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To: Stanislav Shalunov <shalunov@bittorrent.com>
From: Bob Briscoe <rbriscoe@jungle.bt.co.uk>
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Cc: ledbat@ietf.org
Subject: [ledbat] Hard-coded delay target? draft-shalunov-ledbat-congestion-00
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Stas,

This posting repeats what I said in the ledbat=20
w-g meeting in SF, but to the whole list and with=20
added reasoning, given the benefit of more time for thought.

On balance it seems wrong to aim for a hard-coded=20
delay target (25ms). I understand your reasoning=20
for it in the draft (dividing delay between=20
competing flows rather than adding), but there=20
are also stronger scaling reasons against it...

For 1500B* packets, 25ms represents a standing queue of:
    1 pkt @ 512kb/s
   17 pkt @   8Mb/s
   50 pkt @  24Mb/s (some backhaul links today)
  208 pkt @ 100Mb/s (typical PON access rate today)
2080 pkt @   1Gb/s (increasingly backhaul links are GigE)

As bottleneck links get faster (magically this is=20
happening!), we will want to reduce the=20
hard-coded delay (to save on fast buffer memory and the power it requires).

But if the the algo in draft-00 were changed in=20
the future to aim at 10ms (while the installed=20
base all still aimed at 25ms), the new code would=20
always be starved by the installed base.=20
Therefore we would be stuck forever with the=20
number we first thought of: 25ms hard-coded.

Spelling this out longhand, imagine two flows competing:
         Flow N (New)    (10ms target)
         Flow L (Legacy) (25ms target)

L drives delay to 25ms
while both flows have packets to send {
     N reduces rate because delay greater than 10ms
     L sees delay decrease so increases its rate to drive delay to 25ms
}
Result: newer algorithm always starves.

IMHO, the algo needs to aim at minimising the=20
congestion delay, not aim at a fixed target (the=20
draft says the former but does the latter).=20
Section 3 onwards of [Key04] may help, even tho=20
it's in the context of adapting the receive=20
window it's still based on delay sensing. I'd be=20
interested to know whether Microsoft's BITS now=20
implements something like this algo.


Bob

[* =3D This might be fine if packet size scaled=20
with link speeds, but that isn't happening.]

[Key04] Peter Key, Laurent Massouli=E9 and Bing=20
Wang, "Emulating low-priority transport at the=20
application layer: a background transfer service"=20
In: "ACM SIGMETRICS '04/Performance '04: Proceedings of the joint=
 international
conference on Measurement and modeling of=20
computer systems", pp. 118--129=20
<http://research.microsoft.com/apps/pubs/default.aspx?id=3D67381> (2004)




________________________________________________________________
Bob Briscoe,               Networks Research Centre, BT Research=20


From shalunov@bittorrent.com  Thu Apr  2 23:10:20 2009
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Subject: Re: [ledbat] Hard-coded delay target? draft-shalunov-ledbat-congestion-00
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Bob,

The choice of parameters for congestion control has a degree to which
it's arbitrary.  Why does TCP ramp up at a hard-coded 1 MSS/RTT?  Why
not 3.5 or 0.6?  Why does it multiply by 0.5 instead of 7/8 or 0.2 or
0?

If a future version of TCP chooses 0.2 and 0.3 as its favorite lucky
numbers while the installed base is at 1 and 0.5, the consequences are
similar.

While the choice of the particular parametrization for TCP might be
argued to be suboptimal because it doesn't scale with link speed, I'm
not aware of any congestion control mechanism without parameters.

The mechanism we use scales the target queue linearly with link speed,
as you point out.  It's a feature.  It enables the mechanism to work
on a variety of bottleneck speeds.  If you specify the target in
bytes, you lose on fast links because your delay starts getting lower
than scheduling noise and/or PCI bus jitter and you lose on slow links
because the delay is too large for interactive apps.

As choice of parameters goes, the choice of the gain is much more
arbitrary than the choice of the target.  The gain simply matches TCP
in the worst case.  Target is essentially the lowest value than can be
reliably distinguished from 0 on a variety of machines under a variety
of circumstances.

As a practical matter, 25ms is a substantial reduction from the 1RTT
rule of thumb and so it's reasonable to expect that the deployed gear
will have this much delay.  Usually it has far more.  (Even at 10Gb/s
and up.)  If some gear at an atypical bottleneck doesn't have this
much, the behavior degenerates into TCP.

It is also a fairly unsubstantial increase relative to RTT and will
remain such unless progress is made on the speed-of-light problem (or
the planet is shrunk?).

The work you cite never seems to mention minimizing delay as a goal.
It discusses the queuing delay achieved in Theorem 3.  And there's a
parameter that controls it (alpha_b^*).

-- Stas


On Wed, Apr 1, 2009 at 3:46 AM, Bob Briscoe <rbriscoe@jungle.bt.co.uk> wrot=
e:
> Stas,
>
> This posting repeats what I said in the ledbat w-g meeting in SF, but to =
the
> whole list and with added reasoning, given the benefit of more time for
> thought.
>
> On balance it seems wrong to aim for a hard-coded delay target (25ms). I
> understand your reasoning for it in the draft (dividing delay between
> competing flows rather than adding), but there are also stronger scaling
> reasons against it...
>
> For 1500B* packets, 25ms represents a standing queue of:
> =A0 1 pkt @ 512kb/s
> =A017 pkt @ =A0 8Mb/s
> =A050 pkt @ =A024Mb/s (some backhaul links today)
> =A0208 pkt @ 100Mb/s (typical PON access rate today)
> 2080 pkt @ =A0 1Gb/s (increasingly backhaul links are GigE)
>
> As bottleneck links get faster (magically this is happening!), we will wa=
nt
> to reduce the hard-coded delay (to save on fast buffer memory and the pow=
er
> it requires).
>
> But if the the algo in draft-00 were changed in the future to aim at 10ms
> (while the installed base all still aimed at 25ms), the new code would
> always be starved by the installed base. Therefore we would be stuck fore=
ver
> with the number we first thought of: 25ms hard-coded.
>
> Spelling this out longhand, imagine two flows competing:
> =A0 =A0 =A0 =A0Flow N (New) =A0 =A0(10ms target)
> =A0 =A0 =A0 =A0Flow L (Legacy) (25ms target)
>
> L drives delay to 25ms
> while both flows have packets to send {
> =A0 =A0N reduces rate because delay greater than 10ms
> =A0 =A0L sees delay decrease so increases its rate to drive delay to 25ms
> }
> Result: newer algorithm always starves.
>
> IMHO, the algo needs to aim at minimising the congestion delay, not aim a=
t a
> fixed target (the draft says the former but does the latter). Section 3
> onwards of [Key04] may help, even tho it's in the context of adapting the
> receive window it's still based on delay sensing. I'd be interested to kn=
ow
> whether Microsoft's BITS now implements something like this algo.
>
>
> Bob
>
> [* =3D This might be fine if packet size scaled with link speeds, but tha=
t
> isn't happening.]
>
> [Key04] Peter Key, Laurent Massouli=E9 and Bing Wang, "Emulating low-prio=
rity
> transport at the application layer: a background transfer service" In: "A=
CM
> SIGMETRICS '04/Performance '04: Proceedings of the joint international
> conference on Measurement and modeling of computer systems", pp. 118--129
> <http://research.microsoft.com/apps/pubs/default.aspx?id=3D67381> (2004)
>
>
>
>
> ________________________________________________________________
> Bob Briscoe, =A0 =A0 =A0 =A0 =A0 =A0 =A0 Networks Research Centre, BT Res=
earch
>



--=20
Stanislav Shalunov
BitTorrent Inc
shalunov@bittorrent.com

personal: http://shlang.com

From shalunov@bittorrent.com  Thu Apr  2 23:28:51 2009
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Subject: Re: [ledbat] fairness between competing delay flows in draft-shalunov-ledbat-congestion-00.txt
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We have not observed the late-comer's advantage.

As soon as the first flow has a hiccup (for example, because something
needs to wait for a few disk head moves), there will be a gap in the
first flow's traffic, enabling the second flow to measure accurately.

During the session Murari mentioned that in their experience, flows
were also always able to get a reasonable base delay measurement.

Obviously, the more flows, the more uniform the traffic, the more
likely it is that the late comers might be mistaken about the base.
At tens of flows, this is clearly not a problem.  At tens of
thousands, which is what most backbones see for bulk connections, I'd
expect it to start showing.  The coordination problem at tens of
thousands of connections is hard and almost certain to be emergent,
but then extended congestion on backbone links is infrequent.  In the
worst case, it'll all fall back to TCP.

For a small number of flows, the situation most typically encountered
on congested links, base estimation seems to be reliable.


On Tue, Mar 31, 2009 at 5:23 PM, Douglas Leith <Doug.Leith@nuim.ie> wrote:
> Can I ask what the expected/desired behaviour will be when two or more
> ledbat flows share a link ?
> For example, say a first flow starts, ramps us cwnd until it hits the tar=
get
> queueing delay and then a second ledbat flow starts a while later. =A0Its=
 not
> clear to me from the ID how the first flow would alter its cwnd to make
> space for the second ledbat flow (especially if the second flow does not =
use
> slow start, or perhaps has an early exit from slow start).
> Also, is there a danger that, similarly the vegas and related algorithms,
> the later starting flow will have difficulty getting an accurate estimate=
 of
> the base RTT since earlier starting flows maintain a standing queue ?
> Doug
> www.hamilton.ie



--=20
Stanislav Shalunov
BitTorrent Inc
shalunov@bittorrent.com

personal: http://shlang.com

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Subject: Re: [ledbat] Hard-coded delay target? draft-shalunov-ledbat-congestion-00
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On Fri, Apr 3, 2009 at 8:11 AM, Stanislav Shalunov
<shalunov@bittorrent.com>wrote:

> Bob,
>
> The choice of parameters for congestion control has a degree to which
> it's arbitrary.  Why does TCP ramp up at a hard-coded 1 MSS/RTT?


here i have an answer that it is important to have in mind, especially
because of many TCP variants that try a more aggressive probing (some of
them replacing TCP NewReno).

The  increasing at 1 MSS/RTT is the *unique* increase with  the property
that  *only one packet is lost* when  the path capacity is hit. This becaus=
e
of  a feedback lag equal to RTT, i.e. after 1 RTT TCP  knows if the capacit=
y
has been hit. This 1 MSS/RTT increment has the very nice property that only
one lost packet needs to be retransmitted.

TCP variants trying a more aggressive probing experience several packet
losses when the capacity is hit. So what happens is that the throughput
increases but also retransmissions so that you end up with same goodput but
with higher retransmissions, which is not good.


These considerations imply that when setting:

cwnd+=3Dgain*(T_target-T_forward)

it is feasible to choose a low gain in order  to have a TCP less than best
effort but it is difficult  to set a larger gain to have a quick probing in
case of gigabit nets without ending in the problem before mentioned (we
tried it sometime ago and we give up)!

Why
> not 3.5 or 0.6?  Why does it multiply by 0.5 instead of 7/8 or 0.2 or
> 0?


again, here  a less than best effort can be:

cwnd=3Dmin(cwnd/2,BWE*base_rtt)

saverio



> If a future version of TCP chooses 0.2 and 0.3 as its favorite lucky
> numbers while the installed base is at 1 and 0.5, the consequences are
> similar.
>
> While the choice of the particular parametrization for TCP might be
> argued to be suboptimal because it doesn't scale with link speed, I'm
> not aware of any congestion control mechanism without parameters.
>
> The mechanism we use scales the target queue linearly with link speed,
> as you point out.  It's a feature.  It enables the mechanism to work
> on a variety of bottleneck speeds.  If you specify the target in
> bytes, you lose on fast links because your delay starts getting lower
> than scheduling noise and/or PCI bus jitter and you lose on slow links
> because the delay is too large for interactive apps.
>
> As choice of parameters goes, the choice of the gain is much more
> arbitrary than the choice of the target.  The gain simply matches TCP
> in the worst case.  Target is essentially the lowest value than can be
> reliably distinguished from 0 on a variety of machines under a variety
> of circumstances.
>
> As a practical matter, 25ms is a substantial reduction from the 1RTT
> rule of thumb and so it's reasonable to expect that the deployed gear
> will have this much delay.  Usually it has far more.  (Even at 10Gb/s
> and up.)  If some gear at an atypical bottleneck doesn't have this
> much, the behavior degenerates into TCP.
>
> It is also a fairly unsubstantial increase relative to RTT and will
> remain such unless progress is made on the speed-of-light problem (or
> the planet is shrunk?).
>
> The work you cite never seems to mention minimizing delay as a goal.
> It discusses the queuing delay achieved in Theorem 3.  And there's a
> parameter that controls it (alpha_b^*).
>
> -- Stas
>
>
> On Wed, Apr 1, 2009 at 3:46 AM, Bob Briscoe <rbriscoe@jungle.bt.co.uk>
> wrote:
> > Stas,
> >
> > This posting repeats what I said in the ledbat w-g meeting in SF, but t=
o
> the
> > whole list and with added reasoning, given the benefit of more time for
> > thought.
> >
> > On balance it seems wrong to aim for a hard-coded delay target (25ms). =
I
> > understand your reasoning for it in the draft (dividing delay between
> > competing flows rather than adding), but there are also stronger scalin=
g
> > reasons against it...
> >
> > For 1500B* packets, 25ms represents a standing queue of:
> >   1 pkt @ 512kb/s
> >  17 pkt @   8Mb/s
> >  50 pkt @  24Mb/s (some backhaul links today)
> >  208 pkt @ 100Mb/s (typical PON access rate today)
> > 2080 pkt @   1Gb/s (increasingly backhaul links are GigE)
> >
> > As bottleneck links get faster (magically this is happening!), we will
> want
> > to reduce the hard-coded delay (to save on fast buffer memory and the
> power
> > it requires).
> >
> > But if the the algo in draft-00 were changed in the future to aim at 10=
ms
> > (while the installed base all still aimed at 25ms), the new code would
> > always be starved by the installed base. Therefore we would be stuck
> forever
> > with the number we first thought of: 25ms hard-coded.
> >
> > Spelling this out longhand, imagine two flows competing:
> >        Flow N (New)    (10ms target)
> >        Flow L (Legacy) (25ms target)
> >
> > L drives delay to 25ms
> > while both flows have packets to send {
> >    N reduces rate because delay greater than 10ms
> >    L sees delay decrease so increases its rate to drive delay to 25ms
> > }
> > Result: newer algorithm always starves.
> >
> > IMHO, the algo needs to aim at minimising the congestion delay, not aim
> at a
> > fixed target (the draft says the former but does the latter). Section 3
> > onwards of [Key04] may help, even tho it's in the context of adapting t=
he
> > receive window it's still based on delay sensing. I'd be interested to
> know
> > whether Microsoft's BITS now implements something like this algo.
> >
> >
> > Bob
> >
> > [* =3D This might be fine if packet size scaled with link speeds, but t=
hat
> > isn't happening.]
> >
> > [Key04] Peter Key, Laurent Massouli=E9 and Bing Wang, "Emulating
> low-priority
> > transport at the application layer: a background transfer service" In:
> "ACM
> > SIGMETRICS '04/Performance '04: Proceedings of the joint international
> > conference on Measurement and modeling of computer systems", pp. 118--1=
29
> > <http://research.microsoft.com/apps/pubs/default.aspx?id=3D67381> (2004=
)
> >
> >
> >
> >
> > ________________________________________________________________
> > Bob Briscoe,               Networks Research Centre, BT Research
> >
>
>
>
> --
> Stanislav Shalunov
> BitTorrent Inc
> shalunov@bittorrent.com
>
> personal: http://shlang.com
> _______________________________________________
> ledbat mailing list
> ledbat@ietf.org
> https://www.ietf.org/mailman/listinfo/ledbat
>



--=20
Prof. Saverio Mascolo
Dipartimento di Elettrotecnica ed Elettronica
Politecnico di Bari
Via Orabona 4
70125 Bari
Italy
Tel. +39 080 5963621
Fax. +39 080 5963410
email:mascolo@poliba.it <email%3Amascolo@poliba.it>

http://c3lab.poliba.it

--0016368e2bc9a719b60466a50ff7
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><div class=3D"gmail_quote">On Fri, Apr 3, 2009 at 8:11 AM, Stanislav Sh=
alunov <span dir=3D"ltr">&lt;<a href=3D"mailto:shalunov@bittorrent.com" tar=
get=3D"_blank">shalunov@bittorrent.com</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); =
margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">



Bob,<br>
<br>
The choice of parameters for congestion control has a degree to which<br>
it&#39;s arbitrary. =A0Why does TCP ramp up at a hard-coded 1 MSS/RTT? =A0<=
/blockquote><div><br>here i have an answer that it is important to have in =
mind, especially because of many TCP variants that try a more aggressive pr=
obing (some of them replacing TCP NewReno).<br>



<br>The=A0 increasing at 1 MSS/RTT is the <u>unique</u> increase with=A0 th=
e property that=A0 <u>only one packet is lost</u> when=A0 the path capacity=
 is hit. This because of=A0 a feedback lag equal to RTT, i.e. after 1 RTT T=
CP=A0 knows if the capacity has been hit. This 1 MSS/RTT  increment has the=
 very nice property that only one lost packet needs to be retransmitted.<br=
>



<br>TCP variants trying a more aggressive probing experience several packet=
 losses when the capacity is hit. So what happens is that the throughput in=
creases but also retransmissions so that you end up with same goodput but w=
ith higher retransmissions, which is not good.<br>

<br><br>These considerations imply that when setting:<br><br>cwnd+=3Dgain*(=
T_target-T_forward)<br><br>it is feasible to choose a low gain in order=A0 =
to have a TCP less than best effort but it is difficult=A0 to set a larger =
gain to have a quick probing in case of gigabit nets without ending in the =
problem before mentioned (we tried it sometime ago and we give up)!<br>

<br></div><blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid=
 rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

Why<br>
not 3.5 or 0.6? =A0Why does it multiply by 0.5 instead of 7/8 or 0.2 or<br>
0?</blockquote><div><br>again, here=A0 a less than best effort can be:<br><=
br>cwnd=3Dmin(cwnd/2,BWE*base_rtt)<br><br>saverio<br><br><br></div><blockqu=
ote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204=
); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

<br>
If a future version of TCP chooses 0.2 and 0.3 as its favorite lucky<br>
numbers while the installed base is at 1 and 0.5, the consequences are<br>
similar.<br>
<br>
While the choice of the particular parametrization for TCP might be<br>
argued to be suboptimal because it doesn&#39;t scale with link speed, I&#39=
;m<br>
not aware of any congestion control mechanism without parameters.<br>
<br>
The mechanism we use scales the target queue linearly with link speed,<br>
as you point out. =A0It&#39;s a feature. =A0It enables the mechanism to wor=
k<br>
on a variety of bottleneck speeds. =A0If you specify the target in<br>
bytes, you lose on fast links because your delay starts getting lower<br>
than scheduling noise and/or PCI bus jitter and you lose on slow links<br>
because the delay is too large for interactive apps.<br>
<br>
As choice of parameters goes, the choice of the gain is much more<br>
arbitrary than the choice of the target. =A0The gain simply matches TCP<br>
in the worst case. =A0Target is essentially the lowest value than can be<br=
>
reliably distinguished from 0 on a variety of machines under a variety<br>
of circumstances.<br>
<br>
As a practical matter, 25ms is a substantial reduction from the 1RTT<br>
rule of thumb and so it&#39;s reasonable to expect that the deployed gear<b=
r>
will have this much delay. =A0Usually it has far more. =A0(Even at 10Gb/s<b=
r>
and up.) =A0If some gear at an atypical bottleneck doesn&#39;t have this<br=
>
much, the behavior degenerates into TCP.<br>
<br>
It is also a fairly unsubstantial increase relative to RTT and will<br>
remain such unless progress is made on the speed-of-light problem (or<br>
the planet is shrunk?).<br>
<br>
The work you cite never seems to mention minimizing delay as a goal.<br>
It discusses the queuing delay achieved in Theorem 3. =A0And there&#39;s a<=
br>
parameter that controls it (alpha_b^*).<br>
<br>
-- Stas<br>
<div><div></div><div><br>
<br>
On Wed, Apr 1, 2009 at 3:46 AM, Bob Briscoe &lt;<a href=3D"mailto:rbriscoe@=
jungle.bt.co.uk" target=3D"_blank">rbriscoe@jungle.bt.co.uk</a>&gt; wrote:<=
br>
&gt; Stas,<br>
&gt;<br>
&gt; This posting repeats what I said in the ledbat w-g meeting in SF, but =
to the<br>
&gt; whole list and with added reasoning, given the benefit of more time fo=
r<br>
&gt; thought.<br>
&gt;<br>
&gt; On balance it seems wrong to aim for a hard-coded delay target (25ms).=
 I<br>
&gt; understand your reasoning for it in the draft (dividing delay between<=
br>
&gt; competing flows rather than adding), but there are also stronger scali=
ng<br>
&gt; reasons against it...<br>
&gt;<br>
&gt; For 1500B* packets, 25ms represents a standing queue of:<br>
&gt; =A0 1 pkt @ 512kb/s<br>
&gt; =A017 pkt @ =A0 8Mb/s<br>
&gt; =A050 pkt @ =A024Mb/s (some backhaul links today)<br>
&gt; =A0208 pkt @ 100Mb/s (typical PON access rate today)<br>
&gt; 2080 pkt @ =A0 1Gb/s (increasingly backhaul links are GigE)<br>
&gt;<br>
&gt; As bottleneck links get faster (magically this is happening!), we will=
 want<br>
&gt; to reduce the hard-coded delay (to save on fast buffer memory and the =
power<br>
&gt; it requires).<br>
&gt;<br>
&gt; But if the the algo in draft-00 were changed in the future to aim at 1=
0ms<br>
&gt; (while the installed base all still aimed at 25ms), the new code would=
<br>
&gt; always be starved by the installed base. Therefore we would be stuck f=
orever<br>
&gt; with the number we first thought of: 25ms hard-coded.<br>
&gt;<br>
&gt; Spelling this out longhand, imagine two flows competing:<br>
&gt; =A0 =A0 =A0 =A0Flow N (New) =A0 =A0(10ms target)<br>
&gt; =A0 =A0 =A0 =A0Flow L (Legacy) (25ms target)<br>
&gt;<br>
&gt; L drives delay to 25ms<br>
&gt; while both flows have packets to send {<br>
&gt; =A0 =A0N reduces rate because delay greater than 10ms<br>
&gt; =A0 =A0L sees delay decrease so increases its rate to drive delay to 2=
5ms<br>
&gt; }<br>
&gt; Result: newer algorithm always starves.<br>
&gt;<br>
&gt; IMHO, the algo needs to aim at minimising the congestion delay, not ai=
m at a<br>
&gt; fixed target (the draft says the former but does the latter). Section =
3<br>
&gt; onwards of [Key04] may help, even tho it&#39;s in the context of adapt=
ing the<br>
&gt; receive window it&#39;s still based on delay sensing. I&#39;d be inter=
ested to know<br>
&gt; whether Microsoft&#39;s BITS now implements something like this algo.<=
br>
&gt;<br>
&gt;<br>
&gt; Bob<br>
&gt;<br>
&gt; [* =3D This might be fine if packet size scaled with link speeds, but =
that<br>
&gt; isn&#39;t happening.]<br>
&gt;<br>
&gt; [Key04] Peter Key, Laurent Massouli=E9 and Bing Wang, &quot;Emulating =
low-priority<br>
&gt; transport at the application layer: a background transfer service&quot=
; In: &quot;ACM<br>
&gt; SIGMETRICS &#39;04/Performance &#39;04: Proceedings of the joint inter=
national<br>
&gt; conference on Measurement and modeling of computer systems&quot;, pp. =
118--129<br>
&gt; &lt;<a href=3D"http://research.microsoft.com/apps/pubs/default.aspx?id=
=3D67381" target=3D"_blank">http://research.microsoft.com/apps/pubs/default=
.aspx?id=3D67381</a>&gt; (2004)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; ________________________________________________________________<br>
&gt; Bob Briscoe, =A0 =A0 =A0 =A0 =A0 =A0 =A0 Networks Research Centre, BT =
Research<br>
&gt;<br>
<br>
<br>
<br>
</div></div><font color=3D"#888888">--<br>
Stanislav Shalunov<br>
BitTorrent Inc<br>
<a href=3D"mailto:shalunov@bittorrent.com" target=3D"_blank">shalunov@bitto=
rrent.com</a><br>
<br>
personal: <a href=3D"http://shlang.com" target=3D"_blank">http://shlang.com=
</a><br>
</font><div><div></div><div>_______________________________________________=
<br>
ledbat mailing list<br>
<a href=3D"mailto:ledbat@ietf.org" target=3D"_blank">ledbat@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/ledbat" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/ledbat</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><br>-- <br>Prof. Saver=
io Mascolo<br>Dipartimento di Elettrotecnica ed Elettronica<br>Politecnico =
di Bari<br>Via Orabona 4<br>70125 Bari <br>Italy<br>Tel. +39 080 5963621<br=
>



Fax. +39 080 5963410<br><a href=3D"mailto:email%3Amascolo@poliba.it" target=
=3D"_blank">email:mascolo@poliba.it</a><br> <br><a href=3D"http://c3lab.pol=
iba.it" target=3D"_blank">http://c3lab.poliba.it</a><br><br>=A0<br>



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Greetings Saverio,

2009/4/3 Saverio Mascolo <saverio.mascolo@gmail.com>:
> On Fri, Apr 3, 2009 at 8:11 AM, Stanislav Shalunov <shalunov@bittorrent.com>
> wrote:
>>
>> The choice of parameters for congestion control has a degree to which
>> it's arbitrary.  Why does TCP ramp up at a hard-coded 1 MSS/RTT?
>
> The  increasing at 1 MSS/RTT is the _unique_ increase with  the property that
> _only_one_packet_is_lost_ when  the path capacity is hit.

It is the fastest, but increasing by 1 MSS  every 2 RTT also has that
property if there is a single flow.  More importantly, if there are
multiple flows, then 1 MSS/RTT doesn't guarantee a single loss per RTT
for a given flow, and so any benefit from that is lost.

$0.02,
Lachlan

-- 
Lachlan Andrew  Centre for Advanced Internet Architectures (CAIA)
Swinburne University of Technology, Melbourne, Australia
<http://caia.swin.edu.au/cv/landrew> <http://netlab.caltech.edu/lachlan>
Ph +61 3 9214 4837

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2009/4/3 Saverio Mascolo <saverio.mascolo@gmail.com>:
> On Fri, Apr 3, 2009 at 8:11 AM, Stanislav Shalunov
<shalunov@bittorrent.com>
> wrote:
>>
>> The choice of parameters for congestion control has a degree to which
>> it's arbitrary.  Why does TCP ramp up at a hard-coded 1 MSS/RTT?
>
> The  increasing at 1 MSS/RTT is the _unique_ increase with  the
property that
> _only_one_packet_is_lost_ when  the path capacity is hit.

Has anyone taken a stab @ a model in NS? [http://www.isi.edu/nsnam/ns/].
It would
help answer the what-ifs of timing and interactions with other protocols
I think.



From arnaud.legout@inria.fr  Sun Apr  5 12:47:56 2009
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Hi

Don Bowman wrote:
> Has anyone taken a stab @ a model in NS? [http://www.isi.edu/nsnam/ns/].
> It would
> help answer the what-ifs of timing and interactions with other protocols
> I think.
>   
doing simulations involving TCP in ns-2 is simply a bad idea, as ns-3 
has been released.
Why? Because in ns-3 you have the real TCP BSD stack in it. Ns-3 is not 
a simple evolution
of ns-2, it is a complete rewrite with a new design based on the 
experience gathered on ns-2.
Here is the URL for ns-3
http://www.nsnam.org/

Regards,
Arnaud.


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Subject: Re: [ledbat] Hard-coded delay target? draft-shalunov-ledbat-congestion-00
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hi,

what about keeping track of the number of uploading connections N and using
the setting:

on ack reception:
cwnd+=(1/cwnd)*1/N

in any of the tcp connections open by the p2p client? this would make the
p2p application equivalent to one TCP uploading connection.

saverio

On Sat, Apr 4, 2009 at 12:28 PM, Lachlan Andrew <lachlan.andrew@gmail.com>wrote:

> Greetings Saverio,
>
> 2009/4/3 Saverio Mascolo <saverio.mascolo@gmail.com>:
> > On Fri, Apr 3, 2009 at 8:11 AM, Stanislav Shalunov <
> shalunov@bittorrent.com>
> > wrote:
> >>
> >> The choice of parameters for congestion control has a degree to which
> >> it's arbitrary.  Why does TCP ramp up at a hard-coded 1 MSS/RTT?
> >
> > The  increasing at 1 MSS/RTT is the _unique_ increase with  the property
> that
> > _only_one_packet_is_lost_ when  the path capacity is hit.
>
> It is the fastest, but increasing by 1 MSS  every 2 RTT also has that
> property if there is a single flow.  More importantly, if there are
> multiple flows, then 1 MSS/RTT doesn't guarantee a single loss per RTT
> for a given flow, and so any benefit from that is lost.
>
> $0.02,
> Lachlan
>
> --
> Lachlan Andrew  Centre for Advanced Internet Architectures (CAIA)
> Swinburne University of Technology, Melbourne, Australia
> <http://caia.swin.edu.au/cv/landrew> <http://netlab.caltech.edu/lachlan>
> Ph +61 3 9214 4837
>



-- 
Prof. Saverio Mascolo
Dipartimento di Elettrotecnica ed Elettronica
Politecnico di Bari
Via Orabona 4
70125 Bari
Italy
Tel. +39 080 5963621
Fax. +39 080 5963410
email:mascolo@poliba.it <email%3Amascolo@poliba.it>

http://c3lab.poliba.it


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hi,<br><br>what about keeping track of the number of uploading connections =
N and using the setting:<br><br>on ack reception:<br>cwnd+=3D(1/cwnd)*1/N<b=
r><br>in any of the tcp connections open by the p2p client? this would make=
 the p2p application equivalent to one TCP uploading connection.<br>

<br>saverio<br><br><div class=3D"gmail_quote">On Sat, Apr 4, 2009 at 12:28 =
PM, Lachlan Andrew <span dir=3D"ltr">&lt;<a href=3D"mailto:lachlan.andrew@g=
mail.com">lachlan.andrew@gmail.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); marg=
in: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

Greetings Saverio,<br>
<br>
2009/4/3 Saverio Mascolo &lt;<a href=3D"mailto:saverio.mascolo@gmail.com">s=
averio.mascolo@gmail.com</a>&gt;:<br>
<div class=3D"im">&gt; On Fri, Apr 3, 2009 at 8:11 AM, Stanislav Shalunov &=
lt;<a href=3D"mailto:shalunov@bittorrent.com">shalunov@bittorrent.com</a>&g=
t;<br>
&gt; wrote:<br>
&gt;&gt;<br>
</div><div class=3D"im">&gt;&gt; The choice of parameters for congestion co=
ntrol has a degree to which<br>
&gt;&gt; it&#39;s arbitrary. =A0Why does TCP ramp up at a hard-coded 1 MSS/=
RTT?<br>
&gt;<br>
</div><div class=3D"im">&gt; The =A0increasing at 1 MSS/RTT is the _unique_=
 increase with =A0the property that<br>
&gt; _only_one_packet_is_lost_ when =A0the path capacity is hit.<br>
<br>
</div>It is the fastest, but increasing by 1 MSS =A0every 2 RTT also has th=
at<br>
property if there is a single flow. =A0More importantly, if there are<br>
multiple flows, then 1 MSS/RTT doesn&#39;t guarantee a single loss per RTT<=
br>
for a given flow, and so any benefit from that is lost.<br>
<br>
$0.02,<br>
Lachlan<br>
<font color=3D"#888888"><br>
--<br>
Lachlan Andrew =A0Centre for Advanced Internet Architectures (CAIA)<br>
Swinburne University of Technology, Melbourne, Australia<br>
&lt;<a href=3D"http://caia.swin.edu.au/cv/landrew" target=3D"_blank">http:/=
/caia.swin.edu.au/cv/landrew</a>&gt; &lt;<a href=3D"http://netlab.caltech.e=
du/lachlan" target=3D"_blank">http://netlab.caltech.edu/lachlan</a>&gt;<br>
Ph +61 3 9214 4837<br>
</font></blockquote></div><br><br clear=3D"all"><br>-- <br>Prof. Saverio Ma=
scolo<br>Dipartimento di Elettrotecnica ed Elettronica<br>Politecnico di Ba=
ri<br>Via Orabona 4<br>70125 Bari <br>Italy<br>Tel. +39 080 5963621<br>Fax.=
 +39 080 5963410<br>

<a href=3D"mailto:email%3Amascolo@poliba.it">email:mascolo@poliba.it</a><br=
> <br><a href=3D"http://c3lab.poliba.it">http://c3lab.poliba.it</a><br><br>=
<br>=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<br> This message may contain confidential an=
d/or legally privileged information.<br>

 =A0If you are not the intended recipient of the message, please destroy it=
.<br> Any unauthorized dissemination, distribution, or copying of the mater=
ial in<br> this message, and any attachments to the message, is strictly fo=
rbidden.<br>



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Subject: Re: [ledbat] Hard-coded delay target? draft-shalunov-ledbat-congestion-00
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Stas,

Pointing to other protocols that don't scale=20
isn't a good reason for deliberately designing a=20
new one that we know from the start won't scale=20
into the future. But I guess it's up to me /=20
others to come up with a more adaptive approach if we can.

more inline...

At 07:11 03/04/2009, Stanislav Shalunov wrote:
>Bob,
>
>The choice of parameters for congestion control has a degree to which
>it's arbitrary.  Why does TCP ramp up at a hard-coded 1 MSS/RTT?  Why
>not 3.5 or 0.6?  Why does it multiply by 0.5 instead of 7/8 or 0.2 or
>0?
>
>If a future version of TCP chooses 0.2 and 0.3 as its favorite lucky
>numbers while the installed base is at 1 and 0.5, the consequences are
>similar.
>
>While the choice of the particular parametrization for TCP might be
>argued to be suboptimal because it doesn't scale with link speed, I'm
>not aware of any congestion control mechanism without parameters.

I believe it's particularly important to avoid=20
non-dimensionless parameters (ie. to avoid=20
parameters with units). Wherever there is one,=20
I'd rather it was either adaptive, or a parameter=20
under application control. But enough of these generalities...


>The mechanism we use scales the target queue linearly with link speed,
>as you point out.  It's a feature.  It enables the mechanism to work
>on a variety of bottleneck speeds.  If you specify the target in
>bytes, you lose on fast links because your delay starts getting lower
>than scheduling noise and/or PCI bus jitter and you lose on slow links
>because the delay is too large for interactive apps.

All understood. I certainly wasn't proposing we set a buffer target in=
 bytes.

But to me it's short-sighted and defeatist to say=20
"operating systems will never reliably be able to=20
measure delays to finer granularity than Xms".=20
Just because they usually can't today.

The part I don't like is increasing rate to hit a=20
target delay. I'd prefer that the algo=20
continually increases rate (w per window say)=20
whether or not there's delay, and decreases rate=20
whenever it measures delay. Then the two will=20
balance at the *lowest measureable* delay delta.=20
I'd like to eventually see w under app-control=20
through the API (that argument requires another=20
thread). But, if I don't win that argument, w could initially be hard-coded.

This has the property that the more flows compete=20
for the same queue, the more the delay will be. I=20
know you don't like that, but I actually think=20
it's a desirable property. It automatically=20
solves sharing of bottlnecks and problems like=20
the late-comers disadvantage (separate thread=20
with Doug Leith). It's a proven approach in=20
nearly every congestion control we have.

Imagine large numbers of flows share a hard-coded=20
delay target, all with fixed gain. Imagine the=20
measured delay is slightly above target. Each one=20
makes a small rate reduction and gets surprised=20
that the delay has reduced a lot more than they=20
expected (because everyone else made a similar=20
reduction). So they all increase more than they=20
reduced. And the delay shoots up. Result, unstable oscillation?

I know, this depends on the relative RTTs etc...

I'm not saying there *is* a problem. It just=20
seems a rate *increase* towards a hard-coded=20
delay target is unnecessary. We only have to=20
*reduce* in the presence of *any* measurable=20
delay. Doing otherwise moves into unnecessarily=20
uncharted territory that should have warning signs "Here be monsters".

>As choice of parameters goes, the choice of the gain is much more
>arbitrary than the choice of the target.  The gain simply matches TCP
>in the worst case.  Target is essentially the lowest value than can be
>reliably distinguished from 0 on a variety of machines under a variety
>of circumstances.

Understood


>As a practical matter, 25ms is a substantial reduction from the 1RTT
>rule of thumb and so it's reasonable to expect that the deployed gear
>will have this much delay.  Usually it has far more.  (Even at 10Gb/s
>and up.)

I doubt the 1RTT rule of thumb is used much these=20
days. It would require ridiculously large amounts=20
of buffer memory on hi-speed interfaces. Back in=20
2005, when I last asked, I discovered BT folks=20
were generally configuring buffer sizes in a way=20
that grew with about the sq-root of link speed=20
(not deliberately following Nick McKeown's advice=20
that came out around then, this was just=20
happenchance even tho they knew about McKeown's papers).

You might configure about 25ms of buffer on a=20
155Mb/s interface these days. But on a GigE I'm=20
sure a good config would be less. For 10Gb/s=20
interfaces, what's the evidence for them being=20
config'd with far more than 25ms? Assuming 1500B=20
pkt size, this is about 21,000 packets of buffer.

However, the point is that the trend for the=20
future is to reduce buffer memory on faster=20
interfaces, to reduce power requirements, given=20
so much buffer is now believed not to actually be=20
necessary. But I should also re-run my 2005 survey.

>If some gear at an atypical bottleneck doesn't have this
>much, the behavior degenerates into TCP.

That sounds good if you say it quickly. But it's=20
actually a failure. It means all the flows one=20
expected to scavenge quietly in the background,=20
actually compete on equal terms in the foreground.

Do you think of LEDBAT as a solution with=20
long-term relevance for sharing any bottleneck?=20
Or do you think of it as a stop-gap for sharing=20
low speed bottlenecks? I don't use the term=20
stop-gap disparagingly. I just mean something=20
that loses relevance in 5-10years, rather than=20
something that one expects will last as long as TCP has.

>It is also a fairly unsubstantial increase relative to RTT and will
>remain such unless progress is made on the speed-of-light problem (or
>the planet is shrunk?).

I keep on constant alert for breakthroughs in=20
these areas. But I'm currently more worried that=20
a 25ms target is an unnecessarily large choice,=20
given I don't believe we have to make a choice=20
anyway (see alternative approach above).

>The work you cite never seems to mention minimizing delay as a goal.
>It discusses the queuing delay achieved in Theorem 3.  And there's a
>parameter that controls it (alpha_b^*).

Queuing delay is inside the alpha variable. On=20
second thoughts, it's prob not worth using that=20
paper to shed light on anything tho - too much=20
generality is included, which makes it too=20
complex as a basis for discussion here.


Bob


>-- Stas
>
>
>On Wed, Apr 1, 2009 at 3:46 AM, Bob Briscoe <rbriscoe@jungle.bt.co.uk>=
 wrote:
> > Stas,
> >
> > This posting repeats what I said in the=20
> ledbat w-g meeting in SF, but to the
> > whole list and with added reasoning, given the benefit of more time for
> > thought.
> >
> > On balance it seems wrong to aim for a hard-coded delay target (25ms). I
> > understand your reasoning for it in the draft (dividing delay between
> > competing flows rather than adding), but there are also stronger scaling
> > reasons against it...
> >
> > For 1500B* packets, 25ms represents a standing queue of:
> >   1 pkt @ 512kb/s
> >  17 pkt @   8Mb/s
> >  50 pkt @  24Mb/s (some backhaul links today)
> >  208 pkt @ 100Mb/s (typical PON access rate today)
> > 2080 pkt @   1Gb/s (increasingly backhaul links are GigE)
> >
> > As bottleneck links get faster (magically this is happening!), we will=
 want
> > to reduce the hard-coded delay (to save on fast buffer memory and the=
 power
> > it requires).
> >
> > But if the the algo in draft-00 were changed in the future to aim at=
 10ms
> > (while the installed base all still aimed at 25ms), the new code would
> > always be starved by the installed base.=20
> Therefore we would be stuck forever
> > with the number we first thought of: 25ms hard-coded.
> >
> > Spelling this out longhand, imagine two flows competing:
> >        Flow N (New)    (10ms target)
> >        Flow L (Legacy) (25ms target)
> >
> > L drives delay to 25ms
> > while both flows have packets to send {
> >    N reduces rate because delay greater than 10ms
> >    L sees delay decrease so increases its rate to drive delay to 25ms
> > }
> > Result: newer algorithm always starves.
> >
> > IMHO, the algo needs to aim at minimising the=20
> congestion delay, not aim at a
> > fixed target (the draft says the former but does the latter). Section 3
> > onwards of [Key04] may help, even tho it's in the context of adapting=
 the
> > receive window it's still based on delay sensing. I'd be interested to=
 know
> > whether Microsoft's BITS now implements something like this algo.
> >
> >
> > Bob
> >
> > [* =3D This might be fine if packet size scaled with link speeds, but=
 that
> > isn't happening.]
> >
> > [Key04] Peter Key, Laurent Massouli=E9 and Bing Wang, "Emulating=
 low-priority
> > transport at the application layer: a background transfer service" In:=
 "ACM
> > SIGMETRICS '04/Performance '04: Proceedings of the joint international
> > conference on Measurement and modeling of computer systems", pp.=
 118--129
> > <http://research.microsoft.com/apps/pubs/default.aspx?id=3D67381> (2004)
> >
> >
> >
> >
> > ________________________________________________________________
> > Bob Briscoe,               Networks Research Centre, BT Research
> >
>
>
>
>--
>Stanislav Shalunov
>BitTorrent Inc
>shalunov@bittorrent.com
>
>personal: http://shlang.com

________________________________________________________________
Bob Briscoe,               Networks Research Centre, BT Research=20


From lachlan.andrew@gmail.com  Wed Apr  8 15:19:15 2009
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Date: Thu, 9 Apr 2009 08:20:21 +1000
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Greetings Bob,

2009/4/9 Bob Briscoe <rbriscoe@jungle.bt.co.uk>:
>
> At 07:11 03/04/2009, Stanislav Shalunov wrote:
>> If you specify the target in
>> bytes, you lose on fast links because your delay starts getting lower
>> than scheduling noise and/or PCI bus jitter and you lose on slow links
>> because the delay is too large for interactive apps.
>
> All understood. I certainly wasn't proposing we set a buffer target in
> bytes.
>
> But to me it's short-sighted and defeatist to say "operating systems will
> never reliably be able to measure delays to finer granularity than Xms".
> Just because they usually can't today.

It is easy to *measure* delay, the questions is "what is the source of
the delay?"  If delay is a sign of congestion, then it is justifiable
to back off.  If it is just jitter (something falling out of a cache
along the route, perhaps) then the OS has no way to know.

If jitter is consistently going down, then you're right about being
short sighted.  If jitter isn't going down, then we shouldn't design a
protocol which assumes it will.  Does anyone on the list have data
about trends in jitter?

> The part I don't like is increasing rate to hit a target delay. I'd prefer
> that the algo continually increases rate (w per window say) whether or not
> there's delay, and decreases rate whenever it measures delay. Then the two
> will balance at the *lowest measureable* delay delta.

That sounds like Doug's "delay AIMD".  He and I have been discussing
that, and it seems quite sensitive to asymmetric jitter.  If one flow
(with jitter) backs off noticeably when it senses a particular
threshold of delay then the flow without jitter never gets to see the
threshold met, and so never backs off until the jittered flow is
starved, and then the cycle repeats.

That might be avoided if the backoff is very small when delay is
experienced (like Vegas).

Also, the Linux stack has occasional very long delays of many ms;
should it set a "lowest measureable" target which is longer than these
very long delays, because otherwise the might be spurious?

> I'm currently
> more worried that a 25ms target is an unnecessarily large choice, given I
> don't believe we have to make a choice anyway

I agree that 25ms is an unnecessarily large delay, since jitter is
typically in the 1ms range.

$0.02,
Lachlan

-- 
Lachlan Andrew  Centre for Advanced Internet Architectures (CAIA)
Swinburne University of Technology, Melbourne, Australia
<http://caia.swin.edu.au/cv/landrew> <http://netlab.caltech.edu/lachlan>
Ph +61 3 9214 4837

From don@sandvine.com  Wed Apr  8 19:42:56 2009
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Andrew Lachlan wrote:
>2009/4/9 Bob Briscoe <rbriscoe@jungle.bt.co.uk>:
>>
>> At 07:11 03/04/2009, Stanislav Shalunov wrote:
>>> If you specify the target in
>>> bytes, you lose on fast links because your delay starts getting
lower
>>> than scheduling noise and/or PCI bus jitter and you lose on slow
links
>>> because the delay is too large for interactive apps.
>>
>> All understood. I certainly wasn't proposing we set a buffer target
in
>> bytes.
>>
>> But to me it's short-sighted and defeatist to say "operating systems
will
>> never reliably be able to measure delays to finer granularity than
Xms".
>> Just because they usually can't today.
>
>It is easy to *measure* delay, the questions is "what is the source of
>the delay?"  If delay is a sign of congestion, then it is justifiable
>to back off.  If it is just jitter (something falling out of a cache
>along the route, perhaps) then the OS has no way to know.

Watch out for mobile networks where handoffs cause change in delay,
Changes in error correction (that tunnel you are driving through) cause
delay changes, etc. The network may also be able to dynamically shift
bandwidth from upstream to downstream (e.g. wimax, lte).
The latency is also much higher on 3G networks than on fixed.

Also dynamic bandwidth networks don't always have buffering
associated. An example, the 'powerboost' used in cable is a token-credit
model which allows higher than nominal rates to be achieved until
the sender/receiver is out of credits, and then it reverts to nominal
rate. No buffer depth changes occur, the scheduler just changes rate
suddenly. This has sometimes caused oscillations with media applications
that try to 'measure' the rate of the line, and then pick a codec.

 ...

>> I'm currently
>> more worried that a 25ms target is an unnecessarily large choice,
given I
>> don't believe we have to make a choice anyway
>
>I agree that 25ms is an unnecessarily large delay, since jitter is
>typically in the 1ms range.

I think this (1ms) is pretty low for mobile. I also think it's pretty
low for fixed.
Measurements I have done on RTP jitter (e.g. voip) show more like 10ms
of jitter for non-facilities-based voip, and I believe should be
consistent
for TCP.

The proposed initial use of ledbat is e.g. p2p file sharing. In this
application,
the other endpoint is as likely to be on the far side of the world as
the
far side of your street, so I think we will see a pretty wide dynamic
range
in both jitter (from the high hop count) and latency.


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Stas,

OK, taking what Lachlan has said, I now accept we need *a* hard-coded 
threshold below which we assume delay variation isn't necessarily congestion.

1/ The value is TBA. Perhaps 25ms, perhaps 5ms. Whatever we choose 
will always hurt in some cases (e.g. Don's); the choice is about 
which ones are corner cases we can live with. E.g. During mobile 
handover, there will be a period during which the controller gets 
confused. It will sort itself out in time. Is that important? 
Dunno... It's for debate.

I'd say it's something we can live with traded against the benefit of 
exploiting congestion info from delay 'most' of the time.

2/ But this still leaves questions about the core part of the control 
algo. However, these are more appropriate to Doug Lieth's thread on 
competing flows, so I'll shift to there.


Bob


At 03:43 09/04/2009, Don Bowman wrote:
>Andrew Lachlan wrote:
> >2009/4/9 Bob Briscoe <rbriscoe@jungle.bt.co.uk>:
> >>
> >> At 07:11 03/04/2009, Stanislav Shalunov wrote:
> >>> If you specify the target in
> >>> bytes, you lose on fast links because your delay starts getting
>lower
> >>> than scheduling noise and/or PCI bus jitter and you lose on slow
>links
> >>> because the delay is too large for interactive apps.
> >>
> >> All understood. I certainly wasn't proposing we set a buffer target
>in
> >> bytes.
> >>
> >> But to me it's short-sighted and defeatist to say "operating systems
>will
> >> never reliably be able to measure delays to finer granularity than
>Xms".
> >> Just because they usually can't today.
> >
> >It is easy to *measure* delay, the questions is "what is the source of
> >the delay?"  If delay is a sign of congestion, then it is justifiable
> >to back off.  If it is just jitter (something falling out of a cache
> >along the route, perhaps) then the OS has no way to know.
>
>Watch out for mobile networks where handoffs cause change in delay,
>Changes in error correction (that tunnel you are driving through) cause
>delay changes, etc. The network may also be able to dynamically shift
>bandwidth from upstream to downstream (e.g. wimax, lte).
>The latency is also much higher on 3G networks than on fixed.
>
>Also dynamic bandwidth networks don't always have buffering
>associated. An example, the 'powerboost' used in cable is a token-credit
>model which allows higher than nominal rates to be achieved until
>the sender/receiver is out of credits, and then it reverts to nominal
>rate. No buffer depth changes occur, the scheduler just changes rate
>suddenly. This has sometimes caused oscillations with media applications
>that try to 'measure' the rate of the line, and then pick a codec.
>
>  ...
>
> >> I'm currently
> >> more worried that a 25ms target is an unnecessarily large choice,
>given I
> >> don't believe we have to make a choice anyway
> >
> >I agree that 25ms is an unnecessarily large delay, since jitter is
> >typically in the 1ms range.
>
>I think this (1ms) is pretty low for mobile. I also think it's pretty
>low for fixed.
>Measurements I have done on RTP jitter (e.g. voip) show more like 10ms
>of jitter for non-facilities-based voip, and I believe should be
>consistent
>for TCP.
>
>The proposed initial use of ledbat is e.g. p2p file sharing. In this
>application,
>the other endpoint is as likely to be on the far side of the world as
>the
>far side of your street, so I think we will see a pretty wide dynamic
>range
>in both jitter (from the high hop count) and latency.

________________________________________________________________
Bob Briscoe,               Networks Research Centre, BT Research 


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Subject: Re: [ledbat] fairness between competing delay flows in draft-shalunov-ledbat-congestion-00.txt
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Stas,

Point taken about base measurements, but this questions is about the 
motivation for the core part of the control algo:
         cwnd += GAIN * off_target / cwnd. (using the draft's 
terminology: off_target = TARGET - queuing_delay)
Q. Why should the controller's increase and decrease both depend on 
the same formula?

I think this means the shares that competing flows get depend purely 
on the minutiae of the dynamics of each competing flow. That means 
shares depend on arbitrary stuff that sharing shouldn't depend on 
(e.g. whether disk heads move, how many other interrupts are served, 
who arrived first), and there's nothing explicit to control what the 
shares do depend on. However, I assume I must have missed something, 
as I'm sure you've thought of this.

As far as I can work out (haven't had time to work through the 
non-linear case where the small standing queue might disappear 
completely), shares don't explicitly depend on RTT or packet size or 
delayed ack factor (which IMHO they shouldn't). While both the 
acceleration and deceleration of each flow do depend on these factors 
in equal measures (which IMHO they should too). So far so good, but...

...taking RTT as one example, altho the steady state shares won't 
depend directly on RTT, they might depend on second order effects of 
how differences in RTT interact with other random events. If correct, 
that's not good. It's random.

Put another way, repeating the question in my earlier posting:
Q. Why should the controller aim to *increase* delay up to the target?

Instead the window *increase* could operate as a separate continuous 
process (as in AIMD algos). And the threshold only indicates a 
*decrease* is needed if exceeded. Then bottleneck sharing is 
explicitly under control. It is the equilibrium when the two 
decoupled increase & decrease processes balance. 'TARGET' then 
becomes merely a lower threshold for the decrease process, not a 
target for both processes.


Bob

At 07:29 03/04/2009, Stanislav Shalunov wrote:
>We have not observed the late-comer's advantage.
>
>As soon as the first flow has a hiccup (for example, because something
>needs to wait for a few disk head moves), there will be a gap in the
>first flow's traffic, enabling the second flow to measure accurately.
>
>During the session Murari mentioned that in their experience, flows
>were also always able to get a reasonable base delay measurement.
>
>Obviously, the more flows, the more uniform the traffic, the more
>likely it is that the late comers might be mistaken about the base.
>At tens of flows, this is clearly not a problem.  At tens of
>thousands, which is what most backbones see for bulk connections, I'd
>expect it to start showing.  The coordination problem at tens of
>thousands of connections is hard and almost certain to be emergent,
>but then extended congestion on backbone links is infrequent.  In the
>worst case, it'll all fall back to TCP.
>
>For a small number of flows, the situation most typically encountered
>on congested links, base estimation seems to be reliable.
>
>
>On Tue, Mar 31, 2009 at 5:23 PM, Douglas Leith <Doug.Leith@nuim.ie> wrote:
> > Can I ask what the expected/desired behaviour will be when two or more
> > ledbat flows share a link ?
> > For example, say a first flow starts, ramps us cwnd until it hits 
> the target
> > queueing delay and then a second ledbat flow starts a while later.  Its not
> > clear to me from the ID how the first flow would alter its cwnd to make
> > space for the second ledbat flow (especially if the second flow 
> does not use
> > slow start, or perhaps has an early exit from slow start).
> > Also, is there a danger that, similarly the vegas and related algorithms,
> > the later starting flow will have difficulty getting an accurate 
> estimate of
> > the base RTT since earlier starting flows maintain a standing queue ?
> > Doug
> > www.hamilton.ie
>
>
>
>--
>Stanislav Shalunov
>BitTorrent Inc
>shalunov@bittorrent.com
>
>personal: http://shlang.com
>_______________________________________________
>ledbat mailing list
>ledbat@ietf.org
>https://www.ietf.org/mailman/listinfo/ledbat

________________________________________________________________
Bob Briscoe,               Networks Research Centre, BT Research 

