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From owner-ecm@wyvern.aciri.org  Mon Feb  5 13:30:45 2001
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To: "Rob + Helena" <brennanr@iol.ie>
From: Mark Allman <mallman@grc.nasa.gov>
Reply-To: mallman@grc.nasa.gov
cc: ecm@aciri.org
Subject: Re: SCTP congestion control questions 
Organization: NASA GRC/BBN Technologies
Song-of-the-Day: Selling the Drama
Date: Mon, 05 Feb 2001 13:30:06 -0500
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Ugh -- it is not clear that this is the place to discuss this,
but... 

> 1. Should the SCTP fast retransmit proceedure respect cwnd/rwnd
> when retransmitting a packet?
> (Section 6.1 of the spec seems to state yes, but this means that FR
> is not very fast :-) as it must wait for approx 1/2 a window of duplicate
> SACKs to arrive before it can do the retransmit)

It should ingore the rwnd, for sure.  I think that since you are
retransmitting you are simply replacing something in the window, not
adding to it.

I also think you *should* ignore cwnd (I don't remember the SCTP
spec), but RFC 2581 says that you should retransmit -- I don't
believe there is any mention of a check to cwnd.

> 3. The FR proceedure outlined in section 7.2.4 (especially the
> implementation note) seems to lead to multiple retransmits of
> the same packet in the case where there is a large number of
> duplicate ACKs arriving. 
> (counter goes to 4, packet is FR'd, counter is reset, 4+ duplicate
> sacks arrive before 1 rtt => packet is FR'd again.
> This of course also drastically reduces ssthresh and cwnd.)

I agree this seems busted.

> 4. 1 Packet loss with large windows, high bandwidth and long rtts
> can lead to very large bursts of up to half a window of packets.
> Does the spec (or current best practice) address this?  (1 pkt
> lost => cwnd =1/2 ssthresh, receiver buffer fills with 1 window of
> data => a_rwnd = 0, then retransmitted pkt arrives => a_rwnd =
> full size of buffers, when this SACK arrives sender can transmit
> cwnd (ie 1/2 ssthresh) packets immediately)

Again, I am not sure what the spec says, but you should be sending
new data during the "recovery phase".  After half a RTT worth of
duplicate ACKs goes by you should then start sending a new (never
sent) segment per duplicate ACK.  Therefore, when loss recovery ends
you will have outstanding data and therefore no huge burst of loss.
I know this is in the SCTP document in spirit.  I thought it seemed
correct to me when I read it.  

(Also, rwnd can limit the sending of new data during recovery and
cause a big burst.  We might think about using something like CWV
(RFC 2861) in SCTP.bis to prevent such bursts.).

allman


---
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/


From owner-ecm@wyvern.aciri.org  Mon Feb  5 22:52:49 2001
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Date: Mon, 5 Feb 2001 19:50:27 -0800 (PST)
From: Kacheong Poon <Kacheong.Poon@eng.sun.com>
Reply-To: Kacheong Poon <Kacheong.Poon@eng.sun.com>
Subject: Re: SCTP congestion control questions 
To: mallman@grc.nasa.gov
Cc: Rob + Helena <brennanr@iol.ie>, ecm@aciri.org
In-Reply-To: "Your message with ID" <200102051830.NAA19211@guns.lerc.nasa.gov>
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> I also think you *should* ignore cwnd (I don't remember the SCTP
> spec), but RFC 2581 says that you should retransmit -- I don't
> believe there is any mention of a check to cwnd.

One important note on this is the stack should avoid sending a burst
of traffic in retransmission, even it ignores cwnd and rwnd.  For
example, in TCP, SACK info can tell the stack that more than half of
the window's data is lost (only the last 3 segments are received).
The stack should not fast retransmit the whole half window at once.

I remember (I may be mistaken) that the original new Reno which is
based on the work by J. Hoe, the stack uses something similar to "slow
start" in the fast retransmit phase to recover more than 1 lost segment
in 1 RTT.

Anyway, when the spec is not clear, the implementor should use a
good judgement on how the SCTP fast retransmit behavior should be like.
I believe there is enough info on all those RFCs which can be used to
make a good decision.

							K. Poon.
							kcpoon@eng.sun.com




From owner-ecm@wyvern.aciri.org  Tue Feb  6 09:17:14 2001
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To: Kacheong Poon <Kacheong.Poon@eng.sun.com>
From: Mark Allman <mallman@grc.nasa.gov>
Reply-To: mallman@grc.nasa.gov
cc: Rob + Helena <brennanr@iol.ie>, ecm@aciri.org
Subject: Re: SCTP congestion control questions 
Organization: NASA GRC/BBN Technologies
Song-of-the-Day: Ticket to Ride
Date: Tue, 06 Feb 2001 09:17:05 -0500
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> One important note on this is the stack should avoid sending a
> burst of traffic in retransmission, even it ignores cwnd and rwnd.
> For example, in TCP, SACK info can tell the stack that more than
> half of the window's data is lost (only the last 3 segments are
> received).  The stack should not fast retransmit the whole half
> window at once.

Definatly!

> I remember (I may be mistaken) that the original new Reno which is
> based on the work by J. Hoe, the stack uses something similar to
> "slow start" in the fast retransmit phase to recover more than 1
> lost segment in 1 RTT.

As I remember it, SCTP uses SACK by default and therefore doesn't
really have a newreno-like mechanism.

> Anyway, when the spec is not clear, the implementor should use a
> good judgement on how the SCTP fast retransmit behavior should be
> like.  I believe there is enough info on all those RFCs which can
> be used to make a good decision.

Agreed.

allman


---
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/


From owner-ecm@wyvern.aciri.org  Tue Feb  6 12:51:42 2001
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Mark:

Some comments below...

Mark Allman wrote:
> 
> Ugh -- it is not clear that this is the place to discuss this,
> but...
> 
> > 1. Should the SCTP fast retransmit proceedure respect cwnd/rwnd
> > when retransmitting a packet?
> > (Section 6.1 of the spec seems to state yes, but this means that FR
> > is not very fast :-) as it must wait for approx 1/2 a window of duplicate
> > SACKs to arrive before it can do the retransmit)
> 
> It should ingore the rwnd, for sure.  I think that since you are
> retransmitting you are simply replacing something in the window, not
> adding to it.
> 
I don't think the rwnd is ever an issue. Since when you
do a FR you:

A) Add back the packet to the rwnd you have left
and
B) Then go to do the FR..

So you never should hit the problem that the FR is gated
by the rwnd... as long as you FR right away :/

> I also think you *should* ignore cwnd (I don't remember the SCTP
> spec), but RFC 2581 says that you should retransmit -- I don't
> believe there is any mention of a check to cwnd.

I believe this may be broke in the spec. I have re-issued the
bake-off draft to include this item.

> 
> > 3. The FR proceedure outlined in section 7.2.4 (especially the
> > implementation note) seems to lead to multiple retransmits of
> > the same packet in the case where there is a large number of
> > duplicate ACKs arriving.
> > (counter goes to 4, packet is FR'd, counter is reset, 4+ duplicate
> > sacks arrive before 1 rtt => packet is FR'd again.
> > This of course also drastically reduces ssthresh and cwnd.)
> 
> I agree this seems busted.

Yep, this is also listed to be fixed in the BIS (in the
bake-off draft).

> 
> > 4. 1 Packet loss with large windows, high bandwidth and long rtts
> > can lead to very large bursts of up to half a window of packets.
> > Does the spec (or current best practice) address this?  (1 pkt
> > lost => cwnd =1/2 ssthresh, receiver buffer fills with 1 window of
> > data => a_rwnd = 0, then retransmitted pkt arrives => a_rwnd =
> > full size of buffers, when this SACK arrives sender can transmit
> > cwnd (ie 1/2 ssthresh) packets immediately)
> 
> Again, I am not sure what the spec says, but you should be sending
> new data during the "recovery phase".  After half a RTT worth of
> duplicate ACKs goes by you should then start sending a new (never
> sent) segment per duplicate ACK.  Therefore, when loss recovery ends
> you will have outstanding data and therefore no huge burst of loss.
> I know this is in the SCTP document in spirit.  I thought it seemed
> correct to me when I read it.
> 
> (Also, rwnd can limit the sending of new data during recovery and
> cause a big burst.  We might think about using something like CWV
> (RFC 2861) in SCTP.bis to prevent such bursts.).

Two things to note:

A) Yes you should start sending new data (if you have more data
   to send of course) 1/2 window later..

B) I believe the effect of RFC2861 is already present (at least
   in theory). If you do NOT start sending new data, you will
   by definition be "idle", and if you are idle you are to
   begin halving your cwnd value... 

Another thing we could do just to be a bit cautious is to 
do what Kevin/Sally suggestin "Simulation-based Comparisons of
Tahoe, Reno and SACK TCP" .. i.e. have a "maxBurst" value that
will apply after a FR has been acked. This would in theory never
be a problem if A and B are working... so it would be harmless
to add it in as a safety measure... I have done so in the
next release of the ref-implementation :)

R

> 
> allman
> 
> ---
> Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/

-- 
Randall R. Stewart
randall@stewart.chicago.il.us or rrs@cisco.com
815-342-5222 (cell) 815-477-2127 (work)


From owner-ecm@wyvern.aciri.org  Tue Feb  6 12:53:53 2001
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To: Kacheong Poon <Kacheong.Poon@eng.sun.com>
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Kacheong:

A minor comment.

Kacheong Poon wrote:
> 
> > I also think you *should* ignore cwnd (I don't remember the SCTP
> > spec), but RFC 2581 says that you should retransmit -- I don't
> > believe there is any mention of a check to cwnd.
> 
> One important note on this is the stack should avoid sending a burst
> of traffic in retransmission, even it ignores cwnd and rwnd.  For
> example, in TCP, SACK info can tell the stack that more than half of
> the window's data is lost (only the last 3 segments are received).
> The stack should not fast retransmit the whole half window at once.
> 
You are never allowed (according to RFC2960) to send more than
ONE single packet at the time a retransmit is determined... either
by FR or T-O. So for any one feedback event/SACK you should never
respond with more than one RETRANSMIT...


R


> I remember (I may be mistaken) that the original new Reno which is
> based on the work by J. Hoe, the stack uses something similar to "slow
> start" in the fast retransmit phase to recover more than 1 lost segment
> in 1 RTT.
> 
> Anyway, when the spec is not clear, the implementor should use a
> good judgement on how the SCTP fast retransmit behavior should be like.
> I believe there is enough info on all those RFCs which can be used to
> make a good decision.
> 
>                                                         K. Poon.
>                                                         kcpoon@eng.sun.com

-- 
Randall R. Stewart
randall@stewart.chicago.il.us or rrs@cisco.com
815-342-5222 (cell) 815-477-2127 (work)


From owner-ecm@wyvern.aciri.org  Tue Feb  6 13:12:12 2001
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"Randall R. Stewart" wrote:
> 
> Another thing we could do just to be a bit cautious is to
> do what Kevin/Sally suggestin "Simulation-based Comparisons of
> Tahoe, Reno and SACK TCP" .. i.e. have a "maxBurst" value that
> will apply after a FR has been acked. This would in theory never
> be a problem if A and B are working... so it would be harmless
> to add it in as a safety measure... I have done so in the
> next release of the ref-implementation :)

There is a variant of maxburst called burst-or-lose (Touch/Floyd)
that combines the need to avoiding bursts due to retransmissions and
avoiding bursts after idle periods. It was described in 
an old ID, which is still somewhat in process (though not active).
An archived version of the ID is at:

	http://www.isi.edu/touch/pubs/tcpimpl-restart-01.txt

(from an email of mine from back in Sept):

> ...regarding MaxBurst,
> Sally's version was a 'per send call' limit. It assumes that "send" is called
> on every ACK or timeout event (an implementation detail), and limits only
> stretch-ACK related bursts. MaxBurst:
> 
>         - resets the count inside the send-call to N=5
> 
> The variant called Burst or Lose" (in the earlier e-mail) augments MaxBurst
> in a few ways:
> 
>         - it retains the count (N=3 or N=5) between send-call invocations
>         - it resets the count on ACKs (not dupacks, though) and timeouts
>                 this is a 'reset to N' - it doesn't increment beyond N
> 
> The goal of BOL is twofold:
> 
>         - avoid stretch-ACK bursts
>         - avoid bursts due to source stream burstiness (idle then active)
> 
> Joe


From owner-ecm@wyvern.aciri.org  Tue Feb  6 13:25:51 2001
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> A) Yes you should start sending new data (if you have more data
>    to send of course) 1/2 window later..

Right.

> B) I believe the effect of RFC2861 is already present (at least
>    in theory). If you do NOT start sending new data, you will
>    by definition be "idle", and if you are idle you are to
>    begin halving your cwnd value... 

OK.  

> Another thing we could do just to be a bit cautious is to do what
> Kevin/Sally suggestin "Simulation-based Comparisons of Tahoe, Reno
> and SACK TCP" .. i.e. have a "maxBurst" value that will apply
> after a FR has been acked. This would in theory never be a problem
> if A and B are working... so it would be harmless to add it in as
> a safety measure... I have done so in the next release of the
> ref-implementation :)

I have never been convinced that I should be a huge fan of such
blanket mechanisms.  My hit is that if there is a situation that
causes these bursts, we should fix the root cause.

allman


From owner-ecm@wyvern.aciri.org  Tue Feb  6 13:34:17 2001
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Mark Allman wrote:
> 
> > Another thing we could do just to be a bit cautious is to do what
> > Kevin/Sally suggestin "Simulation-based Comparisons of Tahoe, Reno
> > and SACK TCP" .. i.e. have a "maxBurst" value that will apply
> > after a FR has been acked. This would in theory never be a problem
> > if A and B are working... so it would be harmless to add it in as
> > a safety measure... I have done so in the next release of the
> > ref-implementation :)
> 
> I have never been convinced that I should be a huge fan of such
> blanket mechanisms.  My hit is that if there is a situation that
> causes these bursts, we should fix the root cause.

The root cause is that bursting is the result of implementation 
decisions that are not otherwise prohibited
by the protocol.

Using Sally's "don't overspecify", the goal of Burst-or-lose
was to put exactly enough protocol mechanism in place to 
prevent accumulation of 'permission to send', which is otherwise
not prohibited by the protocol.

Joe


From owner-ecm@wyvern.aciri.org  Tue Feb  6 13:43:01 2001
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> The root cause is that bursting is the result of implementation
> decisions that are not otherwise prohibited by the protocol.

Right.  Why not fix these bad decisions?  (Yes, there is a good
chance this is simply philosophical).

allman


---
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/


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Mark Allman wrote:
> 
> > The root cause is that bursting is the result of implementation
> > decisions that are not otherwise prohibited by the protocol.
> 
> Right.  Why not fix these bad decisions?  (Yes, there is a good
> chance this is simply philosophical).

Depends on what is 'wrong'.

If it is the behavior that is wrong (bursting), then 
fix the behavior (burst limit).

I'm not clear there is a fundamental reason behind
causing a burst; it's a symptom, and it seems 
appropriate to treat the symptom.

Otherwise we're back-tracking 'intent' - anything that _might_
result in an implementation that _might_ end up bursting is
deemed 'wrong'.

Sometimes cold medicine is exactly what you want.

Joe


From owner-ecm@wyvern.aciri.org  Tue Feb  6 16:25:58 2001
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CC: mallman@grc.nasa.gov, Rob + Helena <brennanr@iol.ie>, ecm@aciri.org
Subject: Re: SCTP congestion control questions
References: <200102061843.NAA06372@guns.lerc.nasa.gov> <3A80469C.3E756FE4@isi.edu>
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Mark/Joe:

One thing to remember on this "burst" is the
fact that this all supposedly occurs (if I remember
right from reading Rob's work... Rob help my
feeble overloaded mind here :->) .. is it
occurs in one RTT. So, thinking on it, none
of the reductions  may have taken place.. but of
course you would have to have a very strange occurance
in that 

A) The application would not have more data to clock
   out (as you point out Mark).

and

B) Just has the FR's ACK arrives the application would
   have to have this "Burst" of new data. 

These two things seem to me to be unlikely but I don't
see how other than by having a maxBurst you can gate
this strange behavior....


And I am sorry Joe, I have not read your draft :< I have
pulled it down.. but alas  I fear it will be a while before
I can spare the cycles ...

R

Joe Touch wrote:
> 
> Mark Allman wrote:
> >
> > > The root cause is that bursting is the result of implementation
> > > decisions that are not otherwise prohibited by the protocol.
> >
> > Right.  Why not fix these bad decisions?  (Yes, there is a good
> > chance this is simply philosophical).
> 
> Depends on what is 'wrong'.
> 
> If it is the behavior that is wrong (bursting), then
> fix the behavior (burst limit).
> 
> I'm not clear there is a fundamental reason behind
> causing a burst; it's a symptom, and it seems
> appropriate to treat the symptom.
> 
> Otherwise we're back-tracking 'intent' - anything that _might_
> result in an implementation that _might_ end up bursting is
> deemed 'wrong'.
> 
> Sometimes cold medicine is exactly what you want.
> 
> Joe

-- 
Randall R. Stewart
randall@stewart.chicago.il.us or rrs@cisco.com
815-342-5222 (cell) 815-477-2127 (work)


From owner-ecm@wyvern.aciri.org  Tue Feb  6 16:50:22 2001
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Subject: Re: SCTP congestion control questions
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"Randall R. Stewart" wrote:
> 
> Mark/Joe:
> 
> One thing to remember on this "burst" is the
> fact that this all supposedly occurs (if I remember
> right from reading Rob's work... Rob help my
> feeble overloaded mind here :->) .. is it
> occurs in one RTT. So, thinking on it, none
> of the reductions  may have taken place.. but of
> course you would have to have a very strange occurance
> in that
> 
> A) The application would not have more data to clock
>    out (as you point out Mark).
> 
> and
> 
> B) Just has the FR's ACK arrives the application would
>    have to have this "Burst" of new data.
> 
> These two things seem to me to be unlikely but I don't
> see how other than by having a maxBurst you can gate
> this strange behavior....

FWIW, maxburst limits per-event - which means a 
bad implementation can have the application keep
doing send calls, each of which can burst.

The idea of BOL was to gate the behavior with more
of a leaky-bucket - refilled when an ACK arrives, 
and drained when packets are sent. 

There are subtle differences, but they
are basically similar in how they gate things,
and both are proactive, not reactive.

Joe


From owner-ecm@wyvern.aciri.org  Tue Feb  6 19:13:51 2001
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From: "Rob + Helena" <brennanr@iol.ie>
To: "Randall R. Stewart" <randall@stewart.chicago.il.us>,
        "Joe Touch" <touch@isi.edu>
Cc: <mallman@grc.nasa.gov>, <ecm@aciri.org>
Subject: Re: SCTP congestion control questions
Date: Wed, 7 Feb 2001 00:15:47 -0000
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Hi all

----------
> From: Randall R. Stewart <randall@stewart.chicago.il.us>
> 
> Mark/Joe:
> 
> One thing to remember on this "burst" is the
> fact that this all supposedly occurs (if I remember
> right from reading Rob's work... Rob help my
> feeble overloaded mind here :->) .. is it
> occurs in one RTT. So, thinking on it, none
> of the reductions  may have taken place.. but of
> course you would have to have a very strange occurance
> in that 
> 
> A) The application would not have more data to clock
>    out (as you point out Mark).
> 
> and
> 
> B) Just has the FR's ACK arrives the application would
>    have to have this "Burst" of new data. 
> 
> These two things seem to me to be unlikely but I don't
> see how other than by having a maxBurst you can gate
> this strange behavior....

If anyone is interested I have put a figure illustrating a
SCTP burst after recovery at:
http://www.teltec.dcu.ie/~brennanr/sctp.html
This comes from my simulation model of SCTP running
over a long delay/high bandwidth link with a droptail
gateway with short qs.

I think that Randal is correct about the non-decay of cwnd
as in the figure we are sending at least 1 pkt per rtt (in fact the
decay is linked to rto rather than rtt directly and these may be
different after a loss).

On the topic of clocking out more data during recovery:
Note that when FR retransmission occurs there is already
a window of data in flight. This is all out of order so is buffered
at the receiver => a_rwnd is reduced to 0. This means that the
sender is rwnd rather than cwnd limited. When the SACK for
the FR'd pkt arrives the q'd data hasn't yet been sent to the
application so a_rwnd is still 0. No data is in flight so the sender
can then send just one more packet. the SACK for this has a_rwnd
= window size. When the receiver gets this it is no longer rwnd
limited => it can send up to cwnd of data. Cwnd was set to half
of the max window when the FR occured => 1/2 a window of
data is sent.

rgds
rob







From owner-ecm@wyvern.aciri.org  Tue Feb  6 19:13:55 2001
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From: "Rob + Helena" <brennanr@iol.ie>
To: <mallman@grc.nasa.gov>, "Joe Touch" <touch@isi.edu>
Cc: "Randall R. Stewart" <randall@stewart.chicago.il.us>, <ecm@aciri.org>
Subject: Re: SCTP congestion control questions 
Date: Tue, 6 Feb 2001 23:30:14 -0000
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Hi All

----------
> From: Mark Allman <mallman@grc.nasa.gov>
> 
> > The root cause is that bursting is the result of implementation
> > decisions that are not otherwise prohibited by the protocol.
> 
> Right.  Why not fix these bad decisions?  (Yes, there is a good
> chance this is simply philosophical).

I'm not sure what you are implying here, that implementors should
fix their code or that we should fix the spec.

As a relative newcomer to TCP/SCTP congestion control, my (perhaps
naive) opinion is:
if the protocol specification is sufficiently loose that a problem like
bursts
can occur with poor "implementation decisions" then we should develop
a minimally intrusive fix to the protocol specification that prevents this
behaviour for conformant implementations.

This area of protocol stack development is sufficiently obtuse that
these problems won't become obvious until after deployment (or
at least some very rigourous testing).

rgds
rob




From owner-ecm@wyvern.aciri.org  Tue Feb  6 20:30:12 2001
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Date: Tue, 6 Feb 2001 17:29:25 -0800 (PST)
From: Kacheong Poon <Kacheong.Poon@eng.sun.com>
Reply-To: Kacheong Poon <Kacheong.Poon@eng.sun.com>
Subject: Re: SCTP congestion control questions 
To: Rob + Helena <brennanr@iol.ie>
Cc: mallman@grc.nasa.gov, Joe Touch <touch@isi.edu>,
        "Randall R. Stewart" <randall@stewart.chicago.il.us>, ecm@aciri.org
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> As a relative newcomer to TCP/SCTP congestion control, my (perhaps
> naive) opinion is:
> if the protocol specification is sufficiently loose that a problem like
> bursts
> can occur with poor "implementation decisions" then we should develop
> a minimally intrusive fix to the protocol specification that prevents this
> behaviour for conformant implementations.

I share your concern as an implementor.  But sometimes, IMHO BTW, that
there is a gray area.  In this particular case, I believe an implementation
can have several ways to avoid the problem.  And any of those ways will
not cause interoperability problem.  So should a solution be specified in the
protocol as long as the problem is clearly specified?  Or should this be an
implementation choice?

> This area of protocol stack development is sufficiently obtuse that
> these problems won't become obvious until after deployment (or
> at least some very rigourous testing).

Again in this particular case, this is a well known problem such that
most, if not all, TCP stacks out there have some form of remedy.  So 
people implementing SCTP should learn from past experience of TCP.  I agree
that there can be other problems which will not be found without rigorous
testing.  This is a reason why some people think that a transport protocol
needs several years of real life experience to be called stable...

							K. Poon.
							kcpoon@eng.sun.com




From owner-ecm@wyvern.aciri.org  Wed Feb  7 04:32:07 2001
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From: "Rob + Helena" <brennanr@iol.ie>
To: "Kacheong Poon" <Kacheong.Poon@eng.sun.com>
Cc: <mallman@grc.nasa.gov>, "Joe Touch" <touch@isi.edu>,
        "Randall R. Stewart" <randall@stewart.chicago.il.us>, <ecm@aciri.org>
Subject: Re: SCTP congestion control questions 
Date: Wed, 7 Feb 2001 09:31:41 -0000
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----------
> From: Kacheong Poon <Kacheong.Poon@eng.sun.com>
> 
> > As a relative newcomer to TCP/SCTP congestion control, my (perhaps
> > naive) opinion is:
> > if the protocol specification is sufficiently loose that a problem like
> > bursts
> > can occur with poor "implementation decisions" then we should develop
> > a minimally intrusive fix to the protocol specification that prevents
this
> > behaviour for conformant implementations.
> 
> I share your concern as an implementor.  But sometimes, IMHO BTW, that
> there is a gray area.  In this particular case, I believe an
implementation
> can have several ways to avoid the problem.  And any of those ways will
> not cause interoperability problem.  So should a solution be specified in
the
> protocol as long as the problem is clearly specified?  Or should this be
an
> implementation choice?

OK. However I would suggest that the current SCTP spec should include
at least a statement that this is a problem that implementations SHOULD
avoid and perhaps an implementation note giving references to
research or other IETF work which has possible solutions.
> 
> > This area of protocol stack development is sufficiently obtuse that
> > these problems won't become obvious until after deployment (or
> > at least some very rigourous testing).
> 
> Again in this particular case, this is a well known problem such that
> most, if not all, TCP stacks out there have some form of remedy.  So 
> people implementing SCTP should learn from past experience of TCP.  I
agree
> that there can be other problems which will not be found without rigorous
> testing.  This is a reason why some people think that a transport
protocol
> needs several years of real life experience to be called stable...

Agreed for the TCP case. For SCTP there is a new generation of 
implementors/vendors who may not be aware of the full gamut of
TCP research. Also the subtle differences between SCTP CC
and TCP CC can make it hard to see exactly which research is relevant.
This is why I think it would be good to point out the problem in the
current SCTP specification. 

rgds
rob


From owner-ecm@wyvern.aciri.org  Wed Feb  7 08:04:25 2001
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Date: Wed, 07 Feb 2001 07:04:22 -0600
From: "Randall R. Stewart" <randall@stewart.chicago.il.us>
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To: Rob + Helena <brennanr@iol.ie>
CC: Joe Touch <touch@isi.edu>, mallman@grc.nasa.gov, ecm@aciri.org
Subject: Re: SCTP congestion control questions
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Rob + Helena wrote:
> 
> Hi all
> 
> ----------
> > From: Randall R. Stewart <randall@stewart.chicago.il.us>
> >
> > Mark/Joe:
> >
> > One thing to remember on this "burst" is the
> > fact that this all supposedly occurs (if I remember
> > right from reading Rob's work... Rob help my
> > feeble overloaded mind here :->) .. is it
> > occurs in one RTT. So, thinking on it, none
> > of the reductions  may have taken place.. but of
> > course you would have to have a very strange occurance
> > in that
> >
> > A) The application would not have more data to clock
> >    out (as you point out Mark).
> >
> > and
> >
> > B) Just has the FR's ACK arrives the application would
> >    have to have this "Burst" of new data.
> >
> > These two things seem to me to be unlikely but I don't
> > see how other than by having a maxBurst you can gate
> > this strange behavior....
> 
> If anyone is interested I have put a figure illustrating a
> SCTP burst after recovery at:
> http://www.teltec.dcu.ie/~brennanr/sctp.html
> This comes from my simulation model of SCTP running
> over a long delay/high bandwidth link with a droptail
> gateway with short qs.
> 
> I think that Randal is correct about the non-decay of cwnd
> as in the figure we are sending at least 1 pkt per rtt (in fact the
> decay is linked to rto rather than rtt directly and these may be
> different after a loss).
> 
> On the topic of clocking out more data during recovery:
> Note that when FR retransmission occurs there is already
> a window of data in flight. This is all out of order so is buffered
> at the receiver => a_rwnd is reduced to 0. This means that the
> sender is rwnd rather than cwnd limited. When the SACK for
> the FR'd pkt arrives the q'd data hasn't yet been sent to the
> application so a_rwnd is still 0. No data is in flight so the sender
> can then send just one more packet. the SACK for this has a_rwnd
> = window size. When the receiver gets this it is no longer rwnd
> limited => it can send up to cwnd of data. Cwnd was set to half
> of the max window when the FR occured => 1/2 a window of
> data is sent.
> 
Rob:

I think this is very similar to what Kevin and Sally discuss in
"Simulation-based Comparisons of Tahoe,Reno and SACK TCP".
In this paper they mention sticking a "maxBurst" on SACK TCP
to prevent this behaivor...

I think this provides the needed safety value for the obscure
places that you will get these results.. In general the
normal mechanisms that Mark and I have discussed will address
these problems....

If someone else has a better suggestion I would be glad to
hear it :) Mark... any ideas?

R

-- 
Randall R. Stewart
randall@stewart.chicago.il.us or rrs@cisco.com
815-342-5222 (cell) 815-477-2127 (work)


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