguish an
>>>>   increase in delay at the bottleneck from an increase in delay =
elsewhere.
>>>>   Perhaps this should be =93an increase in observed delay=94?
>>> Sure.
>>>=20
>>>> - Requirement 1C requires the congestion controller to react to =
interface
>>>>   changes. Does this apply to changes in the local interface only, =
or to
>>>>   changes in either the local or remote interface? One is a local =
matter,
>>>>   while the other requires signalling, so the distinction may be =
important.
>>> I would say local interface changes should be noticed and both =
directions should take this into account if possible (for example as a =
sender by restarting adaptation from a known startpoint, or increase =
sensitivity/adaptation-rate temporarily in combination with an immediate =
drop in send rate as a precaution).  On the receive side, it can =
indicate an interface change via RTCP, or just let the other side adapt =
if the bottleneck has changed.  (Indicating a change may be preferable =
in order to more-intelligently adapt, but may not be required, and might =
not even help or help much.)
>>>=20
>>> Really I=92m interested to see what the proposals find are the best =
way to handle this problem, without constricting the solution space.
>> Agree, and agree that those are likely mechanisms for doing this.
>>=20
>>> Suggested wording changes?
>> Maybe something like =93It should detect handover between different =
interfaces (e.g., Wi-Fi to 3G) at sender or receiver, and signal these =
to the congestion control algorithm where possible, to help the =
algorithm respond quickly to such path changes=94?
>=20
> Ok, I'll come up with something like that.
>=20
>>=20
>>>> - Requirement 1E states that "the algorithm must not overreact". =
Can you
>>>>   define (even approximately) what overreacting means in this =
context?
>>>>   Without this definition, it's not clear how this requirement =
could be
>>>>   translated into protocol behaviour. Similarly, what is =
"short-term"?
>>> "the algorithm must deal with traffic bursts caused by actions such =
as webpage loading [by an aggressive browser/website combination]"?   =
(with the part in [] being perhaps removable). Again, the definition of =
=93quickly" comes into play.
>> Perhaps =93Short-term bursts of competing traffic (e.g., due to local =
web browser use) can saturate a local bottleneck link, but also clear =
quickly. The algorithm needs to be able respond quickly to such events =
to help prevent queue build-up; it is desirable if it can also quickly =
recover its sending rate when they conclude=94?
>=20
> That sounds pretty good, thanks
>=20
>>=20
>>>> - Requirement 1F states that the algorithm "must avoid too much =
delay
>>>>   buildup during those bursts". This is, at least partly, outside =
the
>>>>   control of the algorithm, since it depends on the behaviour of =
the
>>>>   cross traffic. Perhaps the requirement ought to be that the =
algorithm
>>>>   does not significantly increase the delay buildup during the =
bursts?
>>> Perhaps that=92s a good rewording; that's a tough/painful case since =
they may smoke your router's buffers regardless of what you do, so you =
may want to keep sending at a "normal" rate or close once the algorithm =
realizes it's lost the battle (temporarily).
>> Sure, it=92s difficult. If it=92s a transient burst that fills the =
queue for a short time then disappears, then you want to back off to =
avoid making the situation worse, then continue, but if the competing =
traffic is determined to keep the queue full, you=92re better keeping =
sending and accepting the latency, and let the user hang-up if it gets =
too bad.
>=20
> Right.  Maybe I'll combine your original and some of that.
>=20
>>=20
>>>> - Requirement 2 states that flows should get "roughly equal" shares =
of
>>>>   bandwidth, but gives no definition of =93roughly equal". I =
realise that
>>>>   this is a difficult issue, but some guidance, however =
approximate, would
>>>>   be helpful.
>>> This is "fair" of course.   I'm not sure how I can say much more =
without constraining the solution-space artificially.  And it's worse =
because it's not a single-point measurement; defining it in detail would =
require looking at startup ramp, steady-state, reaction to BW changes =
and cross-traffic (does it remain "roughly equal"), etc.  Down this road =
lives insanity=85
>> Of course.
>>=20
>>> I can try alternate wiggle-words, or some additional somewhat-vague =
phrases.
>> Add a =93It is difficult to give a definition of =91roughly equal=92 =
because=85=94 to make it clear why the definition is imprecise? I=92m =
less concerned that the definitions are vague, than that they=92re vague =
without explanation of the factors that affect them.
>=20
> Sure, sounds good.
>=20
>>=20
>>>> - Requirement 2A is written in terms of "a useful share" of =
bandwidth.
>>>>   Again, defining "useful" would help: perhaps for the flow to ramp =
up to
>>>>   a bandwidth that allows it to support a minimal quality media =
stream?
>>> A "minimal quality media stream" may be 8Kbps (or less!) or it might =
be 8Mbps, so that makes it hard to define here=85.
>> True, but the issue ought to be discussed. Quickly ramping up to get =
a useful VoIP flow is straight-forward (as you say, 8kbps), but it=92s a =
much harder problem for higher rate media. This draft should be setting =
expectations that the ramp-up will take longer the higher your useful =
rate, and that doesn=92t come across now.
>=20
> Ok, how about "a useful share of the bandwidth" and "A useful share =
will depend on the media types involved and total bandwidth available, =
but should allow reasonable communication quickly assuming similar media =
types and required bandwidths".  Also, it's more about the existing =
flows ramping down fast enough so that the new flow (which will likely =
start at a "reasonable minimum" bandwidth) doesn't cause all the flows =
to start seeing delay and/or loss.  Likely applications using this will =
lower-limit the bandwidth and when forced down there will live with =
extra delay (i.e. the compete with greedy TCP case), and if delay is =
egregious or if loss builds up badly as well then it may decide to drop =
that stream or indicate connection failure.
>=20
>>>> - Requirement 3A is unclear. Is the intent that there is no =
requirement
>>>>   that the algorithm compete equally with TCP flows that have =
different
>>>>   RTT?
>>> 3A says "don't be a hog" (likely not a problem anyways, but since =
once an algorithm has "given up" on low delay due to TCP competition you =
don't want it to starve the TCP streams by going CBR for example.)  And =
for long-lived flows, it can=92t compete equally even with flows with =
the same RTT.
>> I got that from 3, but don=92t see how 3A adds to it. I=92d possibly =
suggest removing 3A, since I think it=92s covered elsewhere in 2.
>=20
> ok
>=20
>>=20
>>>> - Requirements 4A, 4B, and 4C seem to describe possible behaviours =
of the
>>>>   system, but the actual requirements are unclear. Requirment 4A =
could be
>>>>   that the algorithm must reach a useful sending rate for audio =
calls fast
>>>>   enough that the initial "Hello" is not clipped; it's not clear =
how this
>>>>   applies to other media.
>>> Those were to push the solutions in the direction of having a =
reasonable start-of-call/stream behavior, which otherwise might get =
ignored and not looked at or optimized for.  I agree they=92re somewhat =
editorial.
>> Pushing solutions in that direction makes sense, but to fit the way =
this draft is structured as a set of requirements, I think these need to =
be rephrased.
>=20
> I'll take a shot at that.
>=20
>>=20
>>>> - Requirement 5 is for "stable" behaviour, but it is not clear what
>>>>   stability means with discontinuous transmission, since the media =
will
>>>>   have an on-off profile that is not stable.
>>> =93must deal with"?
>> Maybe, although might be better to just merge with 5A.
>>=20
>>>> - Requirement 5A lists possible behaviour for the algorithm, but it =
is not
>>>>   clear what requirement is being placed on an implementation. It =
"may
>>>>   adapt more quickly", but is it required to do so? Previous =
bandwidth
>>>>   estimates "may need to be aged or thrown away", but are they =
required
>>>>   to be?
>>> <t>After stream resumption, the algorithm should attempt to regain a =
share of bandwidth as quickly as possible.</t>
>>> ("useful share"?)  *ducks*
>> =93After stream resumption, the algorithm should attempt to rapidly =
regain its previous share of the bandwidth; the aggressiveness with =
which this is done will decay with the length of the pause=94??
>=20
> That's better.  (I'd say "may decay" or "should decay" to avoid =
hard-restricting the solutions).
>=20
>>=20
>>>> - Requirement 6 could be interpreted as requiring general multipath
>>>>   congestion control. Is that the intent, or is the goal only to =
share
>>>>   information between flows between two endpoint when those flows =
share
>>>>   a common bottleneck?
>>> The latter.  I thought it was clear; wording suggestions?
>> =93across multiple RTP streams sent between two endpoints, when those =
RTP streams share a common bottleneck, whether or not those streams are =
multiplexed onto the same ports=94?
>=20
> ok
>=20
>>=20
>>>> - Requirement 6C gives a possible use for information, but it's not =
clear
>>>>   what requirement is being placed on implementations.
>>> Right, it=92s editorial reminding them they can do that.
>> Sure, but it=92s in the requirements section, so needs to be phrased =
as such. Suggest changing =93it should be possible for the application =
to control=94 to =93the algorithm should allow the application to =
control=94
>=20
> ok
>=20
>=20
> --=20
> Randell Jesup -- rjesup a t mozilla d o t com
>=20



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




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Subject: [rmcat] Physical interim meeting (in Zurich or Munich)
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Hi w-g,

as discussed at the last meeting, we would like to hold an intermin 
meeting. We propose a physical meeting beginning of October. This will 
probably be a meeting with two half days, so people can fly in in the 
morning and leave the next afternoon/evening. There are two options. We 
can choose whichever is more suitable for most of the people:

* Zurich, Oct 8/9 (at ETH in the city center)

* Munich, between 6-9 (at NetApp, 20 Minutes out of the city but good 
train connection)

Please let us know as soon as possible if you would be able to attend 
(and which date would be best for you). If we get a sufficient number of 
replies we will set up the logistics next.

Thanks,
your chairs


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Hi agian,

just to clarify the dates; sorry if this was confusing

* Zurich: Oct 8/9 (fixed)

* Munich: one or two days the same week between Oct 6 and Oct 9

Mirja



On 29.08.2014 11:31, Mirja KÃ¼hlewind wrote:
> Hi w-g,
>
> as discussed at the last meeting, we would like to hold an intermin
> meeting. We propose a physical meeting beginning of October. This will
> probably be a meeting with two half days, so people can fly in in the
> morning and leave the next afternoon/evening. There are two options. We
> can choose whichever is more suitable for most of the people:
>
> * Zurich, Oct 8/9 (at ETH in the city center)
>
> * Munich, between 6-9 (at NetApp, 20 Minutes out of the city but good
> train connection)
>
> Please let us know as soon as possible if you would be able to attend
> (and which date would be best for you). If we get a sufficient number of
> replies we will set up the logistics next.
>
> Thanks,
> your chairs
>
>

-- 
------------------------------------------
Dipl.-Ing. Mirja KÃ¼hlewind
Communication Systems Group
Institute TIK, ETH ZÃ¼rich
Gloriastrasse 35, 8092 ZÃ¼rich, Switzerland

Room ETZ G93
phone: +41 44 63 26932
email: mirja.kuehlewind@tik.ee.ethz.ch
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