From owner-tcp-impl@lerc.nasa.gov  Mon May  1 16:53:48 2000
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From: Sally Floyd <floyd@aciri.org>
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>	Would anyone be able to give me a good feel for what
>direction TCP/IP will take with respect to congestion
>control?

You could look at my viewgraphs from a talk last year on
"On the Evolution of End-to-end Congestion Control in the Internet: An
idiosyncratic View", from October 1999, or an earlier talk on
"On the Evolution of End-to-end Congestion Control in the Internet"
from April 1999.  The viewgraphs are pointed-to on my web page at
"http://www.aciri.org/floyd/talks.html".

The section on pages 18-27, from the October talk, mentions briefly
some of the changes in progress, such as ECN, and the section
beginning on page 28 talks about some of the different views for
the long-term evolution of end-to-end congestion control.
(This is elaborated on in the viewgraphs from the April talk.)

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


From owner-tcp-impl@lerc.nasa.gov  Wed May  3 15:35:57 2000
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From owner-tcp-impl@lerc.nasa.gov  Thu May 11 09:53:10 2000
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the TCP Implementation Working Group of the IETF.

	Title		: TCP Problems with Path MTU Discovery
	Author(s)	: K. Lahey
	Filename	: draft-ietf-tcpimpl-pmtud-03.txt
	Pages		: 16
	Date		: 10-May-00
	
This memo catalogs several known TCP implementation problems dealing
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hole problem, stretch ACKs due to confusion between MSS and segment
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From owner-tcp-impl@lerc.nasa.gov  Mon May 15 12:28:39 2000
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We are starting a WG last call on the I-D "TCP Problems with Path
MTU Discovery" (draft-ietf-tcpimpl-pmtud-03.txt).  We plan to submit
the draft to the IESG as a candidate for an Informational RFC.
Please send comments on the draft to the list or the author before
5/31.

Thanks,
allman


From owner-tcp-impl@lerc.nasa.gov  Mon May 22 14:16:29 2000
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From: "Ramesh Shankar" <RSHANKAR@novell.com>
To: <tcp-impl@grc.nasa.gov>
Subject: Slow start question ...
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If the peer to which a TCP connection has been established is on a "direct connected" network (i.e. the route doesn't point to a gateway), why exactly is slow start required? Linux and NetBSD seem to do slow start for this case while FreeBSD doesn't seem to (I haven't picked through the source to verify this though). Since slow start is meant to "probe the available network bandwidth" of a route (involving routers), it seems to me that I really shouldn't have to use slow start for the case mentioned above. 

One of the e-mails in this list mentioned a case where the outbound interface queue was overwhelmed by a burst of packets when slow start was not used. Slow start seems to be a wrong way to solve this problem. This problem could happen even when too many connections are opened, and hence is a fundamental inter-layer flow control problem.

Any insight would be greatly appreciated.

Thanks,

S.R.



From owner-tcp-impl@lerc.nasa.gov  Mon May 22 15:19:15 2000
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Subject: Re: Slow start question ...
To: RSHANKAR@novell.com (Ramesh Shankar)
Date: Mon, 22 May 2000 17:56:30 +0100 (BST)
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> If the peer to which a TCP connection has been established is on a "direct connected" network (i.e. the route doesn't point to a gateway), why exactly is slow start required? Linux and NetBSD seem to do slow start for this case while FreeBSD doesn't seem to (I haven't picked through the source to verify this though). Since slow start is meant to "probe the available network bandwidth" of a route (involving routers), it seems to me that I really shouldn't have to use slow start for the case mentioned above. 

Your definition of direct connected includes things like ATM networks, slow
dialup links, and even IPIP tunnels across the internet.

Are you telling me you dont want slow start on these ?

The fundamental flaw in your argument is to assume the routers are at IP level.
There are resource issues both on all networks including your own buffering
on modems, switches, bridges and other devices below the IP layer. The
slow start procedure makes us behave well to them as well.

Alan



From owner-tcp-impl@lerc.nasa.gov  Mon May 22 15:28:56 2000
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Subject: Re: Slow start question ...
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From: Jamshid Mahdavi <mahdavi@novell.com>
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In-Reply-To: "Ramesh Shankar"'s message of "Mon, 22 May 2000 08:50:48 -0600"
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There is also no way of knowing what the layer 2 topology is.  For
example, you could be sending out of a 100 Mb/s interface, but the
receiver might only be on a 10 Mb/s interface.  These days, pretty
much anything can be bridged together; wireless, DSL, ATM, and more
could all be quietly masquerading as part of your "local ethernet".

Since there can be congestion on any of these transitions, it is still
necessary to do slowstart and congestion avoidance to hosts on the
local network.

--J


"Ramesh Shankar" <RSHANKAR@novell.com> writes:

> If the peer to which a TCP connection has been established is on a
> "direct connected" network (i.e. the route doesn't point to a
> gateway), why exactly is slow start required? Linux and NetBSD seem
> to do slow start for this case while FreeBSD doesn't seem to (I
> haven't picked through the source to verify this though). Since slow
> start is meant to "probe the available network bandwidth" of a route
> (involving routers), it seems to me that I really shouldn't have to
> use slow start for the case mentioned above.
> 
> One of the e-mails in this list mentioned a case where the outbound
> interface queue was overwhelmed by a burst of packets when slow
> start was not used. Slow start seems to be a wrong way to solve this
> problem. This problem could happen even when too many connections
> are opened, and hence is a fundamental inter-layer flow control
> problem.
> 
> Any insight would be greatly appreciated.
> 
> Thanks,
> 
> S.R.


From owner-tcp-impl@lerc.nasa.gov  Mon May 22 19:16:27 2000
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To: Alan Cox <alan@lxorguk.ukuu.org.uk>
Subject: Re: Slow start question ...
Cc: RSHANKAR@novell.com (Ramesh Shankar), tcp-impl@grc.nasa.gov
Date: Mon, 22 May 2000 13:08:57 PDT
From: "Howard Berkey" <howard@be.com>
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Seems like a moot point anyway, since in the case of sending to someone 
on the same LAN segment as yourself (i.e. not routed), slow start will 
be exited quickly if the amount of data being transferred on the 
connection exceeds a couple mtu's in size.

Howard


>> If the peer to which a TCP connection has been established is on a 
"direct connected" network (i.e. the route doesn't point to a gateway), 
why exactly is slow start required? Linux and NetBSD seem to do slow 
start for this case while FreeBSD doesn't seem to (I haven't picked 
through the source to verify this though). Since slow start is meant to 
"probe the available network bandwidth" of a route (involving routers), 
it seems to me that I really shouldn't have to use slow start for the 
case mentioned above. 
>
>Your definition of direct connected includes things like ATM networks, 
slow
>dialup links, and even IPIP tunnels across the internet.
>
>Are you telling me you dont want slow start on these ?
>
>The fundamental flaw in your argument is to assume the routers are at 
IP level.
>There are resource issues both on all networks including your own 
buffering
>on modems, switches, bridges and other devices below the IP layer. The
>slow start procedure makes us behave well to them as well.
>
>Alan
>
>


From owner-tcp-impl@lerc.nasa.gov  Mon May 22 23:17:58 2000
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From: "Ramesh Shankar" <RSHANKAR@novell.com>
To: <tcp-impl@grc.nasa.gov>
Subject: Any successor to the sockets APIs ???
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I am curious to know whether there is any work in progress on a successor to the BSD (or for that matter Winsock) sockets API. The following are the reasons for the question:

- The BSD sockets API does not have a really powerful interface for use with event driven programming. select() and poll() are available, but the performance issues associated with it have been known for sometime now. Winsock has some interesting APIs which seem to help somewhat better than BSD sockets APIs.

- I have been reading about the performance issues with persistent HTTP connection, pipelined requests and how the responses have to be sequential as there is no clear way to demarcate invidual TCP sessions etc (Work by Jeff Mogul and Venkata Padmanabhan). Is there a move towards some kind of APIs like that? (I haven't yet finished the Ph.D. thesis though to understand more details on the nature of APIs).

Any pointers would be greatly appreciated.

S.R.



From owner-tcp-impl@lerc.nasa.gov  Mon May 22 23:32:57 2000
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From: "Ramesh Shankar" <RSHANKAR@novell.com>
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Subject: What is so special about 2*MSL?
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It is not very clear to me what is the purpose of selecting 2*MSL to be the duration a connection which has performed active close has to be in the TIME_WAIT state instead of MSL.

TIME_WAIT serves two different purposes:

1. Rejection old duplicates of a connection from previous incarnation(s).
2. Retransmission of last ACK if ACK sent during connection close got lost. This ensures reliable connection close.

If we were to consider 1, then it is quite clear that waiting for 1 MSL is more than enough for old duplicates to die off.

Now, 2 really depends on the RTT of the connection. If one imagines that it takes 1MSL (i.e. extreme case) for our ACK to reach peer, then in the case our ACK does get lost, by the time the peer retransmits the FIN, we would have anyway exited the TIME_WAIT state. RFC 1185, Appendix A has some insight into the details of the choice, but doesn't go far enough to tell why 2MSL and why not 1MSL or 1.5 MSL etc.

I am pretty sure that there is some historical reason also associated with the choice of this specific value. Some of the references of Appendix A of RFC 1185 are probably very hard to get now (Any idea how I can get them)?

May be the answer is simple like: Be paranoid and wait for more than 1 MSL and it is easy to implement 2 * MSL in programming rather than 1.75 * MSL :-))).

Thanks in advance,

S.R.



From owner-tcp-impl@lerc.nasa.gov  Tue May 23 03:39:18 2000
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Date: Mon, 22 May 2000 21:59:18 -0700
From: Jan Koum <jkb@ethereal.net>
To: Jamshid Mahdavi <mahdavi@novell.com>
Cc: Ramesh Shankar <RSHANKAR@novell.com>, tcp-impl@grc.nasa.gov,
        Matthew Dillon <dillon@apollo.backplane.com>, jayanth@freebsd.org
Subject: Re: Slow start question ...
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looking at freebsd, it seem that slowstart *is* disabled for local
network:

$ sysctl -a|grep slowstart
net.inet.tcp.slowstart_flightsize: 1
net.inet.tcp.local_slowstart_flightsize: 65535

this is probably correct for 95% of the time, since having bridged
wireless and atm are not very common. but you are right, to be 100%
correct, freebsd should probably also have local slowstart enabled.

-- yan

On Mon, May 22, 2000 at 10:15:34AM -0700, Jamshid Mahdavi <mahdavi@novell.com> wrote:
> 
> There is also no way of knowing what the layer 2 topology is.  For
> example, you could be sending out of a 100 Mb/s interface, but the
> receiver might only be on a 10 Mb/s interface.  These days, pretty
> much anything can be bridged together; wireless, DSL, ATM, and more
> could all be quietly masquerading as part of your "local ethernet".
> 
> Since there can be congestion on any of these transitions, it is still
> necessary to do slowstart and congestion avoidance to hosts on the
> local network.
> 
> --J
> 
> 
> "Ramesh Shankar" <RSHANKAR@novell.com> writes:
> 
> > If the peer to which a TCP connection has been established is on a
> > "direct connected" network (i.e. the route doesn't point to a
> > gateway), why exactly is slow start required? Linux and NetBSD seem
> > to do slow start for this case while FreeBSD doesn't seem to (I
> > haven't picked through the source to verify this though). Since slow
> > start is meant to "probe the available network bandwidth" of a route
> > (involving routers), it seems to me that I really shouldn't have to
> > use slow start for the case mentioned above.
> > 
> > One of the e-mails in this list mentioned a case where the outbound
> > interface queue was overwhelmed by a burst of packets when slow
> > start was not used. Slow start seems to be a wrong way to solve this
> > problem. This problem could happen even when too many connections
> > are opened, and hence is a fundamental inter-layer flow control
> > problem.
> > 
> > Any insight would be greatly appreciated.
> > 
> > Thanks,
> > 
> > S.R.


From owner-tcp-impl@lerc.nasa.gov  Tue May 23 07:31:26 2000
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From: Andi Kleen <ak@muc.de>
To: Ramesh Shankar <RSHANKAR@novell.com>
Cc: tcp-impl@grc.nasa.gov
Subject: Re: Any successor to the sockets APIs ???
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On Tue, May 23, 2000 at 05:25:56AM +0200, Ramesh Shankar wrote:
> I am curious to know whether there is any work in progress on a successor to the BSD (or for that matter Winsock) sockets API. The following are the reasons for the question:
> 
> - The BSD sockets API does not have a really powerful interface for use with event driven programming. select() and poll() are available, but the performance issues associated with it have been known for sometime now. Winsock has some interesting APIs which seem to help somewhat better than BSD sockets APIs.

Single Unix defines a replacement: real time signals that carry fd event
notifications. They unfortunately require a non standard extension to set
the SIGIO signal for a fd to be really useful (or another non standard
extension for queued SIGIO). At least in modern Linux it is supported.
poll() is still needed as fallback when signals get lost.
From first tests queued SIGIO with a sigwaitinfo() main loop scales
very nicely for web servers.

-Andi


From owner-tcp-impl@lerc.nasa.gov  Tue May 23 12:13:24 2000
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Subject: Re: Any successor to the sockets APIs ???
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> - The BSD sockets API does not have a really powerful interface for use with event driven programming. select() and poll() are available, but the performance issues associated with it have been known for sometime now. Winsock has some interesting APIs which seem to help somewhat better than BSD sockets APIs.

Linux supports event based real time signalling (kind of vaguely akin to the
VMS AST setup).  There are plenty of people who find poll/select scale nicely.
Under load your syscalls/task count drops which materially improves your
throughput. Thus as the load goes up degradation is good.

Point a big internet client simulation (not crap like specweb) at a threaded
or multiprocess server like apache and then try it with thttpd. Its not the
simple set of equations some people think.

[Apache btw suffers, thttpd just keeps going. apache is faster for small loads]

Alan




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Subject: Re: Slow start question ...
To: howard@be.com
Date: Tue, 23 May 2000 14:26:07 +0100 (BST)
Cc: alan@lxorguk.ukuu.org.uk (Alan Cox), RSHANKAR@novell.com (Ramesh Shankar),
        tcp-impl@grc.nasa.gov
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> Seems like a moot point anyway, since in the case of sending to someone 
> on the same LAN segment as yourself (i.e. not routed), slow start will 
> be exited quickly if the amount of data being transferred on the 
> connection exceeds a couple mtu's in size.

The difference is in that tiny time period starting up if there is congestion
we don't swamp the lan. But yes the only reason to disable slow start is
benchmarketing. For real world use it should be lost in the noise




From owner-tcp-impl@lerc.nasa.gov  Tue May 23 13:24:23 2000
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From: Chuck Lever <cel@monkey.org>
To: tcp-impl@grc.nasa.gov
Cc: RSHANKAR@novell.com
Subject: Re: Any successor to the sockets APIs ???
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hi ramesh & al. -

our research shows that providing a more efficient poll() interface scales
as well as the new RT signals.  niels provos implemented /dev/poll in
Linux (see our Freenix track paper for Usenix 2000).  we benchmarked
thttpd using /dev/poll against phhttpd, an experimental web server that
uses Posix RT signals.

RT signals are still a young programming model, so there are issues with
using them:  portability among operating systems, how to share them among
libraries, lack of support for sharing signal queues among threads, and so
on.  of great concern to us is the use of poll() for handling an RT signal
queue overflow: poll() must be much more efficient to be an effective
overflow remedy.

at this point, i don't think we understand how to use RT signals most
effectively.  but poll is well-understood, so it's easy to change
applications to use /dev/poll and see a significant scalability
improvement.

	- Chuck Lever
--
corporate:	<chuckl@netscape.com>
personal:	<chucklever@bigfoot.com>

The Linux Scalability project:
	http://www.citi.umich.edu/projects/linux-scalability/

Date: Tue, 23 May 2000 10:28:31 +0200
From: Andi Kleen <ak@muc.de>
To: Ramesh Shankar <RSHANKAR@novell.com>
Cc: tcp-impl@grc.nasa.gov
Subject: Re: Any successor to the sockets APIs ???

On Tue, May 23, 2000 at 05:25:56AM +0200, Ramesh Shankar wrote:
> I am curious to know whether there is any work in progress on a successor to the BSD (or for that matter Winsock) sockets API. The following are the reasons for the question:
> 
> - The BSD sockets API does not have a really powerful interface for use with event driven programming. select() and poll() are available, but the performance issues associated with it have been known for sometime now. Winsock has some interesting APIs which seem to help somewhat better than BSD sockets APIs.

Single Unix defines a replacement: real time signals that carry fd event
notifications. They unfortunately require a non standard extension to set
the SIGIO signal for a fd to be really useful (or another non standard
extension for queued SIGIO). At least in modern Linux it is supported.
poll() is still needed as fallback when signals get lost.
>From first tests queued SIGIO with a sigwaitinfo() main loop scales
very nicely for web servers.

- -Andi



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To: Jan Koum <jkb@ethereal.net>
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Subject: Re: Slow start question ...
References: <s928f4f2.050@prv-mail20.provo.novell.com> <yu8xya52bj61.fsf@sabre.sjf.novell.com> <20000522215918.D39617@ethereal.net>
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Which version are you running?  

In FreeBSD 2.2.8, there's a sysctl option 
net.inet.tcp.no_local_slowstart.  It's default value is 0, I
believe, which forces slow start on the local network.  (This option
wasn't present before 2.2.8.)

-Michele

Jan Koum wrote:
> 
> looking at freebsd, it seem that slowstart *is* disabled for local
> network:
> 
> $ sysctl -a|grep slowstart
> net.inet.tcp.slowstart_flightsize: 1
> net.inet.tcp.local_slowstart_flightsize: 65535
> 
> this is probably correct for 95% of the time, since having bridged
> wireless and atm are not very common. but you are right, to be 100%
> correct, freebsd should probably also have local slowstart enabled.
> 
> -- yan
> 
> On Mon, May 22, 2000 at 10:15:34AM -0700, Jamshid Mahdavi <mahdavi@novell.com> wrote:
> >
> > There is also no way of knowing what the layer 2 topology is.  For
> > example, you could be sending out of a 100 Mb/s interface, but the
> > receiver might only be on a 10 Mb/s interface.  These days, pretty
> > much anything can be bridged together; wireless, DSL, ATM, and more
> > could all be quietly masquerading as part of your "local ethernet".
> >
> > Since there can be congestion on any of these transitions, it is still
> > necessary to do slowstart and congestion avoidance to hosts on the
> > local network.
> >
> > --J
> >
> >
> > "Ramesh Shankar" <RSHANKAR@novell.com> writes:
> >
> > > If the peer to which a TCP connection has been established is on a
> > > "direct connected" network (i.e. the route doesn't point to a
> > > gateway), why exactly is slow start required? Linux and NetBSD seem
> > > to do slow start for this case while FreeBSD doesn't seem to (I
> > > haven't picked through the source to verify this though). Since slow
> > > start is meant to "probe the available network bandwidth" of a route
> > > (involving routers), it seems to me that I really shouldn't have to
> > > use slow start for the case mentioned above.
> > >
> > > One of the e-mails in this list mentioned a case where the outbound
> > > interface queue was overwhelmed by a burst of packets when slow
> > > start was not used. Slow start seems to be a wrong way to solve this
> > > problem. This problem could happen even when too many connections
> > > are opened, and hence is a fundamental inter-layer flow control
> > > problem.
> > >
> > > Any insight would be greatly appreciated.
> > >
> > > Thanks,
> > >
> > > S.R.


From owner-tcp-impl@lerc.nasa.gov  Tue May 23 15:27:50 2000
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Subject: Re: Slow start question ...
In-Reply-To: <392AB240.6EF60106@cs.unc.edu> from Michele Clark at "May 23, 2000
 12:30:56 pm"
To: clark@cs.unc.edu
Date: Tue, 23 May 2000 18:57:24 +0200 (CEST)
CC: Jan Koum <jkb@ethereal.net>, tcp-impl@grc.nasa.gov
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> Which version are you running?  
> 
> In FreeBSD 2.2.8, there's a sysctl option 
> net.inet.tcp.no_local_slowstart.  It's default value is 0, I
> believe, which forces slow start on the local network.  (This option
> wasn't present before 2.2.8.)

actually the default is as before so it does not do slow_start
on the local network (i.e. blasts a whole window of data at startup).
I introduced this because experiments with dummynet (where you
usally end up using local machines) would give unrealistic
results otherwise.

	cheers
	luigi

> -Michele
> 
> Jan Koum wrote:
> > 
> > looking at freebsd, it seem that slowstart *is* disabled for local
> > network:
-----------------------------------+-------------------------------------
  Luigi RIZZO, luigi@iet.unipi.it  . Dip. di Ing. dell'Informazione
  http://www.iet.unipi.it/~luigi/  . Universita` di Pisa
  TEL/FAX: +39-050-568.533/522     . via Diotisalvi 2, 56126 PISA (Italy)
  Mobile   +39-347-0373137
-----------------------------------+-------------------------------------


From owner-tcp-impl@lerc.nasa.gov  Tue May 23 15:58:40 2000
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To: Alan Cox <alan@lxorguk.ukuu.org.uk>
Subject: Re: Any successor to the sockets APIs ???
Cc: RSHANKAR@novell.com (Ramesh Shankar), tcp-impl@grc.nasa.gov
Date: Tue, 23 May 2000 10:42:38 PDT
From: "Howard Berkey" <howard@be.com>
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>> - The BSD sockets API does not have a really powerful interface for 
use with event driven programming. select() and poll() are available, 
but the performance issues associated with it have been known for 
sometime now. Winsock has some interesting APIs which seem to help 
somewhat better than BSD sockets APIs.
>
>Linux supports event based real time signalling (kind of vaguely akin 
to the
>VMS AST setup).  There are plenty of people who find poll/select scale 
nicely.
>Under load your syscalls/task count drops which materially improves 
your
>throughput. Thus as the load goes up degradation is good.
>
>Point a big internet client simulation (not crap like specweb) at a 
threaded
>or multiprocess server like apache and then try it with thttpd. Its 
not the
>simple set of equations some people think.
>
>[Apache btw suffers, thttpd just keeps going. apache is faster for 
small loads]
>


Two issues here cause select and poll to appear bad.  The first is 
crappy OS implementations of select;  the second is programmers that 
don't know how to properly use select in a multithreaded environment.

Alan is spot on in that if you know what you are doing, select can be 
quite scalable, provided your OS doesn't have a brain-dead 
implementation (and BSD and Linux have pretty good ones).  If you just 
throw several hundred sockets into select() and sit on it with a single 
thread, of course it will be a poor performer.

FWIW, the select() in the networking implementation currently shipping 
in BeOS is very, very poor.  It's one of the reasons we have rewritten 
the OS networking for the next release.

It also seems to me that programming correctly to select() and poll() 
is far easier and more elegant than doing it asynchronously via SIGIO.  
That opens lots of potential cans of worms. 

Howard

 


From owner-tcp-impl@lerc.nasa.gov  Tue May 23 16:06:54 2000
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Date: Tue, 23 May 2000 10:41:58 -0700
From: Jan Koum <jkb@ethereal.net>
To: Michele Clark <clark@cs.unc.edu>
Cc: tcp-impl@grc.nasa.gov
Subject: Re: Slow start question ...
Message-ID: <20000523104158.E47375@ethereal.net>
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my output is from 4.0 -- i didn't think there was a sysctl in earlier
versions since 3.x doesn't have a sysct:


        /*      
         * Don't force slow-start on local network.
         */     
        if (!in_localaddr(inp->inp_faddr))
                tp->snd_cwnd = mss;

but 2.2.x does seem to have a sysctl:

        /*
         * Don't force slow-start on local network.
         * Make this depend on the sysctl variable below
         */
        if (!no_local_slowstart || !in_localaddr(inp->inp_faddr))
                tp->snd_cwnd = mss;

and 2.1 does not have a sysctl:

        /*
         * Don't force slow-start on local network.
         */
        if (!in_localaddr(inp->inp_faddr))
                tp->snd_cwnd = mss;


with no_local_slowstart set to 0, this evaluates to true:
  (!no_local_slowstart || !in_localaddr(inp->inp_faddr)


so i think 2.2.8 (just like the comment indicates) does not force slow
start on the local network either.

-- yan



On Tue, May 23, 2000 at 12:30:56PM -0400, Michele Clark <clark@cs.unc.edu> wrote:
> 
> Which version are you running?  
> 
> In FreeBSD 2.2.8, there's a sysctl option 
> net.inet.tcp.no_local_slowstart.  It's default value is 0, I
> believe, which forces slow start on the local network.  (This option
> wasn't present before 2.2.8.)
> 
> -Michele
> 
> Jan Koum wrote:
> > 
> > looking at freebsd, it seem that slowstart *is* disabled for local
> > network:
> > 
> > $ sysctl -a|grep slowstart
> > net.inet.tcp.slowstart_flightsize: 1
> > net.inet.tcp.local_slowstart_flightsize: 65535
> > 
> > this is probably correct for 95% of the time, since having bridged
> > wireless and atm are not very common. but you are right, to be 100%
> > correct, freebsd should probably also have local slowstart enabled.
> > 
> > -- yan
> > 
> > On Mon, May 22, 2000 at 10:15:34AM -0700, Jamshid Mahdavi <mahdavi@novell.com> wrote:
> > >
> > > There is also no way of knowing what the layer 2 topology is.  For
> > > example, you could be sending out of a 100 Mb/s interface, but the
> > > receiver might only be on a 10 Mb/s interface.  These days, pretty
> > > much anything can be bridged together; wireless, DSL, ATM, and more
> > > could all be quietly masquerading as part of your "local ethernet".
> > >
> > > Since there can be congestion on any of these transitions, it is still
> > > necessary to do slowstart and congestion avoidance to hosts on the
> > > local network.
> > >
> > > --J
> > >
> > >
> > > "Ramesh Shankar" <RSHANKAR@novell.com> writes:
> > >
> > > > If the peer to which a TCP connection has been established is on a
> > > > "direct connected" network (i.e. the route doesn't point to a
> > > > gateway), why exactly is slow start required? Linux and NetBSD seem
> > > > to do slow start for this case while FreeBSD doesn't seem to (I
> > > > haven't picked through the source to verify this though). Since slow
> > > > start is meant to "probe the available network bandwidth" of a route
> > > > (involving routers), it seems to me that I really shouldn't have to
> > > > use slow start for the case mentioned above.
> > > >
> > > > One of the e-mails in this list mentioned a case where the outbound
> > > > interface queue was overwhelmed by a burst of packets when slow
> > > > start was not used. Slow start seems to be a wrong way to solve this
> > > > problem. This problem could happen even when too many connections
> > > > are opened, and hence is a fundamental inter-layer flow control
> > > > problem.
> > > >
> > > > Any insight would be greatly appreciated.
> > > >
> > > > Thanks,
> > > >
> > > > S.R.


From owner-tcp-impl@lerc.nasa.gov  Tue May 23 19:01:50 2000
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Reply-To: <jspence@zama.net>
From: "John Spence" <jspence@zama.net>
To: <tcp-impl@grc.nasa.gov>
Subject: IPv6 Path MTU Discovery process ...
Date: Tue, 23 May 2000 13:29:09 -0700
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Hello;

I sent an eMail to Jack McGann asking a question about IPv6 Path MTU
Discovery.  I had a question about how it was described, and thought I had
an idea for a better solution.  Mr. McGann answered saying that he didn't
know if that had been considered, and suggested I distribute it in this
forum.

Here is the eMail I sent Mr. McGann - please feel free to contact me for
questions and to let me know what you think.

Best regards;

John Spence

---------------------
Hello Mr. McCann;
I have read with interest "IPv6 Clearly Explained", by Pete Loshin. I read
about the Path MTU Discovery process and I had a question. I went to the
IETF site and researched the RFC, and I have a thought.
The book, and the RFC, describe the process is described as the source node
sending the largest packet permissible on it's own network - essentially the
first hop. If any intervening link cannot handle packets that large, it
discards the packet and sends a ICMPv6 "packet too big" message to the
source. The source node drops the size of the packet to within that reported
limit, and send the packet again. Perhaps it makes the final destination,
perhaps another node requires even a smaller packet size, and generates
another "packet too big" message for the source. This could go on for
awhile.
My question is this. Why not have each node in the path that supports only a
smaller MTU than the source node send the "packet too big" message, but
still forward a "shell packet" on towards the destination, where the "shell
package" contains the minimum MTU so far along that path. Since the goal is
to send a packet all the way to the destination, that information may well
be needed, and it would reduce latency in a scenario where there was a
"stepping down" of the MTU size allowed by intervening links as the packets
approached the destination.
The source node would simply wait, collecting whatever "packet still too
big" ICMP messages it received from the intervening nodes (a node that could
accommodate the current MTU would simply forward the shell packet forward,
but would not reply to the source node), until it received a confirmation
from the true destination that the packet had finally arrives. Looking over
the ICMP "packet too big" messages the source had received, it could now
choose the best possible MTU and proceed with confidence.
Heck - we could even have the "shell packet" continue all the way to the
destination node, being adjusted downward by each link along the way (if
needed) - without generating any ICMP "packet too big" messages - and then
make the true destination node send a message back to the source saying "a
2785byte MTU is cleared all the way through", at which time the real data
transmission would begin. You would probably want the very first node that
determined it needed a smaller packet generate the ICMP "packet too big"
message for the source, just to let the source know it wasn't going to be
clear sailing. It could mark the "shell packet" it sent forward to
downstream nodes so they knew they weren't the first to demand a smaller
MTU.
What do you think? Good idea? I'll bet you guys thought of this - just
trying to help. You at the IETF are doing a great job.
----------------------------------------------------------------------------
-----
John E Spence                    ZAMA NETWORKS, INC.
Senior UNIX Systems Engineer
12101 International Boulevard, Seattle WA 98168-2569
wk 206.352.9262  NEXTEL 206.571.9792 fx 206.352.3918
jspence@zama.net || http://www.zama.net
-------------------------------------------------------------------------



From owner-tcp-impl@lerc.nasa.gov  Tue May 23 20:29:43 2000
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From: Bernie Volz <Volz@ipworks.com>
To: "'jspence@zama.net'" <jspence@zama.net>, tcp-impl@grc.nasa.gov
Subject: RE: IPv6 Path MTU Discovery process ...
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Seems like a reasonable idea ... some comments follow:

I like the idea of having the destination respond with the MTU. It's cleaner
since it avoids a lot of responses and figuring out when to stop waiting for
a responses (since the destination might be down - so you'd get no response)
if you allow potentially all intermediate nodes to response as well.

An interesting question to consider is the cost of doing this ...
intermediate routers now have to be capable of generating these "shell
packets" and end nodes (routers too) would need to understand the "shell
packets" and respond accordingly. How well do the characteristics of this
"shell packet" need to match the original packet? For example, it is always
possible that different types of traffic take different paths.

I think from IPv4 experience, while this idea would be an optimization, I
doubt that there are many MTU drops that typically occur (at least on todays
networks). The value is usually found reasonably fast (with one or two MTU
drops) [I don't have any hard data to back this up, it is just a gut
feeling!].

I do believe the IPv4 system, on which the IPv6 was likely built, was a
"upgrade" and therefore had to work with the existing hardware/software. 

It is probably a bit late to require IPv6 systems to support this revised
technique. But ...

- Bernie Volz


-----Original Message-----
From: John Spence [mailto:jspence@zama.net]
Sent: Tuesday, May 23, 2000 4:29 PM
To: tcp-impl@grc.nasa.gov
Subject: IPv6 Path MTU Discovery process ...



Hello;

I sent an eMail to Jack McGann asking a question about IPv6 Path MTU
Discovery.  I had a question about how it was described, and thought I had
an idea for a better solution.  Mr. McGann answered saying that he didn't
know if that had been considered, and suggested I distribute it in this
forum.

Here is the eMail I sent Mr. McGann - please feel free to contact me for
questions and to let me know what you think.

Best regards;

John Spence

---------------------
Hello Mr. McCann;
I have read with interest "IPv6 Clearly Explained", by Pete Loshin. I read
about the Path MTU Discovery process and I had a question. I went to the
IETF site and researched the RFC, and I have a thought.
The book, and the RFC, describe the process is described as the source node
sending the largest packet permissible on it's own network - essentially the
first hop. If any intervening link cannot handle packets that large, it
discards the packet and sends a ICMPv6 "packet too big" message to the
source. The source node drops the size of the packet to within that reported
limit, and send the packet again. Perhaps it makes the final destination,
perhaps another node requires even a smaller packet size, and generates
another "packet too big" message for the source. This could go on for
awhile.
My question is this. Why not have each node in the path that supports only a
smaller MTU than the source node send the "packet too big" message, but
still forward a "shell packet" on towards the destination, where the "shell
package" contains the minimum MTU so far along that path. Since the goal is
to send a packet all the way to the destination, that information may well
be needed, and it would reduce latency in a scenario where there was a
"stepping down" of the MTU size allowed by intervening links as the packets
approached the destination.
The source node would simply wait, collecting whatever "packet still too
big" ICMP messages it received from the intervening nodes (a node that could
accommodate the current MTU would simply forward the shell packet forward,
but would not reply to the source node), until it received a confirmation
from the true destination that the packet had finally arrives. Looking over
the ICMP "packet too big" messages the source had received, it could now
choose the best possible MTU and proceed with confidence.
Heck - we could even have the "shell packet" continue all the way to the
destination node, being adjusted downward by each link along the way (if
needed) - without generating any ICMP "packet too big" messages - and then
make the true destination node send a message back to the source saying "a
2785byte MTU is cleared all the way through", at which time the real data
transmission would begin. You would probably want the very first node that
determined it needed a smaller packet generate the ICMP "packet too big"
message for the source, just to let the source know it wasn't going to be
clear sailing. It could mark the "shell packet" it sent forward to
downstream nodes so they knew they weren't the first to demand a smaller
MTU.
What do you think? Good idea? I'll bet you guys thought of this - just
trying to help. You at the IETF are doing a great job.
----------------------------------------------------------------------------
-----
John E Spence                    ZAMA NETWORKS, INC.
Senior UNIX Systems Engineer
12101 International Boulevard, Seattle WA 98168-2569
wk 206.352.9262  NEXTEL 206.571.9792 fx 206.352.3918
jspence@zama.net || http://www.zama.net
-------------------------------------------------------------------------


From owner-tcp-impl@lerc.nasa.gov  Tue May 23 22:34:19 2000
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To: Bernie Volz <Volz@ipworks.com>
Cc: tcp-impl@grc.nasa.gov
Subject: Re: IPv6 Path MTU Discovery process ... 
From: Zachary Amsden <zamsden@cthulhu.engr.sgi.com>
In-Reply-To: Your message of "Tue, 23 May 2000 18:00:42 EDT."
             <63D30D6E10CFD11190A90000F805FE8602BEBCCC@lespaul.process.com> 
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> I think from IPv4 experience, while this idea would be an optimization, I
> doubt that there are many MTU drops that typically occur (at least on todays
> networks). The value is usually found reasonably fast (with one or two MTU
> drops) [I don't have any hard data to back this up, it is just a gut
> feeling!].

Yes, but how do you know when you have an MTU drop?  There are plenty of 
black-holing routers out there that won't give you an ICMP_UNCREACH/NEEDFRAG.

-- 
Zachary Amsden  zamsden@engr.sgi.com  3-6919  31-2-510  Core Protocols




From owner-tcp-impl@lerc.nasa.gov  Tue May 23 22:34:28 2000
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Date: Wed, 24 May 2000 01:15:15 +0100 (BST)
From: Lloyd Wood <l.wood@eim.surrey.ac.uk>
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To: John Spence <jspence@zama.net>
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Subject: Re: IPv6 Path MTU Discovery process ...
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On Tue, 23 May 2000, John Spence wrote:

> My question is this. Why not have each node in the path that supports only a
> smaller MTU than the source node send the "packet too big" message, but
> still forward a "shell packet" on towards the destination, where the "shell
> package" contains the minimum MTU so far along that path. Since the goal is
> to send a packet all the way to the destination, that information may well
> be needed, and it would reduce latency in a scenario where there was a
> "stepping down" of the MTU size allowed by intervening links as the packets
> approached the destination.

Er, you've just reinvented partial fragmentation. 

For TCP channels, sending on the first part of the packet payload
doesn't really help overall latency; the overall e.g. ftp transfer
doesn't get completed any faster. Full fragmentation towards the
destination might reduce latency, but introduces other problems...

L.

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



From owner-tcp-impl@lerc.nasa.gov  Tue May 23 22:46:41 2000
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Subject: Re: IPv6 Path MTU Discovery process ...
To: Volz@ipworks.com (Bernie Volz)
Date: Wed, 24 May 2000 01:31:54 +0100 (BST)
Cc: jspence@zama.net ('jspence@zama.net'), tcp-impl@grc.nasa.gov
In-Reply-To: <63D30D6E10CFD11190A90000F805FE8602BEBCCC@lespaul.process.com> from "Bernie Volz" at May 23, 2000 06:00:42 PM
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> An interesting question to consider is the cost of doing this ...
> intermediate routers now have to be capable of generating these "shell
> packets" and end nodes (routers too) would need to understand the "shell
> packets" and respond accordingly. How well do the characteristics of this
> "shell packet" need to match the original packet? For example, it is always
> possible that different types of traffic take different paths.

Also you have to checksum it if you want to trust it - thats more overhead
that people just got off routers on the whole

> doubt that there are many MTU drops that typically occur (at least on todays
> networks). The value is usually found reasonably fast (with one or two MTU
> drops) [I don't have any hard data to back this up, it is just a gut
> feeling!].

MTU drops can make a mess on lossy wireless links and very slow stuff, but
even then I see no material problems



From owner-tcp-impl@lerc.nasa.gov  Wed May 24 10:34:12 2000
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Date: Wed, 24 May 2000 12:57:27 +0100
From: Bogdan Ghita <b.ghita@jack.see.plym.ac.uk>
Organization: NRG
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Hello everybody

- if the content of this e-mail is not relevant to the topic of the
list, then, if possible, tell me if there is any other mailing list
where I could ask the question - 

I am trying to determine the number of lost packets for TCP connections.
I created an emulated 'lossy link' (10% packet loss) using Nistnet, and
downloaded a file through this link. I get the number of 'lost' packets
within a connection from the program, then compare it to my estimates.
The monitoring point is AFTER the lossy link, on the direction of the
download.
I noticed that the nearest estimator is the number of 'inverted'
packets, as in the following example:
12:41:35.990251 50.234.ftp-data > ac.uk.1096: . 5841:7301(1460) ack 1
win 8760 (DF)
12:41:35.990294 ac.uk.1096 > 50.234.ftp-data: . 1:1(0) ack 7301 win
30660 (DF) [tos 0x8] 
12:41:36.294567 141.163.50.234.ftp-data > ac.uk.1096: . 8761:10221(1460)
ack 1 win 8760 (DF)
12:41:36.294616 ac.uk.1096 > 50.234.ftp-data: . 1:1(0) ack 7301 win
30660 (DF) [tos 0x8] 
12:41:38.919563 50.234.ftp-data > ac.uk.1096: P 7301:8761(1460) ack 1
win 8760 (DF)

The packet containing data 7301 - 8761 seems to be inverted but, in
fact, it was lost, and this is a retransmission.

The main problem with counting the inverted packets is that it fails to
see a lost packet if it is at the end of a transmission window (there is
no packet sent after it, no duplicate ACK and the sender retransmits it
when the timer expires), which makes it inaccurate.

I have tried also with the Estimated Timeout, but, because I am not
measuring it right at the sending endpoint, the measurement is not as
acurate as it should be, and it gives me some 'false alarms'.

The question is: are there any good estimators for the number of lost
packets?
Any suggestions / relevant papers are welcome as well.

Than kyou very much for your time

Best regards
Bogdan Ghita


From owner-tcp-impl@lerc.nasa.gov  Wed May 24 14:13:09 2000
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> The question is: are there any good estimators for the number of lost
> packets? Any suggestions / relevant papers are welcome as well.
> 
> Than kyou very much for your time

I would start by counting the duplicate ACKs. There are of course some 
exclude cases but by and large the presence of multiple ACKs is what TCP
uses to indicate loss and that seems to be what you're really interested
in, right?

-- 
Eric A. Hall                                            ehall@ehsco.com
+1-650-685-0557                                    http://www.ehsco.com


From owner-tcp-impl@lerc.nasa.gov  Tue May 30 18:15:51 2000
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From: Hyoung-Kee Choi <hkchoi@cc.gatech.edu>
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I am having a problem with Snoop. Snoop abrubtly finished after
recording 4 million packets although I did not specify the -c option in
the command line. The trace file size is about 510MB after the Snoop
run.

In addition, has anyone sucessfully recorded packets in one file more
than 2GB?

Please teach me how i go over it.


From owner-tcp-impl@lerc.nasa.gov  Wed May 31 06:31:46 2000
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From: "Jens-S. Voeckler" <voeckler@rvs.uni-hannover.de>
To: Hyoung-Kee Choi <hkchoi@cc.gatech.edu>
cc: tcp-impl@grc.nasa.gov
Subject: Re: Snoop maxcount
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On Tue, 30 May 2000, Hyoung-Kee Choi wrote:

]I am having a problem with Snoop. Snoop abrubtly finished after
]recording 4 million packets although I did not specify the -c option in
]the command line. The trace file size is about 510MB after the Snoop
]run.

I am restarting my snoop every night...

]In addition, has anyone sucessfully recorded packets in one file more
]than 2GB?

Using the "largefiles" mount option might be the key here, see mount_ufs.

Le deagh dhùrachd,
Dipl.-Ing. Jens-S. Vöckler (voeckler@rvs.uni-hannover.de)
Institute for Computer Networks and Distributed Systems
University of Hanover, Germany; +49 511 762 4726



