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Subject: Re: [Tsvwg] Re: WGLC  on draft-ietf-tsvwg-prsctp-00
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Hi all:

I just want to summarize where I think we are from the comments
I have seen in this thread:

1) We will modify F5 to be more clear.. it will end up as follows
     (quoting from the XML source):
"
<t hangText="F5)">
 If a decision to "abandon" a chunk is made, no matter how such decision
is made, the appropriate congestion adjustment MUST be made as specified
in RFC2960 if the chunk later would have been marked
for retransmission (e.g. T3-Timeout or Fast Retransmit).
"

2) We have not a consensus either way on if we need to limit the number
     of forward TSN's.. I had put out a proposed text change... and asked
     for comments... Brian and Armando chimed in that no change was
     needed.. I am open to the group opinion... there are 3 basic things
     I have  heard of (onlist and off):
    a) No change
    b) only one FWD-TSN out per RTT (I lean towards this if we do
         anything)
    c) send a FWD-TSN every other SACK

   Opinons????

R


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From exim@www1.ietf.org  Wed Jul  2 02:52:34 2003
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Subject: [Tsvwg] SIP client over SCTP
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Hi,

I'm developping the first SIP over SCTP User Agent in Linux.
For this I use the SIP UA KPhone, and I'm modifying it.

A first version of SIP SCTP client ( KPhone SCTP) is ready !!!

If you are interested, I update my home page 

( http://utenti.lycos.it/ssal/ )

with all the necessary information in order to setup 
the enviroment and test the UA. 
In this page you'll find also the mail address where 
it is possible demand the source code !


Best regards
sal


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Date: Tue, 1 Jul 2003 13:33:58 -0400 (EDT)
From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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Subject: Re: [Tsvwg] Re: WGLC  on draft-ietf-tsvwg-prsctp-00
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On Tue, 1 Jul 2003, Randall R. Stewart (home) wrote:

> 2) We have not a consensus either way on if we need to limit the number
>      of forward TSN's.. I had put out a proposed text change... and asked
>      for comments... Brian and Armando chimed in that no change was
>      needed.. I am open to the group opinion... there are 3 basic things
>      I have  heard of (onlist and off):
>     a) No change
>     b) only one FWD-TSN out per RTT (I lean towards this if we do
>          anything)
>     c) send a FWD-TSN every other SACK

If people feel that there is a need to reduce the number of FWD-TSNs, then
I vote for option b. Note that in doing so, the timeliness of the FWD-TNSs
may make PR-SCTP useless. If the forward path is lossy, then TSNs will get
lost and their corresponding FWD-TSNs will get lost as well. Reducing the
number of FWD-TSNs increases the number of RTTs needed to notify the peer
that a TSN is being abandoned by the sender. Since the sender can bundle
FWD-TSNs with outgoing data, I am not concerned about too many on the
wire.

~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: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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Subject: Re: [Tsvwg] Re: WGLC  on draft-ietf-tsvwg-prsctp-00
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Armando,

a) is the way to go.

I would surely like to see argument here as to why the supporters of b) o=
r c)
think that they are necessary and limiting the number of SACKs is not.  F=
or
example, in b) whare are not SACKs limited to every RTT?  Also, in c), wh=
ere
are not SACKs limited to every other (current) SACK?

--brian

On Tue, 01 Jul 2003, Armando L. Caro Jr. wrote:

> On Tue, 1 Jul 2003, Randall R. Stewart (home) wrote:
>=20
> > 2) We have not a consensus either way on if we need to limit the numb=
er
> >      of forward TSN's.. I had put out a proposed text change... and a=
sked
> >      for comments... Brian and Armando chimed in that no change was
> >      needed.. I am open to the group opinion... there are 3 basic thi=
ngs
> >      I have  heard of (onlist and off):
> >     a) No change
> >     b) only one FWD-TSN out per RTT (I lean towards this if we do
> >          anything)
> >     c) send a FWD-TSN every other SACK
>=20
> If people feel that there is a need to reduce the number of FWD-TSNs, t=
hen
> I vote for option b. Note that in doing so, the timeliness of the FWD-T=
NSs
> may make PR-SCTP useless. If the forward path is lossy, then TSNs will =
get
> lost and their corresponding FWD-TSNs will get lost as well. Reducing t=
he
> number of FWD-TSNs increases the number of RTTs needed to notify the pe=
er
> that a TSN is being abandoned by the sender. Since the sender can bundl=
e
> FWD-TSNs with outgoing data, I am not concerned about too many on the
> wire.
>=20
> ~armando
>=20
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0
>=20
>=20
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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

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Subject: [Tsvwg] I-D ACTION:draft-manner-lrsvp-02.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Localized RSVP
	Author(s)	: J. Manner et al.
	Filename	: draft-manner-lrsvp-02.txt
	Pages		: 17
	Date		: 2003-7-2
	
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-02.txt

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From exim@www1.ietf.org  Thu Jul  3 00:24:32 2003
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Date: Thu, 03 Jul 2003 00:23:02 -0400
From: Cyrus Shaoul <cyrus@ntt-at.com>
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Subject: [Tsvwg] VoIP and Congestive Collapes: QoS vs. Best Effort
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TSVWG list members, 

After lurking for a bit, I wanted to try asking a question of the other
list members. I have been reading the recent IAB draft relating to
congestive collapse of VoIP traffic, called draft-iab-congestion-00.txt,
and I wanted to see if my understanding of the issues relate to the work
of the TSVWG and to the recent draft that Naotaka Morita recently
submitted (draft-morita-tsvwg-pps-00.txt)

As I see it, one way to compare the two is to look at the level of QoS
support in the routers that are assumed. In draft-iab-congestion-00, the
IAB authors assume that all nodes are not capable of Diffserv or other
QoS functions, and all traffic is treated equally with best effort.
Morita's draft assumes that all packet forwarding nodes support Diffserv,
and have configurable PHB. (This is not currently the case, but it may
be the case in the future)

Another comparison is the type of communication targeted. draft-iab-congestion-00
only considers voice, which has a range of 4 to 64kpbs in most cases.
In draft-morita-tsvwg-pps-00.txt, video calls and other broadband
services, around 1Mps per call, are the focus.

A final comparison can be made between the reactive vs. proactive
natures of the solution. One of the conclusions, as I understand them,
of the IAB draft is that in cases of certain specific amounts of congestion
and packet loss, the application should end the call instead of using
any of the remaining resources. I feel that this is a very different
user experience for the one that the VoIP user expects. (Other than
during emergencies, PSTN phone calls rarely terminate in mid-call due to
high call volume). The Morita proposal acts like a preventative measures,
and instead of reacting to congestion, it proactively prevents voice
traffic from ever getting to the point of failing. By using admission
control, the user can be informed of the lack of available bandwidth
before the call begins, reducing the change to the standard telephone user
experience.

Are these two drafts trying to solve different problems? Or are they
solving the same problem in different ways because they begin with
different assumptions?

Thanks for your time, 

Cyrus


Cyrus Shaoul
NTT Advanced Technology Corp.
cyrus@ntt-at.com
http://www.ntt-at.com/



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From exim@www1.ietf.org  Thu Jul  3 01:28:33 2003
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To: Cyrus Shaoul <cyrus@ntt-at.com>
cc: tsvwg@ietf.org
From: Sally Floyd <floyd@icir.org>
Subject: Re: [Tsvwg] VoIP and Congestive Collapes: QoS vs. Best Effort 
Date: Wed, 02 Jul 2003 22:27:30 -0700
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>Are these two drafts trying to solve different problems? 

The two drafts are trying to solve different problems.

The IAB draft, draft-iab-congestion-00.txt, is not trying to say
that best-effort service is a good solution for VoIP traffic, or a
better solution than some form of QoS, or anything of the kind.
The IAB draft is trying to say that today some VoIP traffic in fact
runs as best-effort traffic over one or more links in the best-effort
Internet, and that this is likely to be the case in the near future
as well, and that it is not deemed recommended behavior by the IAB
for these flows to maintain their minimum sending rate in the
presence of persistent, steady-state packet drop rates experienced
by the flow.

We will try to be more clear, in the next round of revisions, that
we are not trying to say that we are recommending best-effort service
as the recommended path for roll-out for VoIP.  We are not recommending
any roll-out path one way or another.  We are observing that in fact,
VoIP traffic some times ends up competing, as best-effort traffic,
with other best-effort traffic in the Internet.

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




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From exim@www1.ietf.org  Thu Jul  3 01:58:32 2003
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From: Jukka MJ Manner <jmanner@cs.Helsinki.FI>
To: tsvwg@ietf.org
Subject: Re: [Tsvwg] I-D ACTION:draft-manner-lrsvp-02.txt
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Dear all,

We have updated our draft on the extension to RSVP to allow for resource
signaling within an access network. The technical specification has not
changed since the previous version of the draft. The main changes were
editorial, e.g. better explanations here and there. We would still be very
interested in hearing comments about the ideas (the concept and the
technical implementation) presented in the draft.

At least me and Markku Kojo will be in Vienna if anyone wants to have chat 
in person.

Regards,
Jukka

On Wed, 2 Jul 2003 Internet-Drafts@ietf.org wrote:

> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> 
> 
> 	Title		: Localized RSVP
> 	Author(s)	: J. Manner et al.
> 	Filename	: draft-manner-lrsvp-02.txt
> 	Pages		: 17
> 	Date		: 2003-7-2
> 	
> 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-02.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-manner-lrsvp-02.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-manner-lrsvp-02.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
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> 		
> 		
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> 


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Sal:

I saw no way to download this code or get a copy.. and I seemed
to get a LOT of adds and such when I hit this site :-<

Is there any place we can get the code at to play with ??

R

loretosa@vodafone.it wrote:

>Hi,
>
>I'm developping the first SIP over SCTP User Agent in Linux.
>For this I use the SIP UA KPhone, and I'm modifying it.
>
>A first version of SIP SCTP client ( KPhone SCTP) is ready !!!
>
>If you are interested, I update my home page 
>
>( http://utenti.lycos.it/ssal/ )
>
>with all the necessary information in order to setup 
>the enviroment and test the UA. 
>In this page you'll find also the mail address where 
>it is possible demand the source code !
>
>
>Best regards
>sal
>
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg
>
>
>
>  
>



_______________________________________________
tsvwg mailing list
tsvwg@ietf.org
https://www1.ietf.org/mailman/listinfo/tsvwg



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Sal:

I saw no way to download this code or get a copy.. and I seemed
to get a LOT of adds and such when I hit this site :-<

Is there any place we can get the code at to play with ??

R

loretosa@vodafone.it wrote:

>Hi,
>
>I'm developping the first SIP over SCTP User Agent in Linux.
>For this I use the SIP UA KPhone, and I'm modifying it.
>
>A first version of SIP SCTP client ( KPhone SCTP) is ready !!!
>
>If you are interested, I update my home page 
>
>( http://utenti.lycos.it/ssal/ )
>
>with all the necessary information in order to setup 
>the enviroment and test the UA. 
>In this page you'll find also the mail address where 
>it is possible demand the source code !
>
>
>Best regards
>sal
>
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg
>
>
>
>  
>



_______________________________________________
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tsvwg@ietf.org
https://www1.ietf.org/mailman/listinfo/tsvwg



From exim@www1.ietf.org  Thu Jul  3 13:03:37 2003
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Hi Randal,


Yes, maybe it's not a very clean web page... you're right.

In any case it's impossible download the software from the site,
as I've written in the message:

I'll send the source code by mail on demand.

best regard
Sal



> Sal:
> 
> I saw no way to download this code or get a copy.. and I seemed
> to get a LOT of adds and such when I hit this site :-<
> 
> Is there any place we can get the code at to play with ??
> 
> R
> 
> loretosa@vodafone.it wrote:
> 
> >Hi,
> >
> >I'm developping the first SIP over SCTP User Agent in Linux.
> >For this I use the SIP UA KPhone, and I'm modifying it.
> >
> >A first version of SIP SCTP client ( KPhone SCTP) is ready !!!
> >
> >If you are interested, I update my home page 
> >
> >( http://utenti.lycos.it/ssal/ )
> >
> >with all the necessary information in order to setup 
> >the enviroment and test the UA. 
> >In this page you'll find also the mail address where 
> >it is possible demand the source code !
> >
> >
> >Best regards
> >sal
> >
> >
> >_______________________________________________
> >tsvwg mailing list
> >tsvwg@ietf.org
> >https://www1.ietf.org/mailman/listinfo/tsvwg
> >
> >
> >
> >  
> >
> 
> 
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 


_______________________________________________
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From exim@www1.ietf.org  Thu Jul  3 19:41:32 2003
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Date: Thu, 03 Jul 2003 19:07:28 -0400
From: Cyrus Shaoul <cyrus@ntt-at.com>
To: Sally Floyd <floyd@icir.org>
Subject: Re[2]: [Tsvwg] VoIP and Congestive Collapes: QoS vs. Best Effort
Cc: tsvwg@ietf.org
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Thanks a lot for clearing this up for me. I have a follow-on thought: I
think that one aspect that the two drafts do share is a need for certain
applications to abide by certain guidelines (for example, calls always
terminate when packet loss becomes unacceptable, or video calls never
exceed or their requested bandwidth utilization after being admitted.)

Do you know if there is a generic way to enforce these requirements on
applications? It seems to me that if the guidelines are not enforceable,
there will be a tendency to descend into the "tragedy of the Commons" in
all these scenarios. (Somebody will always hack a non-compliant version
that ignores packet loss, and then distribute to the Internet User it as
a "SuperPhone" or something like that.) Is this part of the larger
"end-to-end vs. active core" debate? Is this out of the scope of the
TSVWG?

Thanks again, 

Cyrus


On Wed, 02 Jul 2003 22:27:30 -0700
Sally Floyd <floyd@icir.org> wrote:

SF> >Are these two drafts trying to solve different problems? 
SF> 
SF> The two drafts are trying to solve different problems.
SF> 
SF> The IAB draft, draft-iab-congestion-00.txt, is not trying to say
SF> that best-effort service is a good solution for VoIP traffic, or a
SF> better solution than some form of QoS, or anything of the kind.
SF> The IAB draft is trying to say that today some VoIP traffic in fact
SF> runs as best-effort traffic over one or more links in the best-effort
SF> Internet, and that this is likely to be the case in the near future
SF> as well, and that it is not deemed recommended behavior by the IAB
SF> for these flows to maintain their minimum sending rate in the
SF> presence of persistent, steady-state packet drop rates experienced
SF> by the flow.
SF> 
SF> We will try to be more clear, in the next round of revisions, that
SF> we are not trying to say that we are recommending best-effort service
SF> as the recommended path for roll-out for VoIP.  We are not recommending
SF> any roll-out path one way or another.  We are observing that in fact,
SF> VoIP traffic some times ends up competing, as best-effort traffic,
SF> with other best-effort traffic in the Internet.
SF> 
SF> - Sally
SF> http://www.icir.org/floyd/
SF> 
SF> 
SF> 
SF> 
SF> _______________________________________________
SF> tsvwg mailing list
SF> tsvwg@ietf.org
SF> https://www1.ietf.org/mailman/listinfo/tsvwg

Cyrus Shaoul
CEO
Tomigaia Inc.
http://www.tomigaia.com/
cyrus@tomigaia.com
646-645-6881



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From exim@www1.ietf.org  Thu Jul  3 19:45:20 2003
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To: Cyrus Shaoul <cyrus@ntt-at.com>
cc: tsvwg@ietf.org, Ratul Mahajan <ratul@cs.washington.edu>
From: Sally Floyd <floyd@icir.org>
Subject: Re: Re[2]: [Tsvwg] VoIP and Congestive Collapes: QoS vs. Best Effort 
Date: Thu, 03 Jul 2003 16:23:39 -0700
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>Thanks a lot for clearing this up for me. I have a follow-on thought: I
>think that one aspect that the two drafts do share is a need for certain
>applications to abide by certain guidelines (for example, calls always
>terminate when packet loss becomes unacceptable, or video calls never
>exceed or their requested bandwidth utilization after being admitted.)
>
>Do you know if there is a generic way to enforce these requirements on
>applications? It seems to me that if the guidelines are not enforceable,
>there will be a tendency to descend into the "tragedy of the Commons" in
>all these scenarios. (Somebody will always hack a non-compliant version
>that ignores packet loss, and then distribute to the Internet User it as
>a "SuperPhone" or something like that.) Is this part of the larger
>"end-to-end vs. active core" debate? Is this out of the scope of the
>TSVWG?

I agree that the network itself has to offer incentives for flows
to use end-to-end congestion control (or more precisely, mechanisms
for discouraging flows traversing congested paths from not using
end-to-end congestion control.)  This argument was given in a paper
with Kevin Fall on "Promoting the Use of End-to-End Congestion
Control in the Internet" ("http://www.icir.org/floyd/end2end-paper.html"),
and a router mechanism was proposed in a paper with Ratul Mahajan
on "Controlling High-Bandwidth Flows at the Congested Router"
(http://www.cs.washington.edu/homes/ratul/red-pd/).  A related paper
on controlling aggregates from flash crowds or DDoS attacks is a
paper on "Controlling High Bandwidth Aggregates in the Network",
with a number of coauthors, available from "http://www.icir.org/pushback/".

Ratul and I now have an internet-draft-in-progress, not yet submitted,
on "Router Primitives for Protection Against High-Bandwidth Flows
and Aggregates" ("http://www.icir.org/floyd/papers/draft-flows-00c.txt").
Any feedback on this would be appreciated.  This draft is intended
to be rather generic, and to encourage the practical deployment of
any such mechanisms in routers, so that the Internet community can
learn more from the deployment of such mechanisms.  (The current
draft doesn't consider the specific case of VoIP traffic, but that
could easily be added.)

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

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Date: Thu, 03 Jul 2003 20:19:30 -0400
From: Cyrus Shaoul <cyrus@ntt-at.com>
To: tsvwg@ietf.org
Subject: Re[4]: [Tsvwg] VoIP and Congestive Collapes: QoS vs. Best Effort
Cc: Ratul Mahajan <ratul@cs.washington.edu>
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Sally, 

I will read all of the documents you pointed to, and try to post any
feedback to this list.

Thanks, 

Cyrus

On Thu, 03 Jul 2003 16:23:39 -0700
Sally Floyd <floyd@icir.org> wrote:

SF> >Thanks a lot for clearing this up for me. I have a follow-on thought: I
SF> >think that one aspect that the two drafts do share is a need for certain
SF> >applications to abide by certain guidelines (for example, calls always
SF> >terminate when packet loss becomes unacceptable, or video calls never
SF> >exceed or their requested bandwidth utilization after being admitted.)
SF> >
SF> >Do you know if there is a generic way to enforce these requirements on
SF> >applications? It seems to me that if the guidelines are not enforceable,
SF> >there will be a tendency to descend into the "tragedy of the Commons" in
SF> >all these scenarios. (Somebody will always hack a non-compliant version
SF> >that ignores packet loss, and then distribute to the Internet User it as
SF> >a "SuperPhone" or something like that.) Is this part of the larger
SF> >"end-to-end vs. active core" debate? Is this out of the scope of the
SF> >TSVWG?
SF> 
SF> I agree that the network itself has to offer incentives for flows
SF> to use end-to-end congestion control (or more precisely, mechanisms
SF> for discouraging flows traversing congested paths from not using
SF> end-to-end congestion control.)  This argument was given in a paper
SF> with Kevin Fall on "Promoting the Use of End-to-End Congestion
SF> Control in the Internet" ("http://www.icir.org/floyd/end2end-paper.html"),
SF> and a router mechanism was proposed in a paper with Ratul Mahajan
SF> on "Controlling High-Bandwidth Flows at the Congested Router"
SF> (http://www.cs.washington.edu/homes/ratul/red-pd/).  A related paper
SF> on controlling aggregates from flash crowds or DDoS attacks is a
SF> paper on "Controlling High Bandwidth Aggregates in the Network",
SF> with a number of coauthors, available from "http://www.icir.org/pushback/".
SF> 
SF> Ratul and I now have an internet-draft-in-progress, not yet submitted,
SF> on "Router Primitives for Protection Against High-Bandwidth Flows
SF> and Aggregates" ("http://www.icir.org/floyd/papers/draft-flows-00c.txt").
SF> Any feedback on this would be appreciated.  This draft is intended
SF> to be rather generic, and to encourage the practical deployment of
SF> any such mechanisms in routers, so that the Internet community can
SF> learn more from the deployment of such mechanisms.  (The current
SF> draft doesn't consider the specific case of VoIP traffic, but that
SF> could easily be added.)
SF> 
SF> - Sally
SF> http://www.icir.org/floyd/

Cyrus Shaoul
NTT Advanced Technology Corp.
cyrus@ntt-at.com
http://www.ntt-at.com/



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To: Sally Floyd <floyd@icir.org>, Cyrus Shaoul <cyrus@ntt-at.com>
From: "MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp>
Subject: Re: Re[2]: [Tsvwg] VoIP and Congestive Collapes: QoS vs. Best
  Effort 
Cc: tsvwg@ietf.org, Ratul Mahajan <ratul@cs.washington.edu>
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Sally,

I have quickly reviewed the relevant documents.
My understanding is that your point is how to
coordinate best effort traffic regardless of TCP or UDP.
This is to avoid useless usage of the Internet.
One way is to impose self-control mechanism to each terminal
to reduce or to terminate the session
as is descirbed in IAB's concern.
Another way is to detect over-usage flows or aggregates at routers
to get rid of useless ones.

It seems reactive control and assumes single traffic class, i.e., best effort.

My point as is proposed in the following draft is
a kind of preventative control or admission control.
http://www.ietf.org/internet-drafts/draft-morita-tsvwg-pps-00.txt

The similarity is to rely on terminals behavior.

Naotaka

At 16:23 03/07/03 -0700, Sally Floyd wrote:
> >Thanks a lot for clearing this up for me. I have a follow-on thought: I
> >think that one aspect that the two drafts do share is a need for certain
> >applications to abide by certain guidelines (for example, calls always
> >terminate when packet loss becomes unacceptable, or video calls never
> >exceed or their requested bandwidth utilization after being admitted.)
> >
> >Do you know if there is a generic way to enforce these requirements on
> >applications? It seems to me that if the guidelines are not enforceable,
> >there will be a tendency to descend into the "tragedy of the Commons" in
> >all these scenarios. (Somebody will always hack a non-compliant version
> >that ignores packet loss, and then distribute to the Internet User it as
> >a "SuperPhone" or something like that.) Is this part of the larger
> >"end-to-end vs. active core" debate? Is this out of the scope of the
> >TSVWG?
>
>I agree that the network itself has to offer incentives for flows
>to use end-to-end congestion control (or more precisely, mechanisms
>for discouraging flows traversing congested paths from not using
>end-to-end congestion control.)  This argument was given in a paper
>with Kevin Fall on "Promoting the Use of End-to-End Congestion
>Control in the Internet" ("http://www.icir.org/floyd/end2end-paper.html"),
>and a router mechanism was proposed in a paper with Ratul Mahajan
>on "Controlling High-Bandwidth Flows at the Congested Router"
>(http://www.cs.washington.edu/homes/ratul/red-pd/).  A related paper
>on controlling aggregates from flash crowds or DDoS attacks is a
>paper on "Controlling High Bandwidth Aggregates in the Network",
>with a number of coauthors, available from "http://www.icir.org/pushback/".
>
>Ratul and I now have an internet-draft-in-progress, not yet submitted,
>on "Router Primitives for Protection Against High-Bandwidth Flows
>and Aggregates" ("http://www.icir.org/floyd/papers/draft-flows-00c.txt").
>Any feedback on this would be appreciated.  This draft is intended
>to be rather generic, and to encourage the practical deployment of
>any such mechanisms in routers, so that the Internet community can
>learn more from the deployment of such mechanisms.  (The current
>draft doesn't consider the specific case of VoIP traffic, but that
>could easily be added.)
>
>- Sally
>http://www.icir.org/floyd/
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg

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



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From exim@www1.ietf.org  Fri Jul 11 04:17:37 2003
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Hi,

Sourabh Ladha from UDEL is working with us over the summer looking into 
ways of adding Eifel to SCTP. In this context we have taken a closer look 
at the current proposals for spurious retransmit detection schemes. Here 
are some issues we found with draft-blanton-dsack-use-02 ...

A detection scheme that is intended to serve as the basis for reversing cc 
state should not declare an event a spurious retransmit if, in fact, it was 
not. This seemed to be the WG consensus after discussions about earlier 
versions of the Eifel detection algorithm. For that reason the final Eifel 
detection algorithm was extended by step (5) which is motivated in Section 
3.3 of RFC3522 ("Timeout due to loss of all ACKs"). With this extension, we 
believe that the Eifel detection algorithm is reliable, i.e., we are not 
aware of any case where it would declare an event a spurious retransmit if, 
in fact, it was not.
     [The price for that is that now we have some rare cases where spurious
      retransmits can occur that are *not* detected by Eifel.]

It seems that draft-blanton-dsack-use-02 is not that reliable since we 
think that we found the following cases where the algorithm will declare an 
event a spurious retransmit if, in fact, it was not:

1. "Timeout due to loss of all ACKs", see Section 3.3 of RFC3522.

2. Segment N is lost, the ACK for the retransmit of N is lost, and the 
second retransmit of N goes fine (the segment and the ACK for it) arrives.

3. Segment N is lost, and the retransmit of N is duplicated in the network.

Comments?

///Reiner & Sourabh 


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> It seems that draft-blanton-dsack-use-02 is not that reliable
> since we think that we found the following cases where the
> algorithm will declare an event a spurious retransmit if, in fact,
> it was not:
> 
> 1. "Timeout due to loss of all ACKs", see Section 3.3 of RFC3522.

Ah, yes.  So, this can be easily special cased as something like: if
everything is retransmitted and we get DSACK for everything then we
need to reduce cwnd.  (The wording needs careful crafting, but that
spirit, anyway.)

> 2. Segment N is lost, the ACK for the retransmit of N is lost, and
>    the second retransmit of N goes fine (the segment and the ACK
>    for it) arrives.
> 
> 3. Segment N is lost, and the retransmit of N is duplicated in the
>    network.

I believe that long ago when we coded this algorithm in ns that
Ethan and I talked about these (and, I bet there are others).  The
document is definately missing discussion that should be there,
though.

It seems that if we cannot accept these sort of cases (pathologies,
IMO) then we cannot use DSACK for this purpose.  That is, the DSACK
signal is fundementally weak in some respects and so these sorts of
cases are inevitable.  And, note, that at least for case (3) the
document offers *some* protection.  If the sender notices a DSACK
for something that was not retransmitted then it takes the
conservative assuption that since the network is duplicating
segments the DSACK signal is not to be believed.

So, the question really is... can we accept these sorts of cases or
not?  If we cannot accept these cases then DSACK (and duplicate TSN
notification in SCTP) is pretty much useless for reverting
congestion control state.  Opinions?

My own hit is that things like 2 & 3 are pathalogical and should not
be deal breakers for DSACK-based schemes for this purpose.

allman


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

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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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        Reiner Ludwig <Reiner.Ludwig@ericsson.com>,
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Subject: Re: [Tsvwg] Issues with draft-blanton-dsack-use-02 
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I think I agree with Mark...

Item 1 can be handled by a special case.

Item 2 seems like a corner case to me.

Item 3... I would expect that if the network duplicates one segment, the
network will duplicate others. The sender can detect that the network is
duplicating segments, and decide to not use DSACKs.

~armando

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

On Fri, 11 Jul 2003, Mark Allman wrote:

>
> > It seems that draft-blanton-dsack-use-02 is not that reliable
> > since we think that we found the following cases where the
> > algorithm will declare an event a spurious retransmit if, in fact,
> > it was not:
> >
> > 1. "Timeout due to loss of all ACKs", see Section 3.3 of RFC3522.
>
> Ah, yes.  So, this can be easily special cased as something like: if
> everything is retransmitted and we get DSACK for everything then we
> need to reduce cwnd.  (The wording needs careful crafting, but that
> spirit, anyway.)
>
> > 2. Segment N is lost, the ACK for the retransmit of N is lost, and
> >    the second retransmit of N goes fine (the segment and the ACK
> >    for it) arrives.
> >
> > 3. Segment N is lost, and the retransmit of N is duplicated in the
> >    network.
>
> I believe that long ago when we coded this algorithm in ns that
> Ethan and I talked about these (and, I bet there are others).  The
> document is definately missing discussion that should be there,
> though.
>
> It seems that if we cannot accept these sort of cases (pathologies,
> IMO) then we cannot use DSACK for this purpose.  That is, the DSACK
> signal is fundementally weak in some respects and so these sorts of
> cases are inevitable.  And, note, that at least for case (3) the
> document offers *some* protection.  If the sender notices a DSACK
> for something that was not retransmitted then it takes the
> conservative assuption that since the network is duplicating
> segments the DSACK signal is not to be believed.
>
> So, the question really is... can we accept these sorts of cases or
> not?  If we cannot accept these cases then DSACK (and duplicate TSN
> notification in SCTP) is pretty much useless for reverting
> congestion control state.  Opinions?
>
> My own hit is that things like 2 & 3 are pathalogical and should not
> be deal breakers for DSACK-based schemes for this purpose.
>
> allman
>
>
> --
> Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>



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        Reiner Ludwig <Reiner.Ludwig@ericsson.com>,
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Subject: Re: [Tsvwg] Issues with draft-blanton-dsack-use-02 
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> Item 2 seems like a corner case to me.

It may be..though  note that in the example we gave the ACK
for the first retransmit does not necessarily have to be lost. In general
the case can be described that the second retransmit got triggered before
the ACK for the first retransmit arrived. I wonder if SCTP
multihoming with diverse paths (retransmissions going to alternate(s))
could make this case less pathological?

> Item 3... I would expect that if the network duplicates one segment, the
> network will duplicate others. The sender can detect that the network is
> duplicating segments, and decide to not use DSACKs.

Well, in SCTP (multihomed endpoints) retransmissions SHOULD go via
alternate path. If the alternate path is duplicating packets then
the sender may never be able to realize that it is detecting losses as
spurious  retransmits (i.e. the sort of protection currently in the given
in the draft is rendered useless).

I think Mark very well noted in the mail that DSACKs alone are
weak signals and the draft notes that timestamp option may futher be used in
conjunction to refine the algorithm for network duplication scenarios.


-Sourabh



> ~armando
>
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0
>
> On Fri, 11 Jul 2003, Mark Allman wrote:
>
> >
> > > It seems that draft-blanton-dsack-use-02 is not that reliable
> > > since we think that we found the following cases where the
> > > algorithm will declare an event a spurious retransmit if, in fact,
> > > it was not:
> > >
> > > 1. "Timeout due to loss of all ACKs", see Section 3.3 of RFC3522.
> >
> > Ah, yes.  So, this can be easily special cased as something like: if
> > everything is retransmitted and we get DSACK for everything then we
> > need to reduce cwnd.  (The wording needs careful crafting, but that
> > spirit, anyway.)
> >
> > > 2. Segment N is lost, the ACK for the retransmit of N is lost, and
> > >    the second retransmit of N goes fine (the segment and the ACK
> > >    for it) arrives.
> > >
> > > 3. Segment N is lost, and the retransmit of N is duplicated in the
> > >    network.
> >
> > I believe that long ago when we coded this algorithm in ns that
> > Ethan and I talked about these (and, I bet there are others).  The
> > document is definately missing discussion that should be there,
> > though.
> >
> > It seems that if we cannot accept these sort of cases (pathologies,
> > IMO) then we cannot use DSACK for this purpose.  That is, the DSACK
> > signal is fundementally weak in some respects and so these sorts of
> > cases are inevitable.  And, note, that at least for case (3) the
> > document offers *some* protection.  If the sender notices a DSACK
> > for something that was not retransmitted then it takes the
> > conservative assuption that since the network is duplicating
> > segments the DSACK signal is not to be believed.
> >
> > So, the question really is... can we accept these sorts of cases or
> > not?  If we cannot accept these cases then DSACK (and duplicate TSN
> > notification in SCTP) is pretty much useless for reverting
> > congestion control state.  Opinions?
> >
> > My own hit is that things like 2 & 3 are pathalogical and should not
> > be deal breakers for DSACK-based schemes for this purpose.
> >
> > allman
> >
> >
> > --
> > Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
> >
> > _______________________________________________
> > tsvwg mailing list
> > tsvwg@ietf.org
> > https://www1.ietf.org/mailman/listinfo/tsvwg
> >
>
>
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>


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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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        Reiner Ludwig <Reiner.Ludwig@ericsson.com>,
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Subject: Re: [Tsvwg] Issues with draft-blanton-dsack-use-02 
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On Fri, 11 Jul 2003, Sourabh Ladha wrote:

> > Item 2 seems like a corner case to me.
>
> It may be..though  note that in the example we gave the ACK
> for the first retransmit does not necessarily have to be lost. In general
> the case can be described that the second retransmit got triggered before
> the ACK for the first retransmit arrived. I wonder if SCTP
> multihoming with diverse paths (retransmissions going to alternate(s))
> could make this case less pathological?

It may be less pathological with a multihomed SCTP association. However,
what's the loss in making a mistake on an alternate path? The cwnd of
alternate paths generally stays close to 2 MTU due to cwnd degradation (no
matter which algorithm you go with: RFC2960 or imp guide v8).

> > Item 3... I would expect that if the network duplicates one segment, the
> > network will duplicate others. The sender can detect that the network is
> > duplicating segments, and decide to not use DSACKs.
>
> Well, in SCTP (multihomed endpoints) retransmissions SHOULD go via
> alternate path. If the alternate path is duplicating packets then
> the sender may never be able to realize that it is detecting losses as
> spurious  retransmits (i.e. the sort of protection currently in the given
> in the draft is rendered useless).

Ok, this is true...

> I think Mark very well noted in the mail that DSACKs alone are
> weak signals and the draft notes that timestamp option may futher be used in
> conjunction to refine the algorithm for network duplication scenarios.

I agree.

~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: "MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp>
cc: tsvwg@ietf.org
From: Sally Floyd <floyd@icir.org>
Subject: Re: Re[2]: [Tsvwg] VoIP and Congestive Collapes: QoS vs. Best Effort 
Date: Sun, 13 Jul 2003 06:20:34 -0700
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>My point as is proposed in the following draft is
>a kind of preventative control or admission control.
>http://www.ietf.org/internet-drafts/draft-morita-tsvwg-pps-00.txt

Thanks for the pointer.  We are planning to add a brief section in the
document with pointers to efforts towards QoS in the Internet (though the
draft only addresses that traffic that might be competing directly
with best-effort traffic on some congested link).

- Sally

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To: Sourabh Ladha <ladha@mail.eecis.udel.edu>
From: Mark Allman <mallman@grc.nasa.gov>
Reply-To: mallman@grc.nasa.gov
Cc: "Armando L. Caro Jr." <acaro@ACM.ORG>, tsvwg@ietf.org,
        Reiner Ludwig <Reiner.Ludwig@ericsson.com>,
        Ethan Blanton <eblanton@prime.cs.ohiou.edu>
Subject: Re: [Tsvwg] Issues with draft-blanton-dsack-use-02 
In-Reply-To: <Pine.GSO.4.33.0307111335470.7703-100000@louie.udel.edu> 
Organization: BBN Technologies/NASA GRC
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> > Item 2 seems like a corner case to me.
> 
> It may be..though note that in the example we gave the ACK for the
> first retransmit does not necessarily have to be lost. In general
> the case can be described that the second retransmit got triggered
> before the ACK for the first retransmit arrived. I wonder if SCTP
> multihoming with diverse paths (retransmissions going to
> alternate(s)) could make this case less pathological?

If you consider the loss rate to be independent in the two
directions (or, independent on two paths for SCTP) then even a high
loss rate would yield quite low occurances of hitting the case you
mentioned.  Say the loss rate was 10% in both directions.  Now, we
have a 1% chance of falling into this situation.  That seems
pathological to me.

Now, if you broaden things to say that excessive delay imposed on
either the retransmit or the corresponding ACK can also cause us to
fall into this situation (which I agree with) that may increase the
chances.  But, I have no good idea how to quantify the chances of
that.  (My gut says that the chances of getting bit by this remain
low.) 

I'd be interested in hearing if others think that issues 2 & 3 are
showstoppers for this sort of DSACK use (reverting CC on mistakes).

Thanks!

allman


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

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From exim@www1.ietf.org  Tue Jul 15 05:32:29 2003
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From: Stephan Baucke <stephan.baucke@ericsson.com>
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Subject: [Tsvwg] SCTP Timestamps
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Hi all,

we have been thinking about timestamps a la RFC 1323 for SCTP. We see a 
number of compelling uses:

- Allowing for better and more frequent RTT measurements

- Resolving the "path ambiguity" in a multihomed association: If an ACK 
arrives for a chunk that has been retransmitted one or more times on 
alternating paths, it is not possible to safely determine which 
destination the ACK should be credited to (this is briefly mentioned in 
section 8.2 of RFC 2960). This may result in resetting the wrong error 
counter (delaying the detection of an inactive path), or in unfair 
behavior if the congestion window of the wrong path is adapted as a 
result of the ACK.

- Resolving the retransmission ambiguity, i.e. enabling the sender to 
determine whether an ACK refers to the original transmission or a 
retransmission of a chunk. This is a necessary precondition for 
implementing the Eifel algorithm for SCTP (which we are currently 
looking at).


It seems to me there is a pretty strong case for having timestamps in 
SCTP. Unfortunately there are no provisions for a timestamp option 
similar to TCP's in the SCTP common header. There has been a proposal 
(first brought up by Armando, I believe) to introduce a timestamp chunk. 
This is fine for doing RTT measurements, but for the latter two cases 
mentioned above, it would require that the timestamp chunk is always 
bundled with the data chunk it is referring to.

My question is how such a "mandatory bundling" is viewed in the 
community. In some ways it seems to be a break with existing SCTP 
"philosophy", since so far all chunk types are "self contained", and 
there is no known case of mandatory bundling. Is it desirable to use new 
chunk types as a replacement for TCP-like per-packet options? There is 
also the very practical problem of fragmentation. If we want to avoid 
possible refragmentation of data chunks, we would have to at least 
reserve room for the timestamp chunk in each packet (this could be 
negotiated at connection setup). What happens if we want to introduce 
further per-packet information in the future?

Stephan.


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From exim@www1.ietf.org  Tue Jul 15 06:32:29 2003
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Date: Tue, 15 Jul 2003 12:23:32 +0200
Subject: Re: [Tsvwg] SCTP Timestamps
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To: Stephan Baucke <stephan.baucke@ericsson.com>
From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Hi Stephan,

I'm not sure if you want per packet info or per DATA chunk info?

In case of per packet info you could define a new chunk and have it
in the packets.

In case of per DATA chunk info there is the possibility of defining a
new EXT_DATA chunk which is like a DATA chunk but also holds the 
additional
information.

Both ways require an extension of the protocol.

Best regards
Michael

On Tuesday, Jul 15, 2003, at 11:29 Europe/Berlin, Stephan Baucke wrote:

> Hi all,
>
> we have been thinking about timestamps a la RFC 1323 for SCTP. We see 
> a number of compelling uses:
>
> - Allowing for better and more frequent RTT measurements
>
> - Resolving the "path ambiguity" in a multihomed association: If an 
> ACK arrives for a chunk that has been retransmitted one or more times 
> on alternating paths, it is not possible to safely determine which 
> destination the ACK should be credited to (this is briefly mentioned 
> in section 8.2 of RFC 2960). This may result in resetting the wrong 
> error counter (delaying the detection of an inactive path), or in 
> unfair behavior if the congestion window of the wrong path is adapted 
> as a result of the ACK.
>
> - Resolving the retransmission ambiguity, i.e. enabling the sender to 
> determine whether an ACK refers to the original transmission or a 
> retransmission of a chunk. This is a necessary precondition for 
> implementing the Eifel algorithm for SCTP (which we are currently 
> looking at).
>
>
> It seems to me there is a pretty strong case for having timestamps in 
> SCTP. Unfortunately there are no provisions for a timestamp option 
> similar to TCP's in the SCTP common header. There has been a proposal 
> (first brought up by Armando, I believe) to introduce a timestamp 
> chunk. This is fine for doing RTT measurements, but for the latter two 
> cases mentioned above, it would require that the timestamp chunk is 
> always bundled with the data chunk it is referring to.
>
> My question is how such a "mandatory bundling" is viewed in the 
> community. In some ways it seems to be a break with existing SCTP 
> "philosophy", since so far all chunk types are "self contained", and 
> there is no known case of mandatory bundling. Is it desirable to use 
> new chunk types as a replacement for TCP-like per-packet options? 
> There is also the very practical problem of fragmentation. If we want 
> to avoid possible refragmentation of data chunks, we would have to at 
> least reserve room for the timestamp chunk in each packet (this could 
> be negotiated at connection setup). What happens if we want to 
> introduce further per-packet information in the future?
>
> Stephan.
>
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>


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Subject: Re: [Tsvwg] SCTP Timestamps
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Stephan,

"Better and more frequent RTT measurements"?

Better how?  More frequent?  Not significantly so, except
perhaps during special (read: rare) anomalies I would think.

Clear the wrong error counter?  RFC 2960 says:

  If this ambiguity is undesirable, the transmitter may choose
  not to clear the error counter if the last chunk was a
  retransmission.

Where is the congestion window of the wrong path adapted as a
result of an ACK?

I think this could only be as a result of some really strange
situation.  Besides, not crediting a destination cwnd for
retransmitted packets would be more conservative than
crediting, and is therefore, permitted.  The same applies, if
you don't like possibly crediting the wrong destination for
the retransmitted packet, don't credit any destination for a
retransmitted packet.  It shouldn't really matter, some
destination got dinged really bad for slipping a RTO in the
first place.  This exponential back off far overwhelms any
meager linear growth from a few retransmitted packets.

"Unfair behavior"?

Fairness between paths is not of consideration.  Congestion
fairness between TCP connections and SCTP associations is.
Only the primary gets credited when all is well and the other
paths suck slow start, even if heartbeats show superb RTT.
How fair is that?

What do you mean "refragmentation [sic] of data chunks"?

Once fragmentation of data is made into data chunks, they
cannot be re-fragmented.  See RFC 2960 Section 6.9:

  Note: Once a message is fragmented it cannot be re-fragmented.
  Instead if the PMTU has been reached, then IP fragmentation must be
  used. ...

Perhaps Eifel an TSN wrap on Googleplex-bytes-per-second paths
might be reasons for timestamps, but the above don't sound
like justifications.

If you want to take a really accurate RTT measurement in SCTP,
send a heartbeat and place whatever timestamps and identifiers
you want in the heartbeat data.

I find sending a heartbeat just after Path.Max.Retrans has
been exceeded to be quite useful for quickly restoring paths
to which you were simply too cruel, and particularly when
Path.Max.Retrans is set to 0.

--brian

On Tue, 15 Jul 2003, Stephan Baucke wrote:

> Hi all,
>=20
> we have been thinking about timestamps a la RFC 1323 for SCTP. We see a=
=20
> number of compelling uses:
>=20
> - Allowing for better and more frequent RTT measurements
>=20
> - Resolving the "path ambiguity" in a multihomed association: If an ACK=
=20
> arrives for a chunk that has been retransmitted one or more times on=20
> alternating paths, it is not possible to safely determine which=20
> destination the ACK should be credited to (this is briefly mentioned in=
=20
> section 8.2 of RFC 2960). This may result in resetting the wrong error=20
> counter (delaying the detection of an inactive path), or in unfair=20
> behavior if the congestion window of the wrong path is adapted as a=20
> result of the ACK.
>=20
> - Resolving the retransmission ambiguity, i.e. enabling the sender to=20
> determine whether an ACK refers to the original transmission or a=20
> retransmission of a chunk. This is a necessary precondition for=20
> implementing the Eifel algorithm for SCTP (which we are currently=20
> looking at).
>=20
>=20
> It seems to me there is a pretty strong case for having timestamps in=20
> SCTP. Unfortunately there are no provisions for a timestamp option=20
> similar to TCP's in the SCTP common header. There has been a proposal=20
> (first brought up by Armando, I believe) to introduce a timestamp chunk=
.=20
> This is fine for doing RTT measurements, but for the latter two cases=20
> mentioned above, it would require that the timestamp chunk is always=20
> bundled with the data chunk it is referring to.
>=20
> My question is how such a "mandatory bundling" is viewed in the=20
> community. In some ways it seems to be a break with existing SCTP=20
> "philosophy", since so far all chunk types are "self contained", and=20
> there is no known case of mandatory bundling. Is it desirable to use ne=
w=20
> chunk types as a replacement for TCP-like per-packet options? There is=20
> also the very practical problem of fragmentation. If we want to avoid=20
> possible refragmentation of data chunks, we would have to at least=20
> reserve room for the timestamp chunk in each packet (this could be=20
> negotiated at connection setup). What happens if we want to introduce=20
> further per-packet information in the future?
>=20
> Stephan.
>=20
>=20
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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

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Transport Area WG (tsvwg)

Friday, July 18 at 0900-1130
=============================

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

0900 Charter Discussion (chairs)
0910 General Area Review (chairs)

0925 Fast TCP (Cheng Jin)

0950 Gbit TCP (Sally Floyd)

1010 NewReno to PS (Sally Floyd)

1030 Eifel Response (Reiner Ludwig)

1045 MH-PHB (Naotaka Morita)

1100 Update on PR-SCTP (Randall Stewart)

1115 Update on F-RTO (Psi Sarolahti)

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Subject: Re: [Tsvwg] SCTP Timestamps
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Brian F. G. Bidulock wrote:

> "Better and more frequent RTT measurements"?
> 
> Better how?  More frequent?  Not significantly so, except
> perhaps during special (read: rare) anomalies I would think.

Timestamps would enable us to do RTTM (RTT measurement on every packet), 
as described in RFC 1323. Together with an appropriate timer 
calculation, this has the potential for an RTO that more accurately 
follows the RTT. Also, since timestamps resolve the retransmission 
ambiguity, we don't have to use Karn's algorithm. Karn's algorithm 
together with SCTP's retransmission policy has been shown to cause an 
overly conservative RTO in certain multihomed scenarios (see 
http://www.cis.udel.edu/~acaro/papers/2003.milcom.acaro.pdf)

> Clear the wrong error counter?  RFC 2960 says:
> 
>   If this ambiguity is undesirable, the transmitter may choose
>   not to clear the error counter if the last chunk was a
>   retransmission.

This helps, but can result in a premature failure detection on the 
secondary path (which is usually only used for retransmissions).

> Where is the congestion window of the wrong path adapted as a
> result of an ACK?

If an ACK is credited to a wrong path, the CWND on that path will be 
falsely increased where it should be decreased.

> I think this could only be as a result of some really strange
> situation.

Consider the following example:
1) Chunk A is sent over the primary path
2) A spurious timeout occurs on the primary path
3) Chunk A is retransmitted over the secondary path
4) The packet is lost on the secondary path
5) Finally, the SACK arrives for the original transmission over the 
pirmary path

According to RFC 2960, the SACK will be credited to the secondary path, 
since that's where the last retransmission took place. So, we will 
increase its congestion window where we should really halve it or set it 
to <=2*MTU, since a packet loss occured on that path.

 >  Besides, not crediting a destination cwnd for
> retransmitted packets would be more conservative than
> crediting, and is therefore, permitted.  The same applies, if
> you don't like possibly crediting the wrong destination for
> the retransmitted packet, don't credit any destination for a
> retransmitted packet.

We would still fail to reduce the cwnd on one path.

> "Unfair behavior"?
> 
> Fairness between paths is not of consideration.  Congestion
> fairness between TCP connections and SCTP associations is.

Of course I was referring to the latter case.

> What do you mean "refragmentation [sic] of data chunks"?
> 
> Once fragmentation of data is made into data chunks, they
> cannot be re-fragmented.  See RFC 2960 Section 6.9:
> 
>   Note: Once a message is fragmented it cannot be re-fragmented.
>   Instead if the PMTU has been reached, then IP fragmentation must be
>   used. ...

Yes. That's why we would run into problems if we were to use new chunk 
types as per-packet options without reserving space for them in every 
packet, regardless whether the option is actually used in all packets or 
only a part of them.

> Perhaps Eifel an TSN wrap on Googleplex-bytes-per-second paths
> might be reasons for timestamps, but the above don't sound
> like justifications.
> 
> If you want to take a really accurate RTT measurement in SCTP,
> send a heartbeat and place whatever timestamps and identifiers
> you want in the heartbeat data.

Heartbeats might be too infrequent for some scenarios.

Stephan.


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Date: Tue, 15 Jul 2003 09:16:31 -0400 (EDT)
Subject: Re: [Tsvwg] SCTP Timestamps
To: bidulock@openss7.org
Cc: "Stephan Baucke" <stephan.baucke@ericsson.com>, tsvwg@ietf.org
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Brian,

> "Better and more frequent RTT measurements"?
>
> Better how?  More frequent?  Not significantly so, except
> perhaps during special (read: rare) anomalies I would think.

I hope that you know the answer to "More Frequent?". As far as better is
concerned, I think this has been argued before on the list as well as in
research papers. You can probably look for RTTM in RFC1323 and Phil Karn's
1987 paper which brings out the Retransmission-Ambuiguity problem and on
the other side Allman's 1999 SIGCOMM paper. I wonder if you are also
questioning timestamps in TCP ?


> Clear the wrong error counter?  RFC 2960 says:
>
>   If this ambiguity is undesirable, the transmitter may choose
            ^^^^^^^^^
>   not to clear the error counter if the last chunk was a
>   retransmission.

I thought the point was that with timestamps you can avoid this ambuiguity.


> Where is the congestion window of the wrong path adapted as a
> result of an ACK?
>
> I think this could only be as a result of some really strange
> situation.  Besides, not crediting a destination cwnd for
> retransmitted packets would be more conservative than
> crediting, and is therefore, permitted.  The same applies, if
> you don't like possibly crediting the wrong destination for
> the retransmitted packet, don't credit any destination for a
> retransmitted packet.  It shouldn't really matter, some
> destination got dinged really bad for slipping a RTO in the
> first place.  This exponential back off far overwhelms any
> meager linear growth from a few retransmitted packets.
>
> "Unfair behavior"?
>
> Fairness between paths is not of consideration.  Congestion
> fairness between TCP connections and SCTP associations is.
> Only the primary gets credited when all is well and the other
> paths suck slow start, even if heartbeats show superb RTT.
> How fair is that?

I do not think Stephan's meant or even mentioned "fairness between paths".
I believe he meant that the "Retransmission-Ambuiguity" may cause the
wrong destination to be credited for cwnd. A simplistic situation is: If
you have a timeout and retransmit on the alternate, then you do not know
on the reciept of an ACK if it corresponds to the retransmission or the
original (which may have got delayed). Try adapting this to cwnd crediting
as well and you will see what Stephan referred to. I believe you can find
some action related to this at 
http://www.cis.udel.edu/~iyengar/publications/
(papers: [1] Making SCTP More Robust to Changeover [2] SCTP Congestion
Window Overgrowth During Changeover)


> What do you mean "refragmentation [sic] of data chunks"?
>
> Once fragmentation of data is made into data chunks, they
> cannot be re-fragmented.  See RFC 2960 Section 6.9:
>
>   Note: Once a message is fragmented it cannot be re-fragmented.
>   Instead if the PMTU has been reached, then IP fragmentation must be
>   used. ...

I agree. Maybe the issue was that for mandatory bundling of timestamp
chunks one will have to take into consideration the association-PMTU as
well as the timestamp-chunk size before fragmenting.


 -Sourabh

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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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To: Stephan Baucke <stephan.baucke@ericsson.com>
Cc: <tsvwg@ietf.org>
Subject: Re: [Tsvwg] SCTP Timestamps
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On Tue, 15 Jul 2003, Stephan Baucke wrote:

> My question is how such a "mandatory bundling" is viewed in the
> community. In some ways it seems to be a break with existing SCTP
> "philosophy", since so far all chunk types are "self contained", and
> there is no known case of mandatory bundling.

First, I don't see a need to make it mandatory. The sender can choose to
bundle timestamps with all packets containing data and/or sack chunks. If
the sender chooses not to include timestamps, then the sender must follow
the conservative behavior anytime an ambiguous scenario arises.

Second, I think the ECNE and CWR chunks are examples of chunk types that,
if used, need to be in every packet. The sender simply needs to reserve
space for these chunks.

~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] SCTP Timestamps
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On Tue, 15 Jul 2003, Stephan Baucke wrote:

> Brian F. G. Bidulock wrote:
>
> > "Better and more frequent RTT measurements"?
> >
> > Better how?  More frequent?  Not significantly so, except
> > perhaps during special (read: rare) anomalies I would think.
>
> Timestamps would enable us to do RTTM (RTT measurement on every packet),
> as described in RFC 1323. Together with an appropriate timer
> calculation, this has the potential for an RTO that more accurately
> follows the RTT.

"Together with an appropriate timer calculation" is an important clause.
As mentioned in RFC2960, Mark Allman and Vern Paxson show that measuring
RTTs more frequently than once per RTT does not yield any significant
benefit. Also, if measurements are made more often, RTO.Alpha and RTO.Beta
should be adjusted, but it is still an open question how to adjust the RTO
calculation.

Allman, M. and Paxson, V., "On Estimating End-to-End Network Path
Properties", Proc. SIGCOMM'99, 1999.

But with multihoming, timestamps can be useful for alternate paths as you
allude to below.

> Also, since timestamps resolve the retransmission
> ambiguity, we don't have to use Karn's algorithm. Karn's algorithm
> together with SCTP's retransmission policy has been shown to cause an
> overly conservative RTO in certain multihomed scenarios (see
> http://www.cis.udel.edu/~acaro/papers/2003.milcom.acaro.pdf)

Also, see
http://www.cis.udel.edu/~acaro/papers/2003.icon.acaro.pdf

or the full tech report
http://www.cis.udel.edu/~acaro/papers/2003.TR2003-05.acaro.pdf

> > Clear the wrong error counter?  RFC 2960 says:
> >
> >   If this ambiguity is undesirable, the transmitter may choose
> >   not to clear the error counter if the last chunk was a
> >   retransmission.
>
> This helps, but can result in a premature failure detection on the
> secondary path (which is usually only used for retransmissions).

Yes. Also the point is, if we can eliminate the ambiguity to improve
performance, then we should. Should we not??

> > Where is the congestion window of the wrong path adapted as a
> > result of an ACK?
>
> If an ACK is credited to a wrong path, the CWND on that path will be
> falsely increased where it should be decreased.

Timestamps may help alleviate this problem, but I think the SFR-CACC
algorithm is a better solution. The SFR-CACC algorithm requires
changes only at the sender side without extra bits on the wire.

See:
http://www.cis.udel.edu/~acaro/papers/2003.spects.iyengar.pdf
http://www.cis.udel.edu/~acaro/papers/2002.sci.iyengar.pdf

Jana Iyengar has also made more progress with the algorithm to prevent the
same issue from occuring with cycling changeovers. Plus, the improved
algorithm is actually simpler. Unfortunately, he doesn't have any
documents yet which explain the modified algorithm.

> > I think this could only be as a result of some really strange
> > situation.
>
> Consider the following example:
> 1) Chunk A is sent over the primary path
> 2) A spurious timeout occurs on the primary path
> 3) Chunk A is retransmitted over the secondary path
> 4) The packet is lost on the secondary path
> 5) Finally, the SACK arrives for the original transmission over the
> pirmary path
>
> According to RFC 2960, the SACK will be credited to the secondary path,
> since that's where the last retransmission took place. So, we will
> increase its congestion window where we should really halve it or set it
> to <=2*MTU, since a packet loss occured on that path.

Your scenario is very similar to the example scenario which Jana uses,
except his scenario involves a changeover which causes spurious fast
retransmits.

>  >  Besides, not crediting a destination cwnd for
> > retransmitted packets would be more conservative than
> > crediting, and is therefore, permitted.  The same applies, if
> > you don't like possibly crediting the wrong destination for
> > the retransmitted packet, don't credit any destination for a
> > retransmitted packet.
>
> We would still fail to reduce the cwnd on one path.

Yes, that's right. The CACC problem causes spurious retransmits on the
alternate/new path and incorrect cwnd growth on the alternate/new path
...even if the retransmissions on the alternate/new path are lost!

> > "Unfair behavior"?
> >
> > Fairness between paths is not of consideration.  Congestion
> > fairness between TCP connections and SCTP associations is.
>
> Of course I was referring to the latter case.

Obviously, incorrectly increasing the cwnd as described above is not fair
to TCP, TCP-friendly, Internet-friendly, or whatever you want to call it.

> Yes. That's why we would run into problems if we were to use new chunk
> types as per-packet options without reserving space for them in every
> packet, regardless whether the option is actually used in all packets or
> only a part of them.

I am not sure I understand the problem here. I know there is some
inefficiency with reserving space for control chunks, but since timestamps
will be essentially in every packet... what's the problem?

> > Perhaps Eifel an TSN wrap on Googleplex-bytes-per-second paths
> > might be reasons for timestamps, but the above don't sound
> > like justifications.
> >
> > If you want to take a really accurate RTT measurement in SCTP,
> > send a heartbeat and place whatever timestamps and identifiers
> > you want in the heartbeat data.
>
> Heartbeats might be too infrequent for some scenarios.

Yes. You _could_ decrease the HB.interval, but then the sender would
introduce more "useless" packets on the wire... which I don't particularly
like.

~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] SCTP Timestamps
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> On Tue, 15 Jul 2003, Stephan Baucke wrote:
>
> > My question is how such a "mandatory bundling" is viewed in the
> > community. In some ways it seems to be a break with existing SCTP
> > "philosophy", since so far all chunk types are "self contained", and
> > there is no known case of mandatory bundling.
>
> First, I don't see a need to make it mandatory. The sender can choose to
> bundle timestamps with all packets containing data and/or sack chunks. If
> the sender chooses not to include timestamps, then the sender must follow
> the conservative behavior anytime an ambiguous scenario arises.

Of course if adopted, the timestamp chunk parameter will have to be
defined to be negotiated in the INIT/INIT-ACK. By mandatory the reference
was that if both endpoints agree on timestamp then the timestamp chunk
MUST be bundled with all packets containing DATA/SACK chunk(s). Your
clarification below addresses that concern, I think.

> Second, I think the ECNE and CWR chunks are examples of chunk types that,
> if used, need to be in every packet. The sender simply needs to reserve
> space for these chunks.
>

-Sourabh


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In a private discussion with Sourabh, he pointed out to me that the CACC
algorithm will not solve the problem presented below by Stephan. Although
the side effects of the scenario are similar, the CACC relies on Gap Ack
information to solve the problem. My mistake... So I take back what I say,
and agree that timestamps are a good solution.

Sorry for the confusion.

~armando

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

On Tue, 15 Jul 2003, Armando L. Caro Jr. wrote:

> > > Where is the congestion window of the wrong path adapted as a
> > > result of an ACK?
> >
> > If an ACK is credited to a wrong path, the CWND on that path will be
> > falsely increased where it should be decreased.
>
> Timestamps may help alleviate this problem, but I think the SFR-CACC
> algorithm is a better solution. The SFR-CACC algorithm requires
> changes only at the sender side without extra bits on the wire.
>
> See:
> http://www.cis.udel.edu/~acaro/papers/2003.spects.iyengar.pdf
> http://www.cis.udel.edu/~acaro/papers/2002.sci.iyengar.pdf
>
> Jana Iyengar has also made more progress with the algorithm to prevent the
> same issue from occuring with cycling changeovers. Plus, the improved
> algorithm is actually simpler. Unfortunately, he doesn't have any
> documents yet which explain the modified algorithm.
>
> > > I think this could only be as a result of some really strange
> > > situation.
> >
> > Consider the following example:
> > 1) Chunk A is sent over the primary path
> > 2) A spurious timeout occurs on the primary path
> > 3) Chunk A is retransmitted over the secondary path
> > 4) The packet is lost on the secondary path
> > 5) Finally, the SACK arrives for the original transmission over the
> > pirmary path
> >
> > According to RFC 2960, the SACK will be credited to the secondary path,
> > since that's where the last retransmission took place. So, we will
> > increase its congestion window where we should really halve it or set it
> > to <=2*MTU, since a packet loss occured on that path.
>
> Your scenario is very similar to the example scenario which Jana uses,
> except his scenario involves a changeover which causes spurious fast
> retransmits.
>
> >  >  Besides, not crediting a destination cwnd for
> > > retransmitted packets would be more conservative than
> > > crediting, and is therefore, permitted.  The same applies, if
> > > you don't like possibly crediting the wrong destination for
> > > the retransmitted packet, don't credit any destination for a
> > > retransmitted packet.
> >
> > We would still fail to reduce the cwnd on one path.
>
> Yes, that's right. The CACC problem causes spurious retransmits on the
> alternate/new path and incorrect cwnd growth on the alternate/new path
> ...even if the retransmissions on the alternate/new path are lost!
>
> > > "Unfair behavior"?
> > >
> > > Fairness between paths is not of consideration.  Congestion
> > > fairness between TCP connections and SCTP associations is.
> >
> > Of course I was referring to the latter case.
>
> Obviously, incorrectly increasing the cwnd as described above is not fair
> to TCP, TCP-friendly, Internet-friendly, or whatever you want to call it.


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Subject: Re: [Tsvwg] SCTP Timestamps
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Hi,

Pls see me comments below...

Sourabh Ladha wrote:
> 
> > On Tue, 15 Jul 2003, Stephan Baucke wrote:
> >
> > > My question is how such a "mandatory bundling" is viewed in the
> > > community. In some ways it seems to be a break with existing SCTP
> > > "philosophy", since so far all chunk types are "self contained", and
> > > there is no known case of mandatory bundling.
> >
> > First, I don't see a need to make it mandatory. The sender can choose to
> > bundle timestamps with all packets containing data and/or sack chunks. If
> > the sender chooses not to include timestamps, then the sender must follow
> > the conservative behavior anytime an ambiguous scenario arises.
> 
> Of course if adopted, the timestamp chunk parameter will have to be
> defined to be negotiated in the INIT/INIT-ACK. By mandatory the reference
> was that if both endpoints agree on timestamp then the timestamp chunk
> MUST be bundled with all packets containing DATA/SACK chunk(s). Your
> clarification below addresses that concern, I think.

Whenever I see a "MUST" used (or being suggested to be used) in the
spec, I'd like to understand what would be the error handling strategy
if this "MUST" is violated. Since you are suggesting a "MUST" clause
here, would you immediately severe the communication with a peer that
had agreed to use timestamp feature but you subsequently receive a
packet from it without the timestamp chunk bundled? If your answer is
"yes", then I'd guess that you somehow must view this missing timestamp
as an unrecoverable protocol error which can do potential harm to your
endpoint or the network if not stopping the communication right away.
But this is not obvious to me, so please explain. If your answer is
"no", then I'd question why you want to use a "MUST" here.

Another question: since the timestamp information (however it would be
defined for your purposes) seems to only have local significance to the
data sender. If so, this info can be kept opaque to the peer/receiver.
Since the SCTP heartbeat (it has been clarified that the HB can be sent
at different frequency or even "manually" triggered by the ULP whenever
one is needed) has the capability to carry and return an opaque data
blob, I don't see why you need to define a new timestamp chunk? 

> > Second, I think the ECNE and CWR chunks are examples of chunk types that,
> > if used, need to be in every packet. The sender simply needs to reserve
> > space for these chunks.

To reserve room for control chunks is part of the SCTP fragmentation
code present today in *every* implementation that supports
fragmentation. So this should not be a problem and should add no
complexity.

-Qiaobing

> >
> 
> -Sourabh
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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From exim@www1.ietf.org  Tue Jul 15 19:38:29 2003
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From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] SCTP Timestamps
To: Stephan Baucke <stephan.baucke@ericsson.com>
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On Tuesday, July 15, 2003, stephan.baucke@ericsson.com wrote:

>My question is how such a "mandatory bundling" is viewed in the
>community. In some ways it seems to be a break with existing
>SCTP "philosophy", since so far all chunk types are "self
>contained", and there is no known case of mandatory bundling.
>Is it desirable to use new chunk types as a replacement for
>TCP-like per-packet options? There is also the very practical
>problem of fragmentation. If we want to avoid possible
>refragmentation of data chunks, we would have to at least
>reserve room for the timestamp chunk in each packet (this could
>be negotiated at connection setup). What happens if we want to
>introduce further per-packet information in the future?

With RDMAP/DDP over SCTP requiring that such a chunk be included
in *every* packet would guarantee that it was included in none.

The effective PMTU is reported to the DDP layer, which is then
responsible for segmenting DDP messages to conform with the
PMTU. It is unlikely that anyone would configure their stack
to accept a reduction in the data carrying PMTU for a protocol
that involves bulk data transfer.

But there would be frequent times when such a chunk could 
piggy-back "for free" in an SCTP packet that was ready to
ship but not full. While not as useful as time information
included in every packet, it is more likely to be provided.

I would recomend allowing the sending stack to apply its
own criteria as to when to slip one in, and then simply
have the receiving side determine whatever it can from
the data actually provided. If complete info has to be provided,
fine, you haven't lost anything. If partial info can be useful,
allowing the sender to include it on a "standby" basis could
be a better solution.

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Subject: [Tsvwg] viewgraphs for the tsvwg meeting
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I am giving two presentations at the tsvwg meeting, and my viewgraphs
are on-line, just FYI.

The talks are on:
HighSpeed TCP for Large Congestion Windows
http://www.icir.org/floyd/talks/hstcp-Jul03.pdf

and:

Moving NewReno from Experimental to Proposed Standard?
http://www.icir.org/floyd/talks/newreno-Jul03.pdf

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




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> Whenever I see a "MUST" used (or being suggested to be used) in the
> spec, I'd like to understand what would be the error handling strategy
> if this "MUST" is violated. Since you are suggesting a "MUST" clause
> here, would you immediately severe the communication with a peer that
> had agreed to use timestamp feature but you subsequently receive a
> packet from it without the timestamp chunk bundled? If your answer is
> "yes", then I'd guess that you somehow must view this missing timestamp
> as an unrecoverable protocol error which can do potential harm to your
> endpoint or the network if not stopping the communication right away.
> But this is not obvious to me, so please explain. If your answer is
> "no", then I'd question why you want to use a "MUST" here.

Hi Qiaobing,

My answer would be "no"..I did not mean the spec MUST while writing the
email :(. However I am glad that you brought it up, since it would be
interesting, and maybe relevant for timestamps, to know what you all had
thought of ECNE and CWR bundling rules.
(The spec does not mention these 2 chunks in detail).

>
> Another question: since the timestamp information (however it would be
> defined for your purposes) seems to only have local significance to the
> data sender. If so, this info can be kept opaque to the peer/receiver.
> Since the SCTP heartbeat (it has been clarified that the HB can be sent
> at different frequency or even "manually" triggered by the ULP whenever
> one is needed) has the capability to carry and return an opaque data
> blob, I don't see why you need to define a new timestamp chunk?

I remember earlier having thought of using HB's for timestamp purposes.
The problem is that while timestamp information is useful for the sender, it
does requires intelligence at the reciever. I.E. The reciever will have to
be aware  which timestamp to echo (e.x. In case a duplicate chunk, in case of
delayed SACKS, etc [please see rfc1323). And so the need for defining a
timestamp chunk.


-Sourabh


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Armando L. Caro Jr. wrote:

>>My question is how such a "mandatory bundling" is viewed in the
>>community. In some ways it seems to be a break with existing SCTP
>>"philosophy", since so far all chunk types are "self contained", and
>>there is no known case of mandatory bundling.
> 
> First, I don't see a need to make it mandatory. The sender can choose to
> bundle timestamps with all packets containing data and/or sack chunks. If
> the sender chooses not to include timestamps, then the sender must follow
> the conservative behavior anytime an ambiguous scenario arises.
> 
> Second, I think the ECNE and CWR chunks are examples of chunk types that,
> if used, need to be in every packet. The sender simply needs to reserve
> space for these chunks.

Good point, I didn't think of ECNE/CWR. Shouldn't this be in the 
implementer's guide? I mean, if an option is negotiated at startup time 
that may later take away room in packets containing data chunks, this 
must be taken into account when calculating the available room for data 
chunks (and when reporting the MULPDU to protocols such as DDP?).


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Armando L. Caro Jr. wrote:

>>Timestamps would enable us to do RTTM (RTT measurement on every packet),
>>as described in RFC 1323. Together with an appropriate timer
>>calculation, this has the potential for an RTO that more accurately
>>follows the RTT.
> 
> "Together with an appropriate timer calculation" is an important clause.

Yes :-)

>>Yes. That's why we would run into problems if we were to use new chunk
>>types as per-packet options without reserving space for them in every
>>packet, regardless whether the option is actually used in all packets or
>>only a part of them.
> 
> I am not sure I understand the problem here. I know there is some
> inefficiency with reserving space for control chunks, but since timestamps
> will be essentially in every packet... what's the problem?

Hm, ECNE/CWR chunks will not be in every packet. Has the issue of 
reserving room for those been addressed? I'm also thinking about more 
per-packet options that might be introduced in the future; should each 
of them be allowed to take away some more room?


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Caitlin Bestler wrote:

>>My question is how such a "mandatory bundling" is viewed in the
>>community. In some ways it seems to be a break with existing
>>SCTP "philosophy", since so far all chunk types are "self
>>contained", and there is no known case of mandatory bundling.
>>Is it desirable to use new chunk types as a replacement for
>>TCP-like per-packet options? There is also the very practical
>>problem of fragmentation. If we want to avoid possible
>>refragmentation of data chunks, we would have to at least
>>reserve room for the timestamp chunk in each packet (this could
>>be negotiated at connection setup). What happens if we want to
>>introduce further per-packet information in the future?
> 
> With RDMAP/DDP over SCTP requiring that such a chunk be included
> in *every* packet would guarantee that it was included in none.
> 
> The effective PMTU is reported to the DDP layer, which is then
> responsible for segmenting DDP messages to conform with the
> PMTU. It is unlikely that anyone would configure their stack
> to accept a reduction in the data carrying PMTU for a protocol
> that involves bulk data transfer.

Caitlin,

I'm not that familiar with RDDP, but my understanding is that the LLP 
(in this case SCTP) reports the MULPDU to DDP. Besides a small loss in 
efficiency due to increased overhead, is there any problem with taking 
control chunks into account when calculating the MULPDU? How is this 
handled if TCP (where options also take away from the MULPDU) is used as 
the LLP?

Stephan.


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>>> My question is how such a "mandatory bundling" is viewed in the
>>> community. In some ways it seems to be a break with existing SCTP
>>> "philosophy", since so far all chunk types are "self contained", and
>>> there is no known case of mandatory bundling.
>>
>>
>> First, I don't see a need to make it mandatory. The sender can choose to
>> bundle timestamps with all packets containing data and/or sack chunks. If
>> the sender chooses not to include timestamps, then the sender must follow
>> the conservative behavior anytime an ambiguous scenario arises.
>>
>> Second, I think the ECNE and CWR chunks are examples of chunk types that,
>> if used, need to be in every packet. The sender simply needs to reserve
>> space for these chunks.
> 
> Good point, I didn't think of ECNE/CWR. Shouldn't this be in the 
> implementer's guide? I mean, if an option is negotiated at startup time 
> that may later take away room in packets containing data chunks, this 
> must be taken into account when calculating the available room for data 
> chunks (and when reporting the MULPDU to protocols such as DDP?).

As Sourabh just pointed out to me, the ECNE/CWR chunks in fact do *not* 
have to be bundled with any other chunk, since they contain a TSN that 
acts as a pointer to the packet where the CE bit was originally set. So 
this is not a precedent for "mandatory bundling" either.

Another idea: Couldn't we use the same method for timestamps, i.e. the 
timestamp chunk would contain the TSN of a chunk in the packet that this 
timestamp is referring to?

Stephan.


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From exim@www1.ietf.org  Wed Jul 16 09:53:27 2003
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From: Pasi Sarolahti <sarolaht@cs.Helsinki.FI>
To: Stephan Baucke <stephan.baucke@ericsson.com>
cc: tsvwg@ietf.org
Subject: Re: [Tsvwg] SCTP Timestamps
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Hi,

I believe that timestamps are useful as an option for SCTP, but there 
might be cases where the user would not like to use them due to additional 
overhead (like with some telnet-kind of application protocol with small 
payloads). Just a quick note below inline, FWIW

On Tue, 15 Jul 2003, Stephan Baucke wrote:

> Timestamps would enable us to do RTTM (RTT measurement on every packet), 
> as described in RFC 1323. Together with an appropriate timer 
> calculation, this has the potential for an RTO that more accurately 
> follows the RTT.

Timestamps are not a requirement for this. If the sender stores the
transmission times of segments, it can measure RTT for every
non-retransmitted segment. This is a small overhead to the sender's
TCP/SCTP control data, but few bytes per segment should not be a problem
for modern computers. The retransmission ambiguity exists, but unless
retransmissions are very frequent, I wouldn't think it's too bad a
problem.

> Also, since timestamps resolve the retransmission 
> ambiguity, we don't have to use Karn's algorithm. Karn's algorithm 
> together with SCTP's retransmission policy has been shown to cause an 
> overly conservative RTO in certain multihomed scenarios (see 
> http://www.cis.udel.edu/~acaro/papers/2003.milcom.acaro.pdf)

[Sorry, I can't read the paper right now, but will do it as soon as 
possible]

Was overly conservative RTO due to Karn's algorithm or due to only one RTT
sample per round-trip?

Thanks,
- Pasi


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On Wed, 16 Jul 2003, Sourabh Ladha wrote:

> I remember earlier having thought of using HB's for timestamp purposes.
> The problem is that while timestamp information is useful for the sender, it
> does requires intelligence at the reciever. I.E. The reciever will have to
> be aware  which timestamp to echo (e.x. In case a duplicate chunk, in case of
> delayed SACKS, etc [please see rfc1323). And so the need for defining a
> timestamp chunk.

Two more points why HBs may not work well:

1. The sender would send HBs in almost every packet, which could make the
failure detection algorithm trickier.

2. If data transmission is bi-directional, then a packet which has SACK
and DATA chunks will need two HBs. On the other hand, a single Timestamp
chunk is needed since it has a "timestamp" and "timestamp echo" field.

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
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On Wed, 16 Jul 2003, Stephan Baucke wrote:

> Hm, ECNE/CWR chunks will not be in every packet. Has the issue of
> reserving room for those been addressed? I'm also thinking about more
> per-packet options that might be introduced in the future; should each
> of them be allowed to take away some more room?

That's a tricky situation. It is really an implementation issue, but it
probably makes sense to reserve room for chunks that must be bundled.
Others which are self contained don't have to be bundled if they don't
fit.

~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 Wed, 16 Jul 2003, Stephan Baucke wrote:

> As Sourabh just pointed out to me, the ECNE/CWR chunks in fact do *not*
> have to be bundled with any other chunk, since they contain a TSN that
> acts as a pointer to the packet where the CE bit was originally set. So
> this is not a precedent for "mandatory bundling" either.
>
> Another idea: Couldn't we use the same method for timestamps, i.e. the
> timestamp chunk would contain the TSN of a chunk in the packet that this
> timestamp is referring to?

You could, but that would make the chunk significantly larger. The TSN
info would be needed for the timestamp and timestamp echo. Also, the TSN
info would have to be a range of timestamps. So the resulting timestamp
chunk would be something like:

 ------------------
| TIMESTAMP Header |
 ------------------
|    timestamp     |
 ------------------
|    Start TSN     |
 ------------------
|    End TSN       |
 ------------------
|  timestamp echo  |
 ------------------
|    Start TSN     |
 ------------------
|    End TSN       |
 ------------------

That's a total of 28 bytes! Or, you could break them up and have a
timestamp chunk and timestamp echo chunk. But then 4 bytes are wasted in
packets that bundle both, which may happen often with bi-directional data.

~armando

0--                                                  --0
| Armando L. Caro Jr.      |  Protocol Engineering Lab |
| www.cis.udel.edu/~acaro  |    University of Delaware |
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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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Subject: Re: [Tsvwg] SCTP Timestamps
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On Wed, 16 Jul 2003, Pasi Sarolahti wrote:

> Was overly conservative RTO due to Karn's algorithm or due to only one RTT
> sample per round-trip?

Karn's algorithm (specifically in multihomed SCTP associations).

~armando

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


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Date: Wed, 16 Jul 2003 11:18:59 -0500
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] SCTP Timestamps
To: Stephan Baucke <stephan.baucke@ericsson.com>
Cc: tsvwg@ietf.org
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On Wednesday, July 16, 2003, stephan.baucke@ericsson.com wrote:

>I'm not that familiar with RDDP, but my understanding is that the LLP 
>(in this case SCTP) reports the MULPDU to DDP. Besides a small loss in 
>efficiency due to increased overhead, is there any problem with taking 
>control chunks into account when calculating the MULPDU? How is this 
>handled if TCP (where options also take away from the MULPDU) is used as 
>the LLP?

Yes, the DDP layer can account for this overhead. But it would be to
the DDP applications advantage to refuse the option rather than to
reduce its effective bandwidth. DDP applications are generally
concerned with bulk data transfer. I believe that SCTP is far superior
to TCP for DDP. But there is already an issue that at first glance
the overhead of SCTP appears larger (it isn't when you understand that
it can be shared across multiple streams). But I would not want to make
those statistics appear worse by adding "just one more option".
I would certainly *not* enable such a chunk by default (because I
wouldn't want it to be part of my benchmark statistics).

At the tail end of a given exchange, however, there will always be
a message that needs to be sent without delay. For DDP it is the
tail segment of an Untagged Message. It would be highly probable
that those packets would have room for a "standby" Data Chunk with
the additional data you desired.

I suspect this would be true for *many* applications, and you may
find stacks willing to support opportunistic low-priority sending
of Data Chunks. The only thing unique is that the Data Chunk is
being filled in at placement time, not when it is placed in the queue.

My suggestion is to allow that type of strategy to supply the data
you want on an opportunistic basis, because there will be data
transfer applications that will resist yielding any of their
bandwidth when asked to do so on *every* packet. And it would
be very easy for them to insist on a stack that at least pretended
not to know what this option was, especially since they would have
no incentive to upgrade to such an option.


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Subject: RE: [Tsvwg] SCTP Timestamps
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From: "Duke, Martin" <martin.duke@boeing.com>
To: "Armando L. Caro Jr." <acaro@acm.org>,
        "Sourabh Ladha" <ladha@mail.eecis.udel.edu>
Cc: "Qiaobing Xie" <Qiaobing.Xie@motorola.com>,
        "Stephan Baucke" <stephan.baucke@ericsson.com>, <tsvwg@ietf.org>
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> On Wed, 16 Jul 2003, Sourabh Ladha wrote:
>=20
> > I remember earlier having thought of using HB's for=20
> timestamp purposes.
> > The problem is that while timestamp information is useful=20
> for the sender, it
> > does requires intelligence at the reciever. I.E. The=20
> reciever will have to
> > be aware  which timestamp to echo (e.x. In case a duplicate=20
> chunk, in case of
> > delayed SACKS, etc [please see rfc1323). And so the need=20
> for defining a
> > timestamp chunk.

I'm not sure the concerns in RFC 1323 apply here.  There are two cases =
where the most recent=20
received timestamp isn't sent.  I would summarize them as follows:

A) To capture the delayed ACK timer in the RTT measurement, ack the 1st =
unacknowledged packet: since HEARTBEAT-ACKs have a 1:1 relationship to =
HEARTBEATs, this isn't really relevant.  If you like, one could delay =
the HB-ACK by the maximum delayed ACK time (500 ms), or add it at the =
measuring endpoint.

B) If there's a "hole" in sequence numbers, assume congestion and echo =
an old timestamp to be conservative: The value of this is arguable in =
any case, but as HBs don't have sequence numbers anyway there's no way =
for the receiver to detect a "hole."  You'll still get an accurate =
measurement of the RTT, rather than an inflated one produced in an =
effort to be "conservative."

C) DATA that is previously fully acknowledged: some argue that this is =
actually a bug in the RFC, but either way, I'm not convinced that this =
can cause a significant problem with HBs.  A HB duplicated in the =
network and long-delayed might trigger a single extremely high =
measurement, but this is a pathological case, and will hardly have =
catastrophic effects.

Then Armando wrote:
> Two more points why HBs may not work well:
>=20
> 1. The sender would send HBs in almost every packet, which=20
> could make the
> failure detection algorithm trickier.
>=20
> 2. If data transmission is bi-directional, then a packet=20
> which has SACK
> and DATA chunks will need two HBs. On the other hand, a=20
> single Timestamp
> chunk is needed since it has a "timestamp" and "timestamp echo" field.
>=20
> ~armando
>=20

1)  There will have to be a change to the treatment of HBs sent out =
bundled with DATA.  It's probably impractical to have a timer running =
for each one, and to wait 30sec (or whatever) for the timer to expire.  =
However, we want to keep these features for testing paths that aren't =
busy, so I would suggest that the timer is not run for HBs bundled with =
data; the data being sent on that path is fully adequate for testing the =
path for failure.

2) Assuming the MTU is ~1500 bytes, you're talking about a 12? Byte HB =
and 12 byte HB-ACK, vs. a 10 byte TS.  So it's a 0.7% vs. 1.6% overhead, =
whereas using HBs requires no negotiation and stack modifications only =
on the side that wishes to do the measuring.  I think this outweighs a =
few bytes of overhead.

-Martin

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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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        Qiaobing Xie <Qiaobing.Xie@motorola.com>,
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Subject: RE: [Tsvwg] SCTP Timestamps
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Martin,

You present some valid points (I especially like argument 1), but there is
still one problem. If the overloaded functionality of HBs are used without
negotiation, then the receiver will not delay HB-ACKs. Since HBs would be
bundled with each packet containing DATA chunks, the result will be
similar to that of turning off the delayed ack algorithm for SACKs.

I do like the idea of sender only changes. To avoid the above problem, the
HBs can be bundled every other packet to avoid so many HB-ACKs.

Like I said above, I like your argument 1. I was also thinking about
something like this... bundled HBs don't need timers or state, and if lost
the sender doesn't care.  Since DATA is bundled with the HBs, the data
would also be lost and that would help towards failure detection.

At the same time, HBs can still go to idle destinations the same way.
These HBs have timers and contribute towards failure detection.


All this works fine for gathering more RTT measurements for alternate
paths, but more complexity is needed to solve the problem that Stephan and
Sourabh brought up. They want a more robust method of detecting spurious
retransmits. HBs could be used, but the sender has to do more work at
putting the puzzle together. Since negotiation would not used, the
receiver may not bundle HB-ACKs with SACKs. Thus, it may not always be
possible to correlate timestamps in HB-ACKs with newly acked TSNs in
SACKs.  Also, if HBs are bundled with every other packet, then not all
spurious retransmits can be detected with equal accuracy. Some TSNs will
not have timestamps.

~armando

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

On Wed, 16 Jul 2003, Duke, Martin wrote:

> > On Wed, 16 Jul 2003, Sourabh Ladha wrote:
> >
> > > I remember earlier having thought of using HB's for
> > timestamp purposes.
> > > The problem is that while timestamp information is useful
> > for the sender, it
> > > does requires intelligence at the reciever. I.E. The
> > reciever will have to
> > > be aware  which timestamp to echo (e.x. In case a duplicate
> > chunk, in case of
> > > delayed SACKS, etc [please see rfc1323). And so the need
> > for defining a
> > > timestamp chunk.
>
> I'm not sure the concerns in RFC 1323 apply here.  There are two cases where the most recent
> received timestamp isn't sent.  I would summarize them as follows:
>
> A) To capture the delayed ACK timer in the RTT measurement, ack the 1st unacknowledged packet: since HEARTBEAT-ACKs have a 1:1 relationship to HEARTBEATs, this isn't really relevant.  If you like, one could delay the HB-ACK by the maximum delayed ACK time (500 ms), or add it at the measuring endpoint.
>
> B) If there's a "hole" in sequence numbers, assume congestion and echo an old timestamp to be conservative: The value of this is arguable in any case, but as HBs don't have sequence numbers anyway there's no way for the receiver to detect a "hole."  You'll still get an accurate measurement of the RTT, rather than an inflated one produced in an effort to be "conservative."
>
> C) DATA that is previously fully acknowledged: some argue that this is actually a bug in the RFC, but either way, I'm not convinced that this can cause a significant problem with HBs.  A HB duplicated in the network and long-delayed might trigger a single extremely high measurement, but this is a pathological case, and will hardly have catastrophic effects.
>
> Then Armando wrote:
> > Two more points why HBs may not work well:
> >
> > 1. The sender would send HBs in almost every packet, which
> > could make the
> > failure detection algorithm trickier.
> >
> > 2. If data transmission is bi-directional, then a packet
> > which has SACK
> > and DATA chunks will need two HBs. On the other hand, a
> > single Timestamp
> > chunk is needed since it has a "timestamp" and "timestamp echo" field.
> >
> > ~armando
> >
>
> 1)  There will have to be a change to the treatment of HBs sent out
> bundled with DATA.  It's probably impractical to have a timer running
> for each one, and to wait 30sec (or whatever) for the timer to expire.
> However, we want to keep these features for testing paths that aren't
> busy, so I would suggest that the timer is not run for HBs bundled
> with data; the data being sent on that path is fully adequate for
> testing the path for failure.
>
> 2) Assuming the MTU is ~1500 bytes, you're talking about a 12? Byte HB
> and 12 byte HB-ACK, vs. a 10 byte TS.  So it's a 0.7% vs. 1.6%
> overhead, whereas using HBs requires no negotiation and stack
> modifications only on the side that wishes to do the measuring.  I
> think this outweighs a few bytes of overhead.
>
> -Martin
>



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Subject: RE: [Tsvwg] SCTP Timestamps
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	Hi all!

	I have been following this topic and I have one comment or two...

> From: ext Armando L. Caro Jr. [mailto:acaro@mail.eecis.udel.edu]
> Sent: Wednesday, 16 July, 2003 20:35

...

> All this works fine for gathering more RTT measurements for alternate
> paths, but more complexity is needed to solve the problem=20
> that Stephan and
> Sourabh brought up. They want a more robust method of=20
> detecting spurious
> retransmits. HBs could be used, but the sender has to do more work at
> putting the puzzle together. Since negotiation would not used, the
> receiver may not bundle HB-ACKs with SACKs. Thus, it may not always be
> possible to correlate timestamps in HB-ACKs with newly acked TSNs in

	But... The HB-ACK itself acts as the SACK, you don't need the SACK at =
all... I'm not sure whether I'm missing anything... :-/

	You might include in the HB information the TSN, and the time when it =
was sent, as it has been already said. That uniquely identifies the =
HB-ACK when it arrives as having been produced after the reception of =
one specific DATA chunk...

	So you send your DATA+HB and wait for either the SACK or the HB-ACK. =
Any of these (or better, both in the same datagram) will tell you about =
the reception of the DATA chunk.

	I think this might be a way to solve the problem... The cost, however, =
would be 16-20 bytes per DATA chunk (depending on how many bytes takes =
the time information). In any case 4 more bytes (the ones of the =
Heartbeat Information header) than using a special chunk...

	You could always include, let's say, the last 24 bits of the TSN plus 8 =
bits (or less) indicating how many times you have already retransmitted =
this TSN, instead of including the time value. This would also uniquely =
identify the HB-ACK... Now that I think about this, a specific chunk =
might use the Chunk Flags for this, so only 8 extra bytes would be =
needed...

	Or even better, you don't need at all to include the TSN inside the HB, =
just an unique identifier, as in the IP header...

	BR Iv=E1n Arias Rodr=EDguez

> SACKs.  Also, if HBs are bundled with every other packet, then not all
> spurious retransmits can be detected with equal accuracy.=20
> Some TSNs will
> not have timestamps.
>=20
> ~armando
>=20
> 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] SCTP Timestamps
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Caitlin,

> My suggestion is to allow that type of strategy to supply the data
> you want on an opportunistic basis, because there will be data
> transfer applications that will resist yielding any of their
> bandwidth when asked to do so on *every* packet. And it would
> be very easy for them to insist on a stack that at least pretended
> not to know what this option was, especially since they would have
> no incentive to upgrade to such an option.
>

While what you say regarding overheads and bulk data transfer is indeed
true, much of the purpose of timestamps is lost if we can only send them
on an *oppurtunistic basis* (see some of the motivations listed in the
first email by Stephan). Personally, I think the best which can be done is
to disable timestamps if an app/stack feels that it can achieve better
throughput by doing so.

 -Sourabh



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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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Subject: RE: [Tsvwg] SCTP Timestamps
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Hei Ivan!!

Yes, you are right. The HB-ACK becomes like another SACK. I like these
ideas. Since the information contained in the HB is actually sender
implementation specific, each implementation can embed what ever info they
find most useful. It's up to that implementation to be effecient with
overhead tradeoffs.

I guess the only concern left is avoiding too many HB-ACKs on the wire.
I'm not sure how much Eifel detection will by affected by reducing the
number of bundled HBs.

~armando

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

On Wed, 16 Jul 2003 Ivan.Arias-Rodriguez@nokia.com wrote:

> 	Hi all!
>
> 	I have been following this topic and I have one comment or two...
>
> > From: ext Armando L. Caro Jr. [mailto:acaro@mail.eecis.udel.edu]
> > Sent: Wednesday, 16 July, 2003 20:35
>
> ...
>
> > All this works fine for gathering more RTT measurements for alternate
> > paths, but more complexity is needed to solve the problem
> > that Stephan and
> > Sourabh brought up. They want a more robust method of
> > detecting spurious
> > retransmits. HBs could be used, but the sender has to do more work at
> > putting the puzzle together. Since negotiation would not used, the
> > receiver may not bundle HB-ACKs with SACKs. Thus, it may not always be
> > possible to correlate timestamps in HB-ACKs with newly acked TSNs in
>
> 	But... The HB-ACK itself acts as the SACK, you don't need the SACK at al=
l... I'm not sure whether I'm missing anything... :-/
>
> 	You might include in the HB information the TSN, and the time when it wa=
s sent, as it has been already said. That uniquely identifies the HB-ACK wh=
en it arrives as having been produced after the reception of one specific D=
ATA chunk...
>
> 	So you send your DATA+HB and wait for either the SACK or the HB-ACK. Any=
 of these (or better, both in the same datagram) will tell you about the re=
ception of the DATA chunk.
>
> 	I think this might be a way to solve the problem... The cost, however, w=
ould be 16-20 bytes per DATA chunk (depending on how many bytes takes the t=
ime information). In any case 4 more bytes (the ones of the Heartbeat Infor=
mation header) than using a special chunk...
>
> 	You could always include, let's say, the last 24 bits of the TSN plus 8 =
bits (or less) indicating how many times you have already retransmitted thi=
s TSN, instead of including the time value. This would also uniquely identi=
fy the HB-ACK... Now that I think about this, a specific chunk might use th=
e Chunk Flags for this, so only 8 extra bytes would be needed...
>
> 	Or even better, you don't need at all to include the TSN inside the HB, =
just an unique identifier, as in the IP header...
>
> 	BR Iv=E1n Arias Rodr=EDguez
>
> > SACKs.  Also, if HBs are bundled with every other packet, then not all
> > spurious retransmits can be detected with equal accuracy.
> > Some TSNs will
> > not have timestamps.
> >
> > ~armando
> >
> > 0--                                                  --0
> > | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> > | www.cis.udel.edu/~acaro  |    University of Delaware |
> > 0--                                                  --0
>



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> -----Original Message-----
> From: ext Armando L. Caro Jr. [mailto:acaro@mail.eecis.udel.edu]
> Sent: Wednesday, 16 July, 2003 21:21
>=20
> Hei Ivan!!

	Hola Armando! See my comments below...

> Yes, you are right. The HB-ACK becomes like another SACK. I like these
> ideas. Since the information contained in the HB is actually sender
> implementation specific, each implementation can embed what=20
> ever info they
> find most useful. It's up to that implementation to be effecient with
> overhead tradeoffs.

	Exactly, but I would say that a 12 bytes HB would make it... I don't =
think you should include anything else but a 32-bit identifier of the HB =
sent...

	At the sender you can keep all the needed information in order to make =
sense out of a received HB-ACK, such as TSNs included in the datagram =
and time sent...

> I guess the only concern left is avoiding too many HB-ACKs on=20
> the wire.
> I'm not sure how much Eifel detection will by affected by reducing the
> number of bundled HBs.

	Well... You only need one HB per datagram, not per DATA chunk. But they =
are still 12 bytes you have to be ready to reserve in every packet =
sent... Or... no, maybe not...

	You are already reserve 16 bytes for the SACK chunk. So, if you don't =
need to include the SACK, let's say that the HB travels "for free".

	Moreover, I'm wondering if including such "timestamp HB" only when =
retransmitting something would be enough... This might make things more =
difficult to the data sender, since the peer could send the SACK and =
then the HB-ACK in a separate datagram, and the HB might even get lost =
and then... :-/ But probably this situation won't be worse than the =
current problem about losing a SACK...

	Also the SACK might come before the HB-ACK, so the sender should wait =
for some time before being sure that the HB-ACK won't arrive --> the =
SACK acknowledged the original datagram sent and not the retransmission =
(unless the HB-ACK was lost, of course).

	In the most common case you would be able to include a HB in every =
other datagram (when you don't include the SACK). If only one peer is =
sending data, then everything is perfect, you always have space... Of =
course, if there are TSNs lost, you might include SACKs in every =
datagram and then you won't be able to include HBs... However, in this =
case the SACK will be longer than 16 bytes, so you have the problem of =
space anyway and maybe you don't lack 12 bytes to put your HB inside the =
datagram, but have MTU-(16+12) extra bytes ... :-/

	My 0.02$

	BR Iv=E1n Arias Rodr=EDguez

> ~armando
>=20
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0

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On Wed, 16 Jul 2003 Ivan.Arias-Rodriguez@nokia.com wrote:

> > I guess the only concern left is avoiding too many HB-ACKs on
> > the wire.
> > I'm not sure how much Eifel detection will by affected by reducing the
> > number of bundled HBs.
>
> 	Well... You only need one HB per datagram, not per DATA chunk.
> But they are still 12 bytes you have to be ready to reserve in every
> packet sent... Or... no, maybe not...

Althogh packet reservation is an issue, that is not what my concern was
about. If a HB is included in each packet that contains DATA chunks, then
the receiver will send HB-ACKs for each of these packets. Thus, the point
of the delayed sack algorithm is lost.

~armando

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


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	Sourabh,

> -----Original Message-----
> From: ext Sourabh Ladha [mailto:ladha@mail.eecis.udel.edu]
> Sent: Wednesday, 16 July, 2003 21:44
>=20
> > Yes, you are right. The HB-ACK becomes like another SACK. I=20
> like these
> > ideas. Since the information contained in the HB is actually sender
> > implementation specific, each implementation can embed what=20
> ever info they
> > find most useful. It's up to that implementation to be=20
> effecient with
> > overhead tradeoffs.
>=20
> I do not agree at all if this is a "clean" way to proceed if=20
> a timestamp
> information per packet is required. The cost that you incur by
> this is a whole new definition of SACK, HB-ACK and rules=20
> associated with
> these. (For example you would still need SACKs along with=20
> HB-ACKs if an
> out of order, duplicate chunk is recieved)

	Hmm... Why? SACK will still be a SACK, and HB-ACK will still be what it =
was...

	The only thing is that you know you sent the HB together with some =
number of TSNs. If you receive the HB-ACK it means that the peer got the =
datagram, and thus he also got the DATA chunks...

	Of course you won't do anything you normally do when receiving the SACK =
before you actually do that...

	As I have said in a mail I have just sent, the problem will arise when =
the peer doesn't bundle the SACK and HB-ACK and the SACK arrives and the =
HB-ACK does not... But at the same point, not bundling the SACK and the =
HB-ACK chunks in the same datagram might even be a plus sometimes: if =
the problem wasn't a lost DATA but a lost SACK, the HB-ACK might =
"survive" and still reach the data sender...

	Probably only including the HB when retransmitting (also, in this =
situation I think you don't need to include a SACK normally and thus you =
can use its 16 bytes) and hoping the peer not sending the SACK and =
HB-ACK in different datagrams and that only the HB-ACK is lost, would be =
enough... However, if you are retransmitting, that might mean that it is =
more likely that packets get lost in the way (even in the return =
path)...

> I still think if needed an optional timestamp chunk is the=20
> best way to go
> if such an information is required.

	That might be optimal, but think that being compatible with current =
SCTP implementations is a big plus... Maybe you even want to have both =
solutions to the problem... :-/

	BR Iv=E1n Arias Rodr=EDguez

> -Sourabh
>=20
>=20

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> -----Original Message-----
> From: ext Armando L. Caro Jr. [mailto:acaro@mail.eecis.udel.edu]
> Sent: Wednesday, 16 July, 2003 21:55
>=20
> On Wed, 16 Jul 2003 Ivan.Arias-Rodriguez@nokia.com wrote:
>=20
> > > I guess the only concern left is avoiding too many HB-ACKs on
> > > the wire.
> > > I'm not sure how much Eifel detection will by affected by=20
> reducing the
> > > number of bundled HBs.
> >
> > 	Well... You only need one HB per datagram, not per DATA chunk.
> > But they are still 12 bytes you have to be ready to reserve in every
> > packet sent... Or... no, maybe not...
>=20
> Althogh packet reservation is an issue, that is not what my=20
> concern was
> about. If a HB is included in each packet that contains DATA=20
> chunks, then
> the receiver will send HB-ACKs for each of these packets.=20
> Thus, the point
> of the delayed sack algorithm is lost.

	Yep, you are right... :-(

	Do you think it would be enough just including the HB when =
retransmitting?

	BR Iv=E1n Arias Rodr=EDguez

> ~armando
>=20
> 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] SCTP Timestamps
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On Wed, 16 Jul 2003 Ivan.Arias-Rodriguez@nokia.com wrote:

> 	Do you think it would be enough just including the HB when
> retransmitting?

Since there is no guarantee that the HB-ACK will be bundled with a SACK,
it may get ambiguious for the sender. When a SACK arrives without a
HB-ACK, the sender doesn't know if the SACK is (1) for the original
transmission, (2) for a retransmission but the HB-ACK is on the way, or
(3) for a retransmission but the HB-ACK was lost.

~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] SCTP Timestamps
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On Wed, 16 Jul 2003, Sourabh Ladha wrote:

> I do not agree at all if this is a "clean" way to proceed if a timestamp
> information per packet is required. The cost that you incur by
> this is a whole new definition of SACK, HB-ACK and rules associated with
> these. (For example you would still need SACKs along with HB-ACKs if an
> out of order, duplicate chunk is recieved)
>
> I still think if needed an optional timestamp chunk is the best way to go
> if such an information is required.

Like I said, using HBs is good enough for gathering more RTT measurements
on alternate paths, but _may_ not be good enough for Eifel detection. I
think that using HBs would be almost as good as the timestamp chunk if we
could figure out how to solve the HB-ACKing every packet problem. I do
agree that an optional timestamp chunk is the best way, because the
timestamp is guaranteed to be bundled. However, if we can get most of the
gains with just the HBs, that would be a big plus. It's always nice when
only one end has to be modified to get the improvment.

~armando

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



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> -----Original Message-----
> From: ext Armando L. Caro Jr. [mailto:acaro@mail.eecis.udel.edu]
> Sent: Wednesday, 16 July, 2003 22:22
>=20
> On Wed, 16 Jul 2003 Ivan.Arias-Rodriguez@nokia.com wrote:
>=20
> > 	Do you think it would be enough just including the HB when
> > retransmitting?
>=20
> Since there is no guarantee that the HB-ACK will be bundled=20
> with a SACK,
> it may get ambiguious for the sender. When a SACK arrives without a
> HB-ACK, the sender doesn't know if the SACK is (1) for the original
> transmission, (2) for a retransmission but the HB-ACK is on=20
> the way, or
> (3) for a retransmission but the HB-ACK was lost.

	Yes... The point would be that when being in (2) you should wait for a =
while before you consider that you are either in (3) or (1).

	Since you must bundle the control chunk (i.e., HB) before the DATA =
ones, (2) might not be that common because the HB-ACK should be sent =
before the SACK (or maybe some implementation will not)... :-/

	BR Iv=E1n Arias Rodr=EDguez

> ~armando
>=20
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0
>=20
>=20

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> Yes, you are right. The HB-ACK becomes like another SACK. I like these
> ideas. Since the information contained in the HB is actually sender
> implementation specific, each implementation can embed what ever info they
> find most useful. It's up to that implementation to be effecient with
> overhead tradeoffs.

I do not agree at all if this is a "clean" way to proceed if a timestamp
information per packet is required. The cost that you incur by
this is a whole new definition of SACK, HB-ACK and rules associated with
these. (For example you would still need SACKs along with HB-ACKs if an
out of order, duplicate chunk is recieved)

I still think if needed an optional timestamp chunk is the best way to go
if such an information is required.


-Sourabh


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Date: Wed, 16 Jul 2003 15:13:36 -0500
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] SCTP Timestamps
To: Sourabh Ladha <ladha@mail.eecis.udel.edu>
Cc: Stephan Baucke <stephan.baucke@ericsson.com>, tsvwg@ietf.org
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On Wednesday, July 16, 2003, ladha@mail.eecis.udel.edu wrote:

>Caitlin,
>
>> My suggestion is to allow that type of strategy to supply the data
>> you want on an opportunistic basis, because there will be data
>> transfer applications that will resist yielding any of their
>> bandwidth when asked to do so on *every* packet. And it would
>> be very easy for them to insist on a stack that at least pretended
>> not to know what this option was, especially since they would have
>> no incentive to upgrade to such an option.
>>
>
>While what you say regarding overheads and bulk data transfer
>is indeed true, much of the purpose of timestamps is lost if we
>can only send them on an *oppurtunistic basis* (see some of the
>motivations listed in the first email by Stephan). Personally,
>I think the best which can be done is to disable timestamps if
>an app/stack feels that it can achieve better throughput by
>doing so.
>

There are some assumptions being made here that I am not following.

If I am following the thread, the purpose of timestamping is to
allow frequent ongoing measurement of the RTT along each path.
Further, knowing the current RTT for each path, it will be
possible to disambiguate certain Acks. This leads to better
decisions on when to shift paths and/or do a fast retransmit.

Is that correct?

If so, why is timestamp data required in every packet?

MPEG transport maintains a distributed clock over unreliable
jittered networks even in the face of frequent packet drops.
I am not following why a timestamp must *always* be present
to allow measurements to be made accurately.

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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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        Stephan Baucke <stephan.baucke@ericsson.com>, <tsvwg@ietf.org>
Subject: Re: [Tsvwg] SCTP Timestamps
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Caitlin,

There are actually 2 motivations for timestamps.

The original motivation was to be get rid of Karn's algorithm and be to
use retransmissions for RTT estimation. This purpose does not need
timestamps in every single packet, but the more the better.

The second motivation was to use timestamps for Eifel detection. A
spurious timeout retransmit can be detected if the retransmitted TSN gets
acked with a timestamp earlier than the timestamp bundled with the
retransmission. This purpose does require a timestamp in each packet (I
think).

~armando

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

On Wed, 16 Jul 2003, Caitlin Bestler wrote:

> There are some assumptions being made here that I am not following.
>
> If I am following the thread, the purpose of timestamping is to
> allow frequent ongoing measurement of the RTT along each path.
> Further, knowing the current RTT for each path, it will be
> possible to disambiguate certain Acks. This leads to better
> decisions on when to shift paths and/or do a fast retransmit.
>
> Is that correct?
>
> If so, why is timestamp data required in every packet?
>
> MPEG transport maintains a distributed clock over unreliable
> jittered networks even in the face of frequent packet drops.
> I am not following why a timestamp must *always* be present
> to allow measurements to be made accurately.



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> There are actually 2 motivations for timestamps.
>
> The original motivation was to be get rid of Karn's algorithm and be to
> use retransmissions for RTT estimation. This purpose does not need
> timestamps in every single packet, but the more the better.
>
> The second motivation was to use timestamps for Eifel detection. A
> spurious timeout retransmit can be detected if the retransmitted TSN gets
> acked with a timestamp earlier than the timestamp bundled with the
> retransmission. This purpose does require a timestamp in each packet (I
> think).
>


Another motivation for having timestamp per packet or maybe just for
retransmissions (either way may require you to have room reserved for
timestamp chunk in every packet) was to allow correct crediting of cwnd
(i.e. in particular avoid congestion window overgrowth) by eliminating
"retransmission ambuiguity".


-Sourabh




>
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0
>
> On Wed, 16 Jul 2003, Caitlin Bestler wrote:
>
> > There are some assumptions being made here that I am not following.
> >
> > If I am following the thread, the purpose of timestamping is to
> > allow frequent ongoing measurement of the RTT along each path.
> > Further, knowing the current RTT for each path, it will be
> > possible to disambiguate certain Acks. This leads to better
> > decisions on when to shift paths and/or do a fast retransmit.
> >
> > Is that correct?
> >
> > If so, why is timestamp data required in every packet?
> >
> > MPEG transport maintains a distributed clock over unreliable
> > jittered networks even in the face of frequent packet drops.
> > I am not following why a timestamp must *always* be present
> > to allow measurements to be made accurately.
>
>
>


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Subject: RE: [Tsvwg] SCTP Timestamps
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> From: Armando L. Caro Jr. [mailto:acaro@mail.eecis.udel.edu]
>=20
> Like I said, using HBs is good enough for gathering more RTT=20
> measurements
> on alternate paths, but _may_ not be good enough for Eifel=20
> detection. I
> think that using HBs would be almost as good as the timestamp=20
> chunk if we
> could figure out how to solve the HB-ACKing every packet problem. I do
> agree that an optional timestamp chunk is the best way, because the
> timestamp is guaranteed to be bundled. However, if we can get=20
> most of the
> gains with just the HBs, that would be a big plus. It's=20
> always nice when
> only one end has to be modified to get the improvment.
>=20
> ~armando

We seem to be trying to solve multiple problems with one fix, although =
their needs diverge somewhat.

1) More frequent RTT measurements (more often than 1/cwnd): obviously =
does not require every packet to have a HB, as Caitlin points out.

2) Allow RTT measurements on retransmissions: HB only required on =
retransmissions.

3) Detect spurious timeouts by specifying if original or retransmission =
was ACKed:  There's no need to be perfect here.  First, on multihomed =
connections the ACK will normally come along the same path as the DATA, =
allowing some differentiation.  Second, only initial transmissions =
require HB; if a HB-ACK comes back for it, you can restore the initial =
cwnd.  If you don't like a HB-ACK after every packet, there's a =
tradeoff: more packet density or less assurance of Eifel detection?  It =
would be an interesting study to see the dynamics of this tradeoff in =
different loss or delay environments.

I think HBs are superior to a new TS option for several reasons:

A) A cooperative peer will wait long enough to bundle its HB-ACKs with =
SACK.  An uncooperative peer will flood the network with separate =
HB-ACKs.  With timestamps, a cooperative peer will behave as with HBs, =
but an uncooperative peer will do nothing.

B) If the peer turns out to flood the network, the sender can even =
adjust the frequency of HBs to compensate.

C) 100% reliable detection of spurious timeouts is unnecessary.

D) Changes to the stack (limited to one side) are quite minimal: =
eliminate the HB timer for timestamp-related heartbeats, store the =
mapping of TSNs to timestamps, possibly do some dynamic rate control, =
possibly add 500ms to RTT measurements to take delayed ACKs into =
account.  And they're entirely borne by the measuring endpoint.  The =
peer has no choice but to comply if they are conforming to the =
specification.

That said, messing with SACK to avoid duplication with HB-ACK is a bad =
idea in my opinion.  It requires changes in the peer implementation and =
may open up a whole range of unforeseen consequences.

Martin

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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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Subject: RE: [Tsvwg] SCTP Timestamps
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On Wed, 16 Jul 2003, Duke, Martin wrote:

> We seem to be trying to solve multiple problems with one fix, although
> their needs diverge somewhat.
>
> 1) More frequent RTT measurements (more often than 1/cwnd): obviously
> does not require every packet to have a HB, as Caitlin points out.

This is not a problem in my opinion. More frequent RTT measurements has
not been shown to be necessary or even useful.

> 2) Allow RTT measurements on retransmissions: HB only required on
> retransmissions.

As I mentioned in a reply to Ivan... Unless HBs are bundled with SACKs,
this technique doesn't work without heuristics. To make sure the peer
bundles HBs with SACKs, you might as well use Timestamp chunks.

> 3) Detect spurious timeouts by specifying if original or
> retransmission was ACKed:  There's no need to be perfect here.

Nobody is talking about being perfect, but we would like to have _some_
accuracy.

> First, on multihomed connections the ACK will normally come along the
> same path as the DATA, allowing some differentiation.

No, you don't know that. SACKs are sent to the IP address from which data
was sent, which means that the receiver may use a different return path.
Imagine the following:

       +- A1                     B1 -+
Host A |                             | Host B
       +- A2                     B2 -+

Data may be sent from A1 to B1, but sacks may be sent B2 to A1. That
follows the specs rules, but doesn't have the same return path.

> Second, only initial transmissions require HB; if a HB-ACK comes back
> for it, you can restore the initial cwnd.

Maybe it doesn't matter, but what if the first  retransmission spuriously
times out.

> If you don't like a HB-ACK after every packet, there's a tradeoff:
> more packet density or less assurance of Eifel detection?

Well, sending HBs with every other packet brings down the detection to at
most 50%. Not very good. What is good enough? 90%? So send HBs with 9 of
10 packets. I don't think I want to go down that road.

> I think HBs are superior to a new TS option for several reasons:
>
> A) A cooperative peer will wait long enough to bundle its HB-ACKs with
> SACK. An uncooperative peer will flood the network with separate
> HB-ACKs.  With timestamps, a cooperative peer will behave as with HBs,
> but an uncooperative peer will do nothing.

How can a peer be cooperative without negotiation? And if there is
negotiation, then why not use timestamps? (I assume we all agree that if
negotiation is used, then timestamps are the best way to go.)

> B) If the peer turns out to flood the network, the sender can even
> adjust the frequency of HBs to compensate.

How does the sender know when the peer is flooding the network?

> C) 100% reliable detection of spurious timeouts is unnecessary.

Maybe not, but then what is necessary? The purpose of timestamps is to
move away from heuristics for robustness. But the HB idea seems to add
more heuristics.

> D) Changes to the stack (limited to one side) are quite minimal:
> eliminate the HB timer for timestamp-related heartbeats, store the
> mapping of TSNs to timestamps, possibly do some dynamic rate control,
> possibly add 500ms to RTT measurements to take delayed ACKs into
> account.  And they're entirely borne by the measuring endpoint.  The
> peer has no choice but to comply if they are conforming to the
> specification.

All these changes are fine, but I don't feel that it adds robustness to
Eifel detection. The only way I can see to add robustness is to bundle
timestamp echos with sacks... doesn't matter if they are HB-ACKs or
Timestamp chunks.

> That said, messing with SACK to avoid duplication with HB-ACK is a bad
> idea in my opinion.  It requires changes in the peer implementation
> and may open up a whole range of unforeseen consequences.

Hmmm... I don't follow this point. Are you arguing against your previous
arguments? HB-ACK seems to be morphing into SACK with a timestamp. Is it
not?

~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 exim@www1.ietf.org  Wed Jul 16 18:11:26 2003
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From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] SCTP Timestamps
To: "Armando L. Caro Jr." <acaro@acm.org>
Cc: Sourabh Ladha <ladha@mail.eecis.udel.edu>,
        Stephan Baucke <stephan.baucke@ericsson.com>, tsvwg@ietf.org
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On Wednesday, July 16, 2003, acaro@mail.eecis.udel.edu wrote:

>The second motivation was to use timestamps for Eifel
>detection. A spurious timeout retransmit can be detected if the
>retransmitted TSN gets acked with a timestamp earlier than the
>timestamp bundled with the retransmission. This purpose does
>require a timestamp in each packet (I think).

It is still possible to infer a possible timestamp range for
when a packet was sent by interpolation from earlier and later
packets, particularly if the reception time of each packet is
captured for use by SCTP.

This is especially true for sustained transmits of large
messages that span multiple SCTP packets (which is precisely
when it would be valuable to omit the timing data). If we
assume that for the duration of a "sustained transmit" that
the sender will be paced by the congestion and buffer windows
then the timing of each packet becomes very predictable on
a statistical basis.

These aren't necessarily easy algorithms to work out, but they
can be *implemented* cheaply. Video on Demand projects had very
simple firmware algorithms to recover an MPEG Transport clock
even when transmitted over a heavily jittered ATM network. The
recovery algorithm could even be simple enough to be done in
hardware or 8-bit processors.

I am not certain that this is worthwhile, but the assumption
a packet without a timestamp was a complete mystery kind of
leapt out at me. MPEG Transport systems survive quite well
on periodic timestamps, and they have much more precise
timing requirements. Recovering an imputed clock with
sufficient resolution to differentiate between two possible
acks of the same TSN should be quite doable.
 

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        martin.duke@boeing.com, stephan.baucke@ericsson.com, tsvwg@ietf.org
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With a new optional chunk, you still need to pay the cost for adding
negotiation at init, new procedures, changes to the receiver for using
this feature. This IMO would cost even more.

-Qiaobing

Sourabh Ladha wrote:
> 
> > Yes, you are right. The HB-ACK becomes like another SACK. I like these
> > ideas. Since the information contained in the HB is actually sender
> > implementation specific, each implementation can embed what ever info they
> > find most useful. It's up to that implementation to be effecient with
> > overhead tradeoffs.
> 
> I do not agree at all if this is a "clean" way to proceed if a timestamp
> information per packet is required. The cost that you incur by
> this is a whole new definition of SACK, HB-ACK and rules associated with
> these. (For example you would still need SACKs along with HB-ACKs if an
> out of order, duplicate chunk is recieved)
> 
> I still think if needed an optional timestamp chunk is the best way to go
> if such an information is required.
> 
> -Sourabh

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Armando,

Would the alternative approach (i.e., introducing a new timestamp and
echo chunk) change this situation? I seem to see no difference. Maybe I
missed something.

-Qiaobing

"Armando L. Caro Jr." wrote:
> 
> On Wed, 16 Jul 2003 Ivan.Arias-Rodriguez@nokia.com wrote:
> 
> > > I guess the only concern left is avoiding too many HB-ACKs on
> > > the wire.
> > > I'm not sure how much Eifel detection will by affected by reducing the
> > > number of bundled HBs.
> >
> >       Well... You only need one HB per datagram, not per DATA chunk.
> > But they are still 12 bytes you have to be ready to reserve in every
> > packet sent... Or... no, maybe not...
> 
> Althogh packet reservation is an issue, that is not what my concern was
> about. If a HB is included in each packet that contains DATA chunks, then
> the receiver will send HB-ACKs for each of these packets. Thus, the point
> of the delayed sack algorithm is lost.
> 
> ~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: Janardhan Iyengar <iyengar@mail.eecis.udel.edu>
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To: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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        "Armando L. Caro Jr." <acaro@acm.org>,
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Subject: Re: [Tsvwg] SCTP Timestamps
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Hi all,

I just got through the entire thread; an attempt to consolidate broadly
the solutions being discussed - there seem to be two possibilities:

1. Negotiation during association setup.
Upside: Timestamps option can be negotiated.
Downside: Receiver knowledge required; new chunk types required (TS,
          TS-ECHO).

2. No negotiation during setup.
Upside: Sender-side mods only; use existing chunk types (HB, HB-ACK).
Downside: Possible increased traffic on return path; heuristics required
	  to detect spurious retransmits.

As far as either of these solutions is concerned, the overhead of
reserving space in datagrams for either Timestamps or HBs exists; the
sender has to take into consideration space needed for either a TS or HB
chunk when fragmenting user data. IMHO, the difference in bit overhead
between the two options is trivial. In fact, if negotiation is done, the
sender can fragment data accordingly, i.e, reserve space or not in the
datagram.

Given that space is available in the datagram, the sender has to figure
out reliable mechanisms of using HB-ACK information to reliably infer
spurious retransmits. Since the receiver's response is not known for
certain, the sender may have to use heuristics to infer this information.

I'm not certain why a TS option was not proposed, given that it exists for
TCP (maybe because of packetization problems?). This point biases my view,
because it seems to me that a TS chunk can have uses beyond the immediate
Eifel algorithm on hand. (I may be wrong here; but not having a TS option
seems like removing an option that has existed with TCP; I do not see a
replacement in SCTP). Further, if a TS chunk is used for Eifel, the sender
_may_ be certain about the response from the receiver, _possibly_ free
from ambiguity.

I am leaning towards specifying a TS option; I do not think that option
negotiation at setup should become a bottleneck. At the same time, Ivan's
comment also seems appropriate that it is a good idea to have an inference
mechanism in case the receiver does not recognize the TS option.  So,
maybe two algorithms should be considered.

In case of purely sender-side mods, I have a thought in mind:

On detection of a loss through an RTO, the sender sends a HB bundled with
the next three datagrams that are sent. These HBs can carry whatever
information is required by the sender to disambiguate the HB-ACKs from one
another, and other timing information as well. If the sender receives
alternating SACKs and HB-ACKs or SACKs bundled with HBs, the sender should
be able to infer that the SACKs received were for rtxs. Three HBs (or half
cwnd) may be good enough in the face of loss on the return path. Of
course, this is another heuristic.

Caitlin's point about Video on Demand servers sounds interesting, but I
have a feeling that the solution needs more accurate time measurements in
the kernel. I'm in no capacity to comment on the complexity involved here.
Maybe somebody more capable could comment?

Comments? (I apologize for the lack of brevity)

regards,
jana

On Wed, 16 Jul 2003, Qiaobing Xie wrote:

> 7bit
> MIME-Version: 1.0
> Content-Type: text/plain; charset="us-ascii"
> Content-Transfer-Encoding: 7bit
> 7bit
>
> With a new optional chunk, you still need to pay the cost for adding
> negotiation at init, new procedures, changes to the receiver for using
> this feature. This IMO would cost even more.
>
> -Qiaobing
>
> Sourabh Ladha wrote:
> >
> > > Yes, you are right. The HB-ACK becomes like another SACK. I like these
> > > ideas. Since the information contained in the HB is actually sender
> > > implementation specific, each implementation can embed what ever info they
> > > find most useful. It's up to that implementation to be effecient with
> > > overhead tradeoffs.
> >
> > I do not agree at all if this is a "clean" way to proceed if a timestamp
> > information per packet is required. The cost that you incur by
> > this is a whole new definition of SACK, HB-ACK and rules associated with
> > these. (For example you would still need SACKs along with HB-ACKs if an
> > out of order, duplicate chunk is recieved)
> >
> > I still think if needed an optional timestamp chunk is the best way to go
> > if such an information is required.
> >
> > -Sourabh
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>

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





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Subject: Re: [Tsvwg] SCTP Timestamps
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Timestamps, in of itself, doesn't make a difference. The difference is
when the receiver is aware of what the sender is trying to do. If the
sender is aware of the timestamps (or HBs), then the receiver can be sure
to bundle the timestamp echos only with SACKs.

~armando

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

On Wed, 16 Jul 2003, Qiaobing Xie wrote:

> Armando,
>
> Would the alternative approach (i.e., introducing a new timestamp and
> echo chunk) change this situation? I seem to see no difference. Maybe I
> missed something.
>
> -Qiaobing
>
> "Armando L. Caro Jr." wrote:
> >
> > On Wed, 16 Jul 2003 Ivan.Arias-Rodriguez@nokia.com wrote:
> >
> > > > I guess the only concern left is avoiding too many HB-ACKs on
> > > > the wire.
> > > > I'm not sure how much Eifel detection will by affected by reducing the
> > > > number of bundled HBs.
> > >
> > >       Well... You only need one HB per datagram, not per DATA chunk.
> > > But they are still 12 bytes you have to be ready to reserve in every
> > > packet sent... Or... no, maybe not...
> >
> > Althogh packet reservation is an issue, that is not what my concern was
> > about. If a HB is included in each packet that contains DATA chunks, then
> > the receiver will send HB-ACKs for each of these packets. Thus, the point
> > of the delayed sack algorithm is lost.
> >
> > ~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: Wed, 16 Jul 2003 21:28:46 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
Cc: Qiaobing Xie <Qiaobing.Xie@motorola.com>, Ivan.Arias-Rodriguez@nokia.com,
        martin.duke@boeing.com, ladha@mail.eecis.udel.edu,
        stephan.baucke@ericsson.com, tsvwg@ietf.org
Subject: Re: [Tsvwg] SCTP Timestamps
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Armando,

Stange.  The DATA chunk might be difficult to extend without defining a
new EDATA chunk that contains a TS or rather complicated bundling rules f=
or
bundling a some chunk containing TS with sent data, but the SACK chunk is
easy to extend.  There is no reason why any number of TLVs couldn't be ta=
cked
onto the end of a SACK.

So why aggravate fragmentation and bundling on the way back?  Just tack a
TS TLV on the end of the SACK.  Didon't someone already define a TS TLV
(parameter not chunk).

Just a thought.

--brian

On Wed, 16 Jul 2003, Armando L. Caro Jr. wrote:

> Timestamps, in of itself, doesn't make a difference. The difference is
> when the receiver is aware of what the sender is trying to do. If the
> sender is aware of the timestamps (or HBs), then the receiver can be su=
re
> to bundle the timestamp echos only with SACKs.
>=20
> ~armando
>=20
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0
>=20
> On Wed, 16 Jul 2003, Qiaobing Xie wrote:
>=20
> > Armando,
> >
> > Would the alternative approach (i.e., introducing a new timestamp and
> > echo chunk) change this situation? I seem to see no difference. Maybe=
 I
> > missed something.
> >
> > -Qiaobing
> >
> > "Armando L. Caro Jr." wrote:
> > >
> > > On Wed, 16 Jul 2003 Ivan.Arias-Rodriguez@nokia.com wrote:
> > >
> > > > > I guess the only concern left is avoiding too many HB-ACKs on
> > > > > the wire.
> > > > > I'm not sure how much Eifel detection will by affected by reduc=
ing the
> > > > > number of bundled HBs.
> > > >
> > > >       Well... You only need one HB per datagram, not per DATA chu=
nk.
> > > > But they are still 12 bytes you have to be ready to reserve in ev=
ery
> > > > packet sent... Or... no, maybe not...
> > >
> > > Althogh packet reservation is an issue, that is not what my concern=
 was
> > > about. If a HB is included in each packet that contains DATA chunks=
, then
> > > the receiver will send HB-ACKs for each of these packets. Thus, the=
 point
> > > of the delayed sack algorithm is lost.
> > >
> > > ~armando
> > >
> > > 0--                                                  --0
> > > | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> > > | www.cis.udel.edu/~acaro  |    University of Delaware |
> > > 0--                                                  --0
> >
>=20
>=20
>=20
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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

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Date: Wed, 16 Jul 2003 21:38:05 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Sourabh Ladha <ladha@mail.eecis.udel.edu>
Cc: "Armando L. Caro Jr." <acaro@acm.org>, Ivan.Arias-Rodriguez@nokia.com,
        martin.duke@boeing.com, Qiaobing.Xie@motorola.com,
        stephan.baucke@ericsson.com, tsvwg@ietf.org
Subject: Re: [Tsvwg] SCTP Timestamps
Message-ID: <20030716213805.A10144@openss7.org>
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Sourabh,

I agree that HEARTBEAT/ACK is not a good way to do this.  Although the
opaque data in the chunk looks attractive, there are rather narrow rules
for the handling of HB/ACK that have a rather different purpose.  SCTP
relies on the HB mechanism for recovery of unavailable destinations in
the multihomed environment and is also important for detection of lost
ABORTs.  There are some items concerning HB that was discovered in intero=
p
testing that showed that rather special rules were required to accomodate
implementations that could not (or did not) control source address vs. th=
ose
that did.

I think that by overloading HB/ACK for the purpose of doing more frequent
RTT measurements would break the original purpose of HB/ACK: testing the
availability of desintations, keeping RTO current on idle destinations, a=
nd
detecting lost ABORT chunks.

I too would prefer to see a different chunk, (TS chunk) bundled with all
outgoing packets containing DATA and use an extension of the SACK chunk t=
o=20
return pertinent timestamps.  (SACK is the right place to put it on the
return anyways.)

--brian

On Wed, 16 Jul 2003, Sourabh Ladha wrote:

>=20
> > Yes, you are right. The HB-ACK becomes like another SACK. I like thes=
e
> > ideas. Since the information contained in the HB is actually sender
> > implementation specific, each implementation can embed what ever info=
 they
> > find most useful. It's up to that implementation to be effecient with
> > overhead tradeoffs.
>=20
> I do not agree at all if this is a "clean" way to proceed if a timestam=
p
> information per packet is required. The cost that you incur by
> this is a whole new definition of SACK, HB-ACK and rules associated wit=
h
> these. (For example you would still need SACKs along with HB-ACKs if an
> out of order, duplicate chunk is recieved)
>=20
> I still think if needed an optional timestamp chunk is the best way to =
go
> if such an information is required.
>=20
>=20
> -Sourabh
>=20
>=20
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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

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Subject: Re: [Tsvwg] SCTP Timestamps
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On Wed, 16 Jul 2003, Brian F. G. Bidulock wrote:

> Armando,
>
> Stange.  The DATA chunk might be difficult to extend without defining a
> new EDATA chunk that contains a TS or rather complicated bundling rules for
> bundling a some chunk containing TS with sent data, but the SACK chunk is
> easy to extend.  There is no reason why any number of TLVs couldn't be tacked
> onto the end of a SACK.
>
> So why aggravate fragmentation and bundling on the way back?  Just tack a
> TS TLV on the end of the SACK.  Didon't someone already define a TS TLV
> (parameter not chunk).

Is that legal? Also, how does it alleviate frag/bundling issues?

~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: Wed, 16 Jul 2003 22:23:58 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
Cc: Qiaobing Xie <Qiaobing.Xie@motorola.com>, Ivan.Arias-Rodriguez@nokia.com,
        martin.duke@boeing.com, ladha@mail.eecis.udel.edu,
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Subject: Re: [Tsvwg] SCTP Timestamps
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Armando,

On Wed, 16 Jul 2003, Armando L. Caro Jr. wrote:

> On Wed, 16 Jul 2003, Brian F. G. Bidulock wrote:
>=20
> > Armando,
> >
> > Stange.  The DATA chunk might be difficult to extend without defining=
 a
> > new EDATA chunk that contains a TS or rather complicated bundling rul=
es for
> > bundling a some chunk containing TS with sent data, but the SACK chun=
k is
> > easy to extend.  There is no reason why any number of TLVs couldn't b=
e tacked
> > onto the end of a SACK.
> >
> > So why aggravate fragmentation and bundling on the way back?  Just ta=
ck a
> > TS TLV on the end of the SACK.  Didon't someone already define a TS T=
LV
> > (parameter not chunk).
>=20
> Is that legal? Also, how does it alleviate frag/bundling issues?

No, no, I didn't mean that it was legal now.  I meant that it is quite ea=
sy to
extend the SACK chunk because it has an internal length (number of gaps,
number of dups) that indicate the length of the content of the existing c=
hunk.
Anything after the last dup TSN could be pretty much ignored.

Certainly an extension document would be required (one would anyway) and =
a
negotiation procedure for backwards compatibility.

The frag bundling it solves is what to do, for example, when a TS is bund=
led
with the wrong SACK.  Also, nowhere in 2960 does it say that two SACKs ca=
n't
be bundled in the same packet, although it is probably not done.  Having =
the
value in the SACK inexhorably binds it to the SACK to which it correspond=
s,
extinguishing all issues as to where what is in what message with respect=
 to
the SACK.

It was just a thought.

--brian

>=20
> ~armando
>=20
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0

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

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From: Pasi Sarolahti <sarolaht@cs.Helsinki.FI>
To: "Armando L. Caro Jr." <acaro@acm.org>
cc: tsvwg@ietf.org
Subject: Re: [Tsvwg] SCTP Timestamps
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Hi,

On Wed, 16 Jul 2003, Armando L. Caro Jr. wrote:

> The second motivation was to use timestamps for Eifel detection. A
> spurious timeout retransmit can be detected if the retransmitted TSN gets
> acked with a timestamp earlier than the timestamp bundled with the
> retransmission. This purpose does require a timestamp in each packet (I
> think).

Aside from this timestamp discussion, but since spurious RTOs were 
mentioned: We have proposed a scheme for detecting spurious RTOs without 
timestamps in draft-sarolahti-tsvwg-tcp-frto-04.txt. I solicited for 
comments on it in early June, but unfortunately got no answer. It would be 
nice to hear whether the SCTP community would find that useful.

The idea in short is that after an RTO the sender would try to send some
new data and check if there are acks of non-retransmitted segments
arriving after that, in which case the RTO is considered to be spurious.
This should work with SCTP as well as with TCP as far as I can see. I
believe Markku Kojo will have a short presentation about the draft in the
tsvwg meeting.

Thanks,
- Pasi



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Date: Thu, 17 Jul 2003 08:27:55 -0400 (EDT)
Subject: Re: [Tsvwg] SCTP Timestamps
To: "Pasi Sarolahti" <sarolaht@cs.Helsinki.FI>
Cc: "Armando L. Caro Jr." <acaro@acm.org>, tsvwg@ietf.org
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Pasi,

I had read the FRTO document some time back. On the SCTP front I was
wondering if the following are applicable:

* Multihoming: If the delay spike on the primary path is greater than (1
RTO(primary) + 2 RTT(alternate)), I would believe that FRTO will not be
able to detect a timeout spurious. (complying with the current spec that
retransmissions SHOULD go to alternate destination)

* Chunk Bundling: FRTO assumes that on a timeout the sender will
retransmit only one segment("I think chunk will be appropriate in SCTP
terminology"). This may not be true in SCTP. If the association is
multihomed and the PMTU's on the primary and alternate path differ by a
factor of two or more then, then I think the sender may be able to
retransmit more than one unacknowledged chunk on a timeout (bundled in a
single packet). This can also be true if the app is writing small chunks
such that on a timeout more than one of such chunks can be bundled in a
single packet and retransmitted. I remember having noticed a couple of
side effects of this which could lead FRTO to declare the timeout spurious
while it was not.

Comments?
(Others shoot in if I missed something)

A question I have is:
Section 3- step 3(b): When you refer to an ACK which acknowledges data
(above spurious threshhold or previously unacknowledged), do you refer to
acknowledging {via cum-ack point} or {via SACK blocks} or {either of these
two}?


-Sourabh




> -----Original Message-----
> From: Pasi Sarolahti [mailto:sarolaht@cs.Helsinki.FI]
> Sent: Thursday, July 17, 2003 12:52 PM
> To: Armando L. Caro Jr.
> Cc: tsvwg@ietf.org
> Subject: Re: [Tsvwg] SCTP Timestamps
>
>
> Hi,
>
> On Wed, 16 Jul 2003, Armando L. Caro Jr. wrote:
>
> > The second motivation was to use timestamps for Eifel detection. A
> > spurious timeout retransmit can be detected if the
> retransmitted TSN gets
> > acked with a timestamp earlier than the timestamp bundled with the
> > retransmission. This purpose does require a timestamp in
> each packet (I
> > think).
>
> Aside from this timestamp discussion, but since spurious RTOs were
> mentioned: We have proposed a scheme for detecting spurious
> RTOs without
> timestamps in draft-sarolahti-tsvwg-tcp-frto-04.txt. I solicited for
> comments on it in early June, but unfortunately got no
> answer. It would be
> nice to hear whether the SCTP community would find that useful.
>
> The idea in short is that after an RTO the sender would try
> to send some
> new data and check if there are acks of non-retransmitted segments
> arriving after that, in which case the RTO is considered to
> be spurious.
> This should work with SCTP as well as with TCP as far as I can see. I
> believe Markku Kojo will have a short presentation about the
> draft in the
> tsvwg meeting.
>
> Thanks,
> - Pasi
>
>
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>


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Armando L. Caro Jr. wrote:

>>Another idea: Couldn't we use the same method for timestamps, i.e. the
>>timestamp chunk would contain the TSN of a chunk in the packet that this
>>timestamp is referring to?
> 
> You could, but that would make the chunk significantly larger. The TSN
> info would be needed for the timestamp and timestamp echo. Also, the TSN
> info would have to be a range of timestamps. So the resulting timestamp
> chunk would be something like:

You could probably get away with just one TSN, analog to the ECNW/CWR 
chunks. But I admit that this solution isn't ideal anyway for Eifel or 
the disambiguating of retransmissions; e.g. it gets difficult if the 
timestamp chunk is sent separately and gets lost or reordered in 
relation to the corresponding data packet.

Stephan.


>  ------------------
> | TIMESTAMP Header |
>  ------------------
> |    timestamp     |
>  ------------------
> |    Start TSN     |
>  ------------------
> |    End TSN       |
>  ------------------
> |  timestamp echo  |
>  ------------------
> |    Start TSN     |
>  ------------------
> |    End TSN       |
>  ------------------
> 
> That's a total of 28 bytes! Or, you could break them up and have a
> timestamp chunk and timestamp echo chunk. But then 4 bytes are wasted in
> packets that bundle both, which may happen often with bi-directional data.
> 
> ~armando


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Caitlin Bestler wrote:

>>I'm not that familiar with RDDP, but my understanding is that the LLP 
>>(in this case SCTP) reports the MULPDU to DDP. Besides a small loss in 
>>efficiency due to increased overhead, is there any problem with taking 
>>control chunks into account when calculating the MULPDU? How is this 
>>handled if TCP (where options also take away from the MULPDU) is used as 
>>the LLP?
> 
> Yes, the DDP layer can account for this overhead. But it would be to
> the DDP applications advantage to refuse the option rather than to
> reduce its effective bandwidth. DDP applications are generally
> concerned with bulk data transfer. I believe that SCTP is far superior
> to TCP for DDP. But there is already an issue that at first glance
> the overhead of SCTP appears larger (it isn't when you understand that
> it can be shared across multiple streams). But I would not want to make
> those statistics appear worse by adding "just one more option".
> I would certainly *not* enable such a chunk by default (because I
> wouldn't want it to be part of my benchmark statistics).

I see your point. I think the best solution would be to simply let 
applications deactivate timestamps if desired. However, while the 
overhead may be significant for high performance bulk transfer, there 
are many other applications (e.g. in the signaling area) where it 
doesn't matter all that much.

> My suggestion is to allow that type of strategy to supply the data
> you want on an opportunistic basis, because there will be data
> transfer applications that will resist yielding any of their
> bandwidth when asked to do so on *every* packet. And it would
> be very easy for them to insist on a stack that at least pretended
> not to know what this option was, especially since they would have
> no incentive to upgrade to such an option.

Only sending the timestamp on an opportunistic basis would degrade the 
effectiveness of Eifel or algorithms to disambiguate the "path 
ambiguity". However, it may be good enough for those purposes to send 
timestamps with every packet that contains retransmitted chunks.

Stephan.




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Subject: Re: [Tsvwg] SCTP Timestamps
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Caitlin Bestler wrote:

>>The second motivation was to use timestamps for Eifel
>>detection. A spurious timeout retransmit can be detected if the
>>retransmitted TSN gets acked with a timestamp earlier than the
>>timestamp bundled with the retransmission. This purpose does
>>require a timestamp in each packet (I think).
> 
> It is still possible to infer a possible timestamp range for
> when a packet was sent by interpolation from earlier and later
> packets, particularly if the reception time of each packet is
> captured for use by SCTP.

The algorithms for the disambiguation of retransmissions are meant to 
counter problems caused by anomalies such as sudden delay spikes, packet 
reordering or, in the case of SCTP, retransmissions alternating between 
paths with different RTTs and separate path estimators. I believe it is 
next to impossible to "guess" the timestamps with a good hitrate under 
those conditions.

Stephan.


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From: Stephan Baucke <stephan.baucke@ericsson.com>
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Subject: Re: [Tsvwg] SCTP Timestamps
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Caitlin Bestler wrote:

> There are some assumptions being made here that I am not following.
> 
> If I am following the thread, the purpose of timestamping is to
> allow frequent ongoing measurement of the RTT along each path.
> Further, knowing the current RTT for each path, it will be
> possible to disambiguate certain Acks. This leads to better
> decisions on when to shift paths and/or do a fast retransmit.
> 
> Is that correct?

Caitlin,

getting better knowledge of the RTT is only one of the aspects. Eifel 
basically works by marking packets and ACKs in a way that allows the 
sender to distiguish whether an ACK refers to the original transmission 
or a retransmission. For things like disambiguating the path the chunk 
being acknowledged was sent over, it needs to be able to identify which 
(re)transmission it was.

> If so, why is timestamp data required in every packet?
> 
> MPEG transport maintains a distributed clock over unreliable
> jittered networks even in the face of frequent packet drops.
> I am not following why a timestamp must *always* be present
> to allow measurements to be made accurately.

It's not really about maintaining a common clock. There are other 
possible ways to achieve the goals mentioned above (e.g. marking packets 
with dedicated flags), but the timestamps have worked out nicely for 
TCP, and seem attractive to me because they can be used to solve several 
problems.

Stephan.


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Date: Thu, 17 Jul 2003 09:17:36 -0500
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] SCTP Timestamps
To: Stephan Baucke <stephan.baucke@ericsson.com>
Cc: tsvwg@ietf.org
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On Thursday, July 17, 2003, stephan.baucke@ericsson.com wrote:

>Only sending the timestamp on an opportunistic basis would degrade the 
>effectiveness of Eifel or algorithms to disambiguate the "path 
>ambiguity". However, it may be good enough for those purposes to send 
>timestamps with every packet that contains retransmitted chunks.

The chunk size was already determined on the first transmit. There
won't be any more room on the retransmit.


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Date: Thu, 17 Jul 2003 09:22:43 -0500
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] SCTP Timestamps
To: Stephan Baucke <stephan.baucke@ericsson.com>
Cc: tsvwg@ietf.org
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On Thursday, July 17, 2003, stephan.baucke@ericsson.com wrote:

>The algorithms for the disambiguation of retransmissions
>are meant to counter problems caused by anomalies such as
>sudden delay spikes, packet reordering or, in the case of
>SCTP, retransmissions alternating between paths with
>different RTTs and separate path estimators. I believe it
>is next to impossible to "guess" the timestamps with a
>good hitrate under those conditions.

You don't need to predict them. But you could easily detect
that they are out of the predictable range for a
retransmit-free stable path. That is, you could once a
history on the path was established.

For sporadic traffic this type of approach would indeed be
ineffective, but it is when you are transferring on a
sustained basis that you will be most concerned about not
inserting timestamps. Adding a small timestamp to sporadic
traffic will seldom be objectionable.

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At 18:25 11.07.2003, Mark Allman wrote:
> > 2. Segment N is lost, the ACK for the retransmit of N is lost, and
> >    the second retransmit of N goes fine (the segment and the ACK
> >    for it) arrives.
> >
> > 3. Segment N is lost, and the retransmit of N is duplicated in the
> >    network.
>
>[...]
>
>So, the question really is... can we accept these sorts of cases or
>not?  If we cannot accept these cases then DSACK (and duplicate TSN
>notification in SCTP) is pretty much useless for reverting
>congestion control state.  Opinions?
>
>My own hit is that things like 2 & 3 are pathalogical and should not
>be deal breakers for DSACK-based schemes for this purpose.

Since there hasn't been much traffic on this, here is my hit ...

I think that issue 3 is well taken care off in the current I-D.

But, I think that issue 2 needs to be addressed. With issue 2 we again have 
the retransmission ambiguity problem: the sender can't tell whether the 
DSACK corresponds to the 1st or the 2nd retransmit, and hence shouldn't 
make any conclusions.

So, in general dsack-use should be disabled once the second retransmit of 
the oldest outstanding segment has been sent.

///Reiner 


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From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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CC: Stephan Baucke <stephan.baucke@ericsson.com>, tsvwg@ietf.org
Subject: Re: [Tsvwg] SCTP Timestamps
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It really IS an implementation issue and has already been addressed
since the very first implementation (and I'd guess each of the existing
impls does the same... just in its own different way). For example, in
the open source user-land ref impl as I remember it, there are two
queues; one for pending control chunks and the other for pending user
messages. Whenever an outgoing packet is being formed, the bundling and
fragmentation code will always first include all the pending control
chunks and then figure out how much room is still left for user data. In
the calculation, it always count in the data chunk header, SCTP header,
and IP header... It is tricky code... one of the focal points the early
debugging excitement :-)

-Qiaobing

"Armando L. Caro Jr." wrote:
> 
> On Wed, 16 Jul 2003, Stephan Baucke wrote:
> 
> > Hm, ECNE/CWR chunks will not be in every packet. Has the issue of
> > reserving room for those been addressed? I'm also thinking about more
> > per-packet options that might be introduced in the future; should each
> > of them be allowed to take away some more room?
> 
> That's a tricky situation. It is really an implementation issue, but it
> probably makes sense to reserve room for chunks that must be bundled.
> Others which are self contained don't have to be bundled if they don't
> fit.
> 
> ~armando
> 
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0
> 
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On Thu, 17 Jul 2003, Reiner Ludwig wrote:

> Since there hasn't been much traffic on this, here is my hit ...
>
> I think that issue 3 is well taken care off in the current I-D.
>
> But, I think that issue 2 needs to be addressed. With issue 2 we again have
> the retransmission ambiguity problem: the sender can't tell whether the
> DSACK corresponds to the 1st or the 2nd retransmit, and hence shouldn't
> make any conclusions.
>
> So, in general dsack-use should be disabled once the second retransmit of
> the oldest outstanding segment has been sent.

That makes sense. I suppose this problem started the idea of using
timestamps instead?

~armando

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


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Transport Area WG (tsvwg)

Friday, July 18 at 0900-1130
=============================

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


0900 Charter Discussion (chairs)
0910 General Area Review (chairs)

0925 Fast TCP (Steven Low/Cheng Jin)

draft-jin-wei-low-tcp-fast-00.txt 

0955 Highspeed TCP (Sally Floyd)

draft-floyd-tcp-highspeed-03.txt

1015 NewReno to PS (Sally Floyd)

  draft-ietf-tsvwg-newreno-00.txt

1030 Eifel Response (Reiner Ludwig)

draft-ietf-tsvwg-tcp-eifel-response-03.txt

1045 MH-PHB (Naotaka Morita)

   draft-morita-tsvwg-pps-00.txt

1100 Update on PR-SCTP (TBD)

   draft-ietf-tsvwg-prsctp-00.txt

1115 Update on F-RTO (Markku Kojo)

   draft-sarolahti-tsvwg-tcp-frto-04.txt




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To: "Armando L. Caro Jr." <acaro@acm.org>
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        Ivan.Arias-Rodriguez@nokia.com, tsvwg@ietf.org
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I haven't gone thru the all the mails in this thread. My apology if my
comments below overlap with others.

"Armando L. Caro Jr." wrote:
...
> > A) A cooperative peer will wait long enough to bundle its HB-ACKs with
> > SACK. An uncooperative peer will flood the network with separate
> > HB-ACKs.  With timestamps, a cooperative peer will behave as with HBs,
> > but an uncooperative peer will do nothing.
> 
> How can a peer be cooperative without negotiation? And if there is
> negotiation, then why not use timestamps? (I assume we all agree that if
> negotiation is used, then timestamps are the best way to go.)

In the current SCTP, normally an endpoint will not receive a HB with an
outbound SACK pending. So, we could simply add the following rule to get
the behavior you want without negotiation:

 When an endpoint receives a HB while having received data pending 
 for ack, the endpoint should delay sending its HB-ACK till the 
 SACK is sent and the HB-ACK should be bundled with SACK.

Would this solve the problem?

> > C) 100% reliable detection of spurious timeouts is unnecessary.
> 
> Maybe not, but then what is necessary? The purpose of timestamps is to
> move away from heuristics for robustness. But the HB idea seems to add
> more heuristics.

I've heard that 50% detection is not good enough and 100% is unnecessary
(and in practice unachievable), and 90% may be about right... This
sounds all too speculative to me. Are there any real world data from the
field or simulation that shows what's the performance gain vs. various
percentage of detection? Or these are just theoretical speculations? 

Also, there seem to be other efforts to address spurious timeouts (e.g.,
D-SACK and two other methods mentioned in this thread). Is there a
comparison study on them? Or, are we going to build them all into TCP
and SCTP? :-)

regards,
-Qiaobing

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Subject: Re: [Tsvwg] SCTP Timestamps
References: <Pine.GSO.4.33.0307161413020.6553-100000@stimpy.eecis.udel.edu> <Pine.GSO.4.33.0307161425360.21227-100000@louie.udel.edu> <20030716213805.A10144@openss7.org>
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Brian,

You made a good point and you convinced me that a separate TS chunk is
cleaner if this feature is really worthwhile to add.

-Qiaobing

"Brian F. G. Bidulock" wrote:
>=20
> Sourabh,
>=20
> I agree that HEARTBEAT/ACK is not a good way to do this.  Although the
> opaque data in the chunk looks attractive, there are rather narrow rule=
s
> for the handling of HB/ACK that have a rather different purpose.  SCTP
> relies on the HB mechanism for recovery of unavailable destinations in
> the multihomed environment and is also important for detection of lost
> ABORTs.  There are some items concerning HB that was discovered in inte=
rop
> testing that showed that rather special rules were required to accomoda=
te
> implementations that could not (or did not) control source address vs. =
those
> that did.
>=20
> I think that by overloading HB/ACK for the purpose of doing more freque=
nt
> RTT measurements would break the original purpose of HB/ACK: testing th=
e
> availability of desintations, keeping RTO current on idle destinations,=
 and
> detecting lost ABORT chunks.
>=20
> I too would prefer to see a different chunk, (TS chunk) bundled with al=
l
> outgoing packets containing DATA and use an extension of the SACK chunk=
 to
> return pertinent timestamps.  (SACK is the right place to put it on the
> return anyways.)
>=20
> --brian
>=20
> On Wed, 16 Jul 2003, Sourabh Ladha wrote:
>=20
> >
> > > Yes, you are right. The HB-ACK becomes like another SACK. I like th=
ese
> > > ideas. Since the information contained in the HB is actually sender
> > > implementation specific, each implementation can embed what ever in=
fo they
> > > find most useful. It's up to that implementation to be effecient wi=
th
> > > overhead tradeoffs.
> >
> > I do not agree at all if this is a "clean" way to proceed if a timest=
amp
> > information per packet is required. The cost that you incur by
> > this is a whole new definition of SACK, HB-ACK and rules associated w=
ith
> > these. (For example you would still need SACKs along with HB-ACKs if =
an
> > out of order, duplicate chunk is recieved)
> >
> > I still think if needed an optional timestamp chunk is the best way t=
o go
> > if such an information is required.
> >
> >
> > -Sourabh
> >
> >
> > _______________________________________________
> > tsvwg mailing list
> > tsvwg@ietf.org
> > https://www1.ietf.org/mailman/listinfo/tsvwg
>=20
> --
> Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
> bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
> http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
>                         =A6 Therefore  all  progress  depends on the =A6
>                         =A6 unreasonable man. -- George Bernard Shaw =A6

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Date: Thu, 17 Jul 2003 14:05:55 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Qiaobing Xie <Qiaobing.Xie@motorola.com>
Cc: "Armando L. Caro Jr." <acaro@acm.org>,
        "Duke, Martin" <martin.duke@boeing.com>,
        Sourabh Ladha <ladha@mail.eecis.udel.edu>,
        Ivan.Arias-Rodriguez@nokia.com, tsvwg@ietf.org
Subject: Re: [Tsvwg] SCTP Timestamps
Message-ID: <20030717140555.A25067@openss7.org>
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Qiaobing,

On Thu, 17 Jul 2003, Qiaobing Xie wrote:

> I haven't gone thru the all the mails in this thread. My apology if my
> comments below overlap with others.
>=20
> "Armando L. Caro Jr." wrote:
> ...
> > > A) A cooperative peer will wait long enough to bundle its HB-ACKs w=
ith
> > > SACK. An uncooperative peer will flood the network with separate
> > > HB-ACKs.  With timestamps, a cooperative peer will behave as with H=
Bs,
> > > but an uncooperative peer will do nothing.
> >=20
> > How can a peer be cooperative without negotiation? And if there is
> > negotiation, then why not use timestamps? (I assume we all agree that=
 if
> > negotiation is used, then timestamps are the best way to go.)
>=20
> In the current SCTP, normally an endpoint will not receive a HB with an
> outbound SACK pending. So, we could simply add the following rule to ge=
t
> the behavior you want without negotiation:
>=20
>  When an endpoint receives a HB while having received data pending=20
>  for ack, the endpoint should delay sending its HB-ACK till the=20
>  SACK is sent and the HB-ACK should be bundled with SACK.
>=20
> Would this solve the problem?

Would the above not result in delaying ALL HEARTBEAT-ACK chunks?  When
you delay that HEARTBEAT-ACK chunk, cannot an RTO can occur because the
HEARTBEAT-ACK is lost?  Currently the HEARTBEAT-ACK is sent to the
source address in the HEARTBEAT, and the RFC says to send any bundled
DATA to that address too.  Will this not result in sending DATA to the
wrong address (not the primary) most of the time?  Will not HEARTBEAT-ACK=
s
sent testing idle or failed destinations also have to wait when data
acknowledgement is pending for heartbeat acknowledgement?  There is no
way for the receive to know the difference between a normal HEARTBEAT
and a timestamp HEARTBEAT, is there?  The difference lies in the opaque
data.

Please consider using a new chunk to avoid breaking the current heartbeat
mechanism.

--brian

>=20
> > > C) 100% reliable detection of spurious timeouts is unnecessary.
> >=20
> > Maybe not, but then what is necessary? The purpose of timestamps is t=
o
> > move away from heuristics for robustness. But the HB idea seems to ad=
d
> > more heuristics.
>=20
> I've heard that 50% detection is not good enough and 100% is unnecessar=
y
> (and in practice unachievable), and 90% may be about right... This
> sounds all too speculative to me. Are there any real world data from th=
e
> field or simulation that shows what's the performance gain vs. various
> percentage of detection? Or these are just theoretical speculations?=20
>=20
> Also, there seem to be other efforts to address spurious timeouts (e.g.=
,
> D-SACK and two other methods mentioned in this thread). Is there a
> comparison study on them? Or, are we going to build them all into TCP
> and SCTP? :-)
>=20
> regards,
> -Qiaobing
>=20
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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

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From exim@www1.ietf.org  Thu Jul 17 20:34:32 2003
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Date: Thu, 17 Jul 2003 17:33:10 -0700
From: Andrei Gurtov <gurtov@cs.helsinki.fi>
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Subject: [Tsvwg] Slides for Eifel Response in Vienna IETF
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Slides for the presentation on Eifel Response in the TSVWG meeting in 
Vienna IETF are available at:

http://www.cs.helsinki.fi/u/gurtov/reiner/eifel_alg_ietf57.pdf
http://www.cs.helsinki.fi/u/gurtov/reiner/eifel_alg_ietf57.ppt

Also of possible interest is:

A. Gurtov, R. Ludwig, Responding to Spurious Timeouts in TCP, IEEE 
INFOCOM'03, March 2003.
http://www.cs.helsinki.fi/u/gurtov/papers/resp.html

Reiner&Andrei


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Qiaobing Xie wrote:

> I've heard that 50% detection is not good enough and 100% is unnecessary
> (and in practice unachievable), and 90% may be about right... This
> sounds all too speculative to me. Are there any real world data from the
> field or simulation that shows what's the performance gain vs. various
> percentage of detection? Or these are just theoretical speculations? 
> 
> Also, there seem to be other efforts to address spurious timeouts (e.g.,
> D-SACK and two other methods mentioned in this thread). Is there a
> comparison study on them? Or, are we going to build them all into TCP
> and SCTP? :-)

I also want to know why can't (since nobody in this thread even
mentioned it) the duplicate TSN be used to detect spurious
retransmission.  It is already specified in the standard and does
not require any new chunk be defined.  The draw back is not 100%
reliable and it is "slower" to detect the problem.  Is there a
fundamental issue with it such that people in this discussion want
a new chunk to be defined?


---

						K. Poon.
						kacheong.poon@sun.com




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On Thu, 17 Jul 2003, Kacheong Poon wrote:

> I also want to know why can't (since nobody in this thread even
> mentioned it) the duplicate TSN be used to detect spurious
> retransmission.  It is already specified in the standard and does
> not require any new chunk be defined.  The draw back is not 100%
> reliable and it is "slower" to detect the problem.  Is there a
> fundamental issue with it such that people in this discussion want
> a new chunk to be defined?

From what I understand, timestamps can provide a more robust method of
detection.

~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, 18 Jul 2003 01:52:32 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
Cc: Kacheong Poon <kacheong.poon@sun.com>, tsvwg@ietf.org
Subject: Re: [Tsvwg] SCTP Timestamps
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Armando,

What is not robust about the receiver indicating that it received
something multiple times?  The receiver is in the best position to
determine when a spurious retransmission has occured at the sender
(because it receives multiple copies).

I think that is more robust than timestamps that may or may not be
included, might get lost, misbundled, associated with the wrong ack,
etc.

With duplicates, the receiver says "I already received this TSN".
I agree with Kacheong here, but I disagree that it is slower.  In
the RFC receiving duplicates is reason for immediate SACK (whereas
timestamps would not).  Therefore, the dup TSNs are, I believe,
faster, in that the duplicate TSN report is immediate under most
situations.  (See 6.2 RFC 2960).

Furthermore, the duplicate TSN mechanism is already exists, and is
already used to determine that a spurious restransmission has
occurred, possibly due to a lost SACK return path (the most obvious
cause of persistent spurious retransmissions).

Can someone back up the statement of timestamps being "more robust"?

The statement doesn't seem to withstand any scrutiny...

--brian

On Fri, 18 Jul 2003, Armando L. Caro Jr. wrote:

> On Thu, 17 Jul 2003, Kacheong Poon wrote:
>=20
> > I also want to know why can't (since nobody in this thread even
> > mentioned it) the duplicate TSN be used to detect spurious
> > retransmission.  It is already specified in the standard and does
> > not require any new chunk be defined.  The draw back is not 100%
> > reliable and it is "slower" to detect the problem.  Is there a
> > fundamental issue with it such that people in this discussion want
> > a new chunk to be defined?
>=20
> >From what I understand, timestamps can provide a more robust method of
> detection.
>=20
> ~armando
>=20
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0
>=20
>=20
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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

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From: Schruefer Wolfgang <wolfgang.schruefer@siemens.com>
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Subject: Re: [Tsvwg] SCTP Timestamps
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Dear all,

I will suggest an alternative solution, how to enable a sender to determine whether an ACK refers to the original transmission or a retransmission of a chunk.

My thought is to sacrifice one of the unused bits in the Chunk Flags field of the DATA and SACK chunks to solve the problem. The clue is that the value of this bit from the DATA chunk should be echoed by the receiver in the SACK chunk. A feature negotiation at association setup is not necessary, because the sender recognises early if the feature is supported by the peer or not.


A scenario may run like this (let me just call this new bit Mirror Bit or M-bit): 

- A sender sets the M-bit at any value when a DATA chunk is transmitted for the first time and stores the value together with the timestamp.

- In the SACK the receiver mirrors the M-bit for the DATA chunk with TSN == Cumulative TSN Ack.

- When the sender recognises that the M-bit is not echoed correctly (for not retransmitted DATA), it knows that the peer does not support this feature. 

- When the sender needs to retransmit the DATA for the first time it toggles the M-bit and stores the retransmission time.

- For up to one retransmission the sender can 100% resolve the chunk where a SACK refers to.

- When there is a need for further retransmissions I see three alternatives:
  ** The M-bit is toggled again and the sender assumes that the DATA chunk sent two RTO earlier was lost.
  ** The M-bit remains identical in all retransmissions (but e.g. a SACK for this DATA chunk may not be used any more for RTT calculation).
  ** We use two Chunk Flag Bits to mark the last four transmissions.


Is this a way worth while to think about?

Wolfgang











Dear all,

following the Timestamp discussion I asked myself whether there is a need to transmit any timestamp and if there are no cheaper solution. 
What we need in my understanding is a solution for the problem Stephan described at the beginning of the thread:

Resolving the retransmission ambiguity, i.e. enabling the sender to determine whether an ACK refers to the original transmission or a retransmission of a chunk. This implies resolving a "path ambiguity" and allowing better and more frequent RTT measurements.

My thought is to sacrifice one (or two) bit(s) of the unused DATA and SACK Chunk Flags Bits to solve this problem.  The clue is that this bit should be echoed by the receiver in the SACK Chunk Flags. Let me just call this bit Mirror Bit or M-bit.  


A scenario may run like this:
- A sender sets the M-bit at any value when a DATA chunk is transmitted for the first time and stores the value together with the timestamp.

- The receiver mirrors the bit for the DATA chunk with TSN == Cumulative TSN Ack.

- When the sender recognises that the M-bit is not echoed correctly (for not retransmitted DATA), it knows that the peer does not support this feature. I think a feature negotiation at association setup is therefore not necessary.

- When the sender needs to retransmit the DATA for the first time it toggles the M-bit and stores the retransmission time.

- Up to now the sender can always resolve the chunk where a SACK refers to.

- When there is a need for further retransmissions I see three alternatives:
  ** The M-bit is toggled again and the sender assumes that the DATA chunk sent two RTO before was lost.
  ** The M-bit remains identical in all retransmissions (but e.g. a SACK for this DATA chunk may not be used any more for RTT calculation).
  ** We use two Chunk Flag Bits to mark the last four transmissions.


Is this a way worth to think further?

Wolfgang


Viele Gruesse
Wolfgang Schruefer

Siemens ICN CP D NP A8
TEL: +49 89 722-61646 


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Date: Fri, 18 Jul 2003 01:47:14 -0700
Subject: [Tsvwg] the outstanding issues to track for taking NewReno to PS
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From the tsvwg meeting at the IETF, now in progress, here are
the issues to track (in the next week) for WG Last Call,
for the NewReno TCP:

* Add a note to the section on "Implementation issues for the data 
  sender" saying that the sender might want a separate flag to record
  whether it is in the Fast Recovery procedure, for robustness
  with window updates and out-of-order acks.
 
* Add an implementation note about taking care of sequence wrap.
 
* Address the issue raised by Andrei Gurtov about whether
  packets are sent in response to dup acks in those cases
  when Fast Recovery is not invoked (because of bugfix).

- Sally



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Wolfgang,

What about the packet that contains:

  RTx (2) DATA RTx (2) DATA RTx (1) DATA New DATA New DATA

from two separate retransmissions.

What about when the spurious retransmissions are before the
Cummulative Ack point (due to loss of SACKs)?  Won't these
M-bit indications be lost?

Doesn't sound better than existing dup TSN reports in SACKs.

--brian

On Fri, 18 Jul 2003, Schruefer Wolfgang wrote:

> Dear all,
>=20
> I will suggest an alternative solution, how to enable a sender to deter=
mine
> whether an ACK refers to the original transmission or a retransmission =
of a
> chunk.
>=20
> My thought is to sacrifice one of the unused bits in the Chunk Flags fi=
eld
> of the DATA and SACK chunks to solve the problem. The clue is that the =
value
> of this bit from the DATA chunk should be echoed by the receiver in the=
 SACK
> chunk. A feature negotiation at association setup is not necessary, bec=
ause
> the sender recognises early if the feature is supported by the peer or =
not.
>=20
>=20
> A scenario may run like this (let me just call this new bit Mirror Bit =
or
> M-bit):=20
>=20
> - A sender sets the M-bit at any value when a DATA chunk is transmitted=
 for
> the first time and stores the value together with the timestamp.
>=20
> - In the SACK the receiver mirrors the M-bit for the DATA chunk with TS=
N =3D=3D
> Cumulative TSN Ack.
>=20
> - When the sender recognises that the M-bit is not echoed correctly (fo=
r not
> retransmitted DATA), it knows that the peer does not support this featu=
re.=20
>=20
> - When the sender needs to retransmit the DATA for the first time it to=
ggles
> the M-bit and stores the retransmission time.
>=20
> - For up to one retransmission the sender can 100% resolve the chunk wh=
ere a
> SACK refers to.
>=20
> - When there is a need for further retransmissions I see three alternat=
ives:
> ** The M-bit is toggled again and the sender assumes that the DATA chun=
k
> sent two RTO earlier was lost.  ** The M-bit remains identical in all
> retransmissions (but e.g. a SACK for this DATA chunk may not be used an=
y
> more for RTT calculation).  ** We use two Chunk Flag Bits to mark the l=
ast
> four transmissions.
>=20
>=20
> Is this a way worth while to think about?
>=20
> Wolfgang
>=20

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

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>
> I also want to know why can't (since nobody in this thread even
> mentioned it) the duplicate TSN be used to detect spurious
> retransmission.  It is already specified in the standard and does
> not require any new chunk be defined.  The draw back is not 100%
> reliable and it is "slower" to detect the problem.  Is there a
> fundamental issue with it such that people in this discussion want
> a new chunk to be defined?
>

From some of the emails I have seen in this thread, it seems folks have
the wrong impression that timestamps are being proposed for "Eifel
purpose".

I would like to clarify and REPEAT myself again that we were NOT
discussing timestamps for Eifel purposes. Eifel happens to be
ONE of the possible mechanisms which can use timestamps.


-Sourabh


















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


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References: <Pine.GSO.4.33.0307161749350.6553-100000@stimpy.eecis.udel.edu> <3F16F159.34FC8B4C@motorola.com> <3F176037.8090100@sun.com>
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Kacheong Poon wrote:

> I also want to know why can't (since nobody in this thread even
> mentioned it) the duplicate TSN be used to detect spurious
> retransmission.  It is already specified in the standard and does
> not require any new chunk be defined.  The draw back is not 100%
> reliable and it is "slower" to detect the problem.  Is there a
> fundamental issue with it such that people in this discussion want
> a new chunk to be defined?

You have already mentioned one of the reasons: Detection of spurious 
retransmissions using DSACK only works "after the fact", whereas 
something like Eifel detects it early enough to avoid a go-back-n-like 
behavior after a spurious timeout (also see RFC 3522). Also, DSACKs do 
not address the other issues mentioned in this thread, such as the "path 
ambiguity" and RTTM. Note that DSACKs are still useful and may be 
combined with Eifel; I think there is a compelling case for introducing 
optional timestamps even if DSACKs are already utilized.

Stephan.


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References: <3F176037.8090100@sun.com> <Pine.GSO.4.33.0307180000360.13032-100000@stimpy.eecis.udel.edu> <20030718015232.C3547@openss7.org>
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Brian F. G. Bidulock wrote:

> What is not robust about the receiver indicating that it received
> something multiple times?  The receiver is in the best position to
> determine when a spurious retransmission has occured at the sender
> (because it receives multiple copies).
> 
> I think that is more robust than timestamps that may or may not be
> included, might get lost, misbundled, associated with the wrong ack,
> etc.

Well, this is what I was hoping to get out of this disucssion: A good 
way to do timestamps without those problems :-)

> With duplicates, the receiver says "I already received this TSN".
> I agree with Kacheong here, but I disagree that it is slower.  In
> the RFC receiving duplicates is reason for immediate SACK (whereas
> timestamps would not).  Therefore, the dup TSNs are, I believe,
> faster, in that the duplicate TSN report is immediate under most
> situations.  (See 6.2 RFC 2960).

The problem is that with DSACKs the sender is usually only notified 
after he has already spuriously retransmitted all packets he erroneously 
considered lost. With timestamps, the sender can tell that a spurious 
retransmission happened from the first new ACK. This way unnecessary 
go-back-n retransmissions can be avoided in many cases. Please see RFC 
3522 for more.

> Furthermore, the duplicate TSN mechanism is already exists, and is
> already used to determine that a spurious restransmission has
> occurred, possibly due to a lost SACK return path (the most obvious
> cause of persistent spurious retransmissions).
> 
> Can someone back up the statement of timestamps being "more robust"?

For one, the DSACK method is more sensitive to ACK losses.

Stephan.


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>
> What is not robust about the receiver indicating that it received
> determine when a spurious retransmission has occured at the sender
> (because it receives multiple copies).
>
> I think that is more robust than timestamps that may or may not be
> included, might get lost, misbundled, associated with the wrong ack,
> etc.
>
> With duplicates, the receiver says "I already received this TSN".
> I agree with Kacheong here, but I disagree that it is slower.  In
> the RFC receiving duplicates is reason for immediate SACK (whereas
> timestamps would not).  Therefore, the dup TSNs are, I believe,
> faster, in that the duplicate TSN report is immediate under most
> situations.  (See 6.2 RFC 2960).
>
> Furthermore, the duplicate TSN mechanism is already exists, and is
> already used to determine that a spurious restransmission has
> occurred, possibly due to a lost SACK return path (the most obvious
> cause of persistent spurious retransmissions).

hmm..What about packet reordering, packet duplication, delay spikes..we
seem to be missing some basic causes of spurious retransmissions.

>
> Can someone back up the statement of timestamps being "more robust"?

First, we seem to be using the terms Eifel and timestamps in the same
context. This is a fundamental flaw. Are you asking the motivation behind
timestamps or are you asking how Eifel using timestamps is more robust
than the dup-tsn detection.

I would guess the second..from the top of my head, I can think of a
few advantages of Eifel over dup-tsn detection.

* the dup-tsn notification comes much late in the game. You cannot avoid a
GoBackN (which is one of the motivations anyway for having a spurious
retransmission detection scheme) if you had a spurious timeout. (I believe
RFC3522 will have more details on these). Eifel will be able to detect
possibly on the arrival of the first acceptable ACK if the retransmission
was spurious.

* the dup-tsn detection is sensitive to ACK loss: if you lost the ACK
carrying the duptsn you possibly fail the detection.

* the duptsn notification is less robust to packet duplication (in
particular for SCTP). I remember Reiner and I had noted this in some of
the previous emails on a different thread: If the SCTP association is
multihomed and the alternate path is duplicating packets then with the
dup-tsn detection scheme coupled with a response algorithm may end up
making an SCTP non-responsive to congestion.


-Sourabh



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From: Stephan Baucke <stephan.baucke@ericsson.com>
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Subject: Re: [Tsvwg] SCTP Timestamps
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Wolfgang,

this is indeed another way to resolve the retransmission ambiguity (in 
fact the first proposal of the Eifel algorithm for TCP used this 
method). The questions are whether (1) there are header bits available 
for this and (2) whether losing the additional benefits of timestamps 
(RTTM and the ability to distinguish not only between original 
transmission and retransmission, but between an arbitrary number of 
consecutive retransmissions as well) is a serious concern.

Stephan.


Schruefer Wolfgang wrote:

> Dear all,
> 
> I will suggest an alternative solution, how to enable a sender to
> determine whether an ACK refers to the original transmission or a
> retransmission of a chunk.
> 
> My thought is to sacrifice one of the unused bits in the Chunk Flags
> field of the DATA and SACK chunks to solve the problem. The clue is
> that the value of this bit from the DATA chunk should be echoed by
> the receiver in the SACK chunk. A feature negotiation at association
> setup is not necessary, because the sender recognises early if the
> feature is supported by the peer or not.
> 
> 
> A scenario may run like this (let me just call this new bit Mirror
> Bit or M-bit):
> 
> - A sender sets the M-bit at any value when a DATA chunk is
> transmitted for the first time and stores the value together with the
> timestamp.
> 
> - In the SACK the receiver mirrors the M-bit for the DATA chunk with
> TSN == Cumulative TSN Ack.
> 
> - When the sender recognises that the M-bit is not echoed correctly
> (for not retransmitted DATA), it knows that the peer does not support
> this feature.
> 
> - When the sender needs to retransmit the DATA for the first time it
> toggles the M-bit and stores the retransmission time.
> 
> - For up to one retransmission the sender can 100% resolve the chunk
> where a SACK refers to.
> 
> - When there is a need for further retransmissions I see three
> alternatives: ** The M-bit is toggled again and the sender assumes
> that the DATA chunk sent two RTO earlier was lost. ** The M-bit
> remains identical in all retransmissions (but e.g. a SACK for this
> DATA chunk may not be used any more for RTT calculation). ** We use
> two Chunk Flag Bits to mark the last four transmissions.
> 
> 
> Is this a way worth while to think about?
> 
> Wolfgang
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> Dear all,
> 
> following the Timestamp discussion I asked myself whether there is a
> need to transmit any timestamp and if there are no cheaper solution.
>  What we need in my understanding is a solution for the problem
> Stephan described at the beginning of the thread:
> 
> Resolving the retransmission ambiguity, i.e. enabling the sender to
> determine whether an ACK refers to the original transmission or a
> retransmission of a chunk. This implies resolving a "path ambiguity"
> and allowing better and more frequent RTT measurements.
> 
> My thought is to sacrifice one (or two) bit(s) of the unused DATA and
> SACK Chunk Flags Bits to solve this problem.  The clue is that this
> bit should be echoed by the receiver in the SACK Chunk Flags. Let me
> just call this bit Mirror Bit or M-bit.
> 
> 
> A scenario may run like this: - A sender sets the M-bit at any value
> when a DATA chunk is transmitted for the first time and stores the
> value together with the timestamp.
> 
> - The receiver mirrors the bit for the DATA chunk with TSN ==
> Cumulative TSN Ack.
> 
> - When the sender recognises that the M-bit is not echoed correctly
> (for not retransmitted DATA), it knows that the peer does not support
> this feature. I think a feature negotiation at association setup is
> therefore not necessary.
> 
> - When the sender needs to retransmit the DATA for the first time it
> toggles the M-bit and stores the retransmission time.
> 
> - Up to now the sender can always resolve the chunk where a SACK
> refers to.
> 
> - When there is a need for further retransmissions I see three
> alternatives: ** The M-bit is toggled again and the sender assumes
> that the DATA chunk sent two RTO before was lost. ** The M-bit
> remains identical in all retransmissions (but e.g. a SACK for this
> DATA chunk may not be used any more for RTT calculation). ** We use
> two Chunk Flag Bits to mark the last four transmissions.
> 
> 
> Is this a way worth to think further?
> 
> Wolfgang
> 
> 
> Viele Gruesse Wolfgang Schruefer
> 
> Siemens ICN CP D NP A8 TEL: +49 89 722-61646
> 
> 
> _______________________________________________ tsvwg mailing list 
> tsvwg@ietf.org https://www1.ietf.org/mailman/listinfo/tsvwg



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From: Schruefer Wolfgang <wolfgang.schruefer@siemens.com>
To: "'bidulock@openss7.org'" <bidulock@openss7.org>
Cc: tsvwg@ietf.org
Subject: AW: [Tsvwg] SCTP Timestamps
Date: Fri, 18 Jul 2003 11:35:54 +0200
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Brian,

yes, it is normal that M-bit indications a lost, but why do you care =
about spurious retransmissions older than Cumulative Ack Point?
In my implementation I would not care because RTT measurement is done =
only for the DATA chunk at the Cumulative Ack Point.

I think it is better than dup TSN reports, because these arrive =
delayed.

Wolfgang =20



> Von: Brian F. G. Bidulock [mailto:bidulock@openss7.org]
> Gesendet: Freitag, 18. Juli 2003 10:55
>=20
> Wolfgang,
>=20
> What about the packet that contains:
>=20
>   RTx (2) DATA RTx (2) DATA RTx (1) DATA New DATA New DATA
>=20
> from two separate retransmissions.
>=20
> What about when the spurious retransmissions are before the
> Cummulative Ack point (due to loss of SACKs)?  Won't these
> M-bit indications be lost?
>=20
> Doesn't sound better than existing dup TSN reports in SACKs.
>=20
> --brian
>=20
> On Fri, 18 Jul 2003, Schruefer Wolfgang wrote:
>=20
> > Dear all,
> >=20
> > I will suggest an alternative solution, how to enable a=20
> sender to determine
> > whether an ACK refers to the original transmission or a=20
> retransmission of a
> > chunk.
> >=20
> > My thought is to sacrifice one of the unused bits in the=20
> Chunk Flags field
> > of the DATA and SACK chunks to solve the problem. The clue=20
> is that the value
> > of this bit from the DATA chunk should be echoed by the=20
> receiver in the SACK
> > chunk. A feature negotiation at association setup is not=20
> necessary, because
> > the sender recognises early if the feature is supported by=20
> the peer or not.
> >=20
> >=20
> > A scenario may run like this (let me just call this new bit=20
> Mirror Bit or
> > M-bit):=20
> >=20
> > - A sender sets the M-bit at any value when a DATA chunk is=20
> transmitted for
> > the first time and stores the value together with the timestamp.
> >=20
> > - In the SACK the receiver mirrors the M-bit for the DATA=20
> chunk with TSN =3D=3D
> > Cumulative TSN Ack.
> >=20
> > - When the sender recognises that the M-bit is not echoed=20
> correctly (for not
> > retransmitted DATA), it knows that the peer does not=20
> support this feature.=20
> >=20
> > - When the sender needs to retransmit the DATA for the=20
> first time it toggles
> > the M-bit and stores the retransmission time.
> >=20
> > - For up to one retransmission the sender can 100% resolve=20
> the chunk where a
> > SACK refers to.
> >=20
> > - When there is a need for further retransmissions I see=20
> three alternatives:
> > ** The M-bit is toggled again and the sender assumes that=20
> the DATA chunk
> > sent two RTO earlier was lost.  ** The M-bit remains=20
> identical in all
> > retransmissions (but e.g. a SACK for this DATA chunk may=20
> not be used any
> > more for RTT calculation).  ** We use two Chunk Flag Bits=20
> to mark the last
> > four transmissions.
> >=20
> >=20
> > Is this a way worth while to think about?
> >=20
> > Wolfgang
> >=20
>=20
> --=20
> Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =
=A6
> bidulock@openss7.org    =A6 world; the unreasonable one persists in  =
=A6
> http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =
=A6
>                         =A6 Therefore  all  progress  depends on the =
=A6
>                         =A6 unreasonable man. -- George Bernard Shaw =
=A6
>=20

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From: Schruefer Wolfgang <wolfgang.schruefer@siemens.com>
To: "'Stephan Baucke'" <stephan.baucke@ericsson.com>, tsvwg@ietf.org
Subject: AW: [Tsvwg] SCTP Timestamps
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Stephan,
please see my comments below...
Wolfgang

> Von: Stephan Baucke 
> 
> this is indeed another way to resolve the retransmission 
> ambiguity (in 
> fact the first proposal of the Eifel algorithm for TCP used this 
> method). The questions are whether (1) there are header bits 
> available 
> for this 

I can not judge if this is a feature worth to sacrifice a bit, but there are 5 unused Bits in the DATA chunk header... 

> and (2) whether losing the additional benefits of timestamps 
> (RTTM 

Why do we loose RTTM? The timestamps (transmission and retransmission) are stored locally.

> and the ability to distinguish not only between original 
> transmission and retransmission, but between an arbitrary number of 
> consecutive retransmissions as well

Here I doubt that it makes sense to enhance the communication if more than one retransmission is needed for a significant number of packets. 

) is a serious concern.
> 
> Stephan.
> 
> 
> Schruefer Wolfgang wrote:
> 
> > Dear all,
> > 
> > I will suggest an alternative solution, how to enable a sender to
> > determine whether an ACK refers to the original transmission or a
> > retransmission of a chunk.
> > 
> > My thought is to sacrifice one of the unused bits in the Chunk Flags
> > field of the DATA and SACK chunks to solve the problem. The clue is
> > that the value of this bit from the DATA chunk should be echoed by
> > the receiver in the SACK chunk. A feature negotiation at association
> > setup is not necessary, because the sender recognises early if the
> > feature is supported by the peer or not.
> > 
> > 
> > A scenario may run like this (let me just call this new bit Mirror
> > Bit or M-bit):
> > 
> > - A sender sets the M-bit at any value when a DATA chunk is
> > transmitted for the first time and stores the value 
> together with the
> > timestamp.
> > 
> > - In the SACK the receiver mirrors the M-bit for the DATA chunk with
> > TSN == Cumulative TSN Ack.
> > 
> > - When the sender recognises that the M-bit is not echoed correctly
> > (for not retransmitted DATA), it knows that the peer does 
> not support
> > this feature.
> > 
> > - When the sender needs to retransmit the DATA for the first time it
> > toggles the M-bit and stores the retransmission time.
> > 
> > - For up to one retransmission the sender can 100% resolve the chunk
> > where a SACK refers to.
> > 
> > - When there is a need for further retransmissions I see three
> > alternatives: ** The M-bit is toggled again and the sender assumes
> > that the DATA chunk sent two RTO earlier was lost. ** The M-bit
> > remains identical in all retransmissions (but e.g. a SACK for this
> > DATA chunk may not be used any more for RTT calculation). ** We use
> > two Chunk Flag Bits to mark the last four transmissions.
> > 
> > 
> > Is this a way worth while to think about?
> > 
> > Wolfgang
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > Dear all,
> > 
> > following the Timestamp discussion I asked myself whether there is a
> > need to transmit any timestamp and if there are no cheaper solution.
> >  What we need in my understanding is a solution for the problem
> > Stephan described at the beginning of the thread:
> > 
> > Resolving the retransmission ambiguity, i.e. enabling the sender to
> > determine whether an ACK refers to the original transmission or a
> > retransmission of a chunk. This implies resolving a "path ambiguity"
> > and allowing better and more frequent RTT measurements.
> > 
> > My thought is to sacrifice one (or two) bit(s) of the 
> unused DATA and
> > SACK Chunk Flags Bits to solve this problem.  The clue is that this
> > bit should be echoed by the receiver in the SACK Chunk Flags. Let me
> > just call this bit Mirror Bit or M-bit.
> > 
> > 
> > A scenario may run like this: - A sender sets the M-bit at any value
> > when a DATA chunk is transmitted for the first time and stores the
> > value together with the timestamp.
> > 
> > - The receiver mirrors the bit for the DATA chunk with TSN ==
> > Cumulative TSN Ack.
> > 
> > - When the sender recognises that the M-bit is not echoed correctly
> > (for not retransmitted DATA), it knows that the peer does 
> not support
> > this feature. I think a feature negotiation at association setup is
> > therefore not necessary.
> > 
> > - When the sender needs to retransmit the DATA for the first time it
> > toggles the M-bit and stores the retransmission time.
> > 
> > - Up to now the sender can always resolve the chunk where a SACK
> > refers to.
> > 
> > - When there is a need for further retransmissions I see three
> > alternatives: ** The M-bit is toggled again and the sender assumes
> > that the DATA chunk sent two RTO before was lost. ** The M-bit
> > remains identical in all retransmissions (but e.g. a SACK for this
> > DATA chunk may not be used any more for RTT calculation). ** We use
> > two Chunk Flag Bits to mark the last four transmissions.
> > 
> > 
> > Is this a way worth to think further?
> > 
> > Wolfgang
> > 
> > 
> > Viele Gruesse Wolfgang Schruefer
> > 
> > Siemens ICN CP D NP A8 TEL: +49 89 722-61646
> > 
> > 
> > _______________________________________________ tsvwg mailing list 
> > tsvwg@ietf.org https://www1.ietf.org/mailman/listinfo/tsvwg
> 
> 
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 

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From exim@www1.ietf.org  Fri Jul 18 06:22:24 2003
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From: Mark Handley <M.Handley@cs.ucl.ac.uk>
To: tsvwg@ietf.org
Date: Fri, 18 Jul 2003 10:32:29 +0100
Message-ID: <9147.1058520749@eve.cs.ucl.ac.uk>
Subject: [Tsvwg] High-speed TCP
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In followup to today's TSVWG meeting, here's some suggested
boilerplate text for the High-speed TCP Experimental RFC:

  The proposals in this document are experimental.  We believe they
  are safe for deployment in the current Internet, but they do not
  represent a consensus that this is the best method for high-speed
  congestion control.  In particular, we note that alternative
  experimental proposals are likely to be forthcoming, and it is not
  well understood how the proposals in this document will interact
  with such alternative proposals.  

Cheers,
	Mark

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Date: Fri, 18 Jul 2003 04:39:32 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Sourabh Ladha <ladha@mail.eecis.udel.edu>
Cc: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>,
        Kacheong Poon <kacheong.poon@sun.com>, tsvwg@ietf.org
Subject: Re: [Tsvwg] SCTP Timestamps
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Sourabh,

On Fri, 18 Jul 2003, Sourabh Ladha wrote:

> >
> > What is not robust about the receiver indicating that it received
> > determine when a spurious retransmission has occured at the sender
> > (because it receives multiple copies).
> >
> > I think that is more robust than timestamps that may or may not be
> > included, might get lost, misbundled, associated with the wrong ack,
> > etc.
> >
> > With duplicates, the receiver says "I already received this TSN".
> > I agree with Kacheong here, but I disagree that it is slower.  In
> > the RFC receiving duplicates is reason for immediate SACK (whereas
> > timestamps would not).  Therefore, the dup TSNs are, I believe,
> > faster, in that the duplicate TSN report is immediate under most
> > situations.  (See 6.2 RFC 2960).
> >
> > Furthermore, the duplicate TSN mechanism is already exists, and is
> > already used to determine that a spurious restransmission has
> > occurred, possibly due to a lost SACK return path (the most obvious
> > cause of persistent spurious retransmissions).
>=20
> hmm..What about packet reordering, packet duplication, delay spikes..we
> seem to be missing some basic causes of spurious retransmissions.

I didn't mean that it couldn't be used for those, it just isn't at the
moment.  2960 mentions the more obvious, most injurious to SCTP (blindly
sending SACKs to a dead destination address that never receives DATA and
never suffers a timeout except at slower HEARTBEAT intervals.)

>=20
> >
> > Can someone back up the statement of timestamps being "more robust"?
>=20
> First, we seem to be using the terms Eifel and timestamps in the same
> context. This is a fundamental flaw. Are you asking the motivation behi=
nd
> timestamps or are you asking how Eifel using timestamps is more robust
> than the dup-tsn detection.
>=20
> I would guess the second..from the top of my head, I can think of a
> few advantages of Eifel over dup-tsn detection.
>=20
> * the dup-tsn notification comes much late in the game. You cannot avoi=
d a
> GoBackN (which is one of the motivations anyway for having a spurious
> retransmission detection scheme) if you had a spurious timeout. (I beli=
eve
> RFC3522 will have more details on these). Eifel will be able to detect
> possibly on the arrival of the first acceptable ACK if the retransmissi=
on
> was spurious.

Ok. I see that.  But I would say that it is "possible" that timestamps wi=
ll
detect faster, and not necessarily always or even often and certainly not
obviously the case.

>=20
> * the dup-tsn detection is sensitive to ACK loss: if you lost the ACK
> carrying the duptsn you possibly fail the detection.

Lost ACKs can cause Eifel to fail detection on timestamps too.

>=20
> * the duptsn notification is less robust to packet duplication (in
> particular for SCTP). I remember Reiner and I had noted this in some of
> the previous emails on a different thread: If the SCTP association is
> multihomed and the alternate path is duplicating packets then with the
> dup-tsn detection scheme coupled with a response algorithm may end up
> making an SCTP non-responsive to congestion.

Duplicating network paths are easy to detect: a TSN cannot have more
dup-TSns that it was retransmitted.

--brian

>=20
>=20
> -Sourabh
>=20

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

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Subject: Re: AW: [Tsvwg] SCTP Timestamps
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>> and (2) whether losing the additional benefits of timestamps
>> (RTTM
>
> Why do we loose RTTM? The timestamps (transmission and retransmission) are
> stored locally.

I think the whole purpose of using timestamp (even applicable for TCP)
rather than storing timestamps locally for RTTM was to avoid this added
state at the sender.

>
>> and the ability to distinguish not only between original
>> transmission and retransmission, but between an arbitrary number of
>> consecutive retransmissions as well
>
> Here I doubt that it makes sense to enhance the communication if more than
> one retransmission is needed for a significant number of packets.
>

I think I am missing something here? I think if we do intend to "enhance
communication" we do it correctly and for all possible cases we can,
rather than restrict ourselves to the case of "one retransmission". Note
that the consecutive retransmissions that Stephan was referring to can go
on different paths (destinations). I can still think (while using
retransmission bit) scenarios where an SCTP sender can break packet
conservation, do congestion window overgrowth and potentially harm the
network.


-Sourabh

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From: Schruefer Wolfgang <wolfgang.schruefer@siemens.com>
To: "'ladha@mail.eecis.udel.edu'" <ladha@mail.eecis.udel.edu>
Cc: "'Stephan Baucke'" <stephan.baucke@ericsson.com>, tsvwg@ietf.org
Subject: AW: [Tsvwg] SCTP Timestamps
Date: Fri, 18 Jul 2003 16:32:46 +0200
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> >> and (2) whether losing the additional benefits of timestamps
> >> (RTTM
> >
> > Why do we loose RTTM? The timestamps (transmission and 
> retransmission) are
> > stored locally.
> 
> I think the whole purpose of using timestamp (even applicable for TCP)
> rather than storing timestamps locally for RTTM was to avoid 
> this added
> state at the sender.

I am not sure what you mean by 'added state'.
Is it that you want to avoid a locally stored timestamp and instead you store it in the network by a Timstamp ping-pong?
When I compare the costs of both alternatives I can not imagine that a ping-pong is more effective (think of 4 times checksum calculation, reduced payload and net bandwith, ...). 

> 
> >
> >> and the ability to distinguish not only between original
> >> transmission and retransmission, but between an arbitrary number of
> >> consecutive retransmissions as well
> >
> > Here I doubt that it makes sense to enhance the 
> communication if more than
> > one retransmission is needed for a significant number of packets.
> >
> 
> I think I am missing something here? I think if we do intend 
> to "enhance
> communication" we do it correctly and for all possible cases we can,
> rather than restrict ourselves to the case of "one 
> retransmission". Note
> that the consecutive retransmissions that Stephan was 
> referring to can go
> on different paths (destinations). I can still think (while using
> retransmission bit) scenarios where an SCTP sender can break packet
> conservation, do congestion window overgrowth and potentially harm the
> network.

When I go at the beginning of the thread, Stephen showed up the problems:

> - Allowing for better and more frequent RTT measurements
> 
> - Resolving the "path ambiguity" in a multihomed association: If an ACK 
> arrives for a chunk that has been retransmitted one or more times on 
> alternating paths, it is not possible to safely determine which 
> destination the ACK should be credited to (this is briefly mentioned in 
> section 8.2 of RFC 2960). This may result in resetting the wrong error 
> counter (delaying the detection of an inactive path), or in unfair 
> behavior if the congestion window of the wrong path is adapted as a 
> result of the ACK.

>  - Resolving the retransmission ambiguity, i.e. enabling the sender to 
> determine whether an ACK refers to the original transmission or a 
> retransmission of a chunk. This is a necessary precondition for 
> implementing the Eifel algorithm for SCTP (which we are currently 
> looking at).


And all of the points are enhanced significantly by implementation of the retransmission bit, because a single path failure and a single retransmission can be handled 100% correct.
Also for multiple retransmissions the retransmission bit may prevent us from harming the network, because a very late SACK for an original Packet can be identified.
 
If it is considered that one bit is not to be enough, a second bit may be used. I just wanted to express that I think for the problem of spurious retransmissions one retransmission bit seems for me to be enough. 

Wolfgang



> 
> 
> -Sourabh
> 

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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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Subject: Re: AW: [Tsvwg] SCTP Timestamps
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On Fri, 18 Jul 2003, Schruefer Wolfgang wrote:

> If it is considered that one bit is not to be enough, a second bit may
> be used. I just wanted to express that I think for the problem of
> spurious retransmissions one retransmission bit seems for me to be
> enough.

I disagree. Remember that if the peer SCTP endpoint has multiple
destination IP addresses, each retransmission may go on a different path.

So you are saying that it is enough to only be able to detect spurious
retransmissions the first time, which only helps the cwnd and RTO of the
primary destination. If the first retransmission spuriously times out,
then we don't care according to you. We _may_ not care that the cwnd gets
cut unnecessarily, because the cwnd generally stays pretty small on
alternate paths. But, we DO care that the RTO is left doubled, because
there isn't much traffic on an alternate path to bring that RTO down
again. So the next time a timeout occurs (whether spurious or not) on that
alternate path, it will take that much longer.

If you are not following this argument, _please_ read at least section 3
of our paper. It's a short read. The full paper is only 5 pages.
http://www.cis.udel.edu/~acaro/papers/2003.milcom.acaro.pdf


~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] Slides for Eifel Response in Vienna IETF
Date: Fri, 18 Jul 2003 11:28:50 -0500
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As I was going through these slides, I didn't even see a single mention =
of DCLOR, which was the main reason why we had the discussion about =
appropriate congestion response and rate halving.  In fact, just a =
couple of months back, most people (including Reiner Ludwig) were not =
even willing to consider rate halving (I've put links to the e-mails =
that were exchanged on this issues).

The present DCLOR draft already does what the updated Eifel response =
will do (the present draft has not even been updated with these changes =
and the document is being asked for last call). Also, if I can recall, =
people who engaged in the discussion that time did agree to having two =
different response algorithm--one as described in the Eifel response =
--03 draft, and another one which will rate halve every RTO as in DCLOR.

Now, Eifel Response incorporates the work of DCLOR, and makes it look as =
if they are the ones who came with this scheme in the first place (Here =
is a link to Reiner's view in March about Rate Halving: =
http://www1.ietf.org/mail-archive/working-groups/tsvwg/current/msg03819.h=
tml). I had asked the working group chairs in the past about making =
DCLOR a working group draft, but I did not receive any response. To me =
all this seems a little strange, and disheartening.

--yogesh=20

Please follow the thread of this e-mail for the discussion.

http://www1.ietf.org/mail-archive/working-groups/tsvwg/current/msg03813.h=
tml


> -----Original Message-----
> From: ext Andrei Gurtov [mailto:gurtov@cs.helsinki.fi]
> Sent: 17 July, 2003 7:33 PM
> To: tsvwg@ietf.org; Reiner Ludwig
> Subject: [Tsvwg] Slides for Eifel Response in Vienna IETF
>=20
>=20
>=20
> Slides for the presentation on Eifel Response in the TSVWG meeting in=20
> Vienna IETF are available at:
>=20
> http://www.cs.helsinki.fi/u/gurtov/reiner/eifel_alg_ietf57.pdf
> http://www.cs.helsinki.fi/u/gurtov/reiner/eifel_alg_ietf57.ppt
>=20
> Also of possible interest is:
>=20
> A. Gurtov, R. Ludwig, Responding to Spurious Timeouts in TCP, IEEE=20
> INFOCOM'03, March 2003.
> http://www.cs.helsinki.fi/u/gurtov/papers/resp.html
>=20
> Reiner&Andrei
>=20
>=20
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>=20

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I aggregate several replies into one mail to save bandwidth...

Sourabh Ladha wrote:

>>From some of the emails I have seen in this thread, it seems folks have
> the wrong impression that timestamps are being proposed for "Eifel
> purpose".
> 
> I would like to clarify and REPEAT myself again that we were NOT
> discussing timestamps for Eifel purposes. Eifel happens to be
> ONE of the possible mechanisms which can use timestamps.

I guess what people have said so far is that the other purposes
are either not very beneficial or do not necessarily require the
use of timestamp.  That's why I try to concentrate on the one
issue which people think "definitely" require the use of timestamp.

Stephan Baucke wrote:

 > You have already mentioned one of the reasons: Detection of spurious
 > retransmissions using DSACK only works "after the fact", whereas
 > something like Eifel detects it early enough to avoid a go-back-n-like
 > behavior after a spurious timeout (also see RFC 3522).

How about Pasi's F-RTO?  I have read Sourabh's questions about
F-RTO.  But I think F-RTO can be modified, possible making use
of duplicate TSN, for SCTP to address those questions.

 > Also, DSACKs do
 > not address the other issues mentioned in this thread, such as the "path
 > ambiguity" and RTTM. Note that DSACKs are still useful and may be
 > combined with Eifel; I think there is a compelling case for introducing
 > optional timestamps even if DSACKs are already utilized.

I guess many people have pointed out that RTTM is not very
crucial and there are implementation techniques which can make
more RTT measurements without using timestamp.

For the issue of path ambiguity, I guess we need to first
understand why it happens.  And can it be avoided?  One case
mentioned is that a retransmission takes place in an alternate
path causing problem in cwnd growth.  We should first ask if
retransmitting using an alternate path is always a good thing.
There is a reason why RFC 2960 does not require (MUST) an
implementation to always retransmit using alternate path.  I
guess we should first decide when an alternate path be used for
retransmission to minimize the path ambiguity issue.

Another path ambiguity issue mentioned is changeover by the app.
A "simple" way to avoid that is for the implementation to withhold
sending packets before all outstanding packets to the original
primary destination address have been ack'ed.  This has the side
effect of discouraging an app to change often.  Granted, this "simple"
solution prevents a lot of "experimentation" on, for example,
load balancing and has obvious performance implication.  But I think
at this point, there is no solid proposal in making using of alternate 
paths other than retransmission.  So I believe it is OK for now.

I guess we should try to avoid the problems first before trying
to find a solution, timestamp in this case, to the problems.  And
it seems to me that we can actually minimize the chances of the
problems mentioned so far.  I don't see a compelling reason to
introduce a new chunk for timestamp.


---

						K. Poon.
						kacheong.poon@sun.com





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From: Stephan Baucke <stephan.baucke@ericsson.com>
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>> You have already mentioned one of the reasons: Detection of
>> spurious retransmissions using DSACK only works "after the fact",
>> whereas something like Eifel detects it early enough to avoid a
>> go-back-n-like behavior after a spurious timeout (also see RFC
>> 3522).
> 
> How about Pasi's F-RTO?  I have read Sourabh's questions about F-RTO.
> But I think F-RTO can be modified, possible making use of duplicate
> TSN, for SCTP to address those questions.

I have not really looked at F-RTO in depth, but seem to remember that it
only covers part of the cases that e.g. Eifel covers (perhaps Pasi can
comment on that?)

>> Also, DSACKs do not address the other issues mentioned in this
>> thread, such as the "path ambiguity" and RTTM. Note that DSACKs are
>> still useful and may be combined with Eifel; I think there is a
>> compelling case for introducing optional timestamps even if DSACKs
>> are already utilized.
> 
> I guess many people have pointed out that RTTM is not very crucial
> and

I am not convinced of that. While there is evidence questioning the
benefits of timing every packet for modest congestion window sizes 
(Allman/Paxson SIGCOMM '99), there is on the other hand evidence that it
can counter aliasing effects in case of large congestion windows (RFC
1323), and can help to reduce spurious timeouts in case of links with
highly variant RTTs, in particular wireless links (see e.g. RFC 3481,
section 4.8). Timestamps may also enable alternative RTO estimators in
the future.

> and there are implementation techniques which can make more RTT
> measurements without using timestamp.

But do these techniques also allow for the timing of retransmitted
packets? This still seems important to me, especially for obtaining a
better RTO on alternate paths. Without timestamps we would need some
other method to disambiguate retransmissions in order to achieve this.

> For the issue of path ambiguity, I guess we need to first understand
> why it happens.  And can it be avoided?  One case mentioned is that a
> retransmission takes place in an alternate path causing problem in
> cwnd growth.  We should first ask if retransmitting using an
> alternate path is always a good thing. There is a reason why RFC 2960
> does not require (MUST) an implementation to always retransmit using
> alternate path.  I guess we should first decide when an alternate
> path be used for retransmission to minimize the path ambiguity issue.
> 
I guess one could reduce the probability of the "path ambiguity" by
changing the SCTP retransmit policy, but nevertheless it seems sensible
to me to address problems that may arise from the standard policy
recommended by RFC 2960.

> Another path ambiguity issue mentioned is changeover by the app. A
> "simple" way to avoid that is for the implementation to withhold 
> sending packets before all outstanding packets to the original 
> primary destination address have been ack'ed.  This has the side 
> effect of discouraging an app to change often.  Granted, this
> "simple" solution prevents a lot of "experimentation" on, for
> example, load balancing and has obvious performance implication.  But
> I think at this point, there is no solid proposal in making using of
> alternate paths other than retransmission.  So I believe it is OK for
> now.

I think the performance implications would be serious. The failure
detection and failover mechanism are already too conservative for some
applications, and this would make them even more conservative.

> I guess we should try to avoid the problems first before trying to
> find a solution, timestamp in this case, to the problems.

I'll try to consolidate the issues:

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

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

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

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

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

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


References:
[1] RFC 1323

[2] RFC 3481 (and citations [45], [52], [54], [60] therein)

[3] M. Allman, V. Paxson, "On Estimating End-to-End Network Path
Properties", SIGCOMM'99

[4] Armando L. Caro Jr., Paul D. Amer, Randall R. Stewart, "Transport
Layer Multihoming for Fault Tolerance in FCS Networks", MILCOM 2003
http://www.cis.udel.edu/~acaro/papers/2003.milcom.acaro.pdf

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

[6] RFC 3522

[7] draft-ietf-tsvwg-dsack-use-00.txt

[8] draft-sarolahti-tsvwg-tcp-frto-04.txt





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Date: Fri, 18 Jul 2003 14:48:13 +0200
To: Mark Handley <M.Handley@cs.ucl.ac.uk>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: [Tsvwg] High-speed TCP
Cc: tsvwg@ietf.org, Scott Bradner <sob@harvard.edu>
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At 11:32 18.07.2003, Mark Handley wrote:

>In followup to today's TSVWG meeting, here's some suggested
>boilerplate text for the High-speed TCP Experimental RFC:
>
>   The proposals in this document are experimental.  We believe they
>   are safe for deployment in the current Internet, but they do not
>   represent a consensus that this is the best method for high-speed
>   congestion control.  In particular, we note that alternative
>   experimental proposals are likely to be forthcoming, and it is not
>   well understood how the proposals in this document will interact
>   with such alternative proposals.

I think this is good proposal, and I support it.

Just one question: Could anybody, please, explain to me the difference 
between an "experimental RFC with a recommendation for deployment (with or 
without a boilerplate about warnings)" and a "proposed standard RFC"?

Concrete example: Is the IETF recommending the deployment of the many 
TCP-related experimental RFCs (that don't have an explicit recommendation 
for deployment), or not?

I tried to find the answer in RFC2026, but wasn't quite clear. Also, I hear 
that the standards track definition of RFC2026 might not be current IETF 
believe anymore. See also draft-dawkins-pstmt-twostage-00.txt ...

///Reiner  


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Date: Mon, 21 Jul 2003 06:11:25 +0200
To: Yogesh.Swami@nokia.com
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: RE: [Tsvwg] Slides for Eifel Response in Vienna IETF
Cc: tsvwg@ietf.org
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 .nokia.com>
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At 18:28 18.07.2003, Yogesh.Swami@nokia.com wrote:
>Now, Eifel Response incorporates the work of DCLOR, and makes it look as 
>if they are the ones who came with this scheme in the first place

No.

Not to confuse things: the "new" thing about the Eifel Response is that the 
stored congestion control state is decayed every RTO. A simpler version of 
that idea had already been part of the original proposal described in our 
CCR 30/1 paper. It was even included in 
draft-ietf-tsvwg-tcp-eifel-alg-00.txt. However, after some debate, we 
decided to take it out in the following revisions of the I-Ds.

In further discussions, including those in which you were involved, we were 
again convinced to include this feature, but in a form similar to the CC 
state decay algorithm specified in RFC2861.

Still, a typo slipped into the slides we showed at the last IETF. To be 
really compliant with RFC2861, the slides should have said:
     Every RTO (non-backed-off !!) counting from 1st Timeout:
     pipe_prev <- max (3/4 * pipe_prev, SMSS)
instead of
     Every RTO (non-backed-off !!) counting from 1st Timeout:
     pipe_prev <- max (pipe_prev/2, SMSS)

///Reiner


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From: Schruefer Wolfgang <wolfgang.schruefer@siemens.com>
To: "'Armando L. Caro Jr.'" <acaro@acm.org>
Cc: "'ladha@mail.eecis.udel.edu'" <ladha@mail.eecis.udel.edu>,
        "'Stephan Baucke'" <stephan.baucke@ericsson.com>, tsvwg@ietf.org
Subject: RE: AW: [Tsvwg] SCTP Timestamps
Date: Mon, 21 Jul 2003 10:05:06 +0200
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Armando,

> > If it is considered that one bit is not to be enough, a 
> second bit may
> > be used. I just wanted to express that I think for the problem of
> > spurious retransmissions one retransmission bit seems for me to be
> > enough.
> 
> I disagree. Remember that if the peer SCTP endpoint has multiple
> destination IP addresses, each retransmission may go on a 
> different path.
> 
> So you are saying that it is enough to only be able to detect spurious
> retransmissions the first time, which only helps the cwnd and 
> RTO of the
> primary destination. If the first retransmission spuriously times out,
> then we don't care according to you. We _may_ not care that 
> the cwnd gets
> cut unnecessarily, because the cwnd generally stays pretty small on
> alternate paths. But, we DO care that the RTO is left doubled, because
> there isn't much traffic on an alternate path to bring that RTO down
> again. So the next time a timeout occurs (whether spurious or 
> not) on that
> alternate path, it will take that much longer.
> 
> If you are not following this argument, _please_ read at 
> least section 3
> of our paper. It's a short read. The full paper is only 5 pages.
> http://www.cis.udel.edu/~acaro/papers/2003.milcom.acaro.pdf


I studied your paper with high interest, but my conclusion is that a mirrored Retransmission Flag would be a perfect solution for the scenario you describe in the paper.

With the mirrored Retransmission Flag you _can_ recalculate RTO on the second path. You just can not do it for the third or further paths. And here I think that the third path (resp. the a RTO calculation for the third transmission) may not be so important to improve.

Below I copied an extract from your paper, that could be witten similar for a Chunk Flag solution, but without the expense of additional overhead. 

>>>... Solution 3 introduces timestamps into each packet, thus allowing
a sender to disambiguate original transmissions from retransmissions.
By removing retransmission ambiguity, Karn's
algorithm can be eliminated, and successful retransmissions on the alternate path can be used to update the RTT estimate
and keep the RTO value more accurate. Solution 3 provides
more samples for alternate destination(s) to update their RTT
estimate at the expense of additional overhead in each packet...<<<

Wolfgang



  

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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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Subject: RE: AW: [Tsvwg] SCTP Timestamps
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On Mon, 21 Jul 2003, Schruefer Wolfgang wrote:

> > If you are not following this argument, _please_ read at
> > least section 3
> > of our paper. It's a short read. The full paper is only 5 pages.
> > http://www.cis.udel.edu/~acaro/papers/2003.milcom.acaro.pdf
>
>
> I studied your paper with high interest, but my conclusion is that a
> mirrored Retransmission Flag would be a perfect solution for the
> scenario you describe in the paper.
>
> With the mirrored Retransmission Flag you _can_ recalculate RTO on the
> second path. You just can not do it for the third or further paths.
> And here I think that the third path (resp. the a RTO calculation for
> the third transmission) may not be so important to improve.

Well, your one bit solution only works if the sender follows the
recommended mechanism for retransmissions. There hasn't been a consensus
that retransmitting on an alternate path is always the best policy. In
fact, I would argue that it isn't. I believe that for better performance
without affecting failover performance, fast retransmits should go to the
same destination and timeout retransmits should go to an alternate
destination [1].  With such a retransmission policy, one bit would
not be enough. Sure we could use 2 bits, but timestamps just seems
cleaner.

[1] http://www.cis.udel.edu/~acaro/papers/2003.icon.acaro.pdf

~armando

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


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From: Pasi Sarolahti <sarolaht@cs.Helsinki.FI>
To: ladha@mail.eecis.udel.edu
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Hi Sourabh,

Sorry for the delayed response. I have been behind a poor network access,
so my mail reactions are a bit slow currently :-) For the same reason I 
may not have read all the other mails in this thread.

On Thu, 17 Jul 2003 ladha@mail.eecis.udel.edu wrote:

> I had read the FRTO document some time back. On the SCTP front I was
> wondering if the following are applicable:

First, I have to be honest, we have mainly considered F-RTO as a TCP
algorithm and it was only recently when we started thinking that it should
apply on SCTP also. It seems that the special features of SCTP may need to
be discussed a bit more than the draft currently does.

Thanks for bringing up these good points, that definitely need to be
addressed. Nevertheless, I'm still optimistic that the principle of F-RTO
should work with SCTP, too: after an RTO, if an acknowledgement arrives
for a data chunk (or segment) that was sent before the RTO, the
RTO can be declared spurious.

My takes on the following (Remember, I don't have much SCTP knowledge):

> * Multihoming: If the delay spike on the primary path is greater than (1
> RTO(primary) + 2 RTT(alternate)), I would believe that FRTO will not be
> able to detect a timeout spurious. (complying with the current spec that
> retransmissions SHOULD go to alternate destination)

So, the RTO retransmission goes to alternate path and ACK for the
retransmission comes back from the alternate path. Then, the F-RTO sender
transmits one or two chunks that were not transmitted earlier. These
segments should go to primary destination, since they are not
retransmissions, right? If they do, then everything goes as intended,
since the segments will be queued behind other segments that were sent to
the primary address.  In case of spurious timeout, next ACK or SACK should
acknowledge something that was not retransmitted. Maybe I missed something
here?

How would this go with timestamps? ACK (and tsecr) of the retransmission 
will arrive to the sender from the alternate destination before the ACK of 
the original segment, right? So Eifel would not detect a spurious RTO?

> * Chunk Bundling: FRTO assumes that on a timeout the sender will
> retransmit only one segment("I think chunk will be appropriate in SCTP
> terminology"). This may not be true in SCTP. If the association is
> multihomed and the PMTU's on the primary and alternate path differ by a
> factor of two or more then, then I think the sender may be able to
> retransmit more than one unacknowledged chunk on a timeout (bundled in a
> single packet). This can also be true if the app is writing small chunks
> such that on a timeout more than one of such chunks can be bundled in a
> single packet and retransmitted. I remember having noticed a couple of
> side effects of this which could lead FRTO to declare the timeout spurious
> while it was not.

According to the principle I gave above, only acknowledgements of data
chunks (or segment) that were not transmitted after RTO should be used
for declaring the RTO spurious. Chunk bundling is something I did not
consider earlier, so probably this is something that needs to be
explicitly stated when applying the algorithm for SCTP. I think we could
add some kind of "SCTP considerations" section that would address these
kind of issues, plus probably reword the algorithm description to be more
clear on these.

I would be interested to hear about the side effects you mention above in
more detail. False positives are something we definitely want to avoid.  

> A question I have is:
> Section 3- step 3(b): When you refer to an ACK which acknowledges data
> (above spurious threshhold or previously unacknowledged), do you refer to
> acknowledging {via cum-ack point} or {via SACK blocks} or {either of these
> two}?

either of the two. This is worth clarifying in the algorithm. Thanks for 
pointing it out.

- Pasi



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From: Sally Floyd <floyd@icir.org>
Subject: Re: [Tsvwg] High-speed TCP 
Date: Mon, 21 Jul 2003 11:48:31 -0700
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>Just one question: Could anybody, please, explain to me the difference 
>between an "experimental RFC with a recommendation for deployment (with or 
>without a boilerplate about warnings)" and a "proposed standard RFC"?

Actually, my request is not for High-speed TCP to be considered as
an experimental RFC with a recommendation for deployment, but as
an experimental RFC that is judged as safe for deployment, that is,
that someone could deploy (in their own computer, or in their
widely-distributed commercial TCP distribution) *if they wanted
to*.  My own view would be that the likely audience for such
high-congestion-window variants of TCP is small, at the moment, and
that for most TCP users it wouldn't make much different one way or
another whether they used Standard or HighSpeed TCP.

This would have the same formal status of NewReno TCP, RFC 2582,
April 1999, which is also Experimental.  At the time, in 1999, my
memory is that people wanted more operational experience with NewReno
before considering it for Proposed Standard (and at the time, I
thought that there might also be other proposals for NewReno-type
variants of TCP, for those cases where the other end was not
SACK-capable, that would also be Experimental).  It seems plausible
to me that some major changes to TCP pass thought a period of
Experimental before moving on the Proposed Standard, giving the
role of TCP in the Internet.

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


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From: Sally Floyd <floyd@icir.org>
Subject: Re: [Tsvwg] High-speed TCP 
Date: Mon, 21 Jul 2003 12:17:13 -0700
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>In followup to today's TSVWG meeting, here's some suggested
>boilerplate text for the High-speed TCP Experimental RFC:
>
>  The proposals in this document are experimental.  We believe they
>  are safe for deployment in the current Internet, but they do not
>  represent a consensus that this is the best method for high-speed
>  congestion control.  In particular, we note that alternative
>  experimental proposals are likely to be forthcoming, and it is not
>  well understood how the proposals in this document will interact
>  with such alternative proposals.  

That sounds good to me, and I will include it in a revised version
that I submit later this week.  

The only other thing in the revision will be a paragraph later in
the document saying that I don't expect HighSpeed TCP to interact
any more poorly with alternative experimental proposals that would
the N parallel TCP connections commonly used today in the absence
of HighSpeed TCP.  That is, I don't believe that a deployment of
HighSpeed TCP would present any block to the later deployment of
Scalable TCP, FAST TCP, XCP, or any of the other alternative
experimental proposals.

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

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On Monday, July 21, 2003, at 09:35 AM, Armando L. Caro Jr. wrote:
>>
>
> Well, your one bit solution only works if the sender follows the
> recommended mechanism for retransmissions. There hasn't been a 
> consensus
> that retransmitting on an alternate path is always the best policy. In
> fact, I would argue that it isn't. I believe that for better 
> performance
> without affecting failover performance, fast retransmits should go to 
> the
> same destination and timeout retransmits should go to an alternate
> destination [1].  With such a retransmission policy, one bit would
> not be enough. Sure we could use 2 bits, but timestamps just seems
> cleaner.
>
> [1] http://www.cis.udel.edu/~acaro/papers/2003.icon.acaro.pdf
>
> ~armando
>

In the paper cited, you fault the default SCTP behavior for assuming
"that loss indicates either that the destination address used is 
unreachable,
or the destination's network path is congested."

The implication is that the end-to-end algorithm is "giving up too 
quickly"
on a path, and thereby degrading performance.

As that I have seen this objection most often raised when the final hop
is a mobile and/or wireless link, I am tempted to turn that question 
around.

Isn't the real problem that the last hop is giving up on L2 delivery too
quickly, and creating an L3/end-to-end problem for something that
should have been handled locally?

Basically, is each downstream frequency *really* a path? Or should
the collective set of frequencies be viewed as a single L3 path
over which there may be L2 retries?

I designed a two-way CATV system to carry IP traffic through
set-top decoders while I was at Zenith Electronics. Whenever
the set-top needed to change frequency it would be deaf to
the headend for a short period. Realizing that dropping
packets during such a frequency shift would lead to very
poor end-to-end performance, we simply scheduled the
packet to be L2-delivered on a frequency as soon as it
was known, or on all frequencies if the receiver's status
was not known.

We certainly do not want to begin an L2 design debate
here, but I think it is a valid question -- is it a valid assumption
that L2 delivery should be dealing with truly transitory
problems and therefore packet loss really should be
taken to as an indication that the destination address used
is unreachable or the destination's network path is
congested?


Caitlin Bestler - cait@asomi.com - http://asomi.com/


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From exim@www1.ietf.org  Mon Jul 21 17:05:58 2003
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From: Sally Floyd <floyd@icir.org>
Date: Mon, 21 Jul 2003 14:03:37 -0700
Subject: [Tsvwg] draft-ietf-tsvwg-highspeed-00.txt
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Allison and Jon -

I have submitted draft-ietf-tsvwg-highspeed-00.txt and
draft-ietf-tsvwg-slowstart-00.txt to the internet-drafts
editor, and there are local copies on the HSTCP web page
at: 
  http://www.icir.org/floyd/hstcp.html
i.e.,
  http://www.icir.org/floyd/papers/draft-ietf-tsvwg-highspeed-00.txt
  http://www.icir.org/floyd/papers/draft-ietf-tsvwg-slowstart-00.txt

I think these are ready for Working Group Last Call to be
considered for Experimental status.

- Sally

--------------------------------------------------------------------
The diffs for draft-ietf-tsvwg-highspeed-00.txt are below, just FYI:
--------------------------------------------------------------------

...
32c32
< .ds *FN draft-floyd-tcp-highspeed-03
---
> .ds *FN draft-ietf-tsvwg-highspeed-00
...
125a126,139
> Status of this Memo
> .NORMRS
> .LP
> 
> The proposals in this document are experimental.  We believe they
> are safe for deployment in the current Internet, but they do not
> represent a consensus that this is the best method for high-speed
> congestion control.  In particular, we note that alternative
> experimental proposals are likely to be forthcoming, and it is not
> well understood how the proposals in this document will interact
> with such alternative proposals.  
> 
> .NORMRE
> .SH
176a191,198
> Changes from draft-floyd-tcp-highspeed-03.txt:
> 
> Added the section on "Status of this Memo".
> 
> Added a paragraph to the end of the section on "Deployment issues
> of HighSpeed TCP" about possible interactions between HighSpeed TCP
> and other alternative experimental proposals.
> 
1204a1227,1235
> We do not believe that the deployment of HighSpeed TCP would
> serve as a block to the possible deployment of alternate experimental
> protocols for high-speed congestion control, such as
> Scalable TCP, XCP [KHR02], or FAST TCP [JWL03].
> In particular, we don't expect HighSpeed TCP to interact
> any more poorly with alternative experimental proposals that would
> the N parallel TCP connections commonly used today in the absence
> of HighSpeed TCP. 
> 
1518a1550,1554
> [JWL03] Cheng Jin, David X. Wei and Steven H. Low,
> FAST TCP for High-speed Long-distance Networks,
> internet-draft
> draft-jwl-tcp-fast-01.txt, work-in-progress, June 2003.
> 
1521a1558,1561
> 
> [KHR02] Dina Katabi, Mark Handley, and Charlie Rohrs,
> Congestion Control for High Bandwidth-Delay Product Networks,
> SIGCOMM 2002.

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From: Sourabh Ladha <ladha@mail.eecis.udel.edu>
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Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
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Hi Pasi,

Before sending a reply for FRTO-SCTP, I would like to get a bit of
clarification for Step 2 of the SACK-enhanced version of FRTO
(this applies for TCP as well):

"   2) The first acknowledgement after RTO arrives at the sender.

      a) if the cumulative ACK acknowledges all segments up to send_high
         stored in algorithm step 1, the TCP sender SHOULD revert to the
         conventional RTO recovery and it MUST set congestion window to
         no more than 2 * MSS. The sender does not enter step 3 of this
         algorithm.

      b) otherwise, the TCP sender MAY transmit two new segments. If the
         TCP sender does not transmit any previously unsent data, it
         MUST NOT enter step 3 of this algorithm, but revert to the
         conventional RTO recovery."

So I believe that even on the reciept of a dup-ack (with a SACK block) an
FRTO sender follows step 2(b)?

If so, In the event of loss (xxlost) coupled with a delay spike (/\):

A				    B
----------TSN101------>xxlost
----------TSN102--------------/\--->
----------TSN103--------------/\--->
  .
  .
timeout
----------TSN101------------------->
<-----SACK101 (sack block: 102)---(sack generated when TSN 102 reached B)
FRTO sends out TSN104 and TSN105 and enters step 3.
<-----SACK101 (sack block: 103)---(sack generated when TSN 103 reached B)

From your clarification of Step 3(b) below, FRTO will declare the timeout
spurious on the receipt of this SACK?

A false positive?


-Sourabh


> > Section 3- step 3(b): When you refer to an ACK which acknowledges data
> > (above spurious threshhold or previously unacknowledged), do you refer to
> > acknowledging {via cum-ack point} or {via SACK blocks} or {either of these
> > two}?
>
> either of the two. This is worth clarifying in the algorithm. Thanks for
> pointing it out.
>
> - Pasi
>
>
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>


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Dear Reiner,

I'm honored that someone has tripped over our proposal, but it's
just a proposal! I can say that there's some interest in looking at
the standards levels - Scott Bradner is also looking at this -
but 2026 is 2026 for now...

... and feedback is always welcome ...

Spencer

----- Original Message ----- 
From: "Reiner Ludwig" <Reiner.Ludwig@ericsson.com>
To: "Mark Handley" <M.Handley@cs.ucl.ac.uk>
Cc: <tsvwg@ietf.org>; "Scott Bradner" <sob@harvard.edu>
Sent: Friday, July 18, 2003 7:48 AM
Subject: Re: [Tsvwg] High-speed TCP


> At 11:32 18.07.2003, Mark Handley wrote:
>
> >In followup to today's TSVWG meeting, here's some suggested
> >boilerplate text for the High-speed TCP Experimental RFC:
> >
> >   The proposals in this document are experimental.  We believe they
> >   are safe for deployment in the current Internet, but they do not
> >   represent a consensus that this is the best method for high-speed
> >   congestion control.  In particular, we note that alternative
> >   experimental proposals are likely to be forthcoming, and it is not
> >   well understood how the proposals in this document will interact
> >   with such alternative proposals.
>
> I think this is good proposal, and I support it.
>
> Just one question: Could anybody, please, explain to me the difference
> between an "experimental RFC with a recommendation for deployment (with or
> without a boilerplate about warnings)" and a "proposed standard RFC"?
>
> Concrete example: Is the IETF recommending the deployment of the many
> TCP-related experimental RFCs (that don't have an explicit recommendation
> for deployment), or not?
>
> I tried to find the answer in RFC2026, but wasn't quite clear. Also, I
hear
> that the standards track definition of RFC2026 might not be current IETF
> believe anymore. See also draft-dawkins-pstmt-twostage-00.txt ...
>
> ///Reiner


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From: Pasi Sarolahti <sarolaht@cs.Helsinki.FI>
To: Sourabh Ladha <ladha@mail.eecis.udel.edu>
cc: "Armando L. Caro Jr." <acaro@acm.org>, tsvwg@ietf.org
Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
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Hi,

On Mon, 21 Jul 2003, Sourabh Ladha wrote:

[...]
> So I believe that even on the reciept of a dup-ack (with a SACK block) an
> FRTO sender follows step 2(b)?

Yes, and it appears to be wrong, as you presented. The basic algorithm
would not do this and it seems that the SACK-based variant should also
wait for acknowledgement for the RTO-retransmitted segment before
proceeding.

> If so, In the event of loss (xxlost) coupled with a delay spike (/\):
> 
> A				    B
> ----------TSN101------>xxlost
> ----------TSN102--------------/\--->
> ----------TSN103--------------/\--->
>   .
>   .
> timeout
> ----------TSN101------------------->
> <-----SACK101 (sack block: 102)---(sack generated when TSN 102 reached B)
> FRTO sends out TSN104 and TSN105 and enters step 3.
> <-----SACK101 (sack block: 103)---(sack generated when TSN 103 reached B)
> 
> From your clarification of Step 3(b) below, FRTO will declare the timeout
> spurious on the receipt of this SACK?
> 
> A false positive?

Oops... You are right. It seems to be essential to get acknowledgement for
RTO-retransmitted segment before going to step 3. The question is, should
the sender wait (in step 2) until the ack arrives (or until another RTO
occurs), or should it cancel F-RTO immediately on dupack. The basic
algorithm in Section 2 does the latter, but I am wondering if the former
is safe enough alternative. It works better if reordering occurs at the
same time.

Ok, now if the connection is multihomed and delay spike is longer than RTO
(primary) + 2 RTT (alternative) there could be problems with this
scenario. I need to think about this more, but it seems that in multihomed
cases the use of F-RTO needs to limited a bit. It detects a delay spike,
but it may not always detect if the first segment is lost, if the
retransmission goes to different destination than the original, and the
path characteristics are as shown above. However, combining with duplicate
TSNs should work: F-RTO can be used to avoid unnecessary retransmissions
and draft-dsack-use can be used for congestion window undoing.

..or you could use timestamps, once specified. But back to the earlier
mail, wouldn't Eifel have some kind of problems in detecting a real
spurious RTO if retransmissions go to alternative path such as shown
above?

Thanks,
- Pasi


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From: Pasi Sarolahti <sarolaht@cs.Helsinki.FI>
To: Stephan Baucke <stephan.baucke@ericsson.com>
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On Sun, 20 Jul 2003, Stephan Baucke wrote:

> > How about Pasi's F-RTO?  I have read Sourabh's questions about F-RTO.
> > But I think F-RTO can be modified, possible making use of duplicate
> > TSN, for SCTP to address those questions.
> 
> I have not really looked at F-RTO in depth, but seem to remember that it
> only covers part of the cases that e.g. Eifel covers (perhaps Pasi can
> comment on that?)

Yep. F-RTO needs to send in advance at least some bits of unsent data
after the RTO retransmission, which is impossible if 1) sender/receiver
window limits the transmission or 2) there are no unsent data to transmit.
[For case 2) I would argue that detecting spurious RTO is not highly
important, because at least the user won't see the difference.]

These problems are identified in the draft, and there we listed the known
tricks for overcoming them. Some of the solutions may be controversial
(like sending from the tail of the retransmission queue), so we have left
applying them for implementator's decision and separated them from the
algorithm description.

And after the examples shown by Sourabh, I believe detecting spurious
RTOs with multihoming brings interesting questions for both F-RTO and
Eifel that I need to study more thoroughly.

- Pasi


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Date: Tue, 22 Jul 2003 07:15:04 -0400 (EDT)
Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
To: "Pasi Sarolahti" <sarolaht@cs.Helsinki.FI>
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Hi Pasi,

> ..or you could use timestamps, once specified. But back to the earlier
> mail, wouldn't Eifel have some kind of problems in detecting a real
> spurious RTO if retransmissions go to alternative path such as shown
> above?

You are right. I did stumble across this issue. Eifel "for TCP" if ported
"as it is" to SCTP did present these cases where in a multihomed scenario
(with such a setting as described in the earlier mail), Eifel did not
detect a spurious timeout when it was. But the vice-versa was not true,
i.e Eifel did not detect a loss as a spurious timeout. (Note that the case
is less pronounced in the event of a spurious fast-retransmit). However as
you also pointed out, integrating with the dup-tsn notifiation adds some
robustness.


Thanks,
Sourabh

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Subject: [Tsvwg] HS-TCP/FAST-TCP: Kind of Packet Loss?
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As I understand, the motivation given for HS-TCP and FAST-TCP is that since 
the required cwnd to fill the pipe is so high, it would require 
unrealistically low rates of non-congestion-related packet losses.

Now, I wonder whether anybody has studied what kind of 
non-congestion-related packet losses we see in the target regime of 10 
Gbit/s and beyond ...

Do such optical fiber links still do CRC, i.e., are the 
non-congestion-related packet losses mostly due to packets being discarded 
at the link layer? Or are maybe many of such non-congestion-related packet 
losses due to TCP checksum errors, and thus the TCP receiver silently 
discarding the packet?

If it was the latter, one solution could be the definition of an explicit 
TCP NACK that the TCP sender could use to suppress the reduction of cwnd. 
Clearly, this would have some security considerations, but I guess those 
could be fixed, e.g., with a nonce.

Maybe this would be a good reason to disbale CRC on such optical fiber links.

Did I miss something?

///Reiner


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From: Schruefer Wolfgang <wolfgang.schruefer@siemens.com>
To: "'Armando L. Caro Jr.'" <acaro@acm.org>
Cc: "'ladha@mail.eecis.udel.edu'" <ladha@mail.eecis.udel.edu>,
        "'Stephan Baucke'" <stephan.baucke@ericsson.com>, tsvwg@ietf.org
Subject: RE: AW: [Tsvwg] SCTP Timestamps
Date: Tue, 22 Jul 2003 14:48:14 +0200
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Armando,

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

I disagree that the one bit solution works dependent on the mechanism used for retransmission.
The sender is free how to set the Bit. If you like to perform a fast retransmit on the primary path and you want accurate RTT calculation for the alternate path, just set the Bit for the transmission on the alternate path. And the Bit would again solve your problem.

Please see the Mirrored Chunk Flag solution as a cheap tool for any sender to enhance the transmission for its special purpose.

A timestamp solution seems over-engineered to me and I suspect it is used by TCP just because no space for a Header Flag was available (????).

E.g. I can not imagine that a 12 Byte timestamp chunk would be favoured for the usage in Telephone Signaling Systems, when the average packet length is probably only 50 Bytes.
  

Wolfgang

 



 

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From: John Heffner <jheffner@psc.edu>
To: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
cc: <tsvwg@ietf.org>
Subject: Re: [Tsvwg] HS-TCP/FAST-TCP: Kind of Packet Loss?
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On Tue, 22 Jul 2003, Reiner Ludwig wrote:

> As I understand, the motivation given for HS-TCP and FAST-TCP is that since
> the required cwnd to fill the pipe is so high, it would require
> unrealistically low rates of non-congestion-related packet losses.
>
> Now, I wonder whether anybody has studied what kind of
> non-congestion-related packet losses we see in the target regime of 10
> Gbit/s and beyond ...
>
> Do such optical fiber links still do CRC, i.e., are the
> non-congestion-related packet losses mostly due to packets being discarded
> at the link layer? Or are maybe many of such non-congestion-related packet
> losses due to TCP checksum errors, and thus the TCP receiver silently
> discarding the packet?
>
> If it was the latter, one solution could be the definition of an explicit
> TCP NACK that the TCP sender could use to suppress the reduction of cwnd.
> Clearly, this would have some security considerations, but I guess those
> could be fixed, e.g., with a nonce.
>
> Maybe this would be a good reason to disbale CRC on such optical fiber links.
>
> Did I miss something?


I think packet corruption is not currently the greatest problem.  There
are a number of other causes of packet loss, including:
  - intermittent flows starting up (punching holes with slow-start)
  - misbehaving switches
  - burstiness, and a small router buffer

When these things cause cwnd to be halved on the order of once every few
minutes, they can seriously degrade the performance of high-BDP flows.  A
faster response time can really help, and I think is ultimately necessary
if we expect end-to-end congestion control to work in messy networks at
large window sizes.

  -John



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Date: Tue, 22 Jul 2003 12:07:53 -0700
From: Kacheong Poon <kacheong.poon@sun.com>
Organization: Sun Microsystems, Inc.
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To: tsvwg@ietf.org
Subject: Re: [Tsvwg] F-RTO and SCTP (was: SCTP Timestamps)
References: <Pine.LNX.4.44.0307221304550.3045-100000@melkki.cs.Helsinki.FI>
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Pasi Sarolahti wrote:

 > And after the examples shown by Sourabh, I believe detecting spurious
 > RTOs with multihoming brings interesting questions for both F-RTO and
 > Eifel that I need to study more thoroughly.

I guess after all the discussions so far, SCTP implementors should
think about not to follow what is recommended in RFC 2960, timeout
retransmission to an alternate address.  Just a suggestion, it seems
that at least for the first timeout, the retransmission is probably
better sent to the original address.  And when this first retransmission
is also lost leading to another timeout, SCTP should start considering
using an alternate address.

-- 

						K. Poon.
						kacheong.poon@sun.com



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From: Joe Touch <touch@ISI.EDU>
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Subject: Re: [Tsvwg] High-speed TCP
References: <9147.1058520749@eve.cs.ucl.ac.uk>
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Mark Handley wrote:
> In followup to today's TSVWG meeting, here's some suggested
> boilerplate text for the High-speed TCP Experimental RFC:
> 
>   The proposals in this document are experimental.  We believe they
>   are safe for deployment in the current Internet,

Why is this believed? The document states fairly clearly that this sort 
of TCP will receive a disproportionate share of a link if competing with 
conventional TCP.

So far, the discussion seems to qualify 'safe' as 'safe in a closed 
environment' or 'safe if used primarily over very high speed links'. 
That's not particularly 'safe', since these criteria are external to the 
protocol.

Joe


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From: Mark Handley <M.Handley@cs.ucl.ac.uk>
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To: Joe Touch <touch@ISI.EDU>
cc: tsvwg@ietf.org
Subject: Re: [Tsvwg] High-speed TCP 
In-reply-to: Your message of "Wed, 23 Jul 2003 13:41:22 PDT."
             <3F1EF2F2.8020706@isi.edu> 
Date: Wed, 23 Jul 2003 22:06:03 +0100
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>Mark Handley wrote:
>> In followup to today's TSVWG meeting, here's some suggested
>> boilerplate text for the High-speed TCP Experimental RFC:
>> 
>>   The proposals in this document are experimental.  We believe they
>>   are safe for deployment in the current Internet,
>
>Why is this believed? The document states fairly clearly that this sort 
>of TCP will receive a disproportionate share of a link if competing with 
>conventional TCP.

Two points:

 - this was my suggested text, sent to the list for critique.
 - the opinion represented the consensus in the room at the IETF.

Personally I feel it's "safe" in the sense that it will achieve a
disproportionate share only under circumstances under which a
conventional TCP is unable to effectively use the available bandwidth.

But I'm not one of the main advocates of this particular draft, so I'm
not the person to convince otherwise.

Cheers,
	Mark

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From exim@www1.ietf.org  Wed Jul 23 17:17:29 2003
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To: John Heffner <jheffner@psc.edu>
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From: Sally Floyd <floyd@icir.org>
Subject: Re: [Tsvwg] HS-TCP/FAST-TCP: Kind of Packet Loss? 
Date: Wed, 23 Jul 2003 14:16:16 -0700
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>I think packet corruption is not currently the greatest problem.  There
>are a number of other causes of packet loss, including:
>  - intermittent flows starting up (punching holes with slow-start)
>  - misbehaving switches
>  - burstiness, and a small router buffer
>
>When these things cause cwnd to be halved on the order of once every few
>minutes, they can seriously degrade the performance of high-BDP flows.  A
>faster response time can really help, and I think is ultimately necessary
>if we expect end-to-end congestion control to work in messy networks at
>large window sizes.

Yep!  I agree that packet corruption is not the motivating problem here.

- Sally


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From exim@www1.ietf.org  Wed Jul 23 17:49:31 2003
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From: Matt Mathis <mathis@psc.edu>
To: Joe Touch <touch@ISI.EDU>
cc: Transport WG <tsvwg@ietf.org>
Subject: Re: [Tsvwg] High-speed TCP
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There is general agreement that at very large windows (e.g. more than 1000 
packets in flight) TCP is way too sensitive to packet loss, making it 
unreasonably difficult to get line rate on long fast networks.

Fixing this excessive sensitivity to loss intrinsically requires redefining TCP 
friendly.  It is universally understood that during any such transition there 
are likely to be situations where new TCP and old TCP do not play as nicely as 
one would hope.   In the case of Sally's High-speed TCP, we understand the 
likely worse case bound on the unfairness.  In reality, stock TCP does so 
poorly in most of these situations that "fairness" isn't even the point....

The real question that the WG discussed had to do with the possibility of a
partial interim deployment of Sally's HS-TCP causing deployment problems for a
future algorithm, such as XCP or FAST.  Note the real question is not if the
protocols play nicely, but if the users will play nicely.

It would be a_very_bad_thing if HS-TCP proved to be very unfair to
the_right_answer(TM) and at the same time came to be in wide use by innocent
users (say because some large vendor shipped it turned on by default).

As long as HS-TCP is off by default, it will require a certain level of user
awareness to turn it on.  Such users are likely to be aware of the_right_answer
when it becomes available.  Just a moderate level of education will be
sufficient to prevent chronic problems in the (presumably small) regime where
HS-TCP might have an unfair advantage over the_right_answer....

The discussion (and hummed consensus) at the WG was about balancing the value 
of an interim solution and various risks.   Most of the worry seemed to be 
about risk to a future right answer, and not about current TCP.   In any case 
the hums were not even close.

Good enough?

Thanks,
--MM--


On Wed, 23 Jul 2003, Joe Touch wrote:

> Mark Handley wrote:
> > In followup to today's TSVWG meeting, here's some suggested
> > boilerplate text for the High-speed TCP Experimental RFC:
> > 
> >   The proposals in this document are experimental.  We believe they
> >   are safe for deployment in the current Internet,
> 
> Why is this believed? The document states fairly clearly that this sort 
> of TCP will receive a disproportionate share of a link if competing with 
> conventional TCP.
> 
> So far, the discussion seems to qualify 'safe' as 'safe in a closed 
> environment' or 'safe if used primarily over very high speed links'. 
> That's not particularly 'safe', since these criteria are external to the 
> protocol.
> 
> Joe
> 
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 

-- 
--MM--
-------------------------------------------
Matt Mathis      http://www.psc.edu/~mathis
Work:412.268.3319    Home/Cell:412.654.7529
-------------------------------------------





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Matt Mathis wrote:

> There is general agreement that at very large windows (e.g. more than 1000 
> packets in flight) TCP is way too sensitive to packet loss, making it 
> unreasonably difficult to get line rate on long fast networks.
 >
> Fixing this excessive sensitivity to loss intrinsically requires redefining TCP 
> friendly. It is universally understood that during any such transition there
> are likely to be situations where new TCP and old TCP do not play as nicely as 
> one would hope.   In the case of Sally's High-speed TCP, we understand the 
> likely worse case bound on the unfairness.  In reality, stock TCP does so 
> poorly in most of these situations that "fairness" isn't even the point....

There are cases where it is possible that stock TCP is used over short 
fast links and this would be needed for longer paths. Using both on the 
same enterprise could cause significant unfairness, in ways that are not 
considered a reasonable interim step.

> The real question that the WG discussed had to do with the possibility of a
> partial interim deployment of Sally's HS-TCP causing deployment problems for a
> future algorithm, such as XCP or FAST.  Note the real question is not if the
> protocols play nicely, but if the users will play nicely.
> 
> It would be a_very_bad_thing if HS-TCP proved to be very unfair to
> the_right_answer(TM) and at the same time came to be in wide use by innocent
> users (say because some large vendor shipped it turned on by default).
> 
> As long as HS-TCP is off by default, it will require a certain level of user
> awareness to turn it on.  Such users are likely to be aware of the_right_answer
> when it becomes available.  Just a moderate level of education will be
> sufficient to prevent chronic problems in the (presumably small) regime where
> HS-TCP might have an unfair advantage over the_right_answer....

As above, using this protocol over a HS-path might interact very badly 
with other streams that share part but not all of the path, e.g., 
conventional TCP at high speeds near the endpoints (which have a small 
window and no need for this sol'n).

> The discussion (and hummed consensus) at the WG was about balancing the value 
> of an interim solution and various risks.   Most of the worry seemed to be 
> about risk to a future right answer, and not about current TCP.   In any case 
> the hums were not even close.
> 
> Good enough?

I'd be more comfortable with a statement that doesn't so broadly endorse 
experimenting with it on the Internet as a whole without some caveats, 
notably those which appear to be implied.

I.e.,
	- off by default
	- turned on only on applicable paths or environments
	- it is the responsibility of those using this protocol to
	determine whether it is affecting others, and when such effects
	are found, to disable it until they can be corrected

These may be implicit in the concept of 'experimental', but if we're 
saying it's 'safe', it ought to be clear about under what conditions.

Joe



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Pasi,

I had read the FRTO document some time back. On the SCTP front I was wondering if the following are applicable:

* Multihoming: If the delay spike on the primary path is greater than (1 RTO(primary) + 2 RTT(alternate)), I would believe that FRTO will not be able to detect a timeout spurious. (complying with the current spec that retransmissions SHOULD go to alternate destination)

* Chunk Bundling: FRTO assumes that on a timeout the sender will retransmit only one segment("I think chunk will be appropriate in SCTP terminology"). This may not be true in SCTP. If the association is multihomed and the PMTU's on the primary and alternate path differ by a factor of two or more then, then I think the sender may be able to retransmit more than one unacknowledged chunk on a timeout (bundled in a single packet). This can also be true if the app is writing small chunks such that on a timeout more than one of such chunks can be bundled in a single packet and retransmitted. I remember having noticed a couple of side effects of this which could lead FRTO to declare the timeout spurious while it was not. 

Comments?
(Others shoot in if I missed something)

A question I have is:
Section 3- step 3(b): When you refer to an ACK which acknowledges data (above spurious threshhold or previously unacknowledged), do you refer to acknowledging {via cum-ack point} or {via SACK blocks} or {either of these two}?


-Sourabh




> -----Original Message-----
> From: Pasi Sarolahti [mailto:sarolaht@cs.Helsinki.FI]
> Sent: Thursday, July 17, 2003 12:52 PM
> To: Armando L. Caro Jr.
> Cc: tsvwg@ietf.org
> Subject: Re: [Tsvwg] SCTP Timestamps
> 
> 
> Hi,
> 
> On Wed, 16 Jul 2003, Armando L. Caro Jr. wrote:
> 
> > The second motivation was to use timestamps for Eifel detection. A
> > spurious timeout retransmit can be detected if the 
> retransmitted TSN gets
> > acked with a timestamp earlier than the timestamp bundled with the
> > retransmission. This purpose does require a timestamp in 
> each packet (I
> > think).
> 
> Aside from this timestamp discussion, but since spurious RTOs were 
> mentioned: We have proposed a scheme for detecting spurious 
> RTOs without 
> timestamps in draft-sarolahti-tsvwg-tcp-frto-04.txt. I solicited for 
> comments on it in early June, but unfortunately got no 
> answer. It would be 
> nice to hear whether the SCTP community would find that useful.
> 
> The idea in short is that after an RTO the sender would try 
> to send some
> new data and check if there are acks of non-retransmitted segments
> arriving after that, in which case the RTO is considered to 
> be spurious.
> This should work with SCTP as well as with TCP as far as I can see. I
> believe Markku Kojo will have a short presentation about the 
> draft in the
> tsvwg meeting.
> 
> Thanks,
> - Pasi
> 
> 
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 

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From exim@www1.ietf.org  Thu Jul 24 17:19:26 2003
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From: Schruefer Wolfgang <wolfgang.schruefer@siemens.com>
To: tsvwg@ietf.org
Subject: Re: [Tsvwg] SCTP Timestamps (a Chunk Flag alternative)
Date: Thu, 17 Jul 2003 11:32:24 +0200
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Dear all,

following the Timestamp discussion I asked myself whether there is a need to transmit any timestamp and if there are no cheaper solution. Here I will suggest a solution by using a Bit of the DATA and SACK Chunk Flags.

What we need in my understanding is a solution for the problem Stephan described at the beginning of the thread:

Resolving the retransmission ambiguity, i.e. enabling the sender to determine whether an ACK refers to the original transmission or a retransmission of a chunk. This implies resolving a "path ambiguity" and allowing better and more frequent RTT measurements.

My thought is to sacrifice one (or two) bit(s) of the unused DATA and SACK Chunk Flags Bits to solve this problem.  The clue is that this bit should be echoed by the receiver in the SACK Chunk Flags. Let me just call this bit Mirror Bit or M-bit.  


A scenario may run like this:
- A sender sets the M-bit at any value when a DATA chunk is transmitted for the first time and stores the value together with the timestamp.

- The receiver mirrors the bit for the DATA chunk with TSN == Cumulative TSN Ack.

- When the sender recognises that the M-bit is not echoed correctly (for not retransmitted DATA), it knows that the peer does not support this feature. I think a feature negotiation at association setup is therefore not necessary.

- When the sender needs to retransmit the DATA for the first time it toggles the M-bit and stores the retransmission time.

- Up to now the sender can always resolve the chunk where a SACK refers to.

- When there is a need for further retransmissions I see three alternatives:
  ** The M-bit is toggled again and the sender assumes that the DATA chunk sent two RTO before was lost.
  ** The M-bit remains identical in all retransmissions (but e.g. a SACK for this DATA chunk may not be used any more for RTT calculation).
  ** We use two Chunk Flag Bits to mark the last four transmissions.


Is this a way worth to think further?

Wolfgang

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From: Sourabh Ladha <ladha@mail.eecis.udel.edu>
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Talk about reordering and delay spikes..I had send this email a week back
(17th July) :->

-Sourabh

On Thu, 17 Jul 2003, Sourabh Ladha (AC/EED) wrote:

> Pasi,
>
> I had read the FRTO document some time back. On the SCTP front I was wondering if the following are applicable:
>
> * Multihoming: If the delay spike on the primary path is greater than (1 RTO(primary) + 2 RTT(alternate)), I would believe that FRTO will not be able to detect a timeout spurious. (complying with the current spec that retransmissions SHOULD go to alternate destination)
>
> * Chunk Bundling: FRTO assumes that on a timeout the sender will retransmit only one segment("I think chunk will be appropriate in SCTP terminology"). This may not be true in SCTP. If the association is multihomed and the PMTU's on the primary and alternate path differ by a factor of two or more then, then I think the sender may be able to retransmit more than one unacknowledged chunk on a timeout (bundled in a single packet). This can also be true if the app is writing small chunks such that on a timeout more than one of such chunks can be bundled in a single packet and retransmitted. I remember having noticed a couple of side effects of this which could lead FRTO to declare the timeout spurious while it was not.
>
> Comments?
> (Others shoot in if I missed something)
>
> A question I have is:
> Section 3- step 3(b): When you refer to an ACK which acknowledges data (above spurious threshhold or previously unacknowledged), do you refer to acknowledging {via cum-ack point} or {via SACK blocks} or {either of these two}?
>
>
> -Sourabh
>
>
>
>
> > -----Original Message-----
> > From: Pasi Sarolahti [mailto:sarolaht@cs.Helsinki.FI]
> > Sent: Thursday, July 17, 2003 12:52 PM
> > To: Armando L. Caro Jr.
> > Cc: tsvwg@ietf.org
> > Subject: Re: [Tsvwg] SCTP Timestamps
> >
> >
> > Hi,
> >
> > On Wed, 16 Jul 2003, Armando L. Caro Jr. wrote:
> >
> > > The second motivation was to use timestamps for Eifel detection. A
> > > spurious timeout retransmit can be detected if the
> > retransmitted TSN gets
> > > acked with a timestamp earlier than the timestamp bundled with the
> > > retransmission. This purpose does require a timestamp in
> > each packet (I
> > > think).
> >
> > Aside from this timestamp discussion, but since spurious RTOs were
> > mentioned: We have proposed a scheme for detecting spurious
> > RTOs without
> > timestamps in draft-sarolahti-tsvwg-tcp-frto-04.txt. I solicited for
> > comments on it in early June, but unfortunately got no
> > answer. It would be
> > nice to hear whether the SCTP community would find that useful.
> >
> > The idea in short is that after an RTO the sender would try
> > to send some
> > new data and check if there are acks of non-retransmitted segments
> > arriving after that, in which case the RTO is considered to
> > be spurious.
> > This should work with SCTP as well as with TCP as far as I can see. I
> > believe Markku Kojo will have a short presentation about the
> > draft in the
> > tsvwg meeting.
> >
> > Thanks,
> > - Pasi
> >
> >
> >
> > _______________________________________________
> > tsvwg mailing list
> > tsvwg@ietf.org
> > https://www1.ietf.org/mailman/listinfo/tsvwg
> >
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>


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Apologies for/to the old mail that just popped out - there was an even worse than
usual flow of spam on the tsvwg list during IETF - what with IETF and all, it
was hard for Jon and me to see that there were some real messages hiding in there.

Allison

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>>   The proposals in this document are experimental.  We believe they
>>   are safe for deployment in the current Internet,
>
>Why is this believed? The document states fairly clearly that this sort 
>of TCP will receive a disproportionate share of a link if competing with 
>conventional TCP.

The same argument could have been made about NewReno TCP, which has
been Experimental since January 1999.  That is, there are scenarios
where NewReno TCP receives a disproportionate share of a link if
competing with then-conventional Reno TCP, in scenarios where loss
events are likely to consist of multiple packets dropped from a
window of data (e.g., in networks with Drop Tail queue management).
This is due to Reno TCP's often abysmal performance when multiple
packets are dropped from a window of data.  The rough consensus at
the time, as I recall, was that NewReno was considered safe for
deployment in the current Internet even though NewReno flows might
receive a disproportionate share of a link if competing with Reno,
because we didn't want the limitations of Reno TCP to become
preserved in the Internet in perpetuity.  I would suggest that a
similar situation occurs with HighSpeed TCP.

The internet draft about HighSpeed TCP is quite explicit about
the issue of relative fairness with Standard TCP, and it is not
the case the HighSpeed TCP will always receive a disproportionate
share of the link bandwidth when competing with Standard TCP.
In the environments where Standard TCP is clearly capable of
making good use of the existing link bandwidth, e.g., with 
packet drop rates of 10^-2 or 10^-3, HighSpeed TCP does exactly
the same thing as Standard TCP.  Table 5 of the draft shows
how the relative fairness goes in environments with smaller 
packet drop rates:

      Packet Drop Rate P   Fairness  Aggregate Window  Bandwidth
      ------------------   --------  ----------------  ---------
             10^-2            1.0              24        2.8 Mbps
             10^-3            1.0              76        9.1 Mbps
             10^-4            2.2             383       45.9 Mbps
             10^-5            4.7            2174      260.8 Mbps
             10^-6           10.2           13479        1.6 Gbps
             10^-7           22.1           87776       10.5 Gbps

    Table 5: Relative Fairness between the HighSpeed and Standard
    Response Functions.

The arguments of the draft are that the relative fairness of
2.2 with packet drop rates of 10^-4 are not a grave concern, and
that for the lower-packet-drop-rate environments with the more severe
unfairness, Standard TCP doesn't do very well at making use of the
available bandwidth.  From the draft:

"Our judgement would be that ... the benefits of the HighSpeed
response function would outweigh the unfairness that would be
experienced by Standard TCP in this regime."

But this is being brought to the working group for the group to decide.

From later email:
>There are cases where it is possible that stock TCP is used over short 
>fast links and this would be needed for longer paths. Using both on the 
>same enterprise could cause significant unfairness, in ways that are not
>considered a reasonable interim step.

I don't believe this.  Do you have any simulations or experiments
to back this up?  If a HighSpeed TCP and a Standard TCP were competing
over a congested shared link, and the Standard TCP was over a short
fast path, the Standard TCP would induce a rather high packet drop
rate in the shared link, and the HighSpeed TCP would be in the space
where it behaves roughly the same as Standard TCP.

...
>As above, using this protocol over a HS-path might interact very badly 
>with other streams that share part but not all of the path, e.g., 
>conventional TCP at high speeds near the endpoints (which have a small 
>window and no need for this sol'n).

Again, if a long-RTT HighSpeed TCP is competing over a congested
link with a Standard TCP flow that has a small window and a short
RTT, the advantage is more likely to go to the Standard TCP short-RTT
flow.  As is normally the case when short-RTT TCP flows compete
against longer-RTT flows.

...
>I'd be more comfortable with a statement that doesn't so broadly endorse
>experimenting with it on the Internet as a whole without some caveats, 
>notably those which appear to be implied.
>
>I.e.,
>	- off by default
>	- turned on only on applicable paths or environments
>	- it is the responsibility of those using this protocol to
>	determine whether it is affecting others, and when such effects
>	are found, to disable it until they can be corrected
>
>These may be implicit in the concept of 'experimental', but if we're 
>saying it's 'safe', it ought to be clear about under what conditions.

Personally, I think that HighSpeed TCP is perfectly safe to be
deployed on by default, for anyone/implementor who so chooses.  
(My own view is that HighSpeed TCP goes to great lengths to position
itself on the conservative, incrementally-deployable end of the
spectrum.  And that HighSpeed TCP is *more* safe than the N parallel
TCP connections that are often deployed in fact in its place, in
the current Internet in environments where Standard TCP is not up to
making good use of the available bandwidth.) 

But the decision rests with the working group as a whole...

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

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From exim@www1.ietf.org  Thu Jul 24 18:08:01 2003
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To: Matt Mathis <mathis@psc.edu>
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From: Sally Floyd <floyd@icir.org>
Subject: Re: [Tsvwg] High-speed TCP 
Date: Thu, 24 Jul 2003 15:06:21 -0700
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>It would be a_very_bad_thing if HS-TCP proved to be very unfair to
>the_right_answer(TM) and at the same time came to be in wide use by innocent
>users (say because some large vendor shipped it turned on by default).
>
>As long as HS-TCP is off by default, it will require a certain level of user
>awareness to turn it on.  Such users are likely to be aware of the_right_answer
>when it becomes available.  Just a moderate level of education will be
>sufficient to prevent chronic problems in the (presumably small) regime where
>HS-TCP might have an unfair advantage over the_right_answer....

The added text in the revised version of the draft,
draft-ietf-tsvwg-highspeed-00.txt, is as follows:

    We do not believe that the deployment of HighSpeed TCP would serve
    as a block to the possible deployment of alternate experimental
    protocols for high-speed congestion control, such as Scalable TCP,
    XCP [KHR02], or FAST TCP [JWL03].  In particular, we don't expect
    HighSpeed TCP to interact any more poorly with alternative
    experimental proposals that would the N parallel TCP connections
    commonly used today in the absence of HighSpeed TCP.

I would personally expect HighSpeed TCP to interact considerably
*better* with any alternate experimental protocol that would the 
N parallel Standard TCP connections that are in fact deployed in its
place, for some arbitrary value of N. 

(And hopefully HighSpeed TCP would do sufficiently well at making
use of the available bandwidth that people wouldn't feel the need,
or the justification, to use N parallel HighSpeed TCP connections
as a regular thing, for large N.  Or maybe we will manage at some
point to get mechanisms in routers that can take care of such
issues, who knows?)

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




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From exim@www1.ietf.org  Thu Jul 24 18:08:21 2003
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From: Schruefer Wolfgang <wolfgang.schruefer@siemens.com>
To: "'tsvwg@ietf.org'" <tsvwg@ietf.org>
Subject: Re: [Tsvwg] SCTP Timestamps
Date: Thu, 17 Jul 2003 13:06:12 +0200
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Dear all,

following the Timestamp discussion I asked myself whether there is a need to transmit any timestamp and if there are no cheaper solution. Here I will suggest a solution by using a Bit of the DATA and SACK Chunk Flags.

What we need in my understanding is a solution for the problem Stephan described at the beginning of the thread:

Resolving the retransmission ambiguity, i.e. enabling the sender to determine whether an ACK refers to the original transmission or a retransmission of a chunk. This implies resolving a "path ambiguity" and allowing better and more frequent RTT measurements.

My thought is to sacrifice one (or two) bit(s) of the unused DATA and SACK Chunk Flags Bits to solve this problem.  The clue is that this bit should be echoed by the receiver in the SACK Chunk Flags. Let me just call this bit Mirror Bit or M-bit.  


A scenario may run like this:
- A sender sets the M-bit at any value when a DATA chunk is transmitted for the first time and stores the value together with the timestamp.

- The receiver mirrors the bit for the DATA chunk with TSN == Cumulative TSN Ack.

- When the sender recognises that the M-bit is not echoed correctly (for not retransmitted DATA), it knows that the peer does not support this feature. I think a feature negotiation at association setup is therefore not necessary.

- When the sender needs to retransmit the DATA for the first time it toggles the M-bit and stores the retransmission time.

- Up to now the sender can always resolve the chunk where a SACK refers to.

- When there is a need for further retransmissions I see three alternatives:
  ** The M-bit is toggled again and the sender assumes that the DATA chunk sent two RTO before was lost.
  ** The M-bit remains identical in all retransmissions (but e.g. a SACK for this DATA chunk may not be used any more for RTT calculation).
  ** We use two Chunk Flag Bits to mark the last four transmissions.


Is this a way worth to think further?

Wolfgang

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Sally Floyd wrote:
>>>  The proposals in this document are experimental.  We believe they
>>>  are safe for deployment in the current Internet,
>>
>>Why is this believed? The document states fairly clearly that this sort 
>>of TCP will receive a disproportionate share of a link if competing with 
>>conventional TCP.
> 
> 
> The same argument could have been made about NewReno TCP, which has
> been Experimental since January 1999. 

Agreed. Note that the word "safe" does not appear in the NewReno RFC.

...
 > The rough consensus at
> the time, as I recall, was that NewReno was considered safe for
> deployment in the current Internet even though NewReno flows might
> receive a disproportionate share of a link if competing with Reno,
> because we didn't want the limitations of Reno TCP to become
> preserved in the Internet in perpetuity.  I would suggest that a
> similar situation occurs with HighSpeed TCP.

That may be true, but there is no corresponding recommendation of safety 
in the RFC. Declaring experimental protocols safe is a bit 
contradictory, and potentially problematic.

>>From later email:
> 
>>There are cases where it is possible that stock TCP is used over short 
>>fast links and this would be needed for longer paths. Using both on the 
>>same enterprise could cause significant unfairness, in ways that are not
>>considered a reasonable interim step.
> 
> I don't believe this.  Do you have any simulations or experiments
> to back this up?  If a HighSpeed TCP and a Standard TCP were competing
> over a congested shared link, and the Standard TCP was over a short
> fast path, the Standard TCP would induce a rather high packet drop
> rate in the shared link, and the HighSpeed TCP would be in the space
> where it behaves roughly the same as Standard TCP.

My concern is whether these cases have been tested before we make the 
recommendation of 'safe', rather than merely 'experimental.' The burden 
of proof is on the proposed protocol to in fact be safe. While my 
proposed scenario was indeed a thought-experiment, it would be useful if 
some of these typical scenarios were measured before we jumped to 'safe'.

I'm comfortable with the same recommendation as NewReno.

Joe



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>Agreed. Note that the word "safe" does not appear in the NewReno RFC.

While the NewReno RFC didn't have that paragraph, the NewReno RFC
was explicitly considered as acceptable for deployment in the public
Internet.  The point of Mark's proposed paragraph was to say that
although there was no objection to this being deployed in the
Internet, he wanted to make it clear that the WG wasn't endorsing
this as "the best method for high-speed congestion control".

So what if that paragraph said:

"While they may be deployed in the current Internet ... "

instead of: 

"We believe they are safe for deployment in the current Internet ... "

Would that seem better to you?

- Sally

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Sally Floyd wrote:
>>Agreed. Note that the word "safe" does not appear in the NewReno RFC.
> 
> 
> While the NewReno RFC didn't have that paragraph, the NewReno RFC
> was explicitly considered as acceptable for deployment in the public
> Internet.  The point of Mark's proposed paragraph was to say that
> although there was no objection to this being deployed in the
> Internet, he wanted to make it clear that the WG wasn't endorsing
> this as "the best method for high-speed congestion control".
> 
> So what if that paragraph said:
> 
> "While they may be deployed in the current Internet ... "
> 
> instead of: 
> 
> "We believe they are safe for deployment in the current Internet ... "
> 
> Would that seem better to you?

Until we have broader experience, yes.

Joe


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From: Sally Floyd <floyd@icir.org>
Subject: Re: [Tsvwg] High-speed TCP 
Date: Fri, 25 Jul 2003 08:44:54 -0700
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The modified version of draft-floyd-tcp-highspeed-03.txt, now called
draft-ietf-tsvwg-highspeed-00.txt, has already appeared in the
internet-drafts repository, but I have posted a file with the list
of changes to make at:
  http://www.icir.org/floyd/hstcp/draft-ietf-tsvwg-highspeed-00.errata

The contents are given below.

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

-----------------------------------------------------------------------
http://www.icir.org/floyd/hstcp/draft-ietf-tsvwg-highspeed-00.errata:
-----------------------------------------------------------------------

Changes to make to draft-ietf-tsvwg-highspeed-00.txt:

Change:

"The proposals in this document are experimental.  We believe they
are safe for deployment in the current Internet, "

To:

"The proposals in this document are experimental.  While they may be
deployed in the current Internet, "


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--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		: Limited Slow-Start for TCP with Large Congestion 
                          Windows
	Author(s)	: S. Floyd
	Filename	: draft-ietf-tsvwg-slowstart-00.txt
	Pages		: 7
	Date		: 2003-7-25
	
This note describes an optional modification for TCP's slow-start for
use with TCP connections with large congestion windows.  For TCP
connections that are able to use congestion windows of thousands (or
tens of thousands) of MSS-sized segments (for MSS the sender's
MAXIMUM SEGMENT SIZE), the current slow-start procedure can result in
increasing the congestion window by thousands of segments in a single
round-trip time.  Such an increase can easily result in thousands of
packets being dropped in one round-trip time.  This is often counter-
productive for the TCP flow itself, and is also hard on the rest of
the traffic sharing the congested link.  This note describes Limited
Slow-Start as an optional mechanism for limiting the number of
segments by which the congestion window is increased for one window
of data during slow-start, in order to improve performance for TCP
connections with large congestion windows.

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--NextPart

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

	Title		: HighSpeed TCP for Large Congestion Windows
	Author(s)	: S. Floyd
	Filename	: draft-ietf-tsvwg-highspeed-00.txt
	Pages		: 36
	Date		: 2003-7-25
	
This document proposes HighSpeed TCP, a modification to TCP's
congestion control mechanism for use with TCP connections with
large congestion windows.  The congestion control mechanisms
of the current Standard TCP constrains the congestion windows
that can be achieved by TCP in realistic environments.  For
example, for a Standard TCP connection with 1500-byte packets
and a 100 ms round-trip time, achieving a steady-state
throughput of 10 Gbps would require an average congestion
window of 83,333 segments, and a packet drop rate of at most
one congestion event every 5,000,000,000 packets (or
equivalently, at most one congestion event every 1 2/3 hours).
This is widely acknowledged as an unrealistic constraint.  To
address this limitation of TCP, this document proposes
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the wider community.

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From: Andrei Gurtov <gurtov@cs.helsinki.fi>
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Subject: [Tsvwg] reset of backoff with timestamps (RFC2988)
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Several people have noticed that not all TCPs reset the backoff of the retransmit 
timer in go-back-N after a new measurement sample is collected with timestamps. This 
results in unnecessarily long timeouts, which is especially a problem if fast
retransmits are disabled by "bugfix".

RFC2988:

  The only case when TCP can safely take RTT samples from retransmitted
  segments is when the TCP timestamp option [JBB92] is employed, since
  the timestamp option removes the ambiguity garding which instance
  of the data segment triggered the acknowledgment.

....

  Note that after retransmitting, once a new RTT measurement is
  obtained (which can only happen when new data has been sent and
  acknowledged), the computations outlined in section 2 are performed,
  including the computation of RTO, which may result in "collapsing"
  RTO back down after it has been subject to exponential backoff
  (rule 5.5).

The authors of RFC2988 and Sally Floyd agree that the second paragraph should 
also mention that a measurement can be taken earlier with timestamps. Therefore it's 
fine to reset the backoff when a new RTT sample is collected with timestamps.

Does TSV working group feel that it is appropriate? If yes, a following correction 
could be submitted to RFC errata. http://www.rfc-editor.org/errata.html

(which can only happen when new data has been sent and acknowledged *or 
if the timestamp option is used*)

Sally have made a change to ns2 to reflect this update.

Thanks,
Andrei




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--0-1288820232-1059114036=:65148
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Anyone know what is the status of the following draft?
 
draft-snoeren-tcp-migrate-00.txt
 
submitted on TCP migration by Hari Balakrishnan and Alex Snoeren?
 

--0-1288820232-1059114036=:65148
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<DIV>Anyone know what is the status of the following draft?</DIV>
<DIV>&nbsp;</DIV>
<DIV>draft-snoeren-tcp-migrate-00.txt</DIV>
<DIV>&nbsp;</DIV>
<DIV>submitted on TCP migration by Hari Balakrishnan and Alex Snoeren?</DIV>
<DIV>&nbsp;</DIV>
--0-1288820232-1059114036=:65148--

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From exim@www1.ietf.org  Mon Jul 28 19:41:59 2003
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Date: Mon, 28 Jul 2003 16:40:23 -0700
From: Andrei Gurtov <gurtov@cs.helsinki.fi>
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Subject: [Tsvwg] resolving the problem of unnecessary fast retransmits in go-back-N
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In go-back-N recovery TCP DUPACKs can result either from unnecessarily 
retransmitted packets
or from a loss of a retransmitted packet. [Flo94] from RFC2892 shows 
that this ambiguity can cause
unnecessary fast retransmits. A solution called "bugfix" in [F98] 
disables fast retransmits until go-back-N
recovery is completed. However, because the ambiguity is not resolved, 
"bugfix" causes timeouts and very long recovery if packets get lost 
during go-back-N.

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

Until SACK is universally deployed two methods to resolve the DUPACK 
ambiguity might be useful.

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

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

Some algorithm to detect spurious timeouts should be used before 
applying these tests. I'm currently running experiments in ns2 to 
identify any possible corner cases. For example, if ACKs get lost, a 
lost data segment would not be detected by the ACK-heuristic.  However, 
if proposals work at least in common cases, using them is an improvement 
upon the "bugfix". Later I'll send a pointer to simulation results.

Andrei


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Date: Tue, 29 Jul 2003 13:39:23 +0200
To: Andrei Gurtov <gurtov@cs.helsinki.fi>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: [Tsvwg] reset of backoff with timestamps (RFC2988)
Cc: tsvwg@ietf.org
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At 23:42 26.07.2003, Andrei Gurtov wrote:
>Does TSV working group feel that it is appropriate? If yes, a following 
>correction could be submitted to RFC errata. 
>http://www.rfc-editor.org/errata.html
>
>(which can only happen when new data has been sent and acknowledged *or if 
>the timestamp option is used*)

This sounds fine to me. I always understood the text like that, anyway.

What this is saying is that the use of the TCP Timestamps option disables 
both parts of Karn's algorithm. That is, (1) an RTT sample can be taken 
from the ACK for retransmitted data, and (2) the RTO can be collapsed upon 
arrival of that ACK, i.e., it can be recomputed from SRTT and RTTVAR.

In fact, this is how it had already been implemented in BSD4.4-Lite.

///Reiner  


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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		: Limited Slow-Start for TCP with Large Congestion 
                          Windows
	Author(s)	: S. Floyd
	Filename	: draft-ietf-tsvwg-slowstart-00.txt
	Pages		: 7
	Date		: 2003-7-25
	
This note describes an optional modification for TCP's slow-start for
use with TCP connections with large congestion windows.  For TCP
connections that are able to use congestion windows of thousands (or
tens of thousands) of MSS-sized segments (for MSS the sender's
MAXIMUM SEGMENT SIZE), the current slow-start procedure can result in
increasing the congestion window by thousands of segments in a single
round-trip time.  Such an increase can easily result in thousands of
packets being dropped in one round-trip time.  This is often counter-
productive for the TCP flow itself, and is also hard on the rest of
the traffic sharing the congested link.  This note describes Limited
Slow-Start as an optional mechanism for limiting the number of
segments by which the congestion window is increased for one window
of data during slow-start, in order to improve performance for TCP
connections with large congestion windows.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-slowstart-00.txt

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From exim@www1.ietf.org  Tue Jul 29 13:33:52 2003
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From: Andrei Gurtov <gurtov@cs.helsinki.fi>
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Subject: Re: [Tsvwg] reset of backoff with timestamps (RFC2988)
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Reiner Ludwig wrote:

> At 23:42 26.07.2003, Andrei Gurtov wrote:
>
>> Does TSV working group feel that it is appropriate? If yes, a 
>> following correction could be submitted to RFC errata. 
>> http://www.rfc-editor.org/errata.html
>>
>> (which can only happen when new data has been sent and acknowledged 
>> *or if the timestamp option is used*)
>
>
> This sounds fine to me. I always understood the text like that, anyway.
>
> What this is saying is that the use of the TCP Timestamps option 
> disables both parts of Karn's algorithm. That is, (1) an RTT sample 
> can be taken from the ACK for retransmitted data, and (2) the RTO can 
> be collapsed upon arrival of that ACK, i.e., it can be recomputed from 
> SRTT and RTTVAR.
>
> In fact, this is how it had already been implemented in BSD4.4-Lite. 

Updating RTO and reset of the backoff counter are separate issues. For 
example, old ns2 code updated RTO using RTT samples from retransmitted 
segments, but backoff was reset only when go-back-N recovery is completed.

Andrei


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Date: Wed, 30 Jul 2003 13:21:31 +0200
To: Andrei Gurtov <gurtov@cs.helsinki.fi>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: [Tsvwg] reset of backoff with timestamps (RFC2988)
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At 19:32 29.07.2003, Andrei Gurtov wrote:
>>What this is saying is that the use of the TCP Timestamps option disables 
>>both parts of Karn's algorithm. That is, (1) an RTT sample can be taken 
>>from the ACK for retransmitted data, and (2) the RTO can be collapsed 
>>upon arrival of that ACK, i.e., it can be recomputed from SRTT and RTTVAR.
>>
>>In fact, this is how it had already been implemented in BSD4.4-Lite.
>
>Updating RTO and reset of the backoff counter are separate issues.

Exactly. That's what I wanted to say. The first half of Karn's alg. does 
not allow taking an RTT sample from an ACK for retransmitted data while the 
second part of Karn's alg. suggests to *not* collapse the RTO upon that 
ACK, i.e., to reuse the backed-off RTO when restarting the retransmit timer.

And with timestamps none of the two parts are needed any longer.


>For example, old ns2 code updated RTO using RTT samples from retransmitted 
>segments, but backoff was reset only when go-back-N recovery is completed.


... and that was unnecessarily conservative. We had already fixed that in 
our ns2 code. So, the old ns2 code was not "compliant" with the BSD4.4-Lite 
code.

///Reiner


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