From owner-pilc@lerc.nasa.gov  Tue Jul  4 09:59:43 2000
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Date: Tue, 04 Jul 2000 08:54:51 -0400
From: Marie-Jose Montpetit <mariejose.montpetit@prodigy.net>
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I guess with the IETF coming the traffic on the list will grow soon. I
would like to submit a few discussion topics that are closely related to
IP performance over links that exhibit special characteristics. 

I have the impression that our focus up to now has been mostly on the
WAN. How about now looking at the Home Networks, the so called HANs
(Home Area Networks)? While I do believe that what the group has done in
the current drafts applies to the home ethernets and even the wireless
LANs (802.11 and Bluetooth) I think we may want to put our brains
together to make sure it does. I also think that the use of power lines
provided new challenges. While the current implementations are pretty
low bit rate the next generation (read this fall) is moving to the
megabits per second. The way I understand them the link layer of these
networks is the use of error protection mechanisms (FEC and what I would
classify as "fade protection" potentially variable rates) to ensure some
performance in spite of air conditioners or hair dryers and in general a
pretty noisy environment. These mechanisms will have impacts on TCP as
we well know. Do we want to add something in the LINK draft to cover
those or do we believe we have enough right now? There is probably other
issues in the HAN that may need our attention but may also be covered by
other groups.

Anyway my views.

Happy 4th of July!


Marie-Jose



Marie-Jose Montpetit
Nokia Venture Organization
marie-jose.montpetit@nokia.com/marie@mjmontpetit.com
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From owner-pilc@lerc.nasa.gov  Tue Jul  4 10:26:44 2000
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From: "Falk, Aaron" <aaron@PanAmSat.com>
Reply-To: marie@mjmontpetit.com
To: "'Marie-Jose Montpetit '" <mariejose.montpetit@prodigy.net>,
        "'pilc@grc.nasa.gov '" <pilc@grc.nasa.gov>
Subject: RE: New area for the group?
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Marie-Jose-

Well, in the satellite industry we sometimes talk about 'backhoe fades' to
refer to fiber cable breaks but I've never heard of 'hairdryer fades'
before. :^)

On a more serious note, PILC has made a consistant effort to abstract
specific link technologies (wireless, satellite, cable-modem, etc) into
their performance-affecting characteristics (low bit rate, high errors, etc)
rather than focus on the technologies themselves. This allows us to
understand better the performance problems that might be already solved by
other communities versus areas that might deserve more attention. This also
enables us to focus on addressing the performance problems in the most
general fashion.

If you could describe the link characteristics of HANs and any performance
issues they've created (in an email or I-D), this would make it easier for
us to evaluate whether the issues merit working group cycles.

Thanks,

--aaron

-----Original Message-----
From: Marie-Jose Montpetit
To: pilc@grc.nasa.gov
Sent: 7/4/00 8:54 AM
Subject: New area for the group?

I guess with the IETF coming the traffic on the list will grow soon. I
would like to submit a few discussion topics that are closely related to
IP performance over links that exhibit special characteristics. 

I have the impression that our focus up to now has been mostly on the
WAN. How about now looking at the Home Networks, the so called HANs
(Home Area Networks)? While I do believe that what the group has done in
the current drafts applies to the home ethernets and even the wireless
LANs (802.11 and Bluetooth) I think we may want to put our brains
together to make sure it does. I also think that the use of power lines
provided new challenges. While the current implementations are pretty
low bit rate the next generation (read this fall) is moving to the
megabits per second. The way I understand them the link layer of these
networks is the use of error protection mechanisms (FEC and what I would
classify as "fade protection" potentially variable rates) to ensure some
performance in spite of air conditioners or hair dryers and in general a
pretty noisy environment. These mechanisms will have impacts on TCP as
we well know. Do we want to add something in the LINK draft to cover
those or do we believe we have enough right now? There is probably other
issues in the HAN that may need our attention but may also be covered by
other groups.

Anyway my views.

Happy 4th of July!


Marie-Jose



Marie-Jose Montpetit
Nokia Venture Organization
marie-jose.montpetit@nokia.com/marie@mjmontpetit.com
 <<Card for Marie-Jose Montpetit>> 


From owner-pilc@lerc.nasa.gov  Tue Jul  4 12:17:44 2000
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I agree with the previous point that this working group has kept it's
link characteristic analysis general enough to apply to wireless
links.

We do research on these HAN wireless technologies in our
group. This includes IEEE 802.11, Bluetooth, HomeRF, and Hiperlan
(although expensive for home!!). These technologies are used in homes
and other small ad-hoc networks.

The PILC documents on links with errors, and slow links cover all the
aspects involved with these technologies quite well.  They also
correspond with the scientific literature in this area.  The PEP draft
goes further to look at techniques which can be used to improve
performance of these wireless networks as well. This draft did need
some improvement in regard to small wireless networks, but I have
already posted comments. (I can resend to anyone who didn't receive).

If there are more areas missed by these drafts it's best to make
comments on the mailing list?

Regards,
Zach Shelby

----------------------------------------------------------------------
   Zach Shelby - Researcher         Zach.Shelby@vtt.fi
   VTT Elektronics                  Phone   +358 8 551 2164 (Office)
   Kaitoväylä 1, PL 1100            Fax     +358 8 551 2320
   90571 Oulu, Finland              Mobile  +358 40 7796297 
----------------------------------------------------------------------
                    http://paula.oulu.fi/~zdshelby
       Home:   Yliopistokatu 14-707  90570 Oulu, Finland




From owner-pilc@lerc.nasa.gov  Fri Jul  7 08:00:51 2000
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From: Jong Hui Sian <engpac@nus.edu.sg>
To: "'rap@iphighway.com'" <rap@iphighway.com>
Subject: 8th IEEE International Conference on Networks 2000
Date: Fri, 7 Jul 2000 18:57:04 +0800 
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Dear Sir/Madam

Kindly ignore this if you have already registered to participatee in ICON
2000.



CALL FOR PARTICIPATION

* 8th IEEE International Conference on Networks 2000 * 

5 - 8 September 2000

Please visit : http:\\www.comp.nus.edu.sg\~icon
<http://www.comp.nus.edu.sg/~icon> 

 

We are pleased to invite you to attend the 8th IEEE International Conference
On Networks 2000 (ICON 2000) which will be held from September 5- 8, 2000 in
Singapore. It is organized by the IEEE Singapore Computer Chapter and the
Professional Activities Center, Faculty of Engineering, National University
of Singapore with co-operation from the Nanyang Technological University.
Two days of tutorials and two days of technical presentations are planned
for this conference. The conference will also include invited speakers,
exhibitions and special panel discussion sessions. 

The aim of the conference is to provide an international forum for experts
to promote, share and discuss various issues and developments in the broad
field of computer and communication networks.

	* * * 

	

ICON' 2000 will provide a forum to discuss recent advances in computer
networks and communications. Contributions describing original research,
surveys and applications including, but not restricted to, the following
areas are solicited



	

*	Active/intelligent networks 


	

	

*	BISDN and ATM 


	

	

*	Congestion/admission control 


	

	

*	Data traffic engineering and management 


	

	

*	High speed network protocols 


	

	

*	Internet services/applications 


	

	

*	Multicast/broadcast protocols 


	

	

*	Multimedia protocols and networking 


	

	

*	Multiple access technology 


	

	

*	Network architecture and protocol performance 


	

	

*	Network management and control 


	

	

*	Network monitoring 


	

	

*	Converged data and telecom networks 


*	Network security and privacy 


*	Network signaling and connection management 


*	Optical communication networks 


*	Personal communication services 


*	Quality of service 


*	Routing and routing protocols 


*	Storage area networks <outbind://8/san.htm>  


*	Testbeds and measurements 


*	Voice over internet 


*	Virtual private networks 


*	Wireless/mobile/satellite networks and protocols 


*	Pervasive networks 


*	Distributed storage 


	* * * 

	

More details of the conference including the Advance Program and
registration form may be found at our web site
http:\\www.comp.nus.edu.sg\~icon <http://www.comp.nus.edu.sg/~icon> . Please
do not hesitate to contact us should you need further information.

Thank you and looking forward to your attendance to IEEE ICON 2000.

 

Yours sincerely

IEEE ICON 2000 Secretariat

Tel : (65) 874 5113

Fax : (65) 874 5097

E-mail : engicon@nus.edu.sg <mailto:engicon@nus.edu.sg> 



From owner-pilc@lerc.nasa.gov  Thu Jul 13 23:55:09 2000
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To: pilc@grc.nasa.gov
Subject: "MobiCom 2000: Discounted registration ends on July 15" 
Date: Thu, 13 Jul 2000 22:47:25 -0400
From: Ashutosh Dutta <adutta@research.telcordia.com>
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(Our apologies for duplicate transmission)


                ANNOUNCEMENT AND CALL FOR PARTICIPATION

                             MobiCom 2000

            The Sixth Annual International Conference on
                   Mobile Computing and Networking
                        August 6 - 11,  2000

             Seaport Hotel at World Trade Center Boston
                      Boston, Massachusetts, USA

       THE COMPLETE ADVANCE PROGRAM AND REGISTRATION INFORMATION
                            ARE AVAILABLE AT
            ----------------------------------------------
            http://www.research.telcordia.com/mobicom2000/
            ----------------------------------------------

                      Sponsored by ACM SIGMOBILE
            In cooperation with ACM SIGCOMM and SIGMETRICS;
    IEEE Communications Society; the USENIX Association; the IEE (UK);
    the IEICE and IPSJ (Japan); the KICS (Korea); and the IFIP WG 6.3

           With support from:
           
            IBM  Research, HP  Laboratories,  Compaq,Nokia, TRW,  AT&T
            Laboratories Cambridge, Xerox  PARC, 3Com, BBN/GTE, Aether
            Systems, Nortel  Networks, Metricom, Boston Communications
            Group, Toshiba America Research,ThinAirApps


CONFERENCE HIGHLIGHTS
---------------------

o  Keynote Address:

   "Sentient Computing: The Interface is Everywhere," by Andy Hopper,
   Professor, University of Cambridge, and Managing Director, AT&T
   Laboratories Cambridge.

o  Technical Sessions:

   Twenty-eight technical papers will be presented describing
   previously unpublished research on a wide variety of topics in
   mobile computing and wireless networking, including prototype
   systems and networks, location support and data dissemination,
   packet scheduling and channel allocation, mobile internetworking,
   data management in mobile systems, and routing for ad hoc
   networks. Papers in a special session will challenge the community
   with new technologies and visionary applications. This year's
   conference received the largest number of paper submissions ever,
   resulting in a highly selective technical program.

o  Panels:

   We have organized two panels on timely issues with panelists who
   are passionate about these topics.

   P1: Comm'n Sense: Wireless Sensor Networks
       (Moderator: Deborah Estrin, UCLA and USC/ISI)
   P2: The Future Wireless Internet: Gazing Into the Crystal Ball
       (Moderator: Armando Fox, Stanford University)

o  Tutorials:

   There will be eight tutorials on cutting-edge topics:

   T1: Mobile IP for Current and Future Internet
       (David B. Johnson, CMU and Rice University)
   T2: Database Management Systems and Mobile Computing
       (Wang-Chien Lee, GTE Labs, Sandeep K.S. Gupta and Pradeep
        Srimani, Colorado State University)
   T3: Personal Area Networking Over Bluetooth
       (Pravin Bhagwat, AT&T Labs Research)
   T4: Service Discovery and Device Cooperation
       (Golden Richard III, University of New Orleans)
   T5: Energy Efficiency in Mobile Computing and Networking
       (Mani Srivastava, UCLA)
   T6: Mobile Voice over IP
       (Prathima Agrawal, Telcordia, Parmesh Ramanathan, University of
        Wisconsin, and Cormac J. Sreenan, University College Cork)
   T7: Mobile Ad Hoc Networks: Routing, MAC and Transport Issues
       (Nitin Vaidya, Texas A&M University)
   T8: Shaping the User Experience for Handheld Computing
       (Phillip B. Shoemaker, Palm Inc.)

o  Research Demos and Exhibits:

   The conference will also feature research demos from academic and
   industry research groups as well as the newest cutting edge
   products and services from a wealth of companies.

o  Workshops:

   Tackling the dominant issues of the day, the following workshops
   will allow extended considerations of particular topics.

   - Dial-M:  Discrete Algorithms and Methods for Mobile Computing
              and Communications
   - WoWMoM:  Wireless Mobile Multimedia
   - MSWiM:   Modeling Analysis and Simulation of Wireless and Mobile
              Systems
   - MobiHOC: Mobile Ad Hoc Networking and Computing

REGISTRATION
------------

The MobiCom 2000 web pages have all the conference registration and
hotel reservation specific information:

   http://www.research.telcordia.com/mobicom2000/

Important Dates:
   July 7,  2000 --  Discounted hotel reservation deadline.
   July 15, 2000 --  Early conference registration deadline.

Don't delay -- register today for MobiCom 2000. Come, celebrate the
wireless revolution and its convergence with the Internet, on Boston's
historic waterfront. It is THE conference to attend. We look forward to
seeing you in Boston.







From owner-pilc@lerc.nasa.gov  Mon Jul 17 08:08:56 2000
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A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: A TCP profile for W-CDMA: 3G wireless packet service.
	Author(s)	: H. Inamura, T. Ishikawa
	Filename	: draft-inamura-docomo-00.txt
	Pages		: 9
	Date		: 14-Jul-00
	
The third generation, 3G, wireless networks that enable high-speed
data transfer are now being introduced.  Internet access is one of
the most important applications of such networks.  TCP is one of the
key technologies that enable Internet applications for such networks.
A number of TCP optimization techniques have been studied to enhance
the performance of TCP transmission for various wireless
environments.

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This draft is a work item of the Performance Implications of Link Characteristics Working Group of the IETF.

	Title		: Advice for Internet Subnetwork Designers
	Author(s)	: J. Touch, M. Montpetit, J. Mahdavi,
                          G. Montenegro, D. Grossman, G. Fairhurst
	Filename	: draft-ietf-pilc-link-design-03.txt
	Pages		: 21
	Date		: 18-Jul-00
	
This document provides advice to the designers of digital
communication equipment, link layer protocols and packet switched
subnetworks (collectively referred to as subnetworks) who wish to
support the Internet protocols but who may be unfamiliar with the
architecture of the Internet and the implications of their design
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From owner-pilc@lerc.nasa.gov  Wed Jul 19 12:04:42 2000
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Date: Wed, 19 Jul 2000 16:52:33 +0200
To: pilc@grc.nasa.gov
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Comments to "Advice for Internet Subnetwork Designers"
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thanks to the authors for the well written draft!

Some comments below.

///Reiner

1. The subnetwork's retransmission persistency:

A subnetwork's retransmission persistency, i.e., for how long should a link 
layer delay (e.g., in terms of milliseconds) an IP packet in an attempt to 
transmit it over the link before it discards the packet, is an important 
issue for subnetwork designers. However, I feel that the draft is too fague 
on this matter:
"It is important to recognize that a
    subnetwork can go too far in attempting to provide error recovery
    services in the Internet environment.  Subnet reliability should be
    "lightweight", i.e., it only has to be "good enough", *not* perfect."

A more concrete alternative would be to say something along the following 
lines:

Subnets that do not distinguish between delay sensitive (e.g., real-time or 
streaming multimedia) and reliable flows (e.g., TCP, Reliable Multicast) 
should implement a low retransmission persistency (on the order of a few 10 
milliseconds) [IS95] [GSM]. This is necessary to not interfere with the 
delay requirements of delay sensitive flows. On the other hand, subnetworks 
that are capable of separating reliable flows should implement a high 
retransmission persistency for such flows. However, a single IP packet may 
never be delayed by more than the Maximum Segment Lifetime (MSL) of 240 
seconds defined for TCP [RFC1122].

2. Minor detail in the section on "How TCP Works":

There is says: "Queues provide output-buffering on links that are 
momentarily oversubscribed."
However, queues also provide input-buffering on links and in fact packets 
can get dropped there, too, when the host/router is at high processor load.

3. Recovery from Subnetwork Outages:

To the sentence: "If it cannot
    do this, then the subnetwork should hold onto at least some of the
    packets it accepts during an outage and attempt to deliver them when
    the subnetwork comes back up.", I propose to add the following sentence:

However, none of such packet may be held for more than the Maximum Segment 
Lifetime (MSL) of 240 seconds defined for TCP [RFC1122].

Instead of the sentence: " Note that it is *not* necessary to avoid any and 
all packet drops during an outage.", I propose the following refinement:

As long as the queue management policy allows (never queue packets without 
reasonable bounds, e.g., not more than a common bandwidth*delay product) 
the subnetwork should avoid dropping any packets. Beyond that it should 
start dropping from front to give an early congestion signal to the TCP sender.

4. The "Quality of Service, ..." section

Although, I very much like the section and wish we could really implement 
it that way, I don't understand the discussion about the TOS field [RFC1349].

The definitions of the TOS field are obsolete and have been replaced by the 
definition of the differentiated service field [RFC2474 (proposed standard)] :-(



From owner-pilc@lerc.nasa.gov  Wed Jul 19 13:04:16 2000
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Date: Wed, 19 Jul 2000 17:08:37 +0100 (BST)
From: Lloyd Wood <l.wood@eim.surrey.ac.uk>
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On Wed, 19 Jul 2000, Reiner Ludwig wrote:

> 1. The subnetwork's retransmission persistency:
> 
> A subnetwork's retransmission persistency, i.e., for how long should a link 
> layer delay (e.g., in terms of milliseconds) an IP packet in an attempt to 
> transmit it over the link before it discards the packet, is an important 
> issue for subnetwork designers. 

link != subnetwork...

> However, I feel that the draft is too fague 
> on this matter:
> "It is important to recognize that a
>     subnetwork can go too far in attempting to provide error recovery
>     services in the Internet environment.  Subnet reliability should be
>     "lightweight", i.e., it only has to be "good enough", *not* perfect."
> 
> A more concrete alternative would be to say something along the following 
> lines:
> 
> Subnets that do not distinguish between delay sensitive (e.g., real-time or 
> streaming multimedia) and reliable flows (e.g., TCP, Reliable Multicast) 

how would you recommend that such subnetworks distinguish between
these two reliably? Aren't 'delay-sensitive' and 'reliable' entirely
orthogonal criteria? (this leads into a discussion of how e.g.
diffserv specifications of same could be interpreted by the subnet -
arguably out of scope, but see discussion of QoS below.)


> should implement a low retransmission persistency (on the order of a few 10 
> milliseconds) [IS95] [GSM]. This is necessary to not interfere with the 
> delay requirements of delay sensitive flows. On the other hand, subnetworks 
> that are capable of separating reliable flows should implement a high 
> retransmission persistency for such flows.

hey, that would really kill TCP over satellite's performance; RTT
estimates would be thrown right off. Practically, you're getting in
the way of the ramifications of the end-to-end argument; a lot more
careful text would be needed here.

Or, if we can't come up with text that is careful enough (discussing
subnet error rates and contrasting those with the high reliability
expected by IP-based transports in the light of delay overhead
tradeoffs incurred by the subnet), we opt for saying as little as
possible, above.

(there's some relevant text at the end of the
  'Analysis of Link Layer Effects on TCP Performance'
section that could IMO do with moving here.)


> 4. The "Quality of Service, ..." section
> 
> Although, I very much like the section and wish we could really implement 
> it that way, I don't understand the discussion about the TOS field [RFC1349].
> 
> The definitions of the TOS field are obsolete and have been replaced by the 
> definition of the differentiated service field [RFC2474 (proposed standard)] :-(

This was discussed on the mailing list some time back; did
the material get lost in editing? dan's on the author list...

See the thread starting 
http://pilc.grc.nasa.gov/pilc/list/archive/0774.html
and in particular Dan Grossman's text
http://pilc.grc.nasa.gov/pilc/list/archive/0803.html
(which probably needs updating/trimming, the way diffserv terminology
is going...)


Other stuff:

Referencing/discussing ramifications of Bennett's ToN paper in the
packet reordering section seems like a good idea.
'Packet Reordering is not Pathological Network Behavior' Bennett,
Partridge, Sheetman, IEEE/ACM Transactions on Networking,
December 1999, (7):6, pp. 789-798


  Examples of such subnetworks include
  ATM, with a single frame size of 48 bytes plus a 5-byte header,
 
having already referenced AAL encapsulation etc, this example sits
oddly - especially since you're talking about AAL5 several paras
later.

TCP-friendly needs a reference discussing throughput considerations
when first mentioned, and explicitly relating to the 1/sqrt(p) eqn.

   Since ACKs are generally smaller than data
   segments, TCP can tolerate some asymmetry, but as a general rule
   designers of subnetworks should avoid large differences in the
   incoming and outgoing bandwidth.

This only becomes a problem when the forward sending: backchannel acks
capacity ratio exceeds 50:1 [*], but it's worth pointing out that
assumptions about which way TCP is flowing are often invalidated by
pesky users. (Hence ADSL networks banning webservers, napster serving 
etc.)

[*] R. C. Durst, G. J. Miller, E. J. Travis 'TCP extensions for space
    communications', Mobicom '96, Nov '96 - yesyes, conference
    papers are not easily obtainable and not much point in referencing
    in an RFC - that goes for the MONET paper too.

   Using the delayed acknowledgment mechanism {Bra89], which reduces
   the number of ACKs transmitted by the receiver by roughly half, can
   also improve performance by reducing the congestion on the ACK
   channel. These mechanisms should be employed in asymmetric
   networks.

Actually, delacks are recommended use anyway; not much point in giving
endhost hints to subnetwork designers who can't control their
endhosts.

   The other *criterion* for native multicast *is* a link-layer
   filter,


   References of the form RFCnnnn are Internet Request for Comments
   (RFC) documents available online at www.rfc-editor.org.

little point in saying this _in_ an RFC. "Here's how to get this
document you're already reading".

[APS99] should be [RFC2581]

'Security' section is before references, 'Security Considerations' is
after. odd.

cheers,

L.

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









From owner-pilc@lerc.nasa.gov  Thu Jul 20 09:53:58 2000
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From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
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At 17:08 19.07.00 +0100, Lloyd Wood wrote:
> > Subnets that do not distinguish between delay sensitive (e.g., 
> real-time or
> > streaming multimedia) and reliable flows (e.g., TCP, Reliable Multicast)
>
>how would you recommend that such subnetworks distinguish between
>these two reliably?

Good question. The protocol-ID field in the IP header can be used as an 
approximation to at least separate TCP flows from the rest. However, this 
has some problems:
- IPsec overwrites the proto-ID field, and
- other reliable flows (NFS, Reliable Multicast) run on top of UDP and can 
thus not be identified as reliable flows.

It would be so much better to have the old TOS definitions deployed ... 
which unfortunately is not the case.

>Aren't 'delay-sensitive' and 'reliable' entirely
>orthogonal criteria?

Not when it comes to the question about how persistent link layer ARQ 
should be.

> > should implement a low retransmission persistency (on the order of a 
> few 10
> > milliseconds) [IS95] [GSM]. This is necessary to not interfere with the
> > delay requirements of delay sensitive flows. On the other hand, 
> subnetworks
> > that are capable of separating reliable flows should implement a high
> > retransmission persistency for such flows.
>
>hey, that would really kill TCP over satellite's performance;

I don't believe that. Has it ever been shown?

>RTT
>estimates would be thrown right off.

Clearly, the measured RTT may vary more than without link layer ARQ. But 
that's what it should do for such a path because it only reflects the true 
RTT for sending packets reliably e2e. It will lead to a more conservative 
RTO which is exactly what should happen on such a path. Also, note that the 
RTO will drop quite sharply back closer to the "real RTT" after a sudden 
RTT spike so it's not a real problem.

BTW, the draft's proposal to counter link outages (holding on to packets 
until after the outage and then to forward them) is basically the same 
thing as highly persistent link layer ARQ. That mechanism will also cause 
considerable RTT spikes. I don't see a problem with that. It merely 
reflects the path's properties.

///Reiner



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Lloyd Wood wrote:

> On Wed, 19 Jul 2000, Reiner Ludwig wrote:
>
> > 1. The subnetwork's retransmission persistency:
> >
> > A subnetwork's retransmission persistency, i.e., for how long should a link
> > layer delay (e.g., in terms of milliseconds) an IP packet in an attempt to
> > transmit it over the link before it discards the packet, is an important
> > issue for subnetwork designers.
>
> link != subnetwork...
>
> > However, I feel that the draft is too fague
> > on this matter:
> > "It is important to recognize that a
> >     subnetwork can go too far in attempting to provide error recovery
> >     services in the Internet environment.  Subnet reliability should be
> >     "lightweight", i.e., it only has to be "good enough", *not* perfect."
> >
> > A more concrete alternative would be to say something along the following
> > lines:
> >
> > Subnets that do not distinguish between delay sensitive (e.g., real-time or
> > streaming multimedia) and reliable flows (e.g., TCP, Reliable Multicast)
>
> how would you recommend that such subnetworks distinguish between
> these two reliably? Aren't 'delay-sensitive' and 'reliable' entirely
> orthogonal criteria? (this leads into a discussion of how e.g.
> diffserv specifications of same could be interpreted by the subnet -
> arguably out of scope, but see discussion of QoS below.)
>
> > should implement a low retransmission persistency (on the order of a few 10
> > milliseconds) [IS95] [GSM]. This is necessary to not interfere with the
> > delay requirements of delay sensitive flows. On the other hand, subnetworks
> > that are capable of separating reliable flows should implement a high
> > retransmission persistency for such flows.

--- or should we say "better" reliability scheme (a higher persistence is one
option, BUT too high is still dangerous - additional FEC is another option,
hybrid ARQ/FEC schemes are also suitable in some environments, ...).

>
> hey, that would really kill TCP over satellite's performance; RTT
> estimates would be thrown right off. Practically, you're getting in
> the way of the ramifications of the end-to-end argument; a lot more
> careful text would be needed here.
>
> Or, if we can't come up with text that is careful enough (discussing
> subnet error rates and contrasting those with the high reliability
> expected by IP-based transports in the light of delay overhead
> tradeoffs incurred by the subnet), we opt for saying as little as
> possible, above.

It may be difficult to convince me that this document should start proposing
specific solutions, if we do that we run the danger of giving the wrong advice.
At the moment we have advice based on many years of people researching AND
implementing these schemes.  If we move towards what is currently good research,
then we stand the risk of having to be very specific about the types of links we
are considering. I suggest we don't do that - perhaps we should once we have this
draft finished?

>
> (there's some relevant text at the end of the
>   'Analysis of Link Layer Effects on TCP Performance'
> section that could IMO do with moving here.)
>
> > 4. The "Quality of Service, ..." section
> >
> > Although, I very much like the section and wish we could really implement
> > it that way, I don't understand the discussion about the TOS field [RFC1349].
> >
> > The definitions of the TOS field are obsolete and have been replaced by the
> > definition of the differentiated service field [RFC2474 (proposed standard)] :-(
>
> This was discussed on the mailing list some time back; did
> the material get lost in editing? dan's on the author list...
>
> See the thread starting
> http://pilc.grc.nasa.gov/pilc/list/archive/0774.html
> and in particular Dan Grossman's text
> http://pilc.grc.nasa.gov/pilc/list/archive/0803.html
> (which probably needs updating/trimming, the way diffserv terminology
> is going...)
>
> Other stuff:
>
> Referencing/discussing ramifications of Bennett's ToN paper in the
> packet reordering section seems like a good idea.
> 'Packet Reordering is not Pathological Network Behavior' Bennett,
> Partridge, Sheetman, IEEE/ACM Transactions on Networking,
> December 1999, (7):6, pp. 789-798

Excellent idea - this paper at least identifies order/delay are variable
quantities in today's deployed Internet. Perhaps something derived from::

IP networks are allowed to reorder packets. Most link protocols provide in sequence
delivery when used over a single physical link, which guarantees order. Subnets with
alternate
paths do not necessarily provide ordering [Benett]., neither does IP-forwarding when
parallel processing is used [Benett].

My feeling was still that ordering was DESIRABLE, in most cases (for performance, and
expected behaviour of transport protocols). But not required in ANY.

You may disagree :-)



>
>
>   Examples of such subnetworks include
>   ATM, with a single frame size of 48 bytes plus a 5-byte header,
>
> having already referenced AAL encapsulation etc, this example sits
> oddly - especially since you're talking about AAL5 several paras
> later.
>
> TCP-friendly needs a reference discussing throughput considerations
> when first mentioned, and explicitly relating to the 1/sqrt(p) eqn.
>
>    Since ACKs are generally smaller than data
>    segments, TCP can tolerate some asymmetry, but as a general rule
>    designers of subnetworks should avoid large differences in the
>    incoming and outgoing bandwidth.
>
> This only becomes a problem when the forward sending: backchannel acks
> capacity ratio exceeds 50:1 [*], but it's worth pointing out that
> assumptions about which way TCP is flowing are often invalidated by
> pesky users. (Hence ADSL networks banning webservers, napster serving
> etc.)
>
> [*] R. C. Durst, G. J. Miller, E. J. Travis 'TCP extensions for space
>     communications', Mobicom '96, Nov '96 - yesyes, conference
>     papers are not easily obtainable and not much point in referencing
>     in an RFC - that goes for the MONET paper too.
>
>    Using the delayed acknowledgment mechanism {Bra89], which reduces
>    the number of ACKs transmitted by the receiver by roughly half, can
>    also improve performance by reducing the congestion on the ACK
>    channel. These mechanisms should be employed in asymmetric
>    networks.
>
> Actually, delacks are recommended use anyway; not much point in giving
> endhost hints to subnetwork designers who can't control their
> endhosts.

This could be addressed in a revised asymmetry document?

>
>
>    The other *criterion* for native multicast *is* a link-layer
>    filter,
>

I don't agree, because many of the links I have worked with have been single-hop,
and that doesn't imply you have do anything at the link layer except broadcast the
multicast packet - IP-layer filtering would work fine (if it could be done
efficiently).

I may be out-voted though :-)

See my suggestions for a revised multicast section....

(see mail to this list on : Mon, 08 May 2000 15:05:14 +0100

Perhaps it will stimulate wider discussion?



>
>    References of the form RFCnnnn are Internet Request for Comments
>    (RFC) documents available online at www.rfc-editor.org.
>
> little point in saying this _in_ an RFC. "Here's how to get this
> document you're already reading".
>
> [APS99] should be [RFC2581]
>
> 'Security' section is before references, 'Security Considerations' is
> after. odd.
>
> cheers,
>
> L.
>
> <L.Wood@surrey.ac.uk>PGP<http://www.ee.surrey.ac.uk/Personal/L.Wood/>

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



From owner-pilc@lerc.nasa.gov  Thu Jul 20 11:09:13 2000
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On Thu, 20 Jul 2000, Reiner Ludwig wrote:

> >hey, that would really kill TCP over satellite's performance;
> 
> I don't believe that. Has it ever been shown?
> 
> >RTT estimates would be thrown right off.
> 
> Clearly, the measured RTT may vary more than without link layer ARQ. But 
> that's what it should do for such a path because it only reflects the true 
> RTT for sending packets reliably e2e. It will lead to a more conservative 
> RTO which is exactly what should happen on such a path. 

The TCP flow taking that path will compete with other flows of varying
length on the (congested) terrestrial part of the internetwork.

Given TCP's bias against connections with long RTTs (e.g., see 
http://www.cs.berkeley.edu/~tomh/papers/globecom98_abstract.html) this
is often a problem for bulk data transfers.

If the RTTs of flows across your subnetwork are already extremely long
relative to the majority of other flows in the internetwork, you
should think twice before making them any longer imo.


> BTW, the draft's proposal to counter link outages (holding on to packets 
> until after the outage and then to forward them) is basically the same 
> thing as highly persistent link layer ARQ. That mechanism will also cause 
> considerable RTT spikes. I don't see a problem with that. 

A lot depends on the link delay: outage delay ratio.

L.

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








From owner-pilc@lerc.nasa.gov  Thu Jul 20 14:06:01 2000
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Reiner Ludwig wrote:
> 
...
> 
> It would be so much better to have the old TOS definitions deployed ...
> which unfortunately is not the case.
> 

A nice thing about the redefinition of the TOS octet is
that it is certainly possible to encode whatever you
want in the DSCP.  It's more "economical" in use of
bit-patterns than the old definition. I believe that
RFC 2474 gives you enough rope to go do whatever you'd
like.

	Kathie


From owner-pilc@lerc.nasa.gov  Thu Jul 20 15:32:29 2000
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Reiner Ludwig wrote:
	[..]
> It would be so much better to have the old TOS definitions deployed ...
> which unfortunately is not the case.

Elaborating on Kathie's comment, although the DSCP doesn't have
predefined QoS semantics like the old ToS byte, this isn't such a
limitation. Consider a link interface/driver that peaks into the
DSCP as an IP-aware entity. Therefore if it is part of a Diffserv
domain, it ought to be configured for Diffserv QOS just like the
diffserv routers and hosts that surround it. So the advice
to subnetwork designers is "Allow your drivers to map from DSCPs
into 'delay sensitive', etc, classes using operator configured
mappings."   (Clearly serious word mangling is required here, but
hopefully the intent is clear...)

cheers,
gja
________________________________________________________________________
Grenville Armitage                    http://members.home.net/garmitage/
Bell Labs Research Silicon Valley


From owner-pilc@lerc.nasa.gov  Fri Jul 21 03:20:58 2000
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To: Kathleen Nichols <nichols@packetdesign.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
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At 10:04 20.07.00 -0700, Kathleen Nichols wrote:
>A nice thing about the redefinition of the TOS octet is
>that it is certainly possible to encode whatever you
>want in the DSCP.  It's more "economical" in use of
>bit-patterns than the old definition. I believe that
>RFC 2474 gives you enough rope to go do whatever you'd
>like.

The problem I see with the DS field is that every border router is free to 
re-write it. This might make it difficult to mark my IPsec-encoded TCP-flow 
with the QoS requirement "maximize reliability" e2e over the entire path.

///Reiner



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on this topic, I would like to draw your attention to section 4.1 
("Extending the Differentiated Service Framework") of my dissertation which 
you can fetch here 
http://iceberg.cs.berkeley.edu/papers/Ludwig-Diss00/index.html.

It discusses pretty much what you describe, Grenville.

BTW, section 4.1.4 shows measured traces of how "holding on to packets 
during link outages" works so much better than "dropping packets during 
link outages".

///Reiner

At 11:36 20.07.00 -0700, Grenville Armitage wrote:


>Reiner Ludwig wrote:
>         [..]
> > It would be so much better to have the old TOS definitions deployed ...
> > which unfortunately is not the case.
>
>Elaborating on Kathie's comment, although the DSCP doesn't have
>predefined QoS semantics like the old ToS byte, this isn't such a
>limitation. Consider a link interface/driver that peaks into the
>DSCP as an IP-aware entity. Therefore if it is part of a Diffserv
>domain, it ought to be configured for Diffserv QOS just like the
>diffserv routers and hosts that surround it. So the advice
>to subnetwork designers is "Allow your drivers to map from DSCPs
>into 'delay sensitive', etc, classes using operator configured
>mappings."   (Clearly serious word mangling is required here, but
>hopefully the intent is clear...)
>
>cheers,
>gja
>________________________________________________________________________
>Grenville Armitage                    http://members.home.net/garmitage/
>Bell Labs Research Silicon Valley



From owner-pilc@lerc.nasa.gov  Fri Jul 21 04:27:34 2000
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Reiner Ludwig wrote:
> 
> on this topic, I would like to draw your attention to section 4.1
> ("Extending the Differentiated Service Framework") of my dissertation which
> you can fetch here
> http://iceberg.cs.berkeley.edu/papers/Ludwig-Diss00/index.html.
> 
> It discusses pretty much what you describe, Grenville.

Then I am confused, since it was you who wrote:

> > > It would be so much better to have the old TOS definitions deployed ...
> > > which unfortunately is not the case.

To which Kathie and I have observed it is infact better now, since
the DSCP is more flexible. So which situation do you really believe
in - that its good we have the DSCP, or we ought to return to the
ToS semantics?

Also, scanning your chapter 4 there's an odd statement on p.71 where
you suggest adding four *new* DS Field values to represent
four service classes. This is not at all what I was talking about.
The service classes you discuss can be adequately supported using
the existing PHBs (and indeed, Diffserv has always clearly stated that
PHBs/DSCPs do not represent e2e service classes as classically
understood). My previous email outlined that one can make a configured
or signalled association between particular (pre-existing )DSCPs and
the 'service class' that happens to be being mapped onto said DSCPs
on a case by case (and reconfigurable) basis.

cheers,
gja
________________________________________________________________________
Grenville Armitage                    http://members.home.net/garmitage/
Bell Labs Research Silicon Valley


From owner-pilc@lerc.nasa.gov  Fri Jul 21 04:43:54 2000
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Reiner Ludwig wrote:
	[..]
> The problem I see with the DS field is that every border router is free to
> re-write it. This might make it difficult to mark my IPsec-encoded TCP-flow
> with the QoS requirement "maximize reliability" e2e over the entire path.

The DS field (DSCP) doesn't have e2e service class semantics. Expecting
it to do so in a standardized manner suggests a misunderstanding of
Diffserv. However, clearly a network operator might 'know' that they
are using certain PHBs across a certain path to support application
flows having sensitivity to latency, or loss, or whatever. Armed
with this knowledge, the operator then configures certain link
drivers to lookup the DSCP and infer desirable link characteristics.
But that's not the same as formally defining new DSCPs to have
service class semantics. DSCPs defined PHBs - scheduling and drop
precedence semantics.

cheers,
gja


From owner-pilc@lerc.nasa.gov  Fri Jul 21 09:32:49 2000
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From: Reiner Ludwig <Reiner.Ludwig@Ericsson.com>
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
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At 00:42 21.07.00 -0700, Grenville Armitage wrote:
>Reiner Ludwig wrote:
> >
> > on this topic, I would like to draw your attention to section 4.1
> > ("Extending the Differentiated Service Framework") of my dissertation which
> > you can fetch here
> > http://iceberg.cs.berkeley.edu/papers/Ludwig-Diss00/index.html.
> >
> > It discusses pretty much what you describe, Grenville.
>
>Then I am confused, since it was you who wrote:
>
> > > > It would be so much better to have the old TOS definitions deployed ...
> > > > which unfortunately is not the case.
>
>To which Kathie and I have observed it is infact better now, since
>the DSCP is more flexible. So which situation do you really believe
>in - that its good we have the DSCP, or we ought to return to the
>ToS semantics?

What I believe in is that it would be very beneficial for "IP over 
wireless" if we could associate the TOS semantics ("minimize delay", 
"maximize throughput", "maximize reliability", etc.) or more specific 
semantics with certain DSCPs on an e2e basis.

I'm aware of the fact that the DS field (DSCP) as opposed to the TOS byte 
does *not* have e2e service class semantics. This is what I pointed out in 
my reply to Kathie.

I (as a link layer designer) would like to use a flow's DSCP *not only* to 
control scheduling and/or queue management algorithms at the network layer, 
but *also* to control link layer error control algorithms.

E.g., in 3G systems a link layer can dynamically configure many parameters 
(e.g., for FEC, spreading factor, bit rate, ARQ persistency, etc., etc.) on 
a per packet, per flow, or at least on a per flow-class basis. I.e., the 
link layer has means to greatly influence: throughput, delay, reliablity.

For that purpose, it would be extremely helpful if the IP layer (based on 
the DS field) could tell the link layer whether a particular 
packet/flow/flow-class is, e.g.,  delay sensitive or whether it is 
TCP-based bulk data. For the latter, delay is less important but throughput 
and reliability are important QoS requirements.

>Also, scanning your chapter 4 there's an odd statement on p.71 where
>you suggest adding four *new* DS Field values to represent
>four service classes.

I didn't write that they were "*new* DS Field values". I explicitly wrote 
that I borrowed from RFC1349 (Type of Service).

>This is not at all what I was talking about.
>The service classes you discuss can be adequately supported using
>the existing PHBs

Maybe I missed something.

How can you specify with the existing DSCPs/PHBs that a flow (more 
precisely the application on top) is best served with max throughput and 
max reliability?

>(and indeed, Diffserv has always clearly stated that
>PHBs/DSCPs do not represent e2e service classes as classically
>understood).

I didn't say anything else. I only said that I wished that such e2e service 
classes could be defined. As it was possible with the TOS byte.

///Reiner



From owner-pilc@lerc.nasa.gov  Fri Jul 21 13:08:44 2000
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From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>

> ...
> I didn't write that they were "*new* DS Field values". I explicitly wrote 
> that I borrowed from RFC1349 (Type of Service).
>
> >This is not at all what I was talking about.
> >The service classes you discuss can be adequately supported using
> >the existing PHBs
>
> Maybe I missed something.
>
> How can you specify with the existing DSCPs/PHBs that a flow (more 
> precisely the application on top) is best served with max throughput and 
> max reliability?

As you wireless people keep pointing out, you can't maximize both
simultaneously without failing to maximize one or the other.  Do you mean
to maximize throughput or queue position compared to competing traffic
but to not turn off link layer forward error correction or error detection
and retransmission?


> >(and indeed, Diffserv has always clearly stated that
> >PHBs/DSCPs do not represent e2e service classes as classically
> >understood).
>
> I didn't say anything else. I only said that I wished that such e2e service 
> classes could be defined. As it was possible with the TOS byte.


If the people running the edges where the new DS bits are changed
do stupid things, then bad things will happen to the applications that
rely on the new DS bits, or at least they not use the new DS bits.

And in turn, that make make people working on modern wirelss links do as
others did, those who worked on links with bit error rates as bad as modern
wireless links but with speeds 10 times slower than modern wireless links.
1200 bit/sec modems were slow, and they had lots of bit errors.  In not
so ancient days, many people turned of v.42 over v.32 modems.

Those old techniques, besides honoring the IP TOS byte, included assuming
that small packets need lower latency and peeking at UDP and TCP port
numbers.  I've personally written, used, and caused to be shipped users
of a what was once a major UNIX vendorcode that does all three.  I think
Van Jacobson's port snooping queuing is in a vast number of installations.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Fri Jul 21 13:30:26 2000
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Reiner Ludwig wrote:
	[..]
> I (as a link layer designer) would like to use a flow's DSCP *not only* to
> control scheduling and/or queue management algorithms at the network layer,
> but *also* to control link layer error control algorithms.

That's quite independent of statically assigning e2e service class
semantics to DSCPs (new or otherwise). Here's a different way of stating
my suggestion: the link driver can be configured/signalled to know,
on a case by case basis, by the network operator, the service class(es)
of the traffic being mapped into a particular DSCP (AFxy, EF, CSn...),
and thereby infer the desired link error control algorithms. No new
DSCPs (or even old ones) need additional, *statically* standardized 
e2e service semantics. IMHO.

(e.g. in one person's network AF1y might be supporting loss tolerant
traffic, in another person's network AF1y is supporting delay tolerant
traffic, etc, etc.... and this is 'known' as part of the operator's
overall Diffserv network provisioning/configuration process.)

	[..]
> >Also, scanning your chapter 4 there's an odd statement on p.71 where
> >you suggest adding four *new* DS Field values to represent
> >four service classes.
> 
> I didn't write that they were "*new* DS Field values". I explicitly wrote
> that I borrowed from RFC1349 (Type of Service).

http://iceberg.cs.berkeley.edu/papers/Ludwig-Diss00/chapter4_flow_adaptive.pdf
First line of p.71, you write:

"Below, we propose four new DS Field values that allow a differentiation
 between four different service classes, ..."

Perhaps I'm reading the wrong chapter 4.

cheers,
gja
________________________________________________________________________
Grenville Armitage                    http://members.home.net/garmitage/
Bell Labs Research Silicon Valley


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Date: Fri, 21 Jul 2000 10:48:26 -0700
To: Grenville Armitage <gja@dnrc.bell-labs.com>
From: Steve Deering <deering@cisco.com>
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
Cc: Reiner Ludwig <Reiner.Ludwig@ericsson.com>, pilc@grc.nasa.gov
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At 12:42 AM -0700 7/21/00, Grenville Armitage wrote:
>Reiner Ludwig wrote:
> > > > It would be so much better to have the old TOS definitions deployed ...
> > > > which unfortunately is not the case.
>
>To which Kathie and I have observed it is infact better now, since
>the DSCP is more flexible.

In communication protocols, "more flexible" usually means worse, not
better, e.g., more complex, more need for configuration, more ways to
screw up, more ways to be incompatible, etc.

I'm with Reiner on this one.

Steve



From owner-pilc@lerc.nasa.gov  Fri Jul 21 17:34:34 2000
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Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
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Steve Deering wrote:
> 
> At 12:42 AM -0700 7/21/00, Grenville Armitage wrote:
> >Reiner Ludwig wrote:
> > > > > It would be so much better to have the old TOS definitions deployed ...
> > > > > which unfortunately is not the case.
> >
> >To which Kathie and I have observed it is infact better now, since
> >the DSCP is more flexible.
> 
> In communication protocols, "more flexible" usually means worse, not
> better, e.g., more complex, more need for configuration, more ways to
> screw up, more ways to be incompatible, etc.
> 
> I'm with Reiner on this one.
> 
> Steve

And that's why the old TOS definitions were so successful...


From owner-pilc@lerc.nasa.gov  Fri Jul 21 22:04:58 2000
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Steve Deering wrote:
> In communication protocols, "more flexible" usually means worse, not
> better, e.g., more complex, more need for configuration, more ways to
> screw up, more ways to be incompatible, etc.

the bitter lessons of multicast?

L.

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



From owner-pilc@lerc.nasa.gov  Mon Jul 24 07:18:21 2000
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Date: Mon, 24 Jul 2000 12:06:41 +0200
To: Kathleen Nichols <nichols@packetdesign.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
Cc: Steve Deering <deering@cisco.com>,
        Grenville Armitage <gja@dnrc.bell-labs.com>, pilc@grc.nasa.gov
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At 13:39 21.07.00 -0700, Kathleen Nichols wrote:
>Steve Deering wrote:
> >
> > At 12:42 AM -0700 7/21/00, Grenville Armitage wrote:
> > >Reiner Ludwig wrote:
> > > > > > It would be so much better to have the old TOS definitions 
> deployed ...
> > > > > > which unfortunately is not the case.
> > >
> > >To which Kathie and I have observed it is infact better now, since
> > >the DSCP is more flexible.
> >
> > In communication protocols, "more flexible" usually means worse, not
> > better, e.g., more complex, more need for configuration, more ways to
> > screw up, more ways to be incompatible, etc.
> >
> > [...]
>
>And that's why the old TOS definitions were so successful...

One reason for why TOS was never successful might be that in the past there 
was maybe not strong enough a need for it. Today, I think this is very 
different.

The designers of wireless link layers could make transmission across the 
air much more efficient if they knew what kind of flow they were dealing 
with, i.e., which QoS requirements the application on top has.

Unfortunately, many wireless link layers today are one-size-fits-all and 
that can be very inefficient, causing reduced e2e performance and waste of 
radio resources (spectrum, battery power). The only escape from that - 
given that we cannot use the TOS byte - are the tricks that Vernon pointed 
out: port number sniffing, small packets = need for low latency, ... but 
sniffing tricks won't work with IPsec :-(

///Reiner






From owner-pilc@lerc.nasa.gov  Mon Jul 24 07:18:58 2000
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Date: Mon, 24 Jul 2000 12:16:36 +0200
To: Vernon Schryver <vjs@calcite.rhyolite.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
Cc: pilc@grc.nasa.gov
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At 09:55 21.07.00 -0600, Vernon Schryver wrote:
> > How can you specify with the existing DSCPs/PHBs that a flow (more
> > precisely the application on top) is best served with max throughput and
> > max reliability?
>
>As you wireless people keep pointing out, you can't maximize both
>simultaneously without failing to maximize one or the other.

Don't agree with you here.

>Do you mean
>to maximize throughput or queue position compared to competing traffic
>but to not turn off link layer forward error correction or error detection
>and retransmission?

Nothing about qeue position and nothing about competing traffic, i.e., 
nothing about scheduling.

But, I can run my wireless link with maximum error recovery persistency 
(i.e., max reliablity) but with FEC A for telnet to minimize delay and FEC 
B for ftp to maximize throughput.

> > >(and indeed, Diffserv has always clearly stated that
> > >PHBs/DSCPs do not represent e2e service classes as classically
> > >understood).
> >
> > I didn't say anything else. I only said that I wished that such e2e 
> service
> > classes could be defined. As it was possible with the TOS byte.
>
>
>If the people running the edges where the new DS bits are changed
>do stupid things, then bad things will happen to the applications that
>rely on the new DS bits, or at least they not use the new DS bits.

Exactly, and that's the problem I see. Again, I wish we could have DSCPs 
with e2e semantics.

>Those old techniques, besides honoring the IP TOS byte, included assuming
>that small packets need lower latency and peeking at UDP and TCP port
>numbers.  I've personally written, used, and caused to be shipped users
>of a what was once a major UNIX vendorcode that does all three.  I think
>Van Jacobson's port snooping queuing is in a vast number of installations.

True many SLIP and PPP implementations play these tricks. As I said in an 
earlier mail, the problem with sniffing tricks is IPsec which is important 
for many wireless users/operators. Such users easily trade security for 
performance. Too bad because we could have both if we did it right.

///Reiner



From owner-pilc@lerc.nasa.gov  Mon Jul 24 07:42:37 2000
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Date: Mon, 24 Jul 2000 11:53:29 +0200
To: Grenville Armitage <gja@dnrc.bell-labs.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
Cc: pilc@grc.nasa.gov
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At 09:36 21.07.00 -0700, Grenville Armitage wrote:
>Reiner Ludwig wrote:
>         [..]
> > I (as a link layer designer) would like to use a flow's DSCP *not only* to
> > control scheduling and/or queue management algorithms at the network layer,
> > but *also* to control link layer error control algorithms.
>
>That's quite independent of statically assigning e2e service class
>semantics to DSCPs (new or otherwise). Here's a different way of stating
>my suggestion: the link driver can be configured/signalled to know,
>on a case by case basis, by the network operator, the service class(es)
>of the traffic being mapped into a particular DSCP (AFxy, EF, CSn...),
>and thereby infer the desired link error control algorithms. No new
>DSCPs (or even old ones) need additional, *statically* standardized
>e2e service semantics. IMHO.
>
>(e.g. in one person's network AF1y might be supporting loss tolerant
>traffic, in another person's network AF1y is supporting delay tolerant
>traffic, etc, etc.... and this is 'known' as part of the operator's
>overall Diffserv network provisioning/configuration process.)

I do not see how that could work.

Consider this case: I use my wireless PDA to download some streaming media 
from some server in the Internet. How can the operator of the streaming 
server make sure that the DSCP of the streaming media flow has the right 
value when the flow "arrives" at the wireless access network? The "right 
value" should be one that the operator of the wireless access network 
understands as something like "this flow is delay sensitive" or even 
something that is more precise.

Even if the DSCP would not change in transit (which of course it could), 
according to your proposal every operator of a wireless access network 
could have its own definition of which DSCP stands for "this flow is delay 
sensitive". Thus, even if the DSCP would not change in transit, it would at 
least be very difficult for the operator of the streaming server to set the 
right DSCP given that there is more than one wireless access network 
operator in the world.

The problem is that the DS philosophy is not intended for such a scenario. 
The DS framework is intended for 2 bordering domains to agree on what kind 
of traffic each domain accepts to carry from the other. For that purpose 
bilateral agreements can be made including how the DS field should be 
(re-)set at the border routers. This is fine for ensuring QoS across 
domains. I believe this is a great idea and I think that the DS framework 
is a great concept, for the problem it tries to solve.

Still, I believe that the e2e semantics of the TOS byte have quite some 
value, e.g., to solve the problem outline in the example given above. I.e., 
I still see the need for *statically* standardized DSCPs with e2e service 
semantics.

///Reiner



From owner-pilc@lerc.nasa.gov  Mon Jul 24 08:07:50 2000
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Subject: Call for papers of ICN'01 & Final program of ECUMN'00
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Please feel free to circulate this following :

	- call for papers of <bold>ICN'01</bold> (see
<underline><color><param>0000,0000,fefe</param>http://iutsun1.colmar.uha.fr/=
ICN01.html</color></underline>
) AND=20

	- final program of <bold>ECUMN'00</bold>  (see=
 <underline><color><param>0000,0000,fefe</param>http://iutsun1.colmar.uha.fr=
/ECUMN2000.html</color></underline> )

to interested colleagues.

Accept our sincere apologies if you receive multiple copies.

----------------------------------------------------------------------------=
---

<center><bold>     CALL FOR PAPERS=20

IEEE International Conference on Networking=20

ICN'01=20

July 11-13, 2001 - CREF, Colmar, France=20

</bold></center>

GENERAL INFORMATION=20


The 2001 International Conference on Networking (ICN'01) is sponsored by=
 IEEE/Comsoc, IEEE/Computer, by IEE, by SEE and by WSES. ICN'01 is organized=
 by academic, research and industrial societies will be held at the=
 Technical Institute, Colmar, part of the University of Haute Alsace,=
 France, from Wednesday July 11, 2001 to Friday July 13, 2001. The city of=
 Colmar is ideally situated in the eastern part of France, near the German=
 and Swiss borders. The city of Colmar has been working on its own=
 Metropolitan Area Network (MAN) project, called OASICE, including LAN=
 interconnection, PBX interconnection and interactive video. An exhibition=
 illustrating these topics will be organized for industrial companies and=
 development research institutes.=20

In order to encourage closer interaction between academic and industrial ATM=
 research communities, we solicit both academic research papers and=
 industrial contributions.=20


TOPICS OF SPECIAL INTEREST=20


Topics of interest include, but are not limited to the following:=20

 =20

  Communications switching and routing =20

Communications modeling =20

Communications security =20

Computer communications =20

Multimedia and multicast communications =20

Internet environment =20

Network Management and control =20

Quality of Service (IntServ, DiffServ, =85) =20

Wireless Communications (Satellite, WLL, 3G) =20

Voice over IP =20

Java, Tina, Corba architectures Network signaling =20

ATM networks =20

ATM/IP integration =20

Integrated services =20

Traffic engineering =20

Telecommunication networks architectures =20

Performance evaluation, simulation =20

Mobile agents, active and intelligent networks =20

Applications and case studies =20

Protocol design and evaluation =20

MPLS=20



These topics can be discussed in term of concepts, state of the art,=
 standards, implementations, running experiments and applications.=20


INSTRUCTIONS FOR AUTHORS=20


Mail four papers or E-mail preferably in Word 6 format, or alternately a=
 postscript version of a 2000-word extended abstract summarizing an original=
 work. All the manuscripts must be written in English. The top of the first=
 page of each paper should include the title of the paper, authors' name,=
 position, address, telephone and fax numbers, e-mail of the author=
 responsible for correspondence and a list of four keywords. The deadline=
 for submission of all extended abstracts is December 10, 2000 with=
 notification of acceptance by February 20, 2001. Submission of camera-ready=
 paper is by March 30, 2001.=20

Authors of accepted papers will be invited to submit full-length manuscripts=
 for inclusion in the proceedings with ISBN.=20


All submitted papers should be sent to the following address:=20


Pascal LORENZ=20

University of Haute Alsace=20

IUT - Department GTR=20

34 rue du Grillenbreit=20

68008 Colmar, France=20

Phone: 33 (0)389202366 Fax: 33 (0)389202359 Mobile: 33 (0)603658042=20

E-mail: lorenz@colmar.uha.fr=20


Check our Web page at=
 <underline><color><param>0000,0000,fefe</param>http://iutsun1.colmar.uha.fr=
/ICN01.html</color></underline> for the latest information concerning the=
 conference.=20

Best papers will be forwarded for consideration in a special issue of a=
 journal.=20


TUTORIALS AND WORKSHOPS=20


Tutorials and workshops provide overviews of current high interest topics.=
 Proposals for half of full day tutorials are due by December 10, 2000.=20


INTERNATIONAL ADVISORY COMMITTEE=20


R. Addie (Australia) - University of Southern Queensland=20

K. Begain (Jordan) - Mu'tah University=20

A. Benslimane (France) - University of Belfort-Montbeliard=20

B. Bing (Singapore) - Ngee Ann Polytechnic=20

D. Bonjour (France) - CNET=20

A. Brandwajn (USA) - University of California Santa Cruz=20

J.P. Coudreuse (France) =96 Mitsubishi=20

J. Crowcroft (UK) =96 University College London=20

S. Fdida (France) =96 LIP6=20

B. Gavish (USA) - Vanderbilt University=20

H. Guyennet (France) - University of Franche-Comte=20

J. Halpern (USA) =96 Newbridge=20

Z. Hulicki (Poland) =96 University of Cracow=20

R. Israel (France) - IEEE=20

A. Jajszczyk (Poland) - University of Mining & Metallurgy=20

A. Jamalipour (Australia) - University of Sydney=20

S. Kota (USA) - Lockeed Martin=20

D. Kouvatsos (UK) - University of Bradford=20

S. Kumar (USA) =96 Ericsson=20

G.S. Kuo (Taiwan) =96 National Central University=20

F. Le Faucheur (France) - Cisco=20

M. Lee (Korea) =96 Dongshin University=20

P. Lorenz (France) - University of Haute Alsace=20

H.. Mouftah (Canada) - Queen's University=20

G. Omidyar (USA) - Computer Sciences Corp.=20

J.J. Pansiot (France) - University of Strasbourg=20

M. Potts (Switzerland) - Martel=20

Z. Mammeri (France) - University of Toulouse=20

N. Mastorakis (Greece) - Military Institutions of University Education=20

S. Moyer (USA) - Bellcore=20

R. Muraine (France) - Newbridge=20

G. Pujolle (France) - University of Versailles-Saint-Quentin=20

S. Rao (Switzerland) - Ascom=20

A. Reid (UK) - British Telecom=20

S. Ritzenthaler (France) - Newbridge=20

P. Rolin (France) - ENST Bretagne=20

R. Saracco (Italy) - CSELT=20

G. Swallow (USA) - Cisco=20

H. Tobiet (France) =96 Clemessy=20

M. Trehel (France) =96 University of Franche-Comte=20

V.A. Villagra (Spain) =96 University of Madrid=20

E. Vazquez Gallo (Spain) =96 University of Madrid=20

O. Yang (Canada) - University of Ottawa=20


IMPORTANT DATES=20


Extended Abstract due: December 10, 2000=20

Notification of acceptance: February 20, 2001=20

Deadline for full-length camera-ready manuscript: March 30, 2001=20


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

<center><bold>1st IEEE European Conference on Universal Multiservice=
 Networks  ECUMN'2000

 October 2-4, 2000 - CREF, Colmar, France

FINAL PROGRAM

</bold></center>=20

08:30 - 18:30 Conference Registration


<underline>Monday October 2, 2000

</underline>

09:00 - 09:20 Opening Address=20


09:20 - 10:30 Tutorial Session 1

Internet Services over Mobile and Wireless Networks: Architectures and=
 Protocols

G. Omidyar, Computer Sciences Corp., USA ; M.J. Montpetit, Teledesic LLC,=
 USA


10:30 - 11:00 Coffee Break


11:00 - 12:15 Network Architecture and Convergence

Chair: S. Ritzenthaler, Newbridge, France


MPLS and Next Generation Access Network

A. Kankkunen, Integral Access Inc, USA


Deutsche Telekom's View on Network Convergence

L. Falkenhagen Deutsche Telekom AG, Germany


A Functional Architecture for End-to-end Quality-of-Service in a=
 Multi-domain Network

E. Pagani, G.P. Rossi, University of Milano, Italy


12:15 - 13:00 Panel Discussion 1 - Network Evolution and Convergence


13:00 - 14:30 Lunch


14:30 - 16:15 Traffic=20

Chair: M. Villen, Telefonica I+D, Spain


The Relevance of the Bufferless Analysis for Traffic Management in=
 Telecommunication Networks

G. Hasslinger, F. Hartleb, Telekom Innovations-GmbH, Germany ; M. Fiedler,=
 University of Karlskrona/Ronneby, Sweden


Mapping of Loss and Delay Between IntServ and DiffServ

T. Chahed, G. H=E9buterne, C. Fayet, National Institute of=
 Telecommunications, France


Resource Demand of Aggregated Resource Reservations

J. Ehrensberger, Swiss Federal Institute of Technology, Switzerland


Characterization of the MPEG-2 Video Traffic Generated by DVD Applications

A. Chodorek, Kielce University of Technology, Poland ; R. Chodorek,=
 University of Mining and Metallurgy, Poland


14:30 - 16:15 Interworking between narrow band and broadband networks

Chair: S. Chakraborty, Technical University of Helsinki, Finland


Integrating Euro-ISDN with ATM Technology: Interworking Mechanisms and=
 Services Support

L. Mandalos, K. Leonidou, C. Andreopoulos, J. Drakos, S. Koubias, G.=
 Papadopoulos, University of Patras, Greece


Extending the Internet with the Intelligent Network Capabilities

B. El Ouahidi, M. Bouhdadi, University of Rabat, Morocco ; D. Bourget, ENST,=
 France


A Suitable Service Discipline for ATM-Ethernet Interconnection

J.M. Arco, D. Meziat, B. Alarcos, University of Alcala, Spain


Interoperability of ATM-Ethernet Interworking System: Design and Congestion=
 Control

R. Ouni, A. Soudani, S. Nasri, M. Abid, R. Tourki, University of Monastir,=
 Tinisia ; K. Torki, INPG, France


16:15 - 16:45 Coffee Break


16:45 - 18:30 Routing

Chair: P. Chemouil, France Telecom R&D, France


Adaptive Routing and Load Balancing of Ephemeral Connections

M. Heusse, ENST de Bretagne, France


A new Multi-Services Rerouting Algorithm

A. Gueroui, University of Versailles, France ; L. Mokdad University of=
 Dauphine, France ; J. Ben-Othman University of Paris 6, France


Extending Mobile IP-v6 with Multicast to Support Mobile Networks in  IPv6

T. Ernst, C. Castelluccia, INRIA Rh=F4ne-Alpes, France ; H.Y. Lach, Motorola=
 Labs, France


16:45 - 18:30 Interworking between IP and ATM

Chair: M. Potts, Martel, Switzerland


Topology Optimization of IP over ATM

L. Frelechou, M. Osborne, R. Haas, IBM Research, Switzerland


Resource Reservation with Boomerang via Open Interfaces

M. Maliosz, Budapest University of Technology and Economics, Hungary


Efficient Translation of Network Performance Parameters for Transport of IP=
 Packets over Cell-Switched Subnetworks

J. Schmitt, M. Karsten, R. Steinmetz, Darmstadt University of Technology,=
 Germany


Design of Adaptive Access Network and Control Protocol

H. Nakagawa, I. Kazunari, N. Ohta, NTT Information Sharing Platform=
 Laboratories, Japan


19:15 Reception - Town Hall


<underline>Tuesday October 3, 2000

</underline>

09:00 - 10:45 End to end traffic Control (1)

Chair: U. Krieger, Deutsche Telecom, Germany


Guaranteed QoS for TCP flows in ATM-based DiffServ Networks

T. M=FCller, Dresden University of Technology, Germany


Virtual Buffering Strategy for GFR Services in IP/ATM Internetworks

S.C. Hu, Ming Shin Institute of Technology, Taiwan ; P.C. Wang, Y.C. Chen,=
 National Chiao Tung University, Taiwan ; C.T. Chan, Chunghwa Telecom,=
 Taiwan


An Efficient Weight Assignment Process for GPS Servers

G. Urvoy, PRISM, France ; G. Hebuterne, Institut National des=
 T=E9l=E9communications, France


Some Aspects of RTP - TCP Coexistence in Universal Multiservice  Networks

R. Chodorek, University of Mining and Metallurgy, Poland


09:00 - 10:45 Mobile Networks and Mobile Services

Chair: G. Omidyar, Computer Sciences Corp, USA


Voice Enabled Request and Response for Mobile Devices Supporting WAP=
 Protocol: the Constraints

A. Mohan, Himachal Futuristic Communications, India ; A. Mohan, Indian=
 Institute of Technology, India


IP based enhanced Data Casting Services over Radio Broadcast Networks

W. Kellerer, P. Sties, J. Ebersp=E4cher, Munich University of Technology,=
 Germany


Location-Dependant and Value Added Services (VAS) for Mobile Communications

P. Lorenz, University of Haute Alsace, France ; H. Tobiet, NMG, France=20


The IP Revolution and Potential gains for ISP/Mobile/Fixed Telephony=
 Operators in the Liberalization of Telecommunications

D. Vergados, D. Drakoulis, E. Vayias, J. Soldatos, N. Mitrou, National=
 Technical University of Athens, Greece


10:45 - 11:15 Coffee Break


11:15 - 13:00 Multicast

Chair: G. Girardi, CSELT, Italy


A Scalable Fault Tolerant Approach to Core Election in an Inter-Domain=
 Multicast Routing Environment

M.D.Ech-Cherif El Kettani, ENSIAS, Morocco; Y. Souissi, EMI, Morocco


A New Routing Algorithm for Delay-Constrained Dynamic Multicast

T. Asaka, NTT Service Integration Laboratories, Japan ; T. Miyoshi, Y.=
 Tanaka, Waseda University, Japan


Remote Time-Alignment of Interactive Services through Efficient Multicast=
 Algorithms

A. Borella, G. Cancellieri, University of Ancona, Italy


Key Establishment for IGMP Authentication in IP Multicast

T. Hardjono, B. Cain, Nortel Networks, USA


11:15 - 13:00 IP Test-beds

Chair: H. Tobiet, NMG, France


Real-Time Multimedia Services over Internet

A. Benslimane, Technical University of Belfort-Montb=E9liard, France


Telephony over IP: Theoretical Modelling and Lab Experiments

P. Senesi, P. Ferrabone, G. Gritella, R. Rinaldi, M. Siviero, CSELT, Italy


User-domain Multiservice Architecture for Wired and Wireless IP  Networks

L. Cheng, I. Marsic, The State University of New Jersey, USA


A Testbed Environment for the Performance Evaluation of Modular Network=
 Architectures

D. Maggiorini, E. Pagani, G.P. Rossi, University of Milano, Italy


13:00 - 14:30 Lunch


14:30 - 16:15 Modeling

Chair: G. H=E9buterne, INT, France


Estimation of the Renewal Function by Empiral Data - A Bayesian  Approach

N.M. Markovitch, Russian Academy of Sciences, Russia ; U.R. Krieger, T-Nova=
 Deutsche Telekom, Germany


Modeling Access Networks for Quality of Service

F. Duran, T. Lizambri, S. Wakid, National Institute of Standards and=
 Technology, USA ; D.R. Vaman, Megaxess, USA


Managing Wireless Internet Information of Electronic Advertising

E. Rashid, Y. Yoshioka, Hirosaki University, Japan ; Y. Nemoto, T. Nakamura,=
 Tohoku University, Japan


Performance Evaluation of a full Memory Multidestination Protocol for=
 Satellite Based Reliable Multicast with and without Local Recovery

H. Jianhua, K.R. Subramanian, Y. ZongKai, H. Ping, Nanyang Technological=
 University, Singapore


14:30 - 16:15 Tariffs and Bandwidth Allocation

Chair: P. Lorenz, University of Haute Alsace, France


INEDAC Project: A Tool to Calculate Interconnection Tariff based on a=
 Bottom-up Method

J.A. Portilla, K.D. Hackbarth, A.E. Garcia, University of Cantabria, Spain ;=
 R. W=F6hrl, F. Gonzalez, Institut f=FCr Kommunikationsdienste GmbH, Germany


Auction Method and its Performance in a Dynamic Bandwidth Allocation Service

E. Takahashi, Y. Tanaka, Waseda Universiy, Japan


Multivariable Feedforward Plus Feedback Control for Adapting MPEG Video=
 Streams to Variable Channel Bandwidth

H.F. Raynaud, M. Luong, University of Paris Nord, France


Robust Topology for Enterprise Networks against Diverse Tariff Structures

T. Miyoshi, K. Mouri, Y. Tanaka, Waseda University, Japan ; T. Asaka, NTT=
 Service Integration Laboratories, Japan


16:15 - 16:45 Coffee Break


16:45 - 18:30 End to end Traffic Control (2)

Chair: G. H=E9buterne, INT, France


An Architecture of QoS Services for a Core Internet Network over DTM

C.J. Barenco, Polytechnic University of Madrid, Spain ; A.A. Salona, J.I.=
 Moreno, University Carlos III of Madrid, Spain


Congestion Avoidance for Unicast and Multicast Traffic

A. Dracinschi, S. Fdida, University Pierre et Marie Curie, France


MPOA and QoS Support in LIS Internetworking Environments

I. Erturk, Kocaeli University, UK ; E. Stipidis, University of Sussex, UK


A Method for Increasing Throughput based on Packet Striping

F. Jacquet, M. Misson, IUT of Clermont-Ferrand, France


16:45 - 18:30 Advanced Networking

Chair: G.V. Morson, Cambridge Network, England


A New Network Service Environment using Active Network

K. Widoyo, T. Aoki, H. Yasuda, University of Tokyo, Japan


Adaptive Applications over Active Networks: Case Study on Layered Multicast

L. Yamamoto, G. Leduc, University of Li=E8ge, Belgium


Integrated Performance Monitoring of Client/Server Software

C. Steigner, J. Wilke, I. Wulff, University of Koblenz-Landau, Germany


Who needs Addresses ?

V. Guruprasad, IBM, USA


20:00 Gala Dinner - Restaurant Meistermann


<underline>Wednesday October 4, 2000

</underline>

09:00 - 10:00 Tutorial Session 2

Chair: P. Rolin, France Telecom R&D, France


Active Networks and its Management

M. Brunner, NEC Europe Ltd, Germany


10:00- 10:30 Coffee Break


10:30 - 11:45 New Architectures for New Services

Chair: A. Gravey, ENST-Bretagne, France


A Novel Architecture for Efficient Protocol Processing in High Speed=
 Communication Environments

G. Konstantoulakis, V. Nellas, C. Georgopoulos, T. Orphanoudakis, N. Zervos,=
 Ellemedia Technologies, Greece ; M. Steck, Hyperstone electronics GmbH,=
 Germany; D. Verkest, Interuniversity Microelectronics Centre (IMEC),=
 Belgium; G. Doumenis, D.Reisis, University of Athens, Greece; N. Nikolaou,=
 J.A. Sanchez, Lucent Technologies, The Netherlands


Using T=E9l=E9domotis Interface for a new Multiservice Network applied to=
 Monitoring the Elderly

T. Val, E. Campo, IUT of Blagnac, France ; P. Kauffmann, M. Misson,=
 University of Clermont II, France ; P.Y. Danet, France T=E9l=E9com R&D,=
 France


Video and Interactive Internet access in a DVB Network

R. Jaeger, BetaResearch, Germany


11:45 - 13:00 Panel Discussion 2 - New Architectures for New Services


13:00 - 14:30 Lunch


14:30 Visit of Vialis







From owner-pilc@lerc.nasa.gov  Mon Jul 24 10:56:15 2000
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	Mon, 24 Jul 2000 07:53:40 -0600 (MDT)
Date: Mon, 24 Jul 2000 07:53:40 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <200007241353.e6ODren08535@calcite.rhyolite.com>
To: pilc@grc.nasa.gov
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
Sender: owner-pilc@lerc.nasa.gov
Precedence: bulk

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

> ...
> > > How can you specify with the existing DSCPs/PHBs that a flow (more
> > > precisely the application on top) is best served with max throughput and
> > > max reliability?
> >
> >As you wireless people keep pointing out, you can't maximize both
> >simultaneously without failing to maximize one or the other.

> ...
> But, I can run my wireless link with maximum error recovery persistency 
> (i.e., max reliablity) but with FEC A for telnet to minimize delay and FEC 
> B for ftp to maximize throughput.

But you are not maximizing throughput and reliability.  As some of the
wireless people tirelessly point out (especially some with addresses at
Ericsson.com), you could get more throughput by not using any FEC, not to
mention the retransmissions. I assume you mean by "error recovery
persistency."  


> ...
> >If the people running the edges where the new DS bits are changed
> >do stupid things, then bad things will happen to the applications that
> >rely on the new DS bits, or at least they not use the new DS bits.
>
> Exactly, and that's the problem I see. Again, I wish we could have DSCPs 
> with e2e semantics.

You can't pass laws or standards that keep people from doing stupid things.
In other words, why not continue to mark your FTP/TCP/IP packets with 0x08
and their telnet packets with 0x10?  (I trust you know that if you use
BSD UNIX systems, you're already doing that.)  Yes, some people operating
routers will configure them to do stypid things with those values, and
some vendors will build routers that do only stupid things with those
values, but more will do far worse things, such hijacking all of your
HTTP, SMTP, or FTP packets and sending them to redirection proxies
in the name of protecting you from spam or optimizing your WWW experience.


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Mon Jul 24 11:06:27 2000
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	Mon, 24 Jul 2000 08:01:49 -0600 (MDT)
Date: Mon, 24 Jul 2000 08:01:49 -0600 (MDT)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <200007241401.e6OE1nr08870@calcite.rhyolite.com>
To: pilc@grc.nasa.gov
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
Sender: owner-pilc@lerc.nasa.gov
Precedence: bulk

> From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>

> ...
> The designers of wireless link layers could make transmission across the 
> air much more efficient if they knew what kind of flow they were dealing 
> with, i.e., which QoS requirements the application on top has.
>
> Unfortunately, many wireless link layers today are one-size-fits-all and 
> that can be very inefficient, causing reduced e2e performance and waste of 
> radio resources (spectrum, battery power). The only escape from that - 
> given that we cannot use the TOS byte -...

Why can't wireless links use the RFC 1349 values?  Aren't those
4 values  supposed to be grandfathered?

Why can't you trust that people who aren't stupid will configure their
routers to translate the DS values they use inside their networks to mean
low-delay or high throughput back to 0x10 and 0x08 as packets leave their
networks?


Vernon Schryver    vjs@rhyolite.com


From owner-pilc@lerc.nasa.gov  Tue Jul 25 03:19:01 2000
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Date: Tue, 25 Jul 2000 08:19:36 +0200
To: Vernon Schryver <vjs@calcite.rhyolite.com>
From: Reiner Ludwig <Reiner.Ludwig@ericsson.com>
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
Cc: pilc@grc.nasa.gov
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At 08:01 24.07.00 -0600, Vernon Schryver wrote:
>Why can't wireless links use the RFC 1349 values?  Aren't those
>4 values  supposed to be grandfathered?

As mentioned a number of times before: RFC1349 is obsoleted by RFC2474, 
even if BSD still uses it.

>Why can't you trust that people who aren't stupid will configure their
>routers to translate the DS values they use inside their networks to mean
>low-delay or high throughput back to 0x10 and 0x08 as packets leave their
>networks?

Somebody with a "gold card subscription" from her/his ISP will have the 
same DSCP on all of her/his flows independent of whether its ftp, telnet, 
or streaming media. So the last-hop router at the wireless link would have 
a hard time figuring out the right TOS values.

I assume that DSCPs will often have little to do with application layer QoS 
requirements but much more with how much you are willing to pay for your 
traffic.

///Reiner



From owner-pilc@lerc.nasa.gov  Tue Jul 25 09:05:35 2000
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Performance Implications of Link Characteristics Working Group of the IETF.

	Title		: Performance Enhancing Proxies
	Author(s)	: J. Border, M. Kojo, J. Griner, G. Montenegro
	Filename	: draft-ietf-pilc-pep-03.txt
	Pages		: 40
	Date		: 24-Jul-00
	
This document provides a high level overview of Performance Enhancing
Proxies. Different types of Performance Enhancing Proxies are
described as well as the mechanisms used to improve performance. In
addition, motivations for their development and use are described
along with some the consequences of using them, especially in the
context of the Internet.

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From owner-pilc@lerc.nasa.gov  Tue Jul 25 09:21:55 2000
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Performance Implications of Link Characteristics Working Group of the IETF.

	Title		: End-to-end Performance Implications of Slow Links
	Author(s)	: S. Dawkins, G. Montenegro, M. Kojo, V. Magret
	Filename	: draft-ietf-pilc-slow-04.txt
	Pages		: 15
	Date		: 24-Jul-00
	
This document makes performance-related recommendations for users of
network paths that traverse 'very low bit-rate' links.

'Very low bit-rate' implies 'slower than we would like'. This
recommendation may be useful in any network where hosts can saturate
available bandwidth, but the design space for this recommendation
explicitly includes connections that traverse 56 Kb/second modem
links or 4.8 Kb/second wireless access links - both of which are
widely deployed.

This document discusses general-purpose mechanisms. Where
application-specific mechanisms can outperform the relevant
general-purpose mechanism, we point this out and explain why.

This document has some recommendations in common with RFC 2689,
'Providing integrated services over low-bitrate links', especially
in areas like header compression. This document focuses more on
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Performance Implications of Link Characteristics Working Group of the IETF.

	Title		: End-to-end Performance Implications of Links with Errors
	Author(s)	: S. Dawkins, G. Montenegro, M. Kojo, V. Magret, N. Vaidya
	Filename	: draft-ietf-pilc-error-04.txt
	Pages		: 18
	Date		: 24-Jul-00
	
The rapidly-growing Internet is being accessed by an
increasingly wide range of devices over an increasingly wide
variety of links. At least some of these links do not provide
the reliability that hosts expect, and this expansion into
unreliable links causes some Internet protocols, especially TCP
[RFC793], to perform poorly.

Specifically, TCP congestion avoidance procedures [RFC2561], while
appropriate for connections that lose traffic primarily because of
congestion and buffer exhaustion, interact badly with connections
that traverse links with high uncorrected error rates. The result is
that senders may spend an excessive amount of time waiting on
acknowledgements that aren't coming, whether these losses are due to
data losses in the forward path or acknowledgement losses in the
return path, and then, although these losses are not due to
congestion-related buffer exhaustion, the sending TCP then transmits
at substantially reduced traffic levels as it probes the network to
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From owner-pilc@lerc.nasa.gov  Wed Jul 26 19:31:33 2000
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Subject: Final Call for Participation: MobiCom 2000, Aug 6-11, Boston
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                  Final Call for Participation 
                            
                          MobiCom 2000

           The Sixth Annual International Conference on
                  Mobile Computing and Networking
                       August 6 - 11,  2000
           
            Seaport Hotel at World Trade Center Boston
                   Boston, Massachusetts, USA
                    
    The complete program for paper, panel and tutorial sessions, 
    as well as registration and hotel information are available 
    at the conference web site below.

          http://www.research.telcordia.com/mobicom2000/

 Come, celebrate the wireless revolution and its convergence with the
 Internet, on Boston's historic waterfront. We look forward to seeing
 you in MobiCom 2000. 

 Registration by mail or fax will be accepted until July 31.
 Registrations beyond this date will be accepted on site only.





From owner-pilc@lerc.nasa.gov  Sat Jul 29 14:19:18 2000
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Subject: Re: From Lorraine,as promised,I can lower your bill
Date: Sat, 29 Jul 2000 13:38:49
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<html>

<head>
<title>Untitled Document</title>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
</head>

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</div>
</body>
</html>


From owner-pilc@lerc.nasa.gov  Sun Jul 30 21:38:09 2000
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To: Reiner.Ludwig@ericsson.com
CC: gja@dnrc.bell-labs.com, pilc@grc.nasa.gov, karn@ka9q.ampr.org
In-reply-to: <4.3.2.7.0.20000721112423.00af8af0@chapelle.ericsson.se> (message
	from Reiner Ludwig on Fri, 21 Jul 2000 13:44:16 +0200)
Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
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>I (as a link layer designer) would like to use a flow's DSCP *not only* to 
>control scheduling and/or queue management algorithms at the network layer, 
>but *also* to control link layer error control algorithms.

Exactly. Information theory says there are fundamental tradeoffs
between delay and throughput, and it would be nice for the link layer
to have some guidance in making these tradeoffs.

The main parameter is the FEC interleaver blocksize. Larger blocksizes
yield better performance, especially over fading channels, but they
increase delay. So something that can tell the physical layer what
size FEC block to use would be very helpful.

[The idea that larger blocks perform better over noisy channels may
seem strange to those unfamiliar with coding theory. If you think that
larger blocks would have a larger probability of at least one error,
you're right. That's why, on *uncoded* channels, high error rates
dictate smaller blocks. But with interleaved FEC, it's not whether
there are *any* errors in the block, but whether their number exceeds
some correction limit. With uncorrelated channel errors, the number of
errors in any given block follows a binomial distribution; as
blocksize is increased, this becomes a gaussian distribution. Thanks
to the law of large numbers, the standard deviation of this
distribution increases only as the square root of the blocksize. So
the probability that the code will fail, i.e., be overwhelmed by
errors, goes to zero as the blocksize approaches infinity *provided*
that the expected number of errors in a block is within the code's
error correction capability.]

This has gotten me thinking about how TOS should really work in a
wireless link. When you think about it, the old-style qualitative TOS
bits that meant "low delay" "high throughput" or "high reliability"
are really not that orthogonal. Ultimately the *only* link/subnet
characteristic that ever really matters is LATENCY. "High throughput"
is important only insofar as it reduces latency due to queuing delay,
and "high reliability" is important only as it decreases (or avoids)
the latency of an end-to-end retransmission scheme.  In every
situation, the goal is the same: to complete a "transaction" as
quickly as possible.

This suggests that all the link layer ever *really* wants to know is
the application transaction size. Two examples:

A bulk file transfer is a single "large" transaction, i.e., one that
will take many packets to complete.  We assume that just having part
of the file is useless, so we want to finish the whole thing as
quickly as possible. And we assume the file to be much larger than the
bandwidth*delay product of the available network path, so propagation
delays have a small effect on the transaction latency.

Here the link can be allowed to operate on the data in large chunks,
e.g, by using large FEC code blocks that could possibly span many IP
packets. This will increase throughput on a noisy channel and allow
the transaction to complete more quickly -- thus reducing "latency" by
this particular application's definition.

At the other extreme, an interactive session is really a series of
"small" transactions.  Each transaction typically fits into a single
IP packet and must be delivered before the other end can continue.
The transactions may also be small relative to the bandwidth*delay
product of the available network paths.

In this case, reducing "latency" implies getting each packet out as
quickly as possible, even if it means having to use a less efficient
FEC, i.e., one with a smaller blocksize and more overhead to keep an
acceptable probability of loss.

So what about a "TOS" field that simply indicates how many IP packets
make up this application's transaction size? While the analogy isn't
perfect, think of it as a distant cousin of the TCP "push" bit, which
was originally defined to permit implementations to gain efficiency by
aggregating data when immediate delivery wasn't necessary.

I know that redefining the IP TOS fields may be beyond the scope of
our charter, but I've been mulling it over for quite some time and I
wanted to get this idea out there to stimulate some discussion because
it does connect directly to our discussion of what performance-tuning
"knobs" a link/subnetwork should provide to IP and how applications
could use them.

Phil





From owner-pilc@lerc.nasa.gov  Mon Jul 31 09:50:05 2000
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From: "Michael Meyer (EED)" <Michael.Meyer@eed.ericsson.se>
To: "'karn@qualcomm.com'" <karn@qualcomm.com>
Cc: pilc@grc.nasa.gov
Subject: RE: Comments to "Advice for Internet Subnetwork Designers"
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Hi Phil,

In general I like your proposal to provide more information to the link layer. 

>
>>I (as a link layer designer) would like to use a flow's DSCP 
>*not only* to 
>>control scheduling and/or queue management algorithms at the 
>network layer, 
>>but *also* to control link layer error control algorithms.
>
>Exactly. Information theory says there are fundamental tradeoffs
>between delay and throughput, and it would be nice for the link layer
>to have some guidance in making these tradeoffs.
>
>The main parameter is the FEC interleaver blocksize. Larger blocksizes
>yield better performance, especially over fading channels, but they
>increase delay. So something that can tell the physical layer what
>size FEC block to use would be very helpful.
>

There are some more parameters like e.g. FEC code rate and transmission power.

>[The idea that larger blocks perform better over noisy channels may
>seem strange to those unfamiliar with coding theory. If you think that
>larger blocks would have a larger probability of at least one error,
>you're right. That's why, on *uncoded* channels, high error rates
>dictate smaller blocks. But with interleaved FEC, it's not whether
>there are *any* errors in the block, but whether their number exceeds
>some correction limit. With uncorrelated channel errors, the number of
>errors in any given block follows a binomial distribution; as
>blocksize is increased, this becomes a gaussian distribution. Thanks
>to the law of large numbers, the standard deviation of this
>distribution increases only as the square root of the blocksize. So
>the probability that the code will fail, i.e., be overwhelmed by
>errors, goes to zero as the blocksize approaches infinity *provided*
>that the expected number of errors in a block is within the code's
>error correction capability.]

I think that your explanation regarding the coding theory might lead some into the wrong direction. What you describe in your "tutorial" is an ideal scenario. The mobile radio channel is different and therefore infinite blocks lenghts are far from ideal.

The mobile radio channel can be described by a so-called coherence time, which is a measure for the duration during which the channel is correlated. Interleaving is trying to move coded bits describing the same info bit further apart than the coherence time. If that is achieved there is no further gain by increasing the interleaving. 

Interleaving does help to reduce the bit error rate for a channel, but for a given bit error rate, the block error rate is increasing with increasing block length. Thus, there exists an optimal block size. It has to be large enough for efficient interleaving, but small enough for a low block error rate.
>
>This has gotten me thinking about how TOS should really work in a
>wireless link. When you think about it, the old-style qualitative TOS
>bits that meant "low delay" "high throughput" or "high reliability"
>are really not that orthogonal. Ultimately the *only* link/subnet
>characteristic that ever really matters is LATENCY. "High throughput"
>is important only insofar as it reduces latency due to queuing delay,
>and "high reliability" is important only as it decreases (or avoids)
>the latency of an end-to-end retransmission scheme.  In every
>situation, the goal is the same: to complete a "transaction" as
>quickly as possible.
>
>This suggests that all the link layer ever *really* wants to know is
>the application transaction size. Two examples:
>
>A bulk file transfer is a single "large" transaction, i.e., one that
>will take many packets to complete.  We assume that just having part
>of the file is useless, so we want to finish the whole thing as
>quickly as possible. And we assume the file to be much larger than the
>bandwidth*delay product of the available network path, so propagation
>delays have a small effect on the transaction latency.
>
>Here the link can be allowed to operate on the data in large chunks,
>e.g, by using large FEC code blocks that could possibly span many IP
>packets. This will increase throughput on a noisy channel and allow
>the transaction to complete more quickly -- thus reducing "latency" by
>this particular application's definition.

Besides the negative impacts of increased delays for TCP slow start, increasing the interleaving delay might be harmful, because TCP has this bias for connections with a large RTT. This is for me another reason for not having large interleaving blocks.

I don't see big advantages of having many IP packets in one FEC code block for the reasons I explained above. But I do completely agree that the link layer needs more information to perform its job in the best possible way.


/Michael

******************
Michael Meyer
Ericsson Research




From owner-pilc@lerc.nasa.gov  Mon Jul 31 12:05:13 2000
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From: Phil Karn <karn@ka9q.ampr.org>
To: Michael.Meyer@eed.ericsson.se
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Subject: Re: Comments to "Advice for Internet Subnetwork Designers"
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>There are some more parameters like e.g. FEC code rate and transmission power.

Correct. I was assuming these were held constant.

>The mobile radio channel can be described by a so-called coherence
>time, which is a measure for the duration during which the channel is
>correlated. Interleaving is trying to move coded bits describing the
>same info bit further apart than the coherence time. If that is
>achieved there is no further gain by increasing the interleaving.

You got me -- I was oversimplifying too much. I should have said
"channel code block size" rather than "interleaving block size". The
channel code can be a layered combination of one or more FEC code
elements and interleaving. The interleaving may be 1) explicit, 2) an
internal element of a particular code (e.g., a turbo code), or 3) an
implicit property of a large block code, e.g., Reed-Solomon. All that
matters is that the user data bits are effectively "spread" over the
entire channel code block to effectively deal with channel burst
errors.

So assuming that the parameters of each element making up the channel
code are optimally chosen, you can always do better with a larger
channel code block size.

Yes, you do reach a point of diminishing returns with increasing
channel block size, both against Gaussian noise and against fading.
But the improvement is always monotonically increasing with block size.

The diminishing returns point usually occurs first against Gaussian
noise. For modern codes, this is on the order of one or more
kilobits. (The turbo code in the Qualcomm HDR forward link uses a 128
byte block size).

This leaves channel fading. Here the point of diminishing returns
depends heavily on the channel, specifically the coherence time just
as you say. The usual rule of thumb is that you want a code block
transmission time that is at least 10x the coherence time. This can be
a *very* large number when you're transmitting at high speed on a
slowly fading channel. At some point, it's more practical to layer ARQ
above the FEC and just keep trying until the channel fade ends and the
FEC can again deal with it.

The FEC/ARQ tradeoff point depends on factors like the round trip
delay of the channel and the per-packet ARQ overhead. The larger the
FEC channel block size, the lower the overhead of the ARQ header.
Also, if a larger FEC block size can reduce the ARQ retransmission
rate, the higher layer protocols reap the benefit of decreased delay
variance.

What I'm saying is that it would be nice to at least give the physical
layer the opportunity to aggregate relatively large chunks of user
data, possibly comprising several IP packets, if the physical layer
decides that would improve performance over a slow fading channel, and
if the application can deal with the increased short-term latency.

>Besides the negative impacts of increased delays for TCP slow start,
>increasing the interleaving delay might be harmful, because TCP has
>this bias for connections with a large RTT. This is for me another
>reason for not having large interleaving blocks.

I don't quite understand this. My experience is that TCP adapts quite
well to large RTTs, especially if the RTTs are stable and the packet
loss rate is low. TCP window scaling is now pretty widely deployed,
although it may have to be enabled.

What you really want to avoid are the nasty interactions between TCP
and link/physical layer retransmissions that can occur in certain link
operating regions. Consider a physical layer with a long latency and
link layer ARQ. Most packets go through without link level ARQ, so
most of the time the RTT is consistent and TCP shrinks its RTO to just
above the RTT. But whenever the link layer retransmits, the RTT spikes
and there's a spurious TCP retransmission.

If link level retransmissions are rare, then spurious TCP
retransmissions are also rare and their effect on performance is
minimal.  If link level retransmissions are very frequent, TCP sees a
high RTT variance and sets an RTO that is well above the average
value. Again, spurious TCP retransmissions are rare. But things can
degrade badly in the intermediate region, say, every 10th packet. Just
when TCP has dropped its RTO to just above the average RTT, another
link level retransmission occurs and causes TCP to spuriously
retransmit.

Perhaps we could build a better (higher order) RTT model into TCP, or
tune its existing parameters better. But I think it would be better to
fix the physical layers to work better with TCP as it currently
stands.

Phil




From owner-pilc@lerc.nasa.gov  Mon Jul 31 12:19:08 2000
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Phil Karn wrote:
> 
...
> So what about a "TOS" field that simply indicates how many IP packets
> make up this application's transaction size? While the analogy isn't
> perfect, think of it as a distant cousin of the TCP "push" bit, which
> was originally defined to permit implementations to gain efficiency by
> aggregating data when immediate delivery wasn't necessary.
> 
> I know that redefining the IP TOS fields may be beyond the scope of
> our charter, but I've been mulling it over for quite some time and I
> wanted to get this idea out there to stimulate some discussion because
> it does connect directly to our discussion of what performance-tuning
> "knobs" a link/subnetwork should provide to IP and how applications
> could use them.
> 

Phil,
Since the diffserv effort has only recently redefined the IP
TOS fields and was (and is) chartered to do so, I suspect that you
are quite right that it is out of charter for PILC. On the
other hand, it may be quite resonable to define a PHB Group
in PILC. Perhaps you should first determine that what you
want to do is impossible in that framework before rechartering
PILC and deep-sixing RFCs2474/5, RFCx2597/8?

	regards,
		Kathie


From owner-pilc@lerc.nasa.gov  Mon Jul 31 12:41:49 2000
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On Mon, 31 Jul 2000, Phil Karn wrote:

> >Besides the negative impacts of increased delays for TCP slow start,
> >increasing the interleaving delay might be harmful, because TCP has
> >this bias for connections with a large RTT. This is for me another
> >reason for not having large interleaving blocks.

bias _against_ connections with a large RTT when they're competing in
a pipe against connections with smaller RTTs which adapt faster to
changes.

Up the overall RTT of your TCP connections by interleaving, and
expect them to do less well against other connections.


> I don't quite understand this. My experience is that TCP adapts quite
> well to large RTTs, especially if the RTTs are stable and the packet
> loss rate is low. TCP window scaling is now pretty widely deployed,
> although it may have to be enabled.

L.

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




