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Reiner Ludwig wrote:

> At 13:42 18.09.00 -0500, Vincent Magret wrote:
> >Regarding the discussion on the LL-TCP, the draft specifies that it is
> >recommended to use  link layer information when possible. This can only work
> >for the mobile node and the supervisor host. For the correspondent or server,
> >how can this node have access to link layer, since it is not connected or
> >linked to the wireless network?
>
> No entity above the link layer needs to have access to link layer
> information to solve the problem of link outages. Please, check the
> reference to my thesis that I had sent in one of my last mails.

This is one point of view. If no other entity has notion of the link outage, what
are the possible situation:
1. the source entity will continue to behave as is the link was still up (which is
a waste of resources)
2. the link layer will stored packets (hard state, does not allows the mobile node
to got via another entity)
3. the link layer will droppred the packets (pretty much useless)

>
>
> >There is indeed a need of having a message sent from the edge node (i.e. the
> >supervisor node) to the fixed node to inform this latter about the new status
> >of the link.
>
> I don't think so.

Again, it is a point of view.

>

>
>
> >I don't like the idea of having packet store in the supervisor node
>
> I wrote that packets are stored at the link layer sender.

Well that a pure hard-state solution, you don't allow a lot of mobility.

>
> >because it
> >is somehow an hard state situation. the mobile node can not "resume" its
> >connection with another supervisor node.
>
> I don't understand your notion of a supervisor node nor do I think that
> such a node is needed, assuming you mean that this node sits above the link
> layer. All that is needed is a plain old link layer protocol which of
> course is hard state.
>
> ///Reiner

Vincent.



From owner-pilc@lerc.nasa.gov  Thu Oct  5 17:20:24 2000
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From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>
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Reiner Ludwig wrote:

> At 14:01 21.09.00 +0100, Dr G Fairhurst wrote:
> > > ***However***, you have to ensure that the link layer does this _only_ for
> > > the reliable flows such as the TCP-based flows.
> >
> >Well, here I think you need to be much more specific.  What you are proposing
> >may well be of merit where you are able to  differentiate individual flows
> >AND reliably indicate which are TCP and which are UDP. It also implies
> >a sufficiently large link layer retransmission window, so that
> >window does not stall during the time to recover the worst case
> >loss pattern (which could be a challenge on long-delay links (e.g.
> >100 Mbps satellite circuit).
>
> Sure, a link layer retransmission window should be well designed for its
> environment of operation but when it stalls, it stalls. So what? That's not
> any different from a link outage caused by some other reason.

Who said OUTAGE?

There are links out there which do not suffer complete outages for prolonged
periods of tens of seconds (or more), but which do exhibit high rates of
loss: either through bursts of errors or some other error distribution.

Reliable link ARQ with full persistence usually implies you work on retransmitting
the
oldest packet until the link window runs out, or the transmission suceeds.

In traditional link ARQ protocols, this means that you hold other data,
when you are having a rough time forwarding one particular packet.

When the link consists of many TCP flows, this holds all flows. Even those
flows which have sent relatively little data, and are unlikely to be the one
needing the retransmission.

> As Vernon has pointed out in his last mail: if the link layer cannot get
> the packet across the link, TCP won't be able to do that either. So for the
> receiving application it doesn't make a difference whether packets cannot
> be delivered to it because they are held in some link layer or by TCP.

Again, what about the other hosts down-stream which have their packets queued?

> >Recommendations for fully reliable ARQ may also need to consider the impact
> >of alternate network paths which do not pass over the errored wireless link.
>
> We are talking of recommendations for a designer of a specific link layer.
> Why does that require considering the impact of paths that do not pass over
> that very link.

> >[...]
> >While this may be true in these studies, I'd like a clearer understanding of
> >the factors, which influence this before we leap into general conclusions.
> >
> >Many, although I have not examined all of these, refer to cellular
> >phone/wireless
> >links, and not to generic radio links which include satellite, high bandwidth
> >line of sight, very low bit rate, etc.
>
> I forgot to mention the Internet Draft presented by NTT DoCoMo about TCP
> performance over W-CDMA. W-CDMA clearer qualifies as high bandwidth x
> delay. One of their key results (which unfortunately was not on their
> conclusion slide) was that they got the best TCP performance when the link
> layer error recovery persistency was high.

I will be most surprised if this is the case for many TCP flows sharing the link.

> >I think there is evidence that at least some of
> >these
> >links have in the past performed very poorly with reliable ARQ protocols.

> ......
>
> ///Reiner

Gorry Fairhurst

--
------------------------------
http://www.erg.abdn.ac.uk/users/gorry



From owner-pilc@lerc.nasa.gov  Thu Oct  5 19:41:15 2000
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Date: Thu, 5 Oct 2000 15:58:10 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>
> To: Reiner Ludwig <Reiner.Ludwig@ericsson.com>

> When the link consists of many TCP flows, this holds all flows. Even those
> flows which have sent relatively little data, and are unlikely to be the one
> needing the retransmission.
>
> > As Vernon has pointed out in his last mail: if the link layer cannot get
> > the packet across the link, TCP won't be able to do that either. So for the
> > receiving application it doesn't make a difference whether packets cannot
> > be delivered to it because they are held in some link layer or by TCP.
>
> Again, what about the other hosts down-stream which have their packets queued?
> ...

I also tried to point out that ARQ (persistent or otherwise) does not
necessarily implies that all other streams must wait for one stream that
is stuck.  Not even TCP is still limited to go-back-N recovery tactics,
so why would a newfangled radio link layer be so limited?

Isn't it common common wisdom that go-back-N or stop-everything-until-a-
sick-packet-gets-through is ok only when both the bandwidth*delay product
and the probability of error are low?  In other cases, aren't you supposed
to overlap as much as possible, and so have selective ACKs or NACs and
selective retransmissions?


Moreover, by Dr. Fairhurst's assumptions, either
  1. no other packets for any TCP stream are flowing because the link
     is very sick.
  2. this particular packet will soon get through because the problem
     was temporary.
  3. something is very badly broken in the design or implementation of
      the link layer, because one particular packet cannot get through
      but others can.

(#3 is not in the least theoretical to those of us who've fought hardware
and software problems that corrupt some packets but not others.)


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Fri Oct  6 06:32:39 2000
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At 21:34 05.10.00, Dr G Fairhurst wrote:
>Reliable link ARQ with full persistence usually implies you work on 
>retransmitting
>the
>oldest packet until the link window runs out, or the transmission suceeds.
>
>In traditional link ARQ protocols, this means that you hold other data,
>when you are having a rough time forwarding one particular packet.
>
>When the link consists of many TCP flows, this holds all flows. Even those
>flows which have sent relatively little data, and are unlikely to be the one
>needing the retransmission.

You describe a poorly designed link layer ARQ protocol causing the 
head-of-line blocking problem. In this case, the problem is caused only for 
the other *TCP flows* since my argument was that non-TCP flows are treated 
separately.

To avoid that problem, you can allow the link layer to do out-of-order 
delivery of packets belonging to different flows. You should only enforce 
in-order delivery for packets belonging to the same TCP flow to not falsely 
trigger fast retransmit / fast recovery.

Caveat: the link layer must be given the opportunity to distinguish flows.

///Reiner



From owner-pilc@lerc.nasa.gov  Fri Oct  6 08:24:08 2000
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I don't think we differ as much as you may think we do.

Vernon Schryver wrote:

> > From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>
> > To: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
>
> > When the link consists of many TCP flows, this holds all flows. Even those
> > flows which have sent relatively little data, and are unlikely to be the one
> > needing the retransmission.
> >
> > > As Vernon has pointed out in his last mail: if the link layer cannot get
> > > the packet across the link, TCP won't be able to do that either. So for the
> > > receiving application it doesn't make a difference whether packets cannot
> > > be delivered to it because they are held in some link layer or by TCP.
> >
> > Again, what about the other hosts down-stream which have their packets queued?
> > ...
>
> I also tried to point out that ARQ (persistent or otherwise) does not
> necessarily implies that all other streams must wait for one stream that
> is stuck.  Not even TCP is still limited to go-back-N recovery tactics,
> so why would a newfangled radio link layer be so limited?

The Selective Repeat ARQ is definitely recommended (I think you'll find
when we first wrote the words for the draft it does say this already).

Sorry, this debate is less than clear.  I was talking only about WINDOW size,
in many link ARQ schemes there is a fixed modulus or window which
limits the maximum number of packets in flight.  This may be determined by
either amount of buffer memory or size of frame number field (before everyone
shouts and says "but not all" - I know there are exceptions). The HDLC standard
works this way - and so do nearly all variants of it, I have seen.

Out-of-order forwarding is allowed (again in the draft) - although current
recommendations are that this is only allowed when it preserves in-order
delivery within each flow. This still doesn't mean an ARQ protocol can transmit
for ever if it says it is RELIABLE and if there is a packet still unacknowledged
at the low window edge, this can still cause window exhaustion with
a finite numbering space (or available memory).

> Isn't it common common wisdom that go-back-N or stop-everything-until-a-
> sick-packet-gets-through is ok only when both the bandwidth*delay product
> and the probability of error are low?  In other cases, aren't you supposed
> to overlap as much as possible, and so have selective ACKs or NACs and
> selective retransmissions?

Sure you are (the draft says this, I am sure it meant to anyway).

>
> Moreover, by Dr. Fairhurst's assumptions, either
>   1. no other packets for any TCP stream are flowing because the link
>      is very sick.
>   2. this particular packet will soon get through because the problem
>      was temporary.
>   3. something is very badly broken in the design or implementation of
>       the link layer, because one particular packet cannot get through
>       but others can.
>
> (#3 is not in the least theoretical to those of us who've fought hardware
> and software problems that corrupt some packets but not others.)

If I understand you:

1 is what I call an OUTAGE - we have text already which recommends what
to do (hang on to the last few packets if you can and send them once you
are able - not quite the same as recommending persistence!)

2 may happen - but I have seen cases where this can take a long time
- particularly with a big packet and periodic interference.

3. This happens too... and can be bizarre.

The non-persistent argument seems to help with 2,3.  At least it clears out all
that
old link state and any buffers tied up, so that things can resume orderly in the
future.

The persistent argument is equivalent to 1 for an outage.

What I'm trying to find out is what we should recommend when we have a
high (or moderately high) packet loss rate.

If we have only one TCP flow, do we care?
Probably not, as long the ARQ tries (a few attempts) or forever.  The latter may
even be better. Good versions of HDLC could work fine with segmentation.

If we have many flows - are there other issues?
I can think of these two:

1) The ability to classify UDP form TCP seems very desirable (although
difficult with tunnels/encapsulation and fundamentally problematic with IPSEC)

2) I suggest sharing with other TCP flows can become an issue - perhaps less
so if you are end-system connected to a mobile net, but more so if you are
connecting
a remote LAN with many hosts over a wireless (or any RF) link. Several factors
here:
    per-flow state,
    head-of-line block when buffer/window exceeded,
    forwarding delay (out of order delivery may be desirable)

3) Anything else?

> Vernon Schryver    vjs@rhyolite.com

Gorry Fiarhurst.

------------------------------
http://www.erg.abdn.ac.uk/users/gorry



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Date: Fri, 6 Oct 2000 09:32:52 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <200010061532.e96FWqk21100@calcite.rhyolite.com>
To: pilc@grc.nasa.gov
Subject: Re: pilc minutes for IETF 48 - Persistence
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> From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>

> I don't think we differ as much as you may think we do.

I suspect we differ more than either of us thought.


> Sorry, this debate is less than clear.  I was talking only about WINDOW size,
> in many link ARQ schemes there is a fixed modulus or window which
> limits the maximum number of packets in flight.  This may be determined by
> either amount of buffer memory or size of frame number field (before everyone
>shouts and says "but not all" - I know there are exceptions). The HDLC standard
> works this way - and so do nearly all variants of it, I have seen.

Yes, but if you have a whole window's worth of packets that cannot get
through, then the link is very sick.  There's little to talk about besides
radio things like hand-off that should be inivisible to TCP except possibly
for a bunch of lost segments.

If you have one stuck packet that is keeping the window from advancing
and so you cannot get any other data through the link, then either your
window is too small or you have broken hardware or software that cannot
be fixed with ARQ or anything else (i.e. my #3).  There is absolutely
nothing that can or should be done in this case except possibly to change
vendors in a forklift upgrade.


> ...
> > Moreover, by Dr. Fairhurst's assumptions, either
> >   1. no other packets for any TCP stream are flowing because the link
> >      is very sick.
> >   2. this particular packet will soon get through because the problem
> >      was temporary.
> >   3. something is very badly broken in the design or implementation of
> >       the link layer, because one particular packet cannot get through
> >       but others can.

> ...
> 1 is what I call an OUTAGE - we have text already which recommends what
> to do (hang on to the last few packets if you can and send them once you
> are able - not quite the same as recommending persistence!)
>
> 2 may happen - but I have seen cases where this can take a long time
> - particularly with a big packet and periodic interference.
>
> 3. This happens too... and can be bizarre.
>
>The non-persistent argument seems to help with 2,3.  At least it clears out all
> that
> old link state and any buffers tied up, so that things can resume orderly
> in the
> future.

No, in those three cases:
  1. there is nothing to do.  The link is broken and has been broken for
     at least a little while.  Saving packets in the hope that the link
     might return might be good, but is probably a waste of cycles and
     memory, and quite possibly worse.

     You MUST NOT keep any TCP or UDP packet for longer than the MSL.
     You will break things if you somehow keep a packet for a long time
     and then eventually regurgitate it.

  2. this is ordinary bit rot.  If you often find problems with big packets
     and periodic interference, then you have misdesigned the link layer
     packet sizes by making it too big.

  3. there is nothing that can be done in this case except to ship all of
    the hardware and software back to the factory, and possibly to fire
    the poor fellow who messed up the FEC or CRC computation, buffer
    chaining, or whatever.
 

> What I'm trying to find out is what we should recommend when we have a
> high (or moderately high) packet loss rate.
>
> If we have only one TCP flow, do we care?
> Probably not, as long the ARQ tries (a few attempts) or forever.  The
> latter may
> even be better. Good versions of HDLC could work fine with segmentation.
>
> If we have many flows - are there other issues?
> I can think of these two:
>
> 1) The ability to classify UDP form TCP seems very desirable (although
> difficult with tunnels/encapsulation and fundamentally problematic with IPSEC)

It's wise to not attempt things that you know are impossible.  Given the
fact that IPSec exists, the fact that sooner or later people will get
paranoid about their bits, and the fact that people are already trained
to be paranoid about their voices on radios by analog cellular phone
scanners, it is crazy (as in failing to accept reality) to do anything
but assume IPSec makes classifying impossible.

> 2) I suggest sharing with other TCP flows can become an issue - perhaps less
> so if you are end-system connected to a mobile net, but more so if you are
> connecting
> a remote LAN with many hosts over a wireless (or any RF) link. Several factors
> here:
>     per-flow state,
>     head-of-line block when buffer/window exceeded,
>     forwarding delay (out of order delivery may be desirable)

Out of order forwarding is not desirable, but it is always far more
desirable to get the bits through the link than to second guess the upper
layers tolerance to re-ordering.
All IP protocols and applications are absolutely required to tolerate
absolutely all possible re-orderings of IP packets within the MSL.  Any
delaying of later packets merely because earlier packets need link layer
retransmitting would be *WRONG*.  And again, there is the MSL limit
on how long you can delay (including retransmitting) any packet.

When your window or buffer is full, what can you do beside hs head-of-line
block and drop incoming packets?  Mentioning that as an alternative makes
as much sense as mentioning that when the system melts in a fire, it must
stop working.

Per-flow state might be handy, but why would you not start with the sorts
of things that are already being done or considered for other sorts
of routers?  For example, why not do a modified RED?


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Fri Oct  6 21:44:35 2000
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Date: Fri, 6 Oct 2000 18:28:35 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <200010070028.e970SZx11679@calcite.rhyolite.com>
To: pilc@grc.nasa.gov
Subject: Re: pilc minutes for IETF 48
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> From: Reiner Ludwig <Reiner.Ludwig@Ericsson.com>

> ...
> To avoid that problem, you can allow the link layer to do out-of-order 
> delivery of packets belonging to different flows. You should only enforce 
> in-order delivery for packets belonging to the same TCP flow to not falsely 
> trigger fast retransmit / fast recovery.
>
> Caveat: the link layer must be given the opportunity to distinguish flows.

Why go to any trouble to prevent out-of-order delivery?
We are talking only about IP, right?  Not PPP or various bridging protocols?

What is the cost of out-of-order delivery?  As far as I know, IP packet
re-ordering has at most the following costs:

  - broken IP fragmentation code goes crazy
      this is surely not a consideration

  - broken UDP/IP applications go crazy
      they are also surely irrelevant.

  - TCP goes slightly slower, and with sufficient consecutive out-of-order
     segments, can send duplicate ACKs which can cause retransmissions.

Is that last the concern?


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Mon Oct  9 08:47:18 2000
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Date: Mon, 09 Oct 2000 13:30:53 +0200
To: Vernon Schryver <vjs@calcite.rhyolite.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: pilc minutes for IETF 48
Cc: pilc@grc.nasa.gov
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At 02:28 07.10.00, Vernon Schryver wrote:
> > From: Reiner Ludwig <Reiner.Ludwig@Ericsson.com>
>
> > ...
> > To avoid that problem, you can allow the link layer to do out-of-order
> > delivery of packets belonging to different flows. You should only enforce
> > in-order delivery for packets belonging to the same TCP flow to not 
> falsely
> > trigger fast retransmit / fast recovery.
> >
> > Caveat: the link layer must be given the opportunity to distinguish flows.
>
>Why go to any trouble to prevent out-of-order delivery?
>We are talking only about IP, right?  Not PPP or various bridging protocols?
>
>What is the cost of out-of-order delivery?  As far as I know, IP packet
>re-ordering has at most the following costs:
>
>   - broken IP fragmentation code goes crazy
>       this is surely not a consideration
>
>   - broken UDP/IP applications go crazy
>       they are also surely irrelevant.
>
>   - TCP goes slightly slower, and with sufficient consecutive out-of-order
>      segments, can send duplicate ACKs which can cause retransmissions.
>
>Is that last the concern?

Yes, it is.

As soon as the link's capacity goes beyond 4x MTU, any link layer 
retransmission would cause a *spurious* fast retransmit / fast recovery in 
TCP. That assumes that (1) the following 3 packets do not incur a link 
layer retransmission and (2) out-of-order delivery is permitted even within 
a flow.

For some wireless networks this could become a real pain ...

///Reiner



From owner-pilc@lerc.nasa.gov  Mon Oct  9 12:32:01 2000
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Date: Mon, 9 Oct 2000 09:27:17 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <200010091527.e99FRHu01670@calcite.rhyolite.com>
To: pilc@grc.nasa.gov
Subject: Re: pilc minutes for IETF 48
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> From: Reiner Ludwig <Reiner.Ludwig@Ericsson.com>

> ...
> >   - TCP goes slightly slower, and with sufficient consecutive out-of-order
> >      segments, can send duplicate ACKs which can cause retransmissions.
> >
> >Is that last the concern?
>
> Yes, it is.
>
> As soon as the link's capacity goes beyond 4x MTU, any link layer 
> retransmission would cause a *spurious* fast retransmit / fast recovery in 
> TCP. That assumes that (1) the following 3 packets do not incur a link 
> layer retransmission and (2) out-of-order delivery is permitted even within 
> a flow.
>
> For some wireless networks this could become a real pain ...

Has the pain been simulated or measured on real wireless links?

What does it mean to say "link's capacity goes beyond 4x MTU"?  I would
have thought that a link's capacity is measured in bits/second instead of
bits.  What TCP implementation ACK's based on MTU instead of MSS?  What
does the MTU of one link in the path have to do with the MSS?  Do you mean
something about the size of the queues in the radio modem/routers?

In other words, I see the point, but I also wonder if some of the concerns
are as valid as the concerns that lead the WAP people to conclude that
TCP/IP cannot be used on wireless links because we all know that TCP/IP
cannot run on links as slow as 9600 bits/second.  So they've said in the
main IETF list and elsewhere including the trade rags...and never mind
that some of us used TCP over 9600, 2400, and even 1200 bit/sec modems,
and that some of those very fast (for the day) 9600 bit/sec modems had
ARQ and delays that varied between about 120 ms. and 300 ms. depending
on sizes of their queues.  (Telebit Trailblazers with PEP)


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Mon Oct  9 14:02:55 2000
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To: Vernon Schryver <vjs@calcite.rhyolite.com>
Cc: pilc@grc.nasa.gov
Subject: Re: pilc minutes for IETF 48 
In-Reply-To: Message from Vernon Schryver <vjs@calcite.rhyolite.com> 
   of "Fri, 06 Oct 2000 18:28:35 MDT." <200010070028.e970SZx11679@calcite.rhyolite.com> 
Reply-To: sommerfeld@orchard.arlington.ma.us
Date: Mon, 09 Oct 2000 13:04:59 -0400
From: Bill Sommerfeld <sommerfeld@orchard.arlington.ma.us>
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> What is the cost of out-of-order delivery?  As far as I know, IP packet
> re-ordering has at most the following costs:
> 
>   - broken IP fragmentation code goes crazy
>       this is surely not a consideration

related, but slightly different:
	- broken packet filtering code goes crazy when fragments are
	  reordered. 

>   - broken UDP/IP applications go crazy
>       they are also surely irrelevant.
> 
>   - TCP goes slightly slower, and with sufficient consecutive out-of-order
>      segments, can send duplicate ACKs which can cause retransmissions.
> 
> Is that last the concern?
> 
> 
> Vernon Schryver    vjs@rhyolite.com



From owner-pilc@lerc.nasa.gov  Mon Oct  9 23:40:27 2000
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Date: Mon, 9 Oct 2000 11:34:44 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <200010091734.e99HYi004218@calcite.rhyolite.com>
To: pilc@grc.nasa.gov
Subject: Re: pilc minutes for IETF 48
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> From: Bill Sommerfeld <sommerfeld@orchard.arlington.ma.us>

> > What is the cost of out-of-order delivery?  As far as I know, IP packet
> > re-ordering has at most the following costs:
> > 
> >   - broken IP fragmentation code goes crazy
> >       this is surely not a consideration
>
> related, but slightly different:
> 	- broken packet filtering code goes crazy when fragments are
> 	  reordered. 

Yes, but also not a concern, right?


> >   - TCP goes slightly slower, and with sufficient consecutive out-of-order
> >      segments, can send duplicate ACKs which can cause retransmissions.

In thinking about possible responses to my objections to worrying about
that, I've thought of other real life things.  If the worry is that
wireless base stations will have more than 4 MSS's of queue for each TCP
stream (or a common queue for all streams bigger than 4 MSS's), and if
link-layer retransmissions can cause out-of-order delivery and so
fast-recovery/fast-retransmit, then there is an obvious fix that probably
needs doing regardless of wireless ARQ.

4 of today's MSS's is about 6 KBytes, which is a lot more than 0.5
seconds at 14 kbit/sec.  Regardless of wireless ARQ triggering
false fast-retransmit, I wonder if you really want that much queue space.

It is a Bad Thing(tm) for any router to have a queue that is too big.
Too much queue space in the path causes bad things in old fashioned land
lines.  It's important for routers to signal the end points when their
queues are big, and without ECN, that is done by dropping packets.
For an example of the results of that Bad Thing(tm),
 - build a cross country intranet that includes a 56kbit/sec link.
 - configure the routers on the ends of the 56k link to have 60K bytes
     of queue.
 - put hosts on both ends of the path with default TCP windows of at
    least 60KBytes
 - have one or two hosts do bulk transfers (e.g. move 600 MByte CDROM
    imags) while the other hosts do interactive things such as rlogin,
    telnet, DNS, and HTTP.
 - run `ping` to watch latencies for the interactive jobs
You'll find that the interactive jobs experience latencies that vary in
a sawtooth between the speed of light and serialization delays (e.g. < 80
ms) and more than 10 seconds (10,000 ms).  The solution is fix the routers
so that they can tell the bulk transfer hosts to back off, and that is to
reduce the queue sizes at the ends of that 56 kbit/sec link from
10 seconds * 56 Kbit/sec to about 0.5 seconds * 56 kbit/sec.

I've been painfully explicit and used strange values like 56 kbit/sec
because I've "been there and done that" in real life.  It should be easy
to adjust those strange old numbers to match modern wireless links.

Just because DRAM is almost free today does not mean that you should use
GBytes of it for queuing in wireless base stations.  It is important for
wires base stations to signal their congestion, and since you will not
always have ECN to all hosts, you must drop packets, and that means you
must not have too much queing.


Vernon Schryver    vjs@rhyolite.com


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To: Vernon Schryver <vjs@calcite.rhyolite.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: pilc minutes for IETF 48
Cc: pilc@grc.nasa.gov
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At 17:27 09.10.00, Vernon Schryver wrote:
> > As soon as the link's capacity goes beyond 4x MTU, any link layer
> > retransmission would cause a *spurious* fast retransmit / fast recovery in
> > TCP. That assumes that (1) the following 3 packets do not incur a link
> > layer retransmission and (2) out-of-order delivery is permitted even 
> within
> > a flow.
> >
> > For some wireless networks this could become a real pain ...
>
>Has the pain been simulated or measured on real wireless links?

I only know of these two studies (in literature):
1. A Comparison of Mechanisms for Improving TCP Performance over Wireless 
Links. Hari Balakrishnan, Venkat Padmanabhan, Srinivasan Seshan, and Randy 
H. Katz.
2. J.C.R. Bennett, C. Partridge and N. Shectman, Packet Reordering is Not 
Pathological Network Behavior, In IEEE/ACM Transactions on Networking, 
Decem-ber 1999.


>What does it mean to say "link's capacity goes beyond 4x MTU"?  I would
>have thought that a link's capacity is measured in bits/second instead of
>bits.  What TCP implementation ACK's based on MTU instead of MSS?  What
>does the MTU of one link in the path have to do with the MSS?  Do you mean
>something about the size of the queues in the radio modem/routers?

OK, so my wording was a bit loose. With the "link's capacity" I mean the 
number of TCP segments (MSS) that fit into a *link layer RTT*.

///Reiner



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Date: Wed, 11 Oct 2000 15:57:18 +0200
To: Vernon Schryver <vjs@calcite.rhyolite.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: pilc minutes for IETF 48
Cc: pilc@grc.nasa.gov
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At 19:34 09.10.00, Vernon Schryver wrote:
>It is a Bad Thing(tm) for any router to have a queue that is too big.
>Too much queue space in the path causes bad things in old fashioned land
>lines.  It's important for routers to signal the end points when their
>queues are big, and without ECN, that is done by dropping packets.

I fully agree but also think that this is already discussed in the "Advice 
for Internet Subnetwork Designers" ID.


>The solution is fix the routers
>so that they can tell the bulk transfer hosts to back off, and that is to
>reduce the queue sizes at the ends of that 56 kbit/sec link from
>10 seconds * 56 Kbit/sec to about 0.5 seconds * 56 kbit/sec.

If you further restrict this to say that this is the upper limit *per 
flow*, I would fully agree with you again.

The upper limit for the number of bytes allowed for queueing at a link per 
flow before dropping occurs should be "fixed worst-case e2e RTT" x link speed.

You have picked 500 ms for the value of "fixed worst-case RTT". That would 
be fine with me but might be argued ...


>Just because DRAM is almost free today does not mean that you should use
>GBytes of it for queuing in wireless base stations.  It is important for
>wires base stations to signal their congestion, and since you will not
>always have ECN to all hosts, you must drop packets, and that means you
>must not have too much queing.

I think this is already the message of the "Advice for Internet Subnetwork 
Designers" ID. What is still unclear is a rule-of-thumb for the 
dimensioning of the queueing buffer as discussed above.

///Reiner



From owner-pilc@lerc.nasa.gov  Wed Oct 11 11:07:58 2000
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Date: Wed, 11 Oct 2000 08:23:49 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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To: pilc@grc.nasa.gov
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> From: Reiner Ludwig <Reiner.Ludwig@Ericsson.com>

> > > As soon as the link's capacity goes beyond 4x MTU, any link layer
> > > retransmission would cause a *spurious* fast retransmit / fast recovery in
> > > TCP. That assumes that (1) the following 3 packets do not incur a link
> > > layer retransmission and (2) out-of-order delivery is permitted even 
> > within
> > > a flow.
> > >
> > > For some wireless networks this could become a real pain ...

> ...
> OK, so my wording was a bit loose. With the "link's capacity" I mean the 
> number of TCP segments (MSS) that fit into a *link layer RTT*.

Given the speed of light and so forth, that must mean that the near end
of the link has buffering of at least 4x MSS.  The MSS is at least 1500,
so 4X is 6 KBytes, and 6 KBytes at 14000 bit/sec is 3.4 seconds.


] From: Reiner Ludwig <Reiner.Ludwig@Ericsson.com>

] >It is a Bad Thing(tm) for any router to have a queue that is too big.

] ...
] I fully agree but also think that this is already discussed in the "Advice 
] for Internet Subnetwork Designers" ID.

] >The solution is fix the routers
] >so that they can tell the bulk transfer hosts to back off, and that is to
] >reduce the queue sizes at the ends of that 56 kbit/sec link from
] >10 seconds * 56 Kbit/sec to about 0.5 seconds * 56 kbit/sec.
]
] If you further restrict this to say that this is the upper limit *per 
] flow*, I would fully agree with you again.

If each flow gets fair access, then yes.
Otherwise, the limit must be shared by all flows.

] The upper limit for the number of bytes allowed for queueing at a link per 
] flow before dropping occurs should be "fixed worst-case e2e RTT" x link speed.
]
] You have picked 500 ms for the value of "fixed worst-case RTT". That would 
] be fine with me but might be argued ...

I wouldn't argue against any reasonable value.  50 ms is too small.

5000 ms (or 3400 ms) is intolerable to essentially everyone.  See the 25
year old human factors studies on the upper bounds for responses from
timesharing computers.
Many people are unhappy with total (not just per-link) RTT's above 300 ms.


] >Just because DRAM is almost free today does not mean that you should use
] >GBytes of it for queuing in wireless base stations.  It is important for
] >wires base stations to signal their congestion, and since you will not
] >always have ECN to all hosts, you must drop packets, and that means you
] >must not have too much queing.
]
] I think this is already the message of the "Advice for Internet Subnetwork 
] Designers" ID. What is still unclear is a rule-of-thumb for the 
] dimensioning of the queueing buffer as discussed above.

In that case, what is this about a link having so much buffering
that its RTT is 4X MSS?


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Thu Oct 12 10:20:44 2000
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From: "Franco Carducci" <f.carducci@rmd250b.roma.alespazio.it>
To: <pilc@grc.nasa.gov>
References: <200010091734.e99HYi004218@calcite.rhyolite.com>
Subject: ARQ in the physical layer for satellite links
Date: Thu, 12 Oct 2000 15:25:34 +0200
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Hi all,

I suppose the item in subject has been already deeply analysed (pros and
cons): someone could point me to
those evaluations?

Thank you very much
Best regards

*************************************************************
Franco Carducci
Alenia Spazio SpA
Via G. V. Bona 85 - 00156 Roma - Italy
Phone: + (39) 06 4151 457  Fax: +(39)06 4151 471
E-mail: f.carducci@rmmail.alespazio.it
Web:  www.alespazio.it

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



From owner-pilc@lerc.nasa.gov  Thu Oct 12 11:03:02 2000
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Date: Thu, 12 Oct 2000 16:15:41 +0200
To: Vernon Schryver <vjs@calcite.rhyolite.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: pilc minutes for IETF 48
Cc: pilc@grc.nasa.gov
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At 16:23 11.10.00, Vernon Schryver wrote:
> > From: Reiner Ludwig <Reiner.Ludwig@Ericsson.com>
>
> > > > As soon as the link's capacity goes beyond 4x MTU, any link layer
> > > > retransmission would cause a *spurious* fast retransmit / fast 
> recovery in
> > > > TCP. That assumes that (1) the following 3 packets do not incur a link
> > > > layer retransmission and (2) out-of-order delivery is permitted even
> > > within
> > > > a flow.
> > > >
> > > > For some wireless networks this could become a real pain ...
>
> > ...
> > OK, so my wording was a bit loose. With the "link's capacity" I mean the
> > number of TCP segments (MSS) that fit into a *link layer RTT*.
>
>Given the speed of light and so forth, that must mean that the near end
>of the link has buffering of at least 4x MSS.  The MSS is at least 1500,
>so 4X is 6 KBytes, and 6 KBytes at 14000 bit/sec is 3.4 seconds.

Good point. Although I don't agree to the "... must mean that the near end
of the link has buffering of at least 4x MSS".

It means that the link layer sender can have 4x MSS worth of bytes in 
flight, i.e., unacked at the link layer. That sender would not need to have 
any queue at all if it received the next packet from the higher layer (IP) 
just when it has sent out the (last bit of the) preceding packet/frame.

Instead I would say "... the near end of the link should have buffering of 
at least 4x MSS".

The point you raise is that the rule-of-thumb we discussed in the last mail 
does not work on slow links. Thus, I propose the following modification:

The upper limit for the number of bytes allowed for queueing at a link per 
flow before dropping occurs should be

("fixed worst-case e2e RTT excluding the link layer RTT" + "link layer RTT")
x link speed.


>] From: Reiner Ludwig <Reiner.Ludwig@Ericsson.com>
>
>] >It is a Bad Thing(tm) for any router to have a queue that is too big.
>
>] ...
>] I fully agree but also think that this is already discussed in the "Advice
>] for Internet Subnetwork Designers" ID.
>
>] >The solution is fix the routers
>] >so that they can tell the bulk transfer hosts to back off, and that is to
>] >reduce the queue sizes at the ends of that 56 kbit/sec link from
>] >10 seconds * 56 Kbit/sec to about 0.5 seconds * 56 kbit/sec.
>]
>] If you further restrict this to say that this is the upper limit *per
>] flow*, I would fully agree with you again.
>
>If each flow gets fair access, then yes.
>Otherwise, the limit must be shared by all flows.

I agree. In that case, one might need to have some idea of the average 
number of concurrent flows sharing the link.

///Reiner



From owner-pilc@lerc.nasa.gov  Thu Oct 12 22:57:10 2000
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Date: Thu, 12 Oct 2000 09:06:17 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Reiner Ludwig <Reiner.Ludwig@Ericsson.com>

> ...
> > > OK, so my wording was a bit loose. With the "link's capacity" I mean the
> > > number of TCP segments (MSS) that fit into a *link layer RTT*.
> >
> >Given the speed of light and so forth, that must mean that the near end
> >of the link has buffering of at least 4x MSS.  The MSS is at least 1500,
> >so 4X is 6 KBytes, and 6 KBytes at 14000 bit/sec is 3.4 seconds.
>
> Good point. Although I don't agree to the "... must mean that the near end
> of the link has buffering of at least 4x MSS".

I did not use the word "must" to mean "required by standards,"
but to mean "unavoidably implied" by the hypothesis that the
link has 4X of RTT of buffering.


> It means that the link layer sender can have 4x MSS worth of bytes in 
> flight, i.e., unacked at the link layer. That sender would not need to have 
> any queue at all if it received the next packet from the higher layer (IP) 
> just when it has sent out the (last bit of the) preceding packet/frame.
>
> Instead I would say "... the near end of the link should have buffering of 
> at least 4x MSS".

You declare "should" while I infer "MUST NOT" (in the RFC sense).


> The point you raise is that the rule-of-thumb we discussed in the last mail 
> does not work on slow links.

I strongly disagree.  That you don't like the implications is insufficent
reason to amend the rule of thumb.  The reasons for the rule of thumb
are compelling and unavoidable.  Any link with frequent (not only typical)
delays of 500 milliseconds or more will be considered practically useless
by the majority of users.  A link that frequently suffers 3 second delays
will be completely rejected for any and all interactive applications.  It
does not matter why the link delay delay is frequently so high; it matters
only that essentially all IP applications other than email and possibly
instant messaging used by any except an insignificant minority in the 21st
Century are too frustrating to use with such delays.  Even email, when
the words means IMAP or POP, cannot at best barely tolerate 3 second RTT's.

On the other hand, if the hand-offs, fade, or whatever else that would
cause 3 seconds of delay are infrequent, then there are no reasons to need
4x MSS of buffering and still compelling reasons to not have it.


>                              Thus, I propose the following modification:
>
> The upper limit for the number of bytes allowed for queueing at a link per 
> flow before dropping occurs should be
>
> ("fixed worst-case e2e RTT excluding the link layer RTT" + "link layer RTT")
> x link speed.

That makes sense only with the familiar wireless-Telco reasoning based on
the premises that IP history is irrelevant and that anyone except telco
people cares about radio links.  Both premises are contrary to fact. 

10 or 15 years ago people used TCP/IP over 9600 bit/sec modems with 3-5
seconds of buffering.  They hated the results, and so either commonly
reduced the buffering or used only non-interactive applications.

No user wants to care about any link layer's RTT or other characteristics.
Users use TCP/IP for the applications, not to support telcos and telco
suppliers.  The only RTT that matters to users and applications is the
end-to-end RTT.  The reasoning says that people will change their behavior
and expectations to accomodate whatever a consortium of link layer vendors
finds convenient to sell is literally clinically insane, whether used by
the WAP forum now or the ATM Forum 10 years ago.
If you put too much buffering in any router or other box in the path and
so make the path RTT more than about 500 ms, you will do the otherwise
impossible and make people care about your link layer's characteristics.
Unfortunately, they'll care only about staying away from it.
(Yes, the ATM Forum's insanity was more complicated and not as obvious.)

Because of government help, it was once true that whatever was good for
PTT's was by definition good for their customers.  That is no longer true.


> >] If you further restrict this to say that this is the upper limit *per
> >] flow*, I would fully agree with you again.
> >
> >If each flow gets fair access, then yes.
> >Otherwise, the limit must be shared by all flows.
>
> I agree. In that case, one might need to have some idea of the average 
> number of concurrent flows sharing the link.

Why?  Enough memory for 200-300 ms of buffering for as many flows
as can possibly go to a single mobile phone is, if not free, far
cheaper than other things.

The number of flows that a single mobile phone (or IP host on the
far end of a 1985 vintage Telebit Trailblazer using the Packet
Ensemble Protocol (?) or PEP) is not infinite.  Among its bounds
are the reverse bandwidth consumed by TCP ACKs and the serialization
delays each flow suffers due to others.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Fri Oct 13 06:22:04 2000
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Date: Fri, 13 Oct 2000 11:26:41 +0200
To: Vernon Schryver <vjs@calcite.rhyolite.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: pilc minutes for IETF 48
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At 17:06 12.10.00, Vernon Schryver wrote:
> > The point you raise is that the rule-of-thumb we discussed in the last 
> mail
> > does not work on slow links.
>
>I strongly disagree.

That's OK. Then we can hopefully continue this productive (and I really 
mean it) discussion ...


>Any link with frequent (not only typical)
>delays of 500 milliseconds or more will be considered practically useless
>by the majority of users.  A link that frequently suffers 3 second delays
>will be completely rejected for any and all interactive applications. It
>does not matter why the link delay delay is frequently so high; it matters
>only that essentially all IP applications other than email and possibly
>instant messaging used by any except an insignificant minority in the 21st
>Century are too frustrating to use with such delays.  Even email, when
>the words means IMAP or POP, cannot at best barely tolerate 3 second RTT's.

I strongly disagree ... see below ...


>10 or 15 years ago people used TCP/IP over 9600 bit/sec modems with 3-5
>seconds of buffering.  They hated the results, and so either commonly
>reduced the buffering or used only non-interactive applications.

So let's stay with that example since 9600 bit/sec (sync. = 1200 bytes/sec) 
is exactly the ideal (no errors on the link) bit rate you get on 
state-of-the-art wide area wireless networks: IS95 and GSM.

Even if you flame over telco networks so much, if you really need to rely 
on wide area coverage (beyond nation-wide) then these networks are 
state-of-the-art.

When travelling I (and so do many other people) always and frequently use 
GSM to download e-mail from our Intranet and if I really need a piece of 
info I do HTTP. Believe it or not, there are even situations where I have 
to telnet, and it works OK.

I set my MTU to 1500 bytes since I'm mostly doing bulk data stuff and want 
to get highest throughput (I almost never do telnet and bulk data at the 
same time ;-). Now, the time it takes to transmit a 1500 bytes segment and 
the returning 40 bytes ACK over 1200 bytes/sec is about 1300 ms (not 
counting the PPP framing overhead). Like it or not, but for GSM we need to 
add another 200 ms latency (propagation delay if you prefer) *one-way* over 
the GSM hop. We are at an e2e RTT of 1700 ms by now. Adding a worst case 
Internet RTT of 500 ms which is not unlikely gets us to 2200 ms. Because I 
want to make sure that the link is always utilized you must allow me to 
queue at least one packet. Thus, we need to add another 1300 ms for the 
queueing delay of one single packet.
=> a sum of 3500 ms.

But I don't mind because I really need that e-mail to be downloaded and 
it's not so bad after all. Even telnet is acceptable if I really need to 
get something done.

As I write this e-mail, I'm dialing in via a landline modem with 28.800 
bits/sec which should still be a commonly found link speed. Modems usually 
have a *one way* latency of 50 ms (depending on the manufacturer you can 
find much higher numbers!). Doing the same math gets you to a sum of about 
1500 ms. Now I mind even less because 28.8 kbit/sec is OK for e-mail, WWW, 
and telnet.

That's why I can't agree with what you wrote above: "delays of 500 
milliseconds or more will be considered practically useless"

By the way, it's not always the telcos and the modem vendors that are 
responsible for excessive queueing. Check out the BSD operating system. By 
default it happily allocates a 50 packet large output buffer for a PPP 
interface. Given a TCP receiver window of 8 or 16 KBytes, most of those 
bytes will sit in BSD's output buffer. And then we are talking about RTTs 
of several seconds! See, e.g., 
http://iceberg.cs.berkeley.edu/papers/Ludwig-Sigmetrics99/index.html

Yes, I know that there are per route metrics that BSD4.4 uses to avoid 
that. But those metrics are discarded after each PPP session (when the 
route goes down) so that does not help much.

So, you haven't convinced me that the last version of the rule-of-thumb is 
broken.

///Reiner



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the 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, M. Kojo, 
                          V. Magret, N. Vaidya
	Filename	: draft-ietf-pilc-error-05.txt
	Pages		: 20
	Date		: 12-Oct-00
	
The rapidly-growing Internet 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.

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From owner-pilc@lerc.nasa.gov  Fri Oct 13 12:53:52 2000
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Date: Fri, 13 Oct 2000 09:34:09 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Reiner Ludwig <Reiner.Ludwig@Ericsson.com>

> ...
> So let's stay with that example since 9600 bit/sec (sync. = 1200 bytes/sec) 
> is exactly the ideal (no errors on the link) bit rate you get on 
> state-of-the-art wide area wireless networks: IS95 and GSM.
>
> Even if you flame over telco networks so much, if you really need to rely 
> on wide area coverage (beyond nation-wide) then these networks are 
> state-of-the-art.
>
> When travelling I (and so do many other people) always and frequently use 
> GSM to download e-mail from our Intranet and if I really need a piece of 
> info I do HTTP. Believe it or not, there are even situations where I have 
> to telnet, and it works OK.
>
> I set my MTU to 1500 bytes since I'm mostly doing bulk data stuff and want 
> to get highest throughput (I almost never do telnet and bulk data at the 
> same time ;-). Now, the time it takes to transmit a 1500 bytes segment and 
> the returning 40 bytes ACK over 1200 bytes/sec is about 1300 ms (not 
> counting the PPP framing overhead). Like it or not, but for GSM we need to 
> add another 200 ms latency (propagation delay if you prefer) *one-way* over 
> the GSM hop. We are at an e2e RTT of 1700 ms by now. Adding a worst case 
> Internet RTT of 500 ms which is not unlikely gets us to 2200 ms. Because I 
> want to make sure that the link is always utilized you must allow me to 
> queue at least one packet. Thus, we need to add another 1300 ms for the 
> queueing delay of one single packet.
> => a sum of 3500 ms.
>
> But I don't mind because I really need that e-mail to be downloaded and 
> it's not so bad after all. Even telnet is acceptable if I really need to 
> get something done.

It does not matter whether you label such fundamentally broken services
state of the art, by appointment to the king, or anything else.  It is
simply impossible to get people to use such services in the ways that make
your conclusions valid.
Aamong the invalid assumptions are

  - since X is the best that can by done with Y from industry, consortium
   or vendor Z, therefore people want and will pay to do X a lot.  It is
   not only the telcos and telco suppliers that use such reasoning,
   although they perfected it before the computer people arrived.

  - that other than less than an insignificant number of people will ever
   use telnet to talk to an HTTP server or even conceive of the possibility.
   You and I do, but we don't even begin to count.

  - email over such links to such phones needs 4X MSS of buffering in
   the routers.  How big is the screen on your phone, and how much fun
   would 99.99% of users have trying to read an SMTP (or POP or IMAP)
   message that could use 4X MSS?

  - that people have changed since they rejected 500 ms delays 10-15
   years ago and will now embrace them.


> As I write this e-mail, I'm dialing in via a landline modem with 28.800 
> bits/sec which should still be a commonly found link speed. Modems usually 
> have a *one way* latency of 50 ms (depending on the manufacturer you can 
> find much higher numbers!). Doing the same math gets you to a sum of about 
> 1500 ms. Now I mind even less because 28.8 kbit/sec is OK for e-mail, WWW, 
> and telnet.

Yes, and why don't v.34 modems have 30 KBytes or more of buffering?
Why haven't the salescritters that assure us that v.34 modems are really
56 Kbit/sec because we all know that v.42bis gives 4X put another $1 of
buffering in their modems?  One answer is that 90% of consumers cannot
tell the difference between surfing the web at 28-40 kbps with v.34+v.42bis
and using a true 56 kbit/sec, but they do notice that any extra buffering
hurts a lot.


> That's why I can't agree with what you wrote above: "delays of 500 
> milliseconds or more will be considered practically useless"

Agree or not as you like; for anything except email and by anyone except
ubber geeks, it is a fact.  We "have been there and done that."  Delays
of 500 ms or more were found practically useless for interactive applications
10 or 15 years ago by most people.  With their home PC's now running at
6 Mbit/sec and less than 20 ms, they will not and cannot care whether a
GSM phone with 1700 ms is sold by telcos as state of the art or required
by PTT regulations.


> By the way, it's not always the telcos and the modem vendors that are 
> responsible for excessive queueing. Check out the BSD operating system. By 
> default it happily allocates a 50 packet large output buffer for a PPP 
> interface. Given a TCP receiver window of 8 or 16 KBytes, most of those 
> bytes will sit in BSD's output buffer. And then we are talking about RTTs 
> of several seconds! See, e.g., 
> http://iceberg.cs.berkeley.edu/papers/Ludwig-Sigmetrics99/index.html

You're concluding based on too little data.
The commercial PPP code I wrote and caused to be shipped on a system that
used the BSD TCP stack to several 10,000 customers didn't do that.
Moreover, I caused that commmercial UNIX system to ship with a 30 KByte
default TCP window in about 1988 and a 60 K window in about 1993.  (That
is related to my sad tales of woe concerning Ethernet-FR routers with 50
Kbytes of buffering.)
When the company's heavily flogged voice/UDP/IP-multicast application ran
over PPP, it configured a PPP queue length of three hundred (300), but a
tiny MTU and also marked its packets to allow the three kinds of ToS
queuing in the PPP driver to do their things.  (1 kind of ToS queuing was
the IP TOS byte.)
Still more, that particular PPP implementation supported a kind of
multilink that causes packets to be re-ordered, which is where I got my
prejudices about links that re-order and the results on TCP.


> Yes, I know that there are per route metrics that BSD4.4 uses to avoid 
> that. But those metrics are discarded after each PPP session (when the 
> route goes down) so that does not help much.

The 4.4BSD PPP code is very primitive.
Customers forced me to make my PPP code configure routes when links came up.

> So, you haven't convinced me that the last version of the rule-of-thumb is 
> broken.

What you, I, the bosses and sales people at telcos and telco suppliers
think does not matter.
It has been known for 30 years that people start refusing to use
interactive computer applications when delays reach 5 seconds.
See those ancient studies of timesharing systems.

It has also been known for at least 10 years that 500 ms of modem delay
on interactive applications is intolerable to the vast majority of users.
Look at the history of the Telebit corporation.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Sun Oct 15 17:20:16 2000
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I'm not taking sides with neither Mr. Schryver nor Mr. Ludwig
on the issue of how best to optimize TCP delay performance 
over wireless links.

However, I'd like to point out one thing that Mr. Schryver and
many Americans fail to recognize because they may not be witnessing
the explosive growth and the convenience of sending short alpha-
numeric messages (SMS). The lesson is: many applications needn't be 
interactive. Of course Mr. Schryver is right and all the research 
about maximum latency for interactive applications is still valid
today: response times > 500 ms will not be accepted by users.

But email, to cite one example, is NOT interactive. It's only so
when you log on through your home phone line. But a cellular phone is
not quite an ordinary phone. It is also a small computer and it's
always connected (namely through the cellular control channels).
Therefore, the idea is to deliver email directly to the mobile in
which case latency is simply not an issue. Taking another example:
ordering something doesn't necessarily require an immediate feedback 
across the network, either.

Finally, don't forget that a cellular phone is already equipped for
interactive applications when the need arises: just use the voice 
channel. Now, I'm not a firm believer in voice browsing in light of 
all the time I've wasted in recorded-voice menu trees, but I do
think the user experience could be made much more productive than
what is commonplace now. And then there are call centers, of course.

With all this, I am not saying you should not minimise delay and I am
not saying that interactive applications are not important. All I am
saying is that you shouldn't completely dismiss TCP for systems and 
links where you can't meet the 500ms response time requirement. There
may be uses even for that.

Stefan Wiesendanger

DE-TCOM
Swiss Federal Institute 
of Technology, Lausanne



From owner-pilc@lerc.nasa.gov  Sun Oct 15 19:35:26 2000
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Date: Sun, 15 Oct 2000 16:44:27 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: stefan.wiesendanger@epfl.ch

> ...
> However, I'd like to point out one thing that Mr. Schryver and
> many Americans fail to recognize because they may not be witnessing
> the explosive growth and the convenience of sending short alpha-
> numeric messages (SMS).

From my prospective, it's not the "explosive growth of short messages,"
but the "return of short messages."  In the old days (staring in the 1960's
when I first used email), messages were what you could type in a few
seconds on a 10 char/sec model 35 or 33 TTY or 5 char/sec Flexowriter, and
did not have zillions of headers (such as X-Face for pity's sake!), not
to mention MBytes of spread sheets and pictures, fonts, cryptographic
signatures, and complete duplicates of the whole message expanded by
30-100% and encrypted in HTML.  Maybe I'm overstating things a little.
Informal email (i.e.  not to public mailing lists or netnews) was less than
500 but more than 100 bytes.

>                         The lesson is: many applications needn't be 
> interactive  ...

I guess I wasn't clear when I tried to say that 4X MSS worth of buffering
in radio base stations for short text messages that consist of far less
than one MSS is silly.


> But email, to cite one example, is NOT interactive. It's only so
> when you log on through your home phone line. But a cellular phone is
> not quite an ordinary phone. It is also a small computer and it's
> always connected (namely through the cellular control channels).
> Therefore, the idea is to deliver email directly to the mobile in
> which case latency is simply not an issue.

Yes, but "email" is not the really the issue either.  Do those "state of
the art" GSM phones with > 1700 ms of latency also have enough memory to
hold enough bits of email to worry about? Do they have even as much as 4X
MSS=6KBytes?
Are their screens big enough to allow you to read a message longer than
100 bytes?  Your message contained 1893 bytes of text.  Could you have
typed it on a cell phone keypad?  Could you have reviewed it for spelling
and grammar on a cell phone screen?  I couldn't.
Thus, whatever phone text messaging is, it's not modern email.

There are people with lots of real experience with the ultra-short text
messages (50X to 50,000X smaller than 21st Century SMTP) that seem to be
what the GSM and third generation phone people are talking about.  Instead
of using bulk transfer protocols such as TCP/IP, why not look at what the
pager people have been doing for decades?


>                                            Taking another example:
> ordering something doesn't necessarily require an immediate feedback 
> across the network, either.

No, if your order is not instantly acknowledged and you are like most
consumers, you will be very unhappy, will complain, and will try to cancel
or repeat the order.  And never mind how you were able to do the very
interactive job of getting enough information to place your order (e.g.
bus schedule or take-out restaurant menu) over a 1700 ms RTT link without
giving up in frustration.


> Finally, don't forget that a cellular phone is already equipped for
> interactive applications when the need arises: just use the voice 
> channel. Now, I'm not a firm believer in voice browsing in light of 
> all the time I've wasted in recorded-voice menu trees, but I do
> think the user experience could be made much more productive than
> what is commonplace now. And then there are call centers, of course.

That's very true, and it says volumes about the utility and wisdom of
worrying about how many MSS of buffer to put in base stations to make
HTTP/SMTP/TCP/IP work over links with 1700 ms latency.


> With all this, I am not saying you should not minimise delay and I am
> not saying that interactive applications are not important. All I am
> saying is that you shouldn't completely dismiss TCP for systems and 
> links where you can't meet the 500ms response time requirement. There
> may be uses even for that.

As I've repeatedly said, TCP was and even still is used on links with >500
ms latency.  It's just that 500 ms doesn't work for interactive
applications, and essentially all of the proposed uses of IP-to-GSM uses
are interactive, because those "state of the art" phones don't have enough
storage to do anything else.  Almost all of their storage is in the brain
of the user, which is another way of saying the proposed applications are
interactive.

The possible exception is email, but the keyboard, storage, and screen
real estate of phones means that phone-email is something very different
from what we are using here.  Who would want to read any of the messages
in this list on a cell phone's screen?  Who could have prepared even a
fragment of one on a cell phone?


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Mon Oct 16 12:39:17 2000
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Vernon Schryver wrote:

> > From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>
>
> > I don't think we differ as much as you may think we do.
>
> I suspect we differ more than either of us thought.

I wrote this to encourage more debate, pleased you responded.

>
>
> > Sorry, this debate is less than clear.  I was talking only about WINDOW size,
> > in many link ARQ schemes there is a fixed modulus or window which
> > limits the maximum number of packets in flight.  This may be determined by
> > either amount of buffer memory or size of frame number field (before everyone
> >shouts and says "but not all" - I know there are exceptions). The HDLC standard
> > works this way - and so do nearly all variants of it, I have seen.
>
> Yes, but if you have a whole window's worth of packets that cannot get
> through, then the link is very sick.  There's little to talk about besides
> radio things like hand-off that should be inivisible to TCP except possibly
> for a bunch of lost segments.

No, I disagree. Not so sick, for some types of link, just unfortunate: It's
not just cellular mobile we are thinking of.

>
>
> If you have one stuck packet that is keeping the window from advancing
> and so you cannot get any other data through the link, then either your
> window is too small or you have broken hardware or software that cannot
> be fixed with ARQ or anything else (i.e. my #3).  There is absolutely
> nothing that can or should be done in this case except possibly to change
> vendors in a forklift upgrade.

Window is indeed too small (at least in relation to the problem).

A link reset may well fix the problem.  If the protocol engines at each end
reinitialise
(and hopefully log), then you can get them back in sync with no memory issues.

>
>
> > ...
> > > Moreover, by Dr. Fairhurst's assumptions, either
> > >   1. no other packets for any TCP stream are flowing because the link
> > >      is very sick.
> > >   2. this particular packet will soon get through because the problem
> > >      was temporary.
> > >   3. something is very badly broken in the design or implementation of
> > >       the link layer, because one particular packet cannot get through
> > >       but others can.
>
> > ...
> > 1 is what I call an OUTAGE - we have text already which recommends what
> > to do (hang on to the last few packets if you can and send them once you
> > are able - not quite the same as recommending persistence!)
> >
> > 2 may happen - but I have seen cases where this can take a long time
> > - particularly with a big packet and periodic interference.
> >
> > 3. This happens too... and can be bizarre.
> >
> >The non-persistent argument seems to help with 2,3.  At least it clears out all
> > that
> > old link state and any buffers tied up, so that things can resume orderly
> > in the
> > future.
>
> No, in those three cases:
>   1. there is nothing to do.  The link is broken and has been broken for
>      at least a little while.  Saving packets in the hope that the link
>      might return might be good, but is probably a waste of cycles and
>      memory, and quite possibly worse.
>

I'm not convinced the link ARQ platform will always be doing useful work
if the link is not moving data - so I'd like to let people think they could
use the cycles to do something (if they choose to do so). By "worse",
I presume you mean MSL (good point, see below) - are there other bad things
I haven't thought of?

>
>      You MUST NOT keep any TCP or UDP packet for longer than the MSL.
>      You will break things if you somehow keep a packet for a long time
>      and then eventually regurgitate it.

good point. MSL is always to be observed. Incidentally this means that true
reliability (with unlimited retransmissions) is not allowed for IP!

>   2. this is ordinary bit rot.  If you often find problems with big packets
>      and periodic interference, then you have misdesigned the link layer
>      packet sizes by making it too big.
>

Sure, just a badly configured link, or very unfortunate error pattern. Neither
are impossible, and can be found in the Internet.

>
>   3. there is nothing that can be done in this case except to ship all of
>     the hardware and software back to the factory, and possibly to fire
>     the poor fellow who messed up the FEC or CRC computation, buffer
>     chaining, or whatever.
>

Or do a link reset, possibly drop some frames, and continue.

>
> > What I'm trying to find out is what we should recommend when we have a
> > high (or moderately high) packet loss rate.
> >
> > If we have only one TCP flow, do we care?
> > Probably not, as long the ARQ tries (a few attempts) or forever.  The
> > latter may
> > even be better. Good versions of HDLC could work fine with segmentation.
> >
> > If we have many flows - are there other issues?
> > I can think of these two:
> >
> > 1) The ability to classify UDP form TCP seems very desirable (although
> > difficult with tunnels/encapsulation and fundamentally problematic with IPSEC)
>
> It's wise to not attempt things that you know are impossible.  Given the
> fact that IPSec exists, the fact that sooner or later people will get
> paranoid about their bits, and the fact that people are already trained
> to be paranoid about their voices on radios by analog cellular phone
> scanners, it is crazy (as in failing to accept reality) to do anything
> but assume IPSec makes classifying impossible.

I agree unless the IPSEC user chooses to signal the transport class (by some means)

or inhibits IPSEC, then we can not assume an ability to classify the transport
protocol.

>
> > 2) I suggest sharing with other TCP flows can become an issue - perhaps less
> > so if you are end-system connected to a mobile net, but more so if you are
> > connecting
> > a remote LAN with many hosts over a wireless (or any RF) link. Several factors
> > here:
> >     per-flow state,
> >     head-of-line block when buffer/window exceeded,
> >     forwarding delay (out of order delivery may be desirable)
>
> Out of order forwarding is not desirable, but it is always far more
> desirable to get the bits through the link than to second guess the upper
> layers tolerance to re-ordering.
> All IP protocols and applications are absolutely required to tolerate
> absolutely all possible re-orderings of IP packets within the MSL.  Any
> delaying of later packets merely because earlier packets need link layer
> retransmitting would be *WRONG*.  And again, there is the MSL limit
> on how long you can delay (including retransmitting) any packet.

MSL - agreed.  "desirable" - should be "may prove beneficial in some cases"

>
>
> When your window or buffer is full, what can you do beside hs head-of-line
> block and drop incoming packets?  Mentioning that as an alternative makes
> as much sense as mentioning that when the system melts in a fire, it must
> stop working.
>

Can drop from front or back....

>
> Per-flow state might be handy, but why would you not start with the sorts
> of things that are already being done or considered for other sorts
> of routers?  For example, why not do a modified RED?

Has anyone done this?

>
>
> Vernon Schryver    vjs@rhyolite.com

Gorry Fairhurst

------------------------------
http://www.erg.abdn.ac.uk/users/gorry



From owner-pilc@lerc.nasa.gov  Mon Oct 16 14:23:46 2000
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PILC WG,

Due to a personal circumstance, Aaron will be offline for 
some weeks.  Spencer and Aaron have asked that Mark Allman be
viewed as the co-chair pro-tem during this time.  I agree to
this and thank Mark in advance for his efforts.

Allison 




From owner-pilc@lerc.nasa.gov  Mon Oct 16 20:41:14 2000
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From: Bill Yeager <William.Yeager@Eng.Sun.COM>
Reply-To: Bill Yeager <William.Yeager@Eng.Sun.COM>
Subject: Re: pilc minutes for IETF 48
To: Vernon Schryver <vjs@calcite.rhyolite.com>
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| The possible exception is email, but the keyboard, storage, and screen
| real estate of phones means that phone-email is something very different
| from what we are using here.  Who would want to read any of the messages
| in this list on a cell phone's screen?  Who could have prepared even a
| fragment of one on a cell phone?

Exactly the point. Most of those email users who use cell phones will send
very short messages to one another. And, when the messages are not short, ie,
not sent from a cell phone, then any smart email client using IMAP for example
will take advantage of the fetch partial commands and page email messages to
telephones in say, 256 byte chunks.

The above is not a fantasy. It works quite well even with HTTP proxy to a
(shall I dare say) WAP gateway. I did such a product to understand WAP and
wireless, and right, it convinced me that WAP MUST converge to the IETF
standards on the one hand, but that the email service was quite good on the
other hand. The same can be say wrt to calendar access.

We are not talking ftp here or even 4xMSS worth of data. I've used HTTP with
HandWeb on my Palm Pilot talking to a proxy [http <--> imap/smtp/ldap] using
servlets to output compact HTML on the CDPD network here in the bay area. I
could hold a WAP enable telephone in the other hand talking to the same proxy
that was also WML or HDML enabled. Equal performance because the application
was tailored to the device. App designers must be aware of the platform.

This same code has been running over GSM networks in Europe. It works well
because it was designed with the target device in mind. 

So, for example, one can provide the phone with a list of pre-defined
responses that can be set up on the phone or with a web interface. One finger
responses. Make short cuts, take advantage of the smarts in protocols like
IMAP for managing low-bandwidth high latency and noisy connections. 

USE TCP/IP end-2-end!!! That is what moves our industry forward. It is a
unifying force. 

Right, I've been involved in networking since the 60's too. DSN at Nasa-AMES
for the pioneer spacecrafts. Kermit was based on my code (ttyftp which I wrote
at Stanford in the late 70's) and the cisco routers were initially licensed
from Stanford as a derivative of my work, the Sumex-aim etherTip/router code.

And remember, real apps will soon be arriving to the phone. Smart apps. They
will initially require HTTP/TCP/IP end-2-end and soon after use protocols like
IMAP and SMTP. They will be light weight wrt to data use. And, they will use
SSL/TLS for security end-2-end. They will tolerate 2 seconds of RTT, and still
give a great user experience. THAT is cool, very cool.

Finally remember, WAP 2.0 is TCP/IP + basic XHTML + (I'm betting on SSL/TLS
when implemented). There will be required legacy support for WML. But, I'm
again betting on proxy rather than phone resident WML browers for a great deal
of the content.

All of the work in pilc has inspired this. Great arguments! Great results.

Thanks a lot,

Bill


  



From owner-pilc@lerc.nasa.gov  Mon Oct 16 22:07:26 2000
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From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Bill Yeager <William.Yeager@eng.sun.com>

> ...
> Exactly the point. Most of those email users who use cell phones will send
> very short messages to one another. 

like pages or email before 21st Century SMTP.


> ...
> We are not talking ftp here or even 4xMSS worth of data. ...

No, we clearly are talking aobut FTP sized transfers.  If the issue did
not involve large streams of data, then the notion of ARQ blocking other
packets would make no sense.  The issue of re-ordering TCP segments when
there are 4 or more outstanding segments (and so 4xMSS=6000 bytes in
flight) does not make sense up except for big streams.


> ...
> And remember, real apps will soon be arriving to the phone. Smart apps. They
> will initially require HTTP/TCP/IP end-2-end and soon after use protocols like
> IMAP and SMTP. They will be light weight wrt to data use. And, they will use
> SSL/TLS for security end-2-end. They will tolerate 2 seconds of RTT, and still
> give a great user experience. THAT is cool, very cool.

That's not going to happen until phones have reasonable keyboards (or
equivalents), displays, and memory (e.g. attachments to PDA's).  When
that happens, either radio modems will have lower latencies and higher
throughputs or those wonderful applications will still not happen.
The modern Internet would not exist if it were still based on 56k leased
lines connecting IMPs ad TIPs.

No matter how cool things seem to us, users do not tolerate the delays
implied by several 1700 ms round trips for DNS followed by 3 more round
trips to set up an HTTP/TCP/IP session just to get an airline schedule or
restaurant menu so they can start to place an order.  On the other hand,
if those delays and losses go away, much of the PILC proposed improvements
to TCP, questions about ARQ below, and so forth become moot.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Tue Oct 17 14:27:30 2000
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To: pilc@grc.nasa.gov
Cc: "Spencer Dawkins" <spencer.dawkins@fnc.fujitsu.com>,
        "Aaron Falk" <afalk@PanAmSat.com>
Cc: gab@sun.com, kojo@cs.Helsinki.FI, vincent.magret@aud.alcatel.com,
        vaidya@cs.tamu.edu
From: Mark Allman <mallman@grc.nasa.gov>
Reply-To: mallman@grc.nasa.gov
Subject: WG last call for ERROR
Organization: Late Night Hackers, NASA Glenn, Cleveland, Ohio
Song-of-the-Day: Pinball Wizard
Date: Tue, 17 Oct 2000 13:27:55 -0400
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Folks-

I'm told that my first task as "co-chair pro tem" is to issue a WG
last call on the following document...

	Title		: End-to-end Performance Implications of Links with 
                          Errors
	Author(s)	: S. Dawkins, G. Montenegro, M. Kojo, 
                          V. Magret, N. Vaidya
	Filename	: draft-ietf-pilc-error-05.txt
	Pages		: 20
	Date		: 12-Oct-00
	
	http://www.ietf.org/internet-drafts/draft-ietf-pilc-error-05.txt

The document has undergone a previous WG last call, but some
significant new text has been added (section 1.3) and therefore we'd
like to issue another short last call period before forwarding this
document to the IESG.  Unless hearing otherwise, we will assume
there is consensus on this document and that everyone is tired of it
on 10/25/2000 and forward it to the IESG for publication as a
BCP/Informational(***) RFC.  Please take a look at the most recent
version and send comments to the authors or the mailing list
(positive ACKs ("yep, it looks good") to me or the list would also
be appreciated).

*** I have lost track of which kind of document ERROR is tracked to
be.  Can you remind us Spencer?

Thanks,
allman


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


From owner-pilc@lerc.nasa.gov  Tue Oct 17 15:20:31 2000
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"Yep, it looks good."

(I sent a few nit comments to the authors to be addressed before final
publication...)


John


Mark Allman wrote:
> 
> 
> Folks-
> 
> I'm told that my first task as "co-chair pro tem" is to issue a WG
> last call on the following document...
> 
>         Title           : End-to-end Performance Implications of Links with
>                           Errors
>         Author(s)       : S. Dawkins, G. Montenegro, M. Kojo,
>                           V. Magret, N. Vaidya
>         Filename        : draft-ietf-pilc-error-05.txt
>         Pages           : 20
>         Date            : 12-Oct-00
> 
>         http://www.ietf.org/internet-drafts/draft-ietf-pilc-error-05.txt
> 
> The document has undergone a previous WG last call, but some
> significant new text has been added (section 1.3) and therefore we'd
> like to issue another short last call period before forwarding this
> document to the IESG.  Unless hearing otherwise, we will assume
> there is consensus on this document and that everyone is tired of it
> on 10/25/2000 and forward it to the IESG for publication as a
> BCP/Informational(***) RFC.  Please take a look at the most recent
> version and send comments to the authors or the mailing list
> (positive ACKs ("yep, it looks good") to me or the list would also
> be appreciated).
> 
> *** I have lost track of which kind of document ERROR is tracked to
> be.  Can you remind us Spencer?
> 
> Thanks,
> allman
> 
> ---
> http://roland.grc.nasa.gov/~mallman/


From owner-pilc@lerc.nasa.gov  Tue Oct 17 15:38:16 2000
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From: "Dawkins, Spencer" <Spencer.DAWKINS@fnc.fujitsu.com>
To: pilc@grc.nasa.gov
Subject: RE: WG last call for ERROR
Date: Tue, 17 Oct 2000 13:42:08 -0500
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ERROR, SLOW, and LINK were tracked as BCPs in our previous discussions.

Spencer

> -----Original Message-----
> From:	Mark Allman [SMTP:mallman@grc.nasa.gov]
> Sent:	Tuesday, October 17, 2000 12:28 PM
> To:	pilc@grc.nasa.gov
> Cc:	Spencer Dawkins; Aaron Falk; gab@sun.com; kojo@cs.helsinki.fi; vincent.magret@aud.alcatel.com; vaidya@cs.tamu.edu
> Subject:	WG last call for ERROR
> 
>  
> Folks-
> 
> I'm told that my first task as "co-chair pro tem" is to issue a WG
> last call on the following document...
> 
> 	Title		: End-to-end Performance Implications of Links with 
>                           Errors
> 	Author(s)	: S. Dawkins, G. Montenegro, M. Kojo, 
>                           V. Magret, N. Vaidya
> 	Filename	: draft-ietf-pilc-error-05.txt
> 	Pages		: 20
> 	Date		: 12-Oct-00
> 	
> 	http://www.ietf.org/internet-drafts/draft-ietf-pilc-error-05.txt
> 
> The document has undergone a previous WG last call, but some
> significant new text has been added (section 1.3) and therefore we'd
> like to issue another short last call period before forwarding this
> document to the IESG.  Unless hearing otherwise, we will assume
> there is consensus on this document and that everyone is tired of it
> on 10/25/2000 and forward it to the IESG for publication as a
> BCP/Informational(***) RFC.  Please take a look at the most recent
> version and send comments to the authors or the mailing list
> (positive ACKs ("yep, it looks good") to me or the list would also
> be appreciated).
> 
> *** I have lost track of which kind of document ERROR is tracked to
> be.  Can you remind us Spencer?
> 
> Thanks,
> allman
> 
> 
> ---
> http://roland.grc.nasa.gov/~mallman/


From owner-pilc@lerc.nasa.gov  Wed Oct 18 02:05:04 2000
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Subject: Re: pilc minutes for IETF 48 - Persistence
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At 17:31 16.10.00, you wrote:
>good point. MSL is always to be observed. Incidentally this means that true
>reliability (with unlimited retransmissions) is not allowed for IP!

If you check the PILC archive, you will find that I always mentioned the 
MSL as an upper bound for link layer error recovery persistency. In fact, 
the maximum RTO of 64 seconds is a more reasonable upper bound.

///Reiner




From owner-pilc@lerc.nasa.gov  Wed Oct 18 07:07:22 2000
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Subject: Re: pilc minutes for IETF 48 - Persistence
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At 10:40 18.10.00, Dr G Fairhurst wrote:
>Yes, but asserting that MSL/TCP RTO should be considered an upper delay bound,
>aren't you saying you don't think a fully reliable link ARQ (aka HDLC) is a
>good idea?

I'm saying (again) that link layer error recovery (aka ARQ) as, e.g., 
implemented by HDLC, on lossy links or links that suffer transient outages 
is a great idea, *if* that link also implements flow seperation. The 
maximum time the link layer sender tries to re-/transmit a single TCP 
segment should, however, be bound to 64 sec. (TCP's max RTO).

>P.S. 64 secs will usually equate to a very large number of 
>retransmissions...or
>a
>significant link outage.

True.

///Reiner




From owner-pilc@lerc.nasa.gov  Wed Oct 18 07:23:01 2000
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On Wed, 18 Oct 2000, Reiner Ludwig wrote:

> At 17:31 16.10.00, you wrote:
> >good point. MSL is always to be observed. Incidentally this means that true
> >reliability (with unlimited retransmissions) is not allowed for IP!
> 
> If you check the PILC archive, you will find that I always mentioned the 
> MSL as an upper bound for link layer error recovery persistency. In fact, 
> the maximum RTO of 64 seconds is a more reasonable upper bound.

Reasonable for whom?

Having an upper bound much more than several times the link delay
makes little sense. An upper bound that is a significant fraction or
more of the path delay makes little sense. And a telnet session with a
lag of 64 seconds is something I can do quite happily without.

By increasing persistency you're not increasing link
utilisation. You're not helping TCP(-friendly) congestion control
algorithms. What are you doing, exactly?

L.

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




From owner-pilc@lerc.nasa.gov  Wed Oct 18 12:22:16 2000
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Subject: Re: pilc minutes for IETF 48 - Persistence
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Zoran,

Two questions:

A. What is the air link data rate in your case?

B. When you say higher persistency, are you also increasing the aggressiveness
of the link layer recovery? Meaning, if the 3 round recovery scheme
retransmits 1,2 and 3 times then a 5 round one retransmits 1,2,3,4 and 5 times?


Zoran Kostic wrote:
> 
> Lloyd,
> 
> This is an example of why and when it makes sense to
> have an upper delay more than several times the path delay.
> Also, an argument (from yet another individual) in favour
> of "higher" persistency.
> 
> In a cellular network with mobile users, you will often find
> a user whose performance is just fine most of the time.
> Occassionally, it will hit nasty channel conditions, get
> stuck for a short amount of time and get out of trouble
> after not so long. Well, we do not want to break the
> TCP connection while it is stuck, because we know that
> it will get out of trouble. And statisctically, this happens often
> (in the cellular environment)!
> 
> We've run an extensive set of simulations investigating
> interactions between the the cellular environment and TCP.
> Indeed, having very persistent lower layer gives the best results
> from the performance point of view. Having less than 3 or
> 4 retransmissions always screws up things if the channel is
> messy or if the system is highly loaded (many users per cell),
> and infinite persistency gets decently mimicked by
> something on the order of 5-10 retransmission on the RLP
> layer.
> 
> So, as an answer to your question
> "What are you doing, exactly?", I offer the following:
> In the cellular environment, with large number of users,
> cumulative distribution functions expressing system throughput
> and delay show significant improvements if the persistency on
> the lower layers is high. So, we are doing a great deal here !
> 
> Yes, we want persistency. It is not clear that we need to have
> infinite persistency or always as high as 64 seconds, but having
> only 3 retransmission (as in IS99) is suboptimal.
> I actually think that variable persistency (as a function of
> cellular channel conditions) may make sense.
> 
> Zoran


From owner-pilc@lerc.nasa.gov  Wed Oct 18 12:38:34 2000
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Subject: Re: pilc minutes for IETF 48 - Persistence
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Lloyd,

This is an example of why and when it makes sense to
have an upper delay more than several times the path delay.
Also, an argument (from yet another individual) in favour
of "higher" persistency.

In a cellular network with mobile users, you will often find
a user whose performance is just fine most of the time.
Occassionally, it will hit nasty channel conditions, get
stuck for a short amount of time and get out of trouble
after not so long. Well, we do not want to break the
TCP connection while it is stuck, because we know that
it will get out of trouble. And statisctically, this happens often
(in the cellular environment)!

We've run an extensive set of simulations investigating
interactions between the the cellular environment and TCP.
Indeed, having very persistent lower layer gives the best results
from the performance point of view. Having less than 3 or
4 retransmissions always screws up things if the channel is
messy or if the system is highly loaded (many users per cell),
and infinite persistency gets decently mimicked by
something on the order of 5-10 retransmission on the RLP
layer.

So, as an answer to your question
"What are you doing, exactly?", I offer the following:
In the cellular environment, with large number of users,
cumulative distribution functions expressing system throughput
and delay show significant improvements if the persistency on
the lower layers is high. So, we are doing a great deal here !

Yes, we want persistency. It is not clear that we need to have
infinite persistency or always as high as 64 seconds, but having
only 3 retransmission (as in IS99) is suboptimal.
I actually think that variable persistency (as a function of
cellular channel conditions) may make sense.

Zoran




Lloyd Wood wrote:

> On Wed, 18 Oct 2000, Reiner Ludwig wrote:
>
> > At 17:31 16.10.00, you wrote:
> > >good point. MSL is always to be observed. Incidentally this means that true
> > >reliability (with unlimited retransmissions) is not allowed for IP!
> >
> > If you check the PILC archive, you will find that I always mentioned the
> > MSL as an upper bound for link layer error recovery persistency. In fact,
> > the maximum RTO of 64 seconds is a more reasonable upper bound.
>
> Reasonable for whom?
>
> Having an upper bound much more than several times the link delay
> makes little sense. An upper bound that is a significant fraction or
> more of the path delay makes little sense. And a telnet session with a
> lag of 64 seconds is something I can do quite happily without.
>
> By increasing persistency you're not increasing link
> utilisation. You're not helping TCP(-friendly) congestion control
> algorithms. What are you doing, exactly?
>
> L.
>
> <L.Wood@surrey.ac.uk>PGP<http://www.ee.surrey.ac.uk/Personal/L.Wood/>



From owner-pilc@lerc.nasa.gov  Wed Oct 18 13:52:40 2000
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Hi.

> The document has undergone a previous WG last call, but some
> significant new text has been added (section 1.3) and therefore we'd
> like to issue another short last call period before forwarding this
> document to the IESG.  Unless hearing otherwise, we will assume
> there is consensus on this document and that everyone is tired of it
> on 10/25/2000 and forward it to the IESG for publication as a
> BCP/Informational(***) RFC.  Please take a look at the most recent
> version and send comments to the authors or the mailing list
> (positive ACKs ("yep, it looks good") to me or the list would also
> be appreciated).

Positive ACK.

Cya
Matthew

Matthew King.
IP Network Engineer, Cable & Wireless Optus.


From owner-pilc@lerc.nasa.gov  Wed Oct 18 15:50:18 2000
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Subject: Re: pilc minutes for IETF 48 - Persistence
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   Kamesh,

  * The airlink is the one used in EDGE - thereby there is variability
     of the rate between roughly 9 and 59 kbps.

  * The way I think of persistency on the link is the following: Attempt
     to transmit the block no more than X times. So, I do not have the
     concept of rounds, just plain number of  transmission attempts.

     On this layer I also use the mode 0 option, whereby if the channel
     or interference are really bad, I do not even try to retransmit.

   Zoran




Kamesh Medepalli wrote:

> Zoran,
>
> Two questions:
>
> A. What is the air link data rate in your case?
>
> B. When you say higher persistency, are you also increasing the aggressiveness
> of the link layer recovery? Meaning, if the 3 round recovery scheme
> retransmits 1,2 and 3 times then a 5 round one retransmits 1,2,3,4 and 5 times?
>
> Zoran Kostic wrote:
> >
> > Lloyd,
> >
> > This is an example of why and when it makes sense to
> > have an upper delay more than several times the path delay.
> > Also, an argument (from yet another individual) in favour
> > of "higher" persistency.
> >
> > In a cellular network with mobile users, you will often find
> > a user whose performance is just fine most of the time.
> > Occassionally, it will hit nasty channel conditions, get
> > stuck for a short amount of time and get out of trouble
> > after not so long. Well, we do not want to break the
> > TCP connection while it is stuck, because we know that
> > it will get out of trouble. And statisctically, this happens often
> > (in the cellular environment)!
> >
> > We've run an extensive set of simulations investigating
> > interactions between the the cellular environment and TCP.
> > Indeed, having very persistent lower layer gives the best results
> > from the performance point of view. Having less than 3 or
> > 4 retransmissions always screws up things if the channel is
> > messy or if the system is highly loaded (many users per cell),
> > and infinite persistency gets decently mimicked by
> > something on the order of 5-10 retransmission on the RLP
> > layer.
> >
> > So, as an answer to your question
> > "What are you doing, exactly?", I offer the following:
> > In the cellular environment, with large number of users,
> > cumulative distribution functions expressing system throughput
> > and delay show significant improvements if the persistency on
> > the lower layers is high. So, we are doing a great deal here !
> >
> > Yes, we want persistency. It is not clear that we need to have
> > infinite persistency or always as high as 64 seconds, but having
> > only 3 retransmission (as in IS99) is suboptimal.
> > I actually think that variable persistency (as a function of
> > cellular channel conditions) may make sense.
> >
> > Zoran



From owner-pilc@lerc.nasa.gov  Wed Oct 18 16:19:29 2000
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Zoran,

 The reason I asked those questions is for this:

 1. It could be possible that for the TCP window size, MSS, RTT and error
    rate, the max throughput that TCP can achieve is greater than 59Kb/s.
    In that case, you *can* get away by spending more time at the link layer.
    This is because there is some room in the latency before TCP "feels" it.

2. If that is not the case, then it is good that the *extra*
   latency in recovering errored data has indeed paid off. However, there
   clearly needs to be a limit on the latency you are adding. TCP may just
   time out too.

3. Regarding the ARQ scheme itself, clearly the residual error decreases with
   increased retransmissions. Given a fixed number of retransmission, say 5,
   you could choose to retransmit, for e.g. in two ways:
   a. For every NAK the receiver sends retransmit and keep doing it for 
      5 NAKs. If after 5 attempts you still can't recover data, then you 
      leave it to TCP for recovery.
   b. The other option could be once you receive a NAK, send the retrans.
      twice and if you still receive another NAK, send the retransmission 
      3 times. If you still can't recover, then leave it
      up to TCP for recovery.
  The performance of these schemes depends on the TCP window size, RTT, MSS, 
  and ofcourse physical channel [correlated errors or uncorrelated errors]. 
  It is interesting how these many parameters come together to give a 
  sensible result in the end! We have summarized these results in our 
  VTC-2000 Spring  paper.
  
  Infact, as a result of these findings we left the recovery scheme at link
  layer [for CDMA-2000] up to the vendors and the telco operators. We did, 
  however, provide a default scheme in the RLP [Radio Link Protocol] 
  standard.

Kamesh


Zoran Kostic wrote:
> 
>    Kamesh,
> 
>   * The airlink is the one used in EDGE - thereby there is variability
>      of the rate between roughly 9 and 59 kbps.
> 
>   * The way I think of persistency on the link is the following: Attempt
>      to transmit the block no more than X times. So, I do not have the
>      concept of rounds, just plain number of  transmission attempts.
> 
>      On this layer I also use the mode 0 option, whereby if the channel
>      or interference are really bad, I do not even try to retransmit.
> 
>    Zoran
> 
> Kamesh Medepalli wrote:
> 
> > Zoran,
> >
> > Two questions:
> >
> > A. What is the air link data rate in your case?
> >
> > B. When you say higher persistency, are you also increasing the aggressiveness
> > of the link layer recovery? Meaning, if the 3 round recovery scheme
> > retransmits 1,2 and 3 times then a 5 round one retransmits 1,2,3,4 and 5 times?
> >


From owner-pilc@lerc.nasa.gov  Thu Oct 19 01:24:30 2000
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To: Zoran Kostic <kostic@research.att.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: pilc minutes for IETF 48 - Persistence
Cc: pilc@lerc.nasa.gov
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At 20:42 18.10.00, Zoran Kostic wrote:
>   * The way I think of persistency on the link is the following: Attempt
>      to transmit the block no more than X times. So, I do not have the
>      concept of rounds, just plain number of  transmission attempts.

Since (wireless) links have so different properties (frame size, link 
speed, latency, LL-RTT, ....), I suggest to speak of LL-ARQ persistency in 
terms of max. delay an IP packet may incur due to LL-ARQ before it is dropped.

Usually, it's straight forward to translate from "transmit the block no 
more than X times" to the mentioned delay.

IMO, that will make it much easier to talk about these things without 
requiring detailed knowledge of the link in question.

///Reiner




From owner-pilc@lerc.nasa.gov  Thu Oct 19 10:05:22 2000
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> Usually, it's straight forward to translate from "transmit the block no
> more than X times" to the mentioned delay.
> 
> IMO, that will make it much easier to talk about these things without
> requiring detailed knowledge of the link in question.
> 
> ///Reiner


I agree. I was just making a simple comment that within the latency bound 
there are different optimal speeds for different network level and wireless
link level scenarios.

Kamesh


From owner-pilc@lerc.nasa.gov  Fri Oct 20 15:37:01 2000
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Date: Fri, 20 Oct 2000 10:27:34 -0700 (PDT)
From: Bill Yeager <William.Yeager@Eng.Sun.COM>
Reply-To: Bill Yeager <William.Yeager@Eng.Sun.COM>
Subject: Re: pilc minutes for IETF 48
To: Vernon Schryver <vjs@calcite.rhyolite.com>
Cc: pilc@grc.nasa.gov
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Hi,

I don't want this conversation thread to distract from the TCP/IP thread which
is really what pilc is about. I'll note the following, and if you like, we can
continue offline:

| That's not going to happen until phones have reasonable keyboards (or
| equivalents), displays, and memory (e.g. attachments to PDA's).  When
| that happens, either radio modems will have lower latencies and higher
| throughputs or those wonderful applications will still not happen.
| The modern Internet would not exist if it were still based on 56k leased
| lines connecting IMPs ad TIPs.

There are phones with color and detachable keyboards much like a PDA. I-mode
has them. I too recal IMPs and TIPs having been at Stanford U. from 1974 until
1994. I am quite sure we can do a lot with even 9600  baud packet switched
networks. GPRS promises much more, and if we get 30-50kbps that is fine for
mobile phones.

As I said, applications will be different, and data demands also. Yes, there
are plans with 3G to connect laptops into these networks and do ftp. That is
where pilc comes in. 3G (WCDMA) is planned for next year in Japan. 

And, hey, lots of those wonderful Apps are already happening. They do not
depend on holding down the repeat key in a telnet session. Right, I had to
optimize that once to in my TIP code. Interactive games are a natural. Email
is easy. I've done that. Etc ...

| No matter how cool things seem to us, users do not tolerate the delays
| implied by several 1700 ms round trips for DNS followed by 3 more round
| trips to set up an HTTP/TCP/IP session just to get an airline schedule or
| restaurant menu so they can start to place an order.

Interesting. The experience on the AT&T Cellular-I CDPD network in the bay
area was not that bad(9600 baud was what was managed). DNS + HTTP/TCP/IP took
about 3 to 5 seconds on the average to set up. The connect was very quick.
There was good user feed back. I would lay in bed at night, turn on my palm
and read my email using HandWeb, with a servlet proxy, output presentation in
compact HTML, manage the session between me and the email backend (IMAP, LDAP,
SMTP). It was simple to use. Not much of a boot time, etc ... That was almost
two years ago.

Smart apps can manage latency when the net data being transferred is small(3
or 4 segments). 

So, like many of the arguments on this list we agree to disagree. For apps
this is in the spirit of let's see what really happens! I see lots of promise
and very, very soon. My hard fought battles in the trenches made me an
optimist "malgré tout!"

More offline ---

Bill
 



From owner-pilc@lerc.nasa.gov  Fri Oct 20 15:56:16 2000
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Date: Fri, 20 Oct 2000 12:24:36 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <200010201824.e9KIOap15783@calcite.rhyolite.com>
To: pilc@grc.nasa.gov
Subject: Re: pilc minutes for IETF 48
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> From: Bill Yeager <William.Yeager@eng.sun.com>

> ...
> | No matter how cool things seem to us, users do not tolerate the delays
> | implied by several 1700 ms round trips for DNS followed by 3 more round
> | trips to set up an HTTP/TCP/IP session just to get an airline schedule or
> | restaurant menu so they can start to place an order.
>
> Interesting. The experience on the AT&T Cellular-I CDPD network in the bay
> area was not that bad(9600 baud was what was managed). DNS + HTTP/TCP/IP took
> about 3 to 5 seconds on the average to set up. The connect was very quick.
> There was good user feed back. I would lay in bed at night, turn on my palm
> and read my email using HandWeb, with a servlet proxy, output presentation in
> compact HTML, manage the session between me and the email backend (IMAP, LDAP,
> SMTP). It was simple to use. Not much of a boot time, etc ... That was almost
> two years ago.

1700 ms was not my number.  I am not claiming that any wireless systems
have 1700 ms round trip delays, since I have no clue about such things
except what whose who know tell me.  I'm only reasoning from the stated
premises of others who say "X is required because Y," where Y is
consistently extremely long latency and very high error rates.


My fundamental point is that people can't have it both ways:

  1. (a) because radio links have 1700 ms delays and 30% packet loss rates
   (b) things must be done to and below TCP, such as 60 seconds of ARQ.
   (I think 30% loss rates were given by wireless experts in previous
   episodes of this discussion.  ~60 sec ARQ was recently mentioned.)

  2. the same links as in #1 will be used by 1,000,000,000's of happy end
   users of very interactive Internet applications that involve lots of
   data realsoonnow, notwithstanding the implications of 1700 ms delays.
   (1,000,000,000 is also not my number, but appeared in one of the
   earliest episodes.)

If you choose the premise #1(a), then you rule out the conclusions of #2.
If you believe in the conclusions of #2, then you cannot believe in the
premises of #1(a) or you believe they'll become invalid too soon to justify
doing the things repeatedly advocated in this mailing list.  If you don't
believe in #1(b), then you can't advocate #1(b).

It is possible to be sane and support #1(a) and #1(b), provided you give
up #2.  ARQ that retries for 64 seconds (again not my number) makes good
sense for some applications, although not for the applications that have
been advocated.  For example, a PDA with a mere 100 KByte could use such
a link to fetch email while the user is not paying attention, and later,
again in the background, send email responses.  In other words, such a
link can be used just as similar Telebit Trailblazers modem links were
used in the 1980s, and as far as users could tell, it would be as on-line
and interactive as talking to their ISP.

Unfortunately, there is too much telephant thinking in the wireless world
for such insights.  We are stuck with repetitions of #1 and #2, no matter
how crazy that seems.  Telcos and telco supplliers cannot admit the
possibility of smart customer premise equipment today any more than they
could during the Carterphone case 30+ years ago.  There is no hope of the
wireless telcos admitting that a telephone might be more than a very dumb
terminal.  Just as the repeated Ericson complaints and proposals of a few
years ago to disable PPP authentication were based on the belief that it
is impossible for CPE to compute anything in less than large parts of
minutes, the so called state of the art 3G phone is assumed to be little
smarter than a Western Electric 500C.



> Smart apps can manage latency when the net data being transferred is small(3
> or 4 segments). 

Yes, but applications that involve merely 3 or 4 TCP segments have nothing
to do with the issues at hand.  Again, if applications that put 4 or fewer
segments in flight at a time were relevant, then questions of ARQ for one
packet stalling packets or triggering TCP fast retransmissions would never
have come up.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Fri Oct 20 15:56:27 2000
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From: Bill Yeager <William.Yeager@Eng.Sun.COM>
Reply-To: Bill Yeager <William.Yeager@Eng.Sun.COM>
Subject: Re: pilc minutes for IETF 48
To: Vernon Schryver <vjs@calcite.rhyolite.com>
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Hi Vernon,

I think we agree 100% about the underlying TCP issues. I do some work on this
link layer stuff too from time-to-time too. 

Do look at this URL. It tells you what is happening in the App world. I do
believe that the phone viewed as a dumb text terminal is dead or very quickly
dying. We are back in 1983 just before the Mac hit the markets and killed off
the desktop terminal business. The big companies that did not listen are no
longer in the business. 

Change is in the air. It's super stuff:

http://www.washingtonpost.com/wp-dyn/articles/A28462-2000Oct17.html

Bill



From owner-pilc@lerc.nasa.gov  Fri Oct 20 16:39:40 2000
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Date: Fri, 20 Oct 2000 13:16:20 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Bill Yeager <William.Yeager@eng.sun.com>


> I think we agree 100% about the underlying TCP issues.

Probably.
More than that, I think the telco people have come around, except that
they're unwilling to let go of the premise that wireless phones will
be the solid state equivalents of WE 500C's but will still be usable
for all of the applications of a PC or PDA.


>                                                        I do some work on this
> link layer stuff too from time-to-time too. 
>
> Do look at this URL. It tells you what is happening in the App world. I do
> believe that the phone viewed as a dumb text terminal is dead or very quickly
> dying. We are back in 1983 just before the Mac hit the markets and killed off
> the desktop terminal business. The big companies that did not listen are no
> longer in the business. 

I don't think it's a coincidence that the Telebit Trailblazers appeared
about the same time.  The revolutionary idea of the TB was not the fancy
modem protocol, but putting a real computer and vast amounts (for that
era) of memory into a dial-up modem.  Modern "state of the art" wireless
telephones are equivalent to the 3270 emulators for PC's that were so
popular in the same era.  Many people still believed that 3270 still had
a future, or that the future of PC's had something to do with SDLC
connections to glass houses where all of the real computing would be done,
controlled, monitored, billed, etc.


> Change is in the air. It's super stuff:
>
> http://www.washingtonpost.com/wp-dyn/articles/A28462-2000Oct17.html

Yep.

Except that the notion of video phones is just as lame now as it was 25
years ago when AT&T demonstrated them.  The reason video phones are still
not used, after nearly 10 years of cheap or free (i.e. bundled) good,
working products, is that people don't like video calls most of the time.
When humans aren't close enough to smell each other, they prefer the
greater privacy and safety of yelling from a safe distance.  (see recent
literature about what happens to chimpanzees that get too close to the
wrong troop and aren't big enough to keep from being eaten)  In other
words, the people I work with really don't want to see me for the first
several hours of the day before I'm dressed, and vice versa.

See also recent talk about the usefulness of the MBONE to IETF meetings.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Mon Oct 23 13:38:04 2000
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From: "Sotiris Nikoletseas" <nikole@cti.gr>
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Subject:  Deadline approaching -- IPDPS Workshop on Mobile Computing
Date: Mon, 23 Oct 2000 19:31:24 +0300
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Please accept my apologies if you receive multiple
copies of this message.

+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
PRELIMINARY CALL FOR PAPERS
1st  International Workshop on
Parallel and Distributed Computing Issues in Wireless Networks and Mobile
Computing
IPDPS 2001 WORKSHOPS
Hyatt Regency San Francisco April 23-27, 2001
URLs:
http://www.ipdps.org
http://www.computing.edu.au/~kumar/ipdpswpim.html
++++++++++++++++++++++++++++++++++++++++++++++++

General Chair
Sajal K. Das, The University of Texas at Arlington, USA
Program  Co-Chairs
Mohan Kumar, Curtin University of Technology, Perth, Australia and The
University of
Texas at Arlington, USA
Paul Spirakis, Computer Technology Institute, Patras, Greece
*********************************************************
Publicity Co-Chairs
Kwan-Wu Chin, Motorola, Sydney, Australia
Sotiris Nikoletseas, Computer Technology Institute, Patras, Greece
*********************************************************
IMPORTANT DATES:
October 30, 2000            Workshop Paper Due
December 10, 2000       Notification of Acceptance/Rejection
January 15, 2001            Camera-Ready Paper Due

Mobile computing has emerged as an important area of computing today as we
move to the
next millennium. This has been made possible due to the tremendous and
continued growth
of wireless communications and network technology over the past decade,
providing
infrastructures for "anytime anywhere" access to distributed computing
systems and
information repositories. The mobility of users offers new challenges to
seamless
connectivity in a distributed, heterogeneous network of wireline and
wireless
components.  Several techniques and algorithms developed by the parallel and
distributed
computing community can be applied to solve typical wireless networks and
mobile
computing: routing, scheduling, load balancing, cache coherence, information
access, and
QoS provisioning problems. The objective of this workshop is to bring
together
technologists and researchers of international reputation in the areas of
Parallel and
Distributed Computing and Wireless Networks and Mobile Computing in order to
have a
forum for discussions, exchange of ideas and presentations.
Authors are invited to submit original unpublished manuscripts for this
workshop.
Accepted papers will be published in the proceedings (by IEEE CS Press) of
the IPDPS
workshops.

Topics of interest include (but are not limited to):
* Processor Architectures for Mobile and Wearable Computers
* Wireless networks in grid computing applications
* Routing in adhoc networks
* Parallel and distributed techniques for solving problems in mobile
computing
* Distributed techniques for QoS Provisioning in wireless networks
* Distributed caches in mobile computing environments
* Caching, prefetching, pushcaching and replication to enhance information
access in
wireless networks
* Distributed wireless sensor networks
* Routing and location independent information access
* Scheduling issues in mobile computing
* Parallel/distributed algorithms for mobile computing systems
* Distributed Wireless multimedia systems
* Active Networks


SUBMISSION GUIDELINES:
Authors are invited to submit summaries of original unpublished research to
one of the
Workshop Chairs. All submissions will be reviewed by referees. Summaries
should not
exceed ten  (10) single-spaced pages of text using 12-point size type on 8.5
x 11 inch
pages. References, figures, tables, etc. may be included in addition to the
ten pages of
text. The summary should include an abstract of approximately 100 words.
The
corresponding author is requested to include in a cover letter: (1) complete
postal
address; (2) email address; and (3) Phone/fax numbers.

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

Submissions may be by hard copy or by email.  Electronic submissions are
preferred and
should be sent to kumar@cs.curtin.edu.au or spirakis@cti.gr. Authors should
submit the
cover page in ASCII form followed by a PostScript (level 2) file and make
sure that it
will print on a PostScript printer that uses 8.5 x 11 inch (US letter size)
paper.  For
hard copy submissions, authors should submit seven (7) hard copies of their
paper along
with the cover letter to Mohan Kumar or Paul Spirakis.
All manuscripts will be reviewed.  Manuscripts   must be received by October
30, 2000.
Submissions received after the due date or exceeding the length    limit may
not be
considered. Notification of review decisions will be mailed by December 10,
2000.
Camera-ready papers are due January 15, 2001.  Proceedings will be available
at the
Conference.
Workshop Program Co-Chairs
Mohan Kumar
School of Computer Science
Curtin University of Technology
GPO Box U 1987, Perth WA 1987 AUSTRALIA
Fax: +61 8 9266 2819 E-mail: kumar@cs.curtin.edu.au

Paul Spirakis
Director and Senior Researcher
Computer Technology Institute
61 Riga Fereou Str., 26221 Patras, Greece  Patras, Greece
Fax: +30 61 993 973, +30 61 222086 E-mail: spirakis@cti.gr
++++++++++++++++++++++++++++++++++++++++
Technical Program Committee
Somprakash Bandopadhyay, PWC Global, Calcutta,  India
Stefano Basagni, University of Texas at Dallas, USA
Maurizio Bonuccelli University of Pisa, Italy
Azzedine Boukerche, University of North Texas, USA
Luca Cardelli, Microsoft Research, Cambridge, UK
Kwan-Wu Chin, Motorola Australia
Kee Chaing Chua, National University of Singapore
Marco Conti, Italian National Research Council, Italy
Samir Das, University of Cincinatti, USA
Xiaohua Jia, City University of Hong Kong
Jan van Leeuwen University of Utrecht, The Netherlands
John Lui, The Chinese University of Hong Kong
Marios Mavronicolas, University of Cyprus, Cyprus
Sotiris Nikoletseas, Computer Technology Institute, Greece
Cristina Pinnotti, Italian National Research Council
Don Sannella  University of Edinburgh, Scotland
Martha Steenstrup, BBN Technologies, USA
Mukesh Singhal, Ohio State University, USA
Nitin Vaidya, Texas A&M University, USA
Jie Wu, Florida Atlantic University, USA
Yongbing Zhang  University of Tsukuba, Japan
Albert Zomaya University of Western Australia, Perth Australia
----------------------------------------------------------------------------
--------------------







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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		: Performance Enhancing Proxies
	Author(s)	: J. Border, M. Kojo, J. Griner, 
                          G. Montenegro, Z. Shelby
	Filename	: draft-ietf-pilc-pep-04.txt
	Pages		: 45
	Date		: 23-Oct-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
context of the Internet.

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From owner-pilc@lerc.nasa.gov  Thu Oct 26 18:42:19 2000
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From: Alhussein Abouzeid <hussein@ee.washington.edu>
To: pilc@grc.nasa.gov
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Subject: TCP over lossy links
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Hi,

Enclosed is an abstract of a paper (submitted to ToN, Jan. 2000)
re analytical modeling of TCP performance in the presence of 
wireless links, i.e. random, correlated-loss environments. 
This is an extension to our Infocom 2000 paper (available at URL below)  
that attempts to faithfully capture the interplay
between TCP's congestion control algorithm, the channel induced loss behavior and
the dynamics of the transmit queue. 

The primary contribution of the work is modeling the impact of
all the following on TCP performance: 

1. The two modes of packet loss detection: triple duplicates as
well as timeouts and the binary exponential backoff;
2. The variation of the RTT due to queuing;
3. The two congestion avoidance phases of TCP;
4. Channel induced i.i.d and/or correlated loss in addition to possible
packet loss due to buffer overflow.

The papers can be downloaded at

http://students.washington.edu/abouzeid/research.htm

As usual, all comments are greatly appreciated.

Best regards,

Alhussein A. Abouzeid.
 ---------------------------------

 	A Unified Approach to Modeling TCP Performance Over a Lossy Link


                 Alhussein A. Abouzeid, Sumit Roy and Murat Azizoglu
                    Dept. of Electrical Engineering
                    Univ. of Washington Box 352500
                    Seattle, WA 98195


Abstract:

A new, comprehensive analytical framework for modeling the long-term
performance of a single TCP session in the presence of random packet loss
is  presented.
A Markovian model is developed for both memoryless channels (independent,
identically distributed packet loss) and channels with memory (correlated
packet loss) modeled by a two state  continuous time 
Gilbert-Elliott model.
The analytical results (throughput) are validated against those obtained
from ns simulator and shows very good agreement for a wide
range of network  parameters - i.e., low (LANs) and
high (WANs) bandwidth-delay products. Further, the  
throughput for the i.i.d. loss model is found to be relatively 
insensitive to the probability density function (p.d.f) of the loss 
inter-arrival process.
For channels with memory, an empirical rule is presented 
for categorizing the impact of channel dynamics (fading rate) on TCP
performance.







From owner-pilc@lerc.nasa.gov  Mon Oct 30 14:27:26 2000
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From: Segura Ramon <Ramon.Segura@nc3a.nato.int>
To: "'pilc@grc.nasa.gov'" <pilc@grc.nasa.gov>
Subject: ACK decimation using WFQ
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Hello,

Just a few lines in the context of asymmetric networking and forward
throughput limitation caused by ACK congestion. I would like to share a
recent experience on using router standard mechanisms for boosting the
forward throughput beyond the limit established by the available return
channel rate. It's basically a simple "ACK decimation" technique which,
although not being optimum, has proven to increase significantly the forward
throughput for "disadvantaged" receive nodes that use very narrowband and
long-latency satellite links to support the return channel (at 2400 bps or
4800 bps). Using IP/DVB-S on the satellite forward path, the RTT when using
those return channels can be a high as 1500 ms (Inmarsat Mini-M terminal)
and 3000 ms (UHF DAMA satcom radios). The client and server platforms under
test support large windows, Fast Retransmit / Fast Recovery with Selective
Acknowledgement (SACK), extended time stamps, and the 4-segment start-up, as
provided by the OnNet Kernel 4.0 (for Win95/98) and by Windows 2000 TCP/IP
stacks. The target bulk transfer throughput on the forward channel is 2
Mbps.

To cope with the ACK congestion limitation imposed by the narrowband & long
delay return path, we take advantage of the cumulative property of the TCP
ACK's and decimate the return ACK flow by tail-dropping them at the router
serial interface. For that we use cisco's weighted-fair-queuing (WFQ), which
also ensures fairness among the multiple return ACK streams The longer the
ACK queue (i.e. the higher the programmed WFQ drop threshold), the more
noticeable is the effect of TCP forward flow self-clocking/slaving to the
ACK return rate. By setting the WFQ drop threshold to 1, we get one ACK
transmitted out of every K (K-1 packets being dropped off the queue). In our
case, the return channel (2400 bps) can only serve four to five ACK packets
per second (each being 66 bytes long). As expected, the traffic is bursty
during the throughput build-up (i.e. slow-start), but goes and remains
steady when the pipe gets full, or when the generic traffic-shaping limit
for that flow has been reached. Under this configuration, throughputs of 1
Mbps and higher can be achieved and maintained during FTP or HTTP downloads
(@2400 bps return rate, 1500 ms RTT). Without ACK decimation, the return
channel rate would limit the throughput to less than 100 kbps (assuming
delayed ACK's as implemented by Winsock).

Since WFQ creates a queue for each "conversation" or TCP connection,
multiple FTP or HTTP sessions can be established concurrently. Each object
request will go into a different queue, and its ACKs will be "tail-dropped"
independently (and 'fairly'). Despite of the ACK's being severely dropped on
an existing connection, the SYN segments for establishing new connections
will still make it thru additional queues, with little additional delay
('ping' shows an increase of some 100-200 ms compared to an idle channel).
Moreover, the return channel can also be used to upload data on to the
server from the remote client (e.g. FTP), at very low speeds, without
disturbing the downstream-related ACK flow (in that case, the ACK decimation
rate obviously increases). When introducing errors on the forward channel,
we could see SACK behaving quite well, despite the decimation of the return
ACKs (and SACK) at the upstream router interface. Under this ACK-dropping
condition, more (than 3) DUP ACK's or SACK need to be transmitted by the
client to get the threshold number into the server.

Hope this trick may be of help,

ramon

----------------------------------------------------------------------
Ramon Segura
NATO Consultation, Command and Control Agency, NC3A 
Communications Systems Division, SATCOM Branch
The Hague, Netherlands

e-mail: ramon.segura@nc3a.nato.int
Tel. + 31 (0)70 314 2432 / 2448 (Lab)
Fax.+ 31 (0)70 314 2176
----------------------------------------------------------------------



From owner-pilc@lerc.nasa.gov  Mon Oct 30 19:00:10 2000
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Date: Mon, 30 Oct 2000 22:56:52 +0000
From: Mahesh Sooriyabandara <mahesh@erg.abdn.ac.uk>
Organization: University of Aberdeen
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To: Segura Ramon <Ramon.Segura@nc3a.nato.int>
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Hi Ramon,

Segura Ramon wrote:

> Hello,
>
> Just a few lines in the context of asymmetric networking and forward
> throughput limitation caused by ACK congestion. I would like to share a
> recent experience on using router standard mechanisms for boosting the
> forward throughput beyond the limit established by the available return
> channel rate. It's basically a simple "ACK decimation" technique which,
> although not being optimum, has proven to increase significantly the forward
> throughput for "disadvantaged" receive nodes that use very narrowband and
> long-latency satellite links to support the return channel (at 2400 bps or
> 4800 bps). Using IP/DVB-S on the satellite forward path, the RTT when using
> those return channels can be a high as 1500 ms (Inmarsat Mini-M terminal)
> and 3000 ms (UHF DAMA satcom radios). The client and server platforms under
> test support large windows, Fast Retransmit / Fast Recovery with Selective
> Acknowledgement (SACK), extended time stamps, and the 4-segment start-up, as
> provided by the OnNet Kernel 4.0 (for Win95/98) and by Windows 2000 TCP/IP
> stacks. The target bulk transfer throughput on the forward channel is 2
> Mbps.

This sounds quite interesting. We have been doing some work on asymmetric
networks for quite a sometime. Although it seems not like the "optimum"
solution (as you have mentioned) the technique you are proposing seems to be
quite different from other available techniques. To me it looks like a straight
forward in the sense that it uses off the shelf equipment with some parameter
tuning. No need of any sort of end system modifications. I agree with the fact
that this networking technology in your case ( which is again new to apply
these technique) can be benefited by a proper ACK modification scheme.

Although you claim from your experiments that this method works well. I think
we have to raise few questions here. See below.....

> To cope with the ACK congestion limitation imposed by the narrowband & long
> delay return path, we take advantage of the cumulative property of the TCP
> ACK's and decimate the return ACK flow by tail-dropping them at the router
> serial interface. For that we use cisco's weighted-fair-queuing (WFQ), which
> also ensures fairness among the multiple return ACK streams The longer the
> ACK queue (i.e. the higher the programmed WFQ drop threshold), the more
> noticeable is the effect of TCP forward flow self-clocking/slaving to the
> ACK return rate. By setting the WFQ drop threshold to 1, we get one ACK
> transmitted out of every K (K-1 packets being dropped off the queue). In our
> case, the return channel (2400 bps) can only serve four to five ACK packets
> per second (each being 66 bytes long). As expected, the traffic is bursty
> during the throughput build-up (i.e. slow-start), but goes and remains
> steady when the pipe gets full, or when the generic traffic-shaping limit
> for that flow has been reached. Under this configuration, throughputs of 1
> Mbps and higher can be achieved and maintained during FTP or HTTP downloads
> (@2400 bps return rate, 1500 ms RTT). Without ACK decimation, the return
> channel rate would limit the throughput to less than 100 kbps (assuming
> delayed ACK's as implemented by Winsock).
>

Although use of Fair Queuing scheme in the return path is a nice way to ensure
fair queuing ( Some earlier and we have also shown that in our experiments)
"drop from tail" ( not the best way ?) while having a small threshold can leads
to some problems (drop of data packets flow in the return path). As a solution
indeed a larger queue will be required and that can lead to reduce the forward
throughput.

>
> Since WFQ creates a queue for each "conversation" or TCP connection,
> multiple FTP or HTTP sessions can be established concurrently. Each object
> request will go into a different queue, and its ACKs will be "tail-dropped"
> independently (and 'fairly'). Despite of the ACK's being severely dropped on
> an existing connection, the SYN segments for establishing new connections
> will still make it thru additional queues, with little additional delay
> ('ping' shows an increase of some 100-200 ms compared to an idle channel).

Yes I believe because of the per-flow queuing probability of loosing SYN can be
very low.

>
> Moreover, the return channel can also be used to upload data on to the
> server from the remote client (e.g. FTP), at very low speeds, without
> disturbing the downstream-related ACK flow (in that case, the ACK decimation
> rate obviously increases). When introducing errors on the forward channel,
> we could see SACK behaving quite well, despite the decimation of the return
> ACKs (and SACK) at the upstream router interface. Under this ACK-dropping
> condition, more (than 3) DUP ACK's or SACK need to be transmitted by the
> client to get the threshold number into the server.

Even in our experiments we saw that in the presence of losses large number of
DUP ACKs flow in the return path can lead a time out( after FR ). Dropping DUP
ACKs larger than a threshold ( to allow fast retransmission) is necessary to
avoid that ( Some have shown this earlier and we have seen it as well).

I think I agree with your your results with SACK as well. Recent experiments we
did have shown that in a lossy path back to back SACKs can lead to a time out
same as the back to back DUP acks in TCP with out SACK. So we do see the need
of dropping SACK to gain any significant benefit by using it. We are trying to
model this by using a suitable threshold. It will be interesting to see the
effects with SACK deletion.

Any body with some experience on this. I like to hear ....


Regards,


Mahesh.

>
>
> Hope this trick may be of help,
>
> ramon
>
> ----------------------------------------------------------------------
> Ramon Segura
> NATO Consultation, Command and Control Agency, NC3A
> Communications Systems Division, SATCOM Branch
> The Hague, Netherlands
>
> e-mail: ramon.segura@nc3a.nato.int
> Tel. + 31 (0)70 314 2432 / 2448 (Lab)
> Fax.+ 31 (0)70 314 2176
> ----------------------------------------------------------------------



From owner-pilc@lerc.nasa.gov  Mon Oct 30 19:21:05 2000
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From: "Dawkins, Spencer" <Spencer.DAWKINS@fnc.fujitsu.com>
To: "'PILC Mailing List'" <pilc@grc.nasa.gov>
Subject: PEP version 04 now available for WG Last Call
Date: Mon, 30 Oct 2000 17:25:29 -0600
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Version 04 of the Performance Enhancing Proxies Internet-Draft is now available at http://www.ietf.org/internet-drafts/draft-ietf-pilc-pep-04.txt.

The full document details are:
        Title : Performance Enhancing Proxies 
        Author(s) : J. Border, M. Kojo, J. Griner, 
                          G. Montenegro, Z. Shelby 
        Filename : draft-ietf-pilc-pep-04.txt 
        Pages : 45 
        Date : 23-Oct-00 

This draft is a PILC work item, and is slotted as an Informational RFC. It has not been through a previous working group last call. 

We would like to hold a two-week working group last-call on this document, ending on Tuesday, November 14. Please review this document as appropriate, and send questions or comments to the PILC mailing list.

"Yes, it's done" is also an acceptable - in fact, welcome - comment.

Spencer, for Spencer, Mark, and Aaron


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Date: Tue, 31 Oct 2000 13:35:52 +0000
From: Mahesh Sooriyabandara <mahesh@erg.abdn.ac.uk>
Organization: University of Aberdeen
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Subject: Asymmetry draft ?
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Dear Hari,

I was following the PILC mailing list for quite a sometime and wonder
what is going to happen to asymmetry draft. Last thread on this topic
showed that there are lot of disagreements to the draft. For the same
reason I believe that it needs to be revised rather than let it
disappear.

It will be nice to have a document describing the Asymmetry problem and
solutions/(best practices) since there are some disadvantage networks
which can be benefited from it. The draft should address all (  as much
as possible) different asymmetric networks and different techniques. The
former draft may lack this generality. To achieve this contributions
from everybody dealing with different asymmetric networks will be
needed. There are some commercial products out there which uses some
mitigation techniques to improve TCP performance.

We have done series of experiments on this topic and I have seen quite a
lot of techniques ( mainly ACK modification) proposed by various
researchers. Generally each technique is addressing a certain type of
networking technology. FYI following is a list of techniques I have seen
so far:

1) TCP Header Compression    [a]
2) ACK filtering/ Sender Adaptation    [b]
3) ACK Congestion Control    [b]
4) ACK compaction    [c]
5) ACK Companding/ Expanding    [d]
6) Window Prediction Mechanism    [e]
7) Acknowledgement based on Cwnd Estimation    [f]
8) ACK decimation    [h]

[a] Jacobson, V. ,Compressing TCP/IP Headers for Low Speed Serial Lines,
RFC 1144,1990, IETF.

[b] Balakrishnan H, Padmanathan V.N. Katz R. "The effects of asymmetry
on TCP performance" Mobicom Budapest Hungary 1997, ACM/IEEE.

[c] Nihal Samaraweera "Return Link optimization for Internet service
provision using DVB-S networks ACM CCR June 1999.

[d] Johansson G.L , Shakargi, H. Kanljung, C. Kullander, J.
‘ACKNOWLEDGEMENT Compression Rev B’, Technical Report December 1999.

[e] A Calveras, Linare, J. Paradells, J. “Window Prediction Mechanism
For Improving TCP in Wireless Asymmetric Links”, Globecom 98.

[f] Ming-Chit, I.T, Jinsong, D. Wang, W. “Improving TCP Performance Over
Asymmetric Networks” ACM Communication Computer Review, July 2000.

[g] R. Segura “ Asymmetric Networking Techniques For Hybrid Satellite
Communications” NC3A, Technical Note 810. August 2000.

Above techniques have been analysed ( or can be used) in different
asymmetric networking technologies. (like Cable networks , packet radio,
ADSL, DVB-S with return dial-up, Wireless cable modem with dial up
return, DVB satellite forward with narrowband Inmarsat return)

It is obvious that lot of work has been done in this area during past
few years. This may be a correct time to start discussing this to make
use of them and get things correct with the Asymmetry draft. I believe
there are lot of researchers who are willing to put their input in to
this.

Regards,

Mahesh.



