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From: "Long, Darren" <darren.long@uk.thalesgroup.com>
To: "'tsvwg@ietf.org'" <tsvwg@ietf.org>
Date: Fri, 1 Aug 2003 11:01:59 +0100 
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Subject: [Tsvwg] The UDP-Lite Protocol - comment
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Is it not possible that the UDP-Lite header may define an arbitrary region
in the packet payload for checksum protection, and not restrict this region
to starting at the first byte of the payload? I don't think that the current
proposed scheme offers enough flexibility, and think that this capability
would make a valuable addition.

Regards

Darren Long

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To: Kacheong Poon <kacheong.poon@sun.com>
CC: tsvwg@ietf.org
Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
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Kacheong Poon wrote:

> Pasi Sarolahti wrote:
>
> > And after the examples shown by Sourabh, I believe detecting spurious
> > RTOs with multihoming brings interesting questions for both F-RTO and
> > Eifel that I need to study more thoroughly.
>
> I guess after all the discussions so far, SCTP implementors should
> think about not to follow what is recommended in RFC 2960, timeout
> retransmission to an alternate address.  Just a suggestion, it seems
> that at least for the first timeout, the retransmission is probably
> better sent to the original address.  And when this first retransmission
> is also lost leading to another timeout, SCTP should start considering
> using an alternate address.
>
Kacheong:

Slowly catching up on a HUGE backlog of email :-o

As to the above.. I don't agree yet.. Armando has proposed such  to me
and I have yet to be convinced....

It may be that it is better in some cases.. but when you start factoring in
real failures of a NIC you may find that your failure detection starts to
be effected in a adverse way. So far I think Armando has a good point
in that possibly FR's may be best NOT to send to an alternate .. but
T3's I think may still be best sent to an alternate...

Some of Armando's upcoming work will hopefully settle this :>

[now back to catching up on the timestamp thread... and then I will comment
 on that :> ]

R


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Subject: Re: [Tsvwg] SCTP Timestamps
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Brian:

A question or two below on your response here...

Brian F. G. Bidulock wrote:

>Sourabh,
>
>I agree that HEARTBEAT/ACK is not a good way to do this.  Although the
>opaque data in the chunk looks attractive, there are rather narrow rules
>for the handling of HB/ACK that have a rather different purpose.  SCTP
>relies on the HB mechanism for recovery of unavailable destinations in
>the multihomed environment and is also important for detection of lost
>ABORTs.  There are some items concerning HB that was discovered in interop
>testing that showed that rather special rules were required to accomodate
>implementations that could not (or did not) control source address vs. those
>that did.
>
>I think that by overloading HB/ACK for the purpose of doing more frequent
>RTT measurements would break the original purpose of HB/ACK: 
>
I am unclear on these three:

>testing the
>availability of desintations, 
>

>keeping RTO current on idle destinations, 
>

>and
>detecting lost ABORT chunks.
>  
>

If you (as proposed by some) include a HB with every packet carrying data
what about this breaks the above three items?

I would think you would still get an abort if the association was down 
... you would
with just sending data..

You would get a RTO update on whatever path you sent upon.. idle or not..

and you could still test a destination's reachability...

Now I am not yet through this thread ... and I am not convinced either 
way.. but
I just don't get how any of the above three are broken by the HB 
proposal.. not
sure who actually proposed it :>

R

>I too would prefer to see a different chunk, (TS chunk) bundled with all
>outgoing packets containing DATA and use an extension of the SACK chunk to 
>return pertinent timestamps.  (SACK is the right place to put it on the
>return anyways.)
>
>--brian
>
>On Wed, 16 Jul 2003, Sourabh Ladha wrote:
>
>  
>
>>>Yes, you are right. The HB-ACK becomes like another SACK. I like these
>>>ideas. Since the information contained in the HB is actually sender
>>>implementation specific, each implementation can embed what ever info they
>>>find most useful. It's up to that implementation to be effecient with
>>>overhead tradeoffs.
>>>      
>>>
>>I do not agree at all if this is a "clean" way to proceed if a timestamp
>>information per packet is required. The cost that you incur by
>>this is a whole new definition of SACK, HB-ACK and rules associated with
>>these. (For example you would still need SACKs along with HB-ACKs if an
>>out of order, duplicate chunk is recieved)
>>
>>I still think if needed an optional timestamp chunk is the best way to go
>>if such an information is required.
>>
>>
>>-Sourabh
>>
>>
>>_______________________________________________
>>tsvwg mailing list
>>tsvwg@ietf.org
>>https://www1.ietf.org/mailman/listinfo/tsvwg
>>    
>>
>
>  
>



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Subject: RE: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
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-----Original Message-----
From: tsvwg-admin@ietf.org [mailto:tsvwg-admin@ietf.org]On Behalf Of
Randall R. Stewart (home)
Sent: Monday, August 04, 2003 7:19 AM
To: Kacheong Poon
Cc: tsvwg@ietf.org
Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)


Kacheong Poon wrote:

> Pasi Sarolahti wrote:
>
> > And after the examples shown by Sourabh, I believe detecting spurious
> > RTOs with multihoming brings interesting questions for both F-RTO and
> > Eifel that I need to study more thoroughly.
>
> I guess after all the discussions so far, SCTP implementors should
> think about not to follow what is recommended in RFC 2960, timeout
> retransmission to an alternate address.  Just a suggestion, it seems
> that at least for the first timeout, the retransmission is probably
> better sent to the original address.  And when this first retransmission
> is also lost leading to another timeout, SCTP should start considering
> using an alternate address.
>
Kacheong:

Slowly catching up on a HUGE backlog of email :-o

As to the above.. I don't agree yet.. Armando has proposed such  to me
and I have yet to be convinced....

It may be that it is better in some cases.. but when you start factoring in
real failures of a NIC you may find that your failure detection starts to
be effected in a adverse way. So far I think Armando has a good point
in that possibly FR's may be best NOT to send to an alternate .. but
T3's I think may still be best sent to an alternate...

Some of Armando's upcoming work will hopefully settle this :>

[now back to catching up on the timestamp thread... and then I will comment
 on that :> ]

R

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

I think the central question here is what is the assumed
characteristics of each leg of the path. The current SCTP
path algorithm is optimized on the assumption that a drop
represents a problem on the path attempted - congestion
or failure.

So is that a reasonable assumption? And more to the point,
if there are special characteristics of a final hop that
produce more drops, is it appropriate for them to be solved
by end-to-end retransmission?

At least some of the arguments for changing the path
selection algorithm have cited environments where the
final hop is wireless. I do not believe it is a wise
strategy to correct a higher error rate over a wireless
link with an end-to-end retransmit. Wirelss systems are
prone to burst noise. This will be interpeted by the
end-to-end flow control algorithms as severe congestion
and result in excessive rate trimming.

If each hop brings itself to a reasonable standard of
probable delivery, and then the consensus is that a single
drop probably does not represent a problem with a path,
then it might be a good idea to change the algorithm.
But I do not think that changing an end-to-end algorithm
is a good response if the real problem is confined to
a single hop.




Caitlin Bestler - http://asomi.com/


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Stephan/all:

Ok I have now read through this entire thread and I will
use Stephan's consolidated issues to comment on things...

One large META comment I have to this discussion.. One thing
we were VERY careful NOT to do is to require that a Sender Bundle
chunks together. It IS required that a receiver understand how to
"un-bundle" chunks.. but it is NEVER required that a sender bundle
chunks... This is one problem I have with the Time-stamp chunk that
has been talked about so-far.. it requires in some aspects that the
TS-Response be bundled with SACK's .... near as I can tell... this
I do NOT like...

Now it is true that most implementations that I have
tested with DO bundle things together.. but if I remember right there
as been one or two at past bake-offs that did NOT do any bundling... these
may have changed but I hate to see such a restriction show up... i.e. that
we add something that MUST be bundled...

Now on to comments below :>


Stephan Baucke wrote:

>
> I'll try to consolidate the issues:
>
> 1) Increasing the RTT sampling frequency: Benefits are still
> controversial ([1] [2] [3]), could be achieved using either timestamps
> or implementation techniques (at the cost of additional state at the 
> sender) 


Yep, I would imagine that HB's can do a similar job (as mentioned by 
others) to
estimate more often the RTT.. not that I am sure we need this...

>
>
> 2) RTO inflation on alternate paths due to inability to time
> retransmissions [4]: Can be addressed by changed retransmission policy,
> specially timed heartbeats, timestamps, or other method to resolve the
> retransmission ambiguity ("retransmission bits") 

I think this can be solved as well by sending a HB after doing a
retransmission to an alternate path... I put this in the KAME stack
a while ago when Armando and I were "debating" some of the "merits"
of retransmitting to the same address :>

>
>
> 3) Sequence number wrap-around within packet lifetime in high-speed
> networks [1]: This affects SCTP equally as TCP (both use 32-bit seq.
> number spaces), can be addressed by using timestamps (PAWS)

Hmm I do wonder about this... if I am sending 1440 bytes in
each data chunk this means the 32 bit TSN space will need 1440 * 2^32  * 
8(bits)
to wrap... doing a bit of napkin scratch.. on a 10Gigabit ethernet thats
more than 4000 seconds before a sequence wrap... would this really be an
issue?


>
> 3) Incorrect congestion window growth due to crediting of SACKs to the
> wrong destination address or unawareness of failover [5]: Can be
> resolved by timestamps and/or Rhein algorithm (? not sure if these 
> really cover the same problem space; Jana?)

This can be solved with a sender side algorithm.. the KAME stack
includes such a solution that we developed while working on [5]. I don't
see that a TS would be needed for this.. they are an alternative 
method.. but I
don't think are required..


>
> 4) Incorrect error counting due to crediting of SACKs to the wrong
> destination address: Can be addressed using timestamps or other means to
> resolve retransmission ambiguity (preferably for more than one
> retransmission) 

Yes, a TS could do this.. but so could sending a keyed HB a key
for each transmission.. lots of state by the sender and yes there is
the issue of how many HB's are being sent... but I am not
yet convinced about adding a TS chunk...

>
>
> 5) Detection and/or prevention of spurious retransmissions: Detection
> possible using Eifel [6] (either with timestamps or new flags), Dup-TSNs
> [7], or F-RTO [8]. Prevention possible with Eifel and F-RTO (?)


It seems that a TS is needed for [6] but that [7] and [8] (as you point out)
also address this problem... and so I am still not convinced that
we need a TS-Chunk :-0

There may even be some other method of doing an Eifel like detection with
HB's and or some combination of what as been discussed in this thread.. not
sure yet.. I have a lot of catching up to do.. and thinking on this..

But for the moment I don't see a compelling need for a TS-chunk yet...

R

>
>
> References:
> [1] RFC 1323
>
> [2] RFC 3481 (and citations [45], [52], [54], [60] therein)
>
> [3] M. Allman, V. Paxson, "On Estimating End-to-End Network Path
> Properties", SIGCOMM'99
>
> [4] Armando L. Caro Jr., Paul D. Amer, Randall R. Stewart, "Transport
> Layer Multihoming for Fault Tolerance in FCS Networks", MILCOM 2003
> http://www.cis.udel.edu/~acaro/papers/2003.milcom.acaro.pdf
>
> [5] Janardhan R. Iyengar, Armando L. Caro Jr., Paul D. Amer, Gerard J.
> Heinz, Randall R. Stewart, "SCTP Congestion Window Overgrowth During
> Changeover", SCI 2002
> http://www.cis.udel.edu/~acaro/papers/2002.sci.iyengar.pdf
>
> [6] RFC 3522
>
> [7] draft-ietf-tsvwg-dsack-use-00.txt
>
> [8] draft-sarolahti-tsvwg-tcp-frto-04.txt
>
>
>
>
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>
>



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(way late, as usual ... sorry)

> The long-term solution to this problem is SACK (to prevent
> unnecessary retransmissions) and DSACK (to resolve the DUPACK
> ambiguity if unnecessary retransmissions do happen).  However,
> most TCP connections today still do not use SACK.
> http://www.icir.org/floyd/sack-questions.html
> 
> Until SACK is universally deployed two methods to resolve the
> DUPACK ambiguity might be useful.

First, I am not sure I am willing to believe that SACK is not pretty
well deployed.  3 years ago I reported 40% of the web clients that
hit our web server here at NASA were using SACK (and the trend was
growing steadily) [All00].  I don't have good data from that server
right now, but I took a quick look at a server at BBN that I have
been monitoring for the last ~11 days.  In that time I have seen
6545 web connections (yeah, not terribly heavy traffic) and 6183 of
the remote hosts (94%) advertised SACK.  That seems like pretty good
deployment to me.

I am not sure what that says about additional "bugfix" tricks.  Do
we need them?  Do we not need them?  The above data is not
necessarily conclusive or representative.  The changes would add
complexity to a TCP implementation.  On the other hand, not that
much complexity and maybe would provide a decent enough win.  But,
when do we stop patching known suboptimal recovery?  Etc., Etc.,
Etc.    ??????

> ACK-heuristic: if seq(ack_new)-seq(ack_prev)<=2 MSS then trigger
> fast retransmit on 3rd DUPACK This method is based on observation
> that to trigger a false fast retransmit, a TCP sender has to
> retransmit unnecessarily at least 3 adjucent packets. In this case
> there will be a jump in a cumulative ACK at the sender of at least
> three segments.

I am trying to wrap my head around this and cannot yet do so.  Can
you define "ack_new" and "ack_prev"?  Are these the two ACKs before
duplicate ACKs start rolling in?

> Using the timestamp option: if ts_echo(ack_new) == ts (seg_una)
> then trigger fast retransmit on 3rd DUPACK In case of a missing
> segment both RFC1323 or RFC1323bis (expired update draft) require
> that DUPACKs echo a timestamp from the segment below the missing
> one.

Timestamps are less prevalent (by default) than SACK is, I think
(11% of the connection in the BBN trace described above, which is in
the ballpark of what was reported in [All00], too).  So, I am not
sure I think this is a practical approach.

[All00] Mark Allman. A Web Server's View of the Transport Layer, ACM
    Computer Communication Review, 30(5), October 2000.
    http://roland.grc.nasa.gov/~mallman/papers/webobs-ccr.ps

allman


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

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Just a minor comment.

> > 3) Incorrect congestion window growth due to crediting of SACKs to the
> > wrong destination address or unawareness of failover [5]: Can be
> > resolved by timestamps and/or Rhein algorithm (? not sure if these
> > really cover the same problem space; Jana?)
>
> This can be solved with a sender side algorithm.. the KAME stack
> includes such a solution that we developed while working on [5]. I don't
> see that a TS would be needed for this.. they are an alternative
> method.. but I
> don't think are required..


Congestion window overgrowth caused by retransmission ambuiguity will not
be solved by CACC in [5]. I believe that changeover specific
cwnd overgrowth is solved by [5].

-Sourabh


> > [5] Janardhan R. Iyengar, Armando L. Caro Jr., Paul D. Amer, Gerard J.
> > Heinz, Randall R. Stewart, "SCTP Congestion Window Overgrowth During
> > Changeover", SCI 2002
> > http://www.cis.udel.edu/~acaro/papers/2002.sci.iyengar.pdf
> >


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To: "Armando L. Caro Jr." <acaro@acm.org>
CC: Schruefer Wolfgang <wolfgang.schruefer@siemens.com>,
        "'ladha@mail.eecis.udel.edu'" <ladha@mail.eecis.udel.edu>,
        "'Stephan Baucke'" <stephan.baucke@ericsson.com>, tsvwg@ietf.org
Subject: Re: AW: [Tsvwg] SCTP Timestamps
References: <Pine.GSO.4.33.0307211028470.217-100000@stimpy.eecis.udel.edu>
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Armando L. Caro Jr. wrote:

>On Mon, 21 Jul 2003, Schruefer Wolfgang wrote:
>
>  
>
>>>If you are not following this argument, _please_ read at
>>>least section 3
>>>of our paper. It's a short read. The full paper is only 5 pages.
>>>http://www.cis.udel.edu/~acaro/papers/2003.milcom.acaro.pdf
>>>      
>>>
>>I studied your paper with high interest, but my conclusion is that a
>>mirrored Retransmission Flag would be a perfect solution for the
>>scenario you describe in the paper.
>>
>>With the mirrored Retransmission Flag you _can_ recalculate RTO on the
>>second path. You just can not do it for the third or further paths.
>>And here I think that the third path (resp. the a RTO calculation for
>>the third transmission) may not be so important to improve.
>>    
>>
>
>Well, your one bit solution only works if the sender follows the
>recommended mechanism for retransmissions. There hasn't been a consensus
>that retransmitting on an alternate path is always the best policy. In
>fact, I would argue that it isn't. I believe that for better performance
>without affecting failover performance, fast retransmits should go to the
>same destination and timeout retransmits should go to an alternate
>destination [1].  With such a retransmission policy, one bit would
>not be enough. Sure we could use 2 bits, but timestamps just seems
>cleaner.
>  
>
Armando:

A question for you, not that I support the bit approach,  I don't
understand why one bit is not enough.

1) I know where the primary destionation is
2) I know at any time where I last sent the chunk and if
    the chunk was FR'd or not

Between all of these known's I don't see that more than one bit
would be required to detect the FIRST retransmission no matter
if it was sent to the primary or secondary..

R


>[1] http://www.cis.udel.edu/~acaro/papers/2003.icon.acaro.pdf
>
>~armando
>
>0--                                                  --0
>| Armando L. Caro Jr.      |  Protocol Engineering Lab |
>| www.cis.udel.edu/~acaro  |    University of Delaware |
>0--                                                  --0
>
>
>_______________________________________________
>tsvwg mailing list
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>
>
>  
>



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To: Caitlin Bestler <cait@asomi.com>
CC: "Armando L. Caro Jr." <acaro@acm.org>, tsvwg@ietf.org
Subject: Re: [Tsvwg] Meaning of path failure (was re SCTP Timestamps)
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Caitlin Bestler wrote:

>
> On Monday, July 21, 2003, at 09:35 AM, Armando L. Caro Jr. wrote:
>
>>>
>>
>> Well, your one bit solution only works if the sender follows the
>> recommended mechanism for retransmissions. There hasn't been a consensus
>> that retransmitting on an alternate path is always the best policy. In
>> fact, I would argue that it isn't. I believe that for better performance
>> without affecting failover performance, fast retransmits should go to 
>> the
>> same destination and timeout retransmits should go to an alternate
>> destination [1].  With such a retransmission policy, one bit would
>> not be enough. Sure we could use 2 bits, but timestamps just seems
>> cleaner.
>>
>> [1] http://www.cis.udel.edu/~acaro/papers/2003.icon.acaro.pdf
>>
>> ~armando
>>
>
> In the paper cited, you fault the default SCTP behavior for assuming
> "that loss indicates either that the destination address used is 
> unreachable,
> or the destination's network path is congested."
>
> The implication is that the end-to-end algorithm is "giving up too 
> quickly"
> on a path, and thereby degrading performance.
>
> As that I have seen this objection most often raised when the final hop
> is a mobile and/or wireless link, I am tempted to turn that question 
> around.
>
> Isn't the real problem that the last hop is giving up on L2 delivery too
> quickly, and creating an L3/end-to-end problem for something that
> should have been handled locally?
>
> Basically, is each downstream frequency *really* a path? Or should
> the collective set of frequencies be viewed as a single L3 path
> over which there may be L2 retries?
>
> I designed a two-way CATV system to carry IP traffic through
> set-top decoders while I was at Zenith Electronics. Whenever
> the set-top needed to change frequency it would be deaf to
> the headend for a short period. Realizing that dropping
> packets during such a frequency shift would lead to very
> poor end-to-end performance, we simply scheduled the
> packet to be L2-delivered on a frequency as soon as it
> was known, or on all frequencies if the receiver's status
> was not known.
>
> We certainly do not want to begin an L2 design debate
> here, but I think it is a valid question -- is it a valid assumption
> that L2 delivery should be dealing with truly transitory
> problems and therefore packet loss really should be
> taken to as an indication that the destination address used
> is unreachable or the destination's network path is
> congested? 

Caitlin:

As always an interesting comment from you..( yes I have a lot of
other unread email that you have sent to me .. I will get to it I 
promise :>)

Now I understand the logic here and I have two comments...

a) Being able to seperate a L2 drop for NON congestion reasons from the
    "default" assumption would be a nice thing.. I actually will have a 
proposal
    to put forward on this later on when I can get the cycles to clean 
up the drafty
    draft I have in the works somewhere :>

b) If your example L2 environment just delivered the packet late (as I 
assume it
    must have) then what you have is a spurious retransmission.. the 
very thing
    that Eifel is trying to detect...

Just a thought or two :>

R



>
>
> Caitlin Bestler - cait@asomi.com - http://asomi.com/
>
>
> _______________________________________________
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Hi all:

I think I have now gotten all of the minor changes in
to the WG last call on the PR-SCTP document. Please have a
look at the html diff at:

http://www.sctp.org

under the RFC/internet draft tab  you will find at the
bottom a rfc-chgbar/diff that highlights the few
changes..

Please let me know if I got everyones comments.. I would like
to get this in so that we can move the document forward..

Thanks


R
---

Randall Stewart




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Hi, Randy,

Some minor editorial comments:

1. The new implementation note:

"      IMPLEMENTATION NOTE: An implementation may wish to limit the
       number of duplicate FORWARD TSN'S sent by either sending a
       duplicate FORWARD TSN every other SACK or waiting a full RTT
       before sending a duplicate FORWARD TSN.
"

may read better in the following form:

"      IMPLEMENTATION NOTE: An implementation may wish to limit the
       number of duplicate FORWARD TSN chunks it sends by either only
       sending a duplicate FORWARD TSN every other SACK or waiting for
       a full RTT before sending a duplicate FORWARD TSN.
"

2. The use of "forward" and "FORWARD" is inconsistent in a couple of
places.

 - page 12, in para C4)

  "number in the forwarded TSN. ..."

  should be

  "number in the FORWARD TSN. ..."

 - page 13, in para F2)

  "for generating a forward TSN to be ..."

  should be 

  "for generating a FORWARD TSN to be ..."

I am referring to the old page numbers in -00 here but they should be
close or the same in the -01 version you are working on.

-Q

"Randall R. Stewart (home)" wrote:
> 
> Hi all:
> 
> I think I have now gotten all of the minor changes in
> to the WG last call on the PR-SCTP document. Please have a
> look at the html diff at:
> 
> http://www.sctp.org
> 
> under the RFC/internet draft tab  you will find at the
> bottom a rfc-chgbar/diff that highlights the few
> changes..
> 
> Please let me know if I got everyones comments.. I would like
> to get this in so that we can move the document forward..
> 
> Thanks
> 
> R
> ---
> 
> Randall Stewart
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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Date: Tue, 05 Aug 2003 14:12:27 -0500
From: "Randall R. Stewart (home)" <randall@stewart.chicago.il.us>
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To: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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Subject: Re: [Tsvwg] PR-SCTP updated draft
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Qiaobing Xie wrote:

>Hi, Randy,
>
>Some minor editorial comments:
>
>1. The new implementation note:
>
>"      IMPLEMENTATION NOTE: An implementation may wish to limit the
>       number of duplicate FORWARD TSN'S sent by either sending a
>       duplicate FORWARD TSN every other SACK or waiting a full RTT
>       before sending a duplicate FORWARD TSN.
>"
>
>may read better in the following form:
>
>"      IMPLEMENTATION NOTE: An implementation may wish to limit the
>       number of duplicate FORWARD TSN chunks it sends by either only
>       sending a duplicate FORWARD TSN every other SACK or waiting for
>       a full RTT before sending a duplicate FORWARD TSN.
>"
>

Sound good :>

>
>2. The use of "forward" and "FORWARD" is inconsistent in a couple of
>places.
>
> - page 12, in para C4)
>
>  "number in the forwarded TSN. ..."
>
>  should be
>
>  "number in the FORWARD TSN. ..."
>
> - page 13, in para F2)
>
>  "for generating a forward TSN to be ..."
>
>  should be 
>
>  "for generating a FORWARD TSN to be ..."
>
>I am referring to the old page numbers in -00 here but they should be
>close or the same in the -01 version you are working on.
>

Since its XML I don't have page numbers :> but thats ok.. I found them
all :>>

Good catchs.. I will have put up an update with these changes on the
web site :>

R

>
>-Q
>
>"Randall R. Stewart (home)" wrote:
>  
>
>>Hi all:
>>
>>I think I have now gotten all of the minor changes in
>>to the WG last call on the PR-SCTP document. Please have a
>>look at the html diff at:
>>
>>http://www.sctp.org
>>
>>under the RFC/internet draft tab  you will find at the
>>bottom a rfc-chgbar/diff that highlights the few
>>changes..
>>
>>Please let me know if I got everyones comments.. I would like
>>to get this in so that we can move the document forward..
>>
>>Thanks
>>
>>R
>>---
>>
>>Randall Stewart
>>
>>_______________________________________________
>>tsvwg mailing list
>>tsvwg@ietf.org
>>https://www1.ietf.org/mailman/listinfo/tsvwg
>>    
>>
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg
>
>
>  
>



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Randall R. Stewart (home) wrote:

> Slowly catching up on a HUGE backlog of email :-o
> 
> As to the above.. I don't agree yet.. Armando has proposed such  to me
> and I have yet to be convinced....
> 
> It may be that it is better in some cases.. but when you start factoring in
> real failures of a NIC you may find that your failure detection starts to
> be effected in a adverse way. So far I think Armando has a good point
> in that possibly FR's may be best NOT to send to an alternate .. but
> T3's I think may still be best sent to an alternate...

If I remember correctly, the fast retransmit case was discussed
some time ago in either tsvwg or sctp-impl.  The consensus is to
retransmit to the original address.  I thought every implementations
is already doing this...

I guess NIC failure happens very infrequently when compared to
congestion and spurious timeout.  I don't think it is a good
suggestion to design the timeout retransmission around an uncommon
case.  And it seems to me that the correct way to handle a NIC
failure is to use ASCONF to tell the peer to either remove those IP
addresses associated with the failed NIC or to change the primary
path if it is to the failed NIC.  One may ask why such mechanism
is not included in the original spec...




-- 

						K. Poon.
						kacheong.poon@sun.com



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Kacheong Poon wrote:
...
> I guess NIC failure happens very infrequently when compared to
> congestion and spurious timeout.  ...

It depends :-) For example, congestion is only a remote possibility in
many (if not all) commercial telecomm networks which is THE primary use
area for SCTP at present time (e.g., SIGTRAN, 3GPP). This is due to the
over-provisioning of bandwidth in all such networks. However, in those
types of networks, one of the key system requirements is the five 9's
system availability (99.999% availability). The only way to achieve five
9's is to use fault tolerant design including redundant NICs and
cabling. But those extra cost spent on redundant hardware will only be a
waste if fault recovery is too slow. In such cases, retrans to
alternative is very effective in speeding up fault recover.

regards,
-Qiaobing

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Qiaobing Xie wrote:

> It depends :-) For example, congestion is only a remote possibility in
> many (if not all) commercial telecomm networks which is THE primary use
> area for SCTP at present time (e.g., SIGTRAN, 3GPP). This is due to the
> over-provisioning of bandwidth in all such networks. However, in those
> types of networks, one of the key system requirements is the five 9's
> system availability (99.999% availability). The only way to achieve five
> 9's is to use fault tolerant design including redundant NICs and
> cabling. But those extra cost spent on redundant hardware will only be a
> waste if fault recovery is too slow. In such cases, retrans to
> alternative is very effective in speeding up fault recover.

Then in this specific environment (well, SCTP is supposed to
be a general purpose transport protocol afterall), I'd suggest
those SCTP stacks to support ASCONF so that the retransmission
can automatically be made to an alternate address.  This seems
to be a better way to handle it than relying on a "guess" of NIC
failure as there is no need to guess at all.

Is this a problem to have such requirement on those stacks
operating in the telecomm world?  My guess (we are all guessing (-:)
is not...

-- 

						K. Poon.
						kacheong.poon@sun.com



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Kacheong Poon wrote:
> 
> Qiaobing Xie wrote:
> 
> > It depends :-) For example, congestion is only a remote possibility in
> > many (if not all) commercial telecomm networks which is THE primary use
> > area for SCTP at present time (e.g., SIGTRAN, 3GPP). This is due to the
> > over-provisioning of bandwidth in all such networks. However, in those
> > types of networks, one of the key system requirements is the five 9's
> > system availability (99.999% availability). The only way to achieve five
> > 9's is to use fault tolerant design including redundant NICs and
> > cabling. But those extra cost spent on redundant hardware will only be a
> > waste if fault recovery is too slow. In such cases, retrans to
> > alternative is very effective in speeding up fault recover.
> 
> Then in this specific environment (well, SCTP is supposed to
> be a general purpose transport protocol afterall), I'd suggest
> those SCTP stacks to support ASCONF so that the retransmission
> can automatically be made to an alternate address.  This seems
> to be a better way to handle it than relying on a "guess" of NIC
> failure as there is no need to guess at all.
> 
> Is this a problem to have such requirement on those stacks
> operating in the telecomm world?  My guess (we are all guessing (-:)
> is not...

K.

Are we **absolutely** sure that whatever we intended to solve in this
discussion has to include giving up this "retrans to alternative"
behavior which is so key to telecomm folks? Is there a solution that
will let those folks to co-exist in this SCTP world while we are
perfecting its congestion control here?

I am not ready to wave the white flag yet. I'd encourage the CC
perfectionists to continue work and find a solution that will satisfy
the CC behaviors while not sacrificing a very real advantage of SCTP for
telecom folks. Then we would have a WIN-WIN solution and I think we can
do it! (I know I sound a little like a promotional speaker, my apology
:)

regards,
-Qiaobing

> 
> --
> 
>                                                 K. Poon.
>                                                 kacheong.poon@sun.com
> 
> _______________________________________________
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Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
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Kacheong Poon wrote:

> ...I'd suggest
> those SCTP stacks to support ASCONF so that the retransmission
> can automatically be made to an alternate address.  This seems
> to be a better way to handle it than relying on a "guess" of NIC
> failure as there is no need to guess at all.

So you want to rely on an external monitor/detector mechanism at the
peer endpoint that will tell the peer SCTP stack a NIC is bad and then
the remote stack will do a "change primary addr" through ASCONF so that
the local endpoint will start to move the traffic to an alternative dest
addr? Ok, now you have no need to guess since the remote external
detector is doing the "guessing" :-)

But this would essentially remove most of the SCTP's auto path failure
detection / recovery capabilities and take out a big chunk of SCTP's
multi-homing advantage. I think this is too costly for whatever we are
trying to gain here.

regards,
-Qiaobing

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Both Randy and Caitlin's comments below are indeed very interesting. I
wonder if we still have consensus that SCTP is still an end-to-end, L3
transport protocol. If so, I believe the reason we call it end-to-end is
because we assume there is a virtual, structure-less, vairable
bandwidth, end-to-end pipe underneath the L3 transport. 

Now, I start to see people talking about "what if the last hop of the
pipe [between two SCTP ends] is of X or Y nature". To me, at layer 3 I
don't see any internal structure of the L2 pipe execpt of the assumption
that it is an end-to-end pipe. So where is that "last hop" people are
referring to?

If this virtual pipe abstraction no longer holds, then we are in a new
era of layering principle. If it is still valid, then the "last hop" is
invisible to SCTP and needs to remain so. And the nature of the "last
hop" needs to be addressed in L2 groups (e.g., trigtran).

In short, I fail to see how we could succeed by tweaking L3 SCTP to
address some L2 "last hop" characteristics.

regards,
-Qiaobing

"Randall R. Stewart (home)" wrote:
> 
> Caitlin Bestler wrote:
> 
> >
> > On Monday, July 21, 2003, at 09:35 AM, Armando L. Caro Jr. wrote:
> >
> >>>
> >>
> >> Well, your one bit solution only works if the sender follows the
> >> recommended mechanism for retransmissions. There hasn't been a consensus
> >> that retransmitting on an alternate path is always the best policy. In
> >> fact, I would argue that it isn't. I believe that for better performance
> >> without affecting failover performance, fast retransmits should go to
> >> the
> >> same destination and timeout retransmits should go to an alternate
> >> destination [1].  With such a retransmission policy, one bit would
> >> not be enough. Sure we could use 2 bits, but timestamps just seems
> >> cleaner.
> >>
> >> [1] http://www.cis.udel.edu/~acaro/papers/2003.icon.acaro.pdf
> >>
> >> ~armando
> >>
> >
> > In the paper cited, you fault the default SCTP behavior for assuming
> > "that loss indicates either that the destination address used is
> > unreachable,
> > or the destination's network path is congested."
> >
> > The implication is that the end-to-end algorithm is "giving up too
> > quickly"
> > on a path, and thereby degrading performance.
> >
> > As that I have seen this objection most often raised when the final hop
> > is a mobile and/or wireless link, I am tempted to turn that question
> > around.
> >
> > Isn't the real problem that the last hop is giving up on L2 delivery too
> > quickly, and creating an L3/end-to-end problem for something that
> > should have been handled locally?
> >
> > Basically, is each downstream frequency *really* a path? Or should
> > the collective set of frequencies be viewed as a single L3 path
> > over which there may be L2 retries?
> >
> > I designed a two-way CATV system to carry IP traffic through
> > set-top decoders while I was at Zenith Electronics. Whenever
> > the set-top needed to change frequency it would be deaf to
> > the headend for a short period. Realizing that dropping
> > packets during such a frequency shift would lead to very
> > poor end-to-end performance, we simply scheduled the
> > packet to be L2-delivered on a frequency as soon as it
> > was known, or on all frequencies if the receiver's status
> > was not known.
> >
> > We certainly do not want to begin an L2 design debate
> > here, but I think it is a valid question -- is it a valid assumption
> > that L2 delivery should be dealing with truly transitory
> > problems and therefore packet loss really should be
> > taken to as an indication that the destination address used
> > is unreachable or the destination's network path is
> > congested?
> 
> Caitlin:
> 
> As always an interesting comment from you..( yes I have a lot of
> other unread email that you have sent to me .. I will get to it I
> promise :>)
> 
> Now I understand the logic here and I have two comments...
> 
> a) Being able to seperate a L2 drop for NON congestion reasons from the
>     "default" assumption would be a nice thing.. I actually will have a
> proposal
>     to put forward on this later on when I can get the cycles to clean
> up the drafty
>     draft I have in the works somewhere :>
> 
> b) If your example L2 environment just delivered the packet late (as I
> assume it
>     must have) then what you have is a spurious retransmission.. the
> very thing
>     that Eifel is trying to detect...
> 
> Just a thought or two :>
> 
> R
> 
> >
> >
> > Caitlin Bestler - cait@asomi.com - http://asomi.com/
> >
> >
> > _______________________________________________
> > tsvwg mailing list
> > tsvwg@ietf.org
> > https://www1.ietf.org/mailman/listinfo/tsvwg
> >
> >
> 
> _______________________________________________
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On Tuesday, August 5, 2003, at 06:14 PM, Qiaobing Xie wrote:

> In short, I fail to see how we could succeed by tweaking L3 SCTP to
> address some L2 "last hop" characteristics.

It is really a matter of what the best interpretation of
the "end to end" assumption is.

Most of what has been written recently emphasizes that the
L2 layer should not attempt to solve L3 problems.

But it strikes me that you can go too far in the opposite
direction. When a given L2 hop is significantly more likely
to have bursty transmission errors than the Internet as
a whole, I really think we need to question whether those
problems should be solved on and end-to-end basis.

That is especially so if there is a suggestion that the
end-to-end algorithms need to adjust because they react
incorrectly to bursty noise on one hop.

If a typical path does only shows frequent drops when
there are failures and/or congestion, then SCTP assuming
a shift in path remains the correct strategy.



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From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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To: Caitlin Bestler <cait@asomi.com>
CC: "Randall R. Stewart (home)" <randall@stewart.chicago.il.us>,
        "Armando L. Caro Jr." <acaro@acm.org>, tsvwg@ietf.org
Subject: Re: [Tsvwg] Meaning of path failure (was re SCTP Timestamps)
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Caitlin Bestler wrote:
...
> But it strikes me that you can go too far in the opposite
> direction. When a given L2 hop is significantly more likely
> to have bursty transmission errors than the Internet as
> a whole, I really think we need to question whether those
> problems should be solved on and end-to-end basis.
> 
> That is especially so if there is a suggestion that the
> end-to-end algorithms need to adjust because they react
> incorrectly to bursty noise on one hop.

I *think* we are in agreement here :-) 

SCTP CC (as well that of TCP) is designed as an end-to-end mechanism.
Therefore, I'd think any tweak on this end-to-end mechanism will by
definition have an end-to-end impact through the entire pipe. So, if we
can really devise a particular tweak that is "just right" to address the
particular noise/burst pattern of one hop in the pipe, in all the
likelihood, this tweak will be suboptimal or even adversely impact the
other hops in the pipe. And this would not be good.

regards,
-Qiaobing

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From: Andrei Gurtov <gurtov@cs.helsinki.fi>
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>
>
>First, I am not sure I am willing to believe that SACK is not pretty
>well deployed.  3 years ago I reported 40% of the web clients that
>hit our web server here at NASA were using SACK (and the trend was
>growing steadily) [All00].  I don't have good data from that server
>right now, but I took a quick look at a server at BBN that I have
>been monitoring for the last ~11 days.  In that time I have seen
>6545 web connections (yeah, not terribly heavy traffic) and 6183 of
>the remote hosts (94%) advertised SACK.  That seems like pretty good
>deployment to me.
>
I don't claim to know the answer, but what data on Sally's page tells is 
that although many SYNs have a SACK-perm option only few SYN-ACKs have 
it. It may mean that either web servers are using some old OS that do 
not support SACK or that they don't want to bother with extra overhead 
of keeping SACK state.

If SACK is universally deployed why would we be updating NewReno RFC?

>I am not sure what that says about additional "bugfix" tricks.  Do
>we need them?  Do we not need them?  The above data is not
>necessarily conclusive or representative.  The changes would add
>complexity to a TCP implementation.  On the other hand, not that
>much complexity and maybe would provide a decent enough win.  But,
>when do we stop patching known suboptimal recovery?  Etc., Etc.,
>Etc.    ??????
>  
>
I'll send some scenarios next week that might motivate the need for it. 
I think that since implementing them is not a big deal and the NewReno 
draft anyway discusses the issue
it might be a good place to mention them. In the current ns2 snapshot

test-all-tcpOptions timeouts_newreno (shows DUPACKs due to a lost packet)
test-all-newreno reno2 (shows DUPACKs due to duplicate segments )

>>ACK-heuristic: if seq(ack_new)-seq(ack_prev)<=2 MSS then trigger
>>fast retransmit on 3rd DUPACK This method is based on observation
>>that to trigger a false fast retransmit, a TCP sender has to
>>retransmit unnecessarily at least 3 adjucent packets. In this case
>>there will be a jump in a cumulative ACK at the sender of at least
>>three segments.
>>    
>>
>
>I am trying to wrap my head around this and cannot yet do so.  Can
>you define "ack_new" and "ack_prev"?  Are these the two ACKs before
>duplicate ACKs start rolling in?
>
ack_new has the same cum ACK as DUPACKs, ack_prev is the previous ACK before
DUPACKs

>Timestamps are less prevalent (by default) than SACK is, I think
>(11% of the connection in the BBN trace described above, which is in
>the ballpark of what was reported in [All00], too).  So, I am not
>sure I think this is a practical approach.
>
Well, we did publish Eifel based on timestamps. I wouldn't be surprised 
if timeouts due to bugfix are far more common than spurious timeouts in 
the Internet.

Andrei


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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Qiaobing Xie <Qiaobing.Xie@motorola.com>
Cc: Kacheong Poon <kacheong.poon@sun.com>, tsvwg@ietf.org
Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
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Qiaobing,

I agree.  And please do not forget that SCTP was first designed to
meet a set of requirements for transporting telephony signalling
protocols.  If those requirements fail to be met by SCTP, yet another
transport protocol will become necessary again.

--brian

On Tue, 05 Aug 2003, Qiaobing Xie wrote:

> Kacheong Poon wrote:
>=20
> > ...I'd suggest
> > those SCTP stacks to support ASCONF so that the retransmission
> > can automatically be made to an alternate address.  This seems
> > to be a better way to handle it than relying on a "guess" of NIC
> > failure as there is no need to guess at all.
>=20
> So you want to rely on an external monitor/detector mechanism at the
> peer endpoint that will tell the peer SCTP stack a NIC is bad and then
> the remote stack will do a "change primary addr" through ASCONF so that
> the local endpoint will start to move the traffic to an alternative des=
t
> addr? Ok, now you have no need to guess since the remote external
> detector is doing the "guessing" :-)
>=20
> But this would essentially remove most of the SCTP's auto path failure
> detection / recovery capabilities and take out a big chunk of SCTP's
> multi-homing advantage. I think this is too costly for whatever we are
> trying to gain here.
>=20
> regards,
> -Qiaobing
>=20
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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=2D----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On Tuesday 05 August 2003 19:00, Randall R. Stewart (home) wrote:
> Hi all:
>
> I think I have now gotten all of the minor changes in
> to the WG last call on the PR-SCTP document. Please have a
> look at the html diff at:
>
> http://www.sctp.org
>

Hi Randy,

Rule F5 now states:
=2D -----------------------------------------------------------------------=
=2D---
=465) If a decision to "abandon" a chunk is made, no matter how such
decision is made, the appropriate congestion adjustment MUST be
made as specified in RFC2960 if the chunk later would have been
marked for retransmission (e.g. T3-Timeout or Fast Retransmit).
=2D -----------------------------------------------------------------------=
=2D---
That is a little bit difficult to understand, I think. Maybe:
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
=465) If the decision to "abandon" a chunk is made, no matter how such a
decision is made, the appropriate congestion adjustment MUST be
made as specified in RFC2960 if the chunk would have been marked=20
for retransmission later (e.g. either by T3-Timeout or by Fast=20
Retransmit).
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
works better ?

Regards,
Andreas

=2D --=20
Dipl.-Ing. Andreas Jungmaier             =20
Computer Networking Technology Group    =20
University of Duisburg-Essen                      =20
http://www.exp-math.uni-essen.de/~ajung  =20
My PGP Key-ID: D3824AC0            =20

My PGP Key-Fingerprint: 228C 7C2C 3381 2DD4  9998 2812 5C0B 0B04  D382 4AC0
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Subject: Re: [Tsvwg] PR-SCTP updated draft
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Andreas Jungmaier wrote:

>-----BEGIN PGP SIGNED MESSAGE-----
>Hash: SHA1
>
>On Tuesday 05 August 2003 19:00, Randall R. Stewart (home) wrote:
>  
>
>>Hi all:
>>
>>I think I have now gotten all of the minor changes in
>>to the WG last call on the PR-SCTP document. Please have a
>>look at the html diff at:
>>
>>http://www.sctp.org
>>
>>    
>>
>
>Hi Randy,
>
>Rule F5 now states:
>- ---------------------------------------------------------------------------
>F5) If a decision to "abandon" a chunk is made, no matter how such
>decision is made, the appropriate congestion adjustment MUST be
>made as specified in RFC2960 if the chunk later would have been
>marked for retransmission (e.g. T3-Timeout or Fast Retransmit).
>- ---------------------------------------------------------------------------
>That is a little bit difficult to understand, I think. Maybe:
>===========================================================================
>F5) If the decision to "abandon" a chunk is made, no matter how such a
>decision is made, the appropriate congestion adjustment MUST be
>made as specified in RFC2960 if the chunk would have been marked 
>for retransmission later (e.g. either by T3-Timeout or by Fast 
>Retransmit).
>===========================================================================
>works better ?
>  
>
Sounds fine to me :>

I will update it and put another version of the diff up on
my web site..

R

>Regards,
>Andreas
>
>- -- 
>Dipl.-Ing. Andreas Jungmaier              
>Computer Networking Technology Group     
>University of Duisburg-Essen                       
>http://www.exp-math.uni-essen.de/~ajung   
>My PGP Key-ID: D3824AC0             
>
>My PGP Key-Fingerprint: 228C 7C2C 3381 2DD4  9998 2812 5C0B 0B04  D382 4AC0
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To: Qiaobing Xie <Qiaobing.Xie@motorola.com>
CC: Caitlin Bestler <cait@asomi.com>, "Armando L. Caro Jr." <acaro@acm.org>,
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Subject: Re: [Tsvwg] Meaning of path failure (was re SCTP Timestamps)
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Qiaobing:

Yes, I do agree with you... but then do you think
that a router sending a ICMP message that the transport
protocol uses to determine path mtu is bad?

I look at it not as a LAST hop .. but that at ANY point in the
path a router can make a decision to either
1) drop a packet due to congestion
or
2) drop a packet due to corruption

In case <1> we WANT the transport protocol to react
to it and reduce its sending rate. But in case <2> we would
prefer the transport protcol become aware of this drop and to
retransmit it (if retransmission is so desired) WITHOUT reducing
its congestion rate... This in theory could be ANY hop along a
routing chain.. but in most cases would be the last or next to
last hop.. I am thinking satellite transmission for next to last...

Now the problem as always been with reporting a corruption drop that

A) A mallicious receiver could figure out that a TSN is missing and generate
    such a report to get the sender to avoid <1>.
B) A new form of attack could be created that lets a hacker spew forth 
bunches
    of these things to "slow" down everyone else talking to a web server so
   he/she gets better service..

So how can a transport protocol trust the requests?

SCTP of course can help in that if the router has the packet that
it is deciding to drop  it can return it .. and the vtag can be checked
this helps in situation <B> but what about <A>...

The packet corruption may be at a point that the sender can't
verify it? This is true.. and if so it should be tossed...

The new technology we have created at cisco however is a tunneling
software that does some retransmission.. and will give up on
that retransmission think of it as an RLP.. but when it does
give up .. its capable of sending the packet back to the sender..

I will get a draft together on this soon... I am just short of cycles right
now...

I have this up and running and its simulation results for working with
satellites is remarkable.. you can get speeds that normaly take the
disruptive PEP arpproach.. aka where you terminate the TCP/SCTP
at both sides of the satellite to "fool" the end to end protocol into
not realizing the long bandwidth delay by-product.. very ugly...

When I present my draft I will also include some performance
work I have gatherede too...

R

Qiaobing Xie wrote:

>Both Randy and Caitlin's comments below are indeed very interesting. I
>wonder if we still have consensus that SCTP is still an end-to-end, L3
>transport protocol. If so, I believe the reason we call it end-to-end is
>because we assume there is a virtual, structure-less, vairable
>bandwidth, end-to-end pipe underneath the L3 transport. 
>
>Now, I start to see people talking about "what if the last hop of the
>pipe [between two SCTP ends] is of X or Y nature". To me, at layer 3 I
>don't see any internal structure of the L2 pipe execpt of the assumption
>that it is an end-to-end pipe. So where is that "last hop" people are
>referring to?
>
>If this virtual pipe abstraction no longer holds, then we are in a new
>era of layering principle. If it is still valid, then the "last hop" is
>invisible to SCTP and needs to remain so. And the nature of the "last
>hop" needs to be addressed in L2 groups (e.g., trigtran).
>
>In short, I fail to see how we could succeed by tweaking L3 SCTP to
>address some L2 "last hop" characteristics.
>
>regards,
>-Qiaobing
>
>"Randall R. Stewart (home)" wrote:
>  
>
>>Caitlin Bestler wrote:
>>
>>    
>>
>>>On Monday, July 21, 2003, at 09:35 AM, Armando L. Caro Jr. wrote:
>>>
>>>      
>>>
>>>>Well, your one bit solution only works if the sender follows the
>>>>recommended mechanism for retransmissions. There hasn't been a consensus
>>>>that retransmitting on an alternate path is always the best policy. In
>>>>fact, I would argue that it isn't. I believe that for better performance
>>>>without affecting failover performance, fast retransmits should go to
>>>>the
>>>>same destination and timeout retransmits should go to an alternate
>>>>destination [1].  With such a retransmission policy, one bit would
>>>>not be enough. Sure we could use 2 bits, but timestamps just seems
>>>>cleaner.
>>>>
>>>>[1] http://www.cis.udel.edu/~acaro/papers/2003.icon.acaro.pdf
>>>>
>>>>~armando
>>>>
>>>>        
>>>>
>>>In the paper cited, you fault the default SCTP behavior for assuming
>>>"that loss indicates either that the destination address used is
>>>unreachable,
>>>or the destination's network path is congested."
>>>
>>>The implication is that the end-to-end algorithm is "giving up too
>>>quickly"
>>>on a path, and thereby degrading performance.
>>>
>>>As that I have seen this objection most often raised when the final hop
>>>is a mobile and/or wireless link, I am tempted to turn that question
>>>around.
>>>
>>>Isn't the real problem that the last hop is giving up on L2 delivery too
>>>quickly, and creating an L3/end-to-end problem for something that
>>>should have been handled locally?
>>>
>>>Basically, is each downstream frequency *really* a path? Or should
>>>the collective set of frequencies be viewed as a single L3 path
>>>over which there may be L2 retries?
>>>
>>>I designed a two-way CATV system to carry IP traffic through
>>>set-top decoders while I was at Zenith Electronics. Whenever
>>>the set-top needed to change frequency it would be deaf to
>>>the headend for a short period. Realizing that dropping
>>>packets during such a frequency shift would lead to very
>>>poor end-to-end performance, we simply scheduled the
>>>packet to be L2-delivered on a frequency as soon as it
>>>was known, or on all frequencies if the receiver's status
>>>was not known.
>>>
>>>We certainly do not want to begin an L2 design debate
>>>here, but I think it is a valid question -- is it a valid assumption
>>>that L2 delivery should be dealing with truly transitory
>>>problems and therefore packet loss really should be
>>>taken to as an indication that the destination address used
>>>is unreachable or the destination's network path is
>>>congested?
>>>      
>>>
>>Caitlin:
>>
>>As always an interesting comment from you..( yes I have a lot of
>>other unread email that you have sent to me .. I will get to it I
>>promise :>)
>>
>>Now I understand the logic here and I have two comments...
>>
>>a) Being able to seperate a L2 drop for NON congestion reasons from the
>>    "default" assumption would be a nice thing.. I actually will have a
>>proposal
>>    to put forward on this later on when I can get the cycles to clean
>>up the drafty
>>    draft I have in the works somewhere :>
>>
>>b) If your example L2 environment just delivered the packet late (as I
>>assume it
>>    must have) then what you have is a spurious retransmission.. the
>>very thing
>>    that Eifel is trying to detect...
>>
>>Just a thought or two :>
>>
>>R
>>
>>    
>>
>>>Caitlin Bestler - cait@asomi.com - http://asomi.com/
>>>
>>>
>>>_______________________________________________
>>>tsvwg mailing list
>>>tsvwg@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/tsvwg
>>>
>>>
>>>      
>>>
>>_______________________________________________
>>tsvwg mailing list
>>tsvwg@ietf.org
>>https://www1.ietf.org/mailman/listinfo/tsvwg
>>    
>>
>
>_______________________________________________
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>
>
>  
>



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Subject: Re: [Tsvwg] Meaning of path failure (was re SCTP Timestamps)
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Dear Caitlin,

Re:

> Most of what has been written recently emphasizes that the
> L2 layer should not attempt to solve L3 problems.
>
> But it strikes me that you can go too far in the opposite
> direction. When a given L2 hop is significantly more likely
> to have bursty transmission errors than the Internet as
> a whole, I really think we need to question whether those
> problems should be solved on and end-to-end basis.
>
> That is especially so if there is a suggestion that the
> end-to-end algorithms need to adjust because they react
> incorrectly to bursty noise on one hop.

I happen to agree (this was where TRIGTRAN ended up,
oriented to first-hop wireless links because that's the link
where the problem was for users in my part of the Internet).

Having already agreed, "but..."

One problem we ran into was, what happens if you're solving
a specific problem and somebody morphs the topology
underneath you?

In TRIGTRAN, the focus was on wireless WAN links, but
WLAN links are so cheap now that they're built into some
computers by default, so the problem changed from RTO
backoff when there's no coverage to path change when
there's dual coverage.

In TRIGTRAN, we headed off toward "last-hop" link
problems just in time for people to start putting networks
at the other end of the "last-hop" link, so it wasn't a last-hop
link any more.

The pressure was on us to beat end-to-end mechanisms,
without negative impacts even in situations we hadn't
anticipated. This isn't impossible, but it IS a tough sell.

In the SCTP case, what happens if there are TWO
problematic links on the path? You have to make sure
that the two independent repair processes aren't working
at cross purposes and will finish repairs at least as quickly
as end-to-end mechanisms would.

This isn't impossible, but it IS a tough sell.

Spencer

----- Original Message ----- 
From: "Caitlin Bestler" <cait@asomi.com>
To: "Qiaobing Xie" <Qiaobing.Xie@motorola.com>
Cc: "Randall R. Stewart (home)" <randall@stewart.chicago.il.us>;
"Armando L. Caro Jr." <acaro@acm.org>; <tsvwg@ietf.org>
Sent: Tuesday, August 05, 2003 6:28 PM
Subject: Re: [Tsvwg] Meaning of path failure (was re SCTP Timestamps)


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Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
References: <Pine.LNX.4.44.0307221304550.3045-100000@melkki.cs.Helsinki.FI> <3F1D8B89.40502@sun.com> <3F2E4F3A.7020500@stewart.chicago.il.us> <3F3013CB.3020505@sun.com>
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Kacheong Poon wrote:

> Randall R. Stewart (home) wrote:
>
>> Slowly catching up on a HUGE backlog of email :-o
>>
>> As to the above.. I don't agree yet.. Armando has proposed such  to me
>> and I have yet to be convinced....
>>
>> It may be that it is better in some cases.. but when you start 
>> factoring in
>> real failures of a NIC you may find that your failure detection 
>> starts to
>> be effected in a adverse way. So far I think Armando has a good point
>> in that possibly FR's may be best NOT to send to an alternate .. but
>> T3's I think may still be best sent to an alternate...
>
>
> If I remember correctly, the fast retransmit case was discussed
> some time ago in either tsvwg or sctp-impl.  The consensus is to
> retransmit to the original address.  I thought every implementations
> is already doing this... 

Wow.. I must have missed that discussion or forgotten it... Armando
and I have been haggling on this point for a while..

I think we now have definite proof that for FR's you should send to
the original address. For Time-outs however, this is not necessarily 
true. Armando
still has some more simulation to do in this vein and I have also asked him
to take into account the failover i.e. the cases where where you are
transmitting as failed.

I don't see a CC problem in all of this.. at least none that I can find. 
If you
do take a false T-3 you MAY end up retransmitting to an alternate to quickly
and when the ACK's come back credit the alternate... but is this really 
going
to happen? I think most of the times it will NOT..
aka:

Primary------cwnd=N-------------->
Alternate----cwnd=2Mtu---------->

Now assuming N is some larger number (we have been sending a while)... 
and we do
a encounter a delay spike that causes a false T3 on Primary..

You retransmit 1 MTU to Alternate and wait for a SACK.

The SACK arrives (just a bit latter) from your false T3. You then do for the
alternate

If (flight-size > cwnd) {
    if(in-ss)
         cwnd += min(1 mtu, ack-size)
    else
        pba += acksize;
        if(pba > cwnd)
               cwnd += 1 MTU
               pba -= cwnd
               if(pba <=0)
                   pba = 0
}

Now in our example guess what? flight-size = 1MTU (assuming you had
a full MTU or more to retran)... so the "if" would never be entered and
NO cwnd advance would happen.


It is hard for me to imagine a situation where you would really get the
alternate above 1MTU in flight size .... so I doubt there is a problem.. 
in fact
the real issue is the false T3 will have caused a collapse of the
primary... I guess this is what Eifel attempts to fix in TCP....

So

1) I see no CC problem with the current scheme
2) We probably need to NOT retransmit FR's to an alternate
3) It is best, I think, if we schedule a HB on the alternate after
    a retransmission.. Armando is still working on this point but
    if you have any loss on the alternate you need some way of
    collapsing the RTO so scheduling a HB can do that to help
    performance....

I am anxious for Armando to get a bit further in his research and I think
this will shed some additional light on the situation :>

>
>
> I guess NIC failure happens very infrequently when compared to
> congestion and spurious timeout.  I don't think it is a good
> suggestion to design the timeout retransmission around an uncommon
> case.  And it seems to me that the correct way to handle a NIC
> failure is to use ASCONF to tell the peer to either remove those IP
> addresses associated with the failed NIC or to change the primary
> path if it is to the failed NIC.  One may ask why such mechanism
> is not included in the original spec...


Well, we DID want both PR-SCTP and ADD-IP in the base spec... but in order
to not hold things up to long we decided to do them as seperate drafts.. 
and it
is a good thing too.. ADD-IP still needs some more work (in the security 
front) and
PR-SCTP is now 3 years later almost done...

Many implementations do have ASCONF working.. which can help in this
situation... but one would need (as Qiaobing pointed out) some sort of 
L2 trigger
to tell the peer "hey your nic card X failed" so that it could "know" to 
send the
"set-primary" to the alternate. And of course a "generic" trigger, as 
yet, does not
exist. Another area ripe for research...

R

>
>
>
>



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[snip]
> I don't see a CC problem in all of this.. at least none that I can find.
> If you
> do take a false T-3 you MAY end up retransmitting to an alternate to quickly
> and when the ACK's come back credit the alternate... but is this really
> going
> to happen? I think most of the times it will NOT..
> aka:
>
> Primary------cwnd=N-------------->
> Alternate----cwnd=2Mtu---------->
>
> Now assuming N is some larger number (we have been sending a while)...
> and we do
> a encounter a delay spike that causes a false T3 on Primary..
>
> You retransmit 1 MTU to Alternate and wait for a SACK.
>
> The SACK arrives (just a bit latter) from your false T3. You then do for the
> alternate
>
> If (flight-size > cwnd) {
>     if(in-ss)
>          cwnd += min(1 mtu, ack-size)
>     else
>         pba += acksize;
>         if(pba > cwnd)
>                cwnd += 1 MTU
>                pba -= cwnd
>                if(pba <=0)
>                    pba = 0
> }
>
> Now in our example guess what? flight-size = 1MTU (assuming you had
> a full MTU or more to retran)... so the "if" would never be entered and
> NO cwnd advance would happen.

Randall,

What you just described seems to me like a bug rather than a
"feature" (Leave alone the case of spurious timeout for now). In the
example you gave, the SCTP sender would probably need N roundtrips to recover
after a T3 and retransmission starts
off on the alternate (where N packets were in flight (consider all lost)
before you hit a T3 on the primary). This seems an over conservative
(read slow) behavior.

I must be missing something here :-O


-Sourabh



>
>
> It is hard for me to imagine a situation where you would really get the
> alternate above 1MTU in flight size .... so I doubt there is a problem..
> in fact
> the real issue is the false T3 will have caused a collapse of the
> primary... I guess this is what Eifel attempts to fix in TCP....
>
> So
>
> 1) I see no CC problem with the current scheme
> 2) We probably need to NOT retransmit FR's to an alternate
> 3) It is best, I think, if we schedule a HB on the alternate after
>     a retransmission.. Armando is still working on this point but
>     if you have any loss on the alternate you need some way of
>     collapsing the RTO so scheduling a HB can do that to help
>     performance....
>
> I am anxious for Armando to get a bit further in his research and I think
> this will shed some additional light on the situation :>
>


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Sourabh Ladha wrote:

>[snip]
>  
>
>>I don't see a CC problem in all of this.. at least none that I can find.
>>If you
>>do take a false T-3 you MAY end up retransmitting to an alternate to quickly
>>and when the ACK's come back credit the alternate... but is this really
>>going
>>to happen? I think most of the times it will NOT..
>>aka:
>>
>>Primary------cwnd=N-------------->
>>Alternate----cwnd=2Mtu---------->
>>
>>Now assuming N is some larger number (we have been sending a while)...
>>and we do
>>a encounter a delay spike that causes a false T3 on Primary..
>>
>>You retransmit 1 MTU to Alternate and wait for a SACK.
>>
>>The SACK arrives (just a bit latter) from your false T3. You then do for the
>>alternate
>>
>>If (flight-size > cwnd) {
>>    if(in-ss)
>>         cwnd += min(1 mtu, ack-size)
>>    else
>>        pba += acksize;
>>        if(pba > cwnd)
>>               cwnd += 1 MTU
>>               pba -= cwnd
>>               if(pba <=0)
>>                   pba = 0
>>}
>>
>>Now in our example guess what? flight-size = 1MTU (assuming you had
>>a full MTU or more to retran)... so the "if" would never be entered and
>>NO cwnd advance would happen.
>>    
>>
>
>Randall,
>
>What you just described seems to me like a bug rather than a
>"feature" (Leave alone the case of spurious timeout for now). In the
>example you gave, the SCTP sender would probably need N roundtrips to recover
>after a T3 and retransmission starts
>off on the alternate (where N packets were in flight (consider all lost)
>before you hit a T3 on the primary). This seems an over conservative
>(read slow) behavior.
>
>I must be missing something here :-O
>
>
>  
>
Consider for a moment a TCP sender....

At a T3-Timeout it sets cwnd = 1MTU
and then retransmits the lowest un-acked packet it can.

Then when the ack comes back.. it increases its cwnd and send
2 segments.. etc in slow start.

For SCTP the behavior is similar...

When a T3 is hit, you transmit only 1 MTU worth of data
to the alternate (if there is one) and sent the original destination
to 1MTU (marking all un-acked chunks for retransmission).

Now when the SACK comes back for the first retran-packet, you retransmit
up to 2 MTU's to the alternate... (lets assume for the moment you
had N in-flight chunks where  the size was N > 3 MTU)

Then you wait for a SACK... when it comes you increase the cwnd of
the alternate and send 3 MTUs of retranmssion...

In other words.. its NO DIFFERENT than TCP when a T3 hits.. even though
you are using an "alternate" path. This behavior was defined 
conservatively, yes, so
that we would in effect look the same as TCP... one really does not know
if the alternate path is diverse from the primary or not ....

Its no slower than TCP is today...

R



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[so now coming back to your example email]

> I don't see a CC problem in all of this.. at least none that I can find.
> If you
> do take a false T-3 you MAY end up retransmitting to an alternate to quickly
> and when the ACK's come back credit the alternate... but is this really
> going
> to happen? I think most of the times it will NOT..
> aka:
>
> Primary------cwnd=N-------------->
> Alternate----cwnd=2Mtu---------->
>
> Now assuming N is some larger number (we have been sending a while)...
> and we do
> a encounter a delay spike that causes a false T3 on Primary..
>
> You retransmit 1 MTU to Alternate and wait for a SACK.
>
> The SACK arrives (just a bit latter) from your false T3. You then do for the
> alternate
>
> If (flight-size > cwnd) {
>     if(in-ss)
>          cwnd += min(1 mtu, ack-size)
>     else
>         pba += acksize;
>         if(pba > cwnd)
>                cwnd += 1 MTU
>                pba -= cwnd
>                if(pba <=0)
>                    pba = 0
> }
>
> Now in our example guess what? flight-size = 1MTU (assuming you had
> a full MTU or more to retran)... so the "if" would never be entered and
> NO cwnd advance would happen.
>
>
> It is hard for me to imagine a situation where you would really get the
> alternate above 1MTU in flight size .... so I doubt there is a problem..

This generalization was confusing but your latter mail eats this up.

> in fact
> the real issue is the false T3 will have caused a collapse of the
> primary... I guess this is what Eifel attempts to fix in TCP....
> So
>
> 1) I see no CC problem with the current scheme

So the congestion window overgrowth in your example can  occur (and
thats a term I am using because eventually an SCTP sender will credit the
alternate's cwnd for the SACK of the original).

Taking your example where the primary faced a delay spike. When the SCTP
sender hits a T3, it will retransmit 1MTU worth to alternate. Looking at
the GoBackN, as the SACKs for the orginal now start coming in, the SCTP
sender eventually starts crediting the cwnd of the alternate and pumping
more retransmissions to the alternate. This is what I had refereed to as
congestion window overgrowth due to retransmission ambuiguity in a
previous mail. The issue was really that what if the alternate path was
congested/broken (not yet detected by HB). The SCTP sender would not
realize this while doing GoBackNs and pump retransmissions to the
alternate.

As you said Eifel's purpose was to undo the primary's state after this
spurious t/o, but it may also be able to avoid this congestion window
overgrowth in some cases (a +ve side effect).


-Sourabh



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Qiaobing Xie wrote:

> Are we **absolutely** sure that whatever we intended to solve in this
> discussion has to include giving up this "retrans to alternative"
> behavior which is so key to telecomm folks? Is there a solution that
> will let those folks to co-exist in this SCTP world while we are
> perfecting its congestion control here?

I don't understand why we are giving up retransmission to an
alternate address.  The simple change I suggested was to do
the first timeout retransmission to the original address.  SCTP
will still send the second timeout retransmission to an alternate
address.  This is a default behavior I suggested implementors to
use.  Having said that, it does not mean that an SCTP implementation
should always follow the default behavior.  It can just be a
simple switch or an option to change the behavior to send the
first timeout retransmission to an alternate address.

> I am not ready to wave the white flag yet. I'd encourage the CC
> perfectionists to continue work and find a solution that will satisfy
> the CC behaviors while not sacrificing a very real advantage of SCTP for
> telecom folks. Then we would have a WIN-WIN solution and I think we can
> do it! (I know I sound a little like a promotional speaker, my apology
> :)

What I try to say is that if SCTP is a general purpose transport
protocol, the default behavior should not be designed to fit in
a particular kind of environment.  I think you agree with that,
right (-:

[From another email]

 > So you want to rely on an external monitor/detector mechanism at the
 > peer endpoint that will tell the peer SCTP stack a NIC is bad and then
 > the remote stack will do a "change primary addr" through ASCONF so that
 > the local endpoint will start to move the traffic to an alternative dest
 > addr? Ok, now you have no need to guess since the remote external
 > detector is doing the "guessing" :-)

I don't understand.  If the NIC you are referring to above
belongs to the endpoint, why is an external monitor/detector
needed to tell the SCTP stack that the NIC is bad?  Isn't this
the job of the device driver?

 > But this would essentially remove most of the SCTP's auto path failure
 > detection / recovery capabilities and take out a big chunk of SCTP's
 > multi-homing advantage. I think this is too costly for whatever we are
 > trying to gain here.

Assuming we are referring to a NIC belonging to the endpoint where
the SCTP stack lives, I don't think using ASCONF will remove the
auto detection/recovery capabilities of SCTP.  The device driver
tells the IP stack that a NIC is bad, so the IP stack does something.
Isn't this kind of communication default in most OSes nowadays?  If
the stack also supports SCTP, then SCTP can use this info to do
something.  This is quite straightforward.

-- 

						K. Poon.
						kacheong.poon@sun.com



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Sourabh/Stephan:

I have been contemplating the timestamp issue and I am beginning to lean
toward using the already defined HB for this. Could you tell me if
this would work for your eifel algorithm?

First Transmission

----Normal DATA-no-HB------>

----Normal HB-To-Alternate(ts/other stuff)--->

T3-Timeout

---HB(ts/flag=retran#/tsn.. no other stuff)+Bundled-Data------->

If I hit another T3

---HB(ts/flag=retran#/tsn.. no other stuff)+Bundled-Data------>

Now. One of a number of things will occur.

1) The peer does NOT bundle things together... in that case a time-stamp 
chunk
     would have the same problems as a HB-ACK i.e. it would get lost in
     the network from say the SACK that will come back ack'ing the data.

2) A loss occurred and thus the HB-ACK will be sent right away possibly
    bundled with the SACK not showing any duplicates and showing gaps.

3) A loss did NOT occur and a HB-ACK will be bundled with a SACK showing
    a duplicate (if a duplicate is present a SACK is sent right away 
with no delay)

4) A loss occured and there are no gaps so the delay timer is started.. thus
    the HB-ACK would go stand alone.. and then a bit later the SACK would
    be sent based of the delay timer.

Now if the HB-ACK comes bundled with a SACK showing duplicates you have
detected a false retransmission.

In ANY event you can always calculate the RTT since the HB has when it
was sent.. .and if you are really nervous about the missing delay (200ms 
or whatever)
you can add it in .. of course ALL HB's to alternates have this issue as 
well :>

It seems to me that the HB/as long as bundling occurs with the HB-ACK will
accomplish the same purposes... Or what is missing from this detection?

Brian as pointed out that it "changes the nature of the HB" but I don't 
think
so.. since a HB is used to:

1) Test reachablility
and
2) calculate RTT's

And a bundled HB like this is doing both of those... its just that by when
you send it you get more information...

What am I missing?

R







R

---
Randall R. Stewart
randall@stewart.chicago.il.us 815-342-5222 (cell phone)




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Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
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Randy/Armando,

I don't know much detail and the scope of your ongoing
discussion/analysis on this issue. But one of my concerns is that if we
focus the analysis too closely to local behaviors of a single node or
association we may lose our sight to the bigger picture. 

Here is a scenario: a multi-homed SCTP association is sharing a big area
network (or just the Internet) with other TCP, UDP, SCTP, RTP traffic
and is actively passing data across the network. I understand there is
no mechanism in existence yet to guarantee path diversity for this
multi-homed SCTP association (maybe such a mechanism will be produced by
the multi-homing WG soon.. who knows). But there is nothing to prevent
path diversity from naturally occurring either. So, let's assume that
the association has path diversity across the network. 

Now at the SCTP sender suddenly gap SACKs or timeouts start to be
indicated on the primary dest address. This can mean either congestion
or path failure which we have no way to tell for sure. But it surely
indicates *some* problem with the part of the network that the primary
path crosses. Now the issue is how the sender should react. We seem to
be talking about the following options:

a) The "agile" approach - move retrans to the alternative at the first
sign of trouble; this is the current RFC2960 recommendation.

b) The "stubborn" approach - insisting on using the current addr for all
trans and retrans until nothing can get through; 

c) something in-between - e.g., if it is FR, use the same addr, if RTO,
use the alternative.

The main advantage of "stubborn" over "agile" is (to my understanding at
least) that the sender side CC algorithm is 1) less prone to generating
spurious retrans, which translates into less wasted network bandwidth,
and 2) no longer suffers from the ack ambiguity, which translates into
???... (I am still trying to understand the real benefit here).

The benefits of "agile" over "stubborn" are far clearer to me: 1) as I
mentioned in an earlier email, if the *some* problem is caused by an
actual fault in the primary path, the fault recovery speed of the
"agile" approach is much faster than the "stubborn". In fact, the
"stubborn" approach degrades SCTP to TCP equivalent in terms of path
failure detection latency. 2) if the *some* problem is congestion
induced (maybe some one started download huge movie files or a workhorse
router went down in that part of network), wouldn't it still be a very
good thing that the SCTP traffic is swiftly shifted away from the
congested area of the network (assuming path diversity exists here)?
This not only helps the congestion condition to improve quicker in the
affected area, but also lessens the impact on the data sender's
throughput. Imagine that if in the future a portion of the Internet
traffic can perform this type of "agile" multi-homing, how much it could
benefit the network congestion recovery.

IMHO, the *real* benefits of "agile" approach to both the application
and the network more than justify the occasional spurious retrans and
the whatever bad things caused by the ack ambiguity. I hope this wider
impact is factored into your simulation and analysis.

regards,
-Qiaobing

"Randall R. Stewart (home)" wrote:
> 
> Kacheong Poon wrote:
> 
> > Randall R. Stewart (home) wrote:
> >
> >> Slowly catching up on a HUGE backlog of email :-o
> >>
> >> As to the above.. I don't agree yet.. Armando has proposed such  to me
> >> and I have yet to be convinced....
> >>
> >> It may be that it is better in some cases.. but when you start
> >> factoring in
> >> real failures of a NIC you may find that your failure detection
> >> starts to
> >> be effected in a adverse way. So far I think Armando has a good point
> >> in that possibly FR's may be best NOT to send to an alternate .. but
> >> T3's I think may still be best sent to an alternate...
> >
> > If I remember correctly, the fast retransmit case was discussed
> > some time ago in either tsvwg or sctp-impl.  The consensus is to
> > retransmit to the original address.  I thought every implementations
> > is already doing this...
> 
> Wow.. I must have missed that discussion or forgotten it... Armando
> and I have been haggling on this point for a while..
> 
> I think we now have definite proof that for FR's you should send to
> the original address. For Time-outs however, this is not necessarily
> true. Armando
> still has some more simulation to do in this vein and I have also asked him
> to take into account the failover i.e. the cases where where you are
> transmitting as failed.
> 
> I don't see a CC problem in all of this.. at least none that I can find.
> If you
> do take a false T-3 you MAY end up retransmitting to an alternate to quickly
> and when the ACK's come back credit the alternate... but is this really
> going
> to happen? I think most of the times it will NOT..
> aka:
> 
> Primary------cwnd=N-------------->
> Alternate----cwnd=2Mtu---------->
> 
> Now assuming N is some larger number (we have been sending a while)...
> and we do
> a encounter a delay spike that causes a false T3 on Primary..
> 
> You retransmit 1 MTU to Alternate and wait for a SACK.
> 
> The SACK arrives (just a bit latter) from your false T3. You then do for the
> alternate
> 
> If (flight-size > cwnd) {
>     if(in-ss)
>          cwnd += min(1 mtu, ack-size)
>     else
>         pba += acksize;
>         if(pba > cwnd)
>                cwnd += 1 MTU
>                pba -= cwnd
>                if(pba <=0)
>                    pba = 0
> }
> 
> Now in our example guess what? flight-size = 1MTU (assuming you had
> a full MTU or more to retran)... so the "if" would never be entered and
> NO cwnd advance would happen.
> 
> It is hard for me to imagine a situation where you would really get the
> alternate above 1MTU in flight size .... so I doubt there is a problem..
> in fact
> the real issue is the false T3 will have caused a collapse of the
> primary... I guess this is what Eifel attempts to fix in TCP....
> 
> So
> 
> 1) I see no CC problem with the current scheme
> 2) We probably need to NOT retransmit FR's to an alternate
> 3) It is best, I think, if we schedule a HB on the alternate after
>     a retransmission.. Armando is still working on this point but
>     if you have any loss on the alternate you need some way of
>     collapsing the RTO so scheduling a HB can do that to help
>     performance....
> 
> I am anxious for Armando to get a bit further in his research and I think
> this will shed some additional light on the situation :>
> 
> >
> >
> > I guess NIC failure happens very infrequently when compared to
> > congestion and spurious timeout.  I don't think it is a good
> > suggestion to design the timeout retransmission around an uncommon
> > case.  And it seems to me that the correct way to handle a NIC
> > failure is to use ASCONF to tell the peer to either remove those IP
> > addresses associated with the failed NIC or to change the primary
> > path if it is to the failed NIC.  One may ask why such mechanism
> > is not included in the original spec...
> 
> Well, we DID want both PR-SCTP and ADD-IP in the base spec... but in order
> to not hold things up to long we decided to do them as seperate drafts..
> and it
> is a good thing too.. ADD-IP still needs some more work (in the security
> front) and
> PR-SCTP is now 3 years later almost done...
> 
> Many implementations do have ASCONF working.. which can help in this
> situation... but one would need (as Qiaobing pointed out) some sort of
> L2 trigger
> to tell the peer "hey your nic card X failed" so that it could "know" to
> send the
> "set-primary" to the alternate. And of course a "generic" trigger, as
> yet, does not
> exist. Another area ripe for research...
> 
> R
> 
> >
> >
> >
> >
> 
> _______________________________________________
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Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
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Kacheong Poon wrote:
> 
> Qiaobing Xie wrote:
> 
> > Are we **absolutely** sure that whatever we intended to solve in this
> > discussion has to include giving up this "retrans to alternative"
> > behavior which is so key to telecomm folks? Is there a solution that
> > will let those folks to co-exist in this SCTP world while we are
> > perfecting its congestion control here?
> 
> I don't understand why we are giving up retransmission to an
> alternate address.  The simple change I suggested was to do
> the first timeout retransmission to the original address.  SCTP
> will still send the second timeout retransmission to an alternate
> address.  

This is less dramatic. Assuming the sender's RTO_min = 1 sec at the time
of the fault, this still adds 2 sec delay to the fault recovery which
could be significant in some cases. However, does this "just one more
retrans to the current addr" really help you? How?

> This is a default behavior I suggested implementors to
> use.  Having said that, it does not mean that an SCTP implementation
> should always follow the default behavior.  It can just be a
> simple switch or an option to change the behavior to send the
> first timeout retransmission to an alternate address.

But how are we going to tell people when to *not* use that optional
behavior? I'd guess that having faster fault recovery is welcome by
everyone. I used telecomm as an example simply because for them recovery
speed becomes a vital requirement. But in reality, it will be a good
thing for everyone especially for those who have already gone through
the trouble of setting up multi-homing, right?

> 
> > I am not ready to wave the white flag yet. I'd encourage the CC
> > perfectionists to continue work and find a solution that will satisfy
> > the CC behaviors while not sacrificing a very real advantage of SCTP for
> > telecom folks. Then we would have a WIN-WIN solution and I think we can
> > do it! (I know I sound a little like a promotional speaker, my apology
> > :)
> 
> What I try to say is that if SCTP is a general purpose transport
> protocol, the default behavior should not be designed to fit in
> a particular kind of environment.  I think you agree with that,
> right (-:

I guess I would agree if I can understand why the current design does
NOT fit general purpose (also see my other email on this topic).

> 
> [From another email]
> 
>  > So you want to rely on an external monitor/detector mechanism at the
>  > peer endpoint that will tell the peer SCTP stack a NIC is bad and then
>  > the remote stack will do a "change primary addr" through ASCONF so that
>  > the local endpoint will start to move the traffic to an alternative dest
>  > addr? Ok, now you have no need to guess since the remote external
>  > detector is doing the "guessing" :-)
> 
> I don't understand.  If the NIC you are referring to above
> belongs to the endpoint, why is an external monitor/detector
> needed to tell the SCTP stack that the NIC is bad?  Isn't this
> the job of the device driver?

By "external" I referred to anything outside the SCTP stack itself.
Device driver is external and we have know knowledge as to how it really
works to determine a NIC failure, for example.

regards,
-Qiaobing

>  > But this would essentially remove most of the SCTP's auto path failure
>  > detection / recovery capabilities and take out a big chunk of SCTP's
>  > multi-homing advantage. I think this is too costly for whatever we are
>  > trying to gain here.
> 
> Assuming we are referring to a NIC belonging to the endpoint where
> the SCTP stack lives, I don't think using ASCONF will remove the
> auto detection/recovery capabilities of SCTP.  The device driver
> tells the IP stack that a NIC is bad, so the IP stack does something.
> Isn't this kind of communication default in most OSes nowadays?  If
> the stack also supports SCTP, then SCTP can use this info to do
> something.  This is quite straightforward.
> 
> --
> 
>                                                 K. Poon.
>                                                 kacheong.poon@sun.com
> 
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Randy,

Your email below shows perfectly what is the right way to address this
type of L2 problem - it is going to be a collaboration and interaction
between L2 and L3! The router's involvement (dropping the packet) and
feedback (sending back the Vtag) to L3 are the key here. The point I was
trying to make in my original email was that we are unlikely to
correctly address a L2 problem by tweaking L3 alone.

regards,
-Qiaobing

"Randall R. Stewart (home)" wrote:
> 
> Qiaobing:
> 
> Yes, I do agree with you... but then do you think
> that a router sending a ICMP message that the transport
> protocol uses to determine path mtu is bad?
> 
> I look at it not as a LAST hop .. but that at ANY point in the
> path a router can make a decision to either
> 1) drop a packet due to congestion
> or
> 2) drop a packet due to corruption
> 
> In case <1> we WANT the transport protocol to react
> to it and reduce its sending rate. But in case <2> we would
> prefer the transport protcol become aware of this drop and to
> retransmit it (if retransmission is so desired) WITHOUT reducing
> its congestion rate... This in theory could be ANY hop along a
> routing chain.. but in most cases would be the last or next to
> last hop.. I am thinking satellite transmission for next to last...
> 
> Now the problem as always been with reporting a corruption drop that
> 
> A) A mallicious receiver could figure out that a TSN is missing and generate
>     such a report to get the sender to avoid <1>.
> B) A new form of attack could be created that lets a hacker spew forth
> bunches
>     of these things to "slow" down everyone else talking to a web server so
>    he/she gets better service..
> 
> So how can a transport protocol trust the requests?
> 
> SCTP of course can help in that if the router has the packet that
> it is deciding to drop  it can return it .. and the vtag can be checked
> this helps in situation <B> but what about <A>...
> 
> The packet corruption may be at a point that the sender can't
> verify it? This is true.. and if so it should be tossed...
> 
> The new technology we have created at cisco however is a tunneling
> software that does some retransmission.. and will give up on
> that retransmission think of it as an RLP.. but when it does
> give up .. its capable of sending the packet back to the sender..
> 
> I will get a draft together on this soon... I am just short of cycles right
> now...
> 
> I have this up and running and its simulation results for working with
> satellites is remarkable.. you can get speeds that normaly take the
> disruptive PEP arpproach.. aka where you terminate the TCP/SCTP
> at both sides of the satellite to "fool" the end to end protocol into
> not realizing the long bandwidth delay by-product.. very ugly...
> 
> When I present my draft I will also include some performance
> work I have gatherede too...
> 
> R
> 
> Qiaobing Xie wrote:
> 
> >Both Randy and Caitlin's comments below are indeed very interesting. I
> >wonder if we still have consensus that SCTP is still an end-to-end, L3
> >transport protocol. If so, I believe the reason we call it end-to-end is
> >because we assume there is a virtual, structure-less, vairable
> >bandwidth, end-to-end pipe underneath the L3 transport.
> >
> >Now, I start to see people talking about "what if the last hop of the
> >pipe [between two SCTP ends] is of X or Y nature". To me, at layer 3 I
> >don't see any internal structure of the L2 pipe execpt of the assumption
> >that it is an end-to-end pipe. So where is that "last hop" people are
> >referring to?
> >
> >If this virtual pipe abstraction no longer holds, then we are in a new
> >era of layering principle. If it is still valid, then the "last hop" is
> >invisible to SCTP and needs to remain so. And the nature of the "last
> >hop" needs to be addressed in L2 groups (e.g., trigtran).
> >
> >In short, I fail to see how we could succeed by tweaking L3 SCTP to
> >address some L2 "last hop" characteristics.
> >
> >regards,
> >-Qiaobing
> >
> >"Randall R. Stewart (home)" wrote:
> >
> >
> >>Caitlin Bestler wrote:
> >>
> >>
> >>
> >>>On Monday, July 21, 2003, at 09:35 AM, Armando L. Caro Jr. wrote:
> >>>
> >>>
> >>>
> >>>>Well, your one bit solution only works if the sender follows the
> >>>>recommended mechanism for retransmissions. There hasn't been a consensus
> >>>>that retransmitting on an alternate path is always the best policy. In
> >>>>fact, I would argue that it isn't. I believe that for better performance
> >>>>without affecting failover performance, fast retransmits should go to
> >>>>the
> >>>>same destination and timeout retransmits should go to an alternate
> >>>>destination [1].  With such a retransmission policy, one bit would
> >>>>not be enough. Sure we could use 2 bits, but timestamps just seems
> >>>>cleaner.
> >>>>
> >>>>[1] http://www.cis.udel.edu/~acaro/papers/2003.icon.acaro.pdf
> >>>>
> >>>>~armando
> >>>>
> >>>>
> >>>>
> >>>In the paper cited, you fault the default SCTP behavior for assuming
> >>>"that loss indicates either that the destination address used is
> >>>unreachable,
> >>>or the destination's network path is congested."
> >>>
> >>>The implication is that the end-to-end algorithm is "giving up too
> >>>quickly"
> >>>on a path, and thereby degrading performance.
> >>>
> >>>As that I have seen this objection most often raised when the final hop
> >>>is a mobile and/or wireless link, I am tempted to turn that question
> >>>around.
> >>>
> >>>Isn't the real problem that the last hop is giving up on L2 delivery too
> >>>quickly, and creating an L3/end-to-end problem for something that
> >>>should have been handled locally?
> >>>
> >>>Basically, is each downstream frequency *really* a path? Or should
> >>>the collective set of frequencies be viewed as a single L3 path
> >>>over which there may be L2 retries?
> >>>
> >>>I designed a two-way CATV system to carry IP traffic through
> >>>set-top decoders while I was at Zenith Electronics. Whenever
> >>>the set-top needed to change frequency it would be deaf to
> >>>the headend for a short period. Realizing that dropping
> >>>packets during such a frequency shift would lead to very
> >>>poor end-to-end performance, we simply scheduled the
> >>>packet to be L2-delivered on a frequency as soon as it
> >>>was known, or on all frequencies if the receiver's status
> >>>was not known.
> >>>
> >>>We certainly do not want to begin an L2 design debate
> >>>here, but I think it is a valid question -- is it a valid assumption
> >>>that L2 delivery should be dealing with truly transitory
> >>>problems and therefore packet loss really should be
> >>>taken to as an indication that the destination address used
> >>>is unreachable or the destination's network path is
> >>>congested?
> >>>
> >>>
> >>Caitlin:
> >>
> >>As always an interesting comment from you..( yes I have a lot of
> >>other unread email that you have sent to me .. I will get to it I
> >>promise :>)
> >>
> >>Now I understand the logic here and I have two comments...
> >>
> >>a) Being able to seperate a L2 drop for NON congestion reasons from the
> >>    "default" assumption would be a nice thing.. I actually will have a
> >>proposal
> >>    to put forward on this later on when I can get the cycles to clean
> >>up the drafty
> >>    draft I have in the works somewhere :>
> >>
> >>b) If your example L2 environment just delivered the packet late (as I
> >>assume it
> >>    must have) then what you have is a spurious retransmission.. the
> >>very thing
> >>    that Eifel is trying to detect...
> >>
> >>Just a thought or two :>
> >>
> >>R
> >>
> >>
> >>
> >>>Caitlin Bestler - cait@asomi.com - http://asomi.com/
> >>>
> >>>
> >>>_______________________________________________
> >>>tsvwg mailing list
> >>>tsvwg@ietf.org
> >>>https://www1.ietf.org/mailman/listinfo/tsvwg
> >>>
> >>>
> >>>
> >>>
> >>_______________________________________________
> >>tsvwg mailing list
> >>tsvwg@ietf.org
> >>https://www1.ietf.org/mailman/listinfo/tsvwg
> >>
> >>
> >
> >_______________________________________________
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> >
> >
> >
> >

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CC: Sourabh Ladha <ladha@mail.eecis.udel.edu>
Subject: Re: [Tsvwg] SCTP Timestamps
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Randall R. Stewart (home) wrote:

> Sourabh/Stephan:
> 
> I have been contemplating the timestamp issue and I am beginning to lean
> toward using the already defined HB for this. Could you tell me if
> this would work for your eifel algorithm?
> 
> First Transmission
> 
> ----Normal DATA-no-HB------>
> 
> ----Normal HB-To-Alternate(ts/other stuff)--->
> 
> T3-Timeout
> 
> ---HB(ts/flag=retran#/tsn.. no other stuff)+Bundled-Data------->
> 
> If I hit another T3
> 
> ---HB(ts/flag=retran#/tsn.. no other stuff)+Bundled-Data------>
> 
> Now. One of a number of things will occur.
> 
> 1) The peer does NOT bundle things together... in that case a time-stamp 
> chunk
>     would have the same problems as a HB-ACK i.e. it would get lost in
>     the network from say the SACK that will come back ack'ing the data.

I think this would cause a serious problem for Eifel. A SACK for a 
retransmission without a HB (i.e. the HB was not bundled and lost or 
reordered relative to the SACK) would be falsely interpreted as a SACK 
for the original transmission, so Eifel would wrongly assume that the 
timeout was spurios and undo congestion control. This is the worst case.

In essence, if Eifel cannot be sure whether a SACK without HB really 
refers to the original transmission, it can never apply the Eifel response.

> 2) A loss occurred and thus the HB-ACK will be sent right away possibly
>    bundled with the SACK not showing any duplicates and showing gaps.
> 
> 3) A loss did NOT occur and a HB-ACK will be bundled with a SACK showing
>    a duplicate (if a duplicate is present a SACK is sent right away with 
> no delay)
> 
> 4) A loss occured and there are no gaps so the delay timer is started.. 
> thus
>    the HB-ACK would go stand alone.. and then a bit later the SACK would
>    be sent based of the delay timer.
> 
> Now if the HB-ACK comes bundled with a SACK showing duplicates you have
> detected a false retransmission.
> 
> In ANY event you can always calculate the RTT since the HB has when it
> was sent.. .and if you are really nervous about the missing delay (200ms 
> or whatever)
> you can add it in .. of course ALL HB's to alternates have this issue as 
> well :>

I agree that HBs are sufficient for the RTT measurement, but unless 
reliably bundled, they are not sufficient to resolve the retransmission 
  ambiguity.

> It seems to me that the HB/as long as bundling occurs with the HB-ACK will
> accomplish the same purposes... Or what is missing from this detection?

I think some sort of startup negotiation is the best solution so that 
(1) both sides can be sure whether the peer is able to do the bundling, 
and (2) room can be reserved in a timestamp-enabled session for HBs/TSs 
when doing fragmentation. But if we do a negotiation, we can just as 
well use a new chunk type rather then overloading HBs with new 
functionality (and thus avoid changes to the regular HB processing).

Stephan.


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CC: Kacheong Poon <kacheong.poon@sun.com>, tsvwg@ietf.org,
        "Armando L. Caro Jr." <acaro@acm.org>
Subject: Re: [Tsvwg] F-RTO and SCTP
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Qiaobing:

Comments below..

Qiaobing Xie wrote:

>Randy/Armando,
>
>I don't know much detail and the scope of your ongoing
>discussion/analysis on this issue. But one of my concerns is that if we
>focus the analysis too closely to local behaviors of a single node or
>association we may lose our sight to the bigger picture. 
>  
>
There is yet a LOT of analysis to be done I think :>

>Here is a scenario: a multi-homed SCTP association is sharing a big area
>network (or just the Internet) with other TCP, UDP, SCTP, RTP traffic
>and is actively passing data across the network. I understand there is
>no mechanism in existence yet to guarantee path diversity for this
>multi-homed SCTP association (maybe such a mechanism will be produced by
>the multi-homing WG soon.. who knows). But there is nothing to prevent
>path diversity from naturally occurring either. So, let's assume that
>the association has path diversity across the network. 
>
>Now at the SCTP sender suddenly gap SACKs or timeouts start to be
>indicated on the primary dest address. This can mean either congestion
>or path failure which we have no way to tell for sure. But it surely
>indicates *some* problem with the part of the network that the primary
>path crosses. Now the issue is how the sender should react. We seem to
>be talking about the following options:
>
>a) The "agile" approach - move retrans to the alternative at the first
>sign of trouble; this is the current RFC2960 recommendation.
>
>b) The "stubborn" approach - insisting on using the current addr for all
>trans and retrans until nothing can get through; 
>
>c) something in-between - e.g., if it is FR, use the same addr, if RTO,
>use the alternative.
>  
>
I originally was leaning toward <a> as Armando can attest too :>

But some of his work has convinced me that <c> may be a better
choice... the reason being:

1) If you are still getting Sack's that cause you to FR then there
    is no chance that the far end has failed.

2) Armando's work as shown that with even a mild loss rate on the
    alternate (due to congestion) the switched transmission is less 
beneficial
    due to I think mainly the fact that the RTO on the alternate tends to
   inflate over time... since you have very little traffic going there. My
   solution to this was to schedule a HB right away after retransmitting to
   an alternate to help bring down the RTO... this helped some (as Armandos
    simulations show) but not as much as sticking with the original 
destination. However
   Armando never included the concept of failed destination either.. 
thus some more
   work is needed...

3) After looking at all the data, I think because <1> is true in FR case 
then sticking
    with the primary will give the app the best throughput. I do believe 
that retrans
    need to go to the alternate though...

>The main advantage of "stubborn" over "agile" is (to my understanding at
>least) that the sender side CC algorithm is 1) less prone to generating
>spurious retrans, which translates into less wasted network bandwidth,
>and 2) no longer suffers from the ack ambiguity, which translates into
>???... (I am still trying to understand the real benefit here).
>  
>
I don't see how the stubborn approach gets rid of any ack ambiguity.. you
still do not know if the SACK is from the retransmitted data or the 
original.
I think the only ambiguity you may fix is for a cwnd update.. but even that
I could imagine a really really long delay spike that finally gets you
to move to the alternate.. and presto the ambiguity is there... I just
don't see that the stubborn approach helps for the ambiguity issue :>


>The benefits of "agile" over "stubborn" are far clearer to me: 1) as I
>mentioned in an earlier email, if the *some* problem is caused by an
>actual fault in the primary path, the fault recovery speed of the
>"agile" approach is much faster than the "stubborn". In fact, the
>"stubborn" approach degrades SCTP to TCP equivalent in terms of path
>failure detection latency. 2) if the *some* problem is congestion
>induced (maybe some one started download huge movie files or a workhorse
>router went down in that part of network), wouldn't it still be a very
>good thing that the SCTP traffic is swiftly shifted away from the
>congested area of the network (assuming path diversity exists here)?
>This not only helps the congestion condition to improve quicker in the
>affected area, but also lessens the impact on the data sender's
>throughput. Imagine that if in the future a portion of the Internet
>traffic can perform this type of "agile" multi-homing, how much it could
>benefit the network congestion recovery.
>
>IMHO, the *real* benefits of "agile" approach to both the application
>and the network more than justify the occasional spurious retrans and
>the whatever bad things caused by the ack ambiguity. I hope this wider
>impact is factored into your simulation and analysis.
>  
>
We need a lot more work.. and I know Armando will get this out over
the next year or so... I really think that <c> will provide the best
throughput and at the same time the best fault recovery..

R


>regards,
>-Qiaobing
>
>"Randall R. Stewart (home)" wrote:
>  
>
>>Kacheong Poon wrote:
>>
>>    
>>
>>>Randall R. Stewart (home) wrote:
>>>
>>>      
>>>
>>>>Slowly catching up on a HUGE backlog of email :-o
>>>>
>>>>As to the above.. I don't agree yet.. Armando has proposed such  to me
>>>>and I have yet to be convinced....
>>>>
>>>>It may be that it is better in some cases.. but when you start
>>>>factoring in
>>>>real failures of a NIC you may find that your failure detection
>>>>starts to
>>>>be effected in a adverse way. So far I think Armando has a good point
>>>>in that possibly FR's may be best NOT to send to an alternate .. but
>>>>T3's I think may still be best sent to an alternate...
>>>>        
>>>>
>>>If I remember correctly, the fast retransmit case was discussed
>>>some time ago in either tsvwg or sctp-impl.  The consensus is to
>>>retransmit to the original address.  I thought every implementations
>>>is already doing this...
>>>      
>>>
>>Wow.. I must have missed that discussion or forgotten it... Armando
>>and I have been haggling on this point for a while..
>>
>>I think we now have definite proof that for FR's you should send to
>>the original address. For Time-outs however, this is not necessarily
>>true. Armando
>>still has some more simulation to do in this vein and I have also asked him
>>to take into account the failover i.e. the cases where where you are
>>transmitting as failed.
>>
>>I don't see a CC problem in all of this.. at least none that I can find.
>>If you
>>do take a false T-3 you MAY end up retransmitting to an alternate to quickly
>>and when the ACK's come back credit the alternate... but is this really
>>going
>>to happen? I think most of the times it will NOT..
>>aka:
>>
>>Primary------cwnd=N-------------->
>>Alternate----cwnd=2Mtu---------->
>>
>>Now assuming N is some larger number (we have been sending a while)...
>>and we do
>>a encounter a delay spike that causes a false T3 on Primary..
>>
>>You retransmit 1 MTU to Alternate and wait for a SACK.
>>
>>The SACK arrives (just a bit latter) from your false T3. You then do for the
>>alternate
>>
>>If (flight-size > cwnd) {
>>    if(in-ss)
>>         cwnd += min(1 mtu, ack-size)
>>    else
>>        pba += acksize;
>>        if(pba > cwnd)
>>               cwnd += 1 MTU
>>               pba -= cwnd
>>               if(pba <=0)
>>                   pba = 0
>>}
>>
>>Now in our example guess what? flight-size = 1MTU (assuming you had
>>a full MTU or more to retran)... so the "if" would never be entered and
>>NO cwnd advance would happen.
>>
>>It is hard for me to imagine a situation where you would really get the
>>alternate above 1MTU in flight size .... so I doubt there is a problem..
>>in fact
>>the real issue is the false T3 will have caused a collapse of the
>>primary... I guess this is what Eifel attempts to fix in TCP....
>>
>>So
>>
>>1) I see no CC problem with the current scheme
>>2) We probably need to NOT retransmit FR's to an alternate
>>3) It is best, I think, if we schedule a HB on the alternate after
>>    a retransmission.. Armando is still working on this point but
>>    if you have any loss on the alternate you need some way of
>>    collapsing the RTO so scheduling a HB can do that to help
>>    performance....
>>
>>I am anxious for Armando to get a bit further in his research and I think
>>this will shed some additional light on the situation :>
>>
>>    
>>
>>>I guess NIC failure happens very infrequently when compared to
>>>congestion and spurious timeout.  I don't think it is a good
>>>suggestion to design the timeout retransmission around an uncommon
>>>case.  And it seems to me that the correct way to handle a NIC
>>>failure is to use ASCONF to tell the peer to either remove those IP
>>>addresses associated with the failed NIC or to change the primary
>>>path if it is to the failed NIC.  One may ask why such mechanism
>>>is not included in the original spec...
>>>      
>>>
>>Well, we DID want both PR-SCTP and ADD-IP in the base spec... but in order
>>to not hold things up to long we decided to do them as seperate drafts..
>>and it
>>is a good thing too.. ADD-IP still needs some more work (in the security
>>front) and
>>PR-SCTP is now 3 years later almost done...
>>
>>Many implementations do have ASCONF working.. which can help in this
>>situation... but one would need (as Qiaobing pointed out) some sort of
>>L2 trigger
>>to tell the peer "hey your nic card X failed" so that it could "know" to
>>send the
>>"set-primary" to the alternate. And of course a "generic" trigger, as
>>yet, does not
>>exist. Another area ripe for research...
>>
>>R
>>
>>    
>>
>>>
>>>
>>>      
>>>
>>_______________________________________________
>>tsvwg mailing list
>>tsvwg@ietf.org
>>https://www1.ietf.org/mailman/listinfo/tsvwg
>>    
>>
>
>
>  
>



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From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
To: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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> a) The "agile" approach - move retrans to the
> alternative at the first sign of trouble; this is the
> current RFC2960 recommendation.
> 
> b) The "stubborn" approach - insisting on using the
> current addr for all trans and retrans until nothing
> can get through;
> 
> c) something in-between - e.g., if it is FR, use the
> same addr, if RTO, use the alternative.

Actually I think the "stubborn" proposals are more
along the lines of "insisting on using the current addr
until it is obvious that there is really a problem
there". That is, multiple drops would indicate there
really is congestion and the path should be abandoned.
The question is whether presuming that a *single* drop
indicates a problem is appropriate.

But the "agile" versus "stubborn" distinction may be
very close to the real distinctin: "low latency" vs.
"throughput".

-   If your primary concern is low latency, rather
    than throughput, then you want to switch paths on
    the first problem. Not doing so will delay being
    able to communicate over the remaining viable
    path. That delay represents a period with loss of
    contact, which is very unacceptable for a control
    application.
    
    
-   If your primary concern is throughput then
    switching paths because of a single transient drop
    might be an overreaction. The alternate path has
    not been "exercised". Shifting to it will result
    in at least a temporary drop in throughput.
    
    On the other hand, delaying switching might be
    delaying a corrective response to a congestion
    problem. 


That's a distinction based upon usage, not on the
target network. Which is certainly more acceptable.

Just thinking out loud here, but would it be
appropriate to leave this choice to the sender? The
sender could easily know which mode it was in either by
explicit user configuration and/or just observing. (If
the pending Data Chunk queue hasn't hit empty for
several seconds you are probably not being used by a
"low latency" application).

Note: I still believe that such an algorithm has to
make sense based upon "typical" networks and should not
be used to solve special L2 problems. Is there any
hard research as to how often a packet is dropped
for reasons that are neither equipment failure or 
congestion? How high of a "false positive" rate
are we talking about here?





Caitlin Bestler - http://asomi.com/




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From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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        "Armando L. Caro Jr." <acaro@acm.org>
Subject: Re: [Tsvwg] F-RTO and SCTP
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> >...2) if the *some* problem is congestion
> >induced (maybe some one started download huge movie files or a workhorse
> >router went down in that part of network), wouldn't it still be a very
> >good thing that the SCTP traffic is swiftly shifted away from the
> >congested area of the network (assuming path diversity exists here)?
> >This not only helps the congestion condition to improve quicker in the
> >affected area, but also lessens the impact on the data sender's
> >throughput. Imagine that if in the future a portion of the Internet
> >traffic can perform this type of "agile" multi-homing, how much it could
> >benefit the network congestion recovery.
....
> 
> We need a lot more work.. and I know Armando will get this out over
> the next year or so... I really think that <c> will provide the best
> throughput and at the same time the best fault recovery..

One more observation along this line about CC research - The age-old TCP
CC algorithm (as well as that of SCTP and the newer TFRC) is based on a
simple "common sense" strategy: "hit the brake when you see the road
ahead is jammed". Over the years, countless research effort has gone
into figuring out when to hit the brake and how hard to hit the brake
:-) But why sticking to this strategy? It is simply because that is the
ONLY road known to the car driver (i.e., the TCP sender) to reach the
destination! 

Now, SCTP's multi-homing support opens up a new dimension to the CC
world - now the car driver has the knowledge of (possible) alternate
routes to the same destination. Shouldn't this new dimension prompt our
driver to rethink about its strategy and take full advantage of it? Or
should we say "this new dimension does not fit into the age-old strategy
so let's take it out"?

IMHO, CC research community should start thinking about how to best
benefit from multi-homing now, instead of the other way around.

regards,
-Qiaobing

> 
> R
>

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Qiaobing Xie wrote:

>>I don't understand why we are giving up retransmission to an
>>alternate address.  The simple change I suggested was to do
>>the first timeout retransmission to the original address.  SCTP
>>will still send the second timeout retransmission to an alternate
>>address.  
> 
> 
> This is less dramatic. Assuming the sender's RTO_min = 1 sec at the time
> of the fault, this still adds 2 sec delay to the fault recovery which
> could be significant in some cases. However, does this "just one more
> retrans to the current addr" really help you? How?

I believe the second timeout provides a very good indication
that the path to the original address has some issues.  The
primary concerns we have are spurious timeout, congestion,
NIC failure.  I believe the spurious timeout issue is mostly
ruled out if the the timer fires the second time.  If the
first retransmission is also dropped because of congestion,
it will be the same to use the primary and alternate path
as far as congestion control is concerned as ssthresh will
not be a factor anymore.  And the path must be very congested.
If it is a NIC failure, I guess two tries are good enough.

If SCTP retransmits to the original address in the first
timeout, I believe F-RTO will work pretty well in detecting
spurious timeouts without introducing new chunks or new
bundling rules.  To me, this is a big win for F-RTO.  We
don't want to keep changing rules, especially if it requires
both sides to agree.  F-RTO can be done on the sender side
only without changing anything in RFC 2960.  If we really
want to speed detecting NIC failure up, the already defined
ASCONF should work pretty well.  As a matter of fact, it
should work faster than the timeout retransmission.

>>This is a default behavior I suggested implementors to
>>use.  Having said that, it does not mean that an SCTP implementation
>>should always follow the default behavior.  It can just be a
>>simple switch or an option to change the behavior to send the
>>first timeout retransmission to an alternate address.
> 
> 
> But how are we going to tell people when to *not* use that optional
> behavior? I'd guess that having faster fault recovery is welcome by
> everyone. I used telecomm as an example simply because for them recovery
> speed becomes a vital requirement. But in reality, it will be a good
> thing for everyone especially for those who have already gone through
> the trouble of setting up multi-homing, right?

You are making the assumption that the alternate address is
always better here.  I don't believe SCTP has this knowledge.
It is pretty much a guess.  So I suggest people not to use
this optional behavior unless they are pretty sure that their
network setup supports it.  For example, it may work in your
telecomm network.  And if I understand how you put it, the
telecomm network is pretty much separated from the real
Internet such that it can provide that kind of guarantees you
mentioned, no congestion.  So I don't see a problem for all
the SCTP stacks used in that environment to change the behavior.


I guess we should not provide false information to people that
the alternate address is always better than the original
address in the event of retransmission.  Or am I missing something
that you can provide such a proof that the alternate address is
always better in the case of the first timeout?

> I guess I would agree if I can understand why the current design does
> NOT fit general purpose (also see my other email on this topic).

I believe the confusion caused by retransmission to an alternate
address is well discussed.  The current default behavior is only
guaranteed to work well in a specially designed network.  In
general, SCTP does not have this kind of knowledge about the network.
And I believe the simple change not to use alternate address in the
first timeout helps solve the majority of the confusion described with
the help of F-RTO, a mechanism which does not require new chunks or
bundling rules.  To me, this is a very strong reason not to follow
the default behavior described in RFC 2960.  Well, just my $0.02
anyway.  But I hope people will not spread the idea that using
an alternate address is always better, unless the network is designed
that way.  As I mentioned earlier, there may be a proof contrary
to this which I haven't seen.  If it can be proved, I have no issue
using the alternate address in the first timeout.

> By "external" I referred to anything outside the SCTP stack itself.
> Device driver is external and we have know knowledge as to how it really
> works to determine a NIC failure, for example.

Well, I guess you think all those products, commercial and free,
which provide such kind of NIC failure detection are wrong
architecturally and do not work well...


-- 

						K. Poon.
						kacheong.poon@sun.com



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Caitlin Bestler wrote:

> Actually I think the "stubborn" proposals are more
> along the lines of "insisting on using the current addr
> until it is obvious that there is really a problem
> there". That is, multiple drops would indicate there
> really is congestion and the path should be abandoned.
> The question is whether presuming that a *single* drop
> indicates a problem is appropriate.
> 
> But the "agile" versus "stubborn" distinction may be
> very close to the real distinctin: "low latency" vs.
> "throughput".
> 
> -   If your primary concern is low latency, rather
>     than throughput, then you want to switch paths on
>     the first problem. Not doing so will delay being
>     able to communicate over the remaining viable
>     path. That delay represents a period with loss of
>     contact, which is very unacceptable for a control
>     application.

I guess the issue here is whether using the alternate
address will be better.  Firstly, I believe SCTP cannot
know for sure that the paths to the two different addresses
do not cross.  If they in fact are crossed, sending to
the alternate address may have the same problem as well.
I believe all SCTP knows about the alternate address is
that it is reachable at the time SCTP receives the last
HB-ACK.  That is all.  I don't know if SCTP can infer
more info from that.  So blindly retransmitting to an
alternate address may not help in the "low latency"
goal at all.

> -   If your primary concern is throughput then
>     switching paths because of a single transient drop
>     might be an overreaction. The alternate path has
>     not been "exercised". Shifting to it will result
>     in at least a temporary drop in throughput.
>     
>     On the other hand, delaying switching might be
>     delaying a corrective response to a congestion
>     problem. 
> 
> 
> That's a distinction based upon usage, not on the
> target network. Which is certainly more acceptable.

But the issue is if the network actually supports the
claim that the above two mechanisms work as described.

> Just thinking out loud here, but would it be
> appropriate to leave this choice to the sender? The
> sender could easily know which mode it was in either by
> explicit user configuration and/or just observing. (If
> the pending Data Chunk queue hasn't hit empty for
> several seconds you are probably not being used by a
> "low latency" application).

Do you mean by the sender's SCTP code or the app code?
I don't know if it works well for SCTP to determine
the network usage of the app.

> Note: I still believe that such an algorithm has to
> make sense based upon "typical" networks and should not
> be used to solve special L2 problems. Is there any
> hard research as to how often a packet is dropped
> for reasons that are neither equipment failure or 
> congestion? How high of a "false positive" rate
> are we talking about here?

It seems that "typical" means different things to
different people.  So such an algorithm may actually
need to work well in "all kinds of" networks...


-- 

						K. Poon.
						kacheong.poon@sun.com



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Thread-Topic: Updating timestamps (ts_recent) in Linux
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From: "Duke, Martin" <martin.duke@boeing.com>
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Subject: [Tsvwg] Updating timestamps (ts_recent) in Linux
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I've heard rumors that this has been discussed before on tcp-impl, but =
an extensive search found no messages on this topic, so apologies for =
possibly re-treading old ground.

How do people feel about the way Linux decides what timestamp to echo?

Specifically, although the behavior is compliant with RFC 1323, an old =
addendum to the RFC (http://www.kohala.com/start/tcplw-extensions.txt) =
pointed out some odd behavior.

Essentially, if the connection goes down, the subsequent retransmissions =
will have later ts values.  However, if an ACK was lost, the kernel =
currently does not update ts_recent with these values.  If the channel =
comes back up and a retrans succeeds, the client will send an ack with =
an ts echo equal to the old value of ts_recent.

As the difference between the echo and the current time is quite large, =
it=20
will result in an extremely large RTT measurement, triggering a huge =
increase=20
in the RTO.  In an unreliable, bursty link, such as satellites, another =
outage=20
could occur shortly thereafter, and an inappropriately long timeout can =
occur.

The reason for the problem is that ts_recent is only updated if the =
receive window moves to the right.  Ideally, packets with sequence =
numbers to the left of the window edge should trigger an update as well.

There are certainly situations where the current Linux behavior is =
terrible, but I have yet to see an explanation for why this "bug" should =
not be fixed.  I've heard criticisms of the document cited above as a =
whole but have yet to understand why updating ts_recent more frequently =
will break something.  I know that BSD fixed this some time ago (Stevens =
talks about it in his book).

Any thoughts?

Martin Duke
martin.duke@boeing.com

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Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
References: <Pine.LNX.4.44.0307221304550.3045-100000@melkki.cs.Helsinki.FI> <3F1D8B89.40502@sun.com> <3F2E4F3A.7020500@stewart.chicago.il.us> <3F3013CB.3020505@sun.com> <3F3022AA.CF63147F@motorola.com> <3F302502.3050904@sun.com> <3F302E50.1953A739@motorola.com> <3F315350.3050803@sun.com> <3F318120.D983756A@motorola.com> <3F32B176.5040100@sun.com>
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Kacheong Poon wrote:
....
> You are making the assumption that the alternate address is
> always better here.  I don't believe SCTP has this knowledge.
> It is pretty much a guess.  

You emphasized this point several more times below (and did so in your
another email). My comment is that the enter TCP/SCTP CC algorithm is
designed around "making guesses" - or call it the "learn by experience"
principle. If one rejects this principle, nothing in the CC design will
hold. 

In our particular SCTP case here, the sender just had a "bad" experience
with the primary addr and has "no or little experience" with the
alternate addr. No one in his right mind will ever say he's sure that
the alternate address is always better. But it is simply due to this
fundamental "learn by experience" principle that the sender has a reason
to favor the alternate addr ***at that particular moment***. The
reasoning is not based on "always better", rather it is on the "no
experience + optimistism" vs. "fresh bad experience".

> So I suggest people not to use
> this optional behavior unless they are pretty sure that their
> network setup supports it.  For example, it may work in your
> telecomm network.  And if I understand how you put it, the
> telecomm network is pretty much separated from the real
> Internet such that it can provide that kind of guarantees you
> mentioned, no congestion.  So I don't see a problem for all
> the SCTP stacks used in that environment to change the behavior.

As I said before, it's not just about telecomm. Rather, any association
that has path diversity will potentially benefit. This includes
associations across the Internet since there is nothing to prevent path
diversity naturally occurring in the Internet.

> 
> I guess we should not provide false information to people that
> the alternate address is always better than the original
> address in the event of retransmission.  


> ... Or am I missing something
> that you can provide such a proof that the alternate address is
> always better in the case of the first timeout?

ditto.

> 
> > I guess I would agree if I can understand why the current design does
> > NOT fit general purpose (also see my other email on this topic).
> 
> I believe the confusion caused by retransmission to an alternate
> address is well discussed.  The current default behavior is only
> guaranteed to work well in a specially designed network.  In
> general, SCTP does not have this kind of knowledge about the network.

It is quite clear that if you have path diversity you will benefit from
it; But if you don't, will it hurt? This is what I wanted to understand
in my question above.

> And I believe the simple change not to use alternate address in the
> first timeout helps solve the majority of the confusion described with
> the help of F-RTO, a mechanism which does not require new chunks or
> bundling rules.  To me, this is a very strong reason not to follow
> the default behavior described in RFC 2960.  Well, just my $0.02
> anyway. But I hope people will not spread the idea that using
> an alternate address is always better, unless the network is designed
> that way.  As I mentioned earlier, there may be a proof contrary
> to this which I haven't seen.  If it can be proved, I have no issue
> using the alternate address in the first timeout.

ditto.

> 
> > By "external" I referred to anything outside the SCTP stack itself.
> > Device driver is external and we have know knowledge as to how it really
> > works to determine a NIC failure, for example.
> 
> Well, I guess you think all those products, commercial and free,
> which provide such kind of NIC failure detection are wrong
> architecturally and do not work well...

Nothing is wrong about using external detection. I think it's a matter
of design choice. In SCTP, the design decision was made very early to
make it more robust by including built-in failure detection and recovery
mechanism. Now you seem to suggest remove or disallow those built-in
mechanism and instead, to rely on external mechanism for failure
detection. Doing so, aren't you going to make SCTP users' life harder?

regards,
-Qiaobing

> 
> --
> 
>                                                 K. Poon.
>                                                 kacheong.poon@sun.com
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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Date: Fri, 08 Aug 2003 09:50:46 +0200
To: "Duke, Martin" <martin.duke@boeing.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: [Tsvwg] Updating timestamps (ts_recent) in Linux
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At 23:59 07.08.2003, Duke, Martin wrote:
>How do people feel about the way Linux decides what timestamp to echo?
>
>Specifically, although the behavior is compliant with RFC 1323, an old 
>addendum to the RFC (http://www.kohala.com/start/tcplw-extensions.txt) 
>pointed out some odd behavior.

I don't think that LINUX should be blamed for this behavior since afterall 
- as you said - it is compliant with RFC1323. One problem certainly is that 
various RFC1323 update attempts are floating around that suggest different 
TS echoing rules, and therefore different TCPs do different things here.

My personal opinion is that the soon-to-be-established TCP working group 
needs to have a revision of RFC1323 as one of its work items, and the TS 
echoing rules would be one of the issues to be discussed; especially rule 
(B) in Section 3.4 of RFC1323. If I understand you right it is that rule 
(B) that is causing the headaches in your scenario.

We found similar issue with that rule (B). In particular, it prevents the 
TCP sender from sampling the *real* RTT during loss recovery since the 
returning DUPACKs all carry the same old timestamp which leads to an 
inflated RTT.

     [BTW, the ns2 code is not compliant with RFC1323 in this respect!
      At least, rule (B) doesn't seem to be followed.]

One solution would be to change rule (B) to the opposite, i.e., an ACK for 
an out-of-order segment should contain the timestamp from the most recently 
received segment. And in addition, allow the TCP sender to take RTT samples 
from DUPACKs as long as it has data outstanding. The latter is currently 
not allowed due to a rule stated in Section 3.3 of RFC1323:
            A TSecr value received in a segment is used to update the
            averaged RTT measurement only if the segment acknowledges
            some new data, i.e., only if it advances the left edge of the
            send window.

These changes should also take care of the problem you describe, right?

If you're interested, we have a paper in-progress that also discusses these 
issues. A draft of the paper is available here:
http://www.cs.helsinki.fi/u/gurtov/reiner/PeakHopper_draft0207.pdf

///Reiner






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Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
From: Pasi Sarolahti <pasi.sarolahti@nokia.com>
To: ext Kacheong Poon <kacheong.poon@sun.com>
Cc: tsvwg@ietf.org
In-Reply-To: <3F32B176.5040100@sun.com>
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On Thu, 2003-08-07 at 23:07, ext Kacheong Poon wrote:

> I believe the confusion caused by retransmission to an alternate
> address is well discussed.  The current default behavior is only
> guaranteed to work well in a specially designed network.  In
> general, SCTP does not have this kind of knowledge about the network.
> And I believe the simple change not to use alternate address in the
> first timeout helps solve the majority of the confusion described with
> the help of F-RTO, a mechanism which does not require new chunks or
> bundling rules.  To me, this is a very strong reason not to follow
> the default behavior described in RFC 2960.  Well, just my $0.02
> anyway.  But I hope people will not spread the idea that using
> an alternate address is always better, unless the network is designed
> that way.  As I mentioned earlier, there may be a proof contrary
> to this which I haven't seen.  If it can be proved, I have no issue
> using the alternate address in the first timeout.

If I understood it correctly, the first RTO (or T3) retransmission to
alternate address may cause problem for F-RTO and Eifel only if the RTT
of the alternate path is smaller than the RTT of the original path plus
the delay spike, so that the acknowledgement of the retransmission
arrives before the acknowledgements of the delayed original segments.

The problem with F-RTO in setup above was that a possible loss of the
originally transmitted data chunk that was later retransmitted with RTO
may not be detected, if the RTO was caused by a delay spike. So, as far
as I see it, in this case the unnecessary retransmissions can be avoided
with F-RTO, but the congestion control decision may be incorrect
(depending on what kind of congestion control is applied after detecting
a spurious timeout). Hence, with 'SCTP-retransmitting-to-alternate-addr'
the congestion control undoing decision should be made only after the
unnecessary RTO retransmission can be confirmed with a duplicate-TSN.
The benefit of F-RTO would still be that the other unnecessary
retransmissions after RTO can be avoided, regardless of the
characteristics of different paths.

This being said, I believe the issue of whether to do the first
retransmission to original or alternate address is not that important
from the F-RTO standpoint, although making first retransmission to the
original would make things a bit easier. Of course, the other aspects
discussed here are another thing.

- Pasi



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References: <Pine.LNX.4.44.0307221304550.3045-100000@melkki.cs.Helsinki.FI> <3F1D8B89.40502@sun.com> <3F2E4F3A.7020500@stewart.chicago.il.us> <3F3013CB.3020505@sun.com> <3F3022AA.CF63147F@motorola.com> <3F302502.3050904@sun.com> <3F302E50.1953A739@motorola.com> <3F315350.3050803@sun.com> <3F318120.D983756A@motorola.com> <3F32B176.5040100@sun.com>
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Kacheong Poon wrote:

>> By "external" I referred to anything outside the SCTP stack itself.
>> Device driver is external and we have know knowledge as to how it really
>> works to determine a NIC failure, for example.
>
>
> Well, I guess you think all those products, commercial and free,
> which provide such kind of NIC failure detection are wrong
> architecturally and do not work well...


Kacheong:

I don't think this is what Qiaobing meant... I too have
had issues with failure detection in NIC's. The problem is:

1) Some NIC support it, some don't
2) The driver generally has to do some query to see
    that status (active/failed etc)..
3) The driver generally does NOT send this information on
    outside of the driver because there is no general purpose
    way for it to do so.

I have always thought there needed to be some "standard" ethernet or IP
frame the driver could generate to report local changes to the
NIC card... this would give kernel and driver developers a standard
mechanism to pass on to transport protocols etc. And security on
this would not be an issue.. if one comes in from the outside the
driver could silently discard it...

I had a proposal once and even made a driver generate such.. but I never
did anything with it .. not enough cycles  :-0

Oh, well.. maybe someday

R

---

Randall R. Stewart

>
>



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Subject: RE: [Tsvwg] Updating timestamps (ts_recent) in Linux
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> -----Original Message-----
> From: Reiner Ludwig [mailto:Reiner.Ludwig@ericsson.com]
> Sent: Friday, August 08, 2003 12:51 AM
> To: Duke, Martin
> Cc: tsvwg@ietf.org
> Subject: Re: [Tsvwg] Updating timestamps (ts_recent) in Linux
>=20
> especially rule=20
> (B) in Section 3.4 of RFC1323. If I understand you right it=20
> is that rule=20
> (B) that is causing the headaches in your scenario.
>=20
> We found similar issue with that rule (B). In particular, it=20
> prevents the=20
> TCP sender from sampling the *real* RTT during loss recovery=20
> since the=20
> returning DUPACKs all carry the same old timestamp which leads to an=20
> inflated RTT.

Actually, it's not rule (B) but rule (2), in a sense exactly the =
opposite.  If a burst error wipes out a series of ACKs, the sender will =
retransmit a packet to the left of Last.ACK.Sent.  The current test
		SEG.SEQ <=3D Last.ACK.sent < SEG.SEQ + SEG.LEN
will not update ts_recent with the received value.  A better test is
     (SEG.SEQ <=3D Last.ACK.sent) && (ts_val > ts_recent)
since this will allow retransmitted packets to be echoed.

Of course, your problem is more prevalent in single-packet loss =
situations than for burst errors.  I suppose if one combines our =
proposals one ends up with
	if (ts_val > ts_recent)
		ts_recent =3D ts_val
and doesn't pay any attention to sequence numbers at all.

As far as I can figure, these more liberal rules for echoing would =
produce more truthful RTT estimates.  The only counter-argument I've =
heard is that it's best to be conservative, especially in the face of =
packet loss and retransmission.
	In my opinion, however, it's best to be as accurate as possible with =
the RTT, and if you wish to be conservative, do so with the way you =
convert the RTT to the RTO.  To do otherwise is to invite pathological =
behavior in atypical but not uncommon situations.
	I would strongly support any effort to revise RFC 1323 along these =
lines.

Martin


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From: "Geib, Ruediger" <Ruediger.Geib@t-systems.com>
To: morita.naotaka@lab.ntt.co.jp
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Subject: [Tsvwg] Comments on draft-morita-tsvwg-pps
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Hello Naotaka,

I think that pps is a good=20
idea. Having not to maintain QoS signaling=20
state in a backbone router is a pretty=20
good thing to do.

Some comments on contents:

Section 4 suggests two stage monitoring=20
of end-system behaviour. I agree that abuse=20
must be prevented at the source. You don't=20
talk of signaling there, but I guess it=20
could be part of a two stage monitoring=20
solution (you mention however a "declared=20
peak bit rate" - sounds like signaling=20
to me).

Section 5 is about system behaviour in the=20
case of congestion. Would something like=20
an "explicit congestion notification" of=20
the backbone be useful? If end systems=20
could, as a reaction on such a congestion=20
indication, use codecs with lower bit rate=20
for their MF-H traffic, additional calls=20
may be accepted.=20

Section 7.1: I'm not a queuing expert, but=20
why would MF-M traffic see higher jitter?=20
Aren't all MF packet in the same queue and=20
those tagged as MF-M are discarded earlier?=20
The result is higher loss probability,=20
which is intended. But it shouldn't be=20
higher jitter (don't earlier losses make=20
it look like a "shorter" queue?).

Section 7.4 suggests the transmitter not to=20
pause at the edge. ISPs may charge on volume=20
and customers won't be happy if their=20
terminal creates traffic while they aren't=20
active.=20
I understand the issue and I should offer=20
an alternative. Some form of signaling based=20
admission control in combination with=20
measurements?

Section 7.5: The RMONMIB WG maintains some=20
drafts promoting RAQMON. I'm not familiar with=20
details, but your media monitoring server might=20
require similar functionality. If you are=20
interested:
http://www.ietf.org/internet-drafts/draft-ietf-rmonmib-raqmon-framework-=
02.txt

Regards, R=FCdiger


R=FCdiger Geib
T-Systems
Systems Integration
Senior Expert, Technologiezentrum
Am Kavalleriesand 3, 64295 Darmstadt
Germany
Fon: +49 6151 832138
http://www.t-systems.com
=20

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Date: Mon, 11 Aug 2003 13:25:00 +0900
To: "Geib, Ruediger" <Ruediger.Geib@t-systems.com>
From: "MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp>
Subject: Re: [Tsvwg] Comments on draft-morita-tsvwg-pps
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Ruediger,

Thank you so much for your mail.

At 11:24 03/08/08 +0200, Geib, Ruediger wrote:
>Hello Naotaka,
>
>I think that pps is a good
>idea. Having not to maintain QoS signaling
>state in a backbone router is a pretty
>good thing to do.
>
>Some comments on contents:
>
>Section 4 suggests two stage monitoring
>of end-system behaviour. I agree that abuse
>must be prevented at the source. You don't
>talk of signaling there, but I guess it
>could be part of a two stage monitoring
>solution (you mention however a "declared
>peak bit rate" - sounds like signaling
>to me).

Yes, you are correct.  In order to monitor or audit
the terminal behavior, I expect some signaling mechanism
to initiate two stage monitoring.
I am not yet clear enough whether new functions are necessary.
I will provide them in the revised ID, if any.

>Section 5 is about system behaviour in the
>case of congestion. Would something like
>an "explicit congestion notification" of
>the backbone be useful? If end systems
>could, as a reaction on such a congestion
>indication, use codecs with lower bit rate
>for their MF-H traffic, additional calls
>may be accepted.

In section 4, I briefly touch on the possibility of
"explicit congestion notification."
At this stage, just for the sake of the simplicity and being easy to depl=
oyment,
I concentrate on "implicit" notificaiton such as packet loss.

>Section 7.1: I'm not a queuing expert, but
>why would MF-M traffic see higher jitter?
>Aren't all MF packet in the same queue and
>those tagged as MF-M are discarded earlier?
>The result is higher loss probability,
>which is intended. But it shouldn't be
>higher jitter (don't earlier losses make
>it look like a "shorter" queue?).

In case of one queue model, you are right.
But, I want to see the possiblity ot two queue model.
In such case, the jitter of MF-M will be higher than the MF-H.

>Section 7.4 suggests the transmitter not to
>pause at the edge. ISPs may charge on volume
>and customers won't be happy if their
>terminal creates traffic while they aren't
>active.
>I understand the issue and I should offer
>an alternative. Some form of signaling based
>admission control in combination with
>measurements?

There are two ways of charging.
One is simply to count the number of packets.
The other is to measure the session duration, and multipiled by the=20
declared rate.
Personally, I think the latter is easy to implement, but depends on
edge router's capability.

In any case, if the transmitter suspend and resume sending packets with a=
=20
long interval,
it will impact on the mulplexed traffic characteristics.
If such impact is great, the edge router has to terminate the media to av=
oid
the big fluctuation.

>Section 7.5: The RMONMIB WG maintains some
>drafts promoting RAQMON. I'm not familiar with
>details, but your media monitoring server might
>require similar functionality. If you are
>interested:
>http://www.ietf.org/internet-drafts/draft-ietf-rmonmib-raqmon-framework-=
02.txt

Thank you so much.  I will review it.
One of the difference would be to audit the forward RTP flows as well as
backward RTCP flows and check the copprespondence between them.

Naotaka


>Regards, R=1B$B=97E=1B(Biger
>
>
>R=1B$B=97E=1B(Biger Geib
>T-Systems
>Systems Integration
>Senior Expert, Technologiezentrum
>Am Kavalleriesand 3, 64295 Darmstadt
>Germany
>Fon: +49 6151 832138
>http://www.t-systems.com
>
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg

----------------------------------
Naotaka MORITA, Senior Research Engineer, Supervisor
Network Service Systems Laboratories
Information Sharing Laboratory Group
NTT
3-9-11, Midori-Cho, Musashino-Shi, Tokyo 180-8585 JAPAN
Tel. +81-422-59-7464, ISDN-Fax +81-422-60-4012
E-mail: morita.naotaka@lab.ntt.co.jp
MSN messenger: moritanaotaka@hotmail.com
NTT home page: http://www.ntt.co.jp/about/e/index.html=20



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Date: Mon, 11 Aug 2003 14:55:33 +0200
To: "Duke, Martin" <martin.duke@boeing.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: RE: [Tsvwg] Updating timestamps (ts_recent) in Linux
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At 17:37 08.08.2003, Duke, Martin wrote:
>I suppose if one combines our proposals one ends up with
>         if (ts_val > ts_recent)
>                 ts_recent = ts_val
>and doesn't pay any attention to sequence numbers at all.
>
>As far as I can figure, these more liberal rules for echoing would produce 
>more truthful RTT estimates.  The only counter-argument I've heard is that 
>it's best to be conservative, especially in the face of packet loss and 
>retransmission.
>         In my opinion, however, it's best to be as accurate as possible 
> with the RTT, and if you wish to be conservative, do so with the way you 
> convert the RTT to the RTO.  To do otherwise is to invite pathological 
> behavior in atypical but not uncommon situations.

I basically agree, although we should hold on to the special rule that 
deals with delayed ACKs. Otherwise, the sender can badly underestimate the 
RTT, and it seems difficult to accommodate for that in the RTO calculation.

That way, we would end up with:
         if (ts_val > ts_recent) AND ("last data arrival was acked")
                 ts_recent = ts_val

///Reiner 


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
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The chairs would like to announce a two-week working group last call for
comments on draft-ietf-tsvwg-highspeed-00 commencing today (Monday 8/11/03).
While he author has already produced a small editorial fix to the draft
(softening the text about the suitability of these experimental mechanisms
for deployment) in a new version, we'd like to last call the existing text
with the understanding that a modified version with the text in question
(along with any other needed fixes) will be submitted before the document
moves on to IESG review.

Please direct any last call comments associated with the draft to this
mailing list.

Jon Peterson
NeuStar, Inc.


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Date: Tue, 12 Aug 2003 10:40:47 +0200
To: "Duke, Martin" <martin.duke@boeing.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: RE: [Tsvwg] Updating timestamps (ts_recent) in Linux
Cc: tsvwg@ietf.org
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 >
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At 14:55 11.08.2003, Reiner Ludwig wrote:
>At 17:37 08.08.2003, Duke, Martin wrote:
>>I suppose if one combines our proposals one ends up with
>>         if (ts_val > ts_recent)
>>                 ts_recent = ts_val
>>and doesn't pay any attention to sequence numbers at all.
>>
>>As far as I can figure, these more liberal rules for echoing would 
>>produce more truthful RTT estimates.  The only counter-argument I've 
>>heard is that it's best to be conservative, especially in the face of 
>>packet loss and retransmission.
>>         In my opinion, however, it's best to be as accurate as possible 
>> with the RTT, and if you wish to be conservative, do so with the way you 
>> convert the RTT to the RTO.  To do otherwise is to invite pathological 
>> behavior in atypical but not uncommon situations.
>
>I basically agree, although we should hold on to the special rule that 
>deals with delayed ACKs. Otherwise, the sender can badly underestimate the 
>RTT, and it seems difficult to accommodate for that in the RTO calculation.
>
>That way, we would end up with:
>         if (ts_val > ts_recent) AND ("last data arrival was acked")
>                 ts_recent = ts_val

I just realized that a more precise specification of what I meant is this:
         if (ts_val > ts_recent) AND
            ("an ACK or DUPACK was sent in response to the last data arrival")
                 ts_recent = ts_val

///Reiner


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From: David Borman <dab@BSDI.COM>
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To: martin.duke@boeing.com, tsvwg@ietf.org
Subject: Re: [Tsvwg] Updating timestamps (ts_recent) in Linux
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Hi,

Sorry that I haven't responded sooner.  I've submitted the update to
RFC 1323 to internet drafts again, maybe this time we can get it
actually published...

> Date: Thu, 7 Aug 2003 14:59:05 -0700
> From: "Duke, Martin" <martin.duke@boeing.com>
> Subject: [Tsvwg] Updating timestamps (ts_recent) in Linux
>
> I've heard rumors that this has been discussed before on tcp-impl, but an
> extensive search found no messages on this topic, so apologies for possibly
> re-treading old ground.
>
> How do people feel about the way Linux decides what timestamp to echo?
>
> Specifically, although the behavior is compliant with RFC 1323, an old
> addendum to the RFC (http://www.kohala.com/start/tcplw-extensions.txt)
> pointed out some odd behavior.

These changes are in 1323.bis.

> Essentially, if the connection goes down, the subsequent retransmissions
> will have later ts values.  However, if an ACK was lost, the kernel
> currently does not update ts_recent with these values.  If the channel
> comes back up and a retrans succeeds, the client will send an ack with an
> ts echo equal to the old value of ts_recent.
>
> As the difference between the echo and the current time is quite large, it 
> will result in an extremely large RTT measurement, triggering a huge increase 
> in the RTO.  In an unreliable, bursty link, such as satellites, another outage
> could occur shortly thereafter, and an inappropriately long timeout can occur.
>
> The reason for the problem is that ts_recent is only updated if the receive
> window moves to the right.  Ideally, packets with sequence numbers to the
> left of the window edge should trigger an update as well.

For copying TSval from the packet to TS.Recent, rule (2) in RFC 1323 is:
              SEG.SEQ <= Last.ACK.sent < SEG.SEQ + SEG.LEN

This attempted to only update TS.Recent when you received new data.
But that causes the problem with lost ACKs.  In addition, when you
receive an ACK (SEG.LEN == 0), the timestamp that you receive will
not be artificially inflated (since it was just generated by the remote
side when it sent the ACK), so it is OK to update TS.Recent from ACK only
packets (but only if TS.Recent has advanced)

In 1323.bis, rule (2) is:
              SEG.TSval >= TSrecent and SEG.SEQ <= Last.ACK.sent

This change addresses these two issues with RFC 1323:
        (1)  It failed to update TSrecent for a retransmitted segment
             that resulted from a lost ACK.
   
        (2)  It failed if SEG.LEN = 0.

So this modified rule allows TSrecent to be updated whenever a newer
TSrecent is received, except when the packet is out of order.  In that
case, we continue to echo the TS from the latest received packet at or
to the left edge of the window.

> There are certainly situations where the current Linux behavior is terrible,
> but I have yet to see an explanation for why this "bug" should not be fixed.
> I've heard criticisms of the document cited above as a whole but have yet to\
> understand why updating ts_recent more frequently will break something.  I
> know that BSD fixed this some time ago (Stevens talks about it in his book).
>
> Any thoughts?

As a final thought, updating TSrecent more often does not mean that you
can get RTT measurements more often.  It continues to be the case that the
only time you can get a reliable RTT measurement from timestamps is when
when new data is acknowledged.  That is because dup ACK packets can be
spontainiously generated by the remote side to do a window update.  In
that case, taking an RTT measurement can be vastly over inflated.  And
since there is no way to reliably identify whether a received dup ACK
is due to out-of-order data arriving or due to the remote side deciding
to update the window, the best course of action is to only take RTT
measurements when new data is acknowledged.
 
> Martin Duke
> martin.duke@boeing.com

			-David Borman

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Subject: RE: [Tsvwg] Updating timestamps (ts_recent) in Linux
Date: Tue, 12 Aug 2003 09:51:51 -0700
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From: "Duke, Martin" <martin.duke@boeing.com>
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>=20
> As a final thought, updating TSrecent more often does not=20
> mean that you
> can get RTT measurements more often.  It continues to be the=20
> case that the
> only time you can get a reliable RTT measurement from=20
> timestamps is when
> when new data is acknowledged.  That is because dup ACK packets can be
> spontainiously generated by the remote side to do a window update.  In
> that case, taking an RTT measurement can be vastly over inflated.  And
> since there is no way to reliably identify whether a received dup ACK
> is due to out-of-order data arriving or due to the remote=20
> side deciding
> to update the window, the best course of action is to only take RTT
> measurements when new data is acknowledged.
> =20
> 			-David Borman
>=20

David,=20
	Thanks for the insights.

	I whole-heartedly agree with the revision, but I can see two corner =
cases will the measurement could still be massively inflated:

1)      DATA 5, ACK 37, TS 12 -------------> (has long RTO, for whatever =
reason)

		  (lost) --------------- ACK 6, ECHO 12

				<sometime later, data becomes available>

		  <------------------- DATA 37, ACK 6, ECHO 12

2) 	  DATA 3, TS 12 -------------------->  (again, long RTO)

		   (lost) --------------- ACK 4, ECHO 12, rcv window =3D 0

		  <---------------------- ACK 4, ECHO 12, rcv window =3D foo

	I think these situations are probably rare, and am willing to live with =
them, especially since it's still better than the current state of =
things.
	What could we do to avoid them?  The only thing that immediately comes =
to mind is breaking the rule that an endpoint must always include a =
tsval & echo if it's got the option negotiated.  Perhaps, add the =
algorithm
		if  (I already acked this DATA)
			Do not include ts echo     ??
and make sure the other side would know how to handle this. (Indeed, it =
would have no choice on how to handle this-- there's no data to =
manipulate!)
	I could see how this might cause backwards compatibility issues with =
many TCP implementations, although I haven't looked to see.  On the =
other hand, there's something to be said for getting it right the first =
(er, second) time.

-Martin Duke
                 =20

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Hi,

I've just sent an update to RFC 1323 to internet-drafts@ietf.org.
Below is a copy of the document.  Getting an update published and
advancing 1323 from draft to proposed is *way* overdue.  Perhaps we
can get any discussion done in the next couple of months, and then
it can be finished up for the Minneapolis IETF meeting.

			-David Borman

============






Network Working Group                              Network Working Group
Internet-Draft                                               V. Jacobson
Obsoletes: 1323                                            Packet Design
                                                               R. Braden
                                                                     ISI
                                                               D. Borman
                                                Wind River Systems, Inc.
                                                             August 2003


                  TCP Extensions for High Performance

Status of This Memo

   This document is an Internet-Draft and is subject to all provisions
   of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet- Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/1id-abstracts.html

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html

Abstract

   This memo presents a set of TCP extensions to improve performance
   over large bandwidth*delay product paths and to provide reliable
   operation over very high-speed paths.  It defines new TCP options for
   scaled windows and timestamps, which are designed to provide
   compatible interworking with TCP's that do not implement the
   extensions.  The timestamps are used for two distinct mechanisms:
   RTTM (Round Trip Time Measurement) and PAWS (Protect Against Wrapped
   Sequences).  Selective acknowledgments are not included in this memo.

   This memo updates and obsoletes RFC-1323, a Proposed Standard, with
   small clarifications and corrections.




Network Working Group     Expires February 2004                 [Page 1]

Internet-Draft     TCP Extensions for High Performance       August 2003



TABLE OF CONTENTS


   1.  Introduction                                                    2
   2.  TCP Window Scale Option                                         8
   3.  RTTM -- Round-Trip Time Measurement                            11
   4.  PAWS -- Protect Against Wrapped Sequence Numbers               17
   5.  Conclusions and Acknowledgments                                24
   6.  References                                                     25
   APPENDIX A: Implementation Suggestions                             27
   APPENDIX B: Duplicates from Earlier Connection Incarnations        27
   APPENDIX C: Changes from RFC-1072, RFC-1185, RFC-1323              30
   APPENDIX D: Summary of Notation                                    32
   APPENDIX E: Pseudo-code Summary                                    33
   APPENDIX F: Event Processing                                       35
   Security Considerations                                            41
   Authors' Addresses                                                 41

1. INTRODUCTION

   The TCP protocol [Postel81] was designed to operate reliably over
   almost any transmission medium regardless of transmission rate,
   delay, corruption, duplication, or reordering of segments.
   Production TCP implementations currently adapt to transfer rates in
   the range of 100 bps to 10**7 bps and round-trip delays in the range
   1 ms to 100 seconds.  Work on TCP performance has shown that TCP can
   work well over a variety of Internet paths, ranging from 800 Mbit/sec
   I/O channels to 300 bit/sec dial-up modems [Jacobson88a].

   The introduction of fiber optics is resulting in ever-higher
   transmission speeds, and the fastest paths are moving out of the
   domain for which TCP was originally engineered.  This memo defines a
   set of modest extensions to TCP to extend the domain of its
   application to match this increasing network capability.  It is based
   upon and obsoletes RFC-1072 [Jacobson88b] and RFC-1185 [Jacobson90b].

   There is no one-line answer to the question: "How fast can TCP go?".
   There are two separate kinds of issues, performance and reliability,
   and each depends upon different parameters.  We discuss each in turn.

   1.1  TCP Performance

      TCP performance depends not upon the transfer rate itself, but
      rather upon the product of the transfer rate and the round-trip
      delay.  This "bandwidth*delay product" measures the amount of data
      that would "fill the pipe"; it is the buffer space required at
      sender and receiver to obtain maximum throughput on the TCP
      connection over the path, i.e., the amount of unacknowledged data
      that TCP must handle in order to keep the pipeline full.  TCP



Network Working Group     Expires February 2004                 [Page 2]

Internet-Draft     TCP Extensions for High Performance       August 2003


      performance problems arise when the bandwidth*delay product is
      large.  We refer to an Internet path operating in this region as a
      "long, fat pipe", and a network containing this path as an "LFN"
      (pronounced "elephan(t)").

      High-capacity packet satellite channels (e.g., DARPA's Wideband
      Net) are LFN's.  For example, a DS1-speed satellite channel has a
      bandwidth*delay product of 10**6 bits or more; this corresponds to
      100 outstanding TCP segments of 1200 bytes each.  Terrestrial
      fiber-optical paths will also fall into the LFN class; for
      example, a cross-country delay of 30 ms at a DS3 bandwidth
      (45Mbps) also exceeds 10**6 bits.

      There are three fundamental performance problems with the current
      TCP over LFN paths:

      (1)  Window Size Limit

           The TCP header uses a 16 bit field to report the receive
           window size to the sender.  Therefore, the largest window
           that can be used is 2**16 = 65K bytes.

           To circumvent this problem, Section 2 of this memo defines a
           new TCP option, "Window Scale", to allow windows larger than
           2**16.  This option defines an implicit scale factor, which
           is used to multiply the window size value found in a TCP
           header to obtain the true window size.

      (2)  Recovery from Losses

           Packet losses in an LFN can have a catastrophic effect on
           throughput.  In the past, properly-operating TCP
           implementations would cause the data pipeline to drain with
           every packet loss, and require a slow-start action to
           recover.  The Fast Retransmit and Fast Recovery algorithms
           [Jacobson90c] [Stevens97] were introduced, and their combined
           effect was to recover from one packet loss per window,
           without draining the pipeline.  However, more than one packet
           loss per window typically resulted in a retransmission
           timeout and the resulting pipeline drain and slow start.

           Expanding the window size to match the capacity of an LFN
           results in a corresponding increase of the probability of
           more than one packet per window being dropped.  This could
           have a devastating effect upon the throughput of TCP over an
           LFN.  In addition, if a congestion control mechanism based
           upon some form of random dropping were introduced into
           gateways, randomly spaced packet drops would become common,
           possible increasing the probability of dropping more than one
           packet per window.



Network Working Group     Expires February 2004                 [Page 3]

Internet-Draft     TCP Extensions for High Performance       August 2003


           To generalize the Fast Retransmit/Fast Recovery mechanism to
           handle multiple packets dropped per window, selective
           acknowledgments are required.  Unlike the normal cumulative
           acknowledgments of TCP, selective acknowledgments give the
           sender a complete picture of which segments are queued at the
           receiver and which have not yet arrived.  Some evidence in
           favor of selective acknowledgments has been published
           [NBS85], and selective acknowledgments have been included in
           a number of experimental Internet protocols -- VMTP
           [Cheriton88], NETBLT [Clark87], and RDP [Velten84], and
           proposed for OSI TP4 [NBS85].  However, in the non-LFN
           regime, selective acknowledgments reduce the number of
           packets retransmitted but do not otherwise improve
           performance, making their complexity of questionable value.
           However, selective acknowledgments have become much more
           important in the LFN regime.

           RFC-1072 defined a new TCP "SACK" option to send a selective
           acknowledgment.  At the time that RFC-1323 was published,
           important technical issues still had to be worked out
           concerning both the format and semantics of the SACK option,
           so it was omitted from RFC-1323.  SACK has now been published
           as a separate document, RFC-2018 [Mathis96].

      (3)  Round-Trip Measurement

           TCP implements reliable data delivery by retransmitting
           segments that are not acknowledged within some retransmission
           timeout (RTO) interval.  Accurate dynamic determination of an
           appropriate RTO is essential to TCP performance.  RTO is
           determined by estimating the mean and variance of the
           measured round-trip time (RTT), i.e., the time interval
           between sending a segment and receiving an acknowledgment for
           it [Jacobson88a].

           Section 4 introduces a new TCP option, "Timestamps", and then
           defines a mechanism using this option that allows nearly
           every segment, including retransmissions, to be timed at
           negligible computational cost.  We use the mnemonic RTTM
           (Round Trip Time Measurement) for this mechanism, to
           distinguish it from other uses of the Timestamps option.


   1.2 TCP Reliability

      Now we turn from performance to reliability.  High transfer rate
      enters TCP performance through the bandwidth*delay product.
      However, high transfer rate alone can threaten TCP reliability by
      violating the assumptions behind the TCP mechanism for duplicate
      detection and sequencing.



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      An especially serious kind of error may result from an accidental
      reuse of TCP sequence numbers in data segments.  Suppose that an
      "old duplicate segment", e.g., a duplicate data segment that was
      delayed in Internet queues, is delivered to the receiver at the
      wrong moment, so that its sequence numbers falls somewhere within
      the current window.  There would be no checksum failure to warn of
      the error, and the result could be an undetected corruption of the
      data.  Reception of an old duplicate ACK segment at the
      transmitter could be only slightly less serious: it is likely to
      lock up the connection so that no further progress can be made,
      forcing an RST on the connection.

      TCP reliability depends upon the existence of a bound on the
      lifetime of a segment: the "Maximum Segment Lifetime" or MSL.  An
      MSL is generally required by any reliable transport protocol,
      since every sequence number field must be finite, and therefore
      any sequence number may eventually be reused.  In the Internet
      protocol suite, the MSL bound is enforced by an IP-layer
      mechanism, the "Time-to-Live" or TTL field.

      Duplication of sequence numbers might happen in either of two
      ways:

      (1)  Sequence number wrap-around on the current connection

           A TCP sequence number contains 32 bits.  At a high enough
           transfer rate, the 32-bit sequence space may be "wrapped"
           (cycled) within the time that a segment is delayed in queues.

      (2)  Earlier incarnation of the connection

           Suppose that a connection terminates, either by a proper
           close sequence or due to a host crash, and the same
           connection (i.e., using the same pair of sockets) is
           immediately reopened.  A delayed segment from the terminated
           connection could fall within the current window for the new
           incarnation and be accepted as valid.

      Duplicates from earlier incarnations, Case (2), are avoided by
      enforcing the current fixed MSL of the TCP spec, as explained in
      Section 5.3 and Appendix B.   However, case (1), avoiding the
      reuse of sequence numbers within the same connection, requires an
      MSL bound that depends upon the transfer rate, and at high enough
      rates, a new mechanism is required.

      More specifically, if the maximum effective bandwidth at which TCP
      is able to transmit over a particular path is B bytes per second,
      then the following constraint must be satisfied for error-free
      operation:




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          2**31 / B  > MSL (secs)                     [1]

      The following table shows the value for Twrap = 2**31/B in
      seconds, for some important values of the bandwidth B:

           Network       B*8          B         Twrap
                      bits/sec    bytes/sec     secs
           _______     _______     ______       ______

           ARPANET       56kbps       7KBps    3*10**5 (~3.6 days)

           DS1          1.5Mbps     190KBps    10**4 (~3 hours)

           Ethernet      10Mbps    1.25MBps    1700 (~30 mins)

           DS3           45Mbps     5.6MBps    380

           FDDI         100Mbps    12.5MBps    170

           Gigabit        1Gbps     125MBps    17

      It is clear that wrap-around of the sequence space is not a
      problem for 56kbps packet switching or even 10Mbps Ethernets.  On
      the other hand, at DS3 and FDDI speeds, Twrap is comparable to the
      2 minute MSL assumed by the TCP specification [Postel81].  Moving
      towards gigabit speeds, Twrap becomes too small for reliable
      enforcement by the Internet TTL mechanism.

      The 16-bit window field of TCP limits the effective bandwidth B to
      2**16/RTT, where RTT is the round-trip time in seconds
      [McKenzie89].  If the RTT is large enough, this limits B to a
      value that meets the constraint [1] for a large MSL value.  For
      example, consider a transcontinental backbone with an RTT of 60ms
      (set by the laws of physics).  With the bandwidth*delay product
      limited to 64KB by the TCP window size, B is then limited to
      1.1MBps, no matter how high the theoretical transfer rate of the
      path.  This corresponds to cycling the sequence number space in
      Twrap= 2000 secs, which is safe in today's Internet.

      It is important to understand that the culprit is not the larger
      window but rather the high bandwidth.  For example, consider a
      (very large) FDDI LAN with a diameter of 10km.  Using the speed of
      light, we can compute the RTT across the ring as
      (2*10**4)/(3*10**8) = 67 microseconds, and the delay*bandwidth
      product is then 833 bytes.  A TCP connection across this LAN using
      a window of only 833 bytes will run at the full 100mbps and can
      wrap the sequence space in about 3 minutes, very close to the MSL
      of TCP.  Thus, high speed alone can cause a reliability problem
      with sequence number wrap-around, even without extended windows.



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      Watson's Delta-T protocol [Watson81] includes network-layer
      mechanisms for precise enforcement of an MSL.  In contrast, the IP
      mechanism for MSL enforcement is loosely defined and even more
      loosely implemented in the Internet.  Therefore, it is unwise to
      depend upon active enforcement of MSL for TCP connections, and it
      is unrealistic to imagine setting MSL's smaller than the current
      values (e.g., 120 seconds specified for TCP).

      A possible fix for the problem of cycling the sequence space would
      be to increase the size of the TCP sequence number field.  For
      example, the sequence number field (and also the acknowledgment
      field) could be expanded to 64 bits.  This could be done either by
      changing the TCP header or by means of an additional option.

      Section 5 presents a different mechanism, which we call PAWS
      (Protect Against Wrapped Sequence numbers), to extend TCP
      reliability to transfer rates well beyond the foreseeable upper
      limit of network bandwidths.  PAWS uses the TCP Timestamps option
      defined in Section 4 to protect against old duplicates from the
      same connection.

   1.3 Using TCP options

      The extensions defined in this memo all use new TCP options.  We
      must address two possible issues concerning the use of TCP
      options: (1) compatibility and (2) overhead.

      We must pay careful attention to compatibility, i.e., to
      interoperation with existing implementations.  The only TCP option
      defined previously, MSS, may appear only on a SYN segment.  Every
      implementation should (and we expect that most will) ignore
      unknown options on SYN segments.  However, some buggy TCP
      implementation might be crashed by the first appearance of an
      option on a non-SYN segment.  Therefore, for each of the
      extensions defined below, TCP options will be sent on non-SYN
      segments only after an exchange of options on the the SYN segments
      has indicated that both sides understand the extension.
      Furthermore, an extension option will be sent in a <SYN,ACK>
      segment only if the corresponding option was received in the
      initial <SYN> segment.

      A question may be raised about the bandwidth and processing
      overhead for TCP options.  Those options that occur on SYN
      segments are not likely to cause a performance concern.  Opening a
      TCP connection requires execution of significant special-case
      code, and the processing of options is unlikely to increase that
      cost significantly.

      On the other hand, a Timestamps option may appear in any data or
      ACK segment, adding 12 bytes to the 20-byte TCP header.  We



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      believe that the bandwidth saved by reducing unnecessary
      retransmissions will more than pay for the extra header bandwidth.

      There is also an issue about the processing overhead for parsing
      the variable byte-aligned format of options, particularly with a
      RISC-architecture CPU.  Appendix A contains a recommended layout
      of the options in TCP headers to achieve reasonable data field
      alignment.  In the spirit of Header Prediction, a TCP can quickly
      test for this layout and if it is verified then use a fast path.
      Hosts that use this canonical layout will effectively use the
      options as a set of fixed-format fields appended to the TCP
      header.  However, to retain the philosophical and protocol
      framework of TCP options, a TCP must be prepared to parse an
      arbitrary options field, albeit with less efficiency.

      Finally, we observe that most of the mechanisms defined in this
      memo are important for LFN's and/or very high-speed networks.  For
      low-speed networks, it might be a performance optimization to NOT
      use these mechanisms.  A TCP vendor concerned about optimal
      performance over low-speed paths might consider turning these
      extensions off for low-speed paths, or allow a user or
      installation manager to disable them.


2. TCP WINDOW SCALE OPTION

   2.1  Introduction

      The window scale extension expands the definition of the TCP
      window to 32 bits and then uses a scale factor to carry this
      32-bit value in the 16-bit Window field of the TCP header (SEG.WND
      in RFC-793).  The scale factor is carried in a new TCP option,
      Window Scale.  This option is sent only in a SYN segment (a
      segment with the SYN bit on), hence the window scale is fixed in
      each direction when a connection is opened.  (Another design
      choice would be to specify the window scale in every TCP segment.
      It would be incorrect to send a window scale option only when the
      scale factor changed, since a TCP option in an acknowledgement
      segment will not be delivered reliably (unless the ACK happens to
      be piggy-backed on data in the other direction).  Fixing the scale
      when the connection is opened has the advantage of lower overhead
      but the disadvantage that the scale factor cannot be changed
      during the connection.)

      The maximum receive window, and therefore the scale factor, is
      determined by the maximum receive buffer space.  In a typical
      modern implementation, this maximum buffer space is set by default
      but can be overridden by a user program before a TCP connection is
      opened.  This determines the scale factor, and therefore no new
      user interface is needed for window scaling.



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   2.2  Window Scale Option

      The three-byte Window Scale option may be sent in a SYN segment by
      a TCP.  It has two purposes: (1) indicate that the TCP is prepared
      to do both send and receive window scaling, and (2) communicate a
      scale factor to be applied to its receive window.  Thus, a TCP
      that is prepared to scale windows should send the option, even if
      its own scale factor is 1.  The scale factor is limited to a power
      of two and encoded logarithmically, so it may be implemented by
      binary shift operations.


      TCP Window Scale Option (WSopt):

         Kind: 3

         Length: 3 bytes

                +---------+---------+---------+
                | Kind=3  |Length=3 |shift.cnt|
                +---------+---------+---------+


         This option is an offer, not a promise; both sides must send
         Window Scale options in their SYN segments to enable window
         scaling in either direction.  If window scaling is enabled,
         then the TCP that sent this option will right-shift its true
         receive-window values by 'shift.cnt' bits for transmission in
         SEG.WND.  The value 'shift.cnt' may be zero (offering to scale,
         while applying a scale factor of 1 to the receive window).

         This option may be sent in an initial <SYN> segment (i.e., a
         segment with the SYN bit on and the ACK bit off).  It may also
         be sent in a <SYN,ACK> segment, but only if a Window Scale
         option was received in the initial <SYN> segment.  A Window
         Scale option in a segment without a SYN bit should be ignored.

         The Window field in a SYN (i.e., a <SYN> or <SYN,ACK>) segment
         itself is never scaled.

   2.3  Using the Window Scale Option

      A model implementation of window scaling is as follows, using the
      notation of RFC-793 [Postel81]:

      *    All windows are treated as 32-bit quantities for storage in
           the connection control block and for local calculations.
           This includes the send-window (SND.WND) and the receive-
           window (RCV.WND) values, as well as the congestion window.




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      *    The connection state is augmented by two window shift counts,
           Snd.Wind.Scale and Rcv.Wind.Scale, to be applied to the
           incoming and outgoing window fields, respectively.

      *    If a TCP receives a <SYN> segment containing a Window Scale
           option, it sends its own Window Scale option in the <SYN,ACK>
           segment.

      *    The Window Scale option is sent with shift.cnt = R, where R
           is the value that the TCP would like to use for its receive
           window.

      *    Upon receiving a SYN segment with a Window Scale option
           containing shift.cnt = S, a TCP sets Snd.Wind.Scale to S and
           sets Rcv.Wind.Scale to R; otherwise, it sets both
           Snd.Wind.Scale and Rcv.Wind.Scale to zero.

      *    The window field (SEG.WND) in the header of every incoming
           segment, with the exception of SYN segments, is left-shifted
           by Snd.Wind.Scale bits before updating SND.WND:

              SND.WND = SEG.WND << Snd.Wind.Scale

           (assuming the other conditions of RFC793 are met, and using
           the "C" notation "<<" for left-shift).

      *    The window field (SEG.WND) of every outgoing segment, with
           the exception of SYN segments, is right-shifted by
           Rcv.Wind.Scale bits:

              SEG.WND = RCV.WND >> Rcv.Wind.Scale.


      TCP determines if a data segment is "old" or "new" by testing
      whether its sequence number is within 2**31 bytes of the left edge
      of the window, and if it is not, discarding the data as "old".  To
      insure that new data is never mistakenly considered old and vice-
      versa, the left edge of the sender's window has to be at most
      2**31 away from the right edge of the receiver's window.
      Similarly with the sender's right edge and receiver's left edge.
      Since the right and left edges of either the sender's or
      receiver's window differ by the window size, and since the sender
      and receiver windows can be out of phase by at most the window
      size, the above constraints imply that 2 * the max window size
      must be less than 2**31, or

           max window < 2**30

      Since the max window is 2**S (where S is the scaling shift count)
      times at most 2**16 - 1 (the maximum unscaled window), the maximum



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      window is guaranteed to be < 2*30 if S <= 14.  Thus, the shift
      count must be limited to 14 (which allows windows of 2**30 = 1
      Gbyte).  If a Window Scale option is received with a shift.cnt
      value exceeding 14, the TCP should log the error but use 14
      instead of the specified value.

      The scale factor applies only to the Window field as transmitted
      in the TCP header; each TCP using extended windows will maintain
      the window values locally as 32-bit numbers.  For example, the
      "congestion window" computed by Slow Start and Congestion
      Avoidance is not affected by the scale factor, so window scaling
      will not introduce quantization into the congestion window.

3.  RTTM: ROUND-TRIP TIME MEASUREMENT

   3.1  Introduction

      Accurate and current RTT estimates are necessary to adapt to
      changing traffic conditions and to avoid an instability known as
      "congestion collapse" [Nagle84] in a busy network.  However,
      accurate measurement of RTT may be difficult both in theory and in
      implementation.

      Many TCP implementations base their RTT measurements upon a sample
      of one packet per window or less.  While this yields an adequate
      approximation to the RTT for small windows, it results in an
      unacceptably poor RTT estimate for an LFN.  If we look at RTT
      estimation as a signal processing problem (which it is), a data
      signal at some frequency, the packet rate, is being sampled at a
      lower frequency, the window rate.  This lower sampling frequency
      violates Nyquist's criteria and may therefore introduce "aliasing"
      artifacts into the estimated RTT [Hamming77].

      A good RTT estimator with a conservative retransmission timeout
      calculation can tolerate aliasing when the sampling frequency is
      "close" to the data frequency.   For example, with a window of 8
      packets, the sample rate is 1/8 the data frequency -- less than an
      order of magnitude different.  However, when the window is tens or
      hundreds of packets, the RTT estimator may be seriously in error,
      resulting in spurious retransmissions.

      If there are dropped packets, the problem becomes worse.  Zhang
      [Zhang86], Jain [Jain86] and Karn [Karn87] have shown that it is
      not possible to accumulate reliable RTT estimates if retransmitted
      segments are included in the estimate.  Since a full window of
      data will have been transmitted prior to a retransmission, all of
      the segments in that window will have to be ACKed before the next
      RTT sample can be taken.  This means at least an additional
      window's worth of time between RTT measurements and, as the error
      rate approaches one per window of data (e.g., 10**-6 errors per



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      bit for the Wideband satellite network), it becomes effectively
      impossible to obtain a valid RTT measurement.

      A solution to these problems, which actually simplifies the sender
      substantially, is as follows: using TCP options, the sender places
      a timestamp in each data segment, and the receiver reflects these
      timestamps back in ACK segments.  Then a single subtract gives the
      sender an accurate RTT measurement for every ACK segment (which
      will correspond to every other data segment, with a sensible
      receiver).  We call this the RTTM (Round-Trip Time Measurement)
      mechanism.

      It is vitally important to use the RTTM mechanism with big
      windows; otherwise, the door is opened to some dangerous
      instabilities due to aliasing.  Furthermore, the option is
      probably useful for all TCP's, since it simplifies the sender.

   3.2  TCP Timestamps Option

      TCP is a symmetric protocol, allowing data to be sent at any time
      in either direction, and therefore timestamp echoing may occur in
      either direction.  For simplicity and symmetry, we specify that
      timestamps always be sent and echoed in both directions.  For
      efficiency, we combine the timestamp and timestamp reply fields
      into a single TCP Timestamps Option.

      TCP Timestamps Option (TSopt):

         Kind: 8

         Length: 10 bytes

          +-------+-------+---------------------+---------------------+
          |Kind=8 |  10   |   TS Value (TSval)  |TS Echo Reply (TSecr)|
          +-------+-------+---------------------+---------------------+
              1       1              4                     4

         The Timestamps option carries two four-byte timestamp fields.
         The Timestamp Value field (TSval) contains the current value of
         the timestamp clock of the TCP sending the option.

         The Timestamp Echo Reply field (TSecr) is only valid if the ACK
         bit is set in the TCP header; if it is valid, it echos a
         timestamp value that was sent by the remote TCP in the TSval
         field of a Timestamps option.  When TSecr is not valid, its
         value must be zero.  The TSecr value will generally be from the
         most recent Timestamp option that was received; however, there
         are exceptions that are explained below.

         A TCP may send the Timestamps option (TSopt) in an initial



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         <SYN> segment (i.e., a segment containing a SYN bit and no ACK
         bit), and may send a TSopt in other segments only if it
         received a TSopt in the initial <SYN> segment for the
         connection.

   3.3 The RTTM Mechanism

      RTTM places a Timestamps option in every segment, with a TSval
      that is obtained from a (virtual) "timestamp clock".  Values of
      this clock values must be at least approximately proportional to
      real time, in order to measure actual RTT.

      These TSval values are echoed in TSecr values in the reverse
      direction.  The difference between a received TSecr value and the
      current timestamp clock value provides an RTT measurement.

      When timestamps are used, every segment that is received will
      contain a TSecr value; however, these values cannot all be used to
      update the measured RTT.  The following example illustrates why.
      It shows a one-way data flow with segments arriving in sequence
      without loss.  Here A, B, C... represent data blocks occupying
      successive blocks of sequence numbers, and ACK(A),...  represent
      the corresponding cumulative acknowledgments.  The two timestamp
      fields of the Timestamps option are shown symbolically as <TSval=
      x,TSecr=y>.  Each TSecr field contains the value most recently
      received in a TSval field; these echoed values. labelled
      "TS.Recent", are shown in parentheses.

         TCP  A                                            TCP B

       (TS.Recent)                                        (TS.Recent)

      1.  (120)             <A,TSval=1,TSecr=120> --->       (1)

      2.  (125)  <--- <ACK(A),TSval=125,TSecr=1>             (1)

      3.  (125)             <B,TSval=6,TSecr=125> --->       (6)

      4.  (130)  <--- <ACK(B),TSval=130,TSecr=6>             (6)

             . . . ( Pause for 60 timestamp clock ticks ) . . . .


      5.  (130)             <C,TSval=1,TSecr=120> --->       (1)

      6.  (125)  <--- <ACK(A),TSval=125,TSecr=1>             (1)

      4.  (127)  <b,ACK(x),TSval=65,TSecr=127> --->  ...

      5.            ... <--- <y,ACK(A),TSval=191,TSecr=5>    (5)



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         TCP  A                                          TCP B

                        <A,TSval=1,TSecr=120> ------>

             <---- <ACK(A),TSval=127,TSecr=1>

                        <B,TSval=5,TSecr=127> ------>

             <---- <ACK(B),TSval=131,TSecr=5>

             . . . . . . . . . . . . . . . . . . . . . .

                        <C,TSval=65,TSecr=131> ------>

             <---- <ACK(C),TSval=191,TSecr=65>

                        (etc)


      The dotted line marks a pause (60 time units long) in which A had
      nothing to send.  Note that this pause inflates the RTT which B
      could infer from receiving TSecr=131 in data segment C.  Thus, in
      one-way data flows, RTTM in the reverse direction measures a value
      that is inflated by gaps in sending data.  However, the following
      rule prevents a resulting inflation of the measured RTT:

           RTTM Rule: A TSecr value received in a segment is used to
           update the averaged RTT measurement only if the segment
           acknowledges some new data, i.e., only if it advances the
           left edge of the send window.

      Since TCP B is not sending data, the data segment C does not
      acknowledge any new data when it arrives at B.  Thus, the inflated
      RTTM measurement is not used to update B's RTTM measurement.

   3.4  Which Timestamp to Echo

      If more than one Timestamps option is received before a reply
      segment is sent, the TCP must choose only one of the TSvals to
      echo, ignoring the others.  To minimize the state kept in the
      receiver (i.e., the number of unprocessed TSvals), the receiver
      should be required to retain at most one timestamp in the
      connection control block.

      There are three situations to consider:

      (A)  Delayed ACKs.



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           Many TCP's acknowledge only every Kth segment out of a group
           of segments arriving within a short time interval; this
           policy is known generally as "delayed ACKs".  The data-sender
           TCP must measure the effective RTT, including the additional
           time due to delayed ACKs, or else it will retransmit
           unnecessarily.  Thus, when delayed ACKs are in use, the
           receiver should reply with the TSval field from the earliest
           unacknowledged segment.

      (B)  A hole in the sequence space (segment(s) have been lost).

           The sender will continue sending until the window is filled,
           and the receiver may be generating ACKs as these out-of-order
           segments arrive (e.g., to aid "fast retransmit").

           The lost segment is probably a sign of congestion, and in
           that situation the sender should be conservative about
           retransmission.  Furthermore, it is better to overestimate
           than underestimate the RTT.  An ACK for an out-of-order
           segment should therefore contain the timestamp from the most
           recent segment that advanced the window.

           The same situation occurs if segments are re-ordered by the
           network.

      (C)  A filled hole in the sequence space.

           The segment that fills the hole represents the most recent
           measurement of the network characteristics.  On the other
           hand, an RTT computed from an earlier segment would probably
           include the sender's retransmit time-out, badly biasing the
           sender's average RTT estimate.  Thus, the timestamp from the
           latest segment (which filled the hole) must be echoed.

      An algorithm that covers all three cases is described in the
      following rules for Timestamps option processing on a synchronized
      connection:

      (1)  The connection state is augmented with two 32-bit slots:
           TS.Recent holds a timestamp to be echoed in TSecr whenever a
           segment is sent, and Last.ACK.sent holds the ACK field from
           the last segment sent.  Last.ACK.sent will equal RCV.NXT
           except when ACKs have been delayed.

      (2)  If:

              SEG.TSval >= TSrecent and SEG.SEQ <= Last.ACK.sent

           then SEG.TSval is copied to TS.Recent; otherwise, it is
           ignored.



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      (3)  When a TSopt is sent, its TSecr field is set to the current
           TS.Recent value.

      The following examples illustrate these rules.  Here A, B, C...
      represent data segments occupying successive blocks of sequence
      numbers, and ACK(A),...  represent the corresponding
      acknowledgment segments.  Note that ACK(A) has the same sequence
      number as B.  We show only one direction of timestamp echoing, for
      clarity.

      o    Packets arrive in sequence, and some of the ACKs are delayed.

           By Case (A), the timestamp from the oldest unacknowledged
           segment is echoed.

                                                      TS.Recent
                    <A, TSval=1> ------------------->
                                                          1
                    <B, TSval=2> ------------------->
                                                          1
                    <C, TSval=3> ------------------->
                                                          1
                             <---- <ACK(C), TSecr=1>
                    (etc)

      o    Packets arrive out of order, and every packet is
           acknowledged.

           By Case (B), the timestamp from the last segment that
           advanced the left window edge is echoed, until the missing
           segment arrives; it is echoed according to Case (C).  The
           same sequence would occur if segments B and D were lost and
           retransmitted..




















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                                                      TS.Recent
                    <A, TSval=1> ------------------->
                                                          1
                             <---- <ACK(A), TSecr=1>
                                                          1
                    <C, TSval=3> ------------------->
                                                          1
                             <---- <ACK(A), TSecr=1>
                                                          1
                    <B, TSval=2> ------------------->
                                                          2
                             <---- <ACK(C), TSecr=2>
                                                          2
                    <E, TSval=5> ------------------->
                                                          2
                             <---- <ACK(C), TSecr=2>
                                                          2
                    <D, TSval=4> ------------------->
                                                          4
                             <---- <ACK(E), TSecr=4>
                    (etc)





4.  PAWS: PROTECT AGAINST WRAPPED SEQUENCE NUMBERS

   4.1  Introduction

      Section 4.2 describes a simple mechanism to reject old duplicate
      segments that might corrupt an open TCP connection; we call this
      mechanism PAWS (Protect Against Wrapped Sequence numbers).  PAWS
      operates within a single TCP connection, using state that is saved
      in the connection control block.  Section 4.3 and Appendix C
      discuss the implications of the PAWS mechanism for avoiding old
      duplicates from previous incarnations of the same connection.

   4.2  The PAWS Mechanism

      PAWS uses the same TCP Timestamps option as the RTTM mechanism
      described earlier, and assumes that every received TCP segment
      (including data and ACK segments) contains a timestamp SEG.TSval
      whose values are monotone non-decreasing in time.  The basic idea
      is that a segment can be discarded as an old duplicate if it is
      received with a timestamp SEG.TSval less than some timestamp
      recently received on this connection.

      In both the PAWS and the RTTM mechanism, the "timestamps" are
      32-bit unsigned integers in a modular 32-bit space.  Thus, "less



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      than" is defined the same way it is for TCP sequence numbers, and
      the same implementation techniques apply.  If s and t are
      timestamp values, s < t if 0 < (t - s) < 2**31, computed in
      unsigned 32-bit arithmetic.

      The choice of incoming timestamps to be saved for this comparison
      must guarantee a value that is monotone increasing.  For example,
      we might save the timestamp from the segment that last advanced
      the left edge of the receive window, i.e., the most recent in-
      sequence segment.  Instead, we choose the value TS.Recent
      introduced in Section 3.4 for the RTTM mechanism, since using a
      common value for both PAWS and RTTM simplifies the implementation
      of both.  As Section 3.4 explained, TS.Recent differs from the
      timestamp from the last in-sequence segment only in the case of
      delayed ACKs, and therefore by less than one window.  Either
      choice will therefore protect against sequence number wrap-around.

      RTTM was specified in a symmetrical manner, so that TSval
      timestamps are carried in both data and ACK segments and are
      echoed in TSecr fields carried in returning ACK or data segments.
      PAWS submits all incoming segments to the same test, and therefore
      protects against duplicate ACK segments as well as data segments.
      (An alternative un-symmetric algorithm would protect against old
      duplicate ACKs: the sender of data would reject incoming ACK
      segments whose TSecr values were less than the TSecr saved from
      the last segment whose ACK field advanced the left edge of the
      send window.  This algorithm was deemed to lack economy of
      mechanism and symmetry.)

      TSval timestamps sent on {SYN} and {SYN,ACK} segments are used to
      initialize PAWS.  PAWS protects against old duplicate non-SYN
      segments, and duplicate SYN segments received while there is a
      synchronized connection.  Duplicate {SYN} and {SYN,ACK} segments
      received when there is no connection will be discarded by the
      normal 3-way handshake and sequence number checks of TCP.

      It is recommended that RST segments NOT carry timestamps, and that
      RST segments be acceptable regardless of their timestamp.  Old
      duplicate RST segments should be exceedingly unlikely, and their
      cleanup function should take precedence over timestamps.

      4.2.1  Basic PAWS Algorithm

         The PAWS algorithm requires the following processing to be
         performed on all incoming segments for a synchronized
         connection:

         R1)  If there is a Timestamps option in the arriving segment
              and SEG.TSval < TS.Recent and if TS.Recent is valid (see
              later discussion), then treat the arriving segment as not



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              acceptable:

                   Send an acknowledgement in reply as specified in
                   RFC-793 page 69 and drop the segment.

                   Note: it is necessary to send an ACK segment in order
                   to retain TCP's mechanisms for detecting and
                   recovering from half-open connections.  For example,
                   see Figure 10 of RFC-793.


         R2)  If the segment is outside the window, reject it (normal
              TCP processing)

         R3)  If an arriving segment satisfies: SEG.SEQ <= Last.ACK.sent
              (see Section 3.4), then record its timestamp in TS.Recent.

         R4)  If an arriving segment is in-sequence (i.e., at the left
              window edge), then accept it normally.

         R5)  Otherwise, treat the segment as a normal in-window, out-
              of-sequence TCP segment (e.g., queue it for later delivery
              to the user).

         Steps R2, R4, and R5 are the normal TCP processing steps
         specified by RFC-793.

         It is important to note that the timestamp is checked only when
         a segment first arrives at the receiver, regardless of whether
         it is in-sequence or it must be queued for later delivery.
         Consider the following example.

              Suppose the segment sequence: A.1, B.1, C.1, ..., Z.1 has
              been sent, where the letter indicates the sequence number
              and the digit represents the timestamp.  Suppose also that
              segment B.1 has been lost.  The timestamp in TS.TStamp is
              1 (from A.1), so C.1, ..., Z.1 are considered acceptable
              and are queued.  When B is retransmitted as segment B.2
              (using the latest timestamp), it fills the hole and causes
              all the segments through Z to be acknowledged and passed
              to the user.  The timestamps of the queued segments are
              *not* inspected again at this time, since they have
              already been accepted.  When B.2 is accepted, TS.Stamp is
              set to 2.

         This rule allows reasonable performance under loss.  A full
         window of data is in transit at all times, and after a loss a
         full window less one packet will show up out-of-sequence to be
         queued at the receiver (e.g., up to ~2**30 bytes of data); the
         timestamp option must not result in discarding this data.



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         In certain unlikely circumstances, the algorithm of rules R1-R4
         could lead to discarding some segments unnecessarily, as shown
         in the following example:

              Suppose again that segments: A.1, B.1, C.1, ..., Z.1 have
              been sent in sequence and that segment B.1 has been lost.
              Furthermore, suppose delivery of some of C.1, ... Z.1 is
              delayed until AFTER the retransmission B.2 arrives at the
              receiver.  These delayed segments will be discarded
              unnecessarily when they do arrive, since their timestamps
              are now out of date.

         This case is very unlikely to occur.  If the retransmission was
         triggered by a timeout, some of the segments C.1, ... Z.1 must
         have been delayed longer than the RTO time.  This is presumably
         an unlikely event, or there would be many spurious timeouts and
         retransmissions.  If B's retransmission was triggered by the
         "fast retransmit" algorithm, i.e., by duplicate ACKs, then the
         queued segments that caused these ACKs must have been received
         already.

         Even if a segment were delayed past the RTO, the Fast
         Retransmit mechanism [Jacobson90c] will cause the delayed
         packets to be retransmitted at the same time as B.2, avoiding
         an extra RTT and therefore causing a very small performance
         penalty.

         We know of no case with a significant probability of occurrence
         in which timestamps will cause performance degradation by
         unnecessarily discarding segments.

      4.2.2  Timestamp Clock

         It is important to understand that the PAWS algorithm does not
         require clock synchronization between sender and receiver.  The
         sender's timestamp clock is used to stamp the segments, and the
         sender uses the echoed timestamp to measure RTT's.  However,
         the receiver treats the timestamp as simply a monotone-
         increasing serial number, without any necessary connection to
         its clock.  From the receiver's viewpoint, the timestamp is
         acting as a logical extension of the high-order bits of the
         sequence number.

         The receiver algorithm does place some requirements on the
         frequency of the timestamp clock.

         (a)  The timestamp clock must not be "too slow".

              It must tick at least once for each 2**31 bytes sent.  In
              fact, in order to be useful to the sender for round trip



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              timing, the clock should tick at least once per window's
              worth of data, and even with the RFC-1072 window
              extension, 2**31 bytes must be at least two windows.

              To make this more quantitative, any clock faster than 1
              tick/sec will reject old duplicate segments for link
              speeds of ~8 Gbps.  A 1ms timestamp clock will work at
              link speeds up to 8 Tbps (8*10**12) bps!

         (b)  The timestamp clock must not be "too fast".

              Its recycling time must be greater than MSL seconds.
              Since the clock (timestamp) is 32 bits and the worst-case
              MSL is 255 seconds, the maximum acceptable clock frequency
              is one tick every 59 ns.

              However, it is desirable to establish a much longer
              recycle period, in order to handle outdated timestamps on
              idle connections (see Section 4.2.3), and to relax the MSL
              requirement for preventing sequence number wrap-around.
              With a 1 ms timestamp clock, the 32-bit timestamp will
              wrap its sign bit in 24.8 days.  Thus, it will reject old
              duplicates on the same connection if MSL is 24.8 days or
              less.  This appears to be a very safe figure; an MSL of
              24.8 days or longer can probably be assumed by the gateway
              system without requiring precise MSL enforcement by the
              TTL value in the IP layer.

         Based upon these considerations, we choose a timestamp clock
         frequency in the range 1 ms to 1 sec per tick.  This range also
         matches the requirements of the RTTM mechanism, which does not
         need much more resolution than the granularity of the
         retransmit timer, e.g., tens or hundreds of milliseconds.

         The PAWS mechanism also puts a strong monotonicity requirement
         on the sender's timestamp clock.  The method of implementation
         of the timestamp clock to meet this requirement depends upon
         the system hardware and software.

         *    Some hosts have a hardware clock that is guaranteed to be
              monotonic between hardware resets.

         *    A clock interrupt may be used to simply increment a binary
              integer by 1 periodically.

         *    The timestamp clock may be derived from a system clock
              that is subject to being abruptly changed, by adding a
              variable offset value.  This offset is initialized to
              zero.  When a new timestamp clock value is needed, the
              offset can be adjusted as necessary to make the new value



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              equal to or larger than the previous value (which was
              saved for this purpose).


      4.2.3  Outdated Timestamps

         If a connection remains idle long enough for the timestamp
         clock of the other TCP to wrap its sign bit, then the value
         saved in TS.Recent will become too old; as a result, the PAWS
         mechanism will cause all subsequent segments to be rejected,
         freezing the connection (until the timestamp clock wraps its
         sign bit again).

         With the chosen range of timestamp clock frequencies (1 sec to
         1 ms), the time to wrap the sign bit will be between 24.8 days
         and 24800 days.  A TCP connection that is idle for more than 24
         days and then comes to life is exceedingly unusual.  However,
         it is undesirable in principle to place any limitation on TCP
         connection lifetimes.

         We therefore require that an implementation of PAWS include a
         mechanism to "invalidate" the TS.Recent value when a connection
         is idle for more than 24 days.  (An alternative solution to the
         problem of outdated timestamps would be to send keepalive
         segments at a very low rate, but still more often than the
         wrap-around time for timestamps, e.g., once a day.  This would
         impose negligible overhead.  However, the TCP specification has
         never included keepalives, so the solution based upon
         invalidation was chosen.)

         Note that a TCP does not know the frequency, and therefore, the
         wraparound time, of the other TCP, so it must assume the worst.
         The validity of TS.Recent needs to be checked only if the basic
         PAWS timestamp check fails, i.e., only if SEG.TSval <
         TS.Recent.  If TS.Recent is found to be invalid, then the
         segment is accepted, regardless of the failure of the timestamp
         check, and rule R3 updates TS.Recent with the TSval from the
         new segment.

         To detect how long the connection has been idle, the TCP may
         update a clock or timestamp value associated with the
         connection whenever TS.Recent is updated, for example.  The
         details will be implementation-dependent.

      4.2.4  Header Prediction

         "Header prediction" [Jacobson90a] is a high-performance
         transport protocol implementation technique that is most
         important for high-speed links.  This technique optimizes the
         code for the most common case, receiving a segment correctly



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         and in order.  Using header prediction, the receiver asks the
         question, "Is this segment the next in sequence?"  This
         question can be answered in fewer machine instructions than the
         question, "Is this segment within the window?"

         Adding header prediction to our timestamp procedure leads to
         the following recommended sequence for processing an arriving
         TCP segment:

         H1)  Check timestamp (same as step R1 above)

         H2)  Do header prediction: if segment is next in sequence and
              if there are no special conditions requiring additional
              processing, accept the segment, record its timestamp, and
              skip H3.

         H3)  Process the segment normally, as specified in RFC-793.
              This includes dropping segments that are outside the
              window and possibly sending acknowledgments, and queueing
              in-window, out-of-sequence segments.


         Another possibility would be to interchange steps H1 and H2,
         i.e., to perform the header prediction step H2 FIRST, and
         perform H1 and H3 only when header prediction fails.  This
         could be a performance improvement, since the timestamp check
         in step H1 is very unlikely to fail, and it requires interval
         arithmetic on a finite field, a relatively expensive operation.
         To perform this check on every single segment is contrary to
         the philosophy of header prediction.  We believe that this
         change might reduce CPU time for TCP protocol processing by up
         to 5-10% on high-speed networks.

         However, putting H2 first would create a hazard: a segment from
         2**32 bytes in the past might arrive at exactly the wrong time
         and be accepted mistakenly by the header-prediction step.  The
         following reasoning has been introduced [Jacobson90b] to show
         that the probability of this failure is negligible.

              If all segments are equally likely to show up as old
              duplicates, then the probability of an old duplicate
              exactly matching the left window edge is the maximum
              segment size (MSS) divided by the size of the sequence
              space.  This ratio must be less than 2**-16, since MSS
              must be < 2**16; for example, it will be (2**12)/(2**32) =
              2**-20 for an FDDI link.  However, the older a segment is,
              the less likely it is to be retained in the Internet, and
              under any reasonable model of segment lifetime the
              probability of an old duplicate exactly at the left window
              edge must be much smaller than 2**-16.



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              The 16 bit TCP checksum also allows a basic unreliability
              of one part in 2**16.  A protocol mechanism whose
              reliability exceeds the reliability of the TCP checksum
              should be considered "good enough", i.e., it won't
              contribute significantly to the overall error rate.  We
              therefore believe we can ignore the problem of an old
              duplicate being accepted by doing header prediction before
              checking the timestamp.

         However, this probabilistic argument is not universally
         accepted, and the consensus at present is that the performance
         gain does not justify the hazard in the general case.  It is
         therefore recommended that H2 follow H1.


   4.3.  Duplicates from Earlier Incarnations of Connection

      The PAWS mechanism protects against errors due to sequence number
      wrap-around on high-speed connection.  Segments from an earlier
      incarnation of the same connection are also a potential cause of
      old duplicate errors.  In both cases, the TCP mechanisms to
      prevent such errors depend upon the enforcement of a maximum
      segment lifetime (MSL) by the Internet (IP) layer (see Appendix of
      RFC-1185 for a detailed discussion).  Unlike the case of sequence
      space wrap-around, the MSL required to prevent old duplicate
      errors from earlier incarnations does not depend upon the transfer
      rate.  If the IP layer enforces the recommended 2 minute MSL of
      TCP, and if the TCP rules are followed, TCP connections will be
      safe from earlier incarnations, no matter how high the network
      speed.  Thus, the PAWS mechanism is not required for this case.

      We may still ask whether the PAWS mechanism can provide additional
      security against old duplicates from earlier connections, allowing
      us to relax the enforcement of MSL by the IP layer.  Appendix B
      explores this question, showing that further assumptions and/or
      mechanisms are required, beyond those of PAWS.  This is not part
      of the current extension.


5.  CONCLUSIONS AND ACKNOWLEDGMENTS

   This memo presented a set of extensions to TCP to provide efficient
   operation over large-bandwidth*delay-product paths and reliable
   operation over very high-speed paths.  These extensions are designed
   to provide compatible interworking with TCP's that do not implement
   the extensions.

   These mechanisms are implemented using new TCP options for scaled
   windows and timestamps.  The timestamps are used for two distinct
   mechanisms: RTTM (Round Trip Time Measurement) and PAWS (Protect



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   Against Wrapped Sequences).

   The Window Scale option was originally suggested by Mike St. Johns of
   USAF/DCA.  The present form of the option was suggested by Mike
   Karels of UC Berkeley in response to a more cumbersome scheme defined
   by Van Jacobson.  Lixia Zhang helped formulate the PAWS mechanism
   description in RFC-1185.

   Finally, much of this work originated as the result of discussions
   within the End-to-End Task Force on the theoretical limitations of
   transport protocols in general and TCP in particular.  Task force
   members and other on the end2end-interest list have made valuable
   contributions by pointing out flaws in the algorithms and the
   documentation.  Continued discussion and development since the
   publication of RFC 1323 has occurred in the IETF TCP Large Windows
   Working Group.  The authors are grateful for all these contributions.

6.  REFERENCES

      [Braden89] Braden, R., editor, "Requirements for Internet Hosts --
      Communication Layers", RFC 1122, October, 1989 [Clark87]  Clark,
      D., Lambert, M., and L. Zhang, "NETBLT: A Bulk Data Transfer
      Protocol", RFC 998, MIT, March 1987.

      [Garlick77]  Garlick, L., R. Rom, and J. Postel, "Issues in
      Reliable Host-to-Host Protocols", Proc. Second Berkeley Workshop
      on Distributed Data Management and Computer Networks, May 1977.

      [Hamming77]  Hamming, R., "Digital Filters", ISBN 0-13-212571-4,
      Prentice Hall, Englewood Cliffs, N.J., 1977.

      [Cheriton88]  Cheriton, D., "VMTP: Versatile Message Transaction
      Protocol", RFC 1045, Stanford University, February 1988.

      [Jacobson88a] Jacobson, V., "Congestion Avoidance and Control",
      SIGCOMM '88, Stanford, CA., August 1988.

      [Jacobson88b]  Jacobson, V., and R. Braden, "TCP Extensions for
      Long-Delay Paths", RFC-1072, LBL and USC/Information Sciences
      Institute, October 1988.

      [Jacobson90a]  Jacobson, V., "4BSD Header Prediction", ACM
      Computer Communication Review, April 1990.

      [Jacobson90b]  Jacobson, V., Braden, R., and Zhang, L., "TCP
      Extension for High-Speed Paths", RFC-1185, LBL and USC/Information
      Sciences Institute, October 1990.

      [Jacobson90c]  Jacobson, V., "Modified TCP congestion avoidance
      algorithm", Message to end2end-interest mailing list, April 1990.



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      [Jain86]  Jain, R., "Divergence of Timeout Algorithms for Packet
      Retransmissions", Proc. Fifth Phoenix Conf. on Comp. and Comm.,
      Scottsdale, Arizona, March 1986.

      [Karn87]  Karn, P. and C. Partridge, "Estimating Round-Trip Times
      in Reliable Transport Protocols", Proc. SIGCOMM '87, Stowe, VT,
      August 1987.

      [Mathis96] Mathis, M., Mahdavi, J., Floyd, S., and Romanow, A.,
      "TCP Selective Acknowledgment Options", RFC 2018, October, 1996.

      [McKenzie89]  McKenzie, A., "A Problem with the TCP Big Window
      Option", RFC 1110, BBN STC, August 1989.

      [Nagle84]  Nagle, J., "Congestion Control in IP/TCP
      Internetworks", RFC 896, FACC, January 1984.

      [NBS85]  Colella, R., Aronoff, R., and K. Mills, "Performance
      Improvements for ISO Transport", Ninth Data Comm Symposium,
      published in ACM SIGCOMM Comp Comm Review, vol. 15, no. 5,
      September 1985.

      [Postel81]  Postel, J., "Transmission Control Protocol - DARPA
      Internet Program Protocol Specification", RFC 793, DARPA,
      September 1981.

      [Postel83]  Postel, J., "The TCP Maximum Segment Size and Related
      Topics", RFC 879, ISI, November 1983.  [Stevens97] Stevens, W.,
      "TCP Slow Start, Congestion Avoidance, Fast Retransmit, and Fast
      Recovery Algorithms", RFC 2001, NOAO, January 1997.

      [Velten84] Velten, D., Hinden, R., and J. Sax, "Reliable Data
      Protocol", RFC 908, BBN, July 1984.

      [Watson81]  Watson, R., "Timer-based Mechanisms in Reliable
      Transport Protocol Connection Management", Computer Networks, Vol.
      5, 1981.

      [Zhang86]  Zhang, L., "Why TCP Timers Don't Work Well", Proc.
      SIGCOMM '86, Stowe, Vt., August 1986.













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APPENDIX A:  IMPLEMENTATION SUGGESTIONS


   TCP Option Layout

        The following layouts are recommended for sending options on
        non-SYN segments, to achieve maximum feasible alignment of
        32-bit and 64-bit machines.


            +--------+--------+--------+--------+
            |   NOP  |  NOP   |  TSopt |   10   |
            +--------+--------+--------+--------+
            |           TSval  timestamp        |
            +--------+--------+--------+--------+
            |           TSecr  timestamp        |
            +--------+--------+--------+--------+

   Interaction with the TCP Urgent Pointer

        The TCP Urgent pointer, like the TCP window, is a 16 bit value.
        Some of the original discussion for the TCP Window Scale option
        included proposals to increase the Urgent pointer to 32 bits.
        As it turns out, this is unnecessary.  There are two
        observations that should be made:

        (1)  With IP Version 4, the largest amount of TCP data that can
             be sent in a single packet is 65495 bytes (64K - 1 - size
             of fixed IP and TCP headers).

        (2)  Updates to the urgent pointer while the user is in "urgent
             mode" are invisible to the user.

        This means that if the Urgent Pointer points beyond the end of
        the TCP data in the current packet, then the user will remain in
        urgent mode until the next TCP packet arrives.  That packet will
        update the urgent pointer to a new offset, and the user will
        never have left urgent mode.

        Thus, to properly implement the Urgent Pointer, the sending TCP
        only has to check for overflow of the 16 bit Urgent Pointer
        field before filling it in.  If it does overflow, than a value
        of 65535 should be inserted into the Urgent Pointer.

   TCP Options and MSS

        There has been some confusion as to what value should be filled
        in the TCP MSS option when using TCP options.  RFC-879



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        [Postel83] stated:

             The MSS counts only data octets in the segment, it does not
             count the TCP header or the IP header.

        which is unclear about what to do about TCP options.  RFC-1122
        [Braden89] attempted to clarify this in section 4.2.2.6, but
        there still seems to be confusion.

        So, the MSS value to be sent in an MSS option should be equal to
        the effective MTU minus the fixed IP and TCP headers.  Since
        both IP and TCP options are ignored when calculating the value
        for the MSS option, if there are any IP or TCP options to be
        sent in a packet, then the sender must decrease the size of the
        TCP data accordingly.  The reason for this can be seen in the
        following table:

                         +--------------------+--------------------+
                         | MSS is adjusted    | MSS isn't adjusted |
                         | to include options | to include options |
        +----------------+--------------------+--------------------+
        | Sender adjusts | Packets are too    | Packets are the    |
        | length for     | short              | correct length     |
        | options        |                    |                    |
        +----------------+--------------------+--------------------+
        | Sender doesn't | Packets are the    | Packets are too    |
        | adjust length  | correct length     | long.              |
        | for options    |                    |                    |
        +----------------+--------------------+--------------------+

        Since the goal is to not send IP datagrams that have to be
        fragmented, and packets sent with the constraints in the lower
        right of this grid will cause IP fragmentation, the only way to
        guarantee that this doesn't happen is for the data sender to
        decrease the TCP data length by the size of the IP and TCP
        options.  And since the sender will be adjusting the TCP data
        length when sending IP and TCP options, there is no need to
        include the IP and TCP option lengths in the MSS value.


APPENDIX B: DUPLICATES FROM EARLIER CONNECTION INCARNATIONS

   There are two cases to be considered:  (1) a system crashing (and
   losing connection state) and restarting, and (2) the same connection
   being closed and reopened without a loss of host state.  These will
   be described in the following two sections.


   B.1  System Crash with Loss of State




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      TCP's quiet time of one MSL upon system startup handles the loss
      of connection state in a system crash/restart.  For an
      explanation, see for example "When to Keep Quiet" in the TCP
      protocol specification [Postel81].  The MSL that is required here
      does not depend upon the transfer speed.  The current TCP MSL of 2
      minutes seems acceptable as an operational compromise, as many
      host systems take this long to boot after a crash.

      However, the timestamp option may be used to ease the MSL
      requirements (or to provide additional security against data
      corruption).  If timestamps are being used and if the timestamp
      clock can be guaranteed to be monotonic over a system
      crash/restart, i.e., if the first value of the sender's timestamp
      clock after a crash/restart can be guaranteed to be greater than
      the last value before the restart, then a quiet time will be
      unnecessary.

      To dispense totally with the quiet time would require that the
      host clock be synchronized to a time source that is stable over
      the crash/restart period, with an accuracy of one timestamp clock
      tick or better.  We can back off from this strict requirement to
      take advantage of approximate clock synchronization.  Suppose that
      the clock is always re-synchronized to within N timestamp clock
      ticks and that booting (extended with a quiet time, if necessary)
      takes more than N ticks.  This will guarantee monotonicity of the
      timestamps, which can then be used to reject old duplicates even
      without an enforced MSL.

   B.2  Closing and Reopening a Connection

      When a TCP connection is closed, a delay of 2*MSL in TIME-WAIT
      state ties up the socket pair for 4 minutes (see Section 3.5 of
      [Postel81].  Applications built upon TCP that close one connection
      and open a new one (e.g., an FTP data transfer connection using
      Stream mode) must choose a new socket pair each time.  The TIME-
      WAIT delay serves two different purposes:

      (a)  Implement the full-duplex reliable close handshake of TCP.

           The proper time to delay the final close step is not really
           related to the MSL; it depends instead upon the RTO for the
           FIN segments and therefore upon the RTT of the path.  (It
           could be argued that the side that is sending a FIN knows
           what degree of reliability it needs, and therefore it should
           be able to determine the length of the TIME-WAIT delay for
           the FIN's recipient.  This could be accomplished with an
           appropriate TCP option in FIN segments.)

           Although there is no formal upper-bound on RTT, common
           network engineering practice makes an RTT greater than 1



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           minute very unlikely.  Thus, the 4 minute delay in TIME-WAIT
           state works satisfactorily to provide a reliable full-duplex
           TCP close.  Note again that this is independent of MSL
           enforcement and network speed.

           The TIME-WAIT state could cause an indirect performance
           problem if an application needed to repeatedly close one
           connection and open another at a very high frequency, since
           the number of available TCP ports on a host is less than
           2**16.  However, high network speeds are not the major
           contributor to this problem; the RTT is the limiting factor
           in how quickly connections can be opened and closed.
           Therefore, this problem will be no worse at high transfer
           speeds.

      (b)  Allow old duplicate segments to expire.

           To replace this function of TIME-WAIT state, a mechanism
           would have to operate across connections.  PAWS is defined
           strictly within a single connection; the last timestamp is
           TS.Recent is kept in the connection control block, and
           discarded when a connection is closed.

           An additional mechanism could be added to the TCP, a per-host
           cache of the last timestamp received from any connection.
           This value could then be used in the PAWS mechanism to reject
           old duplicate segments from earlier incarnations of the
           connection, if the timestamp clock can be guaranteed to have
           ticked at least once since the old connection was open.  This
           would require that the TIME-WAIT delay plus the RTT together
           must be at least one tick of the sender's timestamp clock.
           Such an extension is not part of the proposal of this RFC.

           Note that this is a variant on the mechanism proposed by
           Garlick, Rom, and Postel [Garlick77], which required each
           host to maintain connection records containing the highest
           sequence numbers on every connection.  Using timestamps
           instead, it is only necessary to keep one quantity per remote
           host, regardless of the number of simultaneous connections to
           that host.













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APPENDIX C: CHANGES FROM RFC-1072, RFC-1185, RFC-1323

   The protocol extensions defined in RFC-1323 document differ in
   several important ways from those defined in RFC-1072 and RFC-1185.

   (a)  SACK has been split off into a separate document, RFC 2018
        [Mathis96].

   (b)  The detailed rules for sending timestamp replies (see Section
        3.4) differ in important ways.  The earlier rules could result
        in an under-estimate of the RTT in certain cases (packets
        dropped or out of order).

   (c)  The same value TS.Recent is now shared by the two distinct
        mechanisms RTTM and PAWS.  This simplification became possible
        because of change (b).

   (d)  An ambiguity in RFC-1185 was resolved in favor of putting
        timestamps on ACK as well as data segments.  This supports the
        symmetry of the underlying TCP protocol.

   (e)  The echo and echo reply options of RFC-1072 were combined into a
        single Timestamps option, to reflect the symmetry and to
        simplify processing.

   (f)  The problem of outdated timestamps on long-idle connections,
        discussed in Section 4.2.2, was realized and resolved.

   (g)  RFC-1185 recommended that header prediction take precedence over
        the timestamp check.  Based upon some scepticism about the
        probabilistic arguments given in Section 4.2.4, it was decided
        to recommend that the timestamp check be performed first.

   (h)  The spec was modified so that the extended options will be sent
        on <SYN,ACK> segments only when they are received in the
        corresponding <SYN> segments.  This provides the most
        conservative possible conditions for interoperation with
        implementations without the extensions.


   In addition to these substantive changes, the present RFC attempts to
   specify the algorithms unambiguously by presenting modifications to
   the Event Processing rules of RFC-793; see Appendix F.

   There are additional changes in this document from RFC-1323.  These
   changes are:

   (a)  The description of which TSecr values can be used to update the
        measured RTT has been clarified.  Specifically, with Timestamps,
        the Karn algorithm [Karn87] is disabled.  The Karn algorithm



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        disables all RTT measurements during retransmission, since it is
        ambiguous whether the ACK is is for the original packet, or the
        retransmitted packet.  With Timestamps, that ambiguity is
        removed since the TSecr in the ACK will contain the TSval from
        which ever data packet made it to the destination.

   (b)  In RFC-1323, section 3.4, step (2) of the algorithm to control
        which timestamp is echoed was incorrect in two regards:

        (1)  It failed to update TSrecent for a retransmitted segment
             that resulted from a lost ACK.

        (2)  It failed if SEG.LEN = 0.

        In the new algorithm, the case of SEG.TSval = TSrecent is
        included for consistency with the PAWS test.

   (c)  One correction was made to the Event Processing Summary in
        Appendix F.  In SEND CALL/ESTABLISHED STATE, RCV.WND is used to
        fill in the SEG.WND value, not SND.WND.

   (d)  New pseudo-code summary has been added in Appendix E.

   (e)  Appendix A has been expanded with information about the TCP MSS
        option and the TCP Urgent Pointer.




























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APPENDIX D: SUMMARY OF NOTATION

   The following notation has been used in this document.

   Options

       WSopt:       TCP Window Scale Option
       TSopt:       TCP Timestamps Option

   Option Fields

       shift.cnt:   Window scale byte in WSopt.
       TSval:       32-bit Timestamp Value field in TSopt.
       TSecr:       32-bit Timestamp Reply field in TSopt.

   Option Fields in Current Segment

       SEG.TSval:   TSval field from TSopt in current segment.
       SEG.TSecr:   TSecr field from TSopt in current segment.
       SEG.WSopt:   8-bit value in WSopt

   Clock Values

       my.TSclock:      Local source of 32-bit timestamp values
       my.TSclock.rate: Period of my.TSclock (1 ms to 1 sec).

   Per-Connection State Variables

       TS.Recent:       Latest received Timestamp
       Last.ACK.sent:   Last ACK field sent

       Snd.TS.OK:       1-bit flag
       Snd.WS.OK:       1-bit flag

       Rcv.Wind.Scale:  Receive window scale power
       Snd.Wind.Scale:  Send window scale power

       Start.Time:      my.TSclock value when segment being timed was
                        sent (used by pre-1323 code).

   Procedure

       Update_SRTT( m ) Procedure to update the smoothed RTT and RTT
                        variance estimates, using the rules of
                        [Jacobson88a], given m, a new RTT measurement.








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APPENDIX E: PSEUDO-CODE SUMMARY

   Create new TCB => {
       Rcv.wind.scale =
             MIN( 14, MAX( 0, floor(log2(receive buffer space)) - 15 ) );
       Snd.wind.scale = 0;
       Last.ACK.sent = 0;
       Snd.TS.OK = Snd.WS.OK = FALSE;
   }


   Send initial {SYN} segment => {

       SEG.WND = MIN( RCV.WND, 65535 );
       Include in segment: TSopt(TSval=my.TSclock, TCecr=0);
       Include in segment: WSopt = Rcv.wind.scale;
   }


   Send {SYN, ACK} segment => {

       SEG.ACK = Last.ACK.sent = RCV.NXT;
       SEG.WND = MIN( RCV.WND, 65535 );
       if (Snd.TS.OK) then
             Include in segment: TSopt(TSval=my.TSclock, TSecr=TS.Recent);
       if (Snd.WS.OK) then
             Include in segment: WSopt = Rcv.wind.scale;
   }


   Receive {SYN} or {SYN,ACK} segment => {

       if (Segment contains TSopt) then {
             TS.Recent = SEG.TSval;
             Snd.TS.OK = TRUE;
             if (is {SYN,ACK} segment) then
                   Update_SRTT(
                            (my.TSclock - SEG.TSecr)/my.TSclock.rate ) ;
       }

       if Segment contains WSopt) then {
             Snd.wind.scale = SEG.WSopt;
             Snd.WS.OK = TRUE;
       }
       else
             Rcv.wind.scale = Snd.wind.scale = 0;
   }






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   Send non-SYN segment => {

       SEG.ACK = Last.ACK.sent = RCV.NXT;
       SEG.WND = MIN( RCV.WND >> Rcv.wind.scale, 65535 );
       if (Snd.TS.OK) then
             Include in segment: TSopt(TSval=my.TSclock, TSecr=TS.Recent);
   }


   Receive non-SYN segment in (state >= ESTABLISHED) => {

       Window = (SEG.WND << Snd.wind.scale);
             /* Use 32-bit 'Window' instead of 16-bit 'SEG.WND'
              * in rest of processing.
              */

       if (Segment contains TSopt) then {
             if (SEG.TSval < TS.Recent && Idle less than 24 days) then {
                   if (Send.TS.OK AND (NOT RST) ) then {
                               /* Timestamp too old =>
                                *    segment is unacceptable.
                                */
                         Send ACK segment;
                         Discard segment and return;
                   }
             }
             else {
                   if (SEG.SEQ =< Last.ACK.sent) then
                               TS.Recent = SEG.TSval;
             }
       }

       if (SEG.ACK > SND.UNA) then {
                    /* (At least part of) first segment in
                     * retransmission queue has been ACKd
                     */
             if (Segment contains TSopt) then
                   Update_SRTT(
                          (my.TSclock - SEG.TSecr)/my.TSclock.rate);
             else
                   Update_SRTT( /* for compatibility */
                          (my.TSclock - Start.Time)/my.TSclock.rate);
       }
   }









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APPENDIX F: EVENT PROCESSING SUMMARY


Event Processing

  OPEN Call

     ...
    An initial send sequence number (ISS) is selected.  Send a SYN
    segment of the form:

        <SEQ=ISS><CTL=SYN><TSval=my.TSclock><WSopt=Rcv.Wind.Scale>

      ...

  SEND Call

    CLOSED STATE (i.e., TCB does not exist)

      ...

    LISTEN STATE

      If the foreign socket is specified, then change the connection
      from passive to active, select an ISS.  Send a SYN segment
      containing the options: <TSval=my.TSclock> and
      <WSopt=Rcv.Wind.Scale>.  Set SND.UNA to ISS, SND.NXT to ISS+1.
      Enter SYN-SENT state. ...

    SYN-SENT STATE
    SYN-RECEIVED STATE

     ...

    ESTABLISHED STATE
    CLOSE-WAIT STATE

      Segmentize the buffer and send it with a piggybacked
      acknowledgment (acknowledgment value = RCV.NXT).  ...

      If the urgent flag is set ...

      If the Snd.TS.OK flag is set, then include the TCP Timestamps
      option <TSval=my.TSclock,TSecr=TS.Recent> in each data segment.

      Scale the receive window for transmission in the segment header:

            SEG.WND = (RCV.WND >> Rcv.Wind.Scale).




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  SEGMENT ARRIVES

     ...

    If the state is LISTEN then

      first check for an RST

       ...

      second check for an ACK

       ...

      third check for a SYN

        if the SYN bit is set, check the security.  If the ...

         ...

        If the SEG.PRC is less than the TCB.PRC then continue.

        Check for a Window Scale option (WSopt); if one is found, save
        SEG.WSopt in Snd.Wind.Scale and set Snd.WS.OK flag on.
        Otherwise, set both Snd.Wind.Scale and Rcv.Wind.Scale to zero
        and clear Snd.WS.OK flag.

        Check for a TSopt option; if one is found, save SEG.TSval in the
        variable TS.Recent and turn on the Snd.TS.OK bit.

        Set RCV.NXT to SEG.SEQ+1, IRS is set to SEG.SEQ and any other
        control or text should be queued for processing later.  ISS
        should be selected and a SYN segment sent of the form:

          <SEQ=ISS><ACK=RCV.NXT><CTL=SYN,ACK>

        If the Snd.WS.OK bit is on, include a WSopt option
        <WSopt=Rcv.Wind.Scale> in this segment.  If the Snd.TS.OK bit is
        on, include a TSopt <TSval=my.TSclock,TSecr=TS.Recent> in this
        segment.  Last.ACK.sent is set to RCV.NXT.

        SND.NXT is set to ISS+1 and SND.UNA to ISS.  The connection
        state should be changed to SYN-RECEIVED.  Note that any other
        incoming control or data (combined with SYN) will be processed
        in the SYN-RECEIVED state, but processing of SYN and ACK should
        not be repeated.  If the listen was not fully specified (i.e.,
        the foreign socket was not fully specified), then the
        unspecified fields should be filled in now.

      fourth other text or control



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       ...

    If the state is SYN-SENT then


      first check the ACK bit

       ...

      fourth check the SYN bit

         ...

        If the SYN bit is on and the security/compartment and precedence
        are acceptable then, RCV.NXT is set to SEG.SEQ+1, IRS is set to
        SEG.SEQ, and any acknowledgements on the retransmission queue
        which are thereby acknowledged should be removed.

        Check for a Window Scale option (WSopt); if is found, save
        SEG.WSopt in Snd.Wind.Scale; otherwise, set both Snd.Wind.Scale
        and Rcv.Wind.Scale to zero.

        Check for a TSopt option; if one is found, save SEG.TSval in
        variable TS.Recent and turn on the Snd.TS.OK bit in the
        connection control block.  If the ACK bit is set, use my.TSclock
        - SEG.TSecr as the initial RTT estimate.

        If SND.UNA > ISS (our SYN has been ACKed), change the connection
        state to ESTABLISHED, form an ACK segment:

            <SEQ=SND.NXT><ACK=RCV.NXT><CTL=ACK>

        and send it.  If the Snd.Echo.OK bit is on, include a TSopt
        option <TSval=my.TSclock,TSecr=TS.Recent> in this ACK segment.
        Last.ACK.sent is set to RCV.NXT.

        Data or controls which were queued for transmission may be
        included.  If there are other controls or text in the segment
        then continue processing at the sixth step below where the URG
        bit is checked, otherwise return.

        Otherwise enter SYN-RECEIVED, form a SYN,ACK segment:

            <SEQ=ISS><ACK=RCV.NXT><CTL=SYN,ACK>

        and send it.  If the Snd.Echo.OK bit is on, include a TSopt
        option <TSval=my.TSclock,TSecr=TS.Recent> in this segment.  If
        the Snd.WS.OK bit is on, include a WSopt option
        <WSopt=Rcv.Wind.Scale> in this segment.  Last.ACK.sent is set to
        RCV.NXT.



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        If there are other controls or text in the segment, queue them
        for processing after the ESTABLISHED state has been reached,
        return.

      fifth, if neither of the SYN or RST bits is set then drop the
      segment and return.


    Otherwise,

    First, check sequence number

      SYN-RECEIVED STATE
      ESTABLISHED STATE
      FIN-WAIT-1 STATE
      FIN-WAIT-2 STATE
      CLOSE-WAIT STATE
      CLOSING STATE
      LAST-ACK STATE
      TIME-WAIT STATE

        Segments are processed in sequence.  Initial tests on arrival
        are used to discard old duplicates, but further processing is
        done in SEG.SEQ order.  If a segment's contents straddle the
        boundary between old and new, only the new parts should be
        processed.

        Rescale the received window field:

            TrueWindow = SEG.WND << Snd.Wind.Scale,

        and use "TrueWindow" in place of SEG.WND in the following steps.

        Check whether the segment contains a Timestamps option and bit
        Snd.TS.OK is on.  If so:

          If SEG.TSval < TS.Recent, then test whether connection has
          been idle less than 24 days; if both are true, then the
          segment is not acceptable; follow steps below for an
          unacceptable segment.

          If SEG.SEQ is equal to Last.ACK.sent, then save SEG.ECopt in
          variable TS.Recent.


        There are four cases for the acceptability test for an incoming
        segment:

         ...




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        If an incoming segment is not acceptable, an acknowledgment
        should be sent in reply (unless the RST bit is set, if so drop
        the segment and return):

          <SEQ=SND.NXT><ACK=RCV.NXT><CTL=ACK>

        Last.ACK.sent is set to SEG.ACK of the acknowledgment.  If the
        Snd.Echo.OK bit is on, include the Timestamps option
        <TSval=my.TSclock,TSecr=TS.Recent> in this ACK segment.  Set
        Last.ACK.sent to SEG.ACK and send the ACK segment.  After
        sending the acknowledgment, drop the unacceptable segment and
        return.

         ...


    fifth check the ACK field.

      if the ACK bit is off drop the segment and return.

      if the ACK bit is on

       ...


        ESTABLISHED STATE

          If SND.UNA < SEG.ACK =< SND.NXT then, set SND.UNA <- SEG.ACK.
          Also compute a new estimate of round-trip time.  If Snd.TS.OK
          bit is on, use my.TSclock - SEG.TSecr; otherwise use the
          elapsed time since the first segment in the retransmission
          queue was sent.  Any segments on the retransmission queue
          which are thereby entirely acknowledged...

           ...


    Seventh, process the segment text.

      ESTABLISHED STATE
      FIN-WAIT-1 STATE
      FIN-WAIT-2 STATE

         ...

        Send an acknowledgment of the form:

          <SEQ=SND.NXT><ACK=RCV.NXT><CTL=ACK>

        If the Snd.TS.OK bit is on, include Timestamps option



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        <TSval=my.TSclock,TSecr=TS.Recent> in this ACK segment.  Set
        Last.ACK.sent to SEG.ACK of the acknowledgment, and send it.
        This acknowledgment should be piggy-backed on a segment being
        transmitted if possible without incurring undue delay.


         ...



Security Considerations

   Security issues are not discussed in this memo.

Authors' Addresses

   Van Jacobson
   Packet Design
   2465 Latham Street
   Mountain View, CA 94040

   EMail: van@packetdesign.com


   Bob Braden
   University of Southern California
   Information Sciences Institute
   4676 Admiralty Way
   Marina del Rey, CA 90292

   Phone: (310) 448-9173
   EMail: Braden@ISI.EDU


   David Borman
   Wind River Systems, Inc.
   1250 Northland Drive, #175
   Mendota Heights, MN 55120-1154

   Phone: (612) 405-8194
   Email: dab@BSDI.COM












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From exim@www1.ietf.org  Wed Aug 13 04:28:36 2003
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Date: Wed, 13 Aug 2003 10:26:33 +0200
To: David Borman <dab@BSDI.COM>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: [Tsvwg] Updating timestamps (ts_recent) in Linux
Cc: martin.duke@boeing.com, tsvwg@ietf.org
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At 17:53 12.08.2003, David Borman wrote:
>As a final thought, updating TSrecent more often does not mean that you
>can get RTT measurements more often.  It continues to be the case that the
>only time you can get a reliable RTT measurement from timestamps is when
>when new data is acknowledged.  That is because dup ACK packets can be
>spontainiously generated by the remote side to do a window update.  In
>that case, taking an RTT measurement can be vastly over inflated.  And
>since there is no way to reliably identify whether a received dup ACK
>is due to out-of-order data arriving or due to the remote side deciding
>to update the window, the best course of action is to only take RTT
>measurements when new data is acknowledged.

This brings up another issue that is currently not defined in any RFC: what 
is a DUPACK?

The BSD code (tcp_input.c) is quite clear about it: an ACK is a DUPACK if
   1. the TCP sender has outstanding data, AND
   2. the ACK number is the largest seen so far, i.e., it acks snd_una, AND
   3. the ACK is a pure ACK, i.e., SEG.LEN == 0, AND
   4. the advertized window info didn't change,
      i.e., this is *not a window update*.

So, we with this definition of a DUPACK it should be save for the TCP 
receiver to echo the most recently received ts_val in DUPACKs, and for the 
TCP sender to use such DUPACKs to get a reliable RTT measurement.

Did I miss something?

///Reiner


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From: David Borman <dab@BSDI.COM>
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To: Reiner.Ludwig@ericsson.com
Subject: Re: [Tsvwg] Updating timestamps (ts_recent) in Linux
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> Date: Wed, 13 Aug 2003 10:26:33 +0200
> From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
> Subject: Re: [Tsvwg] Updating timestamps (ts_recent) in Linux
>
> At 17:53 12.08.2003, David Borman wrote:
> >As a final thought, updating TSrecent more often does not mean that you
> >can get RTT measurements more often.  It continues to be the case that the
> >only time you can get a reliable RTT measurement from timestamps is when
> >when new data is acknowledged.  That is because dup ACK packets can be
> >spontainiously generated by the remote side to do a window update.  In
> >that case, taking an RTT measurement can be vastly over inflated.  And
> >since there is no way to reliably identify whether a received dup ACK
> >is due to out-of-order data arriving or due to the remote side deciding
> >to update the window, the best course of action is to only take RTT
> >measurements when new data is acknowledged.
>
> This brings up another issue that is currently not defined in any RFC: what 
> is a DUPACK?
>
> The BSD code (tcp_input.c) is quite clear about it: an ACK is a DUPACK if
>    1. the TCP sender has outstanding data, AND
>    2. the ACK number is the largest seen so far, i.e., it acks snd_una, AND
>    3. the ACK is a pure ACK, i.e., SEG.LEN == 0, AND
>    4. the advertized window info didn't change,
>       i.e., this is *not a window update*.
>
> So, we with this definition of a DUPACK it should be save for the TCP 
> receiver to echo the most recently received ts_val in DUPACKs, and for the 

Yes, the changed rule (2) in section 3.4 does a much better job of updating
TSrecent.  The only time it doesn't update TSrecent when a newer timestamp
is received, is when a packet arrives out-of-order.

If you always update TSrecent when you get a newer timestamp, then you'd
get this scenario with re-ordering packets (modified from 1323):
                                                      TS.Recent
                    <A, TSval=1> ------------------->
                                                          1
          #1)                <---- <ACK(A), TSecr=1>
                                                          1
                    <C, TSval=3> ------------------->
                                                          3
          #2)                <---- <ACK(A), TSecr=3>
                                                          3
                    <B, TSval=2> ------------------->
                                                          3
          #3)                <---- <ACK(C), TSecr=3>
                                                          3
                    <E, TSval=5> ------------------->
                                                          5
          #4)                <---- <ACK(C), TSecr=5>
                                                          5
                    <D, TSval=4> ------------------->
                                                          5
          #5)                <---- <ACK(E), TSecr=5>

As you can see, the ACK#3 and ACK#5 both are echoing a timestamp
that is newer than the timestamp that was received in the packet
that filled the hole.  This leads to an underestimation for the RTT
for both packets, which are the ones that would be used for RTT
measurement.

In addition, if #3 got lost, then #4 would be seen as the new
ACK and it's shorter RTT value would be used.  With the 1323 rules,
#4 would be echoing TSecr=2, not TSecr=5, leading to an overestimation
of the RTT rather than an underestimation.  The argument in 1323 is
that it's always better to overestimate than underestimate the RTT.

> TCP sender to use such DUPACKs to get a reliable RTT measurement.
> 
> Did I miss something?
>
> ///Reiner

In the above example, if you allow the RTT values from the
DUPACKs to be used (#2 & #4),  you'll be using even shorter RTT
values, skewing the RTT even more.  And if the other side is not
using this new echo scheme, then using the DUPACKs for timing
information could lead wildly overestimating the RTT (too much of
an overestimation is also a bad thing...)

			-David Borman

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To: David Borman <dab@BSDI.COM>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: [Tsvwg] Updating timestamps (ts_recent) in Linux
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At 16:52 13.08.2003, David Borman wrote:
> > This brings up another issue that is currently not defined in any RFC: 
> what
> > is a DUPACK?
> >
> > The BSD code (tcp_input.c) is quite clear about it: an ACK is a DUPACK if
> >    1. the TCP sender has outstanding data, AND
> >    2. the ACK number is the largest seen so far, i.e., it acks snd_una, AND
> >    3. the ACK is a pure ACK, i.e., SEG.LEN == 0, AND
> >    4. the advertized window info didn't change,
> >       i.e., this is *not a window update*.
> >
> > So, we with this definition of a DUPACK it should be save for the TCP
> > receiver to echo the most recently received ts_val in DUPACKs, and for the
>
>Yes, the changed rule (2) in section 3.4 does a much better job of updating
>TSrecent.  The only time it doesn't update TSrecent when a newer timestamp
>is received, is when a packet arrives out-of-order.

And, I still think we can do better ...


>If you always update TSrecent when you get a newer timestamp, then you'd
>get this scenario with re-ordering packets (modified from 1323):
>                                                       TS.Recent
>                     <A, TSval=1> ------------------->
>                                                           1
>           #1)                <---- <ACK(A), TSecr=1>
>                                                           1
>                     <C, TSval=3> ------------------->
>                                                           3
>           #2)                <---- <ACK(A), TSecr=3>
>                                                           3
>                     <B, TSval=2> ------------------->
>                                                           3
>           #3)                <---- <ACK(C), TSecr=3>
>                                                           3
>                     <E, TSval=5> ------------------->
>                                                           5
>           #4)                <---- <ACK(C), TSecr=5>
>                                                           5
>                     <D, TSval=4> ------------------->
>                                                           5
>           #5)                <---- <ACK(E), TSecr=5>
>
>As you can see, the ACK#3 and ACK#5 both are echoing a timestamp
>that is newer than the timestamp that was received in the packet
>that filled the hole.  This leads to an underestimation for the RTT
>for both packets, which are the ones that would be used for RTT
>measurement.

I played with your example for a while, and looked at the following 
scenario: Assume we have a constant (real) RTT of 100 ms, and the TCP 
sender's cwnd allows to send the five packets A - E.
The TCP sender sends A,
10 ms after it sent A, it sends B,
10 ms after it sent B, it sends C,
20 ms after it sent C, it sends D,
10 ms after it sent D, it sends E.

To cause the rordering in your example:
B is held up in the network for 20 ms so that C "overtakes", and
D is held up in the network for 20 ms so that E "overtakes".

As usual the TCP sender (blindly) does 'tcp_now - TSecr' to derive an RTT 
from every ACK (Karn can be safely disabled with timestamps).

The real RTT samples as seen by "somebody traveling in the packet", and as 
they arrive at the TCP sender are:
RTT(A)=100
RTT(C)=100
RTT(B)=120
RTT(E)=100
RTT(D)=120

The RTT samples taken by a TCP sender using the rule I suggested are:
RTT(A)=100
RTT(C)=100
RTT(B)=110(-10)
RTT(E)=100
RTT(D)=110(-10)

The RTT samples taken by a TCP sender using RFC1323 are:
RTT(A)=100
RTT(C)=120(+20)
RTT(B)=120
RTT(E)=140(+40)
RTT(D)=120

So, you are right. The rule I suggested leads to a slight underestimation, 
while RFC1323 - as expected - leads to overestimation.

But, what if we simply forget about TS.Recent and simply let the TCP 
receiver echo TSval received in the segment for which it sends an ACK? 
Then, in your example the TCP sender would see TSecr=1, 3, 2, 5, 4, and in 
the scenario I gave above the RTT samples taken by the TCP sender would be:
RTT(A)=100
RTT(C)=100
RTT(B)=120
RTT(E)=100
RTT(D)=120
=> the true RTTs. No over-/under-estimation.

We would still need to take care of delayed ACKs, though. For that we could 
use TS.Recent. We would store in TS.Recent the TSval from an in-sequence 
data segement for which an ACK is not generated immediately due to delayed 
ACK, and would echo TS.Recent in the delayed ACK.


>In the above example, if you allow the RTT values from the
>DUPACKs to be used (#2 & #4),  you'll be using even shorter RTT
>values, skewing the RTT even more.

I don't think so. Those RTT values would even be the correct values.


>And if the other side is not
>using this new echo scheme, then using the DUPACKs for timing
>information could lead wildly overestimating the RTT (too much of
>an overestimation is also a bad thing...)

A TCP sender could easily detect whether a TCP receiver is using RFC1323 or 
a newer echo scheme: with RFC1323 the TSecr values in subsequent DUPACKs 
don't change.

Apart from these corner cases, I wonder if we are losing the bigger 
picture. The very common case is that the TCP sender receives a series of 
DUPACKs during loss recovery. The recovery phase at least lasts for one RTT 
during which today with RFC1323 the RTT can't be sampled.

Reordering, on the other hand, has so far only been reported to be a rare 
event in the Internet. Studies that did find large reordering, later found 
that this was due to some hardware glitch that then got fixed.

///Reiner





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Subject: RE: [Tsvwg] Updating timestamps (ts_recent) in Linux
Date: Wed, 13 Aug 2003 13:33:41 -0700
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From: "Duke, Martin" <martin.duke@boeing.com>
To: "Reiner Ludwig" <Reiner.Ludwig@ericsson.com>,
        "David Borman" <dab@BSDI.COM>
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>=20
> The real RTT samples as seen by "somebody traveling in the=20
> packet", and as=20
> they arrive at the TCP sender are:
> RTT(A)=3D100
> RTT(C)=3D100
> RTT(B)=3D120
> RTT(E)=3D100
> RTT(D)=3D120
>=20
> The RTT samples taken by a TCP sender using the rule I suggested are:
> RTT(A)=3D100
> RTT(C)=3D100
> RTT(B)=3D110(-10)
> RTT(E)=3D100
> RTT(D)=3D110(-10)
>=20
> The RTT samples taken by a TCP sender using RFC1323 are:
> RTT(A)=3D100
> RTT(C)=3D120(+20)
> RTT(B)=3D120
> RTT(E)=3D140(+40)
> RTT(D)=3D120

Remember that using RFC 1323, only 3 measurements are taken because 2 of =
the ACKs do not ack new data.

Using RFC 1323, the measurements are taken from the 1st, 3rd, and 5th =
ACKs, and are

RTT(A) =3D 100
RTT(B) =3D 120
RTT(D) =3D 120
(Packets C and E do not update ts_recent)

So I don't see what the problem is in your example.

Martin
=20

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Date: Thu, 14 Aug 2003 08:42:42 +0200
To: "Duke, Martin" <martin.duke@boeing.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Cc: tsvwg@ietf.org
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Subject: [Tsvwg] RFC1323.bis [was: Updating timestamps (ts_recent)]
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At 22:33 13.08.2003, Duke, Martin wrote:

> >
> > The real RTT samples as seen by "somebody traveling in the
> > packet", and as
> > they arrive at the TCP sender are:
> > RTT(A)=100
> > RTT(C)=100
> > RTT(B)=120
> > RTT(E)=100
> > RTT(D)=120
> >
> > The RTT samples taken by a TCP sender using the rule I suggested are:
> > RTT(A)=100
> > RTT(C)=100
> > RTT(B)=110(-10)
> > RTT(E)=100
> > RTT(D)=110(-10)
> >
> > The RTT samples taken by a TCP sender using RFC1323 are:
> > RTT(A)=100
> > RTT(C)=120(+20)
> > RTT(B)=120
> > RTT(E)=140(+40)
> > RTT(D)=120
>
>Remember that using RFC 1323, only 3 measurements are taken because 2 of 
>the ACKs do not ack new data.
>
>Using RFC 1323, the measurements are taken from the 1st, 3rd, and 5th 
>ACKs, and are
>
>RTT(A) = 100
>RTT(B) = 120
>RTT(D) = 120
>(Packets C and E do not update ts_recent)

True. And if we would continue to use RFC1323's rule of only taking RTT 
samples from ACKs for new data then with any of the echo schemes only the 
ACKs for A, B, and D got sampled.


>So I don't see what the problem is in your example.

RFC1323's sender rule of only taking RTT samples from ACKs for new data is 
one half of the problem I see. The other half of that problem is RFC1323's 
receiver echo rule (B)).

The combination of both halves of that problem lead to the fact that during 
a typical loss recovery, when an RTT worth of DUPACKs return, the sender 
can't sample the RTT. So far, I haven't seen a good reason for why this has 
to be like that.

I would therefore like to see RFC1323 changed in such a way that (1) the 
TCP sender is allowed to sample the RTT from DUPACKs, and (2) that the TCP 
receiver echos the most recently received TSval in DUPACKs. Clearly, having 
(1) without (2) makes little sense, and vice versa.

The problem you raised is different, but related, and relevant, I believe.

BTW, whatever RFC1323.bis gets created, it might be a good idea to include 
RFC1323.bis-capability-indication in the initial SYN/SYN-ACK handshake.

Another open issue with RFC1323, I think, is the question of whether 
*every* segment MUST carry a TS option once that was negotiated in the 
SYN/SYN-ACK handshake.

///Reiner 


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Hi,

A revision to RFC 1323, TCP Extensions for High Performance, is
currently being discussed on the tsvwg@ietf.org mailing list.
Is this the right place for this discussion, or should it move
over to end2end-interest@postel.org, the End to End mailing list?
(The TCP Large Windows mailing list is no longer active.)

			-David Borman

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To: tsvwg@ietf.org
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Folks-

Ethan and I just revised the i-d per the thread on tsvwg over the
last month.  The two major changes to the document are:

  * if the SACK scoreboard is empty when a DSACK rolls in then we
    say the TCP cannot revert its congestion control state because
    the presumption is that all the ACKs have been lost

  * if the TCP ertransmits something more than once then the TCP
    cannot revert the congestion control state because DSACKs can no
    longer resolve the retransmission ambiguity

We have sent the new version of the i-d to the archives and so it
will appear in the standard places in the next few days.  It is
available at: 

 http://roland.grc.nasa.gov/~mallman/papers/draft-ietf-tsvwg-dup-notify-01.txt

now.

We would appreciate comments on the new version.

Thanks,
allman


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

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Hi Mark/All,

* In the draft the following is about the SACK scoreboard: "We assume the
TCP sender has a data structure to hold selective acknowledgment
information (e.g., as outlined in [RFC3517])."

Traditionally entries lying below the cumulative ACK point are deleted
from the scoreboard. But the DSACK information normally comes after loss
recovery has exited. So how long should the scoreboard entries be
maintained beyond what is required in 3517? (1 ACK after loss recovery
exits in Fast retransmit; and N Acks after loss recovery exits in a
timeout (GoBackN)?)


* In the Security Considerations, ECN Nonce 3540 is offered as a
protection. But 3540 calls for disabling Nonce Sum checks for DupAcks. Is
the draft suggesting some extension to 3540 that NS be checked for DupAcks
carrying DSACK?

It may be helpful to be more explicit on the above in the ID.


Thanks


On Thu, 14 Aug 2003, Mark Allman wrote:

>
> Folks-
>
> Ethan and I just revised the i-d per the thread on tsvwg over the
> last month.  The two major changes to the document are:
>
>   * if the SACK scoreboard is empty when a DSACK rolls in then we
>     say the TCP cannot revert its congestion control state because
>     the presumption is that all the ACKs have been lost
>
>   * if the TCP ertransmits something more than once then the TCP
>     cannot revert the congestion control state because DSACKs can no
>     longer resolve the retransmission ambiguity
>
> We have sent the new version of the i-d to the archives and so it
> will appear in the standard places in the next few days.  It is
> available at:
>
>  http://roland.grc.nasa.gov/~mallman/papers/draft-ietf-tsvwg-dup-notify-01.txt
>
> now.
>
> We would appreciate comments on the new version.
>
> Thanks,
> allman
>
>
> --
> Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>


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Hello, everyone.

We have recently submitted a paper to HOTNET on delay-based congestion
control algorithms. We present a case for delay based congestion control
algorithms in the absence of explicit congestion feedback.  Even though
the advantages of delay based algorithms are small in low-speed networks,
they are decisive in high-speed networks.

The paper is at:

http://netlab.caltech.edu/pub/papers/hotnet2003.pdf

Comments and suggestions are greatly appreciated,

Cheng


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From exim@www1.ietf.org  Mon Aug 18 13:49:33 2003
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To: tsvwg@ietf.org, "Ethan Blanton" <eblanton@cs.purdue.edu>,
        Reiner Ludwig <Reiner.Ludwig@ericsson.com>
From: Mark Allman <mallman@grc.nasa.gov>
Reply-To: mallman@grc.nasa.gov
Subject: Re: [Tsvwg] Issues with draft-blanton-dsack-use-02 
Organization: BBN Technologies/NASA GRC
Song-of-the-Day: Imagine
Date: Mon, 18 Aug 2003 13:47:48 -0400
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> Ethan and I just revised the i-d per the thread on tsvwg over the
> last month.  The two major changes to the document are:
> 
>   * if the SACK scoreboard is empty when a DSACK rolls in then we
>     say the TCP cannot revert its congestion control state because
>     the presumption is that all the ACKs have been lost
> 
>   * if the TCP ertransmits something more than once then the TCP
>     cannot revert the congestion control state because DSACKs can no
>     longer resolve the retransmission ambiguity
> 
> We have sent the new version of the i-d to the archives and so it
> will appear in the standard places in the next few days.  It is
> available at: 
> 
>  [BUSTED URL]
> 
> now.

Folks-

I misnamed the i-d for some reason.  In any case, the name of the
I-D in question is "draft-ietf-tsvwg-dsack-use-01.txt".  It should
pop up in the archives soon.  You can also slurp a copy from:

http://roland.grc.nasa.gov/~mallman/papers/draft-ietf-tsvwg-dsack-use-01.txt

Sorry for the confusion,
allman


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

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Transport Area Working Group Working Group of the IETF.

	Title		: Using TCP DSACKs and SCTP Duplicate TSNs to Detect 
                          Spurious Retransmissions
	Author(s)	: E. Blanton, M. Allman
	Filename	: draft-ietf-tsvwg-dsack-use-01.txt
	Pages		: 7
	Date		: 2003-8-18
	
TCP and SCTP provide notification of duplicate segment receipt
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document presents a conservative method of using this information to
identify unnecessary retransmissions for various applications.

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
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	Title		: Using TCP DSACKs and SCTP Duplicate TSNs to Detect 
                          Spurious Retransmissions
	Author(s)	: E. Blanton, M. Allman
	Filename	: draft-ietf-tsvwg-dsack-use-01.txt
	Pages		: 7
	Date		: 2003-8-18
	
TCP and SCTP provide notification of duplicate segment receipt
through DSACK and Duplicate TSN notification, respectively. This
document presents a conservative method of using this information to
identify unnecessary retransmissions for various applications.

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From exim@www1.ietf.org  Tue Aug 19 14:18:32 2003
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'David Borman'" <dab@BSDI.COM>, end2end-interest@postel.org,
        tsvwg@ietf.org
Subject: RE: [Tsvwg] RFC 1323 revision
Date: Tue, 19 Aug 2003 14:16:26 -0400
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I believe that the TSVWG is a reasonable place to discuss this work (at
least until the long-expected TCP WG is formed), and that an IETF WG mailing
list is the right place to manage documents that aspire to become
standards-track RFCs. Of course, pursuing this document in the TSVWG would
not preclude analysis on the venerable e2e list, which of course harbors
numerous non-overlapping experts who could provide valuable input on this
work.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: David Borman [mailto:dab@BSDI.COM]
> Sent: Thursday, August 14, 2003 6:27 AM
> To: end2end-interest@postel.org; tsvwg@ietf.org
> Subject: [Tsvwg] RFC 1323 revision
> 
> 
> Hi,
> 
> A revision to RFC 1323, TCP Extensions for High Performance, is
> currently being discussed on the tsvwg@ietf.org mailing list.
> Is this the right place for this discussion, or should it move
> over to end2end-interest@postel.org, the End to End mailing list?
> (The TCP Large Windows mailing list is no longer active.)
> 
> 			-David Borman
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Transport Area Working Group Working Group of the IETF.

	Title		: SCTP Partial Reliability Extension
	Author(s)	: R. Stewart et al.
	Filename	: draft-ietf-tsvwg-prsctp-01.txt
	Pages		: 27
	Date		: 2003-8-22
	
This memo describes an extension to the Stream Control Transmission
Protocol (SCTP) RFC2960 [5] that allows an SCTP endpoint to signal to
its peer that it should move the cumulative ack point forward.  When
both sides of an SCTP association support this extension, it can be
used by an SCTP implementation to provide partially reliable data
transmission service to an upper layer protocol.  This memo describes
(1) the protocol extensions, which consist of a new parameter for
INIT and INIT ACK, and a new FORWARD TSN chunk type (2) one example
partially reliable service that can be provided to the upper layer
via this mechanism.

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From exim@www1.ietf.org  Fri Aug 22 12:28:28 2003
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Subject: [Tsvwg] new revision of early retransmit available
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Folks-

We have just sent a new version of the Early Retransmit internet
draft to the secratariet.  It should appear in the archives in a
couple of days.  It is available from:

 http://roland.grc.nasa.gov/~mallman/papers/draft-allman-tcp-early-rexmt-02.txt

now.

The largest change in the document is that it now includes a variant
that uses SACK information to make ER more robust.  The problem that
we have found is that the instances where ER will help are windowed
quite a bit when it just counts duplicate ACKs.

First consider two states the reciever could be in....

  * S1: the receiver has ACKed all data received
  * S2: the receiver has received a packet but is delaying its ACK
    for another arrival (or the timer)

When the receiver is in S1 and an out-of-order segment arrives a dup
ACK will be triggered and ER will work nicely (or, as nicely as
possible).

When the receiver is in S2 and an out-of-order segment arrives the
receiver will send an immediate ACK, but it will not be a duplicate
-- but, rather it will cover previously unACKed data.  In this case,
ER is not in a happy place because it expects a duplicate ACK from
all but one of the segments outstanding.

So, we (meaning Josh, really) figured out that if we leveraged SACK
information we could just tell when all but one packet has arrived
at the receiver and key on that for ER, rather than try to count
duplicate ACKs.  (Because even if the ACK for out-of-order data is
cumulatively ACKs previously unACKed data (i.e., is not a dup ACK)
it will still have SACK information indicating what packet triggered
the ACK.)  This scheme seems more robust.

At the moment both schemes are in the draft.  But, we are seriously
considering ripping out the version that does not use SACK on the
grounds that if you're not using SACK and you are interested in good
loss recovery in 2003 then you really can't be all that interested.
(Recent measurements: I am seeing > 90% of web clients support SACK
at a web server I am watching.  I ran tbit against 61,000+ web
servers and 65% support SACK.)

We'd be interested in getting feedback on any of the above issues
(and the text of the draft itself, of course).

Thanks!

allman

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From exim@www1.ietf.org  Sun Aug 24 05:46:32 2003
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Subject: Re: [Tsvwg] new revision of early retransmit available
From: Mika Liljeberg <mika.liljeberg@welho.com>
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Mark,

The SACK variant looks a lot more robust compared to the ack counting
variant. I would feel fairly confident in putting this into an
implementation. I tend to agree that today there's really no excuse to
not do SACK if one is at all interested in performance, so I think a
SACK only variant would be fine.

One suggestion for improvement, though.

Standard Nagle will prevent any smaller than MSS sized segment of new
data from being sent unless all the preceding segments have been acked
first. This means that limited transmit typically cannot be applied with
the last segment of a data burst and, in particular, at the end of the
connection. 

The draft discusses the problems observed with limited transmit and a
small congestion window but doesn't mention the interaction with Nagle
anywhere. I would be good to give some attention to this as well. Early
retransmit can help in both cases.

	MikaL

On Fri, 2003-08-22 at 19:27, Mark Allman wrote:
>  Folks-
> 
> We have just sent a new version of the Early Retransmit internet
> draft to the secratariet.  It should appear in the archives in a
> couple of days.  It is available from:
> 
>  http://roland.grc.nasa.gov/~mallman/papers/draft-allman-tcp-early-rexmt-02.txt
> 
> now.
> 
> The largest change in the document is that it now includes a variant
> that uses SACK information to make ER more robust.  The problem that
> we have found is that the instances where ER will help are windowed
> quite a bit when it just counts duplicate ACKs.
> 
> First consider two states the reciever could be in....
> 
>   * S1: the receiver has ACKed all data received
>   * S2: the receiver has received a packet but is delaying its ACK
>     for another arrival (or the timer)
> 
> When the receiver is in S1 and an out-of-order segment arrives a dup
> ACK will be triggered and ER will work nicely (or, as nicely as
> possible).
> 
> When the receiver is in S2 and an out-of-order segment arrives the
> receiver will send an immediate ACK, but it will not be a duplicate
> -- but, rather it will cover previously unACKed data.  In this case,
> ER is not in a happy place because it expects a duplicate ACK from
> all but one of the segments outstanding.
> 
> So, we (meaning Josh, really) figured out that if we leveraged SACK
> information we could just tell when all but one packet has arrived
> at the receiver and key on that for ER, rather than try to count
> duplicate ACKs.  (Because even if the ACK for out-of-order data is
> cumulatively ACKs previously unACKed data (i.e., is not a dup ACK)
> it will still have SACK information indicating what packet triggered
> the ACK.)  This scheme seems more robust.
> 
> At the moment both schemes are in the draft.  But, we are seriously
> considering ripping out the version that does not use SACK on the
> grounds that if you're not using SACK and you are interested in good
> loss recovery in 2003 then you really can't be all that interested.
> (Recent measurements: I am seeing > 90% of web clients support SACK
> at a web server I am watching.  I ran tbit against 61,000+ web
> servers and 65% support SACK.)
> 
> We'd be interested in getting feedback on any of the above issues
> (and the text of the draft itself, of course).
> 
> Thanks!
> 
> allman
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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Gentlemen,

This is a good document, I am glad to see this kind of document
being worked on.  I understand that this must have been a huge
undertaking, but I have some very quick requests.  Could you improve
the ToC, for example, listing the RFCs as subsections?  Also,
the formatting of the document is confusing - I believe that
all paragraph text should be indented - which it is not; however
text copied from existing RFCs should probably be doubly indented
(it is currently indented a single time, for example in 3.3.1)

I hope this is not overly nit-picky, but my first attempt at=20
reading this document was not successful as the current formatting
got in the way of understanding what is being discussed.

thanks,
John

> -----Original Message-----
> From: ext Pekka Savola [mailto:pekkas@netcore.fi]
> Sent: 22 August, 2003 11:09
> To: tsvwg@ietf.org
> Cc: bob@thefinks.com; Soininen Jonne (NET/Helsinki);
> andreas.bergstrom@hiof.no
> Subject: WG Last Call: three draft-ietf-v6ops-ipv4survey-*01 documents
> (fwd)
>=20
>=20
> FYI;
>=20
> Your oxen are being gutted and you're invited to the bloodfest.
>=20
> In other words..
>=20
> Feedback & Review is sought.  Please take a look at the=20
> transport IPv4=20
> survey document:
>                                                              =20
>                                     =20
> http://www.ietf.org/internet-drafts/draft-ietf-v6ops-ipv4surve
> y-trans-01.txt
>=20
> It's not too long; please send feedback.  In particular, it=20
> would be good
> to identify specifications which have incorrect information, or
> specifications which are not longer relevant and could be moved to
> historic (if someone bothered to do that, but that's a=20
> different topic),
> or the like.
>=20
> Thanks,
>  Pekka, Jonne & Bob
>  v6ops co-chairs
>=20
> ---------- Forwarded message ----------
> Date: Fri, 22 Aug 2003 10:38:49 +0300 (EEST)
> From: Pekka Savola <pekkas@netcore.fi>
> To: v6ops@ops.ietf.org
> Cc: bob@thefinks.com, jonne.soininen@nokia.com,=20
> cesar.olvera@consulintel.es,
>      andreas.bergstrom@hiof.no
> Subject: WG Last Call: three draft-ietf-v6ops-ipv4survey-*01 documents
>=20
> Hi all,
>=20
> This is a WG Last Call for comments on sending the following the next
> three "Survey of IPv4 Addresses in Currently Deployed IETF Standards"
> documents to the IESG for consideration as Informational RFCs:
>=20
> Survey of IPv4 Addresses in Currently Deployed IETF Routing=20
> Area Standards
>  =20
http://www.ietf.org/internet-drafts/draft-ietf-v6ops-ipv4survey-routing-0=
1.txt

Survey of IPv4 Addresses in Currently Deployed IETF Security Area =
Standards
  =
http://www.ietf.org/internet-drafts/draft-ietf-v6ops-ipv4survey-sec-01.tx=
t

Survey of IPv4 Addresses in Currently Deployed IETF Transport Area=20
Standards
 =20
http://www.ietf.org/internet-drafts/draft-ietf-v6ops-ipv4survey-trans-01.=
txt

Please review these documents carefully, and send your feedback to the
list (editorial modifications may also be sent to the editors, Cesar for
the first, Andreas for the other two).  Please also indicate whether or
not you believe that this document is ready to go to the IESG.  Silence
does NOT indicate consent.  Unless sufficient support is demonstrated on
the list, the documents will not be send to the IESG.

The last call will end in 3 weeks, on 12th September.

Pekka, Jonne & Bob







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Hi all,

Further comments:

Section 4 doesn't contain any Draft Standards ... I assume this is an =
oversite.

In section 5, it might be better not to list the specs that have no IPv4 =
dependencies,
or better yet, move them all to an appendix and list them in tabular =
format.

References are a little sparse, too - you may want to consider putting =
all
RFCs listed into the References as well.

thanks,
John

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Hi Pekka,

> > Could you improve
> > the ToC, for example, listing the RFCs as subsections? =20
>=20
> ToC could get quite long, compare to ...-ipv4survey-apps-01.  It may =
not=20
> necessarily make it clearer (IMHO).  It also may be more difficult to=20
> generate a "full ToC", depends on the tools used.

OK - I was just looking for an easier way to find particular
RFCs / protocols, as most readers won't be experts on all of
the RFCs included.
=20
> Personally I don't have a strong opinion either way.
>=20
> > Also,
> > the formatting of the document is confusing - I believe that
> > all paragraph text should be indented - which it is not;=20
>=20
> Agreed, I don't know how difficult this would be given the methods =
used to=20
> edit the documents.

See my above comment.
=20
> > however
> > text copied from existing RFCs should probably be doubly indented
> > (it is currently indented a single time, for example in 3.3.1)
>=20
> I agree but this could cause problems, if the double-intending would =
push=20
> the length beyond 72 characters, there could be problems..

Reformatting might be OK, depending upon the tools used - again, just
looking for a way to improve readability.

> > I hope this is not overly nit-picky, but my first attempt at=20
> > reading this document was not successful as the current formatting
> > got in the way of understanding what is being discussed.
>=20
> Yep.
>=20
> > Section 4 doesn't contain any Draft Standards ... I assume=20
> this is an
> > oversite.
>=20
> Maybe Andreas forgot to add the sentence saying that there are no =
Draft=20
> Standards here (or if there actually are, those are missing =
somewhere).

OK

> > In section 5, it might be better not to list the specs that have no =
IPv4
> > dependencies, or better yet, move them all to an appendix and list =
them
> > in tabular format.
>=20
> We're trying to list these documents under section 7.  Hopefully this =
is=20
> clear enough, as we'd hope to avoid restructuring Phil's legacy too=20
> much..?

OK, just a suggestion to improve readability.
=20
> > References are a little sparse, too - you may want to=20
> consider putting
> > all RFCs listed into the References as well.
>=20
> I disagree, but not strongly; IMHO, it should be apparent that the=20
> specific RFC's are all "unofficial" RFC's .. the list in the end would =

> just get very, very long for not significant benefit (one that I could =
see=20
> anyway).

That is fine will me.

I'll try to read the document tomorrow for technical comments.

John

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From exim@www1.ietf.org  Tue Aug 26 15:40:41 2003
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Organization: Cisco Systems Inc
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Subject: [Tsvwg] overlapping TCP segments during reassembly
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Hi

Is there any guidelines outlined by any TCP RFC's  for determining what
data in the sequence space to keep and what to drop during TCP out of
order packet reassembly when there are overlaps? The BSD code tries to
trim the incoming segment when there is an overlap with a previous
segment in sequence space, and trims the already queued segments when
there is an overlap with succeeding segments in sequence space. I was
curious abt the motivation behind this strategy.

Murali


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From: Pekka Savola <pekkas@netcore.fi>
To: john.loughney@nokia.com
cc: tsvwg@ietf.org, <bob@thefinks.com>, <Jonne.Soininen@nokia.com>,
        <andreas.bergstrom@hiof.no>
In-Reply-To: <DADF50F5EC506B41A0F375ABEB320636A8B27E@esebe023.ntc.nokia.com>
Message-ID: <Pine.LNX.4.44.0308251635120.14530-100000@netcore.fi>
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Subject: [Tsvwg] RE: WG Last Call: three draft-ietf-v6ops-ipv4survey-*01 documents
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Thanks for review.. (the document in question is the Transport Area 
document).

I'm responding to the both emails here.

On Mon, 25 Aug 2003 john.loughney@nokia.com wrote:
> This is a good document, I am glad to see this kind of document
> being worked on.  I understand that this must have been a huge
> undertaking, but I have some very quick requests.  

Ok..

> Could you improve
> the ToC, for example, listing the RFCs as subsections?  

ToC could get quite long, compare to ...-ipv4survey-apps-01.  It may not 
necessarily make it clearer (IMHO).  It also may be more difficult to 
generate a "full ToC", depends on the tools used.

Personally I don't have a strong opinion either way.

> Also,
> the formatting of the document is confusing - I believe that
> all paragraph text should be indented - which it is not; 

Agreed, I don't know how difficult this would be given the methods used to 
edit the documents.

> however
> text copied from existing RFCs should probably be doubly indented
> (it is currently indented a single time, for example in 3.3.1)

I agree but this could cause problems, if the double-intending would push 
the length beyond 72 characters, there could be problems..
 
> I hope this is not overly nit-picky, but my first attempt at 
> reading this document was not successful as the current formatting
> got in the way of understanding what is being discussed.

Yep.

> Section 4 doesn't contain any Draft Standards ... I assume this is an
> oversite.

Maybe Andreas forgot to add the sentence saying that there are no Draft 
Standards here (or if there actually are, those are missing somewhere).

> In section 5, it might be better not to list the specs that have no IPv4
> dependencies, or better yet, move them all to an appendix and list them
> in tabular format.

We're trying to list these documents under section 7.  Hopefully this is 
clear enough, as we'd hope to avoid restructuring Phil's legacy too 
much..?

> References are a little sparse, too - you may want to consider putting
> all RFCs listed into the References as well.

I disagree, but not strongly; IMHO, it should be apparent that the 
specific RFC's are all "unofficial" RFC's .. the list in the end would 
just get very, very long for not significant benefit (one that I could see 
anyway).

Thoughts?

> > -----Original Message-----
> > From: ext Pekka Savola [mailto:pekkas@netcore.fi]
> > Sent: 22 August, 2003 11:09
> > To: tsvwg@ietf.org
> > Cc: bob@thefinks.com; Soininen Jonne (NET/Helsinki);
> > andreas.bergstrom@hiof.no
> > Subject: WG Last Call: three draft-ietf-v6ops-ipv4survey-*01 documents
> > (fwd)
> > 
> > 
> > FYI;
> > 
> > Your oxen are being gutted and you're invited to the bloodfest.
> > 
> > In other words..
> > 
> > Feedback & Review is sought.  Please take a look at the 
> > transport IPv4 
> > survey document:
> >                                                               
> >                                      
> > http://www.ietf.org/internet-drafts/draft-ietf-v6ops-ipv4surve
> > y-trans-01.txt
> > 
> > It's not too long; please send feedback.  In particular, it 
> > would be good
> > to identify specifications which have incorrect information, or
> > specifications which are not longer relevant and could be moved to
> > historic (if someone bothered to do that, but that's a 
> > different topic),
> > or the like.
> > 
> > Thanks,
> >  Pekka, Jonne & Bob
> >  v6ops co-chairs
> > 
> > ---------- Forwarded message ----------
> > Date: Fri, 22 Aug 2003 10:38:49 +0300 (EEST)
> > From: Pekka Savola <pekkas@netcore.fi>
> > To: v6ops@ops.ietf.org
> > Cc: bob@thefinks.com, jonne.soininen@nokia.com, 
> > cesar.olvera@consulintel.es,
> >      andreas.bergstrom@hiof.no
> > Subject: WG Last Call: three draft-ietf-v6ops-ipv4survey-*01 documents
> > 
> > Hi all,
> > 
> > This is a WG Last Call for comments on sending the following the next
> > three "Survey of IPv4 Addresses in Currently Deployed IETF Standards"
> > documents to the IESG for consideration as Informational RFCs:
> > 
> > Survey of IPv4 Addresses in Currently Deployed IETF Routing 
> > Area Standards
> >   
> http://www.ietf.org/internet-drafts/draft-ietf-v6ops-ipv4survey-routing-01.txt
> 
> Survey of IPv4 Addresses in Currently Deployed IETF Security Area Standards
>   http://www.ietf.org/internet-drafts/draft-ietf-v6ops-ipv4survey-sec-01.txt
> 
> Survey of IPv4 Addresses in Currently Deployed IETF Transport Area 
> Standards
>   
> http://www.ietf.org/internet-drafts/draft-ietf-v6ops-ipv4survey-trans-01.txt
> 
> Please review these documents carefully, and send your feedback to the
> list (editorial modifications may also be sent to the editors, Cesar for
> the first, Andreas for the other two).  Please also indicate whether or
> not you believe that this document is ready to go to the IESG.  Silence
> does NOT indicate consent.  Unless sufficient support is demonstrated on
> the list, the documents will not be send to the IESG.
> 
> The last call will end in 3 weeks, on 12th September.
> 
> Pekka, Jonne & Bob
> 
> 
> 
> 
> 
> 

-- 
Pekka Savola                 "You each name yourselves king, yet the
Netcore Oy                    kingdom bleeds."
Systems. Networks. Security. -- George R.R. Martin: A Clash of Kings



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From exim@www1.ietf.org  Wed Aug 27 00:42:54 2003
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To: john.loughney@nokia.com
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From: Sally Floyd <floyd@icir.org>
Subject: Re: [Tsvwg] WGLC for draft-ietf-tsvwg-highspeed-00
Date: Tue, 26 Aug 2003 14:34:55 -0700
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Since the two-week working group last call for this ended yesterday,
and I haven't seen any comments on the mailing list, I will submit
the revised draft of draft-ietf-tsvwg-highspeed-00 with the changes
that were already agreed-upon (appended below), and
draft-ietf-tsvwg-highspeed and draft-ietf-tsvwg-slowstart
should then be ready to be forwarded to the IESG for IESG Last Call.

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


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

The web page at 
"http://www.icir.org/floyd/hstcp/draft-ietf-tsvwg-highspeed-00.errata"
contains the changes that have already been agreed upon, that I am
going to submit now:

 Changes to make to draft-ietf-tsvwg-highspeed-00.txt:
 
 Change:
 
 "The proposals in this document are experimental.  We believe they
 are safe for deployment in the current Internet, "
 
 To:
 
 "The proposals in this document are experimental.  While they may be
 deployed in the current Internet, "


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Subject: Re: [Tsvwg] overlapping TCP segments during reassembly
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On Monday, August 25, 2003, at 08:56 PM, Murali Bashyam wrote:

> Hi
>
> Is there any guidelines outlined by any TCP RFC's  for determining what
> data in the sequence space to keep and what to drop during TCP out of
> order packet reassembly when there are overlaps? The BSD code tries to
> trim the incoming segment when there is an overlap with a previous
> segment in sequence space, and trims the already queued segments when
> there is an overlap with succeeding segments in sequence space. I was
> curious abt the motivation behind this strategy.
>
> Murali
>

Certain things are implied:

1) overlaps indicate retransmissions.

2) multiple valid segments must all have the same value
    for any given byte in the TCP stream, i.e. the
    retransmit can only correct transmission errors
    it cannot "fix" the data.

3) you can only deliver a given byte to the TCP consumer once.

So, if you have a buffered but undelivered sequence, should
you replace it with new content? Well, since you know the
values are the same, why not optimize the copy by simply
not performing it?




Caitlin Bestler - cait@asomi.com - http://asomi.com/


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> Date: Mon, 25 Aug 2003 18:56:48 -0700
> From: Murali Bashyam <mbashyam@cisco.com>
> Subject: [Tsvwg] overlapping TCP segments during reassembly
...
> Is there any guidelines outlined by any TCP RFC's  for determining what
> data in the sequence space to keep and what to drop during TCP out of
> order packet reassembly when there are overlaps? The BSD code tries to
> trim the incoming segment when there is an overlap with a previous
> segment in sequence space, and trims the already queued segments when
> there is an overlap with succeeding segments in sequence space. I was
> curious abt the motivation behind this strategy.

In the BSD code, it is less work to trim from the front of an mbuf
chain than from the tail of an mbuf chain, so when the front overlaps
it's easier to trim the new segment, and when the tail overlaps it's
easier to trim the succeeding segments.  See m_adj() in uipc_mbuf.c.

			-David Borman

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Transport Area Working Group Working Group of the IETF.

	Title		: HighSpeed TCP for Large Congestion Windows
	Author(s)	: S. Floyd
	Filename	: draft-ietf-tsvwg-highspeed-01.txt,.ps
	Pages		: 36
	Date		: 2003-8-27
	
This document proposes HighSpeed TCP, a modification to TCP's
congestion control mechanism for use with TCP connections with
large congestion windows.  The congestion control mechanisms
of the current Standard TCP constrains the congestion windows
that can be achieved by TCP in realistic environments.  For
example, for a Standard TCP connection with 1500-byte packets
and a 100 ms round-trip time, achieving a steady-state
throughput of 10 Gbps would require an average congestion
window of 83,333 segments, and a packet drop rate of at most
one congestion event every 5,000,000,000 packets (or
equivalently, at most one congestion event every 1 2/3 hours).
This is widely acknowledged as an unrealistic constraint.  To
address this limitation of TCP, this document proposes
HighSpeed TCP, and solicits experimentation and feedback from
the wider community.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-highspeed-01.txt

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Transport Area Working Group Working Group of the IETF.

	Title		: HighSpeed TCP for Large Congestion Windows
	Author(s)	: S. Floyd
	Filename	: draft-ietf-tsvwg-highspeed-01.txt,.ps
	Pages		: 36
	Date		: 2003-8-27
	
This document proposes HighSpeed TCP, a modification to TCP's
congestion control mechanism for use with TCP connections with
large congestion windows.  The congestion control mechanisms
of the current Standard TCP constrains the congestion windows
that can be achieved by TCP in realistic environments.  For
example, for a Standard TCP connection with 1500-byte packets
and a 100 ms round-trip time, achieving a steady-state
throughput of 10 Gbps would require an average congestion
window of 83,333 segments, and a packet drop rate of at most
one congestion event every 5,000,000,000 packets (or
equivalently, at most one congestion event every 1 2/3 hours).
This is widely acknowledged as an unrealistic constraint.  To
address this limitation of TCP, this document proposes
HighSpeed TCP, and solicits experimentation and feedback from
the wider community.

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Caitlin Bestler wrote:

> On Monday, August 25, 2003, at 08:56 PM, Murali Bashyam wrote:
>
> > Hi
> >
> > Is there any guidelines outlined by any TCP RFC's  for determining what
> > data in the sequence space to keep and what to drop during TCP out of
> > order packet reassembly when there are overlaps? The BSD code tries to
> > trim the incoming segment when there is an overlap with a previous
> > segment in sequence space, and trims the already queued segments when
> > there is an overlap with succeeding segments in sequence space. I was
> > curious abt the motivation behind this strategy.
> >
> > Murali
> >
>
> Certain things are implied:
>
> 1) overlaps indicate retransmissions.
>
> 2) multiple valid segments must all have the same value
>     for any given byte in the TCP stream, i.e. the
>     retransmit can only correct transmission errors
>     it cannot "fix" the data.
>
> 3) you can only deliver a given byte to the TCP consumer once.
>
> So, if you have a buffered but undelivered sequence, should
> you replace it with new content? Well, since you know the
> values are the same, why not optimize the copy by simply
> not performing it?

Right,  based on these assumptions, that was the policy i was leaning
towards, can i simply apply all the trimming  i need to do on the  incoming
segment and always honour the already buffered data? Would this end up
violating any delivery semantics?

Murali

>
>
> Caitlin Bestler - cait@asomi.com - http://asomi.com/
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
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