From owner-pilc@lerc.nasa.gov  Wed Mar  1 09:26:16 2000
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More clarification...

Joe Touch wrote:

> > From owner-pilc@lerc.nasa.gov  Mon Feb 28 08:05:06 2000
> > Date: Mon, 28 Feb 2000 15:06:57 +0100
> > From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>
> > CC: "'pilc@grc.nasa.gov'" <pilc@grc.nasa.gov>,
> >         "Phil Karn (E-mail)" <karn@qualcomm.com>
> > Subject: Re: Draft on Advice for Internet Subnetwork Designers
> >
> > Here are some comments on Link Level Issues:
> >
> > "Falk, Aaron" wrote:
> >
> > > Folks--
> > >
> > > A very important piece of work for PILC is the LINK draft that Phil Karn is
> > > editing.
> > > http://www.ietf.org/internet-drafts/draft-ietf-pilc-link-design-01.txt
> >
> > PAGE 4: MTU Size
> >
> > It may be useful to note that using IPSEC in tunnel mode can increase
> > the size of a packet as it travels across the network.  This can result
> > in a smaller MTU.
>
> Doesn't IPSEC in any mode do this? (tunnel or transport?)
>
> (provided it's added after the source - and, in the case
> of overlay networks, [paradoxically] transport may be
> more useful)

Sure, you are correct. But, I seem to remember the draft mentioning 1500B MTU.
Now, since this is a link designer's draft, what we are presumably interested in
is frame payload data size... So, if normal IP is used that is maximum 1500B.
If IPSEC is used end-to-end that is still 1500B of frame payload, even though
MTU is reduced. But with tunnel mode along part of the path, the rest of the
path outside the tunnel now sees a reduced MTU - or the tunneled path needs
to support a larger frame size.  Just saying 1500B is not a universal MTU anymore.

(P.S. GRE encapsulation, etc also have this effect).

>
> > The IPv4 packet header contains a header checksum that verifies the integrity
> > of the header. This checksum was originally design to protect from router
> > software errors, however this function is no longer necessary.
>
> Not so much necessary, as not worth the cost.
>

Better wording.


>
> > IPv6 does
> > not therefore implement a network level integrity checksum. The role of the
> > IP checksum has therefore been subsumed into the TCP and UDP
> > checksums.
>
> More into IPSEC than into TCP and UDP, no?
>

IPSEC is one way, it is not (I think) mandatory, so TCP/UDP may still get
bad packets which they have to reject....

> Joe



From owner-pilc@lerc.nasa.gov  Wed Mar  1 09:46:49 2000
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I'm following up the thread on ARQ, sorry for the delay in getting back:

Lloyd Wood wrote:

> On Mon, 28 Feb 2000, Dr G Fairhurst wrote:
>
> > "Falk, Aaron" wrote:
> >
> > > Folks--
> > >
> > > A very important piece of work for PILC is the LINK draft that Phil Karn is
> > > editing.
> > > http://www.ietf.org/internet-drafts/draft-ietf-pilc-link-design-01.txt
> >

> > (2)  Link layer protocols should provide adequate protection from
> > transmission errors.
>
> It's a tradeoff dependent on the characteristics and use of the link,
> since the end-to-end checksum catches a number of errors. If the
> reader thinks 'adequate protection' means 100% reliability means ARQ,
> you've got problems.

Adequate protection form errors, in my thinking,  means that PDUs delivered
end-to-end should not contain minimal errors.  The UDP/TCP validation
using the checksum is designed to detect computer manipulation errors
(e.g. DMA under-runs, pointer corruption, etc), not to detect random
(or burst) channel errors.  A more powerful CRC is required in this case.
This is a different issue to non-delivery. Point 3 below.


> How about:
>
>  Link layer protocols should provide adequate protection from
>  transmission errors. This is necessary since:
>  1.  there is as yet no internetwork framework for detecting or
>      handling link transmission errors effectively
>  2.  it is desirable from a performance viewpoint to avoid triggering
>      congestion control algorithms needlessly as result of errored
>      link-layer frames being discarded.
>  3.  the TCP and UDP checksums are neither strong nor foolproof.
>
> >  Whereas in previous cases, small CRCs or no CRC
> > (in the case of SLIP) have been tolerated.  A CRC-32 is desirable for
> > each packet (as in Ethernet).

>
> ...or as in ATM AAL5. The best choice of CRC might be dependent on
> advertised MTU and link characteristics; probably need a note on
> transparent fragmentation here. I can't think of any references as to
> why CRC-32 is considered adequate/sufficient offhand, but some would
> be nice.

In fact we have two ways (which may be used together).  The first is add a CRC per

segment (frame) as X.25 did. The second is add a CRC per PDU as ATM did.

The reference to CRC evaluation for ATM is

"Stone, Jonathon, Greenwood, Michael, and Partridge, Craig, Performance of CRC's
over Real Data. IEEE/ACM Transactions on Networking, 1998. 6(5): p. 529-543"

A possible reference for the thought behind per-PDU CRC rather than per segment
CRC is in:

"Wang, Zheng and Crowcroft, Jon, SEAL Detects Cell Misordering. IEEE Network,
1992. 6(4): p. 8-9"

On points 1,2, Reliable or Partial Reliable delivery implies the use of ARQ,
bit level (physical) FEC, Packet level FEC, or some hybrid of these.

>
> > I think it may be wise to note that there are two approaches to TCP
> > friendly link ARQ.  One has been put forward by Balakrishnan:
> >
> > "H. Balakrishnan, V.N. Padmanabhan, S Seshan, and R H Katz, A
> > Comparison of Mechanisms for Improving TCP Performance over
> > Wireless Links. IEEE Trans Net, 1997. 5(6): p. 756-759."
> >
> > In this case, TCP headers are read and the information used to tune
> > the link protocol.
>
> this breaks layering. Ugh. And it's a fat lot of use with e.g. IPSec.
> Not my first or recommended choice.

I'm not sure I like it, but we should be aware there are many ways of doing it,
ranging
from "looking" at headers, which is the most benign. To delaying packets,
modifying
headers and inserting extra packets, which are more pro-active.  People do this
kind
of thing, and given the right technique for the correctly identified problem may
get
big benefits.  There are problems if TCP changes, IPSEC is used (could we need a
proxy?), ECN is deployed, etc, etc.


I agree there is a warning attached here. I don't like going down this path either
:-)

> > The other approach is to bound link behaviour to match TCP
> > "requirements" i.e. A receiver may calculate the variation
> > of the recovery delay and then give up, if the variation
> > which will be introduced is more than that can be tolerated
> > by the TCP transport layer.
>
> and this assumes that the link in question is the only link in the
> path doing such calculations. ugh again.
>

To get best performance this is correct. The penalty for doing too much
retransmission is poorer link performance.  There are far fewer impacts
on TCP end-to-end behaviour.

>
> Really, you should imo be saying that if the variation is any more
> than an order of magnitude less than the limit of transparency to TCP,
> give up on the resend requests. If it's likely to be regularly more
> than an order of magnitude less than the transparency limit, be very
> skeptical of ARQ.

What do you mean by "limit of transparency"?

>
> (500ms RTO gives 50ms per-link limit, which is probably a bit on the
>  high side, given that the draft talks about 10-20ms delay already.
>  But two orders of magnitude would be 5ms. I have to stop myself
>  thinking in binary magnitudes too...)
>
> cheers,
>
> L.
>
> <L.Wood@surrey.ac.uk>PGP<http://www.ee.surrey.ac.uk/Personal/L.Wood/>

Best wishes,

Gorry Fairhurst.



From owner-pilc@lerc.nasa.gov  Wed Mar  1 14:14:27 2000
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From: Joe Touch <touch@ISI.EDU>
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> From gorry@erg.abdn.ac.uk  Wed Mar  1 05:12:19 2000
> Date: Wed, 01 Mar 2000 13:30:30 +0100
> From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>
> To: touch <touch@ISI.EDU>
> CC: pilc@grc.nasa.gov
> Subject: Re: Draft on Advice for Internet Subnetwork Designers
> 
> More clarification...
> 
> Joe Touch wrote:
>
> > > "Falk, Aaron" wrote:
> > >
> > > PAGE 4: MTU Size
> > >
> > > It may be useful to note that using IPSEC in tunnel mode can increase
> > > the size of a packet as it travels across the network.  This can result
> > > in a smaller MTU.
> >
> > Doesn't IPSEC in any mode do this? (tunnel or transport?)
> >
> > (provided it's added after the source - and, in the case
> > of overlay networks, [paradoxically] transport may be
> > more useful)
> 
> Sure, you are correct. But, I seem to remember the draft mentioning 1500B MTU.
> Now, since this is a link designer's draft, what we are presumably interested in
> is frame payload data size... So, if normal IP is used that is maximum 1500B.
> If IPSEC is used end-to-end that is still 1500B of frame payload, even though
> MTU is reduced. But with tunnel mode along part of the path, the rest of the
> path outside the tunnel now sees a reduced MTU - or the tunneled path needs
> to support a larger frame size.  Just saying 1500B is not a
> universal MTU anymore.

It doesn't matter where IPSEC is used - E2E or tunnel. Either
way eats into the MTU. The PILC doc is talking about link MTUs;
everything except the link header consumes that.

> > > IPv6 does
> > > not therefore implement a network level integrity checksum. The role of the
> > > IP checksum has therefore been subsumed into the TCP and UDP
> > > checksums.
> >
> > More into IPSEC than into TCP and UDP, no?
> 
> IPSEC is one way, it is not (I think) mandatory, so TCP/UDP may still get
> bad packets which they have to reject....

Support for IPSEC is mandatory in IPv6, which is where the IP checksum
has been removed. Its _use_ is not mandatory. 

The role of the IP checksum was to check the IP header. The role of
the TCP/UDP checksums is to check the data. If you want to protect/sum
the header, you should (and thus, above, can) use IPSEC.

Joe


From owner-pilc@lerc.nasa.gov  Wed Mar  1 14:42:09 2000
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On Wed, 1 Mar 2000, Joe Touch wrote:
> Gorry wrote:
> > Joe Touch wrote:
> > > Gorry wrote:
> > > > IPv6 does
> > > > not therefore implement a network level integrity checksum. The role of the
> > > > IP checksum has therefore been subsumed into the TCP and UDP
> > > > checksums.
> > >
> > > More into IPSEC than into TCP and UDP, no?

No.


> > IPSEC is one way, it is not (I think) mandatory, so TCP/UDP may still get
> > bad packets which they have to reject....

Yes. Of course, since the TCP/UDP checksum is weak, there's a chance
that a bad packet may slip through.


> Support for IPSEC is mandatory in IPv6

oh, whoopee. This Will Ensure IPSEC's Success.


> which is where the IP checksum
> has been removed. Its _use_ is not mandatory. 
> 
> The role of the IP checksum was to check the IP header. The role of
> the TCP/UDP checksums is to check the data.

Sorry, no. The TCP/UDP checksums still check the integrity of
significant parts of the IP header, even with IPv6. Remember the
"pseudo-header"?

Read RFC2460 section 8, page 26.

It is this check which adequately makes up for the IP header checksum
(which would have been a pain in IPv6, what with extension headers).


> If you want to protect/sum
> the header, you should (and thus, above, can) use IPSEC.

Sorry, no. Use of IPSec is not necessary. Useful header integrity
checks come with the standard UDP/TCP pseudo-header check in IPv6, and
including pseudo-headers in checksums is an expected feature of any
IPv6 transport protocol.

L.

personal doubt is a powerful and accurate weapon.

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






From owner-pilc@lerc.nasa.gov  Wed Mar  1 14:59:17 2000
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From: Joe Touch <touch@ISI.EDU>
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Subject: Re: Draft on Advice for Internet Subnetwork Designers
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> From l.wood@eim.surrey.ac.uk  Wed Mar  1 10:46:34 2000
> Date: Wed, 1 Mar 2000 18:46:28 +0000 (GMT)
> From: Lloyd Wood <l.wood@eim.surrey.ac.uk>
> To: Joe Touch <touch@ISI.EDU>
> cc: gorry@erg.abdn.ac.uk, pilc@grc.nasa.gov
> Subject: Re: Draft on Advice for Internet Subnetwork Designers
> X-no-archive: yes
> 
> On Wed, 1 Mar 2000, Joe Touch wrote:
> > Gorry wrote:
> > > Joe Touch wrote:
> > > > Gorry wrote:
> > > > > IPv6 does
> > > > > not therefore implement a network level integrity checksum. The role of the
> > > > > IP checksum has therefore been subsumed into the TCP and UDP
> > > > > checksums.
> > > >
> > > > More into IPSEC than into TCP and UDP, no?
> 
> No.
> 
> > > IPSEC is one way, it is not (I think) mandatory, so TCP/UDP may still get
> > > bad packets which they have to reject....
> 
> Yes. Of course, since the TCP/UDP checksum is weak, there's a chance
> that a bad packet may slip through.
> 
> 
...
> > The role of the IP checksum was to check the IP header. The role of
> > the TCP/UDP checksums is to check the data.
> 
> Sorry, no. The TCP/UDP checksums still check the integrity of
> significant parts of the IP header, even with IPv6. Remember the
> "pseudo-header"?

It certainly checks the pseudo header. Not the header.
There are components of the header not in the pseudoheader
(flags, options, etc.)

> > If you want to protect/sum
> > the header, you should (and thus, above, can) use IPSEC.
> 
> Sorry, no. Use of IPSec is not necessary. Useful header integrity
> checks come with the standard UDP/TCP pseudo-header check in IPv6, and
> including pseudo-headers in checksums is an expected feature of any
> IPv6 transport protocol.

Not if you are concerned about the integrity of the IP header -
all (or at least the end-to-end part, including flags and E2E options)
of it.

Joe


From owner-pilc@lerc.nasa.gov  Wed Mar  1 15:47:08 2000
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On Wed, 1 Mar 2000, Joe Touch wrote:

> > > If you want to protect/sum
> > > the header, you should (and thus, above, can) use IPSEC.
> > 
> > Sorry, no. Use of IPSec is not necessary. Useful header integrity
> > checks come with the standard UDP/TCP pseudo-header check in IPv6, and
> > including pseudo-headers in checksums is an expected feature of any
> > IPv6 transport protocol.
> 
> Not if you are concerned about the integrity of the IP header -
> all (or at least the end-to-end part, including flags and E2E options)
> of it.

If you're using AH or ESP in tunnel mode, you'll get all the IP
header. Otherwise, yes, AH in transport mode gives a check on
everything bar mutable fields, while ESP is useless in this regard.

Header checksumming isn't an intended explicit design feature of
IPSec, surely? More a side-effect of authentication...

[http://www.counterpane.com/ipsec.html questions header verification,
AH and transport mode entirely, and points out problems with AH header
authentication.]

Doesn't the complexity for IPSec-based header protection more than
outweigh the incremental value of any gains over the pseudo-header
check?

L.

I'm pretty sure that header checksumming is _not_ what IPSec is _for_.
What is IPSec for?

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




From owner-pilc@lerc.nasa.gov  Wed Mar  1 18:09:31 2000
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Subject: Re: Draft on Advice for Internet Subnetwork Designers 
In-reply-to: Your message of "Thu, 24 Feb 2000 12:02:08 EST."
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Comments:

1) Shortly before the Washington meeting, I sent a suggestion to change the 
Introduction/Overview section as follows.  

> ...seminal paper "End-to-End Arguments in System Design" [SRC81]. This paper 
> argued that many -- if not most -- network functions are best implemented on 
> an end-to-end basis, i.e., at the higher protocol layers.  Duplicating these 
> functions at the lower levels can be needlessly redundant or harmful. 
> However,  there are also cases where duplication can be justified by 
> considerations such  as performance, availability or security.  Examples 
> might include link layer  retransmission on an unusually lossy channel, e.g., 
> mobile radio, encryption  of the entire link layer protocol over a shared 
> medium, or  redundant paths at the link layer for fault recovery.  
> Duplication of functions should be done only based on an understanding of 
> system level implications, including any side-effects on the higher layers.

I still believe rather strongly that this is more appropriate than the current 
wording.  Since it has been quite a while since the last discussion of the 
document on the mailing list,  a reminder seems appropriate.

2) Some initial thoughts on what we should do with the empty section on 
buffering, congestion and flow control.  Let's discuss on the list, and I'll 
draft text after the discussion:
a) [Following owes a great deal to a side-discussion I had with Mark Allman] 
TCP needs lots of buffering.  An ability to buffer bursts of up to (fixed 
delay)*(bottleneck rate) per TCP connection in each node is ideal if 
impractical.  Problem is that neither the fixed delay nor the bottleneck rate 
is necessarily known at any node.  A good approximation is fixed delay of the 
link * link bandwidth.  For links that have variable delays, the approximation 
should be based on the max.  Also, the maximum number of TCP connections per 
node is very difficult to determine a priori, and is not subject to any kind 
of admission control (except perhaps at a PEP).  A SWAG has to suffice.  Some 
kind of mechanism to measure delay on the link can be useful in autotuning 
buffer sizes.  In general, more buffers are better than fewer.  For example, 
TCP performance in very early ATM switch implementations suffered horribly 
from the small buffer sizes.

b) There is a rough consensus (not unanimous) in the research community that 
active queue management is important to improving fairness, link utilization 
and throughput. Reference RFC 2309.  Followed by a brief introduction to 
active queue management, with RED as an example. An important part of the 
concept is that active queue management attempts to keep average queue depths 
short, while accommodating short term bursts.  It may be advisable for subnets 
that perform queueing and discarding to consider implementing some form of 
active queue management, especially if large aggregates of TCP connections are 
likely to share the same queue.  It may be less effective in the case, for 
example, of an ATM switch that implements per-VC queueuing.  Note, 
incidentally, that active queue management algorithms in general are highly 
sensitive to parameter settings and reference Borden and  Fioriu paper.  

c) If the subnet does fragmentation and reassembly, make sure that if it 
becomes necessary to discard, then discard entire frames, not just scattered 
fragments. For ATM, this specifically means using EPD/PPD, and reference ATM 
Forum TM 4.0 specification.   Reference Romanow paper. 

d) If the subnet includes flow control that shapes the rate at the edge of the 
subnet to match a share of the bottleneck bandwidth in the subnet (e.g., ATM 
ABR), this has the effect of pushing congestion to the edge of the subnet.  In 
such a case, adequate buffering is needed at the edge of the subnet (see a 
above).  Whether or not there is benefit in this kind of flow control is 
controversial, and there have been simulation studies that go both ways.  
Anecdotally, some large networks have used ABR sucessfully to carry IP 
traffic, but no published results.

e) Flow control mechanisms that mark but do not shape frames in excess of fair 
share of the bottleneck rate (e.g., frame relay FECN and BECN) are at best 
useful only in the context of IP ECN (RFC 2481).  New subnets would do well to 
have flow control mechanisms that have that semantics (i.e., feedback signal 
based on EWMA buffer size, feedback set probabilistically proportional to 
difference between two thresholds, or always set when second threshold 
exceeded) so that feedback could be 'bubbled up' from the subnet layer to the 
IP header.  

3) I have some ideas on QoS, which I'll try to organize and send out over the 
next few days.


  




From owner-pilc@lerc.nasa.gov  Wed Mar  1 19:48:15 2000
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From: Hari Balakrishnan <hari@lcs.mit.edu>
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Subject: PILC ASYM document
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Reviving an old, old thread about draft-ietf-pilc-asym-00.txt (Balakrishnan & 
Padmanabhan):

On Mon, 22 Nov 1999 10:23:46 -0600, Spencer Dawkins wrote:

> Van Jacobson: RFC 1144 header compression, in one-byte mode (as opposed to
> five-six-byte mode), handles 3000:1 asymmetry. Are we seeing connection
> paths that are more asymmetric than this? What's the point?

This comment was from the PILC minutes of the Washington DC IETF (11/10/99).

It has come to our attention from Aaron and Spencer (chairs of the working group) that we had perhaps not clarified what we meant by asymmetry, and that Van's comment above implied that a push-back was needed on this document.

Venkat and I don't believe that this assessment is necessarily correct for the following reasons:

1. The document draft-ietf-pilc-asym-00.txt isn't about bandwidth asymmetry alone; it defines (perhaps inelegantly) "asymmetry" to be all cases when the characteristics of one direction "significantly" impact performance in the other.  It goes on to describe several kinds of interactions, including bandwidth asymmetry (exacerbated in the presence of bi-directional traffic), media-access asymmetry that causes latency variability, error/loss rate asymmetry.  All of these are _real_ and not just of academic interest.

2. If you have a bandwidth-asymmetric network and ACKs aren't going across a loss-prone path, VJ header compression solves the problem.  Completely.  The draft says as much.

3. But there are important, real cases when things aren't quite so simple:

   (i) Media-access.  Many MAC protocols are RTS/CTS based, requiring a synchronization handshake before communication, with each failed RTS triggering an exponential backoff followed by retry.  This avoids unnecessary data collisions that may otherwise result.  Examples include many packet radio networks and the 802.11 spec.  In such networks, it has been observed that latency can be highly variable, and as a result data and ACKs interact badly.  The consequence is that the RTO becomes very large because rttvar is large.  When a timeout occurs, the idle times are prohibitively large.  In such situations, performance would improve a lot more from *infrequent* acks (and of course you need methods to deal with this at the sender, as described in the draft) than from *smaller* (header-compressed) ACKs.  (Infrequent ACKs need to be dealt with without increasing burstiness, and the draft talks about this at length.)

   (ii) Bi-directional traffic.  I didn't think this was such a common occurrence, but I recently received this message from a DSL customer, which I repeat verbatim below.  (The customer recently graduated from MIT.)

--------------------------------------------------------------------
Date: Mon, 31 Jan 2000 11:47:05 -0800 (PST)
From: nelson@monkey.org (Nelson Minar)
To: 6.892-students@MIT.EDU
Subject: assymetry and TCP performance?

I thought I was so smart after taking Hari's class last semester, but I'm having a hard time diagnosing a simple problem.

I'm on PacBell DSL here in California. It's pretty good, 1.5 Mbits/sec downstream and 128 kbits/sec upstream or so. Reliable.

Except one problem - when someone is downloading a file off of my web server, saturating the 128kbits/sec upstream link, then all my performance goes to hell. Pinging the gateway takes 2 seconds round trip, and things like DNS or web page fetches are either terrible performance or just don't work.

I'd expect things to be a bit sluggish, but not anywhere near this bad. Is there anything simple written about assymetry and TCP performance? I have some guesses about what's going on, but I'd like to understand this problem better.

Thanks!
  Nelson
--------------------------------------------------------------------

	One might think that this is an uncommon situation, but if you look at the way the Internet is evolving, nodes serving information from people's homes are becoming increasingly commonplace and are likely to dramatically increase in the next few years.  I predict that situations like Nelson's above will not be rare in the near future, when we start putting all our computers and myriad other devices in our homes on the Internet.

   (iii) Loss rate asymmetry: Some of the techniques (e.g., sender adaptation) described in the draft handle this case as well, which is observed in many real systems including mobile ones where end stations transmit at much lower power and contend for reverse slots whereas the forward channel is from a high-powered base station and is therefore relatively loss-free.

Given these situations, we believe that they are worth clearly documenting in a draft.  I don't believe that these are of purely academic interest.  

We are very interested in your reactions to this.

Thank you!

Hari




From owner-pilc@lerc.nasa.gov  Wed Mar  1 19:55:57 2000
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From: Marie-Jose Montpetit <marie@teledesic.com>
To: "'Dan Grossman'" <dan@dma.isg.mot.com>
Cc: "'pilc@grc.nasa.gov'" <pilc@grc.nasa.gov>,
        "Phil Karn (E-mail)"
	 <karn@qualcomm.com>,
        "Falk, Aaron" <afalk@PanAmSat.com>
Subject: RE: Draft on Advice for Internet Subnetwork Designers 
Date: Wed, 1 Mar 2000 15:50:38 -0800 
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My own comments below.

-----Original Message-----
From: Dan Grossman [mailto:dan@dma.isg.mot.com]
Sent: Wednesday, March 01, 2000 2:17 PM
To: Falk, Aaron
Cc: 'pilc@grc.nasa.gov'; Phil Karn (E-mail)
Subject: Re: Draft on Advice for Internet Subnetwork Designers 


Comments:

1) Shortly before the Washington meeting, I sent a suggestion to change the 
Introduction/Overview section as follows.  

> ...seminal paper "End-to-End Arguments in System Design" [SRC81]. This
paper 
> argued that many -- if not most -- network functions are best implemented
on 
> an end-to-end basis, i.e., at the higher protocol layers.  Duplicating
these 
> functions at the lower levels can be needlessly redundant or harmful. 
> However,  there are also cases where duplication can be justified by 
> considerations such  as performance, availability or security.  Examples 
> might include link layer  retransmission on an unusually lossy channel,
e.g., 
> mobile radio, encryption  of the entire link layer protocol over a shared 
> medium, or  redundant paths at the link layer for fault recovery.  
> Duplication of functions should be done only based on an understanding of 
> system level implications, including any side-effects on the higher
layers.

I still believe rather strongly that this is more appropriate than the
current 
wording.  Since it has been quite a while since the last discussion of the 
document on the mailing list,  a reminder seems appropriate.

I agree. Strangely enough I was also trying to come up with words for the
introduction as it was apparent from the disucssions in Washington that the
purpose of the LINK draft was not fully understood. While the actual wording
is probably variable from one person to another I think the message should
be clear that the LINK draft is there to help to have a system view hence to
help sub-network designers get a flavor for the end to end aspects of
deploying IP subnets and make sure these subnets integrate in the IP Network
of Networks.


2) Some initial thoughts on what we should do with the empty section on 
buffering, congestion and flow control.  Let's discuss on the list, and I'll

draft text after the discussion:
a) [Following owes a great deal to a side-discussion I had with Mark Allman]

TCP needs lots of buffering.  An ability to buffer bursts of up to (fixed 
delay)*(bottleneck rate) per TCP connection in each node is ideal if 
impractical.  Problem is that neither the fixed delay nor the bottleneck
rate 
is necessarily known at any node.  A good approximation is fixed delay of
the 
link * link bandwidth.  For links that have variable delays, the
approximation 
should be based on the max.  Also, the maximum number of TCP connections per

node is very difficult to determine a priori, and is not subject to any kind

of admission control (except perhaps at a PEP).  A SWAG has to suffice.
Some 
kind of mechanism to measure delay on the link can be useful in autotuning 
buffer sizes.  In general, more buffers are better than fewer.  For example,

TCP performance in very early ATM switch implementations suffered horribly 
from the small buffer sizes.

Yes but if I remember this is also when the traffic statistics were
basically ill undestood. While I do agree that more buffering is better than
less, I would also raise the issue of buffering delays especially if a path
goes through more than one subnet. Then do you have to implement some queue
management at each point?

b) There is a rough consensus (not unanimous) in the research community that

active queue management is important to improving fairness, link utilization

and throughput. Reference RFC 2309.  Followed by a brief introduction to 
active queue management, with RED as an example. An important part of the 
concept is that active queue management attempts to keep average queue
depths 
short, while accommodating short term bursts.  It may be advisable for
subnets 
that perform queueing and discarding to consider implementing some form of 
active queue management, especially if large aggregates of TCP connections
are 
likely to share the same queue.  It may be less effective in the case, for 
example, of an ATM switch that implements per-VC queueuing.  Note, 
incidentally, that active queue management algorithms in general are highly 
sensitive to parameter settings and reference Borden and  Fioriu paper.  

Again I agree. But do my question qould be to which level of detail we want
to go? And is active queue management a link issue?

c) If the subnet does fragmentation and reassembly, make sure that if it 
becomes necessary to discard, then discard entire frames, not just scattered

fragments. For ATM, this specifically means using EPD/PPD, and reference ATM

Forum TM 4.0 specification.   Reference Romanow paper. 

Yes and I think we could even suggest that the subnet data unit (SDU?) have
some markings for this.

d) If the subnet includes flow control that shapes the rate at the edge of
the 
subnet to match a share of the bottleneck bandwidth in the subnet (e.g., ATM

ABR), this has the effect of pushing congestion to the edge of the subnet.
In 
such a case, adequate buffering is needed at the edge of the subnet (see a 
above).  Whether or not there is benefit in this kind of flow control is 
controversial, and there have been simulation studies that go both ways.  
Anecdotally, some large networks have used ABR sucessfully to carry IP 
traffic, but no published results.

This also has links into the subnets that use bandwidth on demand and yes
the buffering is an issue again. I have seen BTW RSVP to ATM results (S.
Giordano) but not ABR.

e) Flow control mechanisms that mark but do not shape frames in excess of
fair 
share of the bottleneck rate (e.g., frame relay FECN and BECN) are at best 
useful only in the context of IP ECN (RFC 2481).  New subnets would do well
to 
have flow control mechanisms that have that semantics (i.e., feedback signal

based on EWMA buffer size, feedback set probabilistically proportional to 
difference between two thresholds, or always set when second threshold 
exceeded) so that feedback could be 'bubbled up' from the subnet layer to
the 
IP header.  

Yes. How about some bubbling up of air interface conditions into control
messages or cost functions?

3) I have some ideas on QoS, which I'll try to organize and send out over
the 
next few days.

Any link will have to address the resource negociations/reservations and
some of the diffserv issues; the active queue management you addressed
already.

Marie-Jose

  




From owner-pilc@lerc.nasa.gov  Wed Mar  1 21:46:06 2000
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> From: Hari Balakrishnan <hari@lcs.mit.edu>

> ...
> 3. But there are important, real cases when things aren't quite so simple:
>
>    (i) Media-access.  Many MAC protocols are RTS/CTS based,
> requiring a synchronization handshake before communication, with
> each failed RTS triggering an exponential backoff followed by retry.
> ...

I don't understand the a priori relevance of such considerations.
TCP is not a MAC protocol, and in general is supposed to be blithely 
ignorant of oddities in the link layer.


> ...
>    (ii) Bi-directional traffic.  I didn't think this was such a
> common occurrence, but I recently received this message from a DSL
> customer, which I repeat verbatim below.  (The customer recently
> graduated from MIT.)

> ...
> Except one problem - when someone is downloading a file off of
> my web server, saturating the 128kbits/sec upstream link, then all
> my performance goes to hell. Pinging the gateway takes 2 seconds
> round trip, ...

The problems there are irrelevant to the draft.  First, no matter what
you do, you can't get more than 128 kbit/sec from a 128 kbit/sec link.
Second, such experiences show that saturating a link with mixtures of
interactive and bulk traffic through with routers that no any notion of
priority queues is a bad idea, which is not exactly a new discovery.

>       One might think that this is an uncommon situation, but if you
> look at the way the Internet is evolving, nodes serving information from
> people's homes are becoming increasingly commonplace and are likely to
> dramatically increase in the next few years.  I predict that situations
> like Nelson's above will not be rare in the near future, when we start
> putting all our computers and myriad other devices in our homes on the
> Internet.

Those problems are not new.  In the good old days, many of us suffered
exactly such situations as Nelson's, except that our links were 10 or 50
times slower.  Please consider ancient asymmetric dial-up modems including
those from (I think) Courier and modems that were sort-of assymetric like
Telebits.


>    (iii) Loss rate asymmetry: Some of the techniques (e.g., sender
> adaptation) described in the draft handle this case as well, which
> is observed in many real systems including mobile ones where end
> stations transmit at much lower power and contend for reverse slots
> whereas the forward channel is from a high-powered base station
> and is therefore relatively loss-free.

That sounds like the old argument that the upper layers must be
modified for every new link layer that comes along, regardless of 
whether how different it really is (think about Telebits).  
Some people reflexively resist that argument.


> Given these situations, we believe that they are worth clearly
> documenting in a draft. ...

Anyone can write about almost anything in an offishal fur shure Internet
Draft, as demonstrated by draft-terrell-ip-spec-ipv7-ipv8-addr-cls-*.txt.
Besides, that draft-ietf-pilc-asym-00.txt exists is not controversial.
The interesting question is what happens to a draft after it has been
published.  How people are supposed to respond to this draft?  Is it
intended to become a Proposed Standard or a BCP?   Or would it become
just another Informational RFC?  Or is it a call to start some experiments?

It seems that draft-ietf-pilc-asym-00.txt has two parts.  One is a
discussion of various problems, and the other advocates several significant
de facto changes to TCP.  It's hard to argue with a discussion of problems,
but I'm not convinced of the wisdom of the advocated changes.  For example,
the document is not convincing about obvious worries about consequences
of reconstructing ACK's, such as distorting the RTT measurement.  (Yes,
I've read the end of section 5.2.2.)

For another example, I'm not convinced by section 5.1.5 that Acks-first
scheduling does not require the gateway to explicitly identify or maintain
state for individual TCP connections, if it wants to avoid re-ordering
TCP segments in a single stream that contain data and those that don't.

Please also note that the draft fails to mention by now ancient experience
with some of its advocated changes to TCP, changes that have not been
enthusiastically adopted.  (Yes, I'm once again for the umpteenth time
pointing at the ACK compression mechanisms used on a bunch of well known
cable modem systems with 10 Mbit/sec downstream rates and 10-30 Kbit/sec
upstream rates.)


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Wed Mar  1 23:27:06 2000
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To: Marie-Jose Montpetit <marie@teledesic.com>
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Subject: Re: Draft on Advice for Internet Subnetwork Designers
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Another comment -- one point that was raised quite a while ago, but remains
overlooked in IP6 is that dependence on TCP/UDP checksums is irrelevant to
routers.  The IP checksum, now absent in default IP6 pkts raises the
possibility of bad router memory creating downstream problems for routers that
cache route paths via address analysis.  Any one router or level 3 switch with
some bad buffer/register mem can wreak havoc in later router decision making.
This seems to be one of a few curious steps backward taken in IP6.

Alex


Marie-Jose Montpetit wrote:
> 
> My own comments below.
> 
> -----Original Message-----
> From: Dan Grossman [mailto:dan@dma.isg.mot.com]
> Sent: Wednesday, March 01, 2000 2:17 PM
> To: Falk, Aaron
> Cc: 'pilc@grc.nasa.gov'; Phil Karn (E-mail)
> Subject: Re: Draft on Advice for Internet Subnetwork Designers
> 
> Comments:
> 
> 1) Shortly before the Washington meeting, I sent a suggestion to change the
> Introduction/Overview section as follows.
> 
> > ...seminal paper "End-to-End Arguments in System Design" [SRC81]. This
> paper
> > argued that many -- if not most -- network functions are best implemented
> on
> > an end-to-end basis, i.e., at the higher protocol layers.  Duplicating
> these
> > functions at the lower levels can be needlessly redundant or harmful.
> > However,  there are also cases where duplication can be justified by
> > considerations such  as performance, availability or security.  Examples
> > might include link layer  retransmission on an unusually lossy channel,
> e.g.,
> > mobile radio, encryption  of the entire link layer protocol over a shared
> > medium, or  redundant paths at the link layer for fault recovery.
> > Duplication of functions should be done only based on an understanding of
> > system level implications, including any side-effects on the higher
> layers.
> 
> I still believe rather strongly that this is more appropriate than the
> current
> wording.  Since it has been quite a while since the last discussion of the
> document on the mailing list,  a reminder seems appropriate.
> 
> I agree. Strangely enough I was also trying to come up with words for the
> introduction as it was apparent from the disucssions in Washington that the
> purpose of the LINK draft was not fully understood. While the actual wording
> is probably variable from one person to another I think the message should
> be clear that the LINK draft is there to help to have a system view hence to
> help sub-network designers get a flavor for the end to end aspects of
> deploying IP subnets and make sure these subnets integrate in the IP Network
> of Networks.
> 
> 2) Some initial thoughts on what we should do with the empty section on
> buffering, congestion and flow control.  Let's discuss on the list, and I'll
> 
> draft text after the discussion:
> a) [Following owes a great deal to a side-discussion I had with Mark Allman]
> 
> TCP needs lots of buffering.  An ability to buffer bursts of up to (fixed
> delay)*(bottleneck rate) per TCP connection in each node is ideal if
> impractical.  Problem is that neither the fixed delay nor the bottleneck
> rate
> is necessarily known at any node.  A good approximation is fixed delay of
> the
> link * link bandwidth.  For links that have variable delays, the
> approximation
> should be based on the max.  Also, the maximum number of TCP connections per
> 
> node is very difficult to determine a priori, and is not subject to any kind
> 
> of admission control (except perhaps at a PEP).  A SWAG has to suffice.
> Some
> kind of mechanism to measure delay on the link can be useful in autotuning
> buffer sizes.  In general, more buffers are better than fewer.  For example,
> 
> TCP performance in very early ATM switch implementations suffered horribly
> from the small buffer sizes.
> 
> Yes but if I remember this is also when the traffic statistics were
> basically ill undestood. While I do agree that more buffering is better than
> less, I would also raise the issue of buffering delays especially if a path
> goes through more than one subnet. Then do you have to implement some queue
> management at each point?
> 
> b) There is a rough consensus (not unanimous) in the research community that
> 
> active queue management is important to improving fairness, link utilization
> 
> and throughput. Reference RFC 2309.  Followed by a brief introduction to
> active queue management, with RED as an example. An important part of the
> concept is that active queue management attempts to keep average queue
> depths
> short, while accommodating short term bursts.  It may be advisable for
> subnets
> that perform queueing and discarding to consider implementing some form of
> active queue management, especially if large aggregates of TCP connections
> are
> likely to share the same queue.  It may be less effective in the case, for
> example, of an ATM switch that implements per-VC queueuing.  Note,
> incidentally, that active queue management algorithms in general are highly
> sensitive to parameter settings and reference Borden and  Fioriu paper.
> 
> Again I agree. But do my question qould be to which level of detail we want
> to go? And is active queue management a link issue?
> 
> c) If the subnet does fragmentation and reassembly, make sure that if it
> becomes necessary to discard, then discard entire frames, not just scattered
> 
> fragments. For ATM, this specifically means using EPD/PPD, and reference ATM
> 
> Forum TM 4.0 specification.   Reference Romanow paper.
> 
> Yes and I think we could even suggest that the subnet data unit (SDU?) have
> some markings for this.
> 
> d) If the subnet includes flow control that shapes the rate at the edge of
> the
> subnet to match a share of the bottleneck bandwidth in the subnet (e.g., ATM
> 
> ABR), this has the effect of pushing congestion to the edge of the subnet.
> In
> such a case, adequate buffering is needed at the edge of the subnet (see a
> above).  Whether or not there is benefit in this kind of flow control is
> controversial, and there have been simulation studies that go both ways.
> Anecdotally, some large networks have used ABR sucessfully to carry IP
> traffic, but no published results.
> 
> This also has links into the subnets that use bandwidth on demand and yes
> the buffering is an issue again. I have seen BTW RSVP to ATM results (S.
> Giordano) but not ABR.
> 
> e) Flow control mechanisms that mark but do not shape frames in excess of
> fair
> share of the bottleneck rate (e.g., frame relay FECN and BECN) are at best
> useful only in the context of IP ECN (RFC 2481).  New subnets would do well
> to
> have flow control mechanisms that have that semantics (i.e., feedback signal
> 
> based on EWMA buffer size, feedback set probabilistically proportional to
> difference between two thresholds, or always set when second threshold
> exceeded) so that feedback could be 'bubbled up' from the subnet layer to
> the
> IP header.
> 
> Yes. How about some bubbling up of air interface conditions into control
> messages or cost functions?
> 
> 3) I have some ideas on QoS, which I'll try to organize and send out over
> the
> next few days.
> 
> Any link will have to address the resource negociations/reservations and
> some of the diffserv issues; the active queue management you addressed
> already.
> 
> Marie-Jose
>


From owner-pilc@lerc.nasa.gov  Wed Mar  1 23:38:58 2000
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To: Vernon Schryver <vjs@calcite.rhyolite.com>
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Subject: Re: PILC ASYM document 
In-reply-to: Your message of "Wed, 01 Mar 2000 18:55:15 MST"
             <200003020155.SAA12044@calcite.rhyolite.com> 
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Dear Vernon,

On Wed, 01 Mar 2000 18:55:15 MST, you wrote:

> > From: Hari Balakrishnan <hari@lcs.mit.edu>
> 
> > ...
> > 3. But there are important, real cases when things aren't quite so simple:
> >
> >    (i) Media-access.  Many MAC protocols are RTS/CTS based,
> > requiring a synchronization handshake before communication, with
> > each failed RTS triggering an exponential backoff followed by retry.
> > ...
> 
> I don't understand the a priori relevance of such considerations.
> TCP is not a MAC protocol, and in general is supposed to be blithely 
> ignorant of oddities in the link layer.

And indeed it is.  Making it robust to infrequent ACKs is an all-round win.  No 
one is suggesting any link-specific modification to TCP.

> >    (ii) Bi-directional traffic.  I didn't think this was such a
> > common occurrence, but I recently received this message from a DSL
> > customer, which I repeat verbatim below.  (The customer recently
> > graduated from MIT.)
> 
> > ...
> > Except one problem - when someone is downloading a file off of
> > my web server, saturating the 128kbits/sec upstream link, then all
> > my performance goes to hell. Pinging the gateway takes 2 seconds
> > round trip, ...
> 
> The problems there are irrelevant to the draft.  First, no matter what
> you do, you can't get more than 128 kbit/sec from a 128 kbit/sec link.
> Second, such experiences show that saturating a link with mixtures of
> interactive and bulk traffic through with routers that no any notion of
> priority queues is a bad idea, which is not exactly a new discovery.

Gosh, who said anything about interactive traffic!  This happens when you're 
doing bulk traffic in both directions.

The problems are quite relevant to the draft.

> >       One might think that this is an uncommon situation, but if you
> > look at the way the Internet is evolving, nodes serving information from
> > people's homes are becoming increasingly commonplace and are likely to
> > dramatically increase in the next few years.  I predict that situations
> > like Nelson's above will not be rare in the near future, when we start
> > putting all our computers and myriad other devices in our homes on the
> > Internet.
> 
> Those problems are not new.  In the good old days, many of us suffered
> exactly such situations as Nelson's, except that our links were 10 or 50
> times slower.  Please consider ancient asymmetric dial-up modems including
> those from (I think) Courier and modems that were sort-of assymetric like
> Telebits.

Evidently you decided to do nothing about the problem then, same as now.  :)  
(Sorry, couldn't resist, given your general tone.)

> >    (iii) Loss rate asymmetry: Some of the techniques (e.g., sender
> > adaptation) described in the draft handle this case as well, which
> > is observed in many real systems including mobile ones where end
> > stations transmit at much lower power and contend for reverse slots
> > whereas the forward channel is from a high-powered base station
> > and is therefore relatively loss-free.
> 
> That sounds like the old argument that the upper layers must be
> modified for every new link layer that comes along, regardless of 
> whether how different it really is (think about Telebits).  
> Some people reflexively resist that argument.

In general, reflexive reactions aren't well-advised in a scientific discipline. 
Furthermore, many of the solutions advocated in the draft are link-layer ones.  
The changes proposed to TCP itself are all ones that DO NOT degrade TCP 
behavior or performance over any other paths, and often, they improve them.

Ergo, I believe they are a good idea; the solutions are not of the form where 
they improve performance in one situation and degrade it in several other 
places.

> > Given these situations, we believe that they are worth clearly
> > documenting in a draft. ...
> 
> Anyone can write about almost anything in an offishal fur shure Internet
> Draft, as demonstrated by draft-terrell-ip-spec-ipv7-ipv8-addr-cls-*.txt.
> Besides, that draft-ietf-pilc-asym-00.txt exists is not controversial.
> The interesting question is what happens to a draft after it has been
> published.  How people are supposed to respond to this draft?  Is it
> intended to become a Proposed Standard or a BCP?   Or would it become
> just another Informational RFC?  Or is it a call to start some experiments?

Frankly, whether or not the draft exists makes no difference to me personally.  
However, I think that not being aware of some important issues (not merely 
academic ones) that affect TCP performance over some classes of real networks 
is not a good situation.

It isn't a call for more experiments necessarily, although that would be 
welcome.  I assumed it would be informational.  I don't know for sure though, I 
suppose it also depends on the WG chairs.
 
Hari Balakrishnan
http://wind.lcs.mit.edu/~hari/




From owner-pilc@lerc.nasa.gov  Thu Mar  2 03:16:03 2000
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Date: Thu, 2 Mar 2000 00:09:28 -0700 (MST)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Hari Balakrishnan <hari@breeze.lcs.mit.edu>

> ...
> > I don't understand the a priori relevance of such considerations.
> > TCP is not a MAC protocol, and in general is supposed to be blithely 
> > ignorant of oddities in the link layer.
>
> And indeed it is.  Making it robust to infrequent ACKs is an all-round win.  No 
> one is suggesting any link-specific modification to TCP.

I don't understand that.  Where do you have infrequent ACKs in an
active TCP connection except where a strange link layer is being
either very smart or too smart by half, and eating ACKs?  What
happened to the notion of "ACK pacing"?
Making TCP robust to infrequent ACKs is a special change particularly for
links that like to preferentially eat some special kinds of upper layer
frames.  It might well be a good idea, but with no offense intended, 
more than academic assurances would be nice before we rush out
and modify all current TCP implementations.


> > > ...
> > > Except one problem - when someone is downloading a file off of
> > > my web server, saturating the 128kbits/sec upstream link, then all
> > > my performance goes to hell. Pinging the gateway takes 2 seconds
> > > round trip, ...
> > 
> > The problems there are irrelevant to the draft.  First, no matter what
> > you do, you can't get more than 128 kbit/sec from a 128 kbit/sec link.
> > Second, such experiences show that saturating a link with mixtures of
> > interactive and bulk traffic through with routers that no any notion of
> > priority queues is a bad idea, which is not exactly a new discovery.
>
> Gosh, who said anything about interactive traffic!  This happens when you're 
> doing bulk traffic in both directions.

Nelson's only quantitative performance measurement (that I recall) involved
interactive stuff, his 2 second pings.  He didn't say anything (as I
recall) in terms of bits/sec that showed the downstream pipe was not full.
It might not have been full, and one would hope an MIT alumni would do a
complete set of measurements.  However, the left coast does affect people.
I've even known people on the right coast within a rifle shot of MIT to
blow off a little private steam in wildly overstated complaints about lame
ISP services.


>  ...
> > Those problems are not new.  In the good old days, many of us suffered
> > exactly such situations as Nelson's, except that our links were 10 or 50
> > times slower.  Please consider ancient asymmetric dial-up modems including
> > those from (I think) Courier and modems that were sort-of assymetric like
> > Telebits.
>
> Evidently you decided to do nothing about the problem then, same as now.  :)  
> (Sorry, couldn't resist, given your general tone.)

Actually, that's wrong.  We have various link layer things including data
compression, VJ header compression, and various priority queueing at least
partly because of the old problems.  Note that the things we do have
received a lot of real life testing and serious measuring before they were
documented in RFC's.

Besides, one of my favorite sayings goes something like "anyone can always
find something to add to anything."

Some of what you propose might be very good things.  One or two meet my
personal prejudices for what's right and needed dead on.  However, as far
as I can tell, you've offered only academic or qualitative and anecdotal
support for them.  In other words, what kind of RFC should your draft
should become?


> > >    (iii) Loss rate asymmetry: Some of the techniques (e.g., sender
> > > adaptation) described in the draft handle this case as well, which
> > > is observed in many real systems including mobile ones where end
> > > stations transmit at much lower power and contend for reverse slots
> > > whereas the forward channel is from a high-powered base station
> > > and is therefore relatively loss-free.
> > 
> > That sounds like the old argument that the upper layers must be
> > modified for every new link layer that comes along, regardless of 
> > whether how different it really is (think about Telebits).  
> > Some people reflexively resist that argument.
>
> In general, reflexive reactions aren't well-advised in a scientific discipline

In general, that is false.  This isn't a scientific discipline, but if it
were, Occam's Razor would be a reputable reflexive reaction.  In an
engineering discipline like our's, Occam's Razor as well as the rule of
thumb that says you don't change things without good, quantitative reasons
are wise reflexes.


>Furthermore, many of the solutions advocated in the draft are link-layer ones

The whole point of your proposals is to affect TCP.  Most of your proposals
("solutions"?--sheesh!) might large effects on TCP, assume TCP has certain
behaviors (e.g. RTT measurement won't be trashed by ACK compression or
performance won't be zapped by re-ordered TCP segments), or assume TCP
will be changed at least slightly (e.g. made "robust to infrequent ACKs").


> The changes proposed to TCP itself are all ones that DO NOT degrade TCP 
> behavior or performance over any other paths, and often, they improve them.
>
> Ergo, I believe they are a good idea; the solutions are not of the form where 
> they improve performance in one situation and degrade it in several other 
> places.

I'm sure you think so.  Over the years I too have had wonderful ideas that
I was sure would have lots of wonderful results and no bad effects.  I
was even right on the second half about one or two of them.   In other
words, if you're proposing a BCP or STD, it's far too early.  Still, the
ideas would be interesting to test seriously, and I don't mean a couple
of master's thesis simulations suitable for IEEE publication.  (I hope no
one has done any of those for these ideas since I don't intend any
particular offense by that point.)

And characterizing your proposals as "solutions" again?  I'd go for
"proposed solutions", "possible solutions," or even "likely solutions",
but you're a *long* way from bald "solutions".  You haven't even offered
convincing evidence for all of your characterizations of the problems!


> ..
> > The interesting question is what happens to a draft after it has been
> > published.  How people are supposed to respond to this draft?  Is it
> > intended to become a Proposed Standard or a BCP?   Or would it become
> > just another Informational RFC?  Or is it a call to start some experiments?
>
> Frankly, whether or not the draft exists makes no difference to me personally
> However, I think that not being aware of some important issues (not merely 
> academic ones) that affect TCP performance over some classes of real networks 
> is not a good situation.
>
> It isn't a call for more experiments necessarily, although that would be 
> welcome.  I assumed it would be informational.  I don't know for sure though
> I suppose it also depends on the WG chairs.

Urging serious experiments of the ideas is a good idea.  However, you keep
talking about "solutions," which would be strong for an official working
group Informational RFC.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Thu Mar  2 04:37:31 2000
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	(message from Marie-Jose Montpetit on Wed, 1 Mar 2000 15:50:38 -0800)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
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Thanks for all the comments. I'll work on incorporating them into the
draft this weekend after I get back from the IAB workshop.

Phil


From owner-pilc@lerc.nasa.gov  Thu Mar  2 12:49:58 2000
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Date: Thu, 02 Mar 2000 15:43:15 +0100
From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>
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To: Joe Touch <touch@ISI.EDU>, pilc@grc.nasa.gov
Subject: Re: Draft on Advice for Internet Subnetwork Designers
References: <200003011811.KAA21226@boreas.isi.edu>
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I know this is in danger of getting really side-tracked, and what you say is
interesting, but
the subnet link designer is interested in subnet payload size, not just MTU.  What I
meant to
say was there are conditions under which the payload size is reduced, because a PMTU
smaller than the link MTU was selected by PMTU discovery.

Gorry

Joe Touch wrote:

> > From gorry@erg.abdn.ac.uk  Wed Mar  1 05:12:19 2000
> > Date: Wed, 01 Mar 2000 13:30:30 +0100
> > From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>
> > To: touch <touch@ISI.EDU>
> > CC: pilc@grc.nasa.gov
> > Subject: Re: Draft on Advice for Internet Subnetwork Designers
> >
> > More clarification...
> >
> > Joe Touch wrote:
> >
> > > > "Falk, Aaron" wrote:
> > > >
> > > > PAGE 4: MTU Size
> > > >
> > > > It may be useful to note that using IPSEC in tunnel mode can increase
> > > > the size of a packet as it travels across the network.  This can result
> > > > in a smaller MTU.
> > >
> > > Doesn't IPSEC in any mode do this? (tunnel or transport?)
> > >
> > > (provided it's added after the source - and, in the case
> > > of overlay networks, [paradoxically] transport may be
> > > more useful)
> >
> > Sure, you are correct. But, I seem to remember the draft mentioning 1500B MTU.
> > Now, since this is a link designer's draft, what we are presumably interested in
> > is frame payload data size... So, if normal IP is used that is maximum 1500B.
> > If IPSEC is used end-to-end that is still 1500B of frame payload, even though
> > MTU is reduced. But with tunnel mode along part of the path, the rest of the
> > path outside the tunnel now sees a reduced MTU - or the tunneled path needs
> > to support a larger frame size.  Just saying 1500B is not a
> > universal MTU anymore.
>
> It doesn't matter where IPSEC is used - E2E or tunnel. Either
> way eats into the MTU. The PILC doc is talking about link MTUs;
> everything except the link header consumes that.
>
> > > > IPv6 does
> > > > not therefore implement a network level integrity checksum. The role of the
> > > > IP checksum has therefore been subsumed into the TCP and UDP
> > > > checksums.
> > >
> > > More into IPSEC than into TCP and UDP, no?
> >
> > IPSEC is one way, it is not (I think) mandatory, so TCP/UDP may still get
> > bad packets which they have to reject....
>
> Support for IPSEC is mandatory in IPv6, which is where the IP checksum
> has been removed. Its _use_ is not mandatory.
>
> The role of the IP checksum was to check the IP header. The role of
> the TCP/UDP checksums is to check the data. If you want to protect/sum
> the header, you should (and thus, above, can) use IPSEC.
>
> Joe



From owner-pilc@lerc.nasa.gov  Thu Mar  2 15:45:25 2000
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From: "Falk, Aaron" <afalk@PanAmSat.com>
To: "'Vernon Schryver'" <vjs@calcite.rhyolite.com>
Cc: pilc@grc.nasa.gov
Subject: RE: PILC ASYM document
Date: Thu, 2 Mar 2000 14:26:51 -0500 
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Vernon wrote:

> ...
> The interesting question is what happens to a draft after it has been
> published.  How people are supposed to respond to this draft?  Is it
> intended to become a Proposed Standard or a BCP?   Or would it become
> just another Informational RFC?  Or is it a call to start 
> some experiments?

Vernon-

The ASYM document is in our charter to be published as a BCP. This choice
was made before a document was written and the assumption was that it would
contain a) a description of the problems, b) proposed well-understood
mitigations (implemented at the layer of your choice), and possibly,
well-caveated proposals for future protocol development. The PILC charter
does not include developing any new protocols. However, if we find something
that we believe should be developed, we can approach the IESG about
rechartering the working group.

The working group has several options: 

- the topic can be discarded as not relevant and the draft allowed to expire
(or submitted as a non-wg draft by the authors), 

- the draft can be submitted by the working group as an Informational or BCP
RFC, 

- and/or we can seek to develop some protocol changes and an STD. 

A key to determining what will happen will be how large the consituency is
and deep the need is. Hence this discussion...

Before the working group could take on developing a new protocol, a
compelling argument needs to be made for the need of such an activity. Hari
and Venkat have made their case that this is a real problem and I'm looking
to the mail list for validation that the problem merits further definition
and the proposals merit further development by this working group.

I appreciate your comments and invite more.

--aaron


From owner-pilc@lerc.nasa.gov  Thu Mar  2 18:35:20 2000
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From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: "Falk, Aaron" <afalk@PanAmSat.com>

> ...
> The ASYM document is in our charter to be published as a BCP. This choice
> was made before a document was written and the assumption was that it would
> contain a) a description of the problems, b) proposed well-understood
> mitigations (implemented at the layer of your choice), and possibly,
> well-caveated proposals for future protocol development. The PILC charter
> does not include developing any new protocols. However, if we find something
> that we believe should be developed, we can approach the IESG about
> rechartering the working group.

It is clear Hari's document cannot conceivably be (part of) a Best Current
Practices in the immediate future.  I think Hari agrees with that, despite
his unqualified uses of "problem" and "solution."  Before ACK regenerating,
ACK-only scheduling, or the other ideas could be listed in a BCP, we need
real testing to see what they break.  (Claims that nothing would break
are obvious nonsense; every worthwhile change in a system as big as the
Internet breaks something that someone somewhere values.   Worse, no one
knows enough about the system to predict accurately what will break.)

The many ADSL networks provide many test beds for some hacks on free
UNIX and Linux kernels.  Some artificial throttling and random filtering
might ADLS links similar to wireless links.  If the ideas are safe
enough, users of modified systems would not notice the change.


> The working group has several options: 
>
> - the topic can be discarded as not relevant and the draft allowed to expire
> (or submitted as a non-wg draft by the authors), 
>
> - the draft can be submitted by the working group as an Informational or BCP
> RFC, 
>
> - and/or we can seek to develop some protocol changes and an STD. 
>
> A key to determining what will happen will be how large the consituency is
> and deep the need is. Hence this discussion...

The issues (note I didn't say "problems") listed in Hari's document have
long had a significant consituency outside the IETF.  (Yes, I'm yet again
refering in part to cable modem games with ACKs.)  I think the issues are
more worthy than many other IETF activities, such as the zillions of MIB's.


> Before the working group could take on developing a new protocol, a
> compelling argument needs to be made for the need of such an activity. Hari
> and Venkat have made their case that this is a real problem and I'm looking
> to the mail list for validation that the problem merits further definition
> and the proposals merit further development by this working group.
>
> I appreciate your comments and invite more.

The practice followed by Van Jacobson, Sally Floyd, and others has worked
pretty well over the years.  They do lots of simulating, testing, and more
than prototype implementing before publishing anything, or at least before
announcing solutions or proposing IETF sanctioned changes in widely
deployed implementations.  A recent example is the change in the initial
TCP window, which was done at least partly under the auspices of the TCP
Implementors working group.  (I don't mean to say anything about which WG
should own which territory, but only offer a relevant mailinglist archive.)


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Thu Mar  2 19:27:38 2000
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To: Marie-Jose Montpetit <marie@teledesic.com>
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        "Phil Karn (E-mail)" <karn@qualcomm.com>,
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Subject: Re: Draft on Advice for Internet Subnetwork Designers 
In-reply-to: Your message of "Wed, 01 Mar 2000 15:50:38 EST."
             <25D0C66E6D25D311B2AC0008C7913EE07B8100@tdmail2.teledesic.com> 
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> a) [Following owes a great deal to a side-discussion I had with Mark Allman]
> 
> TCP needs lots of buffering.  An ability to buffer bursts of up to (fixed 
> delay)*(bottleneck rate) per TCP connection in each node is ideal if 
> impractical.  Problem is that neither the fixed delay nor the bottleneck
> rate 
> is necessarily known at any node.  A good approximation is fixed delay of
> the 
> link * link bandwidth.  For links that have variable delays, the
> approximation 
> should be based on the max.  Also, the maximum number of TCP connections per
> 
> node is very difficult to determine a priori, and is not subject to any kind
> 
> of admission control (except perhaps at a PEP).  A SWAG has to suffice.
> Some 
> kind of mechanism to measure delay on the link can be useful in autotuning 
> buffer sizes.  In general, more buffers are better than fewer.  For example,
> 
> TCP performance in very early ATM switch implementations suffered horribly 
> from the small buffer sizes.
> 
> Yes but if I remember this is also when the traffic statistics were
> basically ill undestood. While I do agree that more buffering is better than
> less, I would also raise the issue of buffering delays especially if a path
> goes through more than one subnet. Then do you have to implement some queue
> management at each point?
> 
The model I've been working with (especially since that side discussion) is 
that if the subnet does queueing (buffering, scheduling, discarding), then the 
queueing systems in the subnet has exactly the same effect as queues in 
routers.  This may be worth stating.  Thus, recommendations about having a 
window size worth of buffers per TCP connection,  using active queue 
management, etc. in the subnet would follow from similar recommendations at 
the IP layer.  

As with IP layer forwarding, the objective is not to have large buffers that 
are always full, but rather the capability to absorb large bursts.  Since many 
of the kinds of subnets we're talking about will be (or will contain) the 
bottleneck link, one might expect that an entire window's worth of data might 
easily wind up at a single node in the subnet.
 
> b) There is a rough consensus (not unanimous) in the research community that
> 
> active queue management is important to improving fairness, link utilization
> 
> and throughput. Reference RFC 2309.  Followed by a brief introduction to 
> active queue management, with RED as an example. An important part of the 
> concept is that active queue management attempts to keep average queue
> depths 
> short, while accommodating short term bursts.  It may be advisable for
> subnets 
> that perform queueing and discarding to consider implementing some form of 
> active queue management, especially if large aggregates of TCP connections
> are 
> likely to share the same queue.  It may be less effective in the case, for 
> example, of an ATM switch that implements per-VC queueuing.  Note, 
> incidentally, that active queue management algorithms in general are highly 
> sensitive to parameter settings and reference Borden and  Fioriu paper.  
> 
> Again I agree. But do my question qould be to which level of detail we want
> to go? And is active queue management a link issue?
Probably not much more detail than the sketch.  Details could be by reference.
> 
> c) If the subnet does fragmentation and reassembly, make sure that if it 
> becomes necessary to discard, then discard entire frames, not just scattered
> 
> fragments. For ATM, this specifically means using EPD/PPD, and reference ATM
> 
> Forum TM 4.0 specification.   Reference Romanow paper. 
> 
> Yes and I think we could even suggest that the subnet data unit (SDU?) have
> some markings for this.
Can you explain?
> 
> d) If the subnet includes flow control that shapes the rate at the edge of
> the 
> subnet to match a share of the bottleneck bandwidth in the subnet (e.g., ATM
> 
> ABR), this has the effect of pushing congestion to the edge of the subnet.
> In 
> such a case, adequate buffering is needed at the edge of the subnet (see a 
> above).  Whether or not there is benefit in this kind of flow control is 
> controversial, and there have been simulation studies that go both ways.  
> Anecdotally, some large networks have used ABR sucessfully to carry IP 
> traffic, but no published results.
> 
> This also has links into the subnets that use bandwidth on demand and yes
> the buffering is an issue again. I have seen BTW RSVP to ATM results (S.
> Giordano) but not ABR.
RSVP to ATM is an unrelated issue.  Raj Jain and some of his students did a 
lot of simulation papers on this a few years ago.  So did Marty Borden and 
Nanying Yin (I think).  And there were some advocacy papers in the ATM Forum.  
I think it's fair to say that there was no consensus among the researchers in 
this area.  In the meantime, I have heard that there have been some sucessful 
deployments, but the network operators involved have not published their 
results.
> e) Flow control mechanisms that mark but do not shape frames in excess of
> fair 
> share of the bottleneck rate (e.g., frame relay FECN and BECN) are at best 
> useful only in the context of IP ECN (RFC 2481).  New subnets would do well
> to 
> have flow control mechanisms that have that semantics (i.e., feedback signal
> 
> based on EWMA buffer size, feedback set probabilistically proportional to 
> difference between two thresholds, or always set when second threshold 
> exceeded) so that feedback could be 'bubbled up' from the subnet layer to
> the 
> IP header.  
> 
> Yes. How about some bubbling up of air interface conditions into control
> messages or cost functions?
I'm not sure what you're suggesting.  The semantics that I wanted to 'bubble 
up' is identical to that of the ECN in RFC 2481 (see my observation above). 
Thus, a queue in the subnet could provide a filtered indication to the TCP 
layer that it was congested just as if were a queue in a router.   What 
semantics in an air interface could map to any existing or potential semantics 
in IP?
 

> Marie-Jose
Dan



From owner-pilc@lerc.nasa.gov  Thu Mar  2 20:59:02 2000
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From: Venkat Padmanabhan <padmanab@microsoft.com>
To: "'Vernon Schryver'" <vjs@calcite.rhyolite.com>, pilc@grc.nasa.gov
Subject: RE: PILC ASYM document
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> >    (i) Media-access.  Many MAC protocols are RTS/CTS based,
> > requiring a synchronization handshake before communication, with
> > each failed RTS triggering an exponential backoff followed by retry.
> > ...
> 
> I don't understand the a priori relevance of such considerations.
> TCP is not a MAC protocol, and in general is supposed to be blithely 
> ignorant of oddities in the link layer.

Unless there are aspects of TCP that interact poorly with the
link's characteristics. Media access asymmetry is often a fundamental
artifact of the hub-and-spokes configuration of the network. It is
often a lot less expensive for the hub (e.g., a wireless base station 
or cable modem head-end equipment) to transmit packets to the (distributed)
end-stations than it is for the latter to transmit to the hub (due to
contention). In the context of a downstream TCP data transfer (i.e.,
in the direction from the hub to an end-station), this characteristic
of the link would make the transmission of ack packets by the end-station
expensive, regardless of the size of those packets.

Hari and I have observed this problem via simulations and actual
measurements of a packet radio network. (See sections 5 and 7.2 of
http://www.research.microsoft.com/~padmanab/papers/monet99-tcpasym.ps.gz).
We also have anecdotal evidence of similar problems in cable modem
systems. We were hoping to hear experiences of other people confirming
or refuting this. Barbed emails heavy in sarcasm unfortunately drown
out whatever technical point was sought to be conveyed.

Regarding your implication that TCP should be ignorant of the oddities
of the link layer, an analogy might illustrate why this argument isn't
necessarily right. HTTP/1.0 (an application-level protocol) ignored 
artifacts of the underlying TCP transport protocol (e.g., slow start)
and opened a new TCP connection for each request. Consequently, performance
suffered.

> The problems there are irrelevant to the draft.  First, no matter what
> you do, you can't get more than 128 kbit/sec from a 128 kbit/sec link.

I would second your remark about the 128 Kbps link, but I don't see
why it is relevant. No one is claiming that a 128 Kbps link could/should
be driven faster than 128 Kbps. The point is that congestion on the
128 Kbps uplink resulted in poor utilization of the 1.5 Mbps downlink.

> Second, such experiences show that saturating a link with mixtures of
> interactive and bulk traffic through with routers that no any 
> notion of
> priority queues is a bad idea, which is not exactly a new discovery.

What we are talking about of here is not "mixtures of interactive
and bulk traffic". Rather it is the interference between the acks of
a downstream bulk transfer with data packets of an upstream bulk transfer.  

> That sounds like the old argument that the upper layers must be
> modified for every new link layer that comes along, regardless of 
> whether how different it really is (think about Telebits).  
> Some people reflexively resist that argument.

I have no idea what Telebits is, but regardless we are not arguing 
that the upper layers must be modified for every new link layer.
Rather we have attempted to document a set of problems related to
asymmetry and an approach to addressing these that has a common 
thread: reducing the frequency of acks -- a generalization of
delayed acks, if you will -- and making the sender robust against
disruptions in the ack stream. These have been derived from our
experience/experiments with a few network technologies. Whether to 
turn any of this into a proposed standard would depend on whether 
there is a consensus on their relevance in a wider setting 

> Is it intended to become a Proposed Standard or a BCP?   Or would it
become
> just another Informational RFC?  Or is it a call to start 
> some experiments?
> 

Good question. As Aaron states, the PILC WG's current charter precludes
turning this into a proposed standard (of course, I realize that this
technicality is not the only hurdle in the way!). If there is sufficient
consensus and supplementary/supporting evidence (e.g., experiences with
other network technologies), then (parts of) the document can
be turned into a BCP. If not, I guess it will be an Informational RFC.


Venkat Padmanabhan
Microsoft Research
padmanab@microsoft.com
http://www.research.microsoft.com/~padmanab


From owner-pilc@lerc.nasa.gov  Fri Mar  3 06:32:07 2000
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On Thu, 2 Mar 2000, Venkat Padmanabhan wrote:

> Regarding your implication that TCP should be ignorant of the oddities
> of the link layer, an analogy might illustrate why this argument isn't
> necessarily right. HTTP/1.0 (an application-level protocol) ignored 
> artifacts of the underlying TCP transport protocol (e.g., slow start)
> and opened a new TCP connection for each request. Consequently, performance
> suffered.

That analogy is misleading.

HTTP _always_ runs over TCP, so there are widespread gains from
optimising interaction between the two. It's a global (a World-Wide,
even) optimisation with few tradeoffs.

TCP does not always run over e.g. some braindead ARQ-to-infinity link
layer. Optimising TCP for such a special case would prove detrimental
to interactions with other, different, link layers. It's a local
optimisation with many tradeoffs.

Making TCP independent of link layer oddities strives for the general
case of global optimisation. Vernon is correct.

L.

the Internet: increasingly a bunch of isolated local optimisations
with undesirable global interactions.

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



From owner-pilc@lerc.nasa.gov  Fri Mar  3 06:47:56 2000
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The well proven approach taken by Sally Floyd, et al (just using the name
quoted
on the list, no offence to Sally) is a good one for some problems.  But it
does not
always work.  The problem with PILC as a whole is that the issues being
discussed
are (as Phil Karn put it) from two different communities. The two need to talk
to
each other in moderate tones of voice, so that we understand what TCP is
actually
asking for. I disagree that TCP "works well" over all links.  If you define
"well" as
meaning doesn't fail to delivery (eventually) then that's OK, but normally
that's
not my thinking.  TCP is end-to-end, but link designers need to take on board
what is really happening.  Once the subnet designers document is clear, I
suggest
we will probably see the need for the ASYM document.

I'd like to see the ASYM draft opened up a little in terms of discussing
different
approaches... and getting a clearer problem statement, then input from the
cable
people, radio people, the DVB people, and anyone else.  My concern is, until
we get the first ID into some shape, we attract a lot of confusing comments
every
time something is mentioned (this isn't new/ this isn't needed/etc).

I recommend we keep the document as an ID and wait for the SUBNET DESIGNERS
draft to reach stability first (next draft?), then look discuss the approaches
in detail at Pittsburgh....

Gorry Fairhurst.




From owner-pilc@lerc.nasa.gov  Fri Mar  3 10:46:03 2000
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From: "Falk, Aaron" <afalk@PanAmSat.com>
To: "'Vernon Schryver'" <vjs@calcite.rhyolite.com>
Cc: pilc@grc.nasa.gov
Subject: RE: PILC ASYM document
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Vernon wrote: 
> 
> The practice followed by Van Jacobson, Sally Floyd, and 
> others has worked
> pretty well over the years.  They do lots of simulating, 
> testing, and more
> than prototype implementing before publishing anything, or at 
> least before
> announcing solutions or proposing IETF sanctioned changes in widely
> deployed implementations.  A recent example is the change in 
> the initial
> TCP window, which was done at least partly under the auspices 
> of the TCP
> Implementors working group.  

Vernon-

You've made an excellent point. Actually, getting researchers of the caliber
you mention is part of the goals of this group. (Sally, Van: are you
listening?) However, to get networking researchers engaged in a problem
there needs to be a clear articulation of exactly what the problem is. Part
of that discussion would be to propose some candidate solutions. This is why
PILC isn't developing any protocols -- we are supposed to be bringing
together user/developer communities
_who_may_not_be_experienced_in_Internet_protocol_R_&_D_ to document the
problems they see on links of interest. One possible outcome is that an
Informational RFC is published. Getting to that point will require a rough
consensus on the issues. The dialog will likely be of greater value than the
document. These dialogs can have profound influence decisions on IETF
activities in the future.

Some might say (& I heard it about TCPSAT), that this isn't relevant work
for the IETF, since the IETF develops protocols for the Internet. My
understanding is that the IETF's charter is not quite that narrow. PILC is
doing the advance work to identify whether a new protocol(s) needs to be
developed. Part of that process is requirements definition (such as is going
on in ECM). But before you can define requirements you need agreement on
what the issues are and whether they require fixing. *That* is what we are
doing now. 

(I might add that TCPSAT was very effective in giving the satellite
community a clue on issues surrounding TCP performance. This was/is a
community where most designers didn't have a background in Internet
protocols and didn't understand the performance/design tradeoffs. To a
certain extent, PILC is a generalization of the TCPSAT goals to a wider
community of interest.)

--aaron


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Date: Fri, 3 Mar 2000 09:57:31 -0500 (EST)
From: mukul goyal <mukul@cis.ohio-state.edu>
To: Lloyd Wood <l.wood@eim.surrey.ac.uk>
cc: Venkat Padmanabhan <padmanab@microsoft.com>,
        "'Vernon Schryver'" <vjs@calcite.rhyolite.com>, pilc@grc.nasa.gov
Subject: RE: PILC ASYM document
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> TCP does not always run over e.g. some braindead ARQ-to-infinity link
> layer. Optimising TCP for such a special case would prove detrimental
> to interactions with other, different, link layers. It's a local
> optimisation with many tradeoffs.
>

Lloyd, could you clarify this further? I think that draft talks about
ideas like restricting the number of packets sent in a burst and making
sure that window increase is independent of number of ACKs received. I
think it will be a _global_ optimization and not a _local_ one. I might be
missing something but can you provide some example scenarios where such a
thing will not be good.
  
> Making TCP independent of link layer oddities strives for the general
> case of global optimisation. Vernon is correct.

I thought that's what draft talking about: making TCP independent of link
layer oddities.

Mukul





From owner-pilc@lerc.nasa.gov  Fri Mar  3 11:33:37 2000
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Date: Fri, 3 Mar 2000 10:20:44 -0500 (EST)
From: mukul goyal <mukul@cis.ohio-state.edu>
cc: pilc@grc.nasa.gov
Subject: Self-Clocking (RE: PILC ASYM document)
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Hi All,

THis is a diversion from the topic on hand but I am confused about the
meaning of the term "self-clocking". Recently, I posted this query on
tcpsat list and Dr. Partridge sent a reply which I am pasting below:

Date: Sat, 26 Feb 2000 16:01:55 -0500
From: Craig Partridge <craig@bbn.com>
To: mukul goyal <mukul@cis.ohio-state.edu>
Cc: tcpsat@grc.nasa.gov
Subject: Re: Lack of TCP self-clocking in practical networks 


    I feel that TCP's self-clocking as illustrated in Van Jacobson's 1988
    paper exists only if bottleneck link is a link of lower capacity (i.e.
    bandwidth) than other links. However, if bottleneck link has higher
    capacity than other links (i.e. the link is bottlenecked because a
    large number of flows are using it simulateneously), there is no
    self-clocking any more. 

I don't think it is that simple

Self clocking always occurs -- you inject a packet  in response to an ack
and the ack is telling you when the network has capacity for your new
packet.

The issue raised in your note is whether the spacing of the acks reflects
the available bandwidth at the bottleneck.  In most cases the answer is
that it reflects bandwidth only imperfectly.  Ack compression, queueing
disciplines, etc., all conspire to muddy the spacing.

Craig 

****************************************************

This gives me an impression that self-clocking refers to the property
that a new packet will be injected into the network when an old one has
left it and ACK basically conveyes this information to the sender.

Another interpretation of self-clocking that I have seen in many places
including PILC ASYM draft is that temporal spacing of ACKS reflect the
_bottleneck_ bandwidth in forward (or reverse) direction.

FInally, self-clocking might be interpreted as TCP's property that
temporal spacing of ACKs reflect _available_ bandwidth at bottleneck for a
flow. (I know that this is not true.)

I would like to know what is the accepted definition of Self-Clocking and
if some IETF document talks about it.

It might be a good idea to have a document that lists accepted definition
of concepts like this.

Thanks,
Mukul




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Good points Gorry.  We need to enter this millennium with a little less
bureaucratic defensiveness about the "existence proof" that is the current
Internet protocol/infrastructure.  Passing over its security pit, if we look
at TCP, in particular and in the light of day, we see a 'transport' that is
incapable of distinguishing important, performance-degrading effects.  For
instance, its response to packet loss being identical whether the packet was
errored or congested out of existence causes real, costly delays.  These
events increase as it is run more often over media for which it was never
designed.  The increasing dependence of real business, rather than subsidized
research, on the Internet is already revealing how influential the $ side of
the equation is becoming.  Engineering should follow this lead.

Alex


Dr G Fairhurst wrote:
> 
> The well proven approach taken by Sally Floyd, et al (just using the name
> quoted
> on the list, no offence to Sally) is a good one for some problems.  But it
> does not
> always work.  The problem with PILC as a whole is that the issues being
> discussed
> are (as Phil Karn put it) from two different communities. The two need to talk
> to
> each other in moderate tones of voice, so that we understand what TCP is
> actually
> asking for. I disagree that TCP "works well" over all links.  If you define
> "well" as
> meaning doesn't fail to delivery (eventually) then that's OK, but normally
> that's
> not my thinking.  TCP is end-to-end, but link designers need to take on board
> what is really happening.  Once the subnet designers document is clear, I
> suggest
> we will probably see the need for the ASYM document.
> 
> I'd like to see the ASYM draft opened up a little in terms of discussing
> different
> approaches... and getting a clearer problem statement, then input from the
> cable
> people, radio people, the DVB people, and anyone else.  My concern is, until
> we get the first ID into some shape, we attract a lot of confusing comments
> every
> time something is mentioned (this isn't new/ this isn't needed/etc).
> 
> I recommend we keep the document as an ID and wait for the SUBNET DESIGNERS
> draft to reach stability first (next draft?), then look discuss the approaches
> in detail at Pittsburgh....
> 
> Gorry Fairhurst.


From owner-pilc@lerc.nasa.gov  Fri Mar  3 15:00:58 2000
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From: Vernon Schryver <vjs@calcite.rhyolite.com>
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} From: mukul goyal <mukul@cis.ohio-state.edu>

} THis is a diversion from the topic on hand ...

it seems relevant to me.

} It might be a good idea to have a document that lists accepted definition
} of concepts like this.  

More of a definition of ACK clock, pacing, etc. would be good,
especially in a document that mandates or at least encourage
practices that, as Craig Partridge said, "muddy the spacing".

  ....


From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>

] The well proven approach taken by Sally Floyd, et al (just using
] the name quoted on the list, no offence to Sally) is a good one
] for some problems.  But it does not always work.

Yes, doing "lots of simulating, testing and more than prototype
implementing before publishing anything, or at least before announcing
solutions or proposing IETF sanctioned changes in widely deployed
implementations" does not always work.  Failure is part of inventing.
However, defining protocols (e.g. ACK regeneration) without doing all of
that other stuff is almost certain to fail.

Examining relevant history (e.g. asymmetric dial-up modems carrying
TCP/IP/SLIP and TCP/IP/PPP) and long standing, current industry practice
(e.g. ACK compression in the metropolitan cable TV modem systems of
Hybrid Systems installed for the last 6-10 years) is as good in an
academic paper as in an official WG RFC, even if only Informational.

  ....


> From: Venkat Padmanabhan <padmanab@microsoft.com>

> ...
> We also have anecdotal evidence of similar problems in cable modem
> systems. We were hoping to hear experiences of other people confirming
> or refuting this.

How would such experiences be other than still more anectotal evidence
that could neither confirm nor refute the draft?  Don't we need some
prototype implementing and serious testing before advocating ACK
regeneration or compression?  (Not to mention re-ordering the TCP
segment stream by moving pure ACKs ahead of data)


>                   Barbed emails heavy in sarcasm unfortunately drown
> out whatever technical point was sought to be conveyed.

I haven't intend as much barbed sarcasm as seems to have been read.  But
if you keep twisting my tail by talking about a BCP openly based on a few
tests, some thinking, a few anecdotes, and not even a nod at historical
or current industry practice, I'll start meaning it.


> Regarding your implication that TCP should be ignorant of the oddities
> of the link layer, an analogy might illustrate why this argument isn't
> necessarily right.  HTTP/1.0 (an application-level protocol) ignored 
> artifacts of the underlying TCP transport protocol (e.g., slow start)
> and opened a new TCP connection for each request. Consequently, performance
> suffered.

That's a very good example.  Would it have been better to modify TCP/IP
to fit HTTP/1.0 or to modify HTTP/1.0 to fit TCP/IP?  Do you think a BCP
recommending that slow start be turned off for HTTP traffic should have
been written instead of the work on HTTP/1.1?  Do you think that the
tests that supported increasing the intial TCP window were not needed,
that it would have been wise to mandate that change without what passes
for due consideration in the IETF, running code?


> ...
> What we are talking about of here is not "mixtures of interactive
> and bulk traffic". Rather it is the interference between the acks of
> a downstream bulk transfer with data packets of an upstream bulk transfer.  

Did I fail to see the numbers in Hari's message from measurements that
supported the hypothesis that the downstream pipe was not full?  I didn't
keep a copy of the original message, and I remember only a mention of
ping as support for the hypothesis.  Do you agree that the 2 second ping
measurement contradicts or at best shows nothing about the hypothesis?


> ...
> I have no idea what Telebits is, but regardless we are not arguing 
> that the upper layers must be modified for every new link layer.

You really should look at Telebit and Courier modems.  You might find
that many of the problems listed in the document are 10 or 15 years
old.  That's not to say that they have already been solved or shouldn't
finally be solved, but that you might then have more than anecdotes.


> Rather we have attempted to document a set of problems related to
> asymmetry and an approach to addressing these that has a common 
> thread: reducing the frequency of acks -- a generalization of
> delayed acks, if you will -- and making the sender robust against
> disruptions in the ack stream. These have been derived from our
> experience/experiments with a few network technologies. Whether to 
> turn any of this into a proposed standard would depend on whether 
> there is a consensus on their relevance in a wider setting 

If "disruptions in the ack stream" did not mean intentional, consistent
filtering mandated by the same BCP, who could disagree with that?


> > Is it intended to become a Proposed Standard or a BCP?   Or would it
> become
> > just another Informational RFC?  Or is it a call to start 
> > some experiments?
>
> Good question. As Aaron states, the PILC WG's current charter precludes
> turning this into a proposed standard (of course, I realize that this
> technicality is not the only hurdle in the way!).

I don't see why that should be a hurdle.  The IETF is not yet the IEEE
where very heavy duty politicing happens before a committee gets to
own its turf, or the ANSI where the most weighty deliberations in any
committee concern defending its turf from other committees.  The IETF
is supposed to be about "running code" even before "rough concensus."
Let's have some running code, and then worry about the official documents.


>                                                   If there is sufficient
> consensus and supplementary/supporting evidence (e.g., experiences with
> other network technologies), then (parts of) the document can
> be turned into a BCP. ...

How could the WG publish a Best Current Practice about things that are
not Current Practice and are not known to be harmless, not to mention
Best?  As far as I can tell, no one has even started to demonstrate that
the various ideas for modifying the ACK stream do more good than harm.
For example, if ACK compression as great an idea as you think and my
prejudices agree, then why is it as if the large scale commercial
installations that use ACK compression do not exist?  Why do so many
knowledgable people only grimace when you mention ACK compression?

It's early and unnecessary to write a BCP that might be turn out to say
the equivalent of "disable slow start on all hosts because otherwise
HTTP/1.0 works poorly."

>                       If not, I guess it will be an Informational RFC.

What's wrong with doing some prototype implementing and serious testing
before worrying about publishing an official document or whether it should
be Informational, STD, or BCP?   The point of the IETF is not merely
publishing documents about ideas for protocol changes, but "running code."


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Fri Mar  3 15:11:15 2000
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Restoring cut text here. Venkat wrote:

| Regarding your implication that TCP should be ignorant of the oddities
| of the link layer, an analogy might illustrate why this argument isn't
| necessarily right. HTTP/1.0 (an application-level protocol) ignored 
| artifacts of the underlying TCP transport protocol (e.g., slow start)
| and opened a new TCP connection for each request. Consequently, performance
| suffered.

I responded: 
$ That analogy is misleading.
$
$ HTTP _always_ runs over TCP, so there are widespread gains from
$ optimising interaction between the two. It's a global (a World-Wide,
$ even) optimisation with few tradeoffs.


On Fri, 3 Mar 2000, mukul goyal wrote:
                          
> > TCP does not always run over e.g. some braindead ARQ-to-infinity link
> > layer. Optimising TCP for such a special case would prove detrimental
> > to interactions with other, different, link layers. It's a local
> > optimisation with many tradeoffs.
> 
> Lloyd, could you clarify this further? 

My comments are purely a response to Venkat's comments, and should not
be construed as comments on the contents of the draft.

> I thought that's what draft talking about: making TCP independent of link
> layer oddities.

what the draft and the list talk about can be very separate things.

L.

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






From owner-pilc@lerc.nasa.gov  Fri Mar  3 20:03:55 2000
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From: Marie-Jose Montpetit <marie@teledesic.com>
To: "Falk, Aaron" <afalk@PanAmSat.com>,
        "'Vernon Schryver'"
	 <vjs@calcite.rhyolite.com>
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Subject: RE: PILC ASYM document
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I would also stress that while I never believed we should change the
Internet protocols to support new links with asymmetry or delay or noise (in
fact all pieces of the bigger pie), I think it is important to have the
different communities at least sharing the issues and have the people
designing the links (the subnets) be aware of what is running end to end
over them.

Marie-Jose
-----Original Message-----
From: Falk, Aaron [mailto:afalk@PanAmSat.com]
Sent: Friday, March 03, 2000 6:03 AM
To: 'Vernon Schryver'
Cc: pilc@grc.nasa.gov
Subject: RE: PILC ASYM document


Vernon wrote: 
> 
> The practice followed by Van Jacobson, Sally Floyd, and 
> others has worked
> pretty well over the years.  They do lots of simulating, 
> testing, and more
> than prototype implementing before publishing anything, or at 
> least before
> announcing solutions or proposing IETF sanctioned changes in widely
> deployed implementations.  A recent example is the change in 
> the initial
> TCP window, which was done at least partly under the auspices 
> of the TCP
> Implementors working group.  

Vernon-

You've made an excellent point. Actually, getting researchers of the caliber
you mention is part of the goals of this group. (Sally, Van: are you
listening?) However, to get networking researchers engaged in a problem
there needs to be a clear articulation of exactly what the problem is. Part
of that discussion would be to propose some candidate solutions. This is why
PILC isn't developing any protocols -- we are supposed to be bringing
together user/developer communities
_who_may_not_be_experienced_in_Internet_protocol_R_&_D_ to document the
problems they see on links of interest. One possible outcome is that an
Informational RFC is published. Getting to that point will require a rough
consensus on the issues. The dialog will likely be of greater value than the
document. These dialogs can have profound influence decisions on IETF
activities in the future.

Some might say (& I heard it about TCPSAT), that this isn't relevant work
for the IETF, since the IETF develops protocols for the Internet. My
understanding is that the IETF's charter is not quite that narrow. PILC is
doing the advance work to identify whether a new protocol(s) needs to be
developed. Part of that process is requirements definition (such as is going
on in ECM). But before you can define requirements you need agreement on
what the issues are and whether they require fixing. *That* is what we are
doing now. 

(I might add that TCPSAT was very effective in giving the satellite
community a clue on issues surrounding TCP performance. This was/is a
community where most designers didn't have a background in Internet
protocols and didn't understand the performance/design tradeoffs. To a
certain extent, PILC is a generalization of the TCPSAT goals to a wider
community of interest.)

--aaron


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Subject: DIAL M for Mobility (Second Call for Papers)
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       (Please accept our apologies if you receive multiple copies.) 

                       SECOND CALL FOR PAPERS

           4th International Workshop on  Discrete Algorithms and
               Methods for Mobile Computing & Communications
                         (DIALM for Mobility)

              August 11, 2000  Boston, Massachusetts, USA
                  In conjunction with ACM MobiCom 2000
                      Sponsored by ACM SIGMOBILE

                   Submission Deadline:  April  25, 2000

       Workshop URL : http://www.eecis.udel.edu/~elloyd/dialm.d/home.html

SCOPE:

  Mobile computing and communications devices such as portable phones,
laptops and palmtops will have an enormous impact on our lifestyle over
the next several decades. The introduction of mobility raises a number
of new research issues. This workshop is devoted to discrete algorithms
and methods in the context of mobile and wireless computing and
communications. The workshop is intended to serve as a forum for open
discussions and lively debate, and to foster cooperation among
practitioners and theoreticians. The workshop encourages submission of
papers based on work-in-progress. Proposals for panels designed to
generate lively discussions are also invited.

  Contributions are solicited in all areas related to mobile computing and
communications where discrete algorithms and methods are utilized,
including, but not limited to:

    distributed algorithms     frequency allocation
    scheduling                 location tracking
    site allocation            multihop packet radio networks
    wireless networks          synchronization
    cryptography and security  error correcting codes
    handover (handoff)         telecommunications
    modeling                   optimization
    routing                    satellite communication



PROGRAM CHAIR:   Errol L. Lloyd
                 Department of Computer and Information Sciences
                 University of Delaware
                 Newark, DE, USA
                 Email: elloyd@udel.edu

PROGRAM COMMITTEE

       Amotz Bar-Noy, AT&T Labs and Tel Aviv University, Israel
       Anthony Ephremides, University of Maryland, USA 
       Aura Ganz, University of Massachusetts, USA 
       Juraj Hromkovic, RWTH Aachen, Germany 
       Bo Li, Hong Kong University of Science and Technology, Hong Kong
       Jason Yi-Bing Lin, National Chiao-Tung University, ROC 
       Errol Lloyd, University of Delaware,USA -- Program Chair 
       Madhav Marathe, Los Alamos National Lab 
       Marina Papatriantafilou, Chalmers Univ. of Technology, Sweden
       Stephane Perennes, INRIA, France 
       Cynthia Phillips, Sandia National Laboratories, USA 
       Balaji Raghavachari, University of Texas at Dallas, USA 
       S.S. Ravi, SUNY Albany, USA -- Publicity Chair 
       Andrea Richa, Arizona State University, USA 
       Aravind Srinivasan, Bell Labs, USA 
       Martha Steenstrup, BBN, USA 
       Subhash Suri, Washington University, USA 
       Eli Upfal, Brown University, USA 
       Peter Widmayer, ETH Zurich, Switzerland 

STEERING COMMITTEE

       Ian Akyildiz, Georgia Tech, USA 
       Maurizio Bonuccelli, University of Pisa, Italy (Chair) 
       Afonso Ferreira, CNRS - I3S - INRIA - Sophia Antipolis, France
       Arunabha Sen, Arizona State University, USA. 


SUBMISSION GUIDELINES:

   All paper submissions and panel proposals will be handled
electronically. Authors should e-mail a PostScript version of their full
paper to elloyd@udel.edu. The font size should be at least 10 pt. The
paper should not be longer than 10 single spaced pages using a standard
10point font. Panel proposals may be submitted in ASCII or PostScript
format. A panel proposal should include panel topic, sample questions
the panel would address, and proposed panel members and moderator.

   Since deadlines overlap, dual submission of papers to MobiCom and DIALM
is encouraged.  Any paper accepted for MobiCom will automatically be
removed from consideration for DIALM.

   To ensure that the PostScript versions of the papers can be
printed, use PostScript version 2 or later. Also ensure that the paper
fits on "US Letter" size paper (8.5X11 inches). Reference only standard
Adobe printer fonts (i.e., Courier, Times, Roman, or Helvetica); other
fonts may be used but must be included in the PostScript file.

   Authors will receive a short email ack of their submission within two
days of submission.  If this is not received, the authors should contact
the Program Chair immediately. Authors should separately email the
title, authors, mailing address, and abstract to elloyd@udel.edu.

   All accepted papers will appear (7 to 10 pages per paper) in the
workshop proceedings, which will be published by ACM.

IMPORTANT DATES:

    Submissions due        :  April 25, 2000
    Acceptance notification:  May 15, 2000
    Final paper due        :  June 1, 2000
    Workshop               :  August 11, 2000



From owner-pilc@lerc.nasa.gov  Mon Mar  6 05:43:06 2000
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Unknown and even rejected by many vendors and operators only a few =
months ago, SIP is on the brink of making a definitive name for itself.=20
Visit the SIP 2000 International Conference programme:
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From owner-pilc@lerc.nasa.gov  Mon Mar  6 16:27:59 2000
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From: Venkat Padmanabhan <padmanab@microsoft.com>
To: "'L.Wood@eim.surrey.ac.uk'" <L.Wood@eim.surrey.ac.uk>
Cc: pilc@grc.nasa.gov
Subject: RE: PILC ASYM document
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> That analogy is misleading.
> 
> HTTP _always_ runs over TCP, so there are widespread gains from
> optimising interaction between the two. It's a global (a World-Wide,
> even) optimisation with few tradeoffs.
> 

Good point. A more appropriate analogy may be TCP header compression 
over slow speed links wherein transport-layer specific knowledge is
exploited in the link layer of a particular class of links. I would place
techniques such as ack filtering/reconstruction (discussed in the draft)
in the same category -- link layer algorithms that utilize TCP-specific
knowledge without necessarily modifying TCP itself. In that sense,
I believe these are bang on target for PILC; basically, designers
of such links may need to spinkle some TCP-specific magic dust to
have their links perform well with unmodified end-to-end TCP.
(N.B.: The draft surveys a whole range of alternatives, some of which
do involve mods to the e2e TCP.) 

> TCP does not always run over e.g. some braindead ARQ-to-infinity link
> layer. Optimising TCP for such a special case would prove detrimental
> to interactions with other, different, link layers. It's a local
> optimisation with many tradeoffs.

Yes, it's a local optimization. But it may well be possible to 
confine it to just the problematic link (for instance, by using 
an ack reconstructor to undo the effects of ack filtering) thereby
minimizing/eliminating adverse interactions with e2e TCP.

By the way, as I mentioned in my previous message (as also during 
my presentation at the DC IETF), the problem of upstream packets
(acks in the case of a downstream transfer) being expensive to send 
may be a fundamental consequence of hub-and-spokes network architectures
rather than an artifact of braindead link layers.

-Venkat

Venkat Padmanabhan
Microsoft Research
padmanab@microsoft.com
http://www.research.microsoft.com/~padmanab




From owner-pilc@lerc.nasa.gov  Mon Mar  6 16:38:02 2000
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  >> From: Venkat Padmanabhan <padmanab@microsoft.com>
  ...
  >> in the same category -- link layer algorithms that utilize TCP-specific
  >> knowledge without necessarily modifying TCP itself. In that sense,
  >> I believe these are bang on target for PILC; basically, designers
  >> of such links may need to spinkle some TCP-specific magic dust to
  >> have their links perform well with unmodified end-to-end TCP.
  >> (N.B.: The draft surveys a whole range of alternatives, some of which
  >> do involve mods to the e2e TCP.) 

Have we exhausted alternatives that do NOT require the link
layer to understand TCP?

IMHO it is a great idea to have layers communicate information
to each other, but it does not seem a good idea to require the
link layer to understand the transport protocol.

I hope we can agree on this.

For instance, it seems reasonable to design a link layer with,
say, 2 service classes (such as, never retransmit, versus
retransmit X times) -- then each packet sent to the link layer
could be tagged with the appropriate service class to be used
for that packet

The above alternative would be preferable over requiring the
link layer to take different actions depending on whether
a packet belongs to, say, TCP or UDP.

In my opinion, before this group endorses alternatives that
require link layer to understand transport protocol, a case
needs to be made to show that more reasonable/palatable
alternatives do not exist.

Cheerio.

- nitin



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Subject: Re: Draft on Advice for Internet Subnetwork Designers 
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My thoughts on buffers and queueing didn't seem to generate much as much 
discussion as I'd hoped.  However, since the deadline is upon us, here is some 
text that Phil can insert into the draft.

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

Buffering, flow control and congestion control
Many kinds of subnets have points at which there is contention for link 
transmission resources.  At some of these points, the rate of the link may 
vary.  At such points, there must be a queueing system, including buffering, 
scheduling and a capability to discard excess subnet packets.  These queueing 
points may also be part of a subnet flow control or congestion control scheme.

For the purpose of this discussion, a layered model of the interaction between 
the IP layer and the subnet layer is not helpful.  Instead, we adopt the frame 
of reference of a packet.  At each queueing point -- whether that point is at 
the IP layer or at the subnet layer -- the packet experiences delay for a 
period of time (which depends on competing traffic and the scheduling 
discipline), and is subjected to a local discarding policy.   If there are 
flow control or congestion control mechanisms in the subnet that do not rely 
on TCP, these must operate on the scheduling discipline and/or the discarding 
policy in the subnet or in the routers at the edges of the subnet, since TCP 
has no way to operate on their semantics and IP has no way to  be able to pass 
their semantics to TCP;  however, see below for one exception.

TCP traffic, and especially aggregated TCP traffic, is bursty.  As a result, 
instantanious queue depth at a queueing point can vary dramatically, even in 
nominally stable networks.  For optimal performance, it is desirable that 
packets be dropped in a controlled fashion, rather than due to running out of 
buffers.  As a rule of thumb, nodes should be engineered to have buffering 
available for each TCP connection equal to the product of the maximum 
instantanious rate at which packets can be transmitted on the link (which is 
usually the link rate, but may be further limited by a shaper) and the 
propagation delay of the link.  For links that have variable propagation 
delays (such as having diverse paths, or variable depth interleavers), the 
longest likely propagation delay should be used.   

Unfortunately, since the maximum number of TCP connections at a queueing point 
can neither be known with certainty a priori, nor limited by an admission 
control policy (except, perhaps, at a PEP), buffer sizes in subnet devices can 
at best be engineered based on crude estimates.   In general, it is wise to 
err in favor of too much buffering rather than too little.  It may also be 
useful for subnets to incorporate mechanisms for measuring propagation delay, 
to assist in buffer sizing calculations.

There is a rough consensus in the research community that  active queue 
management is important to improving fairness, link utilization  and 
throughput [RFC2309].   Although there are questions and concerns about the 
efficacy of active queue management (e.g., see [MBDL99]), it is widely 
considered an improvement over tail-drop discard policies.   

One well known example of an active queue management the Random Early 
Detection (RED) algorithm [RED93].  RED maintains an exponential weighted 
moving average of the queue depth.  When this average queue depth is between a 
maximum threshold max_th, and a minimum threshold min_th, packets are dropped 
with a probability which is proportional to the amount by which the average 
queue depth exceeds min_th.  When this average queue depth is equal to max_th, 
the drop probability is equal to a configurable parameter max_p.  When this 
average queue depth is greater than max_th, packets are always dropped.  
Numerous variants on RED appear in the literature, and there are other active 
queue management algorithms which claim advantages over RED in various 
dimensions.

Active queue management algorithms forms a control regime wherein dropped 
packets are treated as a feedback signal.  Randomization of dropping tends to 
break up the observed tendency of TCP windows belonging to different TCP 
connections to become synchronized by correlated drops, and also imposes a 
degree of fairness on those connections which properly implement TCP 
congestion avoidance.    Another property of active queue management 
algorithms which is particularly important to subnet designers is that they 
attempt to keep average queue depths short, while accommodating large short 
term bursts. 

Since TCP neither knows nor cares whether packet loss due to congestive 
discard is due to congestion at the IP layer or in a subnet, it may be 
advisable for subnets  that perform queueing and discarding to consider 
implementing some form of active queue management.  This especially holds if 
large aggregates of TCP connections are likely to share the same queue.  
However, active queue management may be less effective in the case of many 
queues carrying smaller aggregates of TCP connections, as for  example, in an 
ATM switch that implements per-VC queueuing.  

Note,  incidentally, that the performance of active queue management 
algorithms is highly sensitive to settings of configurable parameters, and 
also to factors such as RTT [MBB00][FB00].  

Some subnets, most notably ATM, perform segmentation and reassembly at the 
edge of the subnet, so as to be able to forward subnet packets of size less 
than an MTU.  There are advantages and disadvantages to doing this.  However, 
if this is done, care should be taken in designing discard policies. Subnet 
packets with missing fragments must be destroyed by the subnet, as they are of 
no use to  TCP.  Thus, if the subnet discards random and uncorrelated 
fragments of IP  packets, then the balance of these packets constitute an 
unproductive load on the subnet and can markedly degrade end-to-end 
performance. [RF95]  Therefore, subnets should attempt to discard entire IP 
packets.  If a portion of an IP packet has been forwarded and discarding of 
subnet packets which are fragments of the IP packet becomes unavoidable, then 
either all remaining fragments or all but the fragment marking the end of the 
packet should be discarded.   For ATM subnets, this specifically means using 
Early Packet Discard and Partial Packet Discard [ATMFTM]. 


Some subnets might includes flow control mechanisms that effectively require 
that the rate of traffic flows be shaped as they enter the subnet.  One 
example of such a subnet mechanism is in the ATM Available Bitrate (ABR) 
service category [ATMFTM].   Such flow control mechanisms have the effect of 
making the subnet nearly lossless by pushing congestion into the IP routers 
edges of the subnet.  In such a case, adequate buffering and discard policies 
are needed in these routers to deal with a subnet which appears to have 
dynamically varying bandwidth.   Whether or not there is benefit in this kind 
of flow control is controversial, and there have been numerous simulation and 
analytical studies that go both ways.  It appears that some of the issues that 
lead to such different results include sensitivity to ABR parameters, use of 
binary rather than explicit rate feedback, use (or not) of per-VC queueing, 
and the specific ATM switch algorithms selected for the study.   Anecdotally, 
some large networks have used IP over ABR to carry TCP traffic, and have 
claimed it to be sucessful, but have published no results.
 
Another possible approach to flow control in the subnet would be to work with 
TCP Explicit Congestion Notification (ECN) semantics [RFC2481].  Routers at 
the edges of the subnet, rather than shaping, would set the ECN bit in those 
IP packets that are received in subnet packets that have an ECN indication.  
Nodes in the subnet would need to implement an  active queue management 
protocol which marks subnet packets rather than dropping. However, RFC2481 is 
presently experimental, and TCPs which can use ECN are not widely deployed. 


------------
References 
 
[ATMFTM] The ATM Forum, "Traffic Management Specification, Version 4.0", April 
1996, document af-tm-0056.000 (www.atmforum.com).

[FB00] Firoiu V., and Borden M., "A Study of Active Queue Management for 
Congestion Control" to appear in Infocom 2000

[MBB00] May, M., Bonald, T., and Bolot, J-C., "Analytic Evaluation of RED 
Performance" to appear INFOCOM 2000

[MBDL99] May, M., Bolot, J., Diot, C., and Lyles, B., Reasons not to deploy 
RED, technical report, June 1999.

[RF95] Romanow, A., and Floyd, S., Dynamics of TCP Traffic over ATM Networks. 
IEEE JSAC, V. 13 N. 4, May 1995, p. 633-641.

[RFC2481] Ramakrishan, K. and Floyd S.,  "A Proposal to add Explicit Congestion
          Notification (ECN) to IP" RFC2481 January 1999




From owner-pilc@lerc.nasa.gov  Mon Mar  6 18:13:56 2000
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Dan Grossman wrote:
> 
> My thoughts on buffers and queueing didn't seem to generate much as much
> discussion as I'd hoped.  However, since the deadline is upon us, here is some
> text that Phil can insert into the draft.

Nice text.

cheers,
gja
________________________________________________________________________
Grenville Armitage                            http://mh005.infi.net/~gja
Bell Labs Research Silicon Valley


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From: Phil Karn <karn@ka9q.ampr.org>
To: pilc@grc.nasa.gov
Subject: Kicking TCP
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Last week I attended an IAB workshop on wireless IP issues. One of the
topics of relevance to this group was making TCP work better over
links that can experience long outages.

Way back in the mid-to-late 1980s, when I was writing code to run
TCP/IP over amateur (ham) packet radio, I created something called a
"TCP kick" packet -- a UDP packet that would cause the recipient to
force a retransmission on all of its TCP connections to the sender. I
hooked it into a hot key so you could "kick" a connection by hitting
the key. Given TCP's exponential backoff mechanism and the flakiness
of our radio links, this was a real godsend.

I've been occasionally floating this idea in the IETF (usually
repackaged as an ICMP message), so at the workshop I was asked to
write it up as an Internet Draft. Allison Mankin suggested the term
"Source Encourage" (ICMP SE).

This scheme is designed to improve TCP performance by deliberately
short-circuiting TCP's congestion control in situations where it is
not needed. But in today's Internet, the potential for abuse of such a
scheme for denial of service mischief has to be taken seriously.  So
instead of allowing the ICMP SE message to "kick" any arbitrary set of
TCP connections, I included in it the header of the last packet to be
dropped by the router before the link came back up. The recipient of
the ICMP SE would check the fields to see that they corresponded to a
real TCP connection before forcing a retransmission. This blocks the
"blind" 3rd party denial of service attacks that are so easy to mount
today.

But then I realized that if the router is to hold onto a packet so it
can generate an ICMP SE when the link comes back up, it might as well
just forward that same packet onto its destination. This will get TCP
going again just as surely as the ICMP SE mechanism would.

So I'm thinking of inserting some text into the PILC LINK document
that would recommend that routers driving subnetworks subject to
lengthy outages (e.g., mobile radio) try to hold onto the last IP
packet of each flow when a link goes down and forward it to its
destination when the link comes back up.

A lot of storage would not be needed. One for each TCP connection is
all that's needed, with each packet overwriting the previous
one. (Obviously things are more complicated if IPSEC is in use, so the
router could store just the last packet seen between a given pair of
IP addresses). Simplifying things is the fact that most mobile radio
links carry only a small number of flows, often just one.

Note that this gives nearly all of the benefits claimed for totally
reliable link layer protocols with much less effort. The router would
not have to promise absolute reliability; if it dropped the stored
packet (e.g., due to insufficient memory), TCP would still recover
after the link comes back up, just not as quickly. Like all link layer
reliability mechanisms below TCP, this one is intended merely as
a performance enhancement, not as a reliability guarantee.

Comments?

Phil







From owner-pilc@lerc.nasa.gov  Tue Mar  7 03:54:24 2000
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To: karn@qualcomm.com
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Subject: Re: Kicking TCP 
From: Zachary Amsden <zamsden@cthulhu.engr.sgi.com>
In-Reply-To: Your message of "Mon, 06 Mar 2000 23:27:26 PST."
             <200003070727.XAA19511@homer.ka9q.ampr.org> 
Date: Mon, 06 Mar 2000 23:53:41 -0800
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Yes, kicking TCP is useful in limited situations, but you needn't go to much 
trouble at all to do it - you can easily add a kernel hack to reset the 
retransmission timer on the transmitting socket.  This will trigger 
retranmissions without losing any TCP state, and avoids any protocol 
modifications.

This serves a useful purpose on what I call "blackout" lines - lines of 
communication that temporarily go dead for several seconds and require 
complicated renegotiation (like some wireless links).  On public connections 
with interactive users, having a kick command is a very bad idea from a 
congestion point of view.

A neat solution to this problem would be to cache link characteristics with 
routes, and to tweak TCP strategy routines based on link types - i.e. a 
kick_timeout that limits the maximum retransmission backoff allowed, after 
which the retransmit timer starts it's cycle over again, or stays at the 
maximum.  (Modifying cwnd is also possible, but I wouldn't recommend it 
without analysis).

One would really need to do some serious simulation runs to analyze the 
effects of kicking connections under these circumstances.

If you still can't complete the transmission with this strategy, I can't think 
of any other strategy that would help - if you don't get any ACKs in return, 
kicking is useless.

-- 
Zachary Amsden  zamsden@engr.sgi.com  3-6919  31-2-510  Core Protocols




From owner-pilc@lerc.nasa.gov  Tue Mar  7 07:56:23 2000
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From: Marie-Jose Montpetit <marie@teledesic.com>
To: "'Dan Grossman'" <dan@dma.isg.mot.com>
Cc: "Falk, Aaron" <afalk@PanAmSat.com>,
        "'pilc@grc.nasa.gov'"
	 <pilc@grc.nasa.gov>,
        "Phil Karn (E-mail)" <karn@qualcomm.com>
Subject: RE: Draft on Advice for Internet Subnetwork Designers 
Date: Tue, 7 Mar 2000 03:54:12 -0800 
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On the topic of packet discard especially on those subnet where capacity is
expensive or limited I would add that when possible packets marked form
another subnet should be discarded before transmission on the link.

Actually this discussion raises also the whole issue of reservations. Some
sessions may be reserved so that packet are not randomly discarded but
discarded according to some priority scheme. Do we want to add RSVP or
diffserv issues to the LINK draft? Maybe in a future version?

Marie-Jose
-----Original Message-----
From: Dan Grossman [mailto:dan@dma.isg.mot.com]
Sent: Monday, March 06, 2000 1:58 PM
Cc: Falk, Aaron; 'pilc@grc.nasa.gov'; Phil Karn (E-mail)
Subject: Re: Draft on Advice for Internet Subnetwork Designers 


My thoughts on buffers and queueing didn't seem to generate much as much 
discussion as I'd hoped.  However, since the deadline is upon us, here is
some 
text that Phil can insert into the draft.

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

Buffering, flow control and congestion control
Many kinds of subnets have points at which there is contention for link 
transmission resources.  At some of these points, the rate of the link may 
vary.  At such points, there must be a queueing system, including buffering,

scheduling and a capability to discard excess subnet packets.  These
queueing 
points may also be part of a subnet flow control or congestion control
scheme.

For the purpose of this discussion, a layered model of the interaction
between 
the IP layer and the subnet layer is not helpful.  Instead, we adopt the
frame 
of reference of a packet.  At each queueing point -- whether that point is
at 
the IP layer or at the subnet layer -- the packet experiences delay for a 
period of time (which depends on competing traffic and the scheduling 
discipline), and is subjected to a local discarding policy.   If there are 
flow control or congestion control mechanisms in the subnet that do not rely

on TCP, these must operate on the scheduling discipline and/or the
discarding 
policy in the subnet or in the routers at the edges of the subnet, since TCP

has no way to operate on their semantics and IP has no way to  be able to
pass 
their semantics to TCP;  however, see below for one exception.

TCP traffic, and especially aggregated TCP traffic, is bursty.  As a result,

instantanious queue depth at a queueing point can vary dramatically, even in

nominally stable networks.  For optimal performance, it is desirable that 
packets be dropped in a controlled fashion, rather than due to running out
of 
buffers.  As a rule of thumb, nodes should be engineered to have buffering 
available for each TCP connection equal to the product of the maximum 
instantanious rate at which packets can be transmitted on the link (which is

usually the link rate, but may be further limited by a shaper) and the 
propagation delay of the link.  For links that have variable propagation 
delays (such as having diverse paths, or variable depth interleavers), the 
longest likely propagation delay should be used.   

Unfortunately, since the maximum number of TCP connections at a queueing
point 
can neither be known with certainty a priori, nor limited by an admission 
control policy (except, perhaps, at a PEP), buffer sizes in subnet devices
can 
at best be engineered based on crude estimates.   In general, it is wise to 
err in favor of too much buffering rather than too little.  It may also be 
useful for subnets to incorporate mechanisms for measuring propagation
delay, 
to assist in buffer sizing calculations.

There is a rough consensus in the research community that  active queue 
management is important to improving fairness, link utilization  and 
throughput [RFC2309].   Although there are questions and concerns about the 
efficacy of active queue management (e.g., see [MBDL99]), it is widely 
considered an improvement over tail-drop discard policies.   

One well known example of an active queue management the Random Early 
Detection (RED) algorithm [RED93].  RED maintains an exponential weighted 
moving average of the queue depth.  When this average queue depth is between
a 
maximum threshold max_th, and a minimum threshold min_th, packets are
dropped 
with a probability which is proportional to the amount by which the average 
queue depth exceeds min_th.  When this average queue depth is equal to
max_th, 
the drop probability is equal to a configurable parameter max_p.  When this 
average queue depth is greater than max_th, packets are always dropped.  
Numerous variants on RED appear in the literature, and there are other
active 
queue management algorithms which claim advantages over RED in various 
dimensions.

Active queue management algorithms forms a control regime wherein dropped 
packets are treated as a feedback signal.  Randomization of dropping tends
to 
break up the observed tendency of TCP windows belonging to different TCP 
connections to become synchronized by correlated drops, and also imposes a 
degree of fairness on those connections which properly implement TCP 
congestion avoidance.    Another property of active queue management 
algorithms which is particularly important to subnet designers is that they 
attempt to keep average queue depths short, while accommodating large short 
term bursts. 

Since TCP neither knows nor cares whether packet loss due to congestive 
discard is due to congestion at the IP layer or in a subnet, it may be 
advisable for subnets  that perform queueing and discarding to consider 
implementing some form of active queue management.  This especially holds if

large aggregates of TCP connections are likely to share the same queue.  
However, active queue management may be less effective in the case of many 
queues carrying smaller aggregates of TCP connections, as for  example, in
an 
ATM switch that implements per-VC queueuing.  

Note,  incidentally, that the performance of active queue management 
algorithms is highly sensitive to settings of configurable parameters, and 
also to factors such as RTT [MBB00][FB00].  

Some subnets, most notably ATM, perform segmentation and reassembly at the 
edge of the subnet, so as to be able to forward subnet packets of size less 
than an MTU.  There are advantages and disadvantages to doing this.
However, 
if this is done, care should be taken in designing discard policies. Subnet 
packets with missing fragments must be destroyed by the subnet, as they are
of 
no use to  TCP.  Thus, if the subnet discards random and uncorrelated 
fragments of IP  packets, then the balance of these packets constitute an 
unproductive load on the subnet and can markedly degrade end-to-end 
performance. [RF95]  Therefore, subnets should attempt to discard entire IP 
packets.  If a portion of an IP packet has been forwarded and discarding of 
subnet packets which are fragments of the IP packet becomes unavoidable,
then 
either all remaining fragments or all but the fragment marking the end of
the 
packet should be discarded.   For ATM subnets, this specifically means using

Early Packet Discard and Partial Packet Discard [ATMFTM]. 


Some subnets might includes flow control mechanisms that effectively require

that the rate of traffic flows be shaped as they enter the subnet.  One 
example of such a subnet mechanism is in the ATM Available Bitrate (ABR) 
service category [ATMFTM].   Such flow control mechanisms have the effect of

making the subnet nearly lossless by pushing congestion into the IP routers 
edges of the subnet.  In such a case, adequate buffering and discard
policies 
are needed in these routers to deal with a subnet which appears to have 
dynamically varying bandwidth.   Whether or not there is benefit in this
kind 
of flow control is controversial, and there have been numerous simulation
and 
analytical studies that go both ways.  It appears that some of the issues
that 
lead to such different results include sensitivity to ABR parameters, use of

binary rather than explicit rate feedback, use (or not) of per-VC queueing, 
and the specific ATM switch algorithms selected for the study.
Anecdotally, 
some large networks have used IP over ABR to carry TCP traffic, and have 
claimed it to be sucessful, but have published no results.
 
Another possible approach to flow control in the subnet would be to work
with 
TCP Explicit Congestion Notification (ECN) semantics [RFC2481].  Routers at 
the edges of the subnet, rather than shaping, would set the ECN bit in those

IP packets that are received in subnet packets that have an ECN indication.

Nodes in the subnet would need to implement an  active queue management 
protocol which marks subnet packets rather than dropping. However, RFC2481
is 
presently experimental, and TCPs which can use ECN are not widely deployed. 


------------
References 
 
[ATMFTM] The ATM Forum, "Traffic Management Specification, Version 4.0",
April 
1996, document af-tm-0056.000 (www.atmforum.com).

[FB00] Firoiu V., and Borden M., "A Study of Active Queue Management for 
Congestion Control" to appear in Infocom 2000

[MBB00] May, M., Bonald, T., and Bolot, J-C., "Analytic Evaluation of RED 
Performance" to appear INFOCOM 2000

[MBDL99] May, M., Bolot, J., Diot, C., and Lyles, B., Reasons not to deploy 
RED, technical report, June 1999.

[RF95] Romanow, A., and Floyd, S., Dynamics of TCP Traffic over ATM
Networks. 
IEEE JSAC, V. 13 N. 4, May 1995, p. 633-641.

[RFC2481] Ramakrishan, K. and Floyd S.,  "A Proposal to add Explicit
Congestion
          Notification (ECN) to IP" RFC2481 January 1999




From owner-pilc@lerc.nasa.gov  Tue Mar  7 07:56:26 2000
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From: Marie-Jose Montpetit <marie@teledesic.com>
To: "'Zachary Amsden'" <zamsden@cthulhu.engr.sgi.com>, karn@qualcomm.com
Cc: pilc@grc.nasa.gov
Subject: RE: Kicking TCP 
Date: Tue, 7 Mar 2000 04:02:22 -0800 
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I would tend to agree that some simulations maybe needed.

However the kicking strategy is closely related to something I have been
thinking for a while on how to "advertise" that a link is down or degraded.
ICMP extensions were also our first idea or raising the "cost" of a route
when the link is down (when multiple routes are available).

Maybe what is needed in LINK is a section on suggested mechanisms to survive
capacity degradation?

Marie-Jose

-----Original Message-----
From: Zachary Amsden [mailto:zamsden@cthulhu.engr.sgi.com]
Sent: Monday, March 06, 2000 11:54 PM
To: karn@qualcomm.com
Cc: pilc@grc.nasa.gov
Subject: Re: Kicking TCP 


Yes, kicking TCP is useful in limited situations, but you needn't go to much

trouble at all to do it - you can easily add a kernel hack to reset the 
retransmission timer on the transmitting socket.  This will trigger 
retranmissions without losing any TCP state, and avoids any protocol 
modifications.

This serves a useful purpose on what I call "blackout" lines - lines of 
communication that temporarily go dead for several seconds and require 
complicated renegotiation (like some wireless links).  On public connections

with interactive users, having a kick command is a very bad idea from a 
congestion point of view.

A neat solution to this problem would be to cache link characteristics with 
routes, and to tweak TCP strategy routines based on link types - i.e. a 
kick_timeout that limits the maximum retransmission backoff allowed, after 
which the retransmit timer starts it's cycle over again, or stays at the 
maximum.  (Modifying cwnd is also possible, but I wouldn't recommend it 
without analysis).

One would really need to do some serious simulation runs to analyze the 
effects of kicking connections under these circumstances.

If you still can't complete the transmission with this strategy, I can't
think 
of any other strategy that would help - if you don't get any ACKs in return,

kicking is useless.

-- 
Zachary Amsden  zamsden@engr.sgi.com  3-6919  31-2-510  Core Protocols




From owner-pilc@lerc.nasa.gov  Tue Mar  7 13:14:13 2000
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Nitin,

If the link handles reasonable MTUs and only understands that TCP performs
poorly when packets disappear due to errors, rather than router congestion,
then that should be enough, since that's a TCP and network-layer failing to
manage paths and inform ends well.

Alex


Nitin H Vaidya wrote:
> 
>   >> From: Venkat Padmanabhan <padmanab@microsoft.com>
>   ...
>   >> in the same category -- link layer algorithms that utilize TCP-specific
>   >> knowledge without necessarily modifying TCP itself. In that sense,
>   >> I believe these are bang on target for PILC; basically, designers
>   >> of such links may need to spinkle some TCP-specific magic dust to
>   >> have their links perform well with unmodified end-to-end TCP.
>   >> (N.B.: The draft surveys a whole range of alternatives, some of which
>   >> do involve mods to the e2e TCP.)
> 
> Have we exhausted alternatives that do NOT require the link
> layer to understand TCP?
> 
> IMHO it is a great idea to have layers communicate information
> to each other, but it does not seem a good idea to require the
> link layer to understand the transport protocol.
> 
> I hope we can agree on this.
> 
> For instance, it seems reasonable to design a link layer with,
> say, 2 service classes (such as, never retransmit, versus
> retransmit X times) -- then each packet sent to the link layer
> could be tagged with the appropriate service class to be used
> for that packet
> 
> The above alternative would be preferable over requiring the
> link layer to take different actions depending on whether
> a packet belongs to, say, TCP or UDP.
> 
> In my opinion, before this group endorses alternatives that
> require link layer to understand transport protocol, a case
> needs to be made to show that more reasonable/palatable
> alternatives do not exist.
> 
> Cheerio.
> 
> - nitin


From owner-pilc@lerc.nasa.gov  Tue Mar  7 15:42:28 2000
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Date: Tue, 07 Mar 2000 14:17:18 -0500
To: marie@teledesic.com (Marie-Jose Montpetit)
From: kscott@mitre.org (Keith Scott)
Subject: RE: Kicking TCP 
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We in fact have something akin to this in the Space Communications Protocol 
Standard (SCPS, http://www.scps.org) protocol definition.  We haven't 
tested it on any large scale yet, but the idea was to allow the link layer, 
either through inter-layer signalling or by setting an entry in a MIB, to 
inform the transport layer about the status of an outbound link.  When a 
link goes down, the network layer might reroute, but the transport layer 
could also essentially suspend the retransmission timer and probe (akin to 
persist) to determine when the link becomes available again.  Using an LRU 
queue and something like an ICMP message would allow a router serving many 
flows to inform several (hopefully all) users of a link of that link's status.

 From the SCPS-TP Specification 
(http://www.ccsds.org/blue_books.html).  The reference [10] mentioned is 
the specification for the SCPS network layer.

3.4.3 LINK OUTAGES
When the sending TCP receives an indication of a link outage, either from 
inter-layer
signaling or from a MIB-query, the sending TCP
   a) shall suspend transmission;
   b) shall suspend the retransmission timer;
   c) may continue to queue user data for transmission, to the extent that 
retransmission
         buffer space permits (this is a local implementation option);
   d) shall periodically, at a locally determined rate (recommended to be 
some small
         multiple of the previous retransmission timeout), send a 
link-probe segment,
         consisting of the first octet of unacknowledged data:
       1) if no unacknowledged data exist, no link probe shall be sent;
       2) if data are submitted for transmission during a link outage, the 
sending of link
         probes shall resume;
   e) shall not interpret link probe segments as retransmissions for the 
purposes of
         modification of the retransmission timeout or for invocation of 
slow start or
         congestion avoidance;
   f) may resume normal communication when the sending TCP
       1) receives a 'link restored' or 'link redirect' control message 
(refer to reference [10]);
       2) determines that the link is available from a MIB-query (refer to 
5.2.2.8 of this document);
       3) receives an acknowledgment that arrives at least one RTT after 
the commencement
         of the link outage.


                 --keith

At 04:02 AM 03/07/2000 -0800, Marie-Jose Montpetit wrote:
>I would tend to agree that some simulations may be needed.
>
>However the kicking strategy is closely related to something I have been
>thinking for a while on how to "advertise" that a link is down or degraded.
>ICMP extensions were also our first idea or raising the "cost" of a route
>when the link is down (when multiple routes are available).
>
>Maybe what is needed in LINK is a section on suggested mechanisms to survive
>capacity degradation?
>
>Marie-Jose

--=====================_84782300==_.ALT
Content-Type: text/html; charset="us-ascii"
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<html>
<font size=3D3><br>
We in fact have something akin to this in the Space Communications
Protocol Standard (SCPS,
<a href=3D"http://www.scps.org/" eudora=3D"autourl">http</a><a href=3D"ht=
tp://www.scps.org/" eudora=3D"autourl">://www.scps.org</a>)
protocol definition.&nbsp; We haven't tested it on any large scale yet,
but the idea was to allow the link layer, either through inter-layer
signalling or by setting an entry in a MIB, to inform the transport layer
about the status of an outbound link.&nbsp; When a link goes down, the
network layer might reroute, but the transport layer could also
essentially suspend the retransmission timer and probe (akin to persist)
to determine when the link becomes available again.&nbsp; Using an LRU
queue and something like an ICMP message would allow a router serving
many flows to inform several (hopefully all) users of a link of that
link's status.<br>
<br>
 From the SCPS-TP Specification
(<a href=3D"http://www.ccsds.org/blue_books.html" eudora=3D"autourl">http=
://www.ccsds.org/blue_books.</a><a href=3D"http://www.ccsds.org/blue_book=
s.html" eudora=3D"autourl">html</a>).&nbsp;
The reference [10] mentioned is the specification for the SCPS network
layer.<br>
<br>
3.4.3 LINK OUTAGES<br>
When the sending TCP receives an indication of a link outage, either from
inter-layer<br>
signaling or from a MIB-query, the sending TCP<br>
&nbsp; a) shall suspend transmission;<br>
&nbsp; b) shall suspend the retransmission timer;<br>
&nbsp; c) may continue to queue user data for transmission, to the extent
that retransmission<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>buffer
space permits (this is a local implementation option);<br>
&nbsp; d) shall periodically, at a locally determined rate (recommended
to be some small<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>multiple
of the previous retransmission timeout), send a link-probe segment,<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>consisting
of the first octet of unacknowledged data:<br>
</font><font face=3D"Arial, Helvetica" size=3D3>&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;
1) if no unacknowledged data exist, no link probe shall be sent;<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2) if data are submitted for transmission
during a link outage, the sending of link<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>probes
shall resume;<br>
&nbsp; e) shall not interpret link probe segments as retransmissions for
the purposes of<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>modificati=
on
of the retransmission timeout or for invocation of slow start or<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>congestion
avoidance;<br>
&nbsp; f) may resume normal communication when the sending TCP<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1) receives a =91link restored=92 or =91li=
nk
redirect=92 control message (refer to reference [10]);<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2) determines that the link is available
from a MIB-query (refer to 5.2.2.8 of this document);<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3) receives an acknowledgment that arrives
at least one RTT after the commencement<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>of the
link outage.<br>
<br>
<br>
</font><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-=
tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>--keith<br>
<br>
At 04:02 AM 03/07/2000 -0800, Marie-Jose Montpetit wrote:<br>
<blockquote type=3Dcite cite>I would tend to agree that some simulations
may be needed.<br>
<br>
However the kicking strategy is closely related to something I have
been<br>
thinking for a while on how to &quot;advertise&quot; that a link is down
or degraded.<br>
ICMP extensions were also our first idea or raising the &quot;cost&quot;
of a route<br>
when the link is down (when multiple routes are available).<br>
<br>
Maybe what is needed in LINK is a section on suggested mechanisms to
survive<br>
capacity degradation?<br>
<br>
Marie-Jose</blockquote></html>

--=====================_84782300==_.ALT--




From owner-pilc@lerc.nasa.gov  Tue Mar  7 16:07:11 2000
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To: kscott@mitre.org (Keith Scott)
Cc: pilc@grc.nasa.gov
Subject: Re: Kicking TCP 
From: Zachary Amsden <zamsden@cthulhu.engr.sgi.com>
In-Reply-To: Your message of "Tue, 07 Mar 2000 14:17:18 EST."
             <4.3.2.20000307132541.00b0e5d0@mailsrv2.mitre.org> 
Date: Tue, 07 Mar 2000 11:56:06 -0800
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There is a mechanism for sending redirects within the protocol framework 
itself - ICMP source quench can be used to indicate the loss of a flow, and 
ICMP redirect may be sent as an indicator that the flow has been restored.

-- 
Zachary Amsden  zamsden@engr.sgi.com  3-6919  31-2-510  Core Protocols




From owner-pilc@lerc.nasa.gov  Tue Mar  7 16:31:59 2000
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Date: Tue, 07 Mar 2000 15:07:54 -0500
To: zamsden@cthulhu.engr.sgi.com (Zachary Amsden)
From: kscott@mitre.org (Keith Scott)
Subject: Re: Kicking TCP 
cc: pilc@grc.nasa.gov
In-Reply-To: <200003071956.LAA16193@clock.engr.sgi.com>
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Yes, but if you do this you have to take the hit from the source quench and 
go through slow-start again.  Using our method (the ICMP-style one, say), a 
source gets notified that a link it was using has gone dead and immediately 
suspends its timers and begins probing.  When the link comes up again, 
transmission resumes with no cut in cwnd.  A selective negative 
acknowledgement (SNACK) scheme helps us recover from packets in transit 
when the link went down.  Using SACK to complement SNACK would probably do 
even better.

                 --keith`

At 11:56 AM 03/07/2000 -0800, Zachary Amsden wrote:
>There is a mechanism for sending redirects within the protocol framework
>itself - ICMP source quench can be used to indicate the loss of a flow, and
>ICMP redirect may be sent as an indicator that the flow has been restored.
>
>--
>Zachary Amsden  zamsden@engr.sgi.com  3-6919  31-2-510  Core Protocols
>

--=====================_87817148==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<font size=3>Yes, but if you do this you have to take the hit from the
source quench and go through slow-start again.&nbsp; Using our method
(the ICMP-style one, say), a source gets notified that a link it was
using has gone dead and immediately suspends its timers and begins
probing.&nbsp; When the link comes up again, transmission resumes with no
cut in cwnd.&nbsp; A selective negative acknowledgement (SNACK) scheme
helps us recover from packets in transit when the link went down.&nbsp;
Using SACK to complement SNACK would probably do even better.<br>
<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>--keith`<br>
<br>
At 11:56 AM 03/07/2000 -0800, Zachary Amsden wrote:<br>
<blockquote type=cite cite>There is a mechanism for sending redirects
within the protocol framework <br>
itself - ICMP source quench can be used to indicate the loss of a flow,
and <br>
ICMP redirect may be sent as an indicator that the flow has been
restored.<br>
<br>
-- <br>
Zachary Amsden&nbsp; zamsden@engr.sgi.com&nbsp; 3-6919&nbsp;
31-2-510&nbsp; Core Protocols<br>
<br>
</font></blockquote></html>

--=====================_87817148==_.ALT--




From owner-pilc@lerc.nasa.gov  Tue Mar  7 16:37:24 2000
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Date: Tue, 7 Mar 2000 14:12:58 -0600 (CST)
From: Ravi Prakash <ravip@utdallas.edu>
To: Zachary Amsden <zamsden@cthulhu.engr.sgi.com>
Cc: Keith Scott <kscott@mitre.org>, pilc@grc.nasa.gov
Subject: Re: Kicking TCP 
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Yes, I agree with this. We proposed a similar approach in our WCNC'99
paper (co-authored with Meghana Sahasrabudhe) for mobile ad hoc networks. 
There, we also observed that if a path is disrupted for an extended period
of time, the routing protocol will discover this fact and set the
distance metric to infinity. In ad hoc networks as each node also serves
as a router such an ICMP message can be generated locally. 

Of course, Phil's idea preceded our similar observation by several years.

Thanks,

----Ravi.


On Tue, 7 Mar 2000, Zachary Amsden
wrote:

> There is a mechanism for sending redirects within the protocol framework 
> itself - ICMP source quench can be used to indicate the loss of a flow, and 
> ICMP redirect may be sent as an indicator that the flow has been restored.
> 
> -- 
> Zachary Amsden  zamsden@engr.sgi.com  3-6919  31-2-510  Core Protocols
> 
> 
> 

========================================================================
Ravi Prakash (ravip@utdallas.edu)		www.utdallas.edu/~ravip
Department of Computer Science
The University of Texas at Dallas
Richardson, TX 75083-0688.  Phone: (972) 883-2289,   Fax: (972) 883-2349



From owner-pilc@lerc.nasa.gov  Tue Mar  7 16:48:19 2000
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To: kscott@mitre.org (Keith Scott)
Cc: pilc@grc.nasa.gov
Subject: Re: Kicking TCP 
From: Zachary Amsden <zamsden@cthulhu.engr.sgi.com>
In-Reply-To: Your message of "Tue, 07 Mar 2000 15:07:54 EST."
             <4.3.2.20000307145906.00c53cb0@mailsrv2.mitre.org> 
Date: Tue, 07 Mar 2000 12:41:33 -0800
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> Yes, but if you do this you have to take the hit from the source quench and 
> go through slow-start again.  Using our method (the ICMP-style one, say), a 
> source gets notified that a link it was using has gone dead and immediately 
> suspends its timers and begins probing.  When the link comes up again, 
> transmission resumes with no cut in cwnd.  A selective negative 
> acknowledgement (SNACK) scheme helps us recover from packets in transit 
> when the link went down.  Using SACK to complement SNACK would probably do 
> even better.
> 
>                  --keith

how about using ICMP unreachable with a new code for Network Temporarily 
unavailable - if this is received, TCP should save it's current window, 
temporarily cut the window to zero, and begin sending ACK probes at a regular 
rate.  When it receives an ACK in return (see note below), it restores it's 
old window and resumes transmission.

*Note - you could send some type of message Network Available Again, but if 
the transmitting hosts resends in response to that, it may act too eagerly - 
it may take some time for routes to be renegotiated on the other end of the 
connection, causing the data to be misrouted and lost.  That will result in a 
slow start for TCP, which is what we were trying to avoid.  Getting an ACK in 
response to a probe ensures the network is stable again.

The real question is what to use for the probe timeout value - perhaps a 
function of the computed RTT for the connection?

-- 
Zachary Amsden  zamsden@engr.sgi.com  3-6919  31-2-510  Core Protocols




From owner-pilc@lerc.nasa.gov  Tue Mar  7 17:39:59 2000
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To: kscott@mitre.org (Keith Scott)
cc: zamsden@cthulhu.engr.sgi.com (Zachary Amsden), pilc@grc.nasa.gov
Subject: Re: Kicking TCP 
In-reply-to: Your message of "Tue, 07 Mar 2000 15:07:54 EST."
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From: Dan Grossman <dan@dma.isg.mot.com>
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Seems to me that resuming without changing cwnd assumes that the congestion 
state hasn't changed during the outage.   That seems unlikely.  Wouldn't it be 
better to reduce cwnd to ssthresh?



From owner-pilc@lerc.nasa.gov  Tue Mar  7 19:17:07 2000
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From: Vernon Schryver <vjs@calcite.rhyolite.com>
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Subject: Re: Kicking TCP
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> From: Dan Grossman <dan@dma.isg.mot.com>
>
> Seems to me that resuming without changing cwnd assumes that the congestion 
>state hasn't changed during the outage.   That seems unlikely.  Wouldn't it be 
> better to reduce cwnd to ssthresh?

There is a draft that will soon advance (if it has not already) on the
standards track concerning what to do when the congestion window gets
stale for any of many reasons, including some that cover this situation.


Kicking TCP with a repeated TCP segment to trigger a delayed ACK when a
link comes back sounds plausible.  I bet I could name that tune in 30
lines in a couple of PPP implementations.  I'm not sure why it's useful,
since I would have hoped that reasonable implementations would queue
TCP packets on the far side of the link when it went down, and since the
wireless people seem so interested in monitoring the link state.

Isn't this repeating of TCP segments a very limited form of ARQ? 
Are you sure it wouldn't be better to admit as much and then ask if
partial ARQ wouldn't be just as simple and more generally effective?

Generating an ICMP packet would also be straightforward, if I could
figure out where to send it.  Would you record the last IP address
seen in the router and just send to there in the hope that all TCP
connections are to that host?

Using an ICMP Redirect to kick TCP sounds unlikely to work.  There
is too much deployed code that either sends ICMP Redirects in
completely unrelated situations or that expects to have to deal
with them in unrelated situations.  Then there are the many firewalls
and other filters that discard all ICMP Redirects on (mostly bogus)
security grounds before they could get to the TCP peer.

What with firewall, filters, and NAT boxes, I don't see how you can hope
to get anything to the TCP remote peer often enough to do much good, except
an IP packet that looks very much like other recent IP (i.e. TCP) packets.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Wed Mar  8 03:22:26 2000
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Date: Tue, 7 Mar 2000 23:22:26 -0800
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From: Phil Karn <karn@ka9q.ampr.org>
To: marie@teledesic.com
CC: zamsden@cthulhu.engr.sgi.com, pilc@grc.nasa.gov
In-reply-to: <25D0C66E6D25D311B2AC0008C7913EE07B810F@tdmail2.teledesic.com>
	(message from Marie-Jose Montpetit on Tue, 7 Mar 2000 04:02:22 -0800)
Subject: Re: Kicking TCP
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>However the kicking strategy is closely related to something I have been
>thinking for a while on how to "advertise" that a link is down or degraded.

You can already do that with the ICMP Destination Unreachable (down)
or Source Quench (degraded) messages. But some would claim that no explicit
notification is really needed; packet drops will tell TCP all it needs to
know.

Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 03:22:27 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: dan@dma.isg.mot.com
CC: marie@teledesic.com, pilc@grc.nasa.gov, afalk@PanAmSat.com
In-reply-to: <200003022253.RAA23882@noah.dma.isg.mot.com> (message from Dan
	Grossman on Thu, 02 Mar 2000 17:53:32 -0500)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
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>The model I've been working with (especially since that side discussion) is 
>that if the subnet does queueing (buffering, scheduling, discarding), then the 
>queueing systems in the subnet has exactly the same effect as queues in 
>routers.  This may be worth stating.  Thus, recommendations about having a 

Yes. In addition, a subnet that provides lots of buffering must also
provide QoS hooks that the IP layer can use. Otherwise, meaningful
IP-level QoS will be difficult or impossible to provide.

Consider a subnet that can buffer, say, 1 sec of traffic but only
provides FIFO service. Then 1 second becomes the minimum delay that
the router driving the link can provide to any IP packet regardless of
how important its TOS bits say it is.

I ran into this problem in the early 1990s when dialup modems began to
incorporate error control with internal buffers holding several
seconds of data. Once the router sent something to the modem, it could
not say "hey, take this new packet and put it ahead of the ones I just
sent you". Even though my router gave telnet traffic priority to ftp
traffic, telnet sessions performed poorly because each telnet packet
had to wait for all the FTP data in the modem's internal queue to
drain first.

I'm not sure what to say about this in the draft except that subnets
that implement more than a minimal amount of internal queuing should
probably provide some sort of QoS hook that IP can use to implement
its own QoS mechanisms.

Phil



From owner-pilc@lerc.nasa.gov  Wed Mar  8 03:28:18 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: zamsden@cthulhu.engr.sgi.com
CC: pilc@grc.nasa.gov
In-reply-to: <200003070753.XAA21225@clock.engr.sgi.com> (message from Zachary
	Amsden on Mon, 06 Mar 2000 23:53:41 -0800)
Subject: Re: Kicking TCP
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>Yes, kicking TCP is useful in limited situations, but you needn't go to much 
>trouble at all to do it - you can easily add a kernel hack to reset the 
>retransmission timer on the transmitting socket.  This will trigger 
>retranmissions without losing any TCP state, and avoids any protocol 
>modifications.

Yes, I know it's easy to reset the TCP retransmission timer, but what
would cause that to happen? That was the intent of my ICMP Source Encourage
message.

What I'm really looking for comment on is whether the same goal of
getting TCP going again quickly can be achieved more transparently by
having the subnet hold the last packet of a stream through a link
outage, and forward it normally when the link comes back up.

PHil



From owner-pilc@lerc.nasa.gov  Wed Mar  8 03:32:49 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: zamsden@cthulhu.engr.sgi.com
CC: kscott@mitre.org, pilc@grc.nasa.gov
In-reply-to: <200003071956.LAA16193@clock.engr.sgi.com> (message from Zachary
	Amsden on Tue, 07 Mar 2000 11:56:06 -0800)
Subject: Re: Kicking TCP
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>ICMP redirect may be sent as an indicator that the flow has been restored.

I don't think this is the intended use of ICMP redirect. According to
RFC1122, a redirect will modify a routing table, but it is not one of
the ICMP messages passed upstairs to the transport layer. Are you
proposing that it taken on a new meaning?

I'm not necessarily opposed to such a proposal, but the LINK document
is supposed to concentrate on telling subnet designers what they
should do, complementing the other work in PILC on possible changes to
the Internet protocols to better support unconventional subnetwork layers.
Perhaps your suggestion should be directed to one of the other documents?

Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 03:39:01 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: zamsden@cthulhu.engr.sgi.com
CC: kscott@mitre.org, pilc@grc.nasa.gov
In-reply-to: <200003072041.MAA16815@clock.engr.sgi.com> (message from Zachary
	Amsden on Tue, 07 Mar 2000 12:41:33 -0800)
Subject: Re: Kicking TCP
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>how about using ICMP unreachable with a new code for Network Temporarily 
>unavailable - if this is received, TCP should save it's current window, 
>temporarily cut the window to zero, and begin sending ACK probes at a regular 
>rate.  When it receives an ACK in return (see note below), it restores it's 
>old window and resumes transmission.

I agree with Dan Grossman; I'm not sure you want to save and restore
the old TCP state, as opposed to letting it restart slow-start.

During the outage state, there is probably little harm in letting the
sending TCP probe the link with normal retransmissions. Because each
retransmission will be backed off exponentially, the load will be
limited even if full packets are used in the probes.

Phil





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From: Phil Karn <karn@ka9q.ampr.org>
To: vjs@calcite.rhyolite.com
CC: pilc@grc.nasa.gov
In-reply-to: <200003072326.QAA26350@calcite.rhyolite.com> (message from Vernon
	Schryver on Tue, 7 Mar 2000 16:26:05 -0700 (MST))
Subject: Re: Kicking TCP
Reply-to: karn@qualcomm.com
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>Isn't this repeating of TCP segments a very limited form of ARQ? 
>Are you sure it wouldn't be better to admit as much and then ask if
>partial ARQ wouldn't be just as simple and more generally effective?

It's not quite the same as partial ARQ. I'm assuming that the node
with an interface on the downed link would have an out-of-band
indication from the link level driver whether the link is up or down.
And when it's down, it wouldn't even try to pass a queued packet to
the link driver. It would just hold onto the most recent queued packet
on each flow (e.g., TCP connection), overwriting it with any
subsequent retransmission and wait for the link to come back up. When
it does, it would forward the packet normally. Whether the packet is
just a TCP ack or it also contains data, it should stimulate the
sending TCP into resuming normal operation.

Phil






From owner-pilc@lerc.nasa.gov  Wed Mar  8 03:56:46 2000
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Date: Wed, 8 Mar 2000 00:00:18 -0800
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From: Phil Karn <karn@ka9q.ampr.org>
To: dan@dma.isg.mot.com
CC: afalk@PanAmSat.com, pilc@grc.nasa.gov
In-reply-to: <200003012217.RAA16642@noah.dma.isg.mot.com> (message from Dan
	Grossman on Wed, 01 Mar 2000 17:17:27 -0500)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
Reply-to: karn@qualcomm.com
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>1) Shortly before the Washington meeting, I sent a suggestion to change the 
>Introduction/Overview section as follows.  

>> ...seminal paper "End-to-End Arguments in System Design" [SRC81]. This paper 
>> argued that many -- if not most -- network functions are best implemented on 
>> an end-to-end basis, i.e., at the higher protocol layers.  Duplicating these 
>> functions at the lower levels can be needlessly redundant or harmful. 
>> However,  there are also cases where duplication can be justified by 
>> considerations such  as performance, availability or security.  Examples 
>> might include link layer  retransmission on an unusually lossy channel, e.g., 
>> mobile radio, encryption  of the entire link layer protocol over a shared 
>> medium, or  redundant paths at the link layer for fault recovery.  
>> Duplication of functions should be done only based on an understanding of 
>> system level implications, including any side-effects on the higher layers.

>I still believe rather strongly that this is more appropriate than the current 
>wording.  Since it has been quite a while since the last discussion of the 
>document on the mailing list,  a reminder seems appropriate.

Okay, here's my current proposed version of this paragraph. I've
adopted your language with a few minor changes. What do you think?

A major consideration in the efficient design of any layered
communication network are the appropriate layer(s) in which to
implement a given feature. This issue was first addressed in the
seminal paper "End-to-End Arguments in System Design" [SRC81]. This
paper argued that many -- if not most -- network functions are best
implemented on an end-to-end basis, i.e., at the higher protocol
layers.  Duplicating these functions at the lower levels can be
needlessly redundant or harmful. However, there are cases where
duplication can be justified by performance, availability or security
considerations. Examples include link layer retransmission on an
unusually lossy channel, e.g., mobile radio; link level encryption
intended to thwart traffic analysis; and redundant transmission
links to improve availability. Duplication of functions should be
done only with an understanding of system level implications, including
possible interactions with higher layer mechanisms.

--Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 04:51:10 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: dan@dma.isg.mot.com
CC: afalk@PanAmSat.com, pilc@grc.nasa.gov
In-reply-to: <200003062158.QAA18917@noah.dma.isg.mot.com> (message from Dan
	Grossman on Mon, 06 Mar 2000 16:58:11 -0500)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
Reply-to: karn@qualcomm.com
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>If there are 
>flow control or congestion control mechanisms in the subnet that do not rely 
>on TCP, these must operate on the scheduling discipline and/or the discarding 
>policy in the subnet or in the routers at the edges of the subnet, since TCP 
>has no way to operate on their semantics and IP has no way to  be able to pass 
>their semantics to TCP;  however, see below for one exception.

I am editing your text into the Buffering, Flow Control and Congestion Control
section, but I am having trouble parsing the above paragraph. Can you explain
and/or clarify it?

Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 05:00:52 2000
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Date: Wed, 8 Mar 2000 00:54:31 -0800
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From: Phil Karn <karn@ka9q.ampr.org>
To: pilc@grc.nasa.gov
Subject: Updated LINK draft
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Okay gang, I've updated the current version of the LINK document on my
web page.  It incorporates Dan Grossman's recommended new language on the
end-to-end principle, and it incorporates (in slightly edited form)
Dan Grossman's section on subnetwork buffering, flow control and
congestion control.

I'm still actively digesting the other comments and incorporating them into
the text, but I wanted to update the rather stale version of the document that's
been on my web page since the DC IETF.

The URL is

http://people.qualcomm.com/karn/pilc.txt

--Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 05:20:02 2000
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From: Phil Karn <karn@ka9q.ampr.org>
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In-reply-to: <38BA7E19.CB9D89B1@erg.abdn.ac.uk> (message from Dr G Fairhurst
	on Mon, 28 Feb 2000 15:06:57 +0100)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
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>PAGE 4: MTU Size

>It may be useful to note that using IPSEC in tunnel mode can increase
>the size of a packet as it travels across the network.  This can result
>in a smaller MTU.

I'm not sure what to do about this one. I'm inclined to agree with the
comments that say this is not unique to IPSEC but is a consideration with any
IP-in-IP encapsulation scheme -- all of which are at higher layers than the subnet.

Perhaps some text would be appropriate that says that while 1500 bytes
is an important MTU in the Internet because of the prevalence of
Ethernet, it would be useful to support a somewhat larger size to avoid IP fragmentation
when tunneling (which would include IPSEC) is used. Problem is, this still doesn't
avoid the fragmentation that would occur when the path between the tunnel endpoints
includes one or more Ethernets. So the benefits of a larger MTU are apt to be limited anyway.

Phil



From owner-pilc@lerc.nasa.gov  Wed Mar  8 08:37:45 2000
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Date: Wed, 8 Mar 2000 05:40:10 -0700 (MST)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Phil Karn <karn@ka9q.ampr.org>


> >Isn't this repeating of TCP segments a very limited form of ARQ? 
> >Are you sure it wouldn't be better to admit as much and then ask if
> >partial ARQ wouldn't be just as simple and more generally effective?
>
> It's not quite the same as partial ARQ.

ok, call it "sort of an imitation of a start of ARQ".

>                                         I'm assuming that the node
> with an interface on the downed link would have an out-of-band
> indication from the link level driver whether the link is up or down.
> And when it's down, it wouldn't even try to pass a queued packet to
> the link driver. It would just hold onto the most recent queued packet
> on each flow (e.g., TCP connection), overwriting it with any
> subsequent retransmission and wait for the link to come back up. When
> it does, it would forward the packet normally. Whether the packet is
> just a TCP ack or it also contains data, it should stimulate the
> sending TCP into resuming normal operation.

How does that differ from what a reasonable demand dialing PPP
implementation does the link goes down unexpected, whether because the
peer took it down or the modems (or ISDN line etc.) hiccuped?  A reasonable
PPP implementation queues the most recent N (for some reasonable N>0, but
N>typical window/negotiated MTU is good) and sends them when it can (the
link comes up or back up).  A PPP implementation that does not do this
queuing is a pain to use (or so I think), because it loses the traffic
that caused it to demand dial.  Unfortunately, many demand dialing PPP
implementations are unreasonable by this criterion.

Besides, why would there be any worry about memory space or queuing
mechanisms?  A slow-speed link router would need reasonable queuing memory
and mechanisms to handle at least one TCP window's worth of TCP/IP packets.
Even Microsoft and Sun think a TCP window should be at least 4 or 16K.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Wed Mar  8 09:36:57 2000
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Date: Wed, 08 Mar 2000 08:29:24 -0500
To: vjs@calcite.rhyolite.com (Vernon Schryver), pilc@grc.nasa.gov
From: kscott@mitre.org (Keith Scott)
Subject: Re: Kicking TCP
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At 04:26 PM 03/07/2000 -0700, Vernon Schryver wrote:

[snip]
>Generating an ICMP packet would also be straightforward, if I could
>figure out where to send it.  Would you record the last IP address
>seen in the router and just send to there in the hope that all TCP
>connections are to that host?
[snip]

Our notion was to keep an LRU queue of flows that use that link, and to 
send "Link Outage" messages to each of the sources.

                 --keith

--=====================_150315963==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<font size=3>At 04:26 PM 03/07/2000 -0700, Vernon Schryver wrote:<br>
<br>
</font>[snip]<br>
<blockquote type=cite cite><font size=3>Generating an ICMP packet would
also be straightforward, if I could<br>
figure out where to send it.&nbsp; Would you record the last IP
address<br>
seen in the router and just send to there in the hope that all TCP<br>
connections are to that host?</blockquote>[snip]<br>
<br>
Our notion was to keep an LRU queue of flows that use that link, and to
send &quot;Link Outage&quot; messages to each of the sources.<br>
<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>--keith<br>
</font></html>

--=====================_150315963==_.ALT--




From owner-pilc@lerc.nasa.gov  Wed Mar  8 09:42:28 2000
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        dan@dma.isg.mot.com (Dan Grossman)
From: kscott@mitre.org (Keith Scott)
Subject: Re: Kicking TCP 
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Responding to multiple messages:

Zachary: Your new ICMP message type is very similar to the "Link Outage" in 
the SCPS draft, but using a received ACK instead of a "Link Restored" 
ICMP/MIB Query to notify TCP that the link is back?  OK.  Although if you 
simply 'resotre your old window and resume transmission' you'll have the 
same problem of being too aggressive, no?

The points about being too aggressive on a "Link Restored" are quite 
valid.  In some special cases it's justified (e.g. the link that goes out 
does so for a short period of time then comes back up; routes don't change 
and there aren't too many other flows/enough time that they can eat up the 
bandwidth from suspended flows).  For specific environments like some 
spacecraft comms, this is valid.  Could also be valid for some last-hop 
wireless scenarios (Phil?).

If the link is out for too long, there are a lot of flows sharing routers 
used by suspended flows, etc., then resuming at full speed is probably too 
aggressive (and is bursty regardless), and you would definitely want to 
reduce cwnd or take some other intelligent action.


                 --keith

At 12:41 PM 03/07/2000 -0800, Zachary Amsden wrote:
>how about using ICMP unreachable with a new code for Network Temporarily
>unavailable - if this is received, TCP should save it's current window,
>temporarily cut the window to zero, and begin sending ACK probes at a regular
>rate.  When it receives an ACK in return (see note below), it restores it's
>old window and resumes transmission.
>
>*Note - you could send some type of message Network Available Again, but if
>the transmitting hosts resends in response to that, it may act too eagerly -
>it may take some time for routes to be renegotiated on the other end of the
>connection, causing the data to be misrouted and lost.  That will result in a
>slow start for TCP, which is what we were trying to avoid.  Getting an ACK in
>response to a probe ensures the network is stable again.
>
>The real question is what to use for the probe timeout value - perhaps a
>function of the computed RTT for the connection?
>
>--
>Zachary Amsden  zamsden@engr.sgi.com  3-6919  31-2-510  Core Protocols


At 04:50 PM 03/07/2000 -0500, Dan Grossman wrote:
>Seems to me that resuming without changing cwnd assumes that the congestion
>state hasn't changed during the outage.   That seems unlikely.  Wouldn't 
>it be
>better to reduce cwnd to ssthresh?

--=====================_150678607==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<font size=3>Responding to multiple messages:<br>
<br>
Zachary: Your new ICMP message type is very similar to the &quot;Link
Outage&quot; in the SCPS draft, but using a received ACK instead of a
&quot;Link Restored&quot; ICMP/MIB Query to notify TCP that the link is
back?&nbsp; OK.&nbsp; Although if you simply 'resotre your old window and
resume transmission' you'll have the same problem of being too
aggressive, no?<br>
<br>
The points about being too aggressive on a &quot;Link Restored&quot; are
quite valid.&nbsp; In some special cases it's justified (e.g. the link
that goes out does so for a short period of time then comes back up;
routes don't change and there aren't too many other flows/enough time
that they can eat up the bandwidth from suspended flows).&nbsp; For
specific environments like some spacecraft comms, this is valid.&nbsp;
Could also be valid for some last-hop wireless scenarios (Phil?).<br>
<br>
If the link is out for too long, there are a lot of flows sharing routers
used by suspended flows, etc., then resuming at full speed is probably
too aggressive (and is bursty regardless), and you would definitely want
to reduce cwnd or take some other intelligent action.<br>
<br>
<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>--keith<br>
<br>
At 12:41 PM 03/07/2000 -0800, Zachary Amsden wrote:<br>
<blockquote type=cite cite>how about using ICMP unreachable with a new
code for Network Temporarily <br>
unavailable - if this is received, TCP should save it's current window,
<br>
temporarily cut the window to zero, and begin sending ACK probes at a
regular <br>
rate.&nbsp; When it receives an ACK in return (see note below), it
restores it's <br>
old window and resumes transmission.<br>
<br>
*Note - you could send some type of message Network Available Again, but
if <br>
the transmitting hosts resends in response to that, it may act too
eagerly - <br>
it may take some time for routes to be renegotiated on the other end of
the <br>
connection, causing the data to be misrouted and lost.&nbsp; That will
result in a <br>
slow start for TCP, which is what we were trying to avoid.&nbsp; Getting
an ACK in <br>
response to a probe ensures the network is stable again.<br>
<br>
The real question is what to use for the probe timeout value - perhaps a
<br>
function of the computed RTT for the connection?<br>
<br>
-- <br>
Zachary Amsden&nbsp; zamsden@engr.sgi.com&nbsp; 3-6919&nbsp;
31-2-510&nbsp; Core Protocols<br>
</font></blockquote><br>
<br>
<font size=3>At 04:50 PM 03/07/2000 -0500, Dan Grossman wrote:<br>
<blockquote type=cite cite>Seems to me that resuming without changing
cwnd assumes that the congestion <br>
state hasn't changed during the outage.&nbsp;&nbsp; That seems
unlikely.&nbsp; Wouldn't it be <br>
better to reduce cwnd to ssthresh?<br>
</font></blockquote></html>

--=====================_150678607==_.ALT--




From owner-pilc@lerc.nasa.gov  Wed Mar  8 09:45:12 2000
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To: karn@qualcomm.com
cc: afalk@PanAmSat.com, pilc@grc.nasa.gov
Subject: Re: Draft on Advice for Internet Subnetwork Designers 
In-reply-to: Your message of "Wed, 08 Mar 2000 00:00:18 EST."
             <200003080800.AAA27815@homer.ka9q.ampr.org> 
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Phil,
Looks good.  Thanks.
Dan
> >1) Shortly before the Washington meeting, I sent a suggestion to change the 
> >Introduction/Overview section as follows.  
> 
> >> ...seminal paper "End-to-End Arguments in System Design" [SRC81]. This paper 
> >> argued that many -- if not most -- network functions are best implemented on 
> >> an end-to-end basis, i.e., at the higher protocol layers.  Duplicating these 
> >> functions at the lower levels can be needlessly redundant or harmful. 
> >> However,  there are also cases where duplication can be justified by 
> >> considerations such  as performance, availability or security.  Examples 
> >> might include link layer  retransmission on an unusually lossy channel, e.g., 
> >> mobile radio, encryption  of the entire link layer protocol over a shared 
> >> medium, or  redundant paths at the link layer for fault recovery.  
> >> Duplication of functions should be done only based on an understanding of 
> >> system level implications, including any side-effects on the higher layers.
> 
> >I still believe rather strongly that this is more appropriate than the current 
> >wording.  Since it has been quite a while since the last discussion of the 
> >document on the mailing list,  a reminder seems appropriate.
> 
> Okay, here's my current proposed version of this paragraph. I've
> adopted your language with a few minor changes. What do you think?
> 
> A major consideration in the efficient design of any layered
> communication network are the appropriate layer(s) in which to
> implement a given feature. This issue was first addressed in the
> seminal paper "End-to-End Arguments in System Design" [SRC81]. This
> paper argued that many -- if not most -- network functions are best
> implemented on an end-to-end basis, i.e., at the higher protocol
> layers.  Duplicating these functions at the lower levels can be
> needlessly redundant or harmful. However, there are cases where
> duplication can be justified by performance, availability or security
> considerations. Examples include link layer retransmission on an
> unusually lossy channel, e.g., mobile radio; link level encryption
> intended to thwart traffic analysis; and redundant transmission
> links to improve availability. Duplication of functions should be
> done only with an understanding of system level implications, including
> possible interactions with higher layer mechanisms.
> 
> --Phil




From owner-pilc@lerc.nasa.gov  Wed Mar  8 10:39:56 2000
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To: karn@qualcomm.com
cc: afalk@PanAmSat.com, pilc@grc.nasa.gov
Subject: Re: Draft on Advice for Internet Subnetwork Designers 
In-reply-to: Your message of "Wed, 08 Mar 2000 00:28:33 EST."
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> >If there are 
> >flow control or congestion control mechanisms in the subnet that do not rely 
> >on TCP, these must operate on the scheduling discipline and/or the discarding 
> >policy in the subnet or in the routers at the edges of the subnet, since TCP 
> >has no way to operate on their semantics and IP has no way to  be able to pass 
> >their semantics to TCP;  however, see below for one exception.
> 
> I am editing your text into the Buffering, Flow Control and Congestion Control
> section, but I am having trouble parsing the above paragraph. Can you explain
> and/or clarify it?
> 
> Phil
Yeah, that is a little awkward, and run-on.  Try:

In addition, some subnets may have flow control or congestion control 
mechanisms in addition to packet dropping and reliance on TCP behavior.  Such 
mechanisms can operate on components in the subnet layer, such as schedulers, 
shapers or discarders, and can  affect the operation of IP forwarders at the 
edges of the subnet.  However, with the exception of  RFC2481 explicit 
congestion notification (which will be discussed below), IP has no way to pass 
explicit congestion or flow control signals to TCP, and TCP would not react to 
such signals if they were available. 

Dan



From owner-pilc@lerc.nasa.gov  Wed Mar  8 11:48:10 2000
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Date: Wed, 08 Mar 2000 16:48:37 +0100
To: karn@qualcomm.com
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Why we don't need ICMP SE? (was Re: Kicking TCP)
Cc: pilc@grc.nasa.gov
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At 08:27 3/7/00 , Phil Karn wrote:
>[...]
>But then I realized that if the router is to hold onto a packet so it
>can generate an ICMP SE when the link comes back up, it might as well
>just forward that same packet onto its destination. This will get TCP
>going again just as surely as the ICMP SE mechanism would.

So why not do exactly that?

We have proposed a "link layer version" of that idea in
http://iceberg.cs.berkeley.edu/papers/Ludwig-WINET99/ (the server is 
currently not accessible - that will be fixed soon), and discuss the link 
outage issue in section 5 of that paper. I have additional TCP trace plots 
that show how nicely this appoach works in case of link outages. I can make 
those traces available if people are interested.

If you do this at the link layer instead of the network layer you are free 
to queue more than just a single packet, e.g., you can queue the entire 
flight of packets belonging to a TCP flow if resources allow for that. In 
situations where the link outage was not long enough to trigger a timeout, 
no packets need to retransmitted at all! And remember, with TCP's 
super-conservative retransmission timer you can afford quite some link 
outages without triggering a timeout.

>So I'm thinking of inserting some text into the PILC LINK document
>that would recommend that routers driving subnetworks subject to
>lengthy outages (e.g., mobile radio) try to hold onto the last IP
>packet of each flow when a link goes down and forward it to its
>destination when the link comes back up.

Agree. However, I would leave it open whether this approach is implemented 
at the router level (only buffering a single packet) or the link layer. It 
should also be mentioned that no packet should be buffered longer than the 
Maximum Segment Lifetime (MSL), i.e., 240 seconds, at most.

>Note that this gives nearly all of the benefits claimed for totally
>reliable link layer protocols with much less effort.

Don't agree. With the router level approach you need to change router 
implementation (deployment concern!). The link layer approach can be 
designed into upcoming wireless systems.

>The router would
>not have to promise absolute reliability; if it dropped the stored
>packet (e.g., due to insufficient memory), TCP would still recover
>after the link comes back up, just not as quickly. Like all link layer
>reliability mechanisms below TCP, this one is intended merely as
>a performance enhancement, not as a reliability guarantee.

With Dan's section referencing the e2e argument paper, we know that a 
reliability guarantee can only be provided by an end point, not a link 
layer. I.e., highly persistent link layer error recovery (limited to the 
class of reliable flows to avoid head-of-line blocking for delay-sensitive 
flows) does not provide a reliability guarantee but is also merely a 
performance enhancement.

///Reiner



From owner-pilc@lerc.nasa.gov  Wed Mar  8 11:48:18 2000
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Date: Wed, 8 Mar 2000 08:37:17 -0700 (MST)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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To: pilc@grc.nasa.gov
Subject: Re: Kicking TCP
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> From: kscott@mitre.org (Keith Scott)

> ...
> Our notion was to keep an LRU queue of flows that use that link, and to 
> send "Link Outage" messages to each of the sources.

Which router would send the messages, the near or far side of the link
from the point of view of the kick packet?

If it is the far side, then wouldn't it be a lot simpler and more
effective to either

 1. have the link layer tell the TCP state machines to retransmit any
  unacknowledged TCP bytes by forcing their retransmission timers
  to zero?  This is a layering violation, but I trust most of us
  are Sunday-only worshipers in that religion.

  If you want to get fancy, you could fail to reduce the congestion window.


 2. do what is common (albeit not common enough) and reasonable in in PPP
  implementations, which is to have a reasonable sized transmit queue 
  ( > typical window/MTU), and only send packets when the link seems
  healthy?  Don't you need such a queue to deal with bursts, so that the
  only modification would be to not drop IP packets when the link is
  broken, as is typical in link layer drivers, including Ethernet, FDDI,
  and 802.5?


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Wed Mar  8 11:59:22 2000
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From: Nitin H Vaidya <vaidya@cs.tamu.edu>
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To: karn@qualcomm.com, zamsden@cthulhu.engr.sgi.com
Subject: Re: Kicking TCP
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  >> >how about using ICMP unreachable with a new code for Network Temporarily 
  >> >unavailable - if this is received, TCP should save it's current window, 
  >> >temporarily cut the window to zero, and begin sending ACK probes at a regular 
  >> >rate.  When it receives an ACK in return (see note below), it restores it's 
  >> >old window and resumes transmission.

You might be interested in a paper we wrote on TCP for mobile
ad hoc networks in Mobicom'99, which bahaves similar to what you
suggest.

Regards.

- nitin



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From: Phil Karn <karn@ka9q.ampr.org>
To: dan@dma.isg.mot.com
CC: afalk@PanAmSat.com, pilc@grc.nasa.gov
In-reply-to: <200003081434.JAA18285@noah.dma.isg.mot.com> (message from Dan
	Grossman on Wed, 08 Mar 2000 09:34:40 -0500)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
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>In addition, some subnets may have flow control or congestion control 
>mechanisms in addition to packet dropping and reliance on TCP behavior.  Such 
>mechanisms can operate on components in the subnet layer, such as schedulers, 
>shapers or discarders, and can  affect the operation of IP forwarders at the 
>edges of the subnet.  However, with the exception of  RFC2481 explicit 
>congestion notification (which will be discussed below), IP has no way to pass 
>explicit congestion or flow control signals to TCP, and TCP would not react to 
>such signals if they were available. 

Ah, much better! I'll stick it in.

Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 12:46:39 2000
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From: Phil Karn <karn@ka9q.ampr.org>
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In-reply-to: <38BA7E19.CB9D89B1@erg.abdn.ac.uk> (message from Dr G Fairhurst
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Subject: Re: Draft on Advice for Internet Subnetwork Designers
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Gorry,

>Reference: MPEG-2 or DVB reference.

I'm editing in your text. Can you provide cites to these reference(s)?

Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 12:49:49 2000
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Date: Wed, 08 Mar 2000 10:45:31 -0600
From: Vincent Magret <vincent.magret@usa.alcatel.com>
Organization: Alcatel USA, Inc.
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Why sending a probe when the information could be sent for the node in the
network that monitors the broken link. Just put the TCP sender connection in idle
state for a certain time, after which the connection will be broke down. The
question is then How long?

Vincent

Zachary Amsden wrote:

> > Yes, but if you do this you have to take the hit from the source quench and
> > go through slow-start again.  Using our method (the ICMP-style one, say), a
> > source gets notified that a link it was using has gone dead and immediately
> > suspends its timers and begins probing.  When the link comes up again,
> > transmission resumes with no cut in cwnd.  A selective negative
> > acknowledgement (SNACK) scheme helps us recover from packets in transit
> > when the link went down.  Using SACK to complement SNACK would probably do
> > even better.
> >
> >                  --keith
>
> how about using ICMP unreachable with a new code for Network Temporarily
> unavailable - if this is received, TCP should save it's current window,
> temporarily cut the window to zero, and begin sending ACK probes at a regular
> rate.  When it receives an ACK in return (see note below), it restores it's
> old window and resumes transmission.
>
> *Note - you could send some type of message Network Available Again, but if
> the transmitting hosts resends in response to that, it may act too eagerly -
> it may take some time for routes to be renegotiated on the other end of the
> connection, causing the data to be misrouted and lost.  That will result in a
> slow start for TCP, which is what we were trying to avoid.  Getting an ACK in
> response to a probe ensures the network is stable again.
>
> The real question is what to use for the probe timeout value - perhaps a
> function of the computed RTT for the connection?
>
> --
> Zachary Amsden  zamsden@engr.sgi.com  3-6919  31-2-510  Core Protocols

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From owner-pilc@lerc.nasa.gov  Wed Mar  8 13:47:37 2000
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Perhaps we should be considering at least the FDDI MTU of 4500B as a basis for
future statements?  This would cover all Ethernet-sourced pkts that get
tunnelled.  Routers from the major manuf. have interface specs for which this
is no problem.  Of course, the Ethernet folks are talking of larger pkts as
well.

Alex


Phil Karn wrote:
> 
> >PAGE 4: MTU Size
> 
> >It may be useful to note that using IPSEC in tunnel mode can increase
> >the size of a packet as it travels across the network.  This can result
> >in a smaller MTU.
> 
> I'm not sure what to do about this one. I'm inclined to agree with the
> comments that say this is not unique to IPSEC but is a consideration with any
> IP-in-IP encapsulation scheme -- all of which are at higher layers than the subnet.
> 
> Perhaps some text would be appropriate that says that while 1500 bytes
> is an important MTU in the Internet because of the prevalence of
> Ethernet, it would be useful to support a somewhat larger size to avoid IP fragmentation
> when tunneling (which would include IPSEC) is used. Problem is, this still doesn't
> avoid the fragmentation that would occur when the path between the tunnel endpoints
> includes one or more Ethernets. So the benefits of a larger MTU are apt to be limited anyway.
> 
> Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 13:51:52 2000
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FYI, a related suggestion was for IP to communicate info TCP could use by
passing up the IP ID (v4) via TCP's Urgent Pointer (with URG bit off) and the
receiving TCP would so get the ID of the packet sent, then return that ID when
it Acked that pkt.  The sender keeps a list of IDs vs TCP pkts and then knows
exactly what TCP packet was acked, even if there is a false retransmission
(due to a short timer, etc.) and can avoid going into backoffs simply because
of misjudgments of RTT, etc.  IPv6, of course, has messed this up (no ID).

Alex


Phil Karn wrote:
> 
> >If there are
> >flow control or congestion control mechanisms in the subnet that do not rely
> >on TCP, these must operate on the scheduling discipline and/or the discarding
> >policy in the subnet or in the routers at the edges of the subnet, since TCP
> >has no way to operate on their semantics and IP has no way to  be able to pass
> >their semantics to TCP;  however, see below for one exception.
> 
> I am editing your text into the Buffering, Flow Control and Congestion Control
> section, but I am having trouble parsing the above paragraph. Can you explain
> and/or clarify it?
> 
> Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 14:05:03 2000
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Dan Grossman wrote:
> 
> Phil,
> Looks good.  Thanks.
> Dan
> > >1) Shortly before the Washington meeting, I sent a suggestion to change the
> > >Introduction/Overview section as follows.
> >
> > >> ...seminal paper "End-to-End Arguments in System Design" [SRC81]. This paper
> > >> argued that many -- if not most -- network functions are best implemented on
> > >> an end-to-end basis, i.e., at the higher protocol layers.  Duplicating these
> > >> functions at the lower levels can be needlessly redundant or harmful.
> > >> However,  there are also cases where duplication can be justified by
> > >> considerations such  as performance, availability or security.  Examples
> > >> might include link layer  retransmission on an unusually lossy channel, e.g.,
> > >> mobile radio, encryption  of the entire link layer protocol over a shared
> > >> medium, or  redundant paths at the link layer for fault recovery.
> > >> Duplication of functions should be done only based on an understanding of
> > >> system level implications, including any side-effects on the higher layers.

The paper does not exactly argue that these fns are best implemented
E2E;
it argues that:

	(I call this the E2E Principle:)
	End-to-end services cannot be provided solely by the composition 
	of hop-by-hop services.

	(I call this the E2E Corrolary:)
	Hop-by-hop services may be required for performance, but they enhance, 
	rather than replace, the corresponding end-to-end service.

The issue of whether they are needlessly redundent or harmful stems only
from these.
A HBH implementation is harmful only where it is used to incorrectly
replace an E2E one 
(violating E2E Princ.) or where it fails to enhance the concurrent E2E
version (violating
E2E Corrol.)

"best implemented as E2E" isn't exactly the point; the point is that HBH
is insufficient,
not 'relatively worse'. Duplication at lower layers is strictly a
performance
issue, by my read.

Is there a way of morphing that into the above??

Joe


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Phil Karn wrote:
> 
> >PAGE 4: MTU Size
> 
> >It may be useful to note that using IPSEC in tunnel mode can increase
> >the size of a packet as it travels across the network.  This can result
> >in a smaller MTU.

Many protocols other than IPSEC can reduce the E2E MTU.
IPSEC is one, IPIP tunneling is another (it need not 
include or refer to IPSEC), as are any protocols that augment
the IP header with options.
 
> I'm not sure what to do about this one. I'm inclined to agree with the
> comments that say this is not unique to IPSEC but is a consideration with any
> IP-in-IP encapsulation scheme -- all of which are at higher layers than the subnet.

Any protocol that augments the IP header can cause this, whether by 
encapsulation, option augmentation, etc.
 
> Perhaps some text would be appropriate that says that while 1500 bytes
> is an important MTU in the Internet because of the prevalence of
> Ethernet, it would be useful to support a somewhat larger size to avoid IP fragmentation
> when tunneling (which would include IPSEC) is used. Problem is, this still doesn't
> avoid the fragmentation that would occur when the path between the tunnel endpoints
> includes one or more Ethernets. So the benefits of a larger MTU are apt to be limited anyway.

A useful point may be "don't assume the MTU at all anymore," i.e., it
might be 
worthwhile to have designers of links _and_ upper protocols both drop
such preconceptions
they have about the link.

Joe


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To: karn@qualcomm.com
Cc: kscott@mitre.org, pilc@grc.nasa.gov
Subject: Re: Kicking TCP 
From: Zachary Amsden <zamsden@cthulhu.engr.sgi.com>
In-Reply-To: Your message of "Tue, 07 Mar 2000 23:36:13 PST."
             <200003080736.XAA27464@homer.ka9q.ampr.org> 
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> >how about using ICMP unreachable with a new code for Network Temporarily 
> >unavailable - if this is received, TCP should save it's current window, 
> >temporarily cut the window to zero, and begin sending ACK probes at a regular 
> >rate.  When it receives an ACK in return (see note below), it restores it's 
> >old window and resumes transmission.
> 
> I agree with Dan Grossman; I'm not sure you want to save and restore
> the old TCP state, as opposed to letting it restart slow-start.

That was the intention of my original idea - to reduce cwnd and send periodic 
(rather than backed-off) probes, and do a slow-restart upon receipt of an ACK.

> During the outage state, there is probably little harm in letting the
> sending TCP probe the link with normal retransmissions. Because each
> retransmission will be backed off exponentially, the load will be
> limited even if full packets are used in the probes.

During the outage state, if the router is queuing the data as suggested, upon 
restoration, any queued data will be sent as a burst to the other side, and 
the resulting ACKs will cause the window to re-inflate.  So if you have a 
buffering router, no protocol modifications are really necessary to get you 
going again.  So yes, restoring cwnd to the pre-outage value with a buffering 
router would cause the window to possibly double what the window was before 
congestion, which would be a bad thing.

With no buffering in the router, saving/restoring cwnd via link down/up 
notifications seems like a good idea.  At most only hosts that recognize the 
notification scheme will start retransmitting with prior windows, so hosts 
that don't will just backoff and slow-start.  So there shouldn't be any 
problem with congestion after link up notification if there wasn't any 
congestion when the link went down.

The notification scheme is really just an implementation detail, and I'm just 
wondering whether it can be done within the current framework by being 
creative without breaking too many semantics.

-- 
Zachary Amsden  zamsden@engr.sgi.com  3-6919  31-2-510  Core Protocols




From owner-pilc@lerc.nasa.gov  Wed Mar  8 14:53:58 2000
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To: Joe Touch <touch@ISI.EDU>
cc: karn@qualcomm.com, afalk@PanAmSat.com, pilc@grc.nasa.gov
Subject: Re: Draft on Advice for Internet Subnetwork Designers 
In-reply-to: Your message of "Wed, 08 Mar 2000 10:10:07 EST."
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> 
> 
> Dan Grossman wrote:
> > 
> > Phil,
> > Looks good.  Thanks.
> > Dan
> > > >1) Shortly before the Washington meeting, I sent a suggestion to change the
> > > >Introduction/Overview section as follows.
> > >
> > > >> ...seminal paper "End-to-End Arguments in System Design" [SRC81]. This paper
> > > >> argued that many -- if not most -- network functions are best implemented on
> > > >> an end-to-end basis, i.e., at the higher protocol layers.  Duplicating these
> > > >> functions at the lower levels can be needlessly redundant or harmful.
> > > >> However,  there are also cases where duplication can be justified by
> > > >> considerations such  as performance, availability or security.  Examples
> > > >> might include link layer  retransmission on an unusually lossy channel, e.g.,
> > > >> mobile radio, encryption  of the entire link layer protocol over a shared
> > > >> medium, or  redundant paths at the link layer for fault recovery.
> > > >> Duplication of functions should be done only based on an understanding of
> > > >> system level implications, including any side-effects on the higher layers.
> 
> The paper does not exactly argue that these fns are best implemented
> E2E;
> it argues that:
> 
> 	(I call this the E2E Principle:)
> 	End-to-end services cannot be provided solely by the composition 
> 	of hop-by-hop services.
> 
> 	(I call this the E2E Corrolary:)
> 	Hop-by-hop services may be required for performance, but they enhance, 
> 	rather than replace, the corresponding end-to-end service.
> 
Not just for performance, but otherwise agree thus far.
> The issue of whether they are needlessly redundent or harmful stems only
> from these.
> A HBH implementation is harmful only where it is used to incorrectly
> replace an E2E one 
> (violating E2E Princ.) or where it fails to enhance the concurrent E2E
> version (violating
> E2E Corrol.)
> 
> "best implemented as E2E" isn't exactly the point; the point is that HBH
> is insufficient,
> not 'relatively worse'. 
I agree up to this point.  The old text was a bit of an oversimplification of 
the end-to-end paper.  The underlying issue in end-to-end was the historical 
battle between X.25 with Class 0 transport protocol and IP with TCP (or CLNP 
and Class 4 transport protocol).  X.25 attempted to provide an end-to-end 
reliable service using network layer (hop-by-hop) mechanisms, so that no 
retransmission mechanisms were required in the end systems.  The point was 
that although those kinds of mechanisms could produce an _almost_ reliable 
service at the cost of expensive mechanisms in the network, a _truly_ reliable 
end-to-end service could never be supported using only hop-by-hop mechanisms.  
The end-to-end paper then extends its exposition of this flaw to further argue 
for statelessness as a way of providing end-to-end availability despite 
network failures ("fate sharing"), and in general to call for extremely 
lightweight networks. 

>Duplication at lower layers is strictly a
> performance
> issue, by my read.
Here's where I disagree.  There are hop-by-hop (or, rather) subnet functions 
which are in some sense redundant with end-to-end functions, but which are 
still needed for security and availability, as well as performance.  The 
examples in the new text are fairly generic examples.  Consider the case of 
availability:  various kinds of subnets -- SONET/SDH, ATM, MPLS, optical -- 
provide some kind of failover protection switching.  These are  stateful and 
redundant with IP layer re-routing, and could if improperly used result in 
harmful interactions with the IP layer.  On the other hand, they can 
accomplish a failover in 50 ms, rather than the 30 seconds or more that it 
takes for OSPF to reconverge;  if Five-9s availability is a network objective, 
then these additional mechanisms are needed.

> Is there a way of morphing that into the above??
> 
...[SRC81].   This paper argued that certain kinds of end-to-end services 
cannot be provided solely by the composition of hop-by-hop mechanisms, and 
instead require mechanisms that operate end-to-end. Duplicative presence of 
hop-by-hop mechanisms in the network can be needlessly redundant or even cause 
harmful interactions with end-to-end mechanisms. However...




From owner-pilc@lerc.nasa.gov  Wed Mar  8 15:19:56 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: zamsden@cthulhu.engr.sgi.com
CC: kscott@mitre.org, pilc@grc.nasa.gov
In-reply-to: <200003081841.KAA20340@clock.engr.sgi.com> (message from Zachary
	Amsden on Wed, 08 Mar 2000 10:41:34 -0800)
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>During the outage state, if the router is queuing the data as suggested, upon 
>restoration, any queued data will be sent as a burst to the other side, and 
>the resulting ACKs will cause the window to re-inflate.  So if you have a 

Only if all the data is queued during the outages. I propose having the subnet
queue only the last packet on each flow, or alternatively some small number
of packets N, with the older packets discarded as new ones arrive.

Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 15:20:12 2000
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From: Phil Karn <karn@ka9q.ampr.org>
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CC: dan@dma.isg.mot.com, afalk@PanAmSat.com, pilc@grc.nasa.gov, touch@ISI.EDU
In-reply-to: <38C6977F.35048669@isi.edu> (message from Joe Touch on Wed, 08
	Mar 2000 10:10:07 -0800)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
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>Is there a way of morphing that into the above??

I agree with your presentation. I'll try to come up with the appropriate language.

Phil


From owner-pilc@lerc.nasa.gov  Wed Mar  8 16:45:59 2000
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From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Phil Karn <karn@ka9q.ampr.org>

>>During the outage state, if the router is queuing the data as suggested, upon 
> >restoration, any queued data will be sent as a burst to the other side, and 
> >the resulting ACKs will cause the window to re-inflate.  So if you have a 
>
> Only if all the data is queued during the outages. I propose having the subnet
> queue only the last packet on each flow, or alternatively some small number
> of packets N, with the older packets discarded as new ones arrive.


Again, doesn't any reasonable router queue at least the last N packets
that arrive while its flakey link is down but expected to be restored?


Vernon Schryver    vjs@rhyolite.com


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From owner-pilc@lerc.nasa.gov  Thu Mar  9 04:20:35 2000
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	(message from Marie-Jose Montpetit on Tue, 7 Mar 2000 03:54:12 -0800)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
Reply-to: karn@qualcomm.com
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>On the topic of packet discard especially on those subnet where capacity is
>expensive or limited I would add that when possible packets marked form
>another subnet should be discarded before transmission on the link.

Is this an issue for the subnetwork designer, or for the router that
drives the subnetwork?

Phil


From owner-pilc@lerc.nasa.gov  Thu Mar  9 09:35:06 2000
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From: Michael Orr <orr@flash-networks.com>
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	> Before the working group could take on developing a new protocol,
a
	> compelling argument needs to be made for the need of such an
activity. Hari
	> and Venkat have made their case that this is a real problem and
I'm looking
	> to the mail list for validation that the problem merits further
definition
	> and the proposals merit further development by this working group.

	> --aaron

	Put me down for "yes, it is a real problem, and worth
discussing/solving". 
	I am not quite sure if a new protocol is required, but if we don't
thoroughly 
	discuss, how will we decide? we need to see the trade-offs. 

--
Michael Orr
orr@iname.com





From owner-pilc@lerc.nasa.gov  Thu Mar  9 10:01:02 2000
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From: Marie-Jose Montpetit <marie@teledesic.com>
To: "'karn@qualcomm.com'" <karn@qualcomm.com>
Cc: dan@dma.isg.mot.com, afalk@PanAmSat.com, pilc@grc.nasa.gov
Subject: RE: Draft on Advice for Internet Subnetwork Designers
Date: Thu, 9 Mar 2000 05:48:40 -0800 
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If we start from the premise that the equipment that is connected to the
subnetwork is unaware of the particularities of the subnetwork, which I
think is usually the case, then it is an issue for the designer.

Marie-Jose

> -----Original Message-----
> From:	Phil Karn [SMTP:karn@ka9q.ampr.org]
> Sent:	Thursday, March 09, 2000 12:18 AM
> To:	marie@teledesic.com
> Cc:	dan@dma.isg.mot.com; afalk@PanAmSat.com; pilc@grc.nasa.gov
> Subject:	Re: Draft on Advice for Internet Subnetwork Designers
> 
> >On the topic of packet discard especially on those subnet where capacity
> is
> >expensive or limited I would add that when possible packets marked form
> >another subnet should be discarded before transmission on the link.
> 
> Is this an issue for the subnetwork designer, or for the router that
> drives the subnetwork?
> 
> Phil


From owner-pilc@lerc.nasa.gov  Thu Mar  9 13:58:45 2000
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To: Marie-Jose Montpetit <marie@teledesic.com>
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marie-jose,

not in this case. it is a design decision for the router, not
the link. think ISSLL, IP-ATM, etc, etc.... the IP to link layer
driver is where we implement particular rules for mapping certain
types of (marked) IP packets onto links, subsets of links, or
/dev/null. 

cheers,
gja

Marie-Jose Montpetit wrote:
> 
> If we start from the premise that the equipment that is connected to the
> subnetwork is unaware of the particularities of the subnetwork, which I
> think is usually the case, then it is an issue for the designer.
> 
> Marie-Jose
> 
> > -----Original Message-----
> > From: Phil Karn [SMTP:karn@ka9q.ampr.org]
> > Sent: Thursday, March 09, 2000 12:18 AM
> > To:   marie@teledesic.com
> > Cc:   dan@dma.isg.mot.com; afalk@PanAmSat.com; pilc@grc.nasa.gov
> > Subject:      Re: Draft on Advice for Internet Subnetwork Designers
> >
> > >On the topic of packet discard especially on those subnet where capacity
> > is
> > >expensive or limited I would add that when possible packets marked form
> > >another subnet should be discarded before transmission on the link.
> >
> > Is this an issue for the subnetwork designer, or for the router that
> > drives the subnetwork?
> >
> > Phil

-- 
________________________________________________________________________
Grenville Armitage                            http://mh005.infi.net/~gja
Bell Labs Research Silicon Valley


From owner-pilc@lerc.nasa.gov  Thu Mar  9 13:58:49 2000
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okay. i see the adaptation as part of the router, so perhaps
we're in agreement.

gja

Marie-Jose Montpetit wrote:
> 
> Agreed it is not the "link" as we could understand it in the case of
> wireless for example but surely the "transport" in the subnet. What I meant
> is the router should not (minimally?) be impacted. The adaptation should.
> 
> Marie-Jose
> 
> -----Original Message-----
> From: Grenville Armitage [mailto:gja@lucent.com]
> Sent: Thursday, March 09, 2000 9:41 AM
> To: Marie-Jose Montpetit
> Cc: 'karn@qualcomm.com'; pilc@grc.nasa.gov
> Subject: Re: Draft on Advice for Internet Subnetwork Designers
> 
> marie-jose,
> 
> not in this case. it is a design decision for the router, not
> the link. think ISSLL, IP-ATM, etc, etc.... the IP to link layer
> driver is where we implement particular rules for mapping certain
> types of (marked) IP packets onto links, subsets of links, or
> /dev/null.
> 
> cheers,
> gja
> 
> Marie-Jose Montpetit wrote:
> >
> > If we start from the premise that the equipment that is connected to the
> > subnetwork is unaware of the particularities of the subnetwork, which I
> > think is usually the case, then it is an issue for the designer.
> >
> > Marie-Jose
> >
> > > -----Original Message-----
> > > From: Phil Karn [SMTP:karn@ka9q.ampr.org]
> > > Sent: Thursday, March 09, 2000 12:18 AM
> > > To:   marie@teledesic.com
> > > Cc:   dan@dma.isg.mot.com; afalk@PanAmSat.com; pilc@grc.nasa.gov
> > > Subject:      Re: Draft on Advice for Internet Subnetwork Designers
> > >
> > > >On the topic of packet discard especially on those subnet where
> capacity
> > > is
> > > >expensive or limited I would add that when possible packets marked form
> > > >another subnet should be discarded before transmission on the link.
> > >
> > > Is this an issue for the subnetwork designer, or for the router that
> > > drives the subnetwork?
> > >
> > > Phil
> 
> --
> ________________________________________________________________________
> Grenville Armitage                            http://mh005.infi.net/~gja
> Bell Labs Research Silicon Valley

-- 
________________________________________________________________________
Grenville Armitage                            http://mh005.infi.net/~gja
Bell Labs Research Silicon Valley


From owner-pilc@lerc.nasa.gov  Thu Mar  9 14:33:09 2000
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From: Marie-Jose Montpetit <marie@teledesic.com>
To: "'Grenville Armitage'" <gja@lucent.com>
Cc: "'karn@qualcomm.com'" <karn@qualcomm.com>, pilc@grc.nasa.gov
Subject: RE: Draft on Advice for Internet Subnetwork Designers
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Agreed it is not the "link" as we could understand it in the case of
wireless for example but surely the "transport" in the subnet. What I meant
is the router should not (minimally?) be impacted. The adaptation should.

Marie-Jose

-----Original Message-----
From: Grenville Armitage [mailto:gja@lucent.com]
Sent: Thursday, March 09, 2000 9:41 AM
To: Marie-Jose Montpetit
Cc: 'karn@qualcomm.com'; pilc@grc.nasa.gov
Subject: Re: Draft on Advice for Internet Subnetwork Designers


marie-jose,

not in this case. it is a design decision for the router, not
the link. think ISSLL, IP-ATM, etc, etc.... the IP to link layer
driver is where we implement particular rules for mapping certain
types of (marked) IP packets onto links, subsets of links, or
/dev/null. 

cheers,
gja

Marie-Jose Montpetit wrote:
> 
> If we start from the premise that the equipment that is connected to the
> subnetwork is unaware of the particularities of the subnetwork, which I
> think is usually the case, then it is an issue for the designer.
> 
> Marie-Jose
> 
> > -----Original Message-----
> > From: Phil Karn [SMTP:karn@ka9q.ampr.org]
> > Sent: Thursday, March 09, 2000 12:18 AM
> > To:   marie@teledesic.com
> > Cc:   dan@dma.isg.mot.com; afalk@PanAmSat.com; pilc@grc.nasa.gov
> > Subject:      Re: Draft on Advice for Internet Subnetwork Designers
> >
> > >On the topic of packet discard especially on those subnet where
capacity
> > is
> > >expensive or limited I would add that when possible packets marked form
> > >another subnet should be discarded before transmission on the link.
> >
> > Is this an issue for the subnetwork designer, or for the router that
> > drives the subnetwork?
> >
> > Phil

-- 
________________________________________________________________________
Grenville Armitage                            http://mh005.infi.net/~gja
Bell Labs Research Silicon Valley


From owner-pilc@lerc.nasa.gov  Thu Mar  9 19:18:11 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: vjs@calcite.rhyolite.com
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In-reply-to: <200003081240.FAA14424@calcite.rhyolite.com> (message from Vernon
	Schryver on Wed, 8 Mar 2000 05:40:10 -0700 (MST))
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>How does that differ from what a reasonable demand dialing PPP
>implementation does the link goes down unexpected, whether because the
>peer took it down or the modems (or ISDN line etc.) hiccuped?  A reasonable
>PPP implementation queues the most recent N (for some reasonable N>0, but
>N>typical window/negotiated MTU is good) and sends them when it can (the
>link comes up or back up).  A PPP implementation that does not do this
>queuing is a pain to use (or so I think), because it loses the traffic
>that caused it to demand dial.  Unfortunately, many demand dialing PPP
>implementations are unreasonable by this criterion.

My suggestion really doesn't differ, so perhaps all I need to do is
to insert language into the LINK draft that says essentially this.

Phil


From owner-pilc@lerc.nasa.gov  Thu Mar  9 19:20:16 2000
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>If you do this at the link layer instead of the network layer you are free 
>to queue more than just a single packet, e.g., you can queue the entire 
>flight of packets belonging to a TCP flow if resources allow for that. In 

Well, I guess you could but I wouldn't want a short outage to result
in a lot of duplicate packets spewing out over the link after the link
comes back up. Seems to me you're better off holding onto just the
last packet or two per flow.

>Agree. However, I would leave it open whether this approach is implemented 
>at the router level (only buffering a single packet) or the link layer. It 

I agree with this too.

>should also be mentioned that no packet should be buffered longer than the 
>Maximum Segment Lifetime (MSL), i.e., 240 seconds, at most.

Is the MSL a real thing, or is it just a fiction? In either case, I see
no need to hold onto any packet for longer than the sending TCP's likely
retransmission interval -- except for the very last packet
whose purpose is more to notify TCP that the channel is back up
than to avoid a retransmission. But since the subnet doesn't have any
way to know what TCP's retransmission timeout is, you
have to make a guess.

>>Note that this gives nearly all of the benefits claimed for totally
>>reliable link layer protocols with much less effort.

>Don't agree. With the router level approach you need to change router 
>implementation (deployment concern!). The link layer approach can be 
>designed into upcoming wireless systems.

Okay, I didn't mean to distinguish between a router implementation
and a link implementation of the "hold the last packet" technique,
and I'll try to explain in the draft that either is OK.

>layer. I.e., highly persistent link layer error recovery (limited to the 
>class of reliable flows to avoid head-of-line blocking for delay-sensitive 
>flows) does not provide a reliability guarantee but is also merely a 
>performance enhancement.

Exactly. In this case, the performance enhancement is to get TCP
going again more quickly after the link recovers.

Phil


From owner-pilc@lerc.nasa.gov  Thu Mar  9 19:24:49 2000
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Date: Thu, 9 Mar 2000 16:29:03 -0700 (MST)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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To: pilc@grc.nasa.gov
Subject: Re: Kicking TCP
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> From: Phil Karn <karn@ka9q.ampr.org>

> >How does that differ from what a reasonable demand dialing PPP
> >implementation does the link goes down unexpected, whether because the
> >peer took it down or the modems (or ISDN line etc.) hiccuped?  A reasonable
> >PPP implementation queues the most recent N (for some reasonable N>0, but
> >N>typical window/negotiated MTU is good) ...

> My suggestion really doesn't differ, so perhaps all I need to do is
> to insert language into the LINK draft that says essentially this.

Regardless of whether there might be a new ICMP kick TCP type, I think
the LINK draft should say something like that.  PPP implementations that
don't queue traffic when they know the link is sick are even more common
than they are lame, and they're awfully lame.

Another way they're lame that I haven't mentioned lately is what happens
when you try to start a TCP connection that requires DNS answers.  The
initial UDP packet will wake up your demand dialed link, but if your PPP
code is stupid and discards instead of queuing it, you'll be sitting
waiting a long time after a PPP/ISDN link is up for your DNS resolver to
time out and retry.  Isn't the DNS timeout 30 seconds?  An ISDN link comes
up in 2-5 seconds and a slow v.32bis modem needs only ~28 seconds.

However, someone privately pointed out that there is a difference between
reasonable queuing and something like an ICMP kick TCP message when the
queuing drops the newest instead of oldest traffic and when the queue
is smaller than the traffic while the link is ill.  If the queue overflows
and new traffic is discarded, TCP will suffer a time out.  The old queued
traffic should kick TCP enough to shorten the timeout, but the result
won't be as quick the result of dropping the newest packets.  Worse, if
there are lots of streams and a tiny queue, some TCP connections might
not be kicked at all.

Still, I think reasonable queuing is close enough and needed regardless.
There is no need for a new ICMP kick TCP type, not to mention novel
interpretations of Redirects, new UDP messages, or other things.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Thu Mar  9 19:28:13 2000
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Date: Thu, 9 Mar 2000 15:25:00 -0800
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From: Phil Karn <karn@ka9q.ampr.org>
To: vjs@calcite.rhyolite.com
CC: pilc@grc.nasa.gov
In-reply-to: <200003081537.IAA18001@calcite.rhyolite.com> (message from Vernon
	Schryver on Wed, 8 Mar 2000 08:37:17 -0700 (MST))
Subject: Re: Kicking TCP
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> 1. have the link layer tell the TCP state machines to retransmit any
>  unacknowledged TCP bytes by forcing their retransmission timers
>  to zero?  This is a layering violation, but I trust most of us
>  are Sunday-only worshipers in that religion.

It's more than a technical layering violation, because how would
you do it if the TCP and the link layer are on separate machines?
You'd need to define some communication mechanism, which was the
intent of my ICMP SE message.

> 2. do what is common (albeit not common enough) and reasonable in in PPP
>  implementations, which is to have a reasonable sized transmit queue 
>  ( > typical window/MTU), and only send packets when the link seems
>  healthy?  Don't you need such a queue to deal with bursts, so that the
>  only modification would be to not drop IP packets when the link is
>  broken, as is typical in link layer drivers, including Ethernet, FDDI,
>  and 802.5?

Again, I agree. I will add text that encourages the subnetwork designer
to use any internal information it may have about the state of the
link or subnetwork to block the acceptance of IP packets if it is clear
that those packets would be dropped, and to hold onto the most recent
packets that it may have accepted before the link/subnet went down
and to try to deliver them after the link/subnet comes back up.

Phil




From owner-pilc@lerc.nasa.gov  Thu Mar  9 19:45:18 2000
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Date: Thu, 9 Mar 2000 15:47:12 -0800
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From: Phil Karn <karn@ka9q.ampr.org>
To: cannara@attglobal.net
CC: pilc@grc.nasa.gov
In-reply-to: <38C69427.927B0842@attglobal.net> (message from Cannara on Wed,
	08 Mar 2000 09:55:51 -0800)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
Reply-to: karn@qualcomm.com
References: <200003062158.QAA18917@noah.dma.isg.mot.com> <200003080828.AAA27974@homer.ka9q.ampr.org> <38C69427.927B0842@attglobal.net>
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>FYI, a related suggestion was for IP to communicate info TCP could use by
>passing up the IP ID (v4) via TCP's Urgent Pointer (with URG bit off) and the
>receiving TCP would so get the ID of the packet sent, then return that ID when
>it Acked that pkt.  The sender keeps a list of IDs vs TCP pkts and then knows

Ugh, what a layer-violating hack! I think a better way already exists
to deal with the false retransmission problem: TCP header timestamps.

Phil


From owner-pilc@lerc.nasa.gov  Thu Mar  9 19:45:48 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: touch@ISI.EDU
CC: pilc@grc.nasa.gov
In-reply-to: <38C6994A.2216D26D@isi.edu> (message from Joe Touch on Wed, 08
	Mar 2000 10:17:46 -0800)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
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References: <8046A317B5F4D21186610008C7EBB696F055EE@grnexch1.panamsat.com> <38BA7E19.CB9D89B1@erg.abdn.ac.uk> <200003080900.BAA28435@homer.ka9q.ampr.org> <38C6994A.2216D26D@isi.edu>
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>A useful point may be "don't assume the MTU at all anymore," i.e., it
>might be worthwhile to have designers of links _and_ upper protocols
>both drop such preconceptions they have about the link.

That's a good idea. Not only does tunneling mess up assumptions about
MTUs and the distribution of packet sizes, but so do the newer TCP
header options (timestamps, etc).

phil


From owner-pilc@lerc.nasa.gov  Thu Mar  9 19:47:40 2000
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Date: Thu, 9 Mar 2000 16:55:44 -0700 (MST)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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To: pilc@grc.nasa.gov
Subject: Re: Kicking TCP
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> From: Phil Karn <karn@ka9q.ampr.org>

> ...
> > 1. have the link layer tell the TCP state machines to retransmit any
> >  unacknowledged TCP bytes by forcing their retransmission timers
> >  to zero? ...

> It's more than a technical layering violation, because how would
> you do it if the TCP and the link layer are on separate machines?
> You'd need to define some communication mechanism, which was the
> intent of my ICMP SE message.

Yes, but in many cases (e.g. mobile phones) the host and the router
are pretty close to each other.  But I agree that reasonable queuing
next to the link is far better.


> ...
> Again, I agree. I will add text that encourages the subnetwork designer
> to use any internal information it may have about the state of the
> link or subnetwork to block the acceptance of IP packets if it is clear
> that those packets would be dropped, and to hold onto the most recent
> packets that it may have accepted before the link/subnet went down
> and to try to deliver them after the link/subnet comes back up.

That sound great.

Something I forget to mention in my previous note is that private
cooresondent pointed out that RED and other fancy queuing disciplines
tend to encourage dropping the most recent instead oldest packets.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Thu Mar  9 19:49:05 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: touch@ISI.EDU
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In-reply-to: <38C6977F.35048669@isi.edu> (message from Joe Touch on Wed, 08
	Mar 2000 10:10:07 -0800)
Subject: Re: Draft on Advice for Internet Subnetwork Designers
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Okay, here's the current wording of the end-to-end principle section.
Any more comments? --Phil

A major consideration in the efficient design of any layered
communication network are the appropriate layer(s) in which to
implement a given feature. This issue was first addressed in the
seminal paper "End-to-End Arguments in System Design" [SRC81]. This
paper argued that many functions can be implemented properly *only* on
an end-to-end basis, i.e., at the higher protocol layers, outside the
subnetwork. These functions include ensuring the reliable delivery of
data and the protection of information through cryptography.
Duplicating these functions at the lower protocol layers (i.e., within
the subnetwork) can be needlessly redundant or even harmful to cost
and performance.

However, partial duplication of functionality in a lower layer
can *sometimes* be justified by performance, security or availability
considerations. Examples include link layer retransmission on an
unusually lossy channel, e.g., mobile radio; link level encryption
intended to thwart traffic analysis; and redundant transmission links
to improve availability. Duplication of protocol function should be done only
with an understanding of system level implications, including possible
interactions with higher layer mechanisms.


From owner-pilc@lerc.nasa.gov  Thu Mar  9 21:30:44 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: pilc@grc.nasa.gov, iab-wireless-workshop@ietf.org
Subject: Wording for "Recovery from Subnetwork Outages"
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Here's the text I just wrote for the PILC LINK document that discusses
subnetwork mechanisms that could speed up TCP's recovery from a
temporary subnetwork outage. I put it in a new section titled
"Recovery from Subnetwork Outages". This is the "implicit" alternative
to the ICMP Source Encourage message that I discussed earlier.

--Phil

Some types of subnetworks, particularly mobile radio, are subject to
frequent temporary outages. For example, an active cellular data user
may drive or walk into an area (such as a tunnel) that is out of range
of any base station. No packets will be successfully delivered until
the user returns to an area with coverage.

The Internet protocols currently provide no way for a subnetwork to
explicitly notify an upper layer protocol (e.g., TCP) that it is
experiencing an outage, as distinguished from severe congestion.
Under these circumstances TCP will, after each unsuccessful
retransmission, wait even longer before trying again; this is its
"exponential backoff" algorithm. And since there is also currently no
way for a subnetwork to explicitly notify TCP when it is again
operational, TCP will not discover this until its next retransmission
attempt. If TCP has backed off, this may take some time.  This can
lead to extremely poor TCP performance over such subnetworks.

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

Note that it is *not* necessary to avoid any and all packet drops
during an outage. The purpose of holding onto a packet during an
outage, either in the subnetwork or at the IP layer, is so that its
eventual delivery can implicitly notify TCP that the subnetwork is
again operational.  This is to enhance performance, not to ensure
reliability -- a task that as discussed earlier can only be done
properly on an end-to-end basis.

Only a single packet per TCP connection need be held in this way;
subsequent packets can either replace the existing packet, or they can
be discarded. Either way, TCP will quickly recover from these
additional losses once the flow resumes.

Because it would be a layering violation for IP or a subnetwork to
look at the TCP headers of the packets it carries (which would in any
event be impossible if IPSEC encryption is in use), it would be
reasonable for the IP or subnetwork layers to choose, as a design
parameter, some small number of packets that it will retain during an
outage.






From owner-pilc@lerc.nasa.gov  Thu Mar  9 23:43:09 2000
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Phil, this wasn't suggested as a "thing to do", but an illustration of what
could be done, even just experimentally or by simulation, to demonstrate the
utility of addressing TCP's blind side.  If timestamps can be trusted, then
maybe that's a way.  In any case, considerable performance in real situations
is sacrificed because we've gone years without taking significant action to
help TCP avoid making costly mistakes.  "Ugh" is indeed the right word for
those behaviors.

Alex


Phil Karn wrote:
> 
> >FYI, a related suggestion was for IP to communicate info TCP could use by
> >passing up the IP ID (v4) via TCP's Urgent Pointer (with URG bit off) and the
> >receiving TCP would so get the ID of the packet sent, then return that ID when
> >it Acked that pkt.  The sender keeps a list of IDs vs TCP pkts and then knows
> 
> Ugh, what a layer-violating hack! I think a better way already exists
> to deal with the false retransmission problem: TCP header timestamps.
> 
> Phil


From owner-pilc@lerc.nasa.gov  Thu Mar  9 23:43:14 2000
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References: <200003081340.IAA15226@noah.dma.isg.mot.com> <38C6977F.35048669@isi.edu> <200003092345.PAA09786@homer.ka9q.ampr.org>
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Sounds good Phil, and I'd add something a bit stronger in relation to
higher-layers that, in fact, are not fully capable of offering reliability and
performance under circumstances that the lower layers can do better on.  This
relates to the problems TCP has with noncongestive loss, etc.

Alex


Phil Karn wrote:
> 
> Okay, here's the current wording of the end-to-end principle section.
> Any more comments? --Phil
> 
> A major consideration in the efficient design of any layered
> communication network are the appropriate layer(s) in which to
> implement a given feature. This issue was first addressed in the
> seminal paper "End-to-End Arguments in System Design" [SRC81]. This
> paper argued that many functions can be implemented properly *only* on
> an end-to-end basis, i.e., at the higher protocol layers, outside the
> subnetwork. These functions include ensuring the reliable delivery of
> data and the protection of information through cryptography.
> Duplicating these functions at the lower protocol layers (i.e., within
> the subnetwork) can be needlessly redundant or even harmful to cost
> and performance.
> 
> However, partial duplication of functionality in a lower layer
> can *sometimes* be justified by performance, security or availability
> considerations. Examples include link layer retransmission on an
> unusually lossy channel, e.g., mobile radio; link level encryption
> intended to thwart traffic analysis; and redundant transmission links
> to improve availability. Duplication of protocol function should be done only
> with an understanding of system level implications, including possible
> interactions with higher layer mechanisms.


From owner-pilc@lerc.nasa.gov  Fri Mar 10 02:36:27 2000
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>Something I forget to mention in my previous note is that private
>cooresondent pointed out that RED and other fancy queuing disciplines
>tend to encourage dropping the most recent instead oldest packets.

Did you mean to say that? Before RED, the usual scheme was to drop all
new packets when the queue filled up. With RED, you pick a packet at
random from the queue, so the older packets can get dropped too.

It seems to me that simply implementing RED in the IP router that
drives a link subject to outages will pretty much implement my
suggestion as long as the subnet doesn't take anything out of the IP
send queue as long as its down. During the outage, RED will continue
to manage the IP queue as new packets arrive. When the outage is over,
the survivors on the queue will be sent. Hopefully that will include
at least one packet per pending TCP connection.

Fortunately, most "flaky links" rarely carry more than one active TCP
connection at a time, so this should usually be the case.

Phil



From owner-pilc@lerc.nasa.gov  Fri Mar 10 02:53:42 2000
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Phil Karn wrote:
> 
> >Something I forget to mention in my previous note is that private
> >cooresondent pointed out that RED and other fancy queuing disciplines
> >tend to encourage dropping the most recent instead oldest packets.
> 
> Did you mean to say that? Before RED, the usual scheme was to drop all
> new packets when the queue filled up. With RED, you pick a packet at
> random from the queue, so the older packets can get dropped too.

Not exactly - RED just randomizes the drop/insert decision made
by the queue manager for a newly arrived packet. To the best
of my knowledge, there's nothing inherent in the RED scheme that
requires randomly choosing a packet currently *in* the queue for
dropping. (A minor optimization is to use the arrival of a new
packet to trigger a random decision to drop from front...)

But that's an aside - the observation that queues have enough space
to hold a windows worth of packets from the active flows (either
because the queue is big, or the number of flows is small) makes
the issue moot wrt kicking TCP.

cheers,
the-private-correspondent :)


From owner-pilc@lerc.nasa.gov  Fri Mar 10 03:24:55 2000
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From: Phil Karn <karn@ka9q.ampr.org>
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I just added this section in the "Reliability and Error Control" section.
I will keep editing the LINK document, adding new text and comments as
I get them, until tomorrow when I will send a snapshot to the ID editor
with enough time to make the deadline... --Phil

The TCP, UDP and IP protocols all use a simple 16-bit 1's complement
checksum algorithm to detect corrupted packets. The IP checksum
protects only the IP header, while the TCP and UDP checksums protect
both the TCP/UDP header and any user data. In TCP the checksum is
mandatory while it is optional in UDP.

These checksums are not very strong from a coding theory standpoint.
But they are easy to compute in software, and various proposals to
replace them with stronger checksums have failed. Experience has shown
that nearly all subnetworks that carry the Internet protocols have a
very low undetected error rate, so the existing Internet checksums are
rarely called upon to catch an error. Most of the errors caught by the
Internet checksum are generally the result of hardware or software
failure than of a subnetwork transmission error. This lowers the
motivation to overcome the performance and especially the backward
compatibility problems of a new, stronger Internet checksum.

The typical IP subnetwork implements an internal error detection code that
discards packets failing this check instead of delivering them to
the IP layer. A 16-bit cyclic redundancy check (CRC) is usually the
minimum, and this is known to be considerably stronger than the 16-bit
standard Internet checksum. The Point-to-Point Protocol (PPP) uses a
16-bit CRC, often in conjunction with the V.42bis error control
provided in a typical dialup modem. 802.3/Ethernet uses a 32-bit CRC
that is considerably stronger, and many subnetworks (notably dialup
modem, mobile radio and satellite) also incorporate forward error
correction, often in hardware.

Any new subnetwork designed to carry IP should therefore provide error
detection at least as strong as a 16-bit CRC and preferably stronger.
Note that while this will achieve a very low undetected packet error
rate, this will not (and need not) achieve a very low packet loss rate
as the Internet protocols are better suited to dealing with lost
packets than with corrupted packets.


From owner-pilc@lerc.nasa.gov  Fri Mar 10 07:54:57 2000
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Phil:

 >> Only a single packet per TCP connection need be held in this way; 
 >> subsequent packets can either replace the existing packet, or they can 
 >> be discarded. Either way, TCP will quickly recover from these 
 >> additional losses once the flow resumes. 

One implication of the above statement -- in the case when
a single packet is held -- would be that it can
be *any* single packet, not necessarily the earliest one.

Why is that enough to kickstart TCP?
One out-of-order (data) packet will trigger one dupack.
Not enough to result in a fast retransmit.

Unless something else is modified, if only packet
is going to be held, it should be the earliest data packet.
On reverse path, it is OK to hold any one TCP ACK due
to the cummulative nature.
On the forward path, if the earliest data packet is
dropped, then need to hold at least three other data
packets, not just one, to result in fast retransmit, instead
of a (long) wait for timeout.

Am I missing something? ... it is rather early Friday morning
for me, and hopefully, I can be forgiven in that case -:)

Assuming I am sufficiently awake ...
Preferential drop-from-front is better for
* TCP ACKs in all cases (errors or congstion) 
* TCP Data when packets must be dropped due to congestion,
  (but not necessarily in case of errors).

Of course, if TCP sender was earlier informed that the
channel is dead, then even one dupack would be enough
to kickstart the sender.

Cheerio.

- nitin



From owner-pilc@lerc.nasa.gov  Fri Mar 10 09:41:22 2000
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From: Markku Kojo <kojo@cs.Helsinki.FI>
To: karn@qualcomm.com
cc: vjs@calcite.rhyolite.com, pilc@grc.nasa.gov
Subject: Re: Kicking TCP
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On Thu, 9 Mar 2000, Phil Karn wrote:

> >How does that differ from what a reasonable demand dialing PPP
> >implementation does the link goes down unexpected, whether because the
> >peer took it down or the modems (or ISDN line etc.) hiccuped?  A reasonable
> >PPP implementation queues the most recent N (for some reasonable N>0, but
> >N>typical window/negotiated MTU is good) and sends them when it can (the
> >link comes up or back up).  A PPP implementation that does not do this
> >queuing is a pain to use (or so I think), because it loses the traffic
> >that caused it to demand dial.  Unfortunately, many demand dialing PPP
> >implementations are unreasonable by this criterion.
> 
> My suggestion really doesn't differ, so perhaps all I need to do is
> to insert language into the LINK draft that says essentially this.

Agreed. But just in case someone may interpret the statement
above differently: PPP is not used on all links/subnetworks which
have this problem; there are/will be number of (wireless) links
having the link outage problem but which are not employing PPP. 
PPP is link layer to me. And, if the actual link is of "serial type"
on which PPP is needed, then this is obviously best implemented in
PPP, otherwise this should be addressed in the link design.

Maybe this was already clear to everyone ...

/Markku




From owner-pilc@lerc.nasa.gov  Fri Mar 10 09:49:08 2000
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Subject: RE: CRCs, Checksums and Error Detection
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After the "stronger" in the last sentence maybe you want to add:

"...stronger to survive cluster of errors that can happen in many
subnetworks."

> -----Original Message-----
> From:	Phil Karn [SMTP:karn@ka9q.ampr.org]
> Sent:	Thursday, March 09, 2000 11:29 PM
> To:	pilc@grc.nasa.gov
> Subject:	CRCs, Checksums and Error Detection
> 
> 
> I just added this section in the "Reliability and Error Control" section.
> I will keep editing the LINK document, adding new text and comments as
> I get them, until tomorrow when I will send a snapshot to the ID editor
> with enough time to make the deadline... --Phil
> 
> The TCP, UDP and IP protocols all use a simple 16-bit 1's complement
> checksum algorithm to detect corrupted packets. The IP checksum
> protects only the IP header, while the TCP and UDP checksums protect
> both the TCP/UDP header and any user data. In TCP the checksum is
> mandatory while it is optional in UDP.
> 
> These checksums are not very strong from a coding theory standpoint.
> But they are easy to compute in software, and various proposals to
> replace them with stronger checksums have failed. Experience has shown
> that nearly all subnetworks that carry the Internet protocols have a
> very low undetected error rate, so the existing Internet checksums are
> rarely called upon to catch an error. Most of the errors caught by the
> Internet checksum are generally the result of hardware or software
> failure than of a subnetwork transmission error. This lowers the
> motivation to overcome the performance and especially the backward
> compatibility problems of a new, stronger Internet checksum.
> 
> The typical IP subnetwork implements an internal error detection code that
> discards packets failing this check instead of delivering them to
> the IP layer. A 16-bit cyclic redundancy check (CRC) is usually the
> minimum, and this is known to be considerably stronger than the 16-bit
> standard Internet checksum. The Point-to-Point Protocol (PPP) uses a
> 16-bit CRC, often in conjunction with the V.42bis error control
> provided in a typical dialup modem. 802.3/Ethernet uses a 32-bit CRC
> that is considerably stronger, and many subnetworks (notably dialup
> modem, mobile radio and satellite) also incorporate forward error
> correction, often in hardware.
> 
> Any new subnetwork designed to carry IP should therefore provide error
> detection at least as strong as a 16-bit CRC and preferably stronger.
> Note that while this will achieve a very low undetected packet error
> rate, this will not (and need not) achieve a very low packet loss rate
> as the Internet protocols are better suited to dealing with lost
> packets than with corrupted packets.


From owner-pilc@lerc.nasa.gov  Fri Mar 10 10:11:57 2000
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On Thu, 9 Mar 2000, Phil Karn wrote:

> The TCP, UDP and IP protocols all use a simple 16-bit 1's complement
> checksum algorithm to detect corrupted packets. The IP checksum
> protects only the IP header, while the TCP and UDP checksums protect
> both the TCP/UDP header and any user data. In TCP the checksum is
> mandatory while it is optional in UDP.

Fine for v4, but needs something about IPv6's different behaviour - no
IP checksum, and the UDP checksum is no longer optional (UDP Lite
proposals aside).

L.

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



From owner-pilc@lerc.nasa.gov  Fri Mar 10 12:27:52 2000
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Subject: Re: Draft on Advice for Internet Subnetwork Designers 
In-reply-to: Your message of "Thu, 09 Mar 2000 15:45:17 EST."
             <200003092345.PAA09786@homer.ka9q.ampr.org> 
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> Okay, here's the current wording of the end-to-end principle section.
> Any more comments? --Phil
> 
> A major consideration in the efficient design of any layered
> communication network are the appropriate layer(s) in which to
> implement a given feature. This issue was first addressed in the
Suggest changing 'feature' to 'function'.
I don't believe that it was first addressed in the Saltzer, Reed and Clark 
paper.  If I remember, this was an ongoing debate, and Saltzer, Reed and Clark 
captured one set of views.  
> seminal paper "End-to-End Arguments in System Design" [SRC81]. This
> paper argued that many functions can be implemented properly *only* on
> an end-to-end basis, i.e., at the higher protocol layers, outside the
> subnetwork. These functions include ensuring the reliable delivery of
I think Joe's point was that certain end-to-end services such as reliable 
delivery and
confidentiality cannot be provided solely by composition of hop-by hop 
services.
> data and the protection of information through cryptography.
try... confidentiality and message integrity
> Duplicating these functions at the lower protocol layers (i.e., within
> the subnetwork) can be needlessly redundant or even harmful to cost
> and performance.
> 
> However, partial duplication of functionality in a lower layer
> can *sometimes* be justified by performance, security or availability
why the extra emphasis?
> considerations. Examples include link layer retransmission on an
> unusually lossy channel, e.g., mobile radio; link level encryption
> intended to thwart traffic analysis; and redundant transmission links
> to improve availability. Duplication of protocol function should be done only
> with an understanding of system level implications, including possible
> interactions with higher layer mechanisms.




From owner-pilc@lerc.nasa.gov  Fri Mar 10 12:57:57 2000
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At 00:22 3/10/00 , Phil Karn wrote:
> >If you do this at the link layer instead of the network layer you are free
> >to queue more than just a single packet, e.g., you can queue the entire
> >flight of packets belonging to a TCP flow if resources allow for that. In
>
>Well, I guess you could but I wouldn't want a short outage to result
>in a lot of duplicate packets spewing out over the link after the link
>comes back up. Seems to me you're better off holding onto just the
>last packet or two per flow.

You are right in the case where "standard" TCP gets a spurious timeout 
because the sender basically does a go-back-N. However, short link outages 
will often _not_ trigger a (spurious) timeout. In those cases it would be 
better to buffer all packets that arrive at the link during the outage. So 
I guess it's hard to say what is best to do given that the link layer has 
little knowledge of the sender's RTO. You're right for "standard" TCP, we 
have to do some guessing.

> >should also be mentioned that no packet should be buffered longer than the
> >Maximum Segment Lifetime (MSL), i.e., 240 seconds, at most.
>
>Is the MSL a real thing, or is it just a fiction?

For example, RFC1122 uses the MSL as an upper limit for the reassembly 
timeout value in the case of IP fragmentation.

>In either case, I see
>no need to hold onto any packet for longer than the sending TCP's likely
>retransmission interval -- except for the very last packet
>whose purpose is more to notify TCP that the channel is back up
>than to avoid a retransmission. But since the subnet doesn't have any
>way to know what TCP's retransmission timeout is, you
>have to make a guess.

Basically agree. See above. Note though that with a TCP that is a bit more 
robust (http://iceberg.cs.berkeley.edu/papers/Ludwig-Eifel-Alg/index.html), 
i.e., that does not do a go-back-N after a spurious timeout, it is always 
better to buffer all packets.

///Reiner


From owner-pilc@lerc.nasa.gov  Fri Mar 10 13:01:50 2000
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Subject: Re: CRCs, Checksums and Error Detection 
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> 
> The TCP, UDP and IP protocols all use a simple 16-bit 1's complement
> checksum algorithm to detect corrupted packets. The IP checksum
> protects only the IP header, while the TCP and UDP checksums protect
> both the TCP/UDP header and any user data. In TCP the checksum is
> mandatory while it is optional in UDP.
> 
> These checksums are not very strong from a coding theory standpoint.
> But they are easy to compute in software, and various proposals to
> replace them with stronger checksums have failed. Experience has shown
> that nearly all subnetworks that carry the Internet protocols have a
> very low undetected error rate, so the existing Internet checksums are
> rarely called upon to catch an error. Most of the errors caught by the
> Internet checksum are generally the result of hardware or software
> failure than of a subnetwork transmission error. This lowers the
> motivation to overcome the performance and especially the backward
> compatibility problems of a new, stronger Internet checksum.
> 
Aside:  wouldn't strict application of the end-to-end principle indicate that
strong end-to-end mechanisms are needed to provide assurance of accurate data 
transfer? (Don't bother to answer this until after the deadline :-)

> The typical IP subnetwork implements an internal error detection code that
> discards packets failing this check instead of delivering them to
> the IP layer. A 16-bit cyclic redundancy check (CRC) is usually the
> minimum, and this is known to be considerably stronger than the 16-bit
> standard Internet checksum. The Point-to-Point Protocol (PPP) uses a
> 16-bit CRC, often in conjunction with the V.42bis error control
The correct reference is to PPP over HDLC Framing, RFC 1622.  The 16 bit CRC
is the default, and a 32 bit CRC is an LCP option.  

> provided in a typical dialup modem. 802.3/Ethernet uses a 32-bit CRC
Ethernet is but one example.  Others that come to mind include AAL5/ATM, 
FDDI, Token Ring, PPP over SONET/SDH.

> that is considerably stronger, and many subnetworks (notably dialup
> modem, mobile radio and satellite) also incorporate forward error
> correction, often in hardware.
> 
> Any new subnetwork designed to carry IP should therefore provide error
> detection at least as strong as a 16-bit CRC and preferably stronger.
I think we should make a stronger statement, and say 'at least as strong as
as the CRC-32 specified in ISO/IEC 3309:1991(E), "Information Technology -
         Telecommunications and information exchange between systems -
         High-level data link control (HDLC) procedures - Frame
         structure", International Organization For Standardization,
         Fourth edition 1991-06-01.
The reason for being specific is that other 32-bit CRCs exist (but are not 
widely implemented) which do not have as good error detection properties.

You might also want to mention something to the effect that if the subnet 
designer chooses to invent their own error detecting codes, they need to be 
very careful about their coding theory, and it's probably wiser to just reuse 
the CRC-32 defined for HDLC.

You can also mention that CRCs do not have very good error detection 
properties in the presence of error bursts or deletions longer than the 
generator polynomial (or is it the generator polynomial - 1?).  Use a CRC to 
match and/or include other ways (such as a length field) to detect deletions 
if they are possible in the subnet.

> Note that while this will achieve a very low undetected packet error
> rate, this will not (and need not) achieve a very low packet loss rate
> as the Internet protocols are better suited to dealing with lost
> packets than with corrupted packets.




From owner-pilc@lerc.nasa.gov  Fri Mar 10 13:06:47 2000
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Cannara wrote:
> 
> Sounds good Phil, and I'd add something a bit stronger in relation to
> higher-layers that, in fact, are not fully capable of offering reliability and
> performance under circumstances that the lower layers can do better on.  This
> relates to the problems TCP has with noncongestive loss, etc.

Is this saying that the lower layer is required to help the higher,
or that the lower layer replaces the higher?

WRT the latter, E2E asserts that HBH may be required for performance, 
but not correctness. Is there a counterexample?

Joe


From owner-pilc@lerc.nasa.gov  Fri Mar 10 13:09:49 2000
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Dan Grossman wrote:
> 
> > Okay, here's the current wording of the end-to-end principle section.
> > Any more comments? --Phil
> >
> > A major consideration in the efficient design of any layered
> > communication network are the appropriate layer(s) in which to
> > implement a given feature. This issue was first addressed in the
> Suggest changing 'feature' to 'function'.
> I don't believe that it was first addressed in the Saltzer, Reed and Clark
> paper.  If I remember, this was an ongoing debate, and Saltzer, Reed and Clark
> captured one set of views.

IMO, "captured" == "addressed" (or close enough).

> > seminal paper "End-to-End Arguments in System Design" [SRC81]. This
> > paper argued that many functions can be implemented properly *only* on
> > an end-to-end basis, i.e., at the higher protocol layers, outside the
> > subnetwork. These functions include ensuring the reliable delivery of
> I think Joe's point was that certain end-to-end services such as reliable
> delivery and
> confidentiality cannot be provided solely by composition of hop-by hop
> services.

The *only* above appears to suffice; it might be more clear to
include the caveat. Further, SRC81 didn't say 'some functions'
so much as 'many _end-to-end_ functions'.

> > data and the protection of information through cryptography.
> try... confidentiality and message integrity
> > Duplicating these functions at the lower protocol layers (i.e., within
> > the subnetwork) can be needlessly redundant or even harmful to cost
> > and performance.
> >
> > However, partial duplication of functionality in a lower layer
> > can *sometimes* be justified by performance, security or availability
> why the extra emphasis?

These two paragraphs appear to contradict each other. It may be
useful to unify them into a single statement that:

	Where HBH is implemented without consideration of E2E,
	it can be needlessly redundent or even harmful to cost and performance.
	However, HBH services can be designed to assist, rather
	than replace or ignore, their E2E counterparts. In this latter
	case, they can improve the performance of the E2E function.

Joe


From owner-pilc@lerc.nasa.gov  Fri Mar 10 13:21:23 2000
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Phil Karn wrote:
> 
> I just added this section in the "Reliability and Error Control" section.
> I will keep editing the LINK document, adding new text and comments as
> I get them, until tomorrow when I will send a snapshot to the ID editor
> with enough time to make the deadline... --Phil
> 
> The TCP, UDP and IP protocols all use a simple 16-bit 1's complement
> checksum algorithm to detect corrupted packets. The IP checksum
> protects only the IP header, while the TCP and UDP checksums protect
> both the TCP/UDP header and any user data. In TCP the checksum is
> mandatory while it is optional in UDP.
>
> These checksums are not very strong from a coding theory standpoint.
> But they are easy to compute in software, and various proposals to
> replace them with stronger checksums have failed. Experience has shown
> that nearly all subnetworks that carry the Internet protocols have a
> very low undetected error rate, so the existing Internet checksums are
> rarely called upon to catch an error. Most of the errors caught by the
> Internet checksum are generally the result of hardware or software
> failure than of a subnetwork transmission error. This lowers the
> motivation to overcome the performance and especially the backward
> compatibility problems of a new, stronger Internet checksum.

My understanding is that these checksums either "are intended to"
or "in practice usually catch only" software errors, not link errors.
On the original nodes, they were for memory corruption, etc. Does 
anyone recall whether they were ever intended to find link errors too?

(if not, the wording in paragraph 1 might be softened)

> Any new subnetwork designed to carry IP should therefore provide error
> detection at least as strong as a 16-bit CRC and preferably stronger.
> Note that while this will achieve a very low undetected packet error
> rate, this will not (and need not) achieve a very low packet loss rate
> as the Internet protocols are better suited to dealing with lost
> packets than with corrupted packets.

This appears to assert false positives are better than false negatives,
e.g., that turning the "drop" knob lower is bad when it lets through
bad packets. If so, it may be useful to state it directly.

Joe


From owner-pilc@lerc.nasa.gov  Fri Mar 10 13:22:59 2000
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>One implication of the above statement -- in the case when
>a single packet is held -- would be that it can
>be *any* single packet, not necessarily the earliest one.

>Why is that enough to kickstart TCP?
>One out-of-order (data) packet will trigger one dupack.
>Not enough to result in a fast retransmit.

Uh...yes. Let me mull over this.

Phil


From owner-pilc@lerc.nasa.gov  Fri Mar 10 13:28:54 2000
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Wow, I never thought the wording of a section on the end-to-end principle
would generate so much discussion!

I worded the current sections the way I did because of my observation
that many subnetwork designers do not understand that services such as
reliable delivery can only be properly implemented on an end-to-end basis,
so they go ahead and put reliability into the subnet without any regard
to whether it hurts or helps overall performance -- because they're thinking
about reliability, not performance.

With the ID deadline coming up in a few hours, I can't spend much more time
on this point. I'll maybe make one more pass to try to handle your comments,
but I have to move on. Now if we want to pick it up again after the deadline,
I suppose we could...

Phil



From owner-pilc@lerc.nasa.gov  Fri Mar 10 13:35:36 2000
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Regarding MTU assumptions...
Gorry Fairhurst and I went a round on this off-line;
here are our thoughts.

Joe

--------

Issue:

The concept of link MTU becomes fuzzy or nonexistent in today's
Internet.
The use of security, tunneling, and other header augmentation
over subpaths can substantially affect
the size of link MTU, even while a connection is in progress.

As a result, references to specific link MTU sizes, e.g., 1500B, may
not be appropriate. A designer attempting to achieve that size on
a single link will certainly have their efforts thwarted by IPSEC or
IPIP tunneling, among others.

In other words - even IP is a link layer, in IPSEC and in IPIP.
The best thing a link designer can do about MTU is _assume it can
change_.


From owner-pilc@lerc.nasa.gov  Fri Mar 10 13:49:02 2000
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From: Phil Karn <karn@ka9q.ampr.org>
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>The correct reference is to PPP over HDLC Framing, RFC 1622.  The 16 bit CRC

That's 1662, not 1622.

Phil


From owner-pilc@lerc.nasa.gov  Fri Mar 10 13:51:45 2000
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>Aside:  wouldn't strict application of the end-to-end principle indicate that
>strong end-to-end mechanisms are needed to provide assurance of accurate data 
>transfer? (Don't bother to answer this until after the deadline :-)

I'll answer it anyway. Yes, they do. But this doesn't really argue for
changing TCP. 

For some time I've thought it would be a good idea for application
layer protocols like FTP, SMTP and HTTP to exchange MD5 (or similar)
hashes of their transferred files as a final end-to-end integrity
check. (I implemented a XMD5 command in my own FTP server many years
ago).  TCP is often not implemented in a sufficiently end-to-end
fashion, as in some OSes you can have truncated files that aren't
indicated as such to the user.

Phil



From owner-pilc@lerc.nasa.gov  Fri Mar 10 14:12:20 2000
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>You might also want to mention something to the effect that if the subnet 
>designer chooses to invent their own error detecting codes, they need to be 
>very careful about their coding theory, and it's probably wiser to just reuse 
>the CRC-32 defined for HDLC.

There are even stronger alternatives, so I don't want to specifically
mandate the 32-bit CRC. E.g., in concatenated coding systems with a
Reed-Solomon outer code, the usual number of parity bytes chosen for
the RS code will provide an extremely low undetected error rate, much
lower than for a 32-bit CRC. These codes are frequently used on
satellite channels.

>You can also mention that CRCs do not have very good error detection 
>properties in the presence of error bursts or deletions longer than the 
>generator polynomial (or is it the generator polynomial - 1?).  Use a CRC to 

Well, the worst-case failure rate of a CRC is generally on the order
of 1/2^N, where N is the number of bits in the remainder. As I recall,
they do better when the number of errors is less than N, i.e., they
detect them all. So yeah, CRCs don't do as well on long error bursts,
but that's relative to the short bursts where they're 100% effective.

I think 1/2^32 (worst case) is small enough to suit us, don't you
think?

Phil





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From: Dan Grossman <dan@dma.isg.mot.com>
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> >You might also want to mention something to the effect that if the subnet 
> >designer chooses to invent their own error detecting codes, they need to be 
> >very careful about their coding theory, and it's probably wiser to just reuse 
> >the CRC-32 defined for HDLC.
> 
> There are even stronger alternatives, so I don't want to specifically
> mandate the 32-bit CRC. E.g., in concatenated coding systems with a
> Reed-Solomon outer code, the usual number of parity bytes chosen for
> the RS code will provide an extremely low undetected error rate, much
> lower than for a 32-bit CRC. These codes are frequently used on
> satellite channels.
Point taken.  On the other hand, I vividly recollect an ancient war where the 
CRC in question was so poorly designed that it could not detect any odd-length 
error burst.  So, yeah, wiser to just reuse an existing error detecting code.
> 
> >You can also mention that CRCs do not have very good error detection 
> >properties in the presence of error bursts or deletions longer than the 
> >generator polynomial (or is it the generator polynomial - 1?).  Use a CRC to 
> 
> Well, the worst-case failure rate of a CRC is generally on the order
> of 1/2^N, where N is the number of bits in the remainder. As I recall,
> they do better when the number of errors is less than N, i.e., they
> detect them all. So yeah, CRCs don't do as well on long error bursts,
> but that's relative to the short bursts where they're 100% effective.
I didn't intend to get to that level of detail, but now that you've written it 
down, maybe it would be better to put it in the draft.
> I think 1/2^32 (worst case) is small enough to suit us, don't you
> think?
Sure.  1/2^16 probably isn't, though.  At least if it's my financial data 
that's being protected.

Dan 
> Phil
> 
> 
> 




From owner-pilc@lerc.nasa.gov  Fri Mar 10 15:17:06 2000
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From: Phil Karn <karn@ka9q.ampr.org>
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>of my knowledge, there's nothing inherent in the RED scheme that
>requires randomly choosing a packet currently *in* the queue for
>dropping. (A minor optimization is to use the arrival of a new

I thought it did, but it's been a while since I read up on it.

Phil


From owner-pilc@lerc.nasa.gov  Fri Mar 10 15:29:25 2000
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Subject: Re: Draft on Advice for Internet Subnetwork Designers
References: <200003011811.KAA21226@boreas.isi.edu> <38BE7D3E.71079764@erg.abdn.ac.uk> <38BEB03F.4B7D352A@isi.edu> <38BF7021.D1246081@erg.abdn.ac.uk> <38BFED49.4929A6AB@isi.edu> <38C00C62.AF841B3E@isi.edu> <38C8FC80.6C3BC7BB@erg.abdn.ac.uk> <38C930B6.2F06A0AE@isi.edu>
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Good.  Why not add that the MTU at the lowest layer should be large enough to
handle all known tunnellings of all other bottom layers.  Otherwise,
fragmentation must be handled there, as it has been for FDDI/Enet or
Token/Enet bridging?

Phil's comment about MD5, etc. completion checks is a good recommendation, and
the session layer has that as a logical responsibility, to be done often
enough that it fits users' timescales of activity.

Alex


Joe Touch wrote:
> 
> Regarding MTU assumptions...
> Gorry Fairhurst and I went a round on this off-line;
> here are our thoughts.
> 
> Joe
> 
> --------
> 
> Issue:
> 
> The concept of link MTU becomes fuzzy or nonexistent in today's
> Internet.
> The use of security, tunneling, and other header augmentation
> over subpaths can substantially affect
> the size of link MTU, even while a connection is in progress.
> 
> As a result, references to specific link MTU sizes, e.g., 1500B, may
> not be appropriate. A designer attempting to achieve that size on
> a single link will certainly have their efforts thwarted by IPSEC or
> IPIP tunneling, among others.
> 
> In other words - even IP is a link layer, in IPSEC and in IPIP.
> The best thing a link designer can do about MTU is _assume it can
> change_.


From owner-pilc@lerc.nasa.gov  Fri Mar 10 15:59:27 2000
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Since there've long been many experiences in networking outside the Internet
protocols, even before them, it's often been assumed, in my engineering
experience, that each layer does its every best to assure delivery to the next
node, without sacrificing available performance (note lack of saying: 
"bandwidth", which is erroneously used these days).

So, Enet for example, makes better use of a shared medium than does Token, at
most common usage levels, but it guarantees (within 32-bits of CRC) that
received data are right.  Its CSMA/CD is faster at resolving contention
without loss delays than Token and Token is better than any transport,
including TCP.  Token Ring even had "assured delivery" in the IBM spec, but
not implemented.  Imagine TCP on Enet -- if a collision were to go unsensed,
then we'd be back to Aloha Net, waiting 100s of milliseconds for TCP (NCP,
SPP...) to recover the loss that Enet recovers in a millisecond.  This
obviously makes no sense, which is what Abramson and Metcalfe determined
decades ago and solved by adding swift, effective contention handling for
layer 1.

The e-to-e principles are interesting, but in a real sense, bureaucratic: 
"work with TCP, it's our old buddy and we worked on it for all these
years..."  No layer is fully independent of those below, and those below have
the purpose of delivering services to layers above that are useful to these
upper layers.  There's no 3rd tablet from the mount that says "Thou shalt not
communicate between protocol layers", especially when things can be made to
perform better and more reliably by doing so.  Service to Mankind is
engineering's purpose, right?

So, since TCP counts and tracks individual packets (ok, segments), and has
algorithms pasted on over the years to help router (ok, gateway) queues avoid
embarassment, why should anyone balk at giving it more help from any layer
below -- its backoff algorithm is there to help the network-layer queues, not
just the end-station's slowness.  We've already seen a good deal of mucking
about among TCP/IP layers and even removal of useful services, such as the IP
ID in v6, apparently because no one really understood all the implications. 
We don't need many np simulations to understand that TCP needs help to perform
reasonably in a busy, lossy network. 

Oh, what's "HBH" -- should we buy or sell?  Thanks,

Alex


Joe Touch wrote:
> 
> Cannara wrote:
> >
> > Sounds good Phil, and I'd add something a bit stronger in relation to
> > higher-layers that, in fact, are not fully capable of offering reliability and
> > performance under circumstances that the lower layers can do better on.  This
> > relates to the problems TCP has with noncongestive loss, etc.
> 
> Is this saying that the lower layer is required to help the higher,
> or that the lower layer replaces the higher?
> 
> WRT the latter, E2E asserts that HBH may be required for performance,
> but not correctness. Is there a counterexample?
> 
> Joe


From owner-pilc@lerc.nasa.gov  Fri Mar 10 15:59:28 2000
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Date: Fri, 10 Mar 2000 12:52:27 -0700 (MST)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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To: gja@lucent.com, karn@qualcomm.com
Subject: Re: Kicking TCP
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> From: Phil Karn <karn@ka9q.ampr.org>

> >of my knowledge, there's nothing inherent in the RED scheme that
> >requires randomly choosing a packet currently *in* the queue for
> >dropping. (A minor optimization is to use the arrival of a new
>
> I thought it did, but it's been a while since I read up on it.

The idea behind random drop (or so I think, and my recent words make my
thoughts very suspect) is to make the probability that any stream loses
a packet increase with the number of packets that the stream has in the
queue.  If you follow the easiest and old tactic of always dropping the
newest packet, then greedy bursty streams get an unfair advantage.

It's not very random to always drop the oldest or the newest, and not
much better to choose randomly between the oldest and the newest packet
(unless your queue has depth of 1, but no one cares about that case).


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Fri Mar 10 16:08:24 2000
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Date: Fri, 10 Mar 2000 15:22:54 -0500
To: Vernon Schryver <vjs@calcite.rhyolite.com>
From: William D Ivancic <William.D.Ivancic@lerc.nasa.gov>
Subject: Re: Kicking TCP
Cc: gja@lucent.com, karn@qualcomm.com, pilc@grc.nasa.gov,
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Regarding WRED, I believe you randomly drop at the input to the queue.  You
start dropping slowly and randomly at a predefined threshold and continue
dropping more aggressively as the queue gets closer to its high limit at
which point you take some other action (drop, reset precedence, put into
another queue, or whatever else the specification allows and the router is
capable of).

Will Ivancic  wivancic@grc.nasa.gov

At 12:52 PM 3/10/00 -0700, Vernon Schryver wrote:
>> From: Phil Karn <karn@ka9q.ampr.org>
>
>> >of my knowledge, there's nothing inherent in the RED scheme that
>> >requires randomly choosing a packet currently *in* the queue for
>> >dropping. (A minor optimization is to use the arrival of a new
>>
>> I thought it did, but it's been a while since I read up on it.
>
>The idea behind random drop (or so I think, and my recent words make my
>thoughts very suspect) is to make the probability that any stream loses
>a packet increase with the number of packets that the stream has in the
>queue.  If you follow the easiest and old tactic of always dropping the
>newest packet, then greedy bursty streams get an unfair advantage.
>
>It's not very random to always drop the oldest or the newest, and not
>much better to choose randomly between the oldest and the newest packet
>(unless your queue has depth of 1, but no one cares about that case).
>
>
>Vernon Schryver    vjs@rhyolite.com



From owner-pilc@lerc.nasa.gov  Fri Mar 10 16:08:55 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: pilc@grc.nasa.gov
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I've written this text to kick off the section on QoS, fairness vs performance,
and congestion signalling. --Phil

The IP header includes a 1-byte TOS (type of service) field. This
field is broken into two parts: a 3-bit precedence field that
indicates the relative importance of this packet compared to others,
and a 4-bit "qualitative tradeoff" field to indicate if low delay,
high throughput, high reliability or low cost is qualitatively more
important for this particular packet. One TOS bit is reserved.

Although the TOS field has not been heavily used in the past, it is
becoming increasingly important to the provision of Quality of Service
(QoS) features necessary for services such as Voice over IP (VoIP).

Many subnetwork designers are faced with inherent tradeoffs between
delay, throughput, reliability and cost. When this is the case, the
subnetwork should provide an interface to IP that allows the IP TOS
bits to control these tradeoffs on a per-packet basis. For example,
when a subnetwork implements a hop-by-hop retransmission scheme to
improve reliability, this invariably comes at the cost of increased
delay variance. An IP packet presented to this subnetwork with the
"low delay" bit set in its TOS field should disable hop-by-hop
retransmission within the subnetwork for that particular
packet. Another example is link forward error correction, which can
increase reliability at the expense of throughput without
significantly affecting delay.


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Vernon Schryver wrote:

> > From: Phil Karn <karn@ka9q.ampr.org>
>
> > ...
> > > 1. have the link layer tell the TCP state machines to retransmit any
> > >  unacknowledged TCP bytes by forcing their retransmission timers
> > >  to zero? ...
>
> > It's more than a technical layering violation, because how would
> > you do it if the TCP and the link layer are on separate machines?
> > You'd need to define some communication mechanism, which was the
> > intent of my ICMP SE message.
>
> Yes, but in many cases (e.g. mobile phones) the host and the router
> are pretty close to each other.  But I agree that reasonable queuing
> next to the link is far better.

What about the case where the mobile regain connectivity with another router
(of Base station)? Then aren't we loosing all the advantage of queueing?

Vincent

>
>
> > ...
> > Again, I agree. I will add text that encourages the subnetwork designer
> > to use any internal information it may have about the state of the
> > link or subnetwork to block the acceptance of IP packets if it is clear
> > that those packets would be dropped, and to hold onto the most recent
> > packets that it may have accepted before the link/subnet went down
> > and to try to deliver them after the link/subnet comes back up.
>
> That sound great.
>
> Something I forget to mention in my previous note is that private
> cooresondent pointed out that RED and other fancy queuing disciplines
> tend to encourage dropping the most recent instead oldest packets.
>
> Vernon Schryver    vjs@rhyolite.com

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From owner-pilc@lerc.nasa.gov  Fri Mar 10 17:35:50 2000
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>So, since TCP counts and tracks individual packets (ok, segments), and has
>algorithms pasted on over the years to help router (ok, gateway) queues avoid
>embarassment, why should anyone balk at giving it more help from any layer
>below -- its backoff algorithm is there to help the network-layer queues, not

That's fine, as long as the added lower-level functionality really does
help performance overall. I've seen too many overly complicated subnetwork
designs where this isn't the case, though.

Phil


From owner-pilc@lerc.nasa.gov  Fri Mar 10 17:38:56 2000
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Here's the text I just added to kick off the mobility section. Since the
deadline is looming, I'm going to submit the LINK document as it currently
stands, and update (again) the copy on my web page. --Phil

Internet users are increasingly mobile. Not only are many Internet
nodes laptop computers, but pocket organizers and mobile embedded
systems are also becoming nodes on the Internet. These nodes may
connect to many different access points on the Internet over time, and
they expect this to be largely transparent to their activities.
Except when they are not connected to the Internet at all, and for
performance differences when they are connected, they expect that
everything will "just work" regardless of their current Internet
attachment point or local subnetwork technology.

Mobility can be provided at any of several layers in the Internet
protocol stack, and there is ongoing debate as to which are the most
appropriate and efficient. Mobility is already an feature of certain
application layer protocols; the Post Office Protocol (POP) [RFC1939]
and the Internet Message Access Protocol (IMAP) [RFC2060] were created
specifically to provide mobility in the receipt of electronic mail.

Mobility can also be provided at the IP layer [RFC2002]. This
mechanism provides greater transparency, viz., IP addresses that
remain fixed as the nodes move, but at the cost of potentially
significant network overhead and increased delay because of the
non-optimum network routing and tunneling involved.

Some subnetworks may provide internal mobility, transparent to IP, as
a feature of their own internal routing mechanisms. To the extent that
these simplify routing at the IP layer, reduce the need for mechanisms
like Mobile IP, or exploit mechanisms unique to the subnetwork, this
is generally desirable. This is especially true when the subnetwork
covers a relatively small geographic area and the users move rapidly
between the attachment points within that area.

However, if the subnetwork is physically large and connects to other
parts of the Internet at multiple geographic points, care should be
taken to optimize the wide-area routing of packets between nodes on
the external Internet and nodes on the subnet. This is generally done
with "nearest exit" routing strategies. Because a given subnetwork may
be unaware of the actual physical location of a destination on another
subnetwork, it simply routes packets bound for the other subnetwork to
the nearest gateway between the two. This implies some awareness of IP
addressing and routing within the subnetwork. The subnetwork may wish
to use IP routing internally for wide area routing and restrict
subnetwork-specific routing to constrained geographic areas where the
effects of suboptimal routing are minimized.


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On Fri, 10 Mar 2000, Phil Karn wrote:

> The IP header includes a 1-byte TOS (type of service) field. This
> field is broken into two parts: a 3-bit precedence field that
> indicates the relative importance of this packet compared to others,
> and a 4-bit "qualitative tradeoff" field to indicate if low delay,
> high throughput, high reliability or low cost is qualitatively more
> important for this particular packet. One TOS bit is reserved.

Whoa. That all plays complete havoc with diffserv: 6-bit DSCP, and two
bits reserved. The TOS definitions as above aren't widely used, and
the diffserv architecture is effectively rewriting the field.

> Although the TOS field has not been heavily used in the past, it is
> becoming increasingly important to the provision of Quality of Service
> (QoS) features necessary for services such as Voice over IP (VoIP).

Very true - but not in its form as a TOS field per se.


> Many subnetwork designers are faced with inherent tradeoffs between
> delay, throughput, reliability and cost. When this is the case, the
> subnetwork should provide an interface to IP that allows the IP TOS
> bits to control these tradeoffs on a per-packet basis. For example,
> when a subnetwork implements a hop-by-hop retransmission scheme to
> improve reliability, this invariably comes at the cost of increased
> delay variance. An IP packet presented to this subnetwork with the
> "low delay" bit set in its TOS field should disable hop-by-hop
> retransmission within the subnetwork for that particular
> packet.

IMO we *really* don't want to encourage link designers to look at some
DSCP bit and assume low delay based on the assigned value when it is
an artefact of numbering and has nothing to do with low delay;
designing a link to interact well with some overblown QoS framework
sinking in its own complexity is a whole new ball game, and imo is
probably outside the scope of this draft.


> Another example 

of improving link reliability - but this is somewhat orthogonal to the
QoS discussion above, yes? You're not suggesting varying FEC depending
on QoS settings, surely...

> is link forward error correction, which can
> increase reliability at the expense of throughput without
> significantly affecting delay.

L.

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




From owner-pilc@lerc.nasa.gov  Fri Mar 10 18:09:20 2000
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> From: Phil Karn <karn@ka9q.ampr.org>

>            ...  An IP packet presented to this subnetwork with the
> "low delay" bit set in its TOS field should disable hop-by-hop
> retransmission within the subnetwork for that particular
> packet. Another example is link forward error correction, which can
> increase reliability at the expense of throughput without
> significantly affecting delay.


Why turn off retransmission for "low delay"?  The application protocols
I can think of off hand that currently and commonly use "low delay" are
NTP/UDP (as of a fairly recent change to (x?)ntpd), `timed` or TSP/UDP
(another time protocol), telnet, and rlogind.  I know of some other
applications, but they're much less common and some are even proprietary,
but follow those two families.  Turning off retransmission for the UDP
applications is sounds like a good idea, but not good for the TCP
applications.  For them, you want to retransmit to avoid the longer
end-to-end timeout, and perhaps move them higher in your hop-by-hop
retransmission queue.  In other words, turning off hop-by-hop
retransmissions for "low delay" sounds either too specific a remedy or to
general.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Fri Mar 10 18:15:27 2000
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>Whoa. That all plays complete havoc with diffserv: 6-bit DSCP, and two
>bits reserved. The TOS definitions as above aren't widely used, and
>the diffserv architecture is effectively rewriting the field.

Okay, I wasn't aware of that, and I guess it's time to read up more on
diffserv. Care to contribute some text? (I've already submitted the draft,
but we can continue to edit the document for the next round).

>IMO we *really* don't want to encourage link designers to look at some
>DSCP bit and assume low delay based on the assigned value when it is

What I wanted to say is that when a subnetwork designer is faced with a
design tradeoff between delay/reliability/throughput/cost, instead of
just picking an operating point he should provide "knobs" that IP can
diddle to make the tradeoffs on a per-packet basis.

>of improving link reliability - but this is somewhat orthogonal to the
>QoS discussion above, yes? You're not suggesting varying FEC depending
>on QoS settings, surely...

Actually, yes. It makes quite a bit of sense in some environments,
particularly those using spread spectrum/CDMA. Varying the FEC code
rate and varying the power are closely related approaches. The
Qualcomm HDR system uses a constant channel symbol rate and
transmitter power and varies the modulation type and FEC code rate
(and the user throughput) as a function of signal-to-interference
ratio. Nothing says we couldn't bias those settings according to QoS
bits if that were useful.

Phil


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Cannara wrote:
> 
> Good.  Why not add that the MTU at the lowest layer should be large enough to
> handle all known tunnellings of all other bottom layers.

This can't be determined a-priori, i.e., by the link designer.

Big is good; if not big (or anyway), then either support fragmentation, 
or preferably support PMTU. Don't do anything that assumes knowledge of
the MTU before it can be determined.

Joe


From owner-pilc@lerc.nasa.gov  Fri Mar 10 19:08:16 2000
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On Fri, 10 Mar 2000, Phil Karn wrote:

> >Whoa. That all plays complete havoc with diffserv: 6-bit DSCP, and two
> >bits reserved. The TOS definitions as above aren't widely used, and
> >the diffserv architecture is effectively rewriting the field.
> 
> Okay, I wasn't aware of that, and I guess it's time to read up more on
> diffserv. Care to contribute some text? (I've already submitted the draft,
> but we can continue to edit the document for the next round).

okay. taking the general form of what you wrote...

The IP header includes a type-of-service field [RFC791]. This was not
widely used, and has been redefined as the 'codepoint', identifying a
set of specified delivery requirements, in the differentiated services
architecture [RFC2474]. This field is becoming increasingly important
to the provision of Quality of Service (QoS) features necessary for a
variety of services.

Before considering using any labelling in this field to improve link
performance, the subnetwork MUST be aware of the definition and use of
the field within the local network. It is the responsibility of the
local network to control the access to, and use of, this field.

Many subnetwork designers are faced with inherent tradeoffs between
reliability, delay, throughput, and cost. When this is the case, the
subnetwork should use the codepoint to map the locally-understood QoS
specifications to a set of tradeoff choices on a per-packet basis.

For example, a subnetwork might implement a hop-by-hop retransmission
scheme to improve reliability. An IP packet presented to this
subnetwork with a codepoint indicating a low reliability requirement,
via a high drop precedence setting, might not be subject to
retransmission. Packets marked for a low drop precedence requirement,
indicating preferential treatment, would be more likely to be held and
retransmitted if required.

L.

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








From owner-pilc@lerc.nasa.gov  Fri Mar 10 22:27:49 2000
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From: Phil Karn <karn@qualcomm.com>
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Hi all,

I got an automated bounce message from the Internet Drafts coordinator
saying that my submission of the PILC LINK draft came after the
deadline, even though I *know* I got it in before 5pm EST.

I sent two copies, one from home and one from work. I sent the copy
from work when I noticed a problem with the path through Time Warner
from home. The copy from work went right through and completed about
five minutes before the deadline. Hopefully the bounce came from the
earlier home copy that got stuck.

If not, this is just a bureaucratic annoyance, and we can continue to
work from the copy on my personal web page:

http://people.qualcomm.com/karn/pilc.txt

--Phil



From owner-pilc@lerc.nasa.gov  Sun Mar 12 14:06:47 2000
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In-reply-to: <200003081651.IAA01772@homer.ka9q.ampr.org> (message from Phil
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Phil,

> Reference: MPEG-2 or DVB reference.

I did'nt see Gorry's answer, and did'nt see the references either in
draft-ietf-pilc-link-design-02.txt
so if you still need that

DVB reference is

  ETSI EN 301 192 V1.2.1 (1999-06)
 
MPEG-2 reference is

  ISO/IEC 13818

Patrick.


From owner-pilc@lerc.nasa.gov  Mon Mar 13 10:08:28 2000
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Reply-To: mallman@grc.nasa.gov
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Organization: Late Night Hackers, NASA Glenn, Cleveland, Ohio
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Folks-

The updated PILC I-Ds are available from:

    http://pilc.grc.nasa.gov/pilc/

From what I have heard from the co-chairs, these will be in the
normal Internet-Draft archives as soon as they make it through the
queue.

allman


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


From owner-pilc@lerc.nasa.gov  Mon Mar 13 18:51:57 2000
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References: <200003081340.IAA15226@noah.dma.isg.mot.com> <38C6977F.35048669@isi.edu> <200003092345.PAA09786@homer.ka9q.ampr.org> <38C86F9E.3F8BD601@attglobal.net> <38C92C5F.798967C@isi.edu> <38C953AD.734E99D7@attglobal.net> <200003102137.NAA19256@homer.ka9q.ampr.org>
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Yes, that goes without saying Phil.

Alex


Phil Karn wrote:
> 
> >So, since TCP counts and tracks individual packets (ok, segments), and has
> >algorithms pasted on over the years to help router (ok, gateway) queues avoid
> >embarassment, why should anyone balk at giving it more help from any layer
> >below -- its backoff algorithm is there to help the network-layer queues, not
> 
> That's fine, as long as the added lower-level functionality really does
> help performance overall. I've seen too many overly complicated subnetwork
> designs where this isn't the case, though.
> 
> Phil


From owner-pilc@lerc.nasa.gov  Mon Mar 13 20:32:52 2000
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	(message from Marie-Jose Montpetit on Thu, 9 Mar 2000 05:48:40 -0800)
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>> >On the topic of packet discard especially on those subnet where capacity
>> is
>> >expensive or limited I would add that when possible packets marked form
>> >another subnet should be discarded before transmission on the link.
>> 
>> Is this an issue for the subnetwork designer, or for the router that
>> drives the subnetwork?

>If we start from the premise that the equipment that is connected to the
>subnetwork is unaware of the particularities of the subnetwork, which I
>think is usually the case, then it is an issue for the designer.

I asked the question because I assume by "marked packet" you meant marking
at the IP layer, e.g., with ECN. That implies that it's up to an IP
router to drop the packet if it wishes, since subnetworks know nothing
about IP (or about other subnetworks, for that matter).

Phil


From owner-pilc@lerc.nasa.gov  Tue Mar 14 02:02:06 2000
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To: gorry@erg.abdn.ac.uk
cc: pilc@grc.nasa.gov
From: Sally Floyd <floyd@aciri.org>
Subject: Re: PILC ASYM document 
Date: Mon, 13 Mar 2000 22:11:34 -0800
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>I disagree that TCP "works well" over all links.  If you define
>"well" as
>meaning doesn't fail to delivery (eventually) then that's OK, but normally
>that's
>not my thinking.  TCP is end-to-end, but link designers need to take on board
>what is really happening.  Once the subnet designers document is clear, I
>suggest
>we will probably see the need for the ASYM document.

Yes, I agree completely about the need for the ASYM document.  I
think the fact that TCP currently lacks congestion control for
traffic on the return path (for the case where the TCP flow's
traffic on the return path consists only of ACK packets) is a
significant weakness that happens to manifest particularly strongly
in the case of asymmetric links.  It seems to me that adding some
form of congestion control for this return-path traffic will at
some point become necessary.  I don't view congestion control for
ACK traffic as contorting TCP to accomodate the oddities of some
defective link layer.  I view it as addressing a fundamental
requirement of the transport protocol to have effective end-to-end
congestion control, as much as possible, for all of its traffic.

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



From owner-pilc@lerc.nasa.gov  Tue Mar 14 02:19:26 2000
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From: Sally Floyd <floyd@aciri.org>
Subject: Re: Kicking TCP 
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Phil -

>>of my knowledge, there's nothing inherent in the RED scheme that
>>requires randomly choosing a packet currently *in* the queue for
>>dropping. (A minor optimization is to use the arrival of a new
>
>I thought it did, but it's been a while since I read up on it.

Nope.  The basic RED scheme randomly drops arriving packets.  You
could have variants, if you wanted, where you randomly made the
decision to drop a packet when a new packet arrived at the queue,
but where instead of dropping the arriving packet, you dropped some
other packet instead (the packet at the head of the queue, or some
random packet in the queue).  But I haven't really looked at any
of these options.  (I was also under the impression that dropping
a packet in the middle of the queue was likely to be more expensive
for the router than dropping the arriving packet...)

I have a brief note about "RED with Drop From Front" on the RED
web page at "http://www.aciri.org/floyd/red.html".

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


From owner-pilc@lerc.nasa.gov  Tue Mar 14 02:51:53 2000
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In general, we should not take a single transport protocol (TCP or any other)
out of any family of protocols that carry data, and give it the responsibility
of network "congestion control".  This was done historically because TCP was
dominant in the old Internet and so it could be used to effectively help
routers with their queueing, when it was apparent proper network-layer design
for high loads hadn't been done.  Today, the mix of traffic continues to
change in ways that increasingly expose this bandaid.  Addressing the real
problem at the network layer will be the key, as others have suggested.  

Imagining that TCP has an obligation and capability to control congestion
remains an engineering mistake.  The TCPs people actually use and see
operational every day are extremely inefficient, in comparison with what could
legitimately expect of the years of effort that went into its various
algorithimic components.  Part of this inefficiency is certainly laid to
default parameter settings (timeouts, etc.).  However, the fundamental flaw in
the logic is that TCP has insufficient information to make proper decisions
and is hampered further by provider policies on packet delivery.  The network
layer will need to manage its queues and drops and provide help to higher
layers, not depend on them.  Fortunately, the problems with TCP/IP's current
interwining become extremely evident on the kinds of links discussed here. 
That should be our impetus to deal with the problems effectively rather than
bureaucratically.

Alex


Sally Floyd wrote:
> 
> >I disagree that TCP "works well" over all links.  If you define
> >"well" as
> >meaning doesn't fail to delivery (eventually) then that's OK, but normally
> >that's
> >not my thinking.  TCP is end-to-end, but link designers need to take on board
> >what is really happening.  Once the subnet designers document is clear, I
> >suggest
> >we will probably see the need for the ASYM document.
> 
> Yes, I agree completely about the need for the ASYM document.  I
> think the fact that TCP currently lacks congestion control for
> traffic on the return path (for the case where the TCP flow's
> traffic on the return path consists only of ACK packets) is a
> significant weakness that happens to manifest particularly strongly
> in the case of asymmetric links.  It seems to me that adding some
> form of congestion control for this return-path traffic will at
> some point become necessary.  I don't view congestion control for
> ACK traffic as contorting TCP to accomodate the oddities of some
> defective link layer.  I view it as addressing a fundamental
> requirement of the transport protocol to have effective end-to-end
> congestion control, as much as possible, for all of its traffic.
> 
> - Sally


From owner-pilc@lerc.nasa.gov  Tue Mar 14 13:33:51 2000
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To: cannara@attglobal.net
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From: Sally Floyd <floyd@aciri.org>
Subject: Re: PILC ASYM document 
Date: Tue, 14 Mar 2000 09:14:36 -0800
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>Imagining that TCP has an obligation and capability to control congestion
>remains an engineering mistake.  

Odd.  The lessons that I would take from the last twelve years of
TCP congestion control are that transport-level, end-to-end congestion
control is remarkably robust; that it has successfully prevented
congestion collapse and enabled the evolution of the Internet for
the past twelve years; and that transport-level congestion control
coupled with a minimal network level allows a great heterogeneity
of link-level technologies and higher-level applications as well
as the evolution of other transport-level congestion control
mechanisms for diverse classes of applications (e.g., unreliable
unicast, semi-reliable streaming multimedia, unreliable and reliable
multicast, etc.).

The further lessons that I would take from the last twelve years of
TCP congestion control would be that TCP and other transport
protocols can and should be made even more robust to a variety of
behaviors in the underlying network (e.g., reordered packets due
to link-level transmissions, congestion on the return path); and
that adding some forms of explicit communication between the
transport layer and higher or lower layers (e.g., Explicit Congestion
Notification) can be of significant benefit.

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


From owner-pilc@lerc.nasa.gov  Tue Mar 14 14:42:05 2000
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From: Marie-Jose Montpetit <marie@teledesic.com>
To: "'karn@qualcomm.com'" <karn@qualcomm.com>
Cc: dan@dma.isg.mot.com, afalk@PanAmSat.com, pilc@grc.nasa.gov
Subject: RE: Draft on Advice for Internet Subnetwork Designers
Date: Tue, 14 Mar 2000 10:31:51 -0800
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I don't want to get in a nomenclature argument but by point was that if the
subnetwork provides a transport layer and that transport layer itself has
some queue management then some of the discard can be done ahead of the
router.

Marie-Jose

-----Original Message-----
From: Phil Karn [mailto:karn@ka9q.ampr.org]
Sent: Monday, March 13, 2000 4:44 PM
To: marie@teledesic.com
Cc: dan@dma.isg.mot.com; afalk@PanAmSat.com; pilc@grc.nasa.gov
Subject: Re: Draft on Advice for Internet Subnetwork Designers


>> >On the topic of packet discard especially on those subnet where capacity
>> is
>> >expensive or limited I would add that when possible packets marked form
>> >another subnet should be discarded before transmission on the link.
>> 
>> Is this an issue for the subnetwork designer, or for the router that
>> drives the subnetwork?

>If we start from the premise that the equipment that is connected to the
>subnetwork is unaware of the particularities of the subnetwork, which I
>think is usually the case, then it is an issue for the designer.

I asked the question because I assume by "marked packet" you meant marking
at the IP layer, e.g., with ECN. That implies that it's up to an IP
router to drop the packet if it wishes, since subnetworks know nothing
about IP (or about other subnetworks, for that matter).

Phil


From owner-pilc@lerc.nasa.gov  Tue Mar 14 17:59:25 2000
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On Tue, 14 Mar 2000, Sally Floyd wrote:

> Odd.  The lessons that I would take from the last twelve years of
> TCP congestion control are that transport-level, end-to-end congestion
> control is remarkably robust

so robust (in the self-perpetuating sense) that improvements to it
such as rate-based pacing cannot be incrementally deployed. (You can
only incrementally deploy mechanisms that are equally or more
aggressive for network capacity - bit of a quandary, that.)


> that it has successfully prevented
> congestion collapse and enabled the evolution of the Internet for
> the past twelve years; and that transport-level congestion control
> coupled with a minimal network level allows a great heterogeneity
> of link-level technologies and higher-level applications as well
> as the evolution of other transport-level congestion control
> mechanisms for diverse classes of applications (e.g., unreliable
> unicast, semi-reliable streaming multimedia, unreliable and reliable
> multicast, etc.).

To me, the growing emphasis on TCP-friendliness for these mechanisms
indicates the limited scope for evolution available. The mechanisms
need to display the same level of aggression as TCP to coexist well
with TCP.

Transport-level congestion control is certainly a success, but I
suspect that improvements are being boxed into a corner - an
evolutionary dead end, if you will - by the installed
base. Ever-decreasing returns?

(To extend the evolution analogy slightly further: TCP is a very
 very successful species of dinosaur, set to dominate the planet
 for millions of years. There are no global extermination events
 planned; there are currently few evolutionary pressures on TCP in the
 'congestion control' competition space. Now, if there was a reason to
 deploy queuing mechanisms favouring e.g. rate-based pacing, which
 would be equivalent to altering the global temperature...)


> The further lessons that I would take from the last twelve years of
> TCP congestion control would be that TCP and other transport
> protocols can and should be made even more robust to a variety of
> behaviors in the underlying network (e.g., reordered packets due
> to link-level transmissions, congestion on the return path)

hear, hear.

> and
> that adding some forms of explicit communication between the
> transport layer and higher or lower layers (e.g., Explicit Congestion
> Notification) can be of significant benefit.

although TCP seems to have been doing just fine without that thus far.

L.

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



From owner-pilc@lerc.nasa.gov  Tue Mar 14 17:59:57 2000
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To: L.Wood@eim.surrey.ac.uk
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From: Sally Floyd <floyd@aciri.org>
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Lloyd -

>To me, the growing emphasis on TCP-friendliness for these mechanisms
>indicates the limited scope for evolution available. The mechanisms
>need to display the same level of aggression as TCP to coexist well
>with TCP.

I agree that the requirement for TCP-friendliness limits the scope
for evolution of best-effort traffic competing in FIFO queues with
TCP.  I think of diffserv as the mechanism that will allow
transport-level evolution past TCP-friendliness.  E.g., I could
imagine a class of traffic, labeled by the diffserv field, that is
segregated from best-effort TCP, and that operates by its own rules.
E.g., a class for congestion-based pricing; a class for per-flow
scheduling and users optimizing their own utility functions; a
class for flows all using class-X-compatible congestion control,
etc.  (I don't know what the mechanism is for ensuring that all
routers along the path understand diffserv class X, or something
else appropriate, but I imagine that will come...)

I don't understand what prevents the incremental deployment of 
rate-based pacing, by the way.  Are you suggesting that rate-based
TCP flows competing with regular TCP flows would face starvation?
I wouldn't think so.

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



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From owner-pilc@lerc.nasa.gov  Wed Mar 15 05:04:44 2000
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This may be controversial, but I'd like to add that the use of "subnetwork" is
misleading and a better name would help.  Subnetwork could be applied at any
layer, including the application, so it doesn't reflect the meaning of a
perhaps purely physical medium, a tunnelled path, etc.  Since we already
confuse students with "gateway" and "subnet" in the TCP/IP world, I hope we
can get a better term.

Alex


Marie-Jose Montpetit wrote:
> 
> I don't want to get in a nomenclature argument but by point was that if the
> subnetwork provides a transport layer and that transport layer itself has
> some queue management then some of the discard can be done ahead of the
> router.
> 
> Marie-Jose
[clip]


From owner-pilc@lerc.nasa.gov  Wed Mar 15 07:17:35 2000
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On Tue, 14 Mar 2000, Sally Floyd wrote:

> >To me, the growing emphasis on TCP-friendliness for these mechanisms
> >indicates the limited scope for evolution available. The mechanisms
> >need to display the same level of aggression as TCP to coexist well
> >with TCP.
> 
> I agree that the requirement for TCP-friendliness limits the scope
> for evolution of best-effort traffic competing in FIFO queues with
> TCP.  I think of diffserv as the mechanism that will allow
> transport-level evolution past TCP-friendliness.  

Yes - however, making diffserv adequately TCP-friendly (how do you
handle the asymmetry of most TCP connections? What happens when
responsive TCP and unresponsive UDP-based traffic share a class?) is a
problem in itself.


> I don't understand what prevents the incremental deployment of 
> rate-based pacing, by the way.

There's currently a lack of clear performance benefit as motivation.
No performance benefit or slight degradation == no incentive to
change.


> Are you suggesting that rate-based
> TCP flows competing with regular TCP flows would face starvation?
> I wouldn't think so.

Aggarwal et al's paper had Reno consistently outperforming rate-based
pacing (end of section D) from slow start with large buffer
sizes. That's IMO sufficient disincentive to deploy rate-based pacing
as is.

http://www.cs.washington.edu/homes/savage/
Understanding the Performance of TCP Pacing,
Amit Aggarwal, Stefan Savage, and Tom Anderson,
to appear in Proceedings of the 2000 IEEE Infocom
Conference, Tel-Aviv, Israel, March, 2000.

L.

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



From owner-pilc@lerc.nasa.gov  Wed Mar 15 10:40:31 2000
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Date: Wed, 15 Mar 2000 09:37:51 -0500 (EST)
From: mukul goyal <mukul@cis.ohio-state.edu>
cc: pilc@grc.nasa.gov
Subject: Re: PILC ASYM document 
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On Wed, 15 Mar 2000, Lloyd Wood wrote:

> On Tue, 14 Mar 2000, Sally Floyd wrote:
> 
> > >To me, the growing emphasis on TCP-friendliness for these mechanisms
> > >indicates the limited scope for evolution available. The mechanisms
> > >need to display the same level of aggression as TCP to coexist well
> > >with TCP.
> > 
> > I agree that the requirement for TCP-friendliness limits the scope
> > for evolution of best-effort traffic competing in FIFO queues with
> > TCP.  I think of diffserv as the mechanism that will allow
> > transport-level evolution past TCP-friendliness.  
> 
> Yes - however, making diffserv adequately TCP-friendly (how do you
> handle the asymmetry of most TCP connections? What happens when
> responsive TCP and unresponsive UDP-based traffic share a class?) is a
> problem in itself.

I think that Sally was suggesting that some traffic classes in diffserv
can be used by non-TCP-friendly traffic.

Just for information, the other problem regarding TCP and UDP sharing a
traffic class has been studied in some details. We have an Internet Draft
on the issue. Our finding is that you have to mark 'excess' traffic of TCP
and UDP flows differently in order to achieve fair distribution of excess
resources. (Here, excess means unreserved resources.) The draft can be
seen at:
http://www.cis.ohio-state.edu/~jain/ietf/afstdy.htm

I think there is a draft/paper on the issue from Nortel folks also (N
Seddigh et al).





From owner-pilc@lerc.nasa.gov  Wed Mar 15 11:40:19 2000
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[This is now off-topic for pilc; discussion ends here.]

On Wed, 15 Mar 2000, mukul goyal wrote:
> On Wed, 15 Mar 2000, Lloyd Wood wrote:
> | On Tue, 14 Mar 2000, Sally Floyd wrote:
> | > On Tue, 14 Mar 2000, Lloyd Wood wrote:
> | > |To me, the growing emphasis on TCP-friendliness for these mechanisms
> | > |indicates the limited scope for evolution available. The mechanisms
> | > |need to display the same level of aggression as TCP to coexist well
> | > |with TCP.
> | > 
> | > I agree that the requirement for TCP-friendliness limits the scope
> | > for evolution of best-effort traffic competing in FIFO queues with
> | > TCP.  I think of diffserv as the mechanism that will allow
> | > transport-level evolution past TCP-friendliness.  
> | 
> | Yes - however, making diffserv adequately TCP-friendly (how do you
> | handle the asymmetry of most TCP connections? What happens when
> | responsive TCP and unresponsive UDP-based traffic share a class?) is a
> | problem in itself.
> 
> I think that Sally was suggesting that some traffic classes in diffserv
> can be used by non-TCP-friendly traffic.
> 
> Just for information, the other problem regarding TCP and UDP sharing a
> traffic class has been studied in some details. We have an Internet Draft
> on the issue. Our finding is that you have to mark 'excess' traffic of TCP
> and UDP flows differently in order to achieve fair distribution of excess
> resources.

Oh, draft-goyal-diffserv-afstdy-00.txt.

Unlike the position expressed in the conclusions of your draft, we
have shown that it is not necessary for the classifier to distinguish
between TCP and UDP packets in order to meaningfully utilize the three
drop precedences while treating TCP and UDP fairly. (Distinguishing
between TCP and UDP in the network is generally a bad idea, and
then there's IPSec.)

http://www.ee.surrey.ac.uk/Personal/G.Pavlou/Publications/Conference-papers/Andrik-00a-abstract.html

A fair traffic conditioner for the assured service in a differentiated
services Internet 
I. Andrikopoulos, L. Wood, G. Pavlou
to appear in the proceedings of the IEEE International
Conference on Communications (ICC 2000), New Orleans, June 2000. 

L.

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


> (Here, excess means unreserved resources.) The draft can be
> seen at:
> http://www.cis.ohio-state.edu/~jain/ietf/afstdy.htm
> 
> I think there is a draft/paper on the issue from Nortel folks also (N
> Seddigh et al).





From owner-pilc@lerc.nasa.gov  Wed Mar 15 12:12:07 2000
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Date: Wed, 15 Mar 2000 11:24:43 -0500 (EST)
From: mukul goyal <mukul@cis.ohio-state.edu>
To: Lloyd Wood <l.wood@eim.surrey.ac.uk>
cc: pilc@grc.nasa.gov
Subject: Re: PILC ASYM document 
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I guess I will sort this out with Lloyd offline and may be post a summary
of the conclusions reached on this list! 

On Wed, 15 Mar 2000, Lloyd Wood wrote:

> [This is now off-topic for pilc; discussion ends here.]
> 
> On Wed, 15 Mar 2000, mukul goyal wrote:
> > On Wed, 15 Mar 2000, Lloyd Wood wrote:
> > | On Tue, 14 Mar 2000, Sally Floyd wrote:
> > | > On Tue, 14 Mar 2000, Lloyd Wood wrote:
> > | > |To me, the growing emphasis on TCP-friendliness for these mechanisms
> > | > |indicates the limited scope for evolution available. The mechanisms
> > | > |need to display the same level of aggression as TCP to coexist well
> > | > |with TCP.
> > | > 
> > | > I agree that the requirement for TCP-friendliness limits the scope
> > | > for evolution of best-effort traffic competing in FIFO queues with
> > | > TCP.  I think of diffserv as the mechanism that will allow
> > | > transport-level evolution past TCP-friendliness.  
> > | 
> > | Yes - however, making diffserv adequately TCP-friendly (how do you
> > | handle the asymmetry of most TCP connections? What happens when
> > | responsive TCP and unresponsive UDP-based traffic share a class?) is a
> > | problem in itself.
> > 
> > I think that Sally was suggesting that some traffic classes in diffserv
> > can be used by non-TCP-friendly traffic.
> > 
> > Just for information, the other problem regarding TCP and UDP sharing a
> > traffic class has been studied in some details. We have an Internet Draft
> > on the issue. Our finding is that you have to mark 'excess' traffic of TCP
> > and UDP flows differently in order to achieve fair distribution of excess
> > resources.
> 
> Oh, draft-goyal-diffserv-afstdy-00.txt.
> 
> Unlike the position expressed in the conclusions of your draft, we
> have shown that it is not necessary for the classifier to distinguish
> between TCP and UDP packets in order to meaningfully utilize the three
> drop precedences while treating TCP and UDP fairly. (Distinguishing
> between TCP and UDP in the network is generally a bad idea, and
> then there's IPSec.)
> 
> http://www.ee.surrey.ac.uk/Personal/G.Pavlou/Publications/Conference-papers/Andrik-00a-abstract.html
> 
> A fair traffic conditioner for the assured service in a differentiated
> services Internet 
> I. Andrikopoulos, L. Wood, G. Pavlou
> to appear in the proceedings of the IEEE International
> Conference on Communications (ICC 2000), New Orleans, June 2000. 
> 
> L.
> 
> <L.Wood@surrey.ac.uk>PGP<http://www.ee.surrey.ac.uk/Personal/L.Wood/>
> 
> 
> > (Here, excess means unreserved resources.) The draft can be
> > seen at:
> > http://www.cis.ohio-state.edu/~jain/ietf/afstdy.htm
> > 
> > I think there is a draft/paper on the issue from Nortel folks also (N
> > Seddigh et al).
> 
> 
> 



From owner-pilc@lerc.nasa.gov  Thu Mar 16 16:14:51 2000
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Date: Thu, 16 Mar 2000 21:17:33 +0100
To: karn@qualcomm.com
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: Initial text on QoS
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catching up on old mails ....

Yes, I think the text you have written, Phil, is *really* a step in the 
right direction.

Basically, there are two different link layer design philosophies (this has 
been mentioned before on this list):
1. "one size fits all", i.e., one IP packet is just as good as the next.
2. something what we call "flow adaptive", i.e. distinguishing flow types 
at the link layer for the purpose of adapting link layer error control 
(ARQ, FEC, power control, spreading factor control, ...) to try to best 
meet a flow's application layer QoS requirements.

There are many benefits from the latter: (A) higher e2e performance, (B) 
less load on the Internet because link layer error recovery saves e2e 
retransmissions (C) more efficient utilization of radio resources like 
transmission power (battery), etc., etc.

For flow adaptive links to work, it would be soooo great to have the old 
TOS byte and its definitions - deployed. Unfortunaly this is not the case. 
Moreover, as already noted by Lloyd, diffserv has redefined the TOS byte 
and unfortunately does not allow the required e2e marking of, e.g., 
reliable (TCP, NFS, Reliable-Multicast, ...) vs. delay-senstive (RTP, ...) 
flows.

Maybe PILC can at some point request such CPs when diffserv work goes into 
a new round ...

However, I don't know whether such required e2e markings (that don't change 
once set by the source) are at all compatible with the diffserv model which 
allows that an IP packet's CP can change from domain to domain :-(

///Reiner

At 21:11 10.03.00 , Phil Karn wrote:
>I've written this text to kick off the section on QoS, fairness vs 
>performance,
>and congestion signalling. --Phil
>
>The IP header includes a 1-byte TOS (type of service) field. This
>field is broken into two parts: a 3-bit precedence field that
>indicates the relative importance of this packet compared to others,
>and a 4-bit "qualitative tradeoff" field to indicate if low delay,
>high throughput, high reliability or low cost is qualitatively more
>important for this particular packet. One TOS bit is reserved.
>
>Although the TOS field has not been heavily used in the past, it is
>becoming increasingly important to the provision of Quality of Service
>(QoS) features necessary for services such as Voice over IP (VoIP).
>
>Many subnetwork designers are faced with inherent tradeoffs between
>delay, throughput, reliability and cost. When this is the case, the
>subnetwork should provide an interface to IP that allows the IP TOS
>bits to control these tradeoffs on a per-packet basis. For example,
>when a subnetwork implements a hop-by-hop retransmission scheme to
>improve reliability, this invariably comes at the cost of increased
>delay variance. An IP packet presented to this subnetwork with the
>"low delay" bit set in its TOS field should disable hop-by-hop
>retransmission within the subnetwork for that particular
>packet. Another example is link forward error correction, which can
>increase reliability at the expense of throughput without
>significantly affecting delay.



From owner-pilc@lerc.nasa.gov  Thu Mar 16 16:49:21 2000
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In-reply-to: Your message of "Fri, 10 Mar 2000 12:11:43 EST."
             <200003102011.MAA18720@homer.ka9q.ampr.org> 
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Actually, I had some rather different things in mind for this section.  Sorry 
to be so late with this, I've been thinking about it for a while, and was 
having difficulty organizing my thoughts.  There's more to be said, but enough 
delay (so to speak! :-)

Dan

-----------------------------
The Internet has historically provided a single, undifferentiated best effort 
service. As applications have emerged that have specific quality-of-service 
(QoS) requirements, mechanisms have been, and continue to be developed to 
support them.

QoS, in the broadest sense, is defined as the set of service attributes that 
the provider of a service offers to the user of a service.  These service 
attributes can be categorized as relating to  performance, availability, 
security or cost.   However, this memo, like most treatments of QoS, considers 
only the set of performance attributes.  In the context of the IP and subnet 
layers, the performance attributes of most interest are:
  -- throughput
  -- delay
  -- delay jitter
  -- packet loss ratio
  -- errored packet rate
This memo does not attempt to define these.

In a layered architecture, a layer enhances and/or degrades the service 
provided to it by the layer below, so as to provide a service to the layer 
above.  Layers (especially the IP layer and below) contain impairments which 
degrade performance.  Examples of impairments include variable delays due to 
queueing and scheduling, congestive discard and errored discard.  Mechanisms 
in the layers can enhance one service attribute possibly at the expense of 
another.  For example, a retransmission mechanism in a subnet enhances 
(decreases) the packet loss ratio offered to the IP layer, at the expense of 
degrading (increasing) delay variation.  IP, however, does not contain such 
mechanisms, since strict application of the end-to-end principle relegates 
such functions to the TCP layer and above.

The Internet has historically provided an undifferentiated best effort service 
with no quantifiable service attributes. As applications have emerged that 
have specific performance requirements, mechanisms have been, and continue to 
be developed to support them.  
.in +5
-- IP Integrated Services (Intserv) [RFC1633]  provides finely granular QoS 
guarantees to individual flows.  It uses stateful mechanisms:  a 'soft' state 
is installed in routers along the path using a Reservation Protocol (RSVP) 
[RFC2750], which includes bandwidth reservations.  Reservations are admitted 
or not based on whether the requested resources are available.  State is 
'soft' because its lifetime is limited, and thus must be refreshed 
periodically.  Two integrated services are presently defined for Intserv.  A 
Guaranteed Service (GS) [RFC2212] offers firm bounds on maximum end-to-end 
queueing delay (but not delay jitter) for flows that meet an agreed traffic 
specification (TSpec).  A Controlled Load Service (CLS) offers no quantitative 
guarantees, but promises packet loss and delay closely
approximating the QoS that same flow would receive from an unloaded network.  

The mechanisms used by Intserv, especially RSVP, have been found to have 
scaling problems with respect to wide deployment in the Internet [RFC2208]

-- IP Differentiated Services (DiffServ) [RFC2475] is intended to provides 
coarse-grained QoS guarantees to aggregated flows.  Scalability is expected to 
be realized by confining  stateful mechanisms to the borders of a DiffServ 
domain.  These border nodes mark (or enforce marking) of packets with a 
DiffServ codepoint (DSCP -- the 6 most significant bits of the former TOS byte 
or  Traffic Class byte of the IPv4 or IPv6 datagram, respectively), which 
selects a defined per-hop behavior (PHB).  That PHB defines the forwarding 
behavior which the packet will experience in each node of the DiffServ domain. 
  A collection of packets with the same DSCP crossing a link in a particular 
direction is called a Behavior Aggregate (BA).  Resources are allocated to 
behavior aggregates. 

No per-flow or per-BA signalling protocols are currently defined.  DiffServ 
defines PHBs and not services.  However, PHBs can be used by service providers 
in combination with other mechanisms (such as markers, shapers, policers and 
schedulers) to provide services.

Three PHB groups are defined.  The legacy class selector PHB group [RFC2474] 
provides a relative priority;  this was provided to grandfather legacy use of 
the IP precedence field in the TOS byte of the IPv4 datagram.  The Expedited 
Forwarding (EF) PHB [RFC2598]  requires that the departure rate of a BA's 
packets from any diffserv node must equal or exceed a configurable rate.  The 
EF PHB can be used to build low loss, low latency, low jitter, assured 
bandwidth, end-to-end services through DS domains. The Assured Forwarding (AF) 
PHB group [RFC2597] provides four independently forwarded AF classes.  Within 
each AF class, a packet can be assigned one of three different levels of drop 
precedence.  A DS node does not reorder IP packets of the same microflow if 
they belong to the same AF class.  The AF PHB can be used, for example, to 
provide a so-called Olympic service, which has different levels (gold, silver, 
bronze), offering relatively better or worse delay and loss probability.

Note that services constructed using DiffServ can offer QoS guarantees only on 
BAs, and not on individual microflows.

.in -5

In general, the tradeoff between granularity and scaling tends to favor use of 
Intserv in edge domains (i.e., those most involved in serving hosts) and 
Diffserv in core domains.

Subnet designers should consider whether and how their subnets can support 
Intserv or Diffserv.  The following points should be addressed:
-- If the subnet is designed to provide a constant rate service with no 
contention for subnet resources (e.g., pure TDM/TDMA), IntServ or DiffServ 
mechanisms can be used directly. 
-- Otherwise, means need to be provided to differentiate flows, aggregated 
flows or connections at the subnet layer.  These need to be treated 
independently for purposes of discarding, queueing  and scheduling.  This can 
be done using stateful mechanisms (such as labels or connection identifiers), 
stateless mechanisms analogous to DSCPs or n-tuples within the subnet packet 
header (e.g., source and destination address and SAP).  One or more flows at 
the IP layer can be associated with (or carried over) a flow or connection at 
the subnet layer.
-- Mappings need to be made between the service elements supported by the 
subnet and the Intserv services or Diffserv [service?  BA? PHB? -- things have 
gotten confusing in Diffserv, and maybe we can fill this in after Adelaide]. 
-- As part of those mappings, the subnet needs to be able to offer absolute 
guarantees and possibly relative assurances to the IP layer.  Absolute 
priority mechanisms have been atrractive to subnet designers as they are easy 
to implement.  However, they are not sufficient to provide absolute 
guarantees.  Similarly, in order to support Diffserv class selectors, subnets 
(i.e., ATM) which have offered only absolute guarantees have had to be 
rearchitected to also support relative assurances.
-- As noted elsewhere in this memo, mobility can strongly affect the kinds of 
guarantees that can be offered, and the confidence with which such guarantees 
can be offered.
-- Different service objectives can result in different mechanisms or local 
paramters associated with those objectives.  As an example, if the user of an 
Intserv service is highly intolerant of fixed delay, but more tolerant of 
packet loss, the subnet service might elect not to use FEC, or to use a less 
deep (and thus less robust) interleaver.
-- Considerations on queueing, discard and scheduling discussed elsewhere in 
this memo also affect the QoS offered by the subnet.
 



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This draft is a work item of the Performance Implications of Link Characteristics Working Group of the IETF.

	Title		: TCP Performance Implications of Network Asymmetry
	Author(s)	: H. Balakrishnan, V. Padmanabhan
	Filename	: draft-ietf-pilc-asym-01.txt
	Pages		: 15
	Date		: 15-Mar-00
	
This document describes TCP performance problems that arise because 
of asymmetric effects. These problems arise in several access 
networks, including bandwidth-asymmetric networks and packet radio 
networks, for different underlying reasons. However, the net effect 
on TCP performance is the same in both cases: performance degrades 
significantly because of imperfection and variability in the ACK 
feedback from the receiver to the sender. This document details 
several solutions to these problems, which use a combination of 
local link-layer techniques and end-to-end mechanisms. Solutions to 
the problem of asymmetry are two-pronged: (i) techniques to manage 
the reverse channel used by ACKs, typically using header compression 
or reducing the frequency of TCP ACKs, and (ii) techniques to handle 
this reduced ACK frequency to retain the TCP sender's 
acknowledgment-triggered self-clocking.

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This draft is a work item of the Performance Implications of Link Characteristics Working Group of the IETF.

	Title		: Performance Enhancing Proxies
	Author(s)	: J. Border, G. Montenegro, J. Griner,  M. Kojo
	Filename	: draft-ietf-pilc-pep-02.txt
	Pages		: 39
	Date		: 15-Mar-00
	
This document provides a high level overview of Performance Enhancing
Proxies. Different types of Performance Enhancing Proxies are
described as well as the mechanisms used to improve performance. In
addition, motivations for their development and use are described
along with some the consequences of using them, especially in the
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Performance Implications of Link Characteristics Working Group of the IETF.

	Title		: End-to-end Performance Implications of Slow Links
	Author(s)	: S. Dawkins, G. Montenegro,  V. Magret, M. Kojo
	Filename	: draft-ietf-pilc-slow-03.txt
	Pages		: 14
	Date		: 15-Mar-00
	
This document makes performance-related recommendations for users of
network paths that traverse 'very low bit-rate' links.

'Very low bit-rate' implies 'slower than we would like'. This
recommendation may be useful in any network where hosts can saturate
available bandwidth, but the design space for this recommendation
explicitly includes connections that traverse 56 Kb/second modem
links or 4.8 Kb/second wireless access links - both of which are
widely deployed.

This document discusses general-purpose mechanisms. Where
application-specific mechanisms can outperform the relevant
general-purpose mechanism, we point this out and explain why.

This document has some recommendations in common with RFC 2689,
'Providing integrated services over low-bitrate links', especially
in areas like header compression. This document focuses more on
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This draft is a work item of the Performance Implications of Link Characteristics Working Group of the IETF.

	Title		: End-to-end Performance Implications of Links with 
                          Errors
	Author(s)	: S. Dawkins, G. Montenegro, V. Magret, M. Kojo,
                          N. Vaidya
	Filename	: draft-ietf-pilc-error-03.txt
	Pages		: 17
	Date		: 15-Mar-00
	
The rapidly-growing World Wide Web is being accessed by an
increasingly wide range of devices over an increasingly wide
variety of links. At least some of these links do not provide the
reliability that hosts expect, and this expansion into unreliable
links causes some Internet protocols, especially TCP [RFC793], to
perform poorly.

Specifically, TCP congestion avoidance procedures [RFC2561], while
appropriate for connections that lose traffic primarily because of
congestion and buffer exhaustion, interact badly with connections
that traverse links with high uncorrected error rates. The result is
that senders may spend an excessive amount of time waiting on
acknowledgements that aren't coming, whether these losses are due to
data losses in the forward path or acknowledgement losses in the
return path, and then, although these losses are not due to
congestion-related buffer exhaustion, the sending TCP then transmits
at substantially reduced traffic levels as it probes the network to
determine

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Organization: Motorola Labs
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Dan Grossman wrote:

> Actually, I had some rather different things in mind for this section.  Sorry
> to be so late with this, I've been thinking about it for a while, and was
> having difficulty organizing my thoughts.  There's more to be said, but enough
> delay (so to speak! :-)
>
> Dan
>
> -----------------------------
> The Internet has historically provided a single, undifferentiated best effort
> service. As applications have emerged that have specific quality-of-service
> (QoS) requirements, mechanisms have been, and continue to be developed to
> support them.
>
> QoS, in the broadest sense, is defined as the set of service attributes that
> the provider of a service offers to the user of a service.  These service
> attributes can be categorized as relating to  performance, availability,
> security or cost.   However, this memo, like most treatments of QoS, considers
> only the set of performance attributes.  In the context of the IP and subnet
> layers, the performance attributes of most interest are:
>   -- throughput
>   -- delay
>   -- delay jitter
>   -- packet loss ratio
>   -- errored packet rate
> This memo does not attempt to define these.
>
> In a layered architecture, a layer enhances and/or degrades the service
> provided to it by the layer below, so as to provide a service to the layer
> above.  Layers (especially the IP layer and below) contain impairments which
> degrade performance.  Examples of impairments include variable delays due to
> queueing and scheduling, congestive discard and errored discard.  Mechanisms
> in the layers can enhance one service attribute possibly at the expense of
> another.  For example, a retransmission mechanism in a subnet enhances
> (decreases) the packet loss ratio offered to the IP layer, at the expense of
> degrading (increasing) delay variation.  IP, however, does not contain such
> mechanisms, since strict application of the end-to-end principle relegates
> such functions to the TCP layer and above.
>
> The Internet has historically provided an undifferentiated best effort service
> with no quantifiable service attributes. As applications have emerged that
> have specific performance requirements, mechanisms have been, and continue to
> be developed to support them.
> .in +5
> -- IP Integrated Services (Intserv) [RFC1633]  provides finely granular QoS
> guarantees to individual flows.  It uses stateful mechanisms:  a 'soft' state
> is installed in routers along the path using a Reservation Protocol (RSVP)
> [RFC2750], which includes bandwidth reservations.  Reservations are admitted
> or not based on whether the requested resources are available.  State is
> 'soft' because its lifetime is limited, and thus must be refreshed
> periodically.  Two integrated services are presently defined for Intserv.  A
> Guaranteed Service (GS) [RFC2212] offers firm bounds on maximum end-to-end
> queueing delay (but not delay jitter) for flows that meet an agreed traffic
> specification (TSpec).  A Controlled Load Service (CLS) offers no quantitative
> guarantees, but promises packet loss and delay closely
> approximating the QoS that same flow would receive from an unloaded network.
>
> The mechanisms used by Intserv, especially RSVP, have been found to have
> scaling problems with respect to wide deployment in the Internet [RFC2208]
>
> -- IP Differentiated Services (DiffServ) [RFC2475] is intended to provides
> coarse-grained QoS guarantees to aggregated flows.  Scalability is expected to
> be realized by confining  stateful mechanisms to the borders of a DiffServ
> domain.  These border nodes mark (or enforce marking) of packets with a
> DiffServ codepoint (DSCP -- the 6 most significant bits of the former TOS byte
> or  Traffic Class byte of the IPv4 or IPv6 datagram, respectively), which
> selects a defined per-hop behavior (PHB).  That PHB defines the forwarding
> behavior which the packet will experience in each node of the DiffServ domain.
>   A collection of packets with the same DSCP crossing a link in a particular
> direction is called a Behavior Aggregate (BA).  Resources are allocated to
> behavior aggregates.
>
> No per-flow or per-BA signalling protocols are currently defined.  DiffServ
> defines PHBs and not services.  However, PHBs can be used by service providers
> in combination with other mechanisms (such as markers, shapers, policers and
> schedulers) to provide services.
>
> Three PHB groups are defined.  The legacy class selector PHB group [RFC2474]
> provides a relative priority;  this was provided to grandfather legacy use of
> the IP precedence field in the TOS byte of the IPv4 datagram.  The Expedited
> Forwarding (EF) PHB [RFC2598]  requires that the departure rate of a BA's
> packets from any diffserv node must equal or exceed a configurable rate.  The
> EF PHB can be used to build low loss, low latency, low jitter, assured
> bandwidth, end-to-end services through DS domains. The Assured Forwarding (AF)
> PHB group [RFC2597] provides four independently forwarded AF classes.  Within
> each AF class, a packet can be assigned one of three different levels of drop
> precedence.  A DS node does not reorder IP packets of the same microflow if
> they belong to the same AF class.  The AF PHB can be used, for example, to
> provide a so-called Olympic service, which has different levels (gold, silver,
> bronze), offering relatively better or worse delay and loss probability.
>
> Note that services constructed using DiffServ can offer QoS guarantees only on
> BAs, and not on individual microflows.

-- Core stateless approaches (Dynamic Packet
State<draft-stoica-diffserv-dps-00.txt> ) : Recently there
have been proposals  for supporting per flow commitments to flows using scalable
cores. These approaches may
potentially mature into standards in due course of time, given the richness of
the services they can support.  These
architectures  require edge routers to maintain per-flow state, while the core
routers maintain no per-flow state. In the
data path, edges mark packets of a flow with  the minimum necessary  state
information required by a distributed algorithm implemented
by an island of core routers. The core routers do not have to perform any packet
classification, and update the state
information on the packet header to achieve the committed end to end performance.
They also propose admisison control
algorithms that require no per flow state to support guaranteed services. Such
schemes offer the advantage of supporting per flow
commitments, using a scalable network architecture.

> .in -5
>
> In general, the tradeoff between granularity and scaling tends to favor use of
> Intserv in edge domains (i.e., those most involved in serving hosts) and
> Diffserv in core domains.

(or alternately, core stateless approaches end to end)

>
>
> Subnet designers should consider whether and how their subnets can support
> Intserv or Diffserv.  The following points should be addressed:
> -- If the subnet is designed to provide a constant rate service with no
> contention for subnet resources (e.g., pure TDM/TDMA), IntServ or DiffServ
> mechanisms can be used directly.
> -- Otherwise, means need to be provided to differentiate flows, aggregated
> flows or connections at the subnet layer.  These need to be treated
> independently for purposes of discarding, queueing  and scheduling.  This can
> be done using stateful mechanisms (such as labels or connection identifiers),
> stateless mechanisms analogous to DSCPs or n-tuples within the subnet packet
> header (e.g., source and destination address and SAP).  One or more flows at
> the IP layer can be associated with (or carried over) a flow or connection at
> the subnet layer.
> -- Mappings need to be made between the service elements supported by the
> subnet and the Intserv services or Diffserv [service?  BA? PHB? -- things have
> gotten confusing in Diffserv, and maybe we can fill this in after Adelaide].
> -- As part of those mappings, the subnet needs to be able to offer absolute
> guarantees and possibly relative assurances to the IP layer.  Absolute
> priority mechanisms have been atrractive to subnet designers as they are easy
> to implement.  However, they are not sufficient to provide absolute
> guarantees.  Similarly, in order to support Diffserv class selectors, subnets
> (i.e., ATM) which have offered only absolute guarantees have had to be
> rearchitected to also support relative assurances.
> -- As noted elsewhere in this memo, mobility can strongly affect the kinds of
> guarantees that can be offered, and the confidence with which such guarantees
> can be offered.
> -- Different service objectives can result in different mechanisms or local
> paramters associated with those objectives.  As an example, if the user of an
> Intserv service is highly intolerant of fixed delay, but more tolerant of
> packet loss, the subnet service might elect not to use FEC, or to use a less
> deep (and thus less robust) interleaver.
> -- Considerations on queueing, discard and scheduling discussed elsewhere in
> this memo also affect the QoS offered by the subnet.
>

-Jayanth Mysore,
Motorola Labs



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Performance Implications of Link Characteristics Working Group of the IETF.

	Title		: Advice for Internet Subnetwork Designers
	Author(s)	: P. Karn, J. Touch, J. Mahdavi, D. Grossman, 
                          G. Montenegro, A. Falk, M. Montpetit, G. Fairhurst
	Filename	: draft-ietf-pilc-link-design-02.txt
	Pages		: 21
	Date		: 16-Mar-00
	
This document provides advice to the designers of digital
communication equipment, link layer protocols and packet switched
subnetworks (collectively referred to as subnetworks) who wish to
support the Internet protocols but who may be unfamiliar with the
architecture of the Internet and the implications of their design
choices on the performance and efficiency of the Internet.
This document represents an evolving consensus of the members of the
IETF Performance Implications of Link Characteristics (PILC) working
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This draft is a work item of the Performance Implications of Link Characteristics Working Group of the IETF.

	Title		: TCP Performance Implications of Network Asymmetry
	Author(s)	: H. Balakrishnan, V. Padmanabhan
	Filename	: draft-ietf-pilc-asym-01.txt
	Pages		: 15
	Date		: 15-Mar-00
	
This document describes TCP performance problems that arise because 
of asymmetric effects. These problems arise in several access 
networks, including bandwidth-asymmetric networks and packet radio 
networks, for different underlying reasons. However, the net effect 
on TCP performance is the same in both cases: performance degrades 
significantly because of imperfection and variability in the ACK 
feedback from the receiver to the sender. This document details 
several solutions to these problems, which use a combination of 
local link-layer techniques and end-to-end mechanisms. Solutions to 
the problem of asymmetry are two-pronged: (i) techniques to manage 
the reverse channel used by ACKs, typically using header compression 
or reducing the frequency of TCP ACKs, and (ii) techniques to handle 
this reduced ACK frequency to retain the TCP sender's 
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From owner-pilc@lerc.nasa.gov  Fri Mar 17 10:45:06 2000
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From: "Falk, Aaron" <afalk@PanAmSat.com>
To: "'pilc@grc.nasa.gov'" <pilc@grc.nasa.gov>
Cc: "Spencer Dawkins (E-mail)" <spencer_dawkins@yahoo.com>,
        "'Gabriel Montenegro'" <gab@sun.com>,
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	 <vincent.magret@aud.alcatel.com>,
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Subject: Working group last call for ERROR and SLOW
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draft-ietf-pilc-error-03.txt and draft-ietf-pilc-slow-03.txt are both pretty
mature and may be ready for submission to the IESG as Best Current Practice
RFPs. Let's kick off a working group last call to end around March 31st. If
you have any comments on the drafts, now is the time to get them in.

They are available at:

http://www.ietf.org/internet-drafts/draft-ietf-pilc-error-03.txt

http://www.ietf.org/internet-drafts/draft-ietf-pilc-slow-03.txt

--aaron

---
Aaron Falk
Director - Internet Services Development
PanAmSat Corporation
Greenwich, Connecticut, USA
*	Phone: 	203.861.8326
*	Mobile: 	203.912.5689
*	Fax: 	203.861.8677
*	mailto:afalk@panamsat.com 
*	http://www.panamsat.com



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Peformance Implications of Link Characteristics Working Group
AGENDA

TUESDAY, March 28, 2000
1415-1515  Afternoon Sessions II
Adelaide Convention Centre in Adelaide, Australia
Room 4&5  

5 minutes:	Agenda-bashing
5 minutes: Working Group Status
5 minutes: draft-ietf-pilc-error-03.txt
5 minutes: draft-ietf-pilc-slow-03.txt
10 minutes: draft-ietf-pilc-asym-01.txt
10 minutes:	draft-ietf-pilc-link-design-02.txt
10 minutes:	draft-ietf-pilc-pep-02.txt

For information on IETF-47 see http://www.ietf.org/meetings/IETF-47.html

For copies of drafts see http://www.ietf.org/ids.by.wg/pilc.html


From owner-pilc@lerc.nasa.gov  Fri Mar 17 19:29:43 2000
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Date: Fri, 17 Mar 2000 15:23:33 -0800
From: Joe Touch <touch@ISI.EDU>
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Subject: Announcing X-Bone VPN/overlay software release
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The following may be of interest to PILC - to setup IP testbeds...

-- Joe
-----------


   xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx    
   x                                                          x    
   x           XX    XX      X-BONE Overlay System            x    
   x             X  X                                         x    
   x              XX                Software Release          x    
   x             X  X                                         x    
   x           XX    XX             March 2000                x    
   x                                                          x    
   xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx    
                                                                   
                                                                   
The X-Bone system for automated deployment of VPN / overlay networks
is now publicly available. This is the first major public release, v1.2.
                                                                   
X-Bone dynamically deploys and manages Internet overlays to reduce
configuration effort and increase network component sharing. X-Bone
discovers, configures, and monitors network resources to create
overlays over existing IP networks.
                                                                   
The X-Bone is implemented in Perl, and open source is provided.
                                                                   
The X-Bone can be used for:                                        
                                                                   
      - deploying VPNs                                             
                                                                   
      - sharing lab or wide-area networks                          
         for multiple, concurrent projects                         
         for testing protocols and apps on new topologies          
                                                                   
X-Bone uses two-layer IP in IP tunneled overlays and supports existing
applications and unmodified routing, multicast, and DNS services.
X-Bone also support IPSec within overlays. Applications can use the
X-Bone without modification or recompilation.
                                                                   
The X-Bone is available for the following operating systems:       
                                                                   
      - FreeBSD                                                    
              CAIRN 2.5, 3.*, 3.* + KAME IPsec patches             
                                                                   
      - Linux RedHat                                               
              6.0, 6.0 + NIST Cerberus IPsec patches, 6.1          
                                                                   
The FreeBSD port and Linux RPM have been submitted to the FreeBSD
ports and RedHat Linux RPMs sites; further information and details and
the port and RPM files are currently available at:
                                                                   
      http://www.isi.edu/xbone/                                    
                                                                   
- Joe Touch                                                        
  Project Leader, X-Bone group, USC/ISI                            
  touch@isi.edu                                                    
  http://www.isi.edu/touch                                         
  +1 (310) 448-9151                                                
                                                                   
-------------------------------------------------------------------
 Effort sponsored by the Defense Advanced Research Projects Agency 
 (DARPA)and Air Force Research Laboratory, Air Force Materiel      
 Command, USAF, under agreement number F30602-98-1-0200.           
                                                                   
 Copyright (c) 1998-2000 by the University of Southern California. 
 All rights reserved.                                              
-------------------------------------------------------------------


From owner-pilc@lerc.nasa.gov  Mon Mar 20 18:50:02 2000
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From: "Falk, Aaron" <afalk@PanAmSat.com>
To: "'pilc@grc.nasa.gov'" <pilc@grc.nasa.gov>
Cc: "Spencer Dawkins (E-mail)" <spencer_dawkins@yahoo.com>,
        "'Gabriel Montenegro'" <gab@sun.com>,
        "'vincent.magret@aud.alcatel.com'"
	 <vincent.magret@aud.alcatel.com>,
        "Markku Kojo (E-mail)"
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Subject: Extended: Working group last call for ERROR and SLOW
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Let's extend the last call a week, to 4/7, to make things easier on those
who will be traveling.

--aaron

> 
> draft-ietf-pilc-error-03.txt and draft-ietf-pilc-slow-03.txt 
> are both pretty
> mature and may be ready for submission to the IESG as Best 
> Current Practice
> RFPs. Let's kick off a working group last call to end around 
> March 31st. If
> you have any comments on the drafts, now is the time to get them in.
> 
> They are available at:
> 
> http://www.ietf.org/internet-drafts/draft-ietf-pilc-error-03.txt
> 
> http://www.ietf.org/internet-drafts/draft-ietf-pilc-slow-03.txt
> 
> --aaron
> 


From owner-pilc@lerc.nasa.gov  Thu Mar 30 19:21:25 2000
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To: Xavi <xporrata@menta.net>
cc: pilc@grc.nasa.gov
Subject: Re: Snoop mailing list 
In-reply-to: Your message of "Fri, 31 Mar 2000 01:13:52 +0200"
             <38E3DFB0.5A916A6C@menta.net> 
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Xavi,

As far as I know, there isn't one.

Hari

On Fri, 31 Mar 2000 01:13:52 +0200, you wrote:

> Hi there,
> 
> Excuse me for the slightly off-topic question, I'd be very grateful if
> anybody could tell me if there's a mailing-list or newsgroup devoted to
> the Snoop protocol. Has there been one ever? Any other and perhaps more
> appropriate forum where to ask?
> Thanks in advance,
> 
> Xavi De Porrata
> Dept. de Matemātica Aplicada i Telemātica
> Universitat Politčcnica de Catalunya
> 




From owner-pilc@lerc.nasa.gov  Thu Mar 30 19:21:32 2000
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Date: Fri, 31 Mar 2000 01:13:52 +0200
From: Xavi <xporrata@menta.net>
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Hi there,

Excuse me for the slightly off-topic question, I'd be very grateful if
anybody could tell me if there's a mailing-list or newsgroup devoted to
the Snoop protocol. Has there been one ever? Any other and perhaps more
appropriate forum where to ask?
Thanks in advance,

Xavi De Porrata
Dept. de Matemātica Aplicada i Telemātica
Universitat Politčcnica de Catalunya



From owner-pilc@lerc.nasa.gov  Fri Mar 31 04:45:34 2000
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Date: Fri, 31 Mar 2000 10:34:41 +0200
From: Miguel Sanchez <misan@ieee.org>
Organization: Universidad Politecnica de Valencia
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To: Xavi <xporrata@menta.net>
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--------------80A2044C3640BF75A845B33E
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You can find useful information and publications in here:

http://www.cs.berkeley.edu/~hari/papers/snoop.html

Best,

Miguel

Xavi wrote:

> Hi there,
>
> Excuse me for the slightly off-topic question, I'd be very grateful if
> anybody could tell me if there's a mailing-list or newsgroup devoted to
> the Snoop protocol. Has there been one ever? Any other and perhaps more
> appropriate forum where to ask?
> Thanks in advance,
>
> Xavi De Porrata
> Dept. de Matem=E0tica Aplicada i Telem=E0tica
> Universitat Polit=E8cnica de Catalunya

--
Miguel Sanchez Lopez                            |       misan@ieee.org
Universidad Politecnica de Valencia             | voice:+3496 387 9700
Camino de Vera, 14                              | fax:  +3496 387 7579
46071 VALENCIA (Spain)                          |



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
You can find useful information and publications in here:
<p><a href="http://www.cs.berkeley.edu/~hari/papers/snoop.html">http://www.cs.berkeley.edu/~hari/papers/snoop.html</a>
<p>Best,
<p>Miguel
<p>Xavi wrote:
<blockquote TYPE=CITE>Hi there,
<p>Excuse me for the slightly off-topic question, I'd be very grateful
if
<br>anybody could tell me if there's a mailing-list or newsgroup devoted
to
<br>the Snoop protocol. Has there been one ever? Any other and perhaps
more
<br>appropriate forum where to ask?
<br>Thanks in advance,
<p>Xavi De Porrata
<br>Dept. de Matem&agrave;tica Aplicada i Telem&agrave;tica
<br>Universitat Polit&egrave;cnica de Catalunya</blockquote>

<pre>--&nbsp;
Miguel Sanchez Lopez&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; misan@ieee.org
Universidad Politecnica de Valencia&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | voice:+3496 387 9700&nbsp;&nbsp;
Camino de Vera, 14&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | fax:&nbsp; +3496 387 7579&nbsp;
46071 VALENCIA (Spain)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</pre>
&nbsp;</html>

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