
From nobody Mon May  4 06:16:08 2015
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Subject: Re: [tcpm] Feedback on WG acceptance of draft-zimmermann-tcpm-cubic-01
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I support work on this topic.

I think that such a document should be maintained within TCPM.

Here are my thoughts:

- WG informational draft
It changes TCP, but there is quite a lot of understanding of how cubic
operates. It could document what is currently implemented, and I think
would be useful exactly in this form. My take is that I do not think it
really competes with any planned IETF work - but would introduce an
alternate CC.

A focus on correcting the text to reflect what is implemented could make
this a fast procedure. That would be a major selling point to me.

- EXP?
I think it could be published as EXP.

The classification at first seems odd, in that there may be more
operational deployment experience with CUBIC than many IETF specs.
However, at least it is more "aggressive" in CC than some IETF-defined
specs.

If the intent of EXP status was to flag this as not yet finally perfected
by IETF experience and to show that the IETF could in future improve this
further, then that would seem a reasonable position. (I'm not even saying
I know of a list of improvements to make).

- PS?
If the WG evaluation concludes that the evidence is good, then it could be
a PS, that would need a consensus call. That call is I suggest impossible
to make until we open the document up to detailed scrutiny and debate.

I can see merits in publishing as INFO asap.

Gorry (as an individual)

> <michael.scharf@alcatel-lucent.com> wrote:
>> Dear all,
>>
>> During IETF 91 [1] and on the list there has been strong support for WG
>> adoption of draft-zimmermann-tcpm-cubic [2]. It seems consensus in TCPM
>> to adopt this document.
>>
>> In order to move forward, the chairs seek for additional community
>> guidance regarding the intended status:
>>
>> a) Proposed standard
>>
>> b) Experimental
>>
>> c) Informational
>>
>


From nobody Thu May  7 07:59:41 2015
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From: Nicolas Kuhn <nicolas.kuhn@telecom-bretagne.eu>
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Subject: [tcpm] draft-ietf-tcpm-rtorestart-07
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Hi all,=20

I had a look at this draft-ietf-tcpm-rtorestart-07 draft.=20
I have some minor comments on the document.=20

#############################################################

- "The RTO Restart (RTOR) mechanism described in this document makes the
   RTO slightly more aggressive when the number of outstanding segments
   is too small for fast retransmit to work, in an attempt to enable
   faster loss recovery for all segments while being robust to
   reordering."

The RTO is not aggressive ; but the retransmission scheme ruled by RTO =
expirations is.=20
I would recommend to rephrase that.=20

#############################################################

- =E2=80=9C   The RTO Restart (RTOR) mechanism described in this =
document makes the
   RTO slightly [=E2=80=A6] highly varying RTTs, e.g. mobile networks."

This whole paragraph comes to early in the document, as the reader has =
no clear=20
idea about what RTOR is actually doing. I propose to add that in a =
dedicated sub-
section in Section 5.

#############################################################

- =E2=80=9C 3 RTO Restart Overview=E2=80=9D -> should it not be =E2=80=9C3=
- RTO Overview and rationale of RTOR =E2=80=9C or something equivalent ?
It is confusing, as you do not actually present RTOR.=20

Also, in this section 3:
The first paragraph finishes with =E2=80=9CHence,  in most situations, =
the time before a retransmission is triggered is equal to "RTO + =
RTT=E2=80=9D.=E2=80=9D=20
But the example to illustrate it comes after - I think things should =
moved around to:
1- introduce the classic RTO=20
2- explain the rationale of classic RTO
3- example to show that RTO+RTT is common
4- more parameters could affect that (2 outstanding segments; other =
factors)
5- conclusion on =E2=80=9Chence, RTO+RTT is common for application =
limited traffic"

I would propose to introduce a ToC for this section 3:

3- RTO Overview and rationale of RTOR
  3-1. Experienced RTO is RTO+RTT
  3-2. RTO safety margin
  3-3. Rationale of RTOR: when fast retransmit is not possible

#############################################################

Section 4 : should focus on the description of the algorithm only.=20

I would keep=20
=E2=80=9C=20
This approach makes the RTO more
   aggressive than the standardized approach in [RFC6298] but still
   conforms to the requirement in [RFC6298] that segments must not be
   retransmitted earlier than RTO seconds after their original
   transmission.
"

for the discussion section (at this stage of the reading, the reader =
still does not know exactly what RTOR is actually doing).=20

#############################################################

Section 4: "it will prevent RTOR
   from being more aggressive and potentially causing RTOs instead of
   fast retransmits.=E2=80=9D

More aggressive than what ? Has the "four" being experimentally obtained =
?
I think more discussion is required on this parameter.=20

Regards,=20

Nicolas=


From nobody Fri May  8 02:33:13 2015
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Subject: [tcpm] Last Call: <draft-ietf-tcpm-newcwv-10.txt> (Updating TCP to support Rate-Limited Traffic) to Experimental RFC
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The IESG has received a request from the TCP Maintenance and Minor
Extensions WG (tcpm) to consider the following document:
- 'Updating TCP to support Rate-Limited Traffic'
  <draft-ietf-tcpm-newcwv-10.txt> as Experimental 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 2015-05-22. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   This document provides a mechanism to address issues that arise when
   TCP is used to support traffic that exhibits periods where the
   sending rate is limited by the application rather than the congestion
   window.  It provides an experimental update to TCP that allows a TCP
   sender to restart quickly following a rate-limited interval.  This
   method is expected to benefit applications that send rate-limited
   traffic using TCP, while also providing an appropriate response if
   congestion is experienced.

   It also evaluates the Experimental specification of TCP Congestion
   Window Validation, CWV, defined in RFC 2861, and concludes that RFC
   2861 sought to address important issues, but failed to deliver a
   widely used solution.  This document therefore recommends that the
   status of RFC 2861 is moved from Experimental to Historic, and that
   it is replaced by the current specification.




The file can be obtained via
https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv/

IESG discussion can be tracked via
https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv/ballot/


No IPR declarations have been submitted directly on this I-D.



From nobody Tue May 12 14:21:49 2015
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Subject: Re: [tcpm] [Technical Errata Reported] RFC5925 (4365)
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Hi, all,

Thanks to Michael Sharf for pointing this out.

This can be flagged as "wait for update" because it's not critical to
the operation of the option.

Joe

On 5/12/2015 1:46 PM, RFC Errata System wrote:
> The following errata report has been submitted for RFC5925,
> "The TCP Authentication Option".
> 
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=5925&eid=4365
> 
> --------------------------------------
> Type: Technical
> Reported by: Joe Touch <touch@isi.edu>
> 
> Section: 7.6
> 
> Original Text
> -------------
>    TCP's 4-bit data offset requires that the options end 60 bytes (15
>    32-bit words) after the header begins, including the 20-byte header.
>    This leaves 40 bytes for options, of which 15 are expected in current
>    implementations (listed below), leaving at most 25 for other uses.
>    TCP-AO consumes 16 bytes, leaving 9 bytes for additional SYN options
>    (depending on implementation dependant alignment padding, which could
>    consume another 2 bytes at most).
> 
>    o  SACK permitted (2 bytes) [RFC2018][RFC3517]
> 
>    o  Timestamps (10 bytes) [RFC1323]
> 
>    o  Window scale (3 bytes) [RFC1323]
> 
> 
> Corrected Text
> --------------
>    TCP's 4-bit data offset requires that the options end 60 bytes (15
>    32-bit words) after the header begins, including the 20-byte header.
>    This leaves 40 bytes for options, of which 19 are expected in current
>    implementations (listed below), leaving at most 21 for other uses.
>    TCP-AO consumes 16 bytes, leaving 5 bytes for additional SYN options
>    (depending on implementation dependent alignment padding, which could
>    consume another 2 bytes at most).
> 
>    o  SACK permitted (2 bytes) [RFC2018][RFC3517]
> 
>    o  Timestamps (10 bytes) [RFC1323]
> 
>    o  Window scale (3 bytes) [RFC1323]
> 
>    o  Maximum Segment Size (4 bytes) [RFC793]
> 
> 
> Notes
> -----
> MSS was missing in the original text. New text includes MSS and updates numbers accordingly.
> 
> Also corrects a spelling error (dependant -> dependent), which is non-technical but included in the revised text.
> 
> Instructions:
> -------------
> This erratum is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary. 
> 
> --------------------------------------
> RFC5925 (draft-ietf-tcpm-tcp-auth-opt-11)
> --------------------------------------
> Title               : The TCP Authentication Option
> Publication Date    : June 2010
> Author(s)           : J. Touch, A. Mankin, R. Bonica
> Category            : PROPOSED STANDARD
> Source              : TCP Maintenance and Minor Extensions
> Area                : Transport
> Stream              : IETF
> Verifying Party     : IESG
> 


From nobody Wed May 13 04:53:04 2015
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Subject: Re: [tcpm] Last Call: <draft-ietf-tcpm-newcwv-10.txt> (Updating TCP to support Rate-Limited Traffic) to Experimental RFC
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Hi authors,

I've review this document (missed the WG last call though) and support 
publication of this document.

Find some comments below that I recommend to further improve readability and may 
or may not be addressed in the final version.

Mirja

--------------------

pipeACK (section 4.1 and 4.2 and 4.5.1):
1) Can you maybe add a sentence that explains why FlightSize is not sufficient. 
I understand the difference between the two variables but are there actually 
cases where it makes a big difference if one or the other is used?
2) This is just an idea and does not make a big difference: Instead of saying 
pipeACk is undefined, you could just set it to infinity or 'max' (what every the 
max in the implementation is) and this would always keep you in validated phase.
3) The implementation for pipeACK (based on HighACK) that you propose does not 
take into account that dupACK also acknowledge new data. Is this on purpose? If 
so please state explicitly.
4) Text says "When no meaurements are available, ..."
    Maybe say more explicitly "When no meaurements are available as the 
connection is idle" or "... as no data were sent during the last measurement 
period" or "... no ACK were received...".
5) I would appreciate a forward reference to section 4.5.1 here (as also done 
for other stuff as for cwnd-limited in the next section).
6) Could you also include pseudo code in 4.5.1? I far as I know you have an 
implementation, so it should not be extremely hard to come up with pseudo code 
and I personally think it would be really helpful...
7) nit in last paragraph: s/method defined on [RFC6675]is/ method defined in 
[RFC6675] is/
8) Figure 1 (section 4.5.1): Maybe add a y-axis...?

non-validated phase (section 4.3 and 4.4)
1) Maybe add one sentence saying "A connection is also in non-validated phase if 
pipeACK is zero which means the connection is idle". Saying this explicit would 
help me to understand things better.
I just notice that this might actually not be true because in the section above 
you say that pipeACK is set to undefined if no measurements are available and 
that would mean the connection goes into validated phase... okay I think there 
is some clarification needed here.
2) In section 4.4 you first say
    "cwnd neither grows nor reduces while the sender remains in this phase"
     and then
     "A sender that is cwnd-limited MAY use the standard TCP method to increase 
cwnd".
     Please clarify!
3) In section 4.4.1:
    "A sender that detects a packet-drop MUST record the current
    FlightSize in the variable LossFlightSize ..."
    Is this "packet-drop" or "congestion notification"? I guess the difference 
is that if ECN is used R=0. So that's okay. Please check again that this also 
works correctly for ECN.
4) Is this
    cwnd = (Max(pipeACK,LossFlightSize) - R)/2
    or
    cwnd = (Max(pipeACK,(LossFlightSize - R))/2  [only LossFlightSize - R] ?
    Because the text reads as if it should be the second...
5) "ssthresh is adjusted using the standard TCP method."
    This is not clear to me. Is ssthread set to cwnd/2 before the adjustment or 
to cwnd-1 after the adjustment? The first option would restart the connection in 
Slow Start... Please clarify!
6) "then this can result in the final cwnd after loss recovery being 1/4 of the 
cwnd"
    Shouldn't this be "being at max 1/4 of the cwnd"...?
7) "Subsequent updates to cwnd do not therefore reflect pipeACK history before any
    congestion event."
    Not sure I understand this sentence... which subsequent updates?
8) nit: s/updated during loss recoverySection 4.2/ updated during loss recovery 
as stated in Section 4.2/

Non-validated phase (section 4.4.3 and 5)
1) I would recommend to change the title to
"4.4.3 Adjustment at the end of the Non-Validated Phase (NVP)"
2) "An application that remains in the non-validated phase for a period
    greater than the NVP is required to adjust its congestion control
    state.  If the sender exits the non-validated phase after this
    period, it MUST update the ssthresh:"
    This is not fully clear to me. Do I have to adapt cwnd and ssthresh right 
after NVP or as soon as I leave the non-validated phase after being for at leats 
NVP time in this phase? If I'm not idle but only sending with a low rate, I 
should probably do this adjustment with the next ACK, no...?
3) Further, these adjustments mean that the connection will be in Slow Start. If 
I understand this correctly, this is not a problem as long as the connection 
stays in non-validated as the cwnd will not be further increased. But as soon as 
it leaves non-validated phase it will increase its cwnd as in Slow Start. I 
would like to have this spelled out explicitly.
4) This section does not tell/specify what value NVP should have. Please add a 
sentence that this is on purpose and add a reference to section 5.
5) Section 5 says "A period of five minutes was chosen for this NVP."
    This seeems to be a left over from an older version as nothing was specified 
so far. Please check!



On 08.05.2015 11:33, The IESG wrote:
>
> The IESG has received a request from the TCP Maintenance and Minor
> Extensions WG (tcpm) to consider the following document:
> - 'Updating TCP to support Rate-Limited Traffic'
>    <draft-ietf-tcpm-newcwv-10.txt> as Experimental 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 2015-05-22. Exceptionally, comments may be
> sent to iesg@ietf.org instead. In either case, please retain the
> beginning of the Subject line to allow automated sorting.
>
> Abstract
>
>
>     This document provides a mechanism to address issues that arise when
>     TCP is used to support traffic that exhibits periods where the
>     sending rate is limited by the application rather than the congestion
>     window.  It provides an experimental update to TCP that allows a TCP
>     sender to restart quickly following a rate-limited interval.  This
>     method is expected to benefit applications that send rate-limited
>     traffic using TCP, while also providing an appropriate response if
>     congestion is experienced.
>
>     It also evaluates the Experimental specification of TCP Congestion
>     Window Validation, CWV, defined in RFC 2861, and concludes that RFC
>     2861 sought to address important issues, but failed to deliver a
>     widely used solution.  This document therefore recommends that the
>     status of RFC 2861 is moved from Experimental to Historic, and that
>     it is replaced by the current specification.
>
>
>
>
> The file can be obtained via
> https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv/
>
> IESG discussion can be tracked via
> https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv/ballot/
>
>
> No IPR declarations have been submitted directly on this I-D.
>

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

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


From nobody Wed May 13 08:05:45 2015
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Subject: [tcpm] about synchronization losses in Cubic and draft-zimmermann-tcpm-cubic-01
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Dear all,

FYI and in case it is of interest for the "Discussion" section in 
draft-zimmermann-tcpm-cubic-01, we have published an article where we 
explain how and under which scenarios Cubic suffers from higher loss 
synchronization than standard TCP. Our observations suggest that both 
large RTTs and Fast Convergence mechanism in Cubic catalyze the 
synchronization losses. We also suggest that synchronization is 
unavoidable in high speed versions of TCP and AQM  shall help mitigating 
the problem

Synchronization has been observed in simulations, testbeds and in the 
Internet. Some experimental results are available in the article.

The paper can be downloaded at 
http://www.i3s.unice.fr/~sassatelli/articles/ITC2014.pdf

Thank you!
-- 
Dino L., Lucile S. & Guillaume UK
University Nice Sophia-Antipolis
France


From nobody Wed May 13 14:07:01 2015
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Joe Touch <touch@isi.edu>, RFC Errata System <rfc-editor@rfc-editor.org>,  "mankin@psg.com" <mankin@psg.com>, "rbonica@juniper.net" <rbonica@juniper.net>, "spencerdawkins.ietf@gmail.com" <spencerdawkins.ietf@gmail.com>, "mls.ietf@gmail.com" <mls.ietf@gmail.com>, "nishida@sfc.wide.ad.jp" <nishida@sfc.wide.ad.jp>, "pasi.sarolahti@iki.fi" <pasi.sarolahti@iki.fi>
Thread-Topic: [Technical Errata Reported] RFC5925 (4365)
Thread-Index: AQHQjPTz26oTPS9GGUaJT3EHlHrSpp14tXIAgAGwwzs=
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Indeed, "Hold for Document Update" could be appropriate.=0A=
=0A=
According to [1] "Only errors that could cause implementation or deployment=
 problems or significant confusion should be Verified." and "Things that ar=
e clearly wrong but could not cause an implementation or deployment problem=
 should be Hold for Document Update." To me this errata is more in the latt=
er camp.=0A=
=0A=
Thanks for fixing this!=0A=
=0A=
Michael=0A=
=0A=
[1] http://www.ietf.org/iesg/statement/errata-processing.html=0A=


From nobody Thu May 14 06:39:41 2015
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From: Lawrence Stewart <lstewart@room52.net>
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Subject: [tcpm] Window update handling in the BSD TCP stack
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Hi all,

I've been contemplating the window update code in FreeBSD's TCP input
path (see [1] for full context):

> 2796 	/*
> 2797 	* Update window information.
> 2798 	* Don't look at window if no ACK: TAC's send garbage on first SYN.
> 2799 	*/
> 2800 	if ((thflags & TH_ACK) &&
> 2801 	    (SEQ_LT(tp->snd_wl1, th->th_seq) ||
> 2802 	    (tp->snd_wl1 == th->th_seq && (SEQ_LT(tp->snd_wl2, th->th_ack) ||
> 2803 	    (tp->snd_wl2 == th->th_ack && tiwin > tp->snd_wnd))))) {
> 2804 		/* keep track of pure window updates */
> 2805 		if (tlen == 0 &&
> 2806 		    tp->snd_wl2 == th->th_ack && tiwin > tp->snd_wnd)
> 2807 			TCPSTAT_INC(tcps_rcvwinupd);
> 2808 		tp->snd_wnd = tiwin;
> 2809 		tp->snd_wl1 = th->th_seq;
> 2810 		tp->snd_wl2 = th->th_ack;
> 2811 		if (tp->snd_wnd > tp->max_sndwnd)
> 2812 		tp->max_sndwnd = tp->snd_wnd;
> 2813 		needoutput = 1;
> 2814 	}

The code is documented essentially as above in Stevens Vol II and
reflects RFC 793 pg 72 - it has been around a long time.

I have some concerns that the logic is too restrictive in some
situations, and would like to propose the following for discussion:

>         /*
>          * Update window information if it's an ACK (TAC's send garbage on
>          * first SYN) and:
>          *   - The peer has ACKed (snd_wl2 < th_ack) or sent (snd_wl1 < th_seq)
>          *     new data OR
>          *   - A pure window update (snd_wl1 == th_ack && snd_wl2 == th_seq &&
>          *     tiwn != snd_wnd) with a new timestamp or if not doing timestamps,
>          *     we special case zero window handling to ensure we'll enter persist.
>          */
>         if ((thflags & TH_ACK) &&
>             (SEQ_LT(tp->snd_wl1, th->th_seq) ||
>             SEQ_LT(tp->snd_wl2, th->th_ack) ||
>             (tp->snd_wl1 == th->th_seq && tp->snd_wl2 == th->th_ack &&
>             tiwin != tp->snd_wnd &&
>             (tsrecentupdated || (!tiwin && !(tp->t_flags & TF_RCVD_TSTMP)))))) {
>                 /* keep track of pure window updates */
>                 if (tlen == 0 && tp->snd_wl2 == th->th_ack &&
>                     tiwin != tp->snd_wnd)
>                         TCPSTAT_INC(tcps_rcvwinupd);
>                 tp->snd_wnd = tiwin;
>                 tp->snd_wl1 = th->th_seq;
>                 tp->snd_wl2 = th->th_ack;
>                 if (tp->snd_wnd > tp->max_sndwnd)
>                         tp->max_sndwnd = tp->snd_wnd;
>                 needoutput = 1;
>         }

The tsrecentupdated variable is new, and I default it to 0 but update to
1 anywhere in tcp_do_segment() where tp->ts_recent is set to the
incoming segment's tsval.

The main motivation for the change is correct 0 window handling (the
details of which I'll be vague about as this investigation is related to
a DoS vector I'm working on a fix for).

In the spirit of being liberal with what you accept, why shouldn't we
accept pure window updates that shrink the window if we think they
reflect the most up to date information from the peer (or in the case of
no timestamps, reflect information we could act on to be safe)?

Cheers,
Lawrence

[1]
https://svnweb.freebsd.org/base/head/sys/netinet/tcp_input.c?view=markup#l2796


From nobody Fri May 15 05:33:40 2015
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Subject: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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tcpm folks,

Despite searching and asking around, I cannot find any definitive 
reasoning for the choice of 2*SMSS in eqn (4) in RFC5681.

    ssthresh = max (FlightSize / 2, 2*SMSS)            (4)

In testing a new AQM with multiple flows, now we've got the RTT 
really low, we've found that TCP's minimum window of 2 makes TCP 
ignore the AQM's signals ('cos it always rounds up to 2) so all the 
TCP flows drive up the queueing delay to give themselves a 
large-enough RTT - defeating the AQM.



Bob



________________________________________________________________
Bob Briscoe,                                                  BT 


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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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On Fri, May 15, 2015 at 8:33 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:
> tcpm folks,
>
> Despite searching and asking around, I cannot find any definitive reasoning
> for the choice of 2*SMSS in eqn (4) in RFC5681.
>
>    ssthresh = max (FlightSize / 2, 2*SMSS)            (4)
>
> In testing a new AQM with multiple flows, now we've got the RTT really low,
> we've found that TCP's minimum window of 2 makes TCP ignore the AQM's
> signals ('cos it always rounds up to 2) so all the TCP flows drive up the
> queueing delay to give themselves a large-enough RTT - defeating the AQM.


Delayed ack?


From nobody Fri May 15 14:53:39 2015
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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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On Fri, May 15, 2015 at 5:33 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:
> tcpm folks,
>
> Despite searching and asking around, I cannot find any definitive reasoning
> for the choice of 2*SMSS in eqn (4) in RFC5681.
>
>    ssthresh = max (FlightSize / 2, 2*SMSS)            (4)
>
> In testing a new AQM with multiple flows, now we've got the RTT really low,
> we've found that TCP's minimum window of 2 makes TCP ignore the AQM's
I am confused. min ssthresh != min-cwnd == 1?

> signals ('cos it always rounds up to 2) so all the TCP flows drive up the
> queueing delay to give themselves a large-enough RTT - defeating the AQM.
>
>
>
> Bob
>
>
>
> ________________________________________________________________
> Bob Briscoe,                                                  BT
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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On Fri, 15 May 2015, Bob Briscoe wrote:

> Despite searching and asking around, I cannot find any definitive reasoning
> for the choice of 2*SMSS in eqn (4) in RFC5681.
> 
>    ssthresh = max (FlightSize / 2, 2*SMSS)            (4)
> 
> In testing a new AQM with multiple flows, now we've got the RTT really low,
> we've found that TCP's minimum window of 2 makes TCP ignore the AQM's signals
> ('cos it always rounds up to 2) so all the TCP flows drive up the queueing
> delay to give themselves a large-enough RTT - defeating the AQM.

Does this really matter that much whether ssthresh would be 1 or 2 SMSS? 
Both congestion avoidance and slow start are equally aggressive for the 
first RTT: when TCP start with cwnd=1 and the sender gets the ACK, the
applied increase is cwnd = cwnd + 1 SMSS regardless of the operating
mode.

However, ssthresh == cwnd case is left open so there might be a 
difference depending on implementation:
   "When cwnd and ssthresh are equal, the sender may use either
    slow start or congestion avoidance."
...but I don't see how that would relate to "minimum window of 2" you're 
talking here (and ssthresh is not really "window" anyway).


-- 
 i.


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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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Ilpo & Yuchung,

Sorry, the phrasing of my question was too curt.

I'm not specifically interested in why ssthresh=20
has a min. I only phrased my question that way=20
because the way TCP fast recovery is specified=20
and implemented sets a min for cwnd based on the min for ssthresh.

 From RFC5681:
2.  When the third duplicate ACK is received, a TCP MUST set ssthresh
        to no more than the value given in equation (4).
...
6.  When the next ACK arrives that acknowledges previously
        unacknowledged data, a TCP MUST set cwnd to ssthresh (the value
        set in step 2).  This is termed "deflating" the window.

The only guess I've heard for why it says 2*SMSS=20
in eqn (4) is delayed ACKs (John Heffner, as well=20
as Michael Welzl when I was asking around before=20
I posted this to tcpm). Certainly delayed ACKs=20
will cause segments to bunch in twos even if the=20
window were less than two. However, if the RTT=20
was short enough (or the link congested enough)=20
to need a low window, at least segments could=20
bunch in twos at a lower rate than two per RTT.


I believe TCP would not work against AQMs at low=20
RTTs if the min for ssthresh in Eqn (4) was zero,=20
not 2*SMSS. But there must have been a reason for=20
this minimum at some point in the past - that's what I'm trying to find out.

If there is no strong reason to keep the 2*SMSS=20
minimum, the rule after a timeout would have to=20
be changed too. Specifically, the two changes needed to RFC5681 would be:

CURRENT:
     ssthresh =3D max (FlightSize / 2, 2*SMSS)            (4)
PROPOSED:
     ssthresh =3D FlightSize / 2                          (4)

CURRENT
    upon a timeout [...] cwnd MUST be set to
    no more than the loss window, LW, which equals 1 full-sized segment.
PROPOSED:
    upon a timeout [...] cwnd MUST be set to
      cwnd =3D min(LW, ssthresh)
    where the loss window, LW, which equals 1 full-sized segment.


Bob

At 23:14 15/05/2015, Ilpo J=E4rvinen wrote:
>On Fri, 15 May 2015, Bob Briscoe wrote:
>
> > Despite searching and asking around, I cannot find any definitive=
 reasoning
> > for the choice of 2*SMSS in eqn (4) in RFC5681.
> >
> >    ssthresh =3D max (FlightSize / 2, 2*SMSS)            (4)
> >
> > In testing a new AQM with multiple flows, now we've got the RTT really=
 low,
> > we've found that TCP's minimum window of 2=20
> makes TCP ignore the AQM's signals
> > ('cos it always rounds up to 2) so all the TCP flows drive up the=
 queueing
> > delay to give themselves a large-enough RTT - defeating the AQM.
>
>Does this really matter that much whether ssthresh would be 1 or 2 SMSS?
>Both congestion avoidance and slow start are equally aggressive for the
>first RTT: when TCP start with cwnd=3D1 and the sender gets the ACK, the
>applied increase is cwnd =3D cwnd + 1 SMSS regardless of the operating
>mode.
>
>However, ssthresh =3D=3D cwnd case is left open so there might be a
>difference depending on implementation:
>    "When cwnd and ssthresh are equal, the sender may use either
>     slow start or congestion avoidance."
>...but I don't see how that would relate to "minimum window of 2" you're
>talking here (and ssthresh is not really "window" anyway).
>
>
>--
>  i.

At 22:52 15/05/2015, Yuchung Cheng wrote:
>I am confused. min ssthresh !=3D min-cwnd =3D=3D 1?



________________________________________________________________
Bob Briscoe,                                                  BT=20


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Subject: [tcpm] [Technical Errata Reported] RFC5925 (4365)
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The following errata report has been submitted for RFC5925,
"The TCP Authentication Option".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=5925&eid=4365

--------------------------------------
Type: Technical
Reported by: Joe Touch <touch@isi.edu>

Section: 7.6

Original Text
-------------
   TCP's 4-bit data offset requires that the options end 60 bytes (15
   32-bit words) after the header begins, including the 20-byte header.
   This leaves 40 bytes for options, of which 15 are expected in current
   implementations (listed below), leaving at most 25 for other uses.
   TCP-AO consumes 16 bytes, leaving 9 bytes for additional SYN options
   (depending on implementation dependant alignment padding, which could
   consume another 2 bytes at most).

   o  SACK permitted (2 bytes) [RFC2018][RFC3517]

   o  Timestamps (10 bytes) [RFC1323]

   o  Window scale (3 bytes) [RFC1323]


Corrected Text
--------------
   TCP's 4-bit data offset requires that the options end 60 bytes (15
   32-bit words) after the header begins, including the 20-byte header.
   This leaves 40 bytes for options, of which 19 are expected in current
   implementations (listed below), leaving at most 21 for other uses.
   TCP-AO consumes 16 bytes, leaving 5 bytes for additional SYN options
   (depending on implementation dependent alignment padding, which could
   consume another 2 bytes at most).

   o  SACK permitted (2 bytes) [RFC2018][RFC3517]

   o  Timestamps (10 bytes) [RFC1323]

   o  Window scale (3 bytes) [RFC1323]

   o  Maximum Segment Size (4 bytes) [RFC793]


Notes
-----
MSS was missing in the original text. New text includes MSS and updates numbers accordingly.

Also corrects a spelling error (dependant -> dependent), which is non-technical but included in the revised text.

Instructions:
-------------
This erratum is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
rejected. When a decision is reached, the verifying party (IESG)
can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC5925 (draft-ietf-tcpm-tcp-auth-opt-11)
--------------------------------------
Title               : The TCP Authentication Option
Publication Date    : June 2010
Author(s)           : J. Touch, A. Mankin, R. Bonica
Category            : PROPOSED STANDARD
Source              : TCP Maintenance and Minor Extensions
Area                : Transport
Stream              : IETF
Verifying Party     : IESG


From nobody Mon May 18 01:56:06 2015
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: WG acceptance of draft-zimmermann-tcpm-cubic 
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Date: Mon, 18 May 2015 08:55:43 +0000
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Subject: [tcpm] WG acceptance of draft-zimmermann-tcpm-cubic
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Hi all,

This e-mail conforms the TCPM WG acceptance of draft-zimmermann-tcpm-cubic-=
01. Please resubmit as draft-ietf-tcpm-cubic-00.

According to the list feedback, the TCPM consensus is that this document sh=
ould head for a status as informational RFC.

The chairs believe that it would be very valuable to have an RFC very soon,=
 in order to describe what is widely deployed in the Internet. We observe t=
hat there is also support for a PS document in this space. However, a PS do=
cument would possibly take longer to complete. Therefore, we think we can d=
iscuss a PS document once draft-ietf-tcpm-cubic is mature.

Thanks

Michael, Pasi, Yoshifumi


From nobody Mon May 18 02:12:43 2015
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: Feedback on WG acceptance of draft-bensley-tcpm-dctcp-03
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Subject: Re: [tcpm] Feedback on WG acceptance of draft-bensley-tcpm-dctcp-03
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Hi all,

There have not been many responses to this e-mail. The chairs therefore con=
clude that there is not enough support for adoption of draft-bensley-tcpm-d=
ctcp-03 as TCPM WG item as of now.

Since the TCPM community seems interested in further discussing this topic,=
 the chairs would offer to allocate time for corresponding discussions duri=
ng upcoming TCPM meetings.

Thanks

Michael, Pasi, Yoshifumi


PS: We recently found one possibly interesting study on operational experie=
nce: https://www.usenix.org/system/files/conference/nsdi15/nsdi15-paper-jud=
d.pdf


> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf, Michael
> (Michael)
> Sent: Wednesday, April 22, 2015 10:54 AM
> To: tcpm@ietf.org
> Subject: [tcpm] Feedback on WG acceptance of draft-bensley-tcpm-dctcp-
> 03
>=20
> Dear all,
>=20
> During IETF 89 [1] and on the list there has been a discussion about WG
> adoption of draft-bensley-tcpm-dctcp [2]. The understanding of the
> chairs is that there is no clear consensus up to now.
>=20
> In order to move forward, the chairs seek for community guidance
> regarding the following options to handle this draft:
>=20
> a) Adopt in TCPM as Informational
>=20
> b) Adopt in TCPM as Experimental
>=20
> c) Do not adopt in TCPM now
>=20
> Option c) does not prevent publication outside TCPM, such as
> publication by another working group (e.g., TSVWG), publication on
> Independent Stream, etc.
>=20
> The TCPM chairs believe that adoption of alternative congestion control
> algorithms in TCPM should be backed by a strong consensus. Please also
> review [3] and [4] regarding the difference between Experimental and
> Informational.
>=20
> Please let us know any feedback on a potential WG acceptance of draft-
> bensley-tcpm-dctcp-03 until May 8.
>=20
> Thanks!
>=20
> Michael, Pasi, Yoshifumi
>=20
>=20
> [1] http://www.ietf.org/proceedings/89/minutes/minutes-89-tcpm
>=20
> [2] https://tools.ietf.org/html/draft-bensley-tcpm-dctcp-03
>=20
> [3] RFC 2026, Section 4.2.1, 4.2.2
>=20
> [4] https://www.ietf.org/iesg/informational-vs-experimental.html
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "per.hurtig@kau.se" <per.hurtig@kau.se>, "anna.brunstrom@kau.se" <anna.brunstrom@kau.se>, "apetlund@simula.no" <apetlund@simula.no>, "michawe@ifi.uio.no" <michawe@ifi.uio.no>
Thread-Topic: [tcpm] draft-ietf-tcpm-rtorestart-07
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Date: Mon, 18 May 2015 09:21:54 +0000
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: Bob Briscoe <bob.briscoe@bt.com>
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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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Hi Bob,

I'm not very sure about this.
With your proposal, if FlightSize is zero, cwnd will be zero.
In this case, TCP doesn't have a chance to increase cwnd other than
keep-alive?

Thanks,
--
Yoshi

On Sat, May 16, 2015 at 4:52 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:

> Ilpo & Yuchung,
>
> Sorry, the phrasing of my question was too curt.
>
> I'm not specifically interested in why ssthresh has a min. I only phrased
> my question that way because the way TCP fast recovery is specified and
> implemented sets a min for cwnd based on the min for ssthresh.
>
> From RFC5681:
> 2.  When the third duplicate ACK is received, a TCP MUST set ssthresh
>        to no more than the value given in equation (4).
> ...
> 6.  When the next ACK arrives that acknowledges previously
>        unacknowledged data, a TCP MUST set cwnd to ssthresh (the value
>        set in step 2).  This is termed "deflating" the window.
>
> The only guess I've heard for why it says 2*SMSS in eqn (4) is delayed
> ACKs (John Heffner, as well as Michael Welzl when I was asking around
> before I posted this to tcpm). Certainly delayed ACKs will cause segments
> to bunch in twos even if the window were less than two. However, if the R=
TT
> was short enough (or the link congested enough) to need a low window, at
> least segments could bunch in twos at a lower rate than two per RTT.
>
>
> I believe TCP would not work against AQMs at low RTTs if the min for
> ssthresh in Eqn (4) was zero, not 2*SMSS. But there must have been a reas=
on
> for this minimum at some point in the past - that's what I'm trying to fi=
nd
> out.
>
> If there is no strong reason to keep the 2*SMSS minimum, the rule after a
> timeout would have to be changed too. Specifically, the two changes neede=
d
> to RFC5681 would be:
>
> CURRENT:
>     ssthresh =3D max (FlightSize / 2, 2*SMSS)            (4)
> PROPOSED:
>     ssthresh =3D FlightSize / 2                          (4)
>
> CURRENT
>    upon a timeout [...] cwnd MUST be set to
>    no more than the loss window, LW, which equals 1 full-sized segment.
> PROPOSED:
>    upon a timeout [...] cwnd MUST be set to
>      cwnd =3D min(LW, ssthresh)
>    where the loss window, LW, which equals 1 full-sized segment.
>
>
> Bob
>
>
> At 23:14 15/05/2015, Ilpo J=C3=A4rvinen wrote:
>
>> On Fri, 15 May 2015, Bob Briscoe wrote:
>>
>> > Despite searching and asking around, I cannot find any definitive
>> reasoning
>> > for the choice of 2*SMSS in eqn (4) in RFC5681.
>> >
>> >    ssthresh =3D max (FlightSize / 2, 2*SMSS)            (4)
>> >
>> > In testing a new AQM with multiple flows, now we've got the RTT really
>> low,
>> > we've found that TCP's minimum window of 2 makes TCP ignore the AQM's
>> signals
>> > ('cos it always rounds up to 2) so all the TCP flows drive up the
>> queueing
>> > delay to give themselves a large-enough RTT - defeating the AQM.
>>
>> Does this really matter that much whether ssthresh would be 1 or 2 SMSS?
>> Both congestion avoidance and slow start are equally aggressive for the
>> first RTT: when TCP start with cwnd=3D1 and the sender gets the ACK, the
>> applied increase is cwnd =3D cwnd + 1 SMSS regardless of the operating
>> mode.
>>
>> However, ssthresh =3D=3D cwnd case is left open so there might be a
>> difference depending on implementation:
>>    "When cwnd and ssthresh are equal, the sender may use either
>>     slow start or congestion avoidance."
>> ...but I don't see how that would relate to "minimum window of 2" you're
>> talking here (and ssthresh is not really "window" anyway).
>>
>>
>> --
>>  i.
>>
>
> At 22:52 15/05/2015, Yuchung Cheng wrote:
>
>> I am confused. min ssthresh !=3D min-cwnd =3D=3D 1?
>>
>
>
>
> ________________________________________________________________
> Bob Briscoe,                                                  BT
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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<div dir=3D"ltr">Hi Bob,<div><br></div><div>I&#39;m not very sure about thi=
s.</div><div>With your proposal, if FlightSize is zero, cwnd will be zero.<=
/div><div>In this case, TCP doesn&#39;t have a chance to increase cwnd othe=
r than keep-alive?</div><div><br></div><div>Thanks,</div><div>--</div><div>=
Yoshi<br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sat, =
May 16, 2015 at 4:52 AM, Bob Briscoe <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:bob.briscoe@bt.com" target=3D"_blank">bob.briscoe@bt.com</a>&gt;</span> w=
rote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-styl=
e:solid;padding-left:1ex">Ilpo &amp; Yuchung,<br>
<br>
Sorry, the phrasing of my question was too curt.<br>
<br>
I&#39;m not specifically interested in why ssthresh has a min. I only phras=
ed my question that way because the way TCP fast recovery is specified and =
implemented sets a min for cwnd based on the min for ssthresh.<br>
<br>
>From RFC5681:<br>
2.=C2=A0 When the third duplicate ACK is received, a TCP MUST set ssthresh<=
br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0to no more than the value given in equation (4).=
<br>
...<br>
6.=C2=A0 When the next ACK arrives that acknowledges previously<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0unacknowledged data, a TCP MUST set cwnd to ssth=
resh (the value<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0set in step 2).=C2=A0 This is termed &quot;defla=
ting&quot; the window.<br>
<br>
The only guess I&#39;ve heard for why it says 2*SMSS in eqn (4) is delayed =
ACKs (John Heffner, as well as Michael Welzl when I was asking around befor=
e I posted this to tcpm). Certainly delayed ACKs will cause segments to bun=
ch in twos even if the window were less than two. However, if the RTT was s=
hort enough (or the link congested enough) to need a low window, at least s=
egments could bunch in twos at a lower rate than two per RTT.<br>
<br>
<br>
I believe TCP would not work against AQMs at low RTTs if the min for ssthre=
sh in Eqn (4) was zero, not 2*SMSS. But there must have been a reason for t=
his minimum at some point in the past - that&#39;s what I&#39;m trying to f=
ind out.<br>
<br>
If there is no strong reason to keep the 2*SMSS minimum, the rule after a t=
imeout would have to be changed too. Specifically, the two changes needed t=
o RFC5681 would be:<br>
<br>
CURRENT:<span class=3D""><br>
=C2=A0 =C2=A0 ssthresh =3D max (FlightSize / 2, 2*SMSS)=C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0 =C2=A0 (4)<br></span>
PROPOSED:<br>
=C2=A0 =C2=A0 ssthresh =3D FlightSize / 2=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 (4)<br>
<br>
CURRENT<br>
=C2=A0 =C2=A0upon a timeout [...] cwnd MUST be set to<br>
=C2=A0 =C2=A0no more than the loss window, LW, which equals 1 full-sized se=
gment.<br>
PROPOSED:<br>
=C2=A0 =C2=A0upon a timeout [...] cwnd MUST be set to<br>
=C2=A0 =C2=A0 =C2=A0cwnd =3D min(LW, ssthresh)<br>
=C2=A0 =C2=A0where the loss window, LW, which equals 1 full-sized segment.<=
br>
<br>
<br>
Bob<div><div class=3D"h5"><br>
<br>
At 23:14 15/05/2015, Ilpo J=C3=A4rvinen wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
On Fri, 15 May 2015, Bob Briscoe wrote:<br>
<br>
&gt; Despite searching and asking around, I cannot find any definitive reas=
oning<br>
&gt; for the choice of 2*SMSS in eqn (4) in RFC5681.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 ssthresh =3D max (FlightSize / 2, 2*SMSS)=C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 (4)<br>
&gt;<br>
&gt; In testing a new AQM with multiple flows, now we&#39;ve got the RTT re=
ally low,<br>
&gt; we&#39;ve found that TCP&#39;s minimum window of 2 makes TCP ignore th=
e AQM&#39;s signals<br>
&gt; (&#39;cos it always rounds up to 2) so all the TCP flows drive up the =
queueing<br>
&gt; delay to give themselves a large-enough RTT - defeating the AQM.<br>
<br>
Does this really matter that much whether ssthresh would be 1 or 2 SMSS?<br=
>
Both congestion avoidance and slow start are equally aggressive for the<br>
first RTT: when TCP start with cwnd=3D1 and the sender gets the ACK, the<br=
>
applied increase is cwnd =3D cwnd + 1 SMSS regardless of the operating<br>
mode.<br>
<br>
However, ssthresh =3D=3D cwnd case is left open so there might be a<br>
difference depending on implementation:<br>
=C2=A0 =C2=A0&quot;When cwnd and ssthresh are equal, the sender may use eit=
her<br>
=C2=A0 =C2=A0 slow start or congestion avoidance.&quot;<br>
...but I don&#39;t see how that would relate to &quot;minimum window of 2&q=
uot; you&#39;re<br>
talking here (and ssthresh is not really &quot;window&quot; anyway).<br>
<br>
<br>
--<br>
=C2=A0i.<br>
</blockquote>
<br></div></div><span class=3D"">
At 22:52 15/05/2015, Yuchung Cheng wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
I am confused. min ssthresh !=3D min-cwnd =3D=3D 1?<br>
</blockquote>
<br>
<br>
<br></span>
________________________________________________________________<br>
Bob Briscoe,=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 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 BT <br><div class=3D""><div class=3D=
"h5">
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
</div></div></blockquote></div><br></div></div></div>

--90e6ba475eebf5c441051657eaa8--


From nobody Mon May 18 02:58:50 2015
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Date: Mon, 18 May 2015 11:53:08 +0200
From: Erik Hugne <erik.hugne@ericsson.com>
To: <tcpm@ietf.org>
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Cc: ingemar.s.johansson@ericsson.com
Subject: [tcpm] ip_local_reserved_ports impact on inet_csk_get_port (Linux)
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I raised this on netdev a few weeks ago, but there was no recognition.
Short story; Linux has a sysctl knob (net.ipv4.ip_local_reserved_ports)
that can be used to exclude ports from the random local port assignment
(net.ipv4.ip_local_port_range).
If the initial random port is within the reserved ports subset, the port
will be selected in an incremental/linear fashion.
https://git.kernel.org/cgit/linux/kernel/git/davem/net-next.git/tree/net/ipv4/inet_connection_sock.c#n115

I believe this port reservation is a Linux specific thing, but i was hoping
that i could get a tcpm comment if this is considered ok, or if the port
selection algorithm needs to be smartened up in the case where there are
reserved port ranges.

//E

----- Forwarded message from Erik Hugne <erik.hugne@ericsson.com> -----

Date: Tue, 5 May 2015 14:47:59 +0200
From: Erik Hugne <erik.hugne@ericsson.com>
To: davem@davemloft.net, netdev@vger.kernel.org
Cc: Richard Alpe <richard.alpe@ericsson.com>, Onar Olsen <onar.olsen@ericsson.com>, ying.xue@windriver.com
Subject: tcp: ip_local_reserved_ports impact on inet_csk_get_port()
User-Agent: Mutt/1.5.21 (2010-09-15)

Defining a port range in net.ipv4.ip_local_reserved_ports can cause a linear
and predictable behavior of inet_csk_get_port().
This occurs when smallest_rover = rover = prandom_u32() % remaining + low;
hits a value in the reserved range. The algorithm will then try the next
consecutive port number until a free port is found.

Example:
net.ipv4.ip_local_port_range = 32768	61000
net.ipv4.ip_local_reserved_ports = 35000-61000

This will give ~92% chance that the initial random port will be in the
reserved range, and that the port selection will be done linearly
starting from 32768.
Section 3.3 in RFC6056[1] describes several port selection algorithms, and Linux
seems to follow #1. This does not seem to be the best alternative since
e3826f1e946e ("net: reserve ports for applications using fixed port numbers")

If the local port range is set not to overlap with the reserved ports,
inet_csk_get_port will give a better randomness in the port selection.

//E

[1] https://tools.ietf.org/html/rfc6056


----- End forwarded message -----


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From: "Eggert, Lars" <lars@netapp.com>
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
Thread-Topic: [tcpm] Feedback on WG acceptance of draft-bensley-tcpm-dctcp-03
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Subject: Re: [tcpm] Feedback on WG acceptance of draft-bensley-tcpm-dctcp-03
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Hi,

On 2015-5-18, at 11:12, Scharf, Michael (Michael) <michael.scharf@alcatel-l=
ucent.com> wrote:
> There have not been many responses to this e-mail. The chairs therefore c=
onclude that there is not enough support for adoption of draft-bensley-tcpm=
-dctcp-03 as TCPM WG item as of now.

thanks, we'll investigate the IRTF or Independent stream then. The document=
 is ready to go from our side.

> Since the TCPM community seems interested in further discussing this topi=
c, the chairs would offer to allocate time for corresponding discussions du=
ring upcoming TCPM meetings.

I don't want to sound snarky, but it seems at odds that the WG isn't intere=
sted enough to publish the document, but still wants to discuss the topic. =
Personally, my interest in discussing this further with a group that doesn'=
t want to take on the document is now pretty low.

Lars

>=20
> Thanks
>=20
> Michael, Pasi, Yoshifumi
>=20
>=20
> PS: We recently found one possibly interesting study on operational exper=
ience: https://www.usenix.org/system/files/conference/nsdi15/nsdi15-paper-j=
udd.pdf
>=20
>=20
>> -----Original Message-----
>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf, Michael
>> (Michael)
>> Sent: Wednesday, April 22, 2015 10:54 AM
>> To: tcpm@ietf.org
>> Subject: [tcpm] Feedback on WG acceptance of draft-bensley-tcpm-dctcp-
>> 03
>>=20
>> Dear all,
>>=20
>> During IETF 89 [1] and on the list there has been a discussion about WG
>> adoption of draft-bensley-tcpm-dctcp [2]. The understanding of the
>> chairs is that there is no clear consensus up to now.
>>=20
>> In order to move forward, the chairs seek for community guidance
>> regarding the following options to handle this draft:
>>=20
>> a) Adopt in TCPM as Informational
>>=20
>> b) Adopt in TCPM as Experimental
>>=20
>> c) Do not adopt in TCPM now
>>=20
>> Option c) does not prevent publication outside TCPM, such as
>> publication by another working group (e.g., TSVWG), publication on
>> Independent Stream, etc.
>>=20
>> The TCPM chairs believe that adoption of alternative congestion control
>> algorithms in TCPM should be backed by a strong consensus. Please also
>> review [3] and [4] regarding the difference between Experimental and
>> Informational.
>>=20
>> Please let us know any feedback on a potential WG acceptance of draft-
>> bensley-tcpm-dctcp-03 until May 8.
>>=20
>> Thanks!
>>=20
>> Michael, Pasi, Yoshifumi
>>=20
>>=20
>> [1] http://www.ietf.org/proceedings/89/minutes/minutes-89-tcpm
>>=20
>> [2] https://tools.ietf.org/html/draft-bensley-tcpm-dctcp-03
>>=20
>> [3] RFC 2026, Section 4.2.1, 4.2.2
>>=20
>> [4] https://www.ietf.org/iesg/informational-vs-experimental.html
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Mon May 18 06:11:14 2015
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Subject: Re: [tcpm] ip_local_reserved_ports impact on inet_csk_get_port (Linux)
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On May 18, 11:53am, erik.hugne@ericsson.com (Erik Hugne) wrote:
-- Subject: [tcpm] ip_local_reserved_ports impact on inet_csk_get_port (Linux

| I believe this port reservation is a Linux specific thing, but i was hoping
| that i could get a tcpm comment if this is considered ok, or if the port
| selection algorithm needs to be smartened up in the case where there are
| reserved port ranges.

As another data point, NetBSD also has the port reservation feature, but
does not suffer from the linear scan:

    http://nxr.netbsd.org/xref/src/sys/netinet/portalgo.c

christos


From nobody Mon May 18 08:15:25 2015
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From: John Heffner <johnwheffner@gmail.com>
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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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On Sat, May 16, 2015 at 7:52 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:
> Ilpo & Yuchung,
>
> Sorry, the phrasing of my question was too curt.
>
> I'm not specifically interested in why ssthresh has a min. I only phrased my
> question that way because the way TCP fast recovery is specified and
> implemented sets a min for cwnd based on the min for ssthresh.
>
> From RFC5681:
> 2.  When the third duplicate ACK is received, a TCP MUST set ssthresh
>        to no more than the value given in equation (4).
> ...
> 6.  When the next ACK arrives that acknowledges previously
>        unacknowledged data, a TCP MUST set cwnd to ssthresh (the value
>        set in step 2).  This is termed "deflating" the window.
>
> The only guess I've heard for why it says 2*SMSS in eqn (4) is delayed ACKs
> (John Heffner, as well as Michael Welzl when I was asking around before I
> posted this to tcpm). Certainly delayed ACKs will cause segments to bunch in
> twos even if the window were less than two. However, if the RTT was short
> enough (or the link congested enough) to need a low window, at least
> segments could bunch in twos at a lower rate than two per RTT.

With delayed ack, reducing cwnd to one breaks the ack clock.
Transmission instead will be driven by the delayed ack timer.

  -John


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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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+1

This is also why the original initial window was 2*SMSS.

Joe

On 5/18/2015 8:15 AM, John Heffner wrote:
> With delayed ack, reducing cwnd to one breaks the ack clock.
> Transmission instead will be driven by the delayed ack timer.
> 
>   -John


From nobody Mon May 18 09:40:37 2015
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From: Erik Hugne <erik.hugne@ericsson.com>
To: Christos Zoulas <christos@zoulas.com>
Thread-Topic: [tcpm] ip_local_reserved_ports impact on inet_csk_get_port (Linux)
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Date: Mon, 18 May 2015 16:40:24 +0000
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References: <20150518095308.GA32297@haze>       from Erik Hugne (May 18, 11:53am),<20150518131108.8A54D17FDA8@rebar.astron.com>
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Cc: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] ip_local_reserved_ports impact on inet_csk_get_port (Linux)
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--_000_BB045146AF8EED43BCE05F7F2DB3385311D0114BESESSMB309erics_
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The default algo_bsd does seem to revert to a linear (pick next lower) if t=
he initial port hits a reserved range..

//E

On 18 May 2015 15:11, Christos Zoulas <christos@zoulas.com> wrote:
On May 18, 11:53am, erik.hugne@ericsson.com (Erik Hugne) wrote:
-- Subject: [tcpm] ip_local_reserved_ports impact on inet_csk_get_port (Lin=
ux

| I believe this port reservation is a Linux specific thing, but i was hopi=
ng
| that i could get a tcpm comment if this is considered ok, or if the port
| selection algorithm needs to be smartened up in the case where there are
| reserved port ranges.

As another data point, NetBSD also has the port reservation feature, but
does not suffer from the linear scan:

    http://nxr.netbsd.org/xref/src/sys/netinet/portalgo.c

christos

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<div>
<p dir=3D"ltr">The default algo_bsd does seem to revert to a linear (pick n=
ext lower) if the initial port hits a reserved range..</p>
<p dir=3D"ltr">//E<br>
</p>
<div class=3D"x_quote">On 18 May 2015 15:11, Christos Zoulas &lt;christos@z=
oulas.com&gt; wrote:<br type=3D"attribution">
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On May 18, 11:53am, erik.hugne@ericsson.com (Erik =
Hugne) wrote:<br>
-- Subject: [tcpm] ip_local_reserved_ports impact on inet_csk_get_port (Lin=
ux<br>
<br>
| I believe this port reservation is a Linux specific thing, but i was hopi=
ng<br>
| that i could get a tcpm comment if this is considered ok, or if the port<=
br>
| selection algorithm needs to be smartened up in the case where there are<=
br>
| reserved port ranges.<br>
<br>
As another data point, NetBSD also has the port reservation feature, but<br=
>
does not suffer from the linear scan:<br>
<br>
&nbsp;&nbsp;&nbsp; <a href=3D"http://nxr.netbsd.org/xref/src/sys/netinet/po=
rtalgo.c">http://nxr.netbsd.org/xref/src/sys/netinet/portalgo.c</a><br>
<br>
christos<br>
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On May 18,  4:40pm, erik.hugne@ericsson.com (Erik Hugne) wrote:
-- Subject: Re: [tcpm] ip_local_reserved_ports impact on inet_csk_get_port (L

| The default algo_bsd does seem to revert to a linear (pick next lower) if
| the initial port hits a reserved range..

algo_bsd is the "Traditional BSD Port Selection Algorithm" which is always
linear as described in RFC 6056 section 2.2.

christos


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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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On 5/18/2015 8:18 AM, Joe Touch wrote:
> +1
> 
> This is also why the original initial window was 2*SMSS.

To clarify:

	the initial CWND window spec'd in RFC2001  is 1 SMSS

	I recall reports when RFC2414 was being proposed
	that BSD never really implemented this; they started
	with 2*SMSS

	the interaction with 1*SMSS and delayed ACKs was
	known back in the late 90s (Mark Allman, Sally Floyd, et al.)

	which is why RFC2414 spec'd CWND in terms of an
	even number of SMSS bounded by 4K

In general, CWND with an odd number of SMSS is a bad idea.

Joe

> 
> Joe
> 
> On 5/18/2015 8:15 AM, John Heffner wrote:
>> With delayed ack, reducing cwnd to one breaks the ack clock.
>> Transmission instead will be driven by the delayed ack timer.
>>
>>   -John


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Date: Mon, 18 May 2015 11:44:57 -0700
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From: Matt Mathis <mattmathis@google.com>
To: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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A smidge more detail: it was understood that for efficiency w/ DA we almost
always want cwnd >= 2, however there are some corner cases (persistent
timeouts due to high losses in particular) where this seems a bit unsafe.
As a consequence you get to cwnd=1 in a few cases, but if that one packet
is delivered, you immediately go to cwnd=2.  This balance was carefully
thought out at one point, but may be due to be revisited.

Thanks,
--MM--
The best way to predict the future is to create it.  - Alan Kay

Privacy matters!  We know from recent events that people are using our
services to speak in defiance of unjust governments.   We treat privacy and
security as matters of life and death, because for some users, they are.

On Mon, May 18, 2015 at 11:25 AM, Joe Touch <touch@isi.edu> wrote:

>
>
> On 5/18/2015 8:18 AM, Joe Touch wrote:
> > +1
> >
> > This is also why the original initial window was 2*SMSS.
>
> To clarify:
>
>         the initial CWND window spec'd in RFC2001  is 1 SMSS
>
>         I recall reports when RFC2414 was being proposed
>         that BSD never really implemented this; they started
>         with 2*SMSS
>
>         the interaction with 1*SMSS and delayed ACKs was
>         known back in the late 90s (Mark Allman, Sally Floyd, et al.)
>
>         which is why RFC2414 spec'd CWND in terms of an
>         even number of SMSS bounded by 4K
>
> In general, CWND with an odd number of SMSS is a bad idea.
>
> Joe
>
> >
> > Joe
> >
> > On 5/18/2015 8:15 AM, John Heffner wrote:
> >> With delayed ack, reducing cwnd to one breaks the ack clock.
> >> Transmission instead will be driven by the delayed ack timer.
> >>
> >>   -John
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

--001a113d579629fe4005165f97bb
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<div dir=3D"ltr">A smidge more detail: it was understood that for efficienc=
y w/ DA we almost always want cwnd &gt;=3D 2, however there are some corner=
 cases (persistent timeouts due to high losses in particular) where this se=
ems a bit unsafe.=C2=A0 As a consequence you get to cwnd=3D1 in a few cases=
, but if that one packet is delivered, you immediately go to cwnd=3D2.=C2=
=A0 This balance was carefully thought out at one point, but may be due to =
be revisited.<div><br></div><div class=3D"gmail_extra"><div><div class=3D"g=
mail_signature"><div dir=3D"ltr"><div>Thanks,</div>--MM--<br>The best way t=
o predict the future is to create it. =C2=A0- Alan Kay<br><br>Privacy matte=
rs!=C2=A0 We know from recent events that people are using our services to =
speak in defiance of unjust governments. =C2=A0 We treat privacy and securi=
ty as matters of life and death, because for some users, they are.</div></d=
iv></div>
<br><div class=3D"gmail_quote">On Mon, May 18, 2015 at 11:25 AM, Joe Touch =
<span dir=3D"ltr">&lt;<a href=3D"mailto:touch@isi.edu" target=3D"_blank">to=
uch@isi.edu</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span c=
lass=3D""><br>
<br>
On 5/18/2015 8:18 AM, Joe Touch wrote:<br>
&gt; +1<br>
&gt;<br>
&gt; This is also why the original initial window was 2*SMSS.<br>
<br>
</span>To clarify:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 the initial CWND window spec&#39;d in RFC2001=
=C2=A0 is 1 SMSS<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 I recall reports when RFC2414 was being propose=
d<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 that BSD never really implemented this; they st=
arted<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 with 2*SMSS<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 the interaction with 1*SMSS and delayed ACKs wa=
s<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 known back in the late 90s (Mark Allman, Sally =
Floyd, et al.)<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 which is why RFC2414 spec&#39;d CWND in terms o=
f an<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 even number of SMSS bounded by 4K<br>
<br>
In general, CWND with an odd number of SMSS is a bad idea.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Joe<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt;<br>
&gt; Joe<br>
&gt;<br>
&gt; On 5/18/2015 8:15 AM, John Heffner wrote:<br>
&gt;&gt; With delayed ack, reducing cwnd to one breaks the ack clock.<br>
&gt;&gt; Transmission instead will be driven by the delayed ack timer.<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0-John<br>
<br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
</div></div></blockquote></div><br></div></div>

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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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On Mon, May 18, 2015 at 8:15 AM, John Heffner <johnwheffner@gmail.com> wrote:
>
> On Sat, May 16, 2015 at 7:52 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:
> > Ilpo & Yuchung,
> >
> > Sorry, the phrasing of my question was too curt.
> >
> > I'm not specifically interested in why ssthresh has a min. I only phrased my
> > question that way because the way TCP fast recovery is specified and
> > implemented sets a min for cwnd based on the min for ssthresh.
> >
> > From RFC5681:
> > 2.  When the third duplicate ACK is received, a TCP MUST set ssthresh
> >        to no more than the value given in equation (4).
> > ...
> > 6.  When the next ACK arrives that acknowledges previously
> >        unacknowledged data, a TCP MUST set cwnd to ssthresh (the value
> >        set in step 2).  This is termed "deflating" the window.
> >
> > The only guess I've heard for why it says 2*SMSS in eqn (4) is delayed ACKs
> > (John Heffner, as well as Michael Welzl when I was asking around before I
> > posted this to tcpm). Certainly delayed ACKs will cause segments to bunch in
> > twos even if the window were less than two. However, if the RTT was short
> > enough (or the link congested enough) to need a low window, at least
> > segments could bunch in twos at a lower rate than two per RTT.
>
> With delayed ack, reducing cwnd to one breaks the ack clock.
> Transmission instead will be driven by the delayed ack timer.
Delayed-ack may not be a concern in fast recovery, because the
receiver does not delay ACK when there is out-of-order data in the
queue.

Bob: does reducing ssthresh lower-bound to 1 help your aqm test case?

>
>   -John


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Subject: [tcpm] Milestones changed for tcpm WG
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Changed milestone "Submit document on CUBIC congestion control to the
IESG for publication as Informational RFC", set state to active from
review, accepting new milestone.

URL: https://datatracker.ietf.org/wg/tcpm/charter/


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To: Matt Mathis <mattmathis@google.com>, Joe Touch <touch@isi.edu>, "John Heffner" <johnwheffner@gmail.com>
From: Bob Briscoe <bob.briscoe@bt.com>
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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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--=====================_1459739029==.ALT
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Matt, Joe, John,

Thx for explaining the history.

I am saying what Matt just hinted at. "This decision may need to be=
 revisited."

First, this affects well designed systems, even a=20
broadband network with a good set of local CDNs=20
and AQM installed everywhere, so most usage will=20
see a very low RTT. Not just overloaded, underprovisioned networks with no=
 AQM.

For window, w < 2*SMSS, the sender would need to=20
pace segments at 1/w per RTT (as implemented in=20
Linux for TCP Nice). But only if it also implemented ABC.

Then, at a low RTT, transmissions wouldn't be=20
driven by the delayed ACK timer, they would be=20
driven by the delayed ACK /factor/ (ie the number=20
of segments received before a delayed ACK is=20
emitted). With cwnd =3D 1, after 2 short RTTs, the=20
receiver would release an ACK, (before the DA=20
timer expires), which in turn would release two=20
segments. So it gets an average window of 1, but as 2 segments every 2 RTTs.

If the min cwnd of 2*SMSS has been hit, sending=20
two segments per RTT but pushing up the queue to=20
do so, doesn't make throughput any higher. It=20
just increases the RTT for yourself and everyone else (even with an AQM).

In the light of renewed interest in deploying=20
AQM, we have to review which is more important:
* avoiding TCP being clocked by the delayed ACK=20
mechanism (does this make significant difference to performance?)?
* or avoiding TCP artifically inflating the=20
bottleneck queue (which does make a real=20
difference - it reduces delay, for the user and everyone else)?


Bob

At 19:44 18/05/2015, Matt Mathis wrote:
>A smidge more detail: it was understood that for=20
>efficiency w/ DA we almost always want cwnd >=3D=20
>2, however there are some corner cases=20
>(persistent timeouts due to high losses in=20
>particular) where this seems a bit unsafe.=C2  As=20
>a consequence you get to cwnd=3D1 in a few cases,=20
>but if that one packet is delivered, you=20
>immediately go to cwnd=3D2.=C2  This balance was=20
>carefully thought out at one point, but may be due to be revisited.
>
>Thanks,
>--MM--
>The best way to predict the future is to create it. =C2 - Alan Kay
>
>Privacy matters!=C2  We know from recent events=20
>that people are using our services to speak in=20
>defiance of unjust governments. =C2  We treat=20
>privacy and security as matters of life and=20
>death, because for some users, they are.
>
>On Mon, May 18, 2015 at 11:25 AM, Joe Touch=20
><<mailto:touch@isi.edu>touch@isi.edu> wrote:
>
>
>On 5/18/2015 8:18 AM, Joe Touch wrote:
> > +1
> >
> > This is also why the original initial window was 2*SMSS.
>
>To clarify:
>
>=C2  =C2  =C2  =C2  the initial CWND window spec'd in RFC2001=C2  is 1 SMSS
>
>=C2  =C2  =C2  =C2  I recall reports when RFC2414 was being proposed
>=C2  =C2  =C2  =C2  that BSD never really implemented this; they started
>=C2  =C2  =C2  =C2  with 2*SMSS
>
>=C2  =C2  =C2  =C2  the interaction with 1*SMSS and delayed ACKs was
>=C2  =C2  =C2  =C2  known back in the late 90s (Mark Allman, Sally Floyd,=
 et al.)
>
>=C2  =C2  =C2  =C2  which is why RFC2414 spec'd CWND in terms of an
>=C2  =C2  =C2  =C2  even number of SMSS bounded by 4K
>
>In general, CWND with an odd number of SMSS is a bad idea.
>
>Joe
>
> >
> > Joe
> >
> > On 5/18/2015 8:15 AM, John Heffner wrote:
> >> With delayed ack, reducing cwnd to one breaks the ack clock.
> >> Transmission instead will be driven by the delayed ack timer.
> >>
> >>=C2  =C2 -John
>
>_______________________________________________
>tcpm mailing list
><mailto:tcpm@ietf.org>tcpm@ietf.org
>https://www.ietf.org/mailman/listinfo/tcpm
>

________________________________________________________________
Bob Briscoe,                                                  BT=20
--=====================_1459739029==.ALT
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<body>
Matt, Joe, John,<br><br>
Thx for explaining the history.<br><br>
I am saying what Matt just hinted at. &quot;This decision may need to be
revisited.&quot;<br><br>
First, this affects well designed systems, even a broadband network with
a good set of local CDNs and AQM installed everywhere, so most usage will
see a very low RTT. Not just overloaded, underprovisioned networks with
no AQM.<br><br>
For window, w &lt; 2*SMSS, the sender would need to pace segments at 1/w
per RTT (as implemented in Linux for TCP Nice). But only if it also
implemented ABC.<br><br>
Then, at a low RTT, transmissions wouldn't be driven by the delayed ACK
timer, they would be driven by the delayed ACK /factor/ (ie the number of
segments received before a delayed ACK is emitted). With cwnd =3D 1, after
2 short RTTs, the receiver would release an ACK, (before the DA timer
expires), which in turn would release two segments. So it gets an average
window of 1, but as 2 segments every 2 RTTs.<br><br>
If the min cwnd of 2*SMSS has been hit, sending two segments per RTT but
pushing up the queue to do so, doesn't make throughput any higher. It
just increases the RTT for yourself and everyone else (even with an
AQM).<br><br>
In the light of renewed interest in deploying AQM, we have to review
which is more important:<br>
* avoiding TCP being clocked by the delayed ACK mechanism (does this make
significant difference to performance?)?<br>
* or avoiding TCP artifically inflating the bottleneck queue (which does
make a real difference - it reduces delay, for the user and everyone
else)?<br><br>
<br>
Bob<br><br>
At 19:44 18/05/2015, Matt Mathis wrote:<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">A smidge more detail: it was
understood that for efficiency w/ DA we almost always want cwnd &gt;=3D 2,
however there are some corner cases (persistent timeouts due to high
losses in particular) where this seems a bit unsafe.=C2&nbsp; As a
consequence you get to cwnd=3D1 in a few cases, but if that one packet is
delivered, you immediately go to cwnd=3D2.=C2&nbsp; This balance was
carefully thought out at one point, but may be due to be
revisited.<br><br>
Thanks,<br>
--MM--<br>
The best way to predict the future is to create it. =C2 - Alan Kay<br><br>
Privacy matters!=C2&nbsp; We know from recent events that people are using
our services to speak in defiance of unjust governments. =C2&nbsp; We treat
privacy and security as matters of life and death, because for some
users, they are.<br><br>
On Mon, May 18, 2015 at 11:25 AM, Joe Touch
&lt;<a href=3D"mailto:touch@isi.edu">touch@isi.edu</a>&gt; wrote:<br>

<dl><br><br>

<dd>On 5/18/2015 8:18 AM, Joe Touch wrote:<br>

<dd>&gt; +1<br>

<dd>&gt;<br>

<dd>&gt; This is also why the original initial window was
2*SMSS.<br><br>

<dd>To clarify:<br><br>

<dd>=C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; the initial CWND window spec'd i=
n
RFC2001=C2&nbsp; is 1 SMSS<br><br>

<dd>=C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; I recall reports when RFC2414 wa=
s
being proposed<br>

<dd>=C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; that BSD never really implemente=
d
this; they started<br>

<dd>=C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; with 2*SMSS<br><br>

<dd>=C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; the interaction with 1*SMSS and
delayed ACKs was<br>

<dd>=C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; known back in the late 90s (Mark
Allman, Sally Floyd, et al.)<br><br>

<dd>=C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; which is why RFC2414 spec'd CWND=
 in
terms of an<br>

<dd>=C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; even number of SMSS bounded by
4K<br><br>

<dd>In general, CWND with an odd number of SMSS is a bad idea.<br>
<font color=3D"#888888"><br>

<dd>Joe<br>
</font><br>

<dd>&gt;<br>

<dd>&gt; Joe<br>

<dd>&gt;<br>

<dd>&gt; On 5/18/2015 8:15 AM, John Heffner wrote:<br>

<dd>&gt;&gt; With delayed ack, reducing cwnd to one breaks the ack
clock.<br>

<dd>&gt;&gt; Transmission instead will be driven by the delayed ack
timer.<br>

<dd>&gt;&gt;<br>

<dd>&gt;&gt;=C2&nbsp; =C2 -John<br><br>

<dd>_______________________________________________<br>

<dd>tcpm mailing list<br>

<dd><a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>

<dd><a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" eudora=3D"autourl=
">
https://www.ietf.org/mailman/listinfo/tcpm</a><br><br>

</dl></blockquote>
<x-sigsep><p></x-sigsep>
________________________________________________________________<br>
Bob
Briscoe,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
BT</body>
</html>

--=====================_1459739029==.ALT--


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To: Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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Yuchung,

At 20:07 18/05/2015, Yuchung Cheng wrote:
>Bob: does reducing ssthresh lower-bound to 1 help your aqm test case?

See email just sent to Joe, Matt, John.

I knew we would need pacing for a sub-SMSS window, but I thought we 
could at least do an initial fix to reduce the min cwnd to 1*SMSS 
without pacing. However, people's responses have made me realise that 
we would need pacing in all cases when cwnd < 2*SMSS.

You will know better than me whether pacing would be feasible to add.


Bob


________________________________________________________________
Bob Briscoe,                                                  BT 


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References: <201505151233.t4FCXRxW022847@bagheera.jungle.bt.co.uk> <alpine.DEB.2.02.1505160100440.4699@melkinpaasi.cs.helsinki.fi> <201505161152.t4GBqCmx001121@bagheera.jungle.bt.co.uk> <CABrhC0=FKvgmoJush4s6FMi+X3QjwimY0fdx8iYWW-xT_0aCaw@mail.gmail.com> <555A02C6.5050100@isi.edu> <555A2E8E.4070101@isi.edu> <CAH56bmDxZA01nKJXMnuURgN8Oo001vBFbSBrcpwPwOg5QL4K2A@mail.gmail.com> <201505190954.t4J9s0ms012580@bagheera.jungle.bt.co.uk>
To: Bob Briscoe <bob.briscoe@bt.com>
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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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> On May 19, 2015, at 5:53 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:
>=20
> Matt, Joe, John,
>=20
> Thx for explaining the history.
>=20
> I am saying what Matt just hinted at. "This decision may need to be revisi=
ted."
>=20
> First, this affects well designed systems, even a broadband network with a=
 good set of local CDNs and AQM installed everywhere, so most usage will see=
 a very low RTT. Not just overloaded, underprovisioned networks with no AQM.=

>=20
> For window, w < 2*SMSS, the sender would need to pace segments at 1/w per R=
TT (as implemented in Linux for TCP Nice). But only if it also implemented A=
BC.
>=20
> Then, at a low RTT, transmissions wouldn't be driven by the delayed ACK ti=
mer, they would be driven by the delayed ACK /factor/ (ie the number of segm=
ents received before a delayed ACK is emitted). With cwnd =3D 1, after 2 sho=
rt RTTs, the receiver would release an ACK, (before the DA timer expires), w=
hich in turn would release two segments. So it gets an average window of 1, b=
ut as 2 segments every 2 RTTs.
>=20
> If the min cwnd of 2*SMSS has been hit, sending two segments per RTT but p=
ushing up the queue to do so, doesn't make throughput any higher. It just in=
creases the RTT for yourself and everyone else (even with an AQM).
>=20
> In the light of renewed interest in deploying AQM, we have to review which=
 is more important:
> * avoiding TCP being clocked by the delayed ACK mechanism (does this make s=
ignificant difference to performance?)?
> * or avoiding TCP artifically inflating the bottleneck queue (which does m=
ake a real difference - it reduces delay, for the user and everyone else)?

Another degree of freedom is segment size.  For instance, if the expected fa=
ir share is less than 2 mtu a network operator could reduce mtu.  Or tcp mig=
ht try to be adaptive.

  -John


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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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On Tue, 19 May 2015, John Heffner wrote:

> > On May 19, 2015, at 5:53 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:
> > 
> > Matt, Joe, John,
> > 
> > Thx for explaining the history.
> > 
> > I am saying what Matt just hinted at. "This decision may need to be revisited."
> > 
> > First, this affects well designed systems, even a broadband network with a good set of local CDNs and AQM installed everywhere, so most usage will see a very low RTT. Not just overloaded, underprovisioned networks with no AQM.
> > 
> > For window, w < 2*SMSS, the sender would need to pace segments at 1/w per RTT (as implemented in Linux for TCP Nice). But only if it also implemented ABC.
> > 
> > Then, at a low RTT, transmissions wouldn't be driven by the delayed ACK timer, they would be driven by the delayed ACK /factor/ (ie the number of segments received before a delayed ACK is emitted). With cwnd = 1, after 2 short RTTs, the receiver would release an ACK, (before the DA timer expires), which in turn would release two segments. So it gets an average window of 1, but as 2 segments every 2 RTTs.
> > 
> > If the min cwnd of 2*SMSS has been hit, sending two segments per RTT but pushing up the queue to do so, doesn't make throughput any higher. It just increases the RTT for yourself and everyone else (even with an AQM).
> > 
> > In the light of renewed interest in deploying AQM, we have to review which is more important:
> > * avoiding TCP being clocked by the delayed ACK mechanism (does this make significant difference to performance?)?
> > * or avoiding TCP artifically inflating the bottleneck queue (which does make a real difference - it reduces delay, for the user and everyone else)?
> 
> Another degree of freedom is segment size.  For instance, if the 
> expected fair share is less than 2 mtu a network operator could reduce 
> mtu.  Or tcp might try to be adaptive.

Linux actually already does segment size reduction for the cases where 
receiver advertized window is too small to fit enough packets. However, in 
case of congestion, reducing segment size works to some extent but in the 
end of that road lurks congestion collapse due to header overhead so it 
might not be wise to proceed towards that direction.

Pacing, on the other hard, wouldn't impose similar utilization reductions 
due to overhead increase.


-- 
 i.


From nobody Tue May 19 10:07:28 2015
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To: Bob Briscoe <bob.briscoe@bt.com>, Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] Why is the minimum ssthresh 2*SMSS (rather than 1)?
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It might be useful to take a look at this too:

http://www.computer.org/csdl/proceedings/icnp/1997/8061/00/80610205-abs.html

On 5/19/2015 2:59 AM, Bob Briscoe wrote:
> Yuchung,
> 
> At 20:07 18/05/2015, Yuchung Cheng wrote:
>> Bob: does reducing ssthresh lower-bound to 1 help your aqm test case?
> 
> See email just sent to Joe, Matt, John.
> 
> I knew we would need pacing for a sub-SMSS window, but I thought we
> could at least do an initial fix to reduce the min cwnd to 1*SMSS
> without pacing. However, people's responses have made me realise that we
> would need pacing in all cases when cwnd < 2*SMSS.
> 
> You will know better than me whether pacing would be feasible to add.
> 
> 
> Bob
> 
> 
> ________________________________________________________________
> Bob Briscoe,                                                  BT
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Wed May 20 06:21:05 2015
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Subject: [tcpm] Why is Cubic said to be RTT-fair?
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--047d7bea453c4a1c850516834c84
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Hi,

As per my understanding, unlike other TCP variants, CUBIC claims to be RTT
fair. This is because the window growth rate is independent of the RTT, and
hence the performance is the same as on a low and a high RTT paths. Others
are dependent upon RTT since their windows grow as they receive ACKs from
their peers, while CUBIC doesnt (window growth is a function of time). This
is theory.

In practice, CUBIC (which is the default now on most linux distributions)
also depends upon RTT. I connected to machines on a LAN and did a FTP
transfer. I got a certain bandwidth. Next, i artificially inserted delay
using "tc qdisc" in the path. I increased the delay on the ethernet
interface connecting my linux machine to the LAN to be around 100ms. I
immediately saw that the TCP performance went down.

To bring it up to the same level as before i had to use multiple TCP
streams. To me this means that CUBIC performance is dependent of the RTT.
So why do we call it RTT-fair?

Thanks, Toms

--047d7bea453c4a1c850516834c84
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<div dir=3D"ltr">Hi,<div><br></div>As per my understanding, unlike other TC=
P variants, CUBIC claims to be RTT fair. This is because the window growth =
rate is independent of the RTT, and hence the performance is the same as on=
 a low and a high RTT paths. Others are dependent upon RTT since their wind=
ows grow as they receive ACKs from their peers, while CUBIC doesnt (window =
growth is a function of time). This is theory.<br><br>In practice, CUBIC (w=
hich is the default now on most linux distributions) also depends upon RTT.=
 I connected to machines on a LAN and did a FTP transfer. I got a certain b=
andwidth. Next, i artificially inserted delay using &quot;tc qdisc&quot; in=
 the path. I increased the delay on the ethernet interface connecting my li=
nux machine to the LAN to be around 100ms. I immediately saw that the TCP p=
erformance went down.<br><br>To bring it up to the same level as before i h=
ad to use multiple TCP streams. To me this means that CUBIC performance is =
dependent of the RTT. So why do we call it RTT-fair?<div><br></div><div>Tha=
nks, Toms<div>
</div></div></div>

--047d7bea453c4a1c850516834c84--


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Subject: Re: [tcpm] Why is Cubic said to be RTT-fair?
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On May 20, 2015, at 6:20 AM, Tom Sanders <toms.sanders@gmail.com> wrote:

> Hi,
>=20
> As per my understanding, unlike other TCP variants, CUBIC claims to be =
RTT fair. This is because the window growth rate is independent of the =
RTT, and hence the performance is the same as on a low and a high RTT =
paths. Others are dependent upon RTT since their windows grow as they =
receive ACKs from their peers, while CUBIC doesnt (window growth is a =
function of time). This is theory.
>=20
> In practice, CUBIC (which is the default now on most linux =
distributions) also depends upon RTT. I connected to machines on a LAN =
and did a FTP transfer. I got a certain bandwidth. Next, i artificially =
inserted delay using "tc qdisc" in the path. I increased the delay on =
the ethernet interface connecting my linux machine to the LAN to be =
around 100ms. I immediately saw that the TCP performance went down.
>=20
> To bring it up to the same level as before i had to use multiple TCP =
streams. To me this means that CUBIC performance is dependent of the =
RTT. So why do we call it RTT-fair?

I cannot speak to CUBIC specifically, but I would think you would want =
to make sure you were seeing effects of congestion window and not of the =
classic TCP window.  For example, if the FTP client/server you were =
using makes an explicit setsockopt() to set the socket buffer sizes and =
by extension the classic TCP window, your bumping of the RTT to around =
100ms may have taken the bandwidth down as a consequence of the classic:

Throughput <=3D WindowSize/RTT

Similarly, since you mention Linux, if the FTP client/server didn=92t =
make explicit setsockopt() calls,  the sysctl values for tcp_rmem and =
tcp_wmem may have been such that the auto tuning of the window size =
couldn=92t allow the classic TCP window to grow =93enough.=94

I suspect that the experiment needs to be setup to have the same classic =
window size in both cases, sized per the above to be sufficient to =
achieve the peak throughput in the high RTT case.  Then you will have =
eliminated classic window as a factor.

Also, depending on the version of the Linux kernel you are using, there =
may be some effects from tcp small queues or perhaps (a stretch?) even =
byte queue limits which may not be playing well with netem.  (Just =
guessing there really)

happy benchmarking,

rick jones=


From nobody Wed May 20 10:18:36 2015
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From: Tom Sanders <toms.sanders@gmail.com>
To: Rick Jones <perfgeek@mac.com>
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Subject: Re: [tcpm] Why is Cubic said to be RTT-fair?
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Thanks Rick, this was very helpful.

So if there are no setsockopts, etc then CUBIC's performance, unlike other
TCP variants, should not get impacted by RTTs, right? Is that a fair
statement to make?

Toms

On 20 May 2015 at 19:51, Rick Jones <perfgeek@mac.com> wrote:

>
> On May 20, 2015, at 6:20 AM, Tom Sanders <toms.sanders@gmail.com> wrote:
>
> > Hi,
> >
> > As per my understanding, unlike other TCP variants, CUBIC claims to be
> RTT fair. This is because the window growth rate is independent of the RT=
T,
> and hence the performance is the same as on a low and a high RTT paths.
> Others are dependent upon RTT since their windows grow as they receive AC=
Ks
> from their peers, while CUBIC doesnt (window growth is a function of time=
).
> This is theory.
> >
> > In practice, CUBIC (which is the default now on most linux
> distributions) also depends upon RTT. I connected to machines on a LAN an=
d
> did a FTP transfer. I got a certain bandwidth. Next, i artificially
> inserted delay using "tc qdisc" in the path. I increased the delay on the
> ethernet interface connecting my linux machine to the LAN to be around
> 100ms. I immediately saw that the TCP performance went down.
> >
> > To bring it up to the same level as before i had to use multiple TCP
> streams. To me this means that CUBIC performance is dependent of the RTT.
> So why do we call it RTT-fair?
>
> I cannot speak to CUBIC specifically, but I would think you would want to
> make sure you were seeing effects of congestion window and not of the
> classic TCP window.  For example, if the FTP client/server you were using
> makes an explicit setsockopt() to set the socket buffer sizes and by
> extension the classic TCP window, your bumping of the RTT to around 100ms
> may have taken the bandwidth down as a consequence of the classic:
>
> Throughput <=3D WindowSize/RTT
>
> Similarly, since you mention Linux, if the FTP client/server didn=E2=80=
=99t make
> explicit setsockopt() calls,  the sysctl values for tcp_rmem and tcp_wmem
> may have been such that the auto tuning of the window size couldn=E2=80=
=99t allow
> the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D
>
> I suspect that the experiment needs to be setup to have the same classic
> window size in both cases, sized per the above to be sufficient to achiev=
e
> the peak throughput in the high RTT case.  Then you will have eliminated
> classic window as a factor.
>
> Also, depending on the version of the Linux kernel you are using, there
> may be some effects from tcp small queues or perhaps (a stretch?) even by=
te
> queue limits which may not be playing well with netem.  (Just guessing
> there really)
>
> happy benchmarking,
>
> rick jones




--=20
Toms.

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<div dir=3D"ltr">Thanks Rick, this was very helpful.<div><br></div><div>So =
if there are no setsockopts, etc then CUBIC&#39;s performance, unlike other=
 TCP variants, should not get impacted by RTTs, right? Is that a fair state=
ment to make?</div><div><br></div><div>Toms<div><div class=3D"gmail_extra">=
<br><div class=3D"gmail_quote">On 20 May 2015 at 19:51, Rick Jones <span di=
r=3D"ltr">&lt;<a href=3D"mailto:perfgeek@mac.com" target=3D"_blank">perfgee=
k@mac.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
On May 20, 2015, at 6:20 AM, Tom Sanders &lt;<a href=3D"mailto:toms.sanders=
@gmail.com" target=3D"_blank">toms.sanders@gmail.com</a>&gt; wrote:<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; As per my understanding, unlike other TCP variants, CUBIC claims to be=
 RTT fair. This is because the window growth rate is independent of the RTT=
, and hence the performance is the same as on a low and a high RTT paths. O=
thers are dependent upon RTT since their windows grow as they receive ACKs =
from their peers, while CUBIC doesnt (window growth is a function of time).=
 This is theory.<br>
&gt;<br>
&gt; In practice, CUBIC (which is the default now on most linux distributio=
ns) also depends upon RTT. I connected to machines on a LAN and did a FTP t=
ransfer. I got a certain bandwidth. Next, i artificially inserted delay usi=
ng &quot;tc qdisc&quot; in the path. I increased the delay on the ethernet =
interface connecting my linux machine to the LAN to be around 100ms. I imme=
diately saw that the TCP performance went down.<br>
&gt;<br>
&gt; To bring it up to the same level as before i had to use multiple TCP s=
treams. To me this means that CUBIC performance is dependent of the RTT. So=
 why do we call it RTT-fair?<br>
<br>
I cannot speak to CUBIC specifically, but I would think you would want to m=
ake sure you were seeing effects of congestion window and not of the classi=
c TCP window.=C2=A0 For example, if the FTP client/server you were using ma=
kes an explicit setsockopt() to set the socket buffer sizes and by extensio=
n the classic TCP window, your bumping of the RTT to around 100ms may have =
taken the bandwidth down as a consequence of the classic:<br>
<br>
Throughput &lt;=3D WindowSize/RTT<br>
<br>
Similarly, since you mention Linux, if the FTP client/server didn=E2=80=99t=
 make explicit setsockopt() calls,=C2=A0 the sysctl values for tcp_rmem and=
 tcp_wmem may have been such that the auto tuning of the window size couldn=
=E2=80=99t allow the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D<b=
r>
<br>
I suspect that the experiment needs to be setup to have the same classic wi=
ndow size in both cases, sized per the above to be sufficient to achieve th=
e peak throughput in the high RTT case.=C2=A0 Then you will have eliminated=
 classic window as a factor.<br>
<br>
Also, depending on the version of the Linux kernel you are using, there may=
 be some effects from tcp small queues or perhaps (a stretch?) even byte qu=
eue limits which may not be playing well with netem.=C2=A0 (Just guessing t=
here really)<br>
<br>
happy benchmarking,<br>
<br>
rick jones</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><d=
iv>Toms.</div>
</div></div></div></div>

--001a11c26140c632fd0516869dad--


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Subject: Re: [tcpm] Why is Cubic said to be RTT-fair?
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On 05/20/2015 10:18 AM, Tom Sanders wrote:
> Thanks Rick, this was very helpful.
>
> So if there are no setsockopts, etc then CUBIC's performance, unlike
> other TCP variants, should not get impacted by RTTs, right? Is that a
> fair statement to make?

Like I said, I cannot speak to CUBIC specifically.  But, since it is 
possible to alter the default congestion control under Linux, if you 
have a sufficiently stable test environment, you could run the different 
congestion controls through your added delay and see what you get.

I would suggest setting the likes of net.core.[rw]mem_max to a value 
large enough to allow for "link rate" for your physical network even at 
100ms of RTT, and then run the likes of either iperf or netperf (*) with 
them making the setsockopt()s to that value to eliminate concerns about 
sufficient classic TCP window and/or Linux's autotuning.  Lather, rinse, 
repeat with the different congestion controls set to default via 
net.ipv4.tcp_congestion_control or if using netperf there is the 
test-specific -K option to specify the congestion control to use 
(assuming that functionality hasn't bitrotted)

I believe that iperf can be told to emit interim results, which would 
allow you to see (perhaps) how things "ramp-up" - there is a similar, 
optional feature of netperf which will cause it to emit interim results 
at a specified cadence.

Or you could just take packet traces and run them through the likes of 
tcptrace to get pretty pictures that way.

rick

(*) rather than FTP or some other file transfer because you want to 
eliminate filesystem effects

>
> Toms
>
> On 20 May 2015 at 19:51, Rick Jones <perfgeek@mac.com
> <mailto:perfgeek@mac.com>> wrote:
>
>
>     On May 20, 2015, at 6:20 AM, Tom Sanders <toms.sanders@gmail.com
>     <mailto:toms.sanders@gmail.com>> wrote:
>
>      > Hi,
>      >
>      > As per my understanding, unlike other TCP variants, CUBIC claims
>     to be RTT fair. This is because the window growth rate is
>     independent of the RTT, and hence the performance is the same as on
>     a low and a high RTT paths. Others are dependent upon RTT since
>     their windows grow as they receive ACKs from their peers, while
>     CUBIC doesnt (window growth is a function of time). This is theory.
>      >
>      > In practice, CUBIC (which is the default now on most linux
>     distributions) also depends upon RTT. I connected to machines on a
>     LAN and did a FTP transfer. I got a certain bandwidth. Next, i
>     artificially inserted delay using "tc qdisc" in the path. I
>     increased the delay on the ethernet interface connecting my linux
>     machine to the LAN to be around 100ms. I immediately saw that the
>     TCP performance went down.
>      >
>      > To bring it up to the same level as before i had to use multiple
>     TCP streams. To me this means that CUBIC performance is dependent of
>     the RTT. So why do we call it RTT-fair?
>
>     I cannot speak to CUBIC specifically, but I would think you would
>     want to make sure you were seeing effects of congestion window and
>     not of the classic TCP window.  For example, if the FTP
>     client/server you were using makes an explicit setsockopt() to set
>     the socket buffer sizes and by extension the classic TCP window,
>     your bumping of the RTT to around 100ms may have taken the bandwidth
>     down as a consequence of the classic:
>
>     Throughput <= WindowSize/RTT
>
>     Similarly, since you mention Linux, if the FTP client/server didnt
>     make explicit setsockopt() calls,  the sysctl values for tcp_rmem
>     and tcp_wmem may have been such that the auto tuning of the window
>     size couldnt allow the classic TCP window to grow enough.
>
>     I suspect that the experiment needs to be setup to have the same
>     classic window size in both cases, sized per the above to be
>     sufficient to achieve the peak throughput in the high RTT case.
>     Then you will have eliminated classic window as a factor.
>
>     Also, depending on the version of the Linux kernel you are using,
>     there may be some effects from tcp small queues or perhaps (a
>     stretch?) even byte queue limits which may not be playing well with
>     netem.  (Just guessing there really)
>
>     happy benchmarking,
>
>     rick jones
>
>
>
>
> --
> Toms.
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


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From: Tom Sanders <toms.sanders@gmail.com>
To: Neal Cardwell <ncardwell@google.com>
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Subject: Re: [tcpm] Why is Cubic said to be RTT-fair?
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Hi Neal,

This is odd.

In the original Cubic paper (
http://www4.ncsu.edu/~rhee/export/bitcp/cubic-paper.pdf) by Rhee and Xu,
they explicitly state that the Cubic is independent of RTT.

"The main feature of CUBIC is that its window growth function is defined in
real-time so that its growth will be independent of RTT. Our work was
partially inspired by HTCP [5], whose window growth function is also based
on real time. The congestion epoch period of CUBIC is determined by the
packet loss rate alone. As TCP=E2=80=99s throughput is defined by the packe=
t loss
rate as well as RTT, the throughput of CUBIC is defined by only the packet
loss rate. Thus, when the loss rate is high and/or RTT is short, CUBIC can
operate in a TCP mode. Moreover, since the growth function is independent
of RTT, its RTT fairness is guaranteed as different RTT flows will still
grow their windows at the same rate."

Am i missing something here?

Thanks, Toms

On 20 May 2015 at 23:13, Neal Cardwell <ncardwell@google.com> wrote:

> On Wed, May 20, 2015 at 1:18 PM, Tom Sanders <toms.sanders@gmail.com>
> wrote:
> > Thanks Rick, this was very helpful.
> >
> > So if there are no setsockopts, etc then CUBIC's performance, unlike
> other
> > TCP variants, should not get impacted by RTTs, right? Is that a fair
> > statement to make?
>
> CUBIC, like most TCP congestion control variants, has significant
> aspects that evolve on the scale of RTTs: for example, in slow start,
> loss recovery, or when the CUBIC function is very steep, CUBIC's
> behavior is heavily dependent on RTT (send some packets, wait for an
> ACK, send some more packets,...). So CUBIC will not be
> perfectly"RTT-fair.
>
> I think it might be fair to say something like: once a path is
> saturated with CUBIC flows, as long as loss recovery or send/receive
> buffers are not a bottleneck, then competition between CUBIC flows
> should be more RTT-fair than Reno, in the long run, since for large
> parts of the life cycle of the CUBIC flow the evolution of cwnd is
> constrained by wall clock time rather than ACK arrival.
>
> neal
>
>
> > Toms
> >
> > On 20 May 2015 at 19:51, Rick Jones <perfgeek@mac.com> wrote:
> >>
> >>
> >> On May 20, 2015, at 6:20 AM, Tom Sanders <toms.sanders@gmail.com>
> wrote:
> >>
> >> > Hi,
> >> >
> >> > As per my understanding, unlike other TCP variants, CUBIC claims to =
be
> >> > RTT fair. This is because the window growth rate is independent of
> the RTT,
> >> > and hence the performance is the same as on a low and a high RTT
> paths.
> >> > Others are dependent upon RTT since their windows grow as they
> receive ACKs
> >> > from their peers, while CUBIC doesnt (window growth is a function of
> time).
> >> > This is theory.
> >> >
> >> > In practice, CUBIC (which is the default now on most linux
> >> > distributions) also depends upon RTT. I connected to machines on a
> LAN and
> >> > did a FTP transfer. I got a certain bandwidth. Next, i artificially
> inserted
> >> > delay using "tc qdisc" in the path. I increased the delay on the
> ethernet
> >> > interface connecting my linux machine to the LAN to be around 100ms.=
 I
> >> > immediately saw that the TCP performance went down.
> >> >
> >> > To bring it up to the same level as before i had to use multiple TCP
> >> > streams. To me this means that CUBIC performance is dependent of the
> RTT. So
> >> > why do we call it RTT-fair?
> >>
> >> I cannot speak to CUBIC specifically, but I would think you would want
> to
> >> make sure you were seeing effects of congestion window and not of the
> >> classic TCP window.  For example, if the FTP client/server you were
> using
> >> makes an explicit setsockopt() to set the socket buffer sizes and by
> >> extension the classic TCP window, your bumping of the RTT to around
> 100ms
> >> may have taken the bandwidth down as a consequence of the classic:
> >>
> >> Throughput <=3D WindowSize/RTT
> >>
> >> Similarly, since you mention Linux, if the FTP client/server didn=E2=
=80=99t make
> >> explicit setsockopt() calls,  the sysctl values for tcp_rmem and
> tcp_wmem
> >> may have been such that the auto tuning of the window size couldn=E2=
=80=99t
> allow
> >> the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D
> >>
> >> I suspect that the experiment needs to be setup to have the same class=
ic
> >> window size in both cases, sized per the above to be sufficient to
> achieve
> >> the peak throughput in the high RTT case.  Then you will have eliminat=
ed
> >> classic window as a factor.
> >>
> >> Also, depending on the version of the Linux kernel you are using, ther=
e
> >> may be some effects from tcp small queues or perhaps (a stretch?) even
> byte
> >> queue limits which may not be playing well with netem.  (Just guessing
> there
> >> really)
> >>
> >> happy benchmarking,
> >>
> >> rick jones
> >
> >
> >
> >
> > --
> > Toms.
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> >
>



--=20
Toms.

--047d7bea453c59b49a05168cf2fc
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<div dir=3D"ltr">Hi Neal,<div><br></div><div>This is odd.</div><div><br></d=
iv><div>In the original Cubic paper (<a href=3D"http://www4.ncsu.edu/~rhee/=
export/bitcp/cubic-paper.pdf">http://www4.ncsu.edu/~rhee/export/bitcp/cubic=
-paper.pdf</a>) by Rhee and Xu, they explicitly state that the Cubic is ind=
ependent of RTT.</div><div><br></div><div>&quot;The main feature of
CUBIC is that its window growth function is defined in
real-time so that its growth will be independent of RTT. Our
work was partially inspired by HTCP [5], whose window
growth function is also based on real time. The congestion
epoch period of CUBIC is determined by the packet loss rate
alone. As TCP=E2=80=99s throughput is defined by the packet loss rate as
well as RTT, the throughput of CUBIC is defined by only the
packet loss rate. Thus, when the loss rate is high and/or RTT is
short, CUBIC can operate in a TCP mode. Moreover, since the
growth function is independent of RTT, its RTT fairness is
guaranteed as different RTT flows will still grow their windows
at the same rate.&quot;</div><div><br></div><div>Am i missing something her=
e?</div><div><br></div><div>Thanks, Toms</div></div><div class=3D"gmail_ext=
ra"><br><div class=3D"gmail_quote">On 20 May 2015 at 23:13, Neal Cardwell <=
span dir=3D"ltr">&lt;<a href=3D"mailto:ncardwell@google.com" target=3D"_bla=
nk">ncardwell@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><span class=3D"">On Wed, May 20, 2015 at 1:18 PM, Tom Sanders &lt;<a =
href=3D"mailto:toms.sanders@gmail.com">toms.sanders@gmail.com</a>&gt; wrote=
:<br>
&gt; Thanks Rick, this was very helpful.<br>
&gt;<br>
&gt; So if there are no setsockopts, etc then CUBIC&#39;s performance, unli=
ke other<br>
&gt; TCP variants, should not get impacted by RTTs, right? Is that a fair<b=
r>
&gt; statement to make?<br>
<br>
</span>CUBIC, like most TCP congestion control variants, has significant<br=
>
aspects that evolve on the scale of RTTs: for example, in slow start,<br>
loss recovery, or when the CUBIC function is very steep, CUBIC&#39;s<br>
behavior is heavily dependent on RTT (send some packets, wait for an<br>
ACK, send some more packets,...). So CUBIC will not be<br>
perfectly&quot;RTT-fair.<br>
<br>
I think it might be fair to say something like: once a path is<br>
saturated with CUBIC flows, as long as loss recovery or send/receive<br>
buffers are not a bottleneck, then competition between CUBIC flows<br>
should be more RTT-fair than Reno, in the long run, since for large<br>
parts of the life cycle of the CUBIC flow the evolution of cwnd is<br>
constrained by wall clock time rather than ACK arrival.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
neal<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
&gt; Toms<br>
&gt;<br>
&gt; On 20 May 2015 at 19:51, Rick Jones &lt;<a href=3D"mailto:perfgeek@mac=
.com">perfgeek@mac.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On May 20, 2015, at 6:20 AM, Tom Sanders &lt;<a href=3D"mailto:tom=
s.sanders@gmail.com">toms.sanders@gmail.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; &gt; Hi,<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; As per my understanding, unlike other TCP variants, CUBIC cla=
ims to be<br>
&gt;&gt; &gt; RTT fair. This is because the window growth rate is independe=
nt of the RTT,<br>
&gt;&gt; &gt; and hence the performance is the same as on a low and a high =
RTT paths.<br>
&gt;&gt; &gt; Others are dependent upon RTT since their windows grow as the=
y receive ACKs<br>
&gt;&gt; &gt; from their peers, while CUBIC doesnt (window growth is a func=
tion of time).<br>
&gt;&gt; &gt; This is theory.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; In practice, CUBIC (which is the default now on most linux<br=
>
&gt;&gt; &gt; distributions) also depends upon RTT. I connected to machines=
 on a LAN and<br>
&gt;&gt; &gt; did a FTP transfer. I got a certain bandwidth. Next, i artifi=
cially inserted<br>
&gt;&gt; &gt; delay using &quot;tc qdisc&quot; in the path. I increased the=
 delay on the ethernet<br>
&gt;&gt; &gt; interface connecting my linux machine to the LAN to be around=
 100ms. I<br>
&gt;&gt; &gt; immediately saw that the TCP performance went down.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; To bring it up to the same level as before i had to use multi=
ple TCP<br>
&gt;&gt; &gt; streams. To me this means that CUBIC performance is dependent=
 of the RTT. So<br>
&gt;&gt; &gt; why do we call it RTT-fair?<br>
&gt;&gt;<br>
&gt;&gt; I cannot speak to CUBIC specifically, but I would think you would =
want to<br>
&gt;&gt; make sure you were seeing effects of congestion window and not of =
the<br>
&gt;&gt; classic TCP window.=C2=A0 For example, if the FTP client/server yo=
u were using<br>
&gt;&gt; makes an explicit setsockopt() to set the socket buffer sizes and =
by<br>
&gt;&gt; extension the classic TCP window, your bumping of the RTT to aroun=
d 100ms<br>
&gt;&gt; may have taken the bandwidth down as a consequence of the classic:=
<br>
&gt;&gt;<br>
&gt;&gt; Throughput &lt;=3D WindowSize/RTT<br>
&gt;&gt;<br>
&gt;&gt; Similarly, since you mention Linux, if the FTP client/server didn=
=E2=80=99t make<br>
&gt;&gt; explicit setsockopt() calls,=C2=A0 the sysctl values for tcp_rmem =
and tcp_wmem<br>
&gt;&gt; may have been such that the auto tuning of the window size couldn=
=E2=80=99t allow<br>
&gt;&gt; the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D<br>
&gt;&gt;<br>
&gt;&gt; I suspect that the experiment needs to be setup to have the same c=
lassic<br>
&gt;&gt; window size in both cases, sized per the above to be sufficient to=
 achieve<br>
&gt;&gt; the peak throughput in the high RTT case.=C2=A0 Then you will have=
 eliminated<br>
&gt;&gt; classic window as a factor.<br>
&gt;&gt;<br>
&gt;&gt; Also, depending on the version of the Linux kernel you are using, =
there<br>
&gt;&gt; may be some effects from tcp small queues or perhaps (a stretch?) =
even byte<br>
&gt;&gt; queue limits which may not be playing well with netem.=C2=A0 (Just=
 guessing there<br>
&gt;&gt; really)<br>
&gt;&gt;<br>
&gt;&gt; happy benchmarking,<br>
&gt;&gt;<br>
&gt;&gt; rick jones<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Toms.<br>
&gt;<br>
</div></div><div class=3D"HOEnZb"><div class=3D"h5">&gt; __________________=
_____________________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
&gt;<br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div class=3D"gmail_signature">Toms.</div>
</div>

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Subject: Re: [tcpm] draft-ietf-tcpm-rtorestart-07
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Hi Nicolas,

Thanks a lot for your comments!
See inline for how we suggest to address them.

On 2015-05-07 16:59, Nicolas Kuhn wrote:
> Hi all,
>
> I had a look at this draft-ietf-tcpm-rtorestart-07 draft.
> I have some minor comments on the document.
>
> #############################################################
>
> - "The RTO Restart (RTOR) mechanism described in this document makes the
>   RTO slightly more aggressive when the number of outstanding segments
>   is too small for fast retransmit to work, in an attempt to enable
>   faster loss recovery for all segments while being robust to
>   reordering."
>
> The RTO is not aggressive ; but the retransmission scheme ruled by RTO expirations is.
> I would recommend to rephrase that.

We will change the wording to clarify that the effective RTO becomes 
more aggressive, not the actual RTO calculation. The retransmission 
scheme does not change.

We will apply this clarification also in other parts of the document.

>
> #############################################################
>
> - â   The RTO Restart (RTOR) mechanism described in this document makes the
>   RTO slightly [âĤ] highly varying RTTs, e.g. mobile networks."
>
> This whole paragraph comes to early in the document, as the reader has no clear
> idea about what RTOR is actually doing. I propose to add that in a dedicated sub-
> section in Section 5.
>
> #############################################################

The aim of the paragraph is to introduce both possible gains and 
drawbacks with the scheme early in the document, and then address these 
in more detail later, but we agree that it contains too much detail at 
this point in the document.

As part of the material in this paragraph has been inserted as a result 
of earlier reviews our suggested resolution is to not completely remove 
the paragraph, but to compress it and remove details that are hard to 
understand without knowing the details of the algorithm.

>
> - â 3 RTO Restart Overviewâ -> should it not be â3- RTO Overview and rationale of RTOR â or something equivalent ?
> It is confusing, as you do not actually present RTOR.

Yes, your proposed title is better and we will change the title accordingly.

>
> Also, in this section 3:
> The first paragraph finishes with âHence,  in most situations, the time before a retransmission is triggered is equal to "RTO + RTTâ.â
> But the example to illustrate it comes after - I think things should moved around to:
> 1- introduce the classic RTO
> 2- explain the rationale of classic RTO
> 3- example to show that RTO+RTT is common
> 4- more parameters could affect that (2 outstanding segments; other factors)
> 5- conclusion on âhence, RTO+RTT is common for application limited trafficâ
>
 > I would propose to introduce a ToC for this section 3:
>
> 3- RTO Overview and rationale of RTOR
>  3-1. Experienced RTO is RTO+RTT
>  3-2. RTO safety margin
>  3-3. Rationale of RTOR: when fast retransmit is not possible
>

We will try to rearrange the material in line with your proposed 
outline. However as the section is not so long, we think that explicit 
sections may not be needed.

> #############################################################
>
> Section 4 : should focus on the description of the algorithm only.
>
> I would keep
> â
> This approach makes the RTO more
>   aggressive than the standardized approach in [RFC6298] but still
>   conforms to the requirement in [RFC6298] that segments must not be
>   retransmitted earlier than RTO seconds after their original
>   transmission.
> "
>
> for the discussion section (at this stage of the reading, the reader still does not know exactly what RTOR is actually doing).
>

Agreed, we will move the text.

> #############################################################
>
> Section 4: "it will prevent RTOR
>   from being more aggressive and potentially causing RTOs instead of
>   fast retransmits.â
>
> More aggressive than what ? Has the "four" being experimentally obtained ?
> I think more discussion is required on this parameter.
>

This value was chosen to only enable RTOR when the use of fast 
retransmit for loss recovery isnât possible. We will try and clarify 
this further in the text.

Unless you (or anyone else) have further comments on the proposed 
changes, we will submit a new version with the updates discussed above 
next week.

Thanks again for your comments.

Anna

> Regards,
>
> Nicolas
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
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From nobody Fri May 22 02:40:55 2015
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From: =?utf-8?Q?Mirja_K=C3=BChlewind?= <mirja.kuehlewind@tik.ee.ethz.ch>
In-Reply-To: <655C07320163294895BBADA28372AF5D483AF007@FR712WXCHMBA15.zeu.alcatel-lucent.com>
Date: Fri, 22 May 2015 11:40:50 +0200
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To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
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Subject: Re: [tcpm] Feedback on WG acceptance of draft-bensley-tcpm-dctcp-03
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Hi,

sorry I=E2=80=99ve overlooked the initial email on this. Similar as =
Gorry and Richard, I think this document should be published as =
information (as the document currently states its indented status) and =
there is the expertise in tcpm to review this document. Therefore I'm in =
favor of adopting this as a woking document. As think as this document =
=E2=80=9Aonly=E2=80=98 documents existing and deployed technology, there =
is only some reviews needed to ensure that the document is well readable =
and complete. Therefore there should be basically no technical =
discussion needed and I believe that we could move forward quickly =
within tcpm.

If the document will be adopted in tcp, I=E2=80=99d also volunteer to =
review. Btw., as a comment to the authors, in the last version I=E2=80=99v=
e read, loss handling was not discussed. I think this should be added; =
please see the linux kernel code for this. And that's actually another =
point. I believe this document mainly documents what is implemented in =
Windows. As there is an implementation as part of the Linux kernel in =
the mean time, maybe the author should check this implementation as well =
or make clear what exactly is document in this draft.

Mirja

=20
> Am 18.05.2015 um 11:12 schrieb Scharf, Michael (Michael) =
<michael.scharf@alcatel-lucent.com>:
>=20
> Hi all,
>=20
> There have not been many responses to this e-mail. The chairs =
therefore conclude that there is not enough support for adoption of =
draft-bensley-tcpm-dctcp-03 as TCPM WG item as of now.
>=20
> Since the TCPM community seems interested in further discussing this =
topic, the chairs would offer to allocate time for corresponding =
discussions during upcoming TCPM meetings.
>=20
> Thanks
>=20
> Michael, Pasi, Yoshifumi
>=20
>=20
> PS: We recently found one possibly interesting study on operational =
experience: =
https://www.usenix.org/system/files/conference/nsdi15/nsdi15-paper-judd.pd=
f
>=20
>=20
>> -----Original Message-----
>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Scharf, =
Michael
>> (Michael)
>> Sent: Wednesday, April 22, 2015 10:54 AM
>> To: tcpm@ietf.org
>> Subject: [tcpm] Feedback on WG acceptance of =
draft-bensley-tcpm-dctcp-
>> 03
>>=20
>> Dear all,
>>=20
>> During IETF 89 [1] and on the list there has been a discussion about =
WG
>> adoption of draft-bensley-tcpm-dctcp [2]. The understanding of the
>> chairs is that there is no clear consensus up to now.
>>=20
>> In order to move forward, the chairs seek for community guidance
>> regarding the following options to handle this draft:
>>=20
>> a) Adopt in TCPM as Informational
>>=20
>> b) Adopt in TCPM as Experimental
>>=20
>> c) Do not adopt in TCPM now
>>=20
>> Option c) does not prevent publication outside TCPM, such as
>> publication by another working group (e.g., TSVWG), publication on
>> Independent Stream, etc.
>>=20
>> The TCPM chairs believe that adoption of alternative congestion =
control
>> algorithms in TCPM should be backed by a strong consensus. Please =
also
>> review [3] and [4] regarding the difference between Experimental and
>> Informational.
>>=20
>> Please let us know any feedback on a potential WG acceptance of =
draft-
>> bensley-tcpm-dctcp-03 until May 8.
>>=20
>> Thanks!
>>=20
>> Michael, Pasi, Yoshifumi
>>=20
>>=20
>> [1] http://www.ietf.org/proceedings/89/minutes/minutes-89-tcpm
>>=20
>> [2] https://tools.ietf.org/html/draft-bensley-tcpm-dctcp-03
>>=20
>> [3] RFC 2026, Section 4.2.1, 4.2.2
>>=20
>> [4] https://www.ietf.org/iesg/informational-vs-experimental.html
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri May 22 03:00:38 2015
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From: =?utf-8?Q?Mirja_K=C3=BChlewind?= <mirja.kuehlewind@tik.ee.ethz.ch>
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To: Tom Sanders <toms.sanders@gmail.com>
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Subject: Re: [tcpm] Why is Cubic said to be RTT-fair?
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Hi Tom,

the text below only says that if you have two cubic flows with a =
different base RTT competing at the same bottleneck they=E2=80=99ll =
still converge to share the link equally. This does not mean that if you =
change the RTT strongly during one connection that Cubic is able to =
catch up fast.=20

In fact this is one weakness of Cubic that I have observed. If there are =
change in base delay/rtt or strong increases in the available capacity, =
Cubic needs a rather long time to catch up. This is because cubic has a =
=E2=80=9Atarget value=E2=80=98 to which it increases it sending =
rate/window quickly but then if the capacity cap is not somewhere close =
to this value (as expected) it will probe very conservatively around =
this value and it takes a rather long time until it gets back into a =
phase where it increases its congestion window fast.

So what do you mean by 'I immediately saw that the TCP performance went =
down=E2=80=98. How long do you run your test? And over which period of =
time this the performance go down?

Mirja


> Am 21.05.2015 um 02:51 schrieb Tom Sanders <toms.sanders@gmail.com>:
>=20
> Hi Neal,
>=20
> This is odd.
>=20
> In the original Cubic paper =
(http://www4.ncsu.edu/~rhee/export/bitcp/cubic-paper.pdf) by Rhee and =
Xu, they explicitly state that the Cubic is independent of RTT.
>=20
> "The main feature of CUBIC is that its window growth function is =
defined in real-time so that its growth will be independent of RTT. Our =
work was partially inspired by HTCP [5], whose window growth function is =
also based on real time. The congestion epoch period of CUBIC is =
determined by the packet loss rate alone. As TCP=E2=80=99s throughput is =
defined by the packet loss rate as well as RTT, the throughput of CUBIC =
is defined by only the packet loss rate. Thus, when the loss rate is =
high and/or RTT is short, CUBIC can operate in a TCP mode. Moreover, =
since the growth function is independent of RTT, its RTT fairness is =
guaranteed as different RTT flows will still grow their windows at the =
same rate."
>=20
> Am i missing something here?
>=20
> Thanks, Toms
>=20
> On 20 May 2015 at 23:13, Neal Cardwell <ncardwell@google.com> wrote:
> On Wed, May 20, 2015 at 1:18 PM, Tom Sanders <toms.sanders@gmail.com> =
wrote:
> > Thanks Rick, this was very helpful.
> >
> > So if there are no setsockopts, etc then CUBIC's performance, unlike =
other
> > TCP variants, should not get impacted by RTTs, right? Is that a fair
> > statement to make?
>=20
> CUBIC, like most TCP congestion control variants, has significant
> aspects that evolve on the scale of RTTs: for example, in slow start,
> loss recovery, or when the CUBIC function is very steep, CUBIC's
> behavior is heavily dependent on RTT (send some packets, wait for an
> ACK, send some more packets,...). So CUBIC will not be
> perfectly"RTT-fair.
>=20
> I think it might be fair to say something like: once a path is
> saturated with CUBIC flows, as long as loss recovery or send/receive
> buffers are not a bottleneck, then competition between CUBIC flows
> should be more RTT-fair than Reno, in the long run, since for large
> parts of the life cycle of the CUBIC flow the evolution of cwnd is
> constrained by wall clock time rather than ACK arrival.
>=20
> neal
>=20
>=20
> > Toms
> >
> > On 20 May 2015 at 19:51, Rick Jones <perfgeek@mac.com> wrote:
> >>
> >>
> >> On May 20, 2015, at 6:20 AM, Tom Sanders <toms.sanders@gmail.com> =
wrote:
> >>
> >> > Hi,
> >> >
> >> > As per my understanding, unlike other TCP variants, CUBIC claims =
to be
> >> > RTT fair. This is because the window growth rate is independent =
of the RTT,
> >> > and hence the performance is the same as on a low and a high RTT =
paths.
> >> > Others are dependent upon RTT since their windows grow as they =
receive ACKs
> >> > from their peers, while CUBIC doesnt (window growth is a function =
of time).
> >> > This is theory.
> >> >
> >> > In practice, CUBIC (which is the default now on most linux
> >> > distributions) also depends upon RTT. I connected to machines on =
a LAN and
> >> > did a FTP transfer. I got a certain bandwidth. Next, i =
artificially inserted
> >> > delay using "tc qdisc" in the path. I increased the delay on the =
ethernet
> >> > interface connecting my linux machine to the LAN to be around =
100ms. I
> >> > immediately saw that the TCP performance went down.
> >> >
> >> > To bring it up to the same level as before i had to use multiple =
TCP
> >> > streams. To me this means that CUBIC performance is dependent of =
the RTT. So
> >> > why do we call it RTT-fair?
> >>
> >> I cannot speak to CUBIC specifically, but I would think you would =
want to
> >> make sure you were seeing effects of congestion window and not of =
the
> >> classic TCP window.  For example, if the FTP client/server you were =
using
> >> makes an explicit setsockopt() to set the socket buffer sizes and =
by
> >> extension the classic TCP window, your bumping of the RTT to around =
100ms
> >> may have taken the bandwidth down as a consequence of the classic:
> >>
> >> Throughput <=3D WindowSize/RTT
> >>
> >> Similarly, since you mention Linux, if the FTP client/server =
didn=E2=80=99t make
> >> explicit setsockopt() calls,  the sysctl values for tcp_rmem and =
tcp_wmem
> >> may have been such that the auto tuning of the window size =
couldn=E2=80=99t allow
> >> the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D
> >>
> >> I suspect that the experiment needs to be setup to have the same =
classic
> >> window size in both cases, sized per the above to be sufficient to =
achieve
> >> the peak throughput in the high RTT case.  Then you will have =
eliminated
> >> classic window as a factor.
> >>
> >> Also, depending on the version of the Linux kernel you are using, =
there
> >> may be some effects from tcp small queues or perhaps (a stretch?) =
even byte
> >> queue limits which may not be playing well with netem.  (Just =
guessing there
> >> really)
> >>
> >> happy benchmarking,
> >>
> >> rick jones
> >
> >
> >
> >
> > --
> > Toms.
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> >
>=20
>=20
>=20
> --=20
> Toms.
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri May 22 10:33:41 2015
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-newcwv-11.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 TCP Maintenance and Minor Extensions Working Group of the IETF.

        Title           : Updating TCP to support Rate-Limited Traffic
        Authors         : Godred Fairhurst
                          Arjuna Sathiaseelan
                          Raffaello Secchi
	Filename        : draft-ietf-tcpm-newcwv-11.txt
	Pages           : 25
	Date            : 2015-05-22

Abstract:
   This document provides a mechanism to address issues that arise when
   TCP is used to support traffic that exhibits periods where the
   sending rate is limited by the application rather than the congestion
   window.  It provides an experimental update to TCP that allows a TCP
   sender to restart quickly following a rate-limited interval.  This
   method is expected to benefit applications that send rate-limited
   traffic using TCP, while also providing an appropriate response if
   congestion is experienced.

   It also evaluates the Experimental specification of TCP Congestion
   Window Validation, CWV, defined in RFC 2861, and concludes that RFC
   2861 sought to address important issues, but failed to deliver a
   widely used solution.  This document therefore recommends that the
   status of RFC 2861 is moved from Experimental to Historic, and that
   it is replaced by the current specification.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tcpm-newcwv/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tcpm-newcwv-11

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-newcwv-11


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 Fri May 22 10:38:39 2015
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The document editors have just uploaded a new revision of the CWV draft,
intended to address this issues raised in the IETF LC.  Specific comments
are listed below,

Best wishes,

Gorry & Raffaello

---


Mirja (TCPM)

pipeACK (section 4.1 and 4.2 and 4.5.1):
1) Can you maybe add a sentence that explains why FlightSize is not
sufficient.
I understand the difference between the two variables but are there actually
cases where it makes a big difference if one or the other is used?
- Added to section 1.

2) This is just an idea and does not make a big difference: Instead of saying
pipeACK is undefined, you could just set it to infinity or 'max' (what
every the
max in the implementation is) and this would always keep you in validated
phase.
- Not changed, this is an implementation choice, but setting to infinity
requires care when considering other uses of the pipeACK, e.g., when
calculating  cwnd = (Max(pipeACK,LossFlightSize))/2.

3) The implementation for pipeACK (based on HighACK) that you propose does
not
take into account that dupACK also acknowledge new data. Is this on
purpose? If
so please state explicitly.
- A sender MAY count TCP DupACKs that acknowledge new data when collecting
the pipeACK sample. Other equivalent methods may be used.

4) Text says "When no measurements are available, ..."
    Maybe say more explicitly "When no measurements are available as the
connection is idle" or "... as no data were sent during the last measurement
period" or "... no ACK were received...".
-  (e.g., a sender that has been idle will have sent no data and received
no ACKs)

5) I would appreciate a forward reference to section 4.5.1 here (as also done
for other stuff as for cwnd-limited in the next section).
- Done

6) Could you also include pseudo code in 4.5.1? I far as I know you have an
implementation, so it should not be extremely hard to come up with pseudo
code
and I personally think it would be really helpful...
We dont plan to add this to 4.1, burt there is code available for Linux
and BSD.

7) nit in last paragraph: s/method defined on [RFC6675]is/ method defined in
[RFC6675] is/
- Done

8) Figure 1 (section 4.5.1): Maybe add a y-axis...?
- pipeACK sample (Bytes)

non-validated phase (section 4.3 and 4.4)

1) Maybe add one sentence saying "A connection is also in non-validated
phase if
pipeACK is zero which means the connection is idle". Saying this explicit
would
help me to understand things better.
- Idle isnt quite correct, what it means is no new data was ACKed -
there could be data sent and pending ACK/ We changed 4.5.2 to clarify
this.

I just notice that this might actually not be true because in the section
above
you say that pipeACK is set to undefined if no measurements are available and
that would mean the connection goes into validated phase... okay I think
there
is some clarification needed here.
- see above


2) In section 4.4 you first say
    "cwnd neither grows nor reduces while the sender remains in this phase"
     and then
     "A sender that is cwnd-limited MAY use the standard TCP method to
increase
cwnd".
     Please clarify!
- A TCP sender that enters the non-validated phase preserves the cwnd
(i.e., the cwnd only increases after a sender fully uses the cwnd in this
phase, otherwise the cwnd neither grows nor reduces).

3) In section 4.4.1:
    "A sender that detects a packet-drop MUST record the current
    FlightSize in the variable LossFlightSize ..."
    Is this "packet-drop" or "congestion notification"? I guess the
difference
is that if ECN is used R=0. So that's okay. Please check again that this also
works correctly for ECN.
- Agree, we removed explicit discussion of ECN - We agree R=0 when ECN is
used and there is no loss, but we did not wish to further interpret the
correct response for ECN.

4) Is this
    cwnd = (Max(pipeACK,LossFlightSize) - R)/2
- This is correct and more conservative.
    or
    cwnd = (Max(pipeACK,(LossFlightSize - R))/2  [only LossFlightSize - R] ?
    Because the text reads as if it should be the second...
- Text has been fixed: The loss of segments indicates that the path
capacity was exceeded by at least R, and hence the calculated cwnd is
reduced by at least R before the window is halved.

5) "ssthresh is adjusted using the standard TCP method."
    This is not clear to me. Is ssthresh set to cwnd/2 before the
adjustment or
to cwnd-1 after the adjustment? The first option would restart the
connection in
Slow Start... Please clarify!
- We use the standard method, with no change.

6) "then this can result in the final cwnd after loss recovery being 1/4
of the
cwnd"
    Shouldn't this be "being at max 1/4 of the cwnd"...?
-  at most one quarter of the cwnd

7) "Subsequent updates to cwnd do not therefore reflect pipeACK history
before any
    congestion event."
    Not sure I understand this sentence... which subsequent updates?
- This reduction is conservative, and pipeACK is then reset to undefined,
hence cwnd updates after a congestion event do not depend upon the pipeACK
history before congestion was detected.

8) nit: s/updated during loss recoverySection 4.2/ updated during loss
recovery
as stated in Section 4.2/
- Done

Non-validated phase (section 4.4.3 and 5)
1) I would recommend to change the title to
"4.4.3 Adjustment at the end of the Non-Validated Phase (NVP)"
- Done

2) "An application that remains in the non-validated phase for a period
    greater than the NVP is required to adjust its congestion control
    state.  If the sender exits the non-validated phase after this
    period, it MUST update the ssthresh:"

    This is not fully clear to me. Do I have to adapt cwnd and ssthresh right
after NVP or as soon as I leave the non-validated phase after being for at
least
NVP time in this phase? If I'm not idle, but only sending with a low rate, I
should probably do this adjustment with the next ACK, no...?
- Not sure of the difference, e hope the new intro helps with this.

3) Further, these adjustments mean that the connection will be in Slow
Start. If
I understand this correctly, this is not a problem as long as the connection
stays in non-validated as the cwnd will not be further increased. But as
soon as
it leaves non-validated phase it will increase its cwnd as in Slow Start. I
would like to have this spelled out explicitly.
- ???

4) This section does not tell/specify what value NVP period should have.
Please add a
sentence that this is on purpose and add a reference to section 5.
- We see this problem= as a definition in the info section and added text
to 4.4.

5) Section 5 says "A period of five minutes was chosen for this NVP."
    This seems to be a left over from an older version as nothing was
specified
so far. Please check!
- Fixed

-----------------------------------------------------------------------------------------

Martin (AD)

The comment:
- Section 5:
"   Routing algorithms may modify the network path, disrupting the RTT
    measurement and changing the capacity available to a TCP connection,
    however such changes do not usually occur within a time frame of a
    few minutes."

That is an odd assumption, as a path change can happen at any time,
though it is not very usually under normal conditions. Did actually mean
to say that it is uncommon to happen?
- Routing algorithms may change the the network path that is used by a
transport. Although a change of path can in turn disrupt the RTT
measurement and may result in a change of the capacity available to a TCP
connection, we assume these path changes do not usually occur frequently
(compared to a  time frame of a few minutes).


The two nits:

- Section 4.4.1
"   The pipeACK value is not updated during loss recoverySection 4.2. If
    there is a valid pipeACK value, the new cwnd is adjusted to reflect
    that a non-validated cwnd may be larger than the actual FlightSize,
    or recently used FlightSize (recorded in pipeACK).  The updated cwnd
    therefore prevents overshoot by a sender significantly increasing its
    transmission rate during the recovery period."

Typo in first sentence "recoverySection 4.2"
- Done

- Section 4.5.2.

"   A sender needs to implement a method that detects the cwnd-limited
    condition (see Section 4.4.  This detects a condition where a sender"

closing brackets after 4.4. missing
- Done

-----------------------------------------------------------------------------------------

Sec

The security considerations defer to those of RFC 5681 with the claim that
this document just describes an algorithm for one aspect of the congestion
control procedures and thus that the generic security considerations
apply; this claim seems true.  The security considerations section of RFC
5681 feels a little bit short, but it does seem to cover the relevant
risks, namely that an attacker could make the sender send more slowly or
more quickly, with the latter possibly driving the network into congestion
collapse.  These are issues that protocol designers and implementors must
be aware of (and avoiding congestion collapse is more important).

The authors seem to have taken sufficient care to avoid this algorithm
contributing to congestion collapse, and the concern about an attacker
being able to slow down a sender is not really feasible to mitigate at
this level, so the security considerations of RFC 5681, short as they are,
seem adequate here.  One would hope that implementors know that failing to
follow the specification could lead to congestion collapse, so an explicit
warning is not needed.

The nits are basically all grammatical; I'll include them below and expect
people other than the document authors/shepherd to stop reading here.

-Ben

My apologies in advance if I am applying too much of an American English
perspective to this document written in British English.


In the abstract, "TCP is used to support traffic that [...]", is "support"
the best verb to use (as opposed to "transmit", "carry", "transport",
etc.)?
- Done

The first paragraph of the Introduction might have a little more text to
clarify how the cwnd differs from the FlightSize, for non-experts such as
myself.  (The cwnd is how much the sender is permitted to have
outstanding/unack'd and the FlightSize is how much the sender actually has
outstanding/unack'd?)  Just another sentence after the second one, "The
FlightSize is how many unacknowledged packets/bytes that are currently
outstanding, and the cwnd is the maximum value the sender will allow
FlightSize to take on" would be plenty.
- Added Explanation, and reworked this text.

Introduction, third paragraph, """[CWV] introduced the terminology of
"application limited periods".  This document describes [...]"""  Usually,
in an RFC, "this document" refers to the RFC itself, but in this case,
"this document" seems to be being used to refer to the other document, RFC
2861.  So, the references and text could be made more clear.
- Updated to refer to RFC2861.

A couple sentences later, "or by changing the rate the application sends",
maybe put an "at which" in there somewhere?  Absent context, I would
expect "the rate the application sends" to mean that an application
protocol was conveying a number which is interpreted as a rate in some
fashion.
- Changed

Still in the Introduction, in the summary of the document, paragraph for
section 4, maybe s/does this in a way/does so in a way/ ?  Also, in the
parenthetical at the end of that sentence, I think it reads better as
"including both application-limited and idle senders".
- Changed

Still in the introduction, in the "goals of this update" list, first item,
do bulk transfers consume the cwnd or fill it?
- changed text

I'm not sure that the fifth item is fully grammatical.  Maybe add in an
"to send extra traffic" somewhere?
- changed text

In the sixth item, there's a singular/plural mismatch between flows and
network (unless it's supposed to be "the network").
- changed to benefiting both the flows and other flows sharing the network
path

Section 2, first paragraph, "the behaviour where a sender transmitted at a
rate less than allowed by cwnd", is "where" or "when" better?
- when

Section 2, second paragraph, third sentence: I think it should start
"While this" instead of just "This".  The first comma in this sentence
could also be removed, if desired.
- While this

In section 3 (Terminology), it might be nice to clarify that the listed
terms are new to this document, or in addition to the RFC 5681 terms, or
something like that.  (I.e., "Additionally, the following terminology
...".)
- additional

Section 4.2, last paragraph, a space is missing after the [RFC6675]
citation.
- Done

In section 4.4: if I understand correctly, the sliding window for pipeAck
calculation is smaller than the NVP, so it is not the absolute amount of
data which the sender transmits which can push pipeAck over (1/2)*cwnd,
but rather the rate at which the data is transmitted.
- pipeACK is a measure per RTT (like cwnd)

Also in section 4.4, there is some superficial inconsistency between "a
sender that enters the non-validated phase SHOULD preserve the cwnd" and
the text telling the sender how to reduce cwnd if it encounters congestion
and the (outer) bullet point wherein "a sender determines whether to
increase the cwnd based on ...".  The last paragraph of the section helps
clarify the situation, but it might help readability to clarify the
exceptions to the SHOULD (and the rationale for them) right after the
claim is made.
- Reworked.


In section 4.4.1, there is a missing space after "recovery" and before the
internal reference to Section 4.2.
- Done

Section 4.5.1, comma after "e.g.".
- Done

-----------------------------------------------------------------------------------------

Gen-ART

Nits/editorial comments:

1.        I am not sure that there is a need for the first sentence in
section
1.2
- Moved to ACK section.

2.        In section 4.4.1 MSS is used but not expanded Maximum Segment  Size
(MSS)
- Done



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Subject: Re: [tcpm] Why is Cubic said to be RTT-fair?
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Hi Mirja,

>
>
> the text below only says that if you have two cubic flows with a differen=
t
> base RTT competing at the same bottleneck they=E2=80=99ll still converge =
to share
> the link equally. This does not mean that if you change the RTT strongly
> during one connection that Cubic is able to catch up fast.
>
> In fact this is one weakness of Cubic that I have observed. If there are
> change in base delay/rtt or strong increases in the available capacity,
> Cubic needs a rather long time to catch up. This is because cubic has a
> =E2=80=9Atarget value=E2=80=98 to which it increases it sending rate/wind=
ow quickly but
> then if the capacity cap is not somewhere close to this value (as expecte=
d)
> it will probe very conservatively around this value and it takes a rather
> long time until it gets back into a phase where it increases its congesti=
on
> window fast.
>

Then which congestion control algorithm fares better there? I thought Cubic
was suited for large fat pipes -- bigger bandwidth-delay pipes.


>
> So what do you mean by 'I immediately saw that the TCP performance went
> down=E2=80=98. How long do you run your test? And over which period of ti=
me this
> the performance go down?
>

I did a simple test.

I connect two machines on my 1GB LAN.I do a tcp data transfer test and i
get around ~900Mbps. I then added a delay using qdisc on my outgoing
ethernet interface to simulate a longer RTT. So, if my ping earlier was
around 0.255ms, it then became 120ms. I repeat the test and i see that i
can only send about 200Mbps. I had to use multiple TCP flows to fill up my
1GB link with Cubic. I run this experiment for about 2 minutes, and Cubic
is not able to catch up. I dont have long standing flows, and only need to
send bursts of data, hence did not leave it running for very long.

I was under the impression, after having read the original Cubic paper that
Cubic's window growth rate is independent of RTT and hence wasnt expecting
any difference in the throughput.

Toms.

>
> Mirja
>
>
> > Am 21.05.2015 um 02:51 schrieb Tom Sanders <toms.sanders@gmail.com>:
> >
> > Hi Neal,
> >
> > This is odd.
> >
> > In the original Cubic paper (
> http://www4.ncsu.edu/~rhee/export/bitcp/cubic-paper.pdf) by Rhee and Xu,
> they explicitly state that the Cubic is independent of RTT.
> >
> > "The main feature of CUBIC is that its window growth function is define=
d
> in real-time so that its growth will be independent of RTT. Our work was
> partially inspired by HTCP [5], whose window growth function is also base=
d
> on real time. The congestion epoch period of CUBIC is determined by the
> packet loss rate alone. As TCP=E2=80=99s throughput is defined by the pac=
ket loss
> rate as well as RTT, the throughput of CUBIC is defined by only the packe=
t
> loss rate. Thus, when the loss rate is high and/or RTT is short, CUBIC ca=
n
> operate in a TCP mode. Moreover, since the growth function is independent
> of RTT, its RTT fairness is guaranteed as different RTT flows will still
> grow their windows at the same rate."
> >
> > Am i missing something here?
> >
> > Thanks, Toms
> >
> > On 20 May 2015 at 23:13, Neal Cardwell <ncardwell@google.com> wrote:
> > On Wed, May 20, 2015 at 1:18 PM, Tom Sanders <toms.sanders@gmail.com>
> wrote:
> > > Thanks Rick, this was very helpful.
> > >
> > > So if there are no setsockopts, etc then CUBIC's performance, unlike
> other
> > > TCP variants, should not get impacted by RTTs, right? Is that a fair
> > > statement to make?
> >
> > CUBIC, like most TCP congestion control variants, has significant
> > aspects that evolve on the scale of RTTs: for example, in slow start,
> > loss recovery, or when the CUBIC function is very steep, CUBIC's
> > behavior is heavily dependent on RTT (send some packets, wait for an
> > ACK, send some more packets,...). So CUBIC will not be
> > perfectly"RTT-fair.
> >
> > I think it might be fair to say something like: once a path is
> > saturated with CUBIC flows, as long as loss recovery or send/receive
> > buffers are not a bottleneck, then competition between CUBIC flows
> > should be more RTT-fair than Reno, in the long run, since for large
> > parts of the life cycle of the CUBIC flow the evolution of cwnd is
> > constrained by wall clock time rather than ACK arrival.
> >
> > neal
> >
> >
> > > Toms
> > >
> > > On 20 May 2015 at 19:51, Rick Jones <perfgeek@mac.com> wrote:
> > >>
> > >>
> > >> On May 20, 2015, at 6:20 AM, Tom Sanders <toms.sanders@gmail.com>
> wrote:
> > >>
> > >> > Hi,
> > >> >
> > >> > As per my understanding, unlike other TCP variants, CUBIC claims t=
o
> be
> > >> > RTT fair. This is because the window growth rate is independent of
> the RTT,
> > >> > and hence the performance is the same as on a low and a high RTT
> paths.
> > >> > Others are dependent upon RTT since their windows grow as they
> receive ACKs
> > >> > from their peers, while CUBIC doesnt (window growth is a function
> of time).
> > >> > This is theory.
> > >> >
> > >> > In practice, CUBIC (which is the default now on most linux
> > >> > distributions) also depends upon RTT. I connected to machines on a
> LAN and
> > >> > did a FTP transfer. I got a certain bandwidth. Next, i artificiall=
y
> inserted
> > >> > delay using "tc qdisc" in the path. I increased the delay on the
> ethernet
> > >> > interface connecting my linux machine to the LAN to be around
> 100ms. I
> > >> > immediately saw that the TCP performance went down.
> > >> >
> > >> > To bring it up to the same level as before i had to use multiple T=
CP
> > >> > streams. To me this means that CUBIC performance is dependent of
> the RTT. So
> > >> > why do we call it RTT-fair?
> > >>
> > >> I cannot speak to CUBIC specifically, but I would think you would
> want to
> > >> make sure you were seeing effects of congestion window and not of th=
e
> > >> classic TCP window.  For example, if the FTP client/server you were
> using
> > >> makes an explicit setsockopt() to set the socket buffer sizes and by
> > >> extension the classic TCP window, your bumping of the RTT to around
> 100ms
> > >> may have taken the bandwidth down as a consequence of the classic:
> > >>
> > >> Throughput <=3D WindowSize/RTT
> > >>
> > >> Similarly, since you mention Linux, if the FTP client/server didn=E2=
=80=99t
> make
> > >> explicit setsockopt() calls,  the sysctl values for tcp_rmem and
> tcp_wmem
> > >> may have been such that the auto tuning of the window size couldn=E2=
=80=99t
> allow
> > >> the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D
> > >>
> > >> I suspect that the experiment needs to be setup to have the same
> classic
> > >> window size in both cases, sized per the above to be sufficient to
> achieve
> > >> the peak throughput in the high RTT case.  Then you will have
> eliminated
> > >> classic window as a factor.
> > >>
> > >> Also, depending on the version of the Linux kernel you are using,
> there
> > >> may be some effects from tcp small queues or perhaps (a stretch?)
> even byte
> > >> queue limits which may not be playing well with netem.  (Just
> guessing there
> > >> really)
> > >>
> > >> happy benchmarking,
> > >>
> > >> rick jones
> > >
> > >
> > >
> > >
> > > --
> > > Toms.
> > >
> > > _______________________________________________
> > > tcpm mailing list
> > > tcpm@ietf.org
> > > https://www.ietf.org/mailman/listinfo/tcpm
> > >
> >
> >
> >
> > --
> > Toms.
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
>
>


--=20
Toms.

--047d7b8738d0f9db890516b586a7
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Mirja,<div class=3D"gmail_extra"><div class=3D"gmail_qu=
ote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex"><br><br>
the text below only says that if you have two cubic flows with a different =
base RTT competing at the same bottleneck they=E2=80=99ll still converge to=
 share the link equally. This does not mean that if you change the RTT stro=
ngly during one connection that Cubic is able to catch up fast.<br>
<br>
In fact this is one weakness of Cubic that I have observed. If there are ch=
ange in base delay/rtt or strong increases in the available capacity, Cubic=
 needs a rather long time to catch up. This is because cubic has a =E2=80=
=9Atarget value=E2=80=98 to which it increases it sending rate/window quick=
ly but then if the capacity cap is not somewhere close to this value (as ex=
pected) it will probe very conservatively around this value and it takes a =
rather long time until it gets back into a phase where it increases its con=
gestion window fast.<br></blockquote><div><br></div><div>Then which congest=
ion control algorithm fares better there? I thought Cubic was suited for la=
rge fat pipes -- bigger bandwidth-delay pipes.=C2=A0</div><div>=C2=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
So what do you mean by &#39;I immediately saw that the TCP performance went=
 down=E2=80=98. How long do you run your test? And over which period of tim=
e this the performance go down?<br></blockquote><div><br></div><div>I did a=
 simple test.</div><div><br></div><div>I connect two machines on my 1GB LAN=
.I do a tcp data transfer test and i get around ~900Mbps. I then added a de=
lay using qdisc on my outgoing ethernet interface to simulate a longer RTT.=
 So, if my ping earlier was around 0.255ms, it then became 120ms. I repeat =
the test and i see that i can only send about 200Mbps. I had to use multipl=
e TCP flows to fill up my 1GB link with Cubic. I run this experiment for ab=
out 2 minutes, and Cubic is not able to catch up. I dont have long standing=
 flows, and only need to send bursts of data, hence did not leave it runnin=
g for very long.</div><div><br></div><div>I was under the impression, after=
 having read the original Cubic paper that Cubic&#39;s window growth rate i=
s independent of RTT and hence wasnt expecting any difference in the throug=
hput.</div><div><br></div><div>Toms.</div><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Mirja<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
&gt; Am 21.05.2015 um 02:51 schrieb Tom Sanders &lt;<a href=3D"mailto:toms.=
sanders@gmail.com">toms.sanders@gmail.com</a>&gt;:<br>
&gt;<br>
&gt; Hi Neal,<br>
&gt;<br>
&gt; This is odd.<br>
&gt;<br>
&gt; In the original Cubic paper (<a href=3D"http://www4.ncsu.edu/~rhee/exp=
ort/bitcp/cubic-paper.pdf" target=3D"_blank">http://www4.ncsu.edu/~rhee/exp=
ort/bitcp/cubic-paper.pdf</a>) by Rhee and Xu, they explicitly state that t=
he Cubic is independent of RTT.<br>
&gt;<br>
&gt; &quot;The main feature of CUBIC is that its window growth function is =
defined in real-time so that its growth will be independent of RTT. Our wor=
k was partially inspired by HTCP [5], whose window growth function is also =
based on real time. The congestion epoch period of CUBIC is determined by t=
he packet loss rate alone. As TCP=E2=80=99s throughput is defined by the pa=
cket loss rate as well as RTT, the throughput of CUBIC is defined by only t=
he packet loss rate. Thus, when the loss rate is high and/or RTT is short, =
CUBIC can operate in a TCP mode. Moreover, since the growth function is ind=
ependent of RTT, its RTT fairness is guaranteed as different RTT flows will=
 still grow their windows at the same rate.&quot;<br>
&gt;<br>
&gt; Am i missing something here?<br>
&gt;<br>
&gt; Thanks, Toms<br>
&gt;<br>
&gt; On 20 May 2015 at 23:13, Neal Cardwell &lt;<a href=3D"mailto:ncardwell=
@google.com">ncardwell@google.com</a>&gt; wrote:<br>
&gt; On Wed, May 20, 2015 at 1:18 PM, Tom Sanders &lt;<a href=3D"mailto:tom=
s.sanders@gmail.com">toms.sanders@gmail.com</a>&gt; wrote:<br>
&gt; &gt; Thanks Rick, this was very helpful.<br>
&gt; &gt;<br>
&gt; &gt; So if there are no setsockopts, etc then CUBIC&#39;s performance,=
 unlike other<br>
&gt; &gt; TCP variants, should not get impacted by RTTs, right? Is that a f=
air<br>
&gt; &gt; statement to make?<br>
&gt;<br>
&gt; CUBIC, like most TCP congestion control variants, has significant<br>
&gt; aspects that evolve on the scale of RTTs: for example, in slow start,<=
br>
&gt; loss recovery, or when the CUBIC function is very steep, CUBIC&#39;s<b=
r>
&gt; behavior is heavily dependent on RTT (send some packets, wait for an<b=
r>
&gt; ACK, send some more packets,...). So CUBIC will not be<br>
&gt; perfectly&quot;RTT-fair.<br>
&gt;<br>
&gt; I think it might be fair to say something like: once a path is<br>
&gt; saturated with CUBIC flows, as long as loss recovery or send/receive<b=
r>
&gt; buffers are not a bottleneck, then competition between CUBIC flows<br>
&gt; should be more RTT-fair than Reno, in the long run, since for large<br=
>
&gt; parts of the life cycle of the CUBIC flow the evolution of cwnd is<br>
&gt; constrained by wall clock time rather than ACK arrival.<br>
&gt;<br>
&gt; neal<br>
&gt;<br>
&gt;<br>
&gt; &gt; Toms<br>
&gt; &gt;<br>
&gt; &gt; On 20 May 2015 at 19:51, Rick Jones &lt;<a href=3D"mailto:perfgee=
k@mac.com">perfgeek@mac.com</a>&gt; wrote:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; On May 20, 2015, at 6:20 AM, Tom Sanders &lt;<a href=3D"mailt=
o:toms.sanders@gmail.com">toms.sanders@gmail.com</a>&gt; wrote:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; &gt; Hi,<br>
&gt; &gt;&gt; &gt;<br>
&gt; &gt;&gt; &gt; As per my understanding, unlike other TCP variants, CUBI=
C claims to be<br>
&gt; &gt;&gt; &gt; RTT fair. This is because the window growth rate is inde=
pendent of the RTT,<br>
&gt; &gt;&gt; &gt; and hence the performance is the same as on a low and a =
high RTT paths.<br>
&gt; &gt;&gt; &gt; Others are dependent upon RTT since their windows grow a=
s they receive ACKs<br>
&gt; &gt;&gt; &gt; from their peers, while CUBIC doesnt (window growth is a=
 function of time).<br>
&gt; &gt;&gt; &gt; This is theory.<br>
&gt; &gt;&gt; &gt;<br>
&gt; &gt;&gt; &gt; In practice, CUBIC (which is the default now on most lin=
ux<br>
&gt; &gt;&gt; &gt; distributions) also depends upon RTT. I connected to mac=
hines on a LAN and<br>
&gt; &gt;&gt; &gt; did a FTP transfer. I got a certain bandwidth. Next, i a=
rtificially inserted<br>
&gt; &gt;&gt; &gt; delay using &quot;tc qdisc&quot; in the path. I increase=
d the delay on the ethernet<br>
&gt; &gt;&gt; &gt; interface connecting my linux machine to the LAN to be a=
round 100ms. I<br>
&gt; &gt;&gt; &gt; immediately saw that the TCP performance went down.<br>
&gt; &gt;&gt; &gt;<br>
&gt; &gt;&gt; &gt; To bring it up to the same level as before i had to use =
multiple TCP<br>
&gt; &gt;&gt; &gt; streams. To me this means that CUBIC performance is depe=
ndent of the RTT. So<br>
&gt; &gt;&gt; &gt; why do we call it RTT-fair?<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I cannot speak to CUBIC specifically, but I would think you w=
ould want to<br>
&gt; &gt;&gt; make sure you were seeing effects of congestion window and no=
t of the<br>
&gt; &gt;&gt; classic TCP window.=C2=A0 For example, if the FTP client/serv=
er you were using<br>
&gt; &gt;&gt; makes an explicit setsockopt() to set the socket buffer sizes=
 and by<br>
&gt; &gt;&gt; extension the classic TCP window, your bumping of the RTT to =
around 100ms<br>
&gt; &gt;&gt; may have taken the bandwidth down as a consequence of the cla=
ssic:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Throughput &lt;=3D WindowSize/RTT<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Similarly, since you mention Linux, if the FTP client/server =
didn=E2=80=99t make<br>
&gt; &gt;&gt; explicit setsockopt() calls,=C2=A0 the sysctl values for tcp_=
rmem and tcp_wmem<br>
&gt; &gt;&gt; may have been such that the auto tuning of the window size co=
uldn=E2=80=99t allow<br>
&gt; &gt;&gt; the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I suspect that the experiment needs to be setup to have the s=
ame classic<br>
&gt; &gt;&gt; window size in both cases, sized per the above to be sufficie=
nt to achieve<br>
&gt; &gt;&gt; the peak throughput in the high RTT case.=C2=A0 Then you will=
 have eliminated<br>
&gt; &gt;&gt; classic window as a factor.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Also, depending on the version of the Linux kernel you are us=
ing, there<br>
&gt; &gt;&gt; may be some effects from tcp small queues or perhaps (a stret=
ch?) even byte<br>
&gt; &gt;&gt; queue limits which may not be playing well with netem.=C2=A0 =
(Just guessing there<br>
&gt; &gt;&gt; really)<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; happy benchmarking,<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; rick jones<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; --<br>
&gt; &gt; Toms.<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; tcpm mailing list<br>
&gt; &gt; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"=
_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
&gt; &gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Toms.<br>
&gt; _______________________________________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
<br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div class=3D"gmail_signature">Toms.</div>
</div></div>

--047d7b8738d0f9db890516b586a7--


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From: Gavin McCullagh <gmccullagh@gmail.com>
To: Tom Sanders <toms.sanders@gmail.com>
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Subject: Re: [tcpm] Why is Cubic said to be RTT-fair?
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Hi Tom,

What did you use to create the tcp flows?

Was it iperf or similar?

Gavin
On 23 May 2015 02:16, "Tom Sanders" <toms.sanders@gmail.com> wrote:

> Hi Mirja,
>
>>
>>
>> the text below only says that if you have two cubic flows with a
>> different base RTT competing at the same bottleneck they=E2=80=99ll stil=
l converge
>> to share the link equally. This does not mean that if you change the RTT
>> strongly during one connection that Cubic is able to catch up fast.
>>
>> In fact this is one weakness of Cubic that I have observed. If there are
>> change in base delay/rtt or strong increases in the available capacity,
>> Cubic needs a rather long time to catch up. This is because cubic has a
>> =E2=80=9Atarget value=E2=80=98 to which it increases it sending rate/win=
dow quickly but
>> then if the capacity cap is not somewhere close to this value (as expect=
ed)
>> it will probe very conservatively around this value and it takes a rathe=
r
>> long time until it gets back into a phase where it increases its congest=
ion
>> window fast.
>>
>
> Then which congestion control algorithm fares better there? I thought
> Cubic was suited for large fat pipes -- bigger bandwidth-delay pipes.
>
>
>>
>> So what do you mean by 'I immediately saw that the TCP performance went
>> down=E2=80=98. How long do you run your test? And over which period of t=
ime this
>> the performance go down?
>>
>
> I did a simple test.
>
> I connect two machines on my 1GB LAN.I do a tcp data transfer test and i
> get around ~900Mbps. I then added a delay using qdisc on my outgoing
> ethernet interface to simulate a longer RTT. So, if my ping earlier was
> around 0.255ms, it then became 120ms. I repeat the test and i see that i
> can only send about 200Mbps. I had to use multiple TCP flows to fill up m=
y
> 1GB link with Cubic. I run this experiment for about 2 minutes, and Cubic
> is not able to catch up. I dont have long standing flows, and only need t=
o
> send bursts of data, hence did not leave it running for very long.
>
> I was under the impression, after having read the original Cubic paper
> that Cubic's window growth rate is independent of RTT and hence wasnt
> expecting any difference in the throughput.
>
> Toms.
>
>>
>> Mirja
>>
>>
>> > Am 21.05.2015 um 02:51 schrieb Tom Sanders <toms.sanders@gmail.com>:
>> >
>> > Hi Neal,
>> >
>> > This is odd.
>> >
>> > In the original Cubic paper (
>> http://www4.ncsu.edu/~rhee/export/bitcp/cubic-paper.pdf) by Rhee and Xu,
>> they explicitly state that the Cubic is independent of RTT.
>> >
>> > "The main feature of CUBIC is that its window growth function is
>> defined in real-time so that its growth will be independent of RTT. Our
>> work was partially inspired by HTCP [5], whose window growth function is
>> also based on real time. The congestion epoch period of CUBIC is determi=
ned
>> by the packet loss rate alone. As TCP=E2=80=99s throughput is defined by=
 the packet
>> loss rate as well as RTT, the throughput of CUBIC is defined by only the
>> packet loss rate. Thus, when the loss rate is high and/or RTT is short,
>> CUBIC can operate in a TCP mode. Moreover, since the growth function is
>> independent of RTT, its RTT fairness is guaranteed as different RTT flow=
s
>> will still grow their windows at the same rate."
>> >
>> > Am i missing something here?
>> >
>> > Thanks, Toms
>> >
>> > On 20 May 2015 at 23:13, Neal Cardwell <ncardwell@google.com> wrote:
>> > On Wed, May 20, 2015 at 1:18 PM, Tom Sanders <toms.sanders@gmail.com>
>> wrote:
>> > > Thanks Rick, this was very helpful.
>> > >
>> > > So if there are no setsockopts, etc then CUBIC's performance, unlike
>> other
>> > > TCP variants, should not get impacted by RTTs, right? Is that a fair
>> > > statement to make?
>> >
>> > CUBIC, like most TCP congestion control variants, has significant
>> > aspects that evolve on the scale of RTTs: for example, in slow start,
>> > loss recovery, or when the CUBIC function is very steep, CUBIC's
>> > behavior is heavily dependent on RTT (send some packets, wait for an
>> > ACK, send some more packets,...). So CUBIC will not be
>> > perfectly"RTT-fair.
>> >
>> > I think it might be fair to say something like: once a path is
>> > saturated with CUBIC flows, as long as loss recovery or send/receive
>> > buffers are not a bottleneck, then competition between CUBIC flows
>> > should be more RTT-fair than Reno, in the long run, since for large
>> > parts of the life cycle of the CUBIC flow the evolution of cwnd is
>> > constrained by wall clock time rather than ACK arrival.
>> >
>> > neal
>> >
>> >
>> > > Toms
>> > >
>> > > On 20 May 2015 at 19:51, Rick Jones <perfgeek@mac.com> wrote:
>> > >>
>> > >>
>> > >> On May 20, 2015, at 6:20 AM, Tom Sanders <toms.sanders@gmail.com>
>> wrote:
>> > >>
>> > >> > Hi,
>> > >> >
>> > >> > As per my understanding, unlike other TCP variants, CUBIC claims
>> to be
>> > >> > RTT fair. This is because the window growth rate is independent o=
f
>> the RTT,
>> > >> > and hence the performance is the same as on a low and a high RTT
>> paths.
>> > >> > Others are dependent upon RTT since their windows grow as they
>> receive ACKs
>> > >> > from their peers, while CUBIC doesnt (window growth is a function
>> of time).
>> > >> > This is theory.
>> > >> >
>> > >> > In practice, CUBIC (which is the default now on most linux
>> > >> > distributions) also depends upon RTT. I connected to machines on =
a
>> LAN and
>> > >> > did a FTP transfer. I got a certain bandwidth. Next, i
>> artificially inserted
>> > >> > delay using "tc qdisc" in the path. I increased the delay on the
>> ethernet
>> > >> > interface connecting my linux machine to the LAN to be around
>> 100ms. I
>> > >> > immediately saw that the TCP performance went down.
>> > >> >
>> > >> > To bring it up to the same level as before i had to use multiple
>> TCP
>> > >> > streams. To me this means that CUBIC performance is dependent of
>> the RTT. So
>> > >> > why do we call it RTT-fair?
>> > >>
>> > >> I cannot speak to CUBIC specifically, but I would think you would
>> want to
>> > >> make sure you were seeing effects of congestion window and not of t=
he
>> > >> classic TCP window.  For example, if the FTP client/server you were
>> using
>> > >> makes an explicit setsockopt() to set the socket buffer sizes and b=
y
>> > >> extension the classic TCP window, your bumping of the RTT to around
>> 100ms
>> > >> may have taken the bandwidth down as a consequence of the classic:
>> > >>
>> > >> Throughput <=3D WindowSize/RTT
>> > >>
>> > >> Similarly, since you mention Linux, if the FTP client/server didn=
=E2=80=99t
>> make
>> > >> explicit setsockopt() calls,  the sysctl values for tcp_rmem and
>> tcp_wmem
>> > >> may have been such that the auto tuning of the window size couldn=
=E2=80=99t
>> allow
>> > >> the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D
>> > >>
>> > >> I suspect that the experiment needs to be setup to have the same
>> classic
>> > >> window size in both cases, sized per the above to be sufficient to
>> achieve
>> > >> the peak throughput in the high RTT case.  Then you will have
>> eliminated
>> > >> classic window as a factor.
>> > >>
>> > >> Also, depending on the version of the Linux kernel you are using,
>> there
>> > >> may be some effects from tcp small queues or perhaps (a stretch?)
>> even byte
>> > >> queue limits which may not be playing well with netem.  (Just
>> guessing there
>> > >> really)
>> > >>
>> > >> happy benchmarking,
>> > >>
>> > >> rick jones
>> > >
>> > >
>> > >
>> > >
>> > > --
>> > > Toms.
>> > >
>> > > _______________________________________________
>> > > tcpm mailing list
>> > > tcpm@ietf.org
>> > > https://www.ietf.org/mailman/listinfo/tcpm
>> > >
>> >
>> >
>> >
>> > --
>> > Toms.
>> > _______________________________________________
>> > tcpm mailing list
>> > tcpm@ietf.org
>> > https://www.ietf.org/mailman/listinfo/tcpm
>>
>>
>
>
> --
> Toms.
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>
>

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

<p dir=3D"ltr">Hi Tom,</p>
<p dir=3D"ltr">What did you use to create the tcp flows?</p>
<p dir=3D"ltr">Was it iperf or similar?</p>
<p dir=3D"ltr">Gavin</p>
<div class=3D"gmail_quote">On 23 May 2015 02:16, &quot;Tom Sanders&quot; &l=
t;<a href=3D"mailto:toms.sanders@gmail.com">toms.sanders@gmail.com</a>&gt; =
wrote:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"l=
tr">Hi Mirja,<div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex"><br><br>
the text below only says that if you have two cubic flows with a different =
base RTT competing at the same bottleneck they=E2=80=99ll still converge to=
 share the link equally. This does not mean that if you change the RTT stro=
ngly during one connection that Cubic is able to catch up fast.<br>
<br>
In fact this is one weakness of Cubic that I have observed. If there are ch=
ange in base delay/rtt or strong increases in the available capacity, Cubic=
 needs a rather long time to catch up. This is because cubic has a =E2=80=
=9Atarget value=E2=80=98 to which it increases it sending rate/window quick=
ly but then if the capacity cap is not somewhere close to this value (as ex=
pected) it will probe very conservatively around this value and it takes a =
rather long time until it gets back into a phase where it increases its con=
gestion window fast.<br></blockquote><div><br></div><div>Then which congest=
ion control algorithm fares better there? I thought Cubic was suited for la=
rge fat pipes -- bigger bandwidth-delay pipes.=C2=A0</div><div>=C2=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
So what do you mean by &#39;I immediately saw that the TCP performance went=
 down=E2=80=98. How long do you run your test? And over which period of tim=
e this the performance go down?<br></blockquote><div><br></div><div>I did a=
 simple test.</div><div><br></div><div>I connect two machines on my 1GB LAN=
.I do a tcp data transfer test and i get around ~900Mbps. I then added a de=
lay using qdisc on my outgoing ethernet interface to simulate a longer RTT.=
 So, if my ping earlier was around 0.255ms, it then became 120ms. I repeat =
the test and i see that i can only send about 200Mbps. I had to use multipl=
e TCP flows to fill up my 1GB link with Cubic. I run this experiment for ab=
out 2 minutes, and Cubic is not able to catch up. I dont have long standing=
 flows, and only need to send bursts of data, hence did not leave it runnin=
g for very long.</div><div><br></div><div>I was under the impression, after=
 having read the original Cubic paper that Cubic&#39;s window growth rate i=
s independent of RTT and hence wasnt expecting any difference in the throug=
hput.</div><div><br></div><div>Toms.</div><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<span><font color=3D"#888888"><br>
Mirja<br>
</font></span><div><div><br>
<br>
&gt; Am 21.05.2015 um 02:51 schrieb Tom Sanders &lt;<a href=3D"mailto:toms.=
sanders@gmail.com" target=3D"_blank">toms.sanders@gmail.com</a>&gt;:<br>
&gt;<br>
&gt; Hi Neal,<br>
&gt;<br>
&gt; This is odd.<br>
&gt;<br>
&gt; In the original Cubic paper (<a href=3D"http://www4.ncsu.edu/~rhee/exp=
ort/bitcp/cubic-paper.pdf" target=3D"_blank">http://www4.ncsu.edu/~rhee/exp=
ort/bitcp/cubic-paper.pdf</a>) by Rhee and Xu, they explicitly state that t=
he Cubic is independent of RTT.<br>
&gt;<br>
&gt; &quot;The main feature of CUBIC is that its window growth function is =
defined in real-time so that its growth will be independent of RTT. Our wor=
k was partially inspired by HTCP [5], whose window growth function is also =
based on real time. The congestion epoch period of CUBIC is determined by t=
he packet loss rate alone. As TCP=E2=80=99s throughput is defined by the pa=
cket loss rate as well as RTT, the throughput of CUBIC is defined by only t=
he packet loss rate. Thus, when the loss rate is high and/or RTT is short, =
CUBIC can operate in a TCP mode. Moreover, since the growth function is ind=
ependent of RTT, its RTT fairness is guaranteed as different RTT flows will=
 still grow their windows at the same rate.&quot;<br>
&gt;<br>
&gt; Am i missing something here?<br>
&gt;<br>
&gt; Thanks, Toms<br>
&gt;<br>
&gt; On 20 May 2015 at 23:13, Neal Cardwell &lt;<a href=3D"mailto:ncardwell=
@google.com" target=3D"_blank">ncardwell@google.com</a>&gt; wrote:<br>
&gt; On Wed, May 20, 2015 at 1:18 PM, Tom Sanders &lt;<a href=3D"mailto:tom=
s.sanders@gmail.com" target=3D"_blank">toms.sanders@gmail.com</a>&gt; wrote=
:<br>
&gt; &gt; Thanks Rick, this was very helpful.<br>
&gt; &gt;<br>
&gt; &gt; So if there are no setsockopts, etc then CUBIC&#39;s performance,=
 unlike other<br>
&gt; &gt; TCP variants, should not get impacted by RTTs, right? Is that a f=
air<br>
&gt; &gt; statement to make?<br>
&gt;<br>
&gt; CUBIC, like most TCP congestion control variants, has significant<br>
&gt; aspects that evolve on the scale of RTTs: for example, in slow start,<=
br>
&gt; loss recovery, or when the CUBIC function is very steep, CUBIC&#39;s<b=
r>
&gt; behavior is heavily dependent on RTT (send some packets, wait for an<b=
r>
&gt; ACK, send some more packets,...). So CUBIC will not be<br>
&gt; perfectly&quot;RTT-fair.<br>
&gt;<br>
&gt; I think it might be fair to say something like: once a path is<br>
&gt; saturated with CUBIC flows, as long as loss recovery or send/receive<b=
r>
&gt; buffers are not a bottleneck, then competition between CUBIC flows<br>
&gt; should be more RTT-fair than Reno, in the long run, since for large<br=
>
&gt; parts of the life cycle of the CUBIC flow the evolution of cwnd is<br>
&gt; constrained by wall clock time rather than ACK arrival.<br>
&gt;<br>
&gt; neal<br>
&gt;<br>
&gt;<br>
&gt; &gt; Toms<br>
&gt; &gt;<br>
&gt; &gt; On 20 May 2015 at 19:51, Rick Jones &lt;<a href=3D"mailto:perfgee=
k@mac.com" target=3D"_blank">perfgeek@mac.com</a>&gt; wrote:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; On May 20, 2015, at 6:20 AM, Tom Sanders &lt;<a href=3D"mailt=
o:toms.sanders@gmail.com" target=3D"_blank">toms.sanders@gmail.com</a>&gt; =
wrote:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; &gt; Hi,<br>
&gt; &gt;&gt; &gt;<br>
&gt; &gt;&gt; &gt; As per my understanding, unlike other TCP variants, CUBI=
C claims to be<br>
&gt; &gt;&gt; &gt; RTT fair. This is because the window growth rate is inde=
pendent of the RTT,<br>
&gt; &gt;&gt; &gt; and hence the performance is the same as on a low and a =
high RTT paths.<br>
&gt; &gt;&gt; &gt; Others are dependent upon RTT since their windows grow a=
s they receive ACKs<br>
&gt; &gt;&gt; &gt; from their peers, while CUBIC doesnt (window growth is a=
 function of time).<br>
&gt; &gt;&gt; &gt; This is theory.<br>
&gt; &gt;&gt; &gt;<br>
&gt; &gt;&gt; &gt; In practice, CUBIC (which is the default now on most lin=
ux<br>
&gt; &gt;&gt; &gt; distributions) also depends upon RTT. I connected to mac=
hines on a LAN and<br>
&gt; &gt;&gt; &gt; did a FTP transfer. I got a certain bandwidth. Next, i a=
rtificially inserted<br>
&gt; &gt;&gt; &gt; delay using &quot;tc qdisc&quot; in the path. I increase=
d the delay on the ethernet<br>
&gt; &gt;&gt; &gt; interface connecting my linux machine to the LAN to be a=
round 100ms. I<br>
&gt; &gt;&gt; &gt; immediately saw that the TCP performance went down.<br>
&gt; &gt;&gt; &gt;<br>
&gt; &gt;&gt; &gt; To bring it up to the same level as before i had to use =
multiple TCP<br>
&gt; &gt;&gt; &gt; streams. To me this means that CUBIC performance is depe=
ndent of the RTT. So<br>
&gt; &gt;&gt; &gt; why do we call it RTT-fair?<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I cannot speak to CUBIC specifically, but I would think you w=
ould want to<br>
&gt; &gt;&gt; make sure you were seeing effects of congestion window and no=
t of the<br>
&gt; &gt;&gt; classic TCP window.=C2=A0 For example, if the FTP client/serv=
er you were using<br>
&gt; &gt;&gt; makes an explicit setsockopt() to set the socket buffer sizes=
 and by<br>
&gt; &gt;&gt; extension the classic TCP window, your bumping of the RTT to =
around 100ms<br>
&gt; &gt;&gt; may have taken the bandwidth down as a consequence of the cla=
ssic:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Throughput &lt;=3D WindowSize/RTT<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Similarly, since you mention Linux, if the FTP client/server =
didn=E2=80=99t make<br>
&gt; &gt;&gt; explicit setsockopt() calls,=C2=A0 the sysctl values for tcp_=
rmem and tcp_wmem<br>
&gt; &gt;&gt; may have been such that the auto tuning of the window size co=
uldn=E2=80=99t allow<br>
&gt; &gt;&gt; the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I suspect that the experiment needs to be setup to have the s=
ame classic<br>
&gt; &gt;&gt; window size in both cases, sized per the above to be sufficie=
nt to achieve<br>
&gt; &gt;&gt; the peak throughput in the high RTT case.=C2=A0 Then you will=
 have eliminated<br>
&gt; &gt;&gt; classic window as a factor.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Also, depending on the version of the Linux kernel you are us=
ing, there<br>
&gt; &gt;&gt; may be some effects from tcp small queues or perhaps (a stret=
ch?) even byte<br>
&gt; &gt;&gt; queue limits which may not be playing well with netem.=C2=A0 =
(Just guessing there<br>
&gt; &gt;&gt; really)<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; happy benchmarking,<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; rick jones<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; --<br>
&gt; &gt; Toms.<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; tcpm mailing list<br>
&gt; &gt; <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org<=
/a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"=
_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
&gt; &gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Toms.<br>
&gt; _______________________________________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><b=
r>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
<br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div>Toms.</div>
</div></div>
<br>_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
<br></blockquote></div>

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From: Tom Sanders <toms.sanders@gmail.com>
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Subject: Re: [tcpm] Why is Cubic said to be RTT-fair?
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You can use iperf with parallel streams and the result is the same. In my
case it was my own client/server pushing data over the TCP socket.

I have a theory that the following happens:

When i artificially induce the delay there are some packet losses since the
outgoing ethernet interface has to buffer the packets (to simulate the link
delay). Its probably of these packet losses that the throughput that i get
with one TCP stream is much lower than the available bandwidth. When i
increase the number of TCP streams, for reasons well known, i get better
bandwidth.

I think, and correct me if i am wrong here, if i were to induce latency
WITHOUT packet losses then even one TCP stream would be able to fill up the
network pipe (assuming that the kernel buffers are not the limiting issue).

Thanks, Toms

On 23 May 2015 at 11:06, Gavin McCullagh <gmccullagh@gmail.com> wrote:

> Hi Tom,
>
> What did you use to create the tcp flows?
>
> Was it iperf or similar?
>
> Gavin
> On 23 May 2015 02:16, "Tom Sanders" <toms.sanders@gmail.com> wrote:
>
>> Hi Mirja,
>>
>>>
>>>
>>> the text below only says that if you have two cubic flows with a
>>> different base RTT competing at the same bottleneck they=E2=80=99ll sti=
ll converge
>>> to share the link equally. This does not mean that if you change the RT=
T
>>> strongly during one connection that Cubic is able to catch up fast.
>>>
>>> In fact this is one weakness of Cubic that I have observed. If there ar=
e
>>> change in base delay/rtt or strong increases in the available capacity,
>>> Cubic needs a rather long time to catch up. This is because cubic has a
>>> =E2=80=9Atarget value=E2=80=98 to which it increases it sending rate/wi=
ndow quickly but
>>> then if the capacity cap is not somewhere close to this value (as expec=
ted)
>>> it will probe very conservatively around this value and it takes a rath=
er
>>> long time until it gets back into a phase where it increases its conges=
tion
>>> window fast.
>>>
>>
>> Then which congestion control algorithm fares better there? I thought
>> Cubic was suited for large fat pipes -- bigger bandwidth-delay pipes.
>>
>>
>>>
>>> So what do you mean by 'I immediately saw that the TCP performance went
>>> down=E2=80=98. How long do you run your test? And over which period of =
time this
>>> the performance go down?
>>>
>>
>> I did a simple test.
>>
>> I connect two machines on my 1GB LAN.I do a tcp data transfer test and i
>> get around ~900Mbps. I then added a delay using qdisc on my outgoing
>> ethernet interface to simulate a longer RTT. So, if my ping earlier was
>> around 0.255ms, it then became 120ms. I repeat the test and i see that i
>> can only send about 200Mbps. I had to use multiple TCP flows to fill up =
my
>> 1GB link with Cubic. I run this experiment for about 2 minutes, and Cubi=
c
>> is not able to catch up. I dont have long standing flows, and only need =
to
>> send bursts of data, hence did not leave it running for very long.
>>
>> I was under the impression, after having read the original Cubic paper
>> that Cubic's window growth rate is independent of RTT and hence wasnt
>> expecting any difference in the throughput.
>>
>> Toms.
>>
>>>
>>> Mirja
>>>
>>>
>>> > Am 21.05.2015 um 02:51 schrieb Tom Sanders <toms.sanders@gmail.com>:
>>> >
>>> > Hi Neal,
>>> >
>>> > This is odd.
>>> >
>>> > In the original Cubic paper (
>>> http://www4.ncsu.edu/~rhee/export/bitcp/cubic-paper.pdf) by Rhee and
>>> Xu, they explicitly state that the Cubic is independent of RTT.
>>> >
>>> > "The main feature of CUBIC is that its window growth function is
>>> defined in real-time so that its growth will be independent of RTT. Our
>>> work was partially inspired by HTCP [5], whose window growth function i=
s
>>> also based on real time. The congestion epoch period of CUBIC is determ=
ined
>>> by the packet loss rate alone. As TCP=E2=80=99s throughput is defined b=
y the packet
>>> loss rate as well as RTT, the throughput of CUBIC is defined by only th=
e
>>> packet loss rate. Thus, when the loss rate is high and/or RTT is short,
>>> CUBIC can operate in a TCP mode. Moreover, since the growth function is
>>> independent of RTT, its RTT fairness is guaranteed as different RTT flo=
ws
>>> will still grow their windows at the same rate."
>>> >
>>> > Am i missing something here?
>>> >
>>> > Thanks, Toms
>>> >
>>> > On 20 May 2015 at 23:13, Neal Cardwell <ncardwell@google.com> wrote:
>>> > On Wed, May 20, 2015 at 1:18 PM, Tom Sanders <toms.sanders@gmail.com>
>>> wrote:
>>> > > Thanks Rick, this was very helpful.
>>> > >
>>> > > So if there are no setsockopts, etc then CUBIC's performance, unlik=
e
>>> other
>>> > > TCP variants, should not get impacted by RTTs, right? Is that a fai=
r
>>> > > statement to make?
>>> >
>>> > CUBIC, like most TCP congestion control variants, has significant
>>> > aspects that evolve on the scale of RTTs: for example, in slow start,
>>> > loss recovery, or when the CUBIC function is very steep, CUBIC's
>>> > behavior is heavily dependent on RTT (send some packets, wait for an
>>> > ACK, send some more packets,...). So CUBIC will not be
>>> > perfectly"RTT-fair.
>>> >
>>> > I think it might be fair to say something like: once a path is
>>> > saturated with CUBIC flows, as long as loss recovery or send/receive
>>> > buffers are not a bottleneck, then competition between CUBIC flows
>>> > should be more RTT-fair than Reno, in the long run, since for large
>>> > parts of the life cycle of the CUBIC flow the evolution of cwnd is
>>> > constrained by wall clock time rather than ACK arrival.
>>> >
>>> > neal
>>> >
>>> >
>>> > > Toms
>>> > >
>>> > > On 20 May 2015 at 19:51, Rick Jones <perfgeek@mac.com> wrote:
>>> > >>
>>> > >>
>>> > >> On May 20, 2015, at 6:20 AM, Tom Sanders <toms.sanders@gmail.com>
>>> wrote:
>>> > >>
>>> > >> > Hi,
>>> > >> >
>>> > >> > As per my understanding, unlike other TCP variants, CUBIC claims
>>> to be
>>> > >> > RTT fair. This is because the window growth rate is independent
>>> of the RTT,
>>> > >> > and hence the performance is the same as on a low and a high RTT
>>> paths.
>>> > >> > Others are dependent upon RTT since their windows grow as they
>>> receive ACKs
>>> > >> > from their peers, while CUBIC doesnt (window growth is a functio=
n
>>> of time).
>>> > >> > This is theory.
>>> > >> >
>>> > >> > In practice, CUBIC (which is the default now on most linux
>>> > >> > distributions) also depends upon RTT. I connected to machines on
>>> a LAN and
>>> > >> > did a FTP transfer. I got a certain bandwidth. Next, i
>>> artificially inserted
>>> > >> > delay using "tc qdisc" in the path. I increased the delay on the
>>> ethernet
>>> > >> > interface connecting my linux machine to the LAN to be around
>>> 100ms. I
>>> > >> > immediately saw that the TCP performance went down.
>>> > >> >
>>> > >> > To bring it up to the same level as before i had to use multiple
>>> TCP
>>> > >> > streams. To me this means that CUBIC performance is dependent of
>>> the RTT. So
>>> > >> > why do we call it RTT-fair?
>>> > >>
>>> > >> I cannot speak to CUBIC specifically, but I would think you would
>>> want to
>>> > >> make sure you were seeing effects of congestion window and not of
>>> the
>>> > >> classic TCP window.  For example, if the FTP client/server you wer=
e
>>> using
>>> > >> makes an explicit setsockopt() to set the socket buffer sizes and =
by
>>> > >> extension the classic TCP window, your bumping of the RTT to aroun=
d
>>> 100ms
>>> > >> may have taken the bandwidth down as a consequence of the classic:
>>> > >>
>>> > >> Throughput <=3D WindowSize/RTT
>>> > >>
>>> > >> Similarly, since you mention Linux, if the FTP client/server didn=
=E2=80=99t
>>> make
>>> > >> explicit setsockopt() calls,  the sysctl values for tcp_rmem and
>>> tcp_wmem
>>> > >> may have been such that the auto tuning of the window size couldn=
=E2=80=99t
>>> allow
>>> > >> the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D
>>> > >>
>>> > >> I suspect that the experiment needs to be setup to have the same
>>> classic
>>> > >> window size in both cases, sized per the above to be sufficient to
>>> achieve
>>> > >> the peak throughput in the high RTT case.  Then you will have
>>> eliminated
>>> > >> classic window as a factor.
>>> > >>
>>> > >> Also, depending on the version of the Linux kernel you are using,
>>> there
>>> > >> may be some effects from tcp small queues or perhaps (a stretch?)
>>> even byte
>>> > >> queue limits which may not be playing well with netem.  (Just
>>> guessing there
>>> > >> really)
>>> > >>
>>> > >> happy benchmarking,
>>> > >>
>>> > >> rick jones
>>> > >
>>> > >
>>> > >
>>> > >
>>> > > --
>>> > > Toms.
>>> > >
>>> > > _______________________________________________
>>> > > tcpm mailing list
>>> > > tcpm@ietf.org
>>> > > https://www.ietf.org/mailman/listinfo/tcpm
>>> > >
>>> >
>>> >
>>> >
>>> > --
>>> > Toms.
>>> > _______________________________________________
>>> > tcpm mailing list
>>> > tcpm@ietf.org
>>> > https://www.ietf.org/mailman/listinfo/tcpm
>>>
>>>
>>
>>
>> --
>> Toms.
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>>


--=20
Toms.

--f46d043d67e5bc23b10516bbe731
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">You can use iperf with parallel streams and the result is =
the same. In my case it was my own client/server pushing data over the TCP =
socket.<div><br></div><div>I have a theory that the following happens:</div=
><div><br></div><div>When i artificially induce the delay there are some pa=
cket losses since the outgoing ethernet interface has to buffer the packets=
 (to simulate the link delay). Its probably of these packet losses that the=
 throughput that i get with one TCP stream is much lower than the available=
 bandwidth. When i increase the number of TCP streams, for reasons well kno=
wn, i get better bandwidth.</div><div><br></div><div>I think, and correct m=
e if i am wrong here, if i were to induce latency WITHOUT packet losses the=
n even one TCP stream would be able to fill up the network pipe (assuming t=
hat the kernel buffers are not the limiting issue).</div><div><br></div><di=
v>Thanks, Toms</div></div><div class=3D"gmail_extra"><br><div class=3D"gmai=
l_quote">On 23 May 2015 at 11:06, Gavin McCullagh <span dir=3D"ltr">&lt;<a =
href=3D"mailto:gmccullagh@gmail.com" target=3D"_blank">gmccullagh@gmail.com=
</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir=3D"ltr">Hi =
Tom,</p>
<p dir=3D"ltr">What did you use to create the tcp flows?</p>
<p dir=3D"ltr">Was it iperf or similar?</p><span class=3D"HOEnZb"><font col=
or=3D"#888888">
<p dir=3D"ltr">Gavin</p></font></span><div class=3D"HOEnZb"><div class=3D"h=
5">
<div class=3D"gmail_quote">On 23 May 2015 02:16, &quot;Tom Sanders&quot; &l=
t;<a href=3D"mailto:toms.sanders@gmail.com" target=3D"_blank">toms.sanders@=
gmail.com</a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex"><div dir=3D"ltr">Hi Mirja,<div class=3D"gmail_extra"><div class=3D"gma=
il_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex"><br><br>
the text below only says that if you have two cubic flows with a different =
base RTT competing at the same bottleneck they=E2=80=99ll still converge to=
 share the link equally. This does not mean that if you change the RTT stro=
ngly during one connection that Cubic is able to catch up fast.<br>
<br>
In fact this is one weakness of Cubic that I have observed. If there are ch=
ange in base delay/rtt or strong increases in the available capacity, Cubic=
 needs a rather long time to catch up. This is because cubic has a =E2=80=
=9Atarget value=E2=80=98 to which it increases it sending rate/window quick=
ly but then if the capacity cap is not somewhere close to this value (as ex=
pected) it will probe very conservatively around this value and it takes a =
rather long time until it gets back into a phase where it increases its con=
gestion window fast.<br></blockquote><div><br></div><div>Then which congest=
ion control algorithm fares better there? I thought Cubic was suited for la=
rge fat pipes -- bigger bandwidth-delay pipes.=C2=A0</div><div>=C2=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
So what do you mean by &#39;I immediately saw that the TCP performance went=
 down=E2=80=98. How long do you run your test? And over which period of tim=
e this the performance go down?<br></blockquote><div><br></div><div>I did a=
 simple test.</div><div><br></div><div>I connect two machines on my 1GB LAN=
.I do a tcp data transfer test and i get around ~900Mbps. I then added a de=
lay using qdisc on my outgoing ethernet interface to simulate a longer RTT.=
 So, if my ping earlier was around 0.255ms, it then became 120ms. I repeat =
the test and i see that i can only send about 200Mbps. I had to use multipl=
e TCP flows to fill up my 1GB link with Cubic. I run this experiment for ab=
out 2 minutes, and Cubic is not able to catch up. I dont have long standing=
 flows, and only need to send bursts of data, hence did not leave it runnin=
g for very long.</div><div><br></div><div>I was under the impression, after=
 having read the original Cubic paper that Cubic&#39;s window growth rate i=
s independent of RTT and hence wasnt expecting any difference in the throug=
hput.</div><div><br></div><div>Toms.</div><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<span><font color=3D"#888888"><br>
Mirja<br>
</font></span><div><div><br>
<br>
&gt; Am 21.05.2015 um 02:51 schrieb Tom Sanders &lt;<a href=3D"mailto:toms.=
sanders@gmail.com" target=3D"_blank">toms.sanders@gmail.com</a>&gt;:<br>
&gt;<br>
&gt; Hi Neal,<br>
&gt;<br>
&gt; This is odd.<br>
&gt;<br>
&gt; In the original Cubic paper (<a href=3D"http://www4.ncsu.edu/~rhee/exp=
ort/bitcp/cubic-paper.pdf" target=3D"_blank">http://www4.ncsu.edu/~rhee/exp=
ort/bitcp/cubic-paper.pdf</a>) by Rhee and Xu, they explicitly state that t=
he Cubic is independent of RTT.<br>
&gt;<br>
&gt; &quot;The main feature of CUBIC is that its window growth function is =
defined in real-time so that its growth will be independent of RTT. Our wor=
k was partially inspired by HTCP [5], whose window growth function is also =
based on real time. The congestion epoch period of CUBIC is determined by t=
he packet loss rate alone. As TCP=E2=80=99s throughput is defined by the pa=
cket loss rate as well as RTT, the throughput of CUBIC is defined by only t=
he packet loss rate. Thus, when the loss rate is high and/or RTT is short, =
CUBIC can operate in a TCP mode. Moreover, since the growth function is ind=
ependent of RTT, its RTT fairness is guaranteed as different RTT flows will=
 still grow their windows at the same rate.&quot;<br>
&gt;<br>
&gt; Am i missing something here?<br>
&gt;<br>
&gt; Thanks, Toms<br>
&gt;<br>
&gt; On 20 May 2015 at 23:13, Neal Cardwell &lt;<a href=3D"mailto:ncardwell=
@google.com" target=3D"_blank">ncardwell@google.com</a>&gt; wrote:<br>
&gt; On Wed, May 20, 2015 at 1:18 PM, Tom Sanders &lt;<a href=3D"mailto:tom=
s.sanders@gmail.com" target=3D"_blank">toms.sanders@gmail.com</a>&gt; wrote=
:<br>
&gt; &gt; Thanks Rick, this was very helpful.<br>
&gt; &gt;<br>
&gt; &gt; So if there are no setsockopts, etc then CUBIC&#39;s performance,=
 unlike other<br>
&gt; &gt; TCP variants, should not get impacted by RTTs, right? Is that a f=
air<br>
&gt; &gt; statement to make?<br>
&gt;<br>
&gt; CUBIC, like most TCP congestion control variants, has significant<br>
&gt; aspects that evolve on the scale of RTTs: for example, in slow start,<=
br>
&gt; loss recovery, or when the CUBIC function is very steep, CUBIC&#39;s<b=
r>
&gt; behavior is heavily dependent on RTT (send some packets, wait for an<b=
r>
&gt; ACK, send some more packets,...). So CUBIC will not be<br>
&gt; perfectly&quot;RTT-fair.<br>
&gt;<br>
&gt; I think it might be fair to say something like: once a path is<br>
&gt; saturated with CUBIC flows, as long as loss recovery or send/receive<b=
r>
&gt; buffers are not a bottleneck, then competition between CUBIC flows<br>
&gt; should be more RTT-fair than Reno, in the long run, since for large<br=
>
&gt; parts of the life cycle of the CUBIC flow the evolution of cwnd is<br>
&gt; constrained by wall clock time rather than ACK arrival.<br>
&gt;<br>
&gt; neal<br>
&gt;<br>
&gt;<br>
&gt; &gt; Toms<br>
&gt; &gt;<br>
&gt; &gt; On 20 May 2015 at 19:51, Rick Jones &lt;<a href=3D"mailto:perfgee=
k@mac.com" target=3D"_blank">perfgeek@mac.com</a>&gt; wrote:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; On May 20, 2015, at 6:20 AM, Tom Sanders &lt;<a href=3D"mailt=
o:toms.sanders@gmail.com" target=3D"_blank">toms.sanders@gmail.com</a>&gt; =
wrote:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; &gt; Hi,<br>
&gt; &gt;&gt; &gt;<br>
&gt; &gt;&gt; &gt; As per my understanding, unlike other TCP variants, CUBI=
C claims to be<br>
&gt; &gt;&gt; &gt; RTT fair. This is because the window growth rate is inde=
pendent of the RTT,<br>
&gt; &gt;&gt; &gt; and hence the performance is the same as on a low and a =
high RTT paths.<br>
&gt; &gt;&gt; &gt; Others are dependent upon RTT since their windows grow a=
s they receive ACKs<br>
&gt; &gt;&gt; &gt; from their peers, while CUBIC doesnt (window growth is a=
 function of time).<br>
&gt; &gt;&gt; &gt; This is theory.<br>
&gt; &gt;&gt; &gt;<br>
&gt; &gt;&gt; &gt; In practice, CUBIC (which is the default now on most lin=
ux<br>
&gt; &gt;&gt; &gt; distributions) also depends upon RTT. I connected to mac=
hines on a LAN and<br>
&gt; &gt;&gt; &gt; did a FTP transfer. I got a certain bandwidth. Next, i a=
rtificially inserted<br>
&gt; &gt;&gt; &gt; delay using &quot;tc qdisc&quot; in the path. I increase=
d the delay on the ethernet<br>
&gt; &gt;&gt; &gt; interface connecting my linux machine to the LAN to be a=
round 100ms. I<br>
&gt; &gt;&gt; &gt; immediately saw that the TCP performance went down.<br>
&gt; &gt;&gt; &gt;<br>
&gt; &gt;&gt; &gt; To bring it up to the same level as before i had to use =
multiple TCP<br>
&gt; &gt;&gt; &gt; streams. To me this means that CUBIC performance is depe=
ndent of the RTT. So<br>
&gt; &gt;&gt; &gt; why do we call it RTT-fair?<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I cannot speak to CUBIC specifically, but I would think you w=
ould want to<br>
&gt; &gt;&gt; make sure you were seeing effects of congestion window and no=
t of the<br>
&gt; &gt;&gt; classic TCP window.=C2=A0 For example, if the FTP client/serv=
er you were using<br>
&gt; &gt;&gt; makes an explicit setsockopt() to set the socket buffer sizes=
 and by<br>
&gt; &gt;&gt; extension the classic TCP window, your bumping of the RTT to =
around 100ms<br>
&gt; &gt;&gt; may have taken the bandwidth down as a consequence of the cla=
ssic:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Throughput &lt;=3D WindowSize/RTT<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Similarly, since you mention Linux, if the FTP client/server =
didn=E2=80=99t make<br>
&gt; &gt;&gt; explicit setsockopt() calls,=C2=A0 the sysctl values for tcp_=
rmem and tcp_wmem<br>
&gt; &gt;&gt; may have been such that the auto tuning of the window size co=
uldn=E2=80=99t allow<br>
&gt; &gt;&gt; the classic TCP window to grow =E2=80=9Cenough.=E2=80=9D<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I suspect that the experiment needs to be setup to have the s=
ame classic<br>
&gt; &gt;&gt; window size in both cases, sized per the above to be sufficie=
nt to achieve<br>
&gt; &gt;&gt; the peak throughput in the high RTT case.=C2=A0 Then you will=
 have eliminated<br>
&gt; &gt;&gt; classic window as a factor.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Also, depending on the version of the Linux kernel you are us=
ing, there<br>
&gt; &gt;&gt; may be some effects from tcp small queues or perhaps (a stret=
ch?) even byte<br>
&gt; &gt;&gt; queue limits which may not be playing well with netem.=C2=A0 =
(Just guessing there<br>
&gt; &gt;&gt; really)<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; happy benchmarking,<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; rick jones<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; --<br>
&gt; &gt; Toms.<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; tcpm mailing list<br>
&gt; &gt; <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org<=
/a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"=
_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
&gt; &gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Toms.<br>
&gt; _______________________________________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><b=
r>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
<br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div>Toms.</div>
</div></div>
<br>_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
<br></blockquote></div>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div class=3D"gmail_signature">Toms.</div>
</div>

--f46d043d67e5bc23b10516bbe731--


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From: Rick Jones <perfgeek@mac.com>
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To: Tom Sanders <toms.sanders@gmail.com>
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Subject: Re: [tcpm] Why is Cubic said to be RTT-fair?
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On May 23, 2015, at 1:53 AM, Tom Sanders <toms.sanders@gmail.com> wrote:

> You can use iperf with parallel streams and the result is the same. In =
my case it was my own client/server pushing data over the TCP socket.
>=20
> I have a theory that the following happens:
>=20
> When i artificially induce the delay there are some packet losses =
since the outgoing ethernet interface has to buffer the packets (to =
simulate the link delay). Its probably of these packet losses that the =
throughput that i get with one TCP stream is much lower than the =
available bandwidth. When i increase the number of TCP streams, for =
reasons well known, i get better bandwidth.
>=20
> I think, and correct me if i am wrong here, if i were to induce =
latency WITHOUT packet losses then even one TCP stream would be able to =
fill up the network pipe (assuming that the kernel buffers are not the =
limiting issue).

Not to be overly pedantic, but have you verified the packet losses - =
either through a packet trace on (I would think) the receiver or by =
looking at the netstat statistics on the sender before and after your =
run(s)?  Also, have you validated that indeed there is enough =93classic=94=
 TCP window to allow for 900 Mbit/s on a single stream at 120ms RTT?  My =
rough calculation based on:

Throughput <=3D WindowSize/RTT
900 Mbit/s <=3D WindowSize/0.120s
108 Mbit <=3D WindowSize

suggests a single stream with a 120 ms RTT would require a classic TCP =
window of 13,500,000 bytes or more even without any packet losses.  I =
don=92t think I=92ve seen a net.ipv4.tcp_[rw]mem which defaulted to more =
than 4-6MB (M =3D=3D 1048576) on the Linux systems I=92ve seen =
regardless of how much memory they had in them.

happy benchmarking,

rick jones


>=20
> Thanks, Toms
>=20
> On 23 May 2015 at 11:06, Gavin McCullagh <gmccullagh@gmail.com> wrote:
> Hi Tom,
>=20
> What did you use to create the tcp flows?
>=20
> Was it iperf or similar?
>=20
> Gavin
>=20
> On 23 May 2015 02:16, "Tom Sanders" <toms.sanders@gmail.com> wrote:
> Hi Mirja,
>=20
>=20
> the text below only says that if you have two cubic flows with a =
different base RTT competing at the same bottleneck they=92ll still =
converge to share the link equally. This does not mean that if you =
change the RTT strongly during one connection that Cubic is able to =
catch up fast.
>=20
> In fact this is one weakness of Cubic that I have observed. If there =
are change in base delay/rtt or strong increases in the available =
capacity, Cubic needs a rather long time to catch up. This is because =
cubic has a =82target value=91 to which it increases it sending =
rate/window quickly but then if the capacity cap is not somewhere close =
to this value (as expected) it will probe very conservatively around =
this value and it takes a rather long time until it gets back into a =
phase where it increases its congestion window fast.
>=20
> Then which congestion control algorithm fares better there? I thought =
Cubic was suited for large fat pipes -- bigger bandwidth-delay pipes.=20
> =20
>=20
> So what do you mean by 'I immediately saw that the TCP performance =
went down=91. How long do you run your test? And over which period of =
time this the performance go down?
>=20
> I did a simple test.
>=20
> I connect two machines on my 1GB LAN.I do a tcp data transfer test and =
i get around ~900Mbps. I then added a delay using qdisc on my outgoing =
ethernet interface to simulate a longer RTT. So, if my ping earlier was =
around 0.255ms, it then became 120ms. I repeat the test and i see that i =
can only send about 200Mbps. I had to use multiple TCP flows to fill up =
my 1GB link with Cubic. I run this experiment for about 2 minutes, and =
Cubic is not able to catch up. I dont have long standing flows, and only =
need to send bursts of data, hence did not leave it running for very =
long.
>=20
> I was under the impression, after having read the original Cubic paper =
that Cubic's window growth rate is independent of RTT and hence wasnt =
expecting any difference in the throughput.
>=20
> Toms.
>=20
> Mirja
>=20
>=20
> > Am 21.05.2015 um 02:51 schrieb Tom Sanders <toms.sanders@gmail.com>:
> >
> > Hi Neal,
> >
> > This is odd.
> >
> > In the original Cubic paper =
(http://www4.ncsu.edu/~rhee/export/bitcp/cubic-paper.pdf) by Rhee and =
Xu, they explicitly state that the Cubic is independent of RTT.
> >
> > "The main feature of CUBIC is that its window growth function is =
defined in real-time so that its growth will be independent of RTT. Our =
work was partially inspired by HTCP [5], whose window growth function is =
also based on real time. The congestion epoch period of CUBIC is =
determined by the packet loss rate alone. As TCP=92s throughput is =
defined by the packet loss rate as well as RTT, the throughput of CUBIC =
is defined by only the packet loss rate. Thus, when the loss rate is =
high and/or RTT is short, CUBIC can operate in a TCP mode. Moreover, =
since the growth function is independent of RTT, its RTT fairness is =
guaranteed as different RTT flows will still grow their windows at the =
same rate."
> >
> > Am i missing something here?
> >
> > Thanks, Toms
> >
> > On 20 May 2015 at 23:13, Neal Cardwell <ncardwell@google.com> wrote:
> > On Wed, May 20, 2015 at 1:18 PM, Tom Sanders =
<toms.sanders@gmail.com> wrote:
> > > Thanks Rick, this was very helpful.
> > >
> > > So if there are no setsockopts, etc then CUBIC's performance, =
unlike other
> > > TCP variants, should not get impacted by RTTs, right? Is that a =
fair
> > > statement to make?
> >
> > CUBIC, like most TCP congestion control variants, has significant
> > aspects that evolve on the scale of RTTs: for example, in slow =
start,
> > loss recovery, or when the CUBIC function is very steep, CUBIC's
> > behavior is heavily dependent on RTT (send some packets, wait for an
> > ACK, send some more packets,...). So CUBIC will not be
> > perfectly"RTT-fair.
> >
> > I think it might be fair to say something like: once a path is
> > saturated with CUBIC flows, as long as loss recovery or send/receive
> > buffers are not a bottleneck, then competition between CUBIC flows
> > should be more RTT-fair than Reno, in the long run, since for large
> > parts of the life cycle of the CUBIC flow the evolution of cwnd is
> > constrained by wall clock time rather than ACK arrival.
> >
> > neal
> >
> >
> > > Toms
> > >
> > > On 20 May 2015 at 19:51, Rick Jones <perfgeek@mac.com> wrote:
> > >>
> > >>
> > >> On May 20, 2015, at 6:20 AM, Tom Sanders <toms.sanders@gmail.com> =
wrote:
> > >>
> > >> > Hi,
> > >> >
> > >> > As per my understanding, unlike other TCP variants, CUBIC =
claims to be
> > >> > RTT fair. This is because the window growth rate is independent =
of the RTT,
> > >> > and hence the performance is the same as on a low and a high =
RTT paths.
> > >> > Others are dependent upon RTT since their windows grow as they =
receive ACKs
> > >> > from their peers, while CUBIC doesnt (window growth is a =
function of time).
> > >> > This is theory.
> > >> >
> > >> > In practice, CUBIC (which is the default now on most linux
> > >> > distributions) also depends upon RTT. I connected to machines =
on a LAN and
> > >> > did a FTP transfer. I got a certain bandwidth. Next, i =
artificially inserted
> > >> > delay using "tc qdisc" in the path. I increased the delay on =
the ethernet
> > >> > interface connecting my linux machine to the LAN to be around =
100ms. I
> > >> > immediately saw that the TCP performance went down.
> > >> >
> > >> > To bring it up to the same level as before i had to use =
multiple TCP
> > >> > streams. To me this means that CUBIC performance is dependent =
of the RTT. So
> > >> > why do we call it RTT-fair?
> > >>
> > >> I cannot speak to CUBIC specifically, but I would think you would =
want to
> > >> make sure you were seeing effects of congestion window and not of =
the
> > >> classic TCP window.  For example, if the FTP client/server you =
were using
> > >> makes an explicit setsockopt() to set the socket buffer sizes and =
by
> > >> extension the classic TCP window, your bumping of the RTT to =
around 100ms
> > >> may have taken the bandwidth down as a consequence of the =
classic:
> > >>
> > >> Throughput <=3D WindowSize/RTT
> > >>
> > >> Similarly, since you mention Linux, if the FTP client/server =
didn=92t make
> > >> explicit setsockopt() calls,  the sysctl values for tcp_rmem and =
tcp_wmem
> > >> may have been such that the auto tuning of the window size =
couldn=92t allow
> > >> the classic TCP window to grow =93enough.=94
> > >>
> > >> I suspect that the experiment needs to be setup to have the same =
classic
> > >> window size in both cases, sized per the above to be sufficient =
to achieve
> > >> the peak throughput in the high RTT case.  Then you will have =
eliminated
> > >> classic window as a factor.
> > >>
> > >> Also, depending on the version of the Linux kernel you are using, =
there
> > >> may be some effects from tcp small queues or perhaps (a stretch?) =
even byte
> > >> queue limits which may not be playing well with netem.  (Just =
guessing there
> > >> really)
> > >>
> > >> happy benchmarking,
> > >>
> > >> rick jones
> > >
> > >
> > >
> > >
> > > --
> > > Toms.
> > >
> > > _______________________________________________
> > > tcpm mailing list
> > > tcpm@ietf.org
> > > https://www.ietf.org/mailman/listinfo/tcpm
> > >
> >
> >
> >
> > --
> > Toms.
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
>=20
>=20
>=20
>=20
> --=20
> Toms.
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>=20
>=20
>=20
>=20
> --=20
> Toms.
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: TCP HyStart patch deployment
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Hi

Does anybody know if the TCP Cubic HyStart patch described in the link belo=
w is being used widely ?
It does not seem to be implemented in the latest Linux code.
http://patchwork.ozlabs.org/patch/85945/

Regards
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

"No man has a good enough memory
  to be a successful liar"
    Abraham Lincoln<http://www.brainyquote.com/quotes/authors/a/abraham_lin=
coln.html>
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<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Does anybody know if the TCP Cu=
bic HyStart patch described in the link below is being used widely ?<o:p></=
o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">It does not seem to be implemen=
ted in the latest Linux code.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><a href=3D"http://patchwork.ozlabs.org/patch/85945/"=
><span lang=3D"EN-US">http://patchwork.ozlabs.org/patch/85945/</span></a>
<span lang=3D"EN-US"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Ingemar<o:p></o:p></span></p>
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From: Neal Cardwell <ncardwell@google.com>
To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
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Subject: Re: [tcpm] TCP HyStart patch deployment
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On Tue, May 26, 2015 at 7:38 AM, Ingemar Johansson S
<ingemar.s.johansson@ericsson.com> wrote:
> Does anybody know if the TCP Cubic HyStart patch described in the link below
> is being used widely ?
>
> It does not seem to be implemented in the latest Linux code.
>
> http://patchwork.ozlabs.org/patch/85945/

It appears that roughly a week after that thread started, the
following patch was added to address that issue:

  http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=2b4636a5f8ca547000f6aba24ec1c58f31f4a91d

Then later on the following patch was added to further refine the
sensitivity of Hystart:

  http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=42eef7a0bb0989cd50d74e673422ff98a0ce4d7b

Hope that helps,
neal


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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Neal Cardwell <ncardwell@google.com>
Thread-Topic: [tcpm] TCP HyStart patch deployment
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From: Neal Cardwell <ncardwell@google.com>
To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
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Subject: Re: [tcpm] TCP HyStart patch deployment
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On Tue, May 26, 2015 at 10:09 AM, Ingemar Johansson S
<ingemar.s.johansson@ericsson.com> wrote:
> Thanks for the prompt response.
>
> Noted though that the first suggested first patch wants (http://patchwork=
.ozlabs.org/patch/85945/) to stretch things a bit further than what the lin=
ux.git patches suggest.
>
> I have implemented a HyStart + Cubic (with the linux.git patches) in our =
LTE system simulator. I seem to get issues that HyStart quite often seem to=
 exit too quickly. I have double and triple checked the code but don't find=
 any differences against the Linux code. One possible reason is that LTE in=
troduces an amount of scheduling jitter. Question is if this is what is cau=
sing the problem ?
>
> If I implement the original patch (85945) I seldom get early exits from H=
yStart, on the other hand I get larger overshoots, and that is of course no=
t optimal.

In your tests without patch 85945, is it the HYSTART_ACK_TRAIN or
HYSTART_DELAY mechanism that is triggering early exit of slow start?

Eric Dumazet found HYSTART_ACK_TRAIN triggers early exit often,
particularly if fq pacing is used. You may consider using just
HYSTART_DELAY.

What is the magnitude of the jitter and RTTs for which you are seeing probl=
ems?

The delay threshold of 2*min_rtt in the patch you are citing --
http://patchwork.ozlabs.org/patch/85945/ -- seems to allow a very big
overshoot before exiting slow start.

On the other hand, if you get good results for LTE and non-LTE
simulations for something like:

                        if (ca->curr_rtt > ca->delay_min +
                            HYSTART_DELAY_THRESH(ca->delay_min >> 2)) {
or
                        if (ca->curr_rtt > ca->delay_min +
                            HYSTART_DELAY_THRESH(ca->delay_min >> 1)) {

then that starts to sound potentially applicable for general use.

neal


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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Neal Cardwell <ncardwell@google.com>
Thread-Topic: [tcpm] TCP HyStart patch deployment
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Subject: Re: [tcpm] TCP HyStart patch deployment
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