From owner-pilc@lerc.nasa.gov  Tue Apr  4 19:23:46 2000
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Date: Tue, 04 Apr 2000 16:17:30 -0500
From: Phil Neumiller <neumille@cig.mot.com>
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Folks,

Disclaimer:  I have not read through all 800+ messages in the
PILC archive.  That will take some time.  There are some subjects
of interest to Mobile-IP folks, that is why I cross posted.

I have read through the PILC draft.  It suggests many different 
proven methods for link layer adaptation to IP.  I found errors
of omission rather than of commission.  What could be done
is to expand the current draft (or introduce a new draft) that
specifically addresses problems faced by the implementors of
"wireless Internet".  This document was very TCP specific and
the wireless world is also concerned about RTP among other
things.

I believe that wireless links are sufficiently different (not
to mention increasingly prevalent) that they deserve their own
set of "standard" link layer adaptation techniques.

I suggest that we (enhance the existing) or prepare a draft that
defines "Canonical Wireless Link Layer Adaptation for IP".  In this
draft the important issues of QoS, AAA, and handover could be
addressed in an abstract way, suitable and useable by Wireless LANS
(femto, pico, nano-cellular) and conventional macro-cellular (i.e.
2G, 2.5G, 3G).  I am also hoping that the proposed Bluetooth BOF
planned for Pittsburgh could be generalized in scope to include the
more generic notion of "personal area networking" rather than being
just a link adaptation fest for Bluetooth.  Other link layers such
as 802.11a, 802.11b, HomeRF, Hyperlan, and Hyperlan2 must be
considered world players as well and the more commonality the
better.

I would also hope that the MWIF contribution could be combined with
a previously proposed BOF for open base station protocols.  It is my
hope that the IETF can help unify wireless access by providing
specific link layer hooks that are universal across diverse radio
access networks.

Does this fall into the PILC or Mobile-IP scope or somewhere else?
I am not sure.

Thanks,

Phil QA2757@email.mot.com
847-632-4118


From owner-pilc@lerc.nasa.gov  Tue Apr  4 20:21:57 2000
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Date: Wed, 05 Apr 2000 09:08:33 +1000
From: Andrew Worsley <epaanwo@asac.ericsson.se>
Subject: Re: [MOBILE-IP] Comments on PILC draft
In-reply-to: Message from Phil Neumiller <neumille@CIG.MOT.COM>
 "of Tue, 04 Apr 2000 16:17:30 EST." <38EA5BEA.D50EACE5@cig.mot.com>
To: Phil Neumiller <neumille@cig.mot.com>
Cc: rohc@cdt.luth.se, pilc@lerc.nasa.gov
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  I am not posting this follow up to mobileIP but to the two groups that I
  think it is most relevant to, rohc and pilc. Hand over issues are being
  discussed in the mobileIP group but link layer stuff isn't. My comments are
  about IP over link layer issues. I am quoting the article in full because
  it was not originally posted to these groups.

> Folks,
> 
> Disclaimer:  I have not read through all 800+ messages in the
> PILC archive.  That will take some time.  There are some subjects
> of interest to Mobile-IP folks, that is why I cross posted.
> 
> I have read through the PILC draft.  It suggests many different
> proven methods for link layer adaptation to IP.  I found errors
> of omission rather than of commission.  What could be done
> is to expand the current draft (or introduce a new draft) that
> specifically addresses problems faced by the implementors of
> "wireless Internet".  This document was very TCP specific and
> the wireless world is also concerned about RTP among other
> things.
> 
> I believe that wireless links are sufficiently different (not
> to mention increasingly prevalent) that they deserve their own
> set of "standard" link layer adaptation techniques.
> 
> I suggest that we (enhance the existing) or prepare a draft that
> defines "Canonical Wireless Link Layer Adaptation for IP".  In this
> draft the important issues of QoS, AAA, and handover could be
> addressed in an abstract way, suitable and useable by Wireless LANS
> (femto, pico, nano-cellular) and conventional macro-cellular (i.e.
> 2G, 2.5G, 3G).  I am also hoping that the proposed Bluetooth BOF
> planned for Pittsburgh could be generalized in scope to includethe
> more generic notion of "personal area networking" rather than being
> just a link adaptation fest for Bluetooth.  Other link layers such
> as 802.11a, 802.11b, HomeRF, Hyperlan, and Hyperlan2 must be
> considered world players as well and the more commonality the
> better.

   These issues are being addressed by the Robust header compression working
   group.

      see http://www.ietf.org/html.charters/rohc-charter.html
> 
> I would also hope that the MWIF contribution could be combined with
> a previously proposed BOF for open base station protocols.  It is my
> hope that the IETF can help unify wireless access by providing
> specific link layer hooks that are universal across diverse radio
> access networks.

   This sounds like a worthy goal and I believe some what compatible with
   rohc groups agenda which covers giving guidelines to link layer designers
   and defining compression schemes for specific link technologies.
> 
> Does this fall into the PILC or Mobile-IP scope or somewhere else?
> I am not sure.
> 
> Thanks,
> 
> Phil QA2757@email.mot.com
> 847-632-4118

  The ROHC group's work so far is mainly for RTP and not so much support is
  provided for TCP.

  Correct me if am wrong but the ROHC schemes assume that the packets sent
  have little or no error detection in them but that is how things are wanted
  for Voice over IP over the air as they would rather have corrupted data
  than have the whole packet dropped.

  The draft-jonsson-robust-hc-04.txt working document spells out having
  checksums only of the headers and presumably only dropping packets whoose
  header is corrupted (when it's compressed to a few bytes this is likely to
  be rare). Presumably corrupt packets with good headers are better than no
  packets.

  One question I have about that is that UDP for IPv6 requires mandatory
  checksums. Does this imply that the implementation would still try to use
  the failing packets? It's not clear to me that exactly how this is going to
  work.

  Finally I would like to suggest that a reliable profile transfer where
  there is bidirectional links be produced using a simple Nack
  (Negative acknowledgement) scheme like that described by Phil Karn
  in his paper:

  	http://people.qualcomm.com/karn/papers/usenix.ps.gz
    
  I suggest segmentation using small segments and Nacks.

  There must be a large amount of theory on this and some formulas about it.
  I did some maths and found that surprisingly low segmentation numbers for
  even quite modest BER values.

  I assumed packets were segemented into segements of size f with with two
  bytes of overhead per segment (1 byte CRC and 1 byte count?),
  ignored the Nack traffic (as it's in the other direction?) and got
  for the average amount transmitted (in the one direction) for a packet of
  length D:

  T = ((D/f)+1)*(f + c)/p

    where f = segement size (I called it fragment)
          c = size of segement overhead 
	  p = probability of segment transmission, assuming a constant BER.
	      i.e.  p = 1 - (1 - BER)**(f+c)

    and I assume the last segment is truncated to only contain the remaining
    data, if not you need an upper bound on the segment size say 50 bytes.

    For c = 2, D = 640
    Optimal values for f were:

    BER = .01 	f = 4 p = 0.617290  efficiency = 2.430
    BER = .001 	f = 16 p = 0.865825 efficiency = 1.299
    BER = 0.0001 f = 50 p = 0.959251 efficiency = 1.084
    BER = 1e-05 f = 150 p = 0.987914 efficiency = 1.026

      (efficiency is how much data is transmitted in the one direction
      divided by the size of the original data)

   Admittedly my maths may be off but although surprisingly low, I think it's
   about right. 

    I also came up with the following approximation for the ideal segment
    size :

    	sqrt(c**2 + c/8e)

	which depends only on the segment overhead and the BER value (e).

    The formula appears to work pretty good from my simulation data.
    
    I think that the ROCCO system could use this for TCP support.  I think
    the ideal fragment size calculation could be done on the number of
    packets lost over a period. The result from this calculation could
    be used to update the size of the segement size used.  This would
    allow the system to migrate to an appropriate value for the link
    dynamically as the environment changes. With f values < 200 this
    might only take a few seconds between adjustments.

    It needs a lot of work to turn the idea into a practical scheme, but the
    basic principle seems sound to me. e.g. Issues of frame size, handling
    sequence count wrap around and so on.

    Admittedly a lot of assumptions are made about the distribution of error
    bits but the general principal of using moderate segment sizes and Nacks
    seems to be a very robust approach. i.e. Erring on the side of small
    segments allows people to use large MTUs on their link (to avoid IPv4
    fragmentation problems) and minimise the damage that long or short error
    bursts will do.

    This scheme might actually be useful with some UDP systems. Possibly
    the sending of a packet over this profile as opposed to another more
    lossy profile but lower latency profile could be chosen on a packet stream
    basis?  i.e. Using QoS or protocol TCP vs UDP.

    Finally the selection of what to send over such a reliable profile needs
    to be judged based on the latency of the Nacks. If segments are small
    enough and Nacks can be returned quickly enough then the scheme would
    be practical for time constrained media like voice.

    	Andrew Worsley


From owner-pilc@lerc.nasa.gov  Wed Apr  5 00:00:49 2000
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To: Andrew Worsley <epaanwo@asac.ericsson.se>
From: Mikael Degermark <micke@CS.Arizona.EDU>
Subject: Re: [rohc] Re: [MOBILE-IP] Comments on PILC draft
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Andrew Worsley wrote:

>   This sounds like a worthy goal and I believe some what compatible with
>   rohc groups agenda which covers giving guidelines to link layer designers
>   and defining compression schemes for specific link technologies.

The "guidelines for link layer designers" document developed by ROHC is ONLY 
intended to provide guidelines on what properties the link layer must and should
have in order to run ROHC header compression over it. It is emphatically 
NOT a general-purpose guideline document for wireless link-layer designers.

>  Correct me if am wrong but the ROHC schemes assume that the packets sent
>  have little or no error detection in them

No, that is not an assumption. We do, however, want (some instance of) the ROHC
schemes to work IF there isn't strong error detection on the link layer.

>  but that is how things are wanted
>  for Voice over IP over the air as they would rather have corrupted data

>  than have the whole packet dropped.

Some voice coders want that, yes. Not all, but some that some people care about. 

>  The draft-jonsson-robust-hc-04.txt working document spells out having
>  checksums only of the headers and presumably only dropping packets whoose
>  header is corrupted (when it's compressed to a few bytes this is likely to
>  be rare). Presumably corrupt packets with good headers are better than no
>  packets.

1. The ROCCO checksum covers the original header, before compression. Its
primary purpose is to allow dependable verification of correct decompression.

2. Using that checksum to detect corruption of compressed headers as well will
weaken it for its original purpose. That may or may not be acceptable depending on
a) how resilient the scheme needs to be against production of erroneous headers, and 
b) the error properties of the link. 

>  One question I have about that is that UDP for IPv6 requires mandatory
>  checksums. Does this imply that the implementation would still try to use
>  the failing packets? It's not clear to me that exactly how this is going to
>  work.

1. Even if UDP prevents the packet to be delivered to the application, it will
enhance performance it the header decompression module sees it. 

2. The reason that IPv6 mandates using the UDP checksum is that IPv6 does
not have a header checksum, so in order to ensure correct delivery for UDP, IPv6
had to mandate turning the UDP checksum on. That gives some assurance that
the UDP payload is delivered to the app it is intended for.

3. Applications that want to have corrupted payloads delivered to them, 
clearly cannot use UDP with an enabled checksum to do so. Thus they cannot use
UDP over IPv6. 

4. A transport protocol that optionally allows such applications to receive corrupted

payloads is UDP Lite, but it remains to be seen whether it is acceptable to the IETF.
UDP Lite might be presented to the transport area WG and will be discussed on its
list sometime before the next IETF meeting. 

This is clearly out of scope for the ROHC WG. People who want to discuss issues
related to UDP and UDP Lite should do so on the Transport Area WG mailing list:

General Discussion: tsvwg@ietf.org 
To Subscribe:  tsvwg-request@ietf.org 
In Body: subscribe email_address 
Archive: ftp://ftp.ietf.org/ietf-mail-archive/tsvwg/ 

>  Finally I would like to suggest that a reliable profile transfer where
>  there is bidirectional links be produced using a simple Nack
>  (Negative acknowledgement) scheme like that described by Phil Karn
>  in his paper:
>
>  	http://people.qualcomm.com/karn/papers/usenix.ps.gz
>    
>  I suggest segmentation using small segments and Nacks.

For TCP and most of the non-audio and non-video UDP, I agree that 
the link should be almost error-free. This will be achieved 
by some some combination of (channel) FEC and ARQ. Most links that
ROHC is explicitly intended for can be run in such a mode.
The price paid for this is delay, but that is ok for TCP and some UDP. 

NOTE that the added delay does not primarily come from the ARQ scheme,
but from the increased interleaving needed to make the FEC work well. 

>  There must be a large amount of theory on this and some formulas about it.
>  I did some maths and found that surprisingly low segmentation numbers for
>  even quite modest BER values.

< deletion of some theory...>

>    BER = .01 	f = 4 p = 0.617290  efficiency = 2.430
>    BER = .001 	f = 16 p = 0.865825 efficiency = 1.299
>    BER = 0.0001 f = 50 p = 0.959251 efficiency = 1.084
>    BER = 1e-05 f = 150 p = 0.987914 efficiency = 1.026
>
>      (efficiency is how much data is transmitted in the one direction
>      divided by the size of the original data)
>
>   Admittedly my maths may be off but although surprisingly low, I think it's
>   about right.

I am not surprised. This suggests that performance-enhancing ARQ schemes
over the link should use frames significantly smaller than typical TCP segments. 

<more stuff deleted> 

>    I think that the ROCCO system could use this for TCP support.  I think
>    the ideal fragment size calculation could be done on the number of
>    packets lost over a period. The result from this calculation could
>    be used to update the size of the segement size used.  This would
>    allow the system to migrate to an appropriate value for the link
>    dynamically as the environment changes. With f values < 200 this
>    might only take a few seconds between adjustments.

It seems like a tall order for the ROHC WG to develop an ARQ scheme given that
the radio links explicitly mentioned in the charter already have such schemes. But, 
if the WG so desires, we can ask the ADs to add it to our charter. I would hesitate
to do so before we're done with the tasks currently at hand, though.


>    It needs a lot of work to turn the idea into a practical scheme, but the
>    basic principle seems sound to me. e.g. Issues of frame size, handling
>    sequence count wrap around and so on.

>    Admittedly a lot of assumptions are made about the distribution of error
>    bits but the general principal of using moderate segment sizes and Nacks
>    seems to be a very robust approach. i.e. Erring on the side of small
>    segments allows people to use large MTUs on their link (to avoid IPv4
>    fragmentation problems) and minimise the damage that long or short error
>    bursts will do.
>
>    This scheme might actually be useful with some UDP systems. Possibly
>    the sending of a packet over this profile as opposed to another more
>    lossy profile but lower latency profile could be chosen on a packet stream
>    basis?  i.e. Using QoS or protocol TCP vs UDP.

I agree that different packet streams need different service from the 
radio system. The dividing line is not primarily UDP vs TCP, but rather
interactive audio & video    vs.  traditional TCP & UDP. I also expect that 
some games and other interactive apps will prefer low delay to error-freeness. 

>    Finally the selection of what to send over such a reliable profile needs
>    to be judged based on the latency of the Nacks. If segments are small
>    enough and Nacks can be returned quickly enough then the scheme would
>    be practical for time constrained media like voice.

For the links explicitly mentioned in the ROHC chater, the delay budget for 
interactive voice will not tolerate using such a nack scheme. That is the primary
reason for doing the ROHC work in the first place.

Moreover, if the delay budget allowed using the nack scheme, it would almost
certainly be a better overall system trade-off to improve the performance
of the channel coding by adding interleaving than to deploy the nack scheme.

Mikael Degermark

>    	Andrew Worsley
>---
>Mailing list for Robust Header Compression WG
>Archive: http://www.cdt.luth.se/rohc/
> 


From owner-pilc@lerc.nasa.gov  Wed Apr  5 12:48:42 2000
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To: epaanwo@asac.ericsson.se, neumille@cig.mot.com
Cc: rohc@cdt.luth.se, pilc@lerc.nasa.gov
Subject: RE: [rohc] Re: [MOBILE-IP] Comments on PILC draft
Date: Wed, 5 Apr 2000 10:43:16 -0500 
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> -----Original Message-----
> From: EXT Andrew Worsley [mailto:epaanwo@asac.ericsson.se]
> Sent: Tuesday, April 04, 2000 6:09 PM
> To: Phil Neumiller
> Cc: rohc@cdt.luth.se; pilc@lerc.nasa.gov
> Subject: [rohc] Re: [MOBILE-IP] Comments on PILC draft
> 
......
>   I assumed packets were segemented into segements of size f 
> with with two
>   bytes of overhead per segment (1 byte CRC and 1 byte count?),
>   ignored the Nack traffic (as it's in the other direction?) and got
>   for the average amount transmitted (in the one direction) 
> for a packet of
>   length D:
> 
>   T = ((D/f)+1)*(f + c)/p
> 
>     where f = segement size (I called it fragment)
>           c = size of segement overhead 
> 	  p = probability of segment transmission, assuming a 
> constant BER.
> 	      i.e.  p = 1 - (1 - BER)**(f+c)

Just want to point out a typo here. If p is the probability that
a segment is transmitted successfully, p should equal to
(1 - BER)**(f+c). The p above actually gives the probability of
a failed transmission.

Zhigang Liu

> 
>     and I assume the last segment is truncated to only 
> contain the remaining
>     data, if not you need an upper bound on the segment size 
> say 50 bytes.
> 
>     For c = 2, D = 640
>     Optimal values for f were:
> 
>     BER = .01 	f = 4 p = 0.617290  efficiency = 2.430
>     BER = .001 	f = 16 p = 0.865825 efficiency = 1.299
>     BER = 0.0001 f = 50 p = 0.959251 efficiency = 1.084
>     BER = 1e-05 f = 150 p = 0.987914 efficiency = 1.026
> 
>       (efficiency is how much data is transmitted in the one direction
>       divided by the size of the original data)
> 
......
> 
>     	Andrew Worsley
> ---
> Mailing list for Robust Header Compression WG
> Archive: http://www.cdt.luth.se/rohc/
> 


From owner-pilc@lerc.nasa.gov  Thu Apr  6 15:04:56 2000
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From: "Jan Kullander (ERV)" <jan.kullander@erv.ericsson.se>
To: pilc@grc.nasa.gov
Subject: SLOW draft and TCP Auto-tuning
Date: Thu, 6 Apr 2000 17:00:55 +0200 
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Hi!

As summarised in section 2.5, the auto-tuning technique adjusts the buffer
size based on the utilised window. This entails increased advertised window
until packets are lost. This thus contradicts section 2.3 that recommends
the advertised window to be limited to avoid buffer overflow in the queue to
the bottleneck link. We believe the latter recommendation to be valid for
slow links.

We are however much in favour of TCP auto-tuning since the bw*delay product
could be considerable over some wireless links and a mobile node could roam
between different access links. We would thus like the auto-tuning section
to remain somewhere, somehow in the output from PILC but it is not needed
for the slow links.

Sorry for the late comment.

Haider Shakargi & Jan Kullander

PS
It would be nice to combine auto-tuning with an optional max window
associated with interfaces...
DS



From owner-pilc@lerc.nasa.gov  Thu Apr  6 16:07:01 2000
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From: "Falk, Aaron" <afalk@PanAmSat.com>
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Cc: "Spencer Dawkins (E-mail)" <spencer_dawkins@yahoo.com>,
        "'Gabriel Montenegro'" <gab@sun.com>,
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Subject: Extended (again): Working group last call for ERROR and SLOW
Date: Thu, 6 Apr 2000 14:43:45 -0400 
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I've been requested to extend the last call on SLOW and ERROR to allow some
additional commentary. Let's give things another couple of weeks until 4/21.

Again, the drafts can be found at:
http://www.ietf.org/internet-drafts/draft-ietf-pilc-error-03.txt
http://www.ietf.org/internet-drafts/draft-ietf-pilc-slow-03.txt

Thanks,

--aaron


From owner-pilc@lerc.nasa.gov  Thu Apr  6 22:27:51 2000
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To: "pilc@grc.nasa.gov" <pilc@grc.nasa.gov>
Subject: ERROR draft comments...
From: Jamshid Mahdavi <mahdavi@novell.com>
Date: 06 Apr 2000 17:46:25 -0700
In-Reply-To: "Falk, Aaron"'s message of "Thu, 6 Apr 2000 14:43:45 -0400"
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I've finally gotten around to reading Error, and I have a few
additional comments on it:

> 2.1 Slow Start and Congestion Avoidance [RFC2581]
> 
>    Slow Start and Congestion Avoidance [RFC2581] are essential to
>    the Internet's stability. These mechanisms were designed to
>    accommodate networks that didn't provide explicit congestion
>    notification.

I'd recommend changing the wording slightly here: "These mechanisms
were designed to accomodate networks that didn't provide any form of
explicit congestion control".  (we aren't just talking about ECN
in this sentence.  And ECN in fact uses SS and CA...)

Section 2.2:

>    NewReno [RFC2582] apparently does help a sender
>    better handle partial ACKs and multiple losses in a single
>    window, but at this point is not recommended due to its
>    experimental nature.  Instead, SACK (Selective Acknowledgements)
>    is the preferred mechanism.

Both NewReno and SACK are becoming popular.  I'd say that the bad
thing about NewReno is that it takes N RTTs to recover if there are N
losses.  The good thing about it is that it requires only one side of
the connection to implement the protocol.  SACK appears to be a much
better solution.  Unless there is objection, I'd say recommend SACK
generally and NewReno (in addition, not instead) for situations where
SACK won't work (this would normally be because the person reading
this would only be able to install a solution on the transmitting end
of the connection).  Here is new text:

   Recommendation: Implement Fast Retransmit and Fast Recovery at this
   time. This is a widely-implemented optimization and is currently at
   Proposed Standard level. [RFC2488] recommends implementation of
   Fast Retransmit/Fast Recovery in satellite environments.  In cases
   where SACK (see next section) can not be enabled for both sides of
   a connection, NewReno [RFC2582] may be used by TCP senders to
   better handle partial ACKs and multiple losses in a single window.

Section 2.3:

>    In low-speed, high error-rate environments (for example, the
>    wireless WAN environment), TCP windows are much smaller, and burst
>    errors must be much longer in duration in order to damage multiple
>    segments. Accordingly, the complexity of SACK may not be
>    justifiable, unless there is a high probability of both burst
>    errors and congestion.

This logic doesn't make a lot of sense to me.  I suppose that if your
error rate is high, and you are constantly timing out, SACK is not
going to make any difference.  But if there is congestion too
(i.e. the total error rate is even higher), SACK is going to help even
less.

I'm not an expert, but I would think that in wireless environments,
changes in surroundings (i.e. someone walks in front of the wireless
card) would cause a high probability of correlated errors.  This would
imply to me multiple losses per window, exactly the thing that SACK
is good for.  So if things mostly *don't* suck, SACK will help for the
times when they do.  

If things mostly *do* suck, not much is going to help.

Here is a proposed change/addition:

   In low-speed, high error-rate environments (for example, the
   wireless WAN environment), TCP windows are much smaller, and burst
   errors must be much longer in duration in order to damage multiple
   segments. Accordingly, the complexity of SACK may not be
   justifiable.

   On the other hand, if error rates are generally low but
   occasionally increase due to interference, TCP will have the
   opportunity to increase its window to larger values.  When
   interference occurs, multiple losses within a window are likely to
   occur.  In this case, SACK would provide benefits in speeding the
   recovery and preventing unnecessary extra reduction of window size.

Section 4:

I promised Aaron a note on "explicit notification of transmission
error" based on a discussion Phil and Aaron and I had at IETF.  I'll
try to emit that tomorrow.  It probably doesn't impact the draft once
(except maybe to change this one sentence), but it ought to spark some
interesting discussions.

Section 4.1:

Playing something of the devil's advocate here...  I don't see that
the recommendations are impacted at all by whether connections are
short or long.  Even in the current world with lots of short flows,
sometimes in parallel, we still want people to do SS/CA, FR/FR and
SACK.  I don't think there is any question that it would be of
additional benefit if parallel TCP connections shared congestion state
either through 2140 or ECM.  I don't necessarily think I'd argue that
they are "crucial to the long-term stability of the Internet" (we've
lasted this long, after all!), but they should definitely be
beneficial to performance.

The recent CCR paper by Eggert, Heidemann, and Touch on Ensemble TCP
showed some of the benefits of combining congestion state for http
traffic.  I'd kind of like to see the same work done in an "error"
environment to see what effect it would have on performance...

--Jamshid






From owner-pilc@lerc.nasa.gov  Fri Apr  7 06:00:37 2000
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Date: Fri, 07 Apr 2000 11:05:04 +0200
From: Omar Elloumi <Omar.Elloumi@alcatel.be>
Organization: Alcatel Corporate Research Center
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Jamshid, all
I'm not following this track but just a small comments on NewReno, see
below

Jamshid Mahdavi wrote:

> I've finally gotten around to reading Error, and I have a few
> additional comments on it:
>
>
> Section 2.2:
>
> >    NewReno [RFC2582] apparently does help a sender
> >    better handle partial ACKs and multiple losses in a single
> >    window, but at this point is not recommended due to its
> >    experimental nature.  Instead, SACK (Selective Acknowledgements)
> >    is the preferred mechanism.
>
> Both NewReno and SACK are becoming popular.  I'd say that the bad
> thing about NewReno is that it takes N RTTs to recover if there are N
> losses.  The good thing about it is that it requires only one side of
> the connection to implement the protocol.  SACK appears to be a much
> better solution.  Unless there is objection, I'd say recommend SACK
> generally and NewReno (in addition, not instead) for situations where
> SACK won't work (this would normally be because the person reading
> this would only be able to install a solution on the transmitting end
> of the connection).  Here is new text:
>

NewReno Takes *at most* N RTTs to recover, because since you're
sending two packets for each incoming ACK it may be the case that
you retransmit two lost packet for the same incoming ACK.
I would also say that SACK is *generally* a much better solution. I did
some simulations two years ago and I found some scenarios in which
NewReno outperforms SACK. The reason is simple: SACK is very
aggressive and it may be the case that for heavy congestions, retransmitted

packets can be lost leading to a timeout. While with NewReno which is less
aggressive the probability of loosing a retransmitted packets may be
in some scenarios less than with SACK. Unfortunately I don't have
anymore the simulation results. But I think this makes sens and it
should also be pointed out in some papers.

Kind regards.
--
____________________________________________________________
Omar Elloumi

Alcatel Research - Network Architecture - Traffic Technology

Francis Wellesplein 1, 2018 Antwerp, Belgium

Phone: + 32 3 240 78 33
Fax:   + 32 3 240 99 32
http://www.alcatel.com/crc/
____________________________________________________________

Opinions expressed are those of the sender and do not reflect Alcatel
policy or agreement




From owner-pilc@lerc.nasa.gov  Fri Apr  7 11:32:19 2000
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-----Original Message-----
From: Markku Kojo [mailto:kojo@cs.Helsinki.FI] 
Sent: Friday, April 07, 2000 9:19 AM
To: afalk@PanAmSat.com
Subject: PILC minutes


Aaron,

please find attached the pilc wg meeting minutes that Jim 
and I gathered. Just got an email from Jim saying he is 
still in Australia, so  I am sending these to you. Some 
comments were probably not captured, but I think most
are there.

Rgds,

/Markku


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Aaron opened the meeting at 2:15PM on 3/28/00
Went over the agenda, with no comments.

Defered discussion of assymetric draft, since authors are=20
not at this meeting

WG status:
----------

-Submitted "Long Thin Networks" to IESG in October 99               =20
-Should submit PEP draft as informational to IESG by Nov 00
-ASYM document needs more discussion on the list=20
  -Aaron: ASYM will go away if there is consensus that these        =20
   mechanisms are not needed                                      =20
-Should submit LINK draft to IESG by Nov 00
-SLOW and ERROR are expected to become BCPs during spring 2000,   =20
 now in last call                                                 =20

John Border gave the status on the PEP draft:
---------------------------------------------

-The -01 draft missed the DC meeting deadline but was submitted after
 DC meeting. For the -00 we got valuable comments, but for -01 we   =20
 didn't get any commnets; maybe as a result of being late?          =20

-Looking for input in the following sections
	protocol booster mechanisims
	ACK filtering and regeneration
	Partial ACK mechanisims
	Others??

-Some additional example environments where PEPs are in use may be =20
 added, but only if they illustriate new types or mechanisms not   =20
 already covered                                                   =20

-Need help on completing the section on WAP                        =20

-Mainly looking for input on implications of using PEPS            =20
	End-to-end failure diagnostics
	Multi-homing environments
	QoS transparency
	Security implications
	Others?

	Need further terminology refinment

Comments:

??:  Can get more information on the WAP section from=20
     the IAB worksop BOF and plenary WAP discussion

John?: Some problems between asym and pep draft as what happens=20
       to ASYM is not known
John Border: Leave specific information in asym draft, if it=20
             doesn't go away
Aaron Falk:  ASYM will go away if the mechanisims are not real and
             needed PEP document should only contain mechanisims=20
             that are used or are going to be used, and not just=20
             academic mechanisims

John Border: We have to draw the line somewhere

??:           Are web caches considered peps?
John Border:  There is a fine line to whether they are peps or not
Markku Kojo:  They are considered PEPs in this doc only when they are
              related to specific link characteristic(s), i.e. they =20
              use some specific mechanisms to mitigate problems due =20
              to link behavior                                      =20

Phil Karn gave the status on the link draft:
--------------------------------------------

-Goal of document is to give advice to designers of subnetworks that=20
 intend to carry IP

-Emphasis on end-to-end model, awareness of inter-layer interactions,
 and eliminating redundancy (between what IP already provides and what
 link designers are providing)

-01 draft submitted for nov99 IETF
	Changes include:  address delays, MTUs, error recovery,=20
        connection-oriented subnetworks, framing, Bandwidth on Demand

-02 was submitted for this IETF       =20
	Reflects current versions of mailing list comments         =20
	Added congestion and flow control QoS, multicast, etc.


-Changes to the current draft (since -01)                          =20

	refinement of all sections esp those on impacts on net asym=20
		-end to end philosophy                             =20
		-all sections related to fundamental link design   =20

	added sections
		-QoS management and flow control have some redundent
		 parts compression
		-mobility
		-broadcast/multicast
=09
	Sections needing work
		-QoS & Congestion Control
		-Delay and delay variance characteritics
		-routing & mobility    =20
                   (how much routing should IP provide and how much
                     should the subnetwork provide?)
		-multicast support
		-QoS management

	next steps=20
		-revisit draft
		-seek advice and agree on end to end (QoS ?)      =20

Jamashid:  How much delay can a subnetwork add?  If many people=20
           all take the same advice then the delay can addup
Phil Karn:  No one is adding gratituas delay; give us hooks to tell
            whether the pkt is delay sensitive


??:         Connection oriented subnetworks section is biased might
            want to look at MPLS (MPLS introduces some
            connection-orientinesh to nets)
Phil Karn:  The section advises one to avoid connection oriented
            subnetworks, some of these topics should be added to=20
            congestion management section

Aaron Falk: This is a really important document, and now is the time
            to add new sections or add/correct current sections,=20
            since the document is maturing
Phil Karn:  The more people that add to the document, the more=20
            credibility it will have

Jamashid:   Is there some way to add a short summary of key points
Phil Karn:  Thinks this is a good idea, since some sections are long

??:         What about FDDI switches that fragment IP packets
Phil Karn:  Maybe add this as an example


Gabe Montenagro gave the status of the SLOW and ERROR drafts
------------------------------------------------------------

-Most of the changes are cosmetic
-Changed from explicit corruption notification to explicit=20
 transmission error notification in error draft
-New section were added in both drafts to separate:
	-recommended mechanisms
	-topics for further work
=09
-Both drafts are currently in last call till april 7


-Issues with SLOW draft (as seen by the authors)
	-Need to redo receive and router buffer size example in
	 section 2.3 and address the issues related to the router
	 buffer size more clearly

	-Should we reccomend RED? (we think yes)
	-Should we recomment ECN?
	-Need to beef up MTU section, maybe steal some of link text

-Issues with ERROR draft (as seen by the authors)
	-Need editing on section 1.2 (relationship with LINK draft)
	-Section 2.2 on AIMD is not accurate
	-Section 2.2, why cwnd remains small:
	  -#2 is inaccurate
	  -#3 must be removed, cannot be generalised, applies
	   LTNs only
	-Need to update section 4.0=20
		-on SACK-EXT as it is no onger experimental
		-Delayed DUPACKs section needs clarification
	=09
-Some changes are desirable in both drafts
	Do we need a second last call?
        Aaron: if the changes are substantial


Comments:

Phil Karn:  The wording may not carry over from link document,=20
            since audiences are different (on whether duplicating
            text, using pointers to other pilc docs)

Aaron and John:  Said that we should make it clear that the topics
                 for further work in appendicies are not part of the
                 recommendations
Gabe Montenagro: Maybe a paragraphs stating that the appencicies are=20
                 not the emphasis of the document

Geoff Huston:  Should say active queue management is important and
               use RED as an example of this


------_=_NextPart_000_01BFA09C.7421F2EC--


From owner-pilc@lerc.nasa.gov  Fri Apr  7 15:10:44 2000
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Subject: Re: FW: PILC minutes from IETF-47
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From: Jamshid Mahdavi <mahdavi@novell.com>
Date: 07 Apr 2000 11:04:48 -0700
In-Reply-To: "Falk, Aaron"'s message of "Fri, 7 Apr 2000 10:16:05 -0400"
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Additional notes on ERROR.  I noticed in the meeting comments about
errors in section 2.2 and sure enough there are some.  Here is some
proposed revised text:

Old text:

   After a fast retransmit, a sender invokes the fast recovery
   [RFC2581] algorithm, whereby it invokes congestion avoidance,
   but not slow start from a one-segment congestion window. This also
   saves time.

   It's important to be realistic about the maximum throughput that
   TCP can have over a connection that traverses a high error-rate
   link. Even using Fast Retransmit/Fast Recovery, the sender will
   halve the congestion window each time a window contains one or
   more segments that is lost, and will re-open the window by one
   additional segment for each acknowledgement that is received. If
   a connection path traverses a link that loses one or more segments
   during recovery, the one-half reduction takes place again, this time
   on a reduced congestion window - and this downward spiral will
   continue until the connection is able to recover completely without
   experiencing loss.

   In general, TCP can increase its congestion window beyond the
   delay-bandwidth product. In links with high error rates, the
   TCP window may remain rather small for long periods of time
   due to any of the following reasons:

...

New Text:

   After a fast retransmit, a sender invokes the fast recovery
   [RFC2581] algorithm.  This allows it to remains in congestion
   avoidance, rather than falling back to slow start from a
   one-segment congestion window.  This also saves time.  TCP
   implementations which follow these algorithms as specified in
   RFC2581 are genericly referred to as Reno TCP implementations.

   It's important to be realistic about the maximum throughput that
   TCP can have over a connection that traverses a high error-rate
   link.  Even using Fast Retransmit/Fast Recovery, the sender will
   halve the congestion window each time a window contains one or more
   segments that is lost, and will only re-open the window by one
   additional segment for each subsequent window of data that is sent
   and acknowledged.  For most Reno TCP implementations, multiple
   losses within a single window are very likely to result in a
   retransmission timeout, resulting in further performance
   degradation.  Further, if a connection path traverses a link that
   loses segments in several consecutive windows, the one-half
   reduction takes place each time once for each window of data which
   includes a loss.  This causes a downward spiral which will continue
   until the connection returns to an error-free state, at which time
   it can begin to increase the congestion window at the rate of one
   additional segment per window of data sent and acknowledged.

   In general, TCP is able to increase its congestion window beyond
   the delay-bandwidth product for the connection path.  However, for
   paths with high error rates, TCP's congestion window may remain
   rather small for long periods of time due to one or more of the
   following reasons:

...








From owner-pilc@lerc.nasa.gov  Fri Apr  7 15:11:04 2000
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Subject: ERROR: Apologies for doing these items one at a time...
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From: Jamshid Mahdavi <mahdavi@novell.com>
Date: 07 Apr 2000 11:16:45 -0700
In-Reply-To: "Falk, Aaron"'s message of "Fri, 7 Apr 2000 10:16:05 -0400"
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I hope this doesn't too thoroughly annoy anyone.

Suggested text for items 2 and 3 under section 2.2 in ERROR.

Old Text:

      2. TCP's congestion avoidance strategy is additive-increase,
      multiplicative-decrease, which means that if additional
      errors are encountered during recovery, the effect on the
      congestion window is a "downward spiral" - "reduce by 50 percent,
      recover by 20 percent, reduce by 50 percent due to the next
      error ...".

      3. Often small socket buffers are recommended with high
      error-rate links in order to prevent the RTO from inflating.

New Text:

      2. As described above, TCP's congestion avoidance strategy
      treats packet loss as an indication of congestion.  In links
      with frequent errors (which result in packet corruption/loss),
      this can effectively clamp the window at a low value.

      3. In some cases, links with high error rates are also
      low-bandwidth links.  People may mistakenly believe that by
      reducing socket buffer sizes, they will improve overall
      performance by preventing the RTT from rising (due to
      buffering).  See the <SLOW> document for more details on these
      practices.

I've got several comments on SLOW which this point is tied to.  I plan
to write these up today also.

--Jamshid


From owner-pilc@lerc.nasa.gov  Fri Apr  7 17:14:17 2000
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To: pilc <pilc@grc.nasa.gov>
Subject: ICMP EaTEN message...
From: Jamshid Mahdavi <mahdavi@novell.com>
Date: 07 Apr 2000 13:23:41 -0700
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[Now that it has a cute name, we'll just *have* to implement it :-)].

This is a summary of some discussion we had at the IETF on the concept
of an ICMP message for "Explicit Transmission Error Notification".  I
couldn't think of a good word that starts with "A" but "EaTEN" is
clearly the correct sentiment.

The basic problem, which has been discussed quite a bit in pilc, is
for wireless links.  It is impossible to completely avoid transmission
errors, and people in the WG have argued that in some cases providing
suitable amounts of FEC to reduce errors to a suitable low level is
too costly (for whatever reason).

As outlined in ERROR, losses due to transmission errors have a bad
impact on TCP.  TCP must assume that these losses are caused by
congestion (because there is no information to the contrary), and
therefore back off its congestion window.

The proposed solution, therefore, is to create a mechanism to
communication this information about corruption back to the sending
TCP.  I assume that the receiving wireless node computes a link-layer
checksum and determines that the packet has been corrupted.  Normally
at this point, it would simply drop the packet as it is no longer
forwardable.  In addition to dropping the packet, the proposal is to
send an ICMP EaTEN message back to the origin, in effect saying that
the packet was "eaten".  (The ICMP includes the original IP and TCP
headers).

The sending TCP would respond to this ICMP message in two ways.
First, it would immediately perform a fast retransmit of the corrupted
segment.  Second, because it knows that the loss was *not* due to
congestion, it would *not* reduce cwnd.  This would eliminate the
major detriment (to TCP) of having an error-prone link in the path.

(For security, the considerations discussed in TSVWG for Phil's ICMP
kick-tcp message would be need to be included.  I think this means
matching the ICMP messages to actual packet losses in order to
eliminate various forms of DoS attacks and probably also using some
type of detection for floods of these messages.  This also means that
you don't actually immediately retransmit the corrupted packet, but
still wait until you get 3 dupacks -- treating the ICMP message in an
advisory fashion.)

Note that with this solution, the error-prone hop can be anywhere in
the path; it is not restricted to being an endpoint.  And it works
fine whether there are 1 or 5 or 20 wireless hops in the path.

Cons to the EaTEN ICMP concept:

1. Phil posed the following question: if all this message is going to
   accomplish is to have TCP retransmit the packet, why not just have
   the link layer do this via an ARQ mechanism?  This would get the
   data there quicker, and be more efficient for the network.

2. The other problem is that if the part of the original segment which
   was corrupted included the packet headers, the ICMP might be sent
   to the wrong host or connection.

Since the IETF, I came up with a counter-argument for Phil's point.
If the link layer can accomplish the ARQ within 3 packets of the
original transmission, there is no problem.  However, if it takes
longer than this, it will be out of order by enough to trigger fast
retransmit anyway, which mean that the ARQ had no useful effect.

There are some things we can talk about doing to TCP to make this less
of a problem (basicly delaying fast retransmit under certain
well-defined circumstances), but for long delay links such as
satellites, this might not be possible to overcome.

I don't have a good answer for point 2.  I suppose the security checks
mentioned above would provide some insurance that a misrouted message
didn't do any harm, however.

I pose all of this to the list because I've heard suggestions from
various fronts that such a thing would be useful.  I've also heard a
lot of comments to the effect that if we just design the subnetworks
correctly in the first place, this doesn't need to be a problem.
Personally, I am entirely on the fence and am usually swayed by the
arguments of whoever I happen to have spoken with most recently.

--Jamshid







From owner-pilc@lerc.nasa.gov  Fri Apr  7 20:12:09 2000
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To: pilc <pilc@grc.nasa.gov>
Subject: Comments on SLOW...
From: Jamshid Mahdavi <mahdavi@novell.com>
Date: 07 Apr 2000 16:11:08 -0700
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Note:  This is recommending a lot of changes.  I'm happy to write all
of the text, but I wanted to run the ideas through the group first for
comments before doing all of the writing.

--J

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

I propose that we do some reorganization on section 2 of SLOW.  In
particular, 2.3 and 2.5 kind of go together, so let's move the MTU
stuff before the TCP stuff, and make it section 2.3.

I like the MTU content, but I'd change the phrase "effectively turns
off delayed ACKs".  We TCP weenies tend to equate delayed ACK with ACK
per 2 packets.  We should be more clear, and say that 

    using an MTU that takes more than 200 milliseconds to transmit
    will cause an ACK to be generated for every segment, rather than
    one per two segments as is normal with most TCP implementations.

(The gripe is that the ACKs are still delayed by 200 msec, so they
should still be called delayed ACKs.  They do still have the other
benefits of delaying, such as piggybacking ACKs and data).

Under the TCP sections, I suggest that we reorganize them as follows:

2.4  Interactions with TCP Congestion Avoidance [RFC2581]

*  Keep the description of CA.  

*  However, the current description confuses rate and window.  I'd
   like to clarify this.

*  Explain the issues of increased RTT due to queuing in this section
   also.

2.4.1  Recommendations for network elements connected to SLOW links.

*  This section would include a recommendation to limit queue lengths
   to reasonable values.  This would be minimum of 3 segments (for
   FR/FR), and a maximum based on the delay imposed by the queue.  For
   single user links (i.e. modems), a bandwidth-delay product is
   probably a good value to recommend here.

*  Also include recommendations for RED and ECN in this section.
   Interesting question is whether RED makes sense for single user
   devices (modem pools) with short queues.  This might actually be a
   case where drop-tail is better (or at least equivalent -- how long
   does it take to go over the max threshold if the queue is only 3
   packets?)

2.4.2  Recommendations for TCP implementations

*  In general, I'd say that the management responsibility for SLOW
   connections should be applied in the following order:
   1.  Network queues
   2.  TCP sender
   3.  TCP receiver
   4.  Applications and Sysadmins

   In other words, whatever performance problems we have should first
   be fixed by making the network more suitable to work correctly with
   a standard TCP implementation.  These are the queuing
   recommendations above.

   If the network is messed up (i.e. has queues that can fill up to 3
   seconds worth of data, or no queues at all), the sender TCP is in
   the best position to detect this and compensate for it.  Senders
   can also implement shared congestion state and persistent
   connections which would help because you could remember certain
   parameters from one flow to the next.

   If all of these fail, then the receiver TCP might try to do
   something using the receiver window.

   We should try to make it clear that, only as an absolute last ditch
   effort, should applications or operating system parameters be used
   to make improvements to performance.

My logic on this last point is as follows.  First, applications might
actually have a legitimate application reason to want to set socket
buffer sizes.  This is the actual point in having settable socket
buffer sizes: managing system resources, *not* managing network
resources.  So I don't like the recommendation that apps not set
socket buffers.

Second, a single system default is a pretty poor way to manage network
resources.  For example, if I dial in on my laptop sometimes, and
other times hook up via a 100 Mb/s PC card, what should I do?  Do I
use different values when my modem negotiates 56 kb/s than when it
negotiates 9.6 kb/s?

If we are going to recommend that OS vendors add a "dialup/LAN"
dialog, lets instead recommend that they actually fix the problem in a
way that handles both the SLOW and FAST needs.  This means changes in
the TCP stack along the same lines as autotuning, but more
specificly targetted to this problem.

Section 3: Summary of recommendations.

   Implementors should consider the possibility that a host will be
   directly connected to a low-speed link when choosing default TCP
   receive window sizes.

We should replace this recommendation with something suitable based on
the above discussion.


   Application developers should consider the possibility that an
   application will be used on a host that is directly connected to a
   low-speed link, before increasing the TCP receive window size beyond
   the default for TCP connections used by this application.

I'd replace this with:

   Application developers should not attempt to manually manage
   network bandwidth using socket buffer sizes.  Only in very rare
   circumstances will an application actually know both the bandwidth
   and delay of a path and be able to choose a suitably low (or high)
   value for the socket buffer size to obtain good network
   performance.

I'd also be in favor of recommending the implementation of new APIs
which let applications find out information about how much network
bandwidth (or window or RTT or whatnot) is currently available from
the transport layer.  Perhaps something like what ecm is toying
with...

--Jamshid



From owner-pilc@lerc.nasa.gov  Fri Apr  7 21:33:42 2000
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From: Bora Akyol <akyol@pluris.com>
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Subject: Re: ICMP EaTEN message...
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One concern:

How does a wireless network router know who a corrupted packet belongs to
in a shared medium network so that it can send this message.

Very likely, what this will boil to is sending a message to all TCP
sessions that it has seen recently on this link.

This will work OK for wireless nodes that are transmitting to other nodes,
but for wireline nodes, the sheer number of them that are out there may
"eat" away at the benefits of such a message.

You are much better of giving some sort of piggy back feedback between
wireless nodes involved in the connection to their TCPs if they happen to
be at the same place.

For example:

if you have wireless router A, and wireless node B, then if B got some
sort of feedback on the link quality than it can give explicit feedback to
hosts that it has TCP connections to. If on the other hand two nodes
happen to be connecting through a wireless link, then we are out of luck.

In a nutshell, I think that this idea even though it sounds useful at
first, is really not that useful.

Regards

Bora Akyol

ps. You may want to check on some old research done in the late 70s, early
80s on wireless packet data networks if you have not done so already.

Jamshid Mahdavi wrote:

> [Now that it has a cute name, we'll just *have* to implement it :-)].
>
> This is a summary of some discussion we had at the IETF on the concept
> of an ICMP message for "Explicit Transmission Error Notification".  I
> couldn't think of a good word that starts with "A" but "EaTEN" is
> clearly the correct sentiment.
>
> The basic problem, which has been discussed quite a bit in pilc, is
> for wireless links.  It is impossible to completely avoid transmission
> errors, and people in the WG have argued that in some cases providing
> suitable amounts of FEC to reduce errors to a suitable low level is
> too costly (for whatever reason).
>
> As outlined in ERROR, losses due to transmission errors have a bad
> impact on TCP.  TCP must assume that these losses are caused by
> congestion (because there is no information to the contrary), and
> therefore back off its congestion window.
>
> The proposed solution, therefore, is to create a mechanism to
> communication this information about corruption back to the sending
> TCP.  I assume that the receiving wireless node computes a link-layer
> checksum and determines that the packet has been corrupted.  Normally
> at this point, it would simply drop the packet as it is no longer
> forwardable.  In addition to dropping the packet, the proposal is to
> send an ICMP EaTEN message back to the origin, in effect saying that
> the packet was "eaten".  (The ICMP includes the original IP and TCP
> headers).
>
> The sending TCP would respond to this ICMP message in two ways.
> First, it would immediately perform a fast retransmit of the corrupted
> segment.  Second, because it knows that the loss was *not* due to
> congestion, it would *not* reduce cwnd.  This would eliminate the
> major detriment (to TCP) of having an error-prone link in the path.
>
> (For security, the considerations discussed in TSVWG for Phil's ICMP
> kick-tcp message would be need to be included.  I think this means
> matching the ICMP messages to actual packet losses in order to
> eliminate various forms of DoS attacks and probably also using some
> type of detection for floods of these messages.  This also means that
> you don't actually immediately retransmit the corrupted packet, but
> still wait until you get 3 dupacks -- treating the ICMP message in an
> advisory fashion.)
>
> Note that with this solution, the error-prone hop can be anywhere in
> the path; it is not restricted to being an endpoint.  And it works
> fine whether there are 1 or 5 or 20 wireless hops in the path.
>
> Cons to the EaTEN ICMP concept:
>
> 1. Phil posed the following question: if all this message is going to
>    accomplish is to have TCP retransmit the packet, why not just have
>    the link layer do this via an ARQ mechanism?  This would get the
>    data there quicker, and be more efficient for the network.
>
> 2. The other problem is that if the part of the original segment which
>    was corrupted included the packet headers, the ICMP might be sent
>    to the wrong host or connection.
>
> Since the IETF, I came up with a counter-argument for Phil's point.
> If the link layer can accomplish the ARQ within 3 packets of the
> original transmission, there is no problem.  However, if it takes
> longer than this, it will be out of order by enough to trigger fast
> retransmit anyway, which mean that the ARQ had no useful effect.
>
> There are some things we can talk about doing to TCP to make this less
> of a problem (basicly delaying fast retransmit under certain
> well-defined circumstances), but for long delay links such as
> satellites, this might not be possible to overcome.
>
> I don't have a good answer for point 2.  I suppose the security checks
> mentioned above would provide some insurance that a misrouted message
> didn't do any harm, however.
>
> I pose all of this to the list because I've heard suggestions from
> various fronts that such a thing would be useful.  I've also heard a
> lot of comments to the effect that if we just design the subnetworks
> correctly in the first place, this doesn't need to be a problem.
> Personally, I am entirely on the fence and am usually swayed by the
> arguments of whoever I happen to have spoken with most recently.
>
> --Jamshid



From owner-pilc@lerc.nasa.gov  Mon Apr 10 11:56:02 2000
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From: Jamshid Mahdavi <mahdavi@novell.com>
Date: 10 Apr 2000 07:34:15 -0700
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Bora Akyol <akyol@pluris.com> writes:

> One concern:
> 
> How does a wireless network router know who a corrupted packet belongs to
> in a shared medium network so that it can send this message.
> 
> Very likely, what this will boil to is sending a message to all TCP
> sessions that it has seen recently on this link.

This was not what I had in mind.  Rather, I thought that the router
could simply look at the IP header in the corrupted packet and hope
that the source IP address was not corrupted.  My assumption (not
knowing a lot about wireless) was that there is a reasonably good
chance that four bytes early in the packet would have made it through
OK.  However, perhaps this is not a good assumption...

> You are much better of giving some sort of piggy back feedback between
> wireless nodes involved in the connection to their TCPs if they happen to
> be at the same place.
> 
> For example:
> 
> if you have wireless router A, and wireless node B, then if B got some
> sort of feedback on the link quality than it can give explicit feedback to
> hosts that it has TCP connections to. If on the other hand two nodes
> happen to be connecting through a wireless link, then we are out of
> luck.
> 
> In a nutshell, I think that this idea even though it sounds useful at
> first, is really not that useful.

I'm not completely understanding your argument here, but it does sound
similar to other arguments against doing this sort of thing.  What I
hope to see on the mailing list is some consensus on whether this is
in fact a useful idea or not, and then change the text in section 4 of
the error draft to match.

So far, the score seems to be two votes against :-).

--J


From owner-pilc@lerc.nasa.gov  Mon Apr 10 12:06:05 2000
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On 10 Apr 2000, Jamshid Mahdavi wrote:

> Bora Akyol <akyol@pluris.com> writes:
> 
> > One concern:
> > 
> > How does a wireless network router know who a corrupted packet belongs to
> > in a shared medium network so that it can send this message.
> > 
> > Very likely, what this will boil to is sending a message to all TCP
> > sessions that it has seen recently on this link.
> 
> This was not what I had in mind.  Rather, I thought that the router
> could simply look at the IP header in the corrupted packet and hope
> that the source IP address was not corrupted.  My assumption (not
> knowing a lot about wireless) was that there is a reasonably good
> chance that four bytes early in the packet would have made it through
> OK.  However, perhaps this is not a good assumption...

Depends on whether the IPv4 header checksum passed or not.

L.

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



From owner-pilc@lerc.nasa.gov  Mon Apr 10 12:26:05 2000
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  >> From: Jamshid Mahdavi <mahdavi@novell.com>
  ...
  >> So far, the score seems to be two votes against :-).



Here is a vote partially in favor and partially against -:)

There are two related possibilities for sending
ICMP EaTEN type message, of which only the first one
below seems to be considered in this thread:

* Unnecessary fast retransmits: As you note, the receiver
  may determine that a packet is corrupted, and inform its
  sender (provided that the header info is intact).

  This approach may be beneficial when the channel encounters
  a "random" loss (as opposed to an extended period of bad state).

  However, in such situations, as someone else also pointed out,
  it may be possible to implement link level retransmissions to
  deal with the transmission errors (for large delay*bw links,
  need to ensure that link layer delivers packets "almost" in-order).

* Unnecessary timeouts:
  An alternative situation that is not handled well by link level
  retransmissions is that of extended bad periods (for instance,
  driving through a tunnel with link broken, or losing connectivity
  during hand-off between non-overlapping cells, or a long error
  burst for whatever reason).

  I believe that explicit notifications are of greater value
  in such situations, than when you have occassional packet
  being corrupted by errors.
  In such cases, the ICMP can be generated at the base
  station rather than the wireless receiver, for packets
  destined to the wireless host (fortunately,
  the base station has packet header information available).

  If the lower layers would deliver some packets despite 
  the above events - such as the TCP kickstart proposal - then
  there is less need for a new ICMP even with bursty losses.

The second situation above is also why I do not much like
the name ICMP EaTEN (I agree it is cute, but could misrepresent
the cause of packet loss).

Regards.

- nitin



From owner-pilc@lerc.nasa.gov  Mon Apr 10 15:09:47 2000
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Jamshid

There is no way for the receiver to know which particular pieces of the header
are corrupt when the checksum fails. Therefore you can assume that the source
address is OK.

Overall, this message will not be useful.

Regards

Bora


Jamshid Mahdavi wrote:

> Bora Akyol <akyol@pluris.com> writes:
>
> > One concern:
> >
> > How does a wireless network router know who a corrupted packet belongs to
> > in a shared medium network so that it can send this message.
> >
> > Very likely, what this will boil to is sending a message to all TCP
> > sessions that it has seen recently on this link.
>
> This was not what I had in mind.  Rather, I thought that the router
> could simply look at the IP header in the corrupted packet and hope
> that the source IP address was not corrupted.  My assumption (not
> knowing a lot about wireless) was that there is a reasonably good
> chance that four bytes early in the packet would have made it through
> OK.  However, perhaps this is not a good assumption...
>
> > You are much better of giving some sort of piggy back feedback between
> > wireless nodes involved in the connection to their TCPs if they happen to
> > be at the same place.
> >
> > For example:
> >
> > if you have wireless router A, and wireless node B, then if B got some
> > sort of feedback on the link quality than it can give explicit feedback to
> > hosts that it has TCP connections to. If on the other hand two nodes
> > happen to be connecting through a wireless link, then we are out of
> > luck.
> >
> > In a nutshell, I think that this idea even though it sounds useful at
> > first, is really not that useful.
>
> I'm not completely understanding your argument here, but it does sound
> similar to other arguments against doing this sort of thing.  What I
> hope to see on the mailing list is some consensus on whether this is
> in fact a useful idea or not, and then change the text in section 4 of
> the error draft to match.
>
> So far, the score seems to be two votes against :-).
>
> --J



From owner-pilc@lerc.nasa.gov  Mon Apr 10 21:36:34 2000
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From: William D Ivancic <William.D.Ivancic@lerc.nasa.gov>
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For the router to see the IP header and check sum, the layer 2 protocol
would first have to pass the corrupted layer 2 data to layer 3.  Layer 2
usually has its own check sum.     Assuming this is the case and the IP
checksum does not pass, I would be reluctant to  believe anything in the
header.  It seams to easy to implement denial of services attacks and other
bad things here.

Also,  I've seen ECN used in two forms:  Explicit Congestion Notification
and Explicit Corruption Notification.  These are not interchangeable.  Any
suggestions on renaming Explicate Corruption Notification?  I believe ECN
was first used for Explicit Congestion Notification.

Will Ivancic 


>
>Bora Akyol <akyol@pluris.com> writes:
>
>> One concern:
>> 
>> How does a wireless network router know who a corrupted packet belongs to
>> in a shared medium network so that it can send this message.
>> 
>> Very likely, what this will boil to is sending a message to all TCP
>> sessions that it has seen recently on this link.
>
>This was not what I had in mind.  Rather, I thought that the router
>could simply look at the IP header in the corrupted packet and hope
>that the source IP address was not corrupted.  My assumption (not
>knowing a lot about wireless) was that there is a reasonably good
>chance that four bytes early in the packet would have made it through
>OK.  However, perhaps this is not a good assumption...


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To: William D Ivancic <William.D.Ivancic@lerc.nasa.gov>
Cc: pilc@grc.nasa.gov
Subject: Re: ICMP EaTEN message...
References: <4.1.20000410171641.00a93220@popserve.lerc.nasa.gov>
From: Jamshid Mahdavi <mahdavi@novell.com>
Date: 10 Apr 2000 18:05:40 -0700
In-Reply-To: William D Ivancic's message of "Mon, 10 Apr 2000 17:24:53 -0400"
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William D Ivancic <William.D.Ivancic@lerc.nasa.gov> writes:

> For the router to see the IP header and check sum, the layer 2 protocol
> would first have to pass the corrupted layer 2 data to layer 3.  Layer 2
> usually has its own check sum.     Assuming this is the case and the IP
> checksum does not pass, I would be reluctant to  believe anything in the
> header.  It seams to easy to implement denial of services attacks and other
> bad things here.
> 
> Also,  I've seen ECN used in two forms:  Explicit Congestion Notification
> and Explicit Corruption Notification.  These are not interchangeable.  Any
> suggestions on renaming Explicate Corruption Notification?  I believe ECN
> was first used for Explicit Congestion Notification.

ETEN (or EaTEN) is our attempt to rename Explicit Corruption
Notification, so there isn't a clash of abbreviations.  I added my
thoughts on one way to implement it, with ICMP.  Other suggestions are
also welcome!

ECN should mean Explicit Congestion Notification and nothing else.

--Jamshid


From owner-pilc@lerc.nasa.gov  Tue Apr 11 09:59:18 2000
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From: "Jan Kullander (ERV)" <jan.kullander@erv.ericsson.se>
To: pilc@grc.nasa.gov
Subject: Header compression in the LINK draft (or a new draft?)
Date: Tue, 11 Apr 2000 14:48:13 +0200
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Hi,

It is obvious to a link designer that extra money could be earned if header
compression is made non-transparent (i.e. if e.g. IP ID is generated by the
decompressor, the header checksum is recalculated, not all TCP timestamp
options are transferred etc.). There are however no guidelines describing
consequences (cost) of lost transparency other than general wording. I see a
need for a detailed consequence description of lost transparency for each
specific TCP/UDP/RTP header field. The consequences should be related to the
standards as they are, to current implementations as known and maybe even to
currently proposed standards as they often seem to superimpose new
functionality on existing fields.

I've noticed that the ROCH charter requires the decompressed result to be
semantically identical to the original. This decision has surely been
preceded by discussions that could be documented.

maybe it could be a draft of its own If the detail level of my proposed
documentation doesn't fit in the LINK draft ?

/Jan Kullander

----------------------------
Ericsson Mobile Data Design AB
Jan Kullander
Tel (Direct):  +46 31 344 6057   Fax:  +46 31 344 6033
Email:  Jan.Kullander@erv.ericsson.se
Address:   S:t Sigfridsgatan 89   /   412 66 Gothenburg   /   Sweden



From owner-pilc@lerc.nasa.gov  Tue Apr 11 14:47:59 2000
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To: "Jan Kullander (ERV)" <jan.kullander@erv.ericsson.se>
Cc: pilc@grc.nasa.gov
Subject: Re: Header compression in the LINK draft (or a new draft?)
In-reply-to: Your message of Tue, 11 Apr 2000 14:48:13 PDT.
Date: Tue, 11 Apr 2000 10:42:05 PDT
From: Vern Paxson <vern@ee.lbl.gov>
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> I've noticed that the ROCH charter requires the decompressed result to be
> semantically identical to the original. This decision has surely been
> preceded by discussions that could be documented.

Actually, "semantically identical" was chosen over "bitwise identical"
to allow for some wiggle room, and the hairsplitting arguments about
just what are the semantics have *not* yet been thrashed out.  This
would be a good discussion to have.

		Vern


From owner-pilc@lerc.nasa.gov  Tue Apr 11 22:14:28 2000
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Date: Tue, 11 Apr 2000 18:19:32 -0700 (PDT)
From: Gabriel Montenegro <gab@Eng.Sun.COM>
Reply-To: Gabriel Montenegro <gab@Eng.Sun.COM>
Subject: Re: ICMP EaTEN message...
To: mahdavi@novell.com
Cc: William D Ivancic <William.D.Ivancic@lerc.nasa.gov>, pilc@grc.nasa.gov
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---LA_F2160706241R-4A-955502373=:23136NCE.IlHAFeR
Content-Type: TEXT/PLAIN; CHARSET=US-ASCII

i agree that it is quite hard in practice to use the first of
the two message types as per nitin's classification below:

"Nitin H Vaidya" <vaidya@cs.tamu.edu> said:
> There are two related possibilities for sending
> ICMP EaTEN type message, of which only the first one
> below seems to be considered in this thread:
> 
> * Unnecessary fast retransmits: As you note, the receiver
>   may determine that a packet is corrupted, and inform its
>   sender (provided that the header info is intact).
>   
>   This approach may be beneficial when the channel encounters
>   a "random" loss (as opposed to an extended period of bad state).
...

> * Unnecessary timeouts:
>   An alternative situation that is not handled well by link level
>   retransmissions is that of extended bad periods (for instance,
>   driving through a tunnel with link broken, or losing connectivity
>   during hand-off between non-overlapping cells, or a long error
>   burst for whatever reason).
> 
>   I believe that explicit notifications are of greater value
>   in such situations, than when you have occassional packet
>   being corrupted by errors.
>   In such cases, the ICMP can be generated at the base
>   station rather than the wireless receiver, for packets
>   destined to the wireless host (fortunately,
>   the base station has packet header information available).

couldn't agree more here. i presented such a solution reusing 
an existing (but unused) icmp msg during the TSVWG meeting in
adelaide. there was pushback from van jacobson, steve deering
and geoff huston primarily  on the grounds
that it is easy to launch DOS attacks by using this "host
isolated" icmp unreachable.

the DOS risk can be alleviated by applying the ttl = 255 rule to
limit the DOS to neighboring systems, a mechanism originally
suggested by bob gilligan for use in ipv6 neighbor discovery
and also used in reverse tunneling for mobile ipv4 (rfc2344).
once this is in place, the extended link outage could not be
signalled from several hops away, but thems the breaks. this
would still be useful in the common case in the future in which
cell phones become routers between a cellular network and
a local network (bluetooth, for example). 

i'll be writing this up as an (individual) internet draft for pittsburgh.
for a quick summary see attachment.

"William D Ivancic" <William.D.Ivancic@lerc.nasa.gov> said:
> Also,  I've seen ECN used in two forms:  Explicit Congestion Notification
> and Explicit Corruption Notification.  These are not interchangeable.  Any
> suggestions on renaming Explicate Corruption Notification?  I believe ECN
> was first used for Explicit Congestion Notification.

only the first usage is correct. there is text to clarify this
confusion in the ERROR draft (appendices).

-gabriel

---LA_F2160706241R-4A-955502373=:23136NCE.IlHAFeR
Content-Type: TEXT/plain; name="isolated.txt"; charset=us-ascii
Content-Description: default

Error conditions can be reported
with an ICMP Host Unreachable message.  IP is required to pass
any such messages up to transport layers like UDP and TCP,
and these, in turn, to applications above them [RFC1122].
What is not clearly defined is what code within an ICMP Host
Unreachable message should be used to notify of an error condition.
For conditions of network outage, a currently unused
'host isolated' (code 8) was introduced for routers (actually,
IMP's) to inform hosts of an outage. 
Additionally, [MD95] argues for the application of 'host isolated' for 
notifications emanating from a host's lower layers.

In summary, these notifications to upper layers can originate either from 
within a host itself or in another host altogether. ICMP includes the necessary
information to determine the sender of the notification, as well
as a part of the datagram which encountered the error.

---------------------------------------------------------------
[RFC1122] Robert Braden, RFC-1122, Requirements for Internet hosts-
communication layers, October 1989.

[MD95] Gabriel Montenegro and Steve Drach, "System Isolation and
Network Fast-Fail Capability in Solaris," Second USENIX Symposium
on Mobile and Location-Independent, April 1995.
http://www.usenix.org/publications/library/proceedings/mob95/montenegro.html
                                                                         

---LA_F2160706241R-4A-955502373=:23136NCE.IlHAFeR--


From owner-pilc@lerc.nasa.gov  Wed Apr 12 01:54:34 2000
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To: Gabriel Montenegro <gab@Eng.Sun.COM>
Cc: mahdavi@novell.com, William D Ivancic <William.D.Ivancic@lerc.nasa.gov>,
        pilc@grc.nasa.gov
Subject: Re: ICMP EaTEN message...
In-reply-to: Your message of Tue, 11 Apr 2000 18:19:32 PDT.
Date: Tue, 11 Apr 2000 21:59:12 PDT
From: Vern Paxson <vern@ee.lbl.gov>
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> couldn't agree more here. i presented such a solution reusing 
> an existing (but unused) icmp msg during the TSVWG meeting in
> adelaide. there was pushback from van jacobson, steve deering
> and geoff huston primarily  on the grounds
> that it is easy to launch DOS attacks by using this "host
> isolated" icmp unreachable.

Actually, I think that's misreading their comments.  Their point, I believe,
was instead that there may well be multipathing available to the network,
such that the failure of a particular link just means that routing will
have to change to fix the problem.  It's important not to take today's
model where there's always a single wireless access link and from it
extrapolate that that will always be the architecture.

		Vern


From owner-pilc@lerc.nasa.gov  Wed Apr 12 05:43:09 2000
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From: "Jan Kullander (ERV)" <jan.kullander@erv.ericsson.se>
To: "'Vern Paxson'" <vern@ee.lbl.gov>
Cc: pilc@grc.nasa.gov
Subject: RE: Header compression in the LINK draft (or a new draft?)
Date: Wed, 12 Apr 2000 10:45:20 +0200
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>> I've noticed that the ROCH charter requires the decompressed result to be
>> semantically identical to the original. This decision has surely been
>> preceded by discussions that could be documented.
>
>Actually, "semantically identical" was chosen over "bitwise identical"
>to allow for some wiggle room, and the hairsplitting arguments about
>just what are the semantics have *not* yet been thrashed out.  This
>would be a good discussion to have.

Not being American I missed that nicety...

Still, I think a documentation on the consequences of breaking the
e2e transparency for each field would be more beneficial than a
general discussion of principles. A ban on non-transparency will
not prevent link designers from doing it but if they know the
consequences, at least there will be fewer surprises.

/Jan Kullander


From owner-pilc@lerc.nasa.gov  Wed Apr 12 11:28:49 2000
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Subject: DIAL M Workshop (Final Call for Papers)
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       (Please accept our apologies if you receive multiple copies.) 

                       FINAL CALL FOR PAPERS

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

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

                   Submission Deadline:  April  25, 2000

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

SCOPE:

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

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

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


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

PROGRAM COMMITTEE

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

STEERING COMMITTEE

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


SUBMISSION GUIDELINES:

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

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

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

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

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

  A special issue of Discrete Applied Mathematics will be devoted to 
a selection of submitted  papers from this workshop. Full versions of
the DIALM papers selected to appear in this special issue of 
Discrete Applied Mathematics will be subjected to the standard 
thorough journal refereeing process before final acceptance for
the journal.


IMPORTANT DATES:

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



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	Wed, 12 Apr 2000 09:21:47 -0700 (PDT)
Date: Wed, 12 Apr 2000 09:21:48 -0700 (PDT)
From: Gabriel Montenegro <gab@Eng.Sun.COM>
Reply-To: Gabriel Montenegro <gab@Eng.Sun.COM>
Subject: Re: ICMP EaTEN message...
To: Vern Paxson <vern@ee.lbl.gov>
Cc: pilc@grc.nasa.gov
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> > couldn't agree more here. i presented such a solution reusing 
> > an existing (but unused) icmp msg during the TSVWG meeting in
> > adelaide. there was pushback from van jacobson, steve deering
> > and geoff huston primarily  on the grounds
> > that it is easy to launch DOS attacks by using this "host
> > isolated" icmp unreachable.
> 
> Actually, I think that's misreading their comments.  Their point, I believe,
> was instead that there may well be multipathing available to the network,
> such that the failure of a particular link just means that routing will
> have to change to fix the problem.  It's important not to take today's
> model where there's always a single wireless access link and from it
> extrapolate that that will always be the architecture.

point taken. actually they made both comments (DOS and multipath).
there may have been others (are there any minutes for tsvwg meetings?).

at any rate, on the multipath comment: there's plenty of occasions
when it is useful, in particular when one looks at the predictions
of how common the following scenario will be:

	cell phone with two interfaces:
		cellular
		bluetooth 
	laptop or pda accessing the internet via the cell phone's
		routing services

in this case (or replace "bluetooth" with "serial cable" etc) there
is no multipath, the cell phone is not only the default but also
the exclusive router for the laptop/pda.

the icmp msg by itself just has enough information about where the
link is down. of course if there is multipath one could take that 
as a hint to go try another route or apply whatever
heuristic may apply. 

-gabriel



From owner-pilc@lerc.nasa.gov  Wed Apr 12 14:14:09 2000
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Date: Wed, 12 Apr 2000 12:14:16 -0500
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Please accept our apologies if you receive multiple copies of this call
for papers
			(FINAL CALL FOR PAPERS)
-------------------------------------------------------------------------
                 The Third ACM International Workshop on

	        WIRELESS MOBILE MULTIMEDIA (WoWMoM-2000)

              (to be held in conjunction with MobiCom-2000)

                      August 11, 2000 (Friday) 
              Seaport Hotel at the World Trade Center
                     Boston, Massachusetts, USA
             
                     Sponsored by ACM SIGMOBILE
         with support from Nortel Networks, Microsoft Research, 
              University of California at Riverside, and 
   DIMACS - The Center for Discrete Mathematics & Theoretical Computer
Science, and
           CReWMaN at the University of Texas at Arlington

             URL: http://www.cse.uta.edu/crew/wowmom-2000/

               SUBMISSION DEADLINE:  April  25, 2000

GENERAL CHAIR

Sajal K. Das, Director                               
Center for Research in Wireless Mobility and Networking (CReWMaN)     
Department of Computer Science and Engineering       Tel: (817) 272-7405
The University of Texas at Arlington                 Fax: (817) 272-3784
Arlington, TX 76019-0015                          E-mail:
das@cse.uta.edu


PROGRAM CO-CHAIRS

Satish K. Tripathi, Dean                       Kalyan Basu, Director
William R. Johnson, Jr. Family Professor       Wireless Technologies
Bourns College of Engineering                  Nortel Networks
University of California                       2201 Lakeside Blvd
Riverside, CA 92521-0425                       Richardson, TX 75083
USA                                            USA

Phone: (909) 787-6374                         Phone : (972) 684-5399
Fax:   (909) 787-3188                         Fax   : (972) 684-3775
E-mail: tripathi@engr.ucr.edu                 E-mail:
kbasu@nortelnetworks.com

-------------------------------------------------------------------------
IMPORTANT DATES:

     Paper Submissions Deadline:        April 25, 2000 
     Acceptance Notification:           June 1, 2000
     Camera-ready Manuscripts:          June 15, 2000
     Workshop                           August 11, 2000
-------------------------------------------------------------------------

SCOPE:
Over the last decade there has been a rapid growth of wireless 
communication technology. Voice communication using cellular phones has 
matured and become a significant part of our life today. In the recent 
past, the emergence of portable computing devices such as notebook
computers and personal digital assistants have led to the opportunity 
of such services as electronic mail, fax and calendar/diary programs 
being provided to mobile or roving users. Observing this trend, it can 
be predicted that the traffic over next generation, high-speed wireless 
networks will be dominated by personal multimedia applications such as 
video/news-on-demand, telemetry services, traveler information systems, 
WWW browsing, and so on. 

        The 3rd Generation Wireless activities are focused through the
effort 
of UMTS and IMT-2000 are considering the ATM technologies for the
network 
implementation. The recent activities at 3GIP forum is actively
considering 
the use of IP technology for the future wireless network. The wideband 
wireless network using IP technology is becoming an important industry
issue 
and the seamless integration of the wireless standards and wireless IP
and 
mobility standard are necessary to create this future network. The role
of 
the different IP technologies like SIP, HTML, XML, Netmeeting for
wireless 
and mobility are not finalized and will require considerable discussion
to 
obtain the resolution. Also, the wireless access techniques like
Bluetooth, 
WAP etc may require seamless integration to the 2.5G (GPRS) and 3G
wireless 
standards.

The objective of this workshop is to provide a forum for 
researchers and technologists to present new ideas and contributions in 
the form of technical paper, panel discussions, as well as online demos 
of service applications or technology related to wireless multimedia.
The issues and challenges for the development of wireless multimedia 
networks not only encompass a broad spectrum of research topics such as 
quality-of-service provisioning, broadband wireless, network support, 
protocols, or handover, but also involves a way to envision the
evolution 
of multimedia networks in the future. 
 
TOPICS:
Technical papers and/or product demos are solicited in all areas 
related to wireless multimedia and its interfaces (or co-existence) with 
the conventional wireline networks. Topics include but are not limited
to: 

        - Third Generation Systems
- Multimedia Applications
        - Fixed or Mobile Wireless Multimedia Access
- Wireless Network Evolution
- Broadband Wireless
- Wireless ATM and Wireless IP
        - Multicasting in Wireless Services
        - Delay and Jitter Management for Multimedia Services
        - Synchronization of Multimedia Wireless services using ATM and
IP 
- Quality-of-Service and Admission Control
- Multimedia Network Architecture and Protocols
- Compression Techniques
- Mobility Management & Handover Issues
        - Routing
- WWW Browsing
        - Telemetry Services
        - Broadcast Data
        - Mobile Agents
-------------------------------------------------------------

SUBMISSION GUIDELINES:

All submissions will be handled electronically. Authors should E-mail 
a Postscript version of their full paper to wowmom-2000@engr.ucr.edu . 
This E-mail address will become available by February 15, 2000. In order 
to ensure that the Postscript versions of the papers can be printed,
authors should be careful that their papers meet the following
restrictions: 

 - Postscript version 2 or later. 
 - Fits properly on "US Letter" size paper (8.5 X 11 inches) 
 - Reference only Computer Modern or standard Adobe printer fonts 
   (i.e. Courier, Times, Roman, or Helvetica); other fonts may be used 
   but must be included in the Postscript file. 
 - No longer than 20 pages including, pictures, tables and references. 

In addition, authors should separately E-mail the title, author names 
and full address, and abstract of their paper to the Program Co-Chairs: 
Satish K. Tripathi at tripathi@engr.ucr.edu and Kalyan Basu at 
kbasu@nortel.com. It is expected that all accepted papers will be 
presented at the workshop. 

The Workshop Proceedings will be published by the ACM Press. Selected
papers 
will be considered for publication in the ACM/Baltzer WINET and MONET
journals.
(WoWMoM'99 and WoWMoM'98 proceedings are available on the ACM digital
library.)

TECHNICAL PROGRAM COMMITTEE

   * Azzedine Boukerche (Univ of North Texas, USA) 
   * Pravin Bhagwat (IBM Watson Research Center, USA)
   * Andrew Campbell (Columbia Univ, USA) 
   * Yanghee Choi (Seoul National Univ, Korea)
   * K.-C. Chua (National University of Singapore)
   * Marco Conti (CNUCE, Pisa, Italy)
   * Subir Das (Telcordia Technologies, USA)
   * Lorenzo Donatiello (Univ of Bologna, Italy)
   * Anthony Joseph (UC Berkeley, USA)
   * Anurag Kumar (Indian Inst of Science, Bangalore)
   * Mohan Kumar (Curtin Univ, Perth, Australia)
   * Giri Mandyam (Nokia Research, Irving, USA)
   * Hiroyuki Morikawa (University of Tokyo, Japan)
   * S. Muthukrishnan (At&T Research)
   * Raja Neogi (Intel, Portland, USA)
   * Venkat Padmanabhan (Microsoft Research, USA)
   * Vasant Prabhu (Univ of Texas at Arlington, USA)
   * Jason Redi (BBN Technologies, USA)
   * Sumit Roy (Univ of Washington, USA)
   * Sanjoy Sen (Nortel Networks, USA)
   * Krishna Sivalingam (Washington State Univ, USA)
   * Mani Srivastava (Univ of California at Los Angeles, USA)
   * Violet Syrotiuk (Univ of Texas at Dallas, USA)
   * Philip Whiting (Lucent/Bell Labs, USA)


STEERING COMMITTEE 

   * Prathima Agrawal (Telcordia)
   * Ian Akyildiz (Georgia Institute of Technology)       
   * Kalyan Basu (Nortel Networks)
   * Maurizio Bonuccelli (University of Pisa)      
   * Imrich Chlamtac (University of Texas at Dallas)         
   * Sajal K. Das (Univ of Texas at Arlington)  
   * Randy Katz (University of California at Berkeley)      
   * Mahmoud Naghshineh (IBM Watson Research)
   * Christopher Rose (Rutgers University)
   * Mischa Schwartz (Columbia University)
   * Satish Tripathi (University of California at Riverside)

FINANCE CHAIR

  * Tom Jacob, University of North Texas

PUBLICITY CO-CHAIRS

  * Dhawal Moghe, Nortel Networks
  * Ramesh Yerraballi, Univ of Texas at Arlington
  * Claude Castelluccia, INRIA Rhone-Alpes 
  * Amitava Mukherjee, Pricewaterhouse Coopers Ltd.


From owner-pilc@lerc.nasa.gov  Wed Apr 12 14:24:47 2000
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Date: Wed, 12 Apr 2000 10:23:24 -0700
From: Bora Akyol <akyol@pluris.com>
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On the other hand, sometimes the laws of physics and financial reality set in.

There may be multi-pathing when devices have two or more different wireless
interfaces, such as a LAN and a WAN interface, but in the general WAN
environment, it is hard to get even decent single path coverage. Moreover, the
existing model of base stations and wireless nodes provides far more control to
the service provider than an ad-hoc packet radio network (even though I would
prefer this). Finally, with fiber coming everywhere but the last mile pretty
soon, the base station model provides good cost benefits.

Hence, it is not easy to assume away the current architecture either.

Also note that the problem is not limited to a wireless node connecting to the
Internet, but there may be two routers that are connected via a wireless link
(as in the connection between SLAC and Stanford for the longest time) and the
end hosts may not have a clue on what to do with the feedback they are getting.

Bora


Vern Paxson wrote:

> > couldn't agree more here. i presented such a solution reusing
> > an existing (but unused) icmp msg during the TSVWG meeting in
> > adelaide. there was pushback from van jacobson, steve deering
> > and geoff huston primarily  on the grounds
> > that it is easy to launch DOS attacks by using this "host
> > isolated" icmp unreachable.
>
> Actually, I think that's misreading their comments.  Their point, I believe,
> was instead that there may well be multipathing available to the network,
> such that the failure of a particular link just means that routing will
> have to change to fix the problem.  It's important not to take today's
> model where there's always a single wireless access link and from it
> extrapolate that that will always be the architecture.
>
>                 Vern



From owner-pilc@lerc.nasa.gov  Thu Apr 13 03:13:47 2000
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Cc: pilc@grc.nasa.gov
Subject: Re: ICMP EaTEN message...
In-reply-to: Your message of Wed, 12 Apr 2000 09:21:48 PDT.
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> point taken. actually they made both comments (DOS and multipath).
> there may have been others (are there any minutes for tsvwg meetings?).

Here's the relevant portion of the draft minutes (which I'll send out
to the tsvwg list shortly).

		Vern



Link Outage ICMP Notification -- Gabriel Montenegro, Sun

Both link-outage and link-outage-over are useful. Link outage was a valid
message at one time, worth revisiting. Easy to send, no state, simple
synchronous response using ICMP which stacks and apps already know about:
use 'host unreachable' message. RFC1122 says IP must pass these messages
to transport. But can't overload existing 'unreachable' codes. Recommend
using code #8: 'host isolated.' Introduced for routers (IMPs, actually).
Contains enough message to identify originating host. Actions could include
local disk caching, activate browser use of caches, freeze timers, etc.
Possible issues include whether errors are hard or soft. Could decide
based on whether message comes from local host or router.

Steve Deering: some links will send 'destination unreachable' messages
    when link goes down. What is an example of a hard error?
Gabe: one in which the upper layers won't retransmit forever. Like unhooking
    ethernet during an NFS transaction.
Steve: would need an AI to guess what the right response is. Give me an
    example when you'd know what to do?

Geoff Houston: what about the denial of service implications? This would
    be a great way to take down a network.
(Steve: minumum ICMP format include IP header plus 8 bytes.)

Jamshid: ICMP net down + ICMP kick won't work to manage TCP because you
    can't be sure you'll get the kick.

Van Jacobson: because one router can't reach a destination doesn't mean
    the packets won't reach the destination. Most vendors tend to misinterpret
    this information. Many firewalls filter out these ICMP messages.
    Suggest you don't do this because you can't know enough about your
    topology except for very simple networks.

Steve: one use for ICMP message is to print out a diagnostic when application
    finally quits.


From owner-pilc@lerc.nasa.gov  Fri Apr 14 07:34:03 2000
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From: Sampo A Syreeni <ssyreeni@cc.helsinki.fi>
Reply-To: Sampo Syreeni <decoy@iki.fi>
To: Bora Akyol <akyol@pluris.com>
cc: pilc@grc.nasa.gov
Subject: Re: ICMP EaTEN message...
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On Wed, 12 Apr 2000, Bora Akyol wrote:

>There may be multi-pathing when devices have two or more different wireless
>interfaces, such as a LAN and a WAN interface, but in the general WAN
>environment, it is hard to get even decent single path coverage.

GSM/GPRS plus LEO satellite capability would be a good example, at least
in Europe. The service is becoming available.

Sampo Syreeni <decoy@iki.fi>, aka decoy, student/math/Helsinki university



From owner-pilc@lerc.nasa.gov  Sat Apr 15 23:24:48 2000
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To: Bora Akyol <akyol@pluris.com>
cc: pilc <pilc@grc.nasa.gov>
From: Sally Floyd <floyd@aciri.org>
Subject: Re: ICMP EaTEN message... 
Date: Sat, 15 Apr 2000 19:09:07 -0700
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>How does a wireless network router know who a corrupted packet belongs to
>in a shared medium network so that it can send this message.
 
One possibility for ETEN might be the following:  

Assume that the link-level protocol uses FEC and/or link-level
retransmission, as a good link-level protocol for a lossy link
should, but that, at some point, the link-level protocol decides
to give up on trying to retransmit the corrupted packet.  At this
point, the link-level *receiver* does not necessarily know the
correct packet header for the corrupted packet, but the link-level
*sender* does know the correct header.  So it would seem that the
link-level receiver could tell the link-level sender to send a
short notification message, to the correct destination, that the
link-level has given up trying to correctly transmit the data packet
with this packet header.  (Yes, this short message could get
corrupted also;  in this case, the loss will be interpreted by the
end node as a congestion-related loss.)

One possible scenario for ETEN is for a TCP that has had a single
packet loss, has detected this loss with Fast Retransmit, and has
successfully retransmitted this packet and reduced its congestion
window in half.  If the TCP sender learns, perhaps some time later,
that this loss was in fact due to corruption and not congestion,
then the TCP sender can, after the fact, "undo" its reduction of
the congestion window, by slow-starting back up to its previous
value.

(I don't know the correct response of a TCP sender to the routine
loss of packets due to corruption, but reducing the congestion
window in half in response to a single packet lost due to corruption
seems like overkill to me...)

A second possible scenario for ETEN would be for an unreliable,
unicast transport protocol that uses equation-based congestion
control (e.g., "http://www.aciri.org/tfrc/").  Instead of reducing
the sending rate in half in response to each specific packet loss,
with equation-based congestion control the sender adjusts the
sending rate to the TCP-compatible sending rate that fits the packet
loss rate observed over a longer time interval.  Again, if the
sender learns after the fact that a loss was due to corruption
rather than to congestion, then the sender could adjust its sending
rate appropriately (whatever is the appropriate adjustment of the
sending rate to occasional packet corruption).

Yes?  No?  Did I forget something?  

The ideas are that in cooperation, the link-level sender and the
link-level receiver should know the correct packet header for the
corrupted packet; and that it could be useful for the sender to find
that a loss was due to corruption rather than congestion even if
the sender receives this information some time after it has made
a congestion-control response to the packet loss.

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


From owner-pilc@lerc.nasa.gov  Sun Apr 16 12:06:36 2000
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Subject: Re: ICMP EaTEN message...
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  >> From owner-pilc@lerc.nasa.gov Sat Apr 15 21:20:53 2000
  >> To: Bora Akyol <akyol@pluris.com>
  >> cc: pilc <pilc@grc.nasa.gov>
  >> From: Sally Floyd <floyd@aciri.org>
  >> Subject: Re: ICMP EaTEN message... 
  >> Date: Sat, 15 Apr 2000 19:09:07 -0700
  >> 
  >> >How does a wireless network router know who a corrupted packet belongs to
  >> >in a shared medium network so that it can send this message.
  >>  
  >> One possibility for ETEN might be the following:  
  >> 
  >> Assume that the link-level protocol uses FEC and/or link-level
  >> retransmission, as a good link-level protocol for a lossy link
  >> should, but that, at some point, the link-level protocol decides
  >> to give up on trying to retransmit the corrupted packet.  At this
  >> point, the link-level *receiver* does not necessarily know the
  >> correct packet header for the corrupted packet, but the link-level
  >> *sender* does know the correct header.  So it would seem that the
  >> link-level receiver could tell the link-level sender to send a
  >> short notification message, to the correct destination, that the

The link level sender could do this without having to be told
by the link level receiver.
Is there a disadvantge to link level sender acting on its own?

We have previously suggested a couple of related approaches
(named Partial Ack, and Explicit Bad State Notification), which
differ in the amount of information provided in the feedback
to TCP sender (www.cs.tamu.edu/faculty/vaidya/mobile/).

  >> link-level has given up trying to correctly transmit the data packet
  >> with this packet header.  (Yes, this short message could get
  >> corrupted also;  in this case, the loss will be interpreted by the
  >> end node as a congestion-related loss.)
  >> 
  >> One possible scenario for ETEN is for a TCP that has had a single
  >> packet loss, has detected this loss with Fast Retransmit, and has
  >> successfully retransmitted this packet and reduced its congestion
  >> window in half.  If the TCP sender learns, perhaps some time later,
  >> that this loss was in fact due to corruption and not congestion,
  >> then the TCP sender can, after the fact, "undo" its reduction of
  >> the congestion window, by slow-starting back up to its previous
  >> value.

This sounds like a good idea - if I understand correctly, Reiner
Ludwig's Eiffel algorithm behaves similarly.

  >> (I don't know the correct response of a TCP sender to the routine
  >> loss of packets due to corruption, but reducing the congestion
  >> window in half in response to a single packet lost due to corruption
  >> seems like overkill to me...)
  >> 
  >> A second possible scenario for ETEN would be for an unreliable,
  >> unicast transport protocol that uses equation-based congestion
  >> control (e.g., "http://www.aciri.org/tfrc/").  Instead of reducing
  >> the sending rate in half in response to each specific packet loss,
  >> with equation-based congestion control the sender adjusts the
  >> sending rate to the TCP-compatible sending rate that fits the packet
  >> loss rate observed over a longer time interval.  Again, if the
  >> sender learns after the fact that a loss was due to corruption
  >> rather than to congestion, then the sender could adjust its sending
  >> rate appropriately (whatever is the appropriate adjustment of the
  >> sending rate to occasional packet corruption).
  >> 
  >> Yes?  No?  Did I forget something?  
  >> 
  >> The ideas are that in cooperation, the link-level sender and the
  >> link-level receiver should know the correct packet header for the
  >> corrupted packet; and that it could be useful for the sender to find
  >> that a loss was due to corruption rather than congestion even if
  >> the sender receives this information some time after it has made
  >> a congestion-control response to the packet loss.

This may sounds like nitpicking, but I would suggest we
also consider congestion versus non-congstion losses, rather than
just congestion versus corruption (for instance, packet loss
during hand-off is not a corruption loss, but has similar
impact as a bursty error loss).

- nitin


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


From owner-pilc@lerc.nasa.gov  Mon Apr 17 05:10:35 2000
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Date: Mon, 17 Apr 2000 09:57:26 +0200
To: mahdavi@novell.com
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: ERROR draft comments...
Cc: "pilc@grc.nasa.gov" <pilc@grc.nasa.gov>
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At 02:46 07.04.00 , Jamshid Mahdavi wrote:
>Section 2.3:
>
> >    In low-speed, high error-rate environments (for example, the
> >    wireless WAN environment), TCP windows are much smaller, and burst
> >    errors must be much longer in duration in order to damage multiple
> >    segments. Accordingly, the complexity of SACK may not be
> >    justifiable, unless there is a high probability of both burst
> >    errors and congestion.
>
>This logic doesn't make a lot of sense to me.  I suppose that if your
>error rate is high, and you are constantly timing out, SACK is not
>going to make any difference.  But if there is congestion too
>(i.e. the total error rate is even higher), SACK is going to help even
>less.
>
>I'm not an expert, but I would think that in wireless environments,
>changes in surroundings (i.e. someone walks in front of the wireless
>card) would cause a high probability of correlated errors.  This would
>imply to me multiple losses per window, exactly the thing that SACK
>is good for.  So if things mostly *don't* suck, SACK will help for the
>times when they do.

In http://iceberg.cs.berkeley.edu/papers/Ludwig-WINET99/ we present 
measurements from a low-speed wireless WAN environment (9.6 Kb/s GSM 
circuit-switched data) and we do find the mentioned correlated losses.

This would speak in favor of SACK which we believe is a win anyway (at 
least it never hurts). However, we believe that persistent link layer ARQ 
for the class of reliable flows (TCP, Reliable-Multicast, etc.) is an even 
better solution. Still, SACK helps for correlated congestion losses and on 
wireless links that do not (or cannot, e.g., unidirectional satellite) 
implement link layer ARQ.

Also note, that (unlike stated in the above cited section 2.3) the fact 
that an environment is low-speed has little influence on the TCP sender's 
congestion window if router buffers are large enough. The advertized window 
remains to be the limit and the congestion window will usually get there.

///Reiner



From owner-pilc@lerc.nasa.gov  Mon Apr 17 05:20:08 2000
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Date: Mon, 17 Apr 2000 10:22:28 +0200
To: mahdavi@novell.com
From: Reiner Ludwig <Reiner.Ludwig@eed.ericsson.se>
Subject: Re: ICMP EaTEN message...
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At 22:23 07.04.00 , Jamshid Mahdavi wrote:
>Since the IETF, I came up with a counter-argument for Phil's point.
>If the link layer can accomplish the ARQ within 3 packets of the
>original transmission, there is no problem.  However, if it takes
>longer than this, it will be out of order by enough to trigger fast
>retransmit anyway, which mean that the ARQ had no useful effect.

Link layer ARQ is *commonly* implemented together with the in-order 
delivery property, i.e., no packet can "overtake" the other. Thus, the 
scenario you decribe is unlikely to happen in today's wireless systems 
implementing link layer ARQ.

///Reiner



From owner-pilc@lerc.nasa.gov  Mon Apr 17 08:53:06 2000
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Subject: Re: ICMP EaTEN message...
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I suggest we don't ***exclude*** the possibility of out-of-sequence
delivery
for future "wireless links" (i.e. RF environment).

Someone said, link layer ARQ is *commonly* implemented together with the
in-order
delivery property.... I agree.

Where the link delay is longer (an obvious example is packet satellite)
there
***may*** be advantage in allowing out of sequence delivery (by which I
mean the
receiver ARQ does not buffer correctly received out-of-sequence packets,
but forwards each good
packet on arrival, retransmitting any corrupted packets later
out-of-sequence). We have implemented
such schemes more than once in our lab based on a modified HDLC.,

N.B. Lack of sequencing may also arise from routing over alternate paths
/ links.

Gorry Fairhurst

P.S. I agree with nitin, we should talk about congestion losses (meaning
sender rate is too
high for the moment) with non-congestion losses (i.e. the loss is not a
function
of the sender rate, but is a function of RF environment, access
protocols, etc).



Reiner Ludwig wrote:

> At 22:23 07.04.00 , Jamshid Mahdavi wrote:
> >Since the IETF, I came up with a counter-argument for Phil's point.
> >If the link layer can accomplish the ARQ within 3 packets of the
> >original transmission, there is no problem.  However, if it takes
> >longer than this, it will be out of order by enough to trigger fast
> >retransmit anyway, which mean that the ARQ had no useful effect.
>
> Link layer ARQ is *commonly* implemented together with the in-order
> delivery property, i.e., no packet can "overtake" the other. Thus, the
> scenario you decribe is unlikely to happen in today's wireless systems
> implementing link layer ARQ.
>
> ///Reiner



From owner-pilc@lerc.nasa.gov  Tue Apr 18 08:12:00 2000
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From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>
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To: Nitin H Vaidya <vaidya@cs.tamu.edu>, pilc@grc.nasa.gov
Subject: Re: ICMP EaTEN message...
References: <200004161507.KAA04829@photon.cs.tamu.edu>
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Nitin H Vaidya wrote:

>   >>
>   >> >How does a wireless network router know who a corrupted packet belongs to
>   >> >in a shared medium network so that it can send this message.
>   >>
>   >> One possibility for ETEN might be the following:
>   >>
>   >> Assume that the link-level protocol uses FEC and/or link-level
>   >> retransmission, as a good link-level protocol for a lossy link
>   >> should, but that, at some point, the link-level protocol decides
>   >> to give up on trying to retransmit the corrupted packet.  At this
>   >> point, the link-level *receiver* does not necessarily know the
>   >> correct packet header for the corrupted packet, but the link-level
>   >> *sender* does know the correct header.  So it would seem that the
>   >> link-level receiver could tell the link-level sender to send a
>   >> short notification message, to the correct destination, that the
>
> The link level sender could do this without having to be told
> by the link level receiver.
> Is there a disadvantge to link level sender acting on its own?

What you say is entirely possible - but I guess it is wise to provide
this on a best effort basis - would the "request for the notification message"
take a similar bandwidth to a NAK?

As you say, it is probably only wise to send the notification itself using a
best effort service - or you end-up needing reliability mechanisms.
How valuable all this is will depend, I guess on the processes covering
the probability of loss.  In some cases this a shorter message may have
a better chance of transmission than a longer one. In a general wireless/RF
case, conditions could be very variable and this may not be necessarily so.

I wonder if the problem is that you really need complete information about
all losses being "link related" to inhibit CA. Whereas you only need one
unexplained loss to trigger CA?

> We have previously suggested a couple of related approaches
> (named Partial Ack, and Explicit Bad State Notification), which
> differ in the amount of information provided in the feedback
> to TCP sender (www.cs.tamu.edu/faculty/vaidya/mobile/).
>
>   >> link-level has given up trying to correctly transmit the data packet
>   >> with this packet header.  (Yes, this short message could get
>   >> corrupted also;  in this case, the loss will be interpreted by the
>   >> end node as a congestion-related loss.)
>   >>
>   >> One possible scenario for ETEN is for a TCP that has had a single
>   >> packet loss, has detected this loss with Fast Retransmit, and has
>   >> successfully retransmitted this packet and reduced its congestion
>   >> window in half.  If the TCP sender learns, perhaps some time later,
>   >> that this loss was in fact due to corruption and not congestion,
>   >> then the TCP sender can, after the fact, "undo" its reduction of
>   >> the congestion window, by slow-starting back up to its previous
>   >> value.
>
> This sounds like a good idea - if I understand correctly, Reiner
> Ludwig's Eiffel algorithm behaves similarly.

Have you thought of what would happen if both congestion and link
loss were simultaneously present?  If you have one ICMP EaTEN
message per missing IP fragment (not MSS?) you may be convinced
that all losses are link related?

>   >> (I don't know the correct response of a TCP sender to the routine
>   >> loss of packets due to corruption, but reducing the congestion
>   >> window in half in response to a single packet lost due to corruption
>   >> seems like overkill to me...)
>   >>
>   >> A second possible scenario for ETEN would be for an unreliable,
>   >> unicast transport protocol that uses equation-based congestion
>   >> control (e.g., "http://www.aciri.org/tfrc/").  Instead of reducing
>   >> the sending rate in half in response to each specific packet loss,
>   >> with equation-based congestion control the sender adjusts the
>   >> sending rate to the TCP-compatible sending rate that fits the packet
>   >> loss rate observed over a longer time interval.  Again, if the
>   >> sender learns after the fact that a loss was due to corruption
>   >> rather than to congestion, then the sender could adjust its sending
>   >> rate appropriately (whatever is the appropriate adjustment of the
>   >> sending rate to occasional packet corruption).
>   >>
>   >> Yes?  No?  Did I forget something?

Are you suggesting a proxy/gateway?

>
>   >>
>   >> The ideas are that in cooperation, the link-level sender and the
>   >> link-level receiver should know the correct packet header for the
>   >> corrupted packet; and that it could be useful for the sender to find
>   >> that a loss was due to corruption rather than congestion even if
>   >> the sender receives this information some time after it has made
>   >> a congestion-control response to the packet loss.
>
> This may sounds like nitpicking, but I would suggest we
> also consider congestion versus non-congstion losses, rather than
> just congestion versus corruption (for instance, packet loss
> during hand-off is not a corruption loss, but has similar
> impact as a bursty error loss).
>
> - nitin
>
>   >>
>   >> - Sally
>   >> --------------------------------
>   >> http://www.aciri.org/floyd/
>   >> --------------------------------
>   >>

-Gorry Fairhurst.



From owner-pilc@lerc.nasa.gov  Tue Apr 18 14:20:37 2000
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To: Nitin H Vaidya <vaidya@cs.tamu.edu>
cc: pilc@grc.nasa.gov
From: Sally Floyd <floyd@aciri.org>
Subject: Re: ICMP EaTEN message... 
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Nitin -

>The link level sender could do this without having to be told
>by the link level receiver.
>Is there a disadvantge to link level sender acting on its own?

No.  It is just that the link level sender has to know that the
link level receiver did not correctly receive the earlier transmission
of that packet.

>We have previously suggested a couple of related approaches
>(named Partial Ack, and Explicit Bad State Notification), which
>differ in the amount of information provided in the feedback
>to TCP sender (www.cs.tamu.edu/faculty/vaidya/mobile/).

Thanks, I will look at them.

>This may sounds like nitpicking, but I would suggest we
>also consider congestion versus non-congstion losses, rather than
>just congestion versus corruption (for instance, packet loss
>during hand-off is not a corruption loss, but has similar
>impact as a bursty error loss).

Yes, I agree.

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


From owner-pilc@lerc.nasa.gov  Tue Apr 18 14:27:08 2000
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From: Sally Floyd <floyd@aciri.org>
Subject: Re: ICMP EaTEN message... 
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>I suggest we don't ***exclude*** the possibility of out-of-sequence
>delivery
>for future "wireless links" (i.e. RF environment).

I agree.  One of the purposes of the proposed D-SACK (duplicate-SACK)
extension to SACK, now finishing up its last call before going to
Proposed Standard
("http://search.ietf.org/internet-drafts/draft-floyd-sack-00.txt"),
is to allow the development of mechanisms that make TCP more robust
in the presence of out-of-order delivery of packets.  And TCP is not
likely to be the only transport protocol operating over future "wireless
links"...

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


From owner-pilc@lerc.nasa.gov  Sat Apr 22 05:57:24 2000
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From: myzhai <myzhai@sina.com>
To: pilc@grc.nasa.gov
Subject: about loss rate of wireless link
Date: Sat Apr 22 16:21:37 CST 2000
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Hi,
       I have a question about wireless comm. to ask you.The problem is:
if a wireless user, for example, a wireless phone user establishes a number of connections
to a webserver at the same time, assumed three connections ,one bit rate is 1Mbps, one is
3Mbps and the third is 10Mbps.we only consider the condition that the last hop is wireless or satellite,
then if there are losses occur on the wireless hop for example because of handoff 
or interference, I want to know do the three connections have the same loss rates under the
same condition? How can we model the loss of wireless channel, random loss or bursty loss?

thx
Zhai

______________________________________


===================================================================
ÐÂÀËÃâ·Ñµç×ÓÓÊÏä http://mail.sina.com.cn 



From owner-pilc@lerc.nasa.gov  Mon Apr 24 06:11:29 2000
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From: Dr G Fairhurst <gorry@erg.abdn.ac.uk>
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I'm not sure relates to the Internet Protocol over wireless or satellite...

Basically the answer to what you are asking will depend upon the power budgets for each case,
and the modulation/FEC schemes used.  If all things are equal, the packet loss rate is given by the bit
error rate, the packet (frame) size: Packet Success rate = p = (1-b)**m.  b= ber. m=packet size in bits.

The bit error rate is related to the signal to noise ratio, or bit energy divided by noise.  Naturally for
the same given power, the bit error rate is much smaller if the energy per bit is smaller (i.e.
at higher data rates where bits are smaller).  The situation would be more complex if the different
data rates each used different parameters, or an adaptive system was used.

Interference poses a different problem - generally interference results in a very significant increase
in noise.  Unless an adaptive technique is used, it is likely that a packet sent at any rate would
be corrupted - performance could be independent of rate.

I think you may need to look at electronics/communications/propagation references to get your
answer, I believe the PILC mailing list focusses on Internet protocol issues, not the physical medium.

Best wishes,

Gorry Fairhurst.

myzhai wrote:

> Hi,
>        I have a question about wireless comm. to ask you.The problem is:
> if a wireless user, for example, a wireless phone user establishes a number of connections
> to a webserver at the same time, assumed three connections ,one bit rate is 1Mbps, one is
> 3Mbps and the third is 10Mbps.we only consider the condition that the last hop is wireless or satellite,
> then if there are losses occur on the wireless hop for example because of handoff
> or interference, I want to know do the three connections have the same loss rates under the
> same condition? How can we model the loss of wireless channel, random loss or bursty loss?
>
> thx
> Zhai
>
> ______________________________________
>
> ===================================================================
> ?ÂÀËÃâ·ÑµçxÓÓÊÏä http://mail.sina.com.cn



From owner-pilc@lerc.nasa.gov  Fri Apr 28 02:07:37 2000
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From: "Shirley Tseng" <shirleytseng@earthlink.net>
To: <pilc@grc.nasa.gov>
Subject: IEEE 802.16 Broadband wireless meeting May 1-5
Date: Thu, 27 Apr 2000 22:06:31 -0700
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FYI  - apologies for the late posting.

Shirley Tseng

* SESSION #7 IN MARYLAND
802.16 Working Group on Broadband Wireless Access Session #7
<http://ieee802.org/16/meetings/mtg07> takes place
on 1-5 May 2000 at the Gaithersburg Hilton in Gaithersburg, Maryland,
USA (near Washington, DC). The host is Yunsang Park of Hughes Network
Systems, which is serving as corporate host. The session web site
provides all logistical information and lists the various Calls for
Contributions seeking input documents for the meetings.

ETSI BRAN HIPERACCESS has accepted an invitation from 802.16 to co-locate
their Meeting 18.5 along with 802.16 Session #7. Joint 802.16/HIPERACCESS
Plenary Meetings will be included.

The session host is Yunsang Park, Hughes Network Systems, which is serving
as the corporate host.

Agenda [agenda details depend on responses to Calls for Contributions]


802.16.1 PHY Task Group: Call for Contributions: Session #7

802.16.1 MAC Task Group: Call for Contributions [not published]

Call for Evaluations and Improvements: IEEE 802.16.1, Session #7
802.16.2 Coexistence Task Group: Call for Comments [not published]

802.16.3 (2-11 GHz): Call For Comments on Functional Requirements
Call for Contributions: Requirements for WirelessHUMANTM Systems
Tutorial Open Forum
Registration, Logistics, and







From owner-pilc@lerc.nasa.gov  Sat Apr 29 03:43:39 2000
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