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From: Mark Allman <mallman@icir.org>
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=20
Folks-

A quick note (chair hat off) to let you know that several of us have put
together a quick i-d on one particular strategy for mitigating
micro-bursts within TCP.  We submitted a very rough cut in November, but
have just cleaned it up quite a bit and it was sent to the I-D archives
this morning.  Until it pops out in the standard places you can slurp it
From=20the URL below.  The I-D is:

    Janardhan Iyengar, Mark Allman, Ethan Blanton.  TCP Burst Mitigation
    Through Congestion Window Limiting, January 2006. Internet-Draft
    draft-iyengar-burst-mitigation-01.txt (work in progress).
    http://www.icir.org/mallman/papers/draft-iyengar-burst-mitigation-01.txt

It'd be great to get people experimenting and thinking about whether
this is something that TCP needs or not.  There are some data points on
this contained in the references given in the I-D.  In any case, we'd
like to hear folks' thought on this one.

Thanks!

allman



=2D-=20
Mark Allman -- ICIR/ICSI -- http://www.icir.org/mallman/




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From tcpm-bounces@ietf.org Fri Jan 13 13:02:28 2006
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From: Ted Faber <faber@ISI.EDU>
To: tcpm@ietf.org
Subject: Re: [tcpm] ECN for SYN/ACK draft
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On Thu, Dec 08, 2005 at 02:39:17PM -0800, Ted Faber wrote:
> On Tue, Nov 29, 2005 at 11:40:11AM -0800, Ted Faber wrote:
> > Hi.
> >=20
> > As those of you who were at tsvwg in Vancouver know, Sally Floyd has
> > proposed a draft adding ECN to SYN and SYN/ACK packets, which was not
> > allowed under the original RFC 3168 ECN spec.  The draft has not yet
> > popped out, but there are also relatively-permanent local copies at:
> >=20
> >  http://www.icir.org/floyd/papers/draft-ietf-tsvwg-ecnsyn-00.txt
> >  http://www.icir.org/floyd/papers/draft-kuzmanovic-ecn-syn-00.txt
> [snip]
> >=20
> > If accepted, I'd like to request that we move fairly quickly to get this
> > reviewed and on the publication path.  I think sending it up to the IESG
> > for Proposed Standard before Dallas would be reasonable, but you all get
> > the final say.  If we decide not to accept it, clear reasons need to be
> > given.
> >=20
> > Comments?
>=20
> I realize that this is December and people have end-of-year holidays and
> end-of-semester chores taking their time, but Mark and I need some
> indication that this is a useful thing for us to do or it will go back
> to TSVWG.

OK, one more time.  It's no longer December.  This is a draft that TSVWG
originally took on, but is right in the TCPMi area.  The TSVWG and TCPM
chairs agree this is the right group to do this work, but that doesn't
matter if no one's interested in the work here.  I understand that
people from this group already hummed in TSVWG, but we need a show of
interest to take it on here in TCPM.  Please:

	* If you hummed in TSVWG and think it's good to do the work
	  here, send a note.  (Or if you don't think so, send that
	  note).  This needn't be a long message.
=09
	* Please read the draft and let us know what you think, whether
	  you've read it or not.  Obviously we'd be delighted to see the
	  longer discussion.

If the deafening silence continues, we'll have to send the draft back to
TSVWG and our case for doing new work here will get muddier.  Even if
you think TCPM is a dumb place to do such work, please make that an
actual statement rather than somethign we have to guess from a dead
silence.

Thanks!

--=20
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.=
asc
Unexpected attachment on this mail? See http://www.isi.edu/~faber/FAQ.html#=
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From tcpm-bounces@ietf.org Fri Jan 13 13:13:36 2006
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To: Ted Faber <faber@isi.edu>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] ECN for SYN/ACK draft 
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[speaking as an individual contributer]

I read through this I-D a week or so ago (along with the SIGCOMM paper
From=20which the idea came).  I think the general idea that ECT should be
allowed on the SYN+ACK is a good idea.  So, my opinion is that this
document should be a WG item.

(I am not sold on a host's response when a SYN+ACK is CE-ed by the
network that is currently given in the I-D.  I have iterated a little
with the authors about this and I may have more to say about this aspect
of the I-D in the future.  But, part of the WG process is in engineering
the scheme and so this sort of complaint has no bearing on whether I
think the draft ought to be a WG item.)

[chair hat on]

What Ted said.  More comments, please!

allman




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From tcpm-bounces@ietf.org Sun Jan 15 15:35:57 2006
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To: Ted Faber <faber@ISI.EDU>
Subject: Re: [tcpm] ECN for SYN/ACK draft
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Ted/all,

> 	* If you hummed in TSVWG and think it's good to do the work
> 	  here, send a note.  (Or if you don't think so, send that
> 	  note).  This needn't be a long message.

I somehow missed the last email asking for feedback on whether the draft 
belongs here---I had assumed that the draft was already in this WG. My 
bad, I forgot protocol. I think this work is meaningful as a TCPM WG item.

> 	* Please read the draft and let us know what you think, whether
> 	  you've read it or not.  Obviously we'd be delighted to see the
> 	  longer discussion.

I have read the draft, and it looks good to me except for one part that 
needs clarification. Section 4, last para:

    "As a final step, [ECN+] explored the co-existence of flows that do
    and don't set the ECN-capable codepoint in TCP SYN/ACK packets.  The
    results in [ECN+] confirmed that both types of flows can coexist;
    flows that apply the change improve their end-to-end performance,
    while the performance degradation for flows that don't apply the
    change, as a result of the flows that do apply the change, is
    marginal."

Is this comparison the one in Figure 10 in the SIGCOMM paper? If so, I'm 
not sure if it would be correct to say that the performance degradation 
for flows that don't apply the change is "marginal". I like "gradual" 
instead, as used in the paper. Maybe "gradually significant". Let me know 
if I'm way off.


Thanks!
jana


--------------------
Janardhan R. Iyengar
http://www.cis.udel.edu/~iyengar
Protocol Engineering Lab, University of Delaware


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Subject: Re: [tcpm] ECN for SYN/ACK draft
From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
To: Ted Faber <faber@ISI.EDU>
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I'm one who really intended to read the draft and respond to your call
earlier, but never got to it until now... Sorry!

On Fri, 2006-01-13 at 10:01 -0800, Ted Faber wrote:
> 	* If you hummed in TSVWG and think it's good to do the work
> 	  here, send a note.  (Or if you don't think so, send that
> 	  note).  This needn't be a long message.

I didn't see it a problem to have the draft in tsvwg, because there have
been also other drafts on explicit transport signaling, even when
primarily discussing TCP. But tcpm would be fine for me as well, since
the draft is after all quite specific to TCP, as you said earlier.

> 	* Please read the draft and let us know what you think, whether
> 	  you've read it or not.  Obviously we'd be delighted to see the
> 	  longer discussion.

To me the proposal seems pretty straightforward and well justified, and
I'd be happy to see this going forward without delays.

This sentence got me puzzled in the last paragraph of Section 2:

   "If the sending node (node B) was going to use an
   initial window of one segment, and receives an ECN-Echo packet
   informing it of a Congestion Experienced indication on its SYN/ACK
   packet, the sending node MAY continue to send with an initial window
   of one segment, without waiting for a retransmit timeout."

I don't have opinion for or against this right now, but having TCP
sender not slow down its sending rate in this particular case got me a
bit alarmed. On the other hand, I realize this is a pretty minor thing,
and one motivation of the draft is to avoid retransmit timeouts for
SYN/ACKs, but maybe the draft could then say that this is not thought to
be a significant concern for the network? (Especially since RFC 3168
argues that it is necessary to slow down the sending rate also when
cwnd=3D=3D1 and ECN-Echo arrives)

- Pasi


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From tcpm-bounces@ietf.org Tue Jan 17 21:40:48 2006
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Subject: [tcpm] WGLC: NCR draft-ietf-tcpm-tcp-dcr-06.txt ENDS 27 Jan 2006
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Hi.

I'd like to start a WG last call for the DCR document for protecting TCP
from reordering from Bhandarkar, Reddy, Allman, and Blanton.

	Title		: Improving the Robustness of TCP to
			  Non-Congestion Events
	Authors		: Sumitha Bhandarkar, et al.
	Filename	: draft-ietf-tcpm-tcp-dcr-06.txt
	Pages		: 17
	Date		: November 2005
	http://www.ietf.org/internet-drafts/draft-ietf-tcpm-tcp-dcr-06.txt

This draft is chartered to become Experimental.  I understand that all
the feedback to date, from the list and privately communicated, has been
incorporated into the document.  Further, I believe that the technical
opinions as expressed have been largely in favor of this draft becoming
Experimental.  Now is the time to speak up if you disagree or feel that
there are outstanding technical points that remain unaddressed in the
document.

Because this is starting mid-week, I'm extending the WGLC period for a
couple days.  Final day for comments is Friday 27, 2006 at midnight PST
(but I'm not going to be picky over an hour).

Please take some time to read it.  Please take some time to send
feedback to the list or to me.  As with all TCPM documents, there needs
to be a demonstration that the document has some support from the group,
so even a message that says "ship it" is helpful.

Also note that Mark is one of the authors, so I'm handling all the chair
stuff for this one, including an idnits pass and all the admin.=20

Thanks for your help.


--=20
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.=
asc
Unexpected attachment on this mail? See http://www.isi.edu/~faber/FAQ.html#=
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From tcpm-bounces@ietf.org Wed Jan 18 00:36:55 2006
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To: Janardhan Iyengar <iyengar@mail.eecis.udel.edu>
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Subject: Re: [tcpm] ECN for SYN/ACK draft 
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Janardhan -

Many thanks for the feedback.

>I have read the draft, and it looks good to me except for one part that 
>needs clarification. Section 4, last para:
>
>    "As a final step, [ECN+] explored the co-existence of flows that do
>    and don't set the ECN-capable codepoint in TCP SYN/ACK packets.  The
>    results in [ECN+] confirmed that both types of flows can coexist;
>    flows that apply the change improve their end-to-end performance,
>    while the performance degradation for flows that don't apply the
>    change, as a result of the flows that do apply the change, is
>    marginal."
>
>Is this comparison the one in Figure 10 in the SIGCOMM paper? If so, I'm 
>not sure if it would be correct to say that the performance degradation 
>for flows that don't apply the change is "marginal". I like "gradual" 
>instead, as used in the paper. Maybe "gradually significant". Let me know 
>if I'm way off.

I think that you are quite right.  We will revise that paragraph.
We are also planning on adding some simulations exploring worst-case
scenarios of the addition of ECN-Capability to some SYN/ACK packets
negatively affecting the performance of other traffic, just to be
quite clear about what could and couldn't happen.

Again, many thanks.

- Sally
http://www.icir.org/floyd/

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From tcpm-bounces@ietf.org Wed Jan 18 01:14:48 2006
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To: Pasi Sarolahti <pasi.sarolahti@iki.fi>
From: Sally Floyd <floyd@icir.org>
Subject: Re: [tcpm] ECN for SYN/ACK draft 
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Pasi -

>To me the proposal seems pretty straightforward and well justified, and
>I'd be happy to see this going forward without delays.
>
>This sentence got me puzzled in the last paragraph of Section 2:
>
>   "If the sending node (node B) was going to use an
>   initial window of one segment, and receives an ECN-Echo packet
>   informing it of a Congestion Experienced indication on its SYN/ACK
>   packet, the sending node MAY continue to send with an initial window
>   of one segment, without waiting for a retransmit timeout."
>
>I don't have opinion for or against this right now, but having TCP
>sender not slow down its sending rate in this particular case got me a
>bit alarmed. On the other hand, I realize this is a pretty minor thing,
>and one motivation of the draft is to avoid retransmit timeouts for
>SYN/ACKs, but maybe the draft could then say that this is not thought to
>be a significant concern for the network? (Especially since RFC 3168
>argues that it is necessary to slow down the sending rate also when
>cwnd=1 and ECN-Echo arrives)

Yep, Mark Allman also raised the same point.

At the moment, I have the following text added to the draft:

   Response to ECN-marking of SYN/ACK packets:
   One question is why TCP SYN/ACK packets should be treated differently
   from other packets in terms of the packet sender's response to an
   ECN-marked packet.  Section 6.1.2 of RFC 3168 specifies that when the
   TCP congestion window consists of a single packet and that packet is
   ECN-marked in the network, then the sender must reduce the sending
   rate below one packet per round-trip time, by waiting for one RTO
   before sending another packet.  If the RTO was set to the average
   round-trip time, this would result in halving the sending rate;
   because the RTO is in fact larger than the average round-trip time,
   the sending rate is reduced to less than half of its previous value.

   It might seem at first glance that the comparable response in reply
   to the ECN-marking of the SYN/ACK packet would be for the packet
   sender to reduce its sending rate by waiting for one RTO before
   sending the next packet.  However, this would not result in halving
   the sending rate; the packet sender has only sent one packet, the
   SYN/ACK packet, so there is no sending rate to halve.  If the RTO is
   set to its default initial value of three seconds, this would also
   not reduce the sending rate to one packet every two round-trip times,
   because the default three seconds could be much larger (or, in
   extreme cases, much smaller) than the actual round-trip time.

   One possibility would be for the packet sender to measure the round-
   trip time T between the sending of the SYN/ACK packet and the receipt
   of the acknowledgement, and to reply to the ECN-marking of the
   SYN/ACK packet by waiting T seconds before sending another packet.
   Conceptually, this could be thought of as halving the previous
   sending rate of one packet per round-trip time.  However, because
   there is no `sending rate' associated with the SYN/ACK packet, it
   seems to us just as plausible for the packet sender to set its
   initial congestion window to one packet after the ECN-marking
   of the SYN/ACK packet, and to proceed with a cautious sending rate of
   one packet per round-trip time.  If that packet is ECN-marked or
   dropped, then the sender will wait an RTO before sending another
   packet.

It seems convincing to me, but we are also going to add a
section about worst-case scenarios of the potential bad effects
of this on other traffic, just to be clear.

- Sally
http://www.icir.org/floyd/

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Subject: [tcpm] Last Call: 'A Roadmap for TCP Specification Documents' to 
 Informational RFC 
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The IESG has received a request from the TCP Maintenance and Minor Extensions 
WG to consider the following document:

- 'A Roadmap for TCP Specification Documents '
   <draft-ietf-tcpm-tcp-roadmap-05.txt> as an Informational RFC

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2006-02-01.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-tcp-roadmap-05.txt


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From: Ted Faber <faber@ISI.EDU>
To: tcpm@ietf.org
Subject: Re: [tcpm] ECN for SYN/ACK draft
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In light of the generally positive reaction on the list and in private
mail to the chairs and the complete lack of any negative indications,
TCPM will be taking the ECN for SYN/ACK draft on as a working group
item.  It will be included in the upcoming charter update.

Thanks for your input.

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To: Sally Floyd <floyd@icir.org>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] ECN for SYN/ACK draft 
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[speaking as an individual contributor, obviously]

Sally and I have had a bit of offline discussion about the response to a
CE-ed SYN+ACK.  My intention here is to gather my high-order points for
everyone to consider.

First, I'll react to the new text that Sally posted...

>    It might seem at first glance that the comparable response in reply
>    to the ECN-marking of the SYN/ACK packet would be for the packet
>    sender to reduce its sending rate by waiting for one RTO before
>    sending the next packet.  However, this would not result in halving
>    the sending rate; the packet sender has only sent one packet, the
>    SYN/ACK packet, so there is no sending rate to halve.  

I don't buy this argument.  If we consider two connections ... one that
has an SYN+ACK outstanding and one that has a data segment outstanding
then they have the same rate [*].  I don't understand how there could be
"no sending rate" in either of these cases.  And, it confuses me even
more than there could be a sending rate in one case and no sending rate
in the other case.  Maybe I am missing something here.

[*] In terms of packets / second or packets / RTT --- there could be a
    bps difference because of differing packet sizes.


The way I look at this, is as follows...

  * Exactly when we get to send the first data segment if the
    SYN+ACK is CE-ed is a very minor issue.  Probably you could do
    anything and nothing awful would happen.  Likewise, you could
    probably do anything on the first retransmit of a data segment
    when cwnd=1 in the middle of a connection and nothing awful
    would happen.

  * That said, I like consistency.  I don't like special cases if
    they can be avoided.  I'd rather just uniformly apply our
    algorithms and principles as much as possible.  (That said,
    clearly there are sometimes special cases and deviating when
    there are well laid out special cases seems OK to me.)

  * Therefore, I don't like the notion from the ecnsyn I-D that when
    a SYN+ACK is CE-ed the sender gets to use a cwnd of 1 segment
    and transmit immediately because RFC3168 clearly doesn't allow
    this in other cases when there is only one segment outstanding.

  * It seems dubious to me to standardize as acceptable the practice
    of advertising ECT on a packet when the sender of that ECT has
    no intention of actually slowing down when given a CE by the
    network.

  * Is there a difference in the SYN+ACK getting CE-ed and some
    random data packet getting CE-ed when the cwnd=1?  Yes.  Are the
    differences a reason to transmit immediately?  I don't think so.
    Let me work through two differences I see ...

    (1) If the cwnd is 1 in the middle of a connection there is probably
        a reason for it.  I.e., it has been beat down to 1 segment
        because of congestion.  So, not only do we have a cwnd of 1
        segment, but we have a little more history of congestion.  With
        the SYN+ACK getting CE-ed we do not have that history.

        This is all true.  However, just because there is no visible
        history when the SYN+ACK is CE-ed doesn't mean the network is
        not in the same state as when cwnd=1 in the middle of the
        connection.  We just don't know whether the SYN+ACK just got
        unlucky or the network is in a bad state.  Basically, we're
        presenting one segment to the network and we're getting the same
        signal back.  So, it seems to me that we ought to try to do
        about the same thing.  As with other things we have
        standardized, I think that conservativeness is the right
        approach in the face of the unknown.

    (2) The other difference is that during the 3WHS the RTO is set to
        the default RTO and has nothing to do with the actual network
        path over which the connection is traversing.

        The basic idea behind ECT-ing the SYN+ACK is to get around the
        default RTO (3sec) in cases when ECN could clearly be used.  I
        think this is a very nice idea.  So, simply saying that a CE-ed
        SYN+ACK incurs the default RTO really makes the change proposed
        in this I-D pretty much a moot point.  (It still helps a little
        in that the SYN+ACK wouldn't have to be resent after 3 seconds,
        but ...)  Further, subjecting the connection to an arbitrary
        delay seems pretty dubious.

        If the two response options were "send immediately" or "send
        after the default RTO" then I'd probably lean towards the former
        because the draft would be pretty useless if the latter was
        used, IMO.

        But, there is no reason to limit ourselves to those two options.
        The ACK of the SYN+ACK gives an RTT measurement, so we can be
        much smarter.  E.g., we could actually use this RTT measurement
        to seed the RTO per (2.2) in RFC2988 and then delay for RTO
        seconds before sending the first data segment.  

        There are some people who don't like the idea of seeding the RTO
        during the 3WHS because of the discrepancy between packet sizes
        during the 3WHS and the data transfer.  This certainly makes
        some sense.  But, even if an implementation didn't want to set
        the actual RTO during the 3WHS it could impose a delay per (2.2)
        of RFC2988 on transmission of the first data segment in response
        to a SYN+ACK getting CE-ed.

  * Using either of these methods of imposing delay between getting the
    CE and sending the next data segment seems pretty consistent with
    the way that this is all handled elsewhere to me.  In particular,
    ...

      + It doesn't seem aggressive in the face of the unknown.

      + It delays transmission very similarly to RFC3168.  I.e., we
        respond to the same signal from about the same amount of load in
        about the same way in all cases.

      + It does not impose the penalty of an arbitrary default RTO on
        connections that get the SYN+ACK CE-ed.

      + It very likely improves the situation over dropping the
        SYN+ACK.

    So, my vote is for a delay that is based on RFC2988 and the measured
    RTT.

    (And, I am happy to keep discussing this if someone has another
    approach that might help me understand the difference of these cases
    and why immediate transmission is an OK special case during the
    3WHS.)

allman




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From: Lars Eggert <lars.eggert@netlab.nec.de>
Subject: Re: [tcpm] WGLC: NCR draft-ietf-tcpm-tcp-dcr-06.txt ENDS 27 Jan 2006
Date: Fri, 20 Jan 2006 18:27:55 +0100
To: Ted Faber <faber@ISI.EDU>
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Hi,

On Jan 18, 2006, at 3:39, Ted Faber wrote:
>
> I'd like to start a WG last call for the DCR document for  
> protecting TCP
> from reordering from Bhandarkar, Reddy, Allman, and Blanton.
...
> Please take some time to read it.  Please take some time to send
> feedback to the list or to me.  As with all TCPM documents, there  
> needs
> to be a demonstration that the document has some support from the  
> group,
> so even a message that says "ship it" is helpful.

I gave it a quick read and I think it is in pretty good shape generally.

Given that the text repeatedly states that the impact of this on the  
general Internet is not really known, I assume this is going for the  
"experimental in the sense that we want to experiment with it"-flavor  
of Experimental? If so, stating prominently in the abstract and maybe  
introduction that this is not ready for production use may be a good  
idea.

One question regarding fairness: If there's a DCR flow and a non-DCR  
flow sharing a path that reorders packets a bit, wouldn't the  
reordering tolerance of the DCR flow cause it to steal bandwidth from  
the non-DCR flow? The paper says "it is to be noted that the reason  
for TCP-DCR realizing better throughputs (when packets are reordered)  
is not due to unfairness, but due to correctly recovering from the  
reordering events." Maybe I'm dense, but I don't see the experimental  
data to back this up. Is the throughput for SACK shown in Figure 4  
identical when competing only against other SACK flows? Is the  
dynamic behavior roughly the same?

Nits:

overall, the specification parts of the document could try to use  
RFC2119 terms a bit more explicitly

page 1: "mild reordering" - maybe _minor_ reordering?

page 3/4: "schemes proposed for differentiated services architecture"  
- for _a_ diff. serv. arch.?

page 4: "in the face of packet reordering three duplicate" - in the  
face of packet reordering, three duplicate

(Similarly missing commas in many places throughout the document;  
need to read many sentences twice to grok the meaning.)

page 10: s/algoritms in [RFC3517]/algorithms in [RFC3517]/

page 11: s/that the additonal buffer/that the additional buffer/

page 12/13: section 6 & its related work references could be removed  
for RFC publication

ref BPS99: s/is not pat hological/is not pathological/

ref KM02: s/twostage switche s/twostage switches/

Lars
--
Lars Eggert                                     NEC Network Laboratories


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From tcpm-bounces@ietf.org Fri Jan 20 15:17:49 2006
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To: Lars Eggert <lars.eggert@netlab.nec.de>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] WGLC: NCR draft-ietf-tcpm-tcp-dcr-06.txt ENDS 27 Jan 2006 
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Lars-

Thanks for the comments.  I'll fix up the nits, thanks!

> Given that the text repeatedly states that the impact of this on the
> general Internet is not really known, I assume this is going for the
> "experimental in the sense that we want to experiment with it"-flavor
> of Experimental? 

Yes.

In fact, the really experimental part of it to me is that using
Aggressive Limited Transmit delays the congestion response by an RTT.

(To a lesser degree even the Careful variant of Limited Transmit removes
the standard idle period between detecting "congestion" and resuming
transmission.  But, this seems sort of minor to me.)

> If so, stating prominently in the abstract and maybe introduction that
> this is not ready for production use may be a good idea.

Good call.  I'll add a note.

> One question regarding fairness: 

I am confused about what you're reading here.  I.e., the draft has no
figures, the quote below isn't from the draft and I don't think we even
discuss fairness in the i-d.

> If there's a DCR flow and a non-DCR flow sharing a path that reorders
> packets a bit, wouldn't the reordering tolerance of the DCR flow cause
> it to steal bandwidth from the non-DCR flow?

This is not clear to me.  In fact, it's not clear to me exactly how
fairness comes into this.  If there were 2 flows over some path and no
reordering then we'd expect them to see about the same performance
(about half the bandwidth).  Now, if there was reordering and neither
did NCR then they would each get less than half the bandwidth -- say
they each got 1/4 the bandwidth, just to put some numbers on things.
So, now the reordering is causing a 50% slowdown in the flows and less
than full utilization.  Now, if one of the flows were to use NCR and it
were to get more than 1/4 the bandwidth that would seem fine to me.
And, in fact, the NCR flow could get up to 3/4 of the capacity and I
wouldn't think that was unfair at all --- even though the rates are
quite different.  I.e., the NCR flow is using what is available.  Now,
if the NCR flow ended up with 7/8 of the capacity and drove the non-NCR
flow to 1/8 of the capacity ... well, then there is a question.  In some
sense, the NCR flow is stealing from the non-NCR flow, for sure.  Will
this happen?  I am not sure.  Is there a way to minimize it?  I don't
know.  Is this a natural part of trying to make progress and an
incentive for stacks to combat reordering?  Perhaps.  This could all be
part of the experiment. 

> The paper says "it is to be noted that the reason for TCP-DCR
> realizing better throughputs (when packets are reordered) is not due
> to unfairness, but due to correctly recovering from the reordering
> events." Maybe I'm dense, but I don't see the experimental data to
> back this up. Is the throughput for SACK shown in Figure 4 identical
> when competing only against other SACK flows? Is the dynamic behavior
> roughly the same?

Where is this from?

Thanks again!

allman




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Subject: Re: [tcpm] WGLC: NCR draft-ietf-tcpm-tcp-dcr-06.txt ENDS 27 Jan 2006 
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Hi,

On Jan 20, 2006, at 21:17, Mark Allman wrote:
>> One question regarding fairness:
>
> I am confused about what you're reading here.  I.e., the draft has no
> figures, the quote below isn't from the draft and I don't think we  
> even
> discuss fairness in the i-d.

oops, I meant to add that I got this from [BR04] before I hit  
"send." (I started thinking about the fairness stuff when reading the  
draft and then dug through the refs to find an evaluation.)

>> If there's a DCR flow and a non-DCR flow sharing a path that reorders
>> packets a bit, wouldn't the reordering tolerance of the DCR flow  
>> cause
>> it to steal bandwidth from the non-DCR flow?
>
> This is not clear to me.  In fact, it's not clear to me exactly how
> fairness comes into this.  If there were 2 flows over some path and no
> reordering then we'd expect them to see about the same performance
> (about half the bandwidth).

Yup.

>   Now, if there was reordering and neither
> did NCR then they would each get less than half the bandwidth -- say
> they each got 1/4 the bandwidth, just to put some numbers on things.

Yup.

> So, now the reordering is causing a 50% slowdown in the flows and less
> than full utilization.  Now, if one of the flows were to use NCR  
> and it
> were to get more than 1/4 the bandwidth that would seem fine to me.
> And, in fact, the NCR flow could get up to 3/4 of the capacity and I
> wouldn't think that was unfair at all --- even though the rates are
> quite different.  I.e., the NCR flow is using what is available.

Right - iff the other flow still got its 1/4. I'm wondering if this  
will always (or at least usually) be the case. What if we had a long,  
fat path where the non-DCR flow would need a long time to get back to  
1/2 - could the DCR flow actually delay that even more? Maybe not or  
not by much, but I'd still feel better if there was some sort of  
experimental results.

>   Now,
> if the NCR flow ended up with 7/8 of the capacity and drove the non- 
> NCR
> flow to 1/8 of the capacity ... well, then there is a question.  In  
> some
> sense, the NCR flow is stealing from the non-NCR flow, for sure.  Will
> this happen?  I am not sure.  Is there a way to minimize it?  I don't
> know.  Is this a natural part of trying to make progress and an
> incentive for stacks to combat reordering?  Perhaps.  This could  
> all be
> part of the experiment.

Right, that was sort of what popped up in my head when reading this.

>> The paper says "it is to be noted that the reason for TCP-DCR
>> realizing better throughputs (when packets are reordered) is not due
>> to unfairness, but due to correctly recovering from the reordering
>> events." Maybe I'm dense, but I don't see the experimental data to
>> back this up. Is the throughput for SACK shown in Figure 4 identical
>> when competing only against other SACK flows? Is the dynamic behavior
>> roughly the same?
>
> Where is this from?

Sumitha's Networking 2004 paper [BR04], but I just saw in Section 4  
that the mechanism may have changed since then - that pub was about DCR.

Also, note that I don't think this question is a showstopper for  
going forward with the draft as Experimental. It's one more question  
that should be looked at when experimenting with this.

Lars
--
Lars Eggert                                     NEC Network Laboratories


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From tcpm-bounces@ietf.org Fri Jan 20 16:32:15 2006
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Subject: Re: [tcpm] WGLC: NCR draft-ietf-tcpm-tcp-dcr-06.txt ENDS 27 Jan 2006
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Awesome footwork, Mark!

I've been out of the loop for a while, so can someone
please refresh me on the requirement for fairness.

I understand the requirement to prevent the next
Internet meltdown, but not a requirement for fairness.

Mark Allman wrote:
> 
>>If there's a DCR flow and a non-DCR flow sharing a path that reorders
>>packets a bit, wouldn't the reordering tolerance of the DCR flow cause
>>it to steal bandwidth from the non-DCR flow?
> 
> 
> This is not clear to me.  In fact, it's not clear to me exactly how
> fairness comes into this.  If there were 2 flows over some path and no
> reordering then we'd expect them to see about the same performance
> (about half the bandwidth).  Now, if there was reordering and neither
> did NCR then they would each get less than half the bandwidth -- say
> they each got 1/4 the bandwidth, just to put some numbers on things.
> So, now the reordering is causing a 50% slowdown in the flows and less
> than full utilization.  Now, if one of the flows were to use NCR and it
> were to get more than 1/4 the bandwidth that would seem fine to me.
> And, in fact, the NCR flow could get up to 3/4 of the capacity and I
> wouldn't think that was unfair at all --- even though the rates are
> quite different.  I.e., the NCR flow is using what is available.  Now,
> if the NCR flow ended up with 7/8 of the capacity and drove the non-NCR
> flow to 1/8 of the capacity ... well, then there is a question.  In some
> sense, the NCR flow is stealing from the non-NCR flow, for sure.  Will
> this happen?  I am not sure.  Is there a way to minimize it?  I don't
> know.  Is this a natural part of trying to make progress and an
> incentive for stacks to combat reordering?  Perhaps.  This could all be
> part of the experiment. 
> 
> 
>>The paper says "it is to be noted that the reason for TCP-DCR
>>realizing better throughputs (when packets are reordered) is not due
>>to unfairness, but due to correctly recovering from the reordering
>>events." Maybe I'm dense, but I don't see the experimental data to
>>back this up. Is the throughput for SACK shown in Figure 4 identical
>>when competing only against other SACK flows? Is the dynamic behavior
>>roughly the same?
> 
> 
> Where is this from?
> 
> Thanks again!
> 
> allman


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From tcpm-bounces@ietf.org Fri Jan 20 17:33:57 2006
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Subject: [tcpm] RFC 2581 question
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Hi,

I have a question about when to clear the duplicate ack counter.  RFC  
2581 says:

    The TCP sender SHOULD use the "fast retransmit" algorithm to detect
    and repair loss, based on incoming duplicate ACKs.  The fast
    retransmit algorithm uses the arrival of 3 duplicate ACKs (4
    identical ACKs without the arrival of any other intervening packets)
    as an indication that a segment has been lost.  After receiving 3
    duplicate ACKs, TCP performs a retransmission of what appears to be
    the missing segment, without waiting for the retransmission timer to
    expire.

I'm questioning the description as 3 duplicate ACKs as "4 identical  
ACKs without the arrival of any other intervening packets".  Let's  
take a normal fast retransmit case:

	dupack == 0
	[101,200] ACK(1) ---->
					<---- ACK(200)
	[201,300] ACK(1) ----> LOST
	[301,400] ACK(1) ---->
					<---- ACK(200)
	dupack == 1
	[401,500] ACK(1) ---->
					<---- ACK(200)
	dupack == 2
	[501,600] ACK(1) ---->
					<---- ACK(200)
	dupack == 3 => Fast Retransmit!
	[101,200] ACK(1) ---->
					<---- ACK(600)

Now, let's say the other side decided to send 10 bytes of data in the  
middle:

	dupack == 0
	[101,200] ACK(1) ---->
					<---- ACK(200)
	[201,300] ACK(1) ----> LOST
	[301,400] ACK(1) ---->
					<---- ACK(200)
	dupack == 1
	[401,500] ACK(1) ---->
					<---- ACK(200)
	dupack == 2
					User writes 10 bytes of data
					<---- [1,10] ACK(200)
	Got an intervening packet, so reset:
	dupack = 0
	ACK(10) ------------->
	[501,600] ACK(10) ---->
					<---- ACK(200)
	dupack == 1
	No Fast Retransmit!!!

It seems to me that the dupack counter should only be cleared when an  
ACK for new data is received, whether it is in a pure ACK or  
piggybacked on a data packet.  Just the receipt of something other  
than a dup ack shouldn't cause the duplicate ack count to get reset.

Am I missing something?

			-David Borman


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From tcpm-bounces@ietf.org Sat Jan 21 14:04:41 2006
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David,

> It seems to me that the dupack counter should only be cleared when an ACK for 
> new data is received, whether it is in a pure ACK or piggybacked on a data 
> packet.  Just the receipt of something other than a dup ack shouldn't cause 
> the duplicate ack count to get reset.

I took a quick look at the FreeBSD code, and I think you are right 
(someone correct me if this is not the case). I am also wondering what the 
reason for the clause "without any intervening packets" is.  I also do not 
see how your solution will work without SACK---how can a non-SACK sender 
determine when new data is being acked?

A sender could still "not do anything to the dupack count" when a 
piggybacked ack is received. In other words, the dupack count is changed 
on only pure acks and new acks. That is a fair estimate in this particular 
case, IMHO, that new data is being acked.

Anyone?

- jana

--------------------
Janardhan R. Iyengar 
http://www.cis.udel.edu/~iyengar
Protocol Engineering Lab, University of Delaware


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Subject: Re: [tcpm] WGLC: NCR draft-ietf-tcpm-tcp-dcr-06.txt ENDS 27 Jan 2006
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Mark Allman wrote:
> This is not clear to me.  In fact, it's not clear to me exactly how
> fairness comes into this.  If there were 2 flows over some path and no
> reordering then we'd expect them to see about the same performance
> (about half the bandwidth).  Now, if there was reordering and neither
> did NCR then they would each get less than half the bandwidth -- say
> they each got 1/4 the bandwidth, just to put some numbers on things.
> So, now the reordering is causing a 50% slowdown in the flows and less
> than full utilization.

Wait - you're arguing too far from an assumption, I think.

Does reordering have as bad an effect on a congested link
as on an uncongested link?  Or is it more like a 5% penalty
(congested) and 50% (congested).   If so, there's potential
for NCR to improve the situation for the flow using it to,
say, 3/4 of the bandwidth - leaving the non-NCR user only
getting 95% of 1/4 of the bandwidth.

Possibly this is what you mean later:

 >if the NCR flow ended up with 7/8 of the capacity and drove the non-NCR
 >flow to 1/8 of the capacity ... well, then there is a question.

and I didn't see the steps in your reasoning.  I'd call it unfair,
but justified.

In any case, I'm not arguing against the draft becoming Experimental.

Cheers,
     Jeremy Harris

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From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] WGLC: NCR draft-ietf-tcpm-tcp-dcr-06.txt ENDS 27 Jan 2006 
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> oops, I meant to add that I got this from [BR04] before I hit "send."
> (I started thinking about the fairness stuff when reading the draft
> and then dug through the refs to find an evaluation.)

Thanks.

I looked over the paper and I agree with you that the results in that
paper do not specifically address the question you're asking.

> Sumitha's Networking 2004 paper [BR04], but I just saw in Section 4
> that the mechanism may have changed since then - that pub was about
> DCR.

Right.

> Also, note that I don't think this question is a showstopper for going
> forward with the draft as Experimental. It's one more question that
> should be looked at when experimenting with this.

Great, thanks for the clarifications and the comments!

allman




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> I've been out of the loop for a while, so can someone
> please refresh me on the requirement for fairness.
> 
> I understand the requirement to prevent the next
> Internet meltdown, but not a requirement for fairness.

I am not sure there is a "requirement for fairness".  It's certainly a
little bit of a fuzzy issue in a case like NCR tries to combat.  For
instance, say over a given path two flows can get X bps given the
current congestion and reordering state of the network.  But, if NCR is
used on one flow and that flow gets 3*X bps and the first flow still
gets X bps is that unfair?  For sure the flows have different rates.
So, using something like Jain's fairness index (i.e., just measuring the
spread of rates amongst all flows) might say this is unfair.  But, to
me, neither flow is sending at an inappropriate rate for the given
context.  I.e., the flow sending at X bps cannot send any faster and the
flow sending at 3*X bps is just using what is available.

(Another example of this would be one flow using an advertised window of
Y and another using 5*Y and obtaining better performance.)

Of course, it gets thornier if the flow that gets 3*X bps ends up
stealing some of those bps from the other flow such that it sends at N*X
(for some N < 1).  Now, the reordering robust flow is not just using
spare capacity, but also getting some of its performance boost at the
expense of the non-NCR flow.  We might say that is unfair and say we
have to try to design something into NCR to avoid this - or, largely
mitigate it.  On the other hand, we might decide that the disadvantage
is not great and that this is part of the natural evolutionary path of
TCP, whereby older TCPs don't compete as well as newer TCPs and
therefore, we view this as an incentive to upgrade.  This would be a bit
of engineering judgment after we have a solid set of experiments to look
over, IMO.

allman




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Subject: Re: [tcpm] WGLC: NCR draft-ietf-tcpm-tcp-dcr-06.txt ENDS 27 Jan 2006 
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> Does reordering have as bad an effect on a congested link
> as on an uncongested link?  

My own guess is that on a congested link the congestion would have more
of an impact than reordering.  I think that if reordering ends up being
the dominant phenomenon then the path will not be (as) congested because
of the slow downs from false congestion signals.

But, this would clearly be something to look at with real measurements
and not just wave one's hands over as I am doing.

> In any case, I'm not arguing against the draft becoming Experimental.

Great, thanks!

allman




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On Mon, Jan 23, 2006 at 09:52:50AM -0500, Mark Allman wrote:
>=20
> I am not sure there is a "requirement for fairness".  It's certainly a
> little bit of a fuzzy issue in a case like NCR tries to combat.  For
> instance, say over a given path two flows can get X bps given the
> current congestion and reordering state of the network.  But, if NCR is
> used on one flow and that flow gets 3*X bps and the first flow still
> gets X bps is that unfair?  For sure the flows have different rates.
> So, using something like Jain's fairness index (i.e., just measuring the
> spread of rates amongst all flows) might say this is unfair.  But, to
> me, neither flow is sending at an inappropriate rate for the given
> context.  I.e., the flow sending at X bps cannot send any faster and the
> flow sending at 3*X bps is just using what is available.
>=20
> (Another example of this would be one flow using an advertised window of
> Y and another using 5*Y and obtaining better performance.)
>=20
> Of course, it gets thornier if the flow that gets 3*X bps ends up
> stealing some of those bps from the other flow such that it sends at N*X
> (for some N < 1).  Now, the reordering robust flow is not just using
> spare capacity, but also getting some of its performance boost at the
> expense of the non-NCR flow.  We might say that is unfair and say we
> have to try to design something into NCR to avoid this - or, largely
> mitigate it.  On the other hand, we might decide that the disadvantage
> is not great and that this is part of the natural evolutionary path of
> TCP, whereby older TCPs don't compete as well as newer TCPs and
> therefore, we view this as an incentive to upgrade.  This would be a bit
> of engineering judgment after we have a solid set of experiments to look
> over, IMO.

More compactly, the question is how closely NCR can achieve a Pareto
optimum, or max-min fairness, in competition with other types of flows.

-Wes

--=20
Wesley M. Eddy
Verizon Federal Network Systems

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From tcpm-bounces@ietf.org Mon Jan 23 14:34:45 2006
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Subject: [tcpm] draft-ietf-tcpm-ecnsyn-00
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I have submitted draft-ietf-tcpm-ecnsyn-00.txt to the internet-drafts
editor, replacing draft-ietf-tsvwg-ecnsyn-00.txt.
A copy is available at:
  http://www.icir.org/floyd/papers/draft-ietf-tcpm-ecnsyn-00.txt
  http://www.icir.org/floyd/papers/draft-ietf-tcpm-ecnsyn-00.ps

The changes are as follows:

   Changes from draft-ietf-twvsg-ecnsyn:

   * Changed name of draft to draft-ietf-tcpm-ecnsyn.

   * Added a discussion in Section 3 of "Response to
     ECN-marking of SYN/ACK packets".  Based on
     suggestions from Mark Allman.

   * Added a discussion to the Conclusions about adding
     ECN-capability to relevant set-up packets in other
     protocols.  From a suggestion from Wesley Eddy.

   * Added a description of SYN exchanges with SYN cookies.
     From a suggestion from Wesley Eddy.

   * Added a discussion of one-way data transfers, where the
     host sending the SYN/ACK packet sends no data packets.

   * Minor editing, from feedback from Mark Allman and Janardhan
     Iyengar.

   * Future work: a look at the costs of adding
     ECN-Capability in a worst-case scenario.
     From feedback from Mark Allman and Janardhan Iyengar.

   * Future work: a comparative evaluation of two
     possible responses to an ECN-marked SYN/ACK packet.

Many thanks for the feedback so far!

- Sally
http://www.icir.org/floyd/



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From tcpm-bounces@ietf.org Mon Jan 23 19:19:53 2006
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Subject: RE: [tcpm] RFC 2581 question
Date: Mon, 23 Jan 2006 16:17:34 -0800
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Thread-Topic: [tcpm] RFC 2581 question
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From: "Henderson, Thomas R" <thomas.r.henderson@boeing.com>
To: "Janardhan Iyengar" <iyengar@mail.eecis.udel.edu>,
	"David Borman" <david.borman@windriver.com>
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I seem to recall that, in the course of revising NewReno, we raised some
discussion with Mark about the ambiguity of what constitutes a dupack
(in our discussion at the time, we were noticing that pure window
updates were inappropriately resetting the NewReno dupack counter), and
I believe that he mentioned it as one of the issues to be taken up in
2581bis.

Tom

> -----Original Message-----
> From: Janardhan Iyengar [mailto:iyengar@mail.eecis.udel.edu]=20
> Sent: Saturday, January 21, 2006 11:05 AM
> To: David Borman
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] RFC 2581 question
>=20
> David,
>=20
> > It seems to me that the dupack counter should only be=20
> cleared when an=20
> > ACK for new data is received, whether it is in a pure ACK or=20
> > piggybacked on a data packet.  Just the receipt of something other=20
> > than a dup ack shouldn't cause the duplicate ack count to get reset.
>=20
> I took a quick look at the FreeBSD code, and I think you are=20
> right (someone correct me if this is not the case). I am also=20
> wondering what the reason for the clause "without any=20
> intervening packets" is.  I also do not see how your solution=20
> will work without SACK---how can a non-SACK sender determine=20
> when new data is being acked?
>=20
> A sender could still "not do anything to the dupack count"=20
> when a piggybacked ack is received. In other words, the=20
> dupack count is changed on only pure acks and new acks. That=20
> is a fair estimate in this particular case, IMHO, that new=20
> data is being acked.
>=20
> Anyone?
>=20
> - jana
>=20
> --------------------
> Janardhan R. Iyengar
> http://www.cis.udel.edu/~iyengar
> Protocol Engineering Lab, University of Delaware
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm
>=20

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From tcpm-bounces@ietf.org Mon Jan 23 21:39:43 2006
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From: Ethan Blanton <eblanton@cs.ohiou.edu>
To: Jian Liu <jliu@Brocade.COM>
Subject: Re: [tcpm] TCP duplicate ACK definition and detection, RFC 2581
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Jian Liu spake unto us the following wisdom:
> Summary: duplicate ACK detection is not documented in any ACK; BSD4.4
> (documented in "TCP/IP Illustrated Volume 2 The implementation") may
> not detect duplicate ACK if the traffice is bi-directional; propose a
> simple modification on reseting duplicate ACK count to reliably detect
> 3 duplicate ACK that triggers fast retransmit and fast recovery.

This is well-defined in the next rev of RFC2581.  Announcement
forthcoming...

Ethan

--=20
The laws that forbid the carrying of arms are laws [that have no remedy
for evils].  They disarm only those who are neither inclined nor
determined to commit crimes.
		-- Cesare Beccaria, "On Crimes and Punishments", 1764

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From tcpm-bounces@ietf.org Mon Jan 23 21:51:33 2006
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Date: Mon, 23 Jan 2006 21:51:28 -0500
From: Ethan Blanton <eblanton@cs.ohiou.edu>
To: David Borman <david.borman@windriver.com>
Subject: Re: [tcpm] RFC 2581 question
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Wow, I replied to an ancient email intending to reply to this one.
Just goes to show this question is common.  ;-)

David Borman spake unto us the following wisdom:
> I have a question about when to clear the duplicate ack counter.  RFC =20
> 2581 says:
>=20
>    The TCP sender SHOULD use the "fast retransmit" algorithm to detect
>    and repair loss, based on incoming duplicate ACKs.  The fast
>    retransmit algorithm uses the arrival of 3 duplicate ACKs (4
>    identical ACKs without the arrival of any other intervening packets)
>    as an indication that a segment has been lost.  After receiving 3
>    duplicate ACKs, TCP performs a retransmission of what appears to be
>    the missing segment, without waiting for the retransmission timer to
>    expire.

You are correct that this discription is ambiguous at best, and
incorrect at worst.  There is a forthcoming revision of RFC2581 which
(hopefully) clarifies this issue.  We look forward to comments on the
draft...

Ethan

--=20
The laws that forbid the carrying of arms are laws [that have no remedy
for evils].  They disarm only those who are neither inclined nor
determined to commit crimes.
		-- Cesare Beccaria, "On Crimes and Punishments", 1764

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Hi all,

Mark Allman, Vern Paxson and I are ready to push a revised version of
RFC2581 to the archives, draft-ietf-tcpm-rfc2581bis-00.txt .  This is
a charter goal for this working group, so we would love to have some
feedback on the issue.  Until this hits the archives:

http://www.icir.org/mallman/papers/draft-ietf-tcpm-rfc2581bis-00.txt

Note that the changes in this revision are specifically designed to be
well-understood and non-contentious, as the goal is to move this
document to Draft Standard.  (A summary of changes follows.)  In
general, standards track RFC changes since RFC2581 have been
integrated from elsewhere with very few other changes.  There are a
few clarifications (such as the definition of duplicate acknowledgment
recently discussed on this list), as well.

Please peruse this document when you get a chance (particularly if you
know of a particular clarity issue or outdated text in RFC2581; make
sure we took care of it!) and provide feedback to us.  On-list
feedback is most welcome, to stir community discussion.

7.  Changes Relative to RFC 2581

    A specific definition for "duplicate acknowledgment" has been
    added, based on the definition used by BSD TCP.

    The initial window requirements were changed to allow Larger
    Initial Windows as standardized in [RFC3390].  Additionally, the
    steps to take when an initial window is discovered to be too large
    due to Path MTU Discovery [RFC1191] are detailed.

    The recommended initial value for ssthresh has been changed to say
    that it SHOULD be arbitrarily high, where it was previously MAY.
    This is to provide additional guidance to implementors on the
    matter.

    During slow start, the usage of Appropriate Byte Counting [RFC3465]
    with L=1*SMSS is explicitly recommended.  The method of increasing
    cwnd given in [RFC2581] is still explicitly allowed.  Byte counting
    during congestion avoidance is also recommended, while the method
    from [RFC2581] and other safe methods are still allowed.

    The treatment of ssthresh on retransmission timeout was clarified.
    Specifically, Equation (3) from [RFC2581] was split into Equations
    (4) and (5) in this document.

    The description of fast retransmit and fast recovery has been
    clarified, and the use of Limited Transmit [RFC3042] is now
    recommended.

    The restart window has been changed to min(IW,cwnd) from IW.  This
    behavior was described as "experimental" in [RFC2581].

    It is now recommended that TCP implementors implement an advanced
    loss recovery algorithm conforming to the principles outlined in
    this document.

    The security considerations have been updated to discuss ACK
    division and recommend byte counting as a counter to this attack.

Ethan

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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		: Adding Explicit Congestion Notification (ECN)
                          Capability to TCP's SYN/ACK Packets
	Author(s)	: A. Kuzmanovic, et al.
	Filename	: draft-ietf-tcpm-ecnsyn-00.txt,.ps
	Pages		: 16
	Date		: 2006-1-24
	
   This draft specifies a modification to RFC 3168 to allow TCP SYN/ACK
   packets to be ECN-Capable.  For TCP, RFC 3168 only specified setting
   an ECN-Capable codepoint on data packets, and not on SYN and SYN/ACK
   packets.  However, because of the high cost to the TCP transfer of
   having a SYN/ACK packet dropped, with the resulting retransmit
   timeout, this document is specifying the use of ECN for the SYN/ACK
   packet itself, when sent in response to a SYN packet with the two ECN
   flags set in the TCP header, indicating a willingness to use ECN.
   Setting TCP SYN/ACK packets as ECN-Capable can be of great benefit to
   the TCP connection, avoiding the severe penalty of a retransmit
   timeout for a connection that has not yet started placing a load on
   the network.  The sender of the SYN/ACK packet must respond to an ECN
   mark by reducing its initial congestion window from two, three, or
   four segments to one segment, reducing the subsequent load from that
   connection on the network.

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Subject: Re: [tcpm] draft on congestion window limiting
From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Hi,

Some comments on the burst mitigation draft below...

On Wed, 2006-01-11 at 15:09 -0500, Mark Allman wrote:
>     Janardhan Iyengar, Mark Allman, Ethan Blanton.  TCP Burst Mitigation
>     Through Congestion Window Limiting, January 2006. Internet-Draft
>     draft-iyengar-burst-mitigation-01.txt (work in progress).
>     http://www.icir.org/mallman/papers/draft-iyengar-burst-mitigation-01.=
txt
>=20
> It'd be great to get people experimenting and thinking about whether
> this is something that TCP needs or not.  There are some data points on
> this contained in the references given in the I-D.  In any case, we'd
> like to hear folks' thought on this one.

I believe experimenting with burst mitigation algorithms in real
networks could be possible for many of us. At least Linux has
implemented a similar scheme for years, though I don't know what's the
status of TCP in the latest kernel versions.

(Btw, the Linux scheme is shortly outlined on page 6 of
P. Sarolahti and A. Kuznetsov. "Congestion Control in Linux TCP". In
Proceedings of Usenix/Freenix 2002. Monterey, CA, USA, June 2002.=20
[http://www.cs.helsinki.fi/u/sarolaht/papers/linuxtcp.pdf])

We did some experimentations with Linux TCP on slow links and a limited
set of network paramaters about five years ago, and from what I
remember (with the experience of an undergraduate student), burst
mitigation didn't seem to cause problems, but not huge benefits,
either.

I think it would be worthwhile to discuss the trade-offs a bit
more. Discussion section says that whether or not mitigation is needed
remains open, but it could give some more baits for further
thinking. For instance, while burst mitigation can reduce the stress
on network queues to some extent, much research has been done on how
to utilize a high-delay path as effectively as possible within the
limits of congestion control principles. People working with
high-delay paths might want to evaluate carefully whether
reductions on congestion window are more harmful than the temporary=20
bursts of load on queues.

I was also thinking about the fairness and incentiveness of burst
mitigation as another kind of trade-off. I cannot come up with very good
case that would illustrate this well, but let's assume there was a
receiver with a buggy delayed ack implementation (say, ignoring the
"ack-on-every-second" rule -- I have seen this sometimes), and two
senders on the other side of the network, one implementing burst
mitigation and another not. It would seem that burst mitigation would
give unnecessary advantage to the flow that does not implement it. This
could be a minor effect, but on the other hand, it is not known how
significant the benefit for avoiding micro-bursts is.

But to me the algorithm itself on Section 2 seems straightforward, and
having BLimit to follow initial congestion window equation feels an
appropriate choice. So if there was some consensus that burst
mitigation is generally needed, I could not see what could go
wrong with the algorithm as described in the draft.

Other than that, I'm only able to come up with small nits:

* I was thinking if the beginning of section 2 should emphasize it a bit
more that burst mitigation is in an experimental landscape. Maybe in
addition than just saying "When using CWL, the following steps MUST be
executed", one could also add that CWL is experimental and TCP MAY
implement it, and when using CWL, the following steps MUST be
executed, just to make it clear. Well, I don't know. This is a very
minor thing, and probably a matter of opinon.

* Section 3 begins with:
    CWL makes TCP congestion control more conservative and is therefore
    implicitly allowed by [RFC2581].
Rather than being in the related work section, this note could also be
somewhere in section 2.

Don't know how much these thoughts help, but hopefully they at least
spring up more comments ;-)

- Pasi


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From tcpm-bounces@ietf.org Wed Jan 25 21:32:04 2006
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Subject: RE: [tcpm] RFC 2581 question
Date: Wed, 25 Jan 2006 18:31:27 -0800
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From: "Krejsa, Dan" <dan.krejsa@windriver.com>
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Hi Ethan,

I like the duplicate ACK definition in the new draft, however
I'm not sure the draft addresses David's particular question
regarding the phrase 'without the arrival of any other intervening
packets'.  What is the reason for this restriction?  If
a segment arrives which is not a 'duplicate ACK' according
to the definition, but which does not acknowledge new data,
should the duplicate ACK count really be zeroed as this phrase
seems to imply?

- Dan

-----Original Message-----
From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
Ethan Blanton
Sent: Monday, January 23, 2006 6:51 PM
To: Borman, David
Cc: tcpm@ietf.org
Subject: Re: [tcpm] RFC 2581 question

Wow, I replied to an ancient email intending to reply to this one.
Just goes to show this question is common.  ;-)

David Borman spake unto us the following wisdom:
> I have a question about when to clear the duplicate ack counter.  RFC

> 2581 says:
>=20
>    The TCP sender SHOULD use the "fast retransmit" algorithm to detect
>    and repair loss, based on incoming duplicate ACKs.  The fast
>    retransmit algorithm uses the arrival of 3 duplicate ACKs (4
>    identical ACKs without the arrival of any other intervening
packets)
>    as an indication that a segment has been lost.  After receiving 3
>    duplicate ACKs, TCP performs a retransmission of what appears to be
>    the missing segment, without waiting for the retransmission timer
to
>    expire.

You are correct that this discription is ambiguous at best, and
incorrect at worst.  There is a forthcoming revision of RFC2581 which
(hopefully) clarifies this issue.  We look forward to comments on the
draft...


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Krejsa, Dan spake unto us the following wisdom:
> I like the duplicate ACK definition in the new draft, however
> I'm not sure the draft addresses David's particular question
> regarding the phrase 'without the arrival of any other intervening
> packets'.  What is the reason for this restriction?  If
> a segment arrives which is not a 'duplicate ACK' according
> to the definition, but which does not acknowledge new data,
> should the duplicate ACK count really be zeroed as this phrase
> seems to imply?

Yes, this was brought up to us by another reader off-list ... I
thought this was cleaned up when the dupack definition was added, but
I have now seen that it was not.  I have proposed the following text
for the second paragraph of 3.2:

    The TCP sender SHOULD use the "fast retransmit" algorithm to
    detect and repair loss, based on incoming duplicate ACKs.  The
    fast retransmit algorithm uses the arrival of 3 duplicate ACKs (as
    defined in section 2, without any intervening ACKs which move
    SND.UNA) as an indication that a segment has been lost.  After
    receiving 3 duplicate ACKs, TCP performs a retransmission of what
    appears to be the missing segment, without waiting for the
    retransmission timer to expire.

Does this look better to you?

Ethan

--=20
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for evils].  They disarm only those who are neither inclined nor
determined to commit crimes.
		-- Cesare Beccaria, "On Crimes and Punishments", 1764

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From tcpm-bounces@ietf.org Wed Jan 25 22:00:33 2006
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Ethan Blanton Wrote:
> I like the duplicate ACK definition in the new draft, however
> I'm not sure the draft addresses David's particular question
> regarding the phrase 'without the arrival of any other intervening
> packets'.  What is the reason for this restriction?  If
> a segment arrives which is not a 'duplicate ACK' according
> to the definition, but which does not acknowledge new data,
> should the duplicate ACK count really be zeroed as this phrase
> seems to imply?

  Yes, this was brought up to us by another reader off-list ... I
  thought this was cleaned up when the dupack definition was added, but
  I have now seen that it was not.  I have proposed the following text
  for the second paragraph of 3.2:

    The TCP sender SHOULD use the "fast retransmit" algorithm to
    detect and repair loss, based on incoming duplicate ACKs.  The
    fast retransmit algorithm uses the arrival of 3 duplicate ACKs (as
    defined in section 2, without any intervening ACKs which move
    SND.UNA) as an indication that a segment has been lost.  After
    receiving 3 duplicate ACKs, TCP performs a retransmission of what
    appears to be the missing segment, without waiting for the
    retransmission timer to expire.

  Does this look better to you?

Yup, I think it does!

- Dan

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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		: TCP Congestion Control
	Author(s)	: M. Allman, et al.
	Filename	: draft-ietf-tcpm-rfc2581bis-00.txt
	Pages		: 15
	Date		: 2006-1-26
	
    This document defines TCP's four intertwined congestion control
    algorithms: slow start, congestion avoidance, fast retransmit, and
    fast recovery.  In addition, the document specifies how TCP should
    begin transmission after a relatively long idle period, as well as
    discussing various acknowledgment generation methods.

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Jana -

>We would appreciate it if you could find the time/energy to review this 
>draft. Though we'd prefer a public review (sent to TCPM), we will value 
>your review as much even if you'd rather send it to us in private.

Here it is.  Thanks for writing this draft, and hopefully starting to
get some progress on these issues.

- Sally
http://www.icir.org/floyd/


Title: TCP Burst Mitigation Through Congestion Window Limiting
Authors: Iyengar, Allman, and Blanton.

I liked this draft, and think it is useful.  And, though it makes
a small change, I think that the mechanism proposed in the draft
is a big improvement over "Use It or Lose it".

First the high-level comments, about active queue management;
rate-based pacing; and security considerations:

* The network's tolerance of burstiness:
The first paragraph of the Introduction says the following: 
  "Bursting can stress network queues causing loss in the bursting
  connection as well as in other flows sharing the stressed queues."

I think it would be useful to add that another important change
that is happening in parallel, to some (unknown) extent, is for
congested routers to use some form of Active Queue Management,
allowing queues in the congested routers to accomodate transient
bursts in the arriving traffic.

Reason 1 for using AQM: Congestion control is more effective if
queues don't giving congestion feedback (in the form of packet
drops) in response to small transient bursts in the traffic.

Reason 2 for using AQM: Even if burst mitigation and rate-based
pacing and all other things are done at the end-nodes, there can
still be burstiness in the aggregate traffic inside the network,
and it would be good for queues to be tolerant of this.

* Rate-based pacing:
The document says that rate-based pacing could be quite effective,
but the implementation can be costly (timers and such).  I think
it would be worth it to say that there are many cases where rate-based
pacing should be a big win over CWL, or over MaxBurst, in terms of
traffic dynamics.  E.g., in a short connection with only BLimit + 1 
more packets to send, but when only one ack will be arriving, due
to ACK losses.  Or when dealing with the significant transient
bursts that can be caused by a connection slow-starting up to a
large window.  CWL and MaxBurst don't prevent the transient bursts
of slow-start at all, but rate-based pacing could work well.  

Of course, it is still a question of whether the benefits of
rate-based pacing are worth the costs in terms of implementation
overhead.  And there is still the old paper by Stefan Savage about
rate-based pacing to deal with, about how rate-based pacing isn't
always a win.  But my suggestion would be to be more explicit about
the potential advantages of rate-based pacing, and to say something about it
a little earlier in the document.

(My current assumption is that connections slow-starting up to large
congestion windows have a large effect on the burstiness of the
aggregate traffic.  I forget, however, what some of the papers about
burstiness-on-small-time-scales have to say about this...)

* Security:  
I think that the security issues raised in Section 5 are pretty
important, and could use more discussion.  This is another case
where rate-based pacing would be more robust than CWL, I think.

Details:

* Causes of micro-bursts:
ACK-compression can also be a cause of micro-bursts.  (That is,
ACK-compression can cause micro-bursts as well as macro-bursts.)

* Section 2, step (1):
"the only case where a micro-burst will not occur" ->
"the only case where a micro-burst will not occur, if steps (2)
  and (3) aren't used".

* "BLimit SHOULD be chosen such that bursts are no larger than those
allowed by [RFC3390]."

Why is this?  It is not obvious to me.  For example, four-packet bursts,
for 1500-byte packets, might be perfectly acceptable, for a connection
that has a sufficiently large congestion window to send that many packets.  
And there is a cost to BLimit being at most three 1500-byte packets.

* Maxburst:
"An additional drawback of MaxBurst ...".
What was the first drawback?  Introducing a separate control?  But
the earlier sentence says that "using two different controls may
make sense".

"the two transmission controllers may interact poorly, causing
undesirable side effects."
Do you have any examples of this?  Or citations?  Or anything
else to back this up?

"When BLimit == MaxBurst, CWL and MaxBurst perform similarly [AB05]."
I don't see how CWL and MaxBurst can perform similarly, and at
the same time you can prefer one over the other.  I think it would
be useful to look a little more closely for performance differences,
or to say that there aren't any.

Depending on how it is implemented, MaxBurst could be made to work
quite well with re-ordering, where each duplicate ack packet (possibly
from reordering of data packets) could be responded-to with up to
MaxBurst data packets (if allowed by the congestion window).
(E.g., Aggressive MaxBurst from [AB05].)
Can something like this be done with CWL as well?

* I didn't go back and re-read [AB05]. so if I have forgotten
something that is answered in that document, my apologies...


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On Tue, Jan 17, 2006 at 06:39:19PM -0800, Ted Faber wrote:
>=20
> Hi.
>=20
> I'd like to start a WG last call for the DCR document for protecting TCP
> from reordering from Bhandarkar, Reddy, Allman, and Blanton.
>=20
> 	Title		: Improving the Robustness of TCP to
> 			  Non-Congestion Events
> 	Authors		: Sumitha Bhandarkar, et al.
> 	Filename	: draft-ietf-tcpm-tcp-dcr-06.txt
> 	Pages		: 17
> 	Date		: November 2005
> 	http://www.ietf.org/internet-drafts/draft-ietf-tcpm-tcp-dcr-06.txt
>=20
> This draft is chartered to become Experimental.  I understand that all
> the feedback to date, from the list and privately communicated, has been
> incorporated into the document.  Further, I believe that the technical
> opinions as expressed have been largely in favor of this draft becoming
> Experimental.  Now is the time to speak up if you disagree or feel that
> there are outstanding technical points that remain unaddressed in the
> document.
>=20
> Because this is starting mid-week, I'm extending the WGLC period for a
> couple days.  Final day for comments is Friday 27, 2006 at midnight PST
> (but I'm not going to be picky over an hour).

Having watched the discussion and spoken with Mark about any off-list
interaction, it seems clear to me this dosument has passed WGLC.  Minor
issues were raised and dealt with during the WGLC, but no showstoppers
surfaced.  No one argued that the draft should not move to experimental.

If I've missed comments, or you disagree with my assessment of
consensus, I'd like to hear ASAP.

Thanks to everyone who participated in the creation and review of this
document.

--=20
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.=
asc
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From tcpm-bounces@ietf.org Mon Jan 30 17:17:16 2006
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Pasi,

Thank you for your comments, and for spending your time on this document.

> I believe experimenting with burst mitigation algorithms in real
> networks could be possible for many of us. At least Linux has
> implemented a similar scheme for years, though I don't know what's the
> status of TCP in the latest kernel versions.

Thank you for this bit of information. The scheme mentioned in the paper 
looks like the "Use it or Lose it" scheme mentioned in the draft. I did 
probe the Linux community recently about whether a micro-burst mitigation 
scheme was deployed in Linux. The response I got was that there wasn't any 
such scheme currently deployed, but patches were available. I am not sure 
at this time if there is or isn't a scheme in Linux (I haven't looked at 
the code myself yet). I would be happy if someone could confirm for me 
which one is currently the case.

> I think it would be worthwhile to discuss the trade-offs a bit
> more.

Agree, and Good point. I did have this in my mind, but I should have put 
it in the draft. Thanks for the tradeoff suggestions.

> I was also thinking about the fairness and incentiveness of burst
> mitigation as another kind of trade-off. I cannot come up with very good
> case that would illustrate this well, but let's assume there was a
> receiver with a buggy delayed ack implementation (say, ignoring the
> "ack-on-every-second" rule -- I have seen this sometimes), and two
> senders on the other side of the network, one implementing burst
> mitigation and another not. It would seem that burst mitigation would
> give unnecessary advantage to the flow that does not implement it. This
> could be a minor effect, but on the other hand, it is not known how
> significant the benefit for avoiding micro-bursts is.

As far as the buggy receiver is concerned, I'm not sure that we should aim 
to demonstrate trade-offs in such cases. But, IIUC, your core point is 
that under the same conditions, a sender with burst-mitigation might be at 
a disadvantage with respect to a sender without burst-mitigation. There is 
also the possibility that a sender without burst-mitigation loses packets 
in the network due to the burst when a sender with burst-mitigation does 
not. So, I agree that there is a tradeoff. And that we should write it out 
clearly.

> So if there was some consensus that burst mitigation is generally 
> needed,

We are trying to gather data points on this question too. Our goal in this 
draft is to gauge interest in the issue, and to give a starting point for 
experimentation.

> * I was thinking if the beginning of section 2 should emphasize it a bit
> more that burst mitigation is in an experimental landscape. Maybe in

We did try to emphasize exactly that (just before Section 2, in fact). 
Maybe it did not come through clearly enough.. I'll try to make it 
clearer. Section 2 specifies the mechanics of the algorithm itself and I'm 
not sure if that space should be used for making a statement about the 
experimental status of burst-mitigation. But, point taken.


> * Section 3 begins with:
>    CWL makes TCP congestion control more conservative and is therefore
>    implicitly allowed by [RFC2581].
> Rather than being in the related work section, this note could also be
> somewhere in section 2.

Does Section 4 (Discussion) seem more appropriate?

Thank you for your comments!

regards,
jana

--------------------
Janardhan R. Iyengar
http://www.cis.udel.edu/~iyengar
Protocol Engineering Lab, University of Delaware


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From tcpm-bounces@ietf.org Mon Jan 30 19:51:19 2006
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Subject: RE: [tcpm] RFC2581bis
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Ethan,

I have concerns about setting ssthresh after multiple timeouts (Equation
5).  My initial feeling is that cutting ssthresh in half each time is
too conservative. Let's look at some potential causes of this, and what
we might do with perfect knowledge:

(1) Very severe congestion, causing frequent packet loss due to
	(a) temporary traffic spike: as long as this state exists, we're
highly unlikely to get cwnd to the point where ssthresh matters. Later
recovery will be hampered by a very low ssthresh.  We'd probably prefer
the pre-congestion ssthresh.
	(b) longer-term increase in traffic or reduction in link data
rate: We'd want to reduce ssthresh, but perhaps not asymptotically to
2*MSS.
(2) Link outage -- ideally, ssthresh should not change at all.
(3) Routing change -- preferably, we would reset ssthresh to its default
value to probe the new path.

(1b) is the only case where we'd clearly want to reduce ssthresh, and
it's not clear to me that flightsize/2 is not enough.  So I'm not sure
there's that much upside to being that conservative, and there's
certainly downside in outage-prone wireless and satellite environments.

I understand that we're not engineering to special cases.  However, when
we have the opportunity to refine standards to make TCP a little more
wireless friendly, I think we should do so, especially when costs are
low and benefits are high.  As late as 2003 (and maybe still?), the
Linux stack halved ssthresh only after the first retransmit, and that
didn't appear to have particularly negative effects on internet
performance.  (The Linux stack behavior, and the effect on outage-prone
wireless, are documented in our paper in ACM CCR Jun '04).

So as an alternative, perhaps we should leave ssthresh unchanged after
the first retransmit.

Martin

-----Original Message-----
From: Ethan Blanton [mailto:eblanton@cs.ohiou.edu]=20
Sent: Monday, January 23, 2006 7:23 PM
To: tcpm@ietf.org
Cc: vern@icir.org; mallman@icir.org
Subject: [tcpm] Revising RFC2581: draft-ietf-tcpm-rfc2581bis-00.txt

Hi all,

Mark Allman, Vern Paxson and I are ready to push a revised version of
RFC2581 to the archives, draft-ietf-tcpm-rfc2581bis-00.txt .  This is a
charter goal for this working group, so we would love to have some
feedback on the issue.  Until this hits the archives:

http://www.icir.org/mallman/papers/draft-ietf-tcpm-rfc2581bis-00.txt

Note that the changes in this revision are specifically designed to be
well-understood and non-contentious, as the goal is to move this
document to Draft Standard.  (A summary of changes follows.)  In
general, standards track RFC changes since RFC2581 have been integrated
from elsewhere with very few other changes.  There are a few
clarifications (such as the definition of duplicate acknowledgment
recently discussed on this list), as well.

Please peruse this document when you get a chance (particularly if you
know of a particular clarity issue or outdated text in RFC2581; make
sure we took care of it!) and provide feedback to us.  On-list feedback
is most welcome, to stir community discussion.

7.  Changes Relative to RFC 2581

    A specific definition for "duplicate acknowledgment" has been
    added, based on the definition used by BSD TCP.

    The initial window requirements were changed to allow Larger
    Initial Windows as standardized in [RFC3390].  Additionally, the
    steps to take when an initial window is discovered to be too large
    due to Path MTU Discovery [RFC1191] are detailed.

    The recommended initial value for ssthresh has been changed to say
    that it SHOULD be arbitrarily high, where it was previously MAY.
    This is to provide additional guidance to implementors on the
    matter.

    During slow start, the usage of Appropriate Byte Counting [RFC3465]
    with L=3D1*SMSS is explicitly recommended.  The method of increasing
    cwnd given in [RFC2581] is still explicitly allowed.  Byte counting
    during congestion avoidance is also recommended, while the method
    from [RFC2581] and other safe methods are still allowed.

    The treatment of ssthresh on retransmission timeout was clarified.
    Specifically, Equation (3) from [RFC2581] was split into Equations
    (4) and (5) in this document.

    The description of fast retransmit and fast recovery has been
    clarified, and the use of Limited Transmit [RFC3042] is now
    recommended.

    The restart window has been changed to min(IW,cwnd) from IW.  This
    behavior was described as "experimental" in [RFC2581].

    It is now recommended that TCP implementors implement an advanced
    loss recovery algorithm conforming to the principles outlined in
    this document.

    The security considerations have been updated to discuss ACK
    division and recommend byte counting as a counter to this attack.

Ethan

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