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From: Oliver Korfmacher <okorf@netcs.com>
Message-Id: <9307010739.AA20897@keks.netcs.com>
Subject: Re: Brutal Simplicity
To: Keith Sklower <sklower@vangogh.cs.berkeley.edu>
Date: Thu, 1 Jul 93 9:39:49 MEST
Cc: ietf-ppp@ucdavis.edu, iplpdn@CNRI.Reston.VA.US
In-Reply-To: <199307010204.TAA24191@vangogh.CS.Berkeley.EDU>; from "Keith Sklower" at Jun 30, 93 7:04 pm
X-Mailer: ELM [version 2.2 PL0]
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Guten Morgen --

> Unfortunately it doesn't work so well with old versions of NCSA
> telnet for PC's (which don't understand IP fragmentation), and
> it doesn't work so well with IPX.  (which expects delivery in sequence)
There are more implementations (even commercial ones) having
problems with non-in-sequence delivery, at least performance problems.

> But the real motivation for having sequencing and reliable delivery
> over the links in the IP-only case is to do data compression at
> the link layer.  If you drop something, or get it out of order then
> you screw up the data dictionary and have to start over.
>
> [..]
> If you aren't running LAPB or LAPF
> but you are doing compression and are just trusting the usual good
> service of ISDN, for example, you still will want to know that the
> last fragment you sent on the link to be brought down got there,
> which you accomplish by trading config-requests with the MRCS
> option, (which prevents further traffic on that link but not the
> others), and then you bring down the link by sending LCP
> Terminate-requests and acks.
Yes. We have tested our V.42bis data compression over german
ISDN (which is well, quite good regarding line quality), and
every 1000..10000 datagram is broken if we are using HDLC-framing
only. The problem with LAPB is, that if used over long distance lines,
the end-to-end delay comes to ranges of seconds, which brings down
the throughput (due to too small window sizes).
And: please keep in mind that ISDN is mainly a dial-on-demand
media, so extensive ppp-negotiations in connection setup will definitly
delay the setup time! This is a delay the user directly experiences.

Gruesse,

	Oliver

        Oliver Korfmacher (okorf@netcs.com, whois OK11)
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To: Tom Lyon <Tom.Lyon@eng.sun.com>
Cc: iplpdn@CNRI.Reston.VA.US, westmark@dirac.scri.fsu.edu
Subject: Re: Bandwidth available on DS-3 
In-Reply-To: Your message of "Wed, 30 Jun 93 09:54:07 PDT."
             <9306301654.AA01265@whitsun.eng.sun.com> 
Date: Thu, 01 Jul 93 06:36:28 -0400
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From: Stephen D Crocker <crocker@tis.com>

I'm not sure about the framing of a DS3, but a DS1 carries 1,544,000
bits when all the framing is stripped.  As I recall, the frames are
193 bits, and there are 8,000 frames per second.  When a DS1 line is
subdivided into DS0 channels, the 193 bits is treated as 24 frames of
8 bits with one bit of framing, i.e. 24 * 8 + 1 = 193.

One DS3 channel carries 28 DS1 channels.  28 * 1,544,000 = 43,232,000.
I don't know how much additional framing overhead there is, but 45 Mbs
seems a bit high.


Steve
                 

> Sender:      iplpdn-request@IETF.CNRI.Reston.VA.US
> From:    Tom Lyon <Tom.Lyon@eng.sun.com>
> To:      iplpdn@CNRI.Reston.VA.US, westmark@dirac.scri.fsu.edu
> Date:    Wed, 30 Jun 93 09:54:07 PDT
> Subject: Re: Bandwidth available on DS-3
> 
> 
> A DS3 carries 7 DS2s which carry 4 DS1s which carry 24 DS0s which are 64Kbits.
> So, I'd guess 7*4*24*64000 (not 65536!), which is 43008000 bits/sec.
> The framing required to do all this bumps it up to the commonly heard
> 45Mbps.
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X-Orig-Sender:iplpdn-request@IETF.CNRI.Reston.VA.US
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From: m_fidler%W036_NW@mwmgate1.mitre.org
To: Tom Lyon <Tom.Lyon@eng.sun.com>
Cc: iplpdn@CNRI.Reston.VA.US, westmark@dirac.scri.fsu.edu
Subject: Re: Re: Bandwidth available on DS-3

Well, the channel rate on a T3 is 44,732,000 (not 45m) so the difference
is overhead (to 43.232).  It seems to me that there is an "unframed"
version of T3 (*not* DS3) that is not conformant to the M13 mux framing
outlined here that is probably what we actually use for data and may
differ from all the computations.  Anyone have any insight?

- Mike F.

 >I'm not sure about the framing of a DS3, but a DS1 carries 1,544,000
 >bits when all the framing is stripped.  As I recall, the frames are
 >193 bits, and there are 8,000 frames per second.  When a DS1 line is
 >subdivided into DS0 channels, the 193 bits is treated as 24 frames of
 >8 bits with one bit of framing, i.e. 24 * 8 + 1 = 193.
 >
 >One DS3 channel carries 28 DS1 channels.  28 * 1,544,000 = 43,232,000.
 >I don't know how much additional framing overhead there is, but 45 Mbs
 >seems a bit high.
 >
 >
 >Steve
 >
 >
 >> Sender:      iplpdn-request@IETF.CNRI.Reston.VA.US
 >> From:    Tom Lyon <Tom.Lyon@eng.sun.com>
 >> To:      iplpdn@CNRI.Reston.VA.US, westmark@dirac.scri.fsu.edu
 >> Date:    Wed, 30 Jun 93 09:54:07 PDT
 >> Subject: Re: Bandwidth available on DS-3
 >>
 >>
 >> A DS3 carries 7 DS2s which carry 4 DS1s which carry 24 DS0s which are  
>64Kbits.
 >> So, I'd guess 7*4*24*64000 (not 65536!), which is 43008000 bits/sec.
 >> The framing required to do all this bumps it up to the commonly heard
 >> 45Mbps.

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From: Rick Szmauz 226-6693 LKG2-2/X12 <szmauz@manage.enet.dec.com>
To: iplpdn@CNRI.Reston.VA.US
Cc: szmauz@manage.enet.dec.com
Apparently-To: iplpdn@cnri.reston.va.us
Subject: Re: Re: Bandwidth available on DS-3


Clear channel DS-3 utilizes 1 bit every 85 for DS-3 framing purposes.
Therefore approx. 44.2 Mbps = 44.736 Mbps x 84/85 are available for
DS-3 level payload in a native data link application.  At the DS-3
level, framing is %98.8 efficient.  Of course the efficiency of client 
data link would have to be factored in to provide the overall payload
framing efficiency.

,Rick
szmauz@hifi.lkg.dec.com

____________________________
From:	HIFI::"m_fidler%W036_NW@mwmgate1.mitre.org@dec:.lkg.us1rmc.dnet"  1-JUL-1993 07:55:00.62
To:	Tom Lyon <Tom.Lyon@eng.sun.com>
CC:	iplpdn@CNRI.Reston.VA.US, westmark@dirac.scri.fsu.edu
Subj:	Re: Re: Bandwidth available on DS-3

Well, the channel rate on a T3 is 44,732,000 (not 45m) so the difference
is overhead (to 43.232).  It seems to me that there is an "unframed"
version of T3 (*not* DS3) that is not conformant to the M13 mux framing
outlined here that is probably what we actually use for data and may
differ from all the computations.  Anyone have any insight?

- Mike F.

 >I'm not sure about the framing of a DS3, but a DS1 carries 1,544,000
 >bits when all the framing is stripped.  As I recall, the frames are
 >193 bits, and there are 8,000 frames per second.  When a DS1 line is
 >subdivided into DS0 channels, the 193 bits is treated as 24 frames of
 >8 bits with one bit of framing, i.e. 24 * 8 + 1 = 193.
 >
 >One DS3 channel carries 28 DS1 channels.  28 * 1,544,000 = 43,232,000.
 >I don't know how much additional framing overhead there is, but 45 Mbs
 >seems a bit high.
 >
 >
 >Steve
 >
 >
 >> Sender:      iplpdn-request@IETF.CNRI.Reston.VA.US
 >> From:    Tom Lyon <Tom.Lyon@eng.sun.com>
 >> To:      iplpdn@CNRI.Reston.VA.US, westmark@dirac.scri.fsu.edu
 >> Date:    Wed, 30 Jun 93 09:54:07 PDT
 >> Subject: Re: Bandwidth available on DS-3
 >>
 >>
 >> A DS3 carries 7 DS2s which carry 4 DS1s which carry 24 DS0s which are  
>64Kbits.
 >> So, I'd guess 7*4*24*64000 (not 65536!), which is 43008000 bits/sec.
 >> The framing required to do all this bumps it up to the commonly heard
 >> 45Mbps.


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% From: m_fidler%W036_NW@mwmgate1.mitre.org
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% Cc: iplpdn@CNRI.Reston.VA.US, westmark@dirac.scri.fsu.edu
% Subject: Re: Re: Bandwidth available on DS-3
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From: "k1io, FN42jk" <goldstein@carafe.tay2.dec.com>
To: crocker@tis.com, iplpdn@CNRI.Reston.VA.US
Subject: Re: Bandwidth available on DS-3 
X-Vms-To: SMTP%"crocker@tis.com"
X-Vms-Cc: SMTP%"iplpdn@CNRI.Reston.VA.US"

Well, that's the trouble with asking a question about payload of a 
DS-3.

The real payload of a DS-3 is seven DS-2s, not 28 DS-1s, if you take
the usual literal interpretation.  A DS-2 is 6.312 Mbps; each DS-2 is
of course plesiochronous.  If you were using DS-3 for raw bits you
wouldn't care about DS-1s, a second mux level, would you?  Of course
why care about DS-2's?  But then you have to figure out just how
you're going to use the various overheads at the DS-3 level, plus the
options like C-bit parity, etc.

I think that's why they invented SONET.
(only half :-) needed)
  fred
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From: Chip Sharp 6424 <hhs@teleoscom.com>
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To: Tom.Lyon@eng.sun.com
Cc: iplpdn@CNRI.Reston.VA.US, westmark@dirac.scri.fsu.edu
In-Reply-To: Tom Lyon's message of Wed, 30 Jun 93 09:54:07 PDT <9306301654.AA01265@whitsun.eng.sun.com>
Subject: Bandwidth available on DS-3


The following is correct for the M23 Multiplex format of DS3:


>From: Tom Lyon <Tom.Lyon@eng.sun.com>

>A DS3 carries 7 DS2s which carry 4 DS1s which carry 24 DS0s which are 64Kbits.
>So, I'd guess 7*4*24*64000 (not 65536!), which is 43008000 bits/sec.
>The framing required to do all this bumps it up to the commonly heard
>45Mbps.

However, this includes all the overhead for the 7 DS2's and 4 DS1's.
If you just take the raw bandwidth of the M23 multiplex you get
7 * 6312 kbit/s = 44184000 bit/s.

If you take the raw bandwidth of the DS3 frame format (without the M23
multiplex format) it runs as follows (the frame structure is broken
into Multiframes):

DS3 raw bit rate = 44736000 bit/s (not 45 Mbit/s)
4760 bits per Multiframe 
56 kbit/s overhead per Multiframe

If you subtract the overhead from the total bit count you get 4704
bits per multiframe available to carry data.

Thus to get the total bandwidth, you can divide the total bandwidth by
the number of bits per multiframe to get the number of multiframes per
second and then multiply by 4704 to get the number of bits per second:

	This number comes out to be approximately 44209694 bit/s 

In order to use this bandwidth you have to define a frame structure
within the DS3 frame for your application.  For example, the standards
committees are now working on a mapping for ATM cells into DS3 frames
(which is where I got most of this information).


=======================================================================
Hascall H. Sharp			Teleos Communications, Inc.
System Engineering			2 Meridian Road
					Eatontown, NJ  07724
voice:  +1 908 544 6424
fax:    +1 908 544 9890
email:   hhs@teleoscom.com
========================================================================
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From: Karl Fox <karl@morningstar.com>
Message-Id: <9307011434.AA02238@remora.MorningStar.Com>
To: Oliver Korfmacher <okorf@netcs.com>
Cc: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu, 
    Keith Sklower <sklower@vangogh.cs.berkeley.edu>
Subject: Re: Brutal Simplicity
References: <199307010204.TAA24191@vangogh.CS.Berkeley.EDU>
	<9307010739.AA20897@keks.netcs.com>
Reply-To: Karl Fox <karl@morningstar.com>
Organization: Morning Star Technologies, Inc.

Oliver Korfmacher writes:
> And: please keep in mind that ISDN is mainly a dial-on-demand
> media, so extensive ppp-negotiations in connection setup will definitly
> delay the setup time! This is a delay the user directly experiences.

Then either your users are very perceptive or your PPP implementation is
defective, because I see typical PPP setup times of 600 ms, and this
over V.32/V.42/V.42bis modems which likely have much worse delay
characteristics than your ISDN lines.
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From: "David E. McDysan" <0002806303@mcimail.com>
To: iplpdn <iplpdn@CNRI.Reston.VA.US>
Subject: (Forwarded) Bandwidth available on DS-3
Message-Id: <10930701165801/0002806303MT1EM@mcimail.com>

From ANSI T1.107, section 8 (DS3)

The nominal interface rate is 44.736 Mb/s.

The DS3 signal is partitioned into M-frames of 4760 bits each.

There are 56 frame overhead bits per M-frame (The C-bits, framing and other
overhead bits are here), the rest of the M-frame bits are available for payload,
M13, M23 or "Clear Channel" DS3.

For "Clear Channel" DS3 the available rate is then:

        4760-56
        ------- 44.736 Mb/s = 44.209689 Mb/s
          4760

From the ATM Forum UNI spec, version 2.0, section 2.2.2, when the PLCP overhead
is taken into account, the bit rate available for transport of ATM cells is

         40.704 Mb/s.

Dave McDysan
dmcdysan@mcimail.com
---------------------------------
Forwarded message

via: mailbox 1, dmcdysan on MCI ** Message Received OK      
Date:     Wed Jun 30, 1993  5:57 am  CDT
Source-Date: Wed, 30 Jun 1993 05:47:45 -0400
From:     Jay Westmark
          EMS: INTERNET / MCI ID: 376-5414
          MBX: westmark@dirac.scri.fsu.edu
TO:     * David E. McDysan / MCI ID: 280-6303
TO:       David Greenfield / MCI ID: 402-0699
TO:       RND - RAD NETWORK DEVICES LTD. / MCI ID: 437-3580
TO:       Jim Mollenauer / MCI ID: 478-7822
TO:       Michael E. Conn / MCI ID: 438-7451
TO:       Henry Sinnreich / MCI ID: 249-8337
TO:       Laszlo I. Szerenyi / MCI ID: 432-7219
TO:       Mary Jander / MCI ID: 473-4998
TO:       Scott Brigham / MCI ID: 244-2341
TO:       iplpdn
          EMS: INTERNET / MCI ID: 376-5414
          MBX: iplpdn@cnri.reston.va.us
Subject:  Bandwidth available on DS-3
Message-Id: 53930630105735/0003765414NA3EM
Source-Msg-Id: <199306300947.AA60900@dirac.scri.fsu.edu>


All,

Does anyone know what is the effective usable bandwidth on a clear channel DS-3
after the framing overhead?


-jay westmark

---------------------------------
End forwarded message


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To: Karl Fox <karl@morningstar.com>
cc: Oliver Korfmacher <okorf@netcs.com>, iplpdn@CNRI.Reston.VA.US, 
    ietf-ppp@ucdavis.edu, Keith Sklower <sklower@vangogh.cs.berkeley.edu>
Subject: Re: Brutal Simplicity
In-reply-to: Your message of "Thu, 01 Jul 93 10:34:31 EDT." <9307011434.AA02238@remora.MorningStar.Com>
Date: Thu, 01 Jul 93 23:34:19 +0100
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From: Graham Knight <G.Knight@cs.ucl.ac.uk>
Message-ID:  <9307011825.aa28598@CNRI.Reston.VA.US>

Karl,

	In the way in which we use ISDN, 600ms is a significant time.
We see ISDN call set-ups of the order of 500ms. This is just about
small enough for users to tolerate as an echo delay. Mostly the echo
delay on an ISDN call is unnoticable but, if there has been a pause so that 
the ISDN call has timed out, there will be a delay of the order of 500ms
whilst the ISDN call is re-established. Adding another 600ms to this
delay really does make a difference. Many users perception of the ISDN
will move from "slick" to "clunky".	


						Graham
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From: Karl Fox <karl@morningstar.com>
Message-Id: <9307012348.AA04553@remora.MorningStar.Com>
To: Graham Knight <G.Knight@cs.ucl.ac.uk>
Cc: ietf-ppp@ucdavis.edu, iplpdn@CNRI.Reston.VA.US
Subject: Re: Brutal Simplicity
References: <9307011434.AA02238@remora.MorningStar.Com>
	<9307012225.AA11413@ucdavis.ucdavis.edu>
Reply-To: Karl Fox <karl@morningstar.com>
Organization: Morning Star Technologies, Inc.

Graham Knight writes:
> Karl,
> 
> 	In the way in which we use ISDN, 600ms is a significant time.
> We see ISDN call set-ups of the order of 500ms. This is just about
> small enough for users to tolerate as an echo delay. Mostly the echo
> delay on an ISDN call is unnoticable but, if there has been a pause so that 
> the ISDN call has timed out, there will be a delay of the order of 500ms
> whilst the ISDN call is re-established. Adding another 600ms to this
> delay really does make a difference. Many users perception of the ISDN
> will move from "slick" to "clunky".	

The 1.1s delay doesn't happen on every keystroke, though -- only on the
first keystroke after the idle timer has brought down the connection.
How can the user distinguish the ISDN link reestablishment from simply
being `swapped out'?  How often does that happen, anyway?  Are you
running with a 5 second idle timer?
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From: Oliver Korfmacher <okorf@netcs.com>
Message-Id: <9307020647.AA23628@keks.netcs.com>
Subject: Re: Brutal Simplicity
To: Cheng Chen 6444 <ctc@teleoscom.com>
Date: Fri, 2 Jul 93 8:47:58 MEST
Cc: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu
In-Reply-To: <9307011535.AA24681@teleoscom.com>; from "Cheng     Chen       6444" at Jul 1, 93 11:35 am
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> 
> 
> >Yes. We have tested our V.42bis data compression over german
> >ISDN (which is well, quite good regarding line quality), and
> >every 1000..10000 datagram is broken if we are using HDLC-framing
> >only.
> 
> Oliver,
> 
> Could you please elaborate on the way it is "broken"?  When it's
> broken, is there any possible recovery procedures, or the connection
> has to be taken down?  Thanks.
"broken" means in our context, that the hdlc chip has discarded the
received datagram due to its wrong checksum. Higher protocol layers 
are informed, but in the case of V.42bis with its dynamically changed
tables, there is no possibility to recover, since the de-compressor is
out of synch. We are then going to tear down the link. But anyhow,
our recommendation (to our customers) is to use LAPB is compression
is used.

	Oliver
> 
> Cheng T. Chen
> Teleos Communications, Inc.
> 

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From: Oliver Korfmacher <okorf@netcs.com>
Message-Id: <9307020651.AA23639@keks.netcs.com>
Subject: Re: Brutal Simplicity
To: karl@morningstar.com
Date: Fri, 2 Jul 93 8:51:07 MEST
Cc: okorf@netcs.com, iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu, 
    sklower@vangogh.cs.berkeley.edu
In-Reply-To: <9307011434.AA02238@remora.MorningStar.Com>; from "Karl Fox" at Jul 1, 93 10:34 am
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> 
> Oliver Korfmacher writes:
> > And: please keep in mind that ISDN is mainly a dial-on-demand
> > media, so extensive ppp-negotiations in connection setup will definitly
> > delay the setup time! This is a delay the user directly experiences.
> 
> Then either your users are very perceptive or your PPP implementation is
> defective, because I see typical PPP setup times of 600 ms, and this
> over V.32/V.42/V.42bis modems which likely have much worse delay
> characteristics than your ISDN lines.
As Graham already stated, due to relatively short timeout (in fact,
some customers are using 19 seconds due to the local tariff structure)
the user may percept the delay more than once. If they are using
additional security features like callback or similar, the delay
even doubles.

	Oliver

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To: jas@proteon.com, Jay Westmark <westmark@dirac.scri.fsu.edu>
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From: kaj@cc.bellcore.com
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Subject: Re:  IPX, DECNet, & AppleTalk over FDDI
Cc: iplpdn@CNRI.Reston.VA.US, ecmt1@mail.bellcore.com

At  5:14 PM 6/29/93 -0400, Jay Westmark wrote:

>What about IPX and/or appletalk over frame relay/SMDS?
>
there are specifications by the SMDS Interest Group (SIG)
on at least the following that i'm aware of:
- DEC-IV over SMDS
- OSI/DEC-V over SMDS
- AppleTalk over SMDS (SMDSTalk) (informational spec)
- IPX over SMDS (informational spec)

IP over SMDS is RFC1209.

contact Elysia Tan (SIG WG chair, ecmt1@mail.bellcore.com) for details.

kaj

0------------0--------------0-------------0-------------0-----------0
Kaj Tesink    
pmail Bellcore                          vmail (908) 758-5254
      331 Newman Springs Rd.          faxmail (908) 758-4196
      Red Bank, NJ 07701                email kaj@cc.bellcore.com

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From: Gerry Meyer <gerry@spider.co.uk>
Date: Tue, 6 Jul 93 10:05:11 +0100
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To: sklower@vangogh.cs.berkeley.edu
Subject: re: Simple Multilink
Cc: gerry@spider.co.uk, iplpdn@IETF.CNRI.Reston.VA.US

Keith Sklower and I had some private correspondence, reproduced below.
I have ADDED my response to his last posting as this new message.

Keith wrote:

>Feel free to re-post your questions and my responses to the mailing
>list.  I am trying to make the document reflect more of committee
>consensus than my personal opinion.
>
>    Hi. Me again.
>
>    1) The draft makes no mention of a re-assembly/flush timer on the receive
>       side other than negatively in #6.2.2.
>       Even with the null fragment stuffing (#6.2.1) one will still be required
>
>I am not convinced that a separate re-assembly/flush timer is needed in
>addition to a link-dead timer.  I am still trying to make this a
>``simple'' protocol by not encouraging any options which aren't absolutely
>necessary.  Even if we specify that there should be such a timer, do we
>need a control protocol option to set it?  Wouldn't a setting of
>4 times the transmission + round trip delay for a maximum sized packet
>be a reasonable default?

What is the link dead timer?

I had assumed the null fragment stuffing had been put in as an attempt to do
away with a receiver re-assembly/flush timer.

I would actually prefer to do without the null fragment stuffing because:

- they are not special, they can get lost just like any other datagram so
  a re-assembly timer is in my view needed.

- they result in additional traffic (which costs on X.25 connections).
  I could envisage posting an unfragmented packet down one channel followed
  by null fragments down all channels.

Like you, I am not particularly in favour of negotiating everything - the
value probably has to be manually configured anyway - so why bother to
negotiate it.  It must be derivable from the buffer space allocated and the
line speed.

The real problem with the re-assembly timer is the amount of processing it
can potentially consume.  But if you've got to have it you've got to have it.

>
>    2) Is there some way to get round the double header if a datagram is not
>       being fragmented?
>
>If you have no requirement that the packet not be sequenced, then
>just don't use the fragmentation header!  If you do want it
>sequenced, but not split, then you need to have some kind of
>header saying that there is a sequencing header present.

What I want is sequencing without fragmentation at an acceptable cost
in terms of header.

On X.25 we can support null NLPID (multiplexed) or specific NLPID (non
multiplexed) VCs.  Multiplexed use is preferred on a number of grounds
which I won't go into here.

So currently IP, CLNP have a 1 Octet header and IPX, DECnet etc have an
additional 5 Octet SNAP header.

To provide sequencing in this environment, my interpretation of the
proposal introduces a fragment header NLPID (1) + SNAP (5) + SEQUENCE (2)
+ OFFSET (2) of 10 Octets.

So the leading (or perhaps ONLY) fragment for IP, CLNP will have an 11
octet header and IPX, DECnet etc will have a 16 Octet header.
Subsequent fragments will have a 10 Octet header independent of protocol.

Since I am attracted to NOT fragmenting datagrams I find the leading
fragment header rather too large.

>I had this argument with a guy from gandalf.  He proposed a different
>header than the RFC1294 header which had only two bytes instead of
>4 bytes (a first bit, a done bit, and a minimum of 11 bit sequence
>number, but the sequence number applied to individual fragments).
>Thus, his header is good for both sequencing and fragmentation reassembly
>but at a cost of two bytes per packet instead of 4 bytes.  (the cost
>of the fragmentation/reassembly, is thus 2 **bits** out of the 2 byte
>structure.
>
>With normal ppp type negotiation you could compress the fragmentation
>header to 1 byte (3b) + the fragmentation/sequencing structure (4
>bytes now, 2 bytes his scheme), and then the header for the
>encapsulated packet, which could be a single byte.  So, yes you
>can save 2 bytes per packet by not using the RFC1294 fragmentation
>header.  But, I really, really, really am opposed to using a
>different header for PPP than 1294; saving two bytes per packet is
>just not worth the increase in non-uniformity; on the other hand
>I will glad support you in your attempts to convince the 1294 people
>that they should change their way of doing segmentation.

I am very much in favour of the information being the same wherever it
comes from - so 4 bytes for sequnece number and offset is fine.

>    3) I think it would be worth being more detailed about the
>       packet formats in the different variants - by the use of
>       examples in a future draft.
>
>OK.  Other than the segmentation headers for 1294 and PPP what do
>you suggest? A sample sequenced-but-not-fragmented IP packet?
>

Plus RFC1356 variants with NULL NLPID (IP + SNAP) and specific NLPID.

    Gerry
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From: Dave Carr <dcarr@gandalf.ca>
Message-Id: <9307061609.AA00727@donut.gandalf.ca>
Subject: Simple Multilink Comments (fwd)
To: iplpdn@CNRI.Reston.VA.US
Date: Tue, 6 Jul 1993 12:09:23 -0400 (EDT)
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Content-Length: 3545      

I'm not a regular on this list, but have been active on the PPP
LAPB and Compression lists.  Keith Sklower asked me to cross post
here.

I would like to see some changes to the Simple Multilink draft from
Keith Sklower.  Since Keith's proposal basically does away with the
standard LAPB/multilink which had been proposed by Fred Baker, and 
backed by Dave Rand and myself, I felt the need to stur the stew.

As many of you may know, I like to squeeze every last byte out of the
link.  As we start to add more and more header bytes, things like
VJ compression start to become useless.  Therefore, I propose to
lternate method, saving bytes in the header, while reducing complexity.

I can see some arguments that this will remove any compatability
with RFC1294.  SO WHAT!  I never liked that one either. (Donning
my asbestos suit :-))

At the same time, I also like speed, as I'm sure many of you may
be in favour of.

It is in this light that I propose the following simple multilink 
header.  

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|   HDLC FLAG   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|     0xFF      |      0x03     |     0x00      |     0x3b      |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|M|           Sequence          |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The fragmentation Offset field is deleted.  This is really redundant
information since it can be calculated from the individual fragments.

Instead, the sequence number is bumped by one on each and every
fragment, not on a per frame basis.   The More (M) stolen from X.25
indicates that this packet is not the last fragment.

Sequence number range would be a negotiable parameter.  Like the 
protocol field, the sequence number field size compression can be 
negotiated to reduce the field down to a single byte.

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|   HDLC FLAG   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|     0xFF      |      0x03     |     0x3b      |M|     Seq     |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

For a simple basic rate connection (2B), a sequence number from
0..127 is sufficient to keep the pipe full.  For a larger number
of links, the extended (default) range of 0..32767 is available.

In Keith's proposal, the case of multilink without LAPB is
straightforward.  The reassembly queue can either be an array or
a linked-list of fragments.  When one implements LAPB however, the
receiver may have to hold onto the fragments from many packets
in the window in case of retransmissions.  The only practical
method I can see for this is a linked list of fragments chained
to each receiver ring entry indexed by the sequence number.  This
means walking down the list to insert, as well as checking the
list to check if all fragments have been received for a frame.

From an implmentation standpoint, this proposal IMHO is simpler.
The receiver simply needs a ring buffer, indexed by the sequence 
number (modulo as required).  

Please copy me on any responses to the list, as I haven't requested
to be added to this list.
-- 
Dave Carr               | dcarr@gandalf.ca       | It's what you learn, 
Principal Designer      | TEL (613) 723-6500     | after you know it all,
Gandalf Data Limited    | FAX (613) 226-1717     | that counts. 
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To: Oliver Korfmacher <okorf@netcs.com>
Cc: karl@morningstar.com, iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu, 
    sklower@vangogh.cs.berkeley.edu
Subject: Re: Brutal Simplicity 
In-Reply-To: Your message of "Fri, 02 Jul 1993 08:51:07 EST."
             <9307020651.AA23639@keks.netcs.com> 
Date: Tue, 06 Jul 93 11:12:01 MST
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From: Kevin Paul Herbert <kph@cisco.com>

>As Graham already stated, due to relatively short timeout (in fact,
>some customers are using 19 seconds due to the local tariff structure)
Interesting... now that many of the long distance carriers in the US are
offering 6-second billing, I wonder if people in the US will want to use
dial-on-demand in this way in order to do toll avoidance.

Kevin
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From: Karl Fox <karl@morningstar.com>
Message-Id: <9307070159.AA16614@remora.MorningStar.Com>
To: iplpdn@CNRI.Reston.VA.US
Subject: InvARP format
Reply-To: Karl Fox <karl@morningstar.com>
Organization: Morning Star Technologies, Inc.

What PID do Reverse ARP and Inverse ARP use on a frame relay PVC?  The
latest Multiprotocol over Frame Relay draft contains a nice diagram
showing that normal ARP requests and responses use PID 0x0806, but does
Reverse ARP use 0x0806 or 0x8035 (the Reverse ARP Ethertype)?  What does
Inverse ARP use?
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From: Gerry Meyer <gerry@spider.co.uk>
Date: Wed, 7 Jul 93 09:47:08 +0100
Message-Id: <27579.9307070847@orbweb.spider.co.uk>
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To: dcarr@gandalf.ca, iplpdn@CNRI.Reston.VA.US
Subject: Re: Simple Multilink Comment
Cc: gerry@spider.co.uk


Dave Carr <dcarr@gandalf.ca> wrote:

>I'm not a regular on this list, but have been active on the PPP
>LAPB and Compression lists.  Keith Sklower asked me to cross post
>here.
>
>I would like to see some changes to the Simple Multilink draft from
>Keith Sklower.  Since Keith's proposal basically does away with the
>standard LAPB/multilink which had been proposed by Fred Baker, and
>backed by Dave Rand and myself, I felt the need to stur the stew.
>
>As many of you may know, I like to squeeze every last byte out of the
>link.  As we start to add more and more header bytes, things like
>VJ compression start to become useless.  Therefore, I propose to
>lternate method, saving bytes in the header, while reducing complexity.
>
>I can see some arguments that this will remove any compatability
>with RFC1294.  SO WHAT!  I never liked that one either. (Donning
>my asbestos suit :-))
>
>At the same time, I also like speed, as I'm sure many of you may
>be in favour of.
>
>It is in this light that I propose the following simple multilink
>header.
>
> 0                   1                   2                   3
> 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>+-+-+-+-+-+-+-+-+
>|   HDLC FLAG   |
>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>|     0xFF      |      0x03     |     0x00      |     0x3b      |
>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>|M|           Sequence          |
>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
>The fragmentation Offset field is deleted.  This is really redundant
>information since it can be calculated from the individual fragments.
>
>Instead, the sequence number is bumped by one on each and every
>fragment, not on a per frame basis.   The More (M) stolen from X.25
>indicates that this packet is not the last fragment.

One of the problems I have with Keith's proposal is that unfragmented
packets are penalised in terms of header length.  I like the idea of
increamenting the sequence number on every fragment and using the M-bit.

One small thing to watch out for is that the packets/fragments following
a lost fragment must be discarded up to and including the next one with
the M-bit NOT set.  If the first packet does NOT have the M-bit set it
may be a perfectly valid packet being discarded - probably not a great loss.

>Sequence number range would be a negotiable parameter.  Like the
>protocol field, the sequence number field size compression can be
>negotiated to reduce the field down to a single byte.
>
> 0                   1                   2                   3
> 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>+-+-+-+-+-+-+-+-+
>|   HDLC FLAG   |
>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>|     0xFF      |      0x03     |     0x3b      |M|     Seq     |
>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
>For a simple basic rate connection (2B), a sequence number from
>0..127 is sufficient to keep the pipe full.  For a larger number
>of links, the extended (default) range of 0..32767 is available.

To keep it simple I would NOT allow sequence space negotiation.
I feel it is important to present the SAME sequence/fragment information
in ALL circumstances.  This allows you to combine packets/fragments
from different sources.

>In Keith's proposal, the case of multilink without LAPB is
>straightforward.  The reassembly queue can either be an array or
>a linked-list of fragments.  When one implements LAPB however, the
>receiver may have to hold onto the fragments from many packets
>in the window in case of retransmissions.  The only practical
>method I can see for this is a linked list of fragments chained
>to each receiver ring entry indexed by the sequence number.  This
>means walking down the list to insert, as well as checking the
>list to check if all fragments have been received for a frame.
>
>From an implmentation standpoint, this proposal IMHO is simpler.
>The receiver simply needs a ring buffer, indexed by the sequence
>number (modulo as required).
>
>Please copy me on any responses to the list, as I haven't requested
>to be added to this list.

Perhaps you could elaborate on its use in different environments:
Frame Relay, X.25 and PPP.

    Gerry
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From: Oliver Korfmacher <okorf@netcs.com>
Message-Id: <9307071252.AA02648@keks.netcs.com>
Subject: Re: Brutal Simplicity 
To: Kevin Paul Herbert <kph@cisco.com>
Date: Wed, 7 Jul 93 14:52:57 MEST
Cc: okorf@netcs.com, karl@morningstar.com, iplpdn@CNRI.Reston.VA.US, 
    ietf-ppp@ucdavis.edu, sklower@vangogh.cs.berkeley.edu
In-Reply-To: <199307061812.AA22391@dry.cisco.com>; from "Kevin Paul Herbert" at Jul 06, 93 11:12 am
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> 
> >As Graham already stated, due to relatively short timeout (in fact,
> >some customers are using 19 seconds due to the local tariff structure)
> Interesting... now that many of the long distance carriers in the US are
> offering 6-second billing, I wonder if people in the US will want to use
> dial-on-demand in this way in order to do toll avoidance.
> 
> Kevin
> 
I wonder what happens if the so-called 'Spitzabrechnung' becomes thruth.
This gives fine-granularity connect time accounting (gran.: 1 sec).
Is there any national ISDN which already supports such a thing?

	Oliver

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From: S_SILVERMAN%W035_NW@mwmgate1.mitre.org
To: Oliver Korfmacher <okorf@netcs.com>, Kevin Paul Herbert <kph@cisco.com>
Cc: okorf@netcs.com, karl@morningstar.com, iplpdn@CNRI.Reston.VA.US, 
    ietf-ppp@ucdavis.edu, sklower@vangogh.cs.berkeley.edu
Subject: Re: Brutal Simplicity

>
> >As Graham already stated, due to relatively short timeout (in fact,
> >some customers are using 19 seconds due to the local tariff structure)
> Interesting... now that many of the long distance carriers in the US are
> offering 6-second billing, I wonder if people in the US will want to use
> dial-on-demand in this way in order to do toll avoidance.
>
> Kevin
>
I wonder what happens if the so-called 'Spitzabrechnung' becomes truth.
This gives fine-granularity connect time accounting (gran.: 1 sec).
Is there any national ISDN which already supports such a thing?

	Oliver

          There is a difference between the granularity on the timing
          of a call and the minumum call duration.  The TELCO could
          easily time to 1 second but have a 1 minute minimum. The
          reality is that there is a real cost associated with setting
          up a call.  This eats both signaling bandwidth and processor
          time.  Somehow, this cost must be passed on to the user.

          Steve Silverman   silverma@mitre.org

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From: S_SILVERMAN%W035_NW@mwmgate1.mitre.org
To: Oliver Korfmacher <okorf@netcs.com>, Kevin Paul Herbert <kph@cisco.com>
Cc: okorf@netcs.com, karl@morningstar.com, iplpdn@CNRI.Reston.VA.US, 
    ietf-ppp@ucdavis.edu, sklower@vangogh.cs.berkeley.edu
Subject: Re: Brutal Simplicity

>
> >As Graham already stated, due to relatively short timeout (in fact,
> >some customers are using 19 seconds due to the local tariff structure)
> Interesting... now that many of the long distance carriers in the US are
> offering 6-second billing, I wonder if people in the US will want to use
> dial-on-demand in this way in order to do toll avoidance.
>
> Kevin
>
I wonder what happens if the so-called 'Spitzabrechnung' becomes truth.
This gives fine-granularity connect time accounting (gran.: 1 sec).
Is there any national ISDN which already supports such a thing?

	Oliver

          There is a difference between the granularity on the timing
          of a call and the minumum call duration.  The TELCO could
          easily time to 1 second but have a 1 minute minimum. The
          reality is that there is a real cost associated with setting
          up a call.  This eats both signaling bandwidth and processor
          time.  Somehow, this cost must be passed on to the user.

          Steve Silverman   silverma@mitre.org

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From: Keith Sklower <sklower@vangogh.cs.berkeley.edu>
Message-Id: <199307072327.QAA16820@vangogh.CS.Berkeley.EDU>
To: iplpdn@CNRI.Reston.VA.US
Subject: Is 1356 cast in stone yet?

There is a minor thing I'm wondering about -

1356 says that you if you put a snap header in the call-user-data
then you logically prepend that snap header to each packet received.

On a PVC, real honest-to-goodness call-request packets are
not needed for the network sake.  However, at least one
ISO document (8878 AD2) says that you can make up a call-request,
and put the q bit on it, and pass it through the PVC.  They
do this specifically to transmit the user-connect-data associated
with CONS.  Is it worth adding this technique to 1356?
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X-Orig-Sender:iplpdn-request@IETF.CNRI.Reston.VA.US
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From: Keith Sklower <sklower@vangogh.cs.berkeley.edu>
Message-Id: <199307080052.RAA17374@vangogh.CS.Berkeley.EDU>
To: Internet-Drafts@CNRI.Reston.VA.US
Subject: multi-protocol over ISDN, parameter-negotiation, simple multi-link
Cc: fbaker@acc.com, iplpdn@CNRI.Reston.VA.US

Please find two revisions of previously issued internet-drafts, available
via anonymous ftp at vangogh.cs.berkeley.edu, in the directory pub/ppp-ipldpn:
	draft-ietf-iplpdn-multi-isdn-01.txt
	draft-ietf-iplpdn-para-negotiation-01.txt

(multi-isdn had its name changed at committee request, but
I suggest keeping the ftp name the same to reduce confusion).

Also, in the same directory find a proposal being made to the ppp
and iplpdn commitees jointly:

	draft-ietf-iplpdn-simple-multi-00.txt

This is work which mostly represents committee consensus as of the
last meeting.

``We will take this up at Amsterdam, and there being no objections, will
submit it for advancement to Proposed Standard.'' -- Bill.Simpson
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From: "RND - RAD NETWORK DEVICES LTD." <0004373580@mcimail.com>
To: ietf rip <ietf-rip@xylogics.com>
To: iplpdn <iplpdn@CNRI.Reston.VA.US>
Subject: 
Message-Id: <84930708054748/0004373580NA1EM@mcimail.com>

Hi All
 Are there any applications or NIC drivers that use (initiate) the LLC Type-1
XID and TEST commands. Are there any routers that use it ?

regards
Sefi


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Remove from this List.
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Please remove me from this list.
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From: William Allen Simpson <bill.simpson@um.cc.umich.edu>
Message-Id: <1328.bill.simpson@um.cc.umich.edu>
To: iplpdn@CNRI.Reston.VA.US
Cc: ietf-ppp@ucdavis.edu
Reply-To: bsimpson@morningstar.com
Subject: Re: Brutal Simplicity

I've been a bit appalled by the notion that someone would want to bring
up and tear down a circuit between every character, or even between
every command.

A 6 second or 19 second call is simply ridiculous.  I do not think that
we should bother designing for such silly situations.

It sounds to me as if someone is trying to spoof the telco into giving
them "free" time.  We should not encourage this, and it is likely to be
self defeating in the long run (charges will become more punitive).

The tariff structure here in the States includes a line setup
"installation" charge, a monthly fee, a fee for the first minute (call
setup), and a per 15 second fee after the first minute.  The call setup
charge is usually about SIX TIMES the incremental charge, and can range
as high as TWENTY times.

For standard analog local calls, there is no incremental charge.  Only
the call setup is charged ($0.08-0.25 per call).  Calling every 19
seconds would be extremely foolhardy.  On the contrary, you should NEVER
hang up.

Local connection office equipment is smart enough to know whether your
call was actually connected, so you are not supposed to get charged for
call setup if the call doesn't go through.  This means that you ALWAYS
get charged if your call completes, even for very short periods of time.

Long distance companies (inter-LATA) are always charged for call setup
by the local service provider -- whether or not the actual call goes
through.  For marketing reasons, some will not charge if the call is
less than a minute, because they can't tell if the call completed.  Most
will ALWAYS charge regardless (to recoup the local connection cost).

Looking at the above, there is no reason to call for less than the time
that the incremental charge is less than the average of the cumulative
incremental charge and the setup charge.  In fact, the crossover point
is at least 4 minutes, and often as long as 15 minutes.

For other "special" services, which have the incremental charge, if you
don't use a reasonable amount of time, you are really wasting the
installation and monthly charges, so the few calls you make are actually
*hideously* expensive.


Finally, the theoretical (data transmission only) PPP setup time at
64Kbps sync would be approximately 30 milliseconds.

For 9,600 bps async, it is theoretically as fast as 177 ms.  I measured
in the 200-240 ms range in my implementation.

It's a bit longer for Morningstar, or the NetBlazer, because they have
an extra login step before PPP starts.

Bill.Simpson@um.cc.umich.edu
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--NextPart

A Revised Internet Draft is available from the on-line Internet-Drafts 
directories. This draft is a work item of the IP over Large Public Data 
Networks Working Group of the IETF.                                        

       Title     : Determination of Encapsulation of Multi-protocol 
                   Datagrams in Circuit-switched Environments              
       Author(s) : K. Sklower
       Filename  : draft-ietf-iplpdn-multi-isdn-01.txt
       Pages     : 9

This document enumerates the existing means for transmitting Internet  
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common methods, from which one can determine which method is in use by 
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A Revised Internet Draft is available from the on-line Internet-Drafts 
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Networks Working Group of the IETF.                                        

       Title     : Parameter Negotiation for the Multiprotocol Interconnect
       Author(s) : K. Sklower, C. Frost
       Filename  : draft-ietf-iplpdn-para-negotiation-01.txt
       Pages     : 7

This document describes mechanisms for negotiating operational parameters 
wherever SNAP encapsulation of Internet Protocols is available.  For 
example, it is suitable for use in conjunction with RFC 1294 (Multiprotocol
Over Frame Relay) and RFC 1356 (Multiprotocol Interconnect on X.25 and ISDN
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A New Internet Draft is available from the on-line Internet-Drafts 
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Networks Working Group of the IETF.                                        

       Title     : A Simple Multilink Protocol for Synchronizing the 
                   Transmission of Multi-protocol Datagrams.               
       Author(s) : K. Sklower
       Filename  : draft-ietf-iplpdn-simple-multi-00.txt
       Pages     : 12

This document proposes a method for splitting and recombining datagrams 
across multiple  logical data links.  Such facilities are desirable to 
exploit the potential for increased bandwidth offered by multiple bearer 
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From: Karl Fox <karl@morningstar.com>
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To: William Allen Simpson <bill.simpson@um.cc.umich.edu>
Cc: ietf-ppp@ucdavis.edu, iplpdn@CNRI.Reston.VA.US
Subject: Re: Brutal Simplicity
References: <1328.bill.simpson@um.cc.umich.edu>
Reply-To: Karl Fox <karl@morningstar.com>
Organization: Morning Star Technologies, Inc.

William Allen Simpson writes:
> For 9,600 bps async, it is theoretically as fast as 177 ms.  I measured
> in the 200-240 ms range in my implementation.
> 
> It's a bit longer for Morningstar, or the NetBlazer, because they have
> an extra login step before PPP starts.

We usually recommend the extra login step because it's easier to
understand and set up, but it's not required, and it's not what I use on
my home system.
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To: bsimpson@morningstar.com
Cc: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu
Subject: Re: Brutal Simplicity 
In-Reply-To: Your message of "Wed, 07 Jul 93 12:31:15 EDT."
             <1328.bill.simpson@um.cc.umich.edu> 
Date: Thu, 08 Jul 93 14:34:05 -0400
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X-Mts: smtp

> I've been a bit appalled by the notion that someone would want to bring
> up and tear down a circuit between every character, or even between
> every command.
...
> It sounds to me as if someone is trying to spoof the telco into giving
> them "free" time.  We should not encourage this, and it is likely to be
> self defeating in the long run (charges will become more punitive).


Let me echo that sentiment. Toll avoidance is an ethical issue. The telco folks 
tend to be paranoid about getting ripped off (result, as I understand it, of a 
wave of phone phraud during the late 1960's, when some considered stealing from 
big business a sacrament rather than a sin).  Any hint of making standards that 
manipulate their systems into giving free service sets off alarm bells that, at 
a minimum, will cause them to obstruct progress. Worse, it creates a situation 
where features/capabilities needed to support data - like LLC negotiation in 
ISDN - get wrapped around the axle on the fact that the less progressive 
elements in the telcos believe they can't trust us to not misuse them.  We should be working to build, 
rather than destroy, trust.
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Subject: Re: Is 1356 cast in stone yet?
To:	iplpdn@CNRI.Reston.VA.US
Date:	Thu, 8 Jul 1993 14:39:01 -0400
Cc:	Ragnar Paulson <ragnar@software.group.com>
In-Reply-To: <199307072327.QAA16820@vangogh.CS.Berkeley.EDU>; from "Keith Sklower" at Jul 7, 93 7:27 pm
Reply-to: ragnar@software.group.com
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From:	Ragnar Paulson <ragnar@software.group.com>
Message-ID: <9307081439.aa21049@rosie.software.group.com>

Keith, 

You write:
> 
> There is a minor thing I'm wondering about -
> 
> 1356 says that you if you put a snap header in the call-user-data
> then you logically prepend that snap header to each packet received.
> 
> On a PVC, real honest-to-goodness call-request packets are
> not needed for the network sake.  However, at least one
> ISO document (8878 AD2) says that you can make up a call-request,
> and put the q bit on it, and pass it through the PVC.  They
> do this specifically to transmit the user-connect-data associated
> with CONS.  Is it worth adding this technique to 1356?
> 

This is the first I've heard of this PVC feature, so I don't think it's very
wide spread in network implementations.  It's certainly not CCITT 
conformant which is what most public network providers use.  

Also it seems to me that the nature of a PVC is it is not dynamic,
and thus you don't require the call user data to specify the enscapsulation
being used.  You just configure both ends the same making it a local issue 
to the RFC implementation.

Is this not a simpler and more reasonable approach?

-- 
Ragnar Paulson				email:  ...uunet!tsgfred!ragnar	
The Software Group Limited		or:	ragnar@software.group.com
Phone: 416 856 0238
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Date: Thu, 8 Jul 1993 11:52:22 +0800
To: William Allen Simpson <bill.simpson@um.cc.umich.edu>, 
    iplpdn@CNRI.Reston.VA.US
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From: Brian Lloyd <brian@lloyd.com>
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Subject: Re: Brutal Simplicity
Cc: ietf-ppp@ucdavis.edu

At 12:31 7/7/93 -0400, William Allen Simpson wrote:
>I've been a bit appalled by the notion that someone would want to bring
>up and tear down a circuit between every character, or even between
>every command.
>
>A 6 second or 19 second call is simply ridiculous.  I do not think that
>we should bother designing for such silly situations.
>
>It sounds to me as if someone is trying to spoof the telco into giving
>them "free" time.  We should not encourage this, and it is likely to be
>self defeating in the long run (charges will become more punitive).
>
>The tariff structure here in the States includes a line setup
>"installation" charge, a monthly fee, a fee for the first minute (call
>setup), and a per 15 second fee after the first minute.  The call setup
>charge is usually about SIX TIMES the incremental charge, and can range
>as high as TWENTY times.
>
>For standard analog local calls, there is no incremental charge.  Only
>the call setup is charged ($0.08-0.25 per call).  Calling every 19
>seconds would be extremely foolhardy.  On the contrary, you should NEVER
>hang up.
>
>Local connection office equipment is smart enough to know whether your
>call was actually connected, so you are not supposed to get charged for
>call setup if the call doesn't go through.  This means that you ALWAYS
>get charged if your call completes, even for very short periods of time.
>
>Long distance companies (inter-LATA) are always charged for call setup
>by the local service provider -- whether or not the actual call goes
>through.  For marketing reasons, some will not charge if the call is
>less than a minute, because they can't tell if the call completed.  Most
>will ALWAYS charge regardless (to recoup the local connection cost).
>
>Looking at the above, there is no reason to call for less than the time
>that the incremental charge is less than the average of the cumulative
>incremental charge and the setup charge.  In fact, the crossover point
>is at least 4 minutes, and often as long as 15 minutes.
>
>For other "special" services, which have the incremental charge, if you
>don't use a reasonable amount of time, you are really wasting the
>installation and monthly charges, so the few calls you make are actually
>*hideously* expensive.
>
>
>Finally, the theoretical (data transmission only) PPP setup time at
>64Kbps sync would be approximately 30 milliseconds.
>
>For 9,600 bps async, it is theoretically as fast as 177 ms.  I measured
>in the 200-240 ms range in my implementation.
>
>It's a bit longer for Morningstar, or the NetBlazer, because they have
>an extra login step before PPP starts.
>
>Bill.Simpson@um.cc.umich.edu

Yes, we should be prepared to accommodate short calls because they are not
silly.  The telcos are starting to play games (read, "writing new tariffs")
with call setup costs and billing granularity.  Right now the standard
granularities are on the order of minutes but one second or less billing
granularity is just around the corner (some tariffs already exist).  The
bottom line is "the bottom line."  If call setup costs drop and they start
offering second or sub-second billing granularities it may turn out to be
cost effective to create and tear down links in only a few seconds.

Since we do not know what the future will bring in the way of tariffs, any
design should accommodate a wide variety of call durations, even those
lasting only a few seconds.  Please do not make the mistake of assuming
that the telecommunication industry is static and that today's tariffs will
apply tommorow.

Brian Lloyd                                    BP Lloyd & Associates, Inc.
brian@lloyd.com                                3420 Sudbury Road
brian@mail.barrnet.net                         Cameron Park, CA  95682
(916) 676-1147 - voice                         (916) 676-3442 - fax

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To: Karl Fox <karl@morningstar.com>
Cc: William Allen Simpson <bill.simpson@um.cc.umich.edu>, 
    ietf-ppp@ucdavis.edu, iplpdn@CNRI.Reston.VA.US
Subject: Re: Brutal Simplicity 
In-Reply-To: Your message of "Thu, 08 Jul 1993 13:39:45 EDT."
             <9307081739.AA29270@remora.MorningStar.Com> 
Date: Thu, 08 Jul 1993 16:15:48 -0400
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From: "Larry J. Blunk" <ljb@merit.edu>


> William Allen Simpson writes:
> > For 9,600 bps async, it is theoretically as fast as 177 ms.  I measured
> > in the 200-240 ms range in my implementation.
> > 
> > It's a bit longer for Morningstar, or the NetBlazer, because they have
> > an extra login step before PPP starts.
> 
> We usually recommend the extra login step because it's easier to
> understand and set up, but it's not required, and it's not what I use on
> my home system.

  It'd be nice if the terminal server vendors could get their act together
and support auto-recognition of PPP (Livingston seems to be the only
one doing it at this point).  PAP and CHAP are so much cleaner than
having the user setup some dumb login script (which is specific
to the server that they are dialing into).  How much code does
in take to check for  7E FF 7D 23 .... ???
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From: "k1io, FN42jk" <goldstein@carafe.tay2.dec.com>
To: iplpdn@CNRI.Reston.VA.US
Subject: Re: Brutal Simplicity
X-Vms-To: SMTP%"iplpdn@CNRI.Reston.VA.US"

>I've been a bit appalled by the notion that someone would want to bring
>up and tear down a circuit between every character, or even between
>every command.

>A 6 second or 19 second call is simply ridiculous.  I do not think that
>we should bother designing for such silly situations.

This is not silly at all.  It was silly in the old days of relay-
powered telephone lines.  With all-digital switching (which you
need for ISDN!), it's sensible....

>It sounds to me as if someone is trying to spoof the telco into giving
>them "free" time.  We should not encourage this, and it is likely to be
>self defeating in the long run (charges will become more punitive).

>The tariff structure here in the States includes a line setup
>"installation" charge, a monthly fee, a fee for the first minute (call
>setup), and a per 15 second fee after the first minute.  The call setup
>charge is usually about SIX TIMES the incremental charge, and can range
>as high as TWENTY times.

That is the "traditional" tariff structure.  It still applies to some
tariffs.  New Jersey Bell, for instance, has really low (relatively)
additional-minute rates, but really high first-minute rates.  The 
long distance carriers, on the other hand, charge the same for the 
first and additional minutes, or a penny more for the first.  They
get their per-call surcharges by the rounding-up effect.

A more modern tariff is applied by New England Telephone for intra-LATA
business toll calls in Massachusetts.  The price is quoted as 10.5c
per minute (5.5c after 50 hours/month/account) plus I think a penny a 
call.  Calls are timed to the second.  The bill is based on total
minutes (after adding up the seconds) plus total calls.  Maybe it's
6 cents/call, I don't remember offhand, but it's low.

>For standard analog local calls, there is no incremental charge.  Only
>the call setup is charged ($0.08-0.25 per call).  Calling every 19
>seconds would be extremely foolhardy.  On the contrary, you should NEVER
>hang up.

Even with antique tariffs like NJBell's, it sometimes makes sense to 
hang up, after a prolonged idle...

The point is that the US is beginning to see rates more like Europe's,
and more akin to modern costs.  High call setup fees date back to the
days of operator call setup.  The true cost is in the penny range.
For long distance, transmission time is more expensive.

The long-term future, I posit, will be "fast circuit" communications,
in which bandwidth is allocated for brief (packet-train, perhaps)
intervals of time.  Call setup is becoming cheaper and cheaper as
CPUs get faster.  What you pay your carriers today is more a vestige
of the past than a portent of the future.
   fred 
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Subject: IP option

I have come across what looks like and IP option type 58. I can not find
a reference to it. Does anyone have any idea what it means? Any body know
of a reference?

Thanks

Richard W. Cornetti              | cornetti@wg.com     | It is better to know  
Senior Marketing Engineer        | TEL: (919) 941-5730 | useless things than
Wandel & Goltermann Technologies | FAX: (919) 941-5751 | to know nothing.
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To: bsimpson@morningstar.com
Cc: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu
Subject: Re: Brutal Simplicity 
In-Reply-To: Your message of Wed, 07 Jul 1993 12:31:15 -0400.
             <1328.bill.simpson@um.cc.umich.edu> 
Date: Thu, 08 Jul 1993 16:48:53 -0400
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From: Dory Leifer <leifer@terminator.rs.itd.umich.edu>


Bill - While I agree with many of your points in this message, I want
to point out that there is no hard formula for what makes sense and
that flexibility to implement a given policy is key. I would hate to
make protocol design decisions because of a particular telco pricing
approach. It is even impossible to talk about "tariff structures in
the States" because of the wide range of carriers and carrier services.
I could, for example, quote rate structures that contradict your
assumptions and others that back it.

Dory

> I've been a bit appalled by the notion that someone would want to bring
> up and tear down a circuit between every character, or even between
> every command.
> 
> A 6 second or 19 second call is simply ridiculous.  I do not think that
> we should bother designing for such silly situations.
> 
> It sounds to me as if someone is trying to spoof the telco into giving
> them "free" time.  We should not encourage this, and it is likely to be
> self defeating in the long run (charges will become more punitive).
> 
> The tariff structure here in the States includes a line setup
> "installation" charge, a monthly fee, a fee for the first minute (call
> setup), and a per 15 second fee after the first minute.  The call setup
> charge is usually about SIX TIMES the incremental charge, and can range
> as high as TWENTY times.
> 
> For standard analog local calls, there is no incremental charge.  Only
> the call setup is charged ($0.08-0.25 per call).  Calling every 19
> seconds would be extremely foolhardy.  On the contrary, you should NEVER
> hang up.
> 
> Local connection office equipment is smart enough to know whether your
> call was actually connected, so you are not supposed to get charged for
> call setup if the call doesn't go through.  This means that you ALWAYS
> get charged if your call completes, even for very short periods of time.
> 
> Long distance companies (inter-LATA) are always charged for call setup
> by the local service provider -- whether or not the actual call goes
> through.  For marketing reasons, some will not charge if the call is
> less than a minute, because they can't tell if the call completed.  Most
> will ALWAYS charge regardless (to recoup the local connection cost).
> 
> Looking at the above, there is no reason to call for less than the time
> that the incremental charge is less than the average of the cumulative
> incremental charge and the setup charge.  In fact, the crossover point
> is at least 4 minutes, and often as long as 15 minutes.
> 
> For other "special" services, which have the incremental charge, if you
> don't use a reasonable amount of time, you are really wasting the
> installation and monthly charges, so the few calls you make are actually
> *hideously* expensive.
> 
> 
> Finally, the theoretical (data transmission only) PPP setup time at
> 64Kbps sync would be approximately 30 milliseconds.
> 
> For 9,600 bps async, it is theoretically as fast as 177 ms.  I measured
> in the 200-240 ms range in my implementation.
> 
> It's a bit longer for Morningstar, or the NetBlazer, because they have
> an extra login step before PPP starts.
> 
> Bill.Simpson@um.cc.umich.edu
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To: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu
Subject: Re: Brutal Simplicity 
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while i understand the desire to avoid un-necessary compilications in
order to allow extreme cases to work (like tear down on a per
character basis), please leave the legality, morality, or ethics of
the desire out of it. it is not the intention of the IETF to 
deal with these issues.



thanx.


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To: bsimpson@morningstar.com
cc: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu
Subject: Re: Brutal Simplicity
In-reply-to: Your message of "Wed, 07 Jul 93 12:31:15 EDT." <1328.bill.simpson@um.cc.umich.edu>
Date: Fri, 09 Jul 93 00:01:35 +0100
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From: Graham Knight <G.Knight@cs.ucl.ac.uk>
Message-ID:  <9307081853.aa26382@CNRI.Reston.VA.US>

Bill,

	In the UK we pay 4.935p per unit. Depending on time of day and
distance one unit buys between 220 secs and 19.2 secs for national
calls. For international calls the period drops to 2.19 secs in the
worst case. All charge periods cost the same - including the first one of a
call - so it is definitely worth timing out calls just before the end of
a charge period if the data seems to have dried up. (The 42 pound per
quarter per channel ISDN rental are on top of this).

	I don't think it is crazy for a customer to try to miminise
his/her cost of using a service so timeouts of 19 secs or less will
be used here. We should not design a solution which only makes sense
for the US tariff structure. Actually I don't think your solution does
do this since, as you point out, the PP setup time is small compared to
the ISDN call setup time.

						Graham
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To: bsimpson@morningstar.com
Cc: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu
Subject: Re: Brutal Simplicity
In-Reply-To: Your message of "Wed, 07 Jul 93 12:31:15 EDT." <1328.bill.simpson@um.cc.umich.edu>
Date: Fri, 09 Jul 93 00:01:35 +0100
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From: Graham Knight <G.Knight@cs.ucl.ac.uk>

Bill,

	In the UK we pay 4.935p per unit. Depending on time of day and
distance one unit buys between 220 secs and 19.2 secs for national
calls. For international calls the period drops to 2.19 secs in the
worst case. All charge periods cost the same - including the first one of a
call - so it is definitely worth timing out calls just before the end of
a charge period if the data seems to have dried up. (The 42 pound per
quarter per channel ISDN rental are on top of this).

	I don't think it is crazy for a customer to try to miminise
his/her cost of using a service so timeouts of 19 secs or less will
be used here. We should not design a solution which only makes sense
for the US tariff structure. Actually I don't think your solution does
do this since, as you point out, the PP setup time is small compared to
the ISDN call setup time.

						Graham
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To: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu
Reply-To: marc@ascend.com
Subject: Re: Brutal Simplicity 
In-Reply-To: Your message of "Fri, 09 Jul 1993 00:01:35 BST."
             <9307081853.aa26382@CNRI.Reston.VA.US> 
Date: Thu, 08 Jul 1993 18:13:14 -0700
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From: Marco S Hyman <marc@dumbcat.sf.ca.us>

Graham Knight writes:
 > 	I don't think it is crazy for a customer to try to miminise
 > his/her cost of using a service so timeouts of 19 secs or less will
 > be used here. We should not design a solution which only makes sense
 > for the US tariff structure.

OK, timeouts should be user setable.

 > Actually I don't think your solution does
 > do this since, as you point out, the PP setup time is small compared to
 > the ISDN call setup time.

And ISDN isn't the only digital service, and, at least from my perspective*,
interoperability over all switched digital services is required.  This means
that call setup time, in addition to setup cost, is another important
factor.

Four-wire switched 56 is *pulse* dialed and interoffice calls are likely to
be MF dialed between COs.  When it takes 10 seconds or more to set up a call
you don't want do it every few keystroke, damn the cost!

* I couldn't get BRI at home (PacBell land), nor two-wire switched 56.  The
  only game in town was four-wire switched 56.

// marc
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To: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu
Subject: Predicting the Future -- was Re: Brutal Simplicity 
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From: Frank Kastenholz <kasten@ftp.com>

This discussion seems amazingly irrelevant.  It appears to be all
predicated on assumptions of how subscribers behave in face of tariff
structures from telcos and other service providers. In other words,
if the tariff structure is foo then subscriber's behavior will be bar.

This discussion, then, is predicated on psychology, economics, and public
policy, not technology. These three things are notoriously hard to
predict (especially public policy, which is the domain of politicians
and bureaucrats...).

It seems to me that, since we can not predict with any certainty, what
the tariff environment will be like, we ought not develop a technology
that is limited to any one particular tariff regime, or highly pessimal
towards other tariff regimes.

The technology should allow for lots of short calls, it should allow for
fewer, long calls.



--
Frank Kastenholz
FTP Software
2 High Street
North Andover, Mass. USA 01845
(508)685-4000


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From: Joel Halpern <jmh@anubis.network.com>
Message-Id: <9307092128.AA24654@anubis.network.com>
To: iplpdn@IETF.CNRI.Reston.VA.US
Subject: Simplifying MultiLink

My apologies for not doing this up as a draft, and even further apologies
for not doing it much sooner.

I believe, subject to certain assumptions, that there is a simplication
of Keith's very useful MultiLink proposal, which will allow operation
over mismatched lines, and a range of behaviors, and will be more
implementable on the sending side.  

[The sending complication I am trying to avoid is the necessity for the
	sending process to know what has actually started and finished
	transmission on the actual links, as opposed to what has been
	queued.]

This simplication applies only to unreliable multilink.  This procedure
preserves order, but does not attempt to recover lost frames.  Full LAPB/F
procedures should be used for that problem.  The encapsulation is exactly
as specified by Keith.

The transmitter discipline is simply to use increasing sequence numbers.
No throttling to avoid windows is required.  It is recommended that something
be sent on every working link periodically.  (Thus, there is a minor
drawback if the charge for traffic is significant.)

On the receiver side, assuming for discussion that there are N parallel
links being used to distribute traffic (with our without link fragmentation),
then when a gap is noted in the sequence number:
	1) If a larger number is seen on every link, then the frame
	    with the missing value may be presumed lost
	2) If some link is not receiving traffic, and a time equal to the
	    link transfer (delay + largest bit latency for a frame) elapses,
	    then it may be presumed that the frame was not queued for that
	    link, and test 1 need be applied only to the remaining links.

Comments?
Thank you,
Joel M. Halpern			jmh@network.com
Network Systems Corporation
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Date: Sat, 10 Jul 93 13:43:45 EDT
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From: William Allen Simpson <bill.simpson@um.cc.umich.edu>
Message-Id: <1340.bill.simpson@um.cc.umich.edu>
To: iplpdn@CNRI.Reston.VA.US
Cc: ietf-ppp@ucdavis.edu
Reply-To: bsimpson@morningstar.com
Subject: Predicting the Future

> The technology should allow for lots of short calls, it should allow for
> fewer, long calls.
>
The PPP technology requires per call link setup packets.  The IPLPDN
folks think that (maximum reported) 600 milliseconds is too long.

Should I add a qualification to the PPP LCP introduction, stating:

   This option negotiation typically lasts 600 milliseconds on a low
   speed link, and has been reported to be less than 30 milliseconds on
   medium speed links.	PPP assumes that rapid switching of the link,
   such as might happen with mobility in a cell switched network or
   digital networks with low tariff granularity, is handled by the
   physical-layer, and is transparent to the link-layer.

Bill.Simpson@um.cc.umich.edu
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From: Oliver Korfmacher <okorf@netcs.com>
Message-Id: <9307141532.AA17421@keks.netcs.com>
Subject: Re: Brutal Simplicity
To: bsimpson@morningstar.com
Date: Wed, 14 Jul 93 17:32:17 MEST
Cc: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu
In-Reply-To: <1328.bill.simpson@um.cc.umich.edu>; from "William Allen Simpson" at Jul 7, 93 12:31 pm
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> Finally, the theoretical (data transmission only) PPP setup time at
> 64Kbps sync would be approximately 30 milliseconds.
Can you give the theory in simple words?

My feelings are the same (roughly 50milliseconds), using the following
estimation:

LCP
	send conf req
	send conf ack

NCP
	send conf req
	send conf ack

all theses packets are at least 11 octets without any options,
the receive conf ack/naks are omitted since they may interleave with
sending, agreed?
This gives 44 octets, at 8kbytes/sec minimal 55ms transmission time.

Don't misconceive me - I do not want to be pedanticly, but to
clarify my own understanding.

	Oliver
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From: Fred Baker <fbaker@acc.com>
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To: ietf-ppp@ucdavis.edu, iplpdn@nri.reston.va.us
Subject: Minutes of IPLPDN and PPP Joint Meetings
Cc: dave@mail.bellcore.com, ftev@ftp.com, minutes@nri.reston.va.us

Minutes of the Joint Meetings of the PPP Extentions and the IPLPDN
Working Groups at the 27th IETF, July 12-14 1993.

1.	RFC 1356 X.25

To be recommended as a Draft Standard.

Aware of 6-7 implementations with no inter-operability problems. 

RFC 1294 has already been recommended for advancement to Draft Standard.

2.	Protocol Discrimination

PPP/Q.922 NLPID value coming up. Bill Simpson needs to resend his
justification note to Lyman Chapin, but Lyman has agreed to make this
happen. There is a separate issue with the ISDN Lower Layer
Compatibility Information Element; George Clapp will pursue obtaining a
value indicating PPP.

IP/Circuit Switched Service

The question was seriously discussed whether we in fact need a default
way to send IP over circuit switched services such as ISDN B channel.
It was observed that the question is malformed; we do not need a
default way to send IP over a V.35 or V.11 interface, for example. We
need a way to speak to a peer system at the data link layer, which
might be a Frame Relay or X.25 switch, or a peer host or router.

We already have standards for PPP, Frame Relay, and X.25.

In different contexts, we are willing to run any of the three standards.

This approach is recommended for circuit switched services:

	Systems MUST implement PPP, on the assumption that circuit
	switched communications communications are generally {host or
	router} to {host or router}.

	Systems MAY implement other protocols such as Frame Relay or X.25

The implication here is not that all calls will be initiated with PPP
signaling and encapsulation, but that PPP signaling and encapsulation
will be a universally implemented option.

3.	Multi-link Protocol

Changed the header to one of the following:
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	|M|P|0|0| Sequence Number     |
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

	M - More  - 1 if a non-terminal fragment, 0 if the last fragment
	P - Phase - has the same value on each fragment of a message,
			inverts from message to message
	0 - Reserved, Must be zero
	Sequence Number - 0..4095 fragment sequence number

	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	|F|L|0|0| Sequence Number     |
	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

	F - First - 1 if first fragment in a message
	L - Last  - 1 if last fragment in a message
	0 - Reserved, Must be zero
	Sequence Number - 0..4095 fragment sequence number

Including a link in the multi-link group is done by authenticating
inclusion in the multi-link group and negotiation of the Fragmentation
Protocol Control Protocol (FPCP).

Removing a link from the multi-link group is done by terminating the
FPCP on that link.

In the worst case, receiver recovery from a sequence error (fragment
loss) is done by sending an FPCP Configure Request in the OPEN state on
all links; in most cases, one of the following two conditions is
sufficient to detect and step past the loss of a sequenced fragment:

	receipt of a frame on each link with a successor to the omitted
	sequence number

	the expiration of an implementation specific receipt timer; this
	should be long enough to handle the relevant timing issues

There is a separate LCP negotiation, authentication step, and set of 
Control Protocol negotiations for each link in a multi-link group.

Several other options were considered, including the use of the RFC
1294 fragmentation header, which was agreed on in the March meeting;
1294 provides the same essential features as this but requires four
octets, and additionally provides only compatibility with RFC 1294.

4.	PPP on Frame Relay

We need to have an applicability statement for PPP over Frame Relay, in
view of the existence of RFC 1294. The default encapsulation is as
described in the March IETFs minutes. Various edits were recommended,
which will be included in an updated draft, including collapsing of
Keith Sklowers parameter negotiation document (with attribution as
author) into this document.

LQM SHOULD NOT be used on a Frame Relay DLCI.

5.	PPP on X.25

We need to have an applicability statement for PPP over X.25 in view of
RFCs 877 and 1356. Various edits were recommended, which will be
included in an updated draft. Primary attention should be given to
reducing the size of the X.25 frame.

LQM SHOULD NOT be used in this environment.

The PPP NLPID SHOULD be placed in the call user data rather than being
carried in each frame.

6.	PPP/ISDN

Bill Simpson presented his paper on PPP over ISDN.

PPP must have the same default MRU (and any other defaults) on ISDN as
in other environments. Keith Sklower will publish his IPLPDN document
Determination of Encapsulation of Multi-Protocol Datagrams in Circuit
Switched Environment, and Bill indicates that he would like to copy
some of the technical material from them into this document. It was
decided that he would reference Keiths document.

7.	Parameter Negotiation

Keith and Bill will merge their documents. This document should be
separate from the PPP over <foo> documents, as it is desired to be
placed on the standards track, and the PPP over <foo> documents may not
be placed on that track.

Respectfully submitted

Fred Baker, PPP Chair
George Clapp, IPLPDN Chair
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From: Karl Fox <karl@morningstar.com>
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To: Fred Baker <fbaker@acc.com>
Cc: iplpdn@CNRI.Reston.VA.US, ietf-ppp@ucdavis.edu
Subject: Minutes of IPLPDN and PPP Joint Meetings
References: <9307151303.AA18101@saffron.acc.com>
Reply-To: Karl Fox <karl@morningstar.com>
Organization: Morning Star Technologies, Inc.

Fred Baker writes:
> Minutes of the Joint Meetings of the PPP Extentions and the IPLPDN
> Working Groups at the 27th IETF, July 12-14 1993.
...
> 3.	Multi-link Protocol
...
> There is a separate LCP negotiation, authentication step, and set of 
> Control Protocol negotiations for each link in a multi-link group.

When adding a second circuit to carry, say, IP traffic, do you just run
LCP, authentication, and FPCP, or do you also have to run IPCP?  I would
think you'd just piggyback on the earlier IPCP negotiations and avoid a
second (and potentially conflicting) set of IPCP parameters.
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From: "krishnaswamy,padma" <kri@blitzen.cc.bellcore.com>
To:      
X-To: ~iplpdn
Date:  1 Jul 1993  10:04 EDT
Subject: ds3's


-------------------- begin forwarded message --------------------
>From: blitzen!kri (krishnaswamy,padma)
To: ietf.nri.reston.va.us!iplpdn-request
Date:  1 Jul 1993   8:29 EDT
Subject: Re: Bandwidth available on DS-3 
Posting from memory, I think the question was about how much
data you can have in a DS3.Not all DS3's have to carry 28 DS1's-
especially not clear channel DS3's.
The nominal line rate in a DS-3 is 44.736. The DS3 overhead
functions occupy 1 bit in every 85; the frame is 
106.4 microseconds long- the actual 'data' bit rate is
84/85x44.736=44.210 Mbps
For further and better particulars, there are
a couple of ANSI docs- T1.102 and T1.107

Padma Krishnaswamy
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To: rolc@nsco.network.com, atm@sun.com, Big-Internet@munnari.oz.au, 
    iplpdn@IETF.CNRI.Reston.VA.US
Subject: Routing Over Large Clouds

As a consequence of several discussions at the last IETF, and new
WG/BOF is being formed.

This group will be focussed on the problems of routing over large
switched clouds. 

A charter for this group is being developed with the Routing Area
director.

To facilitate discussion, I have created the e-mail list, and am sending 
out this announcement.  The list is:
	rolc@network.com
Request for addition should be sent to:
	rolc-request@network.com
The Repostory for the Archive of this list, accessible via anonymous FTP is:
	nsco.network.com:/rolc/rolc.arc
Files of interest to the group will be made available there (in addition
to the formal internet-drafts, which of course will be available in the
standard repositories.)

The topics to be addressed by this group (probably) cover all aspects
of routing IP over "large clouds". In this context, large clouds
refer primarily to large public data services, such as future ATM, SMDS,
and Frame Relay (X.25 may also qualify).  In practice a network large
enough that a single router can not peer with all other routers is
subject to the same problems, and should be addressed by the same solutions.
(Thus, it is hoped that this will work on a VERY large bridged ethernet.)

Subtopics to be addressed include:

A) Architectural implications of relaxing the current requirement that
	communicating IP entites have a common NET number.

B) Routing over Demand Circuits.  Gerry Meyers work provides a starting
	point for making more effective use of the communication media
	when one is willing/able to peer.

C) Operating IS-IS over clouds.  Radia Perlman's draft on this should be
	reviewed.

D) Routing and Address Resolution.  This is an outgrowth of much previous
	work, including Directed ARP, Shortcut Routing, NBMA ARP, and
	architectural thinking by several IAB members.  The current
	proposal (to be fleshed out on the email list, with a clear
	proposal soon) is based oround a query/response protocol between
	routers, with MAC addresses carried.  Host operation is possible
	in several different ways (and we need to pick).
    [The number of individuals who have already contributed to this is
	too numerous to name, although some individual credits will
	probably appear in the proposal.  Apologies to those who feel
	that they have contribted much and should be mentioned by name.]

Note: The charter is still to be worked, so ideas for other areas should
be mailed directly to me.

Many people may receive multiple copies of this, but I feel the wide
coverage is needed, since the formal IETF announcement of the WG may be a
bit in the future.

Thank you,
Joel M. Halpern				jmh@network.com
Network Systems Corporation
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    iplpdn@IETF.CNRI.Reston.VA.US, jmh@anubis.network.com
Subject: Re: Routing Over Large Clouds

>This group will be focussed on the problems of routing over large
>switched clouds. 

I think the chief problem of routing over the clouds is brain-damage due to
oxygen deprivation. :-)

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subscribe

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To: Joel Halpern <jmh@anubis.network.com>
Cc: rolc@nsco.network.com, atm@sun.com, Big-Internet@munnari.oz.au, 
    iplpdn@IETF.CNRI.Reston.VA.US
Subject: Re: Routing Over Large Clouds 
In-Reply-To: Your message of "Thu, 22 Jul 93 14:28:33 CDT."
             <9307221928.AA04227@anubis.network.com> 
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From: Simon E Spero <ses@tipper.oit.unc.edu>

Don't forget, you have to circle O'hare fifty times for back-compatibility
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From: Henry Sinnreich <0002498337@mcimail.com>
To: Joel Halpern <jmh@anubis.network.com>
Cc: "David E. McDysan" <0002806303@mcimail.com>
Cc: David Greenfield <0004020699@mcimail.com>
Cc: "RND - RAD NETWORK DEVICES LTD." <0004373580@mcimail.com>
Cc: Jim Mollenauer <0004787822@mcimail.com>
Cc: "Michael E. Conn" <0004387451@mcimail.com>
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Subject: Re: Routing Over Large Clouds
Message-Id: <13930725003531/0002498337MT1EM@mcimail.com>

Please add me to the list - this is a very timely topic !
Thanks,

Henry Sinnreich
MCI Engineering
Richardson, Texas

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From: Keith Sklower <sklower@vangogh.cs.berkeley.edu>
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To: iplpdn@CNRI.Reston.VA.US
Subject: Multilink Discussion redirected

Since, as a consequence of the Amsterdam meeting, the Mutlilink
Protocol proposal uses PPP encapsulations (even for Frame Relay
and X.25), and since iplpdn is going on hiatus soon,
I am directing any more of my comments only to the ietf-ppp@ucdavis.edu list.
I am announcing there the location of a new draft reflecting what
agreement there was in Amsterdam.

'T's been fun!   See y'all later!
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Subject: Comments on Sklower Multilink
To: ietf-ppp@ucdavis.edu
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Keith,
Here are some comments on your draft.
This is quite close to being something I would be happy with.
tc

>
>
>Draft                 Simple Multilink             July 1993
>
>
>INTERNET DRAFT
>Expires: January 31st, 1994
>
	...

>This draft incorporates comments arising at  the  26th  IETF
                                                   ^^^^  
>meeting in Amsterdam.
>
27th IETF

	...

>There  are  proposals  being  advanced   by   communications
>providers  for  providing  synchronization  between multiple
>streams at the bit level;  such  features  are  not  as  yet
>widely  deployed  and  it  may  be useful to have a software
>solution as what may prove to be a stopgap measure.

'Stopgap' unnecessarily denegrates this document.

	...


>
>Furthermore, even if the ISDN  service  providers  guarantee
>bit-synchronization  between  different  switched  circuits,
>there may not be sufficiently flexible hardware  to  combine
>the  multiple  bit streams in arbitrary orders for HDLC bit-
>unstuffing.

There are plenty of time-space switches on the market.
What is problematical is the combination of variable rate
circuits.

	...
>
>6.1.  Crude Description of Overall Intent
'Crude' is self deprecating.


	...

>We suggest that multilink operation can be modeled as a sep-
>arate PPP Network-Layer protocol with its own control proto-
>col, which may be negotiated in the  usual  PPP  manner.   A
>slightly unusually convention (for PPP) might be to identify
                 ^^ 
	...

>Any  changes  to  the state of logical (group) link from the
>default state MUST be made by encapsulating LCP or NCP pack-
>ets inside Fragmentation Protocol packets.  (I.e. By placing
>LCP or NCP packet headers after the fragmentation  headers).
>
LCP packets should never be encapsulated.
In the recommended case, LCP, authentication CP, and the Multilink
control protocol would run over the physical link, all other NCPs
and NPs would run over the bundle. LCP should not be run on the
bundle. In some cases it might be acceptable to run some NCP/NP
pairs on physical links and other NCP/NP pairs on the bundle.
The even more complex case of running the same NCP or NP on both a
bundle and its constituent links would not likey be practical.
The recommended usage may be concisely represented by a table.

	On constituient links	On the bundle

	Link LCP		Bundle NCP
	Authentication NCP (1)	Compression NP/NCP (5)
	Link Quality NCP (1)	NCP/NP_a (6)
	Error Correction (2)
	Compression NP/NCP (3)
	NCP/NP_b (4)

(1) Optional, typical for bundled link
(2) Optional, may be needed by compression
(3) Optional, may reqire error correcting link
(4) Optional, atypical on link
(5) Optional, alternate to (3), still may require correcting links
(6) Optional, typical on bundle

	...

>However, in any case the sender MUST transmit fragments with
>non-decreasing sequence numbers over any constituent circuit
>in  a  multilink  arrangement.  In this section we present a

Increasing. Non-decreasing includes equal values.

	...

>The idea is that the receiver should keep track of the mini-
>mum of the sequence numbers of all fragments received on all
>channels participating in  the  multilink  procedure.  (Call
>this  M).   Every time M advances past a fragment bearing an
>(E)nding bit that  hasn't  otherwise  been  reassembled  and
>sent,  one  can  discard  all fragments with sequence number
>less than M.

So does this mean that as soon as all of packet B is received
that the partial fragments of packet A are discarded? How do you
know, simply on the basis of the sequence numbers, what links
were used to transmit which fragments? 

>The sender should transmit a null  fragment  increasing  the
>sequence  number  for  each  channel when the queue for each
>physical link becomes empty;  otherwise  the  receiver  will
>stall until new packets arrive.

This adds work at the transmitter, without much increase
in reliability. The null fragment will probably be lost for
the same reason as the the missing fragment it is pushing
along. Are the null packets part of some other packet? If not,
they should be some specific protocol.

>As this was intended to be a simple fragmentation/reassembly
>protocol,  we'll  just  mention that it would be possible to

Maybe you should make the goal viability rather than simplicity.

>
>have a more-elaborate timer and/or buffer size based  imple-
>mentations  (like the tradition IP reassembly queues in end-
>system implementations), but we'll leave that to more  ambi-
>tious future extensions.
>
The receiver can implement a buffer wrap scheme as suggested
above, or use a timer, without negotiation with the transmitter.
However, it would be useful to negotiate the maximum relative
window size in units of octets between channels, otherwise the
sender could place 100 fragments in one channel each containing
99% of a complete packet, then send the remaining 1% on another
channel. This is an extreme case, but asymmetric use of channels
is potentially useful. Negotiation of window size is equivalent
to negotiation of a timer, but is perhaps less suggestive of
implementation.

	...

>Precedence
>     FPCP packets may not be exchanged until the  Link  Con-
>     trol  Protocol  has  reached the network-layer Protocol
>     Configuration Negotiation  phase  (i.e.  Authentication
>     and Link Negotiation have concluded).  The FPCP negoti-
>     ation must precede any other network protocol  negotia-
>     tion.

Only for NCP/Nps over the bundle. NCPs on the link
can run in parallel.

>[The  author  of  this proposal realizes the title of it was
>the  ***SIMPLE***  multilink  protocol,  and  would  not  be
>gravely  disappointed if none of the options were permitted,
>but is enumerating them here merely for completeness' sake.]

Simplicity is in the eye of the beholder.

>
>6.4.1.  Sequenced Delivery Option
>
>          Figure 3: Sequenced Delivery Option
>
>            0                   1
>            0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
>           +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>           |     type=1    |  length = 2   |
>           +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
>
>This option requests the system at the other end of the link
>not to deliver packets out of sequence.  In the  absence  of
>reliable delivery, a missing fragment in the packet number N
>would delay the delay the  delivery  of  packet  number  N+1
>until receipt of any fragment for packet N+G, where G is the
>number of channels participating in the multilink procedure.

G should be a parameter. The size of the bundle is a nice default
but too restrictive. This makes a presumption about the distribution
of fragments over links.

>6.4.3.  Maximum Receive Reconstructed Unit
	...
>
>This option advises the peer that the implementation will be
>able to reconstruct a datagram consisting of  the  specified
>number of bytes.
>
?? Can the transmitter negotiate a smaller value?
The receiver could reduce its buffer space allocation accordingly.
	...

>The default value is 8192 bytes.
>
>Note:  this  option  controls the MRU of the logical (group)
>link, and could be negotiated by and  LCP  options  exchange
                                    ^
>inside  fragmentation  packets.  Having a protocol option at
>the FPCP allows  it  to  be  piggy-backed  onto  other  FPCP
>options.
>
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>I oppose the formation of a new working group for this area, unless it
>can be shown that existing groups are incapable of handling this task.

>This was the purpose of the IPLPDN WG.

IPLPDN folded after amsterdam. it is not the job of a WG in the
internet area to be doing work that is clearly in the routing domain.
we have a very nice man (bob, hold your hand up) who deals with the
routing end of the world.

>Extract from IPLPDN charter:

this will not be the first time that a group has selectively
implemented its charter.

>We never did see any *routing* work out of them.  Since you were a
>prominent member of that group, I don't see how making a new working
>group will improve the performance.

bill, even though i would not consider my interactions with joel to
be as great as those i have had with others, i have found him a very
strong and vocal proponent of what he feels is correct, while
maintaining the dignity and respect of both himself and the others
around him in a technical discussion. unfortunately, i have not found
this trait in your interactions with other members of WGs i have
observed you in. i woudl imagine that joel would make a wonderful
chair of this working group, and i have encouraged bob, remember
bob?, to let joel run this new working group.

>I suspect that we will see more turf wars as the new group conflicts
>with the efforts of existing WGs, and with the IPng groups, which are
>also working on the broader routing questions.

i have already suggested to the SIP group that any routing work it
does beyond just porting existing routing protocols, should be
carried out where the routing area (remember bob?) can be sure to
provide them with the guideance that they surely will need, since
routing is canonically a messy subject. (least, i always feel messy
when i leave a routing discussion.) i woudl also imagine that most
turf wars about routing will be lost by the internet area, if it
finds itself one of the sides. so, this shoudl not be a concern.


>I suggest that we disband the IPLPDN group, and focus on building the
>desired routing features into the various IPng proposals.


while i appreciate your feedback, and i have not discussed my position
with my partner, i will strongly resist putting routing work in
internet area WGs, other than that i outlined above (minimal changes
necessary for porting to the new architecture.)


i will keep your comments in mind, along with any other comments this
issue generates, and will be sure to let you know if my position is
changed by the weight of public opinion. as it stands now, the
process for creating this WG seems to be running correctly, since it
allowed you to register your complaint before the process had gone
too far. if there are a substantial number of complaints, i am sure
bob, dave (whom i have only mentioned obliquely as "my partner") and
myself will have a discussion about the viability of our current
course of action. the final resolution will, i am sure, be announced
where ever you read the first announcement.


stev knowles

still one of the internet area directors


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To: bill.simpson@um.cc.umich.edu
Subject: 
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From: Frank Kastenholz <kasten@ftp.com>
Reply-To: kasten@ftp.com
Cc: jmh@anubis.network.com, atm@sun.com, Big-Internet@munnari.oz.au, 
    iplpdn@IETF.CNRI.Reston.VA.US, iesg@CNRI.Reston.VA.US

Bill, at al.

History has shown that large, broadly focussed working groups tend to
be less productive than do tightly focussed, highly specific working
groups.

I think that this working group will be an excellent idea. Its members
can concentrate on their one, well defined, task and there will be a
much higher probability of them producing the right thing in a
reasonable time frame.

 > We never did see any *routing* work out of them.  Since you were a
 > prominent member of that group, I don't see how making a new working
 > group will improve the performance.

These sorts of comments are completely out of order here.

--
Frank Kastenholz
FTP Software
2 High Street
North Andover, Mass. USA 01845
(508)685-4000


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From: Noel Chiappa <jnc@ginger.lcs.mit.edu>
Message-Id: <9307301701.AA28659@ginger.lcs.mit.edu>
To: bill.simpson@um.cc.umich.edu, jmh@anubis.network.com
Subject: Routing Over Large Clouds
Cc: atm@sun.com, big-internet@munnari.oz.au, iesg@CNRI.Reston.VA.US, 
    iplpdn@IETF.CNRI.Reston.VA.US, jnc@ginger.lcs.mit.edu, rolc@network.com

    We never did see any *routing* work out of them. Since you were a
    prominent member of that group, I don't see how making a new working
    group will improve the performance.

I agree with your assessment that the IPLPDN group never got done what the
IESG had wanted it to do when it was created, but the blame for that lies in
many places, including me (as I'll explain), and I think it's really unfair
to beat up one individual member about it, even a vocal and influential one.
If you wanted to beat people up, you should beat up the Chair (for allowing
the group to get sidetracked), the Area Director (for not seeing that the
Chair followed the charter), etc, etc on up the ladder, and me (see below).
Now, in all fairness, all these people are volunteers, serving without
compensation, with other calls on their time, etc, etc, etc, but they are
some of the ones who are supposed to see that this doesn't happen. *All* the
WG members themselves also have to bear some blame; this isn't kindergarden,
and they are expected to help make things work.

    I would agree that IPLPDN has not handled the work assigned to them, but
    was diverted to simpler tasks outside their charter

Here's where I catch some blame. I was serving as Internet AD during the
period when this happened, and I was *supposed* to set up a number of WG's to
do things like define "IP over ATM", "IP over Frame Relay", "IP over ISDN",
etc. I didn't. It wasn't just obviously needed, I actually discussed it with
the IESG, said I would do so, and talked to potential chairs. I just never
followed through (for personal reasons having nothing to do with stuff inside
the IETF). I was "derelict in my duty", to use the technical phrase, and any
brickbats thrown in my direction for this will be deserved.

The inevitable happened: "Nature abhors a vacuum". Without the correct WG's
set up to do this in, people who needed it done drifted to IPLPDN and did it
there. Should 17 different people have seen this, and stepped in? Yup. Did
they try? Yes, actually. Did it do any good? No.


    creating several incompatible schemes for sending multi-protocol datagrams
    over LPDN. This just resulted in turf wars with other multi-protocol
    efforts.

Let's not get into this, but there are lots of people out there who think
bridging from one type of WAN technology to another is a Good Thing, and are
trying to harmonize packet formats to make this easier. I think they are wrong,
but there's no way to stop people trying it. Time will tell...


    I suspect that we will see more turf wars as the new group conflicts with
    the efforts of existing WGs, and with the IPng groups, which are also
    working on the broader routing questions.  I suggest that we disband the
    IPLPDN group, and focus on building the desired routing features into the
    various IPng proposals.

I'm less concerned with turf wars, and more with getting the darn problems
solved. Tying them into IPng will just create an even *bigger* ball of
politics and hassle, and slow down work on these technical problems. Yes,
all the IPng groups will need to make sure their schemes handle these issues,
but there's no more reason to put these things in IPng groups than there is
to bust up, say, multicast work and move it all to IPng groups.

The Right Thing *is* to dissolve IPLPDN, and set up two WG's: one each
*specifically* for i) address resolution mechanisms on large WAN's (since we
have current need for this, plus at least one IPng candidate with a very
determined camp of shortish-fixed-length-internetwork-address advocates), and
ii) routing mechanisms for large WAN's. The charters should *specifically
prohibit* any work on stuff outside the charters, and the Chair's and AD's
have to enforce these.

	Noel
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Date: Fri, 30 Jul 93 15:23:51 EDT
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From: William Allen Simpson <bill.simpson@um.cc.umich.edu>
Message-Id: <922.bill.simpson@um.cc.umich.edu>
To: Noel Chiappa <jnc@ginger.lcs.mit.edu>
Cc: atm@sun.com, big-internet@munnari.oz.au, iesg@CNRI.Reston.VA.US, 
    iplpdn@IETF.CNRI.Reston.VA.US, rolc@network.com
Reply-To: bsimpson@morningstar.com
Subject: Re: Routing Over Large Clouds

> I agree with your assessment that the IPLPDN group never got done what the
> IESG had wanted it to do when it was created, but the blame for that lies in
> many places, including me (as I'll explain), and I think it's really unfair
> to beat up one individual member about it, even a vocal and influential one.

OK.  I'll agree with that.  I like a fellow with the integrity to say
"the buck stops here" (to coin a phrase).  Your humility becomes you. ;-)

In this case, I wanted to make sure this wasn't "I didn't get my way, so
I'm forming another group."  We have too many working groups already,
it's impossible to attend all of the meetings and follow the mailing lists.

Also, it is very important that the other affected groups are taken into
consideration.	Every IPng will be affected.  The "cloud" problem isn't
fundamentally different from the "radio" problem, so mobile-ip has to be
consulted.  People have already been working on the problem.

> I'm less concerned with turf wars, and more with getting the darn problems
> solved. Tying them into IPng will just create an even *bigger* ball of
> politics and hassle, and slow down work on these technical problems. Yes,
> all the IPng groups will need to make sure their schemes handle these issues,
> but there's no more reason to put these things in IPng groups than there is
> to bust up, say, multicast work and move it all to IPng groups.
>
No, but since it may require fundamental paradigm shifts, I think this
should be delayed until we have picked IPng, and then all of the various
existing IETF WGs would work on the problems in that context, the same
as with multicast.

The one positive thing that I note in Stev Knowles' message is that the
WG will be in the Routing rather than the Internet Area.  What the
difference is, I don't know, but hopefully that will be more focused.

Bill.Simpson@um.cc.umich.edu
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To: bsimpson@morningstar.com
CC: jnc@ginger.lcs.mit.edu, atm@sun.com, big-internet@munnari.oz.au, 
    iesg@CNRI.Reston.VA.US, iplpdn@IETF.CNRI.Reston.VA.US, rolc@network.com
In-reply-to: <922.bill.simpson@um.cc.umich.edu>
Subject: Routing Over Large Clouds
Date: Sat, 31 Jul 1993 18:37:11 +0300
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From: Juha Heinanen <Juha.Heinanen@datanet.tele.fi>
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Message-ID:  <9307311136.aa10553@CNRI.Reston.VA.US>

i oppose delaying any IP developments "before we have picked IPng",
since noone knows when that will happen (if ever).  we have god knows
how many IP hosts out there which will have a long life still ahead of
them.

-- juha
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Subject: Routing Over Large Clouds
Date: Sat, 31 Jul 1993 18:37:11 +0300
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From: Juha Heinanen <Juha.Heinanen@datanet.tele.fi>
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Message-ID:  <9307311136.aa01859@IETF.CNRI.Reston.VA.US>

i oppose delaying any IP developments "before we have picked IPng",
since noone knows when that will happen (if ever).  we have god knows
how many IP hosts out there which will have a long life still ahead of
them.

-- juha
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Subject: Routing Over Large Clouds
Date: Sat, 31 Jul 1993 18:37:11 +0300
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From: Juha Heinanen <Juha.Heinanen@datanet.tele.fi>
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i oppose delaying any IP developments "before we have picked IPng",
since noone knows when that will happen (if ever).  we have god knows
how many IP hosts out there which will have a long life still ahead of
them.

-- juha
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Subject: Routing Over Large Clouds
Date: Sat, 31 Jul 1993 18:37:11 +0300
Sender:ietf-archive-request@IETF.CNRI.Reston.VA.US
From: Juha Heinanen <Juha.Heinanen@datanet.tele.fi>
X-Orig-Sender: Juha.Heinanen@datanet.tele.fi

i oppose delaying any IP developments "before we have picked IPng",
since noone knows when that will happen (if ever).  we have god knows
how many IP hosts out there which will have a long life still ahead of
them.

-- juha

