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Subject: Re: [tcpm] TCP consolidation doc 
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Hi all,

> I find myself really wondering what a longer document would buy us.
> It'd be damn hard to write and agree on, I bet.  And, it'd be cumbersome
> to change -- and, I think one lesson we have learned is that TCP is
> always evolving.  So, especially if one puts experimental things into
> the document I think it'd be way hard to make that work.  Think about
> this... say one wanted to include early retransmit and we have this
> mega-doc.  Now we have to open up the document to put in this one
> change.  But, while the document is open people want to throw other
> things in.  Can they?  Do we now have charter lines that say "open
> section X of The TCP Thome only"?

I agree. I was thinking along the same lines myself, and was wondering
whether a document like this could be written up with mass consent
*before* another enhancement pops up and becomes important.

IMHO, the roadmap should be able to provide what, with mass consent, the
"latest and greatest" TCP includes. It seems to me that a practical goal
is a good consolidation doc that would include pointers and have
discussion about interoperations between the different extensions (Sourabh
is looking into something similar at this time) and maybe implementation
notes about possible pitfalls during consolidation (from implementers'
experiences).

Granted that for an implementer, it is a headache to get n documents and
piece together the n extensions/enhancements, and leaves room for error in
the implementation.

regards,
jana

---------------------------------------------------------------
         Visit www.narmada.org, www.indiatogether.org
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Subject: Re: [tcpm] TCP consolidation doc 
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allman wrote:

> I find myself really wondering what a longer document would buy us.
> It'd be damn hard to write and agree on, I bet.

Note the history of aborted/unpublished TCP documents. RFC1323bis, the
whole SACK thing. A long document is doomed before you start. Adding
new/experimental stuff dooms you too.

What you really want is a very short RFC relying heavily on the
normative mechanism, commenting on the current standards-track RFCs as
normative references, saying what is considered important to
implement, what is worthwhile implementing, possibly known
implementation gotchas, etc. This RFC would go BCP, and be updated by
subsequent BCP docs replacing it. You'd expct a yearly or two-yearly
commentary on the state of the art of TCP, really.

Given the existing IETF document focus on TCP and the narrow focus of
this doc on that, it's the one instance where normative references in
a BCP doc actually make sense.

L.

I can't believe I just said that.

<http://www.ee.surrey.ac.uk/Personal/L.Wood/><L.Wood@eim.surrey.ac.uk>

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From: john.loughney@nokia.com
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Subject: RE: [tcpm] TCP consolidation doc 
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Hi Lloyd & others,

> > I find myself really wondering what a longer document would buy us.
> > It'd be damn hard to write and agree on, I bet.
>=20
> Note the history of aborted/unpublished TCP documents. RFC1323bis, the
> whole SACK thing. A long document is doomed before you start. Adding
> new/experimental stuff dooms you too.
>=20
> What you really want is a very short RFC relying heavily on the
> normative mechanism, commenting on the current standards-track RFCs as
> normative references, saying what is considered important to
> implement, what is worthwhile implementing, possibly known
> implementation gotchas, etc. This RFC would go BCP, and be updated by
> subsequent BCP docs replacing it. You'd expct a yearly or two-yearly
> commentary on the state of the art of TCP, really.
>=20
> Given the existing IETF document focus on TCP and the narrow focus of
> this doc on that, it's the one instance where normative references in
> a BCP doc actually make sense.

Because I'm a lazy coder, I always like to look for prior work before
starting something.  On this topic, I'd like to note a draft that I have
been editor on, and wonder if a similar approach to TCP would be one
possible way to tackle this topic. See:

http://www.ietf.org/internet-drafts/draft-ietf-ipv6-node-requirements-08.=
txt

br,
John

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From: Sunay Tripathi <Sunay.Tripathi@eng.sun.com>
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Hi,

AT the end of the TSVWG session on Monday, I was trying to see if people
are interested in discussing TOE (TCP offload Engines) from the
prespective of new technology and issues arising out of it. I am
not sure if there are any protocol level issues yet that need to be
discussed under the scope of IETF (tcpm) charter. I just want to
scope out the interest in this area from implementation and integration
in the host OS point of view.

After the meeting, I already met 5-6 people who are interested in
talking about it so we will try and get together sometime wed/thur
after the plenary. If there are more people interested, Ted can
get us a room or figure out a more structured discussion either this
or next IETF meeting.

Please drop me a note with you time preference if you are interested 
and I'll see what we can rig up. A brief background and possible list 
of things to discuss is listed below.

TOE BAckground
--------------

In last couple of years, the number of vendors selling TOE based hardware
has increased significantly and we at SUN have personally talked to 
about 2 dozen such vendors. The claim is that TOE will be necessary for
10Gb NICs and for emerging technologies like iSCSI, RDMA, etc. These
vendors already have first generation h/w available for Windows and Linux
while some of them have moved on to 2nd generation. On Linux, these 
vendors support TOE by overwriting the S/W stack with their own hooks.
With pretty impressive performance and reduction in CPU utilization claims,
even the end users have started asking (atleast SUN) about TOE support.

Some things to discuss
----------------------

1) Are you working with TOEs in any way?
2) Have you faced any issues with TOE. Someof them are:
        - failover. When TOE runs out of resources or doesn't understand
          some options/features.
        - Old (and possibly buggy implementations) burned in ASIC which
          will be hard to upgrade as new modifications are made to TCP
          (even if they are 2-3 every year).
        - End to End correctness issues like TOE acking a packet but dying
          before the data is transfered to host memory. Or accepting the
          data to be sent out but dying before the packet hits the wire.
3) Any other issues?

Cheers,
Sunay

-- 
Sunay Tripathi
Senior Staff Engineer,
Solaris Kernel Networking,
Sun MicroSystems Inc.

email: sunay@eng.sun.com                 Phone: 650-786-6007 (W)




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From exim@www1.ietf.org  Tue Mar  2 19:39:25 2004
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Well, the interesting thing is we went down this path in the early 1980s
and found that TOEs didn't work.

If I try to distill all that we learned in the 1980s into one question, I
come out with:

    In the 1980s we discovered that communicating with a TOE over a bus
    about a TCP connection was as expensive or more expensive than
    simply handling the TCP connection in the main processor.  What about
    today's TOE designs makes them different?

Craig

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From exim@www1.ietf.org  Thu Mar  4 00:01:34 2004
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From: Sunay Tripathi <Sunay.Tripathi@eng.sun.com>
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Craig,

> Well, the interesting thing is we went down this path in the early 1980s
> and found that TOEs didn't work.
> 
> If I try to distill all that we learned in the 1980s into one question, I
> come out with:
> 
>     In the 1980s we discovered that communicating with a TOE over a bus
>     about a TCP connection was as expensive or more expensive than
>     simply handling the TCP connection in the main processor.  What about
>     today's TOE designs makes them different?

I don't have too much history (1980s predates my career) but there were
some attempts in mid 90s to do the same and compared to that, the
following reasons seem to be the driving force:

1) Extra processor(s) buried in the TOE for networking processing which is
   hidden from the kernel and leaves the host CPU to do more application
   related work. Saves the cost of licences for application which take
   number of CPU into account (oracle is one such application cited).
2) On low end (1-2 CPU) x86 based machines, cost of adding a processor
   is much higher than adding a TOE (I personally haven't verified this).
3) For the up and coming 10Gb NICs, TOE will help saturate the link. Some
   vendors assert that TOE will be required to support 10Gb NICs.
4) Performance reasons. Just the LSO aspect of TOE (sending large chunks of
   data and letting the TOE split it up in mss size pieces) and ack
   coalescing gives a pretty good boost (our own prototypes indicates that
   this is true). The gains are by optimizing data movement and not by
   offloading protocol processing.
5) TOE is necessary for RDMA, iSCSI etc. for layering reasons. I am not
   involved with RDMA so someone who is an expert can probably comment on
   this part.
6) TOE based NIC are already making pretty good headway in embedded space.
   The technology is already maturing so why not use it in broader market.

Note that the above claims are in no particular order of importance and made
my TOE vendors in general. Of these, I personally do agree with 1 and 4 but
that iteself doesn't mean that TOE will make it in general purpose networking.

It would be interesting to see if you and others in the list agree or
disagree with these claims.

Thanks,
Sunay

-- 

Sunay Tripathi
Senior Staff Engineer,
Solaris Kernel Networking,
Sun MicroSystems Inc.

email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)




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From: Sunay Tripathi <Sunay.Tripathi@eng.sun.com>
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In-Reply-To: <000001c401b2$094af3d0$0300a8c0@S2IOtech.com> from Leonid Grossman
 at "Mar 3, 2004 10:29:22 pm"
To: Leonid Grossman <leonid.grossman@s2io.com>
Date: Wed, 3 Mar 2004 23:10:06 -0800 (PST)
CC: "'Sunay Tripathi'" <Sunay.Tripathi@eng.sun.com>, craig@bbn.com,
        tcpm@ietf.org, tsvwg@ietf.org, end2end-interest@postel.org
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<SNIP>
> > 3) For the up and coming 10Gb NICs, TOE will help saturate 
> > the link. Some
> >    vendors assert that TOE will be required to support 10Gb NICs.
> 
> 
> TOE will bring CPU utilization down of course, but 10GbE NIC will
> saturate the link with or without a TOE - much like GbE NICs do it now.

I think the TOE vendors meant that single CPU won't be able to saturate
the 10Gb NIC on a more meaningful workloads than just a simple throughput
tests. Even a 2 CPU machine which wants to do some work also might also
get constrained.

> > 4) Performance reasons. Just the LSO aspect of TOE (sending 
> > large chunks of
> >    data and letting the TOE split it up in mss size pieces) and ack
> >    coalescing gives a pretty good boost (our own prototypes 
> > indicates that
> >    this is true). 
> 
> Just curious - in your testing, how much CPU is freed up by the ack
> coalescing?

We actually always try to run the CPUs to saturation (keep increasing
the load) and we can get a double digit percentage boost on a web
like workload (multiple simultaneous connections). I didn't verify
exactly where the gains were coming from but my feeling was that
more efficient data movement and better locality was contributing to
bulk of the gains.

> >The gains are by optimizing data movement and not by
> >    offloading protocol processing.
> 
> LSO is a 'stateless' offload (much like checksum offload) so one doesn't
> need a TOE to support it; 

It depends. If the kernel does a write based on open window, then yes
you are right. But if you have full TOE and you pass large chunks of
data to TOE (does wonder when sendfile/sendfilev is in play) and let
TOE figure out when to send it, the gains increase. In the later case,
TOE needs teh TCP state.

> most modern NICs (including our 10GbE Adapter) do LSO. Also, for normal
> frames LSO has quite dramatic effect at 10GbE rates, but for Jumbo
> frames performance gain diminishes to ~7%.

Do you have any insight on why you would see a degradation with Jumbo
frames? As long as you are stateless, the jumbograms should show
no degradation. 

Thanks,
Sunay


> 
> 
> > 5) TOE is necessary for RDMA, iSCSI etc. for layering 
> > reasons. I am not
> >    involved with RDMA so someone who is an expert can 
> > probably comment on
> >    this part.
> > 6) TOE based NIC are already making pretty good headway in 
> > embedded space.
> >    The technology is already maturing so why not use it in 
> > broader market.
> > 
> > Note that the above claims are in no particular order of 
> > importance and made my TOE vendors in general. Of these, I 
> > personally do agree with 1 and 4 but that iteself doesn't 
> > mean that TOE will make it in general purpose networking.
> > 
> > It would be interesting to see if you and others in the list 
> > agree or disagree with these claims.
> > 
> > Thanks,
> > Sunay
> > 
> > -- 
> > 
> > Sunay Tripathi
> > Senior Staff Engineer,
> > Solaris Kernel Networking,
> > Sun MicroSystems Inc.
> > 
> > email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)
> > 
> > 
> > 
> > 
> 
> 
> Regards, 
> Leonid Grossman
> www.s2io.com
> 
> 
> > _______________________________________________
> > tsvwg mailing list
> > tsvwg@ietf.org
> > https://www1.ietf.org/mailman/listinfo/tsvwg
> > 
> 


-- 
Sunay Tripathi
Solaris Kernel Networking,
Sun MicroSystems Inc.

email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)




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Leonid,


<SNIP>

> > most modern NICs (including our 10GbE Adapter) do LSO. Also, for normal
> > frames LSO has quite dramatic effect at 10GbE rates, but for Jumbo
> > frames performance gain diminishes to ~7%.

> Do you have any insight on why you would see a degradation with Jumbo
> frames? As long as you are stateless, the jumbograms should show
> no degradation. 

I apologize. Jeremy Harris pointed out that I misread '~' for '-'. So
what you say makes sense now :)

Thanks,
Sunay

-- 
Sunay Tripathi
Senior Staff Engineer,
Solaris Kernel Networking,
Sun MicroSystems Inc.

email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)



----- End of forwarded message (env-from sunay) -----

-- 
Sunay Tripathi
Solaris Kernel Networking,
Sun MicroSystems Inc.

email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)




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Sunay Tripathi <Sunay.Tripathi@eng.sun.com> writes:

> 1) Extra processor(s) buried in the TOE for networking processing which is
>    hidden from the kernel and leaves the host CPU to do more application
>    related work. Saves the cost of licences for application which take
>    number of CPU into account (oracle is one such application cited).

One would hope TCP design would not be guided by Oracle licensing quirks.

> 2) On low end (1-2 CPU) x86 based machines, cost of adding a processor
>    is much higher than adding a TOE (I personally haven't verified this).

It's not obvious why this should necessarily be the case, given that
it is likely that there will continue to be quite a bit more
general-purpose CPUs made than TCP offload engines.

> 3) For the up and coming 10Gb NICs, TOE will help saturate the link. Some
>    vendors assert that TOE will be required to support 10Gb NICs.

Right now, one can do almost 8Gb/s with a single TCP stream over 10GE
(let's say 5 or 6Gb/s with more common hardware).  The limiting factor
is currently host bus speed.  There's nothing (except for compression
on the bus side, but it should be clear we're not going there) that an
offload engine can do about host bus speed.  By the time host busses
faster than 10Gb/s are commonly available, pretty routine x86 box
should handle 10GE saturation.

> 4) Performance reasons. Just the LSO aspect of TOE (sending large chunks of
>    data and letting the TOE split it up in mss size pieces) and ack
>    coalescing gives a pretty good boost (our own prototypes indicates that
>    this is true). The gains are by optimizing data movement and not by
>    offloading protocol processing.

Here I would agree with the unnamed TOE vendor whom you're
paraphrasing (and with Jerry Chu's comments in this thread).  Ethernet
frame size (1500 bytes or even 9kB) is very small; we'd be in a better
world if we could specify the MTU in units of time (the number of CRC
bits would have to scale up as well, of course).  TCP offload engines
could be a kludge that would help to work around this deficiency in
Ethernet.

Note that since even at 10Gb/s one CPU is enough to saturate the link
with 9kB packets, the need for this work-around is not at all
pressing.  Given the potential harmful effects (undetected errors on
the host bus, difficulty in patching stale or buggy TCP code, etc.),
one would probably be better served by concentrating on the deployment
of jumbo frames.  The investment to support jumbo frames has largely
already been made, so why not extract all we can from it first?

-- 
Stanislav Shalunov		http://www.internet2.edu/~shalunov/

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CC: Sunay Tripathi <Sunay.Tripathi@eng.sun.com>, tcpm@ietf.org, tsvwg@ietf.org,
        end2end-interest@postel.org
Subject: Re: [tcpm] Re: Are you interested in TOEs and related issues
References: <200403040455.i244twSG699301@jurassic.eng.sun.com> <86y8qf6bvy.fsf@abel.internet2.edu>
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stanislav shalunov wrote:
> Sunay Tripathi <Sunay.Tripathi@eng.sun.com> writes:
>>2) On low end (1-2 CPU) x86 based machines, cost of adding a processor
>>   is much higher than adding a TOE (I personally haven't verified this).
> 
> 
> It's not obvious why this should necessarily be the case, given that
> it is likely that there will continue to be quite a bit more
> general-purpose CPUs made than TCP offload engines.

It's because cache-coherency is expensive.  x86 boxes have traditionally
avoided playing far into this space.  Looked at from that angle,
TOE is towards the far end of the ncNUMA scale, and coherency
is managed by software.

- Jeremy

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From: "Ehsan Hamadani" <e.hamadani@city.ac.uk>
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Dear all,
 
In TCP-F (TCP feedback based scheme), it has been proposed that when a
specific link becomes invalid, the failure point (FP) will sends Route
Failure Notification (RFN) message to sender and then the sender goes to
snooze mode and waits for Route Re-establishment Notification (RRN)
message. My question here is that, when multiple link failures happen,
does the sender accept the RRN message only from the FP node that has
sent RFN message or it will accept that from any intermediate node than
have learned new route to destination?
 
I would appreciate if someone can help me with that.
 
Regards,
 
Ehsan     
 
 
--------------------------------------------------------------------
Ehsan Hamadani, PhD Student
School of Engineering & Mathematical Science
CITY University
London, EC1V 0HB
United Kingdom
 
Phone:+44(0)2070403886
 
 
 
 

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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Dear all,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>In TCP-F (TCP feedback based scheme), it has been =
proposed
that when a specific link becomes invalid, the failure point (FP) will =
sends
Route Failure Notification (RFN) message to sender and then the sender =
goes to
snooze mode and waits for Route Re-establishment Notification (RRN) =
message. My
question here is that, when multiple link failures happen, does the =
sender
accept the RRN message only from the FP node that has sent RFN message =
or it
will accept that from any intermediate node than have learned new route =
to
destination?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I would appreciate if someone can help me with =
that.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Regards,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Ehsan<span =
style=3D'mso-spacerun:yes'>&nbsp;&nbsp;&nbsp;&nbsp; =
</span><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;mso-no-proof:yes'>-------------------------------------=
-------------------------------</span></font><span
style=3D'mso-no-proof:yes'><o:p></o:p></span></p>

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></font></st2:GivenName><font
 size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:
 yes'> </span></font><st2:Sn><font size=3D2 face=3DArial><span =
style=3D'font-size:
  =
10.0pt;font-family:Arial;mso-no-proof:yes'>Hamadani</span></font></st2:Sn=
><font
 size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:
 yes'>, </span></font><st2:nameSuffix><font size=3D2 face=3DArial><span
  =
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'>PhD</span><=
/font></st2:nameSuffix></st1:PersonName><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:
yes'> Student</span></font><span =
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face=3DArial><span
  =
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n></font></st1:PlaceType><font
 size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:
 yes'> of </span></font><st1:PlaceName><font size=3D2 face=3DArial><span
  =
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'>Engineering=
</span></font></st1:PlaceName></st1:place><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:
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</span></font><st1:PlaceType><font
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span
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<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt;mso-no-proof:yes'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt;mso-no-proof:yes'>&nbsp;</span><o:p></o:p></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

</body>

</html>

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To: Ehsan Hamadani <e.hamadani@city.ac.uk>
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On Sun, Mar 07, 2004 at 09:35:28PM -0000, Ehsan Hamadani wrote:
> Dear all,
>  
> In TCP-F (TCP feedback based scheme), it has been proposed that when a
> specific link becomes invalid, the failure point (FP) will sends Route
> Failure Notification (RFN) message to sender and then the sender goes to
> snooze mode and waits for Route Re-establishment Notification (RRN)
> message. My question here is that, does the sender accept the RRN
> message only from the FP node that has sent RFN message or it will
> accept that from any intermediate node than have learned new route to
> destination?

I'll be needing some context for that.  Point me toward a TCP-F document
or paper.

-- 
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.asc
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The draft minutes from the tcpm meeting at the 59th IETF in Korea are
attached.  Please send comments or corrections to either the list or
Mark and myself.

The minutes are also available from http://www.isi.edu/tcpm/ along with
copies of the presentations.

-- 
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.asc
Unexpected attachment on this mail? See http://www.isi.edu/~faber/FAQ.html#SIG 

--4BlIp4fARb6QCoOq
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TCPM Working Group

Draft Minutes as of 8 Mar 2004

The meeting was chaired by Ted Faber.
These minutes were edited by Ted Faber from notes by Matt Zekauskas.


=====================================================================
1. Overview of the charter and milestones
   --Ted Faber

Ted addressed some slides going over the charter and group goals.
(See slides.)

Why form a new group?
  * This group was created to focus efforts on TCP tweaks.  Take
    revisions and documents that have been "floating around", and
    push them forward.

  * It will also serve as a forum to attract new TCP work where TCP
    expertise can be focused to evaluate the work carefully and move forward.

--
Milestone discussion:
--
FRTO draft (draft-ietf-tsvwg-tcp-frto-01.txt) 
	Discussed and moved fwd, see below.

Roadmap draft (draft-duke-tcp-roadmap-00.txt)
  adopt directly into WG

Ted's impressions:
  draft good in coverage, but needs annotations

  "stuff not used much" the draft calls these "Deprecated TCP
  Extensions".  Ted thinks the discussions of these should include more
  information about why the extensions were deprecated and the lessons
  one can take from them.  Of course editors and group will have the
  final word.

  Ted mentioned that there's a discussion on e2e about a TCP consolidation
  document.  Ted is not clear on the difference between what they're
  asking for and a roadmap, and Ted would rather see that energy in the
  roadmap.

  We're actively looking for editors and contributors.

--
DCR & reordering. (blanton-tcp-reordering-00.txt
	draft-bhandarkar-tcp-dcr-00.txt)

  Need to get with authors and gauge interest.  Ted suspects authors will
  need to drive.

--
1323 revamp: 
  Dave Borman presented, see below.

  Goal is depending on how big a revamp happens either push to Draft
  Standard, or "revalidate" at Proposed Standard.
--
End of Milestone discussion
--
Presentation of other documents that might be of interest to TCPM:

  From the slide:
  
  SACK
	2018 update?  Merge with 2883 (DSACK)?

  Congestion Control 
	Update 2581  (back to PS?)
		merge 3390 (initial cwnd)
		merge 3042 (limited transmit)
		merge 2582 (Reno bugfix/NewReno)
		merge 3465 (App. Byte Count security)
	2861 to PS (cwnd validation)
	2988 to DS (RTO timer comp)
	3517 to PS? into 2018 update? (DSACK loss rec) 
	[NB: after the meeting Kevin Fall pointed out to me that Ted had
	 mischaracterized RFC 3517, which is already a Proposed
	 Standard, and probably doesn't need to move forward any time
	 soon.]

  Again, these will be come work items based on group interest, though
  no one leapt up to take one on.

--
Closed with a "Drivers Wanted" plea.

Allison Mankin encouraged people to volunteer to take things on.  [Thanks]

Sally Floyd: "I wouldn't go so far as to say that I was interested,
 but I agree that it is necessary to do."  
 [Sally's Restatement: I am *willing* to do my share of the work, just not
 necessarily *excited* about having to do it.]

=====================================================================
2. FRTO presentation 
   --Pasi Sarolahti.
   draft: draft-ietf-tsvwg-tcp-frto-01.txt

The draft has been around for a while.  Started in June 2002 as
individual submission, to TSV last october.

detecting spurious RTOs.
  (where a lost or delayed ack causes an RTO time out)

  Send a little new data during retransmit and looks for immediate acks
  of that new data.

  Can be combined with SACK
--
List of editorial changes from last version
--
They have been working for a while... want to submit for pub as Experimental.

Ted called for comments:

Sally Floyd: I have not been paying attention to the discussion on the mailing
list.  Has the discussion converged to consensus?  Are there outstanding
technical points of contention? [Not verbatim]

Pasi: There has been no real discussion since June.  Comments have mostly
editorial, not technical.  There have been some individual nits. nothing
related to algorithm.  I believe that the draft is stable.

Ted says what we're going to do is make another pass for NITs, on tcpm, and
then on ML go to WGLC.  Ted doesn't know whether to ask for a hum or
some other consensus indication in a mixed meeting.

Allison Mankin:  I talked to Jon in the back, who's the responsible AD
nits review is just nits, no humming to do.

Action is to do a nits pass and got to WGLC on the list.

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

3. RFC 1323 update 
   --David Borman

  Large windows & timestamps RFC that has been proposed for way too long.
  at last IETF, gave presentation... this is what's happened since.

Nothing has really changed... no new draft, no discussion.
But we are now part of tcpm wg.  This is a good thing.

Outstanding issues...
--
RTO adjustments
  When using timestamps and calculating RTTs frequently, need to adjust
  weighting factor on estimates.  There seems to be general consensus on
  this issue.
--
Enabling timestamps/PAWS midstream
     Do most current TCP implementations properly handle unknown options
     mid-stream?

     Originally, implementations crash when unknown options appear in
     the middle of a  tcp connection.

     We would have to think about reliability in that context - i.e.
     would allowing timestamp/PAWS renegotiation cause more crashed than
     improved performance?
--
PAWS & DF bit
  Without DF bit set, PAWS would protect 1st fragment not the rest. Only
  the 16-bit IP ID would protect them.

  The pathological case is that the 1st frag arrives, another packet
  fills in hole. then perpetually all assembled incorrectly.
  'TCP cksum should notice'.

  Matt Mathis points out that this is a general ugly problem with IP
  reassembly and points out similar problems with UDP transport.

  Vern Paxson mentions a related problem: every 65000 of those
  collisions will be missed by the TCP checksum.  That's even worse!

  Matt Mathis points out that the real pathological condition is the one
  that Dave mentioned where each contains the data Vern alluded to -
  every packet is misassembled in such a way that the checksum misses
  the corruption.

  He goes on to say that he's meaning to write RFC "Fragmentation
  Considered Really Really Really Harmful" because pathological case
  horrible.

  Someone says that because of probs with MTU discovery, people ignoring
  DF bits -- really bad.

  Ted says that this is all good discussion and I'd like to see it get into
  the new document.

--
RTTM estimate from dup ack.
  discussed on ML (different mailing lists).
  If a connection is idle for long period of time, and 1st ack is lost, 
  when get a dupack, it could have good information for RTT, but that
  means changing what's in document.  Concern: even if get everything right and
  correctly id dupack (potential for random window update coming in) how
  interoperate with existing implementations?  (How do we know that the
  other side giving is good information in a dupack timestamp.  Even though
  the idea is good, you can't do anything with it.

  "not a real big problem out there" -- not clear who said this

  Dave points out that TCP doesn't define what dupack is.  all know one
  when see one, but...  we need a precise definition.

--
non-technical issues:
  Connections with other work:
    HSTCP and others depend on window scaling
    Eifel wants ts option
  Should the revision disucss SACK?
    Originally SACK was part of 1323.  Split originally because SACK
    needed baking)   He thinks SACK is further ahead for stds progress.

  We need to revisit the MUST/SHOULD language.

  what about last 10 yrs of experience?  include clarifications...

draft-jacobson-tsvwg-1323bis-00.txt will expire RSN

This has been discussed on e2e mailing list, but David will attempt to
steer traffic to tcpm.

Ted ask Dave to please summarize to TCPM.

Dave and Ted talk about generating a numbered list of items, so that
we can check items off as we go.

Jon Peterson says the IESG is making issue trackers available to WG chairs.
  Check with Rob Austein... there is one on the psg.com site.

David is mainly looking for things to move along.  He has tried to bring up
the topic of a revised document a couple of times, but it hasn't advanced
in part because we do not having a precise way of saying when an issue
is closed.

Sunay from sun microsystems made an appeal for interest in outboard
protocol processing. [This has since surfaced on the ML and e2e.]

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Hi There,

Could someone please point me to any document about TOE.

Thanks,
Nader


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>
><SNIP>
...
>> >The gains are by optimizing data movement and not by
>> >    offloading protocol processing.
>> 
>> LSO is a 'stateless' offload (much like checksum offload) so one doesn't
>> need a TOE to support it; 
>
>It depends. If the kernel does a write based on open window, then yes
>you are right. But if you have full TOE and you pass large chunks of
>data to TOE (does wonder when sendfile/sendfilev is in play) and let
>TOE figure out when to send it, the gains increase. In the later case,
>TOE needs teh TCP state.
>
>> most modern NICs (including our 10GbE Adapter) do LSO. Also, for normal
>> frames LSO has quite dramatic effect at 10GbE rates, but for Jumbo
>> frames performance gain diminishes to ~7%.

One fundamental performance problem high speed Ethernet imposes on
the hosts is from the tiny size of the standard Ethernet frames.
It's almost like throwing ATM cells at the host 10 years ago but
without any hardware adaptation.

LSO and ack coalescing are the adpatation layer for Ethernet cells.
They can help to alleviate this problem but unfortunately they either
induce, or live on traffic bursts. (Just imagine how useful LSO will
be if the ack-every-other-pkt policy provides only 2-MSS worth of
usable window, and how much back-to-back bursts will be caused by
coaliscing many acks.)

Bursty traffic may be ok for LAN, but is considered evil for WAN.
TOE doesn't suffer this problem (since it is stateful), and may be
a necessary evil until jumbo frames become universal.

Jerry

>
>Do you have any insight on why you would see a degradation with Jumbo
>frames? As long as you are stateless, the jumbograms should show
>no degradation. 
>
>Thanks,
>Sunay
>
>
>> 
>> 
>> > 5) TOE is necessary for RDMA, iSCSI etc. for layering 
>> > reasons. I am not
>> >    involved with RDMA so someone who is an expert can 
>> > probably comment on
>> >    this part.
>> > 6) TOE based NIC are already making pretty good headway in 
>> > embedded space.
>> >    The technology is already maturing so why not use it in 
>> > broader market.
>> > 
>> > Note that the above claims are in no particular order of 
>> > importance and made my TOE vendors in general. Of these, I 
>> > personally do agree with 1 and 4 but that iteself doesn't 
>> > mean that TOE will make it in general purpose networking.
>> > 
>> > It would be interesting to see if you and others in the list 
>> > agree or disagree with these claims.
>> > 
>> > Thanks,
>> > Sunay
>> > 
>> > -- 
>> > 
>> > Sunay Tripathi
>> > Senior Staff Engineer,
>> > Solaris Kernel Networking,
>> > Sun MicroSystems Inc.
>> > 
>> > email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)
>> > 
>> > 
>> > 
>> > 
>> 
>> 
>> Regards, 
>> Leonid Grossman
>> www.s2io.com
>> 
>> 
>> > _______________________________________________
>> > tsvwg mailing list
>> > tsvwg@ietf.org
>> > https://www1.ietf.org/mailman/listinfo/tsvwg
>> > 
>> 
>
>
>-- 
>Sunay Tripathi
>Solaris Kernel Networking,
>Sun MicroSystems Inc.
>
>email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)
>
>
>


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In-Reply-To: Your message of "Mon, 08 Mar 2004 12:42:21 EST."
             <8052E2EA753D144EB906B7A7AA399714022A05B7@mailserv.hatteras.com> 
Date: Mon, 08 Mar 2004 16:50:43 -0500
From: Craig Partridge <craig@bbn.com>
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Subject: [tcpm] Re: [Tsvwg] Re: [e2e] Are you interested in TOEs and related issues
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In message <8052E2EA753D144EB906B7A7AA399714022A05B7@mailserv.hatteras.com>, "S
tephen Suryaputra" writes:

>I think Marko is referring to the situation when there is a congestion.
>Because of that the queue builds up and if the queue space is in term
>of the number of packets, then a bunch of small packets can potentially
>dominate the queue and left little room for large packets eventhough
>there is a space in the buffer to accomodate the large ones.

Well, I tried to nail this down by setting it up as a queueing system
but that's awfully hard.  Here's the cut:

1. Basically you have a general arrival process delivering pieces of work
    of variable size, at arrival rate A subject to the condition that
    if there's an event of size w1 at time t1, then the next event cannot
    arrive sooner than time t1+w1.  [I.e. A(t2,w2) - A(t1,w1) >= w1].

2. You then have a series of queues Q1-Qn attached to servers S1-Sn that
    serve events at a fixed rate R, where R1 <= A (i.e. we can handle
    the maximum rate).

3.  You then have a departure queue D1, and departure server DS, which has
    a service rate D, where D can be defined such that D(w) <= w (that is,
    the output is faster or equivalent speed to the input) or D(w) >= w
    (output is slower).


If D(w) <= w, then I think one can argue simply by inspection that the total
work unit buffering in the system is [max(w)/min(w)]-n.  That is you
simply need enough queueing to handle a queue that develops because a big
packet causes a bunch of little packets to queue behind it (because they
arrive while the big packet is going out).  See Comment (**) below.

I can't find an easy solution for D(w) >= w, because, in general, the queue
can grow without bound.

** If you believe this model, what it says is that if you go with packet
size buffers, and the ratio between the smallest possible packet and
largest possible packet is big, then you can find your queues growing quite
large, due not to classic congestion (where there is too much demand
fighting for one output link) but rather to serialization delays.

Craig

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From: "Leonid Grossman" <leonid.grossman@s2io.com>
To: "'Luigi Rizzo'" <rizzo@icir.org>,
        "'H.K. Jerry Chu'" <Jerry.Chu@eng.sun.com>
Cc: <tcpm@ietf.org>, <tsvwg@ietf.org>, <end2end-interest@postel.org>
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> -----Original Message-----
> From: Luigi Rizzo [mailto:rizzo@icir.org] 
> Sent: Thursday, March 04, 2004 7:12 AM
> To: H.K. Jerry Chu
> Cc: leonid.grossman@s2io.com; tcpm@ietf.org; tsvwg@ietf.org; 
> end2end-interest@postel.org
> Subject: Re: [Tsvwg] Re: [e2e] Are you interested in TOEs and 
> related issues
> 
> 
> On Thu, Mar 04, 2004 at 11:11:36PM -0800, H.K. Jerry Chu wrote: ...
> > Bursty traffic may be ok for LAN, but is considered evil 
> for WAN. TOE 
> > doesn't suffer this problem (since it is stateful), and may be a 
> > necessary evil until jumbo frames become universal.
> 
> actually the interesting observation that this discussion 
> raises is that the frame size/link MTU should be picked large 
> enough to make the end-to-end pps rate bounded (but not too 
> low) for well behaved flows.
> 
> In 1980 the ethernet MTU allowed some 800pps -- going much 
> higher makes little sense anyways, because even if the end 
> nodes can cope with higher rates, the control loops cannot 
> not react fast enough (you have to factor in propagation delays too).
> 
> On the negative side, however, is that that very large MTUs 
> require a lot of buffering to be preallocated on the receive 
> side, and possibly even at the routers (at least, those 
> without hw-assisted forwarding planes).
> 
> I don't have a good answer but going much higher than 
> 16Kbytes MTUs seems unlikely... and at 10Gig this is still 
> close to 100kpps.

I think we'll be stuck at 9Kbytes for a while because of the CRC
protection (or rather the weakening of such) for 16Kbytes frames, at
least for 'simple' errors.
Leonid


> 
> 	cheers
> 	luigi
> 


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>> Bursty traffic may be ok for LAN, but is considered evil for WAN.
>> TOE doesn't suffer this problem (since it is stateful), and may be
>> a necessary evil until jumbo frames become universal.
>
>actually the interesting observation that this discussion raises
>is that the frame size/link MTU should be picked large enough to make the
>end-to-end pps rate bounded (but not too low) for well behaved flows.
>
>In 1980 the ethernet MTU allowed some 800pps -- going much higher makes
>little sense anyways, because even if the end nodes can cope with
>higher rates, the control loops cannot not react fast enough
>(you have to factor in propagation delays too).
>
>On the negative side, however, is that that very large MTUs require
>a lot of buffering to be preallocated on the receive side, and
>possibly even at the routers (at least, those without hw-assisted
>forwarding planes).

It's mainly the TCP receive window, not the PMTU that dictates how much
buffering will be needed. Routers that do cut-through need not
buffer the whole pkt. More significant is the drop of the DRAM price
and increase of the DRAM size making memory a non-issue in many cases
today.

Over the past decade many components involved in providing high-speed
networking have scaled up an order of magnitude. This including link
bandwidth, CPU speed, I/O bus, memory size..., but not the Ethernet MTU
and certain TCP parameters (such as the every-other-pkt acking policy).
This is really hurting the throughput performance of the hosts. IMHO the
amount of burstiness by TCP over WAN should be allowed to scale up an
order of magnitude too. If stretch ACKs are fully adopted into TCP
algorithm (see rfc2525 for a number of issues with stretch acks), one
can use LSO on the transmit side, and per-flow pkt coalescing on the
receive side to provide effectively a simple, stateless AAL5 layer for
the Ethernet "cells" without requiring jumbo frames or complex TOE engine.

BTW, I asked a few transport folks in Minneapolis IETF about how "evil"
is traffic burst in today's enviroment, but did not get any concrete
answer. Perhaps this topic should be discussed in tsvwg or tcpm.

Jerry

Sr. Staff Engineer
Solaris Networking & Security Technology
Sun Microsystems, Inc.

>
>I don't have a good answer but going much higher than 16Kbytes MTUs
>seems unlikely... and at 10Gig this is still close to 100kpps.
>
>	cheers
>	luigi


_______________________________________________
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On Friday 05 March 2004 19:50, H.K. Jerry Chu wrote:
> >> Bursty traffic may be ok for LAN, but is considered evil for WAN.
> >> TOE doesn't suffer this problem (since it is stateful), and may be
> >> a necessary evil until jumbo frames become universal.
> >
> >actually the interesting observation that this discussion raises
> >is that the frame size/link MTU should be picked large enough to
> > make the end-to-end pps rate bounded (but not too low) for well
> > behaved flows.
> >
> >In 1980 the ethernet MTU allowed some 800pps -- going much higher
> > makes little sense anyways, because even if the end nodes can cope
> > with higher rates, the control loops cannot not react fast enough
> > (you have to factor in propagation delays too).
> >
> >On the negative side, however, is that that very large MTUs require
> >a lot of buffering to be preallocated on the receive side, and
> >possibly even at the routers (at least, those without hw-assisted
> >forwarding planes).
>
> It's mainly the TCP receive window, not the PMTU that dictates how
> much buffering will be needed. Routers that do cut-through need not
> buffer the whole pkt. More significant is the drop of the DRAM price
> and increase of the DRAM size making memory a non-issue in many cases
> today.


If the DRAM price alone were the only factor preventing vendors from 
implementing unlimited packet queues, we would already have seen tons 
of such devices on the market over the past 10 years or so. Excessive 
queuing delays on routers are bad, especially for high-speed traffic 
managed by TCP-like congestion control schemes. The queue lengths on 
routers have always been and will always present a tradeoff between a 
smallest usable window for handling traffic bursts, and the desire of 
keeping the queueing delays as low as possible on the other hand.


>
> Over the past decade many components involved in providing high-speed
> networking have scaled up an order of magnitude. This including link
> bandwidth, CPU speed, I/O bus, memory size..., but not the Ethernet
> MTU and certain TCP parameters (such as the every-other-pkt acking
> policy). This is really hurting the throughput performance of the
> hosts. IMHO the amount of burstiness by TCP over WAN should be
> allowed to scale up an order of magnitude too. If stretch ACKs are
> fully adopted into TCP algorithm (see rfc2525 for a number of issues
> with stretch acks), one can use LSO on the transmit side, and
> per-flow pkt coalescing on the receive side to provide effectively a
> simple, stateless AAL5 layer for the Ethernet "cells" without
> requiring jumbo frames or complex TOE engine.
>
> BTW, I asked a few transport folks in Minneapolis IETF about how
> "evil" is traffic burst in today's enviroment, but did not get any
> concrete answer. Perhaps this topic should be discussed in tsvwg or
> tcpm.


Because queues in todays routers have finite maximum lengths, and this 
model is unlikely to change in the forseeable future, excessive traffic 
bursts will be more likely subject to drop-tail policing than other 
kinds of more smoothly shaped traffic. More than that, the bursty 
traffic will not only have less chance of reaching its target with all 
fragments in place, but it will also most probably do much harm to 
other innocent and otherwise well-behaving flows as well.

Marko


>
> Jerry
>
> Sr. Staff Engineer
> Solaris Networking & Security Technology
> Sun Microsystems, Inc.
>
> >I don't have a good answer but going much higher than 16Kbytes MTUs
> >seems unlikely... and at 10Gig this is still close to 100kpps.
> >
> >	cheers
> >	luigi


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



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From: "Alexandre L. Grojsgold" <algold@rnp.br>
To: Marko Zec <zec@tel.fer.hr>
cc: "H.K. Jerry Chu" <Jerry.Chu@eng.sun.com>, tcpm@ietf.org, tsvwg@ietf.org,
        end2end-interest@postel.org
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>
> The problem is that in today's routers the buffer sizes are typically
> accounted in packets, not in bytes. So looking at your example, and
> supposing that the line card buffer is limited to 100 packets, a burst
> of small frames will instantly consume 13% of available buffer slots,
> while jumbos will only use 2%. This is of course no problem if the line
> card can transmit all those frames instantly, but what if it cannot?


So what?

Small packets get more buffer frames, but are also more rapidly sent.
The gueue grows fats, but shrinks fast too. It means that a burst of small
packets will not increase the packet loss, no mather the length of the
buffer (in packets) .

Ok, if the router gets a large amount of packets, representing more bytes
per second than the capacity of the output line, packets will be lost. But
it is a function of the amount of data conveyed by the packets, not of
their individual length.



>
>
> >
> > >> BTW, I asked a few transport folks in Minneapolis IETF about how
> > >> "evil" is traffic burst in today's enviroment, but did not get any
> > >> concrete answer. Perhaps this topic should be discussed in tsvwg
> > >> or tcpm.
> > >
> > >Because queues in todays routers have finite maximum lengths, and
> > > this model is unlikely to change in the forseeable future,
> > > excessive traffic bursts will be more likely subject to drop-tail
> > > policing than other kinds of more smoothly shaped traffic. More
> > > than that, the bursty traffic will not only have less chance of
> > > reaching its target with all fragments in place, but it will also
> > > most probably do much harm to


I think many people tend to thing about IP routers like old circuit
switched protocols switches, like X.25.


In circuit switched networks, flow control is generaly implemented in each
link, each pair of switches exchanging acks and naks, trying to keep data
at the right pace.


With IP, the flow control is end-to-end, performed by TCP. If a router
queues too many packets, it will break TCP stability. It is not an issue
of memory or memory price.


alg.


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Subject: Re: [Tsvwg] Re: [tcpm] Re: [e2e] Are you interested in TOEs and related 
 issues (Resend)
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> TOE has become a buzz-word that needs to be qualified.  Some folks
> include Checksum Offload and Large Segmentation Offload when talking
> about TOE.  When others talk about TOE, they refer to offloading the
> whole TCP/IP stack.

Humbly submitted:

Pinky TOE => Checksum Offload (CKO)
Middle TOE => CKO + LSO (aka TSO or large send)
Big TOE => Full TCP Offload

rick jones
-- 
Wisdom Teeth are impacted, people are affected by the effects of events.
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to post, OR email to raj in cup.hp.com  but NOT BOTH...

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On Mon, Mar 08, 2004 at 10:26:55AM -0800, Rick Jones wrote:
> 
> Pinky TOE => Checksum Offload (CKO)
> Middle TOE => CKO + LSO (aka TSO or large send)
> Big TOE => Full TCP Offload

You forgot these:

Stubbed TOE => disabling TOE so tcpdump/snoop works
Broken TOE  => bug in firmware, difficult to diagnose

-- 
Barney Wolff         http://www.databus.com/bwresume.pdf
I'm available by contract or FT, in the NYC metro area or via the 'Net.

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References: <200403052348.i25NmQdW692336@jurassic.eng.sun.com> <200403061121.37953.zec@tel.fer.hr> <Pine.WNT.4.58.0403071851080.712@manjar>
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Subject: [tcpm] Re: [Tsvwg] Re: [e2e] Are you interested in TOEs and related issues
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On Sunday 07 March 2004 23:06, Alexandre L. Grojsgold wrote:
> > The problem is that in today's routers the buffer sizes are
> > typically accounted in packets, not in bytes. So looking at your
> > example, and supposing that the line card buffer is limited to 100
> > packets, a burst of small frames will instantly consume 13% of
> > available buffer slots, while jumbos will only use 2%. This is of
> > course no problem if the line card can transmit all those frames
> > instantly, but what if it cannot?
>
> So what?
>
> Small packets get more buffer frames, but are also more rapidly sent.
> The gueue grows fats, but shrinks fast too. It means that a burst of
> small packets will not increase the packet loss, no mather the length
> of the buffer (in packets).


This theory is unfortunately wrong, because it ignores the fact that 
when congestions do occur, traffic flows consisting of small packets 
(regardless whether they are bursty or smooth) will use more buffer 
slots then the flows carrying the larger ones, leaving less remaining 
buffer slots available for queuing further packets / bursts.


> Ok, if the router gets a large amount of packets, representing more
> bytes per second than the capacity of the output line, packets will
> be lost. But it is a function of the amount of data conveyed by the
> packets, not of their individual length.


Can you define the function you are talking about more precisely? What 
does it represent (I assume packet loss ratio)? Which argument(s) does 
it take? If this function is as simple as it might sound from your 
statement, why don't you share it with us?


>
> > > >> BTW, I asked a few transport folks in Minneapolis IETF about
> > > >> how "evil" is traffic burst in today's enviroment, but did not
> > > >> get any concrete answer. Perhaps this topic should be
> > > >> discussed in tsvwg or tcpm.
> > > >
> > > >Because queues in todays routers have finite maximum lengths,
> > > > and this model is unlikely to change in the forseeable future,
> > > > excessive traffic bursts will be more likely subject to
> > > > drop-tail policing than other kinds of more smoothly shaped
> > > > traffic. More than that, the bursty traffic will not only have
> > > > less chance of reaching its target with all fragments in place,
> > > > but it will also most probably do much harm to
>
> I think many people tend to thing about IP routers like old circuit
> switched protocols switches, like X.25.
>
> In circuit switched networks, flow control is generaly implemented in
> each link, each pair of switches exchanging acks and naks, trying to
> keep data at the right pace.
>
> With IP, the flow control is end-to-end, performed by TCP. If a
> router queues too many packets, it will break TCP stability. It is
> not an issue of memory or memory price.


Agreed. And that's precisely the reason why bursty traffic is bad - 
since routers are generally tuned to queue only as few packets as 
feasible for normal TCP operation, excessive traffic bursts will have 
more chance of being tail-dropped at router queues.

Marko


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> Draft Minutes as of 8 Mar 2004

<snip>

>  Ted mentioned that there's a discussion on e2e about a TCP consolidation
>  document.  Ted is not clear on the difference between what they're
>  asking for and a roadmap, and Ted would rather see that energy in the
>  roadmap.

<snip>

>  Congestion Control
>	Update 2581  (back to PS?)
>		merge 3390 (initial cwnd)
>		merge 3042 (limited transmit)
>		merge 2582 (Reno bugfix/NewReno)
>		merge 3465 (App. Byte Count security)

These merges are basically what we had in mind with the consolidation
document.

~armando

0--                                              --0
| Armando L. Caro Jr.  |  Protocol Engineering Lab |
| www.armandocaro.net  |    University of Delaware |
0--                                              --0

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On Tue, Mar 09, 2004 at 01:23:13PM -0500, Armando L. Caro Jr. wrote:
> 
> > Draft Minutes as of 8 Mar 2004
> 
> <snip>
> 
> >  Ted mentioned that there's a discussion on e2e about a TCP consolidation
> >  document.  Ted is not clear on the difference between what they're
> >  asking for and a roadmap, and Ted would rather see that energy in the
> >  roadmap.
> 
> <snip>
> 
> >  Congestion Control
> >	Update 2581  (back to PS?)
> >		merge 3390 (initial cwnd)
> >		merge 3042 (limited transmit)
> >		merge 2582 (Reno bugfix/NewReno)
> >		merge 3465 (App. Byte Count security)
> 
> These merges are basically what we had in mind with the consolidation
> document.

I'm *hoping* that means that there's a core group of people interested
in working on an update of RFC2581 (2581bis) document.  If that's so,
it seems like TCPM would be the place to start working on it.

What can I do to encourage such an effort?

-- 
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.asc
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[Sorry I was out travelling and in the meanwhile this thread has
morphed into router design :) Anyway, I hope someone can start a
new thread with the router design subject while we continue on
TOE discussion - Thanks]

David,

> I wouldn't say in general that we found out that TOEs didn't work.  I 
> think a more qualified comment of what we found out was that in an 
> environment where the network is the bottle neck, TOEs did not provide 
> any cost effective benefits; and as CPUs got faster, the TOEs just 
> provided costs without benefits. :-)
> 
> TOE has become a buzz-word that needs to be qualified.  Some folks 
> include Checksum Offload and Large Segmentation Offload when talking 
> about TOE.  When others talk about TOE, they refer to offloading the 
> whole TCP/IP stack.

I think we are talking about offloading the whole stack or to the point
where the NIC needs to keep the state. The stateless offload (cksum
offload and LSO has already been happening for a while).

> With 10GigE, we are in the realm where the network is not the bottle 
> neck.  Even with GigE, the CPU has to do significant work to keep the 
> card fed, leaving little time for doing any application work.  I think 
> there are very few if any CPUs available today that can keep a 10GigE 
> pipe fed, much less handle an inbound 10GigE stream.

Our experience as well. The fastest intel or sparc CPU can saturate the
1GigE but there is not much room left for doing meaningful work by the
application.

> Features such as Checksum Offload and LSO are features that have low 
> impact on the OS and its TCP/IP stack.  Other features such as having a 
> timer to delay and aggregate transmit interrupts also help to reduce 
> the amount of work that the CPU has to do to process the data.
> 
> For me, the question is what can be done in a 10GigE NIC to allow the 
> host CPU to handle both sending and receiving full speed on that 
> interface?  TOE is just one part of that discussion and the danger is 
> that if too much focus is put on TOE, the other aspects of how to 
> improve how the CPU and the NIC communicate will not be properly 
> investigated.

You defintely have to fanout to distribute load across multiple CPUs if
you don't have TOE and CPU speed doesn't scale dramatically in couple of years.
We are developing a dynamic polling/interrupt mechanism to use general
purpose CPU to saturate 10GigE. Using TOE based cards is another possible
approach (you still need to deal with moving a giga byte of data across
your back plane though).

Cheers,
Sunay


> 
> 			-David Borman
> 
> On Mar 3, 2004, at 12:07 AM, Craig Partridge wrote:
> 
> >
> > Well, the interesting thing is we went down this path in the early 
> > 1980s
> > and found that TOEs didn't work.
> >
> > If I try to distill all that we learned in the 1980s into one 
> > question, I
> > come out with:
> >
> >     In the 1980s we discovered that communicating with a TOE over a bus
> >     about a TCP connection was as expensive or more expensive than
> >     simply handling the TCP connection in the main processor.  What 
> > about
> >     today's TOE designs makes them different?
> >
> > Craig
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www1.ietf.org/mailman/listinfo/tcpm
> 


-- 
Sunay Tripathi
Senior Staff Engineer,
Solaris Kernel Networking,
Sun MicroSystems Inc.

email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)




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--=-=-=


> I'm *hoping* that means that there's a core group of people interested
> in working on an update of RFC2581 (2581bis) document.  If that's so,
> it seems like TCPM would be the place to start working on it.
> 
> What can I do to encourage such an effort?

This in on my plate, actually.  (And, has been.)  There is an old, but I
think good, plan of attack.  I'll try to throw it out and let everyone
kick it soon.

(Maybe it'd be nice if the co-chairs actually chatted every once in a
while. :-)

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/




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On Tue, Mar 09, 2004 at 05:03:23PM -0500, Mark Allman wrote:
> 
> > I'm *hoping* that means that there's a core group of people interested
> > in working on an update of RFC2581 (2581bis) document.  If that's so,
> > it seems like TCPM would be the place to start working on it.
> > 
> > What can I do to encourage such an effort?
> 
> This in on my plate, actually.  (And, has been.)  There is an old, but I
> think good, plan of attack.  I'll try to throw it out and let everyone
> kick it soon.
> 
> (Maybe it'd be nice if the co-chairs actually chatted every once in a
> while. :-)

Sounds like I can encourage such an effort by occasionally accidentally
starting a parallel one. :-)

I'm happy to stay out of the way while you compose the 2581 update and
let these guys join in the evaluation of it when it apprears.

Sorry for sending mixed messages.

-- 
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.asc
Unexpected attachment on this mail? See http://www.isi.edu/~faber/FAQ.html#SIG 

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> Sunay Tripathi <Sunay.Tripathi@eng.sun.com> writes:
> 
> > 1) Extra processor(s) buried in the TOE for networking processing which is
> >    hidden from the kernel and leaves the host CPU to do more application
> >    related work. Saves the cost of licences for application which take
> >    number of CPU into account (oracle is one such application cited).
> 
> One would hope TCP design would not be guided by Oracle licensing quirks.

I don't think we are designing TCP here. We are just discussing an
implementation of TCP and obviously costs do play a factor in any
implementation. 

> > 2) On low end (1-2 CPU) x86 based machines, cost of adding a processor
> >    is much higher than adding a TOE (I personally haven't verified this).
> 
> It's not obvious why this should necessarily be the case, given that
> it is likely that there will continue to be quite a bit more
> general-purpose CPUs made than TCP offload engines.
> 
> > 3) For the up and coming 10Gb NICs, TOE will help saturate the link. Some
> >    vendors assert that TOE will be required to support 10Gb NICs.
> 
> Right now, one can do almost 8Gb/s with a single TCP stream over 10GE
> (let's say 5 or 6Gb/s with more common hardware).  The limiting factor
> is currently host bus speed.  There's nothing (except for compression
> on the bus side, but it should be clear we're not going there) that an
> offload engine can do about host bus speed.  By the time host busses
> faster than 10Gb/s are commonly available, pretty routine x86 box
> should handle 10GE saturation.

We have host busses today which can handle a giga bytes plus on the
back plane. Infiniband based machines are already hitting and there
are other technologies (that I can't talk about yet) which will make
the host bus issues irrelevant. 

> > 4) Performance reasons. Just the LSO aspect of TOE (sending large chunks of
> >    data and letting the TOE split it up in mss size pieces) and ack
> >    coalescing gives a pretty good boost (our own prototypes indicates that
> >    this is true). The gains are by optimizing data movement and not by
> >    offloading protocol processing.
> 
> Here I would agree with the unnamed TOE vendor whom you're
> paraphrasing (and with Jerry Chu's comments in this thread).  Ethernet
> frame size (1500 bytes or even 9kB) is very small; we'd be in a better
> world if we could specify the MTU in units of time (the number of CRC
> bits would have to scale up as well, of course).  TCP offload engines
> could be a kludge that would help to work around this deficiency in
> Ethernet.
> 
> Note that since even at 10Gb/s one CPU is enough to saturate the link
> with 9kB packets, the need for this work-around is not at all
> pressing.  Given the potential harmful effects (undetected errors on
> the host bus, difficulty in patching stale or buggy TCP code, etc.),
> one would probably be better served by concentrating on the deployment
> of jumbo frames.  The investment to support jumbo frames has largely
> already been made, so why not extract all we can from it first?

Sure, jumbo frames do deserve attention. Its being discussed right now
in IETF but I am not sure 9k frames would be enough to saturate the
10Gb NIC till the processor speeds also scale significantly. Note that
its not just the ability to saturate the link but you also need to
have some CPU free to do real work as well.

Cheers,
Sunay

> 
> -- 
> Stanislav Shalunov		http://www.internet2.edu/~shalunov/
> 


-- 
Sunay Tripathi
Solaris Kernel Networking,
Sun MicroSystems Inc.

email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)




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On Tuesday 09 March 2004 01:07, Craig Partridge wrote:
> In message <200403090046.15061.zec@tel.fer.hr>, Marko Zec writes:
> >All I was trying to point out in my prior posts is that precisely in
> >such a situation bursty traffic will present a big failure.
>
> Right, but bursts only matter if they cause a queue to build up in a
> router. Putting a burst on a wire doesn't mean a queue in the router
> -- it depends on the speed of the routers inputs and outputs.
>
> Putting this another way, TCP causes a queue in the bottleneck router
> (the spot where A > D).


OK, can we then agree that making such a spot a subject to an additional 
traffic burst can be considered a bad thing?

Marko


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On Monday 08 March 2004 22:50, Craig Partridge wrote:
> In message
> <8052E2EA753D144EB906B7A7AA399714022A05B7@mailserv.hatteras.com>, "S
>
> tephen Suryaputra" writes:
> >I think Marko is referring to the situation when there is a
> > congestion. Because of that the queue builds up and if the queue
> > space is in term of the number of packets, then a bunch of small
> > packets can potentially dominate the queue and left little room for
> > large packets eventhough there is a space in the buffer to
> > accomodate the large ones.
>
> Well, I tried to nail this down by setting it up as a queueing system
> but that's awfully hard.  Here's the cut:
>
> 1. Basically you have a general arrival process delivering pieces of
> work of variable size, at arrival rate A subject to the condition
> that if there's an event of size w1 at time t1, then the next event
> cannot arrive sooner than time t1+w1.  [I.e. A(t2,w2) - A(t1,w1) >=
> w1].
>
> 2. You then have a series of queues Q1-Qn attached to servers S1-Sn
> that serve events at a fixed rate R, where R1 <= A (i.e. we can
> handle the maximum rate).
>
> 3.  You then have a departure queue D1, and departure server DS,
> which has a service rate D, where D can be defined such that D(w) <=
> w (that is, the output is faster or equivalent speed to the input) or
> D(w) >= w (output is slower).
>
>
> If D(w) <= w, then I think one can argue simply by inspection that
> the total work unit buffering in the system is [max(w)/min(w)]-n. 
> That is you simply need enough queueing to handle a queue that
> develops because a big packet causes a bunch of little packets to
> queue behind it (because they arrive while the big packet is going
> out).  See Comment (**) below.
>
> I can't find an easy solution for D(w) >= w, because, in general, the
> queue can grow without bound.


Isn't the TCP's ultimate performance goal precisely the maintenance of 
the equilibrium D(w) = w in the bottleneck queue? If you are a lucky 
owner of just the TCP implementation that has no problem of achieving 
that goal for the particular network configuration, the queue length on 
the bottleneck link will fluctuate - sometimes it will be almost empty, 
sometimes almost full, and every now and then it will even have to drop 
a packet or two, but hopefully not more than that.

All I was trying to point out in my prior posts is that precisely in 
such a situation bursty traffic will present a big failure. If the 
burst arrives when the buffer is almost full, then the majority of the 
packets from the burst will be discarded. Furthermore, if packets of 
smaller size are used, requiring more packets to carry the same amount 
of useful payload, the greater will be the percentage of such packets 
to be dropped, assuming the buffer size on the router is defined by the 
number of packets it can hold, regardless of packet size.

Marko


>
> ** If you believe this model, what it says is that if you go with
> packet size buffers, and the ratio between the smallest possible
> packet and largest possible packet is big, then you can find your
> queues growing quite large, due not to classic congestion (where
> there is too much demand fighting for one output link) but rather to
> serialization delays.
>
> Craig


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--=-=-=


> Sounds like I can encourage such an effort by occasionally
> accidentally starting a parallel one. :-)

Right.  I'd like to say I mentioned this potential effort to you early
on.  But, I am sure it slipped my mind.  Sorry!

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/




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In message <200403090046.15061.zec@tel.fer.hr>, Marko Zec writes:

>Isn't the TCP's ultimate performance goal precisely the maintenance of 
>the equilibrium D(w) = w in the bottleneck queue?

No, TCP's goal is to equal the lesser of A or D (whatever they are).

>All I was trying to point out in my prior posts is that precisely in 
>such a situation bursty traffic will present a big failure.

Right, but bursts only matter if they cause a queue to build up in a router.
Putting a burst on a wire doesn't mean a queue in the router -- it depends
on the speed of the routers inputs and outputs.

Putting this another way, TCP causes a queue in the bottleneck router (the
spot where A > D).

Craig

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As the network gets faster, the user perceived performance issue moves into 
the application layer, except for the special case of remoting bulk disk 
drive accesses.   There is a small "opportunity window" of app requirements 
where the TCP layer is the crucial issue for performance.

It's easy to get focused on unrealistic benchmark cases that ignore the 
complexity of app logic, and the power of general purpose computing 
architectures that can dynamically partition workload rather than 
statically partitioning function into boxes.

I wouldn't waste my time on TOEs.  There are more useful things to work on, 
like making end-to-end secure and resilient communications practical.


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To: "Craig Partridge" <craig@bbn.com>, "Marko Zec" <zec@tel.fer.hr>
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I think that what Marco meant was that when a congestion occurs, i.e., =
the incoming traffic rate is higher then the service rate (the interface =
speed) the potential for congestion also depends on the packet sizes. =
This is so since the queue is not comprised from "bytes" only, but =
rather of buffer descriptors... The queue length in many cases is the =
number of descriptors, and the queue length in bytes is indeed depending =
on the packet sizes enqueued.=20

Thus, for Ethernet interfaces if an interface with speed L (b/s) is =
transmitting a 1500 bytes frame, and several other interfaces (say N =
interfaces) are sending traffic to that the same interface, then at the =
time L/(1500*8) seconds,  there can be either N packets - descriptors =
that needs to be queued, or N*(1500/64) in the worst case... the worst =
case is quite hard on some HW implementations as the number of =
descriptors in many cases is the number of max size packet buffers =
(packets are usually buffered in a fixed size buffers equal to the max =
frame size).

- Tal

-----Original Message-----
From: Craig Partridge [mailto:craig@bbn.com]
Sent: Monday, March 08, 2004 6:15 PM
To: Marko Zec
Cc: tcpm@ietf.org; tsvwg@ietf.org; end2end-interest@postel.org
Subject: Re: [Tsvwg] Re: [e2e] Are you interested in TOEs and related =
issues=20



In message <200403081646.48030.zec@tel.fer.hr>, Marko Zec writes:

>On Sunday 07 March 2004 23:06, Alexandre L. Grojsgold wrote:

>> Small packets get more buffer frames, but are also more rapidly sent.
>> The gueue grows fats, but shrinks fast too. It means that a burst of
>> small packets will not increase the packet loss, no mather the length
>> of the buffer (in packets).
>
>
>This theory is unfortunately wrong, because it ignores the fact that=20
>when congestions do occur, traffic flows consisting of small packets=20
>(regardless whether they are bursty or smooth) will use more buffer=20
>slots then the flows carrying the larger ones, leaving less remaining=20
>buffer slots available for queuing further packets / bursts.

I must be missing something here.  I think Alexandre's generally right.
If the buffers load and empty according to the amount of data in them,
then packet size isn't an issue.

The only situations in which this isn't true, that I can think off the
top of my head, are:

    * If the router has places where it moves all the bytes in the =
packet
      buffer, regardless of how much data the buffer actually contains.
      Since this would cause extraordinary internal bandwidth =
challenges,
      I doubt anyone does it for any but the lowest speed routers.

    * If the router has a deep pipeline and the time at each stage in =
the
      pipeline is set to equal the average packet time rather than the
      minimum packet time.  But most folks carefully design router =
pipelines
      for minimum packet times precisely because of this risk.

So clue me in -- what detail of router design am I missing?

Craig

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From: Sunay Tripathi <Sunay.Tripathi@eng.sun.com>
Message-Id: <200403110223.i2B2NUt4103646@jurassic.eng.sun.com>
Subject: Re: [tcpm] Re: [e2e] Are you interested in TOEs and related  issues
 (Resend)
In-Reply-To: <6.0.1.1.2.20040310095034.03103108@127.0.0.1> from "David P. Reed"
 at "Mar 10, 2004 09:55:18 am"
To: "David P. Reed" <dpreed@reed.com>
Date: Wed, 10 Mar 2004 18:23:30 -0800 (PST)
CC: Sunay Tripathi <Sunay.Tripathi@eng.sun.com>, David Borman <dab@cray.com>,
        Craig Partridge <craig@bbn.com>, tcpm@ietf.org, tsvwg@ietf.org,
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David,

Thanks for clarifying. 

> At 06:29 PM 3/9/2004, David P. Reed wrote:
> >I wouldn't waste my time on TOEs.  There are more useful things to work 
> >on, like making end-to-end secure and resilient communications practical.
> 
> To clarify this, if it seems like a non-sequitur or change in topic, 
> remember that offloading TCP into a separate box locks in any security 
> flaws into the architectural structure you had to create to make it 
> happen.  In practice, this means that no one will have the energy or money 
> to fix the security and resiliency issues in such solutions - which means 
> more kludges in wrap-around devices and interceptors that create the false 
> illusion of security.

Yes, that was one of the first thoughts that came to my mind when I
started looking at this space. Its not just security issues but bugs and
new features. TCP despite being mature still sees couple of RFC or tweaks
a year that needs to be rolled out to existing systems. Apparantly the
TOE vendors had thought about that. The firmware running the protocol code
actually lives on the host and is delivered via exisiting delivery
mechanism (packages in Solaris which can be upgraded). When machine boots
and interface is configured, the protocol firmware is sucked in. So
delivering fix or new revs in pretty similar to exisiting mechanisms.
Also, the source code is still 'C' based so people like me can still
work with it (just need to use some magic in the end to create the
firmware instead of complier). Some vendors we talked to have promised
that they can pretty much suck in the same code that runs on the host
into the card (as long as we can clean some complier specific directives
etc). Although I must admit that this aspect (common source files) of TOE 
has not been investigated by us at all so far.

Cheers,
Sunay

> 
> Communications networking is a systems problem, and ignoring the systems 
> issues by focusing on a narrow problem space is not a good investment of 
> effort.
> 
> 


-- 
Sunay Tripathi
Solaris Kernel Networking,
Sun MicroSystems Inc.

email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)




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> Yes, that was one of the first thoughts that came to my mind when I
> started looking at this space. Its not just security issues but bugs and
> new features. TCP despite being mature still sees couple of RFC or tweaks
> a year that needs to be rolled out to existing systems. Apparantly the
> TOE vendors had thought about that. The firmware running the protocol code
> actually lives on the host and is delivered via exisiting delivery
> mechanism (packages in Solaris which can be upgraded). When machine boots
> and interface is configured, the protocol firmware is sucked in. So
> delivering fix or new revs in pretty similar to exisiting mechanisms.
> Also, the source code is still 'C' based so people like me can still
> work with it (just need to use some magic in the end to create the
> firmware instead of complier). Some vendors we talked to have promised
> that they can pretty much suck in the same code that runs on the host
> into the card (as long as we can clean some complier specific directives
> etc). Although I must admit that this aspect (common source files) of TOE
> has not been investigated by us at all so far.

That seems like a path out when the TOE is based on an embedded CPU, but doesn't
seem like it would work for those TOEs that are ASICs.

And a mostly rhetorical question - strictly in the context of TCP comms and not
RDMA and the like - if one can code tightly enough for an embedded CPU a TCP/IP
stack including comms with the host across the I/O bus with enough performance
to be reasonable for a given link, can't one presumeably make the host's TCP/IP
stack similarly efficient?  If nothing else that seems to be alluded to by the
last sentence in the quoted paragraph.

rick jones
-- 
Wisdom Teeth are impacted, people are affected by the effects of events.
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to post, OR email to raj in cup.hp.com  but NOT BOTH...

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At 10:33 PM 3/4/2004, stanislav shalunov wrote:
>Sunay Tripathi <Sunay.Tripathi@eng.sun.com> writes:
>
> > 1) Extra processor(s) buried in the TOE for networking processing which is
> >    hidden from the kernel and leaves the host CPU to do more application
> >    related work. Saves the cost of licences for application which take
> >    number of CPU into account (oracle is one such application cited).
>
>One would hope TCP design would not be guided by Oracle licensing quirks.
>
> > 2) On low end (1-2 CPU) x86 based machines, cost of adding a processor
> >    is much higher than adding a TOE (I personally haven't verified this).
>
>It's not obvious why this should necessarily be the case, given that
>it is likely that there will continue to be quite a bit more
>general-purpose CPUs made than TCP offload engines.
>
> > 3) For the up and coming 10Gb NICs, TOE will help saturate the link. Some
> >    vendors assert that TOE will be required to support 10Gb NICs.
>
>Right now, one can do almost 8Gb/s with a single TCP stream over 10GE
>(let's say 5 or 6Gb/s with more common hardware).  The limiting factor
>is currently host bus speed.  There's nothing (except for compression
>on the bus side, but it should be clear we're not going there) that an
>offload engine can do about host bus speed.  By the time host busses
>faster than 10Gb/s are commonly available, pretty routine x86 box
>should handle 10GE saturation.
>
> > 4) Performance reasons. Just the LSO aspect of TOE (sending large chunks of
> >    data and letting the TOE split it up in mss size pieces) and ack
> >    coalescing gives a pretty good boost (our own prototypes indicates that
> >    this is true). The gains are by optimizing data movement and not by
> >    offloading protocol processing.
>
>Here I would agree with the unnamed TOE vendor whom you're
>paraphrasing (and with Jerry Chu's comments in this thread).  Ethernet
>frame size (1500 bytes or even 9kB) is very small; we'd be in a better
>world if we could specify the MTU in units of time (the number of CRC
>bits would have to scale up as well, of course).  TCP offload engines
>could be a kludge that would help to work around this deficiency in
>Ethernet.
>
>Note that since even at 10Gb/s one CPU is enough to saturate the link
>with 9kB packets, the need for this work-around is not at all
>pressing.  Given the potential harmful effects (undetected errors on
>the host bus, difficulty in patching stale or buggy TCP code, etc.),
>one would probably be better served by concentrating on the deployment
>of jumbo frames.  The investment to support jumbo frames has largely
>already been made, so why not extract all we can from it first?

Jumbo frames are not a panacea:

- Workloads use a variety of message sizes with many messages being 
significantly smaller than a jumbo frame - applications are not going to 
change that rapidly to expand the amount of information they exchange that 
would benefit from jumbo frames.  This is true for IPC, file and storage 
applications.  Some performance benefits have been seen using jumbo frames 
for services such as a FTP server, but most people don't have a PMTU that 
will support jumbo frames so the benefit is perhaps confined to a data center.

- Jumbo frames cause head-of-line blocking within a fabric so unless one 
supports multi-pathing / traffic segregation as well as multiple ports on 
each endnode, low-latency / hi-pri traffic QoS will be negatively impacted.

- TCP large send also causes QoS problems as many implementations only work 
at the message level rather than the frame level so differentiated services 
/ bandwidth management are much harder to implement as these become 
competing objectives for the point of serialization, i.e. the link.

While it is true that at higher link rates, the impacts on QoS may be 
minimized but the impacts are not eliminated and one would need to do more 
at the fabric level as well as the NIC implementation level to provide a 
reasonable solution.

As for TOE or RDMA, jumbo isn't really a major benefit.  Work is posted for 
transmission / receipt as single operations and the hardware does the rest 
of the work independent of the application / CPU.  The benefits are 
demonstrated using existing technology such as InfiniBand (numerous 
performance analysis papers are publicly available on Linux based 
implementations) and on some early TCP-based protocol off-load prototypes 
which confirm similar savings as found with InfiniBand.  I think it is fair 
to say that for selected applications, protocol stack off-load of at least 
the main data path will result in reduced CPU consumption, reduced memory 
bus and I/O bus consumption, and improved application performance.  Whether 
this is worthwhile or not is really a judgement call and is somewhat a 
function of whether one is changing only one end of the wire ala a TOE or 
both ends ala using a RDMA wire protocol on top of a protocol off-load 
device.  Embedded or host-based (e.g. a server) will see similar resource 
savings on one hand and will pay a price for those savings on the other 
hand, i.e. dealing with state and resource management in what is generally 
implemented as a non-coherent device.

Mike  


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To: Sunay Tripathi <Sunay.Tripathi@eng.sun.com>, David Borman <dab@cray.com>
From: "David P. Reed" <dpreed@reed.com>
Subject: Re: [tcpm] Re: [e2e] Are you interested in TOEs and related
  issues (Resend)
Cc: Craig Partridge <craig@bbn.com>, tcpm@ietf.org, tsvwg@ietf.org,
        end2end-interest@postel.org,
        Sunay Tripathi <Sunay.Tripathi@eng.sun.com>
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At 06:29 PM 3/9/2004, David P. Reed wrote:
>I wouldn't waste my time on TOEs.  There are more useful things to work 
>on, like making end-to-end secure and resilient communications practical.

To clarify this, if it seems like a non-sequitur or change in topic, 
remember that offloading TCP into a separate box locks in any security 
flaws into the architectural structure you had to create to make it 
happen.  In practice, this means that no one will have the energy or money 
to fix the security and resiliency issues in such solutions - which means 
more kludges in wrap-around devices and interceptors that create the false 
illusion of security.

Communications networking is a systems problem, and ignoring the systems 
issues by focusing on a narrow problem space is not a good investment of 
effort.



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--=-=-=


Hello!

As discussed at the meeting last week we'd like to encourage folks to
read and comment on the F-RTO draft: draft-ietf-tsvwg-tcp-frto-01.txt.
Based on the feedback we've received so far it seems like all the large
issues have been working out of this draft and it is near ready to be
forwarded.  We'd like folks to read the document and send any remaining
issues to the list.

Thanks!

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/





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--=-=-=


(catching up)

> The critical observation to make here is as IPv6 addresses are tried
> first (I'm simplifying; it's more complex than that), one must cycle
> through all the addresses, before trying with IPv4.  However, IPv4
> connections might still work! With TCP timeouts, this takes a lot of
> time --- and should be made quicker.

Speaking personally, I think what you have written is fine and would not
block the document on it.

> There is going to be a WG last call on the document soon, aimed for 
> Informational -- just for documenting the issues.  The actual fixes 
> (or not) would have to be done elsewhere (e.g., in TCPM WG ;-).

Speaking as tcpm co-chair, yeah - we'd certainly be game to help out
here.

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/




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To: Pekka Savola <pekkas@netcore.fi>
From: Mark Allman <mallman@icir.org>
Reply-To: mallman@icir.org
Cc: tcpm@ietf.org, v6ops@ops.ietf.org, sebastien.roy@sun.com
Subject: Re: [tcpm] TCP, multiple addresses and soft errors when connecting 
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--=-=-=


(catching up)

> The critical observation to make here is as IPv6 addresses are tried
> first (I'm simplifying; it's more complex than that), one must cycle
> through all the addresses, before trying with IPv4.  However, IPv4
> connections might still work! With TCP timeouts, this takes a lot of
> time --- and should be made quicker.

Speaking personally, I think what you have written is fine and would not
block the document on it.

> There is going to be a WG last call on the document soon, aimed for 
> Informational -- just for documenting the issues.  The actual fixes 
> (or not) would have to be done elsewhere (e.g., in TCPM WG ;-).

Speaking as tcpm co-chair, yeah - we'd certainly be game to help out
here.

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/




--=-=-=
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Subject: Re: [tcpm] TCP, multiple addresses and soft errors when connecting 
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--=-=-=


(catching up)

> The critical observation to make here is as IPv6 addresses are tried
> first (I'm simplifying; it's more complex than that), one must cycle
> through all the addresses, before trying with IPv4.  However, IPv4
> connections might still work! With TCP timeouts, this takes a lot of
> time --- and should be made quicker.

Speaking personally, I think what you have written is fine and would not
block the document on it.

> There is going to be a WG last call on the document soon, aimed for 
> Informational -- just for documenting the issues.  The actual fixes 
> (or not) would have to be done elsewhere (e.g., in TCPM WG ;-).

Speaking as tcpm co-chair, yeah - we'd certainly be game to help out
here.

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/




--=-=-=
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Subject: Re: [tcpm] TCP, multiple addresses and soft errors when connecting 
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--=-=-=


(catching up)

> The critical observation to make here is as IPv6 addresses are tried
> first (I'm simplifying; it's more complex than that), one must cycle
> through all the addresses, before trying with IPv4.  However, IPv4
> connections might still work! With TCP timeouts, this takes a lot of
> time --- and should be made quicker.

Speaking personally, I think what you have written is fine and would not
block the document on it.

> There is going to be a WG last call on the document soon, aimed for 
> Informational -- just for documenting the issues.  The actual fixes 
> (or not) would have to be done elsewhere (e.g., in TCPM WG ;-).

Speaking as tcpm co-chair, yeah - we'd certainly be game to help out
here.

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/




--=-=-=
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To: Pekka Savola <pekkas@netcore.fi>
From: Mark Allman <mallman@icir.org>
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Subject: Re: [tcpm] TCP, multiple addresses and soft errors when connecting 
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--=-=-=


(catching up)

> The critical observation to make here is as IPv6 addresses are tried
> first (I'm simplifying; it's more complex than that), one must cycle
> through all the addresses, before trying with IPv4.  However, IPv4
> connections might still work! With TCP timeouts, this takes a lot of
> time --- and should be made quicker.

Speaking personally, I think what you have written is fine and would not
block the document on it.

> There is going to be a WG last call on the document soon, aimed for 
> Informational -- just for documenting the issues.  The actual fixes 
> (or not) would have to be done elsewhere (e.g., in TCPM WG ;-).

Speaking as tcpm co-chair, yeah - we'd certainly be game to help out
here.

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/




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To: Pekka Savola <pekkas@netcore.fi>, tcpm@ietf.org
From: Fernando Gont <fernando@gont.com.ar>
Subject: Re: [tcpm] TCP, multiple addresses and soft errors when
  connecting
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At 04:52 05/03/2004 +0200, Pekka Savola wrote:

>TCP timeouts when connecting to a destination, but when the
>destination is unreachable (e.g., because you don't have global IPv6
>routes, your first-hop router is not operational, the packets are sent
>on-link by default, etc.), TCP does not abort the connectivity (and
>try the next address quicker -- instead of waiting for the TCP
>timeout).

Not sure if I got it right. Do you mean TCP does not abort the conectivity, 
and thus the *application* cannot try the next address quicker?



>2.3.1.1 TCP Connection Termination
>   One solution is for TCP to abort connections in SYN-SENT or
>    SYN-RECEIVED state when it receives an ICMPv6 Destination Unreachable
>    message.

But what if this scenario arises for all the addresses, because of a 
transient problem?
The application won't cycle again, and will return an error. And the 
*transient* problem could disappear perhaps 5 seconds later...

IMHO, if you had a "higher-level" connect(), this behavior would make 
sense, as you could handle the whole issue, in a transparent way to the 
application.

Does it really make sense to handle the whole connection establishment 
issue inside the app, instead of inside the API itself?

I think that if we're debating this, we're implicitly assuming that 
applications want to connect to say www.example.com, and not one specific 
IP address to which it maps to.

So why not implement such a "higher-level" connect(), and handle all the 
relevant issues inside the API?


>   When [HOSTREQS] was written, most applications would mostly only try
>    one address when establishing communication with a destination.  Not
>    aborting a connection was a sane thing to do if re-trying a single
>    address was a better alternative over quitting the application
>    altogether. With IPv6, and especially on dual stack systems,
>    destinations are often assigned multiple addresses (at least one IPv4
>    and one IPv6 address), and applications iterate through destination
>    addresses when attempting connections.

Doesn't the problem really have to do with having applications performing 
such a low-level action?

Just my two cents,


--
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@acm.org



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From: Fernando Gont <fernando@gont.com.ar>
Subject: Re: [tcpm] TCP, multiple addresses and soft errors when 
  connecting
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At 16:39 20/03/2004 +0200, Pekka Savola wrote:

> > Does it really make sense to handle the whole connection establishment
> > issue inside the app, instead of inside the API itself?
>Those are the core APIs we have today.  We could have better ones, but
>IMHO I think we need to fix problems in the current ones as long as we
>don't have a widely-deployed alternative.

IMHO, there are two points to note:

* If you fix the current ones so that their behavior is "acceptable" to the 
application programmer, then it'll be harder to introduce a new API. People 
is reluctant to change, so they must have a reason for learning a new API. 
If you try to fix the current one, they will find fewer reasons for the change.

* You said "we don't have a widely-deployed alternative". Unless you meant 
"we don't have a well-tested alternative", I'd say that it's a 
chicken-and-egg problem. That means, for an API to be widely deployed, 
people would have to use it. And for people to use it, they should be able 
to find a good reason to do it.

Yes, I'm aware that my proposal my sound as a long-term solution. But, 
IMHO, it will certainly be harder to introduce a new API if you actually 
want to do it "in long term".
I guess "timing" is an issue, here.



> > I think that if we're debating this, we're implicitly assuming that
> > applications want to connect to say www.example.com, and not one specific
> > IP address to which it maps to.
> >
> > So why not implement such a "higher-level" connect(), and handle all the
> > relevant issues inside the API?
>
>Defining such APIs is probably beyond the scope of the work we can do,
>and the things we can fix except in the 5+ year timeframe.. but if I
>get you right, you're proposing that instead of:
>
>  1) getaddrinfo() or gethostbyname() and connect for one address,
>[where, arguably, TCP retry timeout could be warranted -- I don't
>think so myself, but you could argue for it]
>
>  2) getaddrinfo() / gethostbyname() loop to connect to one address
>[where TCP retry timeout could not be warranted]
>
>You'd be proposing to create a new function to accomplish 2) in such a
>manner that it would also react to ICMP messages, etc?

Exactly.

Then, within that function, each individual connect() could react to ICMP 
messages a
as you proposed, but if all addresses fail because of ICMP errors, it could 
let TCP retry in the hope the transiet problem will dissappear.


[....]
> > Doesn't the problem really have to do with having applications performing
> > such a low-level action?
>
>At some point, someone has to decide which behaviour is desirable.
>Isn't this only hiding the same policy decision inside an abstraction
>layer

Yes, but even when the difference may sound subtle, IMHO it's not.
Having that abstraction layer is what let's you choose a policy, without 
having the risk of causing any harm to existing apps.
That way, if you ever need to change the low-level-connect() behavoir for 
whatever reason, you could still do it. As far as the high-level connect() 
behaves the same, there would be no problem.

Think about it. Why should an application programmer care about TCP 
retries, ICMP erros, an the like?
In the same way, why should he care about setting "strange" options such as 
SO_REUSEADDR on his listening socket?
Should *he* really handle byte-ordering issues  by means of htons() and the 
like?


>-- and the same could be achieved by e.g. a setsockopt() flag?

Strictly speaking, you *could* do that. But then you'd have all programmers 
implementing themselves a function you could provide them.
You'd have all them implementing the getaddrinfo()/gethostbyname() loop, 
setting a strange socket option, etc. Why not provide them with a 
well-tested function, and let them spend their time implementing their 
application itself?



>(I don't deny that better APIs would not hurt, but we still seem to
>require low-level C-like primitives, and unless there are additional
>ones implemented there, we still have to fix the problems somehow..)

Well, the low-level ones could be provided as part of the new API.

Or along with the providing a new API, the current one could be modified as 
you suggest. It wouldn't hurt too much if the *low-level* API aborted a 
connection establishment because an ICMP error is received. Being a 
*low-level* API, you'd be suppossed to know the details behind the function.

Best Regards,


--
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@acm.org



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From: Pekka Savola <pekkas@netcore.fi>
To: Fernando Gont <fernando@gont.com.ar>
cc: tcpm@ietf.org, <v6ops@ops.ietf.org>, <sebastien.roy@sun.com>
Subject: Re: [tcpm] TCP, multiple addresses and soft errors when  connecting
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On Fri, 19 Mar 2004, Fernando Gont wrote:
> At 04:52 05/03/2004 +0200, Pekka Savola wrote:
> >TCP timeouts when connecting to a destination, but when the
> >destination is unreachable (e.g., because you don't have global IPv6
> >routes, your first-hop router is not operational, the packets are sent
> >on-link by default, etc.), TCP does not abort the connectivity (and
> >try the next address quicker -- instead of waiting for the TCP
> >timeout).
> 
> Not sure if I got it right. Do you mean TCP does not abort the conectivity, 
> and thus the *application* cannot try the next address quicker?

Yep, that's what I meant (and by application, I mean both the app and
the APIs/libraries it's using).

> >2.3.1.1 TCP Connection Termination
> >   One solution is for TCP to abort connections in SYN-SENT or
> >    SYN-RECEIVED state when it receives an ICMPv6 Destination Unreachable
> >    message.
> 
> But what if this scenario arises for all the addresses, because of a 
> transient problem?
> The application won't cycle again, and will return an error. And the 
> *transient* problem could disappear perhaps 5 seconds later...

Note that this applies to new connections only.

Yes, this is a trade-off.  The assumption is that with big transient
problems, all the addresses {at least of the same address family} are
unreachable, or none of them are.  

So, you either have to choose between having TCP connect() try
retrying for dozens of seconds (hoping the transient to go away) or
failing completely.

The tradeoff recommended was that it's better to fail completely; this
works better for transient and non-transient failures, except for the
case where the transient failure was so short that the TCP retry
mechanism could be the feasible failure recovery mechanism for that.  
I don't think it is..

Obviously, there could be a flag to indicate to the stack whether you 
want this behaviour or not.

> Does it really make sense to handle the whole connection establishment 
> issue inside the app, instead of inside the API itself?

Those are the core APIs we have today.  We could have better ones, but
IMHO I think we need to fix problems in the current ones as long as we
don't have a widely-deployed alternative.
 
> I think that if we're debating this, we're implicitly assuming that 
> applications want to connect to say www.example.com, and not one specific 
> IP address to which it maps to.
> 
> So why not implement such a "higher-level" connect(), and handle all the 
> relevant issues inside the API?

Defining such APIs is probably beyond the scope of the work we can do, 
and the things we can fix except in the 5+ year timeframe.. but if I 
get you right, you're proposing that instead of:

 1) getaddrinfo() or gethostbyname() and connect for one address, 
[where, arguably, TCP retry timeout could be warranted -- I don't 
think so myself, but you could argue for it]

 2) getaddrinfo() / gethostbyname() loop to connect to one address
[where TCP retry timeout could not be warranted]

You'd be proposing to create a new function to accomplish 2) in such a 
manner that it would also react to ICMP messages, etc?

> >   When [HOSTREQS] was written, most applications would mostly only try
> >    one address when establishing communication with a destination.  Not
> >    aborting a connection was a sane thing to do if re-trying a single
> >    address was a better alternative over quitting the application
> >    altogether. With IPv6, and especially on dual stack systems,
> >    destinations are often assigned multiple addresses (at least one IPv4
> >    and one IPv6 address), and applications iterate through destination
> >    addresses when attempting connections.
> 
> Doesn't the problem really have to do with having applications performing 
> such a low-level action?

At some point, someone has to decide which behaviour is desirable.  
Isn't this only hiding the same policy decision inside an abstraction 
layer -- and the same could be achieved by e.g. a setsockopt() flag?

(I don't deny that better APIs would not hurt, but we still seem to 
require low-level C-like primitives, and unless there are additional 
ones implemented there, we still have to fix the problems somehow..)

-- 
Pekka Savola                 "You each name yourselves king, yet the
Netcore Oy                    kingdom bleeds."
Systems. Networks. Security. -- George R.R. Martin: A Clash of Kings


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I'll be submitting the minutes to the IETF for inclusion in the
proceedings on 24 Mar 2004.  I have seen no major corrections on them,
but if you've been keeping corrections to yourself, please let me know.

-- 
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.asc
Unexpected attachment on this mail? See http://www.isi.edu/~faber/FAQ.html#SIG 

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Thread-Topic: [Tsvwg] Re: [e2e] Are you interested in TOEs and related issues
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Hmm, that's kind of funny. My thoughts are just the opposite: jumbo =
frames may be a necessary evil until TOEs become universal.

Pat

-----Original Message-----
From: tsvwg-admin@ietf.org [mailto:tsvwg-admin@ietf.org]On Behalf Of
<SNIP>

Bursty traffic may be ok for LAN, but is considered evil for WAN.
TOE doesn't suffer this problem (since it is stateful), and may be
a necessary evil until jumbo frames become universal.

Jerry

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Subject: RE: [Tsvwg] Re: [tcpm] Re: [e2e] Are you interested in TOEs and related   issues(Resend)
Date: Fri, 26 Mar 2004 16:28:52 -0700
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To: <raj@cup.hp.com>, <tcpm@ietf.org>
Cc: <tsvwg@ietf.org>, <end2end-interest@postel.org>
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I'll take the liberty of responding to your rhetorical question. I don't =
think the question can be answered outside the context of protocols like =
RDMA and iSCSI. A good part of the performance problem for 10 Gig links =
is the data copies required. Protocols like iSCSI and RDMA can allow the =
NIC to place the data into applications buffers. This is similar to the =
function of some Fibre Channel interfaces that allow it higher =
performance. This is a large part of the benefit gained from =
implementing a TOE. Protocols like RDMA and iSCSI require TCP (or an =
alternative transport layer) to be implemented other them which means =
you need a TOE.=20

If you aren't doing placement on the NIC, it isn't clear that there is =
enough benefit from implementing a TOE. Because we want to do placement, =
we need a TOE.

Pat

-----Original Message-----
From: tsvwg-admin@ietf.org [mailto:tsvwg-admin@ietf.org]On Behalf Of
Rick Jones
Sent: Thursday, March 11, 2004 10:49 AM
To: tcpm@ietf.org
Cc: tsvwg@ietf.org; end2end-interest@postel.org
Subject: Re: [Tsvwg] Re: [tcpm] Re: [e2e] Are you interested in TOEs and
related issues(Resend)



> Yes, that was one of the first thoughts that came to my mind when I
> started looking at this space. Its not just security issues but bugs =
and
> new features. TCP despite being mature still sees couple of RFC or =
tweaks
> a year that needs to be rolled out to existing systems. Apparantly the
> TOE vendors had thought about that. The firmware running the protocol =
code
> actually lives on the host and is delivered via exisiting delivery
> mechanism (packages in Solaris which can be upgraded). When machine =
boots
> and interface is configured, the protocol firmware is sucked in. So
> delivering fix or new revs in pretty similar to exisiting mechanisms.
> Also, the source code is still 'C' based so people like me can still
> work with it (just need to use some magic in the end to create the
> firmware instead of complier). Some vendors we talked to have promised
> that they can pretty much suck in the same code that runs on the host
> into the card (as long as we can clean some complier specific =
directives
> etc). Although I must admit that this aspect (common source files) of =
TOE
> has not been investigated by us at all so far.

That seems like a path out when the TOE is based on an embedded CPU, but =
doesn't
seem like it would work for those TOEs that are ASICs.

And a mostly rhetorical question - strictly in the context of TCP comms =
and not
RDMA and the like - if one can code tightly enough for an embedded CPU a =
TCP/IP
stack including comms with the host across the I/O bus with enough =
performance
to be reasonable for a given link, can't one presumeably make the host's =
TCP/IP
stack similarly efficient?  If nothing else that seems to be alluded to =
by the
last sentence in the quoted paragraph.

rick jones
--=20
Wisdom Teeth are impacted, people are affected by the effects of events.
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to post, OR email to raj in cup.hp.com  but NOT BOTH...

_______________________________________________
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From: "Ehsan Hamadani" <e.hamadani@city.ac.uk>
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Dear all,
 
I have the following simple question; however I could not find the
answer in RFC's congestion control document.
 
In the congestion control algorithm, when TCP invokes fast retransmit
and sends the lost segment, which value does the TCP use for "last
segment send" in calculating "effective window"?
For instance, the segment that has been sent before fast retransmits was
20. Now because of receipt of 3 duplicate ACKs for segment 10, sender
retransmit segment 10 and set CWND to 1.
In this case, when another duplicate ACK for segment 10 arrives, and TCP
wants to calculate EffectiveWindow = CWND - (LastByteSent -
LastByteReceived) what is the value for LastByteSent??
Is it 10 or 20??
 
Thanks for your consideration.
 
Regards,
 
Ehsan
 

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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Dear all,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I have the following simple question; however I could =
not
find the answer in RFC&#8217;s congestion control =
document.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>In the congestion control algorithm, when TCP invokes =
fast retransmit
and sends the lost segment, which value does the TCP use for &#8220;last
segment send&#8221; in calculating &#8220;effective =
window&#8221;?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>For instance, the segment that has been sent before =
fast
retransmits was 20. Now because of receipt of 3 duplicate ACKs for =
segment 10,
sender retransmit segment 10 and set CWND to =
1.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>In this case, when another duplicate =
</span></font><st1:stockticker><font
 size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>ACK</span></font></st1:stock=
ticker><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'> for segment
10 arrives, and TCP wants to calculate EffectiveWindow =3D =
</span></font><st1:stockticker><font
 size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>CWN</span></font></st1:stock=
ticker><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>D &#8211;
(LastByteSent &#8211; LastByteReceived) what is the value for =
LastByteSent??<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Is it 10 or 20??<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thanks for your =
consideration.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Regards,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Ehsan<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

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Hi Ehsan,

> In the congestion control algorithm, when TCP invokes fast retransmit
> and sends the lost segment, which value does the TCP use for "last
> segment send" in calculating "effective window"?
> For instance, the segment that has been sent before fast retransmits was
> 20. Now because of receipt of 3 duplicate ACKs for segment 10, sender
> retransmit segment 10 and set CWND to 1.

If you are using RFC 2581, the cwnd will be set to flightsize/2 after a
fast retransmit, not to 1.

> In this case, when another duplicate ACK for segment 10 arrives, and TCP
> wants to calculate EffectiveWindow = CWND - (LastByteSent -
> LastByteReceived) what is the value for LastByteSent??
> Is it 10 or 20??

LastByteSent should really be HighestByteSent, and LastByteReceived should
be HighestByteReceived (assuming no SACKs). That makes the answer 20.

regards,
jana

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        Sunay Tripathi <Sunay.Tripathi@eng.sun.com>
From: David Borman <dab@cray.com>
Subject: Re: [tcpm] Re: [e2e] Are you interested in TOEs and related issues (Resend)
Date: Thu, 4 Mar 2004 16:12:37 +0900
To: Craig Partridge <craig@bbn.com>
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Craig,
I wouldn't say in general that we found out that TOEs didn't work.  I 
think a more qualified comment of what we found out was that in an 
environment where the network is the bottle neck, TOEs did not provide 
any cost effective benefits; and as CPUs got faster, the TOEs just 
provided costs without benefits. :-)

TOE has become a buzz-word that needs to be qualified.  Some folks 
include Checksum Offload and Large Segmentation Offload when talking 
about TOE.  When others talk about TOE, they refer to offloading the 
whole TCP/IP stack.

With 10GigE, we are in the realm where the network is not the bottle 
neck.  Even with GigE, the CPU has to do significant work to keep the 
card fed, leaving little time for doing any application work.  I think 
there are very few if any CPUs available today that can keep a 10GigE 
pipe fed, much less handle an inbound 10GigE stream.

Features such as Checksum Offload and LSO are features that have low 
impact on the OS and its TCP/IP stack.  Other features such as having a 
timer to delay and aggregate transmit interrupts also help to reduce 
the amount of work that the CPU has to do to process the data.

For me, the question is what can be done in a 10GigE NIC to allow the 
host CPU to handle both sending and receiving full speed on that 
interface?  TOE is just one part of that discussion and the danger is 
that if too much focus is put on TOE, the other aspects of how to 
improve how the CPU and the NIC communicate will not be properly 
investigated.

			-David Borman

On Mar 3, 2004, at 12:07 AM, Craig Partridge wrote:

>
> Well, the interesting thing is we went down this path in the early 
> 1980s
> and found that TOEs didn't work.
>
> If I try to distill all that we learned in the 1980s into one 
> question, I
> come out with:
>
>     In the 1980s we discovered that communicating with a TOE over a bus
>     about a TCP connection was as expensive or more expensive than
>     simply handling the TCP connection in the main processor.  What 
> about
>     today's TOE designs makes them different?
>
> Craig
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm


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From: "Leonid Grossman" <leonid.grossman@s2io.com>
To: "'Sunay Tripathi'" <Sunay.Tripathi@eng.sun.com>, <craig@bbn.com>
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> -----Original Message-----
> From: tsvwg-admin@ietf.org [mailto:tsvwg-admin@ietf.org] On 
> Behalf Of Sunay Tripathi
> Sent: Wednesday, March 03, 2004 8:56 PM
> To: craig@bbn.com
> Cc: tcpm@ietf.org; tsvwg@ietf.org; end2end-interest@postel.org
> Subject: [Tsvwg] Re: [e2e] Are you interested in TOEs and 
> related issues
> 
> 
> Craig,
> 
> > Well, the interesting thing is we went down this path in the early 
> > 1980s and found that TOEs didn't work.
> > 
> > If I try to distill all that we learned in the 1980s into one 
> > question, I come out with:
> > 
> >     In the 1980s we discovered that communicating with a 
> TOE over a bus
> >     about a TCP connection was as expensive or more expensive than
> >     simply handling the TCP connection in the main 
> processor.  What about
> >     today's TOE designs makes them different?
> 
> I don't have too much history (1980s predates my career) but 
> there were some attempts in mid 90s to do the same and 
> compared to that, the following reasons seem to be the driving force:
> 
> 1) Extra processor(s) buried in the TOE for networking 
> processing which is
>    hidden from the kernel and leaves the host CPU to do more 
> application
>    related work. Saves the cost of licences for application which take
>    number of CPU into account (oracle is one such application cited).
> 2) On low end (1-2 CPU) x86 based machines, cost of adding a processor
>    is much higher than adding a TOE (I personally haven't 
> verified this).
> 3) For the up and coming 10Gb NICs, TOE will help saturate 
> the link. Some
>    vendors assert that TOE will be required to support 10Gb NICs.


TOE will bring CPU utilization down of course, but 10GbE NIC will
saturate the link with or without a TOE - much like GbE NICs do it now.

> 4) Performance reasons. Just the LSO aspect of TOE (sending 
> large chunks of
>    data and letting the TOE split it up in mss size pieces) and ack
>    coalescing gives a pretty good boost (our own prototypes 
> indicates that
>    this is true). 

Just curious - in your testing, how much CPU is freed up by the ack
coalescing?

>The gains are by optimizing data movement and not by
>    offloading protocol processing.

LSO is a 'stateless' offload (much like checksum offload) so one doesn't
need a TOE to support it; 
most modern NICs (including our 10GbE Adapter) do LSO. Also, for normal
frames LSO has quite dramatic effect at 10GbE rates, but for Jumbo
frames performance gain diminishes to ~7%.


> 5) TOE is necessary for RDMA, iSCSI etc. for layering 
> reasons. I am not
>    involved with RDMA so someone who is an expert can 
> probably comment on
>    this part.
> 6) TOE based NIC are already making pretty good headway in 
> embedded space.
>    The technology is already maturing so why not use it in 
> broader market.
> 
> Note that the above claims are in no particular order of 
> importance and made my TOE vendors in general. Of these, I 
> personally do agree with 1 and 4 but that iteself doesn't 
> mean that TOE will make it in general purpose networking.
> 
> It would be interesting to see if you and others in the list 
> agree or disagree with these claims.
> 
> Thanks,
> Sunay
> 
> -- 
> 
> Sunay Tripathi
> Senior Staff Engineer,
> Solaris Kernel Networking,
> Sun MicroSystems Inc.
> 
> email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)
> 
> 
> 
> 


Regards, 
Leonid Grossman
www.s2io.com


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


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From: Sunay Tripathi <Sunay.Tripathi@eng.sun.com>
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Subject: Re: [Tsvwg] Re: [tcpm] Re: [e2e] Are you interested in TOEs and related
   issues(Resend)
In-Reply-To: <4050B4A6.D346736C@cup.hp.com> from Rick Jones at "Mar 11, 2004
 10:49:10 am"
To: Rick Jones <raj@cup.hp.com>
Date: Thu, 11 Mar 2004 11:31:59 -0800 (PST)
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Rick,

> > Yes, that was one of the first thoughts that came to my mind when I
> > started looking at this space. Its not just security issues but bugs and
> > new features. TCP despite being mature still sees couple of RFC or tweaks
> > a year that needs to be rolled out to existing systems. Apparantly the
> > TOE vendors had thought about that. The firmware running the protocol code
> > actually lives on the host and is delivered via exisiting delivery
> > mechanism (packages in Solaris which can be upgraded). When machine boots
> > and interface is configured, the protocol firmware is sucked in. So
> > delivering fix or new revs in pretty similar to exisiting mechanisms.
> > Also, the source code is still 'C' based so people like me can still
> > work with it (just need to use some magic in the end to create the
> > firmware instead of complier). Some vendors we talked to have promised
> > that they can pretty much suck in the same code that runs on the host
> > into the card (as long as we can clean some complier specific directives
> > etc). Although I must admit that this aspect (common source files) of TOE
> > has not been investigated by us at all so far.
> 
> That seems like a path out when the TOE is based on an embedded CPU, but doesn't
> seem like it would work for those TOEs that are ASICs.

The TOEs I am talking about have 1 or more embedded ARM processor (seems to 
be the more common model these days). 

> And a mostly rhetorical question - strictly in the context of TCP comms and not
> RDMA and the like - if one can code tightly enough for an embedded CPU a TCP/IP
> stack including comms with the host across the I/O bus with enough performance
> to be reasonable for a given link, can't one presumeably make the host's TCP/IP
> stack similarly efficient?  If nothing else that seems to be alluded to by the
> last sentence in the quoted paragraph.

I think Craig did ask this question before and the benefit is not so much
from the protocol processing saving but from the data movement. Below is
a portion of my earlier reply to Craig's email.

"
1) Extra processor(s) buried in the TOE for networking processing which is
   hidden from the kernel and leaves the host CPU to do more application
   related work. Saves the cost of licences for application which take
   number of CPU into account (oracle is one such application cited).
2) On low end (1-2 CPU) x86 based machines, cost of adding a processor
   is much higher than adding a TOE (I personally haven't verified this).
3) For the up and coming 10Gb NICs, TOE will help saturate the link. Some
   vendors assert that TOE will be required to support 10Gb NICs.
4) Performance reasons. Just the LSO aspect of TOE (sending large chunks of
   data and letting the TOE split it up in mss size pieces) and ack
   coalescing gives a pretty good boost (our own prototypes indicates that
   this is true). The gains are by optimizing data movement and not by
   offloading protocol processing.
5) TOE is necessary for RDMA, iSCSI etc. for layering reasons. I am not
   involved with RDMA so someone who is an expert can probably comment on
   this part.
6) TOE based NIC are already making pretty good headway in embedded space.
   The technology is already maturing so why not use it in broader market.
"

Cheers,
Sunay

> 
> rick jones
> -- 
> Wisdom Teeth are impacted, people are affected by the effects of events.
> these opinions are mine, all mine; HP might not want them anyway... :)
> feel free to post, OR email to raj in cup.hp.com  but NOT BOTH...
> 


-- 
Sunay Tripathi
Senior Staff Engineer,
Solaris Kernel Networking,
Sun MicroSystems Inc.

email: sunay@eng.sun.com		 Phone:	650-786-6007 (W)




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Subject: Re: [Tsvwg] Re: [tcpm] Re: [e2e] Are you interested in TOEs and related 
 issues (Resend)
References: <20040302150735.2A2221A9@aland.bbn.com> <510B9F0C-6DAB-11D8-AE57-000A95D7D4C0@cray.com> <404CBAEF.B49A643D@cup.hp.com> <20040308195206.GB44520@pit.databus.com>
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Barney Wolff wrote:
> 
> On Mon, Mar 08, 2004 at 10:26:55AM -0800, Rick Jones wrote:
> >
> > Pinky TOE => Checksum Offload (CKO)
> > Middle TOE => CKO + LSO (aka TSO or large send)
> > Big TOE => Full TCP Offload
> 
> You forgot these:
> 
> Stubbed TOE => disabling TOE so tcpdump/snoop works
> Broken TOE  => bug in firmware, difficult to diagnose

Yes, well... if we are going for full humour (actually, I wasn't just trying to
be funny but still... :)

TOE jam => the kruft of a TOE solution
Hammer TOE => one where the offload is to an embedded CPU
TOE nail => when offloaded security/hardening is included
Athlete's Foot => when a TOE becomes security compromised
Tip TOE => when it operates in a stealth mode
Foot => what you have when more than one TOE are bonded together.
TOE shoes => the shims interfacing a TOE with the rest of the system/stack.

I suppose that had development of the Digital 2114X chips continued, one might
have spoken of a Tip TOE through the Tulips.

ad nauseum :)

rick jones
-- 
Wisdom Teeth are impacted, people are affected by the effects of events.
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to post, OR email to raj in cup.hp.com  but NOT BOTH...

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The argument for offload often takes the premise that host CPU is a
bottleneck at high network speeds, which is often true.  If we can
take that as given, then it is easy to capture the effects of
transport offload on throughput using a back-of-napkin analysis.

One issue is the ratio of host speed to NIC speed for protocol
processing.  The benefit from offload is bounded by the inverse of
that ratio.  In the 1980s, the offload NICs took longer than the host
to process the protocol, and the wire was fast enough to saturate
either of them.  In that case, and only in that case, offload may
actually reduce throughput rather than increase it.

Today, technology improvements and the role of offload in copy
reduction (e.g., RDMA) help to offset this concern.  Jeff Mogul had a
nice paper in the last HotOS called 'TCP offload is a dumb idea whose
time has come' dealing with this point.

One can also draw some interesting conclusions about the maximum
benefit of offload as a function of the ratio of host speed to wire
speed, and of the compute/communicate ratio of the application.
Crudely, the maximum speedup from offload is bounded by the inverse of
both ratios.

Much of the argument about whether offload makes sense boils down to
disagreement about how trends are affecting these various ratios, or
confusion about how these ratios affect potential speedup.

My student Piyush Shivam and I wrote up a 6-page paper with a simple
analysis for the NICELI workshop at SIGCOMM last year, called "On the
Elusive Benefits of Protocol Offload".  It is just Amdahl's Law and
some high-school math.  If you are interested, you can find it at:

http://issg.cs.duke.edu/publications/niceli03.pdf

Regards,
Jeff

http://www.cs.duke.edu/~chase

Craig wrote:
>>Well, the interesting thing is we went down this path in the early 1980s
>>and found that TOEs didn't work.
>>
>>If I try to distill all that we learned in the 1980s into one question, I
>>come out with:
>>
>>    In the 1980s we discovered that communicating with a TOE over a bus
>>    about a TCP connection was as expensive or more expensive than
>>    simply handling the TCP connection in the main processor.  What about
>>    today's TOE designs makes them different?
 >
 >...



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Thread-Topic: [e2e] Re: Are you interested in TOEs and related issues
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From: "Michael Christian" <mchristi@yahoo-inc.com>
To: "stanislav shalunov" <shalunov@internet2.edu>,
        "Sunay Tripathi" <Sunay.Tripathi@eng.sun.com>
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>One would hope TCP design would not be guided by Oracle licensing
quirks.

>> 2) On low end (1-2 CPU) x86 based machines, cost of adding a
processor
>>    is much higher than adding a TOE (I personally haven't verified
this).

>It's not obvious why this should necessarily be the case, given that
>it is likely that there will continue to be quite a bit more
>general-purpose CPUs made than TCP offload engines.

Well, current pricing, with a *huge* negotiated discount, when going
from a 2-proc to 4-proc box is on the order of $20+ grand.  Yes, you do
need to include the hardware, the OS (you need Linux AS vs ES which is
substantially more $$$), and potentially the Oracle licenses (which I
haven't included in the number).

Or, I could buy a $400 TOE.=20

Yeah, CPU's are cheap in the 1-2 range, but when you are TOEing the line
between a 2-proc and a 4-proc, the difference is huge.

-F

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From: Luigi Rizzo <rizzo@icir.org>
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On Thu, Mar 04, 2004 at 11:11:36PM -0800, H.K. Jerry Chu wrote:
...
> Bursty traffic may be ok for LAN, but is considered evil for WAN.
> TOE doesn't suffer this problem (since it is stateful), and may be
> a necessary evil until jumbo frames become universal.

actually the interesting observation that this discussion raises
is that the frame size/link MTU should be picked large enough to make the
end-to-end pps rate bounded (but not too low) for well behaved flows.

In 1980 the ethernet MTU allowed some 800pps -- going much higher makes
little sense anyways, because even if the end nodes can cope with
higher rates, the control loops cannot not react fast enough
(you have to factor in propagation delays too).

On the negative side, however, is that that very large MTUs require
a lot of buffering to be preallocated on the receive side, and
possibly even at the routers (at least, those without hw-assisted
forwarding planes).

I don't have a good answer but going much higher than 16Kbytes MTUs
seems unlikely... and at 10Gig this is still close to 100kpps.

	cheers
	luigi

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From: "H.K. Jerry Chu" <Jerry.Chu@eng.sun.com>
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...
<snip>

>> It's mainly the TCP receive window, not the PMTU that dictates how
>> much buffering will be needed. Routers that do cut-through need not
>> buffer the whole pkt. More significant is the drop of the DRAM price
>> and increase of the DRAM size making memory a non-issue in many cases
>> today.
>
>
>If the DRAM price alone were the only factor preventing vendors from 
>implementing unlimited packet queues, we would already have seen tons 
>of such devices on the market over the past 10 years or so. Excessive 
>queuing delays on routers are bad, especially for high-speed traffic 
>managed by TCP-like congestion control schemes. The queue lengths on 
>routers have always been and will always present a tradeoff between a 
>smallest usable window for handling traffic bursts, and the desire of 
>keeping the queueing delays as low as possible on the other hand.

Assuming pkt switching speed has scaled up too, I don't see why today's
much faster routers with a larger max queue length will cause the
average queuing delay to be any longer than routers 10 years ago.

A perhaps oversimplified comparison can be made between a conforming
traffic using 9K jumbo frames vs a non-conforming, traffic burst of 13
back-to-back regular Ethernet frames. They both require the same amount
of memory bandwidth in the forwarding path (2*9K ~= 13*1500). If all
the forwarding optimization in today's routers has made the switching
speed of 13 consecutive Ethernet "cells" from the same flow similar to
2 back-to-back jumbo frames, why would the former be any more evil than
the latter?

>
>
>>
>> Over the past decade many components involved in providing high-speed
>> networking have scaled up an order of magnitude. This including link
>> bandwidth, CPU speed, I/O bus, memory size..., but not the Ethernet
>> MTU and certain TCP parameters (such as the every-other-pkt acking
>> policy). This is really hurting the throughput performance of the
>> hosts. IMHO the amount of burstiness by TCP over WAN should be
>> allowed to scale up an order of magnitude too. If stretch ACKs are
>> fully adopted into TCP algorithm (see rfc2525 for a number of issues
>> with stretch acks), one can use LSO on the transmit side, and
>> per-flow pkt coalescing on the receive side to provide effectively a
>> simple, stateless AAL5 layer for the Ethernet "cells" without
>> requiring jumbo frames or complex TOE engine.
>>
>> BTW, I asked a few transport folks in Minneapolis IETF about how
>> "evil" is traffic burst in today's enviroment, but did not get any
>> concrete answer. Perhaps this topic should be discussed in tsvwg or
>> tcpm.
>
>
>Because queues in todays routers have finite maximum lengths, and this 
>model is unlikely to change in the forseeable future, excessive traffic 
>bursts will be more likely subject to drop-tail policing than other 
>kinds of more smoothly shaped traffic. More than that, the bursty 
>traffic will not only have less chance of reaching its target with all 
>fragments in place, but it will also most probably do much harm to

Note that these are not IP fragments. Dropping one of them from a burst
is no different from dropping one from any other random flow.

Also don't forget the end2end path covers two endpoints on the host side
so we must consider host side requirement too. This looks like a tug of
war between the host side favoring large burst vs the network side
favoring small burst. Perhaps TOE can be a good bridge between the two.

Rgds,

Jerry

Also 
>other innocent and otherwise well-behaving flows as well.
>
>Marko
>
>
>>
>> Jerry
>>
>> Sr. Staff Engineer
>> Solaris Networking & Security Technology
>> Sun Microsystems, Inc.
>>
>> >I don't have a good answer but going much higher than 16Kbytes MTUs
>> >seems unlikely... and at 10Gig this is still close to 100kpps.
>> >
>> >	cheers
>> >	luigi
>


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To: "H.K. Jerry Chu" <Jerry.Chu@eng.sun.com>
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On Saturday 06 March 2004 17:48, H.K. Jerry Chu wrote:
> ...
> <snip>
>
> >> It's mainly the TCP receive window, not the PMTU that dictates how
> >> much buffering will be needed. Routers that do cut-through need
> >> not buffer the whole pkt. More significant is the drop of the DRAM
> >> price and increase of the DRAM size making memory a non-issue in
> >> many cases today.
> >
> >If the DRAM price alone were the only factor preventing vendors from
> >implementing unlimited packet queues, we would already have seen
> > tons of such devices on the market over the past 10 years or so.
> > Excessive queuing delays on routers are bad, especially for
> > high-speed traffic managed by TCP-like congestion control schemes.
> > The queue lengths on routers have always been and will always
> > present a tradeoff between a smallest usable window for handling
> > traffic bursts, and the desire of keeping the queueing delays as
> > low as possible on the other hand.
>
> Assuming pkt switching speed has scaled up too, I don't see why
> today's much faster routers with a larger max queue length will cause
> the average queuing delay to be any longer than routers 10 years ago.


Congestions and associated queuing delays typically arise due to finite 
line speeds at outgoing network interfaces, which are sometimes offered 
more traffic than they can transmit, so the remaining packets have to 
be either queued or discarded. This phenomenon is well known and is 
largely independent of the switching fabric performance.

So, when such congestions do occur, the max. queuing delay will be 
approximately equal to the max. buffer length (in bits) divided by the 
line card speed. Note that with today's faster line speeds we cannot 
afford to have the similar level of queuing delays as on routers 10 
years ago - those delays have now to be reduced (preferably) 
proportionally to the increase in line rates. And that can't be 
enforced with unlimited buffer lengths.


>
> A perhaps oversimplified comparison can be made between a conforming
> traffic using 9K jumbo frames vs a non-conforming, traffic burst of
> 13 back-to-back regular Ethernet frames. They both require the same
> amount of memory bandwidth in the forwarding path (2*9K ~= 13*1500).
> If all the forwarding optimization in today's routers has made the
> switching speed of 13 consecutive Ethernet "cells" from the same flow
> similar to 2 back-to-back jumbo frames, why would the former be any
> more evil than the latter?


The problem is that in today's routers the buffer sizes are typically 
accounted in packets, not in bytes. So looking at your example, and 
supposing that the line card buffer is limited to 100 packets, a burst 
of small frames will instantly consume 13% of available buffer slots, 
while jumbos will only use 2%. This is of course no problem if the line 
card can transmit all those frames instantly, but what if it cannot?


>
> >> BTW, I asked a few transport folks in Minneapolis IETF about how
> >> "evil" is traffic burst in today's enviroment, but did not get any
> >> concrete answer. Perhaps this topic should be discussed in tsvwg
> >> or tcpm.
> >
> >Because queues in todays routers have finite maximum lengths, and
> > this model is unlikely to change in the forseeable future,
> > excessive traffic bursts will be more likely subject to drop-tail
> > policing than other kinds of more smoothly shaped traffic. More
> > than that, the bursty traffic will not only have less chance of
> > reaching its target with all fragments in place, but it will also
> > most probably do much harm to
>
> Note that these are not IP fragments. Dropping one of them from a
> burst is no different from dropping one from any other random flow.


It _is_ different, because with spiky bursts, you won't be occasionaly 
losing only a packet or two, but you can expect to loose the entire (or 
most of the) burst at once.

Marko


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To: "Craig Partridge" <craig@bbn.com>, "Marko Zec" <zec@tel.fer.hr>
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I think Marko is referring to the situation when there is a congestion.
Because of that the queue builds up and if the queue space is in term
of the number of packets, then a bunch of small packets can potentially
dominate the queue and left little room for large packets eventhough
there is a space in the buffer to accomodate the large ones.

Unless if there is anything I miss.

Cheers,

-----Original Message-----
From: Craig Partridge [mailto:craig@bbn.com]
Sent: Monday, March 08, 2004 11:15 AM
To: Marko Zec
Cc: tcpm@ietf.org; tsvwg@ietf.org; end2end-interest@postel.org
Subject: Re: [Tsvwg] Re: [e2e] Are you interested in TOEs and related
issues=20



In message <200403081646.48030.zec@tel.fer.hr>, Marko Zec writes:

>On Sunday 07 March 2004 23:06, Alexandre L. Grojsgold wrote:

>> Small packets get more buffer frames, but are also more rapidly sent.
>> The gueue grows fats, but shrinks fast too. It means that a burst of
>> small packets will not increase the packet loss, no mather the length
>> of the buffer (in packets).
>
>
>This theory is unfortunately wrong, because it ignores the fact that=20
>when congestions do occur, traffic flows consisting of small packets=20
>(regardless whether they are bursty or smooth) will use more buffer=20
>slots then the flows carrying the larger ones, leaving less remaining=20
>buffer slots available for queuing further packets / bursts.

I must be missing something here.  I think Alexandre's generally right.
If the buffers load and empty according to the amount of data in them,
then packet size isn't an issue.

The only situations in which this isn't true, that I can think off the
top of my head, are:

    * If the router has places where it moves all the bytes in the =
packet
      buffer, regardless of how much data the buffer actually contains.
      Since this would cause extraordinary internal bandwidth =
challenges,
      I doubt anyone does it for any but the lowest speed routers.

    * If the router has a deep pipeline and the time at each stage in =
the
      pipeline is set to equal the average packet time rather than the
      minimum packet time.  But most folks carefully design router =
pipelines
      for minimum packet times precisely because of this risk.

So clue me in -- what detail of router design am I missing?

Craig


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In message <510B9F0C-6DAB-11D8-AE57-000A95D7D4C0@cray.com>, David Borman writes
:

>I wouldn't say in general that we found out that TOEs didn't work.  I 
>think a more qualified comment of what we found out was that in an 
>environment where the network is the bottle neck, TOEs did not provide 
>any cost effective benefits; and as CPUs got faster, the TOEs just 
>provided costs without benefits. :-)

I take a stronger view, but also agree we need to clarify.

>TOE has become a buzz-word that needs to be qualified.  Some folks 
>include Checksum Offload and Large Segmentation Offload when talking 
>about TOE.  When others talk about TOE, they refer to offloading the 
>whole TCP/IP stack.

I understood TOE as being offloading just about the entire stack.

I would agree that experience shows that certain functions can be offfloaded
very effectively, including checksums and some memory management.

I also agree with your thought question -- namely which functions to offload
in a high performance environment -- is the right one.

Craig

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To: Marko Zec <zec@tel.fer.hr>
Cc: tcpm@ietf.org, tsvwg@ietf.org, end2end-interest@postel.org
In-Reply-To: Your message of "Mon, 08 Mar 2004 16:46:47 +0100."
             <200403081646.48030.zec@tel.fer.hr> 
Date: Mon, 08 Mar 2004 11:15:05 -0500
From: Craig Partridge <craig@bbn.com>
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Subject: [tcpm] Re: [Tsvwg] Re: [e2e] Are you interested in TOEs and related issues
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In message <200403081646.48030.zec@tel.fer.hr>, Marko Zec writes:

>On Sunday 07 March 2004 23:06, Alexandre L. Grojsgold wrote:

>> Small packets get more buffer frames, but are also more rapidly sent.
>> The gueue grows fats, but shrinks fast too. It means that a burst of
>> small packets will not increase the packet loss, no mather the length
>> of the buffer (in packets).
>
>
>This theory is unfortunately wrong, because it ignores the fact that 
>when congestions do occur, traffic flows consisting of small packets 
>(regardless whether they are bursty or smooth) will use more buffer 
>slots then the flows carrying the larger ones, leaving less remaining 
>buffer slots available for queuing further packets / bursts.

I must be missing something here.  I think Alexandre's generally right.
If the buffers load and empty according to the amount of data in them,
then packet size isn't an issue.

The only situations in which this isn't true, that I can think off the
top of my head, are:

    * If the router has places where it moves all the bytes in the packet
      buffer, regardless of how much data the buffer actually contains.
      Since this would cause extraordinary internal bandwidth challenges,
      I doubt anyone does it for any but the lowest speed routers.

    * If the router has a deep pipeline and the time at each stage in the
      pipeline is set to equal the average packet time rather than the
      minimum packet time.  But most folks carefully design router pipelines
      for minimum packet times precisely because of this risk.

So clue me in -- what detail of router design am I missing?

Craig

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From: Pekka Savola <pekkas@netcore.fi>
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Subject: [tcpm] TCP, multiple addresses and soft errors when connecting
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Hi all (mainly the TCP guys!),

v6ops WG is in the process of documenting issues which are caused or 
made more severe by the introduction of IPv6.  The document is 
draft-ietf-v6ops-v6onbydefault-01.txt

TCP timeouts when connecting to a destination, but when the 
destination is unreachable (e.g., because you don't have global IPv6 
routes, your first-hop router is not operational, the packets are sent 
on-link by default, etc.), TCP does not abort the connectivity (and 
try the next address quicker -- instead of waiting for the TCP 
timeout).

The critical observation to make here is as IPv6 addresses are tried
first (I'm simplifying; it's more complex than that), one must cycle
through all the addresses, before trying with IPv4.  However, IPv4
connections might still work! With TCP timeouts, this takes a lot of
time --- and should be made quicker.

There is going to be a WG last call on the document soon, aimed for 
Informational -- just for documenting the issues.  The actual fixes 
(or not) would have to be done elsewhere (e.g., in TCPM WG ;-).

Please provide feedback especially on the language used for TCP.  
(Below.)

draft-ietf-v6ops-v6onbydefault-01.txt:

==========
2.3.1 TCP Implications

   In the case of a socket application attempting a connection via TCP,
   it would be unreasonable for the application to block even after the
   system has received notification that the destination address is
   unreachable via an ICMPv6 Destination Unreachable message.

   Following are some ways of solving TCP related delays associated with
   destination unreachability when ICMPv6 errors are generated.

2.3.1.1 TCP Connection Termination

   One solution is for TCP to abort connections in SYN-SENT or
   SYN-RECEIVED state when it receives an ICMPv6 Destination Unreachable
   message.

   [HOSTREQS] document, in section 4.2.3.9., states that TCP MUST NOT
   abort connections when receiving ICMP Destination Unreachable
   messages that indicate "soft errors", where soft errors are defined
   as ICMP codes 0 (network unreachable), 1 (host unreachable), and 5 
   (source route failed), and SHOULD abort connections upon receiving 
   the other codes (which are considered "hard errors").  ICMPv6 didn't
   exist when that document was written, but one could extrapolate the 
   concept of soft errors to ICMPv6 Type 1 codes 0 (no route to
   destination) and 3 (address unreachable), and hard errors to the
   other codes. Thus, it could be argued that a TCP implementation that
   behaves as suggested in this section is in conflict with [HOSTREQS].

   When [HOSTREQS] was written, most applications would mostly only try
   one address when establishing communication with a destination.  Not
   aborting a connection was a sane thing to do if re-trying a single
   address was a better alternative over quitting the application
   altogether. With IPv6, and especially on dual stack systems,
   destinations are often assigned multiple addresses (at least one IPv4
   and one IPv6 address), and applications iterate through destination  
   addresses when attempting connections.

   Since soft errors conditions are those that would entice an
   application to continue iterating to another address, TCP shouldn't 
   make the distinction between ICMPv6 soft errors and hard errors when
   in SYN-SENT or SYN-RECEIVED state.  It should abort a connection in
   those states when receiving any ICMPv6 Destination Unreachable
   message.  When in any other state, TCP would behave as described in 
   [HOSTREQS].

   Many TCP implementations already behave this way, but others do not.
   This should be noted as a best current practice in this case.

   A tangential method of handling the problem in this way would be for
   applications to somehow notify the TCP layer of their preference in
   the matter.  An application could notify TCP that it should abort a
   connection upon receipt of particular ICMPv6 errors.  Similarly, it 
   could notify TCP that it should not abort a connection.  This would 
   allow existing TCP implementations to maintain their status quo at 
   the expense of increased application complexity.

2.3.1.2 Asynchronous Application Notification

   In section 4.2.4.1, [HOSTREQS] states that there MUST be a mechanism
   for reporting soft TCP error conditions to the application. Such a  
   mechanism (assuming one is implemented) could be used by applications
   to cycle through destination addresses.
===========

-- 
Pekka Savola                 "You each name yourselves king, yet the
Netcore Oy                    kingdom bleeds."
Systems. Networks. Security. -- George R.R. Martin: A Clash of Kings



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