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From: Yogesh Prem Swami <yogesh.swami@nokia.com>
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Subject: [Tsvwg] DCLOR draft update
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Hello Folks,

An update for DCLOR is now available from

http://www.ietf.org/internet-drafts/draft-swami-tsvwg-tcp-dclor-01.txt

Since we have decoupled the problem of spurious timeout from subnet
change using the LMDR draft, we no longer require the sender to go to
INIT_WINDOW after a spurious timeout. This draft halves the congestion
every RTO.

Please provide your feedback.

Thanks
Yogesh

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Date: Wed, 02 Apr 2003 12:55:17 -0500
Subject: [Tsvwg] pr-sctp
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In SF I agreed to more beating on PR-SCTP via the mailing list.
So...

In SF Reiner showed a few slides that basically asked the question:
How much "extra" (unnecessary) traffic is acceptable to send?  TCP
naturally sends some.  The coding in some of the RTP protocols
constitutes more than the minimum bits required to send.  Etc.  I
have thoughts on that subject, but that is not what this email is
about.

My point to Randall about PR-SCTP was: should we be thinking about
PR-SCTP in the context of thinking about the points that Reiner
raised?  That is, with PR-SCTP you are sending things into the
network that are then (potentially) not used at the other end.  My
question was/is: If I send some segment X into the network, then
detect that it was lost, but also then decide that I don't care if X
makes it (so, I send an F-TSN), should that count against this "junk
data" counter that Reiner was alluding to?  There are two angles...

  * Phil Conrad and David Black rightly beat me and said something
    like ... but, PR-SCTP is helping you in this case because you
    won't *keep* sending this data that is useless.  And, that is
    true.  (Although, we know we have sent useless data some number
    of times.)

  * But, another question is whether we are providing an incentive
    for apps to dump things into the network quickly because they
    can then "supercede" them if necessary.  So, the tradeoff might
    be for the app to be a bit more careful and thoughtful about
    what is sent versus being quick and careless.  Are we providing
    a mechanism for apps to be more careless about sending data into
    the network?  And, if we are, then I think that thinking about
    some govenor on PR-SCTP in the context of Reiner's thought might
    be something to consider.

I hope that is a bit more clear.  It still may be erroneous thinking
... (so, am I shooting for "clearly erroneous" ... hmmm :).

Thoughts?

allman


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From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: Re: [Tsvwg] pr-sctp
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Mark,

I think you have phrased the issues correctly and presented a
cohesive story.

A comment below on the "second angle".

Michael Ramalho

At 12:55 PM 4/2/2003 -0500, Mark Allman wrote:
>
>In SF I agreed to more beating on PR-SCTP via the mailing list.
>So...
>
>In SF Reiner showed a few slides that basically asked the question:
>How much "extra" (unnecessary) traffic is acceptable to send?  TCP
>naturally sends some.  The coding in some of the RTP protocols
>constitutes more than the minimum bits required to send.  Etc.  I
>have thoughts on that subject, but that is not what this email is
>about.
>
>My point to Randall about PR-SCTP was: should we be thinking about
>PR-SCTP in the context of thinking about the points that Reiner
>raised?  That is, with PR-SCTP you are sending things into the
>network that are then (potentially) not used at the other end.  My
>question was/is: If I send some segment X into the network, then
>detect that it was lost, but also then decide that I don't care if X
>makes it (so, I send an F-TSN), should that count against this "junk
>data" counter that Reiner was alluding to?  There are two angles...
>
>   * Phil Conrad and David Black rightly beat me and said something
>     like ... but, PR-SCTP is helping you in this case because you
>     won't *keep* sending this data that is useless.  And, that is
>     true.  (Although, we know we have sent useless data some number
>     of times.)

[The "useless data" might not be "useless" if it arrived in a timely manner.]

You've basically got this point, though.

But to be emphatic .... under congestion this is exactly the behavior you want.
This is because every bit of information you send that the receiver can't 
profitably
use (because it is stale) is information that is sent *before more useful 
information*.
Therefore the process of re-sending stale information is *delaying* the 
sending of
useful information (potentially making it stale also ... under extended 
congestion
episodes).

Thus the setting of the lifetime parameter of the "timed reliability"
example in the pr-SCTP draft should be tuned to the application's
definition/knowledge of "staleness".

I agree that this adds a degree of definition difficulty in metrics like
"goodput". Is it good only if it is not stale? Clearly "goodput" defined
in such a manner is dependent on the application's definition of
"staleness/goodness" and the congestion condition at the time of
measurement/simulation.

Note: The congestion condition is not simply characterized by loss .... but
additionally by duration (in time) of the congestion event (as this has an
effect on "staleness"). We need to consider the time/delay element in
the congestion episode.


>   * But, another question is whether we are providing an incentive
>     for apps to dump things into the network quickly because they
>     can then "supercede" them if necessary.  So, the tradeoff might
>     be for the app to be a bit more careful and thoughtful about
>     what is sent versus being quick and careless.

One could argue that if transmission bandwidth and app processing
power is almost always plentiful, the "quick and careless" approach
is OK.

But, I'd suggest that making app designers communicate to a PR-SCTP
Service Definitions such as "timed reliability" is, in itself, making the
app designer think about such things .... and creates an environment
whereby the app designer is forced to think about how they desire the
app to perform under less-than-ideal operating conditions (like congestion
episodes, routing changes that result in much larger nominal delays or
high packet loss).

Thus, I happen to believe that we are providing an incentive for app designers
*not* be careless ... but rather to think about overload/congestion scenarios.

>Are we providing
>     a mechanism for apps to be more careless about sending data into
>     the network?  And, if we are, then I think that thinking about
>     some govenor on PR-SCTP in the context of Reiner's thought might
>     be something to consider.

In this thought process we might consider other PR-SCTP service definitions
that are more appropriate to the applications to consider under
overload or congestion scenarios. The creation of other pr-sctp service 
definition
to help "govern" the sending of data would be welcome. However, the F-TSN
mechanism is the only thing required "on the wire" for PR-SCTP.


>I hope that is a bit more clear.  It still may be erroneous thinking
>... (so, am I shooting for "clearly erroneous" ... hmmm :).
>
>Thoughts?
>
>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


Michael A. Ramalho, Ph.D.
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.941.708.4650
Cell: +1.941.544.2844

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Subject: Re: [Tsvwg] pr-sctp
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On Wed, 2 Apr 2003, Michael A. Ramalho wrote:

> But to be emphatic .... under congestion this is exactly the behavior you want.
> This is because every bit of information you send that the receiver can't
> profitably
> use (because it is stale) is information that is sent *before more useful
> information*.
> Therefore the process of re-sending stale information is *delaying* the
> sending of
> useful information (potentially making it stale also ... under extended
> congestion
> episodes).

I agree with the concept of PR-SCTP. I want to emphasize that bundling
FTSNs with data should be done whenever possible. I think that sending a
FTSN in a packet of its own is not much better than re-sending the stale
data. Both will be triggered at the same point. Both will take the same
time to reach the receiver. And if lost, both suffer the same
consequences. The only improvement is that flightsize gets reduced and new
data may be sent.

Section 3.5 F2) emphasizes that bundling should be done, but just
wondering out loud... should PR-SCTP implementations reserve room
for FTSNs in all packets. If data chunks are not fragmented properly
when queued, then SCTP cannot refragment them later when bundling
is needed. The draft says that a sender MAY wait to bundle FTSNs, but is
this wording loose enough to allow SCTP to wait for newly queued data to
bundle with. In other words, data already queued may be fragmented too
large for bundling, but newly queued data may be fragmented small enough
for bundling. ...I don't think I like this idea. Maybe simply reserving
room for some minimum FTSN size is good enough.

> Note: The congestion condition is not simply characterized by loss .... but
> additionally by duration (in time) of the congestion event (as this has an
> effect on "staleness"). We need to consider the time/delay element in
> the congestion episode.

I don't follow how this relates to FTSNs.

> In this thought process we might consider other PR-SCTP service definitions
> that are more appropriate to the applications to consider under
> overload or congestion scenarios. The creation of other pr-sctp service
> definition
> to help "govern" the sending of data would be welcome. However, the F-TSN
> mechanism is the only thing required "on the wire" for PR-SCTP.

I like that the PR-SCTP mechanism is independent of the service
definition. However, I cannot think of applications that could use
anything but timed-reliability.

~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: "Michael A. Ramalho" <mramalho@cisco.com>
From: Mark Allman <mallman@grc.nasa.gov>
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Subject: Re: [Tsvwg] pr-sctp 
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Michael-

> >   * Phil Conrad and David Black rightly beat me and said something
> >     like ... but, PR-SCTP is helping you in this case because you
> >     won't *keep* sending this data that is useless.  And, that is
> >     true.  (Although, we know we have sent useless data some number
> >     of times.)
> 
> [The "useless data" might not be "useless" if it arrived in a
> timely manner.]

(but, if it did then we'd not need to issue an f-tsn for it, either)

> But to be emphatic .... under congestion this is exactly the
> behavior you want.  This is because every bit of information you
> send that the receiver can't profitably use (because it is stale)
> is information that is sent *before more useful information*.

(assuming an endless supply of new information; but I understand
your point)

> Therefore the process of re-sending stale information is
> *delaying* the sending of useful information (potentially making
> it stale also ... under extended congestion episodes).

I agree.  (But, this is orthogonal to my overall point, I think.)

> >   * But, another question is whether we are providing an incentive
> >     for apps to dump things into the network quickly because they
> >     can then "supercede" them if necessary.  So, the tradeoff might
> >     be for the app to be a bit more careful and thoughtful about
> >     what is sent versus being quick and careless.
> 
> One could argue that if transmission bandwidth and app processing
> power is almost always plentiful, the "quick and careless"
> approach is OK.

But, if resources are always plentiful then one could also argue
that there is zero need for *P*R to begin with.

> But, I'd suggest that making app designers communicate to a
> PR-SCTP Service Definitions such as "timed reliability" is, in
> itself, making the app designer think about such things .... and
> creates an environment whereby the app designer is forced to think
> about how they desire the app to perform under less-than-ideal
> operating conditions (like congestion episodes, routing changes
> that result in much larger nominal delays or high packet loss).
> 
> Thus, I happen to believe that we are providing an incentive for
> app designers *not* be careless ... but rather to think about
> overload/congestion scenarios.

And, I think that was exactly Phil and David's point -- that PR-SCTP
in some ways helps the situation because it allows us to terminate
the re-sending of things that are known to be useless.

But, as a Cleveland Brown's coach once said, "there are two sides to
every piece of bologna".  And, I think the flip side very well could
be an incentive to be careless in choosing what to send.  Think
about the app sending the same thing N times and (framed with
application layer sequence numbers or something so that the app
could deal with dups).  Using PR-SCTP there is no incentive against
this.  Using TCP there is a clear incentive (head-of-line blocking
will cause absolutely lousy performance).  Using SCTP there is less
incentive than with TCP (because we can get unordered delivery), but
there is still a cost in terms of retransmissions and cwnd
sacrifice.

I am not sure what the right answer here is.... But, I think both
sides of the piece of bologna need examined.  And, I still think
that this falls within the realm of the sorts of things that Reiner
is trying to get us to think about.

Thanks!

allman


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

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From: Andreas Jungmaier <ajung@exp-math.uni-essen.de>
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Subject: Re: [Tsvwg] pr-sctp
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Hash: SHA1

On Wednesday 02 April 2003 19:55, Mark Allman wrote:
> In SF I agreed to more beating on PR-SCTP via the mailing list.
> So...
>
> In SF Reiner showed a few slides that basically asked the question:
> How much "extra" (unnecessary) traffic is acceptable to send?  TCP
> naturally sends some.  The coding in some of the RTP protocols
> constitutes more than the minimum bits required to send.  Etc.  I
> have thoughts on that subject, but that is not what this email is
> about.

Dear Mark, dear all,

it would be very nice if someone could give a brief wrap-up on
the discussions/presentations that were held in SF on this topic.
(PR-SCTP).

Like me, not all people on this list could make it to SF, and 
especially the PR-SCTP topic is very interesting to us, not only
from our implementation point of view.

Thanks,
best regards,

Andreas

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Date: Thu,  3 Apr 2003 11:47:11 -0600
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] pr-sctp
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On 4/2/03, Mark Allman wrote:


>
> * Phil Conrad and David Black rightly beat me and said
> something like ... but, PR-SCTP is helping you in this
> case because you won't *keep* sending this data that is
> useless.  And, that is true.  (Although, we know we have
> sent useless data some number of times.)
>
> * But, another question is whether we are providing an
> incentive for apps to dump things into the network
> quickly because they can then "supercede" them if
> necessary.  So, the tradeoff might be for the app to be a
> bit more careful and thoughtful about what is sent versus
> being quick and careless.  Are we providing a mechanism
> for apps to be more careless about sending data into the
> network?  And, if we are, then I think that thinking
> about some govenor on PR-SCTP in the context of Reiner's
> thought might be something to consider.
>
The question is whether PR-SCTP aids the applications that
it is intended to support (and whether there are enough of
those applications).

Obviously, any transport protocol can be abused. Consider
the still common practice of setting up multiple TCP
connections to make a session more fault tolerant,
and use of XML for the "benefit" of bypassing firewalls
by masquerading as web traffic. And as many application
levels have proven, you can build almost anything on top of
UDP, including IPv6 networks.

The profile that PR-SCTP addresses is any application that
has a content-sensitive need to retransmit lost packets.
This is most typically related to content that must be
delivered by a certain time to be useful.

But the need could be related to limited source buffering,
rather than a need at the consumer end. A real-time data
capture application, for example, may consider only some raw
data worth retaining for retransmit when the cost is not
capturing new data.

In my mind the prime application for PR-SCTP is streaming
multimedia. The content must be delivered with time
constraints, and it has an inherent structure. Certain data
messages (directory information, directives) MUST be
recieved, other data messages (I frames) will have a major
impact if lost, and the remainder (P,B frames) can be
dropped with very little impact.

Many streaming media servers do duplicate critical content,
but that is more related to support of multicast
environments. PR-SCTP is not suitable for multicast
delivery, but then almost all streaming media deliveries are
really point-to-point anyway.

The comparison with plain SCTP or TCP apparently has already
been covered. Clearly PR-SCTP reduces network load because
it allows content-sensitive prioritization of retransmits.
Only the most important data need be retransmitted, and only
if it will arrive in time. This is classic triage.
Retransmission can be limited to when it is necessary and
when it will not be futile.

The comparison with totally unreliable delivery is more
informative. An application that was merely loss tolerant,
but had no need to ever retansmit portions, would be better
served by a unreliable transport such as UDP or DCP. The
sender can release its buffers after transmitting, and the
receiver knows that what it gets is what it gets. There are
no extra packets (or network bandwidth) used to convey this
information.

More typically, loss of some portion of the data is more
critical to proper functioning of the receiver. There are
two methods to compensate for this:

-- Redundant transmission of critical data: The concern
   being raised seems to fear that PR-SCTP will encourage
   this practice. Compared to use of an unreliable transport
   it actually *discourages* it. The applications that would
   use PR-SCTP will not use TCP or SCTP instead, they will
   use UDP or DCP.
-- Application layer NACKing to selectively request 
   retransmission of critical data. This is part of
   RTP/RTSP now.
   
For typical streaming media applications, the sender has
at least or probably better idea of what data is critical
than the receiver does. PR-SCTP allows it to make these
determinations without an application level ACK or NACK.
With RTP/RTSP there is no need for explicit ACKs, but
that also forces congestion control up to the application
layer.




Caitlin Bestler
http://asomi.com/CaitlinBestler/
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Mark,

Mark Allman wrote:
...
> In SF Reiner showed a few slides that basically asked the question:
> How much "extra" (unnecessary) traffic is acceptable to send?  TCP
> naturally sends some.  The coding in some of the RTP protocols
> constitutes more than the minimum bits required to send.  Etc.  I
> have thoughts on that subject, but that is not what this email is
> about.

Here I have a comment on Reiner's question (which is the base of your
concern on PR-SCTP I think): I can see the point of Reiner's comment
applying to RTP/UDP traffic, but don't see it as much an issue for TCP
or SCTP. 

For RTP/UDP, people sometimes use forward error correction (FEC) to
combat frame loss. Several RFCs have been issued for this in avt. The
idea of FEC is simply to send redundant information for some or all data
frames. Of cause, if the original frame is not lost, this redundant
information becomes "unnecessary", i.e., a waste of network bandwidth.
In the case of RTP/UDP traffic, Reiner raised a good point that the
amount of redundancy the app can add to the original traffic must be
limited since UDP below has no throttle control. For example, a selfish
RTP sender may decide to add 1000% FEC redundancy to his traffic to
minimize his own data loss, at the cost of everyone else.

Things are not the same with TCP or SCTP. The CC in them automatically
puts a limit on how much bandwidth one can get. This changes the whole
situation. It is now more like that the app is given a set amount of
bandwidth and it has to decide *wisely* what portion of this bandwidth
to be used for "unnecessary" data, at the cost of the throughput of his
own "necessary" data. If he wants to do the same 1000% redundancy, he
simply hurts himself badly.

regards,
-Qiaobing
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> It is now more like that the app is given a set amount of
> bandwidth and it has to decide *wisely* what portion of this
> bandwidth to be used for "unnecessary" data, at the cost of the
> throughput of his own "necessary" data. If he wants to do the same
> 1000% redundancy, he simply hurts himself badly.

I don't think he just hurts himself.  Say the app runs at some rate
X bps.  But, for the sake of redundancy the sender runs at 2X bps
and sends all the data twice, which may be fine according to the CC
rate limit.  And, we might even call that "fair" because everyone is
getting a rate of 2X bps.  But, if this connection didn't send all
this needless junk into the network (i.e., only sent at rate X bps)
then there would be X bps of new capacity for others to use to
transmit useful data and get actual work done.  If enough people
send enough redundant data into the network then everyone suffers
because they could all utilize more bandwidth if all the needless
junk was not floating around.

That is not to say we need to get rid of FEC in RTP or *all*
spurious retransmission in TCP or SCTP.  I am just trying to further
motivate my concern (& disagree with your statement above).

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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Mark Allman wrote:
> 
> > It is now more like that the app is given a set amount of
> > bandwidth and it has to decide *wisely* what portion of this
> > bandwidth to be used for "unnecessary" data, at the cost of the
> > throughput of his own "necessary" data. If he wants to do the same
> > 1000% redundancy, he simply hurts himself badly.
> 
> I don't think he just hurts himself.  Say the app runs at some rate
> X bps.  But, for the sake of redundancy the sender runs at 2X bps
> and sends all the data twice, which may be fine according to the CC
> rate limit.  And, we might even call that "fair" because everyone is
> getting a rate of 2X bps.  But, if this connection didn't send all
> this needless junk into the network (i.e., only sent at rate X bps)
> then there would be X bps of new capacity for others to use to
> transmit useful data and get actual work done.  If enough people
> send enough redundant data into the network then everyone suffers
> because they could all utilize more bandwidth if all the needless
> junk was not floating around.
> 
> That is not to say we need to get rid of FEC in RTP or *all*
> spurious retransmission in TCP or SCTP.  I am just trying to further
> motivate my concern (& disagree with your statement above).

Mark,

I can see your point. But this is a little tricky and maybe out (or on
the edge) of tsvwg scope. To some degree, you are questioning whether
some portion of the data passed by the app above is REALLY necessary :-) 

I've had numerous times watching A/V codec designers (I used to be one)
and wondering "do those few extra redundancy bytes in your frame REALLY
make a difference?" They tend to say "surely, you can hear/see the
difference". So, when the app guys REALLY think that using 2X bps gives
them a critical edge, it's hard to argue with them. I guess the best we
in tsvwg could do is just to remind them "please please only put
*necessary* extra bits and not any *unnecessary* extra bits into the
network".

regards,
-Qiaobing


> 
> allman
> 
> --
> Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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> I don't think he just hurts himself.  Say the app runs at some rate
> X bps.  But, for the sake of redundancy the sender runs at 2X bps
> and sends all the data twice, which may be fine according to the CC
> rate limit.  And, we might even call that "fair" because everyone is
> getting a rate of 2X bps.  But, if this connection didn't send all
> this needless junk into the network (i.e., only sent at rate X bps)
> then there would be X bps of new capacity for others to use to
> transmit useful data and get actual work done.  If enough people
> send enough redundant data into the network then everyone suffers
> because they could all utilize more bandwidth if all the needless
> junk was not floating around.


A quick question. How is redundancy defined? Is a text (ascii for example) stream redundant--to me it is, since I can always compress it and reduce the network traffic. So should the transport protocol now go about compressing data ... Why do we need application  layer protocols then. 

--yogesh 
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Subject: Re: [Tsvwg] pr-sctp 
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Mark ... Good volley ... Only one significant point below (and
a requisite amount of cute comments) ... Michael

At 11:35 PM 4/2/2003 -0500, Mark Allman wrote:

>Michael-
>
> > >   * Phil Conrad and David Black rightly beat me and said something
> > >     like ... but, PR-SCTP is helping you in this case because you
> > >     won't *keep* sending this data that is useless.  And, that is
> > >     true.  (Although, we know we have sent useless data some number
> > >     of times.)
> >
> > [The "useless data" might not be "useless" if it arrived in a
> > timely manner.]
>
>(but, if it did then we'd not need to issue an f-tsn for it, either)
>
> > But to be emphatic .... under congestion this is exactly the
> > behavior you want.  This is because every bit of information you
> > send that the receiver can't profitably use (because it is stale)
> > is information that is sent *before more useful information*.
>
>(assuming an endless supply of new information; but I understand
>your point)

BTW - There are a lot of applications with this property. For
example, sensor data applications for roads, factory control,
weather info gathering, temperature control systems, network
monitoring information, etc.


> > Therefore the process of re-sending stale information is
> > *delaying* the sending of useful information (potentially making
> > it stale also ... under extended congestion episodes).
>
>I agree.  (But, this is orthogonal to my overall point, I think.)
>
> > >   * But, another question is whether we are providing an incentive
> > >     for apps to dump things into the network quickly because they
> > >     can then "supercede" them if necessary.  So, the tradeoff might
> > >     be for the app to be a bit more careful and thoughtful about
> > >     what is sent versus being quick and careless.
> >
> > One could argue that if transmission bandwidth and app processing
> > power is almost always plentiful, the "quick and careless"
> > approach is OK.
>
>But, if resources are always plentiful then one could also argue
>that there is zero need for *P*R to begin with.

I agree with your statement; but that is not exactly what I said. I said
"*almost* always" for a reason. In virtually every data characterization
environment I have run into .... the major congestion cases have
been designed to be "rare events". But it is exactly when these
rare events occur (sometimes characterized by fractal modeling)
that you *really want and need* such congestion behavior to
work in an intelligent manner.


> > But, I'd suggest that making app designers communicate to a
> > PR-SCTP Service Definitions such as "timed reliability" is, in
> > itself, making the app designer think about such things .... and
> > creates an environment whereby the app designer is forced to think
> > about how they desire the app to perform under less-than-ideal
> > operating conditions (like congestion episodes, routing changes
> > that result in much larger nominal delays or high packet loss).
> >
> > Thus, I happen to believe that we are providing an incentive for
> > app designers *not* be careless ... but rather to think about
> > overload/congestion scenarios.
>
>And, I think that was exactly Phil and David's point -- that PR-SCTP
>in some ways helps the situation because it allows us to terminate
>the re-sending of things that are known to be useless.
>
>But, as a Cleveland Brown's coach once said, "there are two sides to
>every piece of bologna".  And, I think the flip side very well could
>be an incentive to be careless in choosing what to send.  Think
>about the app sending the same thing N times and (framed with
>application layer sequence numbers or something so that the app
>could deal with dups).  Using PR-SCTP there is no incentive against
>this.

OK ... onto the other side of bologna ... as I partially disagree with
this point when considering other potential pr-SCTP service definitions.
Let me explain.

TCP has infinite persistence (it tries until it dies and blocks
everything following).

PR-SCTP now has a service definition that has a persistence for
a timed lifetime. That is, re-transmit until information stale ... then
forget about it. The "forgetaboutit" mechanism is the F-TSN.

I'd maintain that the app you describe above is begging for
another type of PR-SCTP service definition .... the "try N
times (persistence) service definition".

If these applications are significant ... one could define a
PR-SCTP service definition (addressable through an API that
supported the definition) that told PR-SCTP to "try to resend
this chunk up to N times, then forgetaboutit". Note the mechanism
to "forgetaboutit" *ALREADY EXISTS* in the present PR-SCTP
draft. No on the wire changes are needed. The PR-SCTP
protocol doesn't need modification to accomplish what this
application desires ... only a new API to support the new
service definition.

If such a service definition existed ... there would be *no need*
for the application to send N times via APF as you suggest.
We then let PR-SCTP (the transport ... not the application) do
the "accounting" and it will only send once (if successful) the
majority of the times .... and only sends N times when it was
actually unsuccessful for [N-1] prior times. Thus this
service definition would make PR-SCTP have a "N persistence"
property.

This, in my book, is an incentive for the app to behave
better than sending "N times" in a brute force manner.
It only needs to send a message to the transport layer
once (it may desire to be informed in the event of
unsuccessful transmission ... but that is an issue for
the service definition). This would free the app designer
to be an app designer and not become a true IETF
transport protocol weenie/expert.

We could then deprecate a lot of the UDP ALF mechanisms
previously defined (which do things like try N times
using an exponential backoff mechanism). SIP UDP ALF
comes to mind ... but I don't want to open up that can
of worms.

TCP has "infinite" persistence. UDP has "1 times" persistence.
And PR-SCTP can have "variable" (in time or re-try) persistence.
All the "persistence cases" for "reliable transport" are then
covered.

[Aside: My original suggestion for the name for pr-SCTP
was lp-SCTP (for "limited persistence") ... but was overruled
by co-authors a year ago.]

Do we have a PR-SCTP convert yet?

Probably not ... but I'll keep trying ;-).

>Using TCP there is a clear incentive (head-of-line blocking
>will cause absolutely lousy performance).  Using SCTP there is less
>incentive than with TCP (because we can get unordered delivery), but
>there is still a cost in terms of retransmissions and cwnd
>sacrifice.
>
>I am not sure what the right answer here is.... But, I think both
>sides of the piece of bologna need examined.  And, I still think
>that this falls within the realm of the sorts of things that Reiner
>is trying to get us to think about.
>
>Thanks!
>
>allman
>
>
>--
>Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg


Michael A. Ramalho, Ph.D.
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.941.708.4650
Cell: +1.941.544.2844

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To: "Armando L. Caro Jr." <acaro@acm.org>
From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: Re: [Tsvwg] pr-sctp
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Armando - Two short comments below - Michael Ramalho

At 06:38 PM 4/2/2003 -0500, Armando L. Caro Jr. wrote:
>On Wed, 2 Apr 2003, Michael A. Ramalho wrote:
>
> > But to be emphatic .... under congestion this is exactly the behavior 
> you want.
> > This is because every bit of information you send that the receiver can't
> > profitably
> > use (because it is stale) is information that is sent *before more useful
> > information*.
> > Therefore the process of re-sending stale information is *delaying* the
> > sending of
> > useful information (potentially making it stale also ... under extended
> > congestion
> > episodes).
>
>I agree with the concept of PR-SCTP. I want to emphasize that bundling
>FTSNs with data should be done whenever possible. I think that sending a
>FTSN in a packet of its own is not much better than re-sending the stale
>data. Both will be triggered at the same point. Both will take the same
>time to reach the receiver. And if lost, both suffer the same
>consequences. The only improvement is that flightsize gets reduced and new
>data may be sent.
>
>Section 3.5 F2) emphasizes that bundling should be done, but just
>wondering out loud... should PR-SCTP implementations reserve room
>for FTSNs in all packets. If data chunks are not fragmented properly
>when queued, then SCTP cannot refragment them later when bundling
>is needed. The draft says that a sender MAY wait to bundle FTSNs, but is
>this wording loose enough to allow SCTP to wait for newly queued data to
>bundle with. In other words, data already queued may be fragmented too
>large for bundling, but newly queued data may be fragmented small enough
>for bundling. ...I don't think I like this idea. Maybe simply reserving
>room for some minimum FTSN size is good enough.
>
> > Note: The congestion condition is not simply characterized by loss .... but
> > additionally by duration (in time) of the congestion event (as this has an
> > effect on "staleness"). We need to consider the time/delay element in
> > the congestion episode.
>
>I don't follow how this relates to FTSNs.

It doesn't. It relates to people who like to plot "goodput" vs "packet
error rate". My PR-SCTP "goodput" description of this thread
(which you cut) suggested that the goodput was a function of
staleness ... which was related to the *time duration* of the
congestion episode.

Thus you need three dimensional surface to describe the
PR-SCTP goodput ... one axis "goodput", one axis
"packet error rate" and the remaining axis "congestion
episode duration".


> > In this thought process we might consider other PR-SCTP service definitions
> > that are more appropriate to the applications to consider under
> > overload or congestion scenarios. The creation of other pr-sctp service
> > definition
> > to help "govern" the sending of data would be welcome. However, the F-TSN
> > mechanism is the only thing required "on the wire" for PR-SCTP.
>
>I like that the PR-SCTP mechanism is independent of the service
>definition. However, I cannot think of applications that could use
>anything but timed-reliability.

See my most recent email to Allman ... SIP UDP ALF as an example.

Michael


>~armando
>
>0--                                                  --0
>| Armando L. Caro Jr.      |  Protocol Engineering Lab |
>| www.cis.udel.edu/~acaro  |    University of Delaware |
>0--                                                  --0
>
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Michael A. Ramalho, Ph.D.
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.941.708.4650
Cell: +1.941.544.2844

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From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] redundant data - A TFRC Example
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I'd like to add an example for people to think about ...

If you try to run VoIP over TFRC, you'll most likely be disappointed by the 
quality. The reason is that when A talks, then (usually) B is silent, and 
vice versa. I.e., a talkspurt A>B is followed by a talkspurt B>A is 
followed by a talkspurt A>B, ....

While B is silent and if the talkspurt from A is long enough then B's TFRC 
send rate will decay to zero (or at least often enough below the required 
coder rate). Andrei Gurtov did this experiment and told me how 'broken' 
such a voice conversation becomes.

The "easy fix" is for B to send dummy data (or even potentially useful data 
like pictures, ads, calendar sync data, etc.) for the sole purpose of 
maintain the TFRC rate above the required coder rate. B would be a good 
Internet citizen in the sense that it is always TCP friendly. Still, B is 
sending data it does't have to send necessarily.

Should the IETF set up rules that would limit the sending of such 
potentially useless data? Do we have hope that such rules would be obeyed 
by application and protocol designers?

I could imagine VoIP application designers that would let themselves be 
convinced to move away from uncontrolled UDP to RTP/DCCP, but probably not 
much further. I.e., their view could be: "We ensure that our flows behave 
TCP friendly, but what we do with the available bandwidth is our business."

I'm not saying that I agree to that, but I would understand that view.

///Reiner

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> I can see your point. But this is a little tricky and maybe out
> (or on the edge) of tsvwg scope. To some degree, you are
> questioning whether some portion of the data passed by the app
> above is REALLY necessary :-)

That is absolutely not the question I was asking.  I think that
trying to figure out whether the app is passing new or redundant
data is likely an NP-hard problem.

My wondering is specifically about transports aiding the process of
sending redundant data.  For instance, early retransmit proposes
resending data with less indication that it was lost than fast
retransmit.  Therefore, ER is less robust to reordering than FR.
There are worst cases where ER really wastes network resources by
resending data that does not need to be resent.  That is not an
application->transport issue, rather it is a transport only issue.

Another class of problems is whether the transport provides an
incentive for the app to send lots of data that may accomplish no
useful work.  I am not sure if PR-SCTP does this or not.  But, thus
far, nobody has convinced me that it does not.

allman


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> If such a service definition existed ... there would be *no need*
> for the application to send N times via APF as you suggest.  We
> then let PR-SCTP (the transport ... not the application) do the
> "accounting" and it will only send once (if successful) the
> majority of the times .... and only sends N times when it was
> actually unsuccessful for [N-1] prior times. Thus this service
> definition would make PR-SCTP have a "N persistence" property.
> 
> This, in my book, is an incentive for the app to behave better
> than sending "N times" in a brute force manner.

No need, but maybe a desire.

If I can send something twice from the app and one arrives at the
receiver then by some metrics the performance may be better because
we don't have to wait for (at least) an RTT for the transport to
figure out that the packet was dropped and retransmit.  If we have
to waste our own resources (buffer space, cwnd space, etc.) to
retransmit this packet that is now useless then I think there is
*some* incentive against using such a brute force method.  If we can
say "oh, nevermind on that first segment" and clean up our own
resources then it seems like there is no incentive not to waste
network resources if one wanted lots of redundancy.

allman


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On Fri, 4 Apr 2003, Mark Allman wrote:

> Another class of problems is whether the transport provides an
> incentive for the app to send lots of data that may accomplish no
> useful work.  I am not sure if PR-SCTP does this or not.  But, thus
> far, nobody has convinced me that it does not.

I understand your concern, and I cannot say that I can convince you that
PR-SCTP doesn't face this problem. However, doesn't any
less-than-fully-reliable transport protocol face the same problem?

~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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To: "Armando L. Caro Jr." <acaro@acm.org>, Mark Allman <mallman@grc.nasa.gov>
From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: Re: [Tsvwg] redundant data 
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At 10:44 AM 4/4/2003 -0500, Armando L. Caro Jr. wrote:
>On Fri, 4 Apr 2003, Mark Allman wrote:
>
> > Another class of problems is whether the transport provides an
> > incentive for the app to send lots of data that may accomplish no
> > useful work.  I am not sure if PR-SCTP does this or not.  But, thus
> > far, nobody has convinced me that it does not.
>
>I understand your concern, and I cannot say that I can convince you that
>PR-SCTP doesn't face this problem. However, doesn't any
>less-than-fully-reliable transport protocol face the same problem?
>
>~armando

I too see Mark's lament that the transport may allow (I think encourage
is too strong) the app to send data to it that -  perhaps - will not be
sent (or re-sent if not ACKed). But let's look at the alternative (yet
another side of bologna).

The alternative is to mandate all app designers to be as smart as Mark
[or put your favorite tsvwg member name here] in order that they re-transmit
"transport friendly" (with correct backoffs, etc.) in "limited persistence" 
cases.

Unfortunately, this is too high a burden for most app designers (although
some would be smart enough to do something reasonable).

I think some reasonable, course feedback mechanism should be built
into applications in order to guide the app on the "course rate" of information
sent when the app has significant degrees of rate freedom (most video
codecs, for example).

However, I think it is an unreasonable expectation that application
designers have detailed tsvwg knowledge ... and I think the tradeoff
of letting the application send data to the transport with "lifetimes"
or "N persistence" mechanisms I described in earlier threads is
a reasonable way to let the transport layer take care of the "fine
grained" transport dynamics in a "transport friendly" manner.

Even if the far-end application doesn't provide feedback, the near end
transport can notify the app for chunks it decided to F-TSN over; this
is at least some feedback.

Lastly, I think providing such mechanisms (via service definitions)
actually encourage app designers to  give more thought on what
they desire to transmit under overload/congestion scenarios.

Michael Ramalho

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Subject: Re: [Tsvwg] pr-sctp 
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Michael rightly noted that I wrote a whole lot of ambiguous rubbish
the last time around.  Durned context senssative languages.  So, I
am going to try again...

> If such a service definition existed ... there would be *no need*
> for the application to send N times via APF as you suggest.  We
> then let PR-SCTP (the transport ... not the application) do the
> "accounting" and it will only send once (if successful) the
> majority of the times .... and only sends N times when it was
> actually unsuccessful for [N-1] prior times. Thus this service
> definition would make PR-SCTP have a "N persistence" property.
> 
> This, in my book, is an incentive for the app to behave better
> than sending "N times" in a brute force manner.

No need, but maybe a desire.

Say the app sends the same message to the transport twice.  The
transport then sends both copies out (in two segments).  Now, say
one of these segments gets dropped but the other arrives fine and is
delivered to the receiving app.  By some metrics that offers better
performance than the case when the sending app gives the sending
transport only one copy and lets the transport retransmit as
necessary (which adds an extra ~RTT before the receiving app gets
the data).  If the sender has to waste its own resources (buffer
space, cwnd space, etc.) to retransmit the lost segment (that is now
useless) then I think there is *some* incentive against using such a
brute force method.  If the sender can say "oh, nevermind on that
lost segment" and clean up our own resources then it seems like
there is no incentive not to waste network resources if one wanted
lots of redundancy.

I hope that is a bit more clear.  Sorry for any confusion I caused.

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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Subject: [Tsvwg] Application Scenarios where PR-SCTP reduces "junk data" on the  net
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Dear Colleagues:

As I understand it, (and I might have it wrong)  Mark's concern is that
PR-SCTP might encourage senders to put information on the wire that is
"not very useful"... .what we might call "junk data".    After all, if one is
willing to "give up" on some message,  how important could it have been
in the first place?  Thus, the argument  might go, it is a waste of network
resources to have put that data on the wire in the first place.

I may be putting words in Mark's mouth, so Mark, if that is NOT your argument,
feel free to speak up and correct me.

But if that is the argument, then I think a concept that may be helpful is
to consider the "time varying value of information".   That is, information 
can be very valuable at one point in time, and less valuable at another 
point in time.      That is key idea behind the idea of "timed reliability".   Thus, by
having a transport service that allows us to give up on packets after their 
utility has decreased, we can actually acheive the result of _less_ "junk 
data" going on the wire.

At 09:28 AM 4/4/2003 -0500, Mark Allman wrote:
>Another class of problems is whether the transport provides an
>incentive for the app to send lots of data that may accomplish no
>useful work.  I am not sure if PR-SCTP does this or not.  But, thus
>far, nobody has convinced me that it does not.

Mark, if you are not certain that PR-SCTP does this, I  think rather the burden of proof is on you to show a scenario that PR-SCTP _does_ create such an incentive. 
I think that if you can make that case, we can really have a useful discussion. 

Without a coherent argument to that effect, your technical argument against PR-SCTP seems to be that "it _might_ create an incentive to send junk data, though I can't yet produce the scenario in which that is true".   This doesn't seem to be a very strong technical argument.    

On the other hand, I'd like to present a couple of scenarios, where I think there is a convincing case that PR-SCTP actually achieves the opposite.. that is, to "reduce" the amount of "junk data" flowing on the network.

(Scenario 1): Consider a "situational awareness" application operating over
a wireless IP network.    In a situational awareness application, mobile units
(mounted on vehicles or wearable computers) send periodic status updates
to planners that need to put together a coherent picture of "who is where"
and "how everyone is doing".    Military applications are probably obvious,
 but if you prefer a non-military purpose, consider tracking emergency
vehicles (rescue, fire personnel, gas and electrical repair workers, etc.)
in a disaster situation (e.g. the "big earthquake").

In such a network, packet loss due to both congestion, noise, and link and
router failure may be significant, so it is important to concentrate the 
resources of the network on the most crucial information.    If the application
generates status update messages every 2 seconds, then it might make
sense to put an  "expiration date" on such messages of 3-5 seconds, 
so that if there is a transient network outage of, say 20 seconds... 
not long enough to drop the SCTP association, but long enough to 
create a backlog of update messages, the sender can skip over
the retransmission of some stale messages and get the more timely 
data out the door.

Now, in this application, it is NOT the case that the transmission of any 
of this information is "wasteful" when it is first submitted to the transport 
service... at least not in the usual case of low packet loss.   In that case, the 
constant updates would flow to the receiver in a timely fashion, and give 
a second by second  picture of where each unit in the overall picture is 
located, a picture that may be crucial to saving lives.

However, once the network gets into trouble, and we get into the situation 
where due to a transient outage, a bunch of messages are sitting in the 
transport send  buffer of some unit, and those messages at the head of 
the transmit queue are NOT the most crucial information to transmit at 
that point in time (because they are  stale) PR-SCTP allows
the sender to skip over the information that has now _become_ stale, and 
put the _more_ useful (i.e., more timely) status information on the wire.

(Application 2:)  My understanding is that in 
certain problem  domains that are outside my main area of expertise 
(e.g. signalling transport, ip  storage) there are certain legacy protocols 
that it is desirable now, because of the  ubiquity of IP-based
technology, to place on top of IP networks.  

(Since my knowledge of sigtran and ip storage is very limited, 
I hope those who are more expert in these areas can either confirm
or correct my simplified explanations below.) 

Because it is desired for these protocols to benefit from the congestion 
control, flow control, and retransmission  mechanisms available
in protocols such as TCP and SCTP, these legacy protocols are generally 
implemented on top of TCP and/or SCTP.

However, a problem occurs because many of these protocols
have built into them, legacy retransmission timers of their own.  In some 
cases, these retransmission timers might be, for example, fixed values 
that are an order  of magnitude larger than typical initial RTOs that would 
result from TCP or  SCTP.    (say, for sake of
argument, 5 seconds), and certainly larger than the retransmission that 
might result from a duplicate ack triggered fast retransmit.

However, sometimes, if the RTO gets backed off, the TCP or SCTP RTO might 
be larger than the application layer protocols own RTO.  Now we have 
retransmission occurring  at two different layers, which is an undesirable situation.

Now there are two ways of addressing this probably immediately occur to the 
reader, neither of which is very satisfactory in the real world, as I 
understand it.

One answer to this would be to "fix the application layer protocol" to 
either turn off the retransmissions at this layer.    However, as a 
practical matter, my understanding is that the application layer protocols 
in question are "deeply embedded" in signalling transport and storage 
systems, and that while is "might be nice" to be able to do this, it is 
simply not feasible.

Another approach, is to apply the end-to-end argument, and run these over a 
completely best-effort transport, and rely exclusively on the 
retransmission built into the application layer protocol.    A problem here 
is that the application layer protocol's retransmission strategies don't 
adapt well to the underlying network, not in the way that the 
retransmissions in TCP and SCTP adapt.   (There may be other reasons as 
well that this is not feasible... I'll rely on my colleagues to fill in the 
gaps)

The alternative approach is to use the timed-reliability service provided 
by PR-SCTP to indicate that messages that will be retransmitted by the 
upper layer after, for example, 5 seconds, should be abandonded by the 
transport layer once that 5 seconds has elapsed.   Again, this results in 
an actual _reduction_ in "junk data", i.e. a redundant retransmission of a 
particular message X, which in the absence of the PR-SCTP timed reliabilty 
service, ends up appearing more than once in the reliable data stream.

Therefore, the bottom line is that PR-SCTP helps to keep redundant data,
or data that has become stale off the network.    None of this data was
"junk data" when it was first transmitted, which seems to be one of the
fears or doubts that is being thrown at PR-SCTP.


Regards,
Phill

-- Phillip Conrad, Asst Professor, CIS Dept. Temple University
-- Philadelphia PA 19122    conrad@acm.org

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From: Phillip Conrad <conrad@acm.org>
Subject: Re: [Tsvwg] pr-sctp
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Mark, I agree that this explanation is a lot more clear than your previous ones.    However I still don't think you have made your case, or at least, your case isn't clear to me.  (If it is clear to others, please help me understand it.)

More comments below...

At 11:59 AM 4/4/2003 -0500, Mark Allman wrote:
>
>Michael rightly noted that I wrote a whole lot of ambiguous rubbish
>the last time around.  Durned context senssative languages.  So, I
>am going to try again...
>
>Say the app sends the same message to the transport twice.  The
>transport then sends both copies out (in two segments).  Now, say
>one of these segments gets dropped but the other arrives fine and is
>delivered to the receiving app.  By some metrics that offers better
>performance than the case when the sending app gives the sending
>transport only one copy and lets the transport retransmit as
>necessary (which adds an extra ~RTT before the receiving app gets
>the data).  If the sender has to waste its own resources (buffer
>space, cwnd space, etc.) to retransmit the lost segment (that is now
>useless) then I think there is *some* incentive against using such a
>brute force method.  If the sender can say "oh, nevermind on that
>lost segment" and clean up our own resources then it seems like
>there is no incentive not to waste network resources if one wanted
>lots of redundancy.

You note that there is a disincentive to sending junk data in regular TCP/SCTP (i.e. spending buffer space, cwnd, etc.).  But you imply that this disincentive is NOT present with PR-SCTP.    That's where I think we have a difference in understanding, becuase I claim that the same disincentives to send junk data apply equally to PR-SCTP.  Specifically, I note that: 

* Every chunk you send with PR-SCTP consumes rwnd and cwnd space.    

* PR-SCTP does NOT result in putting any more data on the network than if you did NOT have PR-SCTP.  It is only a question of "which" data goes on the network.

Look at it another way: TCP-friendliness enforces that each transport sender gets some "share" of the bandwidth on the network.   How a given application chooses to use that share is up to that application.  PR-SCTP gives the application more flexibility in how it chooses how to use that share... meaning that it can make decisions on the fly about diverting some of that share from a message that is already in the sequence number space to a new message that is not yet in the sequence number space.

I don't see how that creates an incentive to send junk data.  I'm not saying with 100% certainty that it isn't there, but the more we talk about it, the less worried I get.   Maybe there is an argument that I just haven't seen or understood yet.

Now in response to your comments about "circling the wagons"...   see below. :-)

>At 12:57 PM 4/4/2003 -0500, Mark Allman wrote:
>>I should re-phrase... I think there *is* an incentive for apps to
>>pollute the network with junk data caused by PR-SCTP.  What I am not
>>sure of, yet, is how much we should really care about it.  I.e., how
>>big of a deal is it?  But, I *would advocate* understanding it
>>before sending it out and then waiting for the paper (from Stefan
>>Savage, no doubt!) on how to game SCTP for fun and profit.  (And, I
>>have been somewhat disappointed thus far because I see some folk
>>"circling the wagons" to "protect" SCTP rather than trying to help
>>figure out if this is a real issue or it is just me being paranoid
>>(which is a fine result, BTW).  (My guess - without reading your
>>note - is that you have not done that.))

I don't know if it was deliberate or accidental, but your last phrase is marvelously
ambiguous..  do you mean that your guess is that I have not "circled the wagons", or your guess is that I have not "help[ed] to figure out if this is a real issue"?   (You can leave
this ambiguous if you like... :-) ).

Well, in any case, a accusation that the PR-SCTP advocates are "circling the wagons" sounds a bit light an invitation to up the rhetoric a little bit, but  Mark, since I consider you a friend, I'm not gonna take the bait.    Well, what the heck, maybe I will, trusting that you'll see this as a friendly jab rather than an invitation to a flame war.

I don't think we are circling the wagons at all.  I think we are trying honestly to understand what your fear is that PR-SCTP creates an incentive to send junk data... which, if it could be argued convincingly, would be a reason to pause and reflect.       But so, far, if you'll forgive me for saying so, your criticism that PR-SCTP creates an incentive to send junk data without a clear explanation of how that occurs feels a little bit  like a "fear uncertainty doubt" attack.    That's why I sent my much longer note with some specific application scenarios where I think I've made a case that PR-SCTP does just the opposite: it helps applications _reduce_ junk data on the net.    

The bottom line is that if you are being TCP-friendly, you have a "share" of the network resources.    How you use it is up to you.  If you waste your share, that's your own funeral, because TCP-friendliness limits your impact on others.

How much worse would it be if (in the absence of PR-SCTP) you had to open, for example, two separate connections: say, (1) a TCP or SCTP connection/association for the reliable data, and (2) a TCP-friendly UDP flow, or at somepoint in the future, a DCCP connection for the other data.

Now you have to implement your own reliability on top of connection flow (2) and further, you have increased your share of the bandwidth by having two tcp-friendly flows rather than one.    Seems like a lot more waste to me.  

(Obligatory concession: yes, I know that if we all implemented an end point congestion manager type solution, that last point would be invalidated.   Seems unlikely though.) 

Thanks,
Phill 

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Subject: Re: [Tsvwg] Application Scenarios where PR-SCTP reduces "junk data" on the  net
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Dear all,

one of the main usage scenarios for SCTP is SIGTRAN. The SS7
protocols being transported can not be changed and run their
own timers, ISUP for example.

Looking at this example the following happens:

The ISUP sender sends an ISUP message, starts a timer and
waits for an ISUP answer. If the timer is stopped if the
answer arrives and another message will be send.

But if the answer does not arrive before the timer runs
off ISUP brings down the call by sending a NEW message.
Now it makes no sense for the SCTP layer to retransmit the
first message. It would be better to discard that message and
only deal with the new message.

Assuming that this happens during congestion you are sending
2 messages in case of SCTP and only one in case of PR-SCTP,
because you are not continuing to retransmit the now useless
first message.

You can say that we should change the upper layer, but the intention
of the SIGTRAN WG is exactly the opposite: provide a way of SS7 over
IP WITHOUT changing the SS7 protocols.

This usage scenario was already considered during the work on RFC 2960,
therefore you find a lifetime parameter in the send call description.
The mechanisms in the protocol however, did not make it.

So I think the discussion should not only be on potential ways of
misuse, but also on the scenarions where such a mechanism can be
really helpful. SIGTRAN, in my opinion, is one.

Best regards
Michael


On Friday, Apr 4, 2003, at 19:08 Europe/Berlin, Phillip Conrad wrote:

> Dear Colleagues:
>
> As I understand it, (and I might have it wrong)  Mark's concern is that
> PR-SCTP might encourage senders to put information on the wire that is
> "not very useful"... .what we might call "junk data".    After all, if 
> one is
> willing to "give up" on some message,  how important could it have been
> in the first place?  Thus, the argument  might go, it is a waste of 
> network
> resources to have put that data on the wire in the first place.
>
> I may be putting words in Mark's mouth, so Mark, if that is NOT your 
> argument,
> feel free to speak up and correct me.
>
> But if that is the argument, then I think a concept that may be 
> helpful is
> to consider the "time varying value of information".   That is, 
> information
> can be very valuable at one point in time, and less valuable at another
> point in time.      That is key idea behind the idea of "timed 
> reliability".   Thus, by
> having a transport service that allows us to give up on packets after 
> their
> utility has decreased, we can actually acheive the result of _less_ 
> "junk
> data" going on the wire.
>
> At 09:28 AM 4/4/2003 -0500, Mark Allman wrote:
>> Another class of problems is whether the transport provides an
>> incentive for the app to send lots of data that may accomplish no
>> useful work.  I am not sure if PR-SCTP does this or not.  But, thus
>> far, nobody has convinced me that it does not.
>
> Mark, if you are not certain that PR-SCTP does this, I  think rather 
> the burden of proof is on you to show a scenario that PR-SCTP _does_ 
> create such an incentive.
> I think that if you can make that case, we can really have a useful 
> discussion.
>
> Without a coherent argument to that effect, your technical argument 
> against PR-SCTP seems to be that "it _might_ create an incentive to 
> send junk data, though I can't yet produce the scenario in which that 
> is true".   This doesn't seem to be a very strong technical argument.
>
> On the other hand, I'd like to present a couple of scenarios, where I 
> think there is a convincing case that PR-SCTP actually achieves the 
> opposite.. that is, to "reduce" the amount of "junk data" flowing on 
> the network.
>
> (Scenario 1): Consider a "situational awareness" application operating 
> over
> a wireless IP network.    In a situational awareness application, 
> mobile units
> (mounted on vehicles or wearable computers) send periodic status 
> updates
> to planners that need to put together a coherent picture of "who is 
> where"
> and "how everyone is doing".    Military applications are probably 
> obvious,
>  but if you prefer a non-military purpose, consider tracking emergency
> vehicles (rescue, fire personnel, gas and electrical repair workers, 
> etc.)
> in a disaster situation (e.g. the "big earthquake").
>
> In such a network, packet loss due to both congestion, noise, and link 
> and
> router failure may be significant, so it is important to concentrate 
> the
> resources of the network on the most crucial information.    If the 
> application
> generates status update messages every 2 seconds, then it might make
> sense to put an  "expiration date" on such messages of 3-5 seconds,
> so that if there is a transient network outage of, say 20 seconds...
> not long enough to drop the SCTP association, but long enough to
> create a backlog of update messages, the sender can skip over
> the retransmission of some stale messages and get the more timely
> data out the door.
>
> Now, in this application, it is NOT the case that the transmission of 
> any
> of this information is "wasteful" when it is first submitted to the 
> transport
> service... at least not in the usual case of low packet loss.   In 
> that case, the
> constant updates would flow to the receiver in a timely fashion, and 
> give
> a second by second  picture of where each unit in the overall picture 
> is
> located, a picture that may be crucial to saving lives.
>
> However, once the network gets into trouble, and we get into the 
> situation
> where due to a transient outage, a bunch of messages are sitting in the
> transport send  buffer of some unit, and those messages at the head of
> the transmit queue are NOT the most crucial information to transmit at
> that point in time (because they are  stale) PR-SCTP allows
> the sender to skip over the information that has now _become_ stale, 
> and
> put the _more_ useful (i.e., more timely) status information on the 
> wire.
>
> (Application 2:)  My understanding is that in
> certain problem  domains that are outside my main area of expertise
> (e.g. signalling transport, ip  storage) there are certain legacy 
> protocols
> that it is desirable now, because of the  ubiquity of IP-based
> technology, to place on top of IP networks.
>
> (Since my knowledge of sigtran and ip storage is very limited,
> I hope those who are more expert in these areas can either confirm
> or correct my simplified explanations below.)
>
> Because it is desired for these protocols to benefit from the 
> congestion
> control, flow control, and retransmission  mechanisms available
> in protocols such as TCP and SCTP, these legacy protocols are generally
> implemented on top of TCP and/or SCTP.
>
> However, a problem occurs because many of these protocols
> have built into them, legacy retransmission timers of their own.  In 
> some
> cases, these retransmission timers might be, for example, fixed values
> that are an order  of magnitude larger than typical initial RTOs that 
> would
> result from TCP or  SCTP.    (say, for sake of
> argument, 5 seconds), and certainly larger than the retransmission that
> might result from a duplicate ack triggered fast retransmit.
>
> However, sometimes, if the RTO gets backed off, the TCP or SCTP RTO 
> might
> be larger than the application layer protocols own RTO.  Now we have
> retransmission occurring  at two different layers, which is an 
> undesirable situation.
>
> Now there are two ways of addressing this probably immediately occur 
> to the
> reader, neither of which is very satisfactory in the real world, as I
> understand it.
>
> One answer to this would be to "fix the application layer protocol" to
> either turn off the retransmissions at this layer.    However, as a
> practical matter, my understanding is that the application layer 
> protocols
> in question are "deeply embedded" in signalling transport and storage
> systems, and that while is "might be nice" to be able to do this, it is
> simply not feasible.
>
> Another approach, is to apply the end-to-end argument, and run these 
> over a
> completely best-effort transport, and rely exclusively on the
> retransmission built into the application layer protocol.    A problem 
> here
> is that the application layer protocol's retransmission strategies 
> don't
> adapt well to the underlying network, not in the way that the
> retransmissions in TCP and SCTP adapt.   (There may be other reasons as
> well that this is not feasible... I'll rely on my colleagues to fill 
> in the
> gaps)
>
> The alternative approach is to use the timed-reliability service 
> provided
> by PR-SCTP to indicate that messages that will be retransmitted by the
> upper layer after, for example, 5 seconds, should be abandonded by the
> transport layer once that 5 seconds has elapsed.   Again, this results 
> in
> an actual _reduction_ in "junk data", i.e. a redundant retransmission 
> of a
> particular message X, which in the absence of the PR-SCTP timed 
> reliabilty
> service, ends up appearing more than once in the reliable data stream.
>
> Therefore, the bottom line is that PR-SCTP helps to keep redundant 
> data,
> or data that has become stale off the network.    None of this data was
> "junk data" when it was first transmitted, which seems to be one of the
> fears or doubts that is being thrown at PR-SCTP.
>
>
> Regards,
> Phill
>
> -- Phillip Conrad, Asst Professor, CIS Dept. Temple University
> -- Philadelphia PA 19122    conrad@acm.org
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>

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Reiner Ludwig wrote:
> I'd like to add an example for people to think about ...
> 
> If you try to run VoIP over TFRC, you'll most likely be disappointed by 
> the quality. The reason is that when A talks, then (usually) B is 
> silent, and vice versa. I.e., a talkspurt A>B is followed by a talkspurt 
> B>A is followed by a talkspurt A>B, ....

It seems odd to consider TFRC for this case. VoIP requires communication 
that is decidedly outside the behavior patterns of TCP; making VoIP 
TCP-friendly seems bizzare.

Padding real data with dummy data for the sole purpose of being 
TCP-friendly seems even more bizzare. This would interfere with some 
potentially nice aggregation of VoIP on trunk links.

> The "easy fix" is for B to send dummy data (or even potentially useful 
> data like pictures, ads, calendar sync data, etc.) for the sole purpose 
> of maintain the TFRC rate above the required coder rate. B would be a 
> good Internet citizen in the sense that it is always TCP friendly.

Sending dummy data to quench other TCPs from getting real work done 
isn't being a good Internet citizen; it's a denial-of-service attack, of 
sorts.

The real error here, IMO, is considering TCP-friendly to be equivalent 
to Internet-friendly. There has to be a good definition of the latter, 
and that is what we should really be striving for.

Joe


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At 11:59 AM 4/4/2003 -0500, Mark Allman wrote:

>Michael rightly noted that I wrote a whole lot of ambiguous rubbish
>the last time around.  Durned context senssative languages.  So, I
>am going to try again...
>
> > If such a service definition existed ... there would be *no need*
> > for the application to send N times via APF as you suggest.  We
> > then let PR-SCTP (the transport ... not the application) do the
> > "accounting" and it will only send once (if successful) the
> > majority of the times .... and only sends N times when it was
> > actually unsuccessful for [N-1] prior times. Thus this service
> > definition would make PR-SCTP have a "N persistence" property.
> >
> > This, in my book, is an incentive for the app to behave better
> > than sending "N times" in a brute force manner.
>
>No need, but maybe a desire.
>
>Say the app sends the same message to the transport twice.  The
>transport then sends both copies out (in two segments).  Now, say
>one of these segments gets dropped but the other arrives fine and is
>delivered to the receiving app.  By some metrics that offers better
>performance than the case when the sending app gives the sending
>transport only one copy and lets the transport retransmit as
>necessary (which adds an extra ~RTT before the receiving app gets
>the data).


Ok. Now I get your point. You are specifically looking at "delay
performance" ... not other "performance metrics".

>If the sender has to waste its own resources (buffer
>space, cwnd space, etc.) to retransmit the lost segment (that is now
>useless) then I think there is *some* incentive against using such a
>brute force method.  If the sender can say "oh, nevermind on that
>lost segment" and clean up our own resources then it seems like
>there is no incentive not to waste network resources if one wanted
>lots of redundancy.

You have a point if you desire/need redundancy AND delay performance
less than an RTT. Obviously any (partially) reliable transport
protocol can't do that job without being (knowingly) wasteful
(i.e., sending data twice).

I think these applications - by design necessity - need application
layer design (e.g., FEC on voice packets).

But even the vast majority of non-interactive streaming applications
could benefit from an additional playout delay of a few RTTs .... and
use either the "timed reliability" or "N persistence" flavors of PR-SCTP.
For these applications ... let the transport layer use the resources
(buffer space, etc.) profitably and let it discard "stale information".
To do otherwise, the application needs to get as smart as the
average tsvwg'er.


>I hope that is a bit more clear.  Sorry for any confusion I caused.

Got it this time .. thanks.

Michael Ramalho

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

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Subject: Re: [Tsvwg] redundant data - A TFRC Example
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: ext Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
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Reiner,

On Fri, 2003-04-04 at 02:52, ext Reiner Ludwig wrote:
> I'd like to add an example for people to think about ...
> 
> If you try to run VoIP over TFRC, you'll most likely be disappointed by the 
> quality. The reason is that when A talks, then (usually) B is silent, and 
> vice versa. I.e., a talkspurt A>B is followed by a talkspurt B>A is 
> followed by a talkspurt A>B, ....
> 
> While B is silent and if the talkspurt from A is long enough then B's TFRC 
> send rate will decay to zero (or at least often enough below the required 
> coder rate). Andrei Gurtov did this experiment and told me how 'broken' 
> such a voice conversation becomes.
> 

The conversation will be equally "broken" if you don't follow TFRC in
this case. Sending a lot of data (after waiting for a long time) does
not necessarily mean that the quality will improve--the data also need
to reach the receiver. TFRC tries its best to send only as much data as
will safely reach the sender after a given time. 

What's worse in case of not following TFRC is that it might force some
TCP connections to timeout.

BTW, if a connection has waited for such a long time that it's data rate
is so low, why can't it wait a little longer and buffer the data on the
receiver side before plying it out. This should mitigate the problem if
not solve it sompletely. (Please note rate reduction happends every RTO,
which is several times the RTT.) 

I think TFRC (along with DCCP) does a great job. 

> The "easy fix" is for B to send dummy data (or even potentially useful data 
> like pictures, ads, calendar sync data, etc.) for the sole purpose of 
> maintain the TFRC rate above the required coder rate. B would be a good 
> Internet citizen in the sense that it is always TCP friendly. Still, B is 
> sending data it does't have to send necessarily.
> 

You need to define 'necessary' here. Let's take a different example. Say
there are 10 people on a LAN which are accessing the same web page. This
is also an example of 'unnecessary' data. So should the transport layer
now worry about these things too. If I am sitting on a fiber cable, I
won't be able to tell whether the data for different web clients is
unncessary or whether the data from TFRC is unnecessary. How do we draw
the line here.

> Should the IETF set up rules that would limit the sending of such 
> potentially useless data? Do we have hope that such rules would be obeyed 
> by application and protocol designers?
> 
> I could imagine VoIP application designers that would let themselves be 
> convinced to move away from uncontrolled UDP to RTP/DCCP, but probably not 
> much further. I.e., their view could be: "We ensure that our flows behave 
> TCP friendly, but what we do with the available bandwidth is our business."
> 

That's fine as far as transport layer is concerned. Application
protocols desiners, who're as concerned about bandwidth as transport
layer protocol designers are, will certainly find the best approach (for
example, if the sustainable delay before playout is T = x*RTO, then the
application will be happy not to send any 'unnecessary' data until Xth
RTO. Okay that's not the right calculation, but you get the idea...)

These issues are upto the application to figure out. The transport layer
should not worry about them, IMO. 

--yogesh

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On 4 Apr 2003, Yogesh Prem Swami wrote:

> BTW, if a connection has waited for such a long time that it's data rate
> is so low, why can't it wait a little longer and buffer the data on the
> receiver side before plying it out.

Because his traffic example was VoIP, but as Joe pointed out... VoIP is
probably not a good example.

~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] redundant data - A TFRC Example
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
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On Fri, 2003-04-04 at 14:32, ext Armando L. Caro Jr. wrote:
> On 4 Apr 2003, Yogesh Prem Swami wrote:
> 
> > BTW, if a connection has waited for such a long time that it's data rate
> > is so low, why can't it wait a little longer and buffer the data on the
> > receiver side before plying it out.
> 
> Because his traffic example was VoIP, but as Joe pointed out... VoIP is
> probably not a good example.
> 

To the best of my knowlege, even VoIP data needs to be buffered if for
no other reason than to get rid of jitter. Of course how long it needs
to wait is a different question.

--yogesh


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Subject: Re: [Tsvwg] redundant data - A TFRC Example
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Hi, Yogesh,

--- Yogesh Prem Swami <yogesh.swami@nokia.com> wrote:
> 
> To the best of my knowlege, even VoIP data needs to be
> buffered if for
> no other reason than to get rid of jitter. Of course how long
> it needs
> to wait is a different question.

Yeah, but - "buffering" is relative, as you point out -
buffering streaming traffic for 50-ms for a less-jittery playout
isn't the order of magnitude we talk about for end-to-end
transport roundtrip times, right? 

People start noticing things at about 100-200 ms - if you start
to talk and the other person doesn't stop talking for 400 ms
because the voice stream was buffered, most people find this
disconcerting. It's something closer to "push-to-talk" than
voice traffic we know and love...

Spencer
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Subject: Re: [Tsvwg] redundant data - A TFRC Example
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Believe it or not, I don't always post whatever enters my mind.
I was trying to think of a way to say what Joe said, much better
than any way that I'd come up with, so far:

--- Joe Touch <touch@ISI.EDU> wrote:
> The real error here, IMO, is considering TCP-friendly to be
> equivalent 
> to Internet-friendly. There has to be a good definition of the
> latter, 
> and that is what we should really be striving for.

I'm really curious where a definition of "internet-friendly"
would begin, if it's to be different from "TCP-friendly". I'm
assuming that reducing sending rates in the face of sustained
congestion is the floor we're not willing to go beneath - is
there a higher floor that we all agree on?

Spencer
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Subject: Re: [Tsvwg] redundant data - A TFRC Example
From: Yogesh Prem Swami <yogesh.swami@nokia.com>
To: ext Spencer Dawkins <spencer_dawkins@yahoo.com>
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Hi Spencer,

On Fri, 2003-04-04 at 15:29, ext Spencer Dawkins wrote:
> Hi, Yogesh,
> 
> --- Yogesh Prem Swami <yogesh.swami@nokia.com> wrote:
> > 
> > To the best of my knowlege, even VoIP data needs to be
> > buffered if for
> > no other reason than to get rid of jitter. Of course how long
> > it needs
> > to wait is a different question.
> 
> Yeah, but - "buffering" is relative, as you point out -
> buffering streaming traffic for 50-ms for a less-jittery playout
> isn't the order of magnitude we talk about for end-to-end
> transport roundtrip times, right? 

> People start noticing things at about 100-200 ms - if you start
> to talk and the other person doesn't stop talking for 400 ms
> because the voice stream was buffered, most people find this
> disconcerting. It's something closer to "push-to-talk" than
> voice traffic we know and love...

I completely agree. The point I was trying to make is it's not upto the
transport layer to worry about these things. TFRC says: this is what the
data rate an application will get. It's upto the application to decide
the right strategy to get the data rate it wants. If that requires
sending unnecessary data or buffering them, then that's fine as far as
transport layer is concerned.

--yogesh  

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From: Phillip Conrad <conrad@acm.org>
Subject: Re: [Tsvwg] pr-sctp
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I want to apologize to Mark Allman for inadvertently responding to a private note on the open list.

This was an honest mistake on my part; Mark's note was a response to mine, and so had [tsvwg] in the subject line.   I mistakenly concluded that it was an on-list reply; I should have looked more carefully at the header before responding.

Sincerely,
Phill Conrad

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Spencer Dawkins wrote:
> Believe it or not, I don't always post whatever enters my mind.
> I was trying to think of a way to say what Joe said, much better
> than any way that I'd come up with, so far:
> 
> --- Joe Touch <touch@ISI.EDU> wrote:
> 
>>The real error here, IMO, is considering TCP-friendly to be
>>equivalent 
>>to Internet-friendly. There has to be a good definition of the
>>latter, 
>>and that is what we should really be striving for.
> 
> I'm really curious where a definition of "internet-friendly"
> would begin, if it's to be different from "TCP-friendly". I'm
> assuming that reducing sending rates in the face of sustained
> congestion is the floor we're not willing to go beneath - is
> there a higher floor that we all agree on?
> 
> Spencer

One challenge is the implicit assumption in defining friendliness in 
terms of rate. For some applications, rate is meaningless. For others, 
the timescale is critical - one RTT, multiple RTTs, etc.

TCP analysis generally assumes sustained offered load, with a goal of 
generating stable sending rates over many RTTs. For VoIP, those goals 
don't apply.

Here are some initial definitions of Internet-Friendly:

	- gets 1/Nth of network capacity when there are N flows
		- not more (fair to others)
			implicitly allowing the other N-1
			flows the whole of the capacity
			when there is no data to send
		- not less (fair to self/upper layers)

	- reacts to known network conditions within 1 RTT
		the key here is the term "known" -
		it can't be expected to read minds

	- adds load in a 'gentle' fashion

This might, e.g., allow VoIP to send lots of little bursts, where, e.g., 
each burst is O(10 packets), where the average sending rate over 100's 
of RTTs is stable, but need not be over 1 RTT or even 10's of RTTs.

That wouldn't be TCP-friendly, but maybe that's because TCP expects that 
everyone who backs off does so when the congestion occurs (assuming that 
all connections are constantly active), rather than when they notice it 
(which may be equally reasonable).

Joe

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Date: Sat,  5 Apr 2003 21:24:36 -0600
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] redundant data - A TFRC Example
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On 4/4/03, Joe Touch wrote:

>
>One challenge is the implicit assumption in defining
>friendliness in terms of rate. For some applications, rate
>is meaningless. For others, the timescale is critical -
>one RTT, multiple RTTs, etc.
>
>TCP analysis generally assumes sustained offered load,
>with a goal of generating stable sending rates over many
>RTTs. For VoIP, those goals don't apply.
>
>Here are some initial definitions of Internet-Friendly:
>
>   - gets 1/Nth of network capacity when there are N flows
>       - not more (fair to others)
>           implicitly allowing the other N-1
>           flows the whole of the capacity
>           when there is no data to send
>       - not less (fair to self/upper layers)
>
>   - reacts to known network conditions within 1 RTT
>       the key here is the term "known" -
>       it can't be expected to read minds
>
>   - adds load in a 'gentle' fashion
>
>This might, e.g., allow VoIP to send lots of little
>bursts, where, e.g., each burst is O(10 packets), where
>the average sending rate over 100's of RTTs is stable, but
>need not be over 1 RTT or even 10's of RTTs.
>
>That wouldn't be TCP-friendly, but maybe that's because
>TCP expects that everyone who backs off does so when the
>congestion occurs (assuming that all connections are
>constantly active), rather than when they notice it (which
>may be equally reasonable).
>

ATM QoS differentiates not only the sustained rate, but a
peak rate and latency. I'm not suggesting that IP networks
are capable of dealing with ATM type controls, but that at
least we should recognize the same range of application
bandwidth requirements. As you properly point out, all
current work is horrendously biased to low-latency sustained
throughput.

Caitlin Bestler
http://asomi.com/CaitlinBestler/
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Armando L. Caro Jr. wrote:
> On Wed, 2 Apr 2003, Michael A. Ramalho wrote:
> 
> 
>>But to be emphatic .... under congestion this is exactly the behavior you want.
>>This is because every bit of information you send that the receiver can't
>>profitably
>>use (because it is stale) is information that is sent *before more useful
>>information*.
>>Therefore the process of re-sending stale information is *delaying* the
>>sending of
>>useful information (potentially making it stale also ... under extended
>>congestion
>>episodes).
> 
> 
> I agree with the concept of PR-SCTP. I want to emphasize that bundling
> FTSNs with data should be done whenever possible. I think that sending a
> FTSN in a packet of its own is not much better than re-sending the stale
> data. Both will be triggered at the same point. Both will take the same
> time to reach the receiver. And if lost, both suffer the same
> consequences. The only improvement is that flightsize gets reduced and new
> data may be sent.
> 
> Section 3.5 F2) emphasizes that bundling should be done, but just
> wondering out loud... should PR-SCTP implementations reserve room
> for FTSNs in all packets. If data chunks are not fragmented properly
> when queued, then SCTP cannot refragment them later when bundling
> is needed. The draft says that a sender MAY wait to bundle FTSNs, but is
> this wording loose enough to allow SCTP to wait for newly queued data to
> bundle with. In other words, data already queued may be fragmented too
> large for bundling, but newly queued data may be fragmented small enough
> for bundling. ...I don't think I like this idea. Maybe simply reserving
> room for some minimum FTSN size is good enough.
> 

You have the same issue with SACK's as well... one might be wise
when implementing to set the fragmentation point SCTP uses to
be small enough for a minimal SACK OR a F-TSN..

> 
>>Note: The congestion condition is not simply characterized by loss .... but
>>additionally by duration (in time) of the congestion event (as this has an
>>effect on "staleness"). We need to consider the time/delay element in
>>the congestion episode.
> 
> 
> I don't follow how this relates to FTSNs.
> 
> 
>>In this thought process we might consider other PR-SCTP service definitions
>>that are more appropriate to the applications to consider under
>>overload or congestion scenarios. The creation of other pr-sctp service
>>definition
>>to help "govern" the sending of data would be welcome. However, the F-TSN
>>mechanism is the only thing required "on the wire" for PR-SCTP.
> 
> 
> I like that the PR-SCTP mechanism is independent of the service
> definition. However, I cannot think of applications that could use
> anything but timed-reliability.
> 

I can, one great use would be for applications such as IP-Fix... where
I "want" to get data across on peg counts but if my router that is
reporting the information gets TOO busy I would prefer to drop some
of the peg counts .. too busy being to much memory allocated to the
"peg" counts... at the same time there are things like templates that
MUST make it across...

Note no time is involved.. it has to do with buffering capacity of
the router and busy-ness..

We will be working internally at cisco in getting a prototype up
in IOS that uses PR-SCTP shortly.. if I can ever stop traveling :-0

R

> ~armando
> 
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 
> 


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

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Mark Allman wrote:
> Michael rightly noted that I wrote a whole lot of ambiguous rubbish
> the last time around.  Durned context senssative languages.  So, I
> am going to try again...
> 
> 
>>If such a service definition existed ... there would be *no need*
>>for the application to send N times via APF as you suggest.  We
>>then let PR-SCTP (the transport ... not the application) do the
>>"accounting" and it will only send once (if successful) the
>>majority of the times .... and only sends N times when it was
>>actually unsuccessful for [N-1] prior times. Thus this service
>>definition would make PR-SCTP have a "N persistence" property.
>>
>>This, in my book, is an incentive for the app to behave better
>>than sending "N times" in a brute force manner.
> 
> 
> No need, but maybe a desire.
> 
> Say the app sends the same message to the transport twice.  The
> transport then sends both copies out (in two segments).  Now, say
> one of these segments gets dropped but the other arrives fine and is
> delivered to the receiving app.  By some metrics that offers better
> performance than the case when the sending app gives the sending
> transport only one copy and lets the transport retransmit as
> necessary (which adds an extra ~RTT before the receiving app gets
> the data).  If the sender has to waste its own resources (buffer
> space, cwnd space, etc.) to retransmit the lost segment (that is now
> useless) then I think there is *some* incentive against using such a
> brute force method.  If the sender can say "oh, nevermind on that
> lost segment" and clean up our own resources then it seems like
> there is no incentive not to waste network resources if one wanted
> lots of redundancy.
> 
Mark:

I think I understand your point but I don't see how PR-SCTP encourages
an application designer to send extra data...

Lets say I send two copies of packet X in the hopes of
getting it through with an application sequence number....

Now what have I done with PR-SCTP ... taken up twice the
amount of cwnd space... each copy eats into how much
I can send.

My problem is I am missing the incentive for this bad behavior
that PR-SCTP provides..

The dis-incentive for this behavior you mention is head of
line blocking (for TCP).. but what is my dis-incentive for unordered
SCTP? The same for PR-SCTP, if I send it twice I eat up twice
my cwnd this is the only dis-incentive that I can see and PR-SCTP
does not change compared to SCTP-unordered....

Am I missing a dis-incentive... or to flip the coin.. an incentive
to send extra non-needed data... I don't see it...

If I really am going to send each 'packet X' twice.. I can do the
same thing with SCTP/Un-ordered, UDP or DCP...

I agree that TCP has the extra dis-incentive of head of line blocking,
but semi-ordered SCTP and unordered SCTP do not have this problem.
For example take away unordered SCTP and I the application designer will
setup 100 streams in SCTP. Send copy 1 of Packet X on stream 0, send the
next copy on stream 1. Then send Packet Y on stream 2 and copy on stream 
3.. etc.

Again.. what is my dis-incentive for the above behavior? I eat up
cwnd of my association.

Even in TCP I can do the above... setup multiple connections and
then not only do I get the ability to replicate everything down the
second connection.. I can also get two sets of cwnd's.

Am I missing something in your side of the bolgna? I just can't find
the incentive for bad behavior over what is already out there...

R




> I hope that is a bit more clear.  Sorry for any confusion I caused.
> 
> 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
> 
> 


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

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> I want to apologize to Mark Allman for inadvertently responding to
> a private note on the open list.
> 
> This was an honest mistake on my part; Mark's note was a response
> to mine, and so had [tsvwg] in the subject line.  I mistakenly
> concluded that it was an on-list reply; I should have looked more
> carefully at the header before responding.

Thank you, sir.

allman


--
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To: Joe Touch <touch@ISI.EDU>
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] redundant data - A TFRC Example
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At 21:32 04.04.2003, Joe Touch wrote:
>The real error here, IMO, is considering TCP-friendly to be equivalent to 
>Internet-friendly. There has to be a good definition of the latter, and 
>that is what we should really be striving for.

I agree 100 percent, since that's exactly the point I wanted to raise when 
I presented the few slides at the last TSVWG meeting:
http://www.cs.helsinki.fi/u/gurtov/reiner/TCP_FEC.pdf
http://www.cs.helsinki.fi/u/gurtov/reiner/TCP_FEC.ppt

I just tried to paint a bit in black & white hoping that that would provoke 
some deeper thinking about this :-)

But afterall, if I recall right then VoIP was mentioned as one the 
applications for TFRC in those drafts ...

///Reiner

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Folks-

I do not want to continue to repeat myself on the list.  After
reading through this thread and having some on-list and off-list
dicussions I am left with the following thoughts/clarifications
(/FUD - depending on your viewpoint).

  * I never said and hopefully did not insinuate that there was no
    use/need/application for PR-SCTP.  So, for all of the folks who
    have written me enumerating all sorts of applications: I agree,
    there are compelling apps for PR-SCTP (and, more generally
    partially reliable transports).

    I am not advocating throwing the baby out with the bath water.
    I am wondering about whether we should think about enforcing a
    little more network friendliness because we can.

  * Just because a transport sends at a rate that is dictated by CC
    doesn't mean it's doing so in a terribly useful or friendly
    manner.  Just because a transport is allowed to send at X bps
    should not give it the right to send just anything at X bps.  If
    lots of the traffic is redundant the transport is using network
    resources that could be put to better use (to accomplish useful
    work).  In my opinion, *** being a well-behaved transport is not
    just about sending at the "right" rate. *** (Although, note that
    sending at the "right" rate is *part* of being network
    friendly.)

  * PR-SCTP is network friendly in one sense because it can prevent
    needless retransmits from being sent (i.e., because an
    application can no longer use the data).  This is a Good Thing.

    But, if we're using this PR functionality to squeltch a high
    percentage of messages from being retransmitted then I think we
    turn this Good Thing into a Bad Thing.  Sure, the transport is
    still preventing needless retransmissions, but at the same time
    it is acknowledging that it's cluttering up the network with a
    lot of packets that are not needed.

    IMO, a little PR is fine, but lots of PR is not.  (Where I
    purposely leave "a little" and "a lot" ambiguous because I don't
    know how to precisely define them.)

  * My own personal opinion is there are a number of angles to look
    at how "network friendly" a transport protocol happens to be.
    I believe that one aspect of this is how much useless data the
    transport injects into the network.  

    In this context I believe we need to think about the
    *incentives* a transport protocol offers to applications using
    it.  In addition, we need to look at the *opportunity* the
    transport has to police the sending of redundant data.

    (And, note, that this is not something I have selectivly applied
    to SCTP.  I have explored this in TCP as well.  For instance, in
    work that Ethan Blanton and I have done on making TCP robust to
    reordering one of the goals is to adjust the duplicate ACK
    threshold so that TCP does not pollute the network with lots of
    needless retransmissions that are caused by packet reordering
    not packet loss.  I.e., I am not attempting to disparage only
    SCTP here.  This is a general issue, but I think PR-SCTP is a
    place to apply anti-pollution strategies.)

  * I want to discuss several transports in the context of the above
    bit about incentives and opportunities, since others have
    brought them up.

    If we look at UDP I don't think there is any dis-incentive
    offered by the transport to prevent the application from sending
    lots of redundant data into the network.  If the application
    sends every message twice in case one of them is lost there is
    really no transport-level problem with that (i.e., this behavior
    does not cost the transport anything).  But, the flip side is
    that there is no opportunity for the transport to do anything
    about it.  Each message sent from the application to the
    transport is just transmitted without any sort of accounting,
    data naming, etc.  So, the transport is ignorant of what is
    being sent.  So, while UDP can be abused by applications that
    choose to do so, there is pretty much nothing that UDP can do
    about it.

    Next, look at TCP (and, really, any ordered, reliable
    transport).  TCP provides a huge dis-incentive for applications
    to send multiple copies of a message for redundancy purposes.
    TCP ensures that all copies will arrive -- and in the order the
    were given to the sending transport by the sending application.
    There is head-of-line blocking so there is just no good reason
    to send an application message twice to try to get better
    performance (via some metric).  (Note: TCP itself can be more or
    less careful about what it resends and then starts getting into
    these issues.  For instance, early retransmit allows for the
    resending of data with a less robust loss indicator than we
    currently use.  That *could* have the impact of inserting a
    bunch of useless data into the network in the path that has
    persistent reordering.  This should be prevented.)

    Next up is unordered, reliable SCTP.  In this case, applications
    can send multiple copies of some message for some possible gain.
    Since there is no head-of-line blocking it may make sense
    (performance-wise) for an application to send multiple copies of
    some message to the transport and then throw any extra copies
    that arrive at the receiving application in the bit bucket.  The
    dis-incentive provided by the transport in this case is that the
    extra messages (some of which are useless) take space in
    buffers, take space in the congestion window and take
    (re)transmission opportunities from other data.  And, all
    messages take these resources until they are successfully
    delivered to the receiving transport.  Depending on the
    application these tradeoffs an application designer might think
    these tradeoffs are worth it and the abuse to the network will
    happen.  But, that is not the whole story.  The other side of
    this variant of SCTP is that the transport is "out of the loop"
    on what is redundant and what is not.  Since all the redundancy
    is happening in application land (as in the UDP case) SCTP has
    no real clue about what is going on and therefore has no
    opportunity to police the stream and prevent redundant data.

    Finally, we get to PR-SCTP -- where I believe incentive and
    opportunity finally cross paths.  Think of the application
    passing down multiple copies of the same message to the
    transport with short lifetimes.  So, the job of the extra copies
    of the message is to "back up" the first and get to the
    application if the first is lost (quicker than a retransmission
    of the first could).  Now the transport is giving an incentive
    to the application to do this and has an opportunity to prevent
    excessive (ab)use of it.  The dis-incentive to do this is quite
    similar to the case above for unordered, reliable SCTP -- that
    is, the redundant data takes cwnd space, buffer space and
    transmission opportunities.  But, PR-SCTP goes a step further
    because the application (in effect) gets to tell the transport
    to release the resources quickly.  So, rather than being
    required to use these resources until a loss is repaired they
    can be freed and used for other things.  This is a Good Thing in
    moderation and a Bad Thing in excess, IMO.  In my opinion, the
    dis-incentive for PR-SCTP is *less* than the dis-incentive for
    unordered SCTP.  But, the really important bit that distiguishes
    PR-SCTP from the unordered, reliable SCTP is that PR-SCTP is "in
    the loop" and knows exactly what is going on (i.e., which data
    was sent & lost, but will not be re-sent).  And, hence, PR-SCTP
    can do something about it to prevent excessive (ab)use of the PR
    mechanism.

    So, my opinion is still that PR-SCTP *does* provide *incentive*
    for application abuse *and* yet also has an *opportunity* to
    police it and to ensure that the use of the PR feature is not
    excessive.  And, I believe that having both of those is the
    fundemental discriminator between PR-SCTP and all of our
    previous (popular) transports.

  * It's been suggested that I am advocating things that can be
    accomplished with twisting of other protocols (e.g., multiple
    connection TCP).  Well, I am not going on a long rant here --
    although it is tempting. ;) In any case, fine, I'll admit that
    if you twist and warp things they can be abused.  So, what?
    Does that give us the license to produce lousy protocols?  Did
    the fact that mutliple TCP connections viloates standard TCP CC
    or that UDP does not use CC allow us to design bogus (or no) CC
    into SCTP?  No.  The argument that since we can abuse other
    things we should not have to be careful in the design of PR-SCTP
    is just bogus.  That's the end of my mini-rant -- I am sure you
    can fill in the rest! :)

There is an old tradition in the IETF that says that repeating your
argument over-and-over makes you more-and-more correct.  I don't buy
it (although, alas, sometimes fall into the trap) and so I intend
for this to be my last note in this thread (on tsvwg) because I
don't think I have said much new in here (but, hopefully, in a more
coherent and comprehensive way), or really since the thread
started.  (Besides, if you have made it this far you're all sick of
listening to my bullshit anyway!)

I am intersted in hearing what other folk have to say.

Thanks for listening,
allman


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From mailnull@www1.ietf.org  Mon Apr  7 08:21:00 2003
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From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] redundant data - A TFRC Example
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We must have gone through Reiner's slides REALLY fast. I was a
minute-taker for TSVWG, IIRC, and they didn't sink in during the
meeting at all.

Unfortunate, because they are an excellent jumping-off-point for
the issue we're discussing on this thread.

RFC 3448 includes the following text in the abstract:

   This document specifies TCP-Friendly Rate Control (TFRC). 
TFRC is a
   congestion control mechanism for unicast flows operating in a
best-
   effort Internet environment.  It is reasonably fair when
competing
   for bandwidth with TCP flows, but has a much lower variation
of
   throughput over time compared with TCP, making it more
suitable for
   applications such as telephony or streaming media where a
relatively
   smooth sending rate is of importance.

So it looks like Reiner is remembering correctly that VoIP was a
motivator for TFRC...

Spencer

--- Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se> wrote:
> 
> I agree 100 percent, since that's exactly the point I wanted
> to raise when 
> I presented the few slides at the last TSVWG meeting:
> http://www.cs.helsinki.fi/u/gurtov/reiner/TCP_FEC.pdf
> http://www.cs.helsinki.fi/u/gurtov/reiner/TCP_FEC.ppt
> 
> I just tried to paint a bit in black & white hoping that that
> would provoke 
> some deeper thinking about this :-)
> 
> But afterall, if I recall right then VoIP was mentioned as one
> the 
> applications for TFRC in those drafts ...
> 
> ///Reiner
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Subject: [Tsvwg] Re: pr-sctp -- final thoughts from me
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Folks-

> I do not want to continue to repeat myself on the list.  

I did not mean to discourage on-list chat by the above line, which,
judging from a couple of private emails I may have done.  My point
was that I feel like I am repeating myself and so *I* plan to not
argue point-by-point any longer.  Other SHOULD send thoughts to the
list.  Sorry for any misunderstanding...

allman


--
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Call for Papers
==============

Internetworking 2003, June 22-24, 2003, San Jose, California
http://www.caitr.org/internetworking03/index.htm

REMINDER: Deadline for submissions is April 11, 2003

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

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

Conference Technical Co-chairs: 
- Dr. Maurice Gagnaire, ENST, France 
- Daniel Awduche 

Technical Program Committee of the Internetworking 2003 Conference: 
- Roberto Sabella, Erisson 
- Dr. Moshe Zukerman, Univ. of Melbourne 
- Nada Golmie, NIST 
- Dr. Guy Pujolle, LIP6, France 
- Dr. Samir Tohme, ENST, France 
- Stefano Trumpy, Italian National Research Council 
- Dr. Ibrahim Habib, City Univ. of NY 
- Dr. Vishal Sharma, Metanoia 
- Dr. Parviz Yegani, Cisco Systems 
- Dr. G.S. Kuo 
- Dr. Abbas Jamalipour, Univ. of Sydney 
- Dr. Hussein Mouftah, Univ. of Ottawa 
- James Kempf 
- Elizabeth Rodriguez, Co-chair, IETF Working Group on IP Storage
- Dr. Ferit Yegenoglu, Isocore 
- Dr. Ali Zadeh, George Mason University 
- Tony Przygienda, Co-chair, IETF Working Group on IS-IS for IP Internets
- Ran Canetti, Co-chair, IETF Working Group on Multicast Security


--0-1491182717-1049731467=:26908
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<P>Call for Papers<BR>==============</P>
<P>Internetworking 2003, June 22-24, 2003, San Jose, California<BR><A href="http://www.caitr.org/internetworking03/index.htm">http://www.caitr.org/internetworking03/index.htm</A></P>
<P>REMINDER: Deadline for submissions is April 11, 2003</P>
<P>The Internetworking 2003 Technical Program Committee cordially invites you to submit proposals for original, unpublished presentations focusing on internetworking technologies in the IP, optical, and wireless domains. Summaries not exceeding 250 words can be submitted to <A href="mailto:submissions@caitr.org">submissions@caitr.org</A> for review and possible inclusion in the program, no later than April 11, 2003. Topics of interest include, but are not limited to the following:</P>
<P>- Voice over IP (VoIP)<BR>- IP Video Conferencing<BR>- Storage Area Networks (SANs)<BR>- Unicast and Multicast Routing and Convergence<BR>- QoS Routing<BR>- Network Security and Service Integration<BR>- Operational Support Systems<BR>- Virtual Private Networks<BR>- Internetworking Wireless LANs and 3G Wireless Networks<BR>- IP-based Infrastructure for Wireless Networks<BR>- Internetworking IP and Optical Networks<BR>- Internetworking MPLS with Legacy ATM and Frame Relay Networks<BR>- Transition from IPv4 to IPv6 and interworking<BR>- Pervasive Computing<BR>- High Speed Transport Layer Protocols<BR>- Peer to Peer Networking and Grid Computing<BR>- Video Teleconferencing (VTC) <BR>- 802.11 Hotspots</P>
<P>Conference Technical Co-chairs: <BR>- Dr. Maurice Gagnaire, ENST, France <BR>- Daniel Awduche </P>
<P>Technical Program Committee of the Internetworking 2003 Conference: <BR>- Roberto Sabella, Erisson <BR>- Dr. Moshe Zukerman, Univ. of Melbourne <BR>- Nada Golmie, NIST <BR>- Dr. Guy Pujolle, LIP6, France <BR>- Dr. Samir Tohme, ENST, France <BR>- Stefano Trumpy, Italian National Research Council <BR>- Dr. Ibrahim Habib, City Univ. of NY <BR>- Dr. Vishal Sharma, Metanoia <BR>- Dr. Parviz Yegani, Cisco Systems <BR>- Dr. G.S. Kuo <BR>- Dr. Abbas Jamalipour, Univ. of Sydney <BR>- Dr. Hussein Mouftah, Univ. of Ottawa <BR>- James Kempf <BR>- Elizabeth Rodriguez, Co-chair, IETF Working Group on IP Storage<BR>- Dr. Ferit Yegenoglu, Isocore <BR>- Dr. Ali Zadeh, George Mason University <BR>- Tony Przygienda, Co-chair, IETF Working Group on IS-IS for IP Internets<BR>- Ran Canetti, Co-chair, IETF Working Group on Multicast Security<BR></P>
--0-1491182717-1049731467=:26908--
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From mailnull@www1.ietf.org  Mon Apr  7 13:59:15 2003
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Date: Mon,  7 Apr 2003 12:52:14 -0500
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] pr-sctp -- final thoughts from me
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On 4/7/03, Mark Allman wrote:

> Next, look at TCP (and, really, any ordered, reliable
> transport).  TCP provides a huge dis-incentive for
> applications to send multiple copies of a message for
> redundancy purposes. TCP ensures that all copies will
> arrive -- and in the order the were given to the sending
> transport by the sending application. There is
> head-of-line blocking so there is just no good reason to
> send an application message twice to try to get better
> performance (via some metric).  (Note: TCP itself can be
> more or less careful about what it resends and then
> starts getting into these issues.  For instance, early
> retransmit allows for the resending of data with a less
> robust loss indicator than we currently use.  That
> *could* have the impact of inserting a bunch of useless
> data into the network in the path that has persistent
> reordering.  This should be prevented.)

TCP is actually fully compatabible with this type of
cheating, without any cost to the abusive sender.

The sender simply opens multiple redundant connections
(leaving even less capacity for those pesky other users).
The same TCP segments are then sent on all paths.

Once *any* connection has acked material, the sender simply
discards the buffers. The retention is based on the
*fastest* ack, not the slowest.

Any connection that ultimately needs the now missing data is
simply reset, and then re-established. More network traffic,
but what's that compared to our reckless applications buffer
space? It takes time, so better have *several* spare
connections.

As you already concede, UDP does not have the ability to
detect redundant data transmission by the application.
Therefore it is fully compatible with abusive programs.

Most uses of PR-SCTP would have otherwise used UDP, not
TCP or SCTP. So there is certainly no redudction in the
networks ability to prevent abusive traffic.

Beyond that, what change would be made anyway?

And more importantly, wouldn't it be strictly a local
interface question? The wire messages that invalidate
the need to wait for previously sent data chunks do
not cite a reason.

Whatever safety checks/guidelines might be agreed upon,
they will not be wire visisble. I am also skeptical that
any useful policies could be drafted with any teath beyond
"SHOULD NOT"s. It is very hard to encode real-time
enforcement that NEVER has a false positive. And any
real-time enforcement MUST NOT block ANY legitimate
practice (no matter how strange).

Caitlin Bestler
http://asomi.com/CaitlinBestler/
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Mark,

My comments, as promised. Comments are in-line.

Michael Ramalho

At 08:05 AM 4/7/2003 -0400, Mark Allman wrote:
>
>Folks-
>
>I do not want to continue to repeat myself on the list.  After
>reading through this thread and having some on-list and off-list
>dicussions I am left with the following thoughts/clarifications
>(/FUD - depending on your viewpoint).
>
>   * I never said and hopefully did not insinuate that there was no
>     use/need/application for PR-SCTP.  So, for all of the folks who
>     have written me enumerating all sorts of applications: I agree,
>     there are compelling apps for PR-SCTP (and, more generally
>     partially reliable transports).
>
>     I am not advocating throwing the baby out with the bath water.
>     I am wondering about whether we should think about enforcing a
>     little more network friendliness because we can.
>
>   * Just because a transport sends at a rate that is dictated by CC
>     doesn't mean it's doing so in a terribly useful or friendly
>     manner.  Just because a transport is allowed to send at X bps
>     should not give it the right to send just anything at X bps.

Let me understand you. I find TCP a form of socialism ... each
of N sessions get 1/Nth (on very rough average) of
the (bottleneck) bandwidth.

You desire to go beyond socialism ... and not allow what you
see as *applications created* redundant data ... even if you are
within your 1/Nth ... even, perhaps, if you are within the QoS
bandwidth you paid a service provider for? Many people would
disagree with your point of view here for all "better than best-effort"
application environments. I do note that many, if not the majority,
of sigtran environments are QoS provisioned.

Note: I have no problem with minimizing "redundant data" in
a given layer (your transport optimizations examples below).

>If
>     lots of the traffic is redundant the transport is using network
>     resources that could be put to better use (to accomplish useful
>     work).  In my opinion, *** being a well-behaved transport is not
>     just about sending at the "right" rate. *** (Although, note that
>     sending at the "right" rate is *part* of being network
>     friendly.)

OK. This is a "generic goodness" argument .. but is hard or
impossible to enforce.

If an application has to send twice (or send FEC/redundant
information) in order to meet it's delay performance requirements
and it is otherwise abiding in a responsible manner (e.g., paid
for the bandwidth in a QoS-enabled environment or is otherwise
within it's designed transport bounds) .... who are we to say
that the application isn't "network friendly"?

I, as will be seen below, fully agree with minimizing duplicate
data being sent in a given layer (e.g., wholly within transport layer).


>   * PR-SCTP is network friendly in one sense because it can prevent
>     needless retransmits from being sent (i.e., because an
>     application can no longer use the data).  This is a Good Thing.
>
>     But, if we're using this PR functionality to squeltch a high
>     percentage of messages from being retransmitted then I think we
>     turn this Good Thing into a Bad Thing.  Sure, the transport is
>     still preventing needless retransmissions, but at the same time
>     it is acknowledging that it's cluttering up the network with a
>     lot of packets that are not needed.

Beauty (of data) is in the eye of the beholder. I view most of the
data that pr-sctp decides to do without (because of staleness) *is* data
that we would have liked to have received. But given a choice of getting
"more relevant data now" or "getting the old, potentially irrelevant, data
now ... then the more relevant data even later" ... we choose the former.
That does not mean the data which we decide not to re-transmit was
"clutter" when it was originally sent.


>     IMO, a little PR is fine, but lots of PR is not.  (Where I
>     purposely leave "a little" and "a lot" ambiguous because I don't
>     know how to precisely define them.)
>
>   * My own personal opinion is there are a number of angles to look
>     at how "network friendly" a transport protocol happens to be.
>     I believe that one aspect of this is how much useless data the
>     transport injects into the network.

I agree in general. But note you said "transport layer injects"
(not "application layer injects").


>     In this context I believe we need to think about the
>     *incentives* a transport protocol offers to applications using
>     it.  In addition, we need to look at the *opportunity* the
>     transport has to police the sending of redundant data.

I'm a little confused here. Your next paragraph describes
mechanisms *in the transport layer* to limit the amount of
truly useless (re-transmitted) data sent *by the transport
layer*. I have no problem with such goals and wholeheartedly
agree.

But how does the transport layer know of redundant data sent
to it by the application? Surely the redundant data need not
be identical ... it could have been formatted differently? How
would the transport layer know?

It is not the transport's job to police this!


>     (And, note, that this is not something I have selectivly applied
>     to SCTP.  I have explored this in TCP as well.  For instance, in
>     work that Ethan Blanton and I have done on making TCP robust to
>     reordering one of the goals is to adjust the duplicate ACK
>     threshold so that TCP does not pollute the network with lots of
>     needless retransmissions that are caused by packet reordering
>     not packet loss.  I.e., I am not attempting to disparage only
>     SCTP here.  This is a general issue, but I think PR-SCTP is a
>     place to apply anti-pollution strategies.)
>
>   * I want to discuss several transports in the context of the above
>     bit about incentives and opportunities, since others have
>     brought them up.
>
>     If we look at UDP I don't think there is any dis-incentive
>     offered by the transport to prevent the application from sending
>     lots of redundant data into the network.  If the application
>     sends every message twice in case one of them is lost there is
>     really no transport-level problem with that (i.e., this behavior
>     does not cost the transport anything).  But, the flip side is
>     that there is no opportunity for the transport to do anything
>     about it.  Each message sent from the application to the
>     transport is just transmitted without any sort of accounting,
>     data naming, etc.  So, the transport is ignorant of what is
>     being sent.  So, while UDP can be abused by applications that
>     choose to do so, there is pretty much nothing that UDP can do
>     about it.

Agree.


>     Next, look at TCP (and, really, any ordered, reliable
>     transport).  TCP provides a huge dis-incentive for applications
>     to send multiple copies of a message for redundancy purposes.
>     TCP ensures that all copies will arrive -- and in the order the
>     were given to the sending transport by the sending application.
>     There is head-of-line blocking so there is just no good reason
>     to send an application message twice to try to get better
>     performance (via some metric).

As noted previously ... an application could open multiple
TCP connections, though. Not only would this increase the
probability that the message got there with one-way delay, it
would also double it's "share" of the bandwidth (assuming
two TCP connections) and resources (ports, etc.).

We both agree this is bad - and that there is *nothing* we can
do about it from an *applications enforcement* perspective.
The transport layer cannot be expected to "police" for this
redundant data. Store this thought for use later.

>(Note: TCP itself can be more or
>     less careful about what it resends and then starts getting into
>     these issues.  For instance, early retransmit allows for the
>     resending of data with a less robust loss indicator than we
>     currently use.  That *could* have the impact of inserting a
>     bunch of useless data into the network in the path that has
>     persistent reordering.  This should be prevented.)
>
>     Next up is unordered, reliable SCTP.  In this case, applications
>     can send multiple copies of some message for some possible gain.

Of course, unordered reliable SCTP was designed for messages
that were *uncorrelated* ... you are trying to "abuse" the very
premise of design for unordered service (via 100% correlated
messages). Are you suggesting that because it *can* be
abused ... that it shouldn't exist for the services or applications
that it was designed for? I don't think so ... but I'd like to find
the alternative.

>     Since there is no head-of-line blocking it may make sense
>     (performance-wise) for an application to send multiple copies of
>     some message to the transport and then throw any extra copies
>     that arrive at the receiving application in the bit bucket.

The incentive *only exists* if the SCTP session *isn't presently
using it's fair share*. If it is ... then there is the appropriate
disincentive ..... less throughput/goodput and potentially more
delay of the "other" information.

To me ... this is an even greater *disincentive*. Be "greedy" and
it "hurts you more" during the congestion episode! Exactly
when you need performance ... your greediness hurts you!

I fail to see why you think this is an incentive for bad
applications behavior; as the application is "punished" via
poorer (average) performance when the going gets tough.

I understand your desire to minimize "redundant information"
for generic goodness purposes .... but just like we can't
mandate non use of some of the present day RTP/UDP
tricks ... we can't mandate that applications can't do this. Applications
that do as you suggest can potentially get better performance
during non-congestion periods in exchange for poorer performance
during congestion episodes. But at least they can't use more than
their fair share *during congestion* if they use SCTP. This seems
like pretty optimal behavior ... when the going "gets tough" ... you
protect "the network". When the network has excess capacity, you
live with some bad apples.

>The
>     dis-incentive provided by the transport in this case is that the
>     extra messages (some of which are useless) take space in
>     buffers, take space in the congestion window and take
>     (re)transmission opportunities from other data.  And, all
>     messages take these resources until they are successfully
>     delivered to the receiving transport.  Depending on the
>     application these tradeoffs an application designer might think
>     these tradeoffs are worth it and the abuse to the network will
>     happen.

But ... unlike the multiple TCP case ... it still only uses it's
(singular) fair share of the bandwidth.

>  But, that is not the whole story.  The other side of
>     this variant of SCTP is that the transport is "out of the loop"
>     on what is redundant and what is not.  Since all the redundancy
>     is happening in application land (as in the UDP case) SCTP has
>     no real clue about what is going on and therefore has no
>     opportunity to police the stream and prevent redundant data.

True. It isn't the transport's job to police the data. SCTP doesn't
see the data as "redundant" .... and it still doesn't use more than
it's share of the bandwidth.

>     Finally, we get to PR-SCTP -- where I believe incentive and
>     opportunity finally cross paths.  Think of the application
>     passing down multiple copies of the same message to the
>     transport with short lifetimes.

Call me dense ... but if you have a few RTT's to get the data
across ... I still don't understand why you would do this?

Because ... if you have a few RTT's ... you would only send it
once with the appropriate lifetime.

If you do not have a few RTT's and you really should be
using UDP/DCCP with the appropriate FEC.

Am I wrong in thinking that you are attempting to "abuse"
PR-SCTP in a manner in which no transport protocol can protect
against? If you try to formulate a mechanism in which the
transport polices the application layer information to make
sure that it is not sending redundant information - what you
are suggesting is a clear layer violation.

I have no problem with the TCP fast/quick restart tweaks
you mentioned previously ... as they were fixes entirely
*within* the transport layer.

>So, the job of the extra copies
>     of the message is to "back up" the first and get to the
>     application if the first is lost (quicker than a retransmission
>     of the first could).  Now the transport is giving an incentive
>     to the application to do this and has an opportunity to prevent
>     excessive (ab)use of it.

Even if an "application abuser" did this ... there is still less incentive
than multiple TCP sessions .. and a disincentive during congestion
episodes (as above).

>The dis-incentive to do this is quite
>     similar to the case above for unordered, reliable SCTP -- that
>     is, the redundant data takes cwnd space, buffer space and
>     transmission opportunities.  But, PR-SCTP goes a step further
>     because the application (in effect) gets to tell the transport
>     to release the resources quickly.  So, rather than being
>     required to use these resources until a loss is repaired they
>     can be freed and used for other things.

Yes. By design.

>This is a Good Thing in
>     moderation and a Bad Thing in excess, IMO.  In my opinion, the
>     dis-incentive for PR-SCTP is *less* than the dis-incentive for
>     unordered SCTP.

OK. Now you send multiple identical messages with shorter lifetimes.
Let's look at the two cases.

Case One: Lifetimes greater than a few RTTs. There is *no incentive*
to do this ... as a singular message would make it under reasonable
packet error conditions.

Case Two: Lifetimes on the order of a two RTTs or less. The case for
abuse is there ... but .... PR-SCTP *still* won't use it's fair share of
bandwidth. And (PR-)SCTP is still ignorant of whether the data it
has been passed is redundant.

I fail to see how you can do better than this *without layer violating
application <-> transport policing*.

>But, the really important bit that distiguishes
>     PR-SCTP from the unordered, reliable SCTP is that PR-SCTP is "in
>     the loop" and knows exactly what is going on (i.e., which data
>     was sent & lost, but will not be re-sent).  And, hence, PR-SCTP
>     can do something about it to prevent excessive (ab)use of the PR
>     mechanism.
>
>     So, my opinion is still that PR-SCTP *does* provide *incentive*
>     for application abuse *and* yet also has an *opportunity* to
>     police it and to ensure that the use of the PR feature is not
>     excessive.

Please explain how PR-SCTP has an opportunity to police without
violating the application/transport. I don't quite get this ... and you
haven't provided an example of how the transport layer would
determine "redundant data".

Even if an app "abuses" PR-SCTP in the manner you envision ... it
still doesn't get more than it's bandwidth share.

>And, I believe that having both of those is the
>     fundemental discriminator between PR-SCTP and all of our
>     previous (popular) transports.
>
>   * It's been suggested that I am advocating things that can be
>     accomplished with twisting of other protocols (e.g., multiple
>     connection TCP).  Well, I am not going on a long rant here --
>     although it is tempting. ;) In any case, fine, I'll admit that
>     if you twist and warp things they can be abused.  So, what?
>     Does that give us the license to produce lousy protocols?  Did
>     the fact that mutliple TCP connections viloates standard TCP CC
>     or that UDP does not use CC allow us to design bogus (or no) CC
>     into SCTP?  No.  The argument that since we can abuse other
>     things we should not have to be careful in the design of PR-SCTP
>     is just bogus.  That's the end of my mini-rant -- I am sure you
>     can fill in the rest! :)

I'm not trying to be difficult ... I just do not know how to protect
against what you seem to object to without layer violation.


>There is an old tradition in the IETF that says that repeating your
>argument over-and-over makes you more-and-more correct.  I don't buy
>it (although, alas, sometimes fall into the trap) and so I intend
>for this to be my last note in this thread (on tsvwg) because I
>don't think I have said much new in here (but, hopefully, in a more
>coherent and comprehensive way), or really since the thread
>started.  (Besides, if you have made it this far you're all sick of
>listening to my bullshit anyway!)
>
>I am intersted in hearing what other folk have to say.
>
>Thanks for listening,
>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


Michael A. Ramalho, Ph.D.
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.941.708.4650
Cell: +1.941.544.2844

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Folks-

I lied.  I am replying to this thread.  But, just as a clarification
because I chose some unfortunate language in my original note.

> Please explain how PR-SCTP has an opportunity to police without
> violating the application/transport. I don't quite get this
> ... and you haven't provided an example of how the transport layer
> would determine "redundant data".

It is clear to me from Michael's note that I inter-mixed my examples
and my statements of principle too much.  The term "redundant data"
should have been used in the context of my examples only.  What I
was advocating was the transport layer policing "junk data"
("useless data", "data never delivered to the recieving
application"... pick your favorite).  In the case of PR-SCTP the
transport knows what information this is by what the application
tells the transport and what the transport knows about which bits of
data have been lost, retransmitted, not retransmitted, etc.

I *was not* advocating any sort of new communication between
transport and application.  And, I *was not* advocating that the
transport should try to mine extra information from the application
data passed down.

If the policing scheme I was suggesting is a layer viloation then so
is PR-SCTP (which I do not believe is the case).

allman


--
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[Continuing the slightly off-topic thread...  (I'm await my knuckle
 swat, I suppose).]

> You might find this of interest:
> 
> http://www.acm.org/sigcomm/ccr/archive/1999/oct99/allman2.pdf

The other day I wrote up a very quick web page to collect
documents that give tips on effective research (such as the paper
Aaron pointed out).  The web page is at:

    http://roland.grc.nasa.gov/~mallman/tips/

Additional pointers that I have (inadvertently) left off would be
very much welcome and appreciated.

allman


--
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Date: Tue, 08 Apr 2003 13:09:33 -0400
From: Phillip Conrad <conrad@acm.org>
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Subject: [Tsvwg] 6th SCTP Interoperability Testing Event
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Colleagues,
   The 6th SCTP Interoperability Testing Event (what we used to call by a word pronounced someting like "behkov") will be held Jun 1-8 at the University of Delaware.   

For more details, go to http://pel.cis.udel.edu/interop.  You may also contact Phill Conrad <conrad@acm.org> to be placed on a (moderated, no spam) mailing list specifically related to this event.

Thanks,
Phill Conrad

PS: I announced this "tentatively" at the San Francisco IETF, but the event is now "officially on".  Hope to see you in Newark DE soon!

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Subject: [Tsvwg] SCTP RFC 2960 && draft-ietf-tsvwg-sctpsocket-06.txt
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This is a multipart message in MIME format.
--=_alternative 0000357488256D03_=
Content-Type: text/plain; charset="US-ASCII"

To Whom It May Concern:

I have recently learned about the proposal of the new transport protocol 
SCTP and it shows great promise.  I am also new to the IETF working group 
discussions. 

One of the things that I like about SCTP is that it allows you to use many 
virtualized streams over a single connection.  The possibilities for using 
this to replace TCP and use the virtualized streams to simplfy parsing 
bytes streams is excellent.  However, in my reading I didn't come across 2 
ideas that I thought could add to the flexibility of SCTP and I was 
wondering if any discussion had occured on these topics. 

1.  Dynamically adding and removing the number of virtualized streams on a 
connection.
2.  Prioritizing delivery of specific virtualized streams to enhance 
quality of service.

If they have been discussed for the standard and can somone point me to 
further reading on these topics.

Thanks
Ian Barile
--=_alternative 0000357488256D03_=
Content-Type: text/html; charset="US-ASCII"


<br><font size=2 face="sans-serif">To Whom It May Concern:</font>
<br>
<br><font size=2 face="sans-serif">I have recently learned about the proposal
of the new transport protocol SCTP and it shows great promise. &nbsp;I
am also new to the IETF working group discussions. &nbsp;</font>
<br>
<br><font size=2 face="sans-serif">One of the things that I like about
SCTP is that it allows you to use many virtualized streams over a single
connection. &nbsp;The possibilities for using this to replace TCP and use
the virtualized streams to simplfy parsing bytes streams is excellent.
&nbsp;However, in my reading I didn't come across 2 ideas that I thought
could add to the flexibility of SCTP and I was wondering if any discussion
had occured on these topics. &nbsp;</font>
<br>
<br><font size=2 face="sans-serif">1. &nbsp;Dynamically adding and removing
the number of virtualized streams on a connection.</font>
<br><font size=2 face="sans-serif">2. &nbsp;Prioritizing delivery of specific
virtualized streams to enhance quality of service.</font>
<br>
<br><font size=2 face="sans-serif">If they have been discussed for the
standard and can somone point me to further reading on these topics.</font>
<br>
<br><font size=2 face="sans-serif">Thanks</font>
<br><font size=2 face="sans-serif">Ian Barile</font>
--=_alternative 0000357488256D03_=--
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From: "Armando L. Caro Jr." <acaro@mail.eecis.udel.edu>
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Subject: Re: [Tsvwg] SCTP RFC 2960 && draft-ietf-tsvwg-sctpsocket-06.txt
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On Tue, 8 Apr 2003, Ian Barile wrote:

> bytes streams is excellent.  However, in my reading I didn't come across 2
> ideas that I thought could add to the flexibility of SCTP and I was
> wondering if any discussion had occured on these topics.
>
> 1.  Dynamically adding and removing the number of virtualized streams on a
> connection.

I have heard this idea bounced around, but I don't know if anyone has
looked into this area.

> 2.  Prioritizing delivery of specific virtualized streams to enhance
> quality of service.

Jerry Heinz here at UD is currently investigating the use of prioritized
streams. In particular, he is investigating how such a feature could be
useful to instant messager applications. However, I don't think he has any
papers yet...

~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: Tue, 08 Apr 2003 21:23:11 -0500
From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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To: "Armando L. Caro Jr." <acaro@acm.org>
CC: Ian Barile <ian_barile@symantec.com>, sigtran@ietf.org, tsvwg@ietf.org,
        Gerard Heinz <heinz@cis.udel.edu>
Subject: Re: [Tsvwg] SCTP RFC 2960 && draft-ietf-tsvwg-sctpsocket-06.txt
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"Armando L. Caro Jr." wrote:
..
> > 1.  Dynamically adding and removing the number of virtualized streams on a
> > connection.
> 
> I have heard this idea bounced around, but I don't know if anyone has
> looked into this area.

This was debated at length during the development of SCTP and at the end
we reached the consensus that the complexity of adding this feature
outweighed the value it could bring. For example, if one ever would be
in a situation where additional streams is needed, one could always open
more at the assoc initialization since the cost of maintaining extra
streams is close to null. Similarly, there seems to be no good reason
that one would need to remove stream that are no longer in use.

regards,
-Qiaobing
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From mailnull@www1.ietf.org  Wed Apr  9 05:20:40 2003
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Subject: [Tsvwg] again load sharing
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HI. I am investigating the opportunity of load sharing in SCTP. One of
the thing I not undertand is when the load sharing is effective. There
are situations in which load sharing is effective? 
Second, is there anyone who is attempting to response to this question:
when two links are un-shared? In my opinion two link are unshared when
they do not cross same Point Of Congestion. So, detecting at run-time
this event is a very hard challenge....
Giuseppe De Marco
-- 
Giuseppe De Marco
Departement of Electrical and Information Engineering
University of Salerno Italy SA
via ponte don Melillo 1 Fisciano SA
tel: +39 089 964012
mail: gdemarco@unisa.it

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giuseppe,

On Wed, 09 Apr 2003, giuseppe wrote:

> HI. I am investigating the opportunity of load sharing in SCTP. One of
> the thing I not undertand is when the load sharing is effective. There
> are situations in which load sharing is effective? 

Load sharing is effective in any situation where both the available
bandwidth of the network and the offered load exceeds the available
bandwidth of any one interface.  (Or a similar situation with regard
to latency if latency is of concern.)

OTOH the algorithm for load sharing can have interactions with SCTP
protocol mechanisms (retransmission, gaps, SACKs, bundling) and could
reduce efficiency in some circumstances depdending on network
conditions.

IMO algorithms that more closely correspond to SCTP's "primary"
destination approach will have more success than those that deviate
widely from it.  All other factors equal, a sender should not load-share
to a secondary destination until the primary destiation's congestion
window has been sent.  This or similar procedures should have the effect
of grouping as many sequential TSNs as possible to the same destination,
avoiding excess gap detection and reporting due to difference or
variance in OWT between destinations over which traffic is being
loadshared.

> Second, is there anyone who is attempting to response to this question:
> when two links are un-shared? In my opinion two link are unshared when
> they do not cross same Point Of Congestion. So, detecting at run-time
> this event is a very hard challenge....

SCTP's congestion procedures do not make assuptions about the underlying
network and are resilient to network changes independent of network
topology.

Where is the challenge?

--brian

-- 
Brian F. G. Bidulock    ¦ The reasonable man adapts himself to the ¦
bidulock@openss7.org    ¦ world; the unreasonable one persists in  ¦
http://www.openss7.org/ ¦ trying  to adapt the  world  to himself. ¦
                        ¦ Therefore  all  progress  depends on the ¦
                        ¦ unreasonable man. -- George Bernard Shaw ¦
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Subject: [Fwd: Re: [Tsvwg] again load sharing]
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Subject: Re: [Tsvwg] again load sharing
From: giuseppe <gdemarco@unisa.it>
To: bidulock@openss7.org
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I do not agree totally. Maybe I do not understood the matter.But...
Il mer, 2003-04-09 alle 13:07, Brian F. G. Bidulock ha scritto: 
> giuseppe,
> 
> That does not make SCTP "unfair" to TCP.  Each destination will still
> back off exponentially and grow linearly.  TCP connections can only
> be to one destination.  SCTP multihomed associations in competition with
> TCP connections to the one of its destination will behave the same as
> a TCP connection to that destination.  As long as SCTP acts as a single
> TCP connection to each destination, it will behave no differently than
> a TCP connection made to each destination. 

Sure. But let suppose that in the point (router) R1 arrive one TCP
connection and one SCTP connection (with load sharing turn on). In
steady state SCTP single connection (we mean for connection the couple
(source address, destination address) behaves fairly with TCP
connection. And they equally share the capacity. Same for the other
connection (i.e. the other possible couples).
But if we suppose that all the couples, i.e. all the connections of the
SCTP, traverse the same POC (in this case R1), SCTP grabs more capacity
than TCP, and more, the dropping probability augments . If the common
point traversed were a POC, this "probabily" would not happen and the
load sharing would be "effective".
In my discussion I suppose we are able to fairly select the real path
(i.e. the couples...), maybe by using the source addressing mechanism. 


>  In face SCTP is disadvantaged
> by source address/destination address path mappings.  That is, when
> SCTP has two sources (A and B) and two destinations (C and D) it behaves
> as thought it were two TCP connections, one to each destination.  But
> 4 TCP connections can be formed: A-C, B-C, A-D, B-D.  This means that
> TCP can compete unfairly with SCTP in unbalanced networks or in networks
> where source address based routing is performed (e.g. disjoint networks).
> Where source address based routing is not performed (normal case, common
> points of congestion), the situation is quite fair.
> 
> --brian

We detected that there is an other problem in load sharing with SCTP:
the problem of unique CTSN variable. In fact, the SACK could increment
also the congestion windows associated with paths where there has been a
dropping event...
So, I am waiting your helpufl comment.


> On Wed, 09 Apr 2003, giuseppe wrote:
> 
> > Il mer, 2003-04-09 alle 11:49, Brian F. G. Bidulock ha scritto:
> > > giuseppe,
> > > 
> > > On Wed, 09 Apr 2003, giuseppe wrote:
> > > 
> > > > HI. I am investigating the opportunity of load sharing in SCTP. One of
> > > > the thing I not undertand is when the load sharing is effective. There
> > > > are situations in which load sharing is effective? 
> > > 
> > > Load sharing is effective in any situation where both the available
> > > bandwidth of the network and the offered load exceeds the available
> > > bandwidth of any one interface.  (Or a similar situation with regard
> > > to latency if latency is of concern.)
> > > 
> > > OTOH the algorithm for load sharing can have interactions with SCTP
> > > protocol mechanisms (retransmission, gaps, SACKs, bundling) and could
> > > reduce efficiency in some circumstances depdending on network
> > > conditions.
> > > 
> > > IMO algorithms that more closely correspond to SCTP's "primary"
> > > destination approach will have more success than those that deviate
> > > widely from it.  All other factors equal, a sender should not load-share
> > > to a secondary destination until the primary destiation's congestion
> > > window has been sent.  This or similar procedures should have the effect
> > > of grouping as many sequential TSNs as possible to the same destination,
> > > avoiding excess gap detection and reporting due to difference or
> > > variance in OWT between destinations over which traffic is being
> > > loadshared.
> > > 
> > > > Second, is there anyone who is attempting to response to this question:
> > > > when two links are un-shared? In my opinion two link are unshared when
> > > > they do not cross same Point Of Congestion. So, detecting at run-time
> > > > this event is a very hard challenge....
> > > 
> > > SCTP's congestion procedures do not make assuptions about the underlying
> > > network and are resilient to network changes independent of network
> > > topology.
> > 
> > 
> > > Where is the challenge?
> > > 
> > HI BRian, thanks for the quick response.
> > I mean this. If one want sent data source over all the available paths
> > (i.e. by using load sharing), it could happen that some paths (let
> > suppose 2 for example) share a Point Of Congestion. Thus, sending at a
> > send rate greater than of a single TCP connection sharing the same POC,
> > could lead to an unfairness problem. In practice, we do not reach the
> > well known sqrt(p) relation of the TCP throughput...
> > 
> > > --brian

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From mailnull@www1.ietf.org  Wed Apr  9 08:48:06 2003
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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: giuseppe <gdemarco@unisa.it>
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Subject: Re: [Fwd: Re: [Tsvwg] again load sharing]
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giuseppe,

On Wed, 09 Apr 2003, giuseppe wrote:
> 
> I do not agree totally. Maybe I do not understood the matter.But...
> Il mer, 2003-04-09 alle 13:07, Brian F. G. Bidulock ha scritto: 
> > giuseppe,
> > 
> > That does not make SCTP "unfair" to TCP.  Each destination will still
> > back off exponentially and grow linearly.  TCP connections can only
> > be to one destination.  SCTP multihomed associations in competition with
> > TCP connections to the one of its destination will behave the same as
> > a TCP connection to that destination.  As long as SCTP acts as a single
> > TCP connection to each destination, it will behave no differently than
> > a TCP connection made to each destination. 
> 
> Sure. But let suppose that in the point (router) R1 arrive one TCP
> connection and one SCTP connection (with load sharing turn on). In
> steady state SCTP single connection (we mean for connection the couple
> (source address, destination address) behaves fairly with TCP
> connection. And they equally share the capacity. Same for the other
> connection (i.e. the other possible couples).
> But if we suppose that all the couples, i.e. all the connections of the
> SCTP, traverse the same POC (in this case R1), SCTP grabs more capacity
> than TCP, and more, the dropping probability augments . If the common
> point traversed were a POC, this "probabily" would not happen and the
> load sharing would be "effective".
> In my discussion I suppose we are able to fairly select the real path
> (i.e. the couples...), maybe by using the source addressing mechanism. 

Under congestion during load-sharing, the cwnd of each of N destinations
will fall to 1/Nth of TCP's cwnd to the one destination and, everything
is quite fair.  The only place that SCTP might take advantage in load-
sharing is during slow-start, however, an effective load-sharing algorithm
could avoid loadsharing over multiple destinations in slow-start.

> 
> 
> >  In face SCTP is disadvantaged
> > by source address/destination address path mappings.  That is, when
> > SCTP has two sources (A and B) and two destinations (C and D) it behaves
> > as thought it were two TCP connections, one to each destination.  But
> > 4 TCP connections can be formed: A-C, B-C, A-D, B-D.  This means that
> > TCP can compete unfairly with SCTP in unbalanced networks or in networks
> > where source address based routing is performed (e.g. disjoint networks).
> > Where source address based routing is not performed (normal case, common
> > points of congestion), the situation is quite fair.
> > 
> > --brian
> 
> We detected that there is an other problem in load sharing with SCTP:
> the problem of unique CTSN variable. In fact, the SACK could increment
> also the congestion windows associated with paths where there has been a
> dropping event...

I don't see the problem here: could you describe this further?

--brian

> So, I am waiting your helpufl comment.
> 
> 
> > On Wed, 09 Apr 2003, giuseppe wrote:
> > 
> > > Il mer, 2003-04-09 alle 11:49, Brian F. G. Bidulock ha scritto:
> > > > giuseppe,
> > > > 
> > > > On Wed, 09 Apr 2003, giuseppe wrote:
> > > > 
> > > > > HI. I am investigating the opportunity of load sharing in SCTP. One of
> > > > > the thing I not undertand is when the load sharing is effective. There
> > > > > are situations in which load sharing is effective? 
> > > > 
> > > > Load sharing is effective in any situation where both the available
> > > > bandwidth of the network and the offered load exceeds the available
> > > > bandwidth of any one interface.  (Or a similar situation with regard
> > > > to latency if latency is of concern.)
> > > > 
> > > > OTOH the algorithm for load sharing can have interactions with SCTP
> > > > protocol mechanisms (retransmission, gaps, SACKs, bundling) and could
> > > > reduce efficiency in some circumstances depdending on network
> > > > conditions.
> > > > 
> > > > IMO algorithms that more closely correspond to SCTP's "primary"
> > > > destination approach will have more success than those that deviate
> > > > widely from it.  All other factors equal, a sender should not load-share
> > > > to a secondary destination until the primary destiation's congestion
> > > > window has been sent.  This or similar procedures should have the effect
> > > > of grouping as many sequential TSNs as possible to the same destination,
> > > > avoiding excess gap detection and reporting due to difference or
> > > > variance in OWT between destinations over which traffic is being
> > > > loadshared.
> > > > 
> > > > > Second, is there anyone who is attempting to response to this question:
> > > > > when two links are un-shared? In my opinion two link are unshared when
> > > > > they do not cross same Point Of Congestion. So, detecting at run-time
> > > > > this event is a very hard challenge....
> > > > 
> > > > SCTP's congestion procedures do not make assuptions about the underlying
> > > > network and are resilient to network changes independent of network
> > > > topology.
> > > 
> > > 
> > > > Where is the challenge?
> > > > 
> > > HI BRian, thanks for the quick response.
> > > I mean this. If one want sent data source over all the available paths
> > > (i.e. by using load sharing), it could happen that some paths (let
> > > suppose 2 for example) share a Point Of Congestion. Thus, sending at a
> > > send rate greater than of a single TCP connection sharing the same POC,
> > > could lead to an unfairness problem. In practice, we do not reach the
> > > well known sqrt(p) relation of the TCP throughput...
> > > 
> > > > --brian


-- 
Brian F. G. Bidulock    ¦ The reasonable man adapts himself to the ¦
bidulock@openss7.org    ¦ world; the unreasonable one persists in  ¦
http://www.openss7.org/ ¦ trying  to adapt the  world  to himself. ¦
                        ¦ Therefore  all  progress  depends on the ¦
                        ¦ unreasonable man. -- George Bernard Shaw ¦
_______________________________________________
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Subject: Re: [Fwd: Re: [Tsvwg] again load sharing]
From: giuseppe <gdemarco@unisa.it>
To: "Brian F. G. Bidulock" <bidulock@openss7.org>
Cc: TSWG <tsvwg@ietf.org>
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Brian, let consider this raw figure:



		----- 			------
		|TCP|			|TCP|
		-----			------
		  |			   |
--------	  O------------------------O		-------
|SCTP A|__________|			   |____________|SCTP 1|
|      |                                                |      |
|SCTP B|----------O------------------------O------------|SCTP 2|
--------						--------

Suppose we are in load sharing, i.e. we send out data over A-1 and B-2
jointly.
The RFC say that the congestion window must be incremented when :
1) the variable CSTN at sender is lower than the CTSN-AP
2) new data are acknowledged on the associated path
Let suppose u send 1-2-3-4 on the path A-1 and 5-6-7-8 on the path B-2.
A loss of the packet 5 occurs. A SACK on 1-A, with CSTN-AP=4 and a
GAPstart=6 and a GAPstop=8, will increment the CWND both on the A-1 and
B-2 path. This is the problem to have a unique CSTN.
More, in the figure load sharing is fair. TCP compete fairly with A-1
SCTP.
But what happen if also B-2 traverse R1-R2 path? SCTP grabs more
bandwidth with respect to TCP? To me, yes. Moreover, if R1-R2 is
congested (high p), adding an other connection will further increase p.
 giuseppe

Il mer, 2003-04-09 alle 14:39, Brian F. G. Bidulock ha scritto:
> giuseppe,
> 
> On Wed, 09 Apr 2003, giuseppe wrote:
> > 
> > I do not agree totally. Maybe I do not understood the matter.But...
> > Il mer, 2003-04-09 alle 13:07, Brian F. G. Bidulock ha scritto: 
> > > giuseppe,
> > > 
> > > That does not make SCTP "unfair" to TCP.  Each destination will still
> > > back off exponentially and grow linearly.  TCP connections can only
> > > be to one destination.  SCTP multihomed associations in competition with
> > > TCP connections to the one of its destination will behave the same as
> > > a TCP connection to that destination.  As long as SCTP acts as a single
> > > TCP connection to each destination, it will behave no differently than
> > > a TCP connection made to each destination. 
> > 
> > Sure. But let suppose that in the point (router) R1 arrive one TCP
> > connection and one SCTP connection (with load sharing turn on). In
> > steady state SCTP single connection (we mean for connection the couple
> > (source address, destination address) behaves fairly with TCP
> > connection. And they equally share the capacity. Same for the other
> > connection (i.e. the other possible couples).
> > But if we suppose that all the couples, i.e. all the connections of the
> > SCTP, traverse the same POC (in this case R1), SCTP grabs more capacity
> > than TCP, and more, the dropping probability augments . If the common
> > point traversed were a POC, this "probabily" would not happen and the
> > load sharing would be "effective".
> > In my discussion I suppose we are able to fairly select the real path
> > (i.e. the couples...), maybe by using the source addressing mechanism. 
> 
> Under congestion during load-sharing, the cwnd of each of N destinations
> will fall to 1/Nth of TCP's cwnd to the one destination and, everything
> is quite fair.  The only place that SCTP might take advantage in load-
> sharing is during slow-start, however, an effective load-sharing algorithm
> could avoid loadsharing over multiple destinations in slow-start.
> 
> > 
> > 
> > >  In face SCTP is disadvantaged
> > > by source address/destination address path mappings.  That is, when
> > > SCTP has two sources (A and B) and two destinations (C and D) it behaves
> > > as thought it were two TCP connections, one to each destination.  But
> > > 4 TCP connections can be formed: A-C, B-C, A-D, B-D.  This means that
> > > TCP can compete unfairly with SCTP in unbalanced networks or in networks
> > > where source address based routing is performed (e.g. disjoint networks).
> > > Where source address based routing is not performed (normal case, common
> > > points of congestion), the situation is quite fair.
> > > 
> > > --brian
> > 
> > We detected that there is an other problem in load sharing with SCTP:
> > the problem of unique CTSN variable. In fact, the SACK could increment
> > also the congestion windows associated with paths where there has been a
> > dropping event...
> 
> I don't see the problem here: could you describe this further?
> 
> --brian
> 
> > So, I am waiting your helpufl comment.
> > 
> > 
> > > On Wed, 09 Apr 2003, giuseppe wrote:
> > > 
> > > > Il mer, 2003-04-09 alle 11:49, Brian F. G. Bidulock ha scritto:
> > > > > giuseppe,
> > > > > 
> > > > > On Wed, 09 Apr 2003, giuseppe wrote:
> > > > > 
> > > > > > HI. I am investigating the opportunity of load sharing in SCTP. One of
> > > > > > the thing I not undertand is when the load sharing is effective. There
> > > > > > are situations in which load sharing is effective? 
> > > > > 
> > > > > Load sharing is effective in any situation where both the available
> > > > > bandwidth of the network and the offered load exceeds the available
> > > > > bandwidth of any one interface.  (Or a similar situation with regard
> > > > > to latency if latency is of concern.)
> > > > > 
> > > > > OTOH the algorithm for load sharing can have interactions with SCTP
> > > > > protocol mechanisms (retransmission, gaps, SACKs, bundling) and could
> > > > > reduce efficiency in some circumstances depdending on network
> > > > > conditions.
> > > > > 
> > > > > IMO algorithms that more closely correspond to SCTP's "primary"
> > > > > destination approach will have more success than those that deviate
> > > > > widely from it.  All other factors equal, a sender should not load-share
> > > > > to a secondary destination until the primary destiation's congestion
> > > > > window has been sent.  This or similar procedures should have the effect
> > > > > of grouping as many sequential TSNs as possible to the same destination,
> > > > > avoiding excess gap detection and reporting due to difference or
> > > > > variance in OWT between destinations over which traffic is being
> > > > > loadshared.
> > > > > 
> > > > > > Second, is there anyone who is attempting to response to this question:
> > > > > > when two links are un-shared? In my opinion two link are unshared when
> > > > > > they do not cross same Point Of Congestion. So, detecting at run-time
> > > > > > this event is a very hard challenge....
> > > > > 
> > > > > SCTP's congestion procedures do not make assuptions about the underlying
> > > > > network and are resilient to network changes independent of network
> > > > > topology.
> > > > 
> > > > 
> > > > > Where is the challenge?
> > > > > 
> > > > HI BRian, thanks for the quick response.
> > > > I mean this. If one want sent data source over all the available paths
> > > > (i.e. by using load sharing), it could happen that some paths (let
> > > > suppose 2 for example) share a Point Of Congestion. Thus, sending at a
> > > > send rate greater than of a single TCP connection sharing the same POC,
> > > > could lead to an unfairness problem. In practice, we do not reach the
> > > > well known sqrt(p) relation of the TCP throughput...
> > > > 
> > > > > --brian
> 

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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: giuseppe <gdemarco@unisa.it>
Cc: TSWG <tsvwg@ietf.org>
Subject: Re: [Fwd: Re: [Tsvwg] again load sharing]
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giuseppe,

On Wed, 09 Apr 2003, giuseppe wrote:

> Brian, let consider this raw figure:
> 
> 
> 
> 		----- 			------
> 		|TCP|			|TCP|
> 		-----			------
> 		  |			   |
> --------	  O------------------------O		-------
> |SCTP A|__________|			   |____________|SCTP 1|
> |      |                                                |      |
> |SCTP B|----------O------------------------O------------|SCTP 2|
> --------						--------
> 
> Suppose we are in load sharing, i.e. we send out data over A-1 and B-2
> jointly.
> The RFC say that the congestion window must be incremented when :
> 1) the variable CSTN at sender is lower than the CTSN-AP
> 2) new data are acknowledged on the associated path
> Let suppose u send 1-2-3-4 on the path A-1 and 5-6-7-8 on the path B-2.
> A loss of the packet 5 occurs. A SACK on 1-A, with CSTN-AP=4 and a
> GAPstart=6 and a GAPstop=8, will increment the CWND both on the A-1 and
> B-2 path. This is the problem to have a unique CSTN.

Only the congestion window for the destination to which data was sent
is incremented and then only if the data is cumulatively acked and
the data was not retransmitted.  I still don't see a problem.

> More, in the figure load sharing is fair. TCP compete fairly with A-1
> SCTP.

> But what happen if also B-2 traverse R1-R2 path? SCTP grabs more
> bandwidth with respect to TCP? To me, yes.

No. SCTP's cwnd for each desitination would fall to 1/2 of TCP's cwnd.

> Moreover, if R1-R2 is
> congested (high p), adding an other connection will further increase p.


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

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

I disagree. While many fixed networks are indeed now connected over 
links much faster than slow dialups, there are some important wireless 
situations where conditions are very much like those of the Internet 15 
years ago.

Imagine a self-contained gateway router with 802.11 on one side and a 3G 
(or CDMA, or whatever) cell phone on the other. I envision these things 
being used to provide temporary Internet access to a group in a hotel 
conference room where the hotel doesn't already provide Internet access. 
Here you have a LAN that can potentially generate a lot of traffic being 
linked to the public Internet over a relatively slow (by LAN standards) 
link. It is still important for TCP to be conservative in such 
situations, don't you think?

Phil



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Reiner Ludwig wrote:

> 1. With timestamps, we can also take valid RTTs from retransmitted 
> segments.

One major difference that *does* exist between the Internet of 15 years 
ago and the Internet of today is that TCP header timestamps are now very 
widespread. Nearly all TCPs support them: Linux enables them by default, 
while W2K (and 98, I think) implement them but you have to turn them on.

TCP header timestamps have made my algorithm essentially obsolete, as 
they eliminates the "retransmission ambiguity" problem I was trying to 
solve. You just echo back each timestamp as you get it, and you don't 
care whether the timestamp is on the first transmission of a particular 
segment or its retransmission. I knew at the time I wrote my paper with 
Craig that this was possible, but I was interested in algorithms that 
could work with the existing TCP header.

Header overhead is a non-issue here, IHMO. The savings in unnecessary 
retransmits more than pays back the investment in a larger TCP header.

Phil

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Dear all,

my Siemens e-mail address will no longer work.
Please use

tuexen@fh-muenster.de

in case you want to contact me.

Best regards
Michael

Michael Tuexen   Tel.:   +49 89 722 47210
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Subject: [Tsvwg] RFC 3517 on A Conservative Selective Acknowledgment (SACK)-based Loss Recovery Algorithm for TCP
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3517

        Title:      A Conservative Selective Acknowledgment
                    (SACK)-based Loss Recovery Algorithm for TCP
        Author(s):  E. Blanton, M. Allman, K. Fall, L. Wang
        Status:     Standards Track
        Date:       April 2003
        Mailbox:    eblanton@cs.purdue.edu, mallman@bbn.com,
                    kfall@intel-research.net, lwang0@uky.edu
        Pages:      13
        Characters: 27855
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-allman-tcp-sack-13.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3517.txt


This document presents a conservative loss recovery algorithm
for TCP that is based on the use of the selective acknowledgment
(SACK) TCP option.  The algorithm presented in this document conforms
to the spirit of the current congestion control specification
(RFC 2581), but allows TCP senders to recover more effectively
when multiple segments are lost from a single flight of data.

This document is a product of the Transport Area Working Group of the
IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
of this memo is unlimited.

This announcement is sent to the IETF list and the RFC-DIST list.
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Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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Subject: Re: [Fwd: Re: [Tsvwg] again load sharing]
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi, all,

please see my comments inline:

On Wednesday 09 April 2003 15:24, giuseppe wrote:
>
> 		----- 			------
> 		|TCP|			|TCP|
> 		-----			------
>--------        O------------------------O            --------
>|SCTP A|________|                        |____________|SCTP 1|
>|
>|SCTP B|----------O----------------------O------------|SCTP 2|
>--------                                              --------

> But what happen if also B-2 traverse R1-R2 path? SCTP grabs more
> bandwidth with respect to TCP? To me, yes. Moreover, if R1-R2 is
> congested (high p), adding an other connection will further increase p.

In that configuration, SCTP behaves essentially similar to two 
TCP connections. So, one SCTP association probably grabs more 
bandwidth than one TCP connection. 

However, it still behaves in a fair manner toward the TCP connection.
Also, the SCTP association is bound by the receiver window (RWND), meaning
that there is a limit that is enforced on the SCTP association as a 
whole. 

Now, if you have 2 TCP connections, these are both limited by
their respective receiver windows. So these 2 may, *depending on the
parameters of the network setup*, take more bandwidth than one SCTP
association with dual-homing and load-sharing.

My 2 cents,

Andreas

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

My PGP Key-Fingerprint: 228C 7C2C 3381 2DD4  9998 2812 5C0B 0B04  D382 4AC0
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Hi Andreas/all,

> In that configuration, SCTP behaves essentially similar to two
> TCP connections. So, one SCTP association probably grabs more
> bandwidth than one TCP connection.
>
> However, it still behaves in a fair manner toward the TCP connection.

I don't think I agree. If a single SCTP association behaves as
aggressively as two concurrent TCP connections, I do not think it is being
fair to TCP.  For one, when comparing a single TCP connection with an SCTP
association performing what we call "Concurrent Multipath Transfer (CMT)"
(or load sharing) with a shared bottleneck, a single loss event at the
bottleneck does not result in the same reduced cwnd for the TCP connection
and the SCTP-CMT association.

> Also, the SCTP association is bound by the receiver window (RWND), meaning
> that there is a limit that is enforced on the SCTP association as a
> whole.
>
> Now, if you have 2 TCP connections, these are both limited by
> their respective receiver windows. So these 2 may, *depending on the
> parameters of the network setup*, take more bandwidth than one SCTP
> association with dual-homing and load-sharing.

Agreed. But I think we should be comparing 1 TCP connection with 1 SCTP
association. Or am I missing something here?

Guiseppe: I am currently involved in End-to-End Concurrent Multipath
Transfer at the Transport Layer. With regard to your comment about "unique
CTSN variable", there are a few things that come to mind. But I'm not sure
what you are referring to. If you do have some specific questions
with regard to what you are observing, maybe you could get back to me
offline?

regards,
jana

> My 2 cents,
>
> Andreas
>
> - --
> Dipl.-Ing. Andreas Jungmaier
> Computer Networking Technology Group
> University of Duisburg-Essen
> http://www.exp-math.uni-essen.de/~ajung
> My PGP Key-ID: D3824AC0
>
> My PGP Key-Fingerprint: 228C 7C2C 3381 2DD4  9998 2812 5C0B 0B04  D382 4AC0
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.2.1 (GNU/Linux)
>
> iD8DBQE+lnqiXAsLBNOCSsARAkHvAKCX1+jVWVTrIqvhGjMkFo9vEMXtfQCcCuP7
> lJxY8CujFU6B2LJbNDiY3T8=
> =m+Tj
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
>

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Janardhan R. Iyengar		       http://www.cis.udel.edu/~iyengar
Protocol Engineering Lab			 University of Delaware
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To: tsvwg@ietf.org, minutes@ietf.org
Cc: jon.peterson@neustar.biz
Reply-To: mankin@psg.com
Date: Fri, 11 Apr 2003 17:52:07 +0000
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These are due today it seems, so send corrections to the list, where
our archives can serve as the record.  Thanks to Spencer for the 
note-taking.

Minutes taken by Spencer Dawkins

==================================
Transport Area Working Group TSVWG
==================================

Allison Mankin and Jon Peterson

Scott has been replaced by Jon as AD but will still be active

Kireeti - Status of RSVP change document
	RSVP has become very popular, people outside IETF want to change it
	this causes difficulties for interoperation - how do we prevent breaking RSVP?
		recent code points RFC from completely outside IETF with a lot of pushback
	give RSVP IANA considerations that it's never had before
		previous considerations draft had expired
	give three code points (normal use, experimental use, first-come-first-served)
		really need a "standards action" code point, too
	also need PS RFC to initiate behavior changes
	is this important?

	Andy Malis: this is absolutely needed - 
		IANA is working with no guidelines whatesoever
		even first-come-first-served space is broken
		Bob Braden is now the gatekeeper and things are confusing
	
	Bob Braden has suggested name-based sand boxes - not much reaction - do people care?
	Allison: IESG suggested names not reflect owners for maintainability

	Philip Matthews: IS-IS has the same issues 
		and is dealing with them in the same way
		Split between class numbers and C-types?
	
	will be discussed on TSVWG mailing list

Jukka Manner - Localized RSVP (LRSVP)
	end-to-end QoS is rarely available
	backbones fast and reliable
	access networks are bottleneck congestion points
	using a small enhancement to RSVP that can coexist with RSVP
	Local Indication and Expedited Refresh bits allow coexistence
	Open issues - addressing, asymmetric routing, interworking with end-to-end RSVP, security

	Steve Casner: each end signals its own access network?
	A: end-to-end is preferable, this is fall-back

	Steve Silverman: is "local" on a LAN?
	A: "local" access network

	Tom Hiller: 3GPP does things almost identical to this
		not documented in RFCs because never leaves walled garden

John Bennett - IP measurement OAM problem problem statement

	measure ping, traceroute, etc.
	can't do this because ot ICMP blocking, flow-specific routes, hop-by-hop vs. end-to-end
	want to measure increasingly specific things (take HTTP redirect into account, etc)
	can't figure out which hop is the problem
	need a new protocol that can be processed in fast path - no control
	"polite" protocol
	passes through firewalls, NATs, etc.
	"IP Measurement Protocol" - IPMP Echo, IPMP Information Request
		proposed by Andrew McGregor a couple of years ago
	bennett-00 changes adds redirection, port numbers, TLVs
	mcgregor-00 and bennet-01 also make changes to previous proposal

	Peter Neuman - would like to measure downstream with standardized protocol
	Q: also have a draft we'd like to be considered
	A: looking at general level of interest first	

Sally Floyd - draft-floyd-tcp-highspeed-02.txt
	Quickstart revision is also out for comments
	adding a new response function for TCPs on links with very low error rates
	revision adds discussion of TCP with multiple connections
		Scalable TCP from Tom Kelly
	Adds a section on choosing parameters
	Adds discussion on P-MTU, window scaling, etc.
	Shows simulation results from Evandro de Souza
	in a drop-tail queue? would work better with small P
	Need feedback from community before Experimental RFC
	Limited Slowstart proposal is also related
	Other changes? Higher dupack threshold, etc.
	TCP Quickstart isn't ready to be Experimental RFC - needs more research
	
	Allison: what is the complexity?
	A: maintain a table instead of always cutting in half
	Eric Nordmark: current reference implementation uses scaled math
		cutting in half is easy, cutting by 22.9 percent is harder
	A: could approximate by shifting
	Matt Mathis: uses integer arithmetic
	Reiner: why don't smaller congestion windows get to use this?
	A: need to pick targets for response functions for divergence from standard TCP
		motivation for picking is that TCP is successful now, so small changes
	Matt Mathis: Fairness on the Internet is because of standard response function
		can't let everybody dork with their response function - no fairness
	Q: Not good at translating from drop rates to technologies
	A: starting at 10**-6 packet loss rates, High Speed TCP seems to make a big difference
	Mike from Digital Fountain: have you tried parallel sessions of HSTCP?
	A: yes, pointed to from web page, so far it looks fine
		congestion network would be better off if you're using HSTCP
	Joe Touch: use RSVP for SYN, because we're in slow path anyway
	A: maybe, but not yet
	Joe Touch: point solution for 10-gig Ethernet? more dynamic than you're thinking about
	A: can you talk about this on the list?

        hum to adopt as WG item 

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

	building from Atlanta discussion
	solved real loss during recovery
	what do we do now?

	Randall Stewart: perfectly safe, move it forward
	hum to adopt as WG item

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

	can we recover our congestion window?
	need to prevent massive burst at window reversion time
	need to prevent spurious fast retransmits
	presenting four schemes of increasing complexity
	most conservative scheme is most complexity
	need more data, would like to hear alternative schemes
	spurious RTOs is explicitly out of scope
	Matt Mathis: there are lots of heuristics hidden in real TCPs that will cloud your results
	Mark Allman: should this sort of algorithm be Experimental? They're conservative
	Allison: some RFCs are really experiments, some are really for immediate use
	Sally: Experimental after one real implementation?
	Allison: We use "experimental" to mean several different things
	Sally: Texas A&M proposal could be added to this work
	
Mark Allman - draft-allman-tcp-early-rexmt-00.txt

	No oppositions based on the list traffic, right?
	splitting this work out from Limited Transmit
	less resistent to reordering, but ...
	Spencer: worse cases are bad in macro sense but effect on user may justify this

Sally Floyd - Proposal for newreno standardization
	Newreno is widely deployed and is still experimental - move it forward?
	please identify any major fixes this would require (but not other ideas/changes)

Reiner Ludwig - Redundant data in TCP?
	TCP-friendly with quotas on good and bad bytes?
	please feed back to the mailing list

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

	Phil Cochran - SCTP bakeoff at UDelaware June 1-8 (unofficial)
	PR-SCTP has been around for a while - the first service designed
        by the spec  provides "timed reliability"
	"abandon" stale chunks of data
	still make congestion window (downward) adjustments
	only enabled if both sides understand it!
	three independent implementations, all interop tested
	seven additional implementations coming

	Mark Allman: making it easier to throw useless data into the network?
		should be considered the way we're considering Reiner's proposal
	Randall: will construct a scenario on the list

	David Black: this is the opposite of Reiner's proposal
	Randall: better than I could have said it

	Allison: noticed spec made possible to change the lifetime after it was set,
         suggests interesting race conditions, API complexity in name of flexibility/power...
	Phil: lots of good issues...
		agree that if you can change lifetime values, this creates race conditions
		larger question is about creating lots of new services
		two parts of proposals - on the wire part is what we need to standardize
			policy is internal to a node
	Randall: realistic, can't change this in internal implementations today
	
	Mark: more motivation on the list?
	Randall: unreliable stream didn't simplify anything (per previous connectathon)
	Mark: couldn't you simplify per-stream, not just per-packet?
	Phil: good points for application writers, but these are service definition API issues
	Randall - I could do what you're saying in the current implementation
	David Black: setting timeouts is valuable in some scenario

        Chairs: review to be continued on list.


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From: "Phillip T. Conrad" <conrad@acm.org>
Subject: Re: [Tsvwg] Minutes of TSVWG at IETF 56
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>At 05:52 PM 4/11/2003 +0000, Allison Mankin wrote:
>These are due today it seems, so send corrections to the list, where
>our archives can serve as the record.  Thanks to Spencer for the
>note-taking.
>...

One correction:


...
>Randall Stewart - draft-stewart-tsvwg-prsctp-03.txt
>
>         Phil Cochran - SCTP bakeoff at UDelaware June 1-8 (unofficial)
            ^^^^^^^^^^^^^
>         PR-SCTP has been around for a while - the first service designed
...


Phill Conrad made the SCTP bakeoff announcement, and participated in the 
PR-SCTP discussion (rather than Phil Cochran).   (I have this on good 
authority.)

The minutes are accurate that the bakeoff announcement was "unofficial" at 
the time it was made; however, if you like, we can add a note that the 
bakeoff is now officially "on" for that date, and details are at: 
http://pel.cis.udel.edu/interop

Cheers,
Phill Conrad

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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Janar, all,

please see my comments below - inline:
> > In that configuration, SCTP behaves essentially similar to two
> > TCP connections. So, one SCTP association probably grabs more
> > bandwidth than one TCP connection.
> >
> > However, it still behaves in a fair manner toward the TCP connection.
>
> I don't think I agree. If a single SCTP association behaves as
> aggressively as two concurrent TCP connections, I do not think it is being
> fair to TCP.

Agreed, I have not phrased this precisely enough. I was actually referring
to the TCP friendly congestion control for each SCTP path, i.e. the fact that
if there are n TCP connections and 1 SCTP association with m paths (engaged
in what you nicely call "concurrent multipath transfer") then an additional
TCP connection would obtain another 1/(m+n+1)-th part of the bandwidth.
So clearly, ONE SCTP assoc grabs more than a fair share, but there is still
some "fairness". So: actually we are both d'accord !

Best regards,
Andreas

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

My PGP Key-Fingerprint: 228C 7C2C 3381 2DD4  9998 2812 5C0B 0B04  D382 4AC0
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Hi all,

I had a short presentation of the Localized RSVP extension to RSVP at the
last IETF meeting. The ADs proposed that I should see from the working
group people, whether there is interest in doing something like we have
proposed, to have a mechanism for setting up local resource reservations.
The draft has motivations about when this would be a useful thing. Thus,
if people could read the draft (it is not that long :) and answer to two
questions:

1) Is the concept of local resource reservations a good idea in the first
place?

2) If the above answer was "yes", is our approach using RSVP a good one,
and could be taken further in the TSVWG?

The announcement and a link to the draft is below.

Best regards,
Jukka Manner

---------------------------------------------------------------------
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-01.txt
        Pages           : 16
        Date            : 2003-1-14
        
Guaranteed QoS for multimedia applications is based on reserved
resources in each node on the end-to-end path. Even if the
correspondent node does not support QoS, it would be useful to be
able to reserve at least local resources at the access network,
especially wireless link resources. Additionally, for mobile nodes
the continuity of QoS is disturbed due to end-to-end signaling or
slow re-reservations of resources after each handover. This draft
proposes a simple enhancement to RSVP, so that initial resource
reservations and re-reservations due to host mobility can be done
locally in an access network.

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

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Subject: [Tsvwg] PR-SCTP
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All:

I would like once again to bring up PR-SCTP for discussion...

So far I have received several comments on the document and
am preparing changes based on them. One unresolved
issue is Mark Allmans comments on "junk" data.

Mark and I agree that we have a fundamental difference of opinion
on this...  :-0

I don't think a transport protocol should police the data that
an application sends over it... even if there are hints that
are given saying "you can abandon this data if after some
amount of time you can't get it through". To me this seems to
improve chances of getting out of congestion faster since it
may not be the case that the application as anything else
to say for a while.. and even if it does.. overall you are
reducing the load on the network by the data that you removed
(since classic SCTP would retransmit it until it was sent
  on through).

Mark feels that the parameters given could be used in some
way to enforce on the app that it not use this feature to
much... (Mark jump in and correct me if I mis-state things)...
no matter if it is an abuser (for example sending multiple
copies of data to speed things up) or a valid application.. junk
data is junk data... If you do it to much you should in
some way be penalized... aka make the data reliable or
reset the connection....

Do others feel the same as Mark? Should a transport police
the types of data being sent? ... I guess this falls in
with some of Reiner's work as well (on FEC related use
of the cwnd).. My opinion is obvious :> I am against
this...

Along with the above I would like to get other feedback... I
would like to get this document moved on to last-call possibly
after incorporating all of the comments.. to date...

What does everyone think???

Thanks

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

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From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] PR-SCTP
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OK, let me show how poorly I understand the issue.

If an application gives clues to transport that something isn't
as interesting as something else, how is this different from
per-packet priority marking?

My first take is that I'm not sure how transports know something
is junk, in the general case. I'm actually interested in
understanding more clearly what the axes of "interesting" look
like. I think I'd like to understand this better, before
providing local-optimal feedback.

My understanding of Randall's concept is "deadline discard" - if
you don't deliver this in 800 ms, don't bother? 

My understanding of Reiner's concept is "reduce retransmission
latency by retransmitting" - if you have unACKed data and don't
have anything else new to send, send the old data again?

What are the other concepts that people are thinking about?

Spencer
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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Randall Stewart <randall@stewart.chicago.il.us>
Cc: TSV WG list <tsvwg@ietf.org>, Allison Mankin <mankin@psg.com>,
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Randall,

I am against transports policing junk also.

What other comments are you talking about?  (I only
saw the ladders that Mark started...)

I don't see any problems with the latest draft.

--brian

On Wed, 23 Apr 2003, Randall Stewart wrote:

> All:
> 
> I would like once again to bring up PR-SCTP for discussion...
> 
> So far I have received several comments on the document and
> am preparing changes based on them. One unresolved
> issue is Mark Allmans comments on "junk" data.
> 
> Mark and I agree that we have a fundamental difference of opinion
> on this...  :-0
> 
> I don't think a transport protocol should police the data that
> an application sends over it... even if there are hints that
> are given saying "you can abandon this data if after some
> amount of time you can't get it through". To me this seems to
> improve chances of getting out of congestion faster since it
> may not be the case that the application as anything else
> to say for a while.. and even if it does.. overall you are
> reducing the load on the network by the data that you removed
> (since classic SCTP would retransmit it until it was sent
>   on through).
> 
> Mark feels that the parameters given could be used in some
> way to enforce on the app that it not use this feature to
> much... (Mark jump in and correct me if I mis-state things)...
> no matter if it is an abuser (for example sending multiple
> copies of data to speed things up) or a valid application.. junk
> data is junk data... If you do it to much you should in
> some way be penalized... aka make the data reliable or
> reset the connection....
> 
> Do others feel the same as Mark? Should a transport police
> the types of data being sent? ... I guess this falls in
> with some of Reiner's work as well (on FEC related use
> of the cwnd).. My opinion is obvious :> I am against
> this...
> 
> Along with the above I would like to get other feedback... I
> would like to get this document moved on to last-call possibly
> after incorporating all of the comments.. to date...
> 
> What does everyone think???
> 
> Thanks
> 
> R
> -- 
> Randall R. Stewart
> randall@stewart.chicago.il.us 815-342-5222 (cell phone)
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

-- 
Brian F. G. Bidulock    ¦ The reasonable man adapts himself to the ¦
bidulock@openss7.org    ¦ world; the unreasonable one persists in  ¦
http://www.openss7.org/ ¦ trying  to adapt the  world  to himself. ¦
                        ¦ Therefore  all  progress  depends on the ¦
                        ¦ unreasonable man. -- George Bernard Shaw ¦
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Subject: Re: [Tsvwg] PR-SCTP
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On 4/23/03, Randall Stewart wrote:


>
>Do others feel the same as Mark? Should a transport police
>the types of data being sent? ... I guess this falls in
>with some of Reiner's work as well (on FEC related use
>of the cwnd).. My opinion is obvious :> I am against
>this...
>

Even if you agree that the transport layer should prevent
or at least discourage abuse of network bandwidth by the
application layer, it would not be PR-SCTP specific. Or even
SCTP specific for that matter.

PR-SCTP can never guarantee that payload will *not* be
delivered. Even if you cancel it, the receiving side
might still get the data before the cancellation.

If there is a real concern for transports encouraging junk
data, I would suggest a better target being congestion
control itself. It encourages a "use it or lose it"
mentality concerning bandwidth -- a problem not found
with explicit rate control. I'm not advocating any specific
solution -- but at least its a real problem.

Caitlin Bestler
http://asomi.com/CaitlinBestler/
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My last two notes on this subject from a couple week back stand as
my opinion.  I will not repeat them.  But, I would be interested in
others opinions (especially those we have no heard).  

But, a clarification...

> Mark and I agree that we have a fundamental difference of opinion
> on this...  :-0

Indeed.

> Mark feels that the parameters given could be used in some
> way to enforce on the app that it not use this feature to
> much... (Mark jump in and correct me if I mis-state things)...
> no matter if it is an abuser (for example sending multiple
> copies of data to speed things up) or a valid application.. junk
> data is junk data... If you do it to much you should in
> some way be penalized... aka make the data reliable or
> reset the connection....

Note that "junk data" is maybe not the best way to describe it (even
though I, no doubt, started that).  We can determine data that was
sent, dropped (with good probability) and not retransmitted using
PR-SCTP.  That data **accomplished no useful work and used scarce
network resources**.  (I see this as a pretty non-controversial
opinion.  Do we have agreement on it (i.e., only what I have written
in this paragraph)?)

And, then there is the question of what to do about data being sent
that does not accomplish useful work.

Also, I want to stress that I offered that **excessive** amounts of
data transmitted without accomplishing useful work should be curbed.
I never said it was a reason not to do PR-SCTP.

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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I believe there are actually four (two off-line and one
at the TSV meeting)...

1) The current document uses the term "drop" .. this to some
    in TSV is confusing.. aka a router will "drop" a packet due
    to congestion. Instead the term "abandon" aka the user decides
    to "abandon" a packet.. is being substituted for "drop"

2) There was a typo in the document the word "parameter" is miss-spelled

3) Allison wanted a clearification in the service model presented in the
    document that an application cannot change the lifetime of a
    message after it was sent. This is implied in some places but not
    spelled out... It was also due to some poor wording choices in
    the document as to who is evalutating ... etc:

    So the following clearifications are put in (note the line numbers
    are against my xml file..not the draft)...

887c896
      The sender MAY evaluate the lifetime of messages at anytime.
---
      The sending SCTP MAY evaluate the lifetime of messages at anytime.
891a901,904
TR6)
     The sending application MUST NOT change the lifetime parameter
     once the message is passed to the sending SCTP.

900c913
<     implementor flexibility to evaluate lifetime at any other convenient
---
 >     SCTP implementor flexibility to evaluate lifetime at any other 
convenien




4) Comments at the tsv were made that it was not clear that
    normal cwnd adjustments should be made when timers go off
    even if you decide to abandon a chunk..  So a new F5) will
    be added that specifies:

        If a decision to "abandon" a chunk is made based on a
        retransmission decision (e.g T3-Timeout or Fast Retransmit)
        the appropriate congestion adjustment as specified in RFC2960
        MUST be made before any FORWARD TSN chunk is sent.


These are all I have.. unless others pop up :->

R



Brian F. G. Bidulock wrote:
> Randall,
> 
> I am against transports policing junk also.
> 
> What other comments are you talking about?  (I only
> saw the ladders that Mark started...)
> 
> I don't see any problems with the latest draft.
> 
> --brian
> 
> On Wed, 23 Apr 2003, Randall Stewart wrote:
> 
> 
>>All:
>>
>>I would like once again to bring up PR-SCTP for discussion...
>>
>>So far I have received several comments on the document and
>>am preparing changes based on them. One unresolved
>>issue is Mark Allmans comments on "junk" data.
>>
>>Mark and I agree that we have a fundamental difference of opinion
>>on this...  :-0
>>
>>I don't think a transport protocol should police the data that
>>an application sends over it... even if there are hints that
>>are given saying "you can abandon this data if after some
>>amount of time you can't get it through". To me this seems to
>>improve chances of getting out of congestion faster since it
>>may not be the case that the application as anything else
>>to say for a while.. and even if it does.. overall you are
>>reducing the load on the network by the data that you removed
>>(since classic SCTP would retransmit it until it was sent
>>  on through).
>>
>>Mark feels that the parameters given could be used in some
>>way to enforce on the app that it not use this feature to
>>much... (Mark jump in and correct me if I mis-state things)...
>>no matter if it is an abuser (for example sending multiple
>>copies of data to speed things up) or a valid application.. junk
>>data is junk data... If you do it to much you should in
>>some way be penalized... aka make the data reliable or
>>reset the connection....
>>
>>Do others feel the same as Mark? Should a transport police
>>the types of data being sent? ... I guess this falls in
>>with some of Reiner's work as well (on FEC related use
>>of the cwnd).. My opinion is obvious :> I am against
>>this...
>>
>>Along with the above I would like to get other feedback... I
>>would like to get this document moved on to last-call possibly
>>after incorporating all of the comments.. to date...
>>
>>What does everyone think???
>>
>>Thanks
>>
>>R
>>-- 
>>Randall R. Stewart
>>randall@stewart.chicago.il.us 815-342-5222 (cell phone)
>>
>>_______________________________________________
>>tsvwg mailing list
>>tsvwg@ietf.org
>>https://www1.ietf.org/mailman/listinfo/tsvwg
> 
> 


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

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Mark Allman wrote:
> My last two notes on this subject from a couple week back stand as
> my opinion.  I will not repeat them.  But, I would be interested in
> others opinions (especially those we have no heard).  
> 
> But, a clarification...
> 
> 
>>Mark and I agree that we have a fundamental difference of opinion
>>on this...  :-0
> 
> 
> Indeed.
> 
we surely agree here :>

> 
>>Mark feels that the parameters given could be used in some
>>way to enforce on the app that it not use this feature to
>>much... (Mark jump in and correct me if I mis-state things)...
>>no matter if it is an abuser (for example sending multiple
>>copies of data to speed things up) or a valid application.. junk
>>data is junk data... If you do it to much you should in
>>some way be penalized... aka make the data reliable or
>>reset the connection....
> 
> 
> Note that "junk data" is maybe not the best way to describe it (even
> though I, no doubt, started that). 

It might be nice to have a better term.. but I did not want to
muddy the water by changing terms :-0


  We can determine data that was
> sent, dropped (with good probability) and not retransmitted using
> PR-SCTP.  That data **accomplished no useful work and used scarce
> network resources**.  (I see this as a pretty non-controversial
> opinion.  Do we have agreement on it (i.e., only what I have written
> in this paragraph)?)

Well, Caitlin actually had a good point.. just because you abandon a
chunk does NOT mean it did not get delivered... so it COULD have done
some useful work... you the sender just do not know if it got there
at all (in a timely manner where it was used) or not.... it may
have... with a ACK loss making you think it did not.. or it may
NOT have and then the above would have been true... But I guess
your "good probability" qualification may be saying this :>

> 
> And, then there is the question of what to do about data being sent
> that does not accomplish useful work.
> 
> Also, I want to stress that I offered that **excessive** amounts of
> data transmitted without accomplishing useful work should be curbed.
> I never said it was a reason not to do PR-SCTP.

Thanks for the clearification.. I too would like to hear what
others think..

R


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


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

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> > We can determine data that was sent, dropped (with good
> > probability) and not retransmitted using PR-SCTP.  That data
> > **accomplished no useful work and used scarce network
> > resources**.  (I see this as a pretty non-controversial opinion.
> > Do we have agreement on it (i.e., only what I have written in
> > this paragraph)?)
> 
> Well, Caitlin actually had a good point.. just because you abandon a
> chunk does NOT mean it did not get delivered... so it COULD have done
> some useful work... you the sender just do not know if it got there
> at all (in a timely manner where it was used) or not.... it may
> have... with a ACK loss making you think it did not.. or it may
> NOT have and then the above would have been true... But I guess
> your "good probability" qualification may be saying this :>

Yes, that was my intent.

allman


--
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Laurent:

Thanks for your response.. I am copying the list (hope thats
ok) .. maybe you forgot to do the reply-all think :>

R

Laurent Glaude wrote:
> Randall,
> 
> I support your view that the transport layer should not police the data 
> that an application sends over it. I do not believe that PR-SCTP 
> provides an incentive to send "junk data".
> 
> -Laurent
> 
> Randall Stewart wrote:
> 
>> All:
>>
>> I would like once again to bring up PR-SCTP for discussion...
>>
>> So far I have received several comments on the document and
>> am preparing changes based on them. One unresolved
>> issue is Mark Allmans comments on "junk" data.
>>
>> Mark and I agree that we have a fundamental difference of opinion
>> on this...  :-0
>>
>> I don't think a transport protocol should police the data that
>> an application sends over it... even if there are hints that
>> are given saying "you can abandon this data if after some
>> amount of time you can't get it through". To me this seems to
>> improve chances of getting out of congestion faster since it
>> may not be the case that the application as anything else
>> to say for a while.. and even if it does.. overall you are
>> reducing the load on the network by the data that you removed
>> (since classic SCTP would retransmit it until it was sent
>>  on through).
>>
>> Mark feels that the parameters given could be used in some
>> way to enforce on the app that it not use this feature to
>> much... (Mark jump in and correct me if I mis-state things)...
>> no matter if it is an abuser (for example sending multiple
>> copies of data to speed things up) or a valid application.. junk
>> data is junk data... If you do it to much you should in
>> some way be penalized... aka make the data reliable or
>> reset the connection....
>>
>> Do others feel the same as Mark? Should a transport police
>> the types of data being sent? ... I guess this falls in
>> with some of Reiner's work as well (on FEC related use
>> of the cwnd).. My opinion is obvious :> I am against
>> this...
>>
>> Along with the above I would like to get other feedback... I
>> would like to get this document moved on to last-call possibly
>> after incorporating all of the comments.. to date...
>>
>> What does everyone think???
>>
>> Thanks
>>
>> R
> 
> 
> 
> 


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

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From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: Re: [Tsvwg] PR-SCTP 
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Mark ... A comment or two ... Michael Ramalho

At 12:50 PM 4/23/2003 -0400, Mark Allman wrote:

>My last two notes on this subject from a couple week back stand as
>my opinion.  I will not repeat them.  But, I would be interested in
>others opinions (especially those we have no heard).
>
>But, a clarification...
>
> > Mark and I agree that we have a fundamental difference of opinion
> > on this...  :-0
>
>Indeed.
>
> > Mark feels that the parameters given could be used in some
> > way to enforce on the app that it not use this feature to
> > much... (Mark jump in and correct me if I mis-state things)...
> > no matter if it is an abuser (for example sending multiple
> > copies of data to speed things up) or a valid application.. junk
> > data is junk data... If you do it to much you should in
> > some way be penalized... aka make the data reliable or
> > reset the connection....
>
>Note that "junk data" is maybe not the best way to describe it (even
>though I, no doubt, started that).  We can determine data that was
>sent, dropped (with good probability) and not retransmitted using
>PR-SCTP.  That data **accomplished no useful work and used scarce
>network resources**.  (I see this as a pretty non-controversial
>opinion.  Do we have agreement on it (i.e., only what I have written
>in this paragraph)?)

Yes, I'd agree with the "**" sentence as it stands. However, it is - in
my view - an artful sentence that clouds the issue. (Hey Mark .. don't
get bent out of shape ... I meant this as a compliment ... not everyone
can write like a sentence that :-) ).

The application *did not know a priori* it was using scarce resources - this
became apparent only "after the fact".

Then the application, *given the choice of getting more current data now
rather than getting old data .. then the new data* chose the latter. And
note the latter actually reduced the congestion by the entropy of the
chunk that became "abandoned" (Randall's new descriptive term).


>And, then there is the question of what to do about data being sent
>that does not accomplish useful work.

Obviously, if you knew that a priori ... you would not have sent it!

There are many applications with the property that they monitor
changing parameters. These applications usually send at regular
intervals ... and, in the absence of congestion, would like data
at all the time intervals the data is sent.

PR-SCTP "timed reliability" provides another mechanism for the
application to chose what gets abandoned in the hope that: 1)
it will get more recent data sooner (because it is not behind the
now stale data) and 2) it actually will reduce the congestion
episode by *not* injecting the (now more stale) data into the
network (i.e., by *saving network resources* when you really
need them saved ... during congestion episodes).

I suggested in my earlier posts that I believed this is close to
optimum behavior during congestion episodes (i.e., without a priori
information) ... and you did not refute or challenge that claim.

So I am at a loss to address your concern lacking either:
1) "a priori knowledge on congestion" or
2) a "garbage/relevance" gauge that the transport could use to
determine "excessive/bad/junk/redundant" data.


>Also, I want to stress that I offered that **excessive** amounts of
>data transmitted without accomplishing useful work should be curbed.

I agree that this should be discouraged ... I just don't know
how to enforce/police at the transport level.

>I never said it was a reason not to do PR-SCTP.

Good to hear that.


>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


Michael A. Ramalho, Ph.D.
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.941.708.4650
Cell: +1.941.544.2844

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Subject: Re: [Tsvwg] PR-SCTP
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Spencer Dawkins wrote:
> 
> OK, let me show how poorly I understand the issue.
> 
> If an application gives clues to transport that something isn't
> as interesting as something else, how is this different from
> per-packet priority marking?

To be more precise, the app is not saying that the data is less
interesting. Rather, in the case of PR-SCTP, the apps is simply saying
that *this piece of data is equally interesting/important as any other
at this moment, but it will become less useful after x seconds*

For this reason, I wouldn't call it "junk data". Maybe better to call it
"time sensitive data".

-Qiaobing

> 
> My first take is that I'm not sure how transports know something
> is junk, in the general case. I'm actually interested in
> understanding more clearly what the axes of "interesting" look
> like. I think I'd like to understand this better, before
> providing local-optimal feedback.
> 
> My understanding of Randall's concept is "deadline discard" - if
> you don't deliver this in 800 ms, don't bother?
> 
> My understanding of Reiner's concept is "reduce retransmission
> latency by retransmitting" - if you have unACKed data and don't
> have anything else new to send, send the old data again?
> 
> What are the other concepts that people are thinking about?
> 
> Spencer
> _______________________________________________
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> tsvwg@ietf.org
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From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] PR-SCTP
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OK, that actually helped in Randall's case - I was thinking
about a linear decay, but see that Randall is talking about a
right-angle decay...

I think my question still applies in Reiner's case (the sender
knows that the data could very well have gotten there the first
time, but would rather retransmit than sit idly and wait for a
timer to pop).

Spencer

--- Qiaobing Xie <Qiaobing.Xie@motorola.com> wrote:
> Spencer Dawkins wrote:
> > 
> > OK, let me show how poorly I understand the issue.
> > 
> > If an application gives clues to transport that something
> isn't
> > as interesting as something else, how is this different from
> > per-packet priority marking?
> 
> To be more precise, the app is not saying that the data is
> less
> interesting. Rather, in the case of PR-SCTP, the apps is
> simply saying
> that *this piece of data is equally interesting/important as
> any other
> at this moment, but it will become less useful after x
> seconds*
> 
> For this reason, I wouldn't call it "junk data". Maybe better
> to call it
> "time sensitive data".
> 
> -Qiaobing
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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Spencer Dawkins <spencer_dawkins@yahoo.com>
Cc: tsvwg@ietf.org
Subject: Re: [Tsvwg] PR-SCTP
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Spencer,

Regardless of where you attempt to police FEC or excessive
retransmissions, the application can simply retransmit one
layer up and avoid whatever measures are taken at the lower
layer.

IMO Only a measure that penalizes an application for sending
data would thwart FEC (at the cost of bulk data transfer).

I don't believe that this is a PR-SCTP gating issue at all.
I think it is a general issue outside the scope of PR-SCTP.
I also don't think that PR-SCTP should be delayed in advancing
while disussing this more general question.

My 2 cents.

--brian

On Wed, 23 Apr 2003, Spencer Dawkins wrote:

> OK, that actually helped in Randall's case - I was thinking
> about a linear decay, but see that Randall is talking about a
> right-angle decay...
> 
> I think my question still applies in Reiner's case (the sender
> knows that the data could very well have gotten there the first
> time, but would rather retransmit than sit idly and wait for a
> timer to pop).
> 
> Spencer
> 
> --- Qiaobing Xie <Qiaobing.Xie@motorola.com> wrote:
> > Spencer Dawkins wrote:
> > > 
> > > OK, let me show how poorly I understand the issue.
> > > 
> > > If an application gives clues to transport that something
> > isn't
> > > as interesting as something else, how is this different from
> > > per-packet priority marking?
> > 
> > To be more precise, the app is not saying that the data is
> > less
> > interesting. Rather, in the case of PR-SCTP, the apps is
> > simply saying
> > that *this piece of data is equally interesting/important as
> > any other
> > at this moment, but it will become less useful after x
> > seconds*
> > 
> > For this reason, I wouldn't call it "junk data". Maybe better
> > to call it
> > "time sensitive data".
> > 
> > -Qiaobing
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

-- 
Brian F. G. Bidulock    ¦ The reasonable man adapts himself to the ¦
bidulock@openss7.org    ¦ world; the unreasonable one persists in  ¦
http://www.openss7.org/ ¦ trying  to adapt the  world  to himself. ¦
                        ¦ Therefore  all  progress  depends on the ¦
                        ¦ unreasonable man. -- George Bernard Shaw ¦
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From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] PR-SCTP
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--- "Brian F. G. Bidulock" <bidulock@openss7.org> wrote:
> Spencer,
> 
> Regardless of where you attempt to police FEC or excessive
> retransmissions, the application can simply retransmit one
> layer up and avoid whatever measures are taken at the lower
> layer.
> 
> IMO Only a measure that penalizes an application for sending
> data would thwart FEC (at the cost of bulk data transfer).

Are there only sticks, and no carrots? We've had better luck
getting people to deploy using carrots (congestion avoidance,
HTTP/1.1, CIDR ...)

> 
> I don't believe that this is a PR-SCTP gating issue at all.
> I think it is a general issue outside the scope of PR-SCTP.
> I also don't think that PR-SCTP should be delayed in advancing
> while disussing this more general question.

Agreed.
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Subject: Re: [Tsvwg] PR-SCTP
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Dear all,

I am under the impression that the "discussion" started
by Mark is a "pseudo"-discussion, more for the sake of
having a discussion than making a point.

Suppose three degrees of freedom (as transport protocol): 
	1.TCP, 
	2.plain SCTP, 
	3.PR-SCTP.
For an application that wants to transmit all data reliably,
it will be able to choose from either of the three, and
in the case of PR-SCTP specify an infinite lifetime, i.e.
reliable delivery, of all data (essentially the same as plain
SCTP). Of course, the application may profit from SCTP multi-homing,
especially in the case where congestion occurs, and retransmission
go along a different, otherwise idle path.

Now, things become interesting for applications that make use
of the unreliable delivery feature, and specify a lifetime
for data to be sent. 
Will this be an incentive for applications to sent more data ?
The applications cannot be sure that the data is NOT sent
over the wire, a priori, so any data submitted to the transport
PR-SCTP may or may not be sent, and therefore may, in the worst 
case *for the application*, hurt the application's congestion 
window. I don't think this is an incentive for applications
to send "junk" data (or data that better not be sent).

As retransmissions are more likely to occur in situations
where the network experiences congestion, PR-SCTP may relieve
this congestion compared to plain SCTP or TCP, where TCP will 
insist on retransmitting the data to the same transport
address. Plain SCTP may do the same (in case associations are
single-homed), and will still put more load on the network
than PR-SCTP (in case associations are multi-homed and use distinct
phiysical paths).

So, is PR-SCTP a bad thing ? I do not think so, for the 
above reasons. It will relieve the network over the long run.

My 2 cents.

Andreas

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

My PGP Key-Fingerprint: 228C 7C2C 3381 2DD4  9998 2812 5C0B 0B04  D382 4AC0
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From mailnull@www1.ietf.org  Thu Apr 24 08:22:51 2003
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From: Mark Allman <mallman@grc.nasa.gov>
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Subject: [Tsvwg] early retransmit limits
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Now for something different and the same....

I'm interested in hearing some opinions on what to do about limiting
the use of early retransmit.  We are in the process of updating the
ID on the subject -- the current version is
draft-allman-tcp-early-rexmt-00.txt.  To briefly recap the draft
says that when the amount of outstanding data is < 4 segments and
the TCP sender has no new data to send (ala limited transmit) then
TCP can use ownd-1 dupacks to trigger fast retransmit behavior
rather than waiting for the RTO to fire.  The idea is basically that
in this case TCP knows that it will never get 3 duplicate ACKs and
cannot induce 3 duplicate ACKs with new data.

In the draft we sketch two versions of "worst case" scenarios
whereby one-third of the packets sent into the network are needless
retransmits caused by reordering not loss.  So, the question is
what, if anything, do we do about this worst case.  So, I'd like to
see some discussion / opinions on a range of reasonable options...

OPTION A
--------
Just don't do early retransmit.  The cost, in terms of network
wasteage and/or network wasteage prevention is just not worth it
for the benefits that ER provides.

OPTION B
--------
Ust use early retransmit without worrying about preventing bogus
retransmits.  In other words, the cases where this really breaks and
sends a huge amount of wasteful traffic into the network are
pathelogical and ER is tightly scoped enough that we don't need to
worry about a limit.

OPTION C
--------
Allow ER, but only once per connection.  Most connections
(especially those that will benefit from ER) are short.  So, this
will help lots of connections.  Also, this scheme is easy and cheap
to implement.  But, it will not help long connections through
networks where cwnd is less than 4 segments (due to either low
bandwidth or lots of statistical multiplexing).

OPTION D
--------
Allow some percentage of transmissions to be spurious.  The
deployment path here is sort of two-fold.

(D.1) If the sender and receiver are using timestamps the sender can
    use Eifel to figure out whether an ERed segment was spurious or
    not.  Alternatively, if the receiver sends DSACKs (which the
    sender has to "learn" by seeing one) then we can use those to
    detect spuriously ERed segments.  In either case, we count the
    number of ERed segments that were wrongly sent and make sure
    that this number is some small precent (say 3% -- an order of
    magnitude less than the worst case ER wasteage) of the total
    traffic being sent.

(D.2) In the case when neither timestamps nor DSACKs are used the
    sender can still use ER.  But, the sender must not ER more than
    some small percent (same as above - say 3%) of segments via ER.
    Essentially, this scheme makes the conservative assumption that
    every segment sent with ER is bogus for limiting purposes.
    (And, so, it is clearly more advantageous to the sender to use a
    D.1-like scheme.)

OPTION E
--------
The same D.1, but replace D.2 with OPTION C -- so, if you don't have
fine-grained information about the spuriousness of what you are
sending then you can only send one ER per connection.  (Of course,
this would also have to come with a way to detect that DSACK was
supported -- say, by overlapping the first two packets by one byte
or some such kludginess.)

OPTION x
--------
Some other tightly specified option that someone thinks of.  Send
text.

My personal leaning is towards option B.  (And, I do not speak for
my co-authors who can (and hopefully will) assert their own
preferences.)  One of the problems is that C, D and E only address
and limit the spuriousness for long connections.  But, for
connections that only send 3 packets (2 unique data packets and a
retransmit -- see draft for a better sketch of this "worst case")
none of these mitigations actually helps at all.  Also, I think the
worst case is pathelogical enough (2 (or 3) segment window,
persisten reordering that is just so and a TCP that cannot send new
data) that even without limiting we're not going to see a ton of
spurious retransmits running around the network because of ER.

I'd be thrilled to hear opinions on which option others like best.

Thanks!

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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Date: Thu, 24 Apr 2003 15:23:51 +0200
To: mallman@grc.nasa.gov
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] early retransmit limits
Cc: tsvwg@ietf.org
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At 14:21 24.04.2003, Mark Allman wrote:
>My personal leaning is towards option B.

As I said before, I'm convinced that option B is the right way to go.


>Also, I think the
>worst case is pathelogical enough (2 (or 3) segment window,
>persisten reordering that is just so and a TCP that cannot send new
>data) that even without limiting we're not going to see a ton of
>spurious retransmits running around the network because of ER.

I'm also convinced that this worst-case is so pathological that we 
shouldn't worry about it.

///Reiner 

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On Thu, 24 Apr 2003, Mark Allman wrote:

> OPTION B
> --------
> Ust use early retransmit without worrying about preventing bogus
> retransmits.  In other words, the cases where this really breaks and
> sends a huge amount of wasteful traffic into the network are
> pathelogical and ER is tightly scoped enough that we don't need to
> worry about a limit.

I think I prefer option B, but if there is a concern about too many
spurious retransmits occuring... perhaps one option may be to use a timer
similar to the delayed ack timer. That way, the sender provides the
opportunity for either new data to be sent or the ack for a reordered
segment to arrive.

~armando

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

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hi all,

here are some thoughts

I wouldn't go for options D and E since they need to specify a parameter 
(the percentage)
and in my opinion this introduces some arbitrariness that I'm not 
comfortable with
since we don't know much about what this value should be.
In this sense we are introducing some complexity into the algorithm 
without a
remarkable benefit.

I wouldn't go for setting a hard threshold (option C) since
the aim of ER is to avoid lengthy timeouts for short TCP connections
as well as for interactive applications which send data in bursts.
Setting a hard threshold we  fail in the case of interactive applications.
Maybe we could consider setting a larger hard threshold to provide
some flexibility for long sessions as well, but again what should be the 
value
of this threshold...

as a coauthour I wouldn't go for option A... :-)

So my personal leaning is also towards B.
I expect that systematic high reordering to be pathological behaviour
and ER is in a certain way self-controling since
it would only trigger when the flying number of segments is small.

regards,
urtzi

Mark Allman wrote:

> 
>
>OPTION A
>--------
>Just don't do early retransmit.  The cost, in terms of network
>wasteage and/or network wasteage prevention is just not worth it
>for the benefits that ER provides.
>
>OPTION B
>--------
>Ust use early retransmit without worrying about preventing bogus
>retransmits.  In other words, the cases where this really breaks and
>sends a huge amount of wasteful traffic into the network are
>pathelogical and ER is tightly scoped enough that we don't need to
>worry about a limit.
>
>OPTION C
>--------
>Allow ER, but only once per connection.  Most connections
>(especially those that will benefit from ER) are short.  So, this
>will help lots of connections.  Also, this scheme is easy and cheap
>to implement.  But, it will not help long connections through
>networks where cwnd is less than 4 segments (due to either low
>bandwidth or lots of statistical multiplexing).
>
>OPTION D
>--------
>Allow some percentage of transmissions to be spurious.  The
>deployment path here is sort of two-fold.
>
>(D.1) If the sender and receiver are using timestamps the sender can
>    use Eifel to figure out whether an ERed segment was spurious or
>    not.  Alternatively, if the receiver sends DSACKs (which the
>    sender has to "learn" by seeing one) then we can use those to
>    detect spuriously ERed segments.  In either case, we count the
>    number of ERed segments that were wrongly sent and make sure
>    that this number is some small precent (say 3% -- an order of
>    magnitude less than the worst case ER wasteage) of the total
>    traffic being sent.
>
>(D.2) In the case when neither timestamps nor DSACKs are used the
>    sender can still use ER.  But, the sender must not ER more than
>    some small percent (same as above - say 3%) of segments via ER.
>    Essentially, this scheme makes the conservative assumption that
>    every segment sent with ER is bogus for limiting purposes.
>    (And, so, it is clearly more advantageous to the sender to use a
>    D.1-like scheme.)
>
>OPTION E
>--------
>The same D.1, but replace D.2 with OPTION C -- so, if you don't have
>fine-grained information about the spuriousness of what you are
>sending then you can only send one ER per connection.  (Of course,
>this would also have to come with a way to detect that DSACK was
>supported -- say, by overlapping the first two packets by one byte
>or some such kludginess.)
>
>  
>


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Armando L. Caro Jr. wrote:

>On Thu, 24 Apr 2003, Mark Allman wrote:
>
>I think I prefer option B, but if there is a concern about too many
>spurious retransmits occuring... perhaps one option may be to use a timer
>similar to the delayed ack timer. That way, the sender provides the
>opportunity for either new data to be sent or the ack for a reordered
>segment to arrive.
>
>  
>
this seems to be a reasonable possibility. The value of the timer
could be the same as the one for delayed ACK, couldn't it?

thanks
urtzi



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Armando-

> if there is a concern about too many spurious retransmits
> occuring... perhaps one option may be to use a timer similar to
> the delayed ack timer. That way, the sender provides the
> opportunity for either new data to be sent or the ack for a
> reordered segment to arrive.

I don't follow... Are you saying that we should do Vern Paxson's
trick of using < 3 dupacks, but adding a small timeout as well?  So,
something like "after 2 dupacks + 20 msec we can fast retransmit".
If not, can you try to explain again (maybe with an example)?  I am
just not sure I am following yet.

Thanks!

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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On Thu, 24 Apr 2003, Mark Allman wrote:

> I don't follow... Are you saying that we should do Vern Paxson's
> trick of using < 3 dupacks, but adding a small timeout as well?

I don't know which trick you are referring to, but it sounds like Vern's
trick is similar to what I was thinking...

> So, something like "after 2 dupacks + 20 msec we can fast retransmit".

Yes. I am not sure about the 20ms begin the delay though. Maybe Urtzi's
idea of the same delay as the delayed ack algorithm uses would fine. But
anyway, let's first understand the idea properly...

The idea is to allow early retransmit to occur, but to try being
conservative by allowing the chance for either (a) 3 dupacks or (b) the
hole to be filled in case of reordering.

case (a)

   1. Suppose the sender sends segments 1, 2, and 3.

   2. The receiver receives segments 2 and 3, and sends two dupacks for 1.

   3. The sender receives two dupacks for 1 (i.e., segment 1 is missing).

   4. Since the TCP sender has no other data in flight and no more
      data to send, ER comes into play. However, the sender delays x ms before
      retransmitting.

   5. If within the x ms delay the sending application provides the
      TCP sender with more data to send, ER is not triggered. Instead,
      the TCP sender sends segment 4 and waits for the third dupack
      for segment 1 to arrive. Upon receiving the third dupack for
      segment 1, the TCP can do a "normal" Fast Retransmit.

   6. Otherwise, retransmit segment 1.


case (b)

   1. Sender sends segments 1, 2, and 3.

   2. The receiver receives segments 2, 3, and 1... in that order.

   3. The sender receives two dupacks for 1, which may begin to imply
      that segment 1 is missing.

   4. Again, since the TCP sender has no other data in flight and no
      more data to send, ER comes into play, but delays x ms before
      retransmitting.

   5. If the ack for segment 1 arrives at the sender before the x ms
      delay expires, then the sender infers that a reordering occured
      (as in this example) and no retransmission is necessary.

   6. Otherwise, retransmit segment 1.


~armando

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


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Armando-

Thanks for the clarification.  The ideas makes good sense to me
now.

I'm happy to throw that option on the list as a possibility.  So,
the list is something like:

    option A - no ER
    option B - ER as defined in the draft
    option C - ER, but only once per connection
    option D - use Eifel or DSACKs to ensure that bogus ERs are <
	some small percent of the data sent; if we don't have DSACK
	or timestamps, then just say that ERs cannot amount to more
	than this small percent of the data we send
    option E - use Eifel or DSACKs as option D; if we don't have
	timestamps or DSACKs then revert to option C and ER at most
	once per connection
    option F - ER after ownd - 1 dupacks arrive and some timeout
	occurs (hoping that in the idle time we get an ACK that
	indicates reordering or a new segment to send ala limited
	transmit) 

I still don't like A.  Go figure...

And, I still think C-E are predicated on us thinking that B would
cause too much spurious traffic in the network.  F tries to be more
robust than B (at the cost of a bit of extra machinery).  Is F
enough better that we then don't need to think about C-E?  

I still like B.  But, wait ....

I guess what I'd really like to know is whether folks are worried
about option B.  Several people have indicated on the list (and off
to me today) that they think ER ala B is scoped enough that it is
Just OK.  Before having a long discussion about C-F I guess I'd like
to get a feeling for whether some are concerned about B (which I
think is clearly the easiest path).

Thanks!

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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From: Joseph Ishac <jishac@grc.nasa.gov>
To: tsvwg@ietf.org
Subject: Re: [Tsvwg] early retransmit limits
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On Thu, Apr 24, 2003 at 03:40:13PM -0400, Mark Allman wrote:
>     option A - no ER
>     option B - ER as defined in the draft
>     option C - ER, but only once per connection
>     option D - use Eifel or DSACKs to ensure that bogus ERs are <
> 	some small percent of the data sent; if we don't have DSACK
> 	or timestamps, then just say that ERs cannot amount to more
> 	than this small percent of the data we send
>     option E - use Eifel or DSACKs as option D; if we don't have
> 	timestamps or DSACKs then revert to option C and ER at most
> 	once per connection
>     option F - ER after ownd - 1 dupacks arrive and some timeout
> 	occurs (hoping that in the idle time we get an ACK that
> 	indicates reordering or a new segment to send ala limited
> 	transmit) 
> 
> I still don't like A.  Go figure...
> 
> And, I still think C-E are predicated on us thinking that B would
> cause too much spurious traffic in the network.  F tries to be more
> robust than B (at the cost of a bit of extra machinery).  Is F
> enough better that we then don't need to think about C-E?  
> 
> I still like B.  But, wait ....
> 
> I guess what I'd really like to know is whether folks are worried
> about option B.  Several people have indicated on the list (and off
> to me today) that they think ER ala B is scoped enough that it is
> Just OK.  Before having a long discussion about C-F I guess I'd like
> to get a feeling for whether some are concerned about B (which I
> think is clearly the easiest path).

I can't think a way to really break option B, so I'd vote for it ;)


However, I think it would be beneficial to research the benefit of
options that strengthen ER (~if needed~).  So, if network conditions
change and reordering becomes more prevalent, we would know what the
benefit/impact of using ER would be (in its different forms).

On a side note.  There are studies to try and make TCP more robust to
packet reordering.  One such method is to increase dupthresh, and I'll
use that in the next example.  Suppose that during a connection, we've
grown dupthresh to a value greater than three.  If the connection then
later experiences an ER situation, is ER still valid? (b/c an increased
dupthresh would mean that you've experienced some amount of reordering,
which is non-negligible)

-Joseph
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Subject: Re: [Tsvwg] early retransmit limits
From: "Srijith. K" <srijith@comp.nus.edu.sg>
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To add to option F, the *max* of the delay should be such that you give
just enough time for the ACK of the retransmitted packet to reach the
sender before the timeout occurs.

So, d(max)=RTO-x
where x can be RTT(smooth) or RTT(max) or RTT(prev).

Srijith


> Yes. I am not sure about the 20ms begin the delay though. Maybe Urtzi's
> idea of the same delay as the delayed ack algorithm uses would fine. But
> anyway, let's first understand the idea properly...
>
> The idea is to allow early retransmit to occur, but to try being
> conservative by allowing the chance for either (a) 3 dupacks or (b) the
> hole to be filled in case of reordering.
>
> case (a)
>
>    1. Suppose the sender sends segments 1, 2, and 3.
>
>    2. The receiver receives segments 2 and 3, and sends two dupacks for
> 1.
>
>    3. The sender receives two dupacks for 1 (i.e., segment 1 is
> missing).
>
>    4. Since the TCP sender has no other data in flight and no more
>       data to send, ER comes into play. However, the sender delays x ms
> before retransmitting.
>
>    5. If within the x ms delay the sending application provides the
>       TCP sender with more data to send, ER is not triggered. Instead,
> the TCP sender sends segment 4 and waits for the third dupack for
> segment 1 to arrive. Upon receiving the third dupack for segment
> 1, the TCP can do a "normal" Fast Retransmit.
>
>    6. Otherwise, retransmit segment 1.
>
>
> case (b)
>
>    1. Sender sends segments 1, 2, and 3.
>
>    2. The receiver receives segments 2, 3, and 1... in that order.
>
>    3. The sender receives two dupacks for 1, which may begin to imply
>       that segment 1 is missing.
>
>    4. Again, since the TCP sender has no other data in flight and no
>       more data to send, ER comes into play, but delays x ms before
> retransmitting.
>
>    5. If the ack for segment 1 arrives at the sender before the x ms
>       delay expires, then the sender infers that a reordering occured
> (as in this example) and no retransmission is necessary.
>
>    6. Otherwise, retransmit segment 1.
>
>
> ~armando
>
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0
>
>
> _______________________________________________
> tsvwg mailing list
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> https://www1.ietf.org/mailman/listinfo/tsvwg




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On Thu, 24 Apr 2003, Joseph Ishac wrote:

> I can't think a way to really break option B, so I'd vote for it ;)

Yes, you do... reordering ;-)

> However, I think it would be beneficial to research the benefit of
> options that strengthen ER (~if needed~).  So, if network conditions
> change and reordering becomes more prevalent, we would know what the
> benefit/impact of using ER would be (in its different forms).
>
> On a side note.  There are studies to try and make TCP more robust to
> packet reordering.  One such method is to increase dupthresh, and I'll

Well, my assumption was that the work that people are doing with TCP and
reordering is because reordering _does_ occur. I remember reading at least
one paper which said that reordering wasn't pathological (that was the
title actually). So my question is... is it _that_ unlikely that the
network is reordering, ownd < 4, and the sender doesn't have any more data
to send. Personally, _I_ don't know that we can say that, but maybe
someone else has more insight in this area.

> use that in the next example.  Suppose that during a connection, we've
> grown dupthresh to a value greater than three.  If the connection then
> later experiences an ER situation, is ER still valid? (b/c an increased
> dupthresh would mean that you've experienced some amount of reordering,
> which is non-negligible)

Maybe this problem is where option F would work well. If the dupack
threshold is dynamic based on reordering experienced, then ER comes into
play anytime ownd < dupack threshold. Then by using a delayed ER (option
F), the sender is being extra conservative and taking into consideration
the reordering already experienced.

~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] early retransmit limits
From: "Srijith. K" <srijith@comp.nus.edu.sg>
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Opps.. on second thought:

d(max)=(RTO-time_already_waiting_for_ack) - x

x=RTT(smooth) or RTT(max) or RTT(prev)

Regards,
Srijith

> Yes. I am not sure about the 20ms begin the delay though. Maybe Urtzi's
> idea of the same delay as the delayed ack algorithm uses would fine. But
> anyway, let's first understand the idea properly...
>
> The idea is to allow early retransmit to occur, but to try being
> conservative by allowing the chance for either (a) 3 dupacks or (b) the
> hole to be filled in case of reordering.
>
> case (a)
>
>    1. Suppose the sender sends segments 1, 2, and 3.
>
>    2. The receiver receives segments 2 and 3, and sends two dupacks for
> 1.
>
>    3. The sender receives two dupacks for 1 (i.e., segment 1 is
> missing).
>
>    4. Since the TCP sender has no other data in flight and no more
>       data to send, ER comes into play. However, the sender delays x ms
> before retransmitting.
>
>    5. If within the x ms delay the sending application provides the
>       TCP sender with more data to send, ER is not triggered. Instead,
> the TCP sender sends segment 4 and waits for the third dupack for
> segment 1 to arrive. Upon receiving the third dupack for segment
> 1, the TCP can do a "normal" Fast Retransmit.
>
>    6. Otherwise, retransmit segment 1.
>
>
> case (b)
>
>    1. Sender sends segments 1, 2, and 3.
>
>    2. The receiver receives segments 2, 3, and 1... in that order.
>
>    3. The sender receives two dupacks for 1, which may begin to imply
>       that segment 1 is missing.
>
>    4. Again, since the TCP sender has no other data in flight and no
>       more data to send, ER comes into play, but delays x ms before
> retransmitting.
>
>    5. If the ack for segment 1 arrives at the sender before the x ms
>       delay expires, then the sender infers that a reordering occured
> (as in this example) and no retransmission is necessary.
>
>    6. Otherwise, retransmit segment 1.
>
>
> ~armando
>
> 0--                                                  --0
> | Armando L. Caro Jr.      |  Protocol Engineering Lab |
> | www.cis.udel.edu/~acaro  |    University of Delaware |
> 0--                                                  --0
>
>
> _______________________________________________
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> tsvwg@ietf.org
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On Fri, 25 Apr 2003, Srijith. K wrote:

> To add to option F, the *max* of the delay should be such that you give
> just enough time for the ACK of the retransmitted packet to reach the
> sender before the timeout occurs.
>
> So, d(max)=RTO-x
> where x can be RTT(smooth) or RTT(max) or RTT(prev).

I think that the guidelines used for the delayed ack algorithm is a good
starting point. Recommended: 200ms. Max: 500ms. If delayed ER gets close
to a timeout, then the extra complexity isn't worth the potential gain.

~armando

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

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> However, I think it would be beneficial to research the benefit of
> options that strengthen ER (~if needed~).  So, if network
> conditions change and reordering becomes more prevalent, we would
> know what the benefit/impact of using ER would be (in its
> different forms).

I agree that if reordering became a lot more prevlent then it would
behoove us to re-think ER (both in terms of network resource
wasteage and the cwnd sacrifice that ER causes).  Put another way,
spurious ERs cause a penalty for the connection itself and so there
is an incentive to not spuriously ER.

> On a side note.  There are studies to try and make TCP more robust
> to packet reordering.  One such method is to increase dupthresh,
> and I'll use that in the next example.  Suppose that during a
> connection, we've grown dupthresh to a value greater than three.
> If the connection then later experiences an ER situation, is ER
> still valid? (b/c an increased dupthresh would mean that you've
> experienced some amount of reordering, which is non-negligible)

Maybe.  In the dupthresh adjustment schemes sketched in
draft-blanton-tcp-reordering-00.txt the dupthresh is cwnd-1 at its
maximum.  So, as the window drains dupthresh is reduced.  But,
clearly, we could keep a maxdupthresh that indicates if we have ever
changed dupthresh based on reordering.  But, if we're headed down
that road then I think all we really need to do is look for
duplicate ACKs and no loss and set a flag if we see such an event.
Then we can use that as input to some sort of ER decision.  (A
dupack can be caused by packet replication in the network, but
everything I have seen shows that to be quite rare.)  My own opinion
right now is that this is a bit of overkill for ER because its so
tightly scoped.  But, something to keep in mind for down the road if
ER ends up busting a lot.

Thanks!

allman


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> So my question is... is it _that_ unlikely that the network is
> reordering, ownd < 4, and the sender doesn't have any more data to
> send. Personally, _I_ don't know that we can say that, but maybe
> someone else has more insight in this area.

I would be concerned if I thought that this would cause a whole lot
of extra stuff to be sent into the network.  But, my feeling is that
the situation that breaks ER is actually not going to add a huge
percentage of unnecessary traffic to the network in general.  I bet
you could find some paths where it may be worse that others, but in
the overall scheme of things I don't think it will lead to
problems.  But, without a whole pile of data from a whole pile of
places I cannot say that for sure.  (And, that sounds like a good
argument for experimental to me.)

> Maybe this problem is where option F would work well. If the
> dupack threshold is dynamic based on reordering experienced, then
> ER comes into play anytime ownd < dupack threshold.

Which may be exactly what we have now if the dupack threshold is
tied to cwnd (ala another i-d of mine).  (Note that not all schemes
might have the same properties.)

allman


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From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] early retransmit limits 
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Dear Mark,

You guys understand this stuff more clearly than I do, but my
spin is 

- the only connections affected are either in slow-start or
application-limited, right?

- the biggest burst we would see (on average) is one-half the
dupack threshold, right?

- with a dupack threshold of 3, we're within the burst rate we
see at the beginning of a new connection using initial cwnd=4,
right?

- if we actually DO succeed in losing something on the forward
path, we perform congestion avoidance anyway, right?

If these are true, I don't think we need to worry about the
effect of ER on the network with dupack threshold = 3.

For higher values of dupack threshold, I'm less certain, but the
answer is almost certainly tied to how big the dupack threshold
is allowed to become.

Spencer

--- Mark Allman <mallman@grc.nasa.gov> wrote:
> 
> > So my question is... is it _that_ unlikely that the network
> is
> > reordering, ownd < 4, and the sender doesn't have any more
> data to
> > send. Personally, _I_ don't know that we can say that, but
> maybe
> > someone else has more insight in this area.
> 
> I would be concerned if I thought that this would cause a
> whole lot
> of extra stuff to be sent into the network.  But, my feeling
> is that
> the situation that breaks ER is actually not going to add a
> huge
> percentage of unnecessary traffic to the network in general. 
> I bet
> you could find some paths where it may be worse that others,
> but in
> the overall scheme of things I don't think it will lead to
> problems.  But, without a whole pile of data from a whole pile
> of
> places I cannot say that for sure.  (And, that sounds like a
> good
> argument for experimental to me.)
> 
> > Maybe this problem is where option F would work well. If the
> > dupack threshold is dynamic based on reordering experienced,
> then
> > ER comes into play anytime ownd < dupack threshold.
> 
> Which may be exactly what we have now if the dupack threshold
> is
> tied to cwnd (ala another i-d of mine).  (Note that not all
> schemes
> might have the same properties.)
> 
> allman
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To: Spencer Dawkins <spencer_dawkins@yahoo.com>
From: Mark Allman <mallman@grc.nasa.gov>
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Subject: Re: [Tsvwg] early retransmit limits 
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Spencer-

> - the only connections affected are either in slow-start or
> application-limited, right?

Or, if the advertised window is quite small.

The slow start case may not actually be an issue because you can use
limited transmit if you have new data to send and advertised window
to do it.

> - the biggest burst we would see (on average) is one-half the
> dupack threshold, right?

I don't follow this... ER acts just like a fast retransmit -- only
it requires fewer duplicate ACKs.  So, I don't understand where the
bursting is coming from.

> - with a dupack threshold of 3, we're within the burst rate we
> see at the beginning of a new connection using initial cwnd=4,
> right?

(So, I can't answer this one either.)

> - if we actually DO succeed in losing something on the forward
> path, we perform congestion avoidance anyway, right?

Right.  ER is just an early fast retransmit.  All the CC changes
still apply.  Really, we're just looking at all the information and
judging that we could never fast retransmit if something was lost
and given that then we're readjusting the trigger for fast
retransmit. 

> For higher values of dupack threshold, I'm less certain, but the
> answer is almost certainly tied to how big the dupack threshold
> is allowed to become.

Right.  And, all that would have to be worked out if you wanted to
apply ER to dupthresh > 3 -- but, the current document scopes it to
specifically the current, static dupthresh.

Thanks!

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] early retransmit limits 
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OK, I mispoke...

Me-getting-smarter inline...

Spencer

--- Mark Allman <mallman@grc.nasa.gov> wrote:
> 
> Spencer-
> 
> > - the only connections affected are either in slow-start or
> > application-limited, right?
> 
> Or, if the advertised window is quite small.
> 
> The slow start case may not actually be an issue because you
> can use
> limited transmit if you have new data to send and advertised
> window
> to do it.

OK, so with the current dupack threshold = 3, the amount of
additional traffic we might throw in is still less than 3
segments, right?

> 
> > - the biggest burst we would see (on average) is one-half
> the
> > dupack threshold, right?
> 
> I don't follow this... ER acts just like a fast retransmit --
> only
> it requires fewer duplicate ACKs.  So, I don't understand
> where the
> bursting is coming from.

Sorry, I misused the word. I meant additional traffic that might
not have happened if we'd RTOed instead. (and if I'm still
confused).

> 
> > - with a dupack threshold of 3, we're within the burst rate
> we
> > see at the beginning of a new connection using initial
> cwnd=4,
> > right?
> 
> (So, I can't answer this one either.)

Again, my point was that we are sending a small number of
additional segments (compared to RTO) ...

> 
> > - if we actually DO succeed in losing something on the
> forward
> > path, we perform congestion avoidance anyway, right?
> 
> Right.  ER is just an early fast retransmit.  All the CC
> changes
> still apply.  Really, we're just looking at all the
> information and
> judging that we could never fast retransmit if something was
> lost
> and given that then we're readjusting the trigger for fast
> retransmit. 

So I think we are managing to agree, even though I was
incoherent...

> 
> > For higher values of dupack threshold, I'm less certain, but
> the
> > answer is almost certainly tied to how big the dupack
> threshold
> > is allowed to become.
> 
> Right.  And, all that would have to be worked out if you
> wanted to
> apply ER to dupthresh > 3 -- but, the current document scopes
> it to
> specifically the current, static dupthresh.
> 
> Thanks!
> 
> allman

Thanks for helping me understand better...

Spencer
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> Sorry, I misused the word. I meant additional traffic that might
> not have happened if we'd RTOed instead. (and if I'm still
> confused).

Ah... ok.  So, the amount of additional traffic that ER would inject
into the network over and above what the RTO would is:

  * zero in the case when the segment in question is actually lost

  * likely one segment in the case when reordering fools ER (that
    is, we likely resent a packet that would never have been resent
    via the RTO timer)

Does that make sense?

allman
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Date: Fri, 25 Apr 2003 11:15:54 -0500
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] PR-SCTP
To: Spencer Dawkins <spencer_dawkins@yahoo.com>
Cc: TSV WG list <tsvwg@ietf.org>
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On 4/23/03, Spencer Dawkins wrote:

>If an application gives clues to transport that something
>isn't as interesting as something else, how is this
>different from per-packet priority marking?

Actually PR-SCTP is very different from per-packet priority
marking in several important ways:

-- Per-packet priority marking could be used by network
   elements to drop packets themselves. PR-SCTP only
   drops packets in the sending service, and only drops
   retransmissions.
   
-- Because drops are done by the sending service, the
   set of packets dropped can be done in a service sensitive
   fashion and based upon dynamic information not available
   at the time the packet was originally sent.
   
While at first glance it might seem preferable to allow the
network elements forced to drop packets in the face of
congenstion to be given hints on which packets to drop.
But to be effective, such hints would have to be available
on *most* of the packets going through the element.

PR-SCTP is effective in optimizing an applications response
to network congestion, no matter how many or how few other
competing users are employing the same or similiar
algorithms. Keep in mind that the purpose of dropping
a packet for retransmission will typically be to avoid
holding up more critical data that would otherwise be
behind the retransmitted data.

The sending PR-SCTP service can also consider many factors
in determining which data to abandon retransmission of, not
all of which could be translated to a per-packet priority:

-- When the data must be received to be useful. The primary
   example of this is being streaming media delivery.
   
-- What portion of a useful work unit the dropped data was.
   For example, it makes more sense to retransmit one packet
   that will complete 17 other packets that made it, rather
   than to retransmit a packet that was part of a message
   that had been totally lost.
   
-- The importance of the data to the receiver. Many lossy 
   encoding schemes can employ a pyramid encoding, where
   a few data chunks can give a crude picture, and later
   chunks refine the quality of the image. Obviously 
   there would be little need to retransmit the refining
   chunks in preference to the base chunks.
   
In all likelihood, PR-SCTP services will only support a
comibniation of SCTP message integrity and a timed
expiration. But the draft is quite explicit that the sending
service is free to provide other value added services, and
free to evolve them over time.

So PR-SCTP can be implemented by any two peers, and the
"drop selection" algorithm can be refined by the sending
service at any time.

A packet-priority scheme would sit around for a long time
before network elements actually implemented it, and once
implemented it would be frozen.

So I would say that PR-SCTP is quite a bit friendlier to
both the network and application than packet priority
marking.


 

Caitlin Bestler
http://asomi.com/CaitlinBestler/
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Caitlin,

On Fri, 25 Apr 2003, Caitlin Bestler wrote:

> On 4/23/03, Spencer Dawkins wrote:
> 
> >If an application gives clues to transport that something
> >isn't as interesting as something else, how is this
> >different from per-packet priority marking?
> 
> Actually PR-SCTP is very different from per-packet priority
> marking in several important ways:
> 
> -- Per-packet priority marking could be used by network
>    elements to drop packets themselves. PR-SCTP only
>    drops packets in the sending service, and only drops
>    retransmissions.

Correct me if I'm wrong, but is not a PR-SCTP implementation
also allowed drop packets that have exipired before they are
transmitted for the first time (i.e due to congestion)?

--brian

>    
> -- Because drops are done by the sending service, the
>    set of packets dropped can be done in a service sensitive
>    fashion and based upon dynamic information not available
>    at the time the packet was originally sent.
>    
> While at first glance it might seem preferable to allow the
> network elements forced to drop packets in the face of
> congenstion to be given hints on which packets to drop.
> But to be effective, such hints would have to be available
> on *most* of the packets going through the element.
> 
> PR-SCTP is effective in optimizing an applications response
> to network congestion, no matter how many or how few other
> competing users are employing the same or similiar
> algorithms. Keep in mind that the purpose of dropping
> a packet for retransmission will typically be to avoid
> holding up more critical data that would otherwise be
> behind the retransmitted data.
> 
> The sending PR-SCTP service can also consider many factors
> in determining which data to abandon retransmission of, not
> all of which could be translated to a per-packet priority:
> 
> -- When the data must be received to be useful. The primary
>    example of this is being streaming media delivery.
>    
> -- What portion of a useful work unit the dropped data was.
>    For example, it makes more sense to retransmit one packet
>    that will complete 17 other packets that made it, rather
>    than to retransmit a packet that was part of a message
>    that had been totally lost.
>    
> -- The importance of the data to the receiver. Many lossy 
>    encoding schemes can employ a pyramid encoding, where
>    a few data chunks can give a crude picture, and later
>    chunks refine the quality of the image. Obviously 
>    there would be little need to retransmit the refining
>    chunks in preference to the base chunks.
>    
> In all likelihood, PR-SCTP services will only support a
> comibniation of SCTP message integrity and a timed
> expiration. But the draft is quite explicit that the sending
> service is free to provide other value added services, and
> free to evolve them over time.
> 
> So PR-SCTP can be implemented by any two peers, and the
> "drop selection" algorithm can be refined by the sending
> service at any time.
> 
> A packet-priority scheme would sit around for a long time
> before network elements actually implemented it, and once
> implemented it would be frozen.
> 
> So I would say that PR-SCTP is quite a bit friendlier to
> both the network and application than packet priority
> marking.
> 
> 
>  
> 
> Caitlin Bestler
> http://asomi.com/CaitlinBestler/
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

-- 
Brian F. G. Bidulock    ¦ The reasonable man adapts himself to the ¦
bidulock@openss7.org    ¦ world; the unreasonable one persists in  ¦
http://www.openss7.org/ ¦ trying  to adapt the  world  to himself. ¦
                        ¦ Therefore  all  progress  depends on the ¦
                        ¦ unreasonable man. -- George Bernard Shaw ¦
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Date: Fri, 25 Apr 2003 15:16:37 -0500
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] PR-SCTP
To: bidulock@openss7.org, TSV WG list <tsvwg@ietf.org>
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On 4/25/03, Brian F. G. Bidulock wrote:

>Correct me if I'm wrong, but is not a PR-SCTP
>implementation also allowed drop packets that have
>exipired before they are transmitted for the first time
>(i.e due to congestion)?

It's after a TSN has been assigned. So it would depend on
when the implementation assigned TSNs. But you are correct
that theoretically a service *could* assign a TSN and then
drop it before sending it even once.

Caitlin Bestler
http://asomi.com/CaitlinBestler/
_______________________________________________
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Date: Fri, 25 Apr 2003 13:19:51 -0700 (PDT)
From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] PR-SCTP
To: TSV WG list <tsvwg@ietf.org>
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Dear Caitlin,

I think I knew most of this, but it's a REALLY good summary!
Thanks for the clear explanation. 

Spencer
--- Caitlin Bestler <cait@asomi.com> wrote:
> On 4/23/03, Spencer Dawkins wrote:
> 
> >If an application gives clues to transport that something
> >isn't as interesting as something else, how is this
> >different from per-packet priority marking?
> 
> Actually PR-SCTP is very different from per-packet priority
> marking in several important ways:
> 
> -- Per-packet priority marking could be used by network
>    elements to drop packets themselves. PR-SCTP only
>    drops packets in the sending service, and only drops
>    retransmissions.
>    
> -- Because drops are done by the sending service, the
>    set of packets dropped can be done in a service sensitive
>    fashion and based upon dynamic information not available
>    at the time the packet was originally sent.
>    
> While at first glance it might seem preferable to allow the
> network elements forced to drop packets in the face of
> congenstion to be given hints on which packets to drop.
> But to be effective, such hints would have to be available
> on *most* of the packets going through the element.
> 
> PR-SCTP is effective in optimizing an applications response
> to network congestion, no matter how many or how few other
> competing users are employing the same or similiar
> algorithms. Keep in mind that the purpose of dropping
> a packet for retransmission will typically be to avoid
> holding up more critical data that would otherwise be
> behind the retransmitted data.
> 
> The sending PR-SCTP service can also consider many factors
> in determining which data to abandon retransmission of, not
> all of which could be translated to a per-packet priority:
> 
> -- When the data must be received to be useful. The primary
>    example of this is being streaming media delivery.
>    
> -- What portion of a useful work unit the dropped data was.
>    For example, it makes more sense to retransmit one packet
>    that will complete 17 other packets that made it, rather
>    than to retransmit a packet that was part of a message
>    that had been totally lost.
>    
> -- The importance of the data to the receiver. Many lossy 
>    encoding schemes can employ a pyramid encoding, where
>    a few data chunks can give a crude picture, and later
>    chunks refine the quality of the image. Obviously 
>    there would be little need to retransmit the refining
>    chunks in preference to the base chunks.
>    
> In all likelihood, PR-SCTP services will only support a
> comibniation of SCTP message integrity and a timed
> expiration. But the draft is quite explicit that the sending
> service is free to provide other value added services, and
> free to evolve them over time.
> 
> So PR-SCTP can be implemented by any two peers, and the
> "drop selection" algorithm can be refined by the sending
> service at any time.
> 
> A packet-priority scheme would sit around for a long time
> before network elements actually implemented it, and once
> implemented it would be frozen.
> 
> So I would say that PR-SCTP is quite a bit friendlier to
> both the network and application than packet priority
> marking.
_______________________________________________
tsvwg mailing list
tsvwg@ietf.org
https://www1.ietf.org/mailman/listinfo/tsvwg



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Date: Fri, 25 Apr 2003 14:20:09 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Caitlin Bestler <cait@asomi.com>
Cc: TSV WG list <tsvwg@ietf.org>
Subject: Re: [Tsvwg] PR-SCTP
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Caitlin,

Section 4.1 of draft-stewart-tsvwg-prsctp-03.txt says:

    TR3) Before assigning a TSN to any message, the SCTP send MUST
       evaluate the lifetime of that message.  If it is expired, the SCTP
       sender MUST NOT assign a TSN to that message, but instead SHOULD
       issue a notification to the upper layer and drop the mssage.

So, I suppose it is wrong to say that only retransmissions are
dropped.  Even transmissions will be dropped *before* assigning a TSN.

--brian


On Fri, 25 Apr 2003, Caitlin Bestler wrote:

> On 4/25/03, Brian F. G. Bidulock wrote:
> 
> >Correct me if I'm wrong, but is not a PR-SCTP
> >implementation also allowed drop packets that have
> >exipired before they are transmitted for the first time
> >(i.e due to congestion)?
> 
> It's after a TSN has been assigned. So it would depend on
> when the implementation assigned TSNs. But you are correct
> that theoretically a service *could* assign a TSN and then
> drop it before sending it even once.
> 
> Caitlin Bestler
> http://asomi.com/CaitlinBestler/

-- 
Brian F. G. Bidulock    ¦ The reasonable man adapts himself to the ¦
bidulock@openss7.org    ¦ world; the unreasonable one persists in  ¦
http://www.openss7.org/ ¦ trying  to adapt the  world  to himself. ¦
                        ¦ Therefore  all  progress  depends on the ¦
                        ¦ unreasonable man. -- George Bernard Shaw ¦
_______________________________________________
tsvwg mailing list
tsvwg@ietf.org
https://www1.ietf.org/mailman/listinfo/tsvwg



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	Hi all!

> Caitlin,
> 
> Section 4.1 of draft-stewart-tsvwg-prsctp-03.txt says:
> 
>     TR3) Before assigning a TSN to any message, the SCTP send MUST
>        evaluate the lifetime of that message.  If it is 
> expired, the SCTP
>        sender MUST NOT assign a TSN to that message, but 
> instead SHOULD
>        issue a notification to the upper layer and drop the mssage.
> 
> So, I suppose it is wrong to say that only retransmissions are
> dropped.  Even transmissions will be dropped *before* assigning a TSN.
> 
> --brian

	Well, in fact this is something that the "plain" SCTP implementation might do as well... At least this has been always been my understanding about the use of the "life time" parameter in the SEND primitive in RFC 2960 10.1 E), and the "sinfo_timetolive" field of struct sctp_sndrcvinfo in section 5.2.2 of the SCTP sockets draft...

	So I guess that it is wrong to say that only retransmissions can be dropped, but in fact, the new thing that PR-SCTP adds is dropping retransmissions (a "plain" SCTP peer will never realize you are dropping unsent data).

	BR Iván Arias Rodríguez

> On Fri, 25 Apr 2003, Caitlin Bestler wrote:
> 
> > On 4/25/03, Brian F. G. Bidulock wrote:
> > 
> > >Correct me if I'm wrong, but is not a PR-SCTP
> > >implementation also allowed drop packets that have
> > >exipired before they are transmitted for the first time
> > >(i.e due to congestion)?
> > 
> > It's after a TSN has been assigned. So it would depend on
> > when the implementation assigned TSNs. But you are correct
> > that theoretically a service *could* assign a TSN and then
> > drop it before sending it even once.
> > 
> > Caitlin Bestler
> > http://asomi.com/CaitlinBestler/
> 
> -- 
> Brian F. G. Bidulock    ¦ The reasonable man adapts himself to the ¦
> bidulock@openss7.org    ¦ world; the unreasonable one persists in  ¦
> http://www.openss7.org/ ¦ trying  to adapt the  world  to himself. ¦
>                         ¦ Therefore  all  progress  depends on the ¦
>                         ¦ unreasonable man. -- George Bernard Shaw ¦
> _______________________________________________
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> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 
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Date: Mon, 28 Apr 2003 15:17:31 +0200
To: Phil Karn <karn@qualcomm.com>
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Re: Rethinking Karn's algorithm
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At 03:33 10.04.2003, Phil Karn wrote:
>TCP header timestamps have made my algorithm essentially obsolete, [...]
>
>Header overhead is a non-issue here, IHMO. The savings in unnecessary 
>retransmits more than pays back the investment in a larger TCP header.

catching up ...

I pretty much agree since that is also what we have seen in our investigations.

A different question that was discussed before on the list is the 
exponential backoff; up to ~60 seconds. Don't you think that that is a bit 
too conservative?

///Reiner 

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From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] Re: Rethinking Karn's algorithm
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And, just so I'm not confused and confusing others...

The one notification that we had room-consensus on at the San
Francisco Trigtran BoF was "link up", which makes 

- less sense when you're talking about an RTO that's backed off
to 2-3 seconds, and 

- more sense when you're talking about an RTO that's backed off
to 60 seconds.

What I think Reiner and I are both asking, on two different
mailing lists under two different threads, is "how do we keep
fully-backed-off TCPs from waiting for 30 seconds, on average,
with a human in the loop and a working connection path, before
the TCP figures out that the connection path is working?"

So, if people think the Internet would melt down if the maximum
RTO timer was reduced to 10 seconds, that's interesting; if
people think the Internet would not melt down if it was reduced
to 15 seconds, that's also interesting ...

But I would also like to see other opinions expressed here
(Reiner and I already talk, so ...).

Spencer

--- Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se> wrote:
> A different question that was discussed before on the list is
> the 
> exponential backoff; up to ~60 seconds. Don't you think that
> that is a bit 
> too conservative?
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Spencer-

Nobody answers my questions, so I am using that spirit to not answer
your's, but rather to turn it around on you... :)

> So, if people think the Internet would melt down if the maximum
> RTO timer was reduced to 10 seconds, that's interesting; if people
> think the Internet would not melt down if it was reduced to 15
> seconds, that's also interesting ...

What I am interested in knowing is where you think the line needs to
be drawn to obviate the need for link-up.  If an average wait of 30
seconds is too much what about 15 seconds?  10?  5?  2?

Now I'll sort of answer your question... I believe that the max RTO
should be "significant".  It should not be 3 seconds.  Should it be
60 seconds?  Maybe not.  Is it worth worrying about?  Well, I think
that depends on your answer to the above question.

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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Date: Mon, 28 Apr 2003 14:22:05 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Ivan.Arias-Rodriguez@nokia.com
Cc: cait@asomi.com, tsvwg@ietf.org
Subject: Re: [Tsvwg] PR-SCTP
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Ivan.Arias-Rodriguez,

On Mon, 28 Apr 2003, Ivan.Arias-Rodriguez@nokia.com wrote:

> 	Hi all!
> 
> > Caitlin,
> > 
> > Section 4.1 of draft-stewart-tsvwg-prsctp-03.txt says:
> > 
> >     TR3) Before assigning a TSN to any message, the SCTP send MUST
> >        evaluate the lifetime of that message.  If it is 
> > expired, the SCTP
> >        sender MUST NOT assign a TSN to that message, but 
> > instead SHOULD
> >        issue a notification to the upper layer and drop the mssage.
> > 
> > So, I suppose it is wrong to say that only retransmissions are
> > dropped.  Even transmissions will be dropped *before* assigning a TSN.
> > 
> > --brian
> 
> 	Well, in fact this is something that the "plain" SCTP implementation
> 	might do as well... At least this has been always been my
> 	understanding about the use of the "life time" parameter in the SEND
> 	primitive in RFC 2960 10.1 E), and the "sinfo_timetolive" field of
> 	struct sctp_sndrcvinfo in section 5.2.2 of the SCTP sockets draft...
> 
> 	So I guess that it is wrong to say that only retransmissions can be
> 	dropped, but in fact, the new thing that PR-SCTP adds is dropping
> 	retransmissions (a "plain" SCTP peer will never realize you are
> 	dropping unsent data).
> 
> 	BR Iván Arias Rodríguez

But PR-SCTP "timed reliability" says MUST (RFC 2960 says that lifetimes
are OPTIONAL).  So making this behavior mandatory is another change of
PR-SCTP.

I think that it is important to note when speaking of junk data that,
according to TR3 above, PR-SCTP can, under circumstances of congestion,
discard messages *before* they are sent for the first time, resulting in
even less traffic on the network than provided for by the congestion 
algorithm.  That is, PR-SCTP will exhibit even *more* conservative
congestion avoidance than regular SCTP using "timed-reliability" than
would be exhibited otherwise.

--brian

> 
> > On Fri, 25 Apr 2003, Caitlin Bestler wrote:
> > 
> > > On 4/25/03, Brian F. G. Bidulock wrote:
> > > 
> > > >Correct me if I'm wrong, but is not a PR-SCTP
> > > >implementation also allowed drop packets that have
> > > >exipired before they are transmitted for the first time
> > > >(i.e due to congestion)?
> > > 
> > > It's after a TSN has been assigned. So it would depend on
> > > when the implementation assigned TSNs. But you are correct
> > > that theoretically a service *could* assign a TSN and then
> > > drop it before sending it even once.
> > > 
> > > Caitlin Bestler
> > > http://asomi.com/CaitlinBestler/
> > 
> > -- 
> > Brian F. G. Bidulock    ¦ The reasonable man adapts himself to the ¦
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-- 
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Date: Wed, 30 Apr 2003 10:45:19 +0200
To: Phil Karn <karn@qualcomm.com>
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: [Tsvwg] Re: Rethinking Karn's algorithm
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At 22:13 28.04.2003, Phil Karn wrote:
>Reiner Ludwig wrote:
>>A different question that was discussed before on the list is the 
>>exponential backoff; up to ~60 seconds. Don't you think that that is a 
>>bit too conservative?
>
>Well, I have seen some router queues that get amazingly long...

Are you saying that because we *might* have a *few* broken boxes in the 
Internet all TCPs must "suffer" and do exponential RTO backoff up to 60 
seconds?

I hate to say it, but with claims like yours I can probably kill any 
attempt to modify or add new features to an e2e protocol. To me this is 
only creating "worst-case paranoia", and that's not constructive!

The discussions around packet reordering is going in a similar direction. 
Sure, there could be backbone routers connected by multiple links for 
redundancy, and the load balancing software might not care to keep packets 
of the same flow on the same link. And voila, we have huge reordering 
problems in the Internet; and for TCP.

Now, we could have Best-Current-Practice RFCs that recommend routers: 
"don't queue more than 10 seconds worth of data", "don't throw packets of 
the same flow (or dest. IP addr.) onto different links", etc. in case they 
don't already do that, anyway (which I personally believe is mostly the 
case); then trust that this is the case in most places of the Internet, and 
design our e2e protocols towards that.
Or, we continue to believe that the Internet is a *possibly* crazy 
black-box, and design our e2e protocols with maximum conservativeness.

///Reiner 

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Subject: [Tsvwg] RFC 3522 on The Eifel Detection Algorithm for TCP
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3522

        Title:      The Eifel Detection Algorithm for TCP
        Author(s):  R. Ludwig, M. Meyer
        Status:     Experimental
        Date:       April 2003
        Mailbox:    Reiner.Ludwig@eed.ericsson.se,
                    Michael.Meyer@eed.ericsson.se
        Pages:      14
        Characters: 33729
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-tsvwg-tcp-eifel-alg-07.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3522.txt


The Eifel detection algorithm allows a TCP sender to detect a
posteriori whether it has entered loss recovery unnecessarily.  It
requires that the TCP Timestamps option defined in RFC 1323 be enabled
for a connection.  The Eifel detection algorithm makes use of the fact
that the TCP Timestamps option eliminates the retransmission
ambiguity in TCP.  Based on the timestamp of the first acceptable ACK
that arrives during loss recovery, it decides whether loss recovery
was entered unnecessarily.  The Eifel detection algorithm provides a
basis for future TCP enhancements.  This includes response algorithms
to back out of loss recovery by restoring a TCP sender's congestion
control state.

This document is a product of the Transport Area Working Group Working
Group of the IETF.

This memo defines an Experimental Protocol for the Internet community.
It does not specify an Internet standard of any kind.  Discussion and
suggestions for improvement are requested.  Distribution of this memo
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Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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From mailnull@www1.ietf.org  Wed Apr 30 19:18:48 2003
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Date: Wed, 30 Apr 2003 18:20:29 -0500
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] Re: Rethinking Karn's algorithm
To: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
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On 4/30/03, Reiner Ludwig wrote:

>At 22:13 28.04.2003, Phil Karn wrote:
>>Reiner Ludwig wrote:
>>>A different question that was discussed before on the
>>>list is the exponential backoff; up to ~60 seconds.
>>>Don't you think that that is a bit too conservative?
>>
>>Well, I have seen some router queues that get amazingly
long...
>
>Are you saying that because we *might* have a *few* broken
>boxes in the Internet all TCPs must "suffer" and do
>exponential RTO backoff up to 60 seconds?
>
>I hate to say it, but with claims like yours I can
>probably kill any attempt to modify or add new features to
>an e2e protocol. To me this is only creating "worst-case
>paranoia", and that's not constructive!
>
>The discussions around packet reordering is going in a
>similar direction. Sure, there could be backbone routers
>connected by multiple links for redundancy, and the load
>balancing software might not care to keep packets of the
>same flow on the same link. And voila, we have huge
>reordering problems in the Internet; and for TCP.
>
>Now, we could have Best-Current-Practice RFCs that
>recommend routers: "don't queue more than 10 seconds worth
>of data", "don't throw packets of the same flow (or dest.
>IP addr.) onto different links", etc. in case they don't
>already do that, anyway (which I personally believe is
>mostly the case); then trust that this is the case in most
>places of the Internet, and design our e2e protocols
>towards that. Or, we continue to believe that the Internet
>is a *possibly* crazy black-box, and design our e2e
>protocols with maximum conservativeness.
>

To agree with you even more strongly, consider that for
many applications a human is part of the "backoff
algorithm".

In onther words, how long of a delay will be *tolerated*
before the human consumer initiates corrective action,
such as a new connection, on their own.

Once predictable human responses are factored in, I doubt
that a 60 second backoff will ever truly result in improved
service.

Caitlin Bestler
http://asomi.com/CaitlinBestler/
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To: Caitlin Bestler <cait@asomi.com>
CC: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
Subject: Re: [Tsvwg] Re: Rethinking Karn's algorithm
References: <r01050300-1025-557AC08A7B6211D7A67A003065D48EE0@[192.168.0.2]>
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Caitlin Bestler wrote:
...
> To agree with you even more strongly, consider that for
> many applications a human is part of the "backoff
> algorithm".
> 
> In onther words, how long of a delay will be *tolerated*
> before the human consumer initiates corrective action,
> such as a new connection, on their own.
> 
> Once predictable human responses are factored in, I doubt
> that a 60 second backoff will ever truly result in improved
> service.

Aren't lining up and waiting to be served part of the fun of being
human? :-)

-Qiaobing
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From: Randall Stewart <randall@stewart.chicago.il.us>
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To: Qiaobing Xie <Qiaobing.Xie@motorola.com>
CC: Caitlin Bestler <cait@asomi.com>,
        Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>, tsvwg@ietf.org
Subject: Re: [Tsvwg] Re: Rethinking Karn's algorithm
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Qiaobing Xie wrote:
> Caitlin Bestler wrote:
> ...
> 
>>To agree with you even more strongly, consider that for
>>many applications a human is part of the "backoff
>>algorithm".
>>
>>In onther words, how long of a delay will be *tolerated*
>>before the human consumer initiates corrective action,
>>such as a new connection, on their own.
>>
>>Once predictable human responses are factored in, I doubt
>>that a 60 second backoff will ever truly result in improved
>>service.
> 
> 
> Aren't lining up and waiting to be served part of the fun of being
> human? :-)
> 
> -Qiaobing

Maybe..

But when you give the human another choice... (a different line
to go into) they will run to it... screwing up everything as
murphy puts it..

You can think of a different line as the little stop/reload button
on your web browser... gets a new connection and in cases of
congestion.. guess what.. it will speed things up at times...

So unless we take the "get a new connection aka switch lines" buttons
off of all applications I can't help but agree with Caitlin..

R

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

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Date: Wed, 30 Apr 2003 19:30:44 -0700 (PDT)
From: Spencer Dawkins <spencer_dawkins@yahoo.com>
Subject: Re: [Tsvwg] Re: Rethinking Karn's algorithm
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So I'm thinking that 

- the 60 second max RTO is to save the Internet from connections
between computers that don't have humans in the loop, and 

- the stop/reload button is to save humans in the loop from the
60 second max RTO?

:-}

Spencer

--- Randall Stewart <randall@stewart.chicago.il.us> wrote:
> You can think of a different line as the little stop/reload
> button
> on your web browser... gets a new connection and in cases of
> congestion.. guess what.. it will speed things up at times...
> 
> So unless we take the "get a new connection aka switch lines"
> buttons
> off of all applications I can't help but agree with Caitlin..
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