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Hi All,

The current venue for discussing draft-kompella-rsvp-change-00.txt
(IANA Considerations and change process for RSVP) will be in the
TSVWG (Thu 13:00-15:00 -- if not, I hope either the TSV ADs will
correct me.

For now, let's limit the discussion to the tsvwg mailing list.

Kireeti.
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From: Scott  Bradner <sob@harvard.edu>
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To: ccamp@ops.ietf.org, kireeti@juniper.net, mpls@uu.net, nsis@ietf.org,
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Cc: Jon.peterson@neustar.com, mankin@psg.com
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the tsvwg list will be the main venue 

I will talk with Allison to see if there is time in teh tsvwg meeting
to start the dicussion

Scott


---

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Hi All,

The current venue for discussing draft-kompella-rsvp-change-00.txt
(IANA Considerations and change process for RSVP) will be in the
TSVWG (Thu 13:00-15:00 -- if not, I hope either the TSV ADs will
correct me.

For now, let's limit the discussion to the tsvwg mailing list.

Kireeti.
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http://mailman.isi.edu/mailman/listinfo/rsvp

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Subject: Re: [Tsvwg] early retransmit 
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> - You talk about [All00] and the many short http transfers. I
> don't question that result, but I think that you could maybe put
> it a bit more into perspective. From our recent measurements, we
> find that http 1.1 and especially pipelining are becoming
> increasingly popular. Thus, http transfers should be getting
> longer on average these days.

I'd certainly cite any data on the subject.  I'll go back and read
the text around [All00] and maybe soften a bit.

(The web server used for [All00] did (/does) support HTTP/1.1
persistent connections and pipelining.  However, clients could
certainly be evolving to use such features more.)

> - Concerning the solutions mentioned in the appendix. How about if
> the sender maintained a ratio of ER-based-spurious-retransmits
> over total-number-of-transmitted-segments, assuming something like
> Eifel/DSACK/FRTO-based detection. Then we could say that this
> ratio MUST not be larger than some threshold, say 2 percent. As
> soon as the ratio goes higher, we turn off ER, and turn it on
> again as soon as the ratio goes below the threshold again.

I think this is certainly a reasonable idea... maybe even the most
reasonable idea I have yet heard.  We'll definately add this to some
portion of the id.

Thanks!

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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Subject: RE: [Tsvwg] early retransmit 
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Wouldn't this also be an appropriate suggestion for Fast Retransmit
and Limited Transmit adaptive thresholding (not just ER)?

As an aside:

I have to say that I wonder about the interaction between 
adaptive transport mechanisms like this and mobile handoff 
(at either the Mobile IP or subnetwork layers) that can
significantly twiddle what the transport is looking at.

Which doesn't make Reiner's suggestion UNreasonable, but
does seem like there's still opportunities for refinement.

Spencer

> -----Original Message-----
> From: Mark Allman [mailto:mallman@grc.nasa.gov]
> > - Concerning the solutions mentioned in the appendix. How about if
> > the sender maintained a ratio of ER-based-spurious-retransmits
> > over total-number-of-transmitted-segments, assuming something like
> > Eifel/DSACK/FRTO-based detection. Then we could say that this
> > ratio MUST not be larger than some threshold, say 2 percent. As
> > soon as the ratio goes higher, we turn off ER, and turn it on
> > again as soon as the ratio goes below the threshold again.
> 
> I think this is certainly a reasonable idea... maybe even the most
> reasonable idea I have yet heard.  We'll definately add this to some
> portion of the id.
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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		: Sockets API Extensions for Stream Control Transmission
                          Protocol (SCTP)
	Author(s)	: R. Stewart et al.
	Filename	: draft-ietf-tsvwg-sctpsocket-06.txt
	Pages		: 77
	Date		: 2003-3-3
	
This document describes a mapping of the Stream Control Transmission
Protocol SCTP RFC2960 [8] into a sockets API.  The benefits of this
mapping include compatibility for TCP applications, access to new
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Greeings:

The new version of PR-SCTP is now available...

http://www.ietf.org/internet-drafts/draft-stewart-tsvwg-prsctp-03.txt

Please have a close look at it.. the authors all feel it is fully
baked and ready to move forward... comments would be most
welcome :>

The feature (its been rolling around for quite some time) provides
a partial reliablility. It has been sucessfully tested at the
last SCTP Bakeoff with two implementations.. and many present
said it would be showing up soon in their implementation as
well :->



Thanks

R


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

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Date: Tue,  4 Mar 2003 11:19:46 -0600
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] PR SCTP
To: Randall Stewart <randall@stewart.chicago.il.us>, tsvwg@ietf.org
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>3.3.1 Sending Forward-TSN-Supported param in INIT
>
> If an SCTP endpoint supports the FORWARD TSN chunk, then
> any time it sends an INIT during association
> establishment, it SHOULD include the
> Forward-TSN-supported parameter in the INIT chunk to
> indicate this fact to its peer.

Does an SCTP endpoint supporting this capability mean that
the implementation has this capability, or that the local
ULP has also enabled the capability?

It strikes me that partial reliability is only of use for
a ULP that is explicitly aware of it, and therefore would
enable it locally for the endpoint.

Caitlin Bestler
http://asomi.com/CaitlinBestler/
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Caitlin Bestler wrote:
>>3.3.1 Sending Forward-TSN-Supported param in INIT
>>
>>If an SCTP endpoint supports the FORWARD TSN chunk, then
>>any time it sends an INIT during association
>>establishment, it SHOULD include the
>>Forward-TSN-supported parameter in the INIT chunk to
>>indicate this fact to its peer.
> 
> 
> Does an SCTP endpoint supporting this capability mean that
> the implementation has this capability, or that the local
> ULP has also enabled the capability?
> 

Capability referes to the stack being able to do pr-sctp.. not
the application. The application, would need to send with
a flag to note that the message has a time limit (for example).

If the app was un-aware of the capability then it would
not get used...

> It strikes me that partial reliability is only of use for
> a ULP that is explicitly aware of it, and therefore would
> enable it locally for the endpoint.
> 
Just because the peer endpoints are capable of PR-SCTP does
not mean it would be used by the application...

I think I may be missing your point.. If the app does
not wish to use it.. it won't period.. No matter
if the two stacks support it.

The PR-SCTP document is concentrating on the protocol
aspects and mechanisms.. just like TCP timestamps or
TCP-Sack etc... if the O/S puts an interface to
disable the behavior (such as BSD has to turn on/
off timestamps or SACK via sysctl) that is outside
the scope of this document and would be more apt
to show up in a sockets api like document..

R

> Caitlin Bestler
> http://asomi.com/CaitlinBestler/
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 
> 


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

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Date: Tue,  4 Mar 2003 14:25:32 -0600
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] PR SCTP
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On 3/4/03, Randall Stewart wrote:


>
>I think I may be missing your point.. If the app does
>not wish to use it.. it won't period.. No matter
>if the two stacks support it.
>
I was seeking clarification on the issue of whether a
stack should send this information in all INITs that it
generates, or if it should only do so when there has been
some indication from the application layer that there is
a need for the feature.

In other words, should an INIT being sent for an endpoint
that has never needed partial reliability before and does
not need it now, *change* when this draft is adopted?

Or should the INITs supporting endoints that are unrelated
to partial reliability simply omit this parameter, even
if in fact they have the code to support partial reliability
on other endpoints when the application layer needs it?

Does the capability exist *for an endpoint* because the
library has been upgraded, or should the application layer
be expected to somehow activate the capability?

I believe that applications that do not require new
capabilities would prefer for the advertisements of their
endpoints to remain unchanged even as new features are
defined. If the new features are irrelevant to the
application should they change packets on the wire?


Caitlin Bestler
http://asomi.com/CaitlinBestler/
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The IESG has approved the Internet-Draft 'Robust ECN Signaling with 
Nonces' <draft-ietf-tsvwg-tcp-nonce-04.txt> as an Experimental RFC.  
This document is the product of the Transport Area Working Group. The 
IESG contact persons are Scott Bradner and Allison Mankin. 

Technical Summary

This specification describes the ECN-nonce, an optional addition to 
Explicit Congestion Notification (RFC 3168) that protects against 
accidental or malicious concealment of marked packets from the TCP 
sender. It improves the robustness of congestion control by 
preventing receivers from exploiting ECN to gain an unfair share of 
network bandwidth. The ECN-nonce uses the two ECT codepoints in the 
ECN field of the IP header, and requires a flag in the TCP header. It 
is computationally efficient for both routers and hosts.

Working Group Summary 

The working group supported publishing of this document and the Last 
Call discussion of the document raised no issues with the quality 
of the document. 

Protocol Quality

There are implementations of the nonce algorithm and bits accompanying 
the implementations of RFC 3168, the Explicit Congestion Notification 
Proposed Standard. The specification was reviewed for the IESG by 
Allison Mankin and Randy Bush.


RFC Editor Note:

RFC-Editor, please add the following text after the Abstract:

Statement of Intent:

This specification contains an optional addition to Explicit
Congestion Notification (RFC 3168) improving its robustness
against malicious or accidental concealment of marked packets.
It has not been deployed widely. One goal of publication as an
as an Experimental RFC is to be prudent, and encourage use and
deployment prior to publication in the standards track. Another
consideration is to give time for firewall developers to accept
and recognize the pattern presented by the nonce. It is
the intent of the Transport Area to re-submit this specification
as an IETF Proposed Standard in future after more experience has
been gained.

RFC-Editor, please also expand the acronyms in the abstract on first
appearance:

Explicit Congestion Notification (ECN)
ECN-Capable Transport (ECT)
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CC: tsvwg@ietf.org
Subject: Re: [Tsvwg] PR SCTP
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Caitlin Bestler wrote:
> On 3/4/03, Randall Stewart wrote:
> 
> 
> 
>>I think I may be missing your point.. If the app does
>>not wish to use it.. it won't period.. No matter
>>if the two stacks support it.
>>
> 
> I was seeking clarification on the issue of whether a
> stack should send this information in all INITs that it
> generates, or if it should only do so when there has been
> some indication from the application layer that there is
> a need for the feature.
> 
> In other words, should an INIT being sent for an endpoint
> that has never needed partial reliability before and does
> not need it now, *change* when this draft is adopted?
> 
> Or should the INITs supporting endoints that are unrelated
> to partial reliability simply omit this parameter, even
> if in fact they have the code to support partial reliability
> on other endpoints when the application layer needs it?
> 
> Does the capability exist *for an endpoint* because the
> library has been upgraded, or should the application layer
> be expected to somehow activate the capability?
> 
> I believe that applications that do not require new
> capabilities would prefer for the advertisements of their
> endpoints to remain unchanged even as new features are
> defined. If the new features are irrelevant to the
> application should they change packets on the wire?
> 
> 

I would tend to deem this a user interface issue. I could see
a O/S provider making an option that says:

Don't tell the peer that you support feature X, because I
just don't plain want to do it...

R

> Caitlin Bestler
> http://asomi.com/CaitlinBestler/
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 
> 


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

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To: Randall Stewart <randall@stewart.chicago.il.us>
cc: TSV WG list <tsvwg@ietf.org>
Subject: Re: [Tsvwg] PR SCTP 
In-Reply-To: Message from Randall Stewart <randall@stewart.chicago.il.us> 
   of "Tue, 04 Mar 2003 08:57:35 CST." <3E64BEDF.8090601@stewart.chicago.il.us> 
Date: Tue, 04 Mar 2003 20:18:26 -0800
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We will have some time on the agenda in SF to discuss
PR SCTP as well.  We're going to publish a tentative
agenda for the meeting very shortly.

Allison
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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		: The Eifel Response Algorithm for TCP
	Author(s)	: R. Ludwig, A. Gurtov
	Filename	: draft-ietf-tsvwg-tcp-eifel-response-03.txt
	Pages		: 12
	Date		: 2003-3-4
	
The Eifel response algorithm requires a detection algorithm to detect
a posteriori whether the TCP sender has entered loss recovery
unnecessarily. In response to a spurious timeout it adapts the
retransmission timer to avoid further spurious timeouts, and can
avoid - depending on the detection algorithm - the often unnecessary
go-back-N retransmits that would otherwise be sent. Likewise, it
adapts the duplicate acknowledgement threshold in response to a
spurious fast retransmit. In both cases, the Eifel response algorithm
restores the congestion control state in such a way that packet
bursts are avoided.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-tcp-eifel-response-03.txt

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we have updated the draft that specifies the Eifel response algorithm:
http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-tcp-eifel-response-03.txt

We have followed Pasi Sarolahti's suggestion and made the Eifel response 
algorithm independent of any particular detection algorithm. I.e., it 
should now be possible to also "plug" the Eifel response onto a number of 
alternative detection algorithms including DSACK-based solutions and/or F-RTO.

We would greatly welcome more feedback.

Otherwise, we believe that this draft is ready for WG-last-call to become 
an experimental RFC.

///Reiner

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Subject: Re: [Tsvwg] Eifel-Response-03
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
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Reiner,

I didn't see anything fundamentally different (apart from the timer
calculation) in this draft from the -01 draft, so all my previous
comments about not restoring the congestion window are still valid. The
rationale behind the timer calculation is also not clear to me. But let
me begin with congestion issues. 


-- The bottom line for a spurious timeout is that a spurious timeout is
spurious ONLY in terms of what packets to resend; its not spurious in
terms of congestion. Congestion is not a property of one particular
connection, but its a property of the *entire* system. So the response
algorithm MUST follow the same congestion principles as a normally
timeout connection.

Restoring congestion window after a timeout is WRONG. Just because a
particular TCP connection was waiting for some new ACK does not mean
that rest of the connections in the system would leave the congestion
state of that connection intact. The competing TCP connections will eat
up all the unused resources (buffer space). Do some analysis or run some
tests and see for yourself.

All the papers and documents that you pointed out shows only a SINGLE
TCP connection in the system (there are other CBR traffic, but CBR does
not compete like TCP does. With CBR the congestion state remains
unaltered). Unless you own an ISP, a Backbone, and the  server, you will
have more than one connection in the system.

Conceptually, the *least* a spurious timed out connection can do is to
follow RFC-2861 (and I don't know why it's not a standards track
RFC)--and that is after you discount subnet change causing spurious
timeout. Also, just see how TRFC behaves if it does not receive an ACK
for some time. Why is Eifel response exempted from all these
considerations.

Eifel response will create a havoc in case of subnet change with present
Inter Protocols (specific networks with hop-by-hop flow control may not
have this problem, but such networks are not part of IP). Again, run
some simulations (let me know if you cannot setup a test scenario with
subnet change).

-- In step ReCC, why should the sender stop processing spurious timeout 
response after 3 timeouts. Why the magic number 3? I guess a better
scheme is to just follow RFC-2861 with some modifications.

-- The timer calculation in STO.2:

SRTT <- SRTT + 1/FlightSize * (RTT-SAMPLE - SRTT)

is most disconcerting. Why should SRTT have a component of FLIGHT_SIZE.
BTW, just to refresh your memory, the RTT samples are the round trip
time for the *entire* congestion window--the samples are taken once
every window--and not for every packet. 

Are you implicitly assuming that RTT is a linear function of window
size. Well, RTT is a linear function only if you have a single queue
(one hop); for the rest of the cases, it increases non-linearly
depending upon the queuing discipline of individual router.

Besides, if you are restoring the window which you are, there is no need
to take FlightSize into consideration.

Also, in the later two steps, aren't the values of SRTT and RTTVAR
reset. Could you explain these three steps:

	SRTT <- SRTT + 1/FlightSize * (RTT-SAMPLE - SRTT)
              and set
        SRTT <- RTT-SAMPLE
        
	RTTVAR <- RTT-SAMPLE/2,

Isn't my value of SRTT same as RTT_SAMPLE after all this fancy
calculation? What am I missing here?

-- yogesh


PS:  Folks, there are many more comments in the DCLOR draft 
(draft-swami-tsvwg-tcp-dclor-00.txt) which I cannot possibly write down
here, but please read it for more arguments why Eifel response is not
right.


On Wed, 2003-03-05 at 05:51, ext Reiner Ludwig wrote:
> we have updated the draft that specifies the Eifel response algorithm:
> http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-tcp-eifel-response-03.txt
> 
> We have followed Pasi Sarolahti's suggestion and made the Eifel response 
> algorithm independent of any particular detection algorithm. I.e., it 
> should now be possible to also "plug" the Eifel response onto a number of 
> alternative detection algorithms including DSACK-based solutions and/or F-RTO.
> 
> We would greatly welcome more feedback.
> 
> Otherwise, we believe that this draft is ready for WG-last-call to become 
> an experimental RFC.
> 
> ///Reiner
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: tsvwg@ietf.org
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Hi,

I asked this in last IETF, but let me ask again. Since now we have many
congestion response algorithms, will it make sense to have a well
defined test environment against which authors can run their tests and
get results? The test environment could be an information RFC that can
be used in future too.

In the past, I have seen various drafts and papers where authors want to
evaluate the congestion behavior of their algorithm, but they only use a
single TCP connection in the entire system. This approach sometimes
gives hundreds of percents of improvement and might suit the author
needs, but when these tests are run with multiple connections the
performance goes down significantly--sometimes below the performance of
existing algorithms.

Having spent time trying to model networks, I understand that it's not
always possible to model  a 100% realistic network, but leaving it
totally un-addressed is even worse. Probably it will be useful to have a
set of guidelines: have 10 TCP connections, 5 TFRC based connections
with 5k, 10K, 100K, ... file size, travesing 10 routers with certain
buffer spaces on the path, etc., etc. These numbers could be used from
the Internet measurements (BGP statistics ... ) 

I think whenever we have multiple schemes we must run the tests along
these guidelines and pick the scheme which comes up with the best
compromise in terms of congestion or whaever the working group deems
important. 


--yogesh

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Folks,

Remember our discussion of whether this document should be
Experimental or Proposed Standard?  The kicker when IESG discussed it
was the recollection of the troubles that ensued when ECN first came
out and firewalls blocked it.  The IESG view was that presenting a
sudden Proposed Standard with a new bit pattern in TCP would not
present a stronger case for change to firewall vendors than an RFC
that was Experimental.  So the plan this time was for an explicit
approach: Steve Bellovin will contact firewall and similar developers.
We will think about a Last Call for Proposed Standard when some
openings for nonce by them have been made and reported here.  Others
who have resources to gather data about the experimental use of the
nonce, its interaction with firewalls, etc, or who can work with
firewall, intrusion detection and other such developers, are strongly
encouraged to help and report too.

Here is the final form of the Statement of Intent that will go into
the RFC for the Experimental publication.


 Statement of Intent:

 This specification contains an optional addition to Explicit
 Congestion Notification (RFC 3168) improving its robustness
 against malicious or accidental concealment of marked packets.
 It has not been deployed widely. One goal of publication as an
 as an Experimental RFC is to be prudent, and encourage use and
 deployment prior to publication in the standards track. Another
 consideration is to give time for firewall developers to accept
 and recognize the pattern presented by the nonce. It is
 the intent of the Transport Area to re-submit this specification
 as an IETF Proposed Standard in future after more experience has
 been gained.


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From: Kacheong Poon <poon@cs.wisc.edu>
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Subject: [Tsvwg] Re: [Trigtran] TRIGTRAN BoF at IETF 56
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Included message from "Spencer Dawkins" <sdawkins@cynetanetworks.com>:

>----
>Carl and I will be hosting a second TRIGTRAN BoF at IETF 56.
>
>The working copy of the BoF Agenda is available below (and is 
>being sent to agenda@ietf.org).
>
>We will present issues discussed in the Framework draft, available at 
>http://www.ietf.org/internet-drafts/draft-dawkins-trigtran-framework-00.txt.

I believe there will be a lot more comments if this announcement
is also made to the tsvwg mailing list.  So I cc this reply to
that list.

>This BoF is intended to lead to the formation of a TRIGTRAN working group,
>so most of our time together will be dedicated to answering a series of
>questions to figure out if we're doing the right thing.
>
>Please plan to attend and be opinionated.
>
>Spencer, for Spencer and Carl
>
>p.s. The Problem Statement draft will be available at
>http://www.ietf.org/internet-drafts/draft-dawkins-trigtran-probstmt-01.txt,
>as soon as it's announced. The only changes we've made were updates to
>reflect constraints in the minimal architecture model that we adopted
>at IETF 55.
>
>p.p.s. The meeting minutes from the first BOF are available at
>http://www.ietf.org/proceedings/02nov/239.htm
>
>p.p.p.s. My e-mail address is changing to spencer_dawkins@yahoo.com,
>effective immediately.

....

>AGENDA:
>
>- Agenda Bashing								10 minutes
>
>- Issues identified in TRIGTRAN Framework draft			20 minutes
>	Constraints of Minimal Architecture

Or is it really the constraint of the current Internet
architecture?  Many researchers are already looking at
the next generation Internet and this is probably one
area where they will work on.  So is it really reasonable
for this proposed WG to solve an architecture problem?
Or is what people actually want something not constrained
by the current Internet architecture?

IMHO, trigger is the wrong name to use.  Pardon my English,
but to me, trigger means an immediate response to a certain
event.  But maybe the only thing we can do is to provide
some form of signal, does not necessarily need to be immediate,
to the end points.  Signalling message seems to be a more
appropriate name.

Anyway, it seems to me that what people want here is to have a
way for transport to find out a little bit more about the network
so that it can react more effectively to different network events.
And using signals ("triggers") to do that can be one mechanism.

Should a TRIGTRAN WG be formed now if there can be other mechanisms
to do that?  Should we evaluate them before making a decision on
which one to use, signalling being one?  Another obvious mechanism,
as opposed to active signal, is polling by transport on some
network elements.  I am not suggesting that it is better or worse
(probably worse (-:) than active signal.  But I believe setting
up a WG at this point without even some discussion and evaluation
on other mechanisms is not the appropriate thing to do.

>	Form of Registrations
>	Form of Notifications
>	Allowable Responses to Notifications
>	Deployment Issues
>	Preliminary Security Assessment
>
>- Questions leading to a WG charter					50 minutes

It seems that this should be the first agenda item.  The reason
is that if people don't even think that signal ("triggers") is
the right mechanism to achieve what we want, it does not
make sense to discuss a framework document.  So maybe the two
items should be swapped?

>	- is an end-to-end solution sufficient, 
>		or do we really need help from the network?

Or as mentioned above, can we have reasonable help from
the network without some modifications to the current Internet
architecture?  Is a non-e2e solution sound?

>	- are explicit messages required?
>	- can we agree on the list of canonical triggers?

I believe the above question should be broken up.  We should
first ask if there can be an agreed list of generic network
events that we want transport to be aware of.  And then we
can look at the list and find out which one can be most likely
"told" to transport.  Lastly, we should find the most
appropriate mechanism to do that given the current Internet
architecture.  A WG can be formed to achieve that.  The above
framework document can then be used as one of the documents
describing one mechanism being investigated.

I think that may be a more appropriate WG to be formed instead
of a TRIGTRAN WG.  I will not go to the IETF, so please count my
vote as a NO to the formation of a TRIGTRAN WG, if that question
will actually be asked.

Or is the above problem still a research issue at this moment?

>	- can these triggers be generated by subnetwork technologies?
>	- can we agree on a list of reasonable responses?
>	- who would use TRIGTRAN notifications? (OMA, SIP, etc.)?
>	- who would implement a reasonable TRIGTRAN specification?
>
>- Next Steps leading to a TRIGTRAN working group		40 minutes
>	Volunteers
>	Documents
>	Timeline
>	Land Mines
>----


							K. Poon.
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On 5 Mar 2003, Yogesh Prem Swami wrote:

> Having spent time trying to model networks, I understand that it's not
> always possible to model  a 100% realistic network, but leaving it
> totally un-addressed is even worse. Probably it will be useful to have a
> set of guidelines: have 10 TCP connections, 5 TFRC based connections
> with 5k, 10K, 100K, ... file size, travesing 10 routers with certain
> buffer spaces on the path, etc., etc. These numbers could be used from
> the Internet measurements (BGP statistics ... )

Wouldn't an aggregation of Pareto ON-OFF sources be enough? The
cross-traffic should be self-similar. A few TCP and TFRC connections
probably will not give you the arrival pattern that a router actually
experiences.

I think the idea of an Information RFC as you suggest would be a good
idea, but I'd be really interested in seeing a large group agree on
something like this. :-)

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
0--                                                  --0

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To: Yogesh Prem Swami <yogesh.swami@nokia.com>
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Eifel-Response-03
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At 18:49 05.03.2003, Yogesh Prem Swami wrote:
>The
>rationale behind the timer calculation is also not clear to me.

That point is taken. In the meantime, we have had discussions offline from 
this list about "making the retransmission timer more conservative in 
response to a detected spurious timeout".

Yes, we are convinced. That part needs to be changed.


>Restoring congestion window after a timeout is WRONG.

We have had that discussion before, and you won't convince us just because 
you get more excited about this. Our views and arguments about this issue 
are presented in section 3.4 of the Eifel response draft. You and I have 
different views on this, and that's fine. Let's hear from others instead of 
restating the same opinions over and over again.


The interworking with Congestion Window Validation (CWV) [RFC2861] that you 
brought up, I find relevant, though ...

CWV kicks in if
- the connection has been *idle* for longer than one RTO, or
- the connection has been *application-limited* for longer than one RTO.

In RFC2861, the term *application-limited* is well defined ("When TCP sends 
a packet that does not fill the congestion window, and the TCP send queue 
is empty, then the sender is application-limited."), but unfortunately the 
term *idle* is not so well defined; also not in the CWV pseudo-code. What 
is missing is that *idle* means: "no data is outstanding". This is how V. 
Jacobson defined it in his '92 revision of his SIGCOMM '88 paper (appendix 
C), and that's also how it's implemented in BSD.

With the Eifel response algorithm the situation is different. Once a 
spurious timeout has been detected, the connection had apparently *not* 
been idle before.
Thus, that part of the CWV algorithm should not kick in. The second part of 
the CWV algorithm that deals with *application-limited* connections is 
already implicitly included in the Eifel response algorithm: cwnd is at 
most restored to the current flightsize + IW. This basically follows the 
spirit and intention of the CWV algorithm.

So, I think you have a valid point. In the next revision of the Eifel 
response draft, we should say that the CWV algorithm should be "reset". For 
example, by setting the CWV variables T_last and T_prev to tcpnow.

///Reiner

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Hi guys, 

The new version of the SCTP MIB has been recently published. SCTP-MIB v09 is the result of a long revision process with MIB doctors and SIGTRAN advisors after the WGLC (thanks guys). 

Here you have the a summary of the changes we have made:

- Copyright notice included
- The Boiler plate has been updated.
- Description sections have been reworked
- WG information included in the MODULE-IDENTITY
- Typos
- Removed size limitations from InetAddress. It is instead explained in the DESCRIPTION clause
- sctpAssocRemAddressHBActive object type changed from Integer to Truthvalue
- Values by default added.
- Some objects have been renamed to mantain naming consistency
- Some objects description have been updated/reworded
- In some objects we have added references to the related chapter in RFC2960
- DNS names are recommended in ASCII only. Not UTF.
- Discontinuity time counter added for the whole SCTP layer
- New Module group created: sctpPerAssocStatsGroup included as a Mandatory Group
- References polished and updated

Enjoy the reading.

Cheers // MCarmen and Javier.


> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: miércoles, 05 de marzo de 2003 12:33
> Cc: sigtran@ietf.org
> Subject: [Sigtran] I-D ACTION:draft-ietf-sigtran-sctp-mib-09.txt
> 
> 
> A New Internet-Draft is available from the on-line 
> Internet-Drafts directories.
> This draft is a work item of the Signaling Transport Working 
> Group of the IETF.
> 
> 	Title		: Stream Control Transmission Protocol 
> Management 
>                           Information Base
> 	Author(s)	: J. Pastor, M. Belinchon
> 	Filename	: draft-ietf-sigtran-sctp-mib-09.txt
> 	Pages		: 45
> 	Date		: 2003-3-4
> 	
> The Stream Control Transmission Protocol (SCTP) is a reliable
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> network such as IP, designed to transport PSTN signaling messages
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> applications.
> This memo defines the Management Information Base (MIB) module which
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> implementation of the SCTP.
> 
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On Thu, 6 Mar 2003, Reiner Ludwig wrote:

> At 18:49 05.03.2003, Yogesh Prem Swami wrote:
> >Restoring congestion window after a timeout is WRONG.
>
> We have had that discussion before, and you won't convince us just because
> you get more excited about this. Our views and arguments about this issue
> are presented in section 3.4 of the Eifel response draft. You and I have
> different views on this, and that's fine. Let's hear from others instead of
> restating the same opinions over and over again.

I was ready to say that I agree with Yogesh, but...

> the CWV algorithm that deals with *application-limited* connections is
> already implicitly included in the Eifel response algorithm: cwnd is at
> most restored to the current flightsize + IW. This basically follows the
> spirit and intention of the CWV algorithm.

If cwnd restoration is as explained above, then I think "restoring cwnd"
is fine.

I quickly skimmed the draft and saw on page 5, the step for reverting cwnd
includes the following text:

              If the acceptable ACK has the ECN-Echo flag [RFC3168] set
              OR the TCP sender has already taken more than three
              timeouts for the oldest outstanding segment, then proceed
              to step (DONE),

How was the value three chosen?

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
0--                                                  --0

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Hello Reiner, all,

On Thu, 2003-03-06 at 09:23, ext Reiner Ludwig wrote

> are presented in section 3.4 of the Eifel response draft. You and I have 

Here's the excerpt from 3.4:


	"When a TCP sender enters loss recovery, it also assumes that is 	has
received a congestion indication. In response to that it 	reduces cwnd,
and ssthresh. However, once the TCP sender detects 	that the loss
recovery has been falsely triggered, this 	reduction was unnecessary. In
fact, no congestion signal has 	been received. We therefore believe that
it is safe to revert to 	the previous congestion control state."

If I go by this logic, then why shall I start with a small window size
at the start of the connection? At the start of the connection I have no
congestion indication, so I should be able to send what ever amount of
data I want? 

Are you suggesting we should start with a window of say 10M bytes and
wait until we receive congestion indication? Your logic seems to imply
that.

> The interworking with Congestion Window Validation (CWV) [RFC2861] that you 
> brought up, I find relevant, though ...
> 
> CWV kicks in if
> - the connection has been *idle* for longer than one RTO, or
> - the connection has been *application-limited* for longer than one RTO.
> 
> In RFC2861, the term *application-limited* is well defined ("When TCP sends 
> a packet that does not fill the congestion window, and the TCP send queue 
> is empty, then the sender is application-limited."), but unfortunately the 
> term *idle* is not so well defined; also not in the CWV pseudo-code. What 

idle is any time period greater than RTO. RTO is an statistical upper
bound during which the *end to end state of the network* will remain
*roughly* (no guarantees) the same as far as congestion is concerned.
Beyond that, there is no reason to believe that network will be in the
same state.


> is missing is that *idle* means: "no data is outstanding". This is how V. 
> Jacobson defined it in his '92 revision of his SIGCOMM '88 paper (appendix 
> C), and that's also how it's implemented in BSD.
> 

Congestion control is about the state of the network, not about how you
define certain things. If the state of the network changes because the
sender was not sending data for more than RTO, then that is a valid
candidate for CVW--regardless of the definition used.

In addition, CVS works only when there is no host mobility.

> With the Eifel response algorithm the situation is different. Once a 
> spurious timeout has been detected, the connection had apparently *not* 
> been idle before.

The connection is not idle, but the network has been. Beyond RTO you
don't know what happened in the network.

> Thus, that part of the CWV algorithm should not kick in. The second part of 
> the CWV algorithm that deals with *application-limited* connections is 
> already implicitly included in the Eifel response algorithm: cwnd is at 
> most restored to the current flightsize + IW. This basically follows the 
> spirit and intention of the CWV algorithm.


It doesn't matter whether it's application limited or idle. The simple
rule is: if you have used your share of network resources within RTO,
you can continue using that share. If you have not, you are required to
start from scratch.

But I think it will be good to get some feedback from other people too. 

I strongly disagree with the present response algorithm because it
reduces the over all throughput of the system. Again, I emphsize "the
system" and not one particular TCP connection. We have also put the
results in appendix of the DCLOR draft to see the difference. 

--yogesh

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Subject: Re: [Tsvwg] Eifel-Response-03
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: "Armando L. Caro Jr." <acaro@acm.org>
Cc: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
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Hi,


On Thu, 2003-03-06 at 10:28, ext Armando L. Caro Jr. wrote:

> 
> > the CWV algorithm that deals with *application-limited* connections is
> > already implicitly included in the Eifel response algorithm: cwnd is at
> > most restored to the current flightsize + IW. This basically follows the
> > spirit and intention of the CWV algorithm.
> 
> If cwnd restoration is as explained above, then I think "restoring cwnd"
> is fine.
> 
> I quickly skimmed the draft and saw on page 5, the step for reverting cwnd
> includes the following text:
> 
>               If the acceptable ACK has the ECN-Echo flag [RFC3168] set
>               OR the TCP sender has already taken more than three
>               timeouts for the oldest outstanding segment, then proceed
>               to step (DONE),
> 


The ECN-ECHO doesn't say anything. Consider this case


		 Part-A			        Part-B
	Sender ----------> pkt stalling node ----------> receiver

The Congestion Experience could have been set either on Part-A or on
Part-B, but the ACKs received after the spurious timeout only tell the
state of the network either

a) for the entire path but before RTO, or
b) only Part-B.

In either case, the sender does not know what the current state of
network is on Part-A. It's possible that Part-B was not congested but
Part-A got congestion during the RTO period and you will not receiver
any ECN-Echo in that case.

On the other hand, if the network did get better during the stalling
period, it's better that the sender should start using the resources at
a faster rate. 

Eifel response fails in both the cases. 

--yogesh

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Subject: Re: [Tsvwg] Test environments for congestion response
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: "Armando L. Caro Jr." <acaro@acm.org>
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Hello Armando,


On Wed, 2003-03-05 at 23:10, ext Armando L. Caro Jr. wrote:
> On 5 Mar 2003, Yogesh Prem Swami wrote:
> 
> > Having spent time trying to model networks, I understand that it's not
> > always possible to model  a 100% realistic network, but leaving it
> > totally un-addressed is even worse. Probably it will be useful to have a
> > set of guidelines: have 10 TCP connections, 5 TFRC based connections
> > with 5k, 10K, 100K, ... file size, travesing 10 routers with certain
> > buffer spaces on the path, etc., etc. These numbers could be used from
> > the Internet measurements (BGP statistics ... )
> 
> Wouldn't an aggregation of Pareto ON-OFF sources be enough? The
> cross-traffic should be self-similar. A few TCP and TFRC connections
> probably will not give you the arrival pattern that a router actually
> experiences.
> 

I agree. What is needed is a realistic scenario which can mimic the
network behaviour in a certain way. I am appalled by results which show
300% improvent in TCP performance, but of course the authors were using
loopback addresses for testing their improved schemes. [Just
exaggerating ...]

> I think the idea of an Information RFC as you suggest would be a good
> idea, but I'd be really interested in seeing a large group agree on
> something like this. :-)
> 

Yes, but not defining anything is probably worse. 

--yogesh

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Just a couple of points on Kacheong's note:

Spencer

--- Kacheong Poon <poon@cs.wisc.edu> wrote:
> >AGENDA:
> >
> >- Agenda Bashing								10 minutes
> >
> >- Issues identified in TRIGTRAN Framework draft			20 minutes
> >	Constraints of Minimal Architecture
> 
> Or is it really the constraint of the current Internet
> architecture?  Many researchers are already looking at
> the next generation Internet and this is probably one
> area where they will work on.  So is it really reasonable
> for this proposed WG to solve an architecture problem?
> Or is what people actually want something not constrained
> by the current Internet architecture?

No, this is actually a topic requested by some people who have
asked "what about links that aren't access links?" and similar
questions. It's just explaining why we excluded some topologies
from consideration.

> 
> IMHO, trigger is the wrong name to use.  Pardon my English,
> but to me, trigger means an immediate response to a certain
> event.  But maybe the only thing we can do is to provide
> some form of signal, does not necessarily need to be
> immediate,
> to the end points.  Signalling message seems to be a more
> appropriate name.

You may very well be correct, that we might think of a more
helpful name (although I note, tongue-in-cheek, that Signalling
for Transport/SIGTRAN seems to be previously occuppied - sorry,
Lyndon!)

> 
> Anyway, it seems to me that what people want here is to have a
> way for transport to find out a little bit more about the
> network
> so that it can react more effectively to different network
> events.
> And using signals ("triggers") to do that can be one
> mechanism.

This is my understanding also.

> 
> Should a TRIGTRAN WG be formed now if there can be other
> mechanisms
> to do that?  Should we evaluate them before making a decision
> on
> which one to use, signalling being one?  Another obvious
> mechanism,
> as opposed to active signal, is polling by transport on some
> network elements.  I am not suggesting that it is better or
> worse
> (probably worse (-:) than active signal.  But I believe
> setting
> up a WG at this point without even some discussion and
> evaluation
> on other mechanisms is not the appropriate thing to do.

I'm not sure that I've done a good job of making something
clear, so let me try here:

The framework draft Carl and I put together is (at most) one way
the problem might be addressed. I'm very clear in my mind that
it's not baseline for a working group draft, with as little
discussion as we've had on some of the approaches. 

If $TRIGTRAN (or a more suitable name) is chartered, I'd expect
the charter to include a more rigorous requirements document,
and I'd be asking for other frameworks (maybe not polling, but
I'm sure there's something! perhaps defining a standardized
mechanism for link-up, and end-to-end messaging beyond link-down
and link-up? but I'm getting ahead of where the discussion has
been, at least to date).

So, to be clearer than I have been in the past, Carl and I
aren't tied to a particular working group name, set of
requirements, or set of protocol mechanisms, at this time.
Please send text!

Thanks for your feedback, and for your suggestions on a detailed
way forward.

Spencer, for Spencer and Carl

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From: Ethan Blanton <eblanton@cs.ohiou.edu>
To: Yogesh Prem Swami <yogesh.swami@nokia.com>
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Yogesh Prem Swami spake unto us the following wisdom:
> Here's the excerpt from 3.4:=20
>        "When a TCP sender enters loss recovery, it also assumes
>        that is has received a congestion indication. In response to
>        that it reduces cwnd, and ssthresh. However, once the TCP
>        sender detects that the loss recovery has been falsely
>        triggered, this reduction was unnecessary. In fact, no
>        congestion signal has been received. We therefore believe
>        that it is safe to revert to the previous congestion control
>        state."
>=20
> If I go by this logic, then why shall I start with a small window size
> at the start of the connection? At the start of the connection I have no
> congestion indication, so I should be able to send what ever amount of
> data I want?=20

I don't think this is equivalent at all to what Reiner's draft is
saying.  His draft is more equivalent to saying "As long as I am
continuing with normal TCP transmission and getting normal
acknowledgements, it's OK to use congestion avoidance as usual."  The
basic point here is that we *thought* we saw congestion, but we didn't
-- we busted our own connection by using an apparently bogus RTO
timer.  As another message in this thread indicated, perhaps finding a
way to fix that RTO is called for, but that is orthogonal to the point
you are questioning here.

> Are you suggesting we should start with a window of say 10M bytes and
> wait until we receive congestion indication? Your logic seems to imply
> that.

I'm suggesting (as I agree with Reiner, and I think this is what his
text says) that we should use the accepted TCP congestion avoidance
and congestion control routines, and take advantage of the fact that
we can detect when we made a mistake.

> I strongly disagree with the present response algorithm because it
> reduces the over all throughput of the system. Again, I emphsize "the
> system" and not one particular TCP connection. We have also put the
> results in appendix of the DCLOR draft to see the difference.=20

I think saying it reduces the overall throughput of every system is a
bold claim ... I would believe that it reduces throughput in some
circumstances, but I would like to believe that it improves the
throughput for the connection in question without adversely affecting
the rest of the network in many or most circumstances.

Ethan

--=20
Happiness is a belt-fed weapon.

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Subject: Re: [Tsvwg] Eifel-Response-03
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On 6 Mar 2003, Yogesh Prem Swami wrote:

> Here's the excerpt from 3.4:
>
>
> 	"When a TCP sender enters loss recovery, it also assumes that is 	has
> received a congestion indication. In response to that it 	reduces cwnd,
> and ssthresh. However, once the TCP sender detects 	that the loss
> recovery has been falsely triggered, this 	reduction was unnecessary. In
> fact, no congestion signal has 	been received. We therefore believe that
> it is safe to revert to 	the previous congestion control state."
>
> If I go by this logic, then why shall I start with a small window size
> at the start of the connection? At the start of the connection I have no
> congestion indication, so I should be able to send what ever amount of
> data I want?
>
> Are you suggesting we should start with a window of say 10M bytes and
> wait until we receive congestion indication? Your logic seems to imply
> that.

I agree with you on principal, but I think you are missing something. From
section 3.1 (The Algorithm), the cwnd restoration algorithm is:

		  cwnd <- min (pipe_prev, (FlightSize + IW))
                  ssthresh <- pipe_prev

So this algorithm does use the CWV principles from RFC 2861.

I would suggest to Reiner to reword text references to cwnd restoration so
that the text better reflects the actual algorithm.

(FYI: I will read the response drafts in more detail soon and give my
comments.)

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
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Subject: Re: [Tsvwg] Eifel-Response-03
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: ext Ethan Blanton <eblanton@cs.ohiou.edu>
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Hi,

On Thu, 2003-03-06 at 11:20, ext Ethan Blanton wrote:
> Yogesh Prem Swami spake unto us the following wisdom:
> > Here's the excerpt from 3.4: 
> >        "When a TCP sender enters loss recovery, it also assumes
> >        that is has received a congestion indication. In response to
> >        that it reduces cwnd, and ssthresh. However, once the TCP
> >        sender detects that the loss recovery has been falsely
> >        triggered, this reduction was unnecessary. In fact, no
> >        congestion signal has been received. We therefore believe
> >        that it is safe to revert to the previous congestion control
> >        state."
> > 
> > If I go by this logic, then why shall I start with a small window size
> > at the start of the connection? At the start of the connection I have no
> > congestion indication, so I should be able to send what ever amount of
> > data I want? 
> 
> I don't think this is equivalent at all to what Reiner's draft is
> saying.  His draft is more equivalent to saying "As long as I am
> continuing with normal TCP transmission and getting normal
> acknowledgements, it's OK to use congestion avoidance as usual."  The

That's precisely the point. During the spurious timeout period, the
sender is not getting the ACKs in the regular fashion, so congestion
avoidance makes no sense.  During that period the other TCP connections
in the system were still doing fine and they will use up all the
resources of the network. Please see section 2.2 of DCLOR draft
(draft-swami-tsvwg-tcp-dclor-00.txt) for an analysis.

> basic point here is that we *thought* we saw congestion, but we didn't

we cannot claim that we *didn't see* congestion; the congestion
informaiotn on the first part (the part between the sender and stalling
node) is more than an RTO old so its of little use. All we can claim is
that we don't know what happened during that idle period. 

> I'm suggesting (as I agree with Reiner, and I think this is what his
> text says) that we should use the accepted TCP congestion avoidance
> and congestion control routines, and take advantage of the fact that
> we can detect when we made a mistake.

Yes we can detect congestion after losing packets, but that would be too
late. Consider the case where there are repeated spurious timeouts
(which will happen if there are many mobile users moving from one subnet
to another). In that case, the network will keep losing packets from
random set of senders.

As long as spurious timeouts happen once in a while, it might be
acceptable to restore the window. But once the number of spurious
timeouts increases, the system will the hardly hit.

> 
> > I strongly disagree with the present response algorithm because it
> > reduces the over all throughput of the system. Again, I emphsize "the
> > system" and not one particular TCP connection. We have also put the
> > results in appendix of the DCLOR draft to see the difference. 
> 
> I think saying it reduces the overall throughput of every system is a
> bold claim ... I would believe that it reduces throughput in some

Indeed it's a bold claim, and that's why I am asking the working group
to define a test environment which will consider all possible causes of
spurious timeouts. This will also allow an independent observer to test
each of these schemes.

I think this should be acceptable to everyone.

> circumstances, but I would like to believe that it improves the
> throughput for the connection in question without adversely affecting
> the rest of the network in many or most circumstances.
> 

Could you please provide some data of these circumstances. From my
experiments, I have found the performance to go down (I used the Linux
distribution of Eifel.) in almost all cases except where there are just
one or two TCP connections in the system.

--yogesh

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Armando L. Caro Jr. wrote:
> On Thu, 6 Mar 2003, Reiner Ludwig wrote:
> 
> 
>>At 18:49 05.03.2003, Yogesh Prem Swami wrote:
>>
>>>Restoring congestion window after a timeout is WRONG.
>>
>>We have had that discussion before, and you won't convince us just because
>>you get more excited about this. Our views and arguments about this issue
>>are presented in section 3.4 of the Eifel response draft. You and I have
>>different views on this, and that's fine. Let's hear from others instead of
>>restating the same opinions over and over again.
> 
> 
> I was ready to say that I agree with Yogesh, but...
> 
I like Armando was ready to agree with Yogesh.. then at the
prompting of this set of exchanges I went and read the draft ;->


> 
>>the CWV algorithm that deals with *application-limited* connections is
>>already implicitly included in the Eifel response algorithm: cwnd is at
>>most restored to the current flightsize + IW. This basically follows the
>>spirit and intention of the CWV algorithm.
> 

I think FWIW I am ok with restoring to min(prevfs, (currfs+IW))

But I do have one question..

Would this ever other return anything but currfs+IW?

R

> 
> If cwnd restoration is as explained above, then I think "restoring cwnd"
> is fine.
> 
> I quickly skimmed the draft and saw on page 5, the step for reverting cwnd
> includes the following text:
> 
>               If the acceptable ACK has the ECN-Echo flag [RFC3168] set
>               OR the TCP sender has already taken more than three
>               timeouts for the oldest outstanding segment, then proceed
>               to step (DONE),
> 
> How was the value three chosen?
> 
> ~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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> 
> 


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

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On Thu, 6 Mar 2003, Randall Stewart wrote:

> I think FWIW I am ok with restoring to min(prevfs, (currfs+IW))
>
> But I do have one question..
>
> Would this ever other return anything but currfs+IW?

Yeah, I don't see how prev_flight is ever achieved... unless the
spurious rtx detection is REALLY late.

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
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From: Ethan Blanton <eblanton@cs.ohiou.edu>
To: Yogesh Prem Swami <yogesh.swami@nokia.com>
Cc: ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
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Yogesh Prem Swami spake unto us the following wisdom:
> As long as spurious timeouts happen once in a while, it might be
> acceptable to restore the window. But once the number of spurious
> timeouts increases, the system will the hardly hit.

Now on this point, I *completely* agree.  Our experiments with DSACK
suggested that if these bogus retransmissions (of any kind) are not
curbed, they will become a significant problem in and of themselves.
However, as Reiner has I believe pointed out in another part of this
thread, this is a known problem ... and I would agree with you that it
needs a little bit of face time in this draft or any other draft
addressing the issue of allowing congestion state reversion.
Obviously we do not want to continually take false timeouts and spew
unnecessary retransmits into the network in any event.

If this is the crux of your position, then I am in agreement.

Ethan

--=20
Happiness is a belt-fed weapon.

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From: Mark Allman <mallman@grc.nasa.gov>
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> Yogesh Prem Swami spake unto us the following wisdom:
> > As long as spurious timeouts happen once in a while, it might be
> > acceptable to restore the window. But once the number of spurious
> > timeouts increases, the system will the hardly hit.
> 
> Now on this point, I *completely* agree.  Our experiments with
> DSACK suggested that if these bogus retransmissions (of any kind)
> are not curbed, they will become a significant problem in and of
> themselves.  However, as Reiner has I believe pointed out in
> another part of this thread, this is a known problem ... and I
> would agree with you that it needs a little bit of face time in
> this draft or any other draft addressing the issue of allowing
> congestion state reversion.  Obviously we do not want to
> continually take false timeouts and spew unnecessary retransmits
> into the network in any event.
> 
> If this is the crux of your position, then I am in agreement.

I agree that spuriously retransmitting continuously is a Bad Thing.
But, I want to note that Reiner's draft (& our draft on what to do
in the face of reordering) takes action to attempt to avoid this
situation in the future (i.e., so it doesn't happen continuously).
That is, the RTO or the dupthresh is adjusted in an effort to
prevent continuous bogosity.  And, so I think it is a bit unfair to
argue against Reiner's intent here (if you want to claim that his
algorithms do not go far enough, I think that is fine -- but, I
think that is a different argument all together).

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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To: Yogesh Prem Swami <yogesh.swami@nokia.com>
From: Mark Allman <mallman@grc.nasa.gov>
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> If I go by this logic, then why shall I start with a small window
> size at the start of the connection? At the start of the
> connection I have no congestion indication, so I should be able to
> send what ever amount of data I want?
> 
> Are you suggesting we should start with a window of say 10M bytes
> and wait until we receive congestion indication? Your logic seems
> to imply that.

I disagree with the above analysis.  I hope the following is very
neutral -- I mean it to neither support nor refute your overall
argument (I am still rolling that around in my head).

At the beginning of a connection you have no congestion indication,
indeed.  But, also you have no indication of what the cwnd ought to
be.  When we spuriously timeout we have a cwnd that has been
validated by the network path.  Now, that might or might not be
valid after the suprious timeout--- rather my point si that we have
*some notion* about what cwnd ought to be, as opposed to the start
of a connection when we have none.

allman
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Subject: Re: [Tsvwg] Eifel-Response-03
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: mallman@grc.nasa.gov
Cc: ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
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Hi Marc, all,

On Thu, 2003-03-06 at 13:20, ext Mark Allman wrote:
> > If I go by this logic, then why shall I start with a small window
> > size at the start of the connection? At the start of the
> > connection I have no congestion indication, so I should be able to
> > send what ever amount of data I want?
> > 
> > Are you suggesting we should start with a window of say 10M bytes
> > and wait until we receive congestion indication? Your logic seems
> > to imply that.
> 
> I disagree with the above analysis.  I hope the following is very
> neutral -- I mean it to neither support nor refute your overall
> argument (I am still rolling that around in my head).
> 
> At the beginning of a connection you have no congestion indication,
> indeed.  But, also you have no indication of what the cwnd ought to
> be.  When we spuriously timeout we have a cwnd that has been
> validated by the network path.  Now, that might or might not be
> valid after the suprious timeout--- rather my point si that we have
> *some notion* about what cwnd ought to be, as opposed to the start
> of a connection when we have none.
> 

Yes, I agree that the sender has a some stale information about the
state of the network, but that information, as you note, might not be of
any use. The answer to the question of how good the information actually
is, depends on the how long the spurious timeout lasted

Let us do some simplified calculations (I am using the same calculation
in my DCLOR draft; I apologize if you have already seen this). Lets say
there are M competing connection (apart from the connection which was
interrupted) and each of them has a Round Trip Time or R (this is a
simplifying assumption; I am by no means suggestion it's the case).
Let's also say that that the stalling duration is K*R, where K is some
integer constant (again just a simplified case to the feel for things).
Let's also assume that all the competing connections were doing
congestion aovidance.

In this case while the stalled connection was waiting for
acknowledgements, rest of the M competing connections would increase
their congestoin window 1 segment per RTT. So by the end of the spurious
timeout, the effective buffer space for the defaulting connection will
become:

	cwnd - M*K 


As we can see this number grows linealy with K (that is the stalling
period) and with the number of competing connections. There is NO way
for a TCP sender to know how many competing connections are there in the
system.

Now if I restore the window (in this idealized scenario) then there will
be M*K packet loss. In reality this is not so direct due to
retransmissions etc, but it's close (at least in our emulation which had
differnet uniform distributed RTT values)

All this analysis is true as long as the spurious timeout is not caused
due to user's mobility. With user's mobility, things are worse.

In the DCLOR draft we take the most conservative approach to start with
a small window size and run slow-start. This prevents from packet loss,
and also improves thoughput. This approach is also in sync with RFC-793
which recommends "being conservative in what you send, and liberal in
what you accept."

BTW, it's useful to point out that fast-retransmit, fast-recovery
recovers one packet per RTT. On the other hand, Slow start can recover
packet at a much faster rate if the number of lost packets is large.
There is a lot more information in Janey C. Hoe's MS thesis (Strart Up
dynamics of TCP's congestion control and avoidance schemes)

So coming back to the argument of initial window, a spurious timeout is
no different from a new connection. The information that the sender has
is too old and has no value.

--yogesh

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Marc,


On Thu, 2003-03-06 at 13:02, ext Mark Allman wrote:
> > Yogesh Prem Swami spake unto us the following wisdom:
> > > As long as spurious timeouts happen once in a while, it might be
> > > acceptable to restore the window. But once the number of spurious
> > > timeouts increases, the system will the hardly hit.
> > 
> > Now on this point, I *completely* agree.  Our experiments with
> > DSACK suggested that if these bogus retransmissions (of any kind)
> > are not curbed, they will become a significant problem in and of
> > themselves.  However, as Reiner has I believe pointed out in
> > another part of this thread, this is a known problem ... and I
> > would agree with you that it needs a little bit of face time in
> > this draft or any other draft addressing the issue of allowing
> > congestion state reversion.  Obviously we do not want to
> > continually take false timeouts and spew unnecessary retransmits
> > into the network in any event.
> > 
> > If this is the crux of your position, then I am in agreement.
> 
> I agree that spuriously retransmitting continuously is a Bad Thing.
> But, I want to note that Reiner's draft (& our draft on what to do
> in the face of reordering) takes action to attempt to avoid this

One can avoid (although I'm still skeptical how this can be done without
hurting the thoughput in case of real timeouts) spurious timeouts if the
same sender keeps timing out over and over again. But if the sender's
are new each time--which is most likey the case--then one cannot do
anything. The effect on the network is, however, still the same.


--yogesh

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> > I agree that spuriously retransmitting continuously is a Bad
> > Thing.  But, I want to note that Reiner's draft (& our draft on
> > what to do in the face of reordering) takes action to attempt to
> > avoid this
> 
> One can avoid (although I'm still skeptical how this can be done
> without hurting the thoughput in case of real timeouts) spurious
> timeouts if the same sender keeps timing out over and over
> again.

You want to eat your cake, too?

> But if the sender's are new each time--which is most likey the
> case--then one cannot do anything. The effect on the network is,
> however, still the same.

Good point.

(A theoretical argument is that we could share this information
between connections.  That might help some.  But, in practice we
don't share information -- and, I am not even sure how much it would
help in this case.)

allman


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> -- The bottom line for a spurious timeout is that a spurious
> timeout is spurious ONLY in terms of what packets to resend; its
> not spurious in terms of congestion.
>
> [...]
> 
> Restoring congestion window after a timeout is WRONG. 

I think you bring up an interesting thought.  I am not sure we have
ever actually defined what "congestion" means in an RFC (and, if we
have someone please correct me).  We use drops as implicit
indications of "congestion".  We use ECN bits as explicit
indications of "congestion".  However, that does not preclude the
use of other indications, I would say.

So then the question is: when the retransmission timer fires what
part of that process is the congestion indication?

  * Is it the unexpectedly long time we spent waiting for feedback
    (an ACK) for some segment?

  * Or, is it that we believe something has been dropped (and, so we
    are now retransmitting)?

I *think* I have a hit on this.  But, I am going to roll it around
my head a bit so I can be coherent when I try to explain it.  But,
I'd be interested in hearing other's thoughts, too.

allman


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On 6 Mar 2003, Yogesh Prem Swami wrote:

> Yes, I agree that the sender has a some stale information about the
> state of the network, but that information, as you note, might not be of
> any use. The answer to the question of how good the information actually
> is, depends on the how long the spurious timeout lasted
>
> Let us do some simplified calculations (I am using the same calculation
> in my DCLOR draft; I apologize if you have already seen this). Lets say
> there are M competing connection (apart from the connection which was
> interrupted) and each of them has a Round Trip Time or R (this is a
> simplifying assumption; I am by no means suggestion it's the case).
> Let's also say that that the stalling duration is K*R, where K is some
> integer constant (again just a simplified case to the feel for things).
> Let's also assume that all the competing connections were doing
> congestion aovidance.
>
> In this case while the stalled connection was waiting for
> acknowledgements, rest of the M competing connections would increase
> their congestoin window 1 segment per RTT. So by the end of the spurious
> timeout, the effective buffer space for the defaulting connection will
> become:
>
> 	cwnd - M*K
>
>
> As we can see this number grows linealy with K (that is the stalling
> period) and with the number of competing connections. There is NO way
> for a TCP sender to know how many competing connections are there in the
> system.

I think you have a flaw in your analysis. You are assuming that the
competing flows are synchronized, which is not the case in a real
networks.  I agree with you that the "available bandwidth" for a flow does
fluctuate, but you cannot simply say that one TCP connection reducing its
sending rate will cause others to increase x amount in y amount of time.
The congestion state at routers is bursty... connections may get drops
just because of coincidental burst at the time it decided to send.

~armando

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| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
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On Thu, 6 Mar 2003, Mark Allman wrote:

> > But if the sender's are new each time--which is most likey the
> > case--then one cannot do anything. The effect on the network is,
> > however, still the same.
>
> Good point.
>
> (A theoretical argument is that we could share this information
> between connections.  That might help some.  But, in practice we
> don't share information -- and, I am not even sure how much it would
> help in this case.)

I don't think sharing this information makes much sense. My understanding
is that detecting spurious timeouts is useful in wireless scenarios where
RTTs may fluctuate much more due to L2 rtxs. In these cases, spikes in
RTTs are probably due noise, which is temporary. Thus, it is not something
meaningful to convey to other new connections. For wired netwoks, RTT
spikes are less frequent, but still not meaningful to share.

PLEASE correct me if I am totally off here...

~armando

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| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
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Subject: Re: [Tsvwg] Eifel-Response-03 
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On Thu, 6 Mar 2003, Mark Allman wrote:

> > If I go by this logic, then why shall I start with a small window
> > size at the start of the connection? At the start of the
> > connection I have no congestion indication, so I should be able to
> > send what ever amount of data I want?
> >
> > Are you suggesting we should start with a window of say 10M bytes
> > and wait until we receive congestion indication? Your logic seems
> > to imply that.
>
> I disagree with the above analysis.  I hope the following is very
> neutral -- I mean it to neither support nor refute your overall
> argument (I am still rolling that around in my head).
>
> At the beginning of a connection you have no congestion indication,
> indeed.  But, also you have no indication of what the cwnd ought to
> be.

Now THIS information would be a good idea to share :-)

(Yeah, I know... people have already thought of that)

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
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Subject: Re: [Tsvwg] Eifel-Response-03 
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On Thu, 6 Mar 2003, Mark Allman wrote:

> So then the question is: when the retransmission timer fires what
> part of that process is the congestion indication?
>
>   * Is it the unexpectedly long time we spent waiting for feedback
>     (an ACK) for some segment?
>
>   * Or, is it that we believe something has been dropped (and, so we
>     are now retransmitting)?

Both :-)  It's just that the drop is a clearer indiation (in wired
networks) that there is congestion. Without the loss, a long delay does
imply congestion, but the question is how to act on it. I believe that TCP
Westwood and TCP Vegas provide some direction here.

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
0--                                                  --0

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Subject: Re: [Tsvwg] Eifel-Response-03
From: Pasi Sarolahti <pasi.sarolahti@nokia.com>
To: "Armando L. Caro Jr." <acaro@acm.org>
Cc: Mark Allman <mallman@grc.nasa.gov>,
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On Fri, 2003-03-07 at 03:35, ext Armando L. Caro Jr. wrote:
> On Thu, 6 Mar 2003, Mark Allman wrote:
> 
> > So then the question is: when the retransmission timer fires what
> > part of that process is the congestion indication?
> >
> >   * Is it the unexpectedly long time we spent waiting for feedback
> >     (an ACK) for some segment?
> >
> >   * Or, is it that we believe something has been dropped (and, so we
> >     are now retransmitting)?
> 
> Both :-)  It's just that the drop is a clearer indiation (in wired
> networks) that there is congestion. Without the loss, a long delay does
> imply congestion, but the question is how to act on it. I believe that TCP
> Westwood and TCP Vegas provide some direction here.

I agree. We were suggesting in the F-RTO draft that cwnd & ssthresh
would be reduced by some amount (halving was suggested) after detecting
spurious RTO. Some problem cases were pointed out with that approach
(possible underutilization of the pipe in congestion avoidance,
especially when RTO occurs in the beginning of connection) but I still
think it could be one way to go, at least with some modifications. One
alternative could be, for example, reducing only cwnd and keeping
ssthresh at a higher level, which starts to be quite close to RFC2861, I
think.

In any case, I think it might be useful to have the different potential
response alternatives documented with some background why the particular
response is justifiable. From the reader's perspective it might be
better to have them all in the same covers, but I don't know if folks
here are willing to go that way.

- Pasi


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On Thu, 2003-03-06 at 22:34, ext Mark Allman wrote:

> > But if the sender's are new each time--which is most likey the
> > case--then one cannot do anything. The effect on the network is,
> > however, still the same.
> 
> Good point.
> 
> (A theoretical argument is that we could share this information
> between connections.  That might help some.  But, in practice we
> don't share information -- and, I am not even sure how much it would
> help in this case.)

Btw, that is not necessarily so theoretical. E.g. Linux stores the
recent values of ssthresh, rto-related parameters, observed reordering,
etc. per destination and reuses them as initial values when new
connections start up shortly.

- Pasi


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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		: TCP Extended Statistics MIB
	Author(s)	: M. Mathis, J. Heffner
	Filename	: draft-ietf-tsvwg-tcp-mib-extension-03.txt
	Pages		: 50
	Date		: 2003-3-6
	
This draft describes extended performance statistics for TCP.  They
are designed to use TCP's ideal vantage point to diagnose performance
problems in both the network and the application.  If a network based
application is performing poorly, TCP can determine if the bottleneck
is in the sender, the receiver or the network itself.  If the
bottleneck is in the network, TCP can provide specific information
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Subject: Re: [Tsvwg] Eifel-Response-03 
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> Let us do some simplified calculations 

I'm not yet convinced that your analysis is right because...

  + If the delay spike is on a shared portion of the network then
    chances are multiple connections will suffer the effects.  You
    assume only one connection will.

  + If the delay spike is on a non-shared network then your notion
    that other connections will take the resources once used by the
    transfer that spuriously retransmitted is right, I think.  But,
    in that case, when the transfer that spuriously retransmitted
    figures it out and reverts the CC state, isn't that OK?  That
    is, the other connections have more than their share and the
    connection in question has less than its fair share of the
    shared network resources.  (And, also note that the increased
    traffic is not introduced as a huge burst.)

  + Unfortunately, real networks are not quite as static as you have
    sketched - with connections coming and going, and connections
    experiencing loss (from the non-shared network), and apps
    limiting the sending, and a thousand other things.  So, I think
    it is pretty hard to use a back-of-the-envelope calculation to
    argue that the eifel-response is going to hurt the overall
    system.

> BTW, it's useful to point out that fast-retransmit, fast-recovery
> recovers one packet per RTT. On the other hand, Slow start can
> recover packet at a much faster rate if the number of lost packets
> is large.  There is a lot more information in Janey C. Hoe's MS
> thesis (Strart Up dynamics of TCP's congestion control and
> avoidance schemes)

Or in RFC 2582.

> So coming back to the argument of initial window, a spurious
> timeout is no different from a new connection. The information
> that the sender has is too old and has no value.

So, I am still unconvinced that these two cases are the same.

I might be happy with a scheme that sort of followed RFC 2861 --
i.e., when we detect a spurious retransmit via the RTO we look at
the elapsed time between when sending that retransmit and sending
whatever data segment we sent prior to that.  For each RTO seconds
that have elapsed we reduce cwnd_prev by half.  That places some
value on the previous CC state, but at the same time takes into
account that the connection has not been validating its cwnd for
some time.  (And, if enough time passes the old cwnd information
will be "too old" and have "no value", but that case does not happen
immediately and simply by virtue of the fact that the RTO fired.)

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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Date: Fri, 07 Mar 2003 14:26:41 +0100
To: "Armando L. Caro Jr." <acaro@acm.org>
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Eifel-Response-03
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At 17:28 06.03.2003, Armando L. Caro Jr. wrote:
>I quickly skimmed the draft and saw on page 5, the step for reverting cwnd
>includes the following text:
>
>               If the acceptable ACK has the ECN-Echo flag [RFC3168] set
>               OR the TCP sender has already taken more than three
>               timeouts for the oldest outstanding segment, then proceed
>               to step (DONE),
>
>How was the value three chosen?

Out of the hat. It's up for debate.

The question is: How many spurious timeouts in a row, i.e., for the same 
outstanding sequence number, are OK to still restore congetsion control state?

In an earlier proposal, we suggested "full restore" for the first spurious 
timeout, "half restore" for the second spurious timeout, etc.
(see http://iceberg.cs.berkeley.edu/papers/Ludwig-Eifel-Alg/index.html)

///Reiner

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To: Randall Stewart <randall@stewart.chicago.il.us>
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Eifel-Response-03
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At 18:43 06.03.2003, Randall Stewart wrote:

>I think FWIW I am ok with restoring to min(prevfs, (currfs+IW))
>
>But I do have one question..
>
>Would this ever other return anything but currfs+IW?

Yes, it can. If the TCP sender was in congestion avoidance (FlightSize >= 
ssthresh) when a spurious timeout occurs, no data segments and no ACKs get 
lost, then cwnd == pipe_prev (== value of FlightSize when the timeout 
occurs) after step (ReCC) of the algorithm.

///Reiner


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Subject: Re: [Tsvwg] Eifel-Response-03 
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At 21:11 06.03.2003, Mark Allman wrote:
>So then the question is: when the retransmission timer fires what
>part of that process is the congestion indication?
>
>   * Is it the unexpectedly long time we spent waiting for feedback
>     (an ACK) for some segment?
>
>   * Or, is it that we believe something has been dropped (and, so we
>     are now retransmitting)?
>
>I *think* I have a hit on this.  But, I am going to roll it around
>my head a bit so I can be coherent when I try to explain it.  But,
>I'd be interested in hearing other's thoughts, too.

RFCs 2309, and 2914 only talk about "packet drop" as the congestion signal. 
On the other hand, I certainly agree that the we cannot fully reverse 
congestion control state when the sender hasn't probed the network for some 
time. We thought 3 timeouts in a row should be an OK time. But maybe we 
should go back to the previous approach of halving pipe_prev for every RTO 
(backed-off or not?).

///Reiner

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Subject: [Tsvwg] The Eifel-Response is conservative ...
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... at least compared to the current TCP that after a spurious timeout 
sends (often) useless go-back-Ns in slow-start, and thereby breaking 
'packet conservation'.

With the Eifel-Response + advanced loss recovery, 'packet conservation' is 
preserved *and* useful restransmits are sent.

You make it look as if the Eifel-Response was aggressive. This is not the 
case. For example, if the TCP sender was in congestion avoidance before the 
spurious timeout, then it will simply return to that state with the Eifel 
response. I.e., each of the (late) original ACKs will nicely clock out one 
or two (in case of delayed ACKs) new data segment(s).

///Reiner

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Subject: Re: [Tsvwg] Eifel-Response-03
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: "Armando L. Caro Jr." <acaro@acm.org>
Cc: mallman@grc.nasa.gov, ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>,
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On Thu, 2003-03-06 at 19:17, ext Armando L. Caro Jr. wrote:
> On 6 Mar 2003, Yogesh Prem Swami wrote:
> 
> > Yes, I agree that the sender has a some stale information about the
> > state of the network, but that information, as you note, might not be of
> > any use. The answer to the question of how good the information actually
> > is, depends on the how long the spurious timeout lasted
> >
> > Let us do some simplified calculations (I am using the same calculation
> > in my DCLOR draft; I apologize if you have already seen this). Lets say
> > there are M competing connection (apart from the connection which was
> > interrupted) and each of them has a Round Trip Time or R (this is a
> > simplifying assumption; I am by no means suggestion it's the case).
> > Let's also say that that the stalling duration is K*R, where K is some
> > integer constant (again just a simplified case to the feel for things).
> > Let's also assume that all the competing connections were doing
> > congestion aovidance.
> >
> > In this case while the stalled connection was waiting for
> > acknowledgements, rest of the M competing connections would increase
> > their congestoin window 1 segment per RTT. So by the end of the spurious
> > timeout, the effective buffer space for the defaulting connection will
> > become:
> >
> > 	cwnd - M*K
> >
> >
> > As we can see this number grows linealy with K (that is the stalling
> > period) and with the number of competing connections. There is NO way
> > for a TCP sender to know how many competing connections are there in the
> > system.
> 
> I think you have a flaw in your analysis. You are assuming that the
> competing flows are synchronized, which is not the case in a real
> networks.  I agree with you that the "available bandwidth" for a flow does
> fluctuate, but you cannot simply say that one TCP connection reducing its
> sending rate will cause others to increase x amount in y amount of time.
> The congestion state at routers is bursty... connections may get drops
> just because of coincidental burst at the time it decided to send.
> 

Armando,

There is no flaw in the analysis, although I do agree that the number
M*K is the upper bound--asuuming no non TCP connections. TCP has such a
choatic behavior it's impossible to have a realistic analysis on paper.

The idea behind the analysis is to demonstrate that the congestion state
in the network is not going to remain the same after an RTO, so the
sender cannot start from it left. The analysis also gives a good idea
how the number of connections affects the overall process. If you have
too many connections in the system, chances are your old congestion
state will be of no value at all.

This is also one of the reasons why I asked for a standarad simulation
environment against which we can evaluate different schemes. 


--yogesh

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Subject: Re: [Tsvwg] Eifel-Response-03
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: mallman@grc.nasa.gov
Cc: ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
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Mark,

On Fri, 2003-03-07 at 06:34, ext Mark Allman wrote:
> > Let us do some simplified calculations 
> 
> I'm not yet convinced that your analysis is right because...
> 
>   + If the delay spike is on a shared portion of the network then
>     chances are multiple connections will suffer the effects.  You
>     assume only one connection will.


Well the argument in my analysis assumes that the bottle neck router
(the router that has minumum buffer_space/|different in outgoing link
speed and incoming| ratio on the end to end path) and the point where
packets are stalled are different. That's why there is just one TCP
conenction. This is a very realistic scenario.

Here is figure from the DCLOR draft:

                      C(1) C(2)... C(M)
                        |  |   ... |
                        |  |   ... |
                        |  |   ... |
                        V  V   ... V
                        \  \      /
                         \  \    /
                          \  \  /
                    +------X--X--X---+       +------------------+
        Defaulting  |                |       |                  |
   C(0) ----------->|   Bottleneck   |------>|Buffered packets  |--->
        connection  |   router       |       |                  |
                    +-----X--X----X--+       +------------------+
                          |  |    |
                          |  |    |
                        c(1)c(2) C(M)



The M connections I was talking about go through the bottle neck router.

> 
>   + If the delay spike is on a non-shared network then your notion
>     that other connections will take the resources once used by the
>     transfer that spuriously retransmitted is right, I think.  But,
>     in that case, when the transfer that spuriously retransmitted
>     figures it out and reverts the CC state, isn't that OK?  That
>     is, the other connections have more than their share and the
>     connection in question has less than its fair share of the
>     shared network resources.  (And, also note that the increased
>     traffic is not introduced as a huge burst.)
> 

Yes the increased traffic has not introduced the burst, but the Eifel
response has. 

Also, why should the other traffic which didn't go through spurious
timeout suffer because of the burst caused by Eifel response. Problem
with such bursts is that some connections might get unnecessarily
penalized (for example a connection which was just starting). 

>   + Unfortunately, real networks are not quite as static as you have
>     sketched - with connections coming and going, and connections
>     experiencing loss (from the non-shared network), and apps
>     limiting the sending, and a thousand other things.  So, I think
>     it is pretty hard to use a back-of-the-envelope calculation to
>     argue that the eifel-response is going to hurt the overall
>     system.
> 

Completely agree. I won't bet a penny on that analysis. But the point is
that the congestion state of the system is not going to remain the same.
I gess I got that idea across so the analysis has served its purpose.

> > BTW, it's useful to point out that fast-retransmit, fast-recovery
> > recovers one packet per RTT. On the other hand, Slow start can
> > recover packet at a much faster rate if the number of lost packets
> > is large.  There is a lot more information in Janey C. Hoe's MS
> > thesis (Strart Up dynamics of TCP's congestion control and
> > avoidance schemes)
> 
> Or in RFC 2582.
> 
> > So coming back to the argument of initial window, a spurious
> > timeout is no different from a new connection. The information
> > that the sender has is too old and has no value.
> 
> So, I am still unconvinced that these two cases are the same.
> 
> I might be happy with a scheme that sort of followed RFC 2861 --
> i.e., when we detect a spurious retransmit via the RTO we look at
> the elapsed time between when sending that retransmit and sending
> whatever data segment we sent prior to that.  For each RTO seconds
> that have elapsed we reduce cwnd_prev by half.  That places some
> value on the previous CC state, but at the same time takes into
> account that the connection has not been validating its cwnd for
> some time.  (And, if enough time passes the old cwnd information
> will be "too old" and have "no value", but that case does not happen
> immediately and simply by virtue of the fact that the RTO fired.)


That's precisely my point. Except, this scheme will not work when there
are spurious timeouts due to subnet change. Not compaining too much
though.

I hope we (except Reiner) agree that Eifel Response is Myopic in it's
approach.

--yogesh

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Subject: Re: [Tsvwg] Eifel-Response-03
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
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On Fri, 2003-03-07 at 08:08, ext Reiner Ludwig wrote:
> At 21:11 06.03.2003, Mark Allman wrote:
> >So then the question is: when the retransmission timer fires what
> >part of that process is the congestion indication?
> >
> >   * Is it the unexpectedly long time we spent waiting for feedback
> >     (an ACK) for some segment?
> >
> >   * Or, is it that we believe something has been dropped (and, so we
> >     are now retransmitting)?
> >
> >I *think* I have a hit on this.  But, I am going to roll it around
> >my head a bit so I can be coherent when I try to explain it.  But,
> >I'd be interested in hearing other's thoughts, too.
> 
> RFCs 2309, and 2914 only talk about "packet drop" as the congestion signal. 
> On the other hand, I certainly agree that the we cannot fully reverse 
> congestion control state when the sender hasn't probed the network for some 
> time. We thought 3 timeouts in a row should be an OK time. But maybe we

3 timeouts are too long--it roughly (1+2+4)*(3*RTT) = 21*RTT (I have
replaced RTO with 3RTT for simplicity; that's ofcourse not always the
case). The best approach is to 1/2 the data rate every RTO as Mark (and
RFC 2861) pointed out in his last e-mail.

--yogesh

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Subject: [Tsvwg] Re: The Eifel-Response is conservative ...
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Reiner,

On Fri, 2003-03-07 at 08:22, ext Reiner Ludwig wrote:
> ... at least compared to the current TCP that after a spurious timeout 
> sends (often) useless go-back-Ns in slow-start, and thereby breaking 
> 'packet conservation'.
> With the Eifel-Response + advanced loss recovery, 'packet conservation' is 
> preserved *and* useful restransmits are sent.
> 


Hate to repeat myself,  but Packet conservation is meaningless after a
timeout. To quote from VJ's paper--congestion control and avoidance
(page 2, paragraph 4):

	"By  conservation of packets  we mean that for a connection  in 	
equilibrium , i.e., running stably with a full window of data 	 	 in
transit, the packet flow is what a physicist would call  	 	
conservative ..."

After a timeout, the sender is not "running stably with a full window of
data."

> You make it look as if the Eifel-Response was aggressive. This is not the 

I am not making Eifel Response "look" aggressive--it *is* agressive. And
it's WRONG.

--yogesh


> case. For example, if the TCP sender was in congestion avoidance before the 
> spurious timeout, then it will simply return to that state with the Eifel 
> response. I.e., each of the (late) original ACKs will nicely clock out one 
> or two (in case of delayed ACKs) new data segment(s).
> 
> ///Reiner




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To: Yogesh Prem Swami <yogesh.swami@nokia.com>
From: Mark Allman <mallman@grc.nasa.gov>
Reply-To: mallman@grc.nasa.gov
Cc: ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
Subject: Re: [Tsvwg] Eifel-Response-03 
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I understand the scenario you are assuming now.  Thanks.

> Yes the increased traffic has not introduced the burst, but the
> Eifel response has.

The eifel response calls for sending based on the ACK clock.  And,
the ACKs should be coming back at the rate the bottleneck link can
support (in the return direction).  So, we should not be sending a
huge burst, but rather ACK clocking things out.  (The ACK clock
could be a bit off, I suppose.  But it has to be better than a big
line-rate burst.)

> Also, why should the other traffic which didn't go through
> spurious timeout suffer because of the burst caused by Eifel
> response. Problem with such bursts is that some connections might
> get unnecessarily penalized (for example a connection which was
> just starting).

To flip this around: Why should the connection through the link with
the delay spike have to give up share through the bottleneck?  Why
does the fact that one of the links it traverses uses LL retransmits
(that worked!) mean that it should get less resources through the
bottleneck?  These seem like orthogonal issues to me.

> > I might be happy with a scheme that sort of followed RFC 2861 --
> > i.e., when we detect a spurious retransmit via the RTO we look
> > at the elapsed time between when sending that retransmit and
> > sending whatever data segment we sent prior to that.  For each
> > RTO seconds that have elapsed we reduce cwnd_prev by half.  That
> > places some value on the previous CC state, but at the same time
> > takes into account that the connection has not been validating
> > its cwnd for some time.  (And, if enough time passes the old
> > cwnd information will be "too old" and have "no value", but that
> > case does not happen immediately and simply by virtue of the
> > fact that the RTO fired.)
> 
> That's precisely my point. Except, this scheme will not work when
> there are spurious timeouts due to subnet change. Not compaining
> too much though.

I presume you mean because the new cwnd could have no resemblence to
the previous cwnd.  Is that right?

This is a more general problem though -- if you switch to a subnet
with drastically different properties then I think you're likely to
get into some TCP ugliness whether there are spurious retransmits or
not.

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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Date: Fri, 7 Mar 2003 13:16:00 -0500
From: Ethan Blanton <eblanton@cs.ohiou.edu>
To: Yogesh Prem Swami <yogesh.swami@nokia.com>
Cc: mallman@grc.nasa.gov, ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>,
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Subject: Re: [Tsvwg] Eifel-Response-03
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Yogesh Prem Swami spake unto us the following wisdom:
> I hope we (except Reiner) agree that Eifel Response is Myopic in it's
> approach.

I'm not sure who "we" is at this point, but I do not agree.  I think
the basic plan of attack is valid.

Ethan

-- 
Happiness is a belt-fed weapon.
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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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Subject: Re: [Tsvwg] Eifel-Response-03
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On Fri, 7 Mar 2003, Ethan Blanton wrote:

> Yogesh Prem Swami spake unto us the following wisdom:
> > I hope we (except Reiner) agree that Eifel Response is Myopic in it's
> > approach.
>
> I'm not sure who "we" is at this point, but I do not agree.  I think
> the basic plan of attack is valid.

I too think the Eifel Response is fine. As Mark said in a previous mail,
the cwnd is restored, but the acks are clocking out the data. The ack
clock never went away... just delayed.

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
0--                                                  --0

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Subject: Re: [Tsvwg] Eifel-Response-03
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: "Armando L. Caro Jr." <acaro@acm.org>
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On Fri, 2003-03-07 at 13:08, ext Armando L. Caro Jr. wrote:

> > I'm not sure who "we" is at this point, but I do not agree.  I think
> > the basic plan of attack is valid.
> 
> I too think the Eifel Response is fine. As Mark said in a previous mail,
> the cwnd is restored, but the acks are clocking out the data. The ack
> clock never went away... just delayed.
> 
> ~armando


The ACK clock is how you send new data into the network. Congestion
response is about how you prevent unnecessary packet loss. These two are
different issues. Ack clocking *can* be used to prevent congestion if
these ACKs arrive in a regular fashion. If the ACKs don't arrive in a
regular fashion, as in case of spurious timeout, they don't convey any
information.

If I go back to my previous figure

		Part-A		      Part-B
	sender-------> stalling node -------> receiver

The ACKs are telling me that congestion on Part-B is fine. But is that
sufficient to conclude that congestion on the entire Part-A + Part-B is
fine? To me the answer seems to be no. 

In addition, the ACKs on Part-B will all arrive as a burst--depending
upon how many hops are between stalling node and receiver (stating from
my simulations). 

I still think that a well defined test can help. Based on the
simulations, we can decide the right thing to do. After the tests, the
working group might decide not to give so much importance to congestion
after all, if something else of an algorithm is important.

--yogesh

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To: Yogesh Prem Swami <yogesh.swami@nokia.com>
From: Mark Allman <mallman@grc.nasa.gov>
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        Ethan Blanton <eblanton@cs.ohiou.edu>,
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Subject: Re: [Tsvwg] Eifel-Response-03 
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> The ACK clock is how you send new data into the
> network. Congestion response is about how you prevent unnecessary
> packet loss. These two are different issues. Ack clocking *can* be
> used to prevent congestion if these ACKs arrive in a regular
> fashion. If the ACKs don't arrive in a regular fashion, as in case
> of spurious timeout, they don't convey any information.

I would phrase it "they may not convey any useful information".  On
the other hand, they will serve to prevent a complete line-rate
burst (e.g., send N packets as fast as you can) even in the worst
case.

> I still think that a well defined test can help. Based on the
> simulations, we can decide the right thing to do. 

I bet that with the right simulation I could make either case.
That's the nice thing about simulations! :)

(It's just a damn hard problem to come up with *the* simulation
harness -- probably because there isn't one, IMO.)

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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Date: Fri, 7 Mar 2003 11:44:43 -0800
From: Aaron Falk <falk@isi.edu>
To: Yogesh Prem Swami <yogesh.swami@nokia.com>
Cc: tsvwg@ietf.org
Subject: Re: [Tsvwg] Test environments for congestion response
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Yogesh-

You might find this of interest:

http://www.acm.org/sigcomm/ccr/archive/1999/oct99/allman2.pdf

--aaron

Yogesh Prem Swami wrote:
> Hi,
> 
> I asked this in last IETF, but let me ask again. Since now we have many
> congestion response algorithms, will it make sense to have a well
> defined test environment against which authors can run their tests and
> get results? The test environment could be an information RFC that can
> be used in future too.
> 
> In the past, I have seen various drafts and papers where authors want to
> evaluate the congestion behavior of their algorithm, but they only use a
> single TCP connection in the entire system. This approach sometimes
> gives hundreds of percents of improvement and might suit the author
> needs, but when these tests are run with multiple connections the
> performance goes down significantly--sometimes below the performance of
> existing algorithms.
> 
> Having spent time trying to model networks, I understand that it's not
> always possible to model  a 100% realistic network, but leaving it
> totally un-addressed is even worse. Probably it will be useful to have a
> set of guidelines: have 10 TCP connections, 5 TFRC based connections
> with 5k, 10K, 100K, ... file size, travesing 10 routers with certain
> buffer spaces on the path, etc., etc. These numbers could be used from
> the Internet measurements (BGP statistics ... ) 
> 
> I think whenever we have multiple schemes we must run the tests along
> these guidelines and pick the scheme which comes up with the best
> compromise in terms of congestion or whaever the working group deems
> important. 
> 
> 
> --yogesh
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
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CC: Yogesh Prem Swami <yogesh.swami@nokia.com>, mallman@grc.nasa.gov,
        ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
Subject: Re: [Tsvwg] Eifel-Response-03
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Ethan Blanton wrote:
> Yogesh Prem Swami spake unto us the following wisdom:
> 
>>I hope we (except Reiner) agree that Eifel Response is Myopic in it's
>>approach.
> 
> 
> I'm not sure who "we" is at this point, but I do not agree.  I think
> the basic plan of attack is valid.
> 
> Ethan
> 
Count me NOT in the "we" catagory... I am ok with Eifel's basic
plan of attack as well..

R

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

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I have been told by an author of this document that this document is normative.

Is this the case?

Could someone please point me to the place in the IETF process
where a draft document is considered normative before advancing?

--brian


On Tue, 04 Mar 2003, Internet-Drafts@ietf.org wrote:

> 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		: Stream Control Transmission Protocol (SCTP) 
>                           Implementors Guide
> 	Author(s)	: R. Stewart, L. Ong et al.
> 	Filename	: draft-ietf-tsvwg-sctpimpguide-08.txt
> 	Pages		: 64
> 	Date		: 2003-3-3
> 	
> This document contains a compilation of all defects found up until
> the publishing of this document for the Stream Control Transmission
> Protocol (SCTP) RFC2960 [3].  These defects may be of an editorial or
> technical nature.  This document may be thought of as a companion
> document to be used in the implementation of SCTP to clarify errors
> in the original SCTP document.
> This document updates RFC2960 [3] and text within this document
> supersedes the text found in RFC2960 [3].
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpimpguide-08.txt
> 
> To remove yourself from the IETF Announcement list, send a message to 
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-- 
Brian F. G. Bidulock    ¦ The reasonable man adapts himself to the ¦
bidulock@openss7.org    ¦ world; the unreasonable one persists in  ¦
http://www.openss7.org/ ¦ trying  to adapt the  world  to himself. ¦
                        ¦ Therefore  all  progress  depends on the ¦
                        ¦ unreasonable man. -- George Bernard Shaw ¦
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Date: Mon, 10 Mar 2003 10:53:16 +0100
To: tsvwg@ietf.org
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Subject: [Tsvwg] Dealing with Spurious Retransmits in TCP
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[I have cc:ed end2end-interest@postel.org, but please reply only to 
tsvwg@ietf.org to avoid double postings.]

What about the following rule for those TCPs that are enhanced to detect 
spurious retransmits?

     Those TCPs MUST reduce their cwnd by MSS (alternatively by the amount
     of spuriously retransmitted data) when a spurious retransmit has been
     detected.

The logic would be to say that since the TCP sender has used more than its 
fair share of bandwidth in the past (due to the spurious retransmit), it 
has to pay back for that once the spurious retransmit has been detected.

With that rule, the policies for adapting RTO and DUPTHRESH would become 
less crucial.

Moreover, a TCP sender for interactive applications (assuming that this 
implies that response time is more important than throughput) may then have 
the freedom to use a non-standard retransmission timer (more optimistic; 
not to say more aggressive), and Early Retransmit without concerns.

Taking this a step further, one could argue that as long as the TCP sender 
uses no more than its fair share of bandwidth, it is free to send what it 
wants. For example, everytime an acceptable ACK arrives, and the TCP sender 
does not have new data available, it may retransmit from the queue of 
unacked data.
RTP/UDP-based streams are also allowed to add redundancy in the form of FEC 
and retransmissions when the source rate is lower than the available 
bandwidth, or to trade some of their throughput and responsiveness for an 
increased reliability.
So, couldn't TCP also trade unused throughput for an increased 
responsiveness by adding redundancy in the form of sending potentially 
spurious retransmits?

///Reiner 

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Brian:

I stated to you (I think this is who you refer to) that our intention
is to make it normative whenever it gets incorporated into
a new RFC... You should know that all drafts are not normative
until they finish the IETF process.

Our discussion was in particular about text that you objected to..
my interpretation of your query was that you thought the text
could never be normative... and I just don't think that is true.. since
our intentions are to close the I-G soon...

R

Brian F. G. Bidulock wrote:

>I have been told by an author of this document that this document is normative.
>
>Is this the case?
>
>Could someone please point me to the place in the IETF process
>where a draft document is considered normative before advancing?
>
>--brian
>
>
>On Tue, 04 Mar 2003, Internet-Drafts@ietf.org wrote:
>
>  
>
>>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		: Stream Control Transmission Protocol (SCTP) 
>>                          Implementors Guide
>>	Author(s)	: R. Stewart, L. Ong et al.
>>	Filename	: draft-ietf-tsvwg-sctpimpguide-08.txt
>>	Pages		: 64
>>	Date		: 2003-3-3
>>	
>>This document contains a compilation of all defects found up until
>>the publishing of this document for the Stream Control Transmission
>>Protocol (SCTP) RFC2960 [3].  These defects may be of an editorial or
>>technical nature.  This document may be thought of as a companion
>>document to be used in the implementation of SCTP to clarify errors
>>in the original SCTP document.
>>This document updates RFC2960 [3] and text within this document
>>supersedes the text found in RFC2960 [3].
>>
>>A URL for this Internet-Draft is:
>>http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpimpguide-08.txt
>>
>>To remove yourself from the IETF Announcement list, send a message to 
>>ietf-announce-request with the word unsubscribe in the body of the message.
>>
>>Internet-Drafts are also available by anonymous FTP. Login with the username
>>"anonymous" and a password of your e-mail address. After logging in,
>>type "cd internet-drafts" and then
>>	"get draft-ietf-tsvwg-sctpimpguide-08.txt".
>>
>>A list of Internet-Drafts directories can be found in
>>http://www.ietf.org/shadow.html 
>>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>
>>
>>Internet-Drafts can also be obtained by e-mail.
>>
>>Send a message to:
>>	mailserv@ietf.org.
>>In the body type:
>>	"FILE /internet-drafts/draft-ietf-tsvwg-sctpimpguide-08.txt".
>>	
>>NOTE:	The mail server at ietf.org can return the document in
>>	MIME-encoded form by using the "mpack" utility.  To use this
>>	feature, insert the command "ENCODING mime" before the "FILE"
>>	command.  To decode the response(s), you will need "munpack" or
>>	a MIME-compliant mail reader.  Different MIME-compliant mail readers
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>>Below is the data which will enable a MIME compliant mail reader
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>>
>
>
>
>  
>


-- 
Randall R. Stewart
ITD
Cisco Systems Inc.
rrs@cisco.com 815-342-5222(cell) or 815-477-2127(office)


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Subject: Re: [Tsvwg] Test environments for congestion response
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Yogesh Prem Swami <yogesh.swami@nokia.com> wrote:

> I agree. What is needed is a realistic scenario which can mimic the
> network behaviour in a certain way. I am appalled by results which show
> 300% improvent in TCP performance, but of course the authors were using
> loopback addresses for testing their improved schemes. [Just
> exaggerating ...]

The scenario we've used in the INFOCOM'03 paper is targeted at a specific
environment, namely UMTS-kind of cellular links. Such links have dedicated
per-user buffers and are not likely to have many concurrent TCP or real-time
flows.
Because of the page limit on the paper we have included only the most
important scenarios: uncongested (a single TCP flow) and congested
(TCP+CBR).
However, I agree that it is useful to explore other traffic mix and I'll
provide results later on.

Still, if we assume that our paper demonstrates the best scenario for Eifel
(+300% throughput) and your DCLOR the worst (-10% throughput), I would vote
for deploying Eifel.  Actuallly, since the difference between all schemes,
including the standard TCP, in your tests is below 10%, the scenario does
not look interesting to me at all. Furthermore, your tests are impossible to
verify as the tools you've used are not publicly available.

In the INFOCOM'03 paper we argue why congestion control should be undone
after a spurious timeout. If not, ACKs are ignored after a spurious timeout
which is a poor idea because it drops the thoughput up to ten times and at
the same time *decreases* goodput as more timeouts and hence retransmissions
occur.

We've also made some measurement of TCP&TFRC over vertical handovers
("subnet changes" in your terms) with significant variation of link
characteristics. http://www.cs.helsinki.fi/u/gurtov/papers/vho.html

Andrei

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Subject: Re: [Tsvwg] Dealing with Spurious Retransmits in TCP 
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> Taking this a step further, one could argue that as long as the
> TCP sender uses no more than its fair share of bandwidth, it is
> free to send what it wants.
>
> [...]
> 
> So, couldn't TCP also trade unused throughput for an increased
> responsiveness by adding redundancy in the form of sending
> potentially spurious retransmits?

This seems somewhat dicey to me.  

We took some measurements a couple of years ago here that showed
about 60% of the TCP connections (in and out) of NASA GRC contained
only a single data packet.  That might still hold or be a little
different, but I doubt it's that far fetched today.  So, using your
notion from above and RFC3390 we could send the same packet 4 times
-- which hopefully ensures that one of those packets arrives.
Contrast that to sending the one packet and retransmitting as
necessary.  If you take a high loss rate of like 10% then one out of
ten connections will need a retransmit (& so we'll send 11 data
packets to deliver 10).  In the case of sending redundant data
sketched above we would send 40 packets to deliver 10.

You can penalize folks a little bit for this bad behavior.  But,
that doesn't help the above case where a connection spins up and
down so fast that no penalty is really meaningful.

Now all of the above back-of-the-envelope conjecture might not be
exactly right.  But, the point is that using your notion could open
the door to polluting the network with a whole bunch of data that
potentially gets in the way without accomplishing any useful work.
And, I wonder if there are cases where performance would actually
decrease (when lots and lots of people are firing lots and lots of
redundant data into the network to try to improve their
performance).

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] Dealing with Spurious Retransmits in TCP 
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Mark and I have been having a private conversation on a related
topic, so when I read 

--- Mark Allman <mallman@grc.nasa.gov> wrote:
> 
> We took some measurements a couple of years ago here that
> showed
> about 60% of the TCP connections (in and out) of NASA GRC
> contained
> only a single data packet.  That might still hold or be a
> little
> different, but I doubt it's that far fetched today.  So, using
> your
> notion from above and RFC3390 we could send the same packet 4
> times
> -- which hopefully ensures that one of those packets arrives.
> Contrast that to sending the one packet and retransmitting as
> necessary.  If you take a high loss rate of like 10% then one
> out of
> ten connections will need a retransmit (& so we'll send 11
> data
> packets to deliver 10).  In the case of sending redundant data
> sketched above we would send 40 packets to deliver 10.

and then, a little later on,

> And, I wonder if there are cases where performance would
> actually
> decrease (when lots and lots of people are firing lots and
> lots of
> redundant data into the network to try to improve their
> performance).
> 
> allman

I'm wondering why we're thinking abut TCP at all for these
applications - or is it just too stupid to imagine another
meaningful transport? Because in addition to the problem Mark is
identifying, we also sent something like eight packets in order
to send one packet with data.

An aside - any clue what application we're talking about? I
immediately started thinking about HTTP/1.0 and/or "Connection:
close", but you should have seen at least one packet in each
direction.

Backing up slightly, I can imagine that if Mark was looking at
HTTP/1.0 connections in his previous work, the percentage really
is probably different today (with much wider HTTP/1.1 deployment
and use), but how different is it?

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Subject: Re: [Tsvwg] Eifel-Response-03
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: mallman@grc.nasa.gov
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        Ethan Blanton <eblanton@cs.ohiou.edu>,
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Hello Mark,

[I am quite backlogged on this list; please bear with me]
 
On Fri, 2003-03-07 at 13:44, ext Mark Allman wrote:
> > The ACK clock is how you send new data into the
> > network. Congestion response is about how you prevent unnecessary
> > packet loss. These two are different issues. Ack clocking *can* be
> > used to prevent congestion if these ACKs arrive in a regular
> > fashion. If the ACKs don't arrive in a regular fashion, as in case
> > of spurious timeout, they don't convey any information.
> 
> I would phrase it "they may not convey any useful information".  On

Agreed, that's the right wording. But, the fact that a TCP sender will
have the right information will only be a coincidence. 

One cannot design a congestion response based on coincidences. If that
weren't the case then AIAD (.. additive decrease) would have been
sufficient since in some sunny day scenarios we don't need AIMD. 

> the other hand, they will serve to prevent a complete line-rate
> burst (e.g., send N packets as fast as you can) even in the worst
> case.
> 

Could you please justify your claim for the above scenario. What happens
if all ACKs except the first and the last get lost into the network. Or,
what happens if all the ACKs--instead of the data packets--were stalled
on the first router connected to the sender.

I think in the worst case the sender will still send data at the line
rate. 

Besides, the comparison with line speed is not of relevance here. If
someone had used line-rate that would have been worse, but that does not
tell the sender that using the old data is correct. I never drive my car
faster than a flying airplane, but does that guarantee that I will not
get a speeding ticket. The speeding ticket in this case is posted by the
routers on the e2e path, and the sender does not know anout these speed
limits.

After a spurious timeout, a TCP sender does not have any information
about the state of the network. The only information it has is that the
longer it waits, the lesser it can rely on the old data rate. In my
opinion, an appropriate congestion response will take this into
consideration.

A congestion response must work in all the cases. Put differently, a
congestoin response needs to be slow enough so that even in the worst
case where the network is already congested, the response algorithm will
not add a single bit to the existing level of congestion.

> > I still think that a well defined test can help. Based on the
> > simulations, we can decide the right thing to do. 
> 
> I bet that with the right simulation I could make either case.
> That's the nice thing about simulations! :)
> 

Not if everyone agrees that a particular scenario is the one that most
resembles a real scenario. But I do agree it's a hard problem.

What other options are there? One cannot analyze the problem
mathematically; one cannot run simulation; so shall we have two
experimental RFCs (I don't like this idea more than anyone else), but
I'm running out of options here.  

--yogesh

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Date: Mon, 10 Mar 2003 11:31:20 -0500
From: Ethan Blanton <eblanton@cs.ohiou.edu>
To: Yogesh Prem Swami <yogesh.swami@nokia.com>
Cc: mallman@grc.nasa.gov, "Armando L. Caro Jr." <acaro@acm.org>,
        ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
Subject: Re: [Tsvwg] Eifel-Response-03
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Yogesh Prem Swami spake unto us the following wisdom:
> > the other hand, they will serve to prevent a complete line-rate
> > burst (e.g., send N packets as fast as you can) even in the worst
> > case.
> > 
> 
> Could you please justify your claim for the above scenario. What happens
> if all ACKs except the first and the last get lost into the network. Or,
> what happens if all the ACKs--instead of the data packets--were stalled
> on the first router connected to the sender.
> 
> I think in the worst case the sender will still send data at the line
> rate. 

No.  Because the algorithm sets ssthresh to the previous value of cwnd
... therefore cwnd can open up only as acks arrive, as in initial slow
start.

> After a spurious timeout, a TCP sender does not have any information
> about the state of the network. The only information it has is that the
> longer it waits, the lesser it can rely on the old data rate. In my
> opinion, an appropriate congestion response will take this into
> consideration.

So, let me put this in perspective ... an RTO causes the connection to
go into slow start, Reiner's draft causes the connection to go into
slow start.

> > I bet that with the right simulation I could make either case.
> > That's the nice thing about simulations! :)
> 
> Not if everyone agrees that a particular scenario is the one that most
> resembles a real scenario. But I do agree it's a hard problem.

Never going to happen.  What sort of real scenario?  That's just far
too wide a question.

> What other options are there? One cannot analyze the problem
> mathematically; one cannot run simulation; so shall we have two
> experimental RFCs (I don't like this idea more than anyone else), but
> I'm running out of options here.  

If you mean an experimental RFC for Eifel and an experimental RFC for
DCLOR, I think it's a fantastic solution.

Ethan

-- 
Happiness is a belt-fed weapon.
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Subject: Re: [Tsvwg] Test environments for congestion response
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: ext Andrei Gurtov <gurtov@cs.helsinki.fi>
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Andrei,

On Mon, 2003-03-10 at 07:44, ext Andrei Gurtov wrote:
> Yogesh Prem Swami <yogesh.swami@nokia.com> wrote:
> 
> > I agree. What is needed is a realistic scenario which can mimic the
> > network behaviour in a certain way. I am appalled by results which show
> > 300% improvent in TCP performance, but of course the authors were using
> > loopback addresses for testing their improved schemes. [Just
> > exaggerating ...]
> 
> The scenario we've used in the INFOCOM'03 paper is targeted at a specific
> environment, namely UMTS-kind of cellular links. Such links have dedicated
> per-user buffers and are not likely to have many concurrent TCP or real-time
> flows.

I don't know much about UMTS networks, but I am told that this is left
to implementation. I will confirm this and comment later on ...

> Because of the page limit on the paper we have included only the most
> important scenarios: uncongested (a single TCP flow) and congested
> (TCP+CBR).
> However, I agree that it is useful to explore other traffic mix and I'll
> provide results later on.
> 
> Still, if we assume that our paper demonstrates the best scenario for Eifel
> (+300% throughput) and your DCLOR the worst (-10% throughput), I would vote

I was not talking about your paper in this example. The example I was
giving was an exaggeration to emphasize the fact that results could be
deceiving. 

(I explicitly remember making the point that it was an exaggeration. I
apologize if I gave a different impression--it was not intentional)

> for deploying Eifel.  Actuallly, since the difference between all schemes,
> including the standard TCP, in your tests is below 10%, the scenario does
> not look interesting to me at all. Furthermore, your tests are impossible to
> verify as the tools you've used are not publicly available.

My tests were run on *real networks*, not on NS-2 simulator (I don't
trust NS-2). DCLOR is implemented in Linux. I will make it publically
available by tomorrow end of business.

I also encourage you to run DCLOR in a real whatever network (UMTS/
wireless lan ...) if you have access to it. It won't give a -10%
improvement even in the worst case. But please don't take my word for
it, verify it for yourself.

> 
> In the INFOCOM'03 paper we argue why congestion control should be undone
> after a spurious timeout. If not, ACKs are ignored after a spurious timeout
> which is a poor idea because it drops the thoughput up to ten times and at
> the same time *decreases* goodput as more timeouts and hence retransmissions
> occur.
> 

That happens if you blindly reduce the data rate to half. A clever
implementaion can handle it properly.

> We've also made some measurement of TCP&TFRC over vertical handovers

Please use "subnet change." Not everyone (read Yogesh) knows about fancy
wireless network terminology.
 
> ("subnet changes" in your terms) with significant variation of link
> characteristics. http://www.cs.helsinki.fi/u/gurtov/papers/vho.html
> 

Thanks. This certainly helps.

--yogesh

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> > I would phrase it "they may not convey any useful information".  On
> 
> Agreed, that's the right wording. But, the fact that a TCP sender
> will have the right information will only be a coincidence.

Um.  Not sure I would phrase it that way, but OK.

> One cannot design a congestion response based on coincidences. 

OK.

So, I agree that the worst case is a fairly substantial bursting
behavior (whether it is line rate or slightly less or a lot
less... point is we can burst in a fairly significant way).  And, we
should look at the worst case.

> After a spurious timeout, a TCP sender does not have any
> information about the state of the network. The only information
> it has is that the longer it waits, the lesser it can rely on the
> old data rate. In my opinion, an appropriate congestion response
> will take this into consideration.

I agree with the statement "the longer it waits, the less[er] it can
rely on the old data rate".  But, I do not agree with the first
statement that says just because we RTO means that the previous CC
state information is useless.

And, so, while I am not convinced that Reiner's scheme of reverting
the entire cwnd is right, I do not think that disallowing all
reversion is right either.

But, now I am just repeating...

allman


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> > What other options are there? One cannot analyze the problem
> > mathematically; one cannot run simulation; so shall we have two
> > experimental RFCs (I don't like this idea more than anyone
> > else), but I'm running out of options here.
> 
> If you mean an experimental RFC for Eifel and an experimental RFC
> for DCLOR, I think it's a fantastic solution.

Agreed.
(modulo WG consensus on each independently, of course)

allman


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> So, let me put this in perspective ... an RTO causes the
> connection to go into slow start, Reiner's draft causes the
> connection to go into slow start.

Right.  The difference is in the length of this slow start (from
cwnd_prev/2 (std case) to cwnd_prev (when allowing state reversion))
(and hence the length of the potentially bursty behavior).

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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Subject: Re: [Tsvwg] Eifel-Response-03
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: mallman@grc.nasa.gov
Cc: Ethan Blanton <eblanton@cs.ohiou.edu>,
        "Armando L. Caro Jr." <acaro@acm.org>,
        ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
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On Mon, 2003-03-10 at 10:52, ext Mark Allman wrote:
> > > What other options are there? One cannot analyze the problem
> > > mathematically; one cannot run simulation; so shall we have two
> > > experimental RFCs (I don't like this idea more than anyone
> > > else), but I'm running out of options here.
> > 
> > If you mean an experimental RFC for Eifel and an experimental RFC
> > for DCLOR, I think it's a fantastic solution.
> 
> Agreed.
> (modulo WG consensus on each independently, of course)


Agreed. 

--yogesh

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Date: Mon, 10 Mar 2003 14:27:54 -0700
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: "Randall Stewart (cisco)" <rrs@cisco.com>
Cc: tsvwg@ietf.org, sctp-impl@external.cisco.com
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Randall,

When I said that the IG is not normative, I meant the current
draft.  It is my understanding that an implementation is not
compelled to follow a draft document.

Is that correct?

--brian


On Mon, 10 Mar 2003, Randall Stewart (cisco) wrote:

> Brian:
> 
> I stated to you (I think this is who you refer to) that our intention
> is to make it normative whenever it gets incorporated into
> a new RFC... You should know that all drafts are not normative
> until they finish the IETF process.
> 
> Our discussion was in particular about text that you objected to..
> my interpretation of your query was that you thought the text
> could never be normative... and I just don't think that is true.. since
> our intentions are to close the I-G soon...
> 
> R
> 
> Brian F. G. Bidulock wrote:
> 
> >I have been told by an author of this document that this document is normative.
> >
> >Is this the case?
> >
> >Could someone please point me to the place in the IETF process
> >where a draft document is considered normative before advancing?
> >
> >--brian
> >
> >
> >On Tue, 04 Mar 2003, Internet-Drafts@ietf.org wrote:
> >
> >  
> >
> >>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		: Stream Control Transmission Protocol (SCTP) 
> >>                          Implementors Guide
> >>	Author(s)	: R. Stewart, L. Ong et al.
> >>	Filename	: draft-ietf-tsvwg-sctpimpguide-08.txt
> >>	Pages		: 64
> >>	Date		: 2003-3-3
> >>	
> >>This document contains a compilation of all defects found up until
> >>the publishing of this document for the Stream Control Transmission
> >>Protocol (SCTP) RFC2960 [3].  These defects may be of an editorial or
> >>technical nature.  This document may be thought of as a companion
> >>document to be used in the implementation of SCTP to clarify errors
> >>in the original SCTP document.
> >>This document updates RFC2960 [3] and text within this document
> >>supersedes the text found in RFC2960 [3].
> >>
> >>A URL for this Internet-Draft is:
> >>http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpimpguide-08.txt
> >>
> >>To remove yourself from the IETF Announcement list, send a message to 
> >>ietf-announce-request with the word unsubscribe in the body of the message.
> >>
> >>Internet-Drafts are also available by anonymous FTP. Login with the username
> >>"anonymous" and a password of your e-mail address. After logging in,
> >>type "cd internet-drafts" and then
> >>	"get draft-ietf-tsvwg-sctpimpguide-08.txt".
> >>
> >>A list of Internet-Drafts directories can be found in
> >>http://www.ietf.org/shadow.html 
> >>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >>
> >>
> >>Internet-Drafts can also be obtained by e-mail.
> >>
> >>Send a message to:
> >>	mailserv@ietf.org.
> >>In the body type:
> >>	"FILE /internet-drafts/draft-ietf-tsvwg-sctpimpguide-08.txt".
> >>	
> >>NOTE:	The mail server at ietf.org can return the document in
> >>	MIME-encoded form by using the "mpack" utility.  To use this
> >>	feature, insert the command "ENCODING mime" before the "FILE"
> >>	command.  To decode the response(s), you will need "munpack" or
> >>	a MIME-compliant mail reader.  Different MIME-compliant mail readers
> >>	exhibit different behavior, especially when dealing with
> >>	"multipart" MIME messages (i.e. documents which have been split
> >>	up into multiple messages), so check your local documentation on
> >>	how to manipulate these messages.
> >>		
> >>		
> >>Below is the data which will enable a MIME compliant mail reader
> >>implementation to automatically retrieve the ASCII version of the
> >>Internet-Draft.
> >>    
> >>
> >
> >
> >
> >  
> >
> 
> 
> -- 
> Randall R. Stewart
> ITD
> Cisco Systems Inc.
> rrs@cisco.com 815-342-5222(cell) or 815-477-2127(office)
> 
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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

what you describe is not quite what I meant. Let me try again ...

I'm suggesting that as long as the TCP sender preserves 'packet 
conservation' *and* uses the 'ACK clock' to clock out transmissions, it 
should be the TCP sender's decision of what it sends. In particular, this 
means that also the mentioned Potentially Spurious Retransmits (PSRs) may 
only be clocked out by returning ACKs.

That should take care of one of your concern:
     "And, I wonder if there are cases where performance would actually
     decrease (when lots and lots of people are firing lots and lots of
     redundant data into the network to try to improve their
     performance)."
I.e., if the TCP sender has only 500 bytes to send and uses RFC3390, it 
should *not* be allowed to send 4 back-to-back copies of the same segment; 
only one segment since PSRs may only be clocked out by returning ACKs.

Still, I would like to revise the rule I suggested:
     Those TCPs MUST reduce their cwnd by MSS (alternatively by the amount
     of spuriously retransmitted data) when a spurious *timeout* has been
     detected.
This is, because only in the case of a spurious timeout has the TCP sender 
used more than its fair share of bandwidth in the past (due to the spurious 
retransmit). In the case of a spurious fast retransmit, the TCP sender has 
actually *not* violated 'packet conservation' in the past.

Coming back to your point using an example that should be equivalent to the 
example you gave (short lived TCP connections). Let's assume MSS = 500 
bytes and the TCP sender has 2000 bytes to send. Per RFC3390, it can send 
the 4 segments in the first RTT back-to-back. With the approach I 
suggested, the ACK for segment 1 would clock out a PSR of segment 2, the 
ACK for segment 2 would clock out a PSR of segment 3, etc.

Assuming the TCP sender didn't get any loss/congestion indication, it would 
end up sending 7 segments (3500 bytes) instead of only 4 segments (2000 
bytes). So what? The TCP sender didn't use more than its fair share of 
bandwidth.

I'm not totally familiar with all the stream encodings that are being 
defined for RTP, but I wouldn't be surprised at all if some of those codecs 
added FEC with a ratio of 3500/2000. And for RTP streams we are saying: "as 
long as the entire RTP stream behaves TCP-friendly, we are fine."

Something is not right here. We are being more strict to TCP than to the 
non-TCPs. That's not TCP-friendly! ;-)

Maybe, I'm missing something essential here ... but my current view is that 
the approach I suggested should at least be OK as an option that the 
application could set (or the user if it has to pay per byte for her/his 
access link).

///Reiner

At 15:31 10.03.2003, Mark Allman wrote:

> > Taking this a step further, one could argue that as long as the
> > TCP sender uses no more than its fair share of bandwidth, it is
> > free to send what it wants.
> >
> > [...]
> >
> > So, couldn't TCP also trade unused throughput for an increased
> > responsiveness by adding redundancy in the form of sending
> > potentially spurious retransmits?
>
>This seems somewhat dicey to me.
>
>We took some measurements a couple of years ago here that showed
>about 60% of the TCP connections (in and out) of NASA GRC contained
>only a single data packet.  That might still hold or be a little
>different, but I doubt it's that far fetched today.  So, using your
>notion from above and RFC3390 we could send the same packet 4 times
>-- which hopefully ensures that one of those packets arrives.
>Contrast that to sending the one packet and retransmitting as
>necessary.  If you take a high loss rate of like 10% then one out of
>ten connections will need a retransmit (& so we'll send 11 data
>packets to deliver 10).  In the case of sending redundant data
>sketched above we would send 40 packets to deliver 10.
>
>You can penalize folks a little bit for this bad behavior.  But,
>that doesn't help the above case where a connection spins up and
>down so fast that no penalty is really meaningful.
>
>Now all of the above back-of-the-envelope conjecture might not be
>exactly right.  But, the point is that using your notion could open
>the door to polluting the network with a whole bunch of data that
>potentially gets in the way without accomplishing any useful work.
>And, I wonder if there are cases where performance would actually
>decrease (when lots and lots of people are firing lots and lots of
>redundant data into the network to try to improve their
>performance).
>
>allman
>
>
>--
>Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
>_______________________________________________
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>tsvwg@ietf.org
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Date: Tue, 11 Mar 2003 10:45:23 +0100
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Subject: Re: [Tsvwg] Eifel-Response-03 
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At 17:57 10.03.2003, Mark Allman wrote:

> > So, let me put this in perspective ... an RTO causes the
> > connection to go into slow start, Reiner's draft causes the
> > connection to go into slow start.
>
>Right.  The difference is in the length of this slow start (from
>cwnd_prev/2 (std case) to cwnd_prev (when allowing state reversion))
>(and hence the length of the potentially bursty behavior).

Just to make it precise. After a spurious timeout the Eifel response will 
make the TCP sender either resume transmission in congestion avoidance if 
it had been in congestion avoidance before the timeout, or in slow start if 
it had been in slow start before the timeout. From the draft:
    "We suggest to restore cwnd to the minimum of the previous FlightSize,
    and the current FlightSize plus IW.
    [...]
    In addition, we suggest to restore ssthresh to pipe_prev, i.e., the
    maximum of the previous value of ssthresh and the value that
    FlightSize had when loss recovery was unnecessarily entered. As a
    result, the TCP sender either immediately resumes probing the network
    for more bandwidth in congestion avoidance, or it first slow starts
    until it has reached its previous share of the available bandwidth."

But, yes, Mark is right in what he says about the "difference in the length 
of the slow start" for the case where the TCP sender had been in slow start 
before the timeout.

The motivation for our choice are scenarios of high BxD connections and a 
spurious timeout early in the initial slow start. Note, that such spurious 
timeout is at least not unlikely since in the beginning of a connection, 
SRTT and RTTVAR have not yet converged to "good" values (unless they had 
been initialized from previous or concurrent connections).

In the standard case, ssthresh is "clamped" at a low value. As a result the 
TCP sender is forced to climb up to BxD in congestion avoidance.

With the Eifel response, ssthresh is restored to its previous value (likely 
its initialization value). As a result the TCP sender can resume slow 
starting to ssthresh which ideally is higher than BxD.

///Reiner

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From: "Injong Rhee" <rhee@eos.ncsu.edu>
To: "'Reiner Ludwig'" <Reiner.Ludwig@eed.ericsson.se>, <tsvwg@ietf.org>
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I agree with Reiner. The main problem lies in unnecessary coupling of
recovery and congestion control in TCP. I view CC is an independent from
recovery. Whether it can transmit spurious retransmission or simply do
no-op, it should left to the application or other layer decision.
Therefore, CC must tell the current rate (or window size) that the data
(including retransmission) must be sent at and the recovery layer
decides on whether it should retransmit, forward-error correction or
other data packets. 

Injong Rhee
Computer Science Dept
North Carolina State Univ.
rhee@csc.ncsu.edu
http://www.csc.ncsu.edu/faculty/rhee

-----Original Message-----
From: end2end-interest-admin@postel.org
[mailto:end2end-interest-admin@postel.org] On Behalf Of Reiner Ludwig
Sent: Monday, March 10, 2003 4:53 AM
To: tsvwg@ietf.org
Cc: end2end-interest@postel.org
Subject: [e2e] Dealing with Spurious Retransmits in TCP

[I have cc:ed end2end-interest@postel.org, but please reply only to 
tsvwg@ietf.org to avoid double postings.]

What about the following rule for those TCPs that are enhanced to detect

spurious retransmits?

     Those TCPs MUST reduce their cwnd by MSS (alternatively by the
amount
     of spuriously retransmitted data) when a spurious retransmit has
been
     detected.

The logic would be to say that since the TCP sender has used more than
its 
fair share of bandwidth in the past (due to the spurious retransmit), it

has to pay back for that once the spurious retransmit has been detected.

With that rule, the policies for adapting RTO and DUPTHRESH would become

less crucial.

Moreover, a TCP sender for interactive applications (assuming that this 
implies that response time is more important than throughput) may then
have 
the freedom to use a non-standard retransmission timer (more optimistic;

not to say more aggressive), and Early Retransmit without concerns.

Taking this a step further, one could argue that as long as the TCP
sender 
uses no more than its fair share of bandwidth, it is free to send what
it 
wants. For example, everytime an acceptable ACK arrives, and the TCP
sender 
does not have new data available, it may retransmit from the queue of 
unacked data.
RTP/UDP-based streams are also allowed to add redundancy in the form of
FEC 
and retransmissions when the source rate is lower than the available 
bandwidth, or to trade some of their throughput and responsiveness for
an 
increased reliability.
So, couldn't TCP also trade unused throughput for an increased 
responsiveness by adding redundancy in the form of sending potentially 
spurious retransmits?

///Reiner 


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Yes, this is the *well-known* problem of coupling
between error control and congestion control
in window-based schemes. Rate-based schemes
naturally decouple the two - see Keshav's textbook
for example.

ibrahim



Injong Rhee wrote:
> 
> I agree with Reiner. The main problem lies in unnecessary coupling of
> recovery and congestion control in TCP. I view CC is an independent from
> recovery. Whether it can transmit spurious retransmission or simply do
> no-op, it should left to the application or other layer decision.
> Therefore, CC must tell the current rate (or window size) that the data
> (including retransmission) must be sent at and the recovery layer
> decides on whether it should retransmit, forward-error correction or
> other data packets.
> 
> Injong Rhee
> Computer Science Dept
> North Carolina State Univ.
> rhee@csc.ncsu.edu
> http://www.csc.ncsu.edu/faculty/rhee
> 
> -----Original Message-----
> From: end2end-interest-admin@postel.org
> [mailto:end2end-interest-admin@postel.org] On Behalf Of Reiner Ludwig
> Sent: Monday, March 10, 2003 4:53 AM
> To: tsvwg@ietf.org
> Cc: end2end-interest@postel.org
> Subject: [e2e] Dealing with Spurious Retransmits in TCP
> 
> [I have cc:ed end2end-interest@postel.org, but please reply only to
> tsvwg@ietf.org to avoid double postings.]
> 
> What about the following rule for those TCPs that are enhanced to detect
> 
> spurious retransmits?
> 
>      Those TCPs MUST reduce their cwnd by MSS (alternatively by the
> amount
>      of spuriously retransmitted data) when a spurious retransmit has
> been
>      detected.
> 
> The logic would be to say that since the TCP sender has used more than
> its
> fair share of bandwidth in the past (due to the spurious retransmit), it
> 
> has to pay back for that once the spurious retransmit has been detected.
> 
> With that rule, the policies for adapting RTO and DUPTHRESH would become
> 
> less crucial.
> 
> Moreover, a TCP sender for interactive applications (assuming that this
> implies that response time is more important than throughput) may then
> have
> the freedom to use a non-standard retransmission timer (more optimistic;
> 
> not to say more aggressive), and Early Retransmit without concerns.
> 
> Taking this a step further, one could argue that as long as the TCP
> sender
> uses no more than its fair share of bandwidth, it is free to send what
> it
> wants. For example, everytime an acceptable ACK arrives, and the TCP
> sender
> does not have new data available, it may retransmit from the queue of
> unacked data.
> RTP/UDP-based streams are also allowed to add redundancy in the form of
> FEC
> and retransmissions when the source rate is lower than the available
> bandwidth, or to trade some of their throughput and responsiveness for
> an
> increased reliability.
> So, couldn't TCP also trade unused throughput for an increased
> responsiveness by adding redundancy in the form of sending potentially
> spurious retransmits?
> 
> ///Reiner

--
Ibrahim Matta       Dept of Comp Sci, 111 Cummington St, MCS-271
matta@cs.bu.edu     Assistant Prof, Boston Univ., Boston, MA 02215
Tel: (617)358-1062, Fax: (617)353-6457, URL: www.cs.bu.edu/fac/matta/
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> what you describe is not quite what I meant. Let me try again ...

OK.  But, you constructed one case where we can still send a bunch
of redundant segments into the network without indication that they
were lost -- and, even given the rules you sketch I can come up with
other cases, I think.

> Coming back to your point using an example that should be
> equivalent to the example you gave (short lived TCP
> connections). Let's assume MSS = 500 bytes and the TCP sender has
> 2000 bytes to send. Per RFC3390, it can send the 4 segments in the
> first RTT back-to-back. With the approach I suggested, the ACK for
> segment 1 would clock out a PSR of segment 2, the ACK for segment
> 2 would clock out a PSR of segment 3, etc.
> 
> Assuming the TCP sender didn't get any loss/congestion indication,
> it would end up sending 7 segments (3500 bytes) instead of only 4
> segments (2000 bytes). So what? The TCP sender didn't use more
> than its fair share of bandwidth.

Two things jump to mind...

  * Sure, you are sending at your "fair share" rate-wise.  But, if
    we take a wider view we're sending extra stuff which uses
    network resources and we are essentially spinning our wheels.
    While this practice might help our connection (sometimes) it
    also uses resources that other connections could use to
    accomplish useful work (which we are not in most cases).

  * The reason you would need to send such retransmits would be
    because they were lost because the network is congested (using
    the standard TCP assumption).  So, what is the justification
    here?  "Just in case we're congested I'll throw some extra data
    into the network."  Seems like throwing gas on the fire to me.

> I'm not totally familiar with all the stream encodings that are
> being defined for RTP, but I wouldn't be surprised at all if some
> of those codecs added FEC with a ratio of 3500/2000. And for RTP
> streams we are saying: "as long as the entire RTP stream behaves
> TCP-friendly, we are fine."
> 
> Something is not right here. We are being more strict to TCP than
> to the non-TCPs. That's not TCP-friendly! ;-)

I will likewise plead ignorance here.  But, I think one difference
is that the RTP stuff is not reliable and so FEC is the only way to
protect the bits.  That is just conjecture.  And, FEC that blows up
the size of the data by a factor of ~2 (which I think is the upper
bound of extra stuff you could dump into the network given the
scheme you sketched) seems pretty egregious to me (just from
thinking about the sorts of things folk around here talk about for
satellite channels).  (Although, that might be quite different from
what gets applied in application-land.)

Maybe someone more informed could clue us both in!

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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I know, this PSR (sending Potentially Spurious Retransmit without any clue 
of a loss indication) option for TCP is probably the extreme case of 
maximum redundancy. And I see your points why this might not be so popular. 
Independent of whether that would ever fly, I feel (and that's the point I 
want to make) that we are way too strict to TCP; at least much stricter 
than we are with RTP/UDP-based streams. And that's not fair ;-(

The amount of useful data sent by the current TCP should be pretty close to 
100 percent. The amount of useful data sent by some RTP/UDPs (even if they 
are TCP-friendly) should for certain content encodings be << 100 percent. 
So, I think we should relax a little when it comes to proposals like Early 
Retransmit, and also allow TCP a bit more redundancy (within the bounds of 
packet conservation and the TCP congestion control principles).

Loping back to the Eifel response and alternative schemes, I think that ...

- a spurious timeout is harmful
This is because the resulting spurious retransmit breaks packet 
conservation. One way to address this is to REQUIRE the rule I suggested 
earlier for those TCPs that are enhanced to detect spurious timeouts:
     Those TCPs MUST reduce their cwnd by MSS (alternatively by the amount
     of spuriously retransmitted data) when a spurious timeout has been
     detected.
However, for those TCPs that followed that rule, the retransmission timer 
becomes less crucial. Those TCPs should be free to use whatever 
retransmission timer the implementers choose to use. They MAY use RFC2988, 
but there should be no more IETF restriction that they MUST use RFC2988 or 
any successor. Likewise, (if that rule is used) the resonsibility of how 
the retransmission timer is adapted in response to a detected spurious 
timeout should lie entirely with the implementor, i.e., the TSVWG should at 
most suggest what an implementor MAY do to adapt the retransmission timer.

- a spurious fast retransmit is *not* harmful
This is because the resulting spurious retransmit does *not* break packet 
conservation. Following the same line of argument as above, I would say 
that the resonsibility of how DUPTHRESH is set initially or how it is 
adapted in response to a detected spurious fast retransmit should lie 
entirely with the implementor (except that cwnd MUST be cut in half for a 
DUPACK-based congestion indication), i.e., the TSVWG should at most suggest 
what an implementor MAY do to initialize or adapt the DUPTHRESH (as long as 
DUPTHRESH <= FlightSizeInMSS, I guess). Furthermore, the Early Retransmit 
algorithm would become a totally OK thing to do. Implementors MAY simply 
use it (as long as packet conservation and the TCP congestion control 
principles are preserved).

If that view found consensus in the TSVWG, the problem of how to respond to 
a detected spurious retransmit (spurious timeout or spurious fast 
retransmit) would be reduced to the problem of how to reverse the 
congestion control state.

///Reiner

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Reiner Ludwig wrote:
> The amount of useful data sent by the current TCP should be pretty
> close to 100 percent. The amount of useful data sent by some RTP/UDPs
> (even if they are TCP-friendly) should for certain content encodings
> be << 100 percent. So, I think we should relax a little when it comes
> to proposals like Early Retransmit, and also allow TCP a bit more
> redundancy (within the bounds of packet conservation and the TCP
> congestion control principles).

Is there a pointer to the ratios of TCP to RTP/UDP traffic for the
present and near future?

rick jones
-- 
Wisdom Teeth are impacted, people are affected by the effects of events.
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to post, OR email to raj in cup.hp.com  but NOT BOTH...
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References: <5.1.0.14.0.20030311104535.028037f8@mailhost.eed.ericsson.se> <5.1.0.14.0.20030311204434.00aaaee8@mailhost.eed.ericsson.se> <3E6E4F51.595663CF@hp.com>
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Rick Jones wrote:

>Is there a pointer to the ratios of TCP to RTP/UDP traffic for the
>present and near future?
>
>rick jones
>
you might want to take a look to these papers:

C. Fraleigh, S. Moon, B. Lyles, C. Cotton, M. Khan, D. Moll, R. Rockell, T. Seely, C. Diot,. "Packet-level
Traffic Measurement from the Sprint IP Backbone" <ftp://ftp.sprintlabs.com/diot/ipmon-ieeenetworks-2003.pdf>. To be published in IEEE Network Magazine. 2003

Kevin Thompson, Gregory J. Miller, and Rick Wilder. "Wide-area
Internet traffic patterns and characteristics". IEEE Network, 11(6),
November 1997.

urtzi


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Agenda of the Transport Area Working Group
Thursday 20 March
13:00-15:00

Brief intro  - Chairs        -   5 mins

RSVP-change discussion*

  draft-kompella-rsvp-change-00.txt - Kireeti Kompella - 10 mins.

   *New task for tsvwg - to help the task of IETF
    keeping RSVP protocol registrations coherent.

Measurement discussion*

  draft-bennett-ippm-ipmp-01.txt - John Bennett - 15 mins.

   *An IP measurement protocol, this was considered by
    the IPPM WG, but seen as broader, so sent here for a
    discussion by a larger group of transport folks.

TCP discussions, two returning* and two new drafts:

  draft-floyd-tcp-highspeed-02.txt* - Sally Floyd - 15 mins.

  draft-blanton-dsack-use-02.txt*   - Mark Allman  - 5 mins.

  draft-blanton-tcp-reordering-00.txt - Mark Allman - 15 mins.

  draft-allman-tcp-early-rexmt-00.txt  - Mark Allman - 15 mins.

  Plans for newreno standardization    - Sally Floyd - 5 mins.

SCTP discussion - thorough review of PR-SCTP

  draft-stewart-tsvwg-prsctp-03.txt - Randall Stewart - 35 mins.



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From: "Andrei Gurtov" <gurtov@cs.helsinki.fi>
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Subject: Re: [Tsvwg] early retransmit 
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Mark Allman wrote:

>Mark Allman, Konstantin Avrachenkov, Urtzi Ayesta, Josh
    Blanton. Early Retransmit for TCP, February 2003. Internet-Draft
    draft-allman-tcp-early-rexmt-00.txt (work in progress).

http://roland.grc.nasa.gov/~mallman/papers/draft-allman-tcp-early-rexmt-00.txt

Another possible hack could be as follows. When cwnd<4 do not trigger fast
retransmit even if 3 DUPACKs are received. The rationale is that DUPACKs in
such case come from unnecessary retransmissions in the past (or duplication
which is less likely). They do not indicate a lost packet since there is
unsufficient flight size to trigger enough "good" DUPACKs.

I guess this hack conflicts somehow with LT and ER...

Andrei

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> Another possible hack could be as follows. When cwnd<4 do not
> trigger fast retransmit even if 3 DUPACKs are received. The
> rationale is that DUPACKs in such case come from unnecessary
> retransmissions in the past (or duplication which is less
> likely). They do not indicate a lost packet since there is
> unsufficient flight size to trigger enough "good" DUPACKs.

Ah-ha!

Right, so there are lots of such cases.  But, I think that using
bugfix (ala RFC 2582) alliviates that problem, right?  In which case
it would be good to add a dependancy of bugfix on implementing ER.
Or, are there some corner cases that would not be caught by bugfix?

Thanks!

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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> > Another possible hack could be as follows. When cwnd<4 do not
> > trigger fast retransmit even if 3 DUPACKs are received. The
> > rationale is that DUPACKs in such case come from unnecessary
> > retransmissions in the past (or duplication which is less
> > likely). They do not indicate a lost packet since there is
> > unsufficient flight size to trigger enough "good" DUPACKs.
>
> Ah-ha!
>
> Right, so there are lots of such cases.  But, I think that using
> bugfix (ala RFC 2582) alliviates that problem, right?  In which case
> it would be good to add a dependancy of bugfix on implementing ER.
> Or, are there some corner cases that would not be caught by bugfix?

Right, bugfix prevents this problem. However, denying all fast retransmits
until the recovery point (what the bugfix does) might be too cruel. This
hack is milder and might be more suitable for general use.

As an example against the bugfix consider a handover of a TCP connection
from a network X with 100-KB buffer to a network Y with 10-KB buffer (I
don't argue that this is a common case in the Internet). All outstanding
segments get lost, TCP recovers with go-back-N after a timeout. However,
since fast retransmits are denied, every CA cycle until the recovery point
has to be recovered with a timeout. The throughput is low.

Alternatively, it's possible to construct a case where bugfix is very useful
to prevent unnecessary retransmits. My point is that there is a slippery
tradeoff about bugfix, and I don't know if there is enough consensus to put
it on the standard track.

Also, this hack prevents against duplication in absence of a timeout, while
the bugfix does not. I know duplication is considered rare, but it can occur
frequently due to bicasting over several access links.

Sally has Newreno on the TSVWG agenda in SF and maybe we will hear what is
the plan about bugfix, since it's currently a part of RFC2582. At least this
beast needs a better name.

Either way, I think that ER draft should mention that DUPACKs after a
timeout are not considered.

Andrei

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> > > Another possible hack could be as follows. When cwnd<4 do not
> > > trigger fast retransmit even if 3 DUPACKs are received.	    

So, maybe I don't understand the "hack".  My reading of the above
(from your original note) basically says that we wouldn't use ER.
That is, ER says that if cwnd<4 you can fast retransmit on less
dupacks.  The above says we cannot.  What am I missing?

(And, I agree with you ... our naming scheme is grand.  We have
"bugfix" and now we're talking about adding a "hack" to a spec! :)

allman


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Date: Wed, 12 Mar 2003 11:40:56 -0600
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] Dealing with Spurious Retransmits in TCP 
To: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org,
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On 3/11/03, Reiner Ludwig wrote:

>I know, this PSR (sending Potentially Spurious Retransmit
>without any clue of a loss indication) option for TCP is
>probably the extreme case of maximum redundancy. And I see
>your points why this might not be so popular. Independent
>of whether that would ever fly, I feel (and that's the
>point I want to make) that we are way too strict to TCP;
>at least much stricter than we are with RTP/UDP-based
>streams. And that's not fair ;-(

There are at least two signifigant differences.

First, the retransmission policy is being applied by the ULP
itself. While there is no way to enforce this, typically the
redundant packets are not selected at random. They represent
critical data which will have a disproportionate impact if
lost. For example, an MPEG Stream can more easily tolerate
the loss of P and B frames than I frames.

The retransmission policy may be limited to critical data
and only in response to a retransmit request. In these
scenarios the ULP is prioritizing retransmission based
upon how critical the data is. When faced with network
congestion the ULP responds by dropping less critical
data messages, either on retransmission or entirely.

PSR adds traffic just in case there may be congestion,
and does so at the transport level without any knowledge
of the relative importance of the data to the ULP.

I'm not judging whether or not there are conditions where
this might be valuable, merely that it is far beyond what
RTP/UDP applications do the network.


A second reason some RTP/UDP applications send redundant
packets even with an explicit NACK is that they are laregly
one-way protocols. There are no routine reverse-flow acks.
This may be because the return path is limited in bandwidth,
has a very long round-trip time, or the forward path is
multicast.

In any event, with every RTP/UDP application I am aware of,
any redundant forward packets are minor compared to the
network savings of omitting reverse path aks.

Additionally, I suspect the trend for these applications
will be to encourage their migration to PR-SCTP or DCP
to encourage more uniform congestion control. RTP/UDP
applications apply rate based control, but do so on
their own without central control. This is admittedly
unfair, and should be phased out someday.


Caitlin Bestler
http://asomi.com/CaitlinBestler/
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On 10 Mar 2003, Yogesh Prem Swami wrote:

> On Fri, 2003-03-07 at 13:44, ext Mark Allman wrote:
> > > The ACK clock is how you send new data into the
> > > network. Congestion response is about how you prevent unnecessary
> > > packet loss. These two are different issues. Ack clocking *can* be
> > > used to prevent congestion if these ACKs arrive in a regular
> > > fashion. If the ACKs don't arrive in a regular fashion, as in case
> > > of spurious timeout, they don't convey any information.
> >
> > I would phrase it "they may not convey any useful information".  On
>
> Agreed, that's the right wording. But, the fact that a TCP sender will
> have the right information will only be a coincidence.
>
> One cannot design a congestion response based on coincidences. If that
> weren't the case then AIAD (.. additive decrease) would have been
> sufficient since in some sunny day scenarios we don't need AIMD.

To me, it seems like you are arguing against TCP's feedback mechanism.
Yes, the feedback mechanism _may_ be too slow to convey good/bad
information at times. But the feedback mechanism seems to work fine, and
it is what we have to work with.

Let me flip the perspective on spurious timeouts. Suppose that a spurious
timeout occurs when the RTO is 1 sec. Since the timeout was in fact
spurious, all we can REALLY infer from this is that the RTO was too low.
Ok, that's obvious. Now suppose that instead of the RTO being 1 sec, it
was actually 2 secs. Then the spurious timeout would probably not occur at
all. Thus, we wouldn't be concerned with the cwnd being stale, would we?
The acks would continue clocking out new data as normally done and nobody
would complain because maybe the RTT is really just THAT long.

The fact that as the RTT increases, the feedback mechanism becomes "less
realiable" is a different problem altogether I think. But flows with long
or even highly variable RTTs should not be penalized.

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
0--                                                  --0

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From: "john smith" <johnsmith0302@hotmail.com>
To: rhee@eos.ncsu.edu, Reiner.Ludwig@eed.ericsson.se, tsvwg@ietf.org
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Subject: [Tsvwg] RE: [e2e] Dealing with Spurious Retransmits in TCP
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i think this would have more to do with RWIN than MSS......

eitherways, why not cap RWIN as was previously suggested once here?







>From: "Injong Rhee" <rhee@eos.ncsu.edu>
>To: "'Reiner Ludwig'" <Reiner.Ludwig@eed.ericsson.se>, <tsvwg@ietf.org>
>CC: <end2end-interest@postel.org>
>Subject: RE: [e2e] Dealing with Spurious Retransmits in TCP
>Date: Mon, 10 Mar 2003 11:40:56 -0500
>
>I agree with Reiner. The main problem lies in unnecessary coupling of
>recovery and congestion control in TCP. I view CC is an independent from
>recovery. Whether it can transmit spurious retransmission or simply do
>no-op, it should left to the application or other layer decision.
>Therefore, CC must tell the current rate (or window size) that the data
>(including retransmission) must be sent at and the recovery layer
>decides on whether it should retransmit, forward-error correction or
>other data packets.
>
>Injong Rhee
>Computer Science Dept
>North Carolina State Univ.
>rhee@csc.ncsu.edu
>http://www.csc.ncsu.edu/faculty/rhee
>
>-----Original Message-----
>From: end2end-interest-admin@postel.org
>[mailto:end2end-interest-admin@postel.org] On Behalf Of Reiner Ludwig
>Sent: Monday, March 10, 2003 4:53 AM
>To: tsvwg@ietf.org
>Cc: end2end-interest@postel.org
>Subject: [e2e] Dealing with Spurious Retransmits in TCP
>
>[I have cc:ed end2end-interest@postel.org, but please reply only to
>tsvwg@ietf.org to avoid double postings.]
>
>What about the following rule for those TCPs that are enhanced to detect
>
>spurious retransmits?
>
>      Those TCPs MUST reduce their cwnd by MSS (alternatively by the
>amount
>      of spuriously retransmitted data) when a spurious retransmit has
>been
>      detected.
>
>The logic would be to say that since the TCP sender has used more than
>its
>fair share of bandwidth in the past (due to the spurious retransmit), it
>
>has to pay back for that once the spurious retransmit has been detected.
>
>With that rule, the policies for adapting RTO and DUPTHRESH would become
>
>less crucial.
>
>Moreover, a TCP sender for interactive applications (assuming that this
>implies that response time is more important than throughput) may then
>have
>the freedom to use a non-standard retransmission timer (more optimistic;
>
>not to say more aggressive), and Early Retransmit without concerns.
>
>Taking this a step further, one could argue that as long as the TCP
>sender
>uses no more than its fair share of bandwidth, it is free to send what
>it
>wants. For example, everytime an acceptable ACK arrives, and the TCP
>sender
>does not have new data available, it may retransmit from the queue of
>unacked data.
>RTP/UDP-based streams are also allowed to add redundancy in the form of
>FEC
>and retransmissions when the source rate is lower than the available
>bandwidth, or to trade some of their throughput and responsiveness for
>an
>increased reliability.
>So, couldn't TCP also trade unused throughput for an increased
>responsiveness by adding redundancy in the form of sending potentially
>spurious retransmits?
>
>///Reiner
>
>


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Mark Allman wrote:

>So, maybe I don't understand the "hack".  My reading of the above
>(from your original note) basically says that we wouldn't use ER.
>That is, ER says that if cwnd<4 you can fast retransmit on less
>dupacks.  The above says we cannot.  What am I missing?
>  
>
Maybe the solution is to use ER during normal transmission (when not 
below snd_max after timeout). Otherwise use "bugfix" or "hack".

That does not allow the "hack" to protect against duplication. But since 
ER addressess a much more common case, perhaps it should be used...

Andrei

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Hello folks,

Here's the patch for DCLOR (you need to download linux kernel 2.5.40
separately)

http://people.nokia.net/~yogesh/dclor-2.5.40.tar.gz

Please let me know your experience.

--yogesh

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At 18:40 12.03.2003, Caitlin Bestler wrote:
>In any event, with every RTP/UDP application I am aware of,
>any redundant forward packets are minor compared to the
>network savings of omitting reverse path aks.

But as we move current RTP/UDP unicast applications to RTP/DCCP, we will 
have ACK traffic of probably the same order as in the case of TCP.

///Reiner

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> At 17:28 06.03.2003, Armando L. Caro Jr. wrote:
> >I quickly skimmed the draft and saw on page 5, the step for reverting cwnd
> >includes the following text:
> >
> >               If the acceptable ACK has the ECN-Echo flag [RFC3168] set
> >               OR the TCP sender has already taken more than three
> >               timeouts for the oldest outstanding segment, then proceed
> >               to step (DONE),
> >
> >How was the value three chosen?
> 
> Out of the hat. It's up for debate.
> 
> The question is: How many spurious timeouts in a row, i.e., for
> the same outstanding sequence number, are OK to still restore
> congetsion control state?

My vote would be <= 1 RTO for a full restore.

> In an earlier proposal, we suggested "full restore" for the first
> spurious timeout, "half restore" for the second spurious timeout,
> etc.  (see
> http://iceberg.cs.berkeley.edu/papers/Ludwig-Eifel-Alg/index.html)

I like this.  But, as I offered before the starting point might need
to be variable...  If it has been >= RTO since we last sent
something then you get to restore to only half the previous value.
If it has been <= RTO seconds then you can start by fully
restoring.  (Followed by having on each subsequent RTO.)

allman


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

I'll have a short (very few slides) presentation on the "Localized RSVP"  
draft at the TSVWG session on Thursday afternoon. Allison asked me to give
a pointer to people because this presentation was added to the agenda very
recently.

In summary, we make use of two unused bits in the Session Object and add 
two new messages that are in structure copies of Path and PathTear 
messages. The modification does not prevent or harm standard 
end-to-end RSVP reservations, only adds a new usage scenario.

Regards,
Jukka


        Title           : Localized RSVP
        Author(s)       : J. Manner et al.
        Filename        : draft-manner-lrsvp-01.txt
        Pages           : 16
        Date            : 2003-1-14

Guaranteed QoS for multimedia applications is based on reserved
resources in each node on the end-to-end path. Even if the
correspondent node does not support QoS, it would be useful to be
able to reserve at least local resources at the access network,
especially wireless link resources. Additionally, for mobile nodes
the continuity of QoS is disturbed due to end-to-end signalling or
slow re-reservations of resources after each handover. This draft
proposes a simple enhancement to RSVP, so that initial resource
reservations and re-reservations due to host mobility can be done
locally in an access network.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-manner-lrsvp-01.txt




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Hi,

The recent discussions on this list regarding to spurious RTOs have concerned only TCP, but shouldn't SCTP be considered also, since it does RTO-based retransmissions as well? I don't have any hands-on experience with SCTP, but it seems like detecting spurious RTOs by using F-RTO-like logic (see draft-sarolahti-tsvwg-tcp-frto-03.txt) applied to SCTP TSNs and SACKs should work. Or is it a bad idea? I could write a draftlet, but it would be nice to hear what folks think of it before doing that.

- Pasi
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From: "Andrei Gurtov" <gurtov@cs.helsinki.fi>
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Subject: [Tsvwg] TCP trash-friendliness and TFRC
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Hi,

The TCP rate formulae for TFRC includes only loss rate, RTT (and MSS) as
factors determining the TCP friendly sending rate. Consequently, if a TFRC
flow runs concurrently with a TCP flow over a path with packet reordering,
duplication or delay spikes, TFRC can significantly overshoot TCP. It would
be nice to improve the TCP-friendly rate formulae to reflect reduction of
the TCP rate during these events.

The second thing is that TCP Reno seems to be widely recognized as a
reference rate flow that does not cause congestion collapse under any
circumstances. However, TCP Reno does send a significant amount of
unnecessary data due to (spurious) timeouts and spurious fast retransmits.

We can define TCP trash-friendliness as the amount of unnecessary data sent
by TCP Reno over given network conditions. Since everybody finds TCP Reno
acceptable, other TCPs could implement different features as long as they
remain TCP friendly and trash-friendly.

A first guess about the formulae looks like follows:

t=(Cd*Rd+Cr*Rd+Cs*Rs+Ct*Rt)/w

Where
t - the fraction of unnecessary delivered packets
w - average TCP window (can be computed using rate from TFRC formulae and
RTT)
Rd - rate of "bad"* duplication events
Cd - number of retransmitted segments per duplication event (default 1)
Rr - rate of "bad" reordering events
Cr - number of retransmitted segments per reordering event (default 1)
Rs - rate of "bad" timeouts
Cs - number of retransmitted segments per spurious timeout (default w)
Rt - rate of "good" timeouts (due to several lost packets)
Ct - number of retransmitted segments per good timeout (default w/2)
       (Reno uses go-back-N which retransmits many packets unnecessarily)

*Since not all delay spikes, reordering length or duplication patterns cause
troubles we only account for "bad" cases which do cause retransmissions.

Given this formulae, TCP SACK could for example apply a more aggressive
retransmission timer because its more efficient loss recovery allows to keep
t low.

Furthermore, TCPs detecting spurious timeouts and fast retransmits do not
have to use complicated adaptation schemes for RTO and dupthresh as long as
their t is below Reno's.

Probably, those are completely wild ideas...

Andrei

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From: Mark Handley <mjh@icir.org>
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To: "Andrei Gurtov" <gurtov@cs.helsinki.fi>
cc: tsvwg@ietf.org
Subject: Re: [Tsvwg] TCP trash-friendliness and TFRC 
In-reply-to: Your message of "Wed, 19 Mar 2003 15:10:08 PST."
             <004501c2ee6c$c6a25040$8d828182@gurtoannb2> 
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>The TCP rate formulae for TFRC includes only loss rate, RTT (and MSS) as
>factors determining the TCP friendly sending rate. Consequently, if a TFRC
>flow runs concurrently with a TCP flow over a path with packet reordering,
>duplication or delay spikes, TFRC can significantly overshoot TCP. It would
>be nice to improve the TCP-friendly rate formulae to reflect reduction of
>the TCP rate during these events.

There is absolutely no good reason why TCP can't cope with significant
re-ordering, and there are several good papers on how to solve this
issue.  The correct solution is to fix TCP.  If TFRC beats up TCP
under these circumstances, they that may provide a good incentive to
fix TCP.

Cheers,
	Mark
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Subject: [Tsvwg] viewgraphs for two presentations in tsvwg tomorrow
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HighSpeed TCP and Quick-Start for Fast Long-Distance Networks
tsvwg, IETF, March 2003.

http://www.icir.org/floyd/talks/hstcp-Mar03.ps
http://www.icir.org/floyd/talks/hstcp-Mar03.pdf



Moving NewReno from Experimental to Proposed Standard?
tsvwg, IETF, March 2003.

http://www.icir.org/floyd/talks/newreno-Mar03.ps
http://www.icir.org/floyd/talks/newreno-Mar03.pd

- Sally
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Subject: Re: [Tsvwg] viewgraphs for two presentations in tsvwg tomorrow 
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>Hello.  The following 2 links aren't found.  Perhaps I jumped the gun?

>http://www.icir.org/floyd/talks/newreno-Mar03.ps
>http://www.icir.org/floyd/talks/newreno-Mar03.pd

Sorry, they should be there now:

http://www.icir.org/floyd/talks/newreno-Mar03.pdf

- Sally
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Mark Handley wrote:
...
> There is absolutely no good reason why TCP can't cope with significant
> re-ordering, and there are several good papers on how to solve this
> issue.  The correct solution is to fix TCP.  If TFRC beats up TCP
> under these circumstances, they that may provide a good incentive to
> fix TCP.
> 
> Cheers,
> 	Mark

Fix TCP because TCP-"friendly" rate control beats up TCP?

With "friends" like that, who needs enemies!

Joe


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>Fix TCP because TCP-"friendly" rate control beats up TCP?

No.  Fix TCP because a decent standards-compliant TCP implementation
that didn't trip over its own showlaces in the face of reordering
would beat up existing TCP implementations.

>With "friends" like that, who needs enemies!

:-)

 - Mark
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Subject: RE: [Tsvwg] Spurious RTOs and SCTP
Date: Thu, 20 Mar 2003 19:51:32 +0200
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Hi all,

Actually, this brings up a more broad question - as SCTP and TCP
share (basically) the same mechanisms for congestion control, etc.,
is it worthwhile for TCP mechanisms also to discuss their applicability
for SCTP?  My opinion is that they should.

br,
John

> -----Original Message-----
> From: ext Pasi.Sarolahti@nokia.com [mailto:Pasi.Sarolahti@nokia.com]
> Sent: 19 March, 2003 00:14
> To: tsvwg@ietf.org
> Subject: [Tsvwg] Spurious RTOs and SCTP
> 
> 
> Hi,
> 
> The recent discussions on this list regarding to spurious 
> RTOs have concerned only TCP, but shouldn't SCTP be 
> considered also, since it does RTO-based retransmissions as 
> well? I don't have any hands-on experience with SCTP, but it 
> seems like detecting spurious RTOs by using F-RTO-like logic 
> (see draft-sarolahti-tsvwg-tcp-frto-03.txt) applied to SCTP 
> TSNs and SACKs should work. Or is it a bad idea? I could 
> write a draftlet, but it would be nice to hear what folks 
> think of it before doing that.
> 
> - Pasi
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 
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Times slightly reduced on some talks and two small items added


Transport Area Working Group (tsvwg)

Thursday, March 20 at 1300-1500
=================================

CHAIRS:	Allison Mankin <mankin@psg.com> 
        Jon Peterson <jon.peterson@neustar.biz>

AGENDA:

Brief intro  - Chairs        -   5 mins


RSVP-change discussion

  draft-kompella-rsvp-change-00.txt - Kireeti Kompella - 5 mins.
   New task for tsvwg - to help the task of IETF
    keeping RSVP protocol registrations coherent.

RSVP-localization - Jukka Manner - 10 mins.

IP measurement OAM problem problem statement - John Bennett 10 mins


TCP discussions, two returning and two new drafts:


  draft-floyd-tcp-highspeed-02.txt - Sally Floyd - 15 mins.

  draft-blanton-dsack-use-02.txt   - Ethan Blanton  - 5 mins.

  draft-blanton-tcp-reordering-00.txt - Ethan Blanton - 15 mins.

  draft-allman-tcp-early-rexmt-00.txt  - Mark Allman - 15 mins.


  Proposal for newreno standardization    - Sally Floyd - 5 mins.

  Redundant data in TCP?  - Reiner Ludwig - 5 mins.

SCTP discussion - thorough review of PR-SCTP

  draft-stewart-tsvwg-prsctp-03.txt - Randall Stewart - 25 mins.


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Subject: RE: [Tsvwg] Spurious RTOs and SCTP
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Hi John!

reply inline...

On Thu, 20 Mar 2003 john.loughney@nokia.com wrote:

> Actually, this brings up a more broad question - as SCTP and TCP
> share (basically) the same mechanisms for congestion control, etc.,
> is it worthwhile for TCP mechanisms also to discuss their applicability
> for SCTP?  My opinion is that they should.

I agree. What would be ideal is for new mechanisms to somehow abstract the
problem and solution so that they can be applied to both protocols. It may
not always be possible though, but at least a discussion of applicability
to both protocols is useful.

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
0--                                                  --0

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here are the slides I presented at the meeting:
http://www.cs.helsinki.fi/u/gurtov/reiner/TCP_FEC.pdf
http://www.cs.helsinki.fi/u/gurtov/reiner/TCP_FEC.ppt

(thanks to Andrei Gurtov for providing the web space)

It would be nice to get some feedback from the community on the issue that 
is raised in the slides; in particular on these questions:

   The IETF Mandates Flows to be TCP-Friendly. Fine, but ...

   - For a Flow that is TCP-Friendly, can the IETF Further
     Mandate what "good" and "bad" Packets/Bytes are?

   - If yes, how? Quotas on Potentially Useless Packets/Bytes?

///Reiner

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> > Actually, this brings up a more broad question - as SCTP and TCP
> > share (basically) the same mechanisms for congestion control, etc.,
> > is it worthwhile for TCP mechanisms also to discuss their 
> > applicability for SCTP?  My opinion is that they should.
> 
> I agree. What would be ideal is for new mechanisms to somehow 
> abstract the problem and solution so that they can be applied to both 
> protocols. It may not always be possible though, but at least a
> discussion of applicability to both protocols is useful.

I agree in general. But I see roughly three options:

1) Write the spec so that TCP and SCTP are handled side by side. This could make detailed algorithm descriptions pain, since TCP and SCTP have different set of variables and conventions. Additionally, SCTP has nice features such as SACKs and duplicate TSNs, which are optional in TCP and not always available.
2) Create a dedicated section for TCP and another section for SCTP in the same draft.
3) Have one small document for TCP and another for SCTP (if needed).

I guess you meant something like (1) and John had something closer to (2) in mind?

Considering our F-RTO draft, I feel (1) quite undesirable, as the algorithm description currently refers to TCP-specific parameters, and patching the algorithms with SCTP could be difficult to read. (2) & (3) would seem easier cases, of which I would favour (3), because I believe the TCP F-RTO is in pretty good shape, and I wouldn't want to delay it unnecessarily because of tweaking with the SCTP-side. Besides, it seems somewhat logical to me to have different documents for handling different transport protocols. However, I'd be happy to hear opinions on this.

- Pasi
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On Fri, 21 Mar 2003 Pasi.Sarolahti@nokia.com wrote:

> I agree in general. But I see roughly three options:
>
> 1) Write the spec so that TCP and SCTP are handled side by side. This
> could make detailed algorithm descriptions pain, since TCP and SCTP
> have different set of variables and conventions. Additionally, SCTP
> has nice features such as SACKs and duplicate TSNs, which are optional
> in TCP and not always available.

> 2) Create a dedicated section for TCP and another section for SCTP in
> the same draft.

> 3) Have one small document for TCP and another for SCTP (if needed).
>
> I guess you meant something like (1) and John had something closer to
> (2) in mind?
>
> Considering our F-RTO draft, I feel (1) quite undesirable, as the
> algorithm description currently refers to TCP-specific parameters, and
> patching the algorithms with SCTP could be difficult to read. (2) &

Actually, I had in mind a combination of (1) and (2). A draft can have
the problem statement generic for the two transport protocols, and then
dedicate a separate section for specific solutions for TCP and SCTP.

> (3) would seem easier cases, of which I would favour (3), because I
> believe the TCP F-RTO is in pretty good shape, and I wouldn't want to
> delay it unnecessarily because of tweaking with the SCTP-side.
> Besides, it seems somewhat logical to me to have different documents
> for handling different transport protocols.

I understand your point, but my concern is that we'll end up with some
solutions for TCP and some for SCTP... when all or most solutions can be
applied to both. Then someone has to go through a second pass and apply
the TCP solutions to SCTP and vice-versa. If the solutions are drastically
(or even moderately) different, maybe it makes more sense to have separate
drafts. I think a good example of when both protocols can be addressed in
one draft is when spurious rtxs need to be identified via duplicate tsn
reports. TCP uses DSACKs and SCTP uses DupTSNs, but there is very little
difference in the two.

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
0--                                                  --0

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Date: Fri, 21 Mar 2003 11:25:53 -0600
From: Caitlin Bestler <cait@asomi.com>
Subject: RE: [Tsvwg] Spurious RTOs and SCTP
To: Pasi.Sarolahti@nokia.com
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On 3/21/03, Pasi.Sarolahti@nokia.com wrote:

>> > Actually, this brings up a more broad question - as
SCTP and TCP
>> > share (basically) the same mechanisms for congestion
control, etc.,
>> > is it worthwhile for TCP mechanisms also to discuss
their 
>> > applicability for SCTP?  My opinion is that they
should.
>> 
>> I agree. What would be ideal is for new mechanisms to
>> somehow abstract the problem and solution so that they
>> can be applied to both protocols. It may not always be
>> possible though, but at least a discussion of
>> applicability to both protocols is useful.
>
>I agree in general. But I see roughly three options:
>
>1) Write the spec so that TCP and SCTP are handled side by
>side. This could make detailed algorithm descriptions
>pain, since TCP and SCTP have different set of variables
>and conventions. Additionally, SCTP has nice features such
>as SACKs and duplicate TSNs, which are optional in TCP and
>not always available.
>2) Create a dedicated section for TCP and another section
for SCTP in the same draft.
>3) Have one small document for TCP and another for SCTP (if
needed).
>

Would it be feasible to define a single document which
specified a set of transport neutral terms that would be
compatible with TCP, TCP with SACKs, SCTP and DCP (and the
mapping of those terms to each congestion controlling
transport protocol)?

That way new algorithms can be proposed once, and the
mapping to each transport would be known.

Caitlin Bestler
http://asomi.com/CaitlinBestler/
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Subject: RE: [Tsvwg] Spurious RTOs and SCTP
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> Would it be feasible to define a single document which
> specified a set of transport neutral terms that would be
> compatible with TCP, TCP with SACKs, SCTP and DCP (and the
> mapping of those terms to each congestion controlling
> transport protocol)?
> 
> That way new algorithms can be proposed once, and the
> mapping to each transport would be known.

That may turn out to be difficult. To give an example, TCP DSACK and SCTP dupTSN are not exactly the same. You don't know if the other TCP end point implements DSACK, so one cannot draw very strong conclusions from the lack of DSACK blocks. In SCTP, dupTSNs is mandatory (right?), so stronger algorithms can probably be implemented on top of that.

- Pasi
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Subject: RE: [Tsvwg] Spurious RTOs and SCTP
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At the Triggers for Transport BoF at IETF 56, Allison Mankin
asked Carl Williams and I to propose a charter which included
"link up" notification, and a suggested response function.

Of course, the response function (at this time) must be
specified for at least two or three transports.

I'm just pointing out that we'd ALSO like to see a single
document with these transport mechanisms, because it would make
TRIGTRAN a HECKUVA lot easier for us to knock out.

And I believe Pasi about a single specification being difficult
- the problem is, having a bunch of per-transport specifications
just spreads the work around for anyone who's trying to work on
transport-independent mechanisms...

Spencer, who believes Mark Allman when Mark says it's
unfortunate that we didn't get traction with ECM...

--- Pasi.Sarolahti@nokia.com wrote:
> > Would it be feasible to define a single document which
> > specified a set of transport neutral terms that would be
> > compatible with TCP, TCP with SACKs, SCTP and DCP (and the
> > mapping of those terms to each congestion controlling
> > transport protocol)?
> > 
> > That way new algorithms can be proposed once, and the
> > mapping to each transport would be known.
> 
> That may turn out to be difficult. To give an example, TCP
> DSACK and SCTP dupTSN are not exactly the same. You don't know
> if the other TCP end point implements DSACK, so one cannot
> draw very strong conclusions from the lack of DSACK blocks. In
> SCTP, dupTSNs is mandatory (right?), so stronger algorithms
> can probably be implemented on top of that.
> 
> - Pasi

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Hi,

Does anybody know which all TCP related RFCs are implemented in Windows 98/2000/XP. The same for Solaris, and HP?

--Yogesh 
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Yougesh,

This document may be helpful (starting on page 4) for Windows OS:

	http://rdweb.cns.vt.edu/public/notes/tcpip2000.pdf

This link was found in:

	http://rdweb.cns.vt.edu/public/notes/win2k-tcpip.htm

which also has a link to an MS website.

Hope this helps!

--rr

From:	SMTP%"Yogesh.Swami@nokia.com" 24-MAR-2003 19:23:43.96
To:	tsvwg@ietf.org
CC:	
Subj:	[Tsvwg] RFC status in popular OSes

Hi,

Does anybody know which all TCP related RFCs are implemented in Windows
98/2000/XP. The same for Solaris, and HP?

--Yogesh 
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From mailnull@www1.ietf.org  Tue Mar 25 17:39:41 2003
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Date: Tue, 25 Mar 2003 23:22:46 +0100
To: tsvwg@ietf.org
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Cc: Phil Karn <karn@qualcomm.com>
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Subject: [Tsvwg] Rethinking Karn's algorithm
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instead of an explicit out-of-band "Link Up"-Trigger to jump-start a TCP 
sender that is stuck with a backed-off RTO, we could also rethink Karn's 
algorithm ...

The motivation for the backoff of the RTO is to eventually "catch" a valid 
RTT timing which - according to Karn's algorithm - should not be taken from 
data segments that had been retransmitted (retransmission ambiguity).

But:
1. With timestamps, we can also take valid RTTs from retransmitted segments.
2. Without timestamps and a timeout for segment N, we could delay making
    the RTO more conservative until the ACK for N is received. According to
    Karn's algorithm we should not take an RTT for segment N, anyway. What
    really matters is that the RTO has been made more conservative when
    segment N+1 is sent. The question of how the RTO is made more conservative
    is a separate issue. Probably different alternatives exist.

Either way, we could allow the TCP sender to retransmit the oldest 
outstanding segment every RTO, i.e., without backoff!

There might be an (academic?) issue with persistent congestion, although 
that was not a consideration in Karn's SIGCOMM '87 paper. Still, sending 
one packet every RTO (without backoff) seems sufficiently conservative. 
Anyway, backing off to ultimately 64 seconds seems far too conservative.

///Reiner

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Date: Tue, 25 Mar 2003 19:05:04 -0500 (EST)
From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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To: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Cc: <tsvwg@ietf.org>, Phil Karn <karn@qualcomm.com>
Subject: Re: [Tsvwg] Rethinking Karn's algorithm
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On Tue, 25 Mar 2003, Reiner Ludwig wrote:

> instead of an explicit out-of-band "Link Up"-Trigger to jump-start a TCP
> sender that is stuck with a backed-off RTO, we could also rethink Karn's
> algorithm ...
>
> The motivation for the backoff of the RTO is to eventually "catch" a valid
> RTT timing which - according to Karn's algorithm - should not be taken from
> data segments that had been retransmitted (retransmission ambiguity).
>
> But:
> 1. With timestamps, we can also take valid RTTs from retransmitted segments.
> 2. Without timestamps and a timeout for segment N, we could delay making
>     the RTO more conservative until the ACK for N is received. According to
>     Karn's algorithm we should not take an RTT for segment N, anyway. What
>     really matters is that the RTO has been made more conservative when
>     segment N+1 is sent. The question of how the RTO is made more conservative
>     is a separate issue. Probably different alternatives exist.

Yes I agree.

Although the motivation is different... We have investigated using
Timestamps in SCTP for obtaining a more accurate RTT estimate of alternate
destination paths. In SCTP, the alternate destinations are used only for
retransmissions (and heartbeats).  Hence, the RTO of the alternate
destinations generally are more conservative than necessary... reducing
throughput of the association. We found that using Timestamps
significantly improves performance simply because more RTT measurements
are obtained for the alternate destinations, and thus the RTOs are more
accurate.

The motivation for and a BRIEF description of the SCTP TIMESTAMP
Chunk is in the following paper:

Transport Layer Multihoming For Fault Tolerance In FCS Networks
Armando L. Caro Jr., Paul D. Amer, Randall R. Stewart
CTA 2003, April 2003
www.cis.udel.edu/~acaro/papers

We are currently writing the paper which presents the results. The paper
should be available in a couple of weeks.

> Either way, we could allow the TCP sender to retransmit the oldest
> outstanding segment every RTO, i.e., without backoff!
>
> There might be an (academic?) issue with persistent congestion, although
> that was not a consideration in Karn's SIGCOMM '87 paper. Still, sending
> one packet every RTO (without backoff) seems sufficiently conservative.
> Anyway, backing off to ultimately 64 seconds seems far too conservative.

I agree that 64 seconds (60 secs with SCTP) is much too conservative.
However, I don't understand why you think multiple RTOs without backoff is
ok or even conservative. I think I am missing something here... If the
timeout occurs because the RTO is too small, how will the sender know to
increase the RTO?

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
0--                                                  --0

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Date: Thu, 27 Mar 2003 13:08:14 +0100
To: "Armando L. Caro Jr." <acaro@acm.org>
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Rethinking Karn's algorithm
Cc: tsvwg@ietf.org, Phil Karn <karn@qualcomm.com>
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At 01:05 26.03.2003, Armando L. Caro Jr. wrote:
> > Either way, we could allow the TCP sender to retransmit the oldest
> > outstanding segment every RTO, i.e., without backoff!
> >
> > There might be an (academic?) issue with persistent congestion, although
> > that was not a consideration in Karn's SIGCOMM '87 paper. Still, sending
> > one packet every RTO (without backoff) seems sufficiently conservative.
> > Anyway, backing off to ultimately 64 seconds seems far too conservative.
>
>I agree that 64 seconds (60 secs with SCTP) is much too conservative.
>However, I don't understand why you think multiple RTOs without backoff is
>ok or even conservative. I think I am missing something here...

I didn't mean conservative in terms of adapting the RTO to a potentially 
increased RTT, but conservative in terms of impact on the congestion state 
in the network.
For a potentially severly congested bottleneck, having the end-points send 
one packet per RTO seems sufficiently conservative. I.e., every router in 
today's Internet should at least be able to handle a certain peak number of 
flows (depending on some worst case traffic estimate for that location in 
the topology) each sending with a minimum rate of one packet per RTO.

I guess the times when LANs where interconnected by slow dial-up 
connections are long gone ... and if they still exist then they certainly 
don't pose a threat to the stability of the (public) Internet.


>If the
>timeout occurs because the RTO is too small,

If that was the reason for the timeout, it is quite likely that the ACK for 
the retransmitted segment will arrive shortly after the RTO, and even more 
likely before a second timeout with a non-backed-off RTO.


>how will the sender know [how] to
>increase the RTO?

I guess several options exist. One option would be to allow timing of 
retransmitted segments even without timestamps by taking as the RTT the 
time between sending the original segment and the receipt of the ACK, 
independent of whether that ACK was triggered by the original or some 
retransmit. Alternatively, with the current retransmission timer [RFC2988] 
we could require that when timestamps are not used then SRTT should be 
doubled once the ACK for oldest outstanding segment arrives after a timeout.

///Reiner




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Dear Colleagues:

Our sincere apologies if you receive multiple copies of this announcement.

     CONFERENCE ANNOUNCEMENT AND CALL FOR PRESENTATIONS

                   Internetworking 2003
                      June 22-24, 2003
                   San Jose, California
      In technical cooperation with IEEE, IFIP, and ACM (Pending)


Internetworking 2003 Program Committee cordially invites you to submit proposals for original, unpublished presentations focusing on internetworking technologies in the IP, optical, and wireless domains. Summaries not exceeding 250 words can be submitted to submissions@caitr.org for review and possible inclusion in the program, no later than April 11, 2003. Topics of interest include, but are not limited to the following:

- Voice over IP (VoIP)
- IP Video Conferencing
- Storage Area Networks (SANs)
- Unicast and Multicast Routing and Convergence
- QoS Routing
- Network Security and Service Integration
- Operational Support Systems
- Virtual Private Networks
- Internetworking Wireless LANs and 3G Wireless Networks
- IP-based Infrastructure for Wireless Networks
- Internetworking IP and Optical Networks
- Internetworking MPLS with Legacy ATM and Frame Relay Networks
- Transition from IPv4 to IPv6 and interworking
- Pervasive Computing
- High Speed Transport Layer Protocols
- Peer to Peer Networking and Grid Computing
- Video Teleconferencing (VTC) 
- 802.11 Hotspots


The Internetworking 2003 event in June will include participation from industry, government agencies, and academia. If you need additional technical information, please contact the Technical Cochairs Professor Maurice GAGNAIRE <gagnaire@enst.fr>, or Daniel Awduche <Awduche@awduche.com>.


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<DIV id=message>
<P>Dear Colleagues:</P>
<P>Our sincere apologies if you receive multiple copies of this announcement.</P>
<P>&nbsp;&nbsp;&nbsp;&nbsp; CONFERENCE ANNOUNCEMENT AND CALL FOR PRESENTATIONS</P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Internetworking 2003<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; June 22-24, 2003<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; San Jose, California<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; In technical cooperation with IEEE, IFIP, and ACM (Pending)</P>
<P><BR>Internetworking 2003 Program Committee cordially invites you to submit proposals for original, unpublished presentations focusing on internetworking technologies in the IP, optical, and wireless domains. Summaries not exceeding 250 words can be submitted to <A href="http://mail.yahoo.com/config/login?/ym/Compose?To=submissions@caitr.org" target=_blank>submissions@caitr.org</A> for review and possible inclusion in the program, no later than April 11, 2003. Topics of interest include, but are not limited to the following:</P>
<P>- Voice over IP (VoIP)<BR>- IP Video Conferencing<BR>- Storage Area Networks (SANs)<BR>- Unicast and Multicast Routing and Convergence<BR>- QoS Routing<BR>- Network Security and Service Integration<BR>- Operational Support Systems<BR>- Virtual Private Networks<BR>- Internetworking Wireless LANs and 3G Wireless Networks<BR>- IP-based Infrastructure for Wireless Networks<BR>- Internetworking IP and Optical Networks<BR>- Internetworking MPLS with Legacy ATM and Frame Relay Networks<BR>- Transition from IPv4 to IPv6 and interworking<BR>- Pervasive Computing<BR>- High Speed Transport Layer Protocols<BR>- Peer to Peer Networking and Grid Computing<BR>- Video Teleconferencing (VTC) <BR>- 802.11 Hotspots</P>
<P><BR>The Internetworking 2003 event in June will include participation from industry, government agencies, and academia. If you need additional technical information, please contact the Technical Cochairs Professor Maurice GAGNAIRE &lt;<A href="http://mail.yahoo.com/config/login?/ym/Compose?To=gagnaire@enst.fr" target=_blank>gagnaire@enst.fr</A>&gt;, or Daniel Awduche &lt;<A href="http://mail.yahoo.com/config/login?/ym/Compose?To=Awduche@awduche.com" target=_blank>Awduche@awduche.com</A>&gt;.<BR></P></DIV>
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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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Subject: Re: [Tsvwg] Rethinking Karn's algorithm
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Reiner,

Thanks for the clarification... I like this idea.

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
0--                                                  --0

On Thu, 27 Mar 2003, Reiner Ludwig wrote:

> At 01:05 26.03.2003, Armando L. Caro Jr. wrote:
> > > Either way, we could allow the TCP sender to retransmit the oldest
> > > outstanding segment every RTO, i.e., without backoff!
> > >
> > > There might be an (academic?) issue with persistent congestion, although
> > > that was not a consideration in Karn's SIGCOMM '87 paper. Still, sending
> > > one packet every RTO (without backoff) seems sufficiently conservative.
> > > Anyway, backing off to ultimately 64 seconds seems far too conservative.
> >
> >I agree that 64 seconds (60 secs with SCTP) is much too conservative.
> >However, I don't understand why you think multiple RTOs without backoff is
> >ok or even conservative. I think I am missing something here...
>
> I didn't mean conservative in terms of adapting the RTO to a potentially
> increased RTT, but conservative in terms of impact on the congestion state
> in the network.
> For a potentially severly congested bottleneck, having the end-points send
> one packet per RTO seems sufficiently conservative. I.e., every router in
> today's Internet should at least be able to handle a certain peak number of
> flows (depending on some worst case traffic estimate for that location in
> the topology) each sending with a minimum rate of one packet per RTO.
>
> I guess the times when LANs where interconnected by slow dial-up
> connections are long gone ... and if they still exist then they certainly
> don't pose a threat to the stability of the (public) Internet.
>
>
> >If the
> >timeout occurs because the RTO is too small,
>
> If that was the reason for the timeout, it is quite likely that the ACK for
> the retransmitted segment will arrive shortly after the RTO, and even more
> likely before a second timeout with a non-backed-off RTO.
>
>
> >how will the sender know [how] to
> >increase the RTO?
>
> I guess several options exist. One option would be to allow timing of
> retransmitted segments even without timestamps by taking as the RTT the
> time between sending the original segment and the receipt of the ACK,
> independent of whether that ACK was triggered by the original or some
> retransmit. Alternatively, with the current retransmission timer [RFC2988]
> we could require that when timestamps are not used then SRTT should be
> doubled once the ACK for oldest outstanding segment arrives after a timeout.
>
> ///Reiner
>
>
>
>


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I don't follow this at all...

> The motivation for the backoff of the RTO is to eventually "catch"
> a valid RTT timing which - according to Karn's algorithm - should
> not be taken from data segments that had been retransmitted
> (retransmission ambiguity).
>
> [...]
>
> There might be an (academic?) issue with persistent congestion,
> although that was not a consideration in Karn's SIGCOMM '87 paper.

The Karn and Partridge paper was about managing the RTO estimator
and timer.  The 1988 Jacobson paper motivated the backoff in terms
of congestion control:

    "Another timer mistake is in the backoff after a retransmit: If
    a packet has to be retransmitted more than once, how should the
    retransmits be spaced?  For a transport endpoint embedded in a
    network of unknown topology and with an unknown, unknowable and
    constantly changing population of competing conversations, only
    one scheme has any hope of working---exponential backoff"

My own intuition says that backoff is simply continuing the notion
of halving the sending rate that drives the cwnd adjustments.

allman


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To: mallman@grc.nasa.gov
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Rethinking Karn's algorithm 
Cc: tsvwg@ietf.org, Phil Karn <karn@qualcomm.com>
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At 22:07 27.03.2003, Mark Allman wrote:
>The Karn and Partridge paper was about managing the RTO estimator
>and timer.  The 1988 Jacobson paper motivated the backoff in terms
>of congestion control:
>
>     "Another timer mistake is in the backoff after a retransmit: If
>     a packet has to be retransmitted more than once, how should the
>     retransmits be spaced?  For a transport endpoint embedded in a
>     network of unknown topology and with an unknown, unknowable and
>     constantly changing population of competing conversations, only
>     one scheme has any hope of working---exponential backoff"
>
>My own intuition says that backoff is simply continuing the notion
>of halving the sending rate that drives the cwnd adjustments.

I definitely believe that this was the right approach for the Internet some 
15 years ago. At that time the Internet was really a collection of LANs 
interconnected by slow dial-up lines (potentially even only 
semi-permanent). But today's Internet looks different. For example, quoting 
from 
ftp://ftp.rfc-editor.org/in-notes/internet-drafts/draft-iab-research-funding-00.txt:

   "One of the factors that has blunted the demand for QoS has been the
    transition of the Internet infrastructure from heavy congestion in
    the early 1990s, to overprovisioning in backbones and in many
    international links now."

So, in general I strongly believe that we have to ask ourselves - and not 
only with respect to Karn's algorithm - whether we still need to be so 
ultra-conservative with end-point algorithms.

Concerning Karn's algorithm, I'm not advocating "retransmit every 
non-backed-off RTO forever". We do not want to see the Internet poluted 
with a bunch of half-broken TCP connections that would do that. But, there 
are certainly different alternatives. For example, a TCP sender could 
retransmit every non-backed-off RTO for the first 30 seconds, every 2xRTO 
for the following 60 seconds, every 4xRTO for the following 120 seconds, 
etc. until the sending application gives up, or after some upper bound R2 
as defined in RFC1122. That is also an exponential backoff.

///Reiner

PS.: Apart from Karn's algorithm, I even believe that with the current 
economics behind the operation of today's (best-effort) Internet, it should 
be possible to allow end-points a minimum send rate (e.g., 5 kb/s) as long 
as they get receiver feedback every N RTTs. I think that this is needed if 
we are serious about wide-scale deployment of decent-quality services like 
streaming and VoIP over best-effort.


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> So, in general I strongly believe that we have to ask ourselves -
> and not only with respect to Karn's algorithm - whether we still
> need to be so ultra-conservative with end-point algorithms.

That's a fine question and a fertile area for research maybe.  But,
I'll disagree quite strongly that anything like this is ready for
the IETF.

I don't think you have made the case that the RTO should be the
lower bound on the sending rate.  An RTO of 30 or 60 seconds doesn't
"just happen"; rather it takes lots of repeated losses to make it
happen.  Clearly, in that environment you have a network that is
badly busted in some way (which you have direct evidence of).  I
don't understand the motivation in being terribly aggressive in this
situation.

(And, I am still firmly in the camp that says that the RTO should be
the last resort for loss repair and we should continue to refine
other mechanisms to avoid RTOs.  And, if we can largely do that then
I don't much care how conservative the RTO is.)

allman


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From mailnull@www1.ietf.org  Fri Mar 28 11:38:24 2003
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From: "Heinz Prantner" <heinz.prantner@onlinehome.de>
To: <sigtran@ietf.org>, <tsvwg@ietf.org>, <rserpool@ietf.org>
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actually , this reminds me on something - for those of you I had the chance
to meet on several occasions :

me too, can be reached on a new address: heinz.prantner@onlinehome.de

kind regards,
Heinz


----- Original Message -----
From: "Tuexen Michael" <michael.tuexen@siemens.com>
To: <sigtran@ietf.org>; <tsvwg@ietf.org>; <rserpool@ietf.org>
Sent: Friday, March 28, 2003 2:37 PM
Subject: [Sigtran] New e-mail address


> Dear all,
>
> my Siemens e-mail address will no longer work.
> Please use
>
> tuexen@fh-muenster.de
>
> in case you want to contact me.
>
> Best regards
> Michael
>
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> _______________________________________________
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From: Yogesh Prem Swami <yogesh.swami@nokia.com>
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Subject: [Tsvwg] Lightweight mobility detection and response algorithm
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Hi,

We've submitted a new draft "Lightweight mobility detection and response
algorithm for TCP," which will become available in a couple of days from
IETF web site. The draft is also available from

http://people.nokia.net/~yogesh/draft-swami-tcp-lmdr-00.txt

This draft allows a TCP host to propagate its subnet change information
to its peer and specifies the congestion response that the peer must
follow subsequent to such a change. 

The draft uses one of the bits from the reserved bits field of TCP
header. We would like to have feedback from the working group on whether
we should define a TCP option instead of using this bit. The pros and
cons of the two approach have been identified in the draft. The
mechanism is secure to the extent to sequence number guessing (i.e., as
secure as any other TCP mechanism), but we can make it more secure
(without using any key exchange protocol) if the working group demands
it (it won't be free of cost in terms of number of bytes carried in each
packet).

Any comment, criticism, words of advice, ... are very welcome.

--yogesh


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From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Rethinking Karn's algorithm 
Cc: tsvwg@ietf.org, Phil Karn <karn@qualcomm.com>
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At 16:19 28.03.2003, Mark Allman wrote:
> > So, in general I strongly believe that we have to ask ourselves -
> > and not only with respect to Karn's algorithm - whether we still
> > need to be so ultra-conservative with end-point algorithms.
>
>That's a fine question and a fertile area for research maybe.  But,
>I'll disagree quite strongly that anything like this is ready for
>the IETF.
>
>I don't think you have made the case that the RTO should be the
>lower bound on the sending rate.  An RTO of 30 or 60 seconds doesn't
>"just happen"; rather it takes lots of repeated losses to make it
>happen.  Clearly, in that environment you have a network that is
>badly busted in some way (which you have direct evidence of).  I
>don't understand the motivation in being terribly aggressive in this
>situation.

There is certainly a broad range for viable solutions between "terribly 
aggressive" and "ultra-conservative".

If you think that "Rethinking Karn's algorithm" to make it less 
conservative in *some* way is not ready for the IETF, I wonder about your 
view on an explicit out-of-band "Link Up"-Trigger. How ready is that for 
the IETF?

I mean, *the least* we could do is to discuss a new upper bound for the RTO 
back-off. What's the motivation for the current upper bound of 60 seconds?

///Reiner




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cc: tsvwg@ietf.org, Phil Karn <karn@qualcomm.com>
Subject: Re: [Tsvwg] Rethinking Karn's algorithm 
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> If you think that "Rethinking Karn's algorithm" to make it less
> conservative in *some* way is not ready for the IETF, I wonder
> about your view on an explicit out-of-band "Link Up"-Trigger. How
> ready is that for the IETF?

It's a decent question.  Personally, I think it's OK.  I think we
can craft a transport-level response to be reasonable when receiving
explicit information from the network.  On the other hand, the RTO
backoff is done in the abscence of any information.

> I mean, *the least* we could do is to discuss a new upper bound
> for the RTO back-off. What's the motivation for the current upper
> bound of 60 seconds?

??

The only thing I have to say about this is that the current maximum
is of the right order of magnitude, I think.  Trying to optimize the
value after that is dubious, I think.  (That is, I believe it should
be conservative.  And, if it is then that means that it will be
gross in some situations.)  If we can get explicit information to
cut a very long backoff down then maybe that is OK.

allman


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

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From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Rethinking Karn's algorithm 
Cc: tsvwg@ietf.org, Phil Karn <karn@qualcomm.com>
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At 14:28 31.03.2003, Mark Allman wrote:

> > If you think that "Rethinking Karn's algorithm" to make it less
> > conservative in *some* way is not ready for the IETF, I wonder
> > about your view on an explicit out-of-band "Link Up"-Trigger. How
> > ready is that for the IETF?
>
>It's a decent question.  Personally, I think it's OK.  I think we
>can craft a transport-level response to be reasonable when receiving
>explicit information from the network.

With a less conservative version of Karn's algorithm, we would instead have 
a true end-to-end solution to the problem. That is, with a few changed 
lines of TCP sender code (or even only a single '#define') we would save 
the IETF extra work (a TRIGTRAN WG), and would avoid the deployment issues 
associated with "Link Up"-Triggers & corresponding TCP sender responses.

///Reiner




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> With a less conservative version of Karn's algorithm, we would
> instead have a true end-to-end solution to the problem. 

I doubt it.  My own opinion is that any end2end "solution" that
would obviate the need for a link-up trigger would be too
aggressive.  I.e., there is fundemental tension between (1) probing
often enough to be very responsive on paths that have links that go
down and (2) not sending a whole lot of chud into a network that is
dropping your packets and is quite unresponsive.

> That is, with a few changed lines of TCP sender code (or even only
> a single '#define') we would save the IETF extra work (a TRIGTRAN
> WG), and would avoid the deployment issues associated with "Link
> Up"-Triggers & corresponding TCP sender responses.

IETF resources are not the only costs here.

allman


--
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Date: Mon, 31 Mar 2003 15:31:56 +0200
To: mallman@grc.nasa.gov
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Rethinking Karn's algorithm 
Cc: tsvwg@ietf.org, Phil Karn <karn@qualcomm.com>
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At 15:17 31.03.2003, Mark Allman wrote:
>My own opinion is that any end2end "solution" that
>would obviate the need for a link-up trigger would be too
>aggressive.  I.e., there is fundemental tension between (1) probing
>often enough to be very responsive on paths that have links that go
>down and (2) not sending a whole lot of chud into a network that is
>dropping your packets and is quite unresponsive.

I think, you have nicely summarized the tension that needs to be debated, 
but I would add that (2) needs to be weighed against the extra traffic that 
such triggers would cause. I.e., the traffic caused by all the routers that 
send explicit out-of-band link-up triggers (at some rate?) to a potentially 
large number of flows active in those routers.

[In addition, some of the trigger-receivers might not (cannot) even care, 
e.g., because they don't have any new data to sent, or because they don't 
understand the trigger, or because they never get them because the trigger 
was filtered by a firewall, etc.]

///Reiner 

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Date: Mon, 31 Mar 2003 05:55:36 -0800 (PST)
From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] Rethinking Karn's algorithm 
To: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, mallman@grc.nasa.gov
Cc: tsvwg@ietf.org, Phil Karn <karn@qualcomm.com>
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Hi, Reiner,

I believe that post-IETF 56, what's actually on the table in
TRIGTRAN is end-systems resending one "last packet" at link-up
time, not routers sending triggers.

The considerations you're listing would make explicit requests
for TRIGTRAN coverage more attractive, but what's been expressed
to me is that explicit requests would have to be a LOT more
attractive to require (even edge) routers to maintain
per-connection state - so that's why I've backed off on routers
sending triggers, at this time.

Spencer

--- Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se> wrote:
> 
> I think, you have nicely summarized the tension that needs to
> be debated, 
> but I would add that (2) needs to be weighed against the extra
> traffic that 
> such triggers would cause. I.e., the traffic caused by all the
> routers that 
> send explicit out-of-band link-up triggers (at some rate?) to
> a potentially 
> large number of flows active in those routers.
> 
> [In addition, some of the trigger-receivers might not (cannot)
> even care, 
> e.g., because they don't have any new data to sent, or because
> they don't 
> understand the trigger, or because they never get them because
> the trigger 
> was filtered by a firewall, etc.]
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Date: Mon, 31 Mar 2003 16:12:42 +0200
To: Spencer Dawkins <spencer_dawkins@yahoo.com>
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Rethinking Karn's algorithm 
Cc: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, mallman@grc.nasa.gov,
        tsvwg@ietf.org, Phil Karn <karn@qualcomm.com>
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At 15:55 31.03.2003, Spencer Dawkins wrote:
>I believe that post-IETF 56, what's actually on the table in
>TRIGTRAN is end-systems resending one "last packet" at link-up
>time, not routers sending triggers.

Fine. But don't we have exactly that already in section 8.2 ("Recovery from 
Subnetwork Outages") of draft-ietf-pilc-link-design-13.txt which is a 
soon-to-be-BCP-RFC:

    "[...]
    It is therefore highly desirable that a subnetwork subject to outages
    not silently discard packets during an outage. Ideally, it should
    define an interface to the next higher layer (i.e., IP) that allows
    it to refuse packets during an outage, and to automatically ask IP
    for new packets when it is again able to deliver them. If it cannot
    do this, then the subnetwork should hold onto at least some of the
    packets it accepts during an outage and attempt to deliver them when
    the subnetwork comes back up. When packets are discarded, IP should
    be notified so that the appropriate ICMP messages can be sent.

    Note that it is *not* necessary to completely avoid dropping packets
    during an outage. The purpose of holding onto a packet during an
    outage, either in the subnetwork or at the IP layer, is so that its
    eventual delivery will implicitly notify TCP that the subnetwork is
    again operational. [...]"

And this works for both, hosts and routers.

///Reiner 

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From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] Rethinking Karn's algorithm 
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Hi, Reiner,

Yeah, exactly - what I'm talking about is a more completely
specified variant of this. For instance, we might (this is an
observation, not a suggestion) recommend that host
implementations resend a packet, but not router implementations
(as your posting notes, the recommendation in LINK allows both,
but doesn't prefer one or the other).

Spencer

--- Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se> wrote:
> At 15:55 31.03.2003, Spencer Dawkins wrote:
> >I believe that post-IETF 56, what's actually on the table in
> >TRIGTRAN is end-systems resending one "last packet" at
> link-up
> >time, not routers sending triggers.
> 
> Fine. But don't we have exactly that already in section 8.2
> ("Recovery from 
> Subnetwork Outages") of draft-ietf-pilc-link-design-13.txt
> which is a 
> soon-to-be-BCP-RFC:
> 
>     "[...]
>     It is therefore highly desirable that a subnetwork subject
> to outages
>     not silently discard packets during an outage. Ideally, it
> should
>     define an interface to the next higher layer (i.e., IP)
> that allows
>     it to refuse packets during an outage, and to
> automatically ask IP
>     for new packets when it is again able to deliver them. If
> it cannot
>     do this, then the subnetwork should hold onto at least
> some of the
>     packets it accepts during an outage and attempt to deliver
> them when
>     the subnetwork comes back up. When packets are discarded,
> IP should
>     be notified so that the appropriate ICMP messages can be
> sent.
> 
>     Note that it is *not* necessary to completely avoid
> dropping packets
>     during an outage. The purpose of holding onto a packet
> during an
>     outage, either in the subnetwork or at the IP layer, is so
> that its
>     eventual delivery will implicitly notify TCP that the
> subnetwork is
>     again operational. [...]"
> 
> And this works for both, hosts and routers.
> 
> ///Reiner 
> 

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> I think, you have nicely summarized the tension that needs to be
> debated, but I would add that (2) needs to be weighed against the
> extra traffic that such triggers would cause. I.e., the traffic
> caused by all the routers that send explicit out-of-band link-up
> triggers (at some rate?) to a potentially large number of flows
> active in those routers.
> 
> [In addition, some of the trigger-receivers might not (cannot)
> even care, e.g., because they don't have any new data to sent, or
> because they don't understand the trigger, or because they never
> get them because the trigger was filtered by a firewall, etc.]

Yes, you'd want to consider that.

The trigger traffic is bounded, though, by the number of flows (per
some window of time).  Whereas, it is fairly difficult to quantify
the probing approach in any sort of realistic or meaningful way, it
would seem to me.

allman


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

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cc: "Henderson, Thomas R" <thomas.r.henderson@boeing.com>
From: Sally Floyd <floyd@icir.org>
Date: Mon, 31 Mar 2003 15:29:35 -0800
Subject: [Tsvwg] Moving NewReno from Experimental to Proposed Standard
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At the tsvwg meeting in San Francisco two weeks ago, I gave up
my five-minute slot for discussing "Moving NewReno from Experimental to
Proposed Standard" because of a shortage of time, and said I would
send email to the list instead.  So finally here it is.

(We did take a quick hum at the meeting to verify that people
generally thought that taking NewReno to Proposed was a good idea,
as I recall.)

The viewgraphs for this talk are at:
  http://www.icir.org/floyd/talks/newreno-Mar03.ps
  http://www.icir.org/floyd/talks/newreno-Mar03.pdf

Here is basically what would have been the text part of the talk:


* NewReno was been widely deployed for many years now, and has
been an Experimental RFC since 1999, so it seems time to move
it from Experimental to Proposed. 

* NewReno performs dramatically better than Reno when multiple
packets are dropped from a window of data, and we would recommend
NewReno over Reno, for TCP connections where the other end does not
support SACK.  However, in the interests of full disclosure,
we do know of one scenario where Reno (and Sack) both perform
better than NewReno.  This is when there are reordered packets but
no lost packets, and the sender gets three dup acks from the
reordering alone.  This is illustrated in slides 4, 5, and 6
of the viewgraphs, for Reno, NewReno, and Sack respectively.

* We know of only one change that we want to make to the NewReno RFC.
That RFC describes a Careful and Less Careful variant, and recommends
the Careful variant.  A revision would *require* the Careful instead of 
the Less Careful variant.  The reason for this is illustrated
on slide 8 of the viewgraphs, and has to do with the Less Careful's
quite bad response to a Fast Retransmit that is due to reordering alone.

(The Careful and Less Careful variants are two variants of what is
called "bugfix" in the NS simulator.  The name "bugfix" dates from
the days when Steve McCanne and I were the only users of the NS
simulator, and we had not yet made the simulator publically available,
so I was also less careful in the names that I attached to things...)

* If anyone knows of any other changes that should be made to
the NewReno spec as it goes to Proposed, please let Tom and I 
know.

* For all of the simulations in the viewgraphs, the bottom line
gives the command needed to run that simulation in the NS simulator,
in the directory "tcl/test".

Thanks,
- Sally

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