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<< OUR APOLOGIES FOR MULTIPLE COPIES  >>

                            CALL FOR PAPERS

                Fourth IFIP/IEEE International Conference on
    Management of Multimedia Networks and Services (MMNS'2001)

                       October 29- November 1, 2001
                       Chicago, Illinois, USA
                       (http://www.mnlab.cs.depaul.edu/mmns2001)

IMPORTANT DATES
Submission deadline:  April 20, 2001
Notification of acceptance: July 1, 2001
Final version due:  July 21, 2001
Conference:   October 29 to Nov 1, 2001

The IFIP/IEEE MMNS is the premier IEEE/IFIP conferences focusing on
management of multimedia networks and services. The conference is known
for its high-quality papers from various research communities. Selected
papers
will also be considered for publication as a special issue of the Journal of
High
Speed Networking and Journal of Networks and Information Systems.
MMNS 2001 will provide Travel and Conference Scholarships for students.
Topics of interest include, but are not limited to, the following:

Distributed multimedia service management
Integration of network control and management
Network management models and architectures
Distributed event correlation
Monitoring
QoS management
Multimedia traffic management
Resource, performance and fault management for broadband networks
Configuration management of edge and core services for enabling
multimedia traffic management
Multi-point and multicast services management
Resource management in wireless multimedia
Wireless and mobile network management
Mobility management
Management of ad-hoc networks
Multimedia content protection
Deployment of multimedia services
Active network management
Network programmability for multimedia services
Middleware support for management
Multimedia session management
Packet scheduling and dropping techniques
Multimedia service Engineering
Billing and security for broadband networks
Policy-based management for multimedia service

PAPER SUBMISSION     See <http://www.mnlab.cs.depaul.edu/mmns2001>.





Ehab Al-Shaer, Ph.D.
Assistant Professor,
School of Computer Science, Telecommunications and Information Systems,
DePaul University, Chicago, IL 60604
ehab@cs.depaul.edu




From owner-tcp-impl@lerc.nasa.gov  Mon Mar  5 22:57:49 2001
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From: foo <foo@eek.org>
To: end2end-interest@ISI.EDU, tcp-impl@lerc.nasa.gov
Subject: TEAR.
Message-ID: <20010305194252.L33489@eek.org>
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Does anyone have any experience with or thoughts about TEAR (TCP Emulation
at Receivers) developed by Injong Rhee at NCSU? 

http://www.csc.ncsu.edu/faculty/rhee/export/tear_page/

-Brian


From owner-tcp-impl@lerc.nasa.gov  Tue Mar  6 01:11:43 2001
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Date: Mon, 5 Mar 2001 22:19:54 -0600
From: Min Sik Kim <minskim@cs.utexas.edu>
To: foo <foo@eek.org>
Cc: end2end-interest@ISI.EDU, tcp-impl@lerc.nasa.gov
Subject: Re: [e2e] TEAR.
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* foo <foo@eek.org> [20010305 19:55 CST]:
> Does anyone have any experience with or thoughts about TEAR (TCP Emulation
> at Receivers) developed by Injong Rhee at NCSU? 
> 
> http://www.csc.ncsu.edu/faculty/rhee/export/tear_page/

TEAR is one of the four protocols whose transient behaviors we studied in
our infocom 2001 paper.

    Yang, Kim, and Lam, Transient behaviors of TCP-friendly congestion
    control protocols, in Proc. of INFOCOM 2001.
    http://infocom.ucsd.edu/papers/529-3648348157.pdf

-- 
 Min Sik Kim
 http://www.cs.utexas.edu/users/minskim/


From owner-tcp-impl@lerc.nasa.gov  Tue Mar  6 13:47:44 2001
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Date: 	Sat, 3 Mar 2001 17:55:01 -0600 (CST)
From: Tianbo kuang <kuang@sask.trlabs.ca>
To: foo <foo@eek.org>
cc: end2end-interest@ISI.EDU, tcp-impl@lerc.nasa.gov,
        end2end-interest@postel.org
Subject: Re: [e2e] TEAR.
In-Reply-To: <20010305194252.L33489@eek.org>
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Hi,

I was just about to ask a question about TEAR. It seems unclear to me how
TEAR calculates RTT in the technical report (TEAR: TCP emulation at
receivers - flow control for multimedia streaming). Does the sender
calculate it and send it to the receiver, or does the receiver calculate
it (and how?)? In section 3.5, it does mention under the title _timeout_
that, "this information is embedded in the packet header by the sender".
What does "this information" refer to?

Cheers,

--Tianbo

------------------------------------------------------
  Kuang Tianbo                                         
  TRlabs                                               
  111-116 Research Drive                               
  Saskatoon, Saskatchewan S7N 3R3                      
  Tel: (306) 668-9325(office) (306) 343-9747 (home)                                  
  kuang@sask.trlabs.ca                                 
------------------------------------------------------
On Mon, 5 Mar 2001, foo wrote:

> Date: Mon, 5 Mar 2001 19:42:52 -0600
> From: foo <foo@eek.org>
> To: end2end-interest@ISI.EDU, tcp-impl@lerc.nasa.gov
> Subject: [e2e] TEAR.
> Resent-Date: Mon, 5 Mar 2001 19:48:05 -0600
> Resent-From: foo@eek.org
> Resent-To: end2end-interest@postel.org
> 
> Does anyone have any experience with or thoughts about TEAR (TCP Emulation
> at Receivers) developed by Injong Rhee at NCSU? 
> 
> http://www.csc.ncsu.edu/faculty/rhee/export/tear_page/
> 
> -Brian
> 



From owner-tcp-impl@lerc.nasa.gov  Tue Mar  6 14:16:54 2001
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From: "Injong Rhee" <rhee@eos.ncsu.edu>
To: "Tianbo kuang" <kuang@sask.trlabs.ca>, "foo" <foo@eek.org>
Cc: <end2end-interest@ISI.EDU>, <tcp-impl@lerc.nasa.gov>,
        <end2end-interest@postel.org>
Subject: RE: [e2e] TEAR.
Date: Tue, 6 Mar 2001 12:01:50 -0500
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Hi,

I can't help overhearing, and want to drop a few lines. The RTT calculation
can be done by the sender through receiving the receiver report about the
rate. The receiver sends back the time stamp of the last packet received to
the sender with the sequence number which will be used by the sender to
compute the RTT. This is one way to do it, and there are  many other ways.
For instance, you can use the same way that RTP does or use GPS to
synchronize the clocks and measure the one-way time. Then use the one-way
trip time in place of RTT -- I know in this case that TCP-friendliness may
suffer, but it can at least give some bounded fairness. In fact this removes
back-channel concerns completely from flow control scuh as losses and delays
in the back channels.

Some of nice things about TEAR are that (1) it does not use back channels
much so sutiable for wireless comm; (2) rate control is very smooth; (3)
TCP-friendly over various ranges of bandwidth --- TFRC has some prblems
under very low bandwidth cases.

We have improved TEAR quite bit from the initial work and  TEAR is
incorporated into an MPEG-4 stream player and stream server, and it seems to
give pretty good performance over other existing streaming solutions. Other
areas of exploration are multicast and wireless communication. Sorry I have
not kept you guys up to date about the progress. I got tired of writing
papers and have been digging into writing codes.....Maybe its time to come
out and see the light :-)

Injong


> -----Original Message-----
> From: end2end-interest-admin@postel.org
> [mailto:end2end-interest-admin@postel.org]On Behalf Of Tianbo kuang
> Sent: Saturday, March 03, 2001 6:55 PM
> To: foo
> Cc: end2end-interest@ISI.EDU; tcp-impl@lerc.nasa.gov;
> end2end-interest@postel.org
> Subject: Re: [e2e] TEAR.
>
>
> Hi,
>
> I was just about to ask a question about TEAR. It seems unclear to me how
> TEAR calculates RTT in the technical report (TEAR: TCP emulation at
> receivers - flow control for multimedia streaming). Does the sender
> calculate it and send it to the receiver, or does the receiver calculate
> it (and how?)? In section 3.5, it does mention under the title _timeout_
> that, "this information is embedded in the packet header by the sender".
> What does "this information" refer to?
>
> Cheers,
>
> --Tianbo
>
> ------------------------------------------------------
>   Kuang Tianbo
>   TRlabs
>   111-116 Research Drive
>   Saskatoon, Saskatchewan S7N 3R3
>   Tel: (306) 668-9325(office) (306) 343-9747 (home)
>
>   kuang@sask.trlabs.ca
> ------------------------------------------------------
> On Mon, 5 Mar 2001, foo wrote:
>
> > Date: Mon, 5 Mar 2001 19:42:52 -0600
> > From: foo <foo@eek.org>
> > To: end2end-interest@ISI.EDU, tcp-impl@lerc.nasa.gov
> > Subject: [e2e] TEAR.
> > Resent-Date: Mon, 5 Mar 2001 19:48:05 -0600
> > Resent-From: foo@eek.org
> > Resent-To: end2end-interest@postel.org
> >
> > Does anyone have any experience with or thoughts about TEAR
> (TCP Emulation
> > at Receivers) developed by Injong Rhee at NCSU?
> >
> > http://www.csc.ncsu.edu/faculty/rhee/export/tear_page/
> >
> > -Brian
> >
>
>



From owner-tcp-impl@lerc.nasa.gov  Wed Mar  7 04:11:47 2001
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<< OUR APOLOGIES FOR MULTIPLE COPIES >>

CALL FOR PAPERS for Special Sessions in
the Fourth IFIP/IEEE International Conference on
Management of Multimedia Networks and Services (MMNS'2001)

October 29- November 1, 2001, Chicago, Illinois, USA
(http://www.mnlab.cs.depaul.edu/mmns2001)

IMPORTANT DATES
Submission deadline:    April 20, 2001
Notification of acceptance:  July 1, 2001
Final version due:    July 21, 2001

MMNS has identified a number ``focus topics'' that are of particular
relevance and importance to the field of multimedia network and service
management. MMNS plans to conduct special sessions to address such hot
issues. Papers submitted to these special sessions will go through the
normal submission and review procedure. The goal is to build special
sessions of exceptional papers all dealing with a closely related theme.
The set of focus topics include the following:

* INTERNET TRAFFIC MONITORING:  The focus of this special session
is on techniques and methodologies for collecting Internet traffic
statistics,
descriptions of collection systems implemented and deployed in the
Internet, results from these systems, and analyses of these results
describing how multimedia networks and services can be better managed.
For questions about this special session, contact Kevin Almeroth
(almeroth@cs.ucsb.edu).

* Multimedia in AD-HOC NETWORKS: Networks of the future are envisioned
to integrate the Internet with wireless mobile embedded devices and
ad-hoc networks. Ad-hoc networks are collections of wireless mobile
nodes that can be deployed rapidly without pre-existing or centralized
communication  infrastructure. These networks should support various
types of multimedia traffic, including audio, video, TC-based data,
and sensory data. Due to the nature of ad-hoc networks, number of new
challenges must be addressed in order to facilitate seamless services of
Ad-hoc networks. This includes unicast and multicast routing protocols,
power-aware protocols in almost all layers of the network stack, quality
and guarantee of service provisioning for multimedia traffic,
self-configuring
hierarchies and architectures, address allocation and naming, data
dissemination techniques, efficient mobility support, fault and performance
management tools, performance analysis of end-to-end protocols such as TCP
and reliable multicast over ad-hoc networks. For questions about this
special
session, contact Ahmed Helmy (helmy@ceng.usc.edu)

* RESOURCE MANAGEMENT FOR WIRELESS MULTIMEDIA: Broadband wireless
multimedia services are becoming very popular. This special session will
focus on
architectures and techniques for supporting QoS Internet wireless
multimedia.
The session is looking for original contributions in this demanding area
that
address related issues. This includes integration of personal wireless
multimedia into the Internet, efficient resource management techniques that
consider the limited resources in wireless systems, such as spectrum
resource
and transmitter power, mobility management, bandwidth allocation, location
and handoff procedures, channel access and assignment, call admission
control,
and end-to-end adaptive control for wireless multimedia. For questions about
this special session, contact Raouf Boutaba(rboutaba@bbcr.uwaterloo.ca).

* MOBILE AGENT-BASED NETWORK MANAGEMENT: Research and
development on various forms of agents and mobile agents is continuing to
grow
in a staggering fashion in recent years. Agent-based telecommunication
applications and services such as active networks, Qos and bandwidth
brokers,
E-commerce, information gathering on Internet, and feature interactions are
becoming increasingly popular and are largely contributing to the
development
and to the success of distributed multimedia technology. This session will
focus
on research and development activities related to the field of agents in the
area of
active networks, and QoS agent-based architectures for managing multimedia
networks
and distributed services. The objective is to provide the audiences with a
clear
background into the opportunities and the challenges that the mobile agent
emerging
paradigm brings about For questions about this special session, contact
Ahmed Karmouch (karmouch@site.uottawa.ca), Nazism Agoulmine
(nazim@rp.lip6.fr)
or Allan Marshall (a.marshall@ee.qub.ac)



From owner-tcp-impl@lerc.nasa.gov  Wed Mar  7 04:22:12 2001
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<< OUR APOLOGIES FOR MULTIPLE COPIES >>

CALL FOR PAPERS for Special Sessions in
the Fourth IFIP/IEEE International Conference on
Management of Multimedia Networks and Services (MMNS'2001)

October 29- November 1, 2001, Chicago, Illinois, USA
(http://www.mnlab.cs.depaul.edu/mmns2001)

IMPORTANT DATES
Submission deadline:    April 20, 2001
Notification of acceptance:  July 1, 2001
Final version due:    July 21, 2001

MMNS has identified a number ``focus topics'' that are of particular
relevance and importance to the field of multimedia network and service
management. MMNS plans to conduct special sessions to address such hot
issues. Papers submitted to these special sessions will go through the
normal submission and review procedure. The goal is to build special
sessions of exceptional papers all dealing with a closely related theme.
The set of focus topics include the following:

* INTERNET TRAFFIC MONITORING:  The focus of this special session
is on techniques and methodologies for collecting Internet traffic
statistics,
descriptions of collection systems implemented and deployed in the
Internet, results from these systems, and analyses of these results
describing how multimedia networks and services can be better managed.
For questions about this special session, contact Kevin Almeroth
(almeroth@cs.ucsb.edu).

* Multimedia in AD-HOC NETWORKS: Networks of the future are envisioned
to integrate the Internet with wireless mobile embedded devices and
ad-hoc networks. Ad-hoc networks are collections of wireless mobile
nodes that can be deployed rapidly without pre-existing or centralized
communication  infrastructure. These networks should support various
types of multimedia traffic, including audio, video, TC-based data,
and sensory data. Due to the nature of ad-hoc networks, number of new
challenges must be addressed in order to facilitate seamless services of
Ad-hoc networks. This includes unicast and multicast routing protocols,
power-aware protocols in almost all layers of the network stack, quality
and guarantee of service provisioning for multimedia traffic,
self-configuring
hierarchies and architectures, address allocation and naming, data
dissemination techniques, efficient mobility support, fault and performance
management tools, performance analysis of end-to-end protocols such as TCP
and reliable multicast over ad-hoc networks. For questions about this
special
session, contact Ahmed Helmy (helmy@ceng.usc.edu)

* RESOURCE MANAGEMENT FOR WIRELESS MULTIMEDIA: Broadband wireless
multimedia services are becoming very popular. This special session will
focus on
architectures and techniques for supporting QoS Internet wireless
multimedia.
The session is looking for original contributions in this demanding area
that
address related issues. This includes integration of personal wireless
multimedia into the Internet, efficient resource management techniques that
consider the limited resources in wireless systems, such as spectrum
resource
and transmitter power, mobility management, bandwidth allocation, location
and handoff procedures, channel access and assignment, call admission
control,
and end-to-end adaptive control for wireless multimedia. For questions about
this special session, contact Raouf Boutaba(rboutaba@bbcr.uwaterloo.ca).

* MOBILE AGENT-BASED NETWORK MANAGEMENT: Research and
development on various forms of agents and mobile agents is continuing to
grow
in a staggering fashion in recent years. Agent-based telecommunication
applications and services such as active networks, Qos and bandwidth
brokers,
E-commerce, information gathering on Internet, and feature interactions are
becoming increasingly popular and are largely contributing to the
development
and to the success of distributed multimedia  technology. This session will
focus
on research and development activities related to the field of agents in the
area of
active networks, and QoS agent-based architectures for managing multimedia
networks
and distributed services. The objective is to provide the audiences with a
clear
background into the opportunities and the challenges that the mobile agent
emerging
paradigm brings about For questions about this special session, contact
Ahmed Karmouch (karmouch@site.uottawa.ca), Nazism Agoulmine
(nazim@rp.lip6.fr)
or Allan Marshall (a.marshall@ee.qub.ac)




From owner-tcp-impl@lerc.nasa.gov  Sun Mar 11 14:21:48 2001
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Message-ID: <010701c0aa67$36803860$cbda35d8@danpesta>
From: "Dan Pestana" <danp@digital.net>
To: <david.defelice@grc.nasa.gov>
Subject: PLEASE Consider this Simple, yet Profound Gyroscope Microgravity Concept
Date: Sun, 11 Mar 2001 12:09:07 -0800
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http://www.prowsales.com/gyrospoke/gyrospoke.html

Welcome to Dan Pestana's
"Gyrospoke Hovercraft"
Disclosure Webpage
This email has been created to showcase my=20
invention called the "Gyrospoke."

    =20


      The device consists of 2 sets of 4 levers, strategically and =
symmetrically connected at 8 hinged fulcrums which are fixed pivot =
points. These fulcrums support the load surface and are connected to 2 =
hub planes supporting the 2 sets of spoke levers. Each of the 8 spoke =
levers is equipped with its own motorized gyroscope on the outer end of =
the rimless spoke.=20
      Each spoke is supported midway by a hinged fulcrum leg extending =
from the payload surface above (or below) the levers. The center of the =
device is comprised of 2 hub planes driven together and apart by a =
symmetrical one piston high torque engine. The 2 hubs are connected to =
the load surface by only these fulcrum hinges. Four hub ends of the =
spoke levers are connected to the upper hub plane. The other set of 4 =
hub ends of the spoke levers are connected to the lower hub plane. (The =
hub planes are pushed and pulled apart and together by the =
electronically cycled stroke one piston engine mentioned above.)

      As the gyroscope motors on the outer end of the spokes spin their =
corresponding gyroscopes, an effort or resistance to change is =
introduced at the hinged fulcrums. This effort or force is transmitted =
to the load surface by the supporting hinged legs. An inertial force is =
created by the hub engine's independently self-contained opposition to =
the centrifugal or inertial effort of the 2 sets of four gyroscopes also =
working against each other. The inertial force created is harnessed and =
transmitted to the load. The central opposition is then leveraged by the =
corresponding and relative resistance to the gyroscope's inertial force =
at the fulcrum. The independent hub stroke effort also leverages the 2 =
sets of opposing centrifugal forces at the shared fulcrum hinges. A =
neutral microgravity is expected which will allow the Gyrospoke to =
maintain a levitation or buoyancy on any vertical plane it is located at =
or placed in. This efficient interpretation of inertial force will also =
be useful for a number of other devices needing independent gravity =
(i.e.: braking mechanisms, propulsion and powers supplies).

      I have been working on this concept for 15 years and have many =
notes and several prototypes to support my ownership of its inception. I =
have never able to get past what I thought would be the simplest part of =
the device, that is the construction of the interacting levers between =
the two hub planes. My homemade gyroscopes seemed to work okay, but =
without the two hub planes pushing and pulling symmetrically against =
each other the prototypes are rendered useless.

      I hope to experiment with the direction and speed of the =
gyroscopes' spin once my final prototype is finished. I presume that for =
the sake of stability, it will be wise to the have gyroscopes adjacent =
to each other have symmetrically opposite spins to neutralize their =
inclination to tilt when a load is applied.

      I welcome any person's constructive input.

      I hereby authorize the construction of this device by any person =
wishing to test my theory. Should you build a working device and wish to =
profit financially from its construction, I would be happy to entertain =
a nominal bid for sale of usage to you or your company.

      Dan Pestana
      8002 N. Packwood Ave.
      Tampa, FL 33604-3815
      (813) 931-8036
      danp@digital.net


    =20
=20



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Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4522.1800" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffff00>
<DIV align=3Dcenter><FONT face=3DArial></FONT>&nbsp;</DIV>
<DIV align=3Dcenter><FONT face=3DArial><A=20
href=3D"http://www.prowsales.com/gyrospoke/gyrospoke.html">http://www.pro=
wsales.com/gyrospoke/gyrospoke.html</A></FONT></DIV>
<DIV>
<DIV align=3Dcenter>
<CENTER><FONT face=3DArial></FONT>&nbsp;</CENTER></DIV>
<H1 align=3Dcenter>Welcome to Dan Pestana's<BR>"Gyrospoke=20
Hovercraft"<BR>Disclosure Webpage</H1>
<H3 align=3Dcenter>This&nbsp;email has been created to showcase my =
<BR>invention=20
called the "Gyrospoke."</H3>
<DIV align=3Dcenter>
<CENTER>
<TABLE cellSpacing=3D1 border=3D3>
  <TBODY>
  <TR>
    <TD width=3D"100%">
      <P align=3Dcenter><A=20
      =
href=3D"http://www.prowsales.com/gyrospoke/gyrospoke-big.html"><IMG=20
      height=3D112 =
src=3D"http://www.prowsales.com/gyrospoke/gyrospoke-sm.jpg"=20
      width=3D150 =
border=3D0></A></P></TD></TR></TBODY></TABLE></CENTER></DIV>
<DIV align=3Dcenter>
<CENTER>
<TABLE width=3D"90%" border=3D0>
  <TBODY>
  <TR>
    <TD width=3D"100%"><FONT face=3D"Times New Roman"><BR>The device =
consists of 2=20
      sets of 4 levers, strategically and symmetrically connected at 8 =
hinged=20
      fulcrums which are fixed pivot points. These fulcrums support the =
load=20
      surface and are connected to 2 hub planes supporting the 2 sets of =
spoke=20
      levers. Each of the 8 spoke levers is equipped with its own =
motorized=20
      gyroscope on the outer end of the rimless spoke.</FONT>=20
      <P><FONT face=3D"Times New Roman">Each spoke is supported midway =
by a hinged=20
      fulcrum leg extending from the payload surface above (or below) =
the=20
      levers. The center of the device is comprised of 2 hub planes =
driven=20
      together and apart by a symmetrical one piston high torque engine. =
The 2=20
      hubs are connected to the load surface by only these fulcrum =
hinges. Four=20
      hub ends of the spoke levers are connected to the upper hub plane. =
The=20
      other set of 4 hub ends of the spoke levers are connected to the =
lower hub=20
      plane. (The hub planes are pushed and pulled apart and together by =
the=20
      electronically cycled stroke one piston engine mentioned=20
above.)</FONT></P>
      <P><FONT face=3D"Times New Roman">As the gyroscope motors on the =
outer end=20
      of the spokes spin their corresponding gyroscopes, an effort or =
resistance=20
      to change is introduced at the hinged fulcrums. This effort or =
force is=20
      transmitted to the load surface by the supporting hinged legs. An =
inertial=20
      force is created by the hub engine's independently self-contained=20
      opposition to the centrifugal or inertial effort of the 2 sets of =
four=20
      gyroscopes also working against each other. The inertial force =
created is=20
      harnessed and transmitted to the load. The central opposition is =
then=20
      leveraged by the corresponding and relative resistance to the =
gyroscope's=20
      inertial force at the fulcrum. The independent hub stroke effort =
also=20
      leverages the 2 sets of opposing centrifugal forces at the shared =
fulcrum=20
      hinges. A neutral microgravity is expected which will allow the =
Gyrospoke=20
      to maintain a levitation or buoyancy on any vertical plane it is =
located=20
      at or placed in. This efficient interpretation of inertial force =
will also=20
      be useful for a number of other devices needing independent =
gravity (i.e.:=20
      braking mechanisms, propulsion and powers supplies).</FONT></P>
      <P><FONT face=3D"Times New Roman">I have been working on this =
concept for 15=20
      years and have many notes and several prototypes to support my =
ownership=20
      of its inception. I have never able to get past what I thought =
would be=20
      the simplest part of the device, that is the construction of the=20
      interacting levers between the two hub planes. My homemade =
gyroscopes=20
      seemed to work okay, but without the two hub planes pushing and =
pulling=20
      symmetrically against each other the prototypes are rendered=20
      useless.</FONT></P>
      <P><FONT face=3D"Times New Roman">I hope to experiment with the =
direction=20
      and speed of the gyroscopes' spin once my final prototype is =
finished. I=20
      presume that for the sake of stability, it will be wise to the =
have=20
      gyroscopes adjacent to each other have symmetrically opposite =
spins to=20
      neutralize their inclination to tilt when a load is =
applied.</FONT></P>
      <P><FONT face=3D"Times New Roman">I welcome any person's =
constructive=20
      input.</FONT></P>
      <P><FONT face=3D"Times New Roman">I hereby authorize the =
construction of=20
      this device by any person wishing to test my theory. Should you =
build a=20
      working device and wish to profit financially from its =
construction, I=20
      would be happy to entertain a nominal bid for sale of usage to you =
or your=20
      company.</FONT></P>
      <P><FONT face=3D"Times New Roman">Dan Pestana<BR>8002 N. Packwood=20
      Ave.<BR>Tampa, FL 33604-3815<BR>(813) 931-8036<BR><A=20
      href=3D"mailto:danp@digital.net">danp@digital.net</A></FONT></P>
      <P>&nbsp;</P></TD></TR></TBODY></TABLE><A=20
href=3D"http://www.submitexpress.com/"></A> </CENTER></DIV>
<P align=3Dcenter><FONT =
face=3DArial></FONT>&nbsp;</P></DIV></BODY></HTML>

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------=_NextPart_000_0103_01C0AA24.134F90E0--



From owner-tcp-impl@lerc.nasa.gov  Sun Mar 11 14:38:53 2001
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Message-ID: <007b01c0aa67$ac30b300$b0da35d8@danpesta>
From: "Dan Pestana" <danp@digital.net>
To: <david.defelice@grc.nasa.gov>
Subject: PLEASE Consider this Simple, yet Profound Gyroscope Microgravity Concept
Date: Sun, 11 Mar 2001 12:12:33 -0800
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Welcome to Dan Pestana's "Gyrospoke Hovercraft"
Disclosure Webpage
This email has been created to showcase my
invention called the "Gyrospoke."

http://www.prowsales.com/gyrospoke/gyrospoke.html


The device consists of 2 sets of 4 levers, strategically and symmetrically
connected at 8 hinged fulcrums which are fixed pivot points. These fulcrums
support the load surface and are connected to 2 hub planes supporting the 2
sets of spoke levers. Each of the 8 spoke levers is equipped with its own
motorized gyroscope on the outer end of the rimless spoke.
Each spoke is supported midway by a hinged fulcrum leg extending from the
payload surface above (or below) the levers. The center of the device is
comprised of 2 hub planes driven together and apart by a symmetrical one
piston high torque engine. The 2 hubs are connected to the load surface by
only these fulcrum hinges. Four hub ends of the spoke levers are connected
to the upper hub plane. The other set of 4 hub ends of the spoke levers are
connected to the lower hub plane. (The hub planes are pushed and pulled
apart and together by the electronically cycled stroke one piston engine
mentioned above.)

As the gyroscope motors on the outer end of the spokes spin their
corresponding gyroscopes, an effort or resistance to change is introduced at
the hinged fulcrums. This effort or force is transmitted to the load surface
by the supporting hinged legs. An inertial force is created by the hub
engine's independently self-contained opposition to the centrifugal or
inertial effort of the 2 sets of four gyroscopes also working against each
other. The inertial force created is harnessed and transmitted to the load.
The central opposition is then leveraged by the corresponding and relative
resistance to the gyroscope's inertial force at the fulcrum. The independent
hub stroke effort also leverages the 2 sets of opposing centrifugal forces
at the shared fulcrum hinges. A neutral microgravity is expected which will
allow the Gyrospoke to maintain a levitation or buoyancy on any vertical
plane it is located at or placed in. This efficient interpretation of
inertial force will also be useful for a number of other devices needing
independent gravity (i.e.: braking mechanisms, propulsion and powers
supplies).
I have been working on this concept for 15 years and have many notes and
several prototypes to support my ownership of its inception. I have never
able to get past what I thought would be the simplest part of the device,
that is the construction of the interacting levers between the two hub
planes. My homemade gyroscopes seemed to work okay, but without the two hub
planes pushing and pulling symmetrically against each other the prototypes
are rendered useless.

I hope to experiment with the direction and speed of the gyroscopes' spin
once my final prototype is finished. I presume that for the sake of
stability, it will be wise to the have gyroscopes adjacent to each other
have symmetrically opposite spins to neutralize their inclination to tilt
when a load is applied.

I welcome any person's constructive input.
I hereby authorize the construction of this device by any person wishing to
test my theory. Should you build a working device and wish to profit
financially from its construction, I would be happy to entertain a nominal
bid for sale of usage to you or your company.

Dan Pestana
8002 N. Packwood Ave.
Tampa, FL 33604-3815
(813) 931-8036
danp@digital.net



From owner-tcp-impl@lerc.nasa.gov  Mon Mar 12 07:45:13 2001
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Subject: (no subject)
References: <00ee01c0795d$dd406c00$38072ac2@cs.ucy.ac.cy>
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5th WSES/IEEE WORLD MULTICONFERENCE ON
Circuits, Systems, Communications & Computers (CSCC 2001) 
Location: Rethymnon, CRETE (GREECE) 
JULY 8-15, 2001
(The Island of Crete has FREQUENT DIRECT Connection with all 
European Capitals and big cities in Summer!). 

* 5th World Conference on Circuits,
* 5th World Conference on Systems,
* 5th World Conference on Communications,
* 5th World Conference on Computers,

jointly with the
WSES/IEEE CONFERENCE ON
Signal Processing, Multidimensional Systems, Computational Geometry and
Artificial Vision
(organized by Prof. G.Antoniou)

jointly with the
WSES/IEEE CONFERENCE ON
Neural, Fuzzy and Evolutionary Computation
(organized by Prof. V.Mladenov)

jointly with the
WSES/IEEE CONFERENCE ON
Multirate Systems and Wavelets Analysis
(organized by Prof. B.Suter)

jointly with the
WSES/IEEE CONFERENCE ON
Software and Hardware Engineering
(organized by Prof. O.Panfilov)

jointly with the
WSES/IEEE CONFERENCE ON
Power Engineering
(organized by Prof. L.A.Pecorelli)

jointly with the
WSES CONFERENCE ON
Mathematics and Computers in Physics
(organized by Prof. L.J.Wang)

jointly with the
WSES CONFERENCE ON
Mathematics and Computers in Mechanical Engineerig
(organized by Prof. F.Koumboulis and G.Stavroulakis)


WEB: http://www.worldses.org
[PLEASE, FOLLOW APPROPRIATE LINK]
See also the impressions from the previous conference in 2000.

ALL THE ACCEPTED PAPERS will be published:
1)in the CD-ROM Proceedings (with Search Facilities and Page Numbering) 
as well as 
2)in the Electrical and Computer Engineering International Reference
Book Series of WSES PRESS as Post-Conference Books (Hard cover, velvet
paper, international circulation). 
These will be different International Editions (with different ISBN). 
This material will be ready at the opening of the Multiconference and
will be distributed to the participants. 

Also SPECIAL ISSUES have been scheduled for the journals : (These
special issues will contain only selected papers, not all the papers) 

INTERNATIONAL JOURNAL OF COMPUTER RESEARCH (IJCR) 
INFORMATICA 
INTERNATIONAL JOURNAL OF COMMUNICATIONS AND NETWORKING

More informations: Send a message to me 
or send a message to mbetini@italymail.com
or to cscc@worldses.org

About Crete:
CRETE is a Unique Place in the World: In this magic island, so richly
endowed by Nature and History, everything is in abundance: blond beaches
with
palm-trees, cool gorges, serene coves, palm-forests, mysterious caves,
minoan palaces, paleochristian churches, byzantine churches, islamic
mosques, and of course modern cosmopolitan life.
Crete lies in the crossroads of the three ancient continents:
Europe-Asia-Africa and the Cretan Civilization borrowed elements from
european, asian and african world. Also, Crete, in the minoan era, gave
birth to the first civilization in the west-european sense.
Here, Zeus, the father and king of ancient greek Gods was born.
Many famous Artists and Politicians like El-Greco, Venizelos,
Kazantzakis and Kornaros were born in Crete.

EXTENSION OF THE DEADLINE FOR PAPER SUBMISSION
UNTIL MARCH 20, 2001
- - - - - - - - - - - - - - - - - - - - - - - -


From owner-tcp-impl@lerc.nasa.gov  Tue Mar 13 07:01:09 2001
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Subject: INFOCOM 2001, Anchorage Alaska, 22-26 April 2001  -->  Call for participation
Date: Tue, 13 Mar 2001 11:26:05 +0200
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!!!    PLEASE ACCEPT OUR SICNERE APOLOGIES FOR MULTIPLE COPIES. !!!


              C A L L   F O R    P A R T I C I P A T I O N
                   I E E E   I N F O C O M    2 0 0 1

                    http://www.ieee-infocom.org/2001

                 April 22-26, 2001 - Anchorage, Alaska



TRAIN CHARTER: This year we are chartering a train for our welcome
reception. This train ride will take you through the Kenai fjords
and will include food and drinks. April 24, 6-11 pm. Seats are limited,
and will be given out on the basis of registration date.

CONFERENCE REGISTRATION: The deadline for advance registration
is April 2, 2001 (5pm Eastern Standard Time).
Please register early to ensure that you have
a seat in the train charter. On-line registration is possible at our
website at http://www.ieee-infocom.org/2001 and then clicking on
"Registration Information". Your dinner code is 530T.
Enter this code while registering and you may be a lucky winner of a
$100 dinner at INFOCOM 2001.

HOTEL REGISTRATION: Please book your hotel room early at the Hilton.
Rooms are filling up fast. If the Hilton gets filled up, the Marriott
is our alternative hotel for the conference (same room rate as the Hilton).

TOURS: Several tours of Alaska's natural treasures are planned. Please visit
our website at http://www.ieee-infocom.org/2001 and look for "On-line
tour registration".

AIRLINE TICKETS: Please book your flights to Anchorage early, so that
you can get reasonably priced tickets.

TECHNICAL SESSIONS:
48 technical sessions featuring the latest research and trends in the
field of computer communications and netwroking.

TUTORIALS:
Wireless Home Networks: Technologies and Applications
Quality of Service (QoS) Support for Broadband Wireless Networks
Traffic Engineering in IP/MPLS Networks
TCP Congestion Controls: Algorithms and Models
Video Over IP
Bluetooth and Pervasive Wireless Networking
Control and Management for Optical Networks: An IP-Centric Approach
Principles of Multicast Protocols and Services

PANELS:
All-Optical Networks: Fact or Fiction
Next Generation Wireless Networks: Radio, Networking and Applications
Modeling of the Shrew: the Quest for a "Model" Network Model










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              C A L L   F O R    P A R T I C I P A T I O N
                   I E E E   I N F O C O M    2 0 0 1

                    http://www.ieee-infocom.org/2001

                 April 22-26, 2001 - Anchorage, Alaska



TRAIN CHARTER: This year we are chartering a train for our welcome
reception. This train ride will take you through the Kenai fjords
and will include food and drinks. April 24, 6-11 pm. Seats are limited,
and will be given out on the basis of registration date.

CONFERENCE REGISTRATION: The deadline for advance registration=20
is April 2, 2001 (5pm Eastern Standard Time).=20
Please register early to ensure that you have
a seat in the train charter. On-line registration is possible at our=20
website at http://www.ieee-infocom.org/2001 and then clicking on
"Registration Information". Your dinner code is 530T.
Enter this code while registering and you may be a lucky winner of a
$100 dinner at INFOCOM 2001.=20

HOTEL REGISTRATION: Please book your hotel room early at the Hilton.
Rooms are filling up fast. If the Hilton gets filled up, the Marriott
is our alternative hotel for the conference (same room rate as the =
Hilton).

TOURS: Several tours of Alaska's natural treasures are planned. Please =
visit
our website at http://www.ieee-infocom.org/2001 and look for "On-line
tour registration".

AIRLINE TICKETS: Please book your flights to Anchorage early, so that
you can get reasonably priced tickets.

TECHNICAL SESSIONS:
48 technical sessions featuring the latest research and trends in the
field of computer communications and netwroking.

TUTORIALS:
Wireless Home Networks: Technologies and Applications=20
Quality of Service (QoS) Support for Broadband Wireless Networks=20
Traffic Engineering in IP/MPLS Networks=20
TCP Congestion Controls: Algorithms and Models=20
Video Over IP=20
Bluetooth and Pervasive Wireless Networking=20
Control and Management for Optical Networks: An IP-Centric Approach=20
Principles of Multicast Protocols and Services=20

PANELS:
All-Optical Networks: Fact or Fiction
Next Generation Wireless Networks: Radio, Networking and Applications
Modeling of the Shrew: the Quest for a "Model" Network Model









------=_NextPart_000_063C_01C0ABB0.646AAB80--



From owner-tcp-impl@lerc.nasa.gov  Thu Mar 15 18:15:00 2001
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Reply-To: <karmouch@site.uottawa.ca>
From: "Ahmed Karmouch" <Karmouch@site.uottawa.ca>
To: "karmouch" <karmouch@site.uottawa.ca>
Subject: IEEE/ACM Workshop on Mobile Software Agents-  MATA 2001- Deadline-April 1.
Date: Thu, 15 Mar 2001 08:49:25 -0500
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[My apologies if you receive this more than once]
======================================================
			MATA'2001

  Third International Workshop on Mobile Agents for
	Telecommunication Applications
	August 14-16, 2001, Montréal, CANADA
	  Co-sponsored by IEEE and ACM
	http://www.congresbcu.com/mata01/


CALL FOR PAPERS
================
Technical papers (4500 words maximum) describing previously unpublished,
completed research or work in progress, not currently under review by
another journal or conference, are solicited on the following topics:
Mobile Agent Architecture and Models
Agent Identification, Tracking and Persistence
Agent-based Mobility Management in Mobile Networks
Web Agent Systems
Agent Integration with CORBA and TINA
Active Networks and Mobile Agents
Feature Interaction and Agents
Mobile Agents Communication Language
Security in Mobile Agent Systems
Interactive Multimedia Presentation Agents
Agent-based Electronic Commerce
Agent-based Access to Legacy Services
Managing QoS with agents
Information Discovery and Gathering using agents
Data Mining Agents
Network Management Agents
Policy-based Management using Mobile Agents
Education and Applications of Mobile Agents
Prototypes and Experience with Mobile Agents
Seamless Messaging and Mobile Agents

Accepted papers will be published in the conference proceedings. Papers of
particular merit will be proposed for publication in IEEE Network  Magazine.
All paper submissions should have a cover page containing the title, names,
email address and complete postal addresses (including telephone and fax
numbers) for all authors. Please indicate the main author for the purpose of
correspondence. The cover page should also provide an abstract (150 words
maximum), and a list of keywords. Please include a statement stating that
"when accepted, one of the authors will attend the Workshop to present the
paper".
Submissions should be sent by email to the program chair:

Roch Glitho
Ericcson Research Canada
8400, boul. Décarie
Ville Mont-Royal, Québec, Canada, H4P 2N2
email: roch.glitho@lmc.ericsson.se
Tel.: (514) 345-7000 ext. 2266

IMPORTANT DATES FOR PAPER SUBMISSION
Full paper submission due	April 1, 2001
Notification of acceptance	May 15, 2001
Final paper due	June 15, 2001
=======================================







From owner-tcp-impl@lerc.nasa.gov  Mon Mar 26 07:32:35 2001
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From: rajiv.chakravorty@philips.com
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To: <tcp-impl@grc.nasa.gov>
Subject: query: using two write() calls in succession
Message-ID: <0056890024992624000002L942*@MHS>
Date: Mon, 26 Mar 2001 14:14:02 +0200
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--Boundary=_0.0_=0056890024992624
Content-Type: text/plain; charset=iso-8859-1; name="MEMO 03/26/01 13:09:56"
Content-Disposition: inline
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Hi,=20

This problem is regarding use of  two write() call in succession. Let m=
e explain the=20
problem. In one of the blocking socket implementations, I have used two=
 write calls mainly=20
to detect the state of the TCP connection. I can expect that if the TCP=
 connection has=20
been closed (FIN) by the http server (due to http/1.1  timeout), the us=
e of two write calls in sequence=20
at the client side should detect if the connection was closed. As norma=
lly expected, when I=20
make the first write() call, the server would ackowledge the client wit=
h a RST since=20
the connection was  already closed. Now, when I make my second write() =
call, in=20
succession, the presence of RST at the client TCP should prompt the sec=
ond write()=20
to fail and return with errno to ECONNRESET. I wrote a small code to ch=
eck this behaviour.
I  ran this code in linux 2.2.16 to retrive the index.html page from an=
 intranet web server.
In this code, I made a TCP connection to send and retrieve the index.ht=
ml file normally and=20
then slept for about 20 seconds allowing the web server to timeout.  Af=
ter 20 seconds, after the
web server had timed-out and closed the connection, I again called the =
two writes (breaking the request)
 with the same GET HTTP/1.1 request in succession.=20
I have some observations:

1. The second write() call in succession  does not fail. However, if I =
sleep for 1 second
in between the two writes, the second write() call fails and returns er=
rno with ECONNRESET.
I checked this behaviour running it in Linux2.2.16/ SunOS _5.5.1/ 5.8 a=
nd the behavior remains same.
 However, in Sun OS 5.7, second write() call does not fail at all even =
after giving this delay. (I tried to
 increase the delay, but without any result). In VxWorks 5.2, we could =
detect the error in the second=20
write call without any delay at all.

Q.1.1  Why do we need some delay for the second write() call to fail in=
 most of the systems??
Q.2 .1 Why do we see such inconsistencies in various TCP implementation=
s? (maybe I am missing=20
something)...


2. I checked the tcpdump output of Linux 2.2.16 and I find that when I =
make the second write call
and after the client has received RST from the first write , the client=
 TCP generates it own RST to be=20
sent to the server. I do think that this is not the expected behaviour =
from the TCP. I checked the tcpdump
output running the same code in SunOS 5.7 and I dont find any RST being=
 returned from the client to=20
the server, after it itself has received the RST. This is the normal ex=
pected behavior of TCP, which=20
is somehow not evident in Linux (I have included the tcpdump file; linu=
xdump.txt)

Q.2.1 Is there some problems in the 2.2.16 TCP stack of Linux that prom=
pts it to return RST for an RST=20
when the second write() call is made ??

Maybe I am missing something important or my interpretation of the prob=
lem.=20
Would really appreciate your comments.

Regds,
Rajiv Chakravorty=20

Philips Advanced Systems and Applications Labs (ASA-Labs)
Eindhoven, The Netherlands     rajiv.chakravorty@philips.com

ps: appended, please find the tcpdump(linuxdump.txt) for linux and
the sampe code (testsock2.cpp) code used in this process.
for compilation and final executable:  g++ testsock2.cpp -o testsock -l=
nsl -lsocket
for accessing the server use:  testsock  <webserver>  <file to be retri=
eved>



=

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ZXJybm8pKTsKCQlleGl0KDQpOwoJfQoKCWZwcmludGYoc3Rkb3V0LCAiV3JpdHRlbiAlZCBjaGFy
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SUdQSVBFLCBTSUdfSUdOKTsKCWZwcmludGYoc3RkZXJyLCAiU0lHUElQRSBjYXVnaHRcbiIpOwoJ
cmV0dXJuOwp9Cg==

--Boundary=_0.0_=0056890024992624--


From owner-tcp-impl@lerc.nasa.gov  Mon Mar 26 10:26:02 2001
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From: der Mouse  <mouse@Rodents.Montreal.QC.CA>
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> 1. The second write() call in succession  does not fail. However, if I =
> sleep for 1 second
> in between the two writes, the second write() call fails and returns er=
> rno with ECONNRESET.

This doesn't surprise me.  Without the delay, the first write()
generates an outgoing segment, sure, which draws an RST segment in
response from the peer - but the second write() occurs well before the
RST is received by the sending host.  Even on a fast LAN, this usually
takes at least a millisecond or so, and it's the rare machine that's so
slow as to be unable to do back-to-back writes in less than that.

The rest of your questions I can't help with.

					der Mouse

			       mouse@rodents.montreal.qc.ca
		     7D C8 61 52 5D E7 2D 39  4E F1 31 3E E8 B3 27 4B


From owner-tcp-impl@lerc.nasa.gov  Mon Mar 26 11:25:17 2001
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Subject: Re: query: using two write() calls in succession
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Please find my comments appended.

Regards,
Rajiv Chakravorty




>This doesn't surprise me.  Without the delay, the first write()
>generates an outgoing segment, sure, which draws an RST segment in
>response from the peer - but the second write() occurs well before the
>RST is received by the sending host.  Even on a fast LAN, this usually
>takes at least a millisecond or so, and it's the rare machine that's so
>slow as to be unable to do back-to-back writes in less than that.


Yes, this was my first guess. But it doesnt happen this way. Actually,
I think I am receiving the RST well before I perform the second write. 
Here's the tcpdump output when the code is run on (Linux 2.2.16) without 1 sec 
sleep between two writes (sorry for not enabling the relative time option):

client: wsasa3005
www/server: asaehv.ce.philips.com

17:31:17.185481 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: S 1292046262:1292046262(0) win 32120 <mss 1460,sackOK,timestamp 104441703 0,nop,wscale 0> (DF)
17:31:17.185986 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: S 3010001540:3010001540(0) ack 1292046263 win 8760 <mss 1460> (DF)
17:31:17.186090 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: . 1:1(0) ack 1 win 32120 (DF)
17:31:17.186250 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 1:2(1) ack 1 win 32120 (DF)
17:31:17.233839 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 1:1(0) ack 2 win 8760 (DF)
17:31:17.233891 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 2:34(32) ack 1 win 32120 (DF)
17:31:17.247262 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: P 1:1461(1460) ack 34 win 8760 (DF)
17:31:17.247337 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: . 34:34(0) ack 1461 win 30660 (DF)
17:31:17.249255 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 1461:2921(1460) ack 34 win 8760 (DF)
17:31:17.250033 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: . 34:34(0) ack 2921 win 30660 (DF)
17:31:17.250391 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: P 2921:4243(1322) ack 34 win 8760 (DF)
17:31:17.253254 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 4243:5703(1460) ack 34 win 8760 (DF)
17:31:17.254483 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 5703:7163(1460) ack 34 win 8760 (DF)
17:31:17.254546 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: . 34:34(0) ack 7163 win 27740 (DF)
17:31:17.256139 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 7163:8347(1184) ack 34 win 8760 (DF)
17:31:17.257379 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: P 8347:9807(1460) ack 34 win 8760 (DF)
17:31:17.258604 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: P 9807:11267(1460) ack 34 win 8760 (DF)
17:31:17.258670 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: . 34:34(0) ack 11267 win 26280 (DF)
17:31:17.259607 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: P 11267:12451(1184) ack 34 win 8760 (DF)
17:31:17.260026 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: . 34:34(0) ack 12451 win 25096 (DF)
17:31:17.261449 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 12451:13911(1460) ack 34 win 8760 (DF)
17:31:17.262677 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 13911:15371(1460) ack 34 win 8760 (DF)
17:31:17.262741 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: . 34:34(0) ack 15371 win 23360 (DF)
17:31:17.263905 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 15371:16831(1460) ack 34 win 8760 (DF)
17:31:17.264875 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 16831:17865(1034) ack 34 win 8760 (DF)
17:31:17.266120 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 17865:19325(1460) ack 34 win 8760 (DF)
17:31:17.266203 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: . 34:34(0) ack 19325 win 21900 (DF)
17:31:17.267351 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: P 19325:20785(1460) ack 34 win 8760 (DF)
17:31:17.268485 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: P 20785:21835(1050) ack 34 win 8760 (DF)
17:31:17.270034 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: . 34:34(0) ack 21835 win 19390 (DF)
17:31:33.748842 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: F 21835:21835(0) ack 34 win 8760 (DF)
17:31:33.748906 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: . 34:34(0) ack 21836 win 19389 (DF)
17:31:37.250061 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 34:39(5) ack 21836 win 19389 (DF)
17:31:37.250450 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: R 3010023376:3010023376(0) win 8760 (DF)
17:31:37.251146 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: R 67:67(0) ack 21836 win 32120 (DF)
 

As you can see, the client TCP receives the RST (450-061) = 389 microsecs
after making the first write. I believe this time is sufficient enough for the
client TCP to fail on the second write() call, that was made probably after it had received
the RST. Or is it not ???

The other inconsistent behavior that I was talking about is that you can also see the client also
generating an RST and sending it back to the server after it had received an RST, when the second 
write() call was made slightly probably before 17:31:37.251146.

Please comment.



 
















From owner-tcp-impl@lerc.nasa.gov  Mon Mar 26 13:51:00 2001
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From: Rick Jones <raj@cup.hp.com>
Organization: the Unofficial HP
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Subject: Re: query: using two write() calls in succession
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rajiv.chakravorty@philips.com wrote:
> 
> Hi,
> 
> This problem is regarding use of  two write() call in succession. Let me explain the
> problem. In one of the blocking socket implementations, I have used two write calls
> mainly to detect the state of the TCP connection. I can expect that if the TCP
> connection has been closed (FIN) by the http server (due to http/1.1  timeout), the

Um, isn't the remote's sending of a FIN supposed to generate a
zero-length read to inform the other side that there was at least a
partial shutdown?

> use of two write calls in sequence at the client side should detect if the connection
> was closed. As normally expected, when I make the first write() call, the server
> would ackowledge the client with a RST since the connection was  already closed. Now,
> when I make my second write() call, in succession, the presence of RST at the client
> TCP should prompt the second write() to fail and return with errno to ECONNRESET. I
> wrote a small code to check this behaviour. I  ran this code in linux 2.2.16 to
> retrive the index.html page from an intranet web server. In this code, I made a TCP
> connection to send and retrieve the index.html file normally and then slept for about
> 20 seconds allowing the web server to timeout.  After 20 seconds, after the web
> server had timed-out and closed the connection, I again called the two writes
> (breaking the request) with the same GET HTTP/1.1 request in succession.
> I have some observations:
> 

> 1. The second write() call in succession  does not fail. However, if I sleep for 1
> second in between the two writes, the second write() call fails and returns errno
> with ECONNRESET. I checked this behaviour running it in Linux2.2.16/ SunOS _5.5.1/
> 5.8 and the behavior remains same.  However, in Sun OS 5.7, second write() call does
> not fail at all even after giving this delay. (I tried to  increase the delay, but
> without any result). In VxWorks 5.2, we could detect the error in the second write
> call without any delay at all.

As derMouse pointed-out, once you send the first segment, there is arace
between the other side getting the RST to the sending TCP, and the
sending app getting the second write into TCP.

As for why no delay was required in VxWorks, I'll hazard the guess that
the system running VxWorks was slow enough that the RST arrived before
the second segment could be sent by the app. 

As for why SunOS5.7 never got the ECONNRESET, probably a bug in 5.7.

I still think that paying attention to the zero-length read to detect
remote (graceful) close is best. doing things that elicit RST's is not
such a hot idea.

rick jones
-- 
ftp://ftp.cup.hp.com/dist/networking/misc/rachel/
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to email, OR post, but please do NOT do BOTH...
my email address is raj in the cup.hp.com domain...


From owner-tcp-impl@lerc.nasa.gov  Tue Mar 27 14:35:49 2001
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real products, prototypes, or experimental systems and their performance 
evaluation. In addition to those subscribing to the main theme of the 
conference, contributions are also solicited on the following topics:

- Gb/sec and Tb/sec switching and routing technologies
- High-speed packet processing engines
- Serial link technologies
- xDSL, HFC, and wireless access technologies
- Mobility-enabling technologies
- Seamless appliance interconnectivity
- Network-attached storage devices and interfaces
- Video and voice over packet networks
- Wireless and mobile network interfaces
- Network security
- Next-generation networking chip sets
- Network protocols on a chip
- Low-power networking and interconnect technologies
- Network appliance technologies
- Optical Interconnects

Submission Guidelines:
---------------------------------
Submissions must be in the form of extended summaries and should not 
exceed 5 pages (double-column format). The summary should have 
sufficient technical detail to judge its quality and suitability for 
presentation at the conference. Guidelines for submission are available 
at http://www.hoti.org.
Inquires about the conference should be directed to the TPC chairs.


Important Dates:
-----------------------
 - Submission deadline: April 15, 2001.
 - Notification of acceptance: May 15, 2001.
 - Camera-ready due: July 1, 2001.


Organizing Committee:
---------------------------------
- General Chair: Fred Bauer, Nokia (fred@iprg.nokia.com)

- Technical Program Chairs:
    - Marwan Krunz, University of Arizona  (krunz@ece.arizona.edu)
    - John Lockwood, Washington University in St. Louis 
    (lockwood@arl.wustl.edu)

- Tutorial/Panel: Ibrahim Matta, Boston University (matta@cs.bu.edu)

- Treasurer: Melanie Swan, MS Financial Consulting
    (melanie.swan@wharton.upenn.edu)

- Registration: Bob Wedig, Wedig Consulting (bob@wedig.com)

- Webmaster: Raymond Lee, IP Dynamics (raymondl@ipdynamics.com)

- Local arrangements: Liz Rogers, Liz Rogers Design 
    (liz@lizrogersdesign.com)

- Publicity: Zhi-Li Zhang, University of Minnesota (zhzhang@cs.umn.edu)

Technical Program Committee: 

Anujan Varma, University of California Santa Cruz
Balaji Prabhakar, Stanford University 
Dan Pitt, Nortel Networks
Ian Akyildiz, Georgia Institute of Technology
James Yee, Pacific Broadband
Peter Newman, Ensim
Zhi-Li Zhang, University of Minnesota 
Nina Taft, Sprint Labs 
Christopher Diot,  Sprint Labs 
George Apostolopoulos, Reback Networks 
Andrew Cambell, Columbia University 
Parameswaran Ramanathan, University of Wisconsin
Mohan Kumar, University of Texas at Arlington
Ron Srodawa, Oakland University
James P.G. Sterbenz, BBN
Edward Knightly, Rice University



From owner-tcp-impl@lerc.nasa.gov  Wed Mar 28 06:34:48 2001
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From: rajiv.chakravorty@philips.com
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To: <tcp-impl@grc.nasa.gov>
Subject: Re: query: using two write() calls in succession
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Rick and der Mouse,

I have some doubts regarding your interpretation of the problem. I was 
under the impression that a blocking write (or send) call would block until
it has received an ACK or an RST for data sent and then perform the 2nd write.
I have noticed that this  happens even when I had made two write calls by 
breaking  the request (1+32)  while making the first normal request to the server. 
(check the previous linuxdump.txt file). I have appended relevant parts of it here:

17:31:17.186250 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 1:2(1) ack 1 win 32120 (DF) 
17:31:17.233839 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 1:1(0) ack 2 win 8760 (DF) 
17:31:17.233891 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 2:34(32) ack 1 win 32120 (DF) 

As you can see, when I make my first write call by sending 1 byte of data, it  waits until 
it receives the ACK and then proceed with the second write. That means, in this case write call
blocks until the server or peer TCP ackowledges the data. (unless using non-blocking 
sockets, or using MSG_DONTWAIT option if used with send() call). Also, in this case, I have not 
disabled Nagle's algoritm. Ofcourse, my linux client is a dual P-II terminal. 

After, looking into all this, is it possible that RST can be received even after the application has already
come out from the first write call ?? ...or its actually the tcp stack that's still not able to detect the RST just
before sending the request data for the second write (as you can see it still detects if some delay is given).  
Maybe something to do with silly window syndrome since I am writing small amount of data... ??

Please enlighten me with comments and/or pointers.

Regards,
Rajiv Chakravorty 



From owner-tcp-impl@lerc.nasa.gov  Wed Mar 28 08:44:32 2001
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Date: Wed, 28 Mar 2001 08:28:44 -0500 (EST)
From: der Mouse  <mouse@Rodents.Montreal.QC.CA>
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> I was under the impression that a blocking write (or send) call would
> block until it has received an ACK or an RST for data sent and then
> perform the 2nd write.

This is not the way any implementation I've used works (unless the
first write is larger than the buffering capacity of the socket).

> I have noticed that this happens even when I had made two write calls
> by breaking the request (1+32) while making the first normal request
> to the server.  (check the previous linuxdump.txt file).  I have
> appended relevant parts of it here:

> 17:31:17.186250 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 1:2(1) ack 1 win 32120 (DF) 
> 17:31:17.233839 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 1:1(0) ack 2 win 8760 (DF) 
> 17:31:17.233891 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 2:34(32) ack 1 win 32120 (DF) 

> As you can see, when I make my first write call by sending 1 byte of
> data, it  waits until it receives the ACK and then proceed with the
> second write.

You are confusing the timing of the write() calls with the timing of
the data actually going out onto the wire.

> That means, in this case write call blocks until the server or peer
> TCP ackowledges the data.

No, it means that the TCP does not send the data until then.  It says
nothing about when the write calls complete (except, of course, that
the first write must have completed before the second write's data goes
out).

					der Mouse

			       mouse@rodents.montreal.qc.ca
		     7D C8 61 52 5D E7 2D 39  4E F1 31 3E E8 B3 27 4B


From owner-tcp-impl@lerc.nasa.gov  Wed Mar 28 10:16:57 2001
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From: rajiv.chakravorty@philips.com
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Cc: <tcp-impl@grc.nasa.gov>
Subject: Re: query: using two write() calls in succession
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der Mouse and all,

As indicated by der Mouse, I performed two writes by making two connections to the server. 
I did find that data with the second write call made through the second connection appeared
on the wire before I could receive the RST from the first write (made from the first connection).
This clarifies der Mouse's point.

Thanks,
Rajiv






mouse@Rodents.Montreal.QC.CA@SMTP@lerc.nasa.gov on 03-28-2001 03:49:40 PM
Sent by:	owner-tcp-impl@lerc.nasa.gov
To:	Rajiv Chakravorty/EHV/CE/PHILIPS@EMEA3
cc:	tcp-impl@grc.nasa.gov@SMTP 
Subject:	Re: query: using two write() calls in succession
Classification:	


> I was under the impression that a blocking write (or send) call would
> block until it has received an ACK or an RST for data sent and then
> perform the 2nd write.

This is not the way any implementation I've used works (unless the
first write is larger than the buffering capacity of the socket).

> I have noticed that this happens even when I had made two write calls
> by breaking the request (1+32) while making the first normal request
> to the server.  (check the previous linuxdump.txt file).  I have
> appended relevant parts of it here:

> 17:31:17.186250 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 1:2(1) ack 1 win 32120 (DF)
> 17:31:17.233839 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 1:1(0) ack 2 win 8760 (DF)
> 17:31:17.233891 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 2:34(32) ack 1 win 32120 (DF)

> As you can see, when I make my first write call by sending 1 byte of
> data, it  waits until it receives the ACK and then proceed with the
> second write.

You are confusing the timing of the write() calls with the timing of
the data actually going out onto the wire.

> That means, in this case write call blocks until the server or peer
> TCP ackowledges the data.

No, it means that the TCP does not send the data until then.  It says
nothing about when the write calls complete (except, of course, that
the first write must have completed before the second write's data goes
out).

					der Mouse

			       mouse@rodents.montreal.qc.ca
		     7D C8 61 52 5D E7 2D 39  4E F1 31 3E E8 B3 27 4B




From owner-tcp-impl@lerc.nasa.gov  Wed Mar 28 15:00:04 2001
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Date: Wed, 28 Mar 2001 10:06:36 -0800
From: Rick Jones <raj@cup.hp.com>
Organization: the Unofficial HP
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Subject: Re: query: using two write() calls in succession
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rajiv.chakravorty@philips.com wrote:
> 
> Rick and der Mouse,
> 
> I have some doubts regarding your interpretation of the problem. I was
> under the impression that a blocking write (or send) call would block
> until it has received an ACK or an RST for data sent and then perform
> the 2nd write.

I am afraid your impression, while perhaps natual to the untainted :),
is incorrect. As often as not, a send/write call to a socket will
complete once the data has been copied into the stack. This is well
before it is ACKnowledged by the remote, or even sent out the wire.

> I have noticed that this  happens even when I had made two write calls
> by breaking  the request (1+32)  while making the first normal request
> to the server. (check the previous linuxdump.txt file). I have
> appended relevant parts of it here:
> 
> 17:31:17.186250 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 1:2(1) ack 1 win 32120 (DF)
> 17:31:17.233839 eth0 < asaehv.ce.philips.com.www > wsasa3005.asa-ehv.ce.philips.com.1181: . 1:1(0) ack 2 win 8760 (DF)
> 17:31:17.233891 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 > asaehv.ce.philips.com.www: P 2:34(32) ack 1 win 32120 (DF)
> 
> As you can see, when I make my first write call by sending 1 byte of
> data, it  waits until it receives the ACK and then proceed with the
> second write. That means, in this case write call blocks until the
> server or peer TCP ackowledges the data. (unless using non-blocking
> sockets, or using MSG_DONTWAIT option if used with send() call). Also,
> in this case, I have not disabled Nagle's algoritm. Ofcourse, my linux
> client is a dual P-II terminal.

What you are seeing in that trace is the Nagle algorithm, implying
nothing about the completion semantics of the write() call against a
socket. I suspect that if you were to run a system call trace on your
program at the same time, you would find that the second write() call
happened before the ACK.

> After, looking into all this, is it possible that RST can be received
> even after the application has already come out from the first write
> call ?? ...or its actually the tcp stack that's still not able to
> detect the RST just before sending the request data for the second
> write (as you can see it still detects if some delay is given). Maybe
> something to do with silly window syndrome since I am writing small
> amount of data... ??

Even on a "blocking socket" a send/write completes once the data is
written into the socket buffer. This is independent of the decisions TCP
makes about when to send data onto the wire. 

If the socket buffer is full, the write() call will block, but only
until enough space is freed via ACK's or _prior_ writes to allow the
data of this write() into the buffer.

I guess the fancier way to say it is that writing data into the
connection is asynchronous from that data being send on the wire.

All splitting writes will do is cause your app lication to run afoul of
the nagle algorithm and take more CPU cycles on the system than it
should.

If you want to detect remote connection close (via FIN) use the
zero-bytes on a recv/read semantic. If you want to detect remote
connection death, implement a keepalive handshake in your app.

rick jones
-- 
ftp://ftp.cup.hp.com/dist/networking/misc/rachel/
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to email, OR post, but please do NOT do BOTH...
my email address is raj in the cup.hp.com domain...


From owner-tcp-impl@lerc.nasa.gov  Wed Mar 28 17:27:25 2001
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To: "der Mouse" <mouse@Rodents.Montreal.QC.CA>,
        <rajiv.chakravorty@philips.com>
Cc: <tcp-impl@grc.nasa.gov>
References: <200103281328.IAA02478@Twig.Rodents.Montreal.QC.CA>
Subject: Re: query: using two write() calls in succession
Date: Thu, 29 Mar 2001 03:08:23 +0500
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I unlisted my self from this list long ago, but I still receive mails from
this list every now and then.
May I please be removed
Thanks

----- Original Message -----
From: "der Mouse" <mouse@Rodents.Montreal.QC.CA>
To: <rajiv.chakravorty@philips.com>
Cc: <tcp-impl@grc.nasa.gov>
Sent: Wednesday, March 28, 2001 6:28 PM
Subject: Re: query: using two write() calls in succession


> > I was under the impression that a blocking write (or send) call would
> > block until it has received an ACK or an RST for data sent and then
> > perform the 2nd write.
>
> This is not the way any implementation I've used works (unless the
> first write is larger than the buffering capacity of the socket).
>
> > I have noticed that this happens even when I had made two write calls
> > by breaking the request (1+32) while making the first normal request
> > to the server.  (check the previous linuxdump.txt file).  I have
> > appended relevant parts of it here:
>
> > 17:31:17.186250 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 >
asaehv.ce.philips.com.www: P 1:2(1) ack 1 win 32120 (DF)
> > 17:31:17.233839 eth0 < asaehv.ce.philips.com.www >
wsasa3005.asa-ehv.ce.philips.com.1181: . 1:1(0) ack 2 win 8760 (DF)
> > 17:31:17.233891 eth0 > wsasa3005.asa-ehv.ce.philips.com.1181 >
asaehv.ce.philips.com.www: P 2:34(32) ack 1 win 32120 (DF)
>
> > As you can see, when I make my first write call by sending 1 byte of
> > data, it  waits until it receives the ACK and then proceed with the
> > second write.
>
> You are confusing the timing of the write() calls with the timing of
> the data actually going out onto the wire.
>
> > That means, in this case write call blocks until the server or peer
> > TCP ackowledges the data.
>
> No, it means that the TCP does not send the data until then.  It says
> nothing about when the write calls complete (except, of course, that
> the first write must have completed before the second write's data goes
> out).
>
> der Mouse
>
>        mouse@rodents.montreal.qc.ca
>      7D C8 61 52 5D E7 2D 39  4E F1 31 3E E8 B3 27 4B


_________________________________________________________
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<< OUR APOLOGIES FOR MULTIPLE COPIES >>

CALL FOR PAPERS for Special Sessions in
the Fourth IFIP/IEEE International Conference on
Management of Multimedia Networks and Services (MMNS'2001)

October 29- November 1, 2001, Chicago, Illinois, USA
(http://www.mnlab.cs.depaul.edu/mmns2001)

IMPORTANT DATES
Submission deadline:    April 20, 2001
Notification of acceptance:  July 1, 2001
Final version due:    July 21, 2001

MMNS has identified a number ``focus topics'' that are of particular
relevance and importance to the field of multimedia network and service
management. MMNS plans to conduct special sessions to address such hot
issues. Papers submitted to these special sessions will go through the
normal submission and review procedure. The goal is to build special
sessions of exceptional papers all dealing with a closely related theme.
The set of focus topics include the following:

* INTERNET TRAFFIC MONITORING:  The focus of this special session
is on techniques and methodologies for collecting Internet traffic
statistics,
descriptions of collection systems implemented and deployed in the
Internet, results from these systems, and analyses of these results
describing how multimedia networks and services can be better managed.
For questions about this special session, contact Kevin Almeroth
(almeroth@cs.ucsb.edu).

* Multimedia in AD-HOC NETWORKS: Networks of the future are envisioned
to integrate the Internet with wireless mobile embedded devices and
ad-hoc networks. Ad-hoc networks are collections of wireless mobile
nodes that can be deployed rapidly without pre-existing or centralized
communication  infrastructure. These networks should support various
types of multimedia traffic, including audio, video, TC-based data,
and sensory data. Due to the nature of ad-hoc networks, number of new
challenges must be addressed in order to facilitate seamless services of
Ad-hoc networks. This includes unicast and multicast routing protocols,
power-aware protocols in almost all layers of the network stack, quality
and guarantee of service provisioning for multimedia traffic,
self-configuring hierarchies and architectures, address allocation and
naming,
data dissemination techniques, efficient mobility support, fault and
performance
management tools, performance analysis of end-to-end protocols such as TCP
and reliable multicast over ad-hoc networks. For questions about this
special
session, contact Ahmed Helmy (helmy@ceng.usc.edu)

* RESOURCE MANAGEMENT FOR WIRELESS MULTIMEDIA: Broadband wireless
multimedia services are becoming very popular. This special session will
focus on architectures and techniques for supporting QoS Internet wireless
multimedia. The session is looking for original contributions in this
demanding area
that address related issues. This includes integration of personal wireless
multimedia into the Internet, efficient resource management techniques that
consider the limited resources in wireless systems, such as spectrum
resource and transmitter power, mobility management, bandwidth allocation,
location
and handoff procedures, channel access and assignment, call admission
control, and end-to-end adaptive control for wireless multimedia. For
questions about
this special session, contact Raouf Boutaba(rboutaba@bbcr.uwaterloo.ca).

* MOBILE AGENT-BASED NETWORK MANAGEMENT: Research and
development on various forms of agents and mobile agents is continuing to
grow in a staggering fashion in recent years. Agent-based telecommunication
applications and services such as active networks, Qos and bandwidth
brokers, E-commerce, information gathering on Internet, and feature
interactions are
becoming increasingly popular and are largely contributing to the
development and to the success of distributed multimedia  technology. This
session will
focus on research and development activities related to the field of agents
in the
area of active networks, and QoS agent-based architectures for managing
multimedia
networks and distributed services. The objective is to provide the audiences
with a
clear background into the opportunities and the challenges that the mobile
agent
emerging paradigm brings about For questions about this special session,
contact
Ahmed Karmouch (karmouch@site.uottawa.ca), Nazism Agoulmine
(nazim@rp.lip6.fr)
or Allan Marshall (a.marshall@ee.qub.ac)

Ehab Al-Shaer, Ph.D.
Assistant Professor,
School of Computer Science, Telecommunications and Information Systems,
DePaul University, Chicago, IL 60604
ehab@cs.depaul.edu




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<< OUR APOLOGIES FOR MULTIPLE COPIES  >>

                            CALL FOR PAPERS

                Fourth IFIP/IEEE International Conference on
    Management of Multimedia Networks and Services (MMNS'2001)

                       October 29- November 1, 2001
                       Chicago, Illinois, USA
                       (http://www.mnlab.cs.depaul.edu/mmns2001)

IMPORTANT DATES
    Submission deadline:  April 20, 2001
    Notification of acceptance: July 1, 2001
    Final version due:  July 21, 2001
    Conference:   October 29 to Nov 1, 2001

The IFIP/IEEE MMNS is the premier IEEE/IFIP conferences focusing on
management of multimedia networks and services. The conference is known
for its high-quality papers from various research communities. Selected
papers will also be considered for publication as a special issue of the
Journal
of High Speed Networking and Journal of Networks and Information Systems.
MMNS 2001 will provide Travel and Conference Scholarships for students.
Topics of interest include, but are not limited to, the following:

Distributed multimedia service management
Integration of network control and management
Network management models and architectures
Distributed event correlation
Monitoring
QoS management
Multimedia traffic management
Resource, performance and fault management for broadband networks
Configuration management of edge and core services for enabling
multimedia traffic management
Multi-point and multicast services management
Resource management in wireless multimedia
Wireless and mobile network management
Mobility management
Management of ad-hoc networks
Multimedia content protection
Deployment of multimedia services
Active network management
Network programmability for multimedia services
Middleware support for management
Multimedia session management
Packet scheduling and dropping techniques
Multimedia service Engineering
Billing and security for broadband networks
Policy-based management for multimedia service

PAPER SUBMISSION PROCEDURE     See
<http://www.mnlab.cs.depaul.edu/mmns2001>.





From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 06:40:49 2001
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Subject: initial window size
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hi
 Iam new to this. My doubt is this. With what criteria the intial window for TCP is calculated and set. From the tcp code of
linux, I can interpret that the allowable receive buffer is set to a value 32767. The initial window should be less than this
value. Can any one of u explain me why the initial window is chosen less than 32767 and how is that calculation go.

Thanks
-arun





From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 07:01:57 2001
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Hello arun,

The purpose of a small value for TCP initial window is to limit the
inital 'train' of packets. It is up to each TCP sender to decide the
value, but the higher this value the higher the possibility to end up
with a congestion, because the sender would transmit, without 'probing'
the network, all these packets back-to-back. The main problem is usually
not the TCP receiver not being able to cope with the data, but the path
betwen the sender and the receiver being congested.
This is why the initial window has, preferably, a low value

Bogdan


arun wrote:
> 
> hi
>  Iam new to this. My doubt is this. With what criteria the intial window for TCP is calculated and set. From the tcp code of
> linux, I can interpret that the allowable receive buffer is set to a value 32767. The initial window should be less than this
> value. Can any one of u explain me why the initial window is chosen less than 32767 and how is that calculation go.
> 
> Thanks
> -arun


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It's always been my understanding that the default initial window size is
512K for purposes of flow control (avoiding the overflow of sender's and
receiver's buffers) and to avoid fragmentation of the TCP PDU until the
MTU - and max receive window - can be negotiated.


Dave.



----- Original Message -----
From: "Bogdan Ghita" <b.ghita@jack.see.plym.ac.uk>
To: "arun" <arun@netlab.hcltech.com>
Cc: <tcp-impl@grc.nasa.gov>
Sent: Thursday, March 29, 2001 6:53 AM
Subject: Re: initial window size


> Hello arun,
>
> The purpose of a small value for TCP initial window is to limit the
> inital 'train' of packets. It is up to each TCP sender to decide the
> value, but the higher this value the higher the possibility to end up
> with a congestion, because the sender would transmit, without 'probing'
> the network, all these packets back-to-back. The main problem is usually
> not the TCP receiver not being able to cope with the data, but the path
> betwen the sender and the receiver being congested.
> This is why the initial window has, preferably, a low value
>
> Bogdan
>
>
> arun wrote:
> >
> > hi
> >  Iam new to this. My doubt is this. With what criteria the intial window
for TCP is calculated and set. From the tcp code of
> > linux, I can interpret that the allowable receive buffer is set to a
value 32767. The initial window should be less than this
> > value. Can any one of u explain me why the initial window is chosen less
than 32767 and how is that calculation go.
> >
> > Thanks
> > -arun
>



From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 16:08:59 2001
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From: "Oleg Vishnepolsky" <oleg@inforocket.com>
To: "David St. Germain" <David@StGermain.com>, <tcp-impl@grc.nasa.gov>
Subject: RE: initial window size
Date: Thu, 29 Mar 2001 15:51:49 -0500
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512K is a bit much for initial size, to say the least. In fact 512 bytes (not kbytes) is what is used by a lot of
tcp stacks when establishing connection with non-local hosts. 512 is really conservative IMHO. 
Days when backbone MTUs were that small are gone long time ago.
For hosts on the local net, mileage varies depending on the stack. I used 8K in my stacks
(OS/2 circa 1990, FlexOS circa 93). I found that given prevalent MTUs back then of 1460,
8k gave me best performance.

Oleg Vishnepolsky
-----Original Message-----
From: owner-tcp-impl@lerc.nasa.gov
[mailto:owner-tcp-impl@lerc.nasa.gov]On Behalf Of David St. Germain
Sent: Thursday, March 29, 2001 1:01 PM
To: tcp-impl@grc.nasa.gov
Subject: Re: initial window size


It's always been my understanding that the default initial window size is
512K for purposes of flow control (avoiding the overflow of sender's and
receiver's buffers) and to avoid fragmentation of the TCP PDU until the
MTU - and max receive window - can be negotiated.


Dave.



----- Original Message -----
From: "Bogdan Ghita" <b.ghita@jack.see.plym.ac.uk>
To: "arun" <arun@netlab.hcltech.com>
Cc: <tcp-impl@grc.nasa.gov>
Sent: Thursday, March 29, 2001 6:53 AM
Subject: Re: initial window size


> Hello arun,
>
> The purpose of a small value for TCP initial window is to limit the
> inital 'train' of packets. It is up to each TCP sender to decide the
> value, but the higher this value the higher the possibility to end up
> with a congestion, because the sender would transmit, without 'probing'
> the network, all these packets back-to-back. The main problem is usually
> not the TCP receiver not being able to cope with the data, but the path
> betwen the sender and the receiver being congested.
> This is why the initial window has, preferably, a low value
>
> Bogdan
>
>
> arun wrote:
> >
> > hi
> >  Iam new to this. My doubt is this. With what criteria the intial window
for TCP is calculated and set. From the tcp code of
> > linux, I can interpret that the allowable receive buffer is set to a
value 32767. The initial window should be less than this
> > value. Can any one of u explain me why the initial window is chosen less
than 32767 and how is that calculation go.
> >
> > Thanks
> > -arun
>



From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 16:23:41 2001
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Date: Thu, 29 Mar 2001 16:14:04 -0500 (EST)
From: Martin McSweeney <mmcsween@hopper.math.uwaterloo.ca>
To: Oleg Vishnepolsky <oleg@inforocket.com>
cc: "David St. Germain" <David@StGermain.com>, tcp-impl@grc.nasa.gov
Subject: RE: initial window size
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Hmmm... "a bit much", isn't 512K an extremely large window given any
circumstances?

-martin

Martin McSweeney
mmcsween@styx.uwaterloo.ca


On Thu, 29 Mar 2001, Oleg Vishnepolsky wrote:

> 512K is a bit much for initial size, to say the least. In fact 512 bytes (not kbytes) is what is used by a lot of
> tcp stacks when establishing connection with non-local hosts. 512 is really conservative IMHO. 
> Days when backbone MTUs were that small are gone long time ago.
> For hosts on the local net, mileage varies depending on the stack. I used 8K in my stacks
> (OS/2 circa 1990, FlexOS circa 93). I found that given prevalent MTUs back then of 1460,
> 8k gave me best performance.
> 
> Oleg Vishnepolsky
> -----Original Message-----
> From: owner-tcp-impl@lerc.nasa.gov
> [mailto:owner-tcp-impl@lerc.nasa.gov]On Behalf Of David St. Germain
> Sent: Thursday, March 29, 2001 1:01 PM
> To: tcp-impl@grc.nasa.gov
> Subject: Re: initial window size
> 
> 
> It's always been my understanding that the default initial window size is
> 512K for purposes of flow control (avoiding the overflow of sender's and
> receiver's buffers) and to avoid fragmentation of the TCP PDU until the
> MTU - and max receive window - can be negotiated.
> 
> 
> Dave.
> 
> 
> 
> ----- Original Message -----
> From: "Bogdan Ghita" <b.ghita@jack.see.plym.ac.uk>
> To: "arun" <arun@netlab.hcltech.com>
> Cc: <tcp-impl@grc.nasa.gov>
> Sent: Thursday, March 29, 2001 6:53 AM
> Subject: Re: initial window size
> 
> 
> > Hello arun,
> >
> > The purpose of a small value for TCP initial window is to limit the
> > inital 'train' of packets. It is up to each TCP sender to decide the
> > value, but the higher this value the higher the possibility to end up
> > with a congestion, because the sender would transmit, without 'probing'
> > the network, all these packets back-to-back. The main problem is usually
> > not the TCP receiver not being able to cope with the data, but the path
> > betwen the sender and the receiver being congested.
> > This is why the initial window has, preferably, a low value
> >
> > Bogdan
> >
> >
> > arun wrote:
> > >
> > > hi
> > >  Iam new to this. My doubt is this. With what criteria the intial window
> for TCP is calculated and set. From the tcp code of
> > > linux, I can interpret that the allowable receive buffer is set to a
> value 32767. The initial window should be less than this
> > > value. Can any one of u explain me why the initial window is chosen less
> than 32767 and how is that calculation go.
> > >
> > > Thanks
> > > -arun
> >
> 
> 



From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 16:28:41 2001
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From: "Oleg Vishnepolsky" <oleg@inforocket.com>
To: "Martin McSweeney" <mmcsween@hopper.math.uwaterloo.ca>
Cc: "David St. Germain" <David@StGermain.com>, <tcp-impl@grc.nasa.gov>
Subject: RE: initial window size
Date: Thu, 29 Mar 2001 16:15:40 -0500
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I said "to say the least". Gush, my sense of humor failed me ... Next time I will be dead serious. 

Oleg Vishnepolsky
-----Original Message-----
From: Martin McSweeney [mailto:mmcsween@hopper.math.uwaterloo.ca]
Sent: Thursday, March 29, 2001 4:14 PM
To: Oleg Vishnepolsky
Cc: David St. Germain; tcp-impl@grc.nasa.gov
Subject: RE: initial window size


Hmmm... "a bit much", isn't 512K an extremely large window given any
circumstances?

-martin

Martin McSweeney
mmcsween@styx.uwaterloo.ca


On Thu, 29 Mar 2001, Oleg Vishnepolsky wrote:

> 512K is a bit much for initial size, to say the least. In fact 512 bytes (not kbytes) is what is used by a lot of
> tcp stacks when establishing connection with non-local hosts. 512 is really conservative IMHO. 
> Days when backbone MTUs were that small are gone long time ago.
> For hosts on the local net, mileage varies depending on the stack. I used 8K in my stacks
> (OS/2 circa 1990, FlexOS circa 93). I found that given prevalent MTUs back then of 1460,
> 8k gave me best performance.
> 
> Oleg Vishnepolsky
> -----Original Message-----
> From: owner-tcp-impl@lerc.nasa.gov
> [mailto:owner-tcp-impl@lerc.nasa.gov]On Behalf Of David St. Germain
> Sent: Thursday, March 29, 2001 1:01 PM
> To: tcp-impl@grc.nasa.gov
> Subject: Re: initial window size
> 
> 
> It's always been my understanding that the default initial window size is
> 512K for purposes of flow control (avoiding the overflow of sender's and
> receiver's buffers) and to avoid fragmentation of the TCP PDU until the
> MTU - and max receive window - can be negotiated.
> 
> 
> Dave.
> 
> 
> 
> ----- Original Message -----
> From: "Bogdan Ghita" <b.ghita@jack.see.plym.ac.uk>
> To: "arun" <arun@netlab.hcltech.com>
> Cc: <tcp-impl@grc.nasa.gov>
> Sent: Thursday, March 29, 2001 6:53 AM
> Subject: Re: initial window size
> 
> 
> > Hello arun,
> >
> > The purpose of a small value for TCP initial window is to limit the
> > inital 'train' of packets. It is up to each TCP sender to decide the
> > value, but the higher this value the higher the possibility to end up
> > with a congestion, because the sender would transmit, without 'probing'
> > the network, all these packets back-to-back. The main problem is usually
> > not the TCP receiver not being able to cope with the data, but the path
> > betwen the sender and the receiver being congested.
> > This is why the initial window has, preferably, a low value
> >
> > Bogdan
> >
> >
> > arun wrote:
> > >
> > > hi
> > >  Iam new to this. My doubt is this. With what criteria the intial window
> for TCP is calculated and set. From the tcp code of
> > > linux, I can interpret that the allowable receive buffer is set to a
> value 32767. The initial window should be less than this
> > > value. Can any one of u explain me why the initial window is chosen less
> than 32767 and how is that calculation go.
> > >
> > > Thanks
> > > -arun
> >
> 
> 



From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 16:58:32 2001
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From: Vernon Schryver <vjs@calcite.rhyolite.com>
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> From: Martin McSweeney <mmcsween@hopper.math.uwaterloo.ca>

> Hmmm... "a bit much", isn't 512K an extremely large window given any
> circumstances?

No, it is not so very large if we are not talking about the initial window
size and if you care about going fast.

512 KBytes is only 4 Mbits and a 45 Mbit/sec pipe for only 100 ms. 
A 5,000 mile wire has an RTT of 100 ms. from the speed of light alone.
Assuming zero delay from routers or other bottlenecks, you cannot reach
full speed with a window of only 512K on a 5,000 mile T3.

Even in a 100 Mbit/sec local area network, a window of 512K is not very
large, since 40 ms is not a lot to cover the delays added by a few routers
or bridges and the scheduling and other delays in the hosts at the ends
of a TCP connection.

You almost certainly don't want a tiny 8K window on a 100 Mbit/sec or
faster network, since 8KBytes occupies a 100 Mbit/sec wire for only
640 microseconds.

There are networks that are fast that 512K might be reasonable even
for an initial congestion window.  For example, what is the right
initial congestion window for a 10G Ethernet?


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 16:58:34 2001
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To: arun <arun@netlab.hcltech.com>
From: Mark Allman <mallman@grc.nasa.gov>
Reply-To: mallman@grc.nasa.gov
cc: tcp-impl@grc.nasa.gov
Subject: Re: initial window size 
Organization: BBN Technologies/NASA GRC
Song-of-the-Day: Sweet Emotion
Date: Thu, 29 Mar 2001 16:46:31 -0500
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>  Iam new to this. My doubt is this. With what criteria the intial
> window for TCP is calculated and set. From the tcp code of linux,
> I can interpret that the allowable receive buffer is set to a
> value 32767. The initial window should be less than this
> value. Can any one of u explain me why the initial window is
> chosen less than 32767 and how is that calculation go.

See RFC 2581: 

    The initial window should be 1 or 2 segments (each up to an
    MSS).  (This document also provides the spec. for the congestion
    control algorithms used to increase the window size).

See RFC 2414:

    The initial window may be 3 or 4 segments, depending on the size
    of the segments (basically as long as the 3 or 4 segments do not
    exceed 4380 bytes).

2581 is a proposed standard; 2414 is experimental.

allman


---
Mark Allman -- NASA GRC/BBN -- http://roland.grc.nasa.gov/~mallman/


From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 17:21:53 2001
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Date: Thu, 29 Mar 2001 14:10:48 -0800
From: Joe Touch <touch@ISI.EDU>
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To: Vernon Schryver <vjs@calcite.rhyolite.com>
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Vernon Schryver wrote:
> 
> > From: Martin McSweeney <mmcsween@hopper.math.uwaterloo.ca>
> 
> > Hmmm... "a bit much", isn't 512K an extremely large window given any
> > circumstances?
> 
> No, it is not so very large if we are not talking about the initial window
> size and if you care about going fast.
> 
> 512 KBytes is only 4 Mbits and a 45 Mbit/sec pipe for only 100 ms.
> A 5,000 mile wire has an RTT of 100 ms. from the speed of light alone.
> Assuming zero delay from routers or other bottlenecks, you cannot reach
> full speed with a window of only 512K on a 5,000 mile T3.
> 
> Even in a 100 Mbit/sec local area network, a window of 512K is not very
> large, since 40 ms is not a lot to cover the delays added by a few routers
> or bridges and the scheduling and other delays in the hosts at the ends
> of a TCP connection.
> 
> You almost certainly don't want a tiny 8K window on a 100 Mbit/sec or
> faster network, since 8KBytes occupies a 100 Mbit/sec wire for only
> 640 microseconds.
> 
> There are networks that are fast that 512K might be reasonable even
> for an initial congestion window.  For example, what is the right
> initial congestion window for a 10G Ethernet?

That might depend on whether the connection is local or not. Local
connections (same subnet as an attached interface) ignore the congestion
window in some TCP implementations.

Joe


From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 18:12:31 2001
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From: "Oleg Vishnepolsky" <oleg@inforocket.com>
To: "Vernon Schryver" <vjs@calcite.rhyolite.com>, <tcp-impl@grc.nasa.gov>
Subject: RE: initial window size
Date: Thu, 29 Mar 2001 17:58:10 -0500
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Since TCP stack does not have direct knowledge of the speed, the MTU gives a good starting point.
A while back the reasonable ratio was for the window to be about 6-8 times the MTU size for the
local connections (slow start algorithm for non-local).
Is it still the case these days ? 

Oleg Vishnepolsky
-----Original Message-----
From: owner-tcp-impl@lerc.nasa.gov
[mailto:owner-tcp-impl@lerc.nasa.gov]On Behalf Of Vernon Schryver
Sent: Thursday, March 29, 2001 4:48 PM
To: tcp-impl@grc.nasa.gov
Subject: RE: initial window size



Even in a 100 Mbit/sec local area network, a window of 512K is not very
large, since 40 ms is not a lot to cover the delays added by a few routers
or bridges and the scheduling and other delays in the hosts at the ends
of a TCP connection.

You almost certainly don't want a tiny 8K window on a 100 Mbit/sec or
faster network, since 8KBytes occupies a 100 Mbit/sec wire for only
640 microseconds.

There are networks that are fast that 512K might be reasonable even
for an initial congestion window.  For example, what is the right
initial congestion window for a 10G Ethernet?


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 18:18:07 2001
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Reply-To: "David St. Germain" <David@StGermain.com>
From: "David St. Germain" <David@StGermain.com>
To: <tcp-impl@grc.nasa.gov>
References: <NEBBJFHFADLMLFEFPEHEKEMLDMAA.oleg@inforocket.com>
Subject: Re: initial window size
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You are quite right of course, I meant 512 is what most OS's default to.



Dave.



----- Original Message -----
From: "Oleg Vishnepolsky" <oleg@inforocket.com>
To: "David St. Germain" <David@StGermain.com>; <tcp-impl@grc.nasa.gov>
Sent: Thursday, March 29, 2001 3:51 PM
Subject: RE: initial window size


> 512K is a bit much for initial size, to say the least. In fact 512 bytes
(not kbytes) is what is used by a lot of
> tcp stacks when establishing connection with non-local hosts. 512 is
really conservative IMHO.
> Days when backbone MTUs were that small are gone long time ago.
> For hosts on the local net, mileage varies depending on the stack. I used
8K in my stacks
> (OS/2 circa 1990, FlexOS circa 93). I found that given prevalent MTUs back
then of 1460,
> 8k gave me best performance.
>
> Oleg Vishnepolsky
> -----Original Message-----
> From: owner-tcp-impl@lerc.nasa.gov
> [mailto:owner-tcp-impl@lerc.nasa.gov]On Behalf Of David St. Germain
> Sent: Thursday, March 29, 2001 1:01 PM
> To: tcp-impl@grc.nasa.gov
> Subject: Re: initial window size
>
>
> It's always been my understanding that the default initial window size is
> 512K for purposes of flow control (avoiding the overflow of sender's and
> receiver's buffers) and to avoid fragmentation of the TCP PDU until the
> MTU - and max receive window - can be negotiated.
>
>
> Dave.
>
>
>
> ----- Original Message -----
> From: "Bogdan Ghita" <b.ghita@jack.see.plym.ac.uk>
> To: "arun" <arun@netlab.hcltech.com>
> Cc: <tcp-impl@grc.nasa.gov>
> Sent: Thursday, March 29, 2001 6:53 AM
> Subject: Re: initial window size
>
>
> > Hello arun,
> >
> > The purpose of a small value for TCP initial window is to limit the
> > inital 'train' of packets. It is up to each TCP sender to decide the
> > value, but the higher this value the higher the possibility to end up
> > with a congestion, because the sender would transmit, without 'probing'
> > the network, all these packets back-to-back. The main problem is usually
> > not the TCP receiver not being able to cope with the data, but the path
> > betwen the sender and the receiver being congested.
> > This is why the initial window has, preferably, a low value
> >
> > Bogdan
> >
> >
> > arun wrote:
> > >
> > > hi
> > >  Iam new to this. My doubt is this. With what criteria the intial
window
> for TCP is calculated and set. From the tcp code of
> > > linux, I can interpret that the allowable receive buffer is set to a
> value 32767. The initial window should be less than this
> > > value. Can any one of u explain me why the initial window is chosen
less
> than 32767 and how is that calculation go.
> > >
> > > Thanks
> > > -arun
> >
>
>



From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 18:38:30 2001
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Date: Thu, 29 Mar 2001 16:26:36 -0700 (MST)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
Message-Id: <200103292326.f2TNQaH10763@calcite.rhyolite.com>
To: tcp-impl@grc.nasa.gov
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> From: "Oleg Vishnepolsky" <oleg@inforocket.com>

> Since TCP stack does not have direct knowledge of the speed

As Joe Touch implied, that may be often true, but it is not always true.

>                                  the MTU gives a good starting point.

I don't see what the MTU has to do with link speed.
Yes, I know by heart a bunch of MTU's for various links, but ...

> A while back the reasonable ratio was for the window to be about 6-8 times the MTU size for the
> local connections (slow start algorithm for non-local).
> Is it still the case these days ? 

I doubt it was ever true that 6-8 times the interface MTU was the right
window size.  TCP stacks I've touched have not used anything that tiny
since the mid-1980's, when I jacked up one UNIX vendor's default windows
size to 30K, and that was partly because 32K or more hit sign-bit bugs in
some other vendors' systems.  At the time, my UNIX employer had real support
for only 10 MHz Ethernet.

Disclaimer: my definition of "right" has long been tainted by "goes
fastest according to various benchmarks."


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 18:40:10 2001
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From: "David St. Germain" <David@StGermain.com>
To: <tcp-impl@grc.nasa.gov>
References: <200103292147.f2TLluJ08719@calcite.rhyolite.com> <3AC3B2E8.6C59C446@isi.edu>
Subject: Re: initial window size
Date: Thu, 29 Mar 2001 18:26:05 -0500
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Although I agree with you that this is how one would tweak the window size,
the OS still has to default to something.  With Windows NT 4.0, I know the
default upon installation is 512 (I just recently saw that statistic), I'm
not sure what the other OS's default to upon installation, but I believe the
original RFC recommendation was to start with a default of 512 and use the
slow start algorithm to ramp up.



Dave.



----- Original Message -----
From: "Joe Touch" <touch@ISI.EDU>
To: "Vernon Schryver" <vjs@calcite.rhyolite.com>
Cc: <tcp-impl@grc.nasa.gov>
Sent: Thursday, March 29, 2001 5:10 PM
Subject: Re: initial window size


>
>
> Vernon Schryver wrote:
> >
> > > From: Martin McSweeney <mmcsween@hopper.math.uwaterloo.ca>
> >
> > > Hmmm... "a bit much", isn't 512K an extremely large window given any
> > > circumstances?
> >
> > No, it is not so very large if we are not talking about the initial
window
> > size and if you care about going fast.
> >
> > 512 KBytes is only 4 Mbits and a 45 Mbit/sec pipe for only 100 ms.
> > A 5,000 mile wire has an RTT of 100 ms. from the speed of light alone.
> > Assuming zero delay from routers or other bottlenecks, you cannot reach
> > full speed with a window of only 512K on a 5,000 mile T3.
> >
> > Even in a 100 Mbit/sec local area network, a window of 512K is not very
> > large, since 40 ms is not a lot to cover the delays added by a few
routers
> > or bridges and the scheduling and other delays in the hosts at the ends
> > of a TCP connection.
> >
> > You almost certainly don't want a tiny 8K window on a 100 Mbit/sec or
> > faster network, since 8KBytes occupies a 100 Mbit/sec wire for only
> > 640 microseconds.
> >
> > There are networks that are fast that 512K might be reasonable even
> > for an initial congestion window.  For example, what is the right
> > initial congestion window for a 10G Ethernet?
>
> That might depend on whether the connection is local or not. Local
> connections (same subnet as an attached interface) ignore the congestion
> window in some TCP implementations.
>
> Joe
>



From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 19:01:53 2001
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Date: Thu, 29 Mar 2001 16:51:53 -0700 (MST)
From: Vernon Schryver <vjs@calcite.rhyolite.com>
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To: David@StGermain.com, tcp-impl@grc.nasa.gov
Subject: Re: initial window size
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> From: "David St. Germain" <David@StGermain.com>

> Although I agree with you that this is how one would tweak the window size,
> the OS still has to default to something.  With Windows NT 4.0, I know the
> default upon installation is 512 (I just recently saw that statistic), I'm
> not sure what the other OS's default to upon installation, but I believe the
> original RFC recommendation was to start with a default of 512 and use the
> slow start algorithm to ramp up.

Please consider the difference between initial congestion window size
and window size.  One of the biggest differences is that one is used by
the sender and the other is used by the receiver.  Please also consider
RFC 2414.  In other words, I doubt that even NT 4.0 had a default window
size of 512, although I wouldn't be surprised if NT 4.0 used a initial
congestion window of 1 segment.

Isn't the NT 4.0 initial window size controled by the 
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Tcpip\Parameters\TcpWindowSize 
value?
http://www.winguides.com/registry/display.php/268/ claims the default
is 8760, which is about what I vaguely recall seeing in the registries of
a few Windows boxes.  My recollections put the parameter in a slightly
different registry "hive," "path," or whatever, but I'm probably wrong.

Stevens' "TCP/IP Illustrated Volume 1" has a good description of what
might be meant by "use the slow start algorithm to ramp up," athough it
is to instead of from the configured or default TCP window size.


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 19:40:59 2001
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From: "Oleg Vishnepolsky" <oleg@inforocket.com>
To: "Vernon Schryver" <vjs@calcite.rhyolite.com>, <tcp-impl@grc.nasa.gov>
Subject: RE: initial window size
Date: Thu, 29 Mar 2001 19:26:14 -0500
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The stack has to determine the initial window size, and all it has is the address of the other host, and MSS 
derived from MTU.
Practically speaking, most stacks use a hard-wired default (32k or 8k or whatever). They could a better
job taking MSS into account. 
To your other point, MTU definitely  has to do with the speed and memory on the adaptor. That's why
you will see higher MTUs with higher speeds. 
As far as 6-8 times the MTU being from mid 80s is not correct either. In fact, BSD 4.3 Reno had 2k windows
for local communications, which is even less than that ratio.
6-8 figure is based on my own experiments and measurements done in the early 90s.
-----Original Message-----
From: owner-tcp-impl@lerc.nasa.gov
[mailto:owner-tcp-impl@lerc.nasa.gov]On Behalf Of Vernon Schryver
Sent: Thursday, March 29, 2001 6:27 PM
To: tcp-impl@grc.nasa.gov
Subject: RE: initial window size


> From: "Oleg Vishnepolsky" <oleg@inforocket.com>

> Since TCP stack does not have direct knowledge of the speed

As Joe Touch implied, that may be often true, but it is not always true.

>                                  the MTU gives a good starting point.

I don't see what the MTU has to do with link speed.
Yes, I know by heart a bunch of MTU's for various links, but ...

> A while back the reasonable ratio was for the window to be about 6-8 times the MTU size for the
> local connections (slow start algorithm for non-local).
> Is it still the case these days ? 

I doubt it was ever true that 6-8 times the interface MTU was the right
window size.  TCP stacks I've touched have not used anything that tiny
since the mid-1980's, when I jacked up one UNIX vendor's default windows
size to 30K, and that was partly because 32K or more hit sign-bit bugs in
some other vendors' systems.  At the time, my UNIX employer had real support
for only 10 MHz Ethernet.

Disclaimer: my definition of "right" has long been tainted by "goes
fastest according to various benchmarks."


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 20:10:28 2001
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From: Vernon Schryver <vjs@calcite.rhyolite.com>
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To: tcp-impl@grc.nasa.gov
Subject: RE: initial window size
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> From: "Oleg Vishnepolsky" <oleg@inforocket.com>

> The stack has to determine the initial window size, and all it
> has is the address of the other host, and MSS derived from MTU.

On the contrary, "the stack" is just code.  BSD-style code could
follow the if pointers to find the interface speed as easily
as it now finds the interface MTU to generate the MSS option.

> Practically speaking, most stacks use a hard-wired default (32k
> or 8k or whatever). They could a better job taking MSS into account. 

agreed.

> To your other point, MTU definitely  has to do with the speed
> and memory on the adaptor. That's why
> you will see higher MTUs with higher speeds. 

It is true that as the years have passed, media have become faster
as MTU's generally have grown longer.  However, Ethernet is a rather
important counter example.


> As far as 6-8 times the MTU being from mid 80s is not correct either.
> In fact, BSD 4.3 Reno had 2k windows
> for local communications, which is even less than that ratio.
> 6-8 figure is based on my own experiments and measurements done
> in the early 90s.

So in about 1986 I did not change the default TCP window shipped by one of
the major commercial UNIX vendors to 30K?  And in about 1988 I didn't
change it to 60K?  That's interesting, because I didn't think I'd become
so senile that I'd invent such recollections.  (Note that I don't mean
that I changed it from the default that someone at the factory set.  I
mean that according to those false recollections of mine, I was that
someone at the factory.)

Yes, as a reasult of my enthusiasm for better`ttcp` numbers to give to my
employer's sales people (and my own bosses), I also saw some bug reports
of interoperability problems.  Some were related to bugs in other vendors'
systems uncovered by large mismatches in window sizes.


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 22:33:40 2001
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wow!  that was a nice discussion. I got cleared up with many of my doubts.
But  whatever discussion which went about this is abou the Congestion window
(cwnd). But my basic doubt is the selection of  the send window ( snd.wnd) or
in other terms it is the peer's receive window which is advertised to the
sender . Selecting this receive window might be implementation dependent , but
I like to know what will be criteria and what all parameters affect this
selection. I hope atleast this time Iam clear ..

Thanks
-arun

Vernon Schryver wrote:

> > From: "Oleg Vishnepolsky" <oleg@inforocket.com>
>
> > The stack has to determine the initial window size, and all it
> > has is the address of the other host, and MSS derived from MTU.
>
> On the contrary, "the stack" is just code.  BSD-style code could
> follow the if pointers to find the interface speed as easily
> as it now finds the interface MTU to generate the MSS option.
>
> > Practically speaking, most stacks use a hard-wired default (32k
> > or 8k or whatever). They could a better job taking MSS into account.
>
> agreed.
>
> > To your other point, MTU definitely  has to do with the speed
> > and memory on the adaptor. That's why
> > you will see higher MTUs with higher speeds.
>
> It is true that as the years have passed, media have become faster
> as MTU's generally have grown longer.  However, Ethernet is a rather
> important counter example.
>
> > As far as 6-8 times the MTU being from mid 80s is not correct either.
> > In fact, BSD 4.3 Reno had 2k windows
> > for local communications, which is even less than that ratio.
> > 6-8 figure is based on my own experiments and measurements done
> > in the early 90s.
>
> So in about 1986 I did not change the default TCP window shipped by one of
> the major commercial UNIX vendors to 30K?  And in about 1988 I didn't
> change it to 60K?  That's interesting, because I didn't think I'd become
> so senile that I'd invent such recollections.  (Note that I don't mean
> that I changed it from the default that someone at the factory set.  I
> mean that according to those false recollections of mine, I was that
> someone at the factory.)
>
> Yes, as a reasult of my enthusiasm for better`ttcp` numbers to give to my
> employer's sales people (and my own bosses), I also saw some bug reports
> of interoperability problems.  Some were related to bugs in other vendors'
> systems uncovered by large mismatches in window sizes.
>
> Vernon Schryver    vjs@rhyolite.com



From owner-tcp-impl@lerc.nasa.gov  Thu Mar 29 22:56:39 2001
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> From: arun <arun@netlab.hcltech.com>

> wow!  that was a nice discussion. I got cleared up with many of my doubts.
> But  whatever discussion which went about this is abou the Congestion window
> (cwnd). But my basic doubt is the selection of  the send window ( snd.wnd) or
> in other terms it is the peer's receive window which is advertised to the
> sender . Selecting this receive window might be implementation dependent , but
> I like to know what will be criteria and what all parameters affect this
> selection. I hope atleast this time Iam clear ..

"All of the parameters" covering the initial send or congestion
window cover a lot of ground.  Didn't someone else point out RFC 2414?
What about RFC 2416?  RFC 2001 says "initialization for a given
connection sets cwnd to one segment."

If they're not clear, then about what looking for the string snd_cwnd
in the files netinet/tcp_output.c and netinet/tcp_input.c in the
nearest stash of BSD kernel source?  The NetBSD and FreeBSD source
can be found on the net, at URL's like
http://www.FreeBSD.org/cgi/cvsweb.cgi/src/sys/netinet/


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@lerc.nasa.gov  Fri Mar 30 09:01:48 2001
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On Thu, 29 Mar 2001, Mark Allman wrote:

> See RFC 2581: 
> 
>     The initial window should be 1 or 2 segments (each up to an
>     MSS).  (This document also provides the spec. for the congestion
>     control algorithms used to increase the window size).
> 
> See RFC 2414:
> 
>     The initial window may be 3 or 4 segments, depending on the size
>     of the segments (basically as long as the 3 or 4 segments do not
>     exceed 4380 bytes).
> 
> 2581 is a proposed standard; 2414 is experimental.


Am I right in thinking that we can blame the existence of this on:
a. the MTU of edge system (good old ethernet) not scaling upward?
b. TCP's back-to-back busrtiness behaviour?

Fix either one of those, and an initial window of more than one
segment looks less compelling.

L.

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



From owner-tcp-impl@lerc.nasa.gov  Fri Mar 30 09:32:20 2001
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Subject: Re: initial window size 
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> Am I right in thinking that we can blame the existence of this on:
> a. the MTU of edge system (good old ethernet) not scaling upward?
> b. TCP's back-to-back busrtiness behaviour?
> 
> Fix either one of those, and an initial window of more than one
> segment looks less compelling.

I do not follow.  See RFC 2414 for (what I believe are) a number of
good reasons to use an initial cwnd of more than 1 segment.

allman


From owner-tcp-impl@lerc.nasa.gov  Fri Mar 30 09:32:43 2001
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A couple of comments....


Oleg Vishnapolsky said:
>
> Practically speaking, most stacks use a hard-wired default (32k or
> 8k or whatever).

Yep.  In a study of our web server traffic last year we found that
the median advertised window size was roughly 8KB and the average
was about 18KB in web clients.  These numbers were essentially
invariant over the 18 months of data we examined (I really should
crunch the data since the paper, but haven't found the cycles yet).
Further, for transfers that sent enough data to open their cwnd to
the advertised window we found that 70% were likely performance
limited by the advertised window.  So, I agree with Vernon -- crank
those default socket buffers up (or, use something like autotuned
socket buffers).  More details in:

    Mark Allman. A Web Server's View of the Transport Layer. ACM
    Computer Communication Review, 30(5), October 2000. 
    http://roland.grc.nasa.gov/~mallman/papers/webobs-ccr.ps


Vernon Schryver said:
>
> RFC 2001 says "initialization for a given connection sets cwnd to
> one segment."

RFC 2001 is obsoleted by RFC 2581 -- which allows an initial
congestion window of up to 2 segments (up to MSS bytes in each).

Sally Floyd and Jitu Padhye have been using tbit to determine what
initial cwnd web servers are really using.  You can see their
results at: http://www.aciri.org/tbit/

allman


---
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/


From owner-tcp-impl@lerc.nasa.gov  Fri Mar 30 09:58:58 2001
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From: Lloyd Wood <l.wood@eim.surrey.ac.uk>
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On Fri, 30 Mar 2001, Mark Allman wrote:

> > Am I right in thinking that we can blame the existence of this on:
> > a. the MTU of edge system (good old ethernet) not scaling upward?
> > b. TCP's back-to-back busrtiness behaviour?
> > 
> > Fix either one of those, and an initial window of more than one
> > segment looks less compelling.
> 
> I do not follow.  See RFC 2414 for (what I believe are) a number of
> good reasons to use an initial cwnd of more than 1 segment.

(section 3)

Those reasons are related to the (legacy, dammit!) TCP implementation.

1. is, I think, basically a problem with delayed-ack implementations,
which should ack the first packet immediately before going into delay.
(I'd guess that paced TCP should make the effects of delayed acks less
desirable; acking the first packet strikes me as a good idea
regardless.)

2. is made moot by a larger MTU on edge systems leading to larger path
MTU throughout.

3. as for 1., and the effect on slow-start should be ameliorated by 
byte-counting.

Sections 4 and 5 list the disadvantages of larger initial cwnd - it's
harder to encounter unnecessary retransmits when it's only 1 segment.

L.

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





From owner-tcp-impl@lerc.nasa.gov  Fri Mar 30 10:18:47 2001
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Date: Fri, 30 Mar 2001 08:05:38 -0700 (MST)
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> From: Lloyd Wood <l.wood@eim.surrey.ac.uk>

> ...
> Am I right in thinking that we can blame the existence of this on:
> a. the MTU of edge system (good old ethernet) not scaling upward?
> b. TCP's back-to-back busrtiness behaviour?
>
> Fix either one of those, and an initial window of more than one
> segment looks less compelling.

The 1500 byte MTU of Ethernet has worse consequences than anything
mentioned in RFC 2414, but has nevertheless resisted fixing.  According
to a trade rag I saw this week, Ethernet claims 90% of the LAN market,
which sounds low to me.

The consequences I mean can be seen with your favorite TCP benchmark, a
pair of computers, and a LAN medium that is faster than the computers and
has an MTU of at least 4500.  (By "faster" I mean that the computers cannot
saturate it with a single TCP stream.)  Vary the MTU used by the computers
from 1000 through the medium's maximum while measuring single stream TCP
through-put.  Notice that the local peaks ans valleys that correspond to
various things in your computers, but also notice that throughput increases
linearly or faster with MTU.

Or equivalently, somehow measure total CPU cycles/byte on a fast
pair of computers.

Or look at router performance advertisments in the trade rags and
notice that the honest ones are in packets/second.

In other words, the 1500 byte Ethernet MTU makes bridging among
10, 100, and 100 MHz Ethernet easy, but it does terrible things
to performance.


Vernon Schryver    vjs@rhyolite.com


From owner-tcp-impl@lerc.nasa.gov  Fri Mar 30 10:21:20 2001
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Subject: Re: initial window size 
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OK.  I am getting your points, now...

> 1. is, I think, basically a problem with delayed-ack
> implementations, which should ack the first packet immediately
> before going into delay.  (I'd guess that paced TCP should make
> the effects of delayed acks less desirable; acking the first
> packet strikes me as a good idea regardless.)

So, you're trading one hack for another.  Fine.

> 2. is made moot by a larger MTU on edge systems leading to larger
> path MTU throughout.

I'll buy that.

> 3. as for 1., and the effect on slow-start should be ameliorated
> by byte-counting.

But, byte counting+larger initial window is better yet.  They are
orthogonal, I think.

One more note that was not true when 2414 was written...  With
limited transmit, using an initial window of more than one segment
makes loss recovery more robust and less reliant on the RTO.

allman


---
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/


From owner-tcp-impl@lerc.nasa.gov  Sat Mar 31 02:41:46 2001
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From: "Moncef Boualem" <moncef2@hotmail.com>
To: tcp-impl@lerc.nasa.gov
Subject: LOST CONNECTION
Date: Sat, 31 Mar 2001 02:15:01 -0500
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Hi folks,
An ftp transfer between an NT client and a Sun Solaris server
results in a sudden "LOST CONNECTION" message on the server side.
It all happens after the NT machine initiates the FTP connection and
starts uploading PDF files into the server. Then after several successful 
transfers the connection gets cut. The link between the
two machines is: ethernet-t1-ethernet. Is there anything to
tune up in the TCP stack? thanks.
_________________________________________________________________
Get your FREE download of MSN Explorer at http://explorer.msn.com



From owner-tcp-impl@lerc.nasa.gov  Sat Mar 31 20:33:28 2001
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Message-ID: <001401c0ba48$9e296ae0$babbf018@hal>
From: "Hal F. Gottfried" <hal@gottfriedweb.com>
To: "Moncef Boualem" <moncef2@hotmail.com>, <tcp-impl@lerc.nasa.gov>
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Subject: Re: LOST CONNECTION
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Does this only happen with PDF files or with other files as well ? Can you
see what is happening on the Sun machine  ? What version of Solaris is it
running ?

Hal F. Gottfried
ProTech Training and Technical Services

----- Original Message -----
From: "Moncef Boualem" <moncef2@hotmail.com>
To: <tcp-impl@lerc.nasa.gov>
Sent: Saturday, March 31, 2001 2:15 AM
Subject: LOST CONNECTION


> Hi folks,
> An ftp transfer between an NT client and a Sun Solaris server
> results in a sudden "LOST CONNECTION" message on the server side.
> It all happens after the NT machine initiates the FTP connection and
> starts uploading PDF files into the server. Then after several successful
> transfers the connection gets cut. The link between the
> two machines is: ethernet-t1-ethernet. Is there anything to
> tune up in the TCP stack? thanks.
> _________________________________________________________________
> Get your FREE download of MSN Explorer at http://explorer.msn.com
>
>



