From owner-tcp-impl@cthulhu.engr.sgi.com  Wed Oct  7 18:18:06 1998
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Date: Wed, 07 Oct 1998 15:08:12 -0700
From: "Amy S. Hughes" <ahughes@ISI.EDU>
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Where can I find archives for the tcp-impl mailing list?
The official one listed in the charter,
ftp://ftp.sgi.com/other/tcp-impl/mail.archive 
only has posts upto April 1997.  I need to look for some
more recent than that.

thanks,
Amy S. Hughes


From owner-tcp-impl@cthulhu.engr.sgi.com  Wed Oct  7 19:03:11 1998
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To: "Amy S. Hughes" <ahughes@ISI.EDU>
Cc: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: tcp-impl archives?
In-reply-to: Your message of Wed, 07 Oct 1998 15:08:12 PDT.
Date: Wed, 07 Oct 1998 15:58:02 PDT
From: Vern Paxson <vern@ee.lbl.gov>
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The word from the folks at SGI is that this is going to take some time to
sort out.  If you have a particular search string you want to send me, I'll
see what I have in my private archive, it has 1,000+ tcp-impl messages in it.

		Vern


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To: tcp-impl@cthulhu.engr.sgi.com
Subject: draft minutes for Chicago meeting
Cc: mallman@lerc.nasa.gov
Date: Thu, 08 Oct 1998 15:39:28 PDT
From: Vern Paxson <vern@ee.lbl.gov>
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Sorry these are so late ...  Please send us comments over the next
few days and then we'll finalize them.

		Vern


TCPIMPL minutes
42nd IETF, Chicago 
August 28, 1998

Reported by Evi Nemeth, with editing by the chairs & Sally Floyd.

Agenda:
	1.  Agenda bashing
	2.  Scott Bradner, intellectual property rights
	3.  Status
	4.  Known problems I-D
	5.  Security problems
	6.  RFC 2001 revision (congestion control)
	7.  WG closing after Orlando


-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=

1.  Agenda bashing:

No changes to the agenda.


2.  IPR (intellectual property rights) issues:

Scott Bradner reminded the group that if you know of intellectual
property issues in your company on a topic and don't say so, then
you cannot participate in the discussion and decisions regarding
that topic.  This is outlined in RFC 2028.


3.  Status

The testing tools document is done, RFC 2398.  The larger initial
window documents have been approved by IESG, and the 3 drafts will
soon become RFCs.  Regarding the restart of idle connections, see
draft-ietf-tcpimpl-restart, which Joe Touch reports will soon be
revised for publication.


4.  Known problems

7 new ones have been documented since the LA meeting.  Bill Fenner
described a new bug: if during a bi-directional transfer you are
sending and so is the other end, but you're not reading the incoming
data very fast, you can end up deadlocked with a full buffer.  For
example, a multithreaded client-server where one thread is sending a
lot, another is receiving a lot, but using one tcp connection.  The
fix is to change an unsigned to an int and recognize -1 as a valid
value.  Bill will explain it better and submit it.

There are 3 others which are less serious and also not yet
addressed.  The document will be forwarded to the IESG without
outlining these bugs.


5.  Security problems

There is a list of known problems:
	Predictable initial sequence number
	SYN flooding
	Land attack
Phil Karn noted that the latter two are really denial of service
attacks, and questioned the title of the section.


6.  RFC 2001 revision

High-level sketch of the revisions:
	- removed ambiguities
	- fixes for fast retransmit and fast recovery
	- added discussion of SACK
	- added larger initial window pointer

The discussion of the 2001 revision was a little chaotic at this
point and went back and forth between several topics.  The comments
about each topic have been grouped together for the minutes, and
therefore the comments are somewhat out-of-order.

Sally Floyd described two separate modifications to the Fast Retransmit
and Fast Recovery algorithms in RFC 2001.  The first modification is
the NewReno algorithm, introduced in Janey Hoe's SIGCOMM 96 paper,
which improves TCP's response to a "partial ack" received during Fast
Recovery, acknowledging some but not all of the packets sent before
Fast Recovery was initiated.  The preferred TCP algorithms would be
those of Sack TCP.  However, when the SACK option is not available, the
NewReno algorithm was described as a small but important change to make
to Reno TCP, avoiding Reno TCP's well-documented problems with
retransmit timeouts when multiple packets are dropped from a window of
data.

The second modification described was the bugfix algorithm for avoiding
unnecessary multiple fast retransmits.  This problem occurs in Reno
when, after a retransmit timeout, packets are retransmitted that have
already been received at the receiver.  When the TCP sender receives
three duplicate ACKs acknowledging three retransmitted packets, the
sender can incorrectly interpret this as a new instance of congestion.
Simulations showed that the NewReno algorighm and the bugfix for
avoiding-multiple-fast-retransmits are orthogonal - it is possible to
implement one and not the other.  However, while it is possible to
create scenarios with Reno or NewReno TCP where the bugfix for
avoiding-multiple-fast-retransmits would be helpful, it is not 
possible to create the pathological scenarios that can occur with Tahoe
TCP (e.g., TCP with Fast Retransmit but without Fast Recovery).

Sally's slides can be found at:
"ftp://ftp.ee.lbl.gov/talks/sf-tcpimpl-aug98.ps" (postscript), and
"ftp://ftp.ee.lbl.gov/talks/sf-tcpimpl-aug98.pdf" (PDF).

The chairs think that NewReno is a good thing; folks should implement it
(solaris 2.6 might already) and put out an experimental RFC or include it
(all or part) in 2001.

The decision was made to take this discussion to the mailing list and do an
experimental RFC with NewReno, rather than include it in the bugs list 
in RFC 2001.

There was then discussion of whether to include Sally's Reno
modification for avoiding-multiple-fast-retransmits in the RFC 2001
revision - how much experience with it do we need to include it?

Vern suggested that an experimental document with Sally's modification
could come out at the same time, and be referenced by 2001.

Kacheong Poon (Sun) confirmed that some implementations of NewReno can
behave like stop and go during retransmission (like in Janey Hoe's
paper).  This occurs when multiple packets are dropped from a window of
data, and NewReno TCP recovers by retransmitting at most one dropped
packet per roundtrip time.

Sally said it is possible to implement NewReno with "stop and go"
behavior, but that in an alternate implementation, included as an
option in the NS simulator, the retransmit timer is reset on only the
first retransmission.  In this case, instead of slowly recovering by
retransmitting at most one dropped packet per roundtrip time,
eventually the retransmit timer times out and the sender slow-starts.
The first-order fix for problems with multiple packets dropped from a
window of data is to use Sack, but when Sack is not available, NewReno
with this implementation should not perform worse than Reno.

Sally and Kacheong Poon agreed to confer on different possible
implementations of NewReno.

Phil Karn asked if we want to make TCP more aggressive in the face of
multiple packets dropped.  Sally answered: multiple packets dropped in a
window of data is one instance of congestion.  So cut the window in half,
do one retransmit; if retransmitted packets get lost, then it's more serious
and do slow start.

The RFC 2001 discussion continued with a discussion of ACKing every
second full sized segment being a MUST and not a SHOULD.  A wording
tweak is needed: that ACKing is *at least* every second full-sized
packet, since some systems ACK every segment, and that's allowed.

<There was a comment here about satellite environments which the
 note-taker didn't get recorded - anyone recall it?>

Another issue arose concerning ACK every 2nd full sized segment --
there's no way for the receiver to really know if the segments
arriving are full-sized.  Resolution: loosen the language but word
it so that today's TCPs are ok.

A question regarding definition of 3 duplicate ACKs - must they be
consecutive?  Answer: yes, they need to be consecutive, but it's
rare that they're not, so should not cause an implementation to be
labeled non-conformant.

7.  WG closing after Orlando

2001 is almost done if NewReno is not included.  The plan is to
close the working group after the next meeting (in Orlando).
However, in discussion after adjournment, the issue was raised of
documenting PMTU discovery issues, which may merit keeping the WG
active for one more meeting.


From owner-tcp-impl@cthulhu.engr.sgi.com  Tue Oct 13 19:17:30 1998
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To: mallman@lerc.nasa.gov
cc: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: NewReno and the 2001 Revision 
Date: Tue, 13 Oct 1998 16:10:11 PDT
From: Sally Floyd <floyd@ee.lbl.gov>
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> So, it seems to
>me that the appropriate approach is to outline the general
>principles that should guide the development of SACK-based
>algorithms, in order to prevent such algorithms from hurting the
>network and contributing to congestion collapse.  However, it seems
>less important to try to decide which SACK-based algorithm is the
>best and standardize it (at least at this time).

Actually, I would agree with this approach on a wide range of issues
regarding TCP implementations, not just about SACK algorithms.  I think
it is important to make sure that TCP implementations follow such basic
congestion control principles as slow-start, cutting the congestion
control window in half in response to a packet drop or in response to
Explicit Congestion Notification, etc.  I also think it is important to
list clear bugs such as those catalogued in the tcpimpl documents.
However, I don't know of any reason why TCP responses have to be
standardized in complete detail, even without considering SACK.  There
are a number of areas that it seems to me don't involve
interoperability, dangers of congestion collapse, dangers of
inexcuseably poor performance for the user, etc., and therefore could
perfectly safely be left to the discretion of the implementor.  I am
going to try to write something up making this position more clear...

- Sally


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From owner-tcp-impl@cthulhu.engr.sgi.com  Tue Oct 20 12:29:36 1998
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Date: Tue, 20 Oct 1998 09:14:35 -0700
From: "Eric A. Hall" <ehall@ehsco.com>
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Hi,

I hope this is the right place to ask this question.

Various documents seem to be at odds with each other regarding the
proper behavior for responding to a TCP segment that has an invalid
checksum. Although STD 7 states that the segment should be discarded, it
doesn't state what should happen next.

Some material suggests that the recipient shouldn't do anything, waiting
for the sender to timeout and retransmit. Conversely, other materials
suggest sending an ACK for the last successfully-received segment.
Either method would seem to work, with the benefit depending on the
exact scenario.

The question is: what's the position of the group as to which method is
the enforcable STANDARD method?

Thanks

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Tue Oct 20 12:34:42 1998
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From: Mohit Aron <aron@cs.rice.edu>
Message-Id: <199810201627.LAA16495@cs.rice.edu>
Subject: Re: question about TCP checksums
To: ehall@ehsco.com (Eric A. Hall)
Date: Tue, 20 Oct 1998 11:27:39 -0500 (CDT)
Cc: tcp-impl@cthulhu.engr.sgi.com
In-Reply-To: <362CB6EB.17C81BBD@ehsco.com> from "Eric A. Hall" at Oct 20, 98 09:14:35 am
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> 
> Various documents seem to be at odds with each other regarding the
> proper behavior for responding to a TCP segment that has an invalid
> checksum. Although STD 7 states that the segment should be discarded, it
> doesn't state what should happen next.
> 
> Some material suggests that the recipient shouldn't do anything, waiting
> for the sender to timeout and retransmit. Conversely, other materials
> suggest sending an ACK for the last successfully-received segment.
> Either method would seem to work, with the benefit depending on the
> exact scenario.
> 


If the checksumming fails, then the recipient cannot be sure of which
connection the TCP segment belongs to. The appropriate behavior then should
be not to do anything.



- Mohit Aron
  aron@cs.rice.edu


From owner-tcp-impl@cthulhu.engr.sgi.com  Tue Oct 20 12:51:05 1998
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Date: Tue, 20 Oct 1998 11:42:58 -0500 (CDT)
From: David Borman <dab@bsdi.com>
Message-Id: <199810201642.LAA07555@frantic.bsdi.com>
To: ehall@ehsco.com, tcp-impl@cthulhu.engr.sgi.com
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> Date: Tue, 20 Oct 1998 09:14:35 -0700
> From: "Eric A. Hall" <ehall@ehsco.com>
> Subject: question about TCP checksums
> ...
> Various documents seem to be at odds with each other regarding the
> proper behavior for responding to a TCP segment that has an invalid
> checksum. Although STD 7 states that the segment should be discarded, it
> doesn't state what should happen next.

Nothing, other than noting the fact in local statistics.

> Some material suggests that the recipient shouldn't do anything, waiting
> for the sender to timeout and retransmit. Conversely, other materials

Yes.

> suggest sending an ACK for the last successfully-received segment.
> Either method would seem to work, with the benefit depending on the
> exact scenario.

The "other" scenario is wrong.  Though you know that the IP source and
destination addresses are correct (since they are part of the IP header,
and IP would have tossed the packet if the IP checksum was bad), you
don't know that the TCP source/destination ports are correct, because
the TCP checksum is bad.  Thus, you can't be 100% sure that you would
even know who to send the ACK to.

> The question is: what's the position of the group as to which method is
> the enforcable STANDARD method?

Silently drop the packet, since when you have a TCP checksum error
you can't be 100% sure that you know for which connection the packet
was destined.  TCP will retransmit the dropped data.

> Thanks
>
> -- 
> Eric A. Hall                                            ehall@ehsco.com
> +1-650-685-0557                                    http://www.ehsco.com

			-David Borman, dab@bsdi.com


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To: tcp-impl@cthulhu.engr.sgi.com, rags@bapuji.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
Cc: ahughes@ISI.EDU, touch@ISI.EDU, johnh@ISI.EDU
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> From rags@bapuji.engr.sgi.com Thu Oct 22 11:37:39 1998
> From: rags@bapuji.engr.sgi.com (Raghu Mallena)
> Subject: 'Slow-Start Restart after Idle' draft
> To: tcp-impl@cthulhu.engr.sgi.com
> Date: Thu, 22 Oct 1998 11:36:20 -0700 (PDT)
> Cc: ahughes@ISI.EDU, touch@ISI.EDU, johnh@ISI.EDU,
>         rags@bapuji.engr.sgi.com (Raghu Mallena)
> 
> 
> I have a couple of questions on draft-ietf-tcpimpl-restart-00.txt
> (Issues in TCP Slow-Start Restart After Idle) and I was hoping
> that either the authors(Amy Hughes, Joe Touch, John Hiedemann)
> or someone else on the tcp-impl group who read it or implemented
> it can clarify this for me. 
> 
> My main question is about the Congestion Window Monitoring (CWM)
> solution proposed to avoid bursts.   The draft talks about
> replacing the following lines in tcp_output():
> 
> 	idle = (snd_max == snd_una);
> 	if (idle && now - lastrcv >= rto)
> 		cwnd = 1; 
> 
> with
> 	maxwin = 4 + snd_nxt - snd_una;
> 	if (cwnd > maxwin)
> 		cwnd = maxwin;
> 
> This proposal does burst control as expected,  but I wanted to make 
> sure that it does not undercut the regular slow-start algorithm 
> during normal start up.  It would be really bad if it did  because 
> exponential window growth is highly desirable for high speed LAN 
> connections so that the stable max window size can be reached 
> as soon as possible.
> 
> I took some traces and noticed that the initial slow-start behaviour
> will not be compromised *if* Delayed Ack works as expected.
...
> But what if Delayed Ack does not work as well and you start getting
> one Ack for every N packets with N>2?  

That algorithm does not work well when ACKs are lost, or when
'delayed acks' is not correctly being followed (i.e., sending less
ACKs is not correct behavior; receiving less ACKs may occur, however).


> One reason I can think of why the smooth 2-for-1 Ack flow
> might not happen is because of the way some of the BSD-derived
> stacks are implemented.  

Have you seen this behavior, or is this a thought-experiment?

> In Header Prediction mode, tcp_input() 
> routine does not call tcp_output() directly which means that
> an ACK will not be sent out for the second packet immediately.
> Instead, the input routine wakes up the socket and the
> thread that wakes up then calls tcp_output() after removing
> the data from the socket buffer.   In the mean time,  if there
> were a bunch of other packets that came in,  all of them are
> ACKed *at once* in the output routine. 

'delayed ACKS' would require that there were at least one ACK
for every two full segments received correctly, regardless of
what other data arrived in the meantime. This would result in 
an ACK burst.

> Any comments?  The fact that CWM can backfire if the
> ACKs dont come in smoothly is quite subtle.   Should the draft
> talk about this dependence?   Any real world experience from
> implementors who have incorporated CWM into their stack?

We have a modification in progress, which merges the best aspects
of CWM and Floyd's Maxburst. It avoids the effect you observed above,
where ACK losses cause CWM to reduce its window unnecessarily.

However, it is also clear that most of the algorithms considered
encourage interleaving of send and receive processing. Stalling
that interleave generates bursts; these bursts are what we are trying
to avoid, regardless of the reason.

Joe
 


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 22 15:10:42 1998
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From: rags@bapuji.engr.sgi.com (Raghu Mallena)
Message-Id: <199810221836.LAA03945@bapuji.engr.sgi.com>
Subject: 'Slow-Start Restart after Idle' draft
To: tcp-impl@cthulhu.engr.sgi.com
Date: Thu, 22 Oct 1998 11:36:20 -0700 (PDT)
Cc: ahughes@ISI.EDU, touch@ISI.EDU, johnh@ISI.EDU,
        rags@bapuji.engr.sgi.com (Raghu Mallena)
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I have a couple of questions on draft-ietf-tcpimpl-restart-00.txt
(Issues in TCP Slow-Start Restart After Idle) and I was hoping
that either the authors(Amy Hughes, Joe Touch, John Hiedemann)
or someone else on the tcp-impl group who read it or implemented
it can clarify this for me. 

My main question is about the Congestion Window Monitoring (CWM)
solution proposed to avoid bursts.   The draft talks about
replacing the following lines in tcp_output():

	idle = (snd_max == snd_una);
	if (idle && now - lastrcv >= rto)
		cwnd = 1; 

with
	maxwin = 4 + snd_nxt - snd_una;
	if (cwnd > maxwin)
		cwnd = maxwin;

This proposal does burst control as expected,  but I wanted to make 
sure that it does not undercut the regular slow-start algorithm 
during normal start up.  It would be really bad if it did  because 
exponential window growth is highly desirable for high speed LAN 
connections so that the stable max window size can be reached 
as soon as possible.

I took some traces and noticed that the initial slow-start behaviour
will not be compromised *if* Delayed Ack works as expected.
i.e  as long as we have an ACK coming in for every other packet
sent out (or one ACK per every packet if Delayed Ack is disabled),
then "maxwin" above is always more than "cwnd" (which is the cwnd
computed using Slow Start when the last ACK came in) and we will
still be in the exponential growth mode.   This is because
"snd_nxt - snd_una" roughly corresponds to "cwnd - 2" (2 comes
from the fact that snd_una would have moved by 2 in tcp_input()
when the last delayed ACK came in) and the comparison of 
'cwnd > maxwin' always fails.  

But what if Delayed Ack does not work as well and you start getting
one Ack for every N packets with N>2?   In this case,  the CWM
proposal could lower the cwnd below the existing value and then slow
start starts pushing it up again resulting in an osciallting
behaviour.   In a pathelogical case,  you could get one ACK
for all the outstanding unacknowledged packets which will
make snd_una=snd_nxt and cwnd will now be set all the way down to
4 from whatever its current value is.
 
One reason I can think of why the smooth 2-for-1 Ack flow
might not happen is because of the way some of the BSD-derived
stacks are implemented.   In Header Prediction mode, tcp_input() 
routine does not call tcp_output() directly which means that
an ACK will not be sent out for the second packet immediately.
Instead, the input routine wakes up the socket and the
thread that wakes up then calls tcp_output() after removing
the data from the socket buffer.   In the mean time,  if there
were a bunch of other packets that came in,  all of them are
ACKed *at once* in the output routine.   This could happen
quite often on high speed networks (even more so, if combined
with a relatively slower cpu).

Any comments?  The fact that CWM can backfire if the
ACKs dont come in smoothly is quite subtle.   Should the draft
talk about this dependence?   Any real world experience from
implementors who have incorporated CWM into their stack?

Thanks
Raghu
 


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 22 16:08:32 1998
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From: rags@bapuji.engr.sgi.com (Raghu Mallena)
Message-Id: <199810221953.MAA04092@bapuji.engr.sgi.com>
Subject: Re: 'Slow-Start Restart after Idle' draft
In-Reply-To: <199810221847.LAA21538@rum.isi.edu> from Joe Touch at "Oct 22, 98 11:47:21 am"
To: touch@ISI.EDU (Joe Touch)
Date: Thu, 22 Oct 1998 12:53:20 -0700 (PDT)
Cc: tcp-impl@cthulhu.engr.sgi.com, ahughes@ISI.EDU, touch@ISI.EDU,
        johnh@ISI.EDU
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> 
> > One reason I can think of why the smooth 2-for-1 Ack flow
> > might not happen is because of the way some of the BSD-derived
> > stacks are implemented.  
> 
> Have you seen this behavior, or is this a thought-experiment?

Not a thought-experiment.  I have seen this happen very often while
working on Gigabit Ethernet driver and TCP/IP performance
tuning over it.    If I recollect correctly,   some traces 
showed one ACK for about every 10 packets or so at times.   This
was with no packet loss of any sort.  

> 
> > In Header Prediction mode, tcp_input() 
> > routine does not call tcp_output() directly which means that
> > an ACK will not be sent out for the second packet immediately.
> > Instead, the input routine wakes up the socket and the
> > thread that wakes up then calls tcp_output() after removing
> > the data from the socket buffer.   In the mean time,  if there
> > were a bunch of other packets that came in,  all of them are
> > ACKed *at once* in the output routine. 
> 
> 'delayed ACKS' would require that there were at least one ACK
> for every two full segments received correctly, regardless of
> what other data arrived in the meantime. This would result in 
> an ACK burst.

Host Requirements RFC (rfc1122) that describes Delayed Ack says
that "there SHOULD be an ACK for at least every second segment".
Note that its a SHOULD and not a MUST.  So its not really
incorrect behaviour for stacks to send a single ACK for all
the data received during the time the stack spends in
processing the second segment. (i.e wake up the socket etc
as I described above).    I know atleast two stacks that do
this and I wont be surprised if there are more. ( For reference,
Steven's TCP/IP vol-II describes a similar scheme as well). 

> 
> > Any comments?  The fact that CWM can backfire if the
> > ACKs dont come in smoothly is quite subtle.   Should the draft
> > talk about this dependence?   Any real world experience from
> > implementors who have incorporated CWM into their stack?
> 
> We have a modification in progress, which merges the best aspects
> of CWM and Floyd's Maxburst. It avoids the effect you observed above,
> where ACK losses cause CWM to reduce its window unnecessarily.
> 

Cool!  I will give it a shot when its ready. 

> However, it is also clear that most of the algorithms considered
> encourage interleaving of send and receive processing. Stalling
> that interleave generates bursts; these bursts are what we are trying
> to avoid, regardless of the reason.
> 
> Joe
>  

I am not sure that what I described above can be considered a
burst.   Anyway,  I just wanted to point out the dependency that
CWM has on Delayed Ack implementations so the authors are aware
of it incase this needs a mention in the draft. 

- raghu



From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 22 16:14:57 1998
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From: sm@bossette.engr.sgi.com (Sam Manthorpe)
Message-Id: <199810222005.NAA01200@bossette.engr.sgi.com>
Subject: Re: 'Slow-Start Restart after Idle' draft
To: touch@ISI.EDU (Joe Touch)
Date: Thu, 22 Oct 1998 13:05:05 -0700 (PDT)
Cc: tcp-impl@cthulhu.engr.sgi.com, rags@bapuji.engr.sgi.com, ahughes@ISI.EDU,
        touch@ISI.EDU, johnh@ISI.EDU
In-Reply-To: <199810221847.LAA21538@rum.isi.edu> from "Joe Touch" at Oct 22, 98 11:47:21 am
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Joe Touch wrote:
> 'delayed ACKS' would require that there were at least one ACK
> for every two full segments received correctly, regardless of
> what other data arrived in the meantime. This would result in 
> an ACK burst.

The way I read it RFC-1122 does not `require' this, but rather
recommends it:

            A TCP SHOULD implement a delayed ACK, but an ACK should not
            be excessively delayed; in particular, the delay MUST be
            less than 0.5 seconds, and in a stream of full-sized
            segments there SHOULD be an ACK for at least every second
            segment.

-- Sam

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Sam Manthorpe, SGI.  tel: (650)933-2856 fax: (650)932-2856  sm@engr.sgi.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 22 16:33:05 1998
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Message-Id: <199810222022.QAA05686@guns.lerc.nasa.gov>
To: rags@bapuji.engr.sgi.com (Raghu Mallena)
From: Mark Allman <mallman@lerc.nasa.gov>
Reply-To: mallman@lerc.nasa.gov
cc: touch@ISI.EDU (Joe Touch), tcp-impl@cthulhu.engr.sgi.com, ahughes@ISI.EDU,
        johnh@ISI.EDU
Subject: Re: 'Slow-Start Restart after Idle' draft 
Organization: Late Night Hackers, NASA LeRC, Cleveland, Ohio
Song-of-the-Day: Nobody Told Me
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> Host Requirements RFC (rfc1122) that describes Delayed Ack says
> that "there SHOULD be an ACK for at least every second segment".
> Note that its a SHOULD and not a MUST.  

Actually, I think it is a SHOULD in one place in 1122 and a MUST in
another.  In 2001.bis (draft-ietf-tcpimpl-cong-control-00.txt) we
unambiguously say that you SHOULD implement delayed ACKs and if you
do, you MUST ACK every second full-sized segment (with some hedges
as far as what a full-sized segment is, per Matt Mathis comments to
the list before the last IETF meeting).

allman


---
http://gigahertz.lerc.nasa.gov/~mallman/


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> From rags@bapuji.engr.sgi.com Thu Oct 22 12:54:49 1998
> From: rags@bapuji.engr.sgi.com (Raghu Mallena)
> Subject: Re: 'Slow-Start Restart after Idle' draft
> To: touch@ISI.EDU (Joe Touch)
> Date: Thu, 22 Oct 1998 12:53:20 -0700 (PDT)
> Cc: tcp-impl@cthulhu.engr.sgi.com, ahughes@ISI.EDU, touch@ISI.EDU,
>         johnh@ISI.EDU
> 
> > 
> > > One reason I can think of why the smooth 2-for-1 Ack flow
> > > might not happen is because of the way some of the BSD-derived
> > > stacks are implemented.  
> > 
> > Have you seen this behavior, or is this a thought-experiment?
> 
> Not a thought-experiment.  I have seen this happen very often while
> working on Gigabit Ethernet driver and TCP/IP performance
> tuning over it.    If I recollect correctly,   some traces 
> showed one ACK for about every 10 packets or so at times.   This
> was with no packet loss of any sort.  

This might be an effect of the bundled-transmission of packets
that gigabit ethernet uses.

> > 'delayed ACKS' would require that there were at least one ACK
> > for every two full segments received correctly, regardless of
> > what other data arrived in the meantime. This would result in 
> > an ACK burst.
> 
> Host Requirements RFC (rfc1122) that describes Delayed Ack says
> that "there SHOULD be an ACK for at least every second segment".
> Note that its a SHOULD and not a MUST.  So its not really

This is called "stretch ACK violation"  in draft-ietf-tcpimpl-prob-04.txt,
and there it's listed as a MUST.

It appears that's a bug in the draft, unless it has been
upgraded to a MUST elsewhere (??)

Joe


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[Ignorant mode on.]

> In 2001.bis (draft-ietf-tcpimpl-cong-control-00.txt) we unambiguously
> say that you SHOULD implement delayed ACKs and if you do, you MUST
> ACK every second full-sized segment

My Caldera 1.3 system here sends one ACK for every seven segments, with
no apparent loss in performance.

I take it the benefit of frequent ACKs is most pronounced when timing is
a problem. What is the benefit to increasing the ACK frequency if the
timing is solid?

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


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> From owner-tcp-impl@cthulhu.engr.sgi.com Thu Oct 22 13:53:24 1998
> Date: Thu, 22 Oct 1998 13:48:37 -0700
> From: "Eric A. Hall" <ehall@ehsco.com>
> X-Accept-Language: en
> To: tcp-impl@cthulhu.engr.sgi.com
> Subject: Re: 'Slow-Start Restart after Idle' draft
> 
> 
> [Ignorant mode on.]
> 
> > In 2001.bis (draft-ietf-tcpimpl-cong-control-00.txt) we unambiguously
> > say that you SHOULD implement delayed ACKs and if you do, you MUST
> > ACK every second full-sized segment
> 
> My Caldera 1.3 system here sends one ACK for every seven segments, with
> no apparent loss in performance.
> 
> I take it the benefit of frequent ACKs is most pronounced when timing is
> a problem. What is the benefit to increasing the ACK frequency if the
> timing is solid?

ACK clocking is a pacing-like side-effect of producing ACKs
on a consistent basis.

Sending reduced numbers of ACKs can generate line-rate bursts,
for the same reason as losing them, or allowing the window
to accumulate 'rights to send' without using it.

Joe		


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 22 17:14:50 1998
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> Sending reduced numbers of ACKs can generate line-rate bursts,

"Can" is the operative word here. This assumes that throughput or other
factors will cause the processing cycles to get out-of-synch. But what
if the network is running great? It seems that if the recipient can ACK
every six or seven segments without the two end-points getting out of
synch, it should be allowed to do so.

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 22 17:28:48 1998
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> This might be an effect of the bundled-transmission of packets
> that gigabit ethernet uses.


Gigabit Ethernet only uses bundled-transmission if its running in 
half duplex mode (as far as I know) and there are few if any 
products that support halk-duplex gigabit Ethernet

Scott


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> It seems that if the recipient can ACK every six or seven segments
> without the two end-points getting out of synch, it should be allowed
> to do so.

It seems like the need for ACKs is a function of the RTT and the size of
the receive buffer. ACKs should only be required if (a) the receive
buffer is half it's normal capacity or (b) if the time required to
deliver the ACK would cause the sender to stop transmitting before the
ACK was received.

Fixing the value to two seems kind of arbitrary.

[Ignorant mode is still on, btw].

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 22 17:51:43 1998
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Date: Thu, 22 Oct 1998 15:36:00 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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I doubt the intrinsic burstiness of Gbit Ethernet is responsible for the
reduced ACKs reported here this morning.  While the Ethernet driver might
know that a bunch of IP packets arrived not merely during the servicing
of a single interrupt but were a concatenated burst, the upper layers know
only that a bunch of TCP segments arrived in a short time.  Such things
happen for many causes, including the system being busy dealing with other
work.


> From: Joe Touch <touch@ISI.EDU>

> ...
> Sending reduced numbers of ACKs can generate line-rate bursts,
> ...

Line-rate bursts can be bad, for example when you are talking to a LAN
that is the first hop in a long and slow link.

However, line-rate bursts are necessary, desirable, and inevitable on LANs
if you want to approach the 100% utilization.  That is obvious on token
rings such as FDDI.  If your system has not had the token for a while, it
has received more than one Ack while it was waiting, and your system is
not amone the lame and halt, then it will send a bunch of TCP segments in
a burst at a somewhat more than 98 Mbit/sec.  (The speed of light for
TCP/IP/FDDI on a small ring with 4K segments and 2:1 segments:ACKs is 98
Mbit/sec of user data.)

Between a pair of hosts on <= 100MHz Ethernet, similar effects exist,
although they are perhaps less obvious.  Ignoring those effects, and with
BEB instead of BLAM, you really want line-rate bursts that are long but
not too long, and you want to reduce the number of ACKs to about one per
burst in order to avoid or minimize the effects of the Ethernet capture
effect.

Traffic that consists of alternating line-rate bursts of several dozen
full sized segments in one direction and bursts of half that many ACKs in
the other direction not only looks stupid on a monitor, but is measurably
slower than the obvious alternative.

Because of the line-rate burst facts of life on LANs, any standard that
tries to outlaw line-rate bursts is hopeless in LANs, even if implementors
do not choose to ignore it to improve the benchmark numbers that matter
most.


Vernon Schryver    vjs@rhyolite.com


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To: "Eric A. Hall" <ehall@ehsco.com>
Cc: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
In-reply-to: Your message of Thu, 22 Oct 1998 13:48:37 PDT.
Date: Thu, 22 Oct 1998 15:29:58 PDT
From: Vern Paxson <vern@ee.lbl.gov>
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> I take it the benefit of frequent ACKs is most pronounced when timing is
> a problem. What is the benefit to increasing the ACK frequency if the
> timing is solid?

Along with the points Joe made, two other problems are that infrequent acks
open the congestion window more slowly during slow start, and if acks are
rare then if there's loss along the reverse path, connections are more
likely to suffer from timeouts because they're starved for acks.

		Vern


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To: tcp-impl@cthulhu.engr.sgi.com, vjs@calcite.rhyolite.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> Date: Thu, 22 Oct 1998 15:36:00 -0600 (MDT)
> From: Vernon Schryver <vjs@calcite.rhyolite.com>
> To: tcp-impl@cthulhu.engr.sgi.com
> Subject: Re: 'Slow-Start Restart after Idle' draft
> 
...
> > From: Joe Touch <touch@ISI.EDU>
> 
> > ...
> > Sending reduced numbers of ACKs can generate line-rate bursts,
> > ...
> 
> Line-rate bursts can be bad, for example when you are talking to a LAN
> that is the first hop in a long and slow link.
> 
> However, line-rate bursts are necessary, desirable, and inevitable on LANs
> if you want to approach the 100% utilization.  That is obvious on token
...

The caveat is that these mechanisms, like slow-start restart,
should be disabled on a LAN. We hadn't put that in the latest
draft, but it is in the revision. It's the same check as slow-start -
if the net prefixes match, ignore the code.

Joe


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 22 19:12:24 1998
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From: sm@bossette.engr.sgi.com (Sam Manthorpe)
Message-Id: <199810222300.QAA01993@bossette.engr.sgi.com>
Subject: Re: 'Slow-Start Restart after Idle' draft
To: ehall@ehsco.com (Eric A. Hall)
Date: Thu, 22 Oct 1998 16:00:31 -0700 (PDT)
Cc: tcp-impl@cthulhu.engr.sgi.com
In-Reply-To: <362FA49D.3F1C22B3@ehsco.com> from "Eric A. Hall" at Oct 22, 98 02:33:17 pm
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Eric A. Hall wrote:
> 
> 
> > It seems that if the recipient can ACK every six or seven segments
> > without the two end-points getting out of synch, it should be allowed
> > to do so.
> 
> It seems like the need for ACKs is a function of the RTT and the size of
> the receive buffer. ACKs should only be required if (a) the receive
> buffer is half it's normal capacity or (b) if the time required to
> deliver the ACK would cause the sender to stop transmitting before the
> ACK was received.
>
> Fixing the value to two seems kind of arbitrary.

I beleive the reason for the value two is to avoid the situation where
the rate at which the congestion window increase during slow-start and 
congestion avoidance is limited to one increment every 200ms (i.e. when 
the delayed-ack timer expires).  If we don't force acks and simply rely
on the delayed-ack timer then we can see very serious performance
degradation during slow-start.

-- Sam

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Sam Manthorpe, SGI.  tel: (650)933-2856 fax: (650)932-2856  sm@engr.sgi.com


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Date: Thu, 22 Oct 1998 17:13:11 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> From: Joe Touch <touch@ISI.EDU>

> ...
> > However, line-rate bursts are necessary, desirable, and inevitable on LANs
> > if you want to approach the 100% utilization.  That is obvious on token
> ...
>
> The caveat is that these mechanisms, like slow-start restart,
> should be disabled on a LAN. We hadn't put that in the latest
> draft, but it is in the revision. It's the same check as slow-start -
> if the net prefixes match, ignore the code.

That's a big improvement and I wish it were enough.  I don't think it is.

A pair of hosts can tell only that they are on the same IP network by
looking at prefixes, and cannot even get that right with some flavors of
variable length subnet masking.

If they assume there are no boxes acting as forwarders between them and
turn off the usual mechanisms, then they might be right and they might be
wrong.  Consider the literally millions of hosts that are either bridged
over very slow links (e.g. ISDN TA's) or that have intentionally bogus
network masks and rely on prox-ARP or other hacks at the "PPP server" side
of a modem or ISDN PPP link.   Those cases need slow-start more than many
others on the Internet, and probably more than most of the tests rigs in
which such things are tested, but their network prefixes will tell them
to turn off slow-start when talking to hosts (e.g. DNS, NNTP, and email
servers) at their ISP.

Then consider a pair of hosts that are on different <=100 Mbit/sec LANs,
but where the LANs are interconnected with fast mechanisms of one sort or
another (>= 100 Mbps, e.g. ATM, short-haul SONET, FC, or HIPPI), and where
they should be in fast-break-the-rules mode.

There needs to be a test that detects a super fast, almost zero delay,
practically zero loss link between a pair of hosts to turn off slow-start
etc., but I think those examples show that network prefixesis not good
enough.

Instead of matching network prefixes, why not look at the RTT?  Regardless
of prefixes, if the RTT is less than 10 milliseconds, it's probably a good
idea enter go-fast mode.  If the RTT is more than 10 ms, go-fast mode
won't gain anything and slow-start is a good idea, again regardless of
prefixes.


Vernon Schryver    vjs@rhyolite.com


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> I beleive the reason for the value two is to avoid the situation where
> the rate at which the congestion window increase during slow-start

My comments are in regard to post-start. IE, once slow-start has brought
the two end-points up to full throttle.

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


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From: Keith Owens <kaos@ocs.com.au>
To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft 
In-reply-to: Your message of "Thu, 22 Oct 1998 15:51:39 MST."
             <199810222251.PAA27887@rum.isi.edu> 
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On Thu, 22 Oct 1998 15:51:39 -0700 (PDT), 
Joe Touch <touch@ISI.EDU> wrote:
>[ACK-burst stuff]
>The caveat is that these mechanisms, like slow-start restart,
>should be disabled on a LAN. We hadn't put that in the latest
>draft, but it is in the revision. It's the same check as slow-start -
>if the net prefixes match, ignore the code.

Not having seen the revision yet, what happens if the "LAN" has stubs
behind slower links that are reached using proxy ARP?  Network prefixes
will match, turn off all the congestion mechanisms and overload the
slow links.  Of course we could break our nice flat backbone into lots
of subnets and reprogram all the desktop configurations ...



From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 22 21:16:47 1998
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To: kaos@ocs.com.au, tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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I agree that there is a problem with intra-LAN links, and
the detection thereof.

Whatever that solution, it should be applied equally to
avoiding slow-start and slow-start restart.

The current code makes an assumption, that the network
address indicates intra-LAN if it coincides with an
existing interface address. That assumption generates false
positives and negatives.

Can anyone suggest a general solution?

(in the meantime, I'd consider the "is_local" a placeholder
for a more correct function - and otherwise not address it
separately in the slow-start-restart doc...)

Joe


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 22 21:47:14 1998
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Date: Thu, 22 Oct 1998 19:35:50 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <199810230135.TAA14971@calcite.rhyolite.com>
To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> From: Joe Touch <touch@ISI.EDU>

> I agree that there is a problem with intra-LAN links, and
> the detection thereof.
>
> Whatever that solution, it should be applied equally to
> avoiding slow-start and slow-start restart.
>
> The current code makes an assumption, that the network
> address indicates intra-LAN if it coincides with an
> existing interface address. That assumption generates false
> positives and negatives.
>
> Can anyone suggest a general solution?
> ...

What about what I just said?

You want to avoid slow-start and slow-restart and relax the 2:1
segment:ACK ratio exactly when the hosts are close to each other,
which to say, when the RTT is small.

10 milliseconds is about the right dividing line, since round trips
on reasonable LANs are less than 10 ms, and any long haul link with
an RTT of less than 10 is forced by the speed of light to be fast
and short enough to treat like a LAN.

The only problem with using the measured RTT to know when to drop fancy
stuff is that a new TCP connection does not know it immediately.  To fix
that, use the mechanisms that have been suggested for priming the RTT on
long haul lines.  Cache it however you cache the discovered path MTU.

You might use the round trip time on the 3-way handshake instead of the
smoothed RTT.  Hosts that should start with slow-start off are likely
to receive the SYN-ACK within 10 ms of sending the SYN, and the first
data within 10 ms of sending the SYN-ACK.


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 01:06:54 1998
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From: Luigi Rizzo <luigi@labinfo.iet.unipi.it>
Message-Id: <199810230301.EAA13915@labinfo.iet.unipi.it>
Subject: Re: 'Slow-Start Restart after Idle' draft
To: touch@ISI.EDU (Joe Touch)
Date: Fri, 23 Oct 1998 04:01:02 +0100 (MET)
Cc: tcp-impl@cthulhu.engr.sgi.com, vjs@calcite.rhyolite.com
In-Reply-To: <199810222251.PAA27887@rum.isi.edu> from "Joe Touch" at Oct 22, 98 03:51:20 pm
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> > Line-rate bursts can be bad, for example when you are talking to a LAN
> > that is the first hop in a long and slow link.
> > 
> > However, line-rate bursts are necessary, desirable, and inevitable on LANs
> > if you want to approach the 100% utilization.  That is obvious on token
> ...
> 
> The caveat is that these mechanisms, like slow-start restart,
> should be disabled on a LAN. We hadn't put that in the latest
> draft, but it is in the revision. It's the same check as slow-start -
> if the net prefixes match, ignore the code.

FreeBSD does this as a side effect of the introduction of TTCP, but
this is very bad (and i recently introduced a sysctl variable to
control this behaviour) because in many cases the address block is
further segmented downstream and the sender is unaware of that. Think
of a machine acting as a ppp server by allocating addresses from the
same range as the primary interface and doing proxy arp.

in my opinion, just checking netmasks is very prone to failute and
ignoring slowstart should be explicitly configured if desired.

	cheers
	luigi
-----------------------------+--------------------------------------
Luigi Rizzo                  |  Dip. di Ingegneria dell'Informazione
email: luigi@iet.unipi.it    |  Universita' di Pisa
tel: +39-50-568533           |  via Diotisalvi 2, 56126 PISA (Italy)
fax: +39-50-568522           |  http://www.iet.unipi.it/~luigi/
_____________________________|______________________________________


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 04:10:53 1998
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> From: Vernon Schryver <vjs@calcite.rhyolite.com>
> To: tcp-impl@cthulhu.engr.sgi.com
> Subject: Re: 'Slow-Start Restart after Idle' draft
> 
> > From: Joe Touch <touch@ISI.EDU>
...
> > Can anyone suggest a general solution?
...
> What about what I just said?
> 
> You want to avoid slow-start and slow-restart and relax the 2:1
> segment:ACK ratio exactly when the hosts are close to each other,
> which to say, when the RTT is small.

RTT may be correlated (it is debatable how closely); hopcount is
much more meaningful. If you can get there in 0 hops, then there
is no router, and you shouldn't be doing these tricks. That would 
require a traceroute-like test, out-of-band to the regular TCP
packets.

Closeness is not an issue - if I have a direct satellite
link, the RTT can be 500 ms but I still want to do the bursting,
since there is no router to overload.

Joe


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 06:46:44 1998
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Date: Fri, 23 Oct 1998 12:29:34 +0200 (CEST)
From: Sean Doran <smd@ebone.net>
Message-Id: <199810231029.MAA23878@sean.ebone.net>
To: tcp-impl@cthulhu.engr.sgi.com, vjs@calcite.rhyolite.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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Vernon Schryver wrote:

| A pair of hosts can tell only that they are on the same IP network by
| looking at prefixes, and cannot even get that right with some flavors of
| variable length subnet masking.

Well, yes and no: ARP or ES-IS usually breaks when there
is a serious mask mismatch.

The bigger concern is LISes which are bridged.   

Unfortunately, there is stuff like that out in the world,
and so making assumptions based only on the IP address seems
pretty dangerous.

One might also consider the case where a large server is
on the "outside" of a NAT, and is sending data to a host
on the "inside" (where, for example, it has an RFC-1918 address).
If the server and the NAT are in the same LIS (or on the same LAN)
then disabling congestion-avoidance mechanisms on the assumption
that no non-local infrastructure will be involved is a mistake,
despite the fact that the receiver address, from the perspective
of the "outside" server, is the NAT itself.

Finally, as Vernon notes, even where a LAN really is a LAN and
retransmissions are cheap,  it seems fair to wonder if it's smart
to dump traffic into them in an non-congestion-avoiding way for
the sake of performance.  After all, the medium may be shared by
other users, and they may see a performance decrease as the result
of traffic being blasted into the LAN.

| There needs to be a test that detects a super fast, almost zero delay,
| practically zero loss link between a pair of hosts to turn off slow-start
| etc., but I think those examples show that network prefixesis not good
| enough.

The only thing that seems to make sense would be recent observations
of performance between the two things that may be super-fast-losslessly-
connected to each other.

	Sean.


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From: Sean Doran <smd@ebone.net>
Message-Id: <199810231103.NAA23937@sean.ebone.net>
To: tcp-impl@cthulhu.engr.sgi.com, vjs@calcite.rhyolite.com
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Vernon Schryver wrote:

| Instead of matching network prefixes, why not look at the RTT?  Regardless
| of prefixes, if the RTT is less than 10 milliseconds, it's probably a good
| idea enter go-fast mode.  If the RTT is more than 10 ms, go-fast mode
| won't gain anything and slow-start is a good idea, again regardless of
| prefixes.

At which point do you take the RTT measurement?

Obviously, you also want to rapidly transition into normal mode
the moment you detect any increase in RTT or the moment you see loss.

Hopefully, if there's a mistake and a bunch of congestion is
created by an assumption that it might be safe to be in go-fast
mode, it will remain local and on cheap infrastructure and won't
interfere too badly with other traffic.

Maybe I'm a little too conservative, but 10ms seems pretty long to me.
I can get well past the border and beyond Tele2's router in Malmö
Sweden in less time than that, queues willing (which is thankfully 
fairly often).   Fast hosts blasting traffic to there would be very painful.

	Sean.


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 07:39:46 1998
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From: Sean Doran <smd@ebone.net>
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Vernon Schryver wrote:

| 10 milliseconds is about the right dividing line, since round trips
| on reasonable LANs are less than 10 ms, and any long haul link with
| an RTT of less than 10 is forced by the speed of light to be fast
| and short enough to treat like a LAN.

You have to explain this one to me.

Bear in mind that some links are quiescent most of the time.

Bear in mind also that mileage charges are meaningless when
a border is being crossed.  A circuit crossing the Canada/US
border is going to be roughly 3x the cost of the same length
national circuit on either side of the border.   The factor
of three rapidly becomes a factor of five to twenty in many
other parts of the world.

Treating short expensive links like a LAN is perhaps a bad idea
in some cases.  I wonder if heavily multiplexed but usually fairly
quiet fast lines could lose badly is multiple senders suddenly tried
to treat such a line as being "short and fat enough to treat like a LAN".

A classical example might be a 45Mbps line connected on either end to a
multiplexing router with a half dozen or so 10Mbps ethernet ports,
where the hosts on the ethernets randomly do bulk transfers amongst
themselves.  

	Sean.


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Date: Fri, 23 Oct 1998 14:25:28 +0200 (CEST)
From: Sean Doran <smd@ebone.net>
Message-Id: <199810231225.OAA24217@sean.ebone.net>
To: tcp-impl@cthulhu.engr.sgi.com, touch@ISI.EDU, vjs@calcite.rhyolite.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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Joe Touch wrote:

| RTT may be correlated (it is debatable how closely); hopcount is
| much more meaningful. If you can get there in 0 hops, then there
| is no router, and you shouldn't be doing these tricks. That would 
| require a traceroute-like test, out-of-band to the regular TCP
| packets.

This seems like a good approach, but traceroute-like
tests may not be so simple -- consider the NAT case again:  you
send traffic to the "outside" address of the NAT itself, which may
be hopcount=0, although the traffic passing through the NAT will
be hopcount > 0.

	Sean.


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Date: Fri, 23 Oct 1998 14:45:25 +0200
From: Andi Kleen <ak@muc.de>
To: Sean Doran <smd@ebone.net>, tcp-impl@cthulhu.engr.sgi.com, touch@ISI.EDU,
        vjs@calcite.rhyolite.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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On Fri, Oct 23, 1998 at 02:25:28PM +0200, Sean Doran wrote:
> Joe Touch wrote:
> 
> | RTT may be correlated (it is debatable how closely); hopcount is
> | much more meaningful. If you can get there in 0 hops, then there
> | is no router, and you shouldn't be doing these tricks. That would 
> | require a traceroute-like test, out-of-band to the regular TCP
> | packets.
> 
> This seems like a good approach, but traceroute-like
> tests may not be so simple -- consider the NAT case again:  you
> send traffic to the "outside" address of the NAT itself, which may
> be hopcount=0, although the traffic passing through the NAT will
> be hopcount > 0.

Also it doesn't work for things like LANE over ATM or MPOA. 

-Andi


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Subject: Re: 'Slow-Start Restart after Idle' draft
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> From andi@fred.muc.de Fri Oct 23 05:50:50 1998
> Date: Fri, 23 Oct 1998 14:45:25 +0200
> From: Andi Kleen <ak@muc.de>
> Subject: Re: 'Slow-Start Restart after Idle' draft
> 
> On Fri, Oct 23, 1998 at 02:25:28PM +0200, Sean Doran wrote:
> > Joe Touch wrote:
> > 
> > | RTT may be correlated (it is debatable how closely); hopcount is
> > | much more meaningful. If you can get there in 0 hops, then there
> > | is no router, and you shouldn't be doing these tricks. That would 
> > | require a traceroute-like test, out-of-band to the regular TCP
> > | packets.
> > 
> > This seems like a good approach, but traceroute-like
> > tests may not be so simple -- consider the NAT case again:  you
> > send traffic to the "outside" address of the NAT itself, which may
> > be hopcount=0, although the traffic passing through the NAT will
> > be hopcount > 0.

Agreed.
 
> Also it doesn't work for things like LANE over ATM or MPOA. 

Here I'd argue it works just fine - if you want to make your ATM/MPOA
look like a LAN, then it looks like a LAN, bursts and all. Rate limits
at the ATM level would be required to avoid problems, however.

Joe




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Date: Fri, 23 Oct 1998 08:25:59 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <199810231425.IAA17032@calcite.rhyolite.com>
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> From: Joe Touch <touch@ISI.EDU>

> > You want to avoid slow-start and slow-restart and relax the 2:1
> > segment:ACK ratio exactly when the hosts are close to each other,
> > which to say, when the RTT is small.
>
> RTT may be correlated (it is debatable how closely); hopcount is
> much more meaningful. If you can get there in 0 hops, then there
> is no router, and you shouldn't be doing these tricks. That would 
> require a traceroute-like test, out-of-band to the regular TCP
> packets.

I meant to point out in my first message Thursday that traceroute cannot
detectect all of the forwarding boxes on which there is often congestion
that needs slow start.  Consider bridges.

I disagree that hopcount is more meaningful than RTT.  Whether there are
0 or 10 hops is not what determines whether slow start and ACK pacing
are important, but whether there are hops where congestion should be avoided.
For the bad hop count 0 case, consider a host that is able to run far
faster than 100 Mbit/sec and that is taking 7000 HTTP hits/second, and
also running a a few long term TCP connections to nearby systems, such as
for syslog.  Those local TCP links need slow start and the host needs RED
or similar on its local interfaces because its interfaces are congested.

I claim those cases get more accurate answers by looking at the RTT.


> Closeness is not an issue - if I have a direct satellite
> link, the RTT can be 500 ms but I still want to do the bursting,
> since there is no router to overload.

Don't satellite links almost always involve boxes doing forwarding (as
well as FEC) and visible the count?


Vernon Schryver    vjs@rhyolite.com


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Date: Fri, 23 Oct 1998 08:38:37 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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Subject: Re: 'Slow-Start Restart after Idle' draft
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> From: Sean Doran <smd@ebone.net>

> | A pair of hosts can tell only that they are on the same IP network by
> | looking at prefixes, and cannot even get that right with some flavors of
> | variable length subnet masking.
>
> Well, yes and no: ARP or ES-IS usually breaks when there
> is a serious mask mismatch.

Proxy-ARP is amazingly common--amazing considering what a kludge it is.


> The bigger concern is LISes which are bridged.   

Yes, bridges are fatal for either hop-counting or netmask checks for 
turning on go-fast mode.



> ...
> Finally, as Vernon notes, even where a LAN really is a LAN and
> retransmissions are cheap,  it seems fair to wonder if it's smart
> to dump traffic into them in an non-congestion-avoiding way for
> the sake of performance.  After all, the medium may be shared by
> other users, and they may see a performance decrease as the result
> of traffic being blasted into the LAN.

Agreed.
The Ethernet capture effect is a good (i.e. bad) case to consider.
It need not involve two hosts, but a group of winners and another
group of losers.


> | There needs to be a test that detects a super fast, almost zero delay,
> | practically zero loss link between a pair of hosts to turn off slow-start
> | etc., but I think those examples show that network prefixesis not good
> | enough.
>
> The only thing that seems to make sense would be recent observations
> of performance between the two things that may be super-fast-losslessly-
> connected to each other.

Exactly!  The average (not smoothed), the recent worst-case RTT, or the
time required for the 3-way handshake are good indications of the common
cases where you can use go-fast mode, with 10 ms being a reasonable cut-off.

Note that a router or bridge hop or two likely to push the RTT or 3-way
turn-around above 10 ms.  Or maybe 3-5 ms.  The RTT for small packets on
a quiet 10 MHz Ethernet is less than 2 ms., but the maximum minimum RTT
approaches 3 ms for full sized packets.  5 ms on FDDI tells you that you
have 0 hops and the ring is not eggregiously saturated.


Vernon Schryver    vjs@rhyolite.com


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Date: Fri, 23 Oct 1998 09:28:31 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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(I appologize for sending 3 separate messages instead of checking
everything in my mailbox before replying.)


> From smd@sean.ebone.net Fri Oct 23 05:03:07 1998

> | Instead of matching network prefixes, why not look at the RTT?  Regardless
> ...

> At which point do you take the RTT measurement?

Why not during the 3-way handshake?
Or during the previous connection, cached along with your path-MTU
measurement?


> Obviously, you also want to rapidly transition into normal mode
> the moment you detect any increase in RTT or the moment you see loss.

Agreed.
I think it would go-fast mode would best leave slow-restart enabled at
all times, and only turn off initial-slow-start and enable ACK compression
(or disable ACK pacing).  With the big windows you need to get reasonable
throughput on a LAN, fast-retransmission is quite effective for the rare
loss (due e.g. excessive Ethernet collisions) and mitigates the slow-dwon
from slow-restart.


> ...
> Maybe I'm a little too conservative, but 10ms seems pretty long to me.
> ...

Ok, how about 5 ms?
Or 10 ms divided by the half ratio of the local interface speed to 10 MHhz.
E.g. 10 ms on a 10 MHz Ethernet, 2 ms on FDDI or 100 MHz Ethernet.
weeell, that breaks on HIPPI, which definitely wants go-fast mode.

] From smd@sean.ebone.net Fri Oct 23 05:25:37 1998

] ...
] | 10 milliseconds is about the right dividing line, since round trips
] | on reasonable LANs are less than 10 ms, and any long haul link with
] | an RTT of less than 10 is forced by the speed of light to be fast
] | and short enough to treat like a LAN.
]
] You have to explain this one to me.

My calculator broke.  10 ms is ~2000 miles, not the 2-20 I was (not)thinking.

On the other hand, each forwarding hop at bit rates of less than 45 Mbps
tends to add 1 or more ms., and that tends to weed out the metro-sized
fast WANs.  The wire-occupancy of a 1500 byte packet at 10 MHz is 1.2 ms,
, so 10 ms is not far from the watershed.


] Bear in mind that some links are quiescent most of the time.

And that is fundamental to why go-fast mode is a good thing on LANs.
Conversely, a saturate LAN has a larger RTT and probably wants slow-start.


] Bear in mind also that mileage charges are meaningless when
] a border is being crossed.  A circuit crossing the Canada/US
] border is going to be roughly 3x the cost of the same length
] national circuit on either side of the border.   The factor
] of three rapidly becomes a factor of five to twenty in many
] other parts of the world.

What do monetary charges have to do with go-fast mode?

] Treating short expensive links like a LAN is perhaps a bad idea
] in some cases.

I would have thought that short, expensive links are even more important
candidates for go-fast mode, because their bits are more expensive.  What
matters is whether go-fast mode works better than the now standard way
of pushing bits.


] in some cases.  I wonder if heavily multiplexed but usually fairly
] quiet fast lines could lose badly is multiple senders suddenly tried
] to treat such a line as being "short and fat enough to treat like a LAN".
]
] A classical example might be a 45Mbps line connected on either end to a
] multiplexing router with a half dozen or so 10Mbps ethernet ports,
] where the hosts on the ethernets randomly do bulk transfers amongst
] themselves.  

It depends on the inter-Ethernet traffic.  As long as there are not more
than 4.5 inter-Ethernet bulk transfers happening, you want go-fast mode.

In fact, even when the 45 Mbps interconnect is saturated, you want to turn
off the 2:1 segment:ACK rule to reduce bad results of the Ethernet capture
effect on the individual Ethernets.
The Ethernet capture effect reduces TCP/IP/Ethernet throughput from
the TCP speed of light of about 93% of the raw bit rate to about 68%.
Lots of ACKs are a Very Bad Thing on Ethernets.


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 12:55:58 1998
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From: sm@bossette.engr.sgi.com (Sam Manthorpe)
Message-Id: <199810231644.JAA05688@bossette.engr.sgi.com>
Subject: Re: 'Slow-Start Restart after Idle' draft
To: vjs@calcite.rhyolite.com (Vernon Schryver)
Date: Fri, 23 Oct 1998 09:44:07 -0700 (PDT)
Cc: tcp-impl@cthulhu.engr.sgi.com
In-Reply-To: <199810231425.IAA17032@calcite.rhyolite.com> from "Vernon Schryver" at Oct 23, 98 08:25:59 am
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Vernon Schryver wrote:
> 
> > From: Joe Touch <touch@ISI.EDU>
> 
> > > You want to avoid slow-start and slow-restart and relax the 2:1
> > > segment:ACK ratio exactly when the hosts are close to each other,
> > > which to say, when the RTT is small.
> >
> > RTT may be correlated (it is debatable how closely); hopcount is
> > much more meaningful. If you can get there in 0 hops, then there
> > is no router, and you shouldn't be doing these tricks. That would 
> > require a traceroute-like test, out-of-band to the regular TCP
> > packets.
> 
> I meant to point out in my first message Thursday that traceroute cannot
> detectect all of the forwarding boxes on which there is often congestion
> that needs slow start.  Consider bridges.
> 
> I disagree that hopcount is more meaningful than RTT.  Whether there are
> 0 or 10 hops is not what determines whether slow start and ACK pacing
> are important, but whether there are hops where congestion should be avoided.
> For the bad hop count 0 case, consider a host that is able to run far
> faster than 100 Mbit/sec and that is taking 7000 HTTP hits/second, and
> also running a a few long term TCP connections to nearby systems, such as
> for syslog.  Those local TCP links need slow start and the host needs RED
> or similar on its local interfaces because its interfaces are congested.
> 
> I claim those cases get more accurate answers by looking at the RTT.

It seems both methods have their flaws; how about a routing table
flag that allows go-fast-mode on a route-by-route basis.  This could
be both manually set/cleared and automatically set/cleared by some, as yet
undefined, mechanism.  This also avoids introducing cubersome hop-counting
or RTT estimation mechanisms.  

You would have to be careful about setting go-fast-mode for the local
net if you have proxy-ARP on that net, but you could still hack around
that by setting up specific routes for the proxyarped hosts.  It would
be a bit messy, but that would serve you right ;-)

-- Sam

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Sam Manthorpe, SGI.  tel: (650)933-2856 fax: (650)932-2856  sm@engr.sgi.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 13:13:58 1998
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Date: Fri, 23 Oct 1998 10:59:57 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <199810231659.KAA17788@calcite.rhyolite.com>
To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> From: sm@bossette.engr.sgi.com (Sam Manthorpe)

> ...
> It seems both methods have their flaws; how about a routing table
> flag that allows go-fast-mode on a route-by-route basis.  This could
> be both manually set/cleared and automatically set/cleared by some, as yet
> undefined, mechanism.  This also avoids introducing cubersome hop-counting
> or RTT estimation mechanisms.  
>
> You would have to be careful about setting go-fast-mode for the local
> net if you have proxy-ARP on that net, but you could still hack around
> that by setting up specific routes for the proxyarped hosts.  It would
> be a bit messy, but that would serve you right ;-)

Set asside the question of how a router would compute the go-fast bit when
advertising routes.

On standards committee political grounds, there is absolutely no hope that
any interior routing protocol would ever carry a go-fast bit.  That implies
such that a routing-table bit would be either only a local, purely manual
configuration option, or it would a repository for a bit set by other
means such as hop counting, RTT estimating, or netmask peeking.  Thus,
we're back where we started, having only chosen a place to put the
information in some implementations.

In other words, the "some, as yet undefined, mechanism" by which routing
or anything else would figure out whether to go fast is the crux of the
issue.  Given that mechanism, where you cache its answer is an
implementation detail.


Vernon Schryver    vjs@rhyolite.com


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Reply-To: Venkat Padmanabhan <padmanab@CS.Berkeley.EDU>
From: Venkat Padmanabhan <padmanab@CS.Berkeley.EDU>
To: Vernon Schryver <vjs@calcite.rhyolite.com>
cc: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft 
In-reply-to: Your message of "Fri, 23 Oct 1998 08:25:59 MDT."
             <199810231425.IAA17032@calcite.rhyolite.com> 
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> > From: Joe Touch <touch@ISI.EDU>
> 
> > > You want to avoid slow-start and slow-restart and relax the 2:1
> > > segment:ACK ratio exactly when the hosts are close to each other,
> > > which to say, when the RTT is small.
> >
> > RTT may be correlated (it is debatable how closely); hopcount is
> > much more meaningful. If you can get there in 0 hops, then there
> > is no router, and you shouldn't be doing these tricks. That would 
> > require a traceroute-like test, out-of-band to the regular TCP
> > packets.
> 
> I meant to point out in my first message Thursday that traceroute cannot
> detectect all of the forwarding boxes on which there is often congestion
> that needs slow start.  Consider bridges.
> 
> I disagree that hopcount is more meaningful than RTT.  Whether there are
> 0 or 10 hops is not what determines whether slow start and ACK pacing
> are important, but whether there are hops where congestion should be avoided.
> For the bad hop count 0 case, consider a host that is able to run far
> faster than 100 Mbit/sec and that is taking 7000 HTTP hits/second, and
> also running a a few long term TCP connections to nearby systems, such as
> for syslog.  Those local TCP links need slow start and the host needs RED
> or similar on its local interfaces because its interfaces are congested.
> 
> I claim those cases get more accurate answers by looking at the RTT.
> 
> 

I disagree with your statement that it is irrelevant whether there are 
0 hops or 10 hops. I agree with Joe Touch that having a 0-hop path (that should
actually be "1-hop", but oh well...) is significant. I believe slow
start should be skipped *only if* there is a 0-hop path between source and
destination. Such a path is fundamentally different from all other paths
in the following sense: if the MAC protocol grants the source host access
to the channel, then the source "owns" the channel for a certain length
of time. During this period, the source can be sure that the entire
bandwidth of the channel is exclusively available to it, so there is
no reason to probe for bandwidth a la slow-start. 

The situation is entirely different when a connection traverses more than 
one link (i.e., 1-hop or greater, according to the terminology for this
discussion thread) no matter what the RTT is.

Of course, 0-hop doesn't necessarily mean that it is
okay to skip slow-start (i.e., we can't make an *if and only if* assertion).
A slow receiver could get overwhelmed by a fast sender and drop packets.
This is not fundametally different from congestion at an intermediate
router in the multi-hop case.

Therefore, there is a need for a per-host flag to indicate whether
or not it is okay to skip slow-start (as Sam Manthorpe suggests).
Of course, how these flags would be set is an open issue.

-Venkat





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From: sm@bossette.engr.sgi.com (Sam Manthorpe)
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Subject: Re: 'Slow-Start Restart after Idle' draft
To: vjs@calcite.rhyolite.com (Vernon Schryver)
Date: Fri, 23 Oct 1998 10:40:20 -0700 (PDT)
Cc: tcp-impl@cthulhu.engr.sgi.com
In-Reply-To: <199810231659.KAA17788@calcite.rhyolite.com> from "Vernon Schryver" at Oct 23, 98 10:59:57 am
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Vernon Schryver wrote:
> 
> > From: sm@bossette.engr.sgi.com (Sam Manthorpe)
> 
> > ...
> > It seems both methods have their flaws; how about a routing table
> > flag that allows go-fast-mode on a route-by-route basis.  This could
> > be both manually set/cleared and automatically set/cleared by some, as yet
> > undefined, mechanism.  This also avoids introducing cubersome hop-counting
> > or RTT estimation mechanisms.  
> >
> > You would have to be careful about setting go-fast-mode for the local
> > net if you have proxy-ARP on that net, but you could still hack around
> > that by setting up specific routes for the proxyarped hosts.  It would
> > be a bit messy, but that would serve you right ;-)
> 
> Set asside the question of how a router would compute the go-fast bit when
> advertising routes.
>
> [...]
>
> On standards committee political grounds, there is absolutely no hope that
> any interior routing protocol would ever carry a go-fast bit.  That implies

I can't immediately see any reason why you would even want to communicate
this information in a routing protocol, but I haven't thought about it
in any depth yet.

> In other words, the "some, as yet undefined, mechanism" by which routing
> or anything else would figure out whether to go fast is the crux of the
> issue.  Given that mechanism, where you cache its answer is an
> implementation detail.

I wasn't suggesting this is an alternative to an RTT based, or hop-count 
based or whatever-based mechanism, rather as a supplement to one of these.

The way I interpreted the ongoing discussion is to find _automatic_
mechanisms that would become part of a TCP implementation to determining
whether to `go fast' or not.  The routing table flag suggestion enables
you to manually configure go-fast mode and thus takes care of all
these `special' cases (proxy-ARP, LANE, bridges) that have been cited
as examples of why automated-scheme-x will not work.

It would be an implementational choice of whether you employ some automated
scheme of setting this bit, based on hop counts, RTT or whatever...

-- Sam

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Sam Manthorpe, SGI.  tel: (650)933-2856 fax: (650)932-2856  sm@engr.sgi.com


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Date: Fri, 23 Oct 1998 12:12:16 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> From: Venkat Padmanabhan <padmanab@cs.berkeley.edu>

> ...
> I disagree with your statement that it is irrelevant whether there are 
> 0 hops or 10 hops. I agree with Joe Touch that having a 0-hop path (that should
> actually be "1-hop", but oh well...) is significant. I believe slow
> start should be skipped *only if* there is a 0-hop path between source and
> destination. Such a path is fundamentally different from all other paths
> in the following sense: if the MAC protocol grants the source host access
> to the channel, then the source "owns" the channel for a certain length
> of time. During this period, the source can be sure that the entire
> bandwidth of the channel is exclusively available to it, so there is
> no reason to probe for bandwidth a la slow-start. 

That model of how a LAN works sounds nice and fits the textbooks, but it
is neither entirely accurate nor relevant.  To see its accuracy, consider
CSMA/CD during the first slot-time.  It would be relevant just in case
such ownership affected how TCP works.  The reason for a go-fast mode has
nothing to do with property ownership, but purely with whether TCP/IP runs
faster.  (For TCP, "speed" covers "reliability".)

> The situation is entirely different when a connection traverses more than 
> one link (i.e., 1-hop or greater, according to the terminology for this
> discussion thread) no matter what the RTT is.

Everything is different from everything else, but the only differences
that matter are those that matter.  The difference that matters here is
that paths (not merely LANs) without significant congestion from other
hosts or potential congestion at bottlenecks do not benefit and are harmed
by slow-start, slow-restart, and the 2:1 ACK:segment ratio.  That fact is
the only justification for a fast mode.  That harm has nothing to do with
whether the link is 0/1 or 100 hops long, although it is far less likely
with more hops.  That characterization, "without significant congestion
or bottlenecks," is independent of whether some MAC protocol has granted
some ownership, even where that notion makes sense.

Note that:
    - congestion that ought to be managed with slow-start can happen
       on the sending side of a host-LAN-host link when the sending host
       is much faster than the LAN, and the host-LAN interface is one of
       those bottlenecks.  For a fun, try a bunch of simultaneous ttcp or
       netperf sessions on a fast host using 100BASE-T or FDDI, 60 KByte
       default TCP windows, and less than 1 MByte of output buffering in
       the driver.  For that matter, try it with one ttcp session using
       a MByte TCP window but less than 1 MB drive buffering.

    - a path without congestion or bottlenecks but that has errors is also
       harmed by slow-(re)start.  If you know that there is no other
       traffic and no bottlenecks, then you do NOT want to shrink the
       transmit window when you detect loss, because you know the loss
       had nothing to do with congestion.

    - router and bridges (and therefore hops) tend to go together with
       connecting slower and faster links in a path, and so with
       bottlenecks that need slow-start.  So hops, if they were detectable,
       would be a reasonable go-fast mode switch.  But hops) are not
       detectable (briges, except by looking at the RTT and then only
       much of the time.

    - what I'm saying now is slightly contrary to my claim this morning
       about satellite links.  A link with 500 ms RTT, that is carrying
       only one TCP/IP connection, and where all of the links in the path
       have the same bit rate does not want slow-(re)start or any other
       congestion avoidance mechanism, because there is no congestion to
       avoid (expcept perhaps at the sending host).

       However, a satellite link or any path with an RTT larger than about
       10 ms is likely to involve multiple segments including at least
       one where the bit rate is far slower than the bit rate at the
       sending host, and so a bottleneck that wants slow-start.

> Of course, 0-hop doesn't necessarily mean that it is
> okay to skip slow-start (i.e., we can't make an *if and only if* assertion).
> A slow receiver could get overwhelmed by a fast sender and drop packets.
> This is not fundametally different from congestion at an intermediate
> router in the multi-hop case.

A slow receiver is less likely to produce a fast RTT, and in any case,
had better not be dropping any TCP/IP packets.  That's what the TCP
window is all about.  However, there is the sending-host case.


> Therefore, there is a need for a per-host flag to indicate whether
> or not it is okay to skip slow-start (as Sam Manthorpe suggests).
> Of course, how these flags would be set is an open issue.

A per-host, purely manual, violate-the-standard-and-go-fast flag is always
possible, but is also outside the scope of of a standard.


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 15:10:43 1998
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To: Vernon Schryver <vjs@calcite.rhyolite.com>
Cc: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
In-reply-to: Your message of Fri, 23 Oct 1998 09:28:31 PDT.
Date: Fri, 23 Oct 1998 11:25:04 PDT
From: Vern Paxson <vern@ee.lbl.gov>
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> Lots of ACKs are a Very Bad Thing on Ethernets.

Please back this up with further analysis.  I've put a tcpdump trace in

	ftp://ftp.ee.lbl.gov/.vp-100mbps.trace

This is an rcp along a 100 Mbps fast Ether link, with sufficiently large
windows to fill it.  The receiver for the most part acks every-second,
sometimes acking every third, and there's a periodic glitch in which it
generates stretch acks (doesn't affect the throughput).  Once the connection
reaches steady-state, it sustains upwards of 11 MB/s.

		Vern


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 15:12:44 1998
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From: sm@bossette.engr.sgi.com (Sam Manthorpe)
Message-Id: <199810231859.LAA06319@bossette.engr.sgi.com>
Subject: Re: 'Slow-Start Restart after Idle' draft
To: vjs@calcite.rhyolite.com (Vernon Schryver)
Date: Fri, 23 Oct 1998 11:59:00 -0700 (PDT)
Cc: tcp-impl@cthulhu.engr.sgi.com
In-Reply-To: <199810231826.MAA18096@calcite.rhyolite.com> from "Vernon Schryver" at Oct 23, 98 12:26:34 pm
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Vernon Schryver wrote:
> 
> > From: sm@bossette.engr.sgi.com (Sam Manthorpe)
> 
> > ...
> > I can't immediately see any reason why you would even want to communicate
> > this information in a routing protocol, but I haven't thought about it
> > in any depth yet.
> 
> If it is not sent in routing packets, why should the go-fast-to-this-host
> bit be in the routing table?  I agree that's a nice place to put it, but
> it is not the only place.  The situation is very much like path MTU
> discovery.  Many people put the discovered MTU in the routing table, but
> that is not the only choice, as shown by some code on bonnie.engr.sgi.com.
> 
> 
> > > In other words, the "some, as yet undefined, mechanism" by which routing
> > > or anything else would figure out whether to go fast is the crux of the
> > > issue.  Given that mechanism, where you cache its answer is an
> > > implementation detail.
> >
> > I wasn't suggesting this is an alternative to an RTT based, or hop-count 
> > based or whatever-based mechanism, rather as a supplement to one of these.
> >
> > The way I interpreted the ongoing discussion is to find _automatic_
> > mechanisms that would become part of a TCP implementation to determining
> > whether to `go fast' or not.  The routing table flag suggestion enables
> > you to manually configure go-fast mode and thus takes care of all
> > these `special' cases (proxy-ARP, LANE, bridges) that have been cited
> > as examples of why automated-scheme-x will not work.
> 
> How does putting the per-host go-fast bit in the routing table entry
> for that host (presumably along with other things such as the manual MSS)
> help the proxy-ARP case?  As far as the host that would have the bit in
> its routing table is concerned (assuming a merged ARP-IP routing table),
> all peers look the same regardless of whether proxy-ARP is involved.
> 
> Manual go-fast bits are not practical.  Proofs by example of that claim
> are the manual MSS mechanism used among SGI and Cray boxes, and the
> popularity of the automated path MTU discovery mechanism.
> 
> Are you suggesting that if the MAC source address in an ARP entry does not
> match the contents of the ARP packet (after allowing for bit-order
> nonsense on FDDI), that the go-fast bit would be automatically turned off?
> And on by default otherwise?
> 
> > It would be an implementational choice of whether you employ some automated
> > scheme of setting this bit, based on hop counts, RTT or whatever...
> 
> I am still confused.  Are you saying more than that the 4.4BSD-Lite style
> routing table is a nicer place for a per-host go-fast bit than, for
> example, an independent kernel data structure?  Note that many systems
> do not have the 4.4BSD-Lite style routing table.

I'm thinking in terms of user-interface.  My point was simply that since
there seem to be a multitude of scenarios where the simple go-fast-detection
schemes discussed would fail, it would be desirable for a sysadmin to be able
to override any automatic decision on a net-by-net or host-by-host basis.
It also provides the possibility to enable/disable GFM (go fast mode)
in the complete absence of any automated mechanism.

So my argument is really orthogonal to the discussion on a nice scheme
for detecting when one should use go-fast mode.

Yes, my thinking is definitely BSD biased; how alien OSes would chose
to implement such a scheme would be for them to decide. (the word alien
is in no way meant to be derogatory, I am myself currently a legal alien).

-- Sam

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Sam Manthorpe, SGI.  tel: (650)933-2856 fax: (650)932-2856  sm@engr.sgi.com


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Reply-To: Venkat Padmanabhan <padmanab@CS.Berkeley.EDU>
From: Venkat Padmanabhan <padmanab@CS.Berkeley.EDU>
To: Vernon Schryver <vjs@calcite.rhyolite.com>
cc: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft 
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             <199810231812.MAA18027@calcite.rhyolite.com> 
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> > From: Venkat Padmanabhan <padmanab@cs.berkeley.edu>
> 
> > ...
> > I disagree with your statement that it is irrelevant whether there are 
> > 0 hops or 10 hops. I agree with Joe Touch that having a 0-hop path (that sh
ould
> > actually be "1-hop", but oh well...) is significant. I believe slow
> > start should be skipped *only if* there is a 0-hop path between source and
> > destination. Such a path is fundamentally different from all other paths
> > in the following sense: if the MAC protocol grants the source host access
> > to the channel, then the source "owns" the channel for a certain length
> > of time. During this period, the source can be sure that the entire
> > bandwidth of the channel is exclusively available to it, so there is
> > no reason to probe for bandwidth a la slow-start. 
> 
> That model of how a LAN works sounds nice and fits the textbooks, but it
> is neither entirely accurate nor relevant.  To see its accuracy, consider
> CSMA/CD during the first slot-time.  It would be relevant just in case
> such ownership affected how TCP works.  The reason for a go-fast mode has
> nothing to do with property ownership, but purely with whether TCP/IP runs
> faster.  (For TCP, "speed" covers "reliability".)
> 

I guess I am entirely missing what you are saying here...

What I am saying is that in the 0-hop case, the regulation would be
done by the MAC. TCP needn't. TCP would still have a flow-control
window, though, but that's a function of the socket buffer size set
by applications, and hence orthogonal.

> > The situation is entirely different when a connection traverses more than 
> > one link (i.e., 1-hop or greater, according to the terminology for this
> > discussion thread) no matter what the RTT is.
> 
> Everything is different from everything else, but the only differences
> that matter are those that matter.  The difference that matters here is
> that paths (not merely LANs) without significant congestion from other
> hosts or potential congestion at bottlenecks do not benefit and are harmed
> by slow-start, slow-restart, and the 2:1 ACK:segment ratio.  That fact is
> the only justification for a fast mode.  That harm has nothing to do with
> whether the link is 0/1 or 100 hops long, although it is far less likely
> with more hops.  That characterization, "without significant congestion
> or bottlenecks," is independent of whether some MAC protocol has granted
> some ownership, even where that notion makes sense.
> 

I can't contest the validity of the tautologies in the first sentence ;-)

I was saying that there is *fundamental* difference between 0-hop and
n-hop where n > 0. It is dangerous to assume that a path is lightly
loaded. Since the path is presumably shared by many hosts and no one
host has a complete picture of the load imposed by other hosts, what
is lightly loaded at one instant could be heavily loaded at another
(for instance, due to "flash crowds"). In other words, it is difficult
(in fact, fundametally flawed) to characterize a path as being
"without significant congestion or bottlenecks" given the shared
nature of the network. 

However, in the 0-hop case, the MAC protocol
gives a host exclusive access to the channel for a certain length of
time. There is no competition above the MAC layer, except between
multiple connections on a host. Of course, there is competition
at the MAC layer and we would want it to be resolved fairly, etc.,
but that is an orthogonal issue.

As a performance optimization one certainly might want to take advantage
of times when a multi-hop network is in fact lightly loaded. But I believe such
aggressive use of the network must happen only in conjunction with 
mechanisms that avoid/alleviate disaster when the "lightly loaded"
assumption is incorrect. One possible mechanism is priority dropping
which I discuss in the "TCP fast start" chapter of my thesis:
http://www.cs.berkeley.edu/~padmanab/phd-thesis.html


> Note that:
>     - congestion that ought to be managed with slow-start can happen
>        on the sending side of a host-LAN-host link when the sending host
>        is much faster than the LAN, and the host-LAN interface is one of
>        those bottlenecks.  For a fun, try a bunch of simultaneous ttcp or
>        netperf sessions on a fast host using 100BASE-T or FDDI, 60 KByte
>        default TCP windows, and less than 1 MByte of output buffering in
>        the driver.  For that matter, try it with one ttcp session using
>        a MByte TCP window but less than 1 MB drive buffering.

Sure, it can happen at the sender. I've seen it several times on a
WaveLAN network. It can happen at the receiver, too.

 
> A per-host, purely manual, violate-the-standard-and-go-fast flag is always
> possible, but is also outside the scope of of a standard.

A per-host flag need not be set manually. A host can learn from past 
experience and set it automatically.

-Venkat





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Date: Fri, 23 Oct 1998 12:26:34 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <199810231826.MAA18096@calcite.rhyolite.com>
To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> From: sm@bossette.engr.sgi.com (Sam Manthorpe)

> ...
> I can't immediately see any reason why you would even want to communicate
> this information in a routing protocol, but I haven't thought about it
> in any depth yet.

If it is not sent in routing packets, why should the go-fast-to-this-host
bit be in the routing table?  I agree that's a nice place to put it, but
it is not the only place.  The situation is very much like path MTU
discovery.  Many people put the discovered MTU in the routing table, but
that is not the only choice, as shown by some code on bonnie.engr.sgi.com.


> > In other words, the "some, as yet undefined, mechanism" by which routing
> > or anything else would figure out whether to go fast is the crux of the
> > issue.  Given that mechanism, where you cache its answer is an
> > implementation detail.
>
> I wasn't suggesting this is an alternative to an RTT based, or hop-count 
> based or whatever-based mechanism, rather as a supplement to one of these.
>
> The way I interpreted the ongoing discussion is to find _automatic_
> mechanisms that would become part of a TCP implementation to determining
> whether to `go fast' or not.  The routing table flag suggestion enables
> you to manually configure go-fast mode and thus takes care of all
> these `special' cases (proxy-ARP, LANE, bridges) that have been cited
> as examples of why automated-scheme-x will not work.

How does putting the per-host go-fast bit in the routing table entry
for that host (presumably along with other things such as the manual MSS)
help the proxy-ARP case?  As far as the host that would have the bit in
its routing table is concerned (assuming a merged ARP-IP routing table),
all peers look the same regardless of whether proxy-ARP is involved.

Manual go-fast bits are not practical.  Proofs by example of that claim
are the manual MSS mechanism used among SGI and Cray boxes, and the
popularity of the automated path MTU discovery mechanism.

Are you suggesting that if the MAC source address in an ARP entry does not
match the contents of the ARP packet (after allowing for bit-order
nonsense on FDDI), that the go-fast bit would be automatically turned off?
And on by default otherwise?

> It would be an implementational choice of whether you employ some automated
> scheme of setting this bit, based on hop counts, RTT or whatever...

I am still confused.  Are you saying more than that the 4.4BSD-Lite style
routing table is a nicer place for a per-host go-fast bit than, for
example, an independent kernel data structure?  Note that many systems
do not have the 4.4BSD-Lite style routing table.


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 15:19:17 1998
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Subject: Re: 'Slow-Start Restart after Idle' draft
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> > Lots of ACKs are a Very Bad Thing on Ethernets.
> 
> Please back this up with further analysis.

There's lots of variables that affect the Badness of Frequent ACKs on
Ethernets. Half-duplex or full. Shared or switched. Network load. But
generally speaking, the more frames used for ACKs, the less frames can
be used for data. If you've got plenty of slots, there's no difference.
If you're tight on slots, there's lots of difference. Lots of nodes
sending lots of ACKs will definitely cut your data delivery rate.

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 15:19:19 1998
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Message-ID: <3630CF71.E8A61097@cup.hp.com>
Date: Fri, 23 Oct 1998 11:48:17 -0700
From: Rick Jones <raj@cup.hp.com>
Organization: HP 9000 Network Performance
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Subject: Re: 'Slow-Start Restart after Idle' draft
References: <199810231825.LAA11529@daffy.ee.lbl.gov>
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> > Lots of ACKs are a Very Bad Thing on Ethernets.
> 
> Please back this up with further analysis.  I've put a tcpdump trace in

Well, these days, the cycle processing of an ACK is little different
from the cycle processing of a data packet. Systems now have
copy-avoidance mechanisms, and interface cards with checksum offload
(CKO) functionality.

This makes most if not all of the CPU costs for say a web server
per-packet costs. Reducing the number of packets (eg ACKs) then
increases the capacity of the server.

An arbitrary reduction in ACKs is probably not a good idea. However, a
receiver which is able to figure-out when things are not flowing
smoothly and so ACKs with apropriate frequency at apropriate times would
probably not be such a bad thing.

A fixed choice of an ACK for every two MSS segments is merely a very
simple, straightforward, and "cheap" heuristic to arrive at an ACK rate
apropriate for the traffic conditions. I would certainly like
implementations to be free to use other equally robust mechanisms which
might happen to be cheaper in the free and clear case (such as LANs or
unloaded links). So, if future RFC's can state requirements of behaviour
without forcing a given ACK policy that might be best.

rick jones
-- 
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to email, or post, but please do not do both...
my email address is raj in the cup.hp.com domain...


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Date: Fri, 23 Oct 1998 12:45:39 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <199810231845.MAA18174@calcite.rhyolite.com>
To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> From: Vern Paxson <vern@ee.lbl.gov>
>
> > Lots of ACKs are a Very Bad Thing on Ethernets.
>
> Please back this up with further analysis.  I've put a tcpdump trace in
>
> 	ftp://ftp.ee.lbl.gov/.vp-100mbps.trace
>
> This is an rcp along a 100 Mbps fast Ether link, with sufficiently large
> windows to fill it.  The receiver for the most part acks every-second,
> sometimes acking every third, and there's a periodic glitch in which it
> generates stretch acks (doesn't affect the throughput).  Once the connection
> reaches steady-state, it sustains upwards of 11 MB/s.

I cannot find a file there.

However, I bet that at least one of the following is true, any one of
which is sufficent to sidestep the problem:
  - you used a TCP window size of less than 16 K
  - you used half-duplex MAC chips with BLAM (except that I didn't
     think any were commercially available)
  - you used full-duplex EThernet, which is not really Ethernet since
     it has nothing more to do with CSMA/CD than 100VG-AnyLAN, 802.5,
     HIPP, ATM, FC, or FDDI.
  - you used half-duplex MAC chips that violate 802.3 as the old AMD 7990
     LANCE did, by being "polite" and not transmitting immediately after
     deferring.

If you use a TCP window of >= 32K, 802.3 compliant MAC chips with BEG
instead of BLAM, and fast hosts, you'll get about 8 MByte/sec instead of
11 MByte/sec.

The problem is that if both systems are transmitting about the same
number of packets, and they are using 802.3 compliant MACs with BEB and
not BLAM, then one of them will capture the Ethernet, and the other will
binary-exponential-backoff into the weeds and even suffer excessive
collision errors.  What usually happens is that the TCP sender blasts a
full window, starves for ACKs because the TCP receiver has BEB'ed into
waiting for 512 slot times, and goes to sleep.  Eventually the TCP receiver
wakes up, sends one of its queued ACK's, and everything starts rolling
again, for a while.  If you use a small TCP window, the hiccups still
exist, but they are much smaller.  If the TCP sender does capture the wire
with an 8K window, it runs out of transmit window before the TCP receiver
has backed off to death, and so the TCP receiver is able to get its ACK
through without having the wire being idle for most of 1024 slot times.

If you abuse the 4.3BSD-Reno TCP code to not send as many ACKs, the problem
goes away, at least for TCP windows of 61,440.

I say this based on years of glaring at a <<LOT>> of traces, as well as
having personally abused 4.3BSD-Reno.  I could also tell you about custom
silicon in MAC chips intended by design to violate IEEE 802.3 to deal with
this problem, but it's supposed to be secret.

There is also Mart Molle's extensive writings on the issue.


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 15:34:11 1998
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To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
In-reply-to: Your message of Fri, 23 Oct 1998 12:45:39 PDT.
Date: Fri, 23 Oct 1998 12:24:42 PDT
From: Vern Paxson <vern@ee.lbl.gov>
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> From:  Vernon Schryver <vjs@calcite.rhyolite.com>
>
> > 	ftp://ftp.ee.lbl.gov/.vp-100mbps.trace
> ...
> I cannot find a file there.

Some browsers don't like hidden files - if so, ordinary FTP ought to
fetch it.

> However, I bet that at least one of the following is true, any one of
> which is sufficent to sidestep the problem:
>   - you used a TCP window size of less than 16 K

Receiver offered 17K, sender had about 16K in flight.

>   - you used half-duplex MAC chips with BLAM (except that I didn't
>      think any were commercially available)

I don't know, but we're running stock hardware.

>   - you used full-duplex EThernet, which is not really Ethernet since

No, we have that on one of our machines, but not this one.

>   - you used half-duplex MAC chips that violate 802.3 as the old AMD 7990
>      LANCE did, by being "polite" and not transmitting immediately after
>      deferring.

I don't know - it's fairly new hardware, though.

> The problem is that if both systems are transmitting about the same
> number of packets, and they are using 802.3 compliant MACs with BEB and
> not BLAM, then one of them will capture the Ethernet, and the other will
> binary-exponential-backoff into the weeds and even suffer excessive
> collision errors.

It would appear that acking frequently could actually help with this.  The
window moves in small increments, so there are lulls in the sender's
transmission, during which more acks can be sent.  It seems that infrequent
acking could rather aggravate the capture effect: it leads to large bursts of
back-to-back data packets, so a lot more opportunity for the receiver's
collision timer to back off into the weeds.  I'd be interested in
understanding what's missing from this analysis.

		Vern


From owner-tcp-impl@cthulhu.engr.sgi.com  Fri Oct 23 15:54:27 1998
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Date: Fri, 23 Oct 1998 15:46:40 -0400 (EDT)
From: Scott Bradner <sob@harvard.edu>
Message-Id: <199810231946.PAA17986@newdev.harvard.edu>
To: tcp-impl@cthulhu.engr.sgi.com, vern@ee.lbl.gov
Subject: Re: 'Slow-Start Restart after Idle' draft
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>   - you used half-duplex MAC chips with BLAM (except that I didn't
>      think any were commercially available)  

I had not heard that such chips were in products - has BLAM even been
endorsed by IEEE?

Scott


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Date: Fri, 23 Oct 1998 14:20:57 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <199810232020.OAA18403@calcite.rhyolite.com>
To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> From: Venkat Padmanabhan <padmanab@cs.berkeley.edu>

> ...
> I guess I am entirely missing what you are saying here...

My point is that classic, Van Jacobson TCP congestion avoidance and control
should be applied only where it does more good than harm.


> What I am saying is that in the 0-hop case, the regulation would be
> done by the MAC. TCP needn't. TCP would still have a flow-control
> window, though, but that's a function of the socket buffer size set
> by applications, and hence orthogonal.

That is false in both directions.  First is the case where there is a
bottleneck within a host between it and its LAN that benefits from
slow-start etc.  Second is the case of today's typical bridged network.
If you insist on slow-start whenever there is a forwarder in the path,
then you insist on slow-start practically everywhere.  Host-to-host
MAC-layer flow-control has been almost universally destroyed by so called
"smart hubs" and "switches."


> ...
> I was saying that there is *fundamental* difference between 0-hop and
> n-hop where n > 0.  It is dangerous to assume that a path is lightly
> loaded. Since the path is presumably shared by many hosts and no one
> host has a complete picture of the load imposed by other hosts, what
> is lightly loaded at one instant could be heavily loaded at another
> (for instance, due to "flash crowds").

Those words apply equally to a WAN as to a LAN.  "Flash crowds" happen
on LANs, although they are vastly less frequent than the "switch"
salescritters and trade-rag consultants would like suckers to believe.

>                                        In other words, it is difficult
> (in fact, fundametally flawed) to characterize a path as being
> "without significant congestion or bottlenecks" given the shared
> nature of the network. 

When a path is not "without significant congestion or bottlenecks", then
it obvious is not, and when it is, it is.  How in the world can it be
"fundamentally flawed" to charactize a path as it is?  Do you mean that
the characterization can change?  If so, that problem applies even more
to single-hop LANs than to WANs.  The instant a bulk transfer starts, with
or without go-fast mode, a single hop LAN is no longer "without significant
congestion or bottlenecks" to other traffic.


> However, in the 0-hop case, the MAC protocol
> gives a host exclusive access to the channel for a certain length of
> time. There is no competition above the MAC layer, except between
> multiple connections on a host. Of course, there is competition
> at the MAC layer and we would want it to be resolved fairly, etc.,
> but that is an orthogonal issue.

Those words do not apply to no LAN that I can think of.  They obviously
do not apply to any LAN that involves a shared medium.  "Flash crowds" on
on LANs are familiar to anyone who has looked at an Ethernet or token
ring, not mention implemented applications that did things like update
files on many systems at the stroke of 8 a.m.  Access to the Ethernet
carrier or the 802.5 or FDDI token is just as vulnerable to contention as
access to a leased line or the buffer in a router talking to a leased
line.  Even LANs that are based enirely on non-blocking switches, such as
ATM and HIPPI do not fit your model, because of congestion in the host
interfces.

How do you resolve congestion at the MAC layer without slowing something
down?  If the MAC layer simply slows down but your TCP code keeps pumping,
what must happen?  Hint: my point about what happens with
`ttcp -b 1048576 -l 1048576` between a pair of hosts was not theoretical.
Or consider NFSv3/TCP/IP; it does not take many NFS connections with the
default 32K block size of NFSv3 to overwhelm the buffering in most drivers.


> As a performance optimization one certainly might want to take advantage
> of times when a multi-hop network is in fact lightly loaded. But I believe such
> aggressive use of the network must happen only in conjunction with 
> mechanisms that avoid/alleviate disaster when the "lightly loaded"
> assumption is incorrect. One possible mechanism is priority dropping
> which I discuss in the "TCP fast start" chapter of my thesis:
> http://www.cs.berkeley.edu/~padmanab/phd-thesis.html

Anything involving "dropping" seems irrelevant to the issue at hand.

I favor turning on go-fast mode only when it appears from recent RTT
measurements that it is reasonable, and giving up on lossy networks by
limiting go-fast mode to only turning off slow-start and ACK pacing and
retaining slow-restart.  Turning off slow-start but keeping slow-restart
(which I take to mean standard congestion avoiding TCP/IP but with an
initial full-sized transmit window) has the nice property of responding
appropriately if a "flash crowd" appears.  It has the bad property of
making flash crowds worse, and so should not be used unless there is
evidence things are not bad.  The only way I see to get that evidence is
by looking at the recent RTT.

> ...
> > A per-host, purely manual, violate-the-standard-and-go-fast flag is always
> > possible, but is also outside the scope of of a standard.
>
> A per-host flag need not be set manually. A host can learn from past 
> experience and set it automatically.

Exactly how the host learns and so automatically sets the per-host flag
is the whole point!  The etiology of 0- vs. multi-hop networks might be
interesting, but the issue is how to set that per-host flag automatically!

So far, netmasks, hop counting, and  RTT's have been suggested.  Do
you have another way?

  - netmasks do not work because proxy-ARP, bridges, and VLSM would
     cause false positives with dire consequences on literally millions
     of networks (PPP stubbs connected to home computeres).

  - hop counting is simply impossible.  How does TCP discover how many
     hops are in the path?  Does it delay sending the SYN-ACK until it
     probes with `traceroute`?  How does that detect bridges?
     How can you detect >1 hop paths except with the RTT?

  - RTT's (e.g. from a previous connection or measured during the
     3-way handshake) are imperfect, possibly producing false positives
     on, e.g. OC-12 metro links, or on bursty LANs.   Still, what 
     alternative is there?


Vernon Schryver    vjs@rhyolite.com


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Date: Fri, 23 Oct 1998 14:56:16 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <199810232056.OAA18533@calcite.rhyolite.com>
To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> From: Vern Paxson <vern@ee.lbl.gov>

> ...
> > > 	ftp://ftp.ee.lbl.gov/.vp-100mbps.trace
> > ...
> > I cannot find a file there.
>
> Some browsers don't like hidden files - if so, ordinary FTP ought to
> fetch it.

Netscape delivered garbage, so I tried `ls` via FTP, which denied the file
existed.  Having more than once received large (many MByte) traces by
email, I was a little shy about fetching it blindly.  I just now did
anyway, and found it was < 200KBytes, but it's some kind of binary.  I
guess I could try `tcpdump -r`...yes, I see that produces things.

But you answer the point 

> ...
> >   - you used a TCP window size of less than 16 K
>
> Receiver offered 17K, sender had about 16K in flight.

That's close enough to reduce the problem to tolerable levels.

You might also have BLAM in one of the MAC chips.  Now that I think
back, I think I heard rumors one of the 100 MHz chips has BLAM.


> ...
> > The problem is that if both systems are transmitting about the same
> > number of packets, and they are using 802.3 compliant MACs with BEB and
> > not BLAM, then one of them will capture the Ethernet, and the other will
> > binary-exponential-backoff into the weeds and even suffer excessive
> > collision errors.
>
> It would appear that acking frequently could actually help with this.  The
> window moves in small increments, so there are lulls in the sender's
> transmission, during which more acks can be sent.  It seems that infrequent
> acking could rather aggravate the capture effect: it leads to large bursts of
> back-to-back data packets, so a lot more opportunity for the receiver's
> collision timer to back off into the weeds.  I'd be interested in
> understanding what's missing from this analysis.

In summary, back-to-back packets are not the problem.  The problem is the
collison between to long trains of packets.  If the TCP receiver is not
sending ACKs frequently, then it is less likely to collide with the TCP
sender, and so it's less likely to back-off into the weeds.


If the TCP receiver responds to the second TCP segment with an ACK (and
is fast), it is certain to collide with the third segment.  If it is
unlucky and loses the first 2 or 3 collisions while trying to send that
first ACK, it will probably back-off into the weeds (backoff counter=10)
or wait until the TCP sender stops sending, whichever comes first.  TCP
sender will continue sending and occassionally colliding, as the receiver's
timer occassionally and less and less frequently expires, until it runs
out of window.

With your TCP window of ~17K, the TCP sender will never send more than
~11 packets before starving for an ACK.  That means the receiver will
never suffer more than 8, and at 100 MHz if it is not super fast, probably
not more than 5 or 6 consecutive collisions on any ACK, so its back-off
counter will be only 5, so it will wait only about 16 slot times, which
is less than a packet-time.

Note that the probability of the TCP receiver losing the wire is less than
87% on each ACK.  More than 87% of the time, it will win the first
collision or one of the early subsequent collisions.  (Specifically, on
each packet, it will not be locked out with probability .5+.25+.125+...)

Now consider a TCP window of 60*1024.  That's a collision a train of 40
packets with a train of 20 ACKs under normal rules.  Either the TCP sender
or the TCP receiver is likely to capture the wire, and the other is like
to have a back-off counter equal to 10, and so wait up to 1024 slot times
or the equivalent of 65KBytes of data after the other has stopped sending.
Worse, there's a good chance that the loser will suffer 16 collisions and
drop a packet, and if the loser is the TCP sender, that really hurts.


Now consider reducing the ACK ratio by always delaying ACKs, and not just
after partial segments.  If the TCP receiver does send its first ACK for
the arriving train of segments until 10 or 15 segments have arrived, it
is less likely to be locked off the wire by its own back-off counter. 
Or if its train of ACKs is only 5 intead of 20, it is not nearly as certain
to have one of those ACKs get unlucky and run up its back-off counter.


Note that the point Rick Jones made is also valid, but less dramatic than
the 30% throughput loss due to the Ethernet Capture Effect.  Host
performance is a generally function of packet counts, so more packets
always costs speed, even if they cost no bandwidth, as with full-duplex
media.


Vernon Schryver    vjs@rhyolite.com


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Date: Fri, 23 Oct 1998 15:02:47 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Scott Bradner <sob@harvard.edu>

> >   - you used half-duplex MAC chips with BLAM (except that I didn't
> >      think any were commercially available)  
>
> I had not heard that such chips were in products - has BLAM even been
> endorsed by IEEE?

What does "endorsed by the IEEE" mean?

The last I heard from Mart Molle was quite a while ago and that 802.3v
(u? x? whatever) was chugging along, but that the 10 MHz chip vendors were
not happy with the prospect of either fixing their ancient designs or
shipping slower chips.  Never mind that the really ancient AMD LANCE beats
them all, except when you turn on their various really nasty (e.g. linear
backoff) work-arounds or in a case or two, their emulation of the LANCE
bug.

On reflection, I think I was told at my former job that one of the 100
MHz MACs in one or another of the products would eventually have BLAM.
However, don't take that as gospel, since I was never directly involved,
never even saw a 100 MHz or GHz chip spec, and besides, it might be a secret.


Vernon Schryver    vjs@rhyolite.com


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From: Venkat Padmanabhan <padmanab@CS.Berkeley.EDU>
To: Vernon Schryver <vjs@calcite.rhyolite.com>
cc: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft 
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> 
> > What I am saying is that in the 0-hop case, the regulation would be
> > done by the MAC. TCP needn't. TCP would still have a flow-control
> > window, though, but that's a function of the socket buffer size set
> > by applications, and hence orthogonal.
> 
> That is false in both directions.  First is the case where there is a
> bottleneck within a host between it and its LAN that benefits from
> slow-start etc.  Second is the case of today's typical bridged network.
> If you insist on slow-start whenever there is a forwarder in the path,
> then you insist on slow-start practically everywhere.  Host-to-host
> MAC-layer flow-control has been almost universally destroyed by so called
> "smart hubs" and "switches."

Regarding your first point, you are basically saying that congestion
is can happen at the sender (a point you made in your previous email
and I agreed with in my previous one). But the problem is a local
one that is easy to resolve, and in fact there is a reasonable fix in the
BSD stack. If the interface queue fills up, TCP's call to ip_output()
returns an error code and the sender slows down ("quench") instantaneously.
When there are multiple connections originating from a host, they
should coordinate their actions so that if one of them experiences
a "buffer full" condition, the others should learn from it. (In my
thesis, I worked on doing this in certain special cases.)

Experiencing a "buffer full" condition on your local stack is much
easier to deal with than having your packets lost elsewhere in the
network. For one thing, there is no uncertainty about whether the
buffer is in fact full (and hence dropping packets) or whether there is
only a temporary RTT spike (i.e., large delay but no drops).

Regarding your second point, yes, if there is a forwarder in the
path so that the sender does not have know the load on a LAN
segment on the other side of a bridge than itself, then it should 
do slow start.

> 
> 
> > ...
> > I was saying that there is *fundamental* difference between 0-hop and
> > n-hop where n > 0.  It is dangerous to assume that a path is lightly
> > loaded. Since the path is presumably shared by many hosts and no one
> > host has a complete picture of the load imposed by other hosts, what
> > is lightly loaded at one instant could be heavily loaded at another
> > (for instance, due to "flash crowds").
> 
> Those words apply equally to a WAN as to a LAN.  "Flash crowds" happen
> on LANs, although they are vastly less frequent than the "switch"
> salescritters and trade-rag consultants would like suckers to believe.
> 

You are absolutely right. But the difference I was pointing out was
not that flash crowds happen in one but not in the other. It was that
in the 0-hop case, the MAC guarantees the sender access to the wire
from itself to the receiver. No such guarantees are made for an
n-hop path. 

Of course, as we have agreed, there can still be a problem due to
slowness of the receiver or sender. That is why in my first email,
I said that a 0-hop path was an *only if* requirement but not an
*if and only if* one.

> >                                        In other words, it is difficult
> > (in fact, fundametally flawed) to characterize a path as being
> > "without significant congestion or bottlenecks" given the shared
> > nature of the network. 
> 
> When a path is not "without significant congestion or bottlenecks", then
> it obvious is not, and when it is, it is.  How in the world can it be
> "fundamentally flawed" to charactize a path as it is?  Do you mean that
> the characterization can change?  If so, that problem applies even more
> to single-hop LANs than to WANs.  The instant a bulk transfer starts, with
> or without go-fast mode, a single hop LAN is no longer "without significant
> congestion or bottlenecks" to other traffic.
> 

Yes, I mean it can change, and quite unpredicatably in the worst case.
In the LAN case, as I have explained, the sender knows through *local*
information that the LAN has a high load, so can slow down its
sending rate without aggravating the congestion in the LAN.

> 
> > However, in the 0-hop case, the MAC protocol
> > gives a host exclusive access to the channel for a certain length of
> > time. There is no competition above the MAC layer, except between
> > multiple connections on a host. Of course, there is competition
> > at the MAC layer and we would want it to be resolved fairly, etc.,
> > but that is an orthogonal issue.
> 
> Those words do not apply to no LAN that I can think of.  They obviously
> do not apply to any LAN that involves a shared medium.  "Flash crowds" on
> on LANs are familiar to anyone who has looked at an Ethernet or token
> ring, not mention implemented applications that did things like update
> files on many systems at the stroke of 8 a.m.  Access to the Ethernet
> carrier or the 802.5 or FDDI token is just as vulnerable to contention as
> access to a leased line or the buffer in a router talking to a leased
> line.  Even LANs that are based enirely on non-blocking switches, such as
> ATM and HIPPI do not fit your model, because of congestion in the host
> interfces.

Perhaps you misunderstood what I meant by a "certain length of time".
I was referring to the time granularity of MAC allocations, not minutes
or hours. An example may clarify the issue. Suppose that host A has a single 
TCP connection to host B, and both A and B are on the same LAN segment.
Assuming that the end hosts are not bottlenecks and that TCP flow control
does not kick in, A should be sending to B as fast as its MAC is able
to grab the channel. Of course, at times such as "the stroke of 8 a.m.",
A's MAC will not be able to grab the channel very much because many
others will also be trying to grab the channel, so ... <on to the next point>.

> 
> How do you resolve congestion at the MAC layer without slowing something
> down?  If the MAC layer simply slows down but your TCP code keeps pumping,
> what must happen?  Hint: my point about what happens with
> `ttcp -b 1048576 -l 1048576` between a pair of hosts was not theoretical.
> Or consider NFSv3/TCP/IP; it does not take many NFS connections with the
> default 32K block size of NFSv3 to overwhelm the buffering in most drivers.
> 

My original characterization of TCP and the MAC as being orthogonal was
incorrect. As I have mentioned above, BSD TCP has a "quench" mechanism
that allows TCP to slow down instantly when its local interface queue
is full (for instance, because the MAC is not able to grab the channel very
much). I hope you will agree that dealing with such local
congestion is a whole lot easier than the general problem of dealing
with congestion "somewhere" in the network. I think it makes little
sense to use a general probing algorithm such as slow start is the only 
capacity that you are trying "match" is that of your local link/buffer. 

> 
> > As a performance optimization one certainly might want to take advantage
> > of times when a multi-hop network is in fact lightly loaded. But I believe 
such
> > aggressive use of the network must happen only in conjunction with 
> > mechanisms that avoid/alleviate disaster when the "lightly loaded"
> > assumption is incorrect. One possible mechanism is priority dropping
> > which I discuss in the "TCP fast start" chapter of my thesis:
> > http://www.cs.berkeley.edu/~padmanab/phd-thesis.html
> 
> Anything involving "dropping" seems irrelevant to the issue at hand.
> 
That's a strong statement! I thought we were discussing congestion
avoidance and control, and given that, I don't see how we can ignore
the possibility of packets getting dropped (however undesirable that
may be).

I'll try to explain how I use priority dropping in my answer to your
next question.
 
> > ...
> > > A per-host, purely manual, violate-the-standard-and-go-fast flag is alway
s
> > > possible, but is also outside the scope of of a standard.
> >
> > A per-host flag need not be set manually. A host can learn from past 
> > experience and set it automatically.
> 
> Exactly how the host learns and so automatically sets the per-host flag
> is the whole point!  The etiology of 0- vs. multi-hop networks might be
> interesting, but the issue is how to set that per-host flag automatically!
> 
> So far, netmasks, hop counting, and  RTT's have been suggested.  Do
> you have another way?
> 

The operative part of what I said was "learn from past experience".
Here's the outline of a simple scheme. It makes no assumptions
about hop counts, netmasks, etc.

1. If doing a transfer to another host for the first time in x minutes
(x TBD), do slow start.

2. If you've done a transfer in the past x minutes, start with window size
equal to where you left off the last time. Use timer to smooth out bursts.
Mark all packets, except for the first one, as "low priority". In the best
case, all the packets will go through fine and you will experience the
benefit of fast start. In a worse case (such as the networking having become
much more congested since the last time you used it), priority dropping
with shield (to an extent) other traffic from the ill-effects of your
aggressiveness.

In my work, I also developed some special loss detection/recovery algorithms 
for use during the fast start phase. If your are interested, you should
look at my thesis.

-Venkat




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Date: Fri, 23 Oct 1998 16:37:14 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <199810232237.QAA18911@calcite.rhyolite.com>
To: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> From: Venkat Padmanabhan <padmanab@cs.berkeley.edu>

> ...
> Regarding your second point, yes, if there is a forwarder in the
> path so that the sender does not have know the load on a LAN
> segment on the other side of a bridge than itself, then it should 
> do slow start.

Given the near universal installation of bridges, aren't you saying that
by your reconning, no go-fast switch is possible?  Recall, for example,
that every 100/10BASE-T hub is necessarily a bridge.


> ...
> Of course, as we have agreed, there can still be a problem due to
> slowness of the receiver or sender. That is why in my first email,
> I said that a 0-hop path was an *only if* requirement but not an
> *if and only if* one.

We disagree.  You keep saying that a 0-hop path is almost sacred but
others are vulgar and cannot use a go-fast mode.  Consider a private,
three-segment campus or metro network where the local links are 10 MHz
Ethernet and the long-haul segment is as fast as necessary to guarantee
no slow down, say FDDI or T3.  Such things do exist in real life for
economically sound reasons.  (Many Silicon Valley outfits lease dark
fiber under public streets to connect their buildings).  Please say why
that case is more likely to experience flash crowds than the
128.32.33.00/24 near your desk, and also say how it would be easier for
mercenary.CS.Berkeley.EDU to detect such problems.  (I'm guessing that
there is a "smart hub" or "bridge" on the 128.32.33.00/24 Ethernet, given
the many well known and presumably sometimes busy hosts on it.)  My guess
is that the only way congestion can be detected in either network is by
TCP timeouts or duplicate ACKs, and that congestion is far more likely on
your local Ethernet then on my not so hypothetical network.


> ...
> Yes, I mean it can change, and quite unpredicatably in the worst case.
> In the LAN case, as I have explained, the sender knows through *local*
> information that the LAN has a high load, so can slow down its
> sending rate without aggravating the congestion in the LAN.

Please say how a sender on a LAN can tell that its local LAN has suddenly
gone bad, given the installed base of millions of bridges that we all know
will never look at any new bits in proposed revisions of TCP headers.
There is no hope among those of us who've experienced previous generations
of bridges that they'll even honor old bits they're required by standards
and common sense (e.g. FDDI-Ethernet and IP DF).

Also as I keep saying, the "switch" sales critters have entirely broken
MAC-layer flow control.  They actually use that fact as one of their
selling points, and their suckers--oops actually think it is a good thing.


> ...
> > Anything involving "dropping" seems irrelevant to the issue at hand.
> > 
> That's a strong statement! I thought we were discussing congestion
> avoidance and control, and given that, I don't see how we can ignore
> the possibility of packets getting dropped (however undesirable that
> may be).

We are not discussing new ideas ofr congestion avoidance or management.
We are talking about a sneaky-smart switch in hosts that turns off the
standards-required ACK pacing and/or slow-(re)start.  Yes, packets can be
dropped, but the idea of the go-fast switch is ONLY for cases where packet
losses are low or at least not due to congestion, and only when congestion
is being managed with slow-(re)start, with the need of slow-(re)start for
well smoothed RTT's with requirement from that for the 2:1 ACK:segment
ratio.


> ...
> The operative part of what I said was "learn from past experience".
> Here's the outline of a simple scheme. It makes no assumptions
> about hop counts, netmasks, etc.
>
> 1. If doing a transfer to another host for the first time in x minutes
> (x TBD), do slow start.
>
> 2. If you've done a transfer in the past x minutes, start with window size
> equal to where you left off the last time. Use timer to smooth out bursts.
> Mark all packets, except for the first one, as "low priority". In the best
> case, all the packets will go through fine and you will experience the
> benefit of fast start. In a worse case (such as the networking having become
> much more congested since the last time you used it), priority dropping
> with shield (to an extent) other traffic from the ill-effects of your
> aggressiveness.
> ...

In which bit of the current TCP or IPv4 headers is your low priority
mark?

Given a free hand, a lot of things other than Van Jacobson's old but
extremely cool old kludge of watching for packet loss and guessing about
congestion could be tried.  ECN is an obvious and currently popular idea,
after years of quiet.  However, that's all entirely irrelevant to this
discussion and possibly to this mailing list.  (Recall it's title is
something about TCP implementation).  We are not talking about TCPng, but
how to mitigate the new, stronger language about ACK pacing and
slow-(re)start.

Again, given things that the host that needs to set its own go-fast switch
for traffic to a specified other host, what do you suggest we do?  If you
have a real alternative to looking at the RTT or the netmask, please
describe it.


Vernon Schryver    vjs@rhyolite.com


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From: Scott Bradner <sob@harvard.edu>
Message-Id: <199810240028.UAA18625@newdev.harvard.edu>
To: tcp-impl@cthulhu.engr.sgi.com, vjs@calcite.rhyolite.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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> What does "endorsed by the IEEE" mean?  

passed as a standard

Scott


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Date: Sun, 25 Oct 1998 20:35:53 -0800
From: Michael Ridley <mridley@autobahn.org>
To: Vernon Schryver <vjs@calcite.rhyolite.com>
Cc: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: 'Slow-Start Restart after Idle' draft
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On Fri, Oct 23, 1998 at 02:20:57PM -0600, Vernon Schryver wrote:
> 
>   - RTT's (e.g. from a previous connection or measured during the
>      3-way handshake) are imperfect, possibly producing false positives
>      on, e.g. OC-12 metro links, or on bursty LANs.   Still, what 
>      alternative is there?
> Vernon Schryver    vjs@rhyolite.com

I think I must be missing something here, but why do we care if it's
an OC-12 link, for instance?  Is that a false positive?  The goal
here is to bypass slow start when appropriate and send the full
advertised window so as to maximize use of non-congested links, correct?
Assuming that I haven't gotten really confused in all of this, then
I would think you'd want to send the max window on the OC-12 if it's RTT
was low (often a faster link even than your LAN on both ends).

By stating it's a false positive, you make me think that you'd want slow
start in such a case- why?

Sorry about the late reply on this...

-- 
Michael Ridley
mridley@computer.org



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Date: Mon, 26 Oct 1998 08:01:42 -0700 (MST)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Michael Ridley <mridley@autobahn.org>


> >   - RTT's (e.g. from a previous connection or measured during the
> >      3-way handshake) are imperfect, possibly producing false positives
> >      on, e.g. OC-12 metro links, or on bursty LANs.   Still, what 
> >      alternative is there?


> I think I must be missing something here, but why do we care if it's
> an OC-12 link, for instance?  Is that a false positive?  The goal
> here is to bypass slow start when appropriate and send the full
> advertised window so as to maximize use of non-congested links, correct?
> Assuming that I haven't gotten really confused in all of this, then
> I would think you'd want to send the max window on the OC-12 if it's RTT
> was low (often a faster link even than your LAN on both ends).
>
> By stating it's a false positive, you make me think that you'd want slow
> start in such a case- why?

Yes, I phrased things badly there.

My general claim is that it does not matter whether there is one hop or
many, or whether some of the hops are longer or shorter.  All that matters
is that you want a go-fast mode that violates some of the proposed, even
stronger language enforcing the TCP congestion control and avoidance
mechanisms when there are sufficiently few, opportunites for congestion
at the current instant.  It would be nice if there the go-fast mode and
how and when it is invoked were documented, for all of the standard reasons
standards exist.

I was trying to concede in those words that a fairly short, fast, but
somewhat congested link can appear to have a very small RTT, when the RTT
is measured with a single pair of packets, such as during the 3-way
handshake.

The point of a go-fast mode is the extreme case of what started the
discussions in this mailing list (I think it was here) weeks or months
ago about increasing the initial window for HTTP.  If you have a
semi-private 1 Gbit/sec link connecting hosts using 500KByte windows and
1432 byte MSS to move blocks of data, you really do not want to wait for
175 ACKs before you are using the full window.  You use such large windows
because your bandwidth*delay product is that large or larger, and you are
going at speeds where much of the delay comes from inside your hosts.
For example, if your operating system wakes up your application to dump
to a screen buffer 10 or 20 ms, you want enough data on hand to make it
worth while.  Or consider moving files; most UNIX file systems do not
deliver or sink GBytes/sec, but some do, and there soon will be more.
Given today's cheap Gbit Ethernet and $2000 PC's with 100's of MByte of
RAM, such scenarios are no longer found only among people with $30M CPU's,
or at least with $B atomic particle detectors.  Competative pressures will
at least entice implementors to discard slow-start in such situations.
If they are not conservative in how they turn off slow-start, their code
will also turn it off in the classic cases where it is needed.


Vernon Schryver    vjs@rhyolite.com


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From: Luigi Rizzo <luigi@labinfo.iet.unipi.it>
Message-Id: <199810261603.RAA02603@labinfo.iet.unipi.it>
Subject: tcp initial window and rate pacing...
To: tcp-impl@cthulhu.engr.sgi.com
Date: Mon, 26 Oct 1998 17:03:10 +0100 (MET)
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Hi,

(not sure if this is more for tcpimpl or end2end...)

I have read the recent RFCs (2414,2415,2416) about increasing the
initial window to 4K or so -- all is fine, the situation is not too
different from a regular TCP after one round, etc.

But has anyone considered, in addition to larger windows, also
pacing the generation of the initial burst of packets at a relatively
low rate so that one can an even slightly larger initial window
yet avoiding overflows at the network edges (e.g.  dialup users) ?

The implementation seems trivial (i did it with some 20 lines of code
driven by fasttimo() ) and being able to use slightly larger windows 
makes the connection less prone to ack starvation.
Lacking information the initial rate can be set to a conservative
value so that one can reuse the fasttimo code, or one can use cached
values from previous connections same as it is done with the initial
RTT and RTTVAR

comments ?

	cheers
	luigi
-----------------------------+--------------------------------------
Luigi Rizzo                  |  Dip. di Ingegneria dell'Informazione
email: luigi@iet.unipi.it    |  Universita' di Pisa
tel: +39-50-568533           |  via Diotisalvi 2, 56126 PISA (Italy)
fax: +39-50-568522           |  http://www.iet.unipi.it/~luigi/
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From owner-tcp-impl@cthulhu.engr.sgi.com  Mon Oct 26 14:44:55 1998
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From: Mohit Aron <aron@cs.rice.edu>
Message-Id: <199810261931.NAA08370@cs.rice.edu>
Subject: Re: tcp initial window and rate pacing...
To: luigi@labinfo.iet.unipi.it (Luigi Rizzo)
Date: Mon, 26 Oct 1998 13:31:47 -0600 (CST)
Cc: tcp-impl@cthulhu.engr.sgi.com
In-Reply-To: <199810261603.RAA02603@labinfo.iet.unipi.it> from "Luigi Rizzo" at Oct 26, 98 05:03:10 pm
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> 
> But has anyone considered, in addition to larger windows, also
> pacing the generation of the initial burst of packets at a relatively
> low rate so that one can an even slightly larger initial window
> yet avoiding overflows at the network edges (e.g.  dialup users) ?
> 




I've done some work on this. Please take a look at:
	http://www.cs.rice.edu/~aron/papers/rice-TR98-318.ps



- Mohit


From owner-tcp-impl@cthulhu.engr.sgi.com  Mon Oct 26 16:03:43 1998
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CC: Luigi Rizzo <luigi@labinfo.iet.unipi.it>, tcp-impl@cthulhu.engr.sgi.com
Subject: Re: tcp initial window and rate pacing...
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Mohit Aron wrote:

> >
> > But has anyone considered, in addition to larger windows, also
> > pacing the generation of the initial burst of packets at a relatively
> > low rate so that one can an even slightly larger initial window
> > yet avoiding overflows at the network edges (e.g.  dialup users) ?
> >
>
> I've done some work on this. Please take a look at:
>         http://www.cs.rice.edu/~aron/papers/rice-TR98-318.ps

John Heidemann here at USC/ISI has done some work on rate-pacing.I have
done some work (RFC2140) on TCP control block sharing, which
would allow rate values to be shared over multiple connections to the same
place.
There can, of course, be default initial values, as there are for the
window size.

Papers are available at:
    http://www.isi.edu/lsam/

Joe



From owner-tcp-impl@cthulhu.engr.sgi.com  Mon Oct 26 19:18:16 1998
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To: Luigi Rizzo <luigi@labinfo.iet.unipi.it>
cc: tcp-impl@cthulhu.engr.sgi.com
Subject: Re: tcp initial window and rate pacing... 
In-reply-to: Your message of "Mon, 26 Oct 1998 17:03:10 +0100."
             <199810261603.RAA02603@labinfo.iet.unipi.it> 
Date: Mon, 26 Oct 1998 16:07:33 -0800
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Yes -- we're looking at various pacing schemes and their effects under
contract to NASA.

So far it looks promising but one never knows until one is done.

Thanks!

Craig

In message <199810261603.RAA02603@labinfo.iet.unipi.it>, Luigi Rizzo writes:

>Hi,
>
>(not sure if this is more for tcpimpl or end2end...)
>
>I have read the recent RFCs (2414,2415,2416) about increasing the
>initial window to 4K or so -- all is fine, the situation is not too
>different from a regular TCP after one round, etc.
>
>But has anyone considered, in addition to larger windows, also
>pacing the generation of the initial burst of packets at a relatively
>low rate so that one can an even slightly larger initial window
>yet avoiding overflows at the network edges (e.g.  dialup users) ?
>
>The implementation seems trivial (i did it with some 20 lines of code
>driven by fasttimo() ) and being able to use slightly larger windows 
>makes the connection less prone to ack starvation.
>Lacking information the initial rate can be set to a conservative
>value so that one can reuse the fasttimo code, or one can use cached
>values from previous connections same as it is done with the initial
>RTT and RTTVAR
>
>comments ?
>
>	cheers
>	luigi
>-----------------------------+--------------------------------------
>Luigi Rizzo                  |  Dip. di Ingegneria dell'Informazione
>email: luigi@iet.unipi.it    |  Universita' di Pisa
>tel: +39-50-568533           |  via Diotisalvi 2, 56126 PISA (Italy)
>fax: +39-50-568522           |  http://www.iet.unipi.it/~luigi/
>_____________________________|______________________________________


From owner-tcp-impl@cthulhu.engr.sgi.com  Tue Oct 27 09:40:31 1998
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From: alan@lxorguk.ukuu.org.uk (Alan Cox)
Subject: Re: tcp initial window and rate pacing...
To: luigi@labinfo.iet.unipi.it (Luigi Rizzo)
Date: Tue, 27 Oct 1998 19:39:31 +0000 (GMT)
Cc: tcp-impl@cthulhu.engr.sgi.com
In-Reply-To: <199810261603.RAA02603@labinfo.iet.unipi.it> from "Luigi Rizzo" at Oct 26, 98 05:03:10 pm
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> Lacking information the initial rate can be set to a conservative
> value so that one can reuse the fasttimo code, or one can use cached
> values from previous connections same as it is done with the initial
> RTT and RTTVAR

I played a little with pacing output frames. The overhead on fast links
when trying to do the timer stuff is nasty, on low speed ok. I had horrible
problems with half duplex media though. On AX.25 and irda throughput dropped
in both cases.

Alan



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From: Venkat Padmanabhan <padmanab@microsoft.com>
To: "'luigi@labinfo.iet.unipi.it'" <luigi@labinfo.iet.unipi.it>
Cc: "'tcp-impl@cthulhu.engr.sgi.com'" <tcp-impl@cthulhu.engr.sgi.com>
Subject: Re: tcp initial window and rate pacing...
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> But has anyone considered, in addition to larger windows, also
> pacing the generation of the initial burst of packets at a relatively
> low rate so that one can an even slightly larger initial window
> yet avoiding overflows at the network edges (e.g.  dialup users) ?

I considered this in TCP fast start, which I developed as part of my
thesis (http://www.cs.berkeley.edu/~padmanab/phd-thesis.html).
This also appears in the 1998 Global Internet Conference
(http://www.cs.berkeley.edu/~padmanab/papers/gi98.ps).

I made no assumptions about how often information from the past
would be valid. Therefore, I had two goals: 

1. To reduce latency by reusing window information when such information
is valid.
2. To avoid performance degradation when the information is stale.

I used priority dropping to shield other traffic in case the cached window
is stale (and too large), and some special techniques to quickly detect
and abort a failed fast start attempt. My experimental results showed that 
not doing so could cause much worse performance than with standard 
slow start in certain situations, clearly an undesirable outcome.

-Venkat



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To: Venkat Padmanabhan <padmanab@microsoft.com>
Cc: "'luigi@labinfo.iet.unipi.it'" <luigi@labinfo.iet.unipi.it>,
        "'tcp-impl@cthulhu.engr.sgi.com'" <tcp-impl@cthulhu.engr.sgi.com>
Subject: Re: tcp initial window and rate pacing...
In-reply-to: Your message of Tue, 27 Oct 1998 11:32:20 PST.
Date: Tue, 27 Oct 1998 12:58:46 PST
From: Vern Paxson <vern@ee.lbl.gov>
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Let's please migrate this thread to end2end-interest - thanks.

		Vern


From owner-tcp-impl@cthulhu.engr.sgi.com  Tue Oct 27 19:49:14 1998
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Message-ID: <36366756.8A3D499B@ehsco.com>
Date: Tue, 27 Oct 1998 16:37:42 -0800
From: "Eric A. Hall" <ehall@ehsco.com>
Organization: EHS Company
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Apart from Microsoft, is anybody shipping a TCP stack that incorporates
any of the advanced options? In particular, am looking for products that
use the Window Scale, Timestamp and Selective Acknowledgment options.

Thanks

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Tue Oct 27 19:56:28 1998
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From: sm@bossette.engr.sgi.com (Sam Manthorpe)
Message-Id: <199810280047.QAA35912@bossette.engr.sgi.com>
Subject: Re: shipping implementations w/ advanced options
To: ehall@ehsco.com (Eric A. Hall)
Date: Tue, 27 Oct 1998 16:47:47 -0800 (PST)
Cc: tcp-impl@cthulhu.engr.sgi.com
In-Reply-To: <36366756.8A3D499B@ehsco.com> from "Eric A. Hall" at Oct 27, 98 04:37:42 pm
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Eric A. Hall wrote:
> Apart from Microsoft, is anybody shipping a TCP stack that incorporates
> any of the advanced options? In particular, am looking for products that
> use the Window Scale, Timestamp and Selective Acknowledgment options.

IRIX supports window scale and timestamps (as I think most people do
these days) and will soon support SACK.

-- Sam

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Sam Manthorpe, SGI.  tel: (650)933-2856 fax: (650)932-2856  sm@engr.sgi.com


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From: alan@lxorguk.ukuu.org.uk (Alan Cox)
Subject: Re: shipping implementations w/ advanced options
To: ehall@ehsco.com (Eric A. Hall)
Date: Wed, 28 Oct 1998 01:47:02 +0000 (GMT)
Cc: tcp-impl@cthulhu.engr.sgi.com
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> Apart from Microsoft, is anybody shipping a TCP stack that incorporates
> any of the advanced options? In particular, am looking for products that
> use the Window Scale, Timestamp and Selective Acknowledgment options.

Linux 2.1.x has window scaling, timestamp, and selective ack for both IPv4
and IPv6. The BSD's have been shipping window scaling for a very long
time. 

Alan



From owner-tcp-impl@cthulhu.engr.sgi.com  Tue Oct 27 20:03:21 1998
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Date: Tue, 27 Oct 1998 16:56:42 -0800
From: "Eric A. Hall" <ehall@ehsco.com>
Organization: EHS Company
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Subject: Re: shipping implementations w/ advanced options
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> Linux 2.1.x has window scaling, timestamp, and selective ack for both
> IPv4 and IPv6.

I must have old kernels. Neither RH 5.2 nor COL 1.3 do anything other
than MSS. Thanks.

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Tue Oct 27 20:03:39 1998
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Date: Tue, 27 Oct 1998 16:56:40 -0800
From: Rick Jones <raj@cup.hp.com>
Organization: HP 9000 Network Performance
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Subject: Re: shipping implementations w/ advanced options
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> Apart from Microsoft, is anybody shipping a TCP stack that incorporates
> any of the advanced options? In particular, am looking for products that
> use the Window Scale, Timestamp and Selective Acknowledgment options.

Window Scale and Timestamp functionality have been shipping for HP-UX
since patches for HP-UX 9.05. That goes back several years.

rick jones
-- 
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to email, or post, but please do not do both...
my email address is raj in the cup.hp.com domain...


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From: alan@lxorguk.ukuu.org.uk (Alan Cox)
Subject: Re: shipping implementations w/ advanced options
To: ehall@ehsco.com (Eric A. Hall)
Date: Wed, 28 Oct 1998 02:03:32 +0000 (GMT)
Cc: tcp-impl@cthulhu.engr.sgi.com
In-Reply-To: <36366BC9.78D993D2@ehsco.com> from "Eric A. Hall" at Oct 27, 98 04:56:42 pm
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> > Linux 2.1.x has window scaling, timestamp, and selective ack for both
> > IPv4 and IPv6.
> 
> I must have old kernels. Neither RH 5.2 nor COL 1.3 do anything other
> than MSS. Thanks.

2.1.x does the lot, 2.0.x doesnt do anything funky. 2.0.x is the stable
tree until about December so vendors ship 2.0 (sensibly)

Alan



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Subject: Re: shipping implementations w/ advanced options
To: tcp-impl@cthulhu.engr.sgi.com
Date: Wed, 28 Oct 1998 05:57:44 -0500 (EST)
From: "Mike O'Connor" <mjo@dojo.mi.org>
Reply-To: "Mike O'Connor" <mjo@dojo.mi.org>
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:Apart from Microsoft, is anybody shipping a TCP stack that incorporates
:any of the advanced options? In particular, am looking for products that
:use the Window Scale, Timestamp and Selective Acknowledgment options.

There's a good summary of the information you're looking for at:
http://www.psc.edu/networking/perf_tune.html

-- 
 Michael J. O'Connor | WWW: http://dojo.mi.org/~mjo/ | Email: mjo@dojo.mi.org
 InterNIC WHOIS: MJO | (has my PGP & Geek Code info) | Phone: +1 248-848-4481
 =--==--==--==--==--==--==--==--==--==--==--==--==--==--==--==--==--==--==--=
"You have to use imaginary numbers, like eleventeen..."               -Calvin


From owner-tcp-impl@cthulhu.engr.sgi.com  Wed Oct 28 06:28:20 1998
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Message-ID: <011401be0264$cfe30760$17037e96@barbarella.pd.london.sco.com>
From: Jonathan Webb <jonwe@sco.com>
To: "Eric A. Hall" <ehall@ehsco.com>,
        TCP Implementations <tcp-impl@cthulhu.engr.sgi.com>
Subject: Re: shipping implementations w/ advanced options
Date: Wed, 28 Oct 1998 11:15:45 -0000
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The next release of SCO UnixWare7 will include windows
scale and timestamp.

- Jonathan
________________________________________________

Jonathan Webb                  Architect, Internet Engineering Group
jonwe@sco.com                The Santa Cruz Operation Limited
Tel: +44 1923 813658      Croxley Business Park,
Fax: +44 1923 813804     Hatters Lane,
http://www.sco.com            Watford. WD1 8YN, UK
                                             Registered in England No
2063779
________________________________________________
-----Original Message-----
From: Eric A. Hall <ehall@ehsco.com>
To: TCP Implementations <tcp-impl@cthulhu.engr.sgi.com>
Date: 28 October 1998 00:44
Subject: shipping implementations w/ advanced options


>
>Apart from Microsoft, is anybody shipping a TCP stack that incorporates
>any of the advanced options? In particular, am looking for products that
>use the Window Scale, Timestamp and Selective Acknowledgment options.
>
>Thanks
>
>--
>Eric A. Hall                                            ehall@ehsco.com
>+1-650-685-0557                                    http://www.ehsco.com
>



From owner-tcp-impl@cthulhu.engr.sgi.com  Wed Oct 28 10:48:55 1998
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From: Craig Buffinton <craigb@ftp.com>
Subject: RE: shipping implementations w/ advanced options
Date: Wed, 28 Oct 1998 10:33:58 -0500
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>>Reply to your message of 10/27/98 7:43 PM
>
>Apart from Microsoft, is anybody shipping a TCP stack that incorporates
>any of the advanced options? In particular, am looking for products that
>use the Window Scale, Timestamp and Selective Acknowledgment options.
>

The OnNet Kernel 4.0 from FTP Software, a subsidiary of NetManage incorporates
Window Scale, Timestamp and Selective Acknowledgment (RFCs 1323,  2018, 2001).
 It installs on Win 95/98. Other features: Winsock 2, IPv6, Mobile IP, graphic 
display of TCP/UDP traffic.

BTW: http://www.psc.edu/networking/perf_tune.html is atleast a couple years out of 
date regarding our products.

If you  have any problems / comments please let me know.

Craig Buffinton
Principal Engineer
FTP Software, a subsidiary of NetManage
978-946-5142



From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 29 07:13:56 1998
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DATE: Thu, 29 Oct 1998 06:16:51 -0600
Message-ID: <ydcabhir>
Subject: FREE DEMO of Web Promotion Software that Puts You On Top of 1,000+ Search Engines!
From: hire@hiredhits.com
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Greetings!

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From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 29 08:00:48 1998
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Date: Thu, 29 Oct 1998 07:49:34 -0500 (EST)
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From: "Charles M. Hannum" <root@ihack.net>
To: tcp-impl@cthulhu.engr.sgi.com
Cc: explorer@flame.org, paul@vix.com
Subject: Re: Problem with RFC 1323 and HP-UX
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A user sent me the following tcpdump output, regarding a problem where
a connection between a machine running HP-UX (hp.rc.vix.com) and
NetBSD (xenon.rc.vix.com) would `freeze.'

It appears that the HP box is resetting its time stamp.  (Note that
the time stamp on the 00:19:25.226717 packet is `less than' the time
stamp on the 00:18:05.615634 packet.)  This causes the PAWS check to
fail on all future packets, and they are therefore dropped by the
NetBSD box after sending an ACK.

This would seem to be a bug in the HP-UX implementation of RFC 1323.
It looks to me like HP-UX is using a 16-bit counter and updating it
every .01s, whereas NetBSD is using a 32-bit counter and updating it
every .5s.  The granularity isn't really an issue, but it seems to me
that rolling over at 16 bits is clearly wrong.

For reference, the version of HP-UX in question is:

  HP-UX hpd280 B.11.00 A 9000/820 2006777451 two-user license

Could someone verify that I'm not on drugs here?


00:18:05.445911 hp.rc.vix.com.ssh > xenon.rc.vix.com.65534: P 124148:124240(92) ack 1668 win 32768 <nop,nop,timestamp 62691 204> [tos 0x10]
00:18:05.445998 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: . ack 124240 win 17428 <nop,nop,timestamp 811 62685> [tos 0x10]
00:18:05.589140 hp.rc.vix.com.ssh > xenon.rc.vix.com.65534: P 124240:124284(44) ack 1668 win 32768 <nop,nop,timestamp 62705 204> [tos 0x10]
00:18:05.615634 hp.rc.vix.com.ssh > xenon.rc.vix.com.65534: P 124284:124328(44) ack 1668 win 32768 <nop,nop,timestamp 62708 204> [tos 0x10]
00:18:05.615717 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: . ack 124328 win 17476 <nop,nop,timestamp 811 62705> [tos 0x10]
00:19:25.179785 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: P 1668:1688(20) ack 124328 win 17520 <nop,nop,timestamp 971 62705> [tos 0x10]
00:19:25.226717 hp.rc.vix.com.ssh > xenon.rc.vix.com.65534: P 124328:124364(36) ack 1688 win 32768 <nop,nop,timestamp 5133 971> [tos 0x10]
00:19:25.226806 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: . ack 124328 win 17520 <nop,nop,timestamp 971 62705> [tos 0x10]
00:19:25.736480 hp.rc.vix.com.ssh > xenon.rc.vix.com.65534: P 124328:124364(36) ack 1688 win 32768 <nop,nop,timestamp 5184 971> [tos 0x10]
00:19:25.736560 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: . ack 124328 win 17520 <nop,nop,timestamp 972 62705> [tos 0x10]
00:19:26.163664 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: P 1668:1688(20) ack 124328 win 17520 <nop,nop,timestamp 972 62705> [tos 0x10]
00:19:26.164082 hp.rc.vix.com.ssh > xenon.rc.vix.com.65534: . ack 1688 win 32768 <nop,nop,timestamp 5226 971> [tos 0x10]
00:19:26.164148 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: . ack 124328 win 17520 <nop,nop,timestamp 973 62705> [tos 0x10]
00:19:26.164600 hp.rc.vix.com.ssh > xenon.rc.vix.com.65534: P 124328:124364(36) ack 1688 win 32768 <nop,nop,timestamp 5226 971> [tos 0x10]
00:19:26.164662 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: . ack 124328 win 17520 <nop,nop,timestamp 973 62705> [tos 0x10]
00:19:26.686484 hp.rc.vix.com.ssh > xenon.rc.vix.com.65534: P 124328:124364(36) ack 1688 win 32768 <nop,nop,timestamp 5279 971> [tos 0x10]
00:19:26.686556 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: . ack 124328 win 17520 <nop,nop,timestamp 974 62705> [tos 0x10]
00:19:28.163663 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: P 1668:1688(20) ack 124328 win 17520 <nop,nop,timestamp 976 62705> [tos 0x10]
00:19:28.164080 hp.rc.vix.com.ssh > xenon.rc.vix.com.65534: . ack 1688 win 32768 <nop,nop,timestamp 5426 971> [tos 0x10]
00:19:28.164145 xenon.rc.vix.com.65534 > hp.rc.vix.com.ssh: . ack 124328 win 17520 <nop,nop,timestamp 977 62705> [tos 0x10]



From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 29 11:23:44 1998
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To: "Eric A. Hall" <ehall@ehsco.com>
Cc: TCP Implementations <tcp-impl@cthulhu.engr.sgi.com>
Subject: Re: shipping implementations w/ advanced options
References: <36366756.8A3D499B@ehsco.com>
From: Dave Marquardt <marquard@austin.ibm.com>
Date: 29 Oct 1998 10:11:21 -0600
In-Reply-To: "Eric A. Hall"'s message of "Tue, 27 Oct 1998 16:37:42 -0800"
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"Eric A. Hall" <ehall@ehsco.com> writes:
> Apart from Microsoft, is anybody shipping a TCP stack that incorporates
> any of the advanced options? In particular, am looking for products that
> use the Window Scale, Timestamp and Selective Acknowledgment options.

AIX has had the window scale and timestamp options for several years.

-Dave


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 29 12:21:22 1998
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From: William Gilliam <wag@cup.hp.com>
Message-Id: <199810291713.JAA09958@hpinwag.cup.hp.com>
Subject: Re: Problem with RFC 1323 and HP-UX
To: root@ihack.net (Charles M. Hannum)
Date: Thu, 29 Oct 1998 09:13:46 -0800 (PST)
Cc: tcp-impl@cthulhu.engr.sgi.com
In-Reply-To: <199810291249.HAA10897@zygorthian-space-raiders.mit.edu> from "Charles M. Hannum" at Oct 29, 98 07:49:34 am
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Charles M. Hannum said:

> A user sent me the following tcpdump output, regarding a problem where
> a connection between a machine running HP-UX (hp.rc.vix.com) and
> NetBSD (xenon.rc.vix.com) would `freeze.'
> 
> It appears that the HP box is resetting its time stamp.  (Note that
> the time stamp on the 00:19:25.226717 packet is `less than' the time
> stamp on the 00:18:05.615634 packet.)  This causes the PAWS check to
> fail on all future packets, and they are therefore dropped by the
> NetBSD box after sending an ACK.

Yes, your user has hit an old bug that was fixed quite a while ago.

> This would seem to be a bug in the HP-UX implementation of RFC 1323.
> It looks to me like HP-UX is using a 16-bit counter and updating it
> every .01s, whereas NetBSD is using a 32-bit counter and updating it
> every .5s.  The granularity isn't really an issue, but it seems to me
> that rolling over at 16 bits is clearly wrong.
> 
> For reference, the version of HP-UX in question is:
> 
>   HP-UX hpd280 B.11.00 A 9000/820 2006777451 two-user license
> 
> Could someone verify that I'm not on drugs here?

I'll just say that your interpretation of the tcpdump output is correct.  I'll
also say that this problem was fixed shortly after HP-UX 11.0 was released,
so your customer should probably grab and install the appropriate patch
(the latest cumulative patch is PHNE_16645).


WG


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Date: Thu, 29 Oct 1998 11:41:46 -0800
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Scenario: two PCs with 8k windows on a Token Ring network with 16k
frames. What happens? Send a single 8k segment and then wait for ACK?

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 29 15:12:10 1998
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Message-Id: <9810291159.ZM7756@wli.engr.sgi.com>
Date: Thu, 29 Oct 1998 11:59:20 -0800
In-Reply-To: "Eric A. Hall" <ehall@ehsco.com>
        "MTU greater than rcv.wnd" (Oct 29, 11:41am)
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On Oct 29, 11:41am, Eric A. Hall wrote:
> Subject: MTU greater than rcv.wnd
>
> Scenario: two PCs with 8k windows on a Token Ring network with 16k
> frames. What happens? Send a single 8k segment and then wait for ACK?

if slow_start applies. It sends first MTU, and will get an ACK with updated
rcv.wnd, if the rcv.wnd is 0, the persist timer kick in.




-- 
William Li
wli@sgi.com/650-933-3751



From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 29 15:24:48 1998
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Date: Thu, 29 Oct 1998 12:06:49 -0800
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Organization: EHS Company
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> > Scenario: two PCs with 8k windows on a Token Ring network with 16k
> > frames. What happens? Send a single 8k segment and then wait for
> > ACK?
> 
> if slow_start applies.

I mean beyond slow-start, not that there's any difference in this model.

It seems that if the rcv.wnd is constantly smaller than the MTU, then
the sender will never send more than one segment without having to stop
and wait for an ACK, slow-start or not.

Confirming a suspicion,

Thanks

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@lerc.nasa.gov  Thu Oct 29 15:34:04 1998
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TCP-impl Folks:

The tcp-impl mailing list is moving to NASA LeRC as of right now.
As of several minutes ago, the list of people on the mailing list
was identical at LeRC and SGI.  So, please use
"tcp-impl@lerc.nasa.gov" from now on.  The list archive is available
both as a flat file and a hypermail web index from the following
URL:

    http://tcp-impl.lerc.nasa.gov/tcp-impl

I'd like to thank SGI for hosting the mailing list and archive at
their site for a long time.  We very much appreciate their efforts!
The move is simply an attempt to make things a little easier for
Vern and I to manage.

allman


---
http://gigahertz.lerc.nasa.gov/~mallman/


From owner-tcp-impl@cthulhu.engr.sgi.com  Thu Oct 29 15:36:21 1998
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From: Art Shelest <artshel@microsoft.com>
To: "'Eric A. Hall'" <ehall@ehsco.com>
Cc: TCP Implementations <tcp-impl@cthulhu.engr.sgi.com>
Subject: RE: MTU greater than rcv.wnd
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Will adjusting MSS or MTU down improve the performance?
With any size MSS, you still can only send up to rwin bytes
before stopping and waiting for an ACK.

After slow-start, you have:
Large MTU: send 8KB in one chunk, then wait for an ACK.
Small MTU: send 8KB in small pieces, then wait for an ACK.

Small MTU will reduce performance somewhat because of overhead.

    _Art.

-----Original Message-----
From: Eric A. Hall [mailto:ehall@ehsco.com]
Sent: Thursday, October 29, 1998 12:07 PM
Cc: TCP Implementations
Subject: Re: MTU greater than rcv.wnd



> > Scenario: two PCs with 8k windows on a Token Ring network with 16k
> > frames. What happens? Send a single 8k segment and then wait for
> > ACK?
> 
> if slow_start applies.

I mean beyond slow-start, not that there's any difference in this model.

It seems that if the rcv.wnd is constantly smaller than the MTU, then
the sender will never send more than one segment without having to stop
and wait for an ACK, slow-start or not.

Confirming a suspicion,

Thanks

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@lerc.nasa.gov  Thu Oct 29 16:25:06 1998
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Message-ID: <3638C8D9.CC2DA322@cup.hp.com>
Date: Thu, 29 Oct 1998 11:58:17 -0800
From: Rick Jones <raj@cup.hp.com>
Organization: SNSL
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To: TCP Implementations <tcp-impl@lerc.nasa.gov>
Subject: Re: MTU greater than rcv.wnd
References: <3638C4FA.ED8F11EC@ehsco.com>
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> Scenario: two PCs with 8k windows on a Token Ring network with 16k
> frames. What happens? Send a single 8k segment and then wait for ACK?

That would be my guess as to one correct behaviour. 

HP-UX is supposed to be coded such that for an 8K window a 4K MSS will
be negotiated. In general, HP-UX will negotiate an MSS that is no more
than 1/2 the receive window at time of establishment.

rick jones
-- 
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to email, or post, but please do not do both...
my email address is raj in the cup.hp.com domain...


From owner-tcp-impl@lerc.nasa.gov  Thu Oct 29 17:23:18 1998
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Message-ID: <3638D5CA.F99E1844@ehsco.com>
Date: Thu, 29 Oct 1998 12:53:30 -0800
From: "Eric A. Hall" <ehall@ehsco.com>
Organization: EHS Company
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CC: TCP Implementations <tcp-impl@lerc.nasa.gov>
Subject: Re: MTU greater than rcv.wnd
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> Small MTU will reduce performance somewhat because of overhead.

Right. Obviously the performance solution to this problem is to increase
thw window size on the client.

> With any size MSS, you still can only send up to rwin bytes
> before stopping and waiting for an ACK.

There are other elements here besides performance. In particular, I can
imagine a situation where SWS avoidance causes problems if the rcv.win
and MTU are dramatically different. It would be possible to code
yourself into a corner, although I doubt anybody's made this mistake.

Also, as pointed out from a Sun person, Solaris (as a recipient) will
scale the rcv.wnd size so that it's substantially larger than the MTU,
keeping this from ever happening in the first place. Neat solution.

Just an interesting scenario.

Thanks

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@lerc.nasa.gov  Thu Oct 29 17:34:25 1998
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Date: Thu, 29 Oct 1998 13:01:43 -0800 (PST)
From: Kacheong Poon <Kacheong.Poon@Eng.Sun.COM>
Reply-To: Kacheong Poon <Kacheong.Poon@Eng.Sun.COM>
Subject: RE: MTU greater than rcv.wnd
To: tcp-impl@lerc.nasa.gov
In-Reply-To: "Your message with ID" <3FF8121C9B6DD111812100805F31FC0D048CCE39@RED-MSG-59>
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> Small MTU will reduce performance somewhat because of overhead.

In this particular case, receive window is less than 1*MTU, smaller negotiated
MSS may actually help.  With big MSS, if a segment is lost, the only way to
recover that is a timeout.  But if MSS is smaller, multiple segments can be
sent.  This means that fast retransmit can happen.  Since RTO is usually
bigger than RTT (in BSD's case, it can be much bigger), this definitely helps
in recovery.  But the MSS has to be small enough to allow at least 4 segments
to be sent in a window.  

							K. Poon.
							kcpoon@eng.sun.com




From owner-tcp-impl@lerc.nasa.gov  Thu Oct 29 18:19:39 1998
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From: rags@bapuji.engr.sgi.com (Raghu Mallena)
Message-Id: <199810292144.NAA04291@bapuji.engr.sgi.com>
Subject: Is it really exponential?
To: tcp-impl@lerc.nasa.gov
Date: Thu, 29 Oct 1998 13:44:26 -0800 (PST)
Cc: rags@bapuji.engr.sgi.com (Raghu Mallena)
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I was looking at some traces and noticed that the Slow Start 
algorithm opens the window size by *one* segment size everytime 
an ACK is received.  This is how it was described in Van Jacobson's 
"Congestion Avoidance and Control" SIGCOM'98 paper.   Is there 
any reason why it is designed this way as opposed to opening 
the window by the amount of data that got acked?   With Delayed 
Ack in effect,  the one-segment growth scheme would increase the 
window in roughly 2^(x/2) way as opposed to a true 2^x growth.    
Basically,  RTT*logW (where W is window size in segments) being 
the time to reach steady state as computed in VJ's paper will 
not be true for Delayed Ack.   And this will be even worse 
for the stacks that are lazy about sending Acks i.e send one 
Ack for more than 2 packets (please... i dont mean to re-start 
last weeks discussion on this subject)

However,  a simple one-line fix to open the window by the amount of 
data acked (ack - snd_una) should work well always.   The window would 
still open exponentially and the time to reach full window size would 
still be in the order of RTT*logW, approximately. 

Any reason why it shouldnt be done this way?

- raghu

---
Raghu Mallena
Silicon Graphics



From owner-tcp-impl@lerc.nasa.gov  Thu Oct 29 18:20:13 1998
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Message-Id: <199810292159.QAA11149@guns.lerc.nasa.gov>
To: rags@bapuji.engr.sgi.com (Raghu Mallena)
From: Mark Allman <mallman@lerc.nasa.gov>
Reply-To: mallman@lerc.nasa.gov
cc: tcp-impl@lerc.nasa.gov
Subject: Re: Is it really exponential? 
Organization: Late Night Hackers, NASA LeRC, Cleveland, Ohio
Song-of-the-Day: Lodi
Date: Thu, 29 Oct 1998 16:59:38 -0500
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> However, a simple one-line fix to open the window by the amount of
> data acked (ack - snd_una) should work well always.  The window
> would still open exponentially and the time to reach full window
> size would still be in the order of RTT*logW, approximately.
> 
> Any reason why it shouldnt be done this way?

I think except for the "work well always" part you are right.  I
think generally the fix you suggest works well.  It lowers the
transfer time (of short connections especially).  The size of the
burst of segments in response to an ACK is incraesed a little and
therefore, there can be a slight increase in loss.  The real problem
comes into play in reno-style recovery, where ACKs for large numbers
of un-ACKed data can come in during a slow start after an RTO
timeout.  In this case, you very much *do not* want to generate a
burst that is basically twice what we generate now.

So, a reasonable approach (I think!) is to use a ``limited byte
counting''.  Basically, you should increase the window by the number
of bytes ACKed, up to some limit like 2 segments.  So, when
responding to a correctly behaving delayed ACK receiver you are
getting better growth.  But, you are limiting the size of the burst
that any ACK can generate to one more segment than is currently
allowed in the case of receiving stretch ACK (because of a buggy
delayed ACK implementation or ACK loss).  That seems to be a good
way to do things to me.

For more information, please see my paper on this topic...

    Mark Allman. On the Generation and Use of TCP
    Acknowledgments. ACM Computer Communication Review, 28(5),
    October 1998. 
    http://gigahertz.lerc.nasa.gov/~mallman/papers/acks.ps

allman


From owner-tcp-impl@lerc.nasa.gov  Thu Oct 29 18:23:32 1998
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Message-Id: <199810292203.RAA11210@guns.lerc.nasa.gov>
To: rags@bapuji.engr.sgi.com (Raghu Mallena)
From: Mark Allman <mallman@lerc.nasa.gov>
Reply-To: mallman@lerc.nasa.gov
cc: tcp-impl@lerc.nasa.gov
Subject: Re: Is it really exponential? 
Organization: Late Night Hackers, NASA LeRC, Cleveland, Ohio
Song-of-the-Day: Lodi
Date: Thu, 29 Oct 1998 17:03:38 -0500
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> Any reason why it shouldnt be done this way?

Oh, so the last message was with my research hat on...  I should
also use my IETF hat and point out that byte counting should not be
used during slow start because the spec (2001, 2001.bis)
intentionally does not allow it.  The feeling is that this change
makes TCP more aggressive and we need to study and understand this
change before we unleash it on the network.

allman


From owner-tcp-impl@lerc.nasa.gov  Sat Oct 31 15:34:06 1998
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Message-ID: <363B5918.433881BA@ehsco.com>
Date: Sat, 31 Oct 1998 10:38:16 -0800
From: "Eric A. Hall" <ehall@ehsco.com>
Organization: EHS Company
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To: TCP Implementations <tcp-impl@lerc.nasa.gov>
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Did anybody ever make use of any of the expiremental or legacy TCP
options, such as NAK, Echo and Alternative Checksums? Are these options
seen much today, still?

Thanks

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@lerc.nasa.gov  Sat Oct 31 16:49:09 1998
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To: "Eric A. Hall" <ehall@ehsco.com>
cc: TCP Implementations <tcp-impl@lerc.nasa.gov>
Subject: Re: old TCP options 
In-reply-to: Your message of "Sat, 31 Oct 1998 10:38:16 PST."
             <363B5918.433881BA@ehsco.com> 
Date: Sat, 31 Oct 1998 12:09:59 -0800
From: Craig Partridge <craig@aland.bbn.com>
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Alternative Checksum was, I believe, never used or even implemented.

It was devised as an example during a discussion about whether one could
use a different checksum.

Craig

In message <363B5918.433881BA@ehsco.com>, "Eric A. Hall" writes:

>
>Did anybody ever make use of any of the expiremental or legacy TCP
>options, such as NAK, Echo and Alternative Checksums? Are these options
>seen much today, still?
>
>Thanks
>
>-- 
>Eric A. Hall                                            ehall@ehsco.com
>+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@lerc.nasa.gov  Sat Oct 31 16:58:53 1998
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To: scj2@gs4.revnet.com
Subject: ISM Corp has acquired 4.7 mill to begin production Stock up 100 percent
Date: Sat, 31 Oct 1998 14:47:10 -0600
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Please open the following message in your web browser

    http://gs4.revnet.com/GM/MSGVIEW/MSOHNOPA.HTML
____________________________________________________________
International Shoe Manufacturing Corp Update: 
International Shoe Manufacturing Corp. (Ticker-ISHO) has acquired the final-stage
financing to begin full-scale production at its plants in India. The $4.75 million is being
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The company stated that the financing will be followed up by a $9 million dollar IPO in
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ISHO is in the business of manufacturing athletic footwear for the world’s leading shoe
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The athletic shoe industry is estimated at $14.25 billion a year. The world’s leading shoe
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and marketing organizations that spend hundreds of millions of dollars a year getting their
products sold. They then rely on others to manufacture to their specifications. Almost, if
not all athletic shoe manufacturers are privately owned, benefiting from the hundreds of
millions of dollars spent on advertising by the name-brand companies. The result is an
open purchase order where such manufacturers  literally can sell every pair of shoes they
can produce. A business like this lends itself to being privately held due to the large cash
flow allowing for internal financing. International Shoe Manufacturing Corp. is the only
company known to exist that offers a public investor the opportunity to own a share of this
highly lucrative business in a pure  investment play.

For inquiries please contact the office of the director of investor relations toll free at: 
877-ISM-CORP  (877-476-2677)  or send your e-mail request to nsi@smallcapjournal.com Your request will be handled immediately.  Or write to ISM Corp at P.O. Box 520310 Longwood, Florida 32752

Please visit ISM’s web site at www.ismcorp.net
Safe Harbor for Forward-Looking Statements: Except for historical information contained herein, the statements in this press
release are forward-looking statements that are made pursuant to the safe harbor provisions of the Private Securities Reform Act of
1995.  Forward-looking statements involve known and unknown risks and uncertainties which may cause the company’s actual
results in the future periods to differ materially from forecasted results. These risks and uncertainties include, among other things,
product price volatility, product demand, market competition, risk inherent in the company’s domestic and international operations,
imprecision in estimating product reserves and the company’s ability to replace and expand its holdings.

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