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From: "Eddy, Wesley M. (GRC-MS00)[Verizon]" <wesley.m.eddy@nasa.gov>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Date: Tue, 3 Nov 2009 14:17:30 -0600
Thread-Topic: Authentication Option Combined WGLC
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Subject: [tcpm] Authentication Option Combined WGLC
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David and I would like to start a combined Working Group Last Call on
the two documents related to the TCP Authentication Option to be
submitted for Proposed Standard:

The TCP Authentication Option
http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt

Cryptographic Algorithms for TCP's Authentication Option, TCP-AO
http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt


This WGLC will last 3 weeks, ending on November 24.  Please submit
comments the list.

---------------------------
Wes Eddy
Network & Systems Architect
Verizon FNS / NASA GRC
Office: (216) 433-6682
---------------------------


From Donald.Smith@qwest.com  Tue Nov  3 14:45:55 2009
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From: "Smith, Donald" <Donald.Smith@qwest.com>
To: "'Eddy, Wesley M. (GRC-MS00)[Verizon]'" <wesley.m.eddy@nasa.gov>, "'tcpm@ietf.org'" <tcpm@ietf.org>
Date: Tue, 3 Nov 2009 15:45:57 -0700
Thread-Topic: Authentication Option Combined WGLC
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This makes TCP-AO of little value to protect against spoofed source resets =
within the window size of the current Ack number.


"TCP-AO, like TCP MD5, may inhibit connectionless resets. Such resets
   typically occur after peer crashes, either in response to new
   connection attempts or when data is sent on stale connections; in
   either case, the recovering endpoint may lack the connection key
   required (e.g., if lost during the crash). This may result in time-
   outs, rather than more responsive recovery after such a crash. As
Touch                   Expires April 27, 2010                [Page 39]
Internet-Draft   The TCP Simple Authentication Option      October 2009
   noted in Section 7.2, such cases may also result in persistent TCP
   state for old connections that cannot be cleared, and so
   implementations should be capable of detecting an excess of such
   connections and clearing their state if needed to protect memory
   utilization [Ba09]."


The BIGGEST problem with the current MD5 implementation isn't the algorithm=
 being used its the fact that implementers of BGP took the original md5 sta=
tement about connection resets to mean they didn't have to send resets with=
 AUTH. It was an easy out for them and very few vendors do the lookups need=
ed to do an authenticated reset if no connection state exists in their stat=
e table.

It is not hard for a bgp speaker to first check to see if a packet with src=
_port or dst_port is 179 and coming from a neighbor. In cases where a non-i=
nitial packet is coming from a neighbor if there is no current state for th=
at connection the router has all the information needed (current packet plu=
s M the shared key) to send an authenticated reset.



(coffee !=3D sleep) & (!coffee =3D=3D sleep)
Donald.Smith@qwest.com gcia

> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
> Behalf Of Eddy, Wesley M. (GRC-MS00)[Verizon]
> Sent: Tuesday, November 03, 2009 1:18 PM
> To: tcpm@ietf.org
> Subject: [tcpm] Authentication Option Combined WGLC
>
> David and I would like to start a combined Working Group Last Call on
> the two documents related to the TCP Authentication Option to be
> submitted for Proposed Standard:
>
> The TCP Authentication Option
> http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt
>
> Cryptographic Algorithms for TCP's Authentication Option, TCP-AO
> http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt
>
>
> This WGLC will last 3 weeks, ending on November 24.  Please submit
> comments the list.
>
> ---------------------------
> Wes Eddy
> Network & Systems Architect
> Verizon FNS / NASA GRC
> Office: (216) 433-6682
> ---------------------------
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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From touch@ISI.EDU  Tue Nov  3 15:25:59 2009
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1



Smith, Donald wrote:
> This makes TCP-AO of little value to protect against spoofed source
> resets within the window size of the current Ack number.

TCP-AO will drop spoofed resets, because the signature won't match. How
is that not protected?

(the note below focuses on connectionless resets, e.g., after a reboot,
when state from previous connections - such as the ISNs on which the
traffic keys are based - is lost.)

> "TCP-AO, like TCP MD5, may inhibit connectionless resets. Such resets
>    typically occur after peer crashes, either in response to new
>    connection attempts or when data is sent on stale connections; in
>    either case, the recovering endpoint may lack the connection key
>    required (e.g., if lost during the crash). This may result in time-
>    outs, rather than more responsive recovery after such a crash. As
> Touch                   Expires April 27, 2010                [Page 39]
> Internet-Draft   The TCP Simple Authentication Option      October 2009
>    noted in Section 7.2, such cases may also result in persistent TCP
>    state for old connections that cannot be cleared, and so
>    implementations should be capable of detecting an excess of such
>    connections and clearing their state if needed to protect memory
>    utilization [Ba09]."
> 
> 
> The BIGGEST problem with the current MD5 implementation isn't the
> algorithm being used its the fact that implementers of BGP took the
> original md5 statement about connection resets to mean they didn't have
> to send resets with AUTH. It was an easy out for them and very few
> vendors do the lookups needed to do an authenticated reset if no
> connection state exists in their state table.

If you send resets without the AUTH, one of two things will be true:

- - if the endpoint still thinks the connection is TCP-AO protected, it
will drop the reset silently

- - if the endpoint doesn't think the connection is TCP-AO protected, the
reset will work as usual

> It is not hard for a bgp speaker to first check to see if a packet
> with src_port or dst_port is 179 and coming from a neighbor. In cases
> where a non-initial packet is coming from a neighbor if there is no
> current state for that connection the router has all the information
> needed (current packet plus M the shared key) to send an authenticated
> reset.

That presumes it has the ISNs and the SNEs. Without those - e.g., after
a reboot - the reset MUST be dropped if the connection remains
protected. Such stale connections need to be cleaned up using out of
band techniques (timer based garbage collection).

Note that if you retain the ISNs and SNEs, as well as the MKTs - e.g.,
in persistent storage - then you can recover from a reboot and
authenticate resets fine.

Two questions:

- - does this address your concern?

- - if so, does the doc need to be updated to clarify?

Joe

>> -----Original Message-----
>> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
>> Behalf Of Eddy, Wesley M. (GRC-MS00)[Verizon]
>> Sent: Tuesday, November 03, 2009 1:18 PM
>> To: tcpm@ietf.org
>> Subject: [tcpm] Authentication Option Combined WGLC
>>
>> David and I would like to start a combined Working Group Last Call on
>> the two documents related to the TCP Authentication Option to be
>> submitted for Proposed Standard:
>>
>> The TCP Authentication Option
>> http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt
>>
>> Cryptographic Algorithms for TCP's Authentication Option, TCP-AO
>> http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt
>>
>>
>> This WGLC will last 3 weeks, ending on November 24.  Please submit
>> comments the list.
>>
>> ---------------------------
>> Wes Eddy
>> Network & Systems Architect
>> Verizon FNS / NASA GRC
>> Office: (216) 433-6682
>> ---------------------------
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
> 
> This communication is the property of Qwest and may contain confidential or
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> https://www.ietf.org/mailman/listinfo/tcpm
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From Donald.Smith@qwest.com  Tue Nov  3 15:57:54 2009
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From: "Smith, Donald" <Donald.Smith@qwest.com>
To: "'Joe Touch'" <touch@ISI.EDU>
Date: Tue, 3 Nov 2009 16:58:03 -0700
Thread-Topic: [tcpm] Authentication Option Combined WGLC
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(coffee !=3D sleep) & (!coffee =3D=3D sleep)
Donald.Smith@qwest.com gcia

> -----Original Message-----
> From: Joe Touch [mailto:touch@ISI.EDU]
> Sent: Tuesday, November 03, 2009 4:26 PM
> To: Smith, Donald
> Cc: 'Eddy, Wesley M. (GRC-MS00)[Verizon]'; 'tcpm@ietf.org'
> Subject: Re: [tcpm] Authentication Option Combined WGLC
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
>
>
> Smith, Donald wrote:
> > This makes TCP-AO of little value to protect against spoofed source
> > resets within the window size of the current Ack number.
>
> TCP-AO will drop spoofed resets, because the signature won't
> match. How
> is that not protected?
It will drop the spoofed resets and the connectionless resets too.
Thus many ISPs will not enable it.

>
> (the note below focuses on connectionless resets, e.g., after
> a reboot,
> when state from previous connections - such as the ISNs on which the
> traffic keys are based - is lost.)
ISN may be lost but a keep alive or any other packet from your peer has all=
 the information you need excluding the shared secret.

>
> > "TCP-AO, like TCP MD5, may inhibit connectionless resets.
> Such resets
> >    typically occur after peer crashes, either in response to new
> >    connection attempts or when data is sent on stale connections; in
> >    either case, the recovering endpoint may lack the connection key
> >    required (e.g., if lost during the crash). This may
> result in time-
> >    outs, rather than more responsive recovery after such a crash. As
> > Touch                   Expires April 27, 2010
>   [Page 39]
> > Internet-Draft   The TCP Simple Authentication Option
> October 2009
> >    noted in Section 7.2, such cases may also result in
> persistent TCP
> >    state for old connections that cannot be cleared, and so
> >    implementations should be capable of detecting an excess of such
> >    connections and clearing their state if needed to protect memory
> >    utilization [Ba09]."
> >
> >
> > The BIGGEST problem with the current MD5 implementation isn't the
> > algorithm being used its the fact that implementers of BGP took the
> > original md5 statement about connection resets to mean they
> didn't have
> > to send resets with AUTH. It was an easy out for them and very few
> > vendors do the lookups needed to do an authenticated reset if no
> > connection state exists in their state table.
>
> If you send resets without the AUTH, one of two things will be true:
>
> - - if the endpoint still thinks the connection is TCP-AO
> protected, it
> will drop the reset silently
In a connectionless reset it will think the end point is still protected.
It has no way to know that the other end has rebooted/reset its bgp.

>
> - - if the endpoint doesn't think the connection is TCP-AO
> protected, the
> reset will work as usual
>
> > It is not hard for a bgp speaker to first check to see if a packet
> > with src_port or dst_port is 179 and coming from a
> neighbor. In cases
> > where a non-initial packet is coming from a neighbor if there is no
> > current state for that connection the router has all the information
> > needed (current packet plus M the shared key) to send an
> authenticated
> > reset.
>
> That presumes it has the ISNs and the SNEs. Without those -
> e.g., after
> a reboot - the reset MUST be dropped if the connection remains
It has them. They are in the next packet the bgp speaker that didn't reboot=
/reset sends.
There is even an AUTH so the rebooted bgp speaker "knows" this is coming fr=
om a neighbor with AUTH enabled.

> protected. Such stale connections need to be cleaned up using out of
> band techniques (timer based garbage collection).
>
> Note that if you retain the ISNs and SNEs, as well as the MKTs - e.g.,
> in persistent storage - then you can recover from a reboot and
> authenticate resets fine.
>
> Two questions:
>
> - - does this address your concern?
If you mean storing ISN and SNE (along with MKTs which have to be stored in=
 persistent storage)
this could address it but I don't think its required.
The incoming keep alive or other traffic has all but the MKTs (shared secre=
t).

>
> - - if so, does the doc need to be updated to clarify?
>
> Joe
>
> >> -----Original Message-----
> >> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On
> >> Behalf Of Eddy, Wesley M. (GRC-MS00)[Verizon]
> >> Sent: Tuesday, November 03, 2009 1:18 PM
> >> To: tcpm@ietf.org
> >> Subject: [tcpm] Authentication Option Combined WGLC
> >>
> >> David and I would like to start a combined Working Group
> Last Call on
> >> the two documents related to the TCP Authentication Option to be
> >> submitted for Proposed Standard:
> >>
> >> The TCP Authentication Option
> >> http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt
> >>
> >> Cryptographic Algorithms for TCP's Authentication Option, TCP-AO
> >> http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt
> >>
> >>
> >> This WGLC will last 3 weeks, ending on November 24.  Please submit
> >> comments the list.
> >>
> >> ---------------------------
> >> Wes Eddy
> >> Network & Systems Architect
> >> Verizon FNS / NASA GRC
> >> Office: (216) 433-6682
> >> ---------------------------
> >>
> >> _______________________________________________
> >> tcpm mailing list
> >> tcpm@ietf.org
> >> https://www.ietf.org/mailman/listinfo/tcpm
> >>
> >
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From touch@ISI.EDU  Tue Nov  3 17:26:06 2009
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Hash: SHA1

Hi, Donald,


...
>> > Smith, Donald wrote:
>>> > > This makes TCP-AO of little value to protect against spoofed source
>>> > > resets within the window size of the current Ack number.
>> >
>> > TCP-AO will drop spoofed resets, because the signature won't
>> > match. How
>> > is that not protected?
>
> It will drop the spoofed resets and the connectionless resets too.

That's the point - connectionless resets cannot be distinguished from
spoofed ones. Either both are dropped - which is secure, but can cause
buildup if endpoints restart frequently, or both are allowed - which is
a security hole.

> Thus many ISPs will not enable it.

I don't see this as a configurable option. It's discussed, but IMO both
MUST be dropped.

>> > (the note below focuses on connectionless resets, e.g., after
>> > a reboot,
>> > when state from previous connections - such as the ISNs on which the
>> > traffic keys are based - is lost.)
>
> ISN may be lost but a keep alive or any other packet from your peer
> has all the information you need excluding the shared secret.

The traffic key is computed using the ISN pair and the SNEs. Those are
not available in-band - they're initialized at connection start (ISNs as
exchanged, SNEs to 0) and maintained during the connection. They are not
available in any segment information.

...
>> > If you send resets without the AUTH, one of two things will be true:
>> >
>> > - - if the endpoint still thinks the connection is TCP-AO
>> > protected, it
>> > will drop the reset silently
>
> In a connectionless reset it will think the end point is still protected.
> It has no way to know that the other end has rebooted/reset its bgp.

The packets won't be ACKd - ever. There will be no RST. Eventually, TCP
will time-out and close the connection, since it has data to send and
the other end is failing to respond - see RFC 1122 Sec. 4.2.3.5.

...
>> > That presumes it has the ISNs and the SNEs. Without those -
>> > e.g., after
>> > a reboot - the reset MUST be dropped if the connection remains
>
> It has them. They are in the next packet the bgp speaker that didn't reboot/reset sends.

ISNs are sent during the SYN exchange only. They cannot be determined
from other segments. SNEs are never sent on the wire, and the BGP
speaker may know the SNEs for both ends, but there is no way to
communicate this to the receiver.

> There is even an AUTH so the rebooted bgp speaker "knows" this is coming from a neighbor with AUTH enabled.

Yes, but it doesn't have sufficient information to calculate the traffic
key needed to verify the segment, so the segment will be dropped before
*any* action is taken (including 'knowing' whether the other side has
AUTH enabled - that is on a per-connection basis only, and acting on
*any* non-verified segment would pose a security hole).

>> > protected. Such stale connections need to be cleaned up using out of
>> > band techniques (timer based garbage collection).
>> >
>> > Note that if you retain the ISNs and SNEs, as well as the MKTs - e.g.,
>> > in persistent storage - then you can recover from a reboot and
>> > authenticate resets fine.
>> >
>> > Two questions:
>> >
>> > - - does this address your concern?
>
> If you mean storing ISN and SNE (along with MKTs which have to be stored in persistent storage)
> this could address it but I don't think its required.

As per above, it seems to be. SNEs need to be updated when they
increment too.

Joe
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From carsten.wolff@rwth-aachen.de  Wed Nov  4 04:48:42 2009
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From: Carsten Wolff <carsten.wolff@rwth-aachen.de>
To: Matt Mathis <mathis@psc.edu>
Date: Wed, 4 Nov 2009 13:48:51 +0100
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Cc: tcpm Extensions WG <tcpm@ietf.org>
Subject: [tcpm] Curiosity about draft-mathis-tcp-ratehalving
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Hello Matt, everyone,

I'm curious about draft-mathis-tcp-ratehalving 
(http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt) and why it 
never exceeded version 00 and expired years ago. It seems, several of the 
ideas (fack, ratehalving, the concept of rhcwnd) in this paper made it into 
the Linux TCP implementation in some form, which means they're part of a big 
chunk of today's web servers and other Internet hosts. Why has there been no 
interest in taking this draft further?

Cheers,
Carsten

From rs@netapp.com  Mon Nov  9 04:56:51 2009
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Thread-Topic: [tcpm] Should draft-ietf-tcpm-sack-recovery-entry update RFC 3717 (SACK-TCP)
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Subject: Re: [tcpm] Should draft-ietf-tcpm-sack-recovery-entry update RFC 3717 (SACK-TCP)
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Hi Alexander et al.,

This will be the first post to this group, so excuse me if I act =
inappropriately.

I'm curious about one little tidbit which has been bugging me for the =
better part of the last two monts, and which is closely related with TCP =
SACK operations (thus it might belong to this thread?)


The implicit assumption for TCP fast recovery is, that packet loss =
happens randomly (ie. to different segments each time) with low =
correlation between the drop events. Also, a drop event is used as a =
implicit signal to indicate congestion. So far, so good.

It seems to me, that the focus of most developments has been the =
internet environment - where statistical assumptions like the above =
mentioned arguably hold true.

However, certain high-speed LANs seem to exhibit characteristics, which =
don't play well with these implicit assumptions (uncorrelated packet =
loss) - the smaller the network, the more deviation from an "good =
seasoned" link (exhibiting some form of congestion) is likely to occur.

Also, as has been noted in prior research, many internet routers do use =
more "tcp-friendly" RED or WRED queue policies, over the simplistic =
TailDrop most often encountered in LANs (default policy of L2 switches =
and L3 routers).

In one extreme, I have found a (misbehaving=B4?) TCP stack/host, which =
sends out a burst of segments (4-6) @ 10GbE wirespeed, which immediately =
cause queue buffer overload and TailDrop in the first hop L2 Switch, =
when two such high performance hosts try to establish a high speed =
communication. With other words, the hosts themselves seem to make sure =
that there is a high correlation between TCP (fast) recovery and further =
packet loss.


But what puzzles me the most - even with SACK enabled TCP stacks, =
virtually no implementation can detect / act upon detection of the loss =
of a retransmitted segment during fast recovery. This despite the fact, =
that the stipulations in RFC3517 requires the receiver to make the =
information to detect such an event implicitly available to the sender. =
The first SACK option has to reflect the last segment, which triggered =
this SACK.=20

Together with the scoreboard held at the sender, it should be rather =
easy to find out if the left edge of the lowest hole (relative to stream =
octets) closes.

If that left edge stays constant for "DupThresh" number of ACKs, which =
reduce the overall number of octets in holes (any one hole might close =
due to the retransmitted packets still received), AND the sender =
retransmits beginning with the lowest hole first, this would be a clear =
indication of another segment retransmit loss...

Even a less speedy detection logic would work for SACK-enabled sessions: =
once the fast recovery is finished from the sender's point of view, if =
the receiver still complains about missing segments (indicated by having =
the SACK rightmost edge - in the first slot SACK option - at a segment =
higher than when fast recovery started), another round of fast recovery =
could be invoked, rather than waiting for RTO.

Of course, the first approach would be better for low cwnd sessions with =
only very few segments in transit - and both could be combined with the =
proposed sack recovery speed-ups... (Reducing DupThresh for low cwnd =
sessions / when little data is being sent).



Congestion control should act to this event (it will now, but only one =
RTO later...), and the SACK retransmit vector (HighRxt) reset, using =
LimitedTransmit for sending out the retransmission segments - once cwnd =
+ pipe allows; any retransmitted segments still in the network will =
close their respective SACK holes before the new HighRxt advances to =
them.

And, RTO should be reduce (I guess to nearly zero, between SACK-enabled =
hosts).


I have run numerous tests, to check the behavior of different TCP Stacks =
(FreeBSD 4.2 - 8.0; windows xp, vista, 7, 2003; Linux 2.6.16 and =
others).


All these stacks seem to exhibit this issue; What I don't know yet is =
the percentage of multi-loss segement events triggering RTO - but I =
assume that the majority of RTOs happen because of this.

In LAN environments (ie. 10 GbE over 1 km @ 2 ms latency due to the L2 =
hops in between) featuring relatively few streams, the effect of any =
single RTO can be quite tremendeous - taking considerable theoretical =
bandwidth away from the session (ie. 1 sec minimum RTO equals 1.2 GB; =
even with more recent RTO values around 0.2 - 0.4 sec, each RTO is still =
a few hundred MB "lost" capacity under optimal circumstances.


Nevertheless, I cann't imagine that I am the first one to bring up this =
issue (despite having failed to find any study of this effect). :)


One more clarification, which came up after I looked at the FreeBSD =
implementation of Limited Transmit; this might be a nit-pick, but when =
RFC 3042 is active, shouldn't ABC also be used during LimitedTransmit / =
FastRecovery? (FreeBSD MAIN is increasing cwnd by 1 mss for each new =
ACK, instead for the amount of data in that ack...

Thanks a lot!


Best regards, =20


Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com <BLOCKED::http://www.netapp.com/> =20
Franz-Klein-Gasse 5
1190 Wien=20

*	To: "tcpm at ietf.org <mailto:tcpm@DOMAIN.HIDDEN>  WG Extensions" =
<tcpm at ietf.org <mailto:tcpm@DOMAIN.HIDDEN> >
*	Subject: [tcpm] Should draft-ietf-tcpm-sack-recovery-entry update RFC =
3717 (SACK-TCP)
*	From: Alexander Zimmermann <alexander.zimmermann at =
nets.rwth-aachen.de <mailto:alexander.zimmermann@DOMAIN.HIDDEN> >
*	Date: Wed, 21 Oct 2009 12:22:50 +0200

  _____ =20

Hi folks,

based on the fact that the draft "draft-ietf-tcpm-sack-recovery-entry" =
is adopted as WG item now and intended to be a "standards track" =
document, I would like to start a poll/discussion whether the draft =
should update RFC 3517 or not? Moreover, should we produce a separate =
document or an update of RFC 3517?=20

a) separate document, do not update RFC 3517
b) separate document, update RFC 3517
c) RFC3517bis, obsolete RFC 3517

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann at cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//



From hannemann@i4.informatik.rwth-aachen.de  Mon Nov  9 07:21:16 2009
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Date: Mon, 09 Nov 2009 16:21:15 +0100
From: Arnd Hannemann <hannemann@i4.informatik.rwth-aachen.de>
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Subject: Re: [tcpm] Should draft-ietf-tcpm-sack-recovery-entry update RFC 3717 (SACK-TCP)
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Scheffenegger, Richard schrieb:

> But what puzzles me the most - even with SACK enabled TCP stacks, virtually no
> implementation can detect / act upon detection of the loss of a retransmitted
> segment during fast recovery. This despite the fact, that the stipulations in
> RFC3517 requires the receiver to make the information to detect such an event
> implicitly available to the sender. The first SACK option has to reflect the 
> last segment, which triggered this SACK. 

Some implementations actually do detect lost retransmissions.
(see Linux tcp_mark_lost_retrans())
However, currently only for FACK enabled flows. (Comments indicate,
it would be non-trivial / too-expensive for SACK enabled flows.)

> Together with the scoreboard held at the sender, it should be rather easy to
> find out if the left edge of the lowest hole (relative to stream octets) closes

> If that left edge stays constant for "DupThresh" number of ACKs, which reduce
> the overall number of octets in holes (any one hole might close due to the
> retransmitted packets still received), AND the sender retransmits beginning
> with the lowest hole first, this would be a clear indication of another segment
> retransmit loss...

Sorry, I don't understand. You mean "DupThresh" number of DupACKs? Sack-blocks?
After which point in time? After the first retransmit?
SACK-blocks higher than recovery point? (the latter would make most sense, IMO)

> Even a less speedy detection logic would work for SACK-enabled sessions: once
> the fast recovery is finished from the sender's point of view, if the receiver
> still complains about missing segments (indicated by having the SACK rightmost
> edge - in the first slot SACK option - at a segment higher than when fast
> recovery started), another round of fast recovery could be invoked, rather than
> waiting for RTO.

How can fast recovery be "finished" if the sender is missing packets?

Best regards,
Arnd Hannemann


From alexander.zimmermann@nets.rwth-aachen.de  Mon Nov  9 07:27:23 2009
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From: Alexander Zimmermann <alexander.zimmermann@nets.rwth-aachen.de>
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To: Richard Scheffenegger <rs@netapp.com>
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Subject: [tcpm] Detect Lost Retransmit with SACK
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Hi Richard,

firstly welcome on the list :-)
Since your question in not really related to the poll I change the =20
title...

Comments inline.


Am 09.11.2009 um 13:57 schrieb Scheffenegger, Richard:

>
>
> Hi Alexander et al.,
>
> This will be the first post to this group, so excuse me if I act =20
> inappropriately.
>
> I'm curious about one little tidbit which has been bugging me for =20
> the better part of the last two monts, and which is closely related =20=

> with TCP SACK operations (thus it might belong to this thread?)
>
>
> The implicit assumption for TCP fast recovery is, that packet loss =20
> happens randomly (ie. to different segments each time) with low =20
> correlation between the drop events. Also, a drop event is used as a =20=

> implicit signal to indicate congestion. So far, so good.
>
> It seems to me, that the focus of most developments has been the =20
> internet environment - where statistical assumptions like the above =20=

> mentioned arguably hold true.
>
> However, certain high-speed LANs seem to exhibit characteristics, =20
> which don't play well with these implicit assumptions (uncorrelated =20=

> packet loss) - the smaller the network, the more deviation from an =20
> "good seasoned" link (exhibiting some form of congestion) is likely =20=

> to occur.
>
> Also, as has been noted in prior research, many internet routers do =20=

> use more "tcp-friendly" RED or WRED queue policies, over the =20
> simplistic TailDrop most often encountered in LANs (default policy =20
> of L2 switches and L3 routers).
>
> In one extreme, I have found a (misbehaving=C2=B4?) TCP stack/host, =
which =20
> sends out a burst of segments (4-6) @ 10GbE wirespeed, which =20
> immediately cause queue buffer overload and TailDrop in the first =20
> hop L2 Switch, when two such high performance hosts try to establish =20=

> a high speed communication. With other words, the hosts themselves =20
> seem to make sure that there is a high correlation between TCP =20
> (fast) recovery and further packet loss.
>
>
> But what puzzles me the most - even with SACK enabled TCP stacks, =20
> virtually no implementation can detect / act upon detection of the =20
> loss of a retransmitted segment during fast recovery. This despite =20
> the fact, that the stipulations in RFC3517 requires the receiver to =20=

> make the information to detect such an event implicitly available to =20=

> the sender. The first SACK option has to reflect the last segment, =20
> which triggered this SACK.
>
> Together with the scoreboard held at the sender, it should be rather =20=

> easy to find out if the left edge of the lowest hole (relative to =20
> stream octets) closes.

What do you with "left edge of the lowest hole"? Do you mean SND.UNA? =20=

If ACK covers SND.UNA then it is an cumulative ACK.

>
> If that left edge stays constant for "DupThresh" number of ACKs, =20
> which reduce the overall number of octets in holes (any one hole =20
> might close due to the retransmitted packets still received), AND =20
> the sender retransmits beginning with the lowest hole first, this =20
> would be a clear indication of another segment retransmit loss...

Sorry, I don't understand. If we have 20 segments in flight and one =20
segment gets lost, you will retransmit after 3 DUPACKS the oldest =20
outstanding segment.
Then, assuming no reordering and no further lost, you will get 17 =20
DUPACKS (without Limited Transmit) before your hole is closed.

What do I miss here?

Can you give me an example?

>
> Even a less speedy detection logic would work for SACK-enabled =20
> sessions: once the fast recovery is finished from the sender's point =20=

> of view, if the receiver still complains about missing segments =20
> (indicated by having the SACK rightmost edge - in the first slot =20
> SACK option - at a segment higher than when fast recovery started), =20=

> another round of fast recovery could be invoked, rather than waiting =20=

> for RTO.
>
> Of course, the first approach would be better for low cwnd sessions =20=

> with only very few segments in transit - and both could be combined =20=

> with the proposed sack recovery speed-ups... (Reducing DupThresh for =20=

> low cwnd sessions / when little data is being sent).
>
>
>
> Congestion control should act to this event (it will now, but only =20
> one RTO later...), and the SACK retransmit vector (HighRxt) reset, =20
> using LimitedTransmit for sending out the retransmission segments - =20=

> once cwnd + pipe allows; any retransmitted segments still in the =20
> network will close their respective SACK holes before the new =20
> HighRxt advances to them.
>
> And, RTO should be reduce (I guess to nearly zero, between SACK-=20
> enabled hosts).
>
>
> I have run numerous tests, to check the behavior of different TCP =20
> Stacks (FreeBSD 4.2 - 8.0; windows xp, vista, 7, 2003; Linux 2.6.16 =20=

> and others).
>
>
> All these stacks seem to exhibit this issue; What I don't know yet =20
> is the percentage of multi-loss segement events triggering RTO - but =20=

> I assume that the majority of RTOs happen because of this.
>
> In LAN environments (ie. 10 GbE over 1 km @ 2 ms latency due to the =20=

> L2 hops in between) featuring relatively few streams, the effect of =20=

> any single RTO can be quite tremendeous - taking considerable =20
> theoretical bandwidth away from the session (ie. 1 sec minimum RTO =20
> equals 1.2 GB; even with more recent RTO values around 0.2 - 0.4 =20
> sec, each RTO is still a few hundred MB "lost" capacity under =20
> optimal circumstances.
>
>
> Nevertheless, I cann't imagine that I am the first one to bring up =20
> this issue (despite having failed to find any study of this =20
> effect). :)
>
>
> One more clarification, which came up after I looked at the FreeBSD =20=

> implementation of Limited Transmit; this might be a nit-pick, but =20
> when RFC 3042 is active, shouldn't ABC also be used during =20
> LimitedTransmit / FastRecovery?

Why? One reason for ABC are lying receivers (ACK Division). So, the =20
worst case is Slow-Start...

> (FreeBSD MAIN is increasing cwnd by 1 mss for each new ACK, instead =20=

> for the amount of data in that ack...

What do you describe here? Slow-Start?
RFC 3042 says: "The congestion window (cwnd) MUST NOT be changed when =20=

these new segments are transmitted."

>
> Thanks a lot!
>
>
> Best regards,

Alex

>
>
>
> Richard Scheffenegger
> Field Escalation Engineer
> NetApp Global Support
> NetApp
> +43 1 3676811 3146 Office (2143 3146 - internal)
> +43 676 654 3146 Mobile
> www.netapp.com <BLOCKED::http://www.netapp.com/>
> Franz-Klein-Gasse 5
> 1190 Wien
>
> *	To: "tcpm at ietf.org <mailto:tcpm@DOMAIN.HIDDEN>  WG =
Extensions" =20
> <tcpm at ietf.org <mailto:tcpm@DOMAIN.HIDDEN> >
> *	Subject: [tcpm] Should draft-ietf-tcpm-sack-recovery-entry =
update =20
> RFC 3717 (SACK-TCP)
> *	From: Alexander Zimmermann <alexander.zimmermann at =
nets.rwth-aachen.de=20
>  <mailto:alexander.zimmermann@DOMAIN.HIDDEN> >
> *	Date: Wed, 21 Oct 2009 12:22:50 +0200
>
> _____
>
> Hi folks,
>
> based on the fact that the draft "draft-ietf-tcpm-sack-recovery-=20
> entry" is adopted as WG item now and intended to be a "standards =20
> track" document, I would like to start a poll/discussion whether the =20=

> draft should update RFC 3517 or not? Moreover, should we produce a =20
> separate document or an update of RFC 3517?
>
> a) separate document, do not update RFC 3517
> b) separate document, update RFC 3517
> c) RFC3517bis, obsolete RFC 3517
>
> //
> // Dipl.-Inform. Alexander Zimmermann
> // Department of Computer Science, Informatik 4
> // RWTH Aachen University
> // Ahornstr. 55, 52056 Aachen, Germany
> // phone: (49-241) 80-21422, fax: (49-241) 80-22220
> // email: zimmermann at cs.rwth-aachen.de
> // web: http://www.umic-mesh.net
> //
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//
//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


From rs@netapp.com  Mon Nov  9 09:27:18 2009
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Thread-Topic: Detect Lost Retransmit with SACK
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Alexander Zimmermann" <zimmermann@nets.rwth-aachen.de>
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Subject: Re: [tcpm] Detect Lost Retransmit with SACK
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Hi Alexander,=20

Thanks for the welcome :)=20

I fork another thread with the LimitedTransport||FastRecovery / ABC =
interaction...


I will try to sketch up an example to demonstrate what problem I'm =
trying to address:


Let's assume the cwnd is already open for at least 7 segments, before =
the segment with
sequence number 10000 is the first one to be dropped by the network.

Also, let's assume that FastRetransmit runs from the left edge of the =
leftmost hole=20
(SND.UNA) upwards, and that per ACK only a single segment is sent.=20
             =20

             Triggering    ACK      Left     Right    Left     Right
             Segment                Edge 1   Edge 1   Edge 2   Edge 2

              9000          9000
             10000  (lost)       *
             11000  (lost)
             12000  (lost)
             13000  (lost)
             14000  (lost)
             15000         10000    15000    16000
             16000         10000    15000    17000
             17000         10000    15000    18000

3 ACKs trigger fast retransmit

             10000  (lost again)=20
             11000         10000    11000    12000    15000    18000
             12000         10000    11000    13000    15000    18000
             13000         10000    11000    14000    15000    18000
-> here we have again 3 ACKs indicating a another loss of one of the =
retransmitted=20
packets. The leftmost hole did not change, while the overall number of =
SACKed
octets did decrease for 3 consecutive ACKs (4; 3 and 2 segments marked =
by SACK).

Current behaviour of investigated TCP Stacks:
             14000         10000    11000    18000
(normal transmit resumes)
             18000         10000    11000    19000
             19000         10000    11000    20000
             20000         10000    11000    21000
             21000         10000    11000    22000
             22000         10000    11000    23000
             ::            ::       ::       ::
Eventually, RTO trips off, retransmitting the lost segment; this happens =
RTO later,
followed by slow-start...

             50000         10000    11000    50000
             ::
             ::
             10000         50000

However, this can be somewhere between 0.2 and 1.0 sec later with a =
"fresh" TCP=20
session (no prior connection properties known (cached) by sender). Most =
likely,
the cwnd has filled up already way sooner (as demonstated, the problem =
seems to be
most prominent in Highspeed LANs), so that for nearly as long, no data =
is actually
transmitted.

         =20

Proposed behaviour:=20

             Triggering    ACK      Left     Right    Left     Right
             Segment                Edge 1   Edge 1   Edge 2   Edge 2

              9000          9000
             10000  (lost)       *
             11000  (lost)
             12000  (lost)
             13000  (lost)
             14000  (lost)
             15000         10000    15000    16000
             16000         10000    15000    17000
             17000         10000    15000    18000

3 ACKs trigger fast retransmit

             10000  (lost again) *
             11000         10000    11000    12000    15000    18000
             12000         10000    11000    13000    15000    18000
             13000         10000    11000    14000    15000    18000

Once the ACK + SACK options indicate that the leftmost hole is not =
shrinking,
while the SACKed octets are increasing (to deal with clients which send =
one=20
retransmission segment and one new segment interspaced, or when multiple =
holes
exist which are being filled, or when network reordering occurs, or when =
some=20
wore segments get lost again):=20

Reset the Rexmit vector to the beginning of the Hole-List (SND.UNA), =
clear
counter to count duplicates (just in case one segment gets lost again =
during
retransmit), and keep the DUPACK detection logic armed...

Also, this reaction should not occur before 1 RTT - so ACKs subsequent =
to the=20
three which indicated the "lost again" segment will take care (in the =
typical case)
that no segments are retransmitted needlessly. ACK processing has to =
occur before
deciding which segment (retransmit / new) to send next. Holes will then =
be marked=20
fully retransmitted, before the 2nd retransmission round would advance =
to them).

             10000         14000    15000      18000
             14000         18000
(normal transmit resumes, but with cwnd shrunk by 2 congestion events)
             18000         19000
             19000         20000


And yes, I was unclear with the use of the terminology; I should have =
probably=20
stated "pipe" instead of "cwnd" below, as cwnd is not touched during =
LimitedTransmit /
FastRetransmit...


Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20
Franz-Klein-Gasse 5
1190 Wien=20


-----Original Message-----
From: Alexander Zimmermann [mailto:zimmermann@nets.rwth-aachen.de]=20
Sent: Montag, 9. November 2009 16:26
To: Scheffenegger, Richard
Cc: tcpm@ietf.org Extensions WG
Subject: Detect Lost Retransmit with SACK

Hi Richard,

firstly welcome on the list :-)
Since your question in not really related to the poll I change the =
title...

Comments inline.


Am 09.11.2009 um 13:57 schrieb Scheffenegger, Richard:

>
>
> Hi Alexander et al.,
>
> This will be the first post to this group, so excuse me if I act=20
> inappropriately.
>
> I'm curious about one little tidbit which has been bugging me for the=20
> better part of the last two monts, and which is closely related with=20
> TCP SACK operations (thus it might belong to this thread?)
>
>
> The implicit assumption for TCP fast recovery is, that packet loss =20
> happens randomly (ie. to different segments each time) with low =20
> correlation between the drop events. Also, a drop event is used as a =20
> implicit signal to indicate congestion. So far, so good.
>
> It seems to me, that the focus of most developments has been the =20
> internet environment - where statistical assumptions like the above =20
> mentioned arguably hold true.
>
> However, certain high-speed LANs seem to exhibit characteristics, =20
> which don't play well with these implicit assumptions (uncorrelated =20
> packet loss) - the smaller the network, the more deviation from an =20
> "good seasoned" link (exhibiting some form of congestion) is likely =20
> to occur.
>
> Also, as has been noted in prior research, many internet routers do =20
> use more "tcp-friendly" RED or WRED queue policies, over the =20
> simplistic TailDrop most often encountered in LANs (default policy =20
> of L2 switches and L3 routers).
>
> In one extreme, I have found a (misbehaving=B4?) TCP stack/host, which =
=20
> sends out a burst of segments (4-6) @ 10GbE wirespeed, which =20
> immediately cause queue buffer overload and TailDrop in the first =20
> hop L2 Switch, when two such high performance hosts try to establish =20
> a high speed communication. With other words, the hosts themselves =20
> seem to make sure that there is a high correlation between TCP =20
> (fast) recovery and further packet loss.
>
>
> But what puzzles me the most - even with SACK enabled TCP stacks, =20
> virtually no implementation can detect / act upon detection of the =20
> loss of a retransmitted segment during fast recovery. This despite =20
> the fact, that the stipulations in RFC3517 requires the receiver to =20
> make the information to detect such an event implicitly available to =20
> the sender. The first SACK option has to reflect the last segment, =20
> which triggered this SACK.
>
> Together with the scoreboard held at the sender, it should be rather =20
> easy to find out if the left edge of the lowest hole (relative to =20
> stream octets) closes.

What do you with "left edge of the lowest hole"? Do you mean SND.UNA? =20
If ACK covers SND.UNA then it is an cumulative ACK.

>
> If that left edge stays constant for "DupThresh" number of ACKs, =20
> which reduce the overall number of octets in holes (any one hole =20
> might close due to the retransmitted packets still received), AND =20
> the sender retransmits beginning with the lowest hole first, this =20
> would be a clear indication of another segment retransmit loss...

Sorry, I don't understand. If we have 20 segments in flight and one =20
segment gets lost, you will retransmit after 3 DUPACKS the oldest =20
outstanding segment.
Then, assuming no reordering and no further lost, you will get 17 =20
DUPACKS (without Limited Transmit) before your hole is closed.

What do I miss here?

Can you give me an example?

>
> Even a less speedy detection logic would work for SACK-enabled =20
> sessions: once the fast recovery is finished from the sender's point =20
> of view, if the receiver still complains about missing segments =20
> (indicated by having the SACK rightmost edge - in the first slot =20
> SACK option - at a segment higher than when fast recovery started), =20
> another round of fast recovery could be invoked, rather than waiting =20
> for RTO.
>
> Of course, the first approach would be better for low cwnd sessions =20
> with only very few segments in transit - and both could be combined =20
> with the proposed sack recovery speed-ups... (Reducing DupThresh for =20
> low cwnd sessions / when little data is being sent).
>
>
>
> Congestion control should act to this event (it will now, but only =20
> one RTO later...), and the SACK retransmit vector (HighRxt) reset, =20
> using LimitedTransmit for sending out the retransmission segments - =20
> once cwnd + pipe allows; any retransmitted segments still in the =20
> network will close their respective SACK holes before the new =20
> HighRxt advances to them.
>
> And, RTO should be reduce (I guess to nearly zero, between SACK-=20
> enabled hosts).
>
>
> I have run numerous tests, to check the behavior of different TCP =20
> Stacks (FreeBSD 4.2 - 8.0; windows xp, vista, 7, 2003; Linux 2.6.16 =20
> and others).
>
>
> All these stacks seem to exhibit this issue; What I don't know yet =20
> is the percentage of multi-loss segement events triggering RTO - but =20
> I assume that the majority of RTOs happen because of this.
>
> In LAN environments (ie. 10 GbE over 1 km @ 2 ms latency due to the =20
> L2 hops in between) featuring relatively few streams, the effect of =20
> any single RTO can be quite tremendeous - taking considerable =20
> theoretical bandwidth away from the session (ie. 1 sec minimum RTO =20
> equals 1.2 GB; even with more recent RTO values around 0.2 - 0.4 =20
> sec, each RTO is still a few hundred MB "lost" capacity under =20
> optimal circumstances.
>
>
> Nevertheless, I cann't imagine that I am the first one to bring up =20
> this issue (despite having failed to find any study of this =20
> effect). :)
>
>
> One more clarification, which came up after I looked at the FreeBSD =20
> implementation of Limited Transmit; this might be a nit-pick, but =20
> when RFC 3042 is active, shouldn't ABC also be used during =20
> LimitedTransmit / FastRecovery?

Why? One reason for ABC are lying receivers (ACK Division). So, the =20
worst case is Slow-Start...

> (FreeBSD MAIN is increasing cwnd by 1 mss for each new ACK, instead =20
> for the amount of data in that ack...

What do you describe here? Slow-Start?
RFC 3042 says: "The congestion window (cwnd) MUST NOT be changed when =20
these new segments are transmitted."

>
> Thanks a lot!
>
>
> Best regards,

Alex

>
>
>
> Richard Scheffenegger
> Field Escalation Engineer
> NetApp Global Support
> NetApp
> +43 1 3676811 3146 Office (2143 3146 - internal)
> +43 676 654 3146 Mobile
> www.netapp.com <BLOCKED::http://www.netapp.com/>
> Franz-Klein-Gasse 5
> 1190 Wien
>
> *	To: "tcpm at ietf.org <mailto:tcpm@DOMAIN.HIDDEN>  WG Extensions" =20
> <tcpm at ietf.org <mailto:tcpm@DOMAIN.HIDDEN> >
> *	Subject: [tcpm] Should draft-ietf-tcpm-sack-recovery-entry update =20
> RFC 3717 (SACK-TCP)
> *	From: Alexander Zimmermann <alexander.zimmermann at =
nets.rwth-aachen.de=20
>  <mailto:alexander.zimmermann@DOMAIN.HIDDEN> >
> *	Date: Wed, 21 Oct 2009 12:22:50 +0200
>
>  _____
>
> Hi folks,
>
> based on the fact that the draft "draft-ietf-tcpm-sack-recovery-=20
> entry" is adopted as WG item now and intended to be a "standards =20
> track" document, I would like to start a poll/discussion whether the =20
> draft should update RFC 3517 or not? Moreover, should we produce a =20
> separate document or an update of RFC 3517?
>
> a) separate document, do not update RFC 3517
> b) separate document, update RFC 3517
> c) RFC3517bis, obsolete RFC 3517
>
> //
> // Dipl.-Inform. Alexander Zimmermann
> // Department of Computer Science, Informatik 4
> // RWTH Aachen University
> // Ahornstr. 55, 52056 Aachen, Germany
> // phone: (49-241) 80-21422, fax: (49-241) 80-22220
> // email: zimmermann at cs.rwth-aachen.de
> // web: http://www.umic-mesh.net
> //
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


From rs@netapp.com  Mon Nov  9 09:37:35 2009
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From: "Scheffenegger, Richard" <rs@netapp.com>
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[removed other thread stuff]

Alex,

Again the scenario where the sender is in FastRetransmit or
LimitedTransmit=20
mode, following a congestion (drop) event.

Shouldn't a sender, fully honoring RFC 3042, increase the pipe only by
the amount=20
of newly ACKed / SACKed octets, rather then increasing the pipe by 1*mss
per
received ACK?

If the sender has a decent cwnd, and a misbehaving (not to say lying)
receiver
all of a sudden sends individual SACKs, ACKing one more octet, and
SACKing
every other octet, each in an individual ACK.

This might lure the sender to bloat it's (retransmission / limited
transmit)
pipe to become much larger than cwnd (I'm not sure if all
implementations
limit the pipe to min(cwnd, pipe)....

But really, I think this is mostly a nit-pick; worst case would be a
burst of
traffic, potentially causing momentarily queue overflows in the network,
and=20
(lot's) of retransmissions...

But that later part should be better, when bursty loss can be dealt with
better :)

Regards,

=20


Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20
Franz-Klein-Gasse 5
1190 Wien=20


-----Original Message-----
From: Alexander Zimmermann [mailto:zimmermann@nets.rwth-aachen.de]=20
Sent: Montag, 9. November 2009 16:26
To: Scheffenegger, Richard
Cc: tcpm@ietf.org Extensions WG
Subject: Detect Lost Retransmit with SACK

Hi Richard,

firstly welcome on the list :-)
Since your question in not really related to the poll I change the
title...

Comments inline.


Am 09.11.2009 um 13:57 schrieb Scheffenegger, Richard:

>
> One more clarification, which came up after I looked at the FreeBSD =20
> implementation of Limited Transmit; this might be a nit-pick, but =20
> when RFC 3042 is active, shouldn't ABC also be used during =20
> LimitedTransmit / FastRecovery?

Why? One reason for ABC are lying receivers (ACK Division). So, the =20
worst case is Slow-Start...

> (FreeBSD MAIN is increasing cwnd by 1 mss for each new ACK, instead =20
> for the amount of data in that ack...

What do you describe here? Slow-Start?
RFC 3042 says: "The congestion window (cwnd) MUST NOT be changed when =20
these new segments are transmitted."

>
> Thanks a lot!
>
>
> Best regards,

Alex


From rs@netapp.com  Mon Nov  9 10:06:02 2009
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Hi group,

I forgot to mention the actual testing scenario I was doing, to profile =
all these
TCP stack against.

Basically, I used a userland TCP "forging" tool, where each frame can be =
individually
crafted (content, timing, loss).

My test opens a tcp session (http get request, for simplicity's sake), =
with SACK=20
negotiated and then counts the segments being received, behaving =
(mostly) like a=20
well-behaving TCP client. However, the segments with these numbers:

200, 250, 253, 255, 257, 258, 259, 260, 265, 267

are dropped this number of times they are seen in the stream:

1,   1,   1,   1,   1,   2,   1,   1,   1,   1
=20
The grace period of 200 packets is to have a decently wide open cwnd; =
the drop at
Segment 200 also serves to check if the cwnd is larger than 50 segments =
when the
Burst drop (250-267) occurs, and also to "prime" the SACK scoreboard =
(preventing
the sender from fastpathing). The burst in this case is in the time axis =
and, with
segment number 258, sequence space axis...

None of the TCP Stacks I have investigated so far, were able to recover =
without
a RTO (between 0.2 and 1 sec later; Windows 7 was particularly peculiar, =
as it
starts shifting the original segments after the 2nd or 3rd dropped =
segment;
it seems to retransmit 1/2 1 1 1 1/2 segments if a contingeous hole > 1 =
segment
is being announced by SACK... :) But my code still drops the one =
containing the
258th sequence number again, leading even Win7 to a RTO...



And, on another front, I have checked a few systems in the field (our =
gear
Is run typically in high-speed (1/10 Gbps) LANs; I found one example =
where
Nearly 50% of the retransmissions where followed by a RTO, and even the
Less loaded systems showed a quite high number of RTOs (15-35%) after
Retransmissions.

I assume at this point, that only a minority of the RTOs is "legitimate" =
in=20
the sense that=20
*) TCP Session is not running with SACK
*) Client was forcefully removed from the network (loss of connectivity)

Which leaves probably between 70 and 95% or the RTO events as "burst =
loss"
candidates, where keeping the DUPACK detection armed during =
FastRetransmit
would help.


I will see to it, that I get statistically more relevant data, and also
Put this into context (i.e. total segments transmitted per week vs. =
total=20
retransmitted segments per week vs. retransmit timeout events per week).

(Actually, I got scared at first, when I saw that high-load system =
reporting=20
50% of all retransmissions are followed by RTOs... :) ).


Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20
Franz-Klein-Gasse 5
1190 Wien=20


-----Original Message-----
From: Scheffenegger, Richard=20
Sent: Montag, 9. November 2009 18:27
To: Alexander Zimmermann
Cc: tcpm@ietf.org
Subject: Re: [tcpm] Detect Lost Retransmit with SACK


Hi Alexander,=20

Thanks for the welcome :)=20

I fork another thread with the LimitedTransport||FastRecovery / ABC =
interaction...


I will try to sketch up an example to demonstrate what problem I'm =
trying to address:


Let's assume the cwnd is already open for at least 7 segments, before =
the segment with sequence number 10000 is the first one to be dropped by =
the network.

Also, let's assume that FastRetransmit runs from the left edge of the =
leftmost hole
(SND.UNA) upwards, and that per ACK only a single segment is sent.=20
             =20

             Triggering    ACK      Left     Right    Left     Right
             Segment                Edge 1   Edge 1   Edge 2   Edge 2

              9000          9000
             10000  (lost)       *
             11000  (lost)
             12000  (lost)
             13000  (lost)
             14000  (lost)
             15000         10000    15000    16000
             16000         10000    15000    17000
             17000         10000    15000    18000

3 ACKs trigger fast retransmit

             10000  (lost again)=20
             11000         10000    11000    12000    15000    18000
             12000         10000    11000    13000    15000    18000
             13000         10000    11000    14000    15000    18000
-> here we have again 3 ACKs indicating a another loss of one of the =
retransmitted=20
packets. The leftmost hole did not change, while the overall number of =
SACKed
octets did decrease for 3 consecutive ACKs (4; 3 and 2 segments marked =
by SACK).

Current behaviour of investigated TCP Stacks:
             14000         10000    11000    18000
(normal transmit resumes)
             18000         10000    11000    19000
             19000         10000    11000    20000
             20000         10000    11000    21000
             21000         10000    11000    22000
             22000         10000    11000    23000
             ::            ::       ::       ::
Eventually, RTO trips off, retransmitting the lost segment; this happens =
RTO later,
followed by slow-start...

             50000         10000    11000    50000
             ::
             ::
             10000         50000

However, this can be somewhere between 0.2 and 1.0 sec later with a =
"fresh" TCP=20
session (no prior connection properties known (cached) by sender). Most =
likely,
the cwnd has filled up already way sooner (as demonstated, the problem =
seems to be
most prominent in Highspeed LANs), so that for nearly as long, no data =
is actually
transmitted.

         =20

Proposed behaviour:=20

             Triggering    ACK      Left     Right    Left     Right
             Segment                Edge 1   Edge 1   Edge 2   Edge 2

              9000          9000
             10000  (lost)       *
             11000  (lost)
             12000  (lost)
             13000  (lost)
             14000  (lost)
             15000         10000    15000    16000
             16000         10000    15000    17000
             17000         10000    15000    18000

3 ACKs trigger fast retransmit

             10000  (lost again) *
             11000         10000    11000    12000    15000    18000
             12000         10000    11000    13000    15000    18000
             13000         10000    11000    14000    15000    18000

Once the ACK + SACK options indicate that the leftmost hole is not =
shrinking,
while the SACKed octets are increasing (to deal with clients which send =
one=20
retransmission segment and one new segment interspaced, or when multiple =
holes
exist which are being filled, or when network reordering occurs, or when =
some=20
wore segments get lost again):=20

Reset the Rexmit vector to the beginning of the Hole-List (SND.UNA), =
clear
counter to count duplicates (just in case one segment gets lost again =
during
retransmit), and keep the DUPACK detection logic armed...

Also, this reaction should not occur before 1 RTT - so ACKs subsequent =
to the=20
three which indicated the "lost again" segment will take care (in the =
typical case)
that no segments are retransmitted needlessly. ACK processing has to =
occur before
deciding which segment (retransmit / new) to send next. Holes will then =
be marked=20
fully retransmitted, before the 2nd retransmission round would advance =
to them).

             10000         14000    15000      18000
             14000         18000
(normal transmit resumes, but with cwnd shrunk by 2 congestion events)
             18000         19000
             19000         20000


And yes, I was unclear with the use of the terminology; I should have =
probably=20
stated "pipe" instead of "cwnd" below, as cwnd is not touched during =
LimitedTransmit /
FastRetransmit...


Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20
Franz-Klein-Gasse 5
1190 Wien=20


-----Original Message-----
From: Alexander Zimmermann [mailto:zimmermann@nets.rwth-aachen.de]=20
Sent: Montag, 9. November 2009 16:26
To: Scheffenegger, Richard
Cc: tcpm@ietf.org Extensions WG
Subject: Detect Lost Retransmit with SACK

Hi Richard,

firstly welcome on the list :-)
Since your question in not really related to the poll I change the =
title...

Comments inline.


Am 09.11.2009 um 13:57 schrieb Scheffenegger, Richard:

>
>
> Hi Alexander et al.,
>
> This will be the first post to this group, so excuse me if I act=20
> inappropriately.
>
> I'm curious about one little tidbit which has been bugging me for the=20
> better part of the last two monts, and which is closely related with=20
> TCP SACK operations (thus it might belong to this thread?)
>
>
> The implicit assumption for TCP fast recovery is, that packet loss =20
> happens randomly (ie. to different segments each time) with low =20
> correlation between the drop events. Also, a drop event is used as a =20
> implicit signal to indicate congestion. So far, so good.
>
> It seems to me, that the focus of most developments has been the =20
> internet environment - where statistical assumptions like the above =20
> mentioned arguably hold true.
>
> However, certain high-speed LANs seem to exhibit characteristics, =20
> which don't play well with these implicit assumptions (uncorrelated =20
> packet loss) - the smaller the network, the more deviation from an =20
> "good seasoned" link (exhibiting some form of congestion) is likely =20
> to occur.
>
> Also, as has been noted in prior research, many internet routers do =20
> use more "tcp-friendly" RED or WRED queue policies, over the =20
> simplistic TailDrop most often encountered in LANs (default policy =20
> of L2 switches and L3 routers).
>
> In one extreme, I have found a (misbehaving=B4?) TCP stack/host, which =
=20
> sends out a burst of segments (4-6) @ 10GbE wirespeed, which =20
> immediately cause queue buffer overload and TailDrop in the first =20
> hop L2 Switch, when two such high performance hosts try to establish =20
> a high speed communication. With other words, the hosts themselves =20
> seem to make sure that there is a high correlation between TCP =20
> (fast) recovery and further packet loss.
>
>
> But what puzzles me the most - even with SACK enabled TCP stacks, =20
> virtually no implementation can detect / act upon detection of the =20
> loss of a retransmitted segment during fast recovery. This despite =20
> the fact, that the stipulations in RFC3517 requires the receiver to =20
> make the information to detect such an event implicitly available to =20
> the sender. The first SACK option has to reflect the last segment, =20
> which triggered this SACK.
>
> Together with the scoreboard held at the sender, it should be rather =20
> easy to find out if the left edge of the lowest hole (relative to =20
> stream octets) closes.

What do you with "left edge of the lowest hole"? Do you mean SND.UNA? =20
If ACK covers SND.UNA then it is an cumulative ACK.

>
> If that left edge stays constant for "DupThresh" number of ACKs, =20
> which reduce the overall number of octets in holes (any one hole =20
> might close due to the retransmitted packets still received), AND =20
> the sender retransmits beginning with the lowest hole first, this =20
> would be a clear indication of another segment retransmit loss...

Sorry, I don't understand. If we have 20 segments in flight and one =20
segment gets lost, you will retransmit after 3 DUPACKS the oldest =20
outstanding segment.
Then, assuming no reordering and no further lost, you will get 17 =20
DUPACKS (without Limited Transmit) before your hole is closed.

What do I miss here?

Can you give me an example?

>
> Even a less speedy detection logic would work for SACK-enabled =20
> sessions: once the fast recovery is finished from the sender's point =20
> of view, if the receiver still complains about missing segments =20
> (indicated by having the SACK rightmost edge - in the first slot =20
> SACK option - at a segment higher than when fast recovery started), =20
> another round of fast recovery could be invoked, rather than waiting =20
> for RTO.
>
> Of course, the first approach would be better for low cwnd sessions =20
> with only very few segments in transit - and both could be combined =20
> with the proposed sack recovery speed-ups... (Reducing DupThresh for =20
> low cwnd sessions / when little data is being sent).
>
>
>
> Congestion control should act to this event (it will now, but only =20
> one RTO later...), and the SACK retransmit vector (HighRxt) reset, =20
> using LimitedTransmit for sending out the retransmission segments - =20
> once cwnd + pipe allows; any retransmitted segments still in the =20
> network will close their respective SACK holes before the new =20
> HighRxt advances to them.
>
> And, RTO should be reduce (I guess to nearly zero, between SACK-=20
> enabled hosts).
>
>
> I have run numerous tests, to check the behavior of different TCP =20
> Stacks (FreeBSD 4.2 - 8.0; windows xp, vista, 7, 2003; Linux 2.6.16 =20
> and others).
>
>
> All these stacks seem to exhibit this issue; What I don't know yet =20
> is the percentage of multi-loss segement events triggering RTO - but =20
> I assume that the majority of RTOs happen because of this.
>
> In LAN environments (ie. 10 GbE over 1 km @ 2 ms latency due to the =20
> L2 hops in between) featuring relatively few streams, the effect of =20
> any single RTO can be quite tremendeous - taking considerable =20
> theoretical bandwidth away from the session (ie. 1 sec minimum RTO =20
> equals 1.2 GB; even with more recent RTO values around 0.2 - 0.4 =20
> sec, each RTO is still a few hundred MB "lost" capacity under =20
> optimal circumstances.
>
>
> Nevertheless, I cann't imagine that I am the first one to bring up =20
> this issue (despite having failed to find any study of this =20
> effect). :)
>
>
> One more clarification, which came up after I looked at the FreeBSD =20
> implementation of Limited Transmit; this might be a nit-pick, but =20
> when RFC 3042 is active, shouldn't ABC also be used during =20
> LimitedTransmit / FastRecovery?

Why? One reason for ABC are lying receivers (ACK Division). So, the =20
worst case is Slow-Start...

> (FreeBSD MAIN is increasing cwnd by 1 mss for each new ACK, instead =20
> for the amount of data in that ack...

What do you describe here? Slow-Start?
RFC 3042 says: "The congestion window (cwnd) MUST NOT be changed when =20
these new segments are transmitted."

>
> Thanks a lot!
>
>
> Best regards,

Alex

>
>
>
> Richard Scheffenegger
> Field Escalation Engineer
> NetApp Global Support
> NetApp
> +43 1 3676811 3146 Office (2143 3146 - internal)
> +43 676 654 3146 Mobile
> www.netapp.com <BLOCKED::http://www.netapp.com/>
> Franz-Klein-Gasse 5
> 1190 Wien
>
> *	To: "tcpm at ietf.org <mailto:tcpm@DOMAIN.HIDDEN>  WG Extensions" =20
> <tcpm at ietf.org <mailto:tcpm@DOMAIN.HIDDEN> >
> *	Subject: [tcpm] Should draft-ietf-tcpm-sack-recovery-entry update =20
> RFC 3717 (SACK-TCP)
> *	From: Alexander Zimmermann <alexander.zimmermann at =
nets.rwth-aachen.de=20
>  <mailto:alexander.zimmermann@DOMAIN.HIDDEN> >
> *	Date: Wed, 21 Oct 2009 12:22:50 +0200
>
>  _____
>
> Hi folks,
>
> based on the fact that the draft "draft-ietf-tcpm-sack-recovery-=20
> entry" is adopted as WG item now and intended to be a "standards =20
> track" document, I would like to start a poll/discussion whether the =20
> draft should update RFC 3517 or not? Moreover, should we produce a =20
> separate document or an update of RFC 3517?
>
> a) separate document, do not update RFC 3517
> b) separate document, update RFC 3517
> c) RFC3517bis, obsolete RFC 3517
>
> //
> // Dipl.-Inform. Alexander Zimmermann
> // Department of Computer Science, Informatik 4
> // RWTH Aachen University
> // Ahornstr. 55, 52056 Aachen, Germany
> // phone: (49-241) 80-21422, fax: (49-241) 80-22220
> // email: zimmermann at cs.rwth-aachen.de
> // web: http://www.umic-mesh.net
> //
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//

_______________________________________________
tcpm mailing list
tcpm@ietf.org
https://www.ietf.org/mailman/listinfo/tcpm

From rs@netapp.com  Mon Nov  9 11:39:48 2009
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Thread-Topic: [tcpm] Should draft-ietf-tcpm-sack-recovery-entry update RFC 3717 (SACK-TCP)
Thread-Index: AcphUFwankudnmboRwSABGd1Or8yBAAF0r2g
From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Arnd Hannemann" <hannemann@i4.informatik.rwth-aachen.de>
X-OriginalArrivalTime: 09 Nov 2009 19:39:47.0346 (UTC) FILETIME=[65150F20:01CA6174]
Cc: tcpm@ietf.org
Subject: Re: [tcpm] Should draft-ietf-tcpm-sack-recovery-entry update RFC 3717 (SACK-TCP)
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Hello Arnd,

Thanks for your comment; unfortunately, I'm not supposed to get
inspiration from GPL'ed code :)=20


Do you know which kernel versions feature this? I've checked 2.6.16 and=20
2.6.18 (commercial distributions), but none of the most recent ones...

FACK as described here
http://www.psc.edu/networking/papers/FACKnotes/current/
seems to be a sender-only algorithm; Thus if the receiver is
implementing
SACK, the sender should be able to use it, right?

I just rechecked - the trace does show the hallmark signs of FACK, but
no sign of
a retransmission of a lost retransmitted segment before a
(comparatively) long
Period (assuingly RTO) starting a limited slow start (as the slow start
ACKs
Are still SACKs for a large sequence space)...

(Using CentOS53, kernel 2.6.18 in this quick test).

Hmm... Running a 2nd check (this time with cached session parameters),
the
Linux Kernel seems to do a 2nd sack recovery - even though not optimally
so
(not at the very first possibility w/ 3 SACK-ACKs indicating that it=20
would be possible...). The first trace appears to have run into a RTO
with
The very first dropped segment (and the stream took ages...)

3rd test: 237 msec delay (RTO?), with the last new segment being set 36
ms after
the 2nd drop (so, nearly exactly 200 ms after sending the last new
segment,
And receiveing about 80 uninterrupted ACKs...

So, the linux stack seems to have the capability, but it's not reliably
invoked...

Thanks for pointing this out!

Regarding the other two question - for the first one, please see
The diagram I sent earlier; I hope this clarifies what I mean.

If I am mistaken somewhere, please let me know...


And with the last statement (fastretransmission finished) I referred
To the point in time, when the sender has sent all segments being
flagged
By SACK to have been missing, before noticing (one RTT later) that
another
One might be missing; this might be a special case, though, as I assumed
That retransmission would end prior to the receiver detecting the=20
2nd loss...

I guess a time-flow diagram would help :)

Best regards,


Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20
Franz-Klein-Gasse 5
1190 Wien=20


-----Original Message-----
From: Arnd Hannemann [mailto:hannemann@i4.informatik.rwth-aachen.de]=20
Sent: Montag, 9. November 2009 16:21
To: Scheffenegger, Richard
Cc: tcpm@ietf.org
Subject: Re: [tcpm] Should draft-ietf-tcpm-sack-recovery-entry update
RFC 3717 (SACK-TCP)

Scheffenegger, Richard schrieb:

> But what puzzles me the most - even with SACK enabled TCP stacks,=20
> virtually no implementation can detect / act upon detection of the=20
> loss of a retransmitted segment during fast recovery. This despite the

> fact, that the stipulations in
> RFC3517 requires the receiver to make the information to detect such=20
> an event implicitly available to the sender. The first SACK option has

> to reflect the last segment, which triggered this SACK.

Some implementations actually do detect lost retransmissions.
(see Linux tcp_mark_lost_retrans())
However, currently only for FACK enabled flows. (Comments indicate, it
would be non-trivial / too-expensive for SACK enabled flows.)

> Together with the scoreboard held at the sender, it should be rather=20
> easy to find out if the left edge of the lowest hole (relative to=20
> stream octets) closes

> If that left edge stays constant for "DupThresh" number of ACKs, which

> reduce the overall number of octets in holes (any one hole might close

> due to the retransmitted packets still received), AND the sender=20
> retransmits beginning with the lowest hole first, this would be a=20
> clear indication of another segment retransmit loss...

Sorry, I don't understand. You mean "DupThresh" number of DupACKs?
Sack-blocks?
After which point in time? After the first retransmit?
SACK-blocks higher than recovery point? (the latter would make most
sense, IMO)

> Even a less speedy detection logic would work for SACK-enabled=20
> sessions: once the fast recovery is finished from the sender's point=20
> of view, if the receiver still complains about missing segments=20
> (indicated by having the SACK rightmost edge - in the first slot SACK=20
> option - at a segment higher than when fast recovery started), another

> round of fast recovery could be invoked, rather than waiting for RTO.

How can fast recovery be "finished" if the sender is missing packets?

Best regards,
Arnd Hannemann


From andrew.knutsen@bluecoat.com  Mon Nov  9 11:50:23 2009
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Subject: [tcpm] New Version Notification for draft-knutsen-tcpm-middlebox-discovery-03
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    This new version includes the changes I mentioned in my last email, 
which didn't make it in before the meeting deadline.

    We appreciate your attention to this draft.  We'd like to move it to 
the next step ASAP.

Andrew

A new version of I-D, draft-knutsen-tcpm-middlebox-discovery-03.txt has 
been successfuly submitted by Andrew Knutsen and posted to the IETF 
repository.

Filename:	 draft-knutsen-tcpm-middlebox-discovery
Revision:	 03
Title:		 TCP Option for Transparent Middlebox Discovery
Creation_date:	 2009-11-09
WG ID:		 Independent Submission
Number_of_pages: 10

Abstract:
This document describes a TCP option intended to facilitate
transparent detection of middleboxes (or services playing that role)
along the path of a TCP connection as the connection is made. The
option has no effect if an appropriate middlebox is not on the path.
 



The IETF Secretariat.



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Folks-

> We believe we have addressed all comments we received on the early
> retranmsit ID.  I just went to submit a new version and have missed the
> cut-off.  I have set a bit to submit when the system opens again on
> Nov/9.  However, until then I put the version I was going to submit here
> ...
> 
>   http://www.icir.org/mallman/papers/draft-ietf-tcpm-early-rexmt-02a.txt
> 
> Comments are very welcome.

Since TCPM is not meeting this week Lars got this draft approved for
posting.  So, it is now at the usual place (as "-02" not "-02a).

Our hit is that it seems finished.  Please yell if there is more to do.

allman




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From rs@netapp.com  Mon Nov  9 16:09:09 2009
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From: "Scheffenegger, Richard" <rs@netapp.com>
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Hello Carsten,

This is a very good question indeed.

Keeping the ACK clock running, and avoiding bursts of segments delivered
all at once in a single burst to a L2 LAN network is something
where my interests are too... (as todays hosts are increasingly capable
of oversaturating slightly dated network equipment with full wire speed
traffic).

Is there a procedure to revive / adopt a dead draft?

I have not spent much time on this issue, but noted the burst behavior
After fastretransmit.


As RateHalving extended the scope of Limited Transmit (with the similar
goal), shouldn't this be discussed for inclusion in a RFC 3042bis?

Since certain other congestion algorithms have a beta of less than 0.5=20
(ie. BIC, CUBIC use 0.8) when selecting a new ssthresh, perhaps the
effect
of RateHalfing, but with 2 out of 3 (0.66) policy or 3 out of 4 (0.75)
at the end of the RTT should be investigated? (To be less conservative
When reducing the sending rate).

One extension to the draft might be the discussion on how to deal with
RFC 3465 (ABC) during rate halving; with SACK, ABC could still be
employed,
And a new segment triggered only, when 2*MSS has been covered by the new
SACKs...

Regards,

Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com <BLOCKED::http://www.netapp.com/> =20
Franz-Klein-Gasse 5
1190 Wien=20

*	To: Matt Mathis <mathis at psc.edu <mailto:mathis@DOMAIN.HIDDEN>
>
*	Subject: [tcpm] Curiosity about draft-mathis-tcp-ratehalving
*	From: Carsten Wolff <carsten.wolff at rwth-aachen.de
<mailto:carsten.wolff@DOMAIN.HIDDEN> >
*	Date: Wed, 4 Nov 2009 13:48:51 +0100
  _____ =20

Hello Matt, everyone,

I'm curious about draft-mathis-tcp-ratehalving=20
(http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt) and why
it=20
never exceeded version 00 and expired years ago. It seems, several of
the=20
ideas (fack, ratehalving, the concept of rhcwnd) in this paper made it
into=20
the Linux TCP implementation in some form, which means they're part of a
big=20
chunk of today's web servers and other Internet hosts. Why has there
been no=20
interest in taking this draft further?

Cheers,
Carsten


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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 Maintenance and Minor Extensions Working Group of the IETF.


	Title           : On the implementation of the TCP urgent mechanism
	Author(s)       : F. Gont, A. Yourtchenko
	Filename        : draft-ietf-tcpm-urgent-data-01.txt
	Pages           : 13
	Date            : 2009-11-09

This document analyzes how current TCP implementations process TCP
urgent indications, and how the behavior of some widely-deployed
middle-boxes affect how urgent indications are processed by end
systems.  This document updates the relevant specifications such that
they accommodate current practice in processing TCP urgent
indications, provides advice to applications that make use of the
urgent mechanism, and raises awareness about the reliability of TCP
urgent indications in the current Internet.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-urgent-data-01.txt

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From alexander.zimmermann@nets.rwth-aachen.de  Tue Nov 10 00:39:46 2009
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From: Alexander Zimmermann <alexander.zimmermann@nets.rwth-aachen.de>
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Hi Mark,

give me one or two days more to look over the document again...

Alex

Am 09.11.2009 um 21:50 schrieb Mark Allman:

> 
> Folks-
> 
>> We believe we have addressed all comments we received on the early
>> retranmsit ID.  I just went to submit a new version and have missed the
>> cut-off.  I have set a bit to submit when the system opens again on
>> Nov/9.  However, until then I put the version I was going to submit here
>> ...
>> 
>>  http://www.icir.org/mallman/papers/draft-ietf-tcpm-early-rexmt-02a.txt
>> 
>> Comments are very welcome.
> 
> Since TCPM is not meeting this week Lars got this draft approved for
> posting.  So, it is now at the usual place (as "-02" not "-02a).
> 
> Our hit is that it seems finished.  Please yell if there is more to do.
> 
> allman
> 
> 
> 
> <ATT00001>_______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


From hannemann@i4.informatik.rwth-aachen.de  Tue Nov 10 04:03:41 2009
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Cc: Alexander Zimmermann <Alexander.Zimmermann@nets.rwth-aachen.de>, tcpm@ietf.org
Subject: [tcpm] Detect Lost Retransmit mit SACK
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Hi,

Scheffenegger, Richard schrieb:

> Thanks for your comment; unfortunately, I'm not supposed to get
> inspiration from GPL'ed code :) 
> 
> 
> Do you know which kernel versions feature this? I've checked 2.6.16 and 
> 2.6.18 (commercial distributions), but none of the most recent ones...

At least 2.6.24 (Ubuntu 8.04 LTS) and newer have this.

> 
> FACK as described here
> http://www.psc.edu/networking/papers/FACKnotes/current/
> seems to be a sender-only algorithm; Thus if the receiver is
> implementing
> SACK, the sender should be able to use it, right?

As far as I remember the advantage of FACK for detecting lost
retransmissions is, that you can be sure that after invoking
"Fast Retransmit" you retransmitted all holes up to RecoveryPoint
before sending any new data. Now you can use any ACK containing
a SACK block above RecoveryPoint for an indication of a
lost retransmission. But note, that FACK has serious problems
with reordering.

In my opinion SACK-TCP you would need to track a lot more.
Thinking of it, maybe it would be enough to store highest seq
sent so-far for each retransmit. And then check this for incoming
SACKs individually for each retransmit. Might be still too expensive...

[Shortened E-mail]

> 
> And with the last statement (fastretransmission finished) I referred
> To the point in time, when the sender has sent all segments being
> flagged
> By SACK to have been missing, before noticing (one RTT later) that
> another
> One might be missing; this might be a special case, though, as I assumed
> That retransmission would end prior to the receiver detecting the 
> 2nd loss...

Don't forget that with SACK-TCP the TCP sender will mix retransmissions
and new segments all the time during recovery.
The interesting part is: How does the TCP sender detect, that a
retransmission might be missing? Simply checking against RecoveryPoint
would not work... Or only for the "first hole".

> 
> I guess a time-flow diagram would help :)

Yes please!

Best regards,
Arnd Hannemann


From alexander.zimmermann@nets.rwth-aachen.de  Tue Nov 10 04:50:06 2009
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From: Alexander Zimmermann <alexander.zimmermann@nets.rwth-aachen.de>
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Subject: Re: [tcpm] Detect Lost Retransmit with SACK
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Hi Richard,

I discussed your example with Arnd (he is a line-of-sight colleague). Your "algorithms"
may workwhen you have only one bust lost per cwnd. If you have multiple non-burst
loss (e.g. WLAN), IMHO, is doesn't work.

Am 09.11.2009 um 18:27 schrieb Scheffenegger, Richard:

> 
> Hi Alexander,
> 
> Thanks for the welcome :)
> 
> I fork another thread with the LimitedTransport||FastRecovery / ABC interaction...
> 
> 
> I will try to sketch up an example to demonstrate what problem I'm trying to address:
> 
> 
> Let's assume the cwnd is already open for at least 7 segments, before the segment with
> sequence number 10000 is the first one to be dropped by the network.
> 
> Also, let's assume that FastRetransmit runs from the left edge of the leftmost hole
> (SND.UNA) upwards, and that per ACK only a single segment is sent.
> 
> 
>             Triggering    ACK      Left     Right    Left     Right
>             Segment                Edge 1   Edge 1   Edge 2   Edge 2
> 
>              9000          9000
>             10000  (lost)       *
>             11000  (lost)
>             12000  (lost)
>             13000  (lost)
>             14000  (lost)
>             15000         10000    15000    16000
>             16000         10000    15000    17000
>             17000         10000    15000    18000

Ok, I count 9 segments ;-) Anyway, your example is a little bit strange.
You assume you send 9 segment in a burst. Then your ACK for 9000 will trigger
the segment 18000. Then the 2 DUPACKs  for 15000 and 16000 will trigger 19000,
20000 respectively (Limited Transmit). The third DUPACK will trigger the Fast
Retransmit 10000. Since NextSeq() and pipe allow you retransmit 11000, 12000.

At this point you have to wait since the pipe is full (if I calculate pipe in a rush correctly).
A RTT later you will get 3 Dupacks, even if the 10000 is not lost.

OK, I think you can adjust your example so that it works for the given case, however in
other scenarios (multiple loss, reordering, packet duplication,...) it will be much more
complicated.

I suggest writing examples as Ilpo does in
http://tools.ietf.org/html/draft-ietf-tcpm-sack-recovery-entry-00.
It's much more easier to read.

> 
> 3 ACKs trigger fast retransmit
> 
>             10000  (lost again)
>             11000         10000    11000    12000    15000    18000
>             12000         10000    11000    13000    15000    18000
>             13000         10000    11000    14000    15000    18000
> -> here we have again 3 ACKs indicating a another loss of one of the retransmitted
> packets. The leftmost hole did not change, while the overall number of SACKed
> octets did decrease for 3 consecutive ACKs (4; 3 and 2 segments marked by SACK).
> 

Alex

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


From rs@netapp.com  Tue Nov 10 05:33:00 2009
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Thread-Topic: Detect Lost Retransmit mit SACK
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Arnd Hannemann" <hannemann@i4.informatik.rwth-aachen.de>
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Subject: Re: [tcpm] Detect Lost Retransmit mit SACK
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Hi Arnd,

Let's think about what would be required to track a lost retransmitted
segment.

A) the sender can only know about this one RTT after it send out the
first=20
   retransmission at the earliest.

  A1) With TSOpt, they could be used to "remember" when Retransmission
started;=20
      When an ACKs with TSEcn > (TSOpt + RTT) is seen by the sender, it
can=20
      re-arm the DUPACK detector.

  A2) With SACK, tracking SND.NXT and HOLEBYTES (number of octets in all
current holes)=20
      can be employed to track RTT (they need to be initialized when the
first=20
      retransmitted segment is sent). When an ACK contains a SACK for =
>=3D
SND.NXT, or the=20
      HOLEBYTES are smaller than when retransmission started, the DUPACK
detector can=20
      be re-armed.

  A3) If neither TSOpt nor SACK is used, revert to current behavior (RTO
timeout to
      detect a lost retransmit).

B) DUPACK detector would need to be enhanced over what is currently in
the RFCs (but=20
   I think most of that is already in Drafts, ie.=20
   http://tools.ietf.org/html/draft-jarvinen-tcpm-sack-recovery-entry-01
).

 B1) With SACK enabled, fully duplicate ACKs (ACK and SACK have been
seen before, and=20
     prev_HOLEBYTES =3D=3D HOLEBYTES) can be discarded; lost ACKs would
count for two=20
     (prev_HOLEBYTES < HOLEBYTES - SMSS).

 B2) With TSOpt, if another RTT passes without advancing SND.UNA,=20
     increase DupAcks +=3D DupThresh=20

 When DupThresh # of ACKs are received, reset HighRxt =3D SND.UNA, =
DupAcks
=3D 0; *)


There are two comments I would like to make:

A) Latency: if the sender first sends out the entire range of holes,
before
   reacting to a detectable lost retransmitted packet, the latency
observed
   by the application on the receiver side will increase needlessly. It
might
   take a number of RTTs under certain circumstances, before all the
holes
   have been retransmitted once...=20

B) Complexity: As soon as any congestion event is encountered, TCP won't
be in
   fastpath mode any longer; the added computational complexity to
(reliably, see my
   check on FACK with Linux 2.6.18 yesterday) trigger another
retransmission
   of an already retransmitted segment is neglegible, compared to
current best
   practise (waiting for RTO), when you need to move data around as
quickly (low
   latency AND high bandwidth) as possible. As mentioned, my background
is not
   so much the global internet (with statistically valid implict
assumptions), but
   rather High-Speed, short Range, (lossy) LANs, with only very few
active TCP
   sessions at any one time. The hosts I deal with might have as few as
10-20 TCP
   sessions, of which only 1-4 are pushing data, but those might run
over dedicated
   10GbE ports (overloading L2 switches in the process, and exhibiting
Burst Loss
   (correlated packet loss) ).


Thus my view might deviate from a "typical" view in certain aspects. I
might even be=20
in error and would like to hear, why some things cann't work like
depicted. However,=20
I would place more emphasis on the timely recovery  of any lost (or
re-lost) segment,=20
over making sure that no single segment is re-transmitted needlessly;
better retransmit=20
a little (!) too much too soon, instead of waiting for RTO :) I want to
point out the=20
"little" though, as overwhelming a already lossy network with too much
retransmissions=20
(ie. Setting RTO =3D 0/1 tick, with a TCP clock runnig in msec), will =
only
cause more loss...






*) reducing the segment size for this generation of retransmission might
also be=20
advisable; I have seen odd gear (but that's actually a long time ago),
which simply=20
would choke on certain bit-patterns. Reducing the mss might break such
bit/byte=20
patterns, so that the always-lost segment now looks differently and
makes it through...=20

But really, this shouldn't be necessary any more, I think.



Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20
Franz-Klein-Gasse 5
1190 Wien=20


-----Original Message-----
From: Arnd Hannemann [mailto:hannemann@i4.informatik.rwth-aachen.de]=20
Sent: Dienstag, 10. November 2009 13:04
To: Scheffenegger, Richard
Cc: tcpm@ietf.org; Alexander Zimmermann
Subject: Detect Lost Retransmit mit SACK

Hi,

Scheffenegger, Richard schrieb:

> Thanks for your comment; unfortunately, I'm not supposed to get
> inspiration from GPL'ed code :)=20
>=20
>=20
> Do you know which kernel versions feature this? I've checked 2.6.16
and=20
> 2.6.18 (commercial distributions), but none of the most recent ones...

At least 2.6.24 (Ubuntu 8.04 LTS) and newer have this.

>=20
> FACK as described here
> http://www.psc.edu/networking/papers/FACKnotes/current/
> seems to be a sender-only algorithm; Thus if the receiver is
> implementing
> SACK, the sender should be able to use it, right?

As far as I remember the advantage of FACK for detecting lost
retransmissions is, that you can be sure that after invoking
"Fast Retransmit" you retransmitted all holes up to RecoveryPoint
before sending any new data. Now you can use any ACK containing
a SACK block above RecoveryPoint for an indication of a
lost retransmission. But note, that FACK has serious problems
with reordering.

In my opinion SACK-TCP you would need to track a lot more.
Thinking of it, maybe it would be enough to store highest seq
sent so-far for each retransmit. And then check this for incoming
SACKs individually for each retransmit. Might be still too expensive...

[Shortened E-mail]

>=20
> And with the last statement (fastretransmission finished) I referred
> To the point in time, when the sender has sent all segments being
> flagged
> By SACK to have been missing, before noticing (one RTT later) that
> another
> One might be missing; this might be a special case, though, as I
assumed
> That retransmission would end prior to the receiver detecting the=20
> 2nd loss...

Don't forget that with SACK-TCP the TCP sender will mix retransmissions
and new segments all the time during recovery.
The interesting part is: How does the TCP sender detect, that a
retransmission might be missing? Simply checking against RecoveryPoint
would not work... Or only for the "first hole".

>=20
> I guess a time-flow diagram would help :)

Yes please!

Best regards,
Arnd Hannemann


From rs@netapp.com  Tue Nov 10 06:34:21 2009
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Thanks Alex,

I will try to give my example in another form: (RWND =3D 40000;=20
the timing might not be perfectly depicted).

      ACK         Transmitted       Received    ACK Sent
      Received    Segment           Segment     (Including SACK Blocks)
=09

                      0-  999
                   1000- 1999
						    0-  999=09
								 1000
	 1000		                   1000- 1999
			 2000- 2999                    2000
	 2000        3000- 3999
			 4000- 4999		 2000- 2999
	             5000- 5999				 3000
	 3000		 6000- 6999		 3000- 3999
			 7000- 7999				 4000
	 4000		 8000- 8999		 4000- 4999
			 9000- 9999				 5000
	 5000		10000-10999		 5000- 5999
			11000-11999				 6000
	 6000		12000-12999		 6000- 6999
			13000-13999				 7000
	 7000					 7000- 7999
			14000-14999				 8000
	 8000					 8000- 8999
			15000-15999				 9000
	 9000					 9000-10000
			16000-16999				10000
	10000					(dropped)
			17000-17999
						(dropped)
			.		=09
						(dropped)
			.		=09
						(dropped)
			.
						(dropped)
			.=09
						15000-15999
								10000,
SACK=3D15k-16k
	10000, SACK=3D15k-16k		16000-16999
			18000-18999				10000,
SACK=3D15k-17k
	10000, SACK=3D15k-17k		17000-17999
			19000-19999				10000,
SACK=3D15k-18k
	10000, SACK=3D15k-18k		.
                  10000-10999
						18000-18999
								10000,
SACK=3D15k-19k
	10000, SACK=3D15k-19k		19000-19999
			11000-11999				10000,
SACK=3D15k-20k
	10000, SACK=3D15k-20k		(dropped)
			12000-12999	=09
						11000-11999
								10000,
SACK=3D11k-12k;15k-20k
	10000, SACK=3D11k-12k;15k-20k	12000-12999
			13000-13999				10000,
SACK=3D11k-13k;15k-20k
	10000, SACK=3D11k-13k;15k-20k
			14000-14999		13000-13999
								10000,
SACK=3D11k-14k;15k-20k
	10000, SACK=3D11k-14k;15k-20k	14000-14999
!*A			20000-20999				10000,
SACK=3D11k-20k
!	10000, SACK=3D11k-20k
!			21000-21999		20000-20999
!			22000-22999				10000,
SACK=3D11k-21k
!	10000, SACK=3D11k-21k
!			23000-23999		21000-21999
!			24000-24999				10000,
SACK=3D11k-22k
!	10000, SACK=3D11k-22k		22000-22999
!			25000-25999				10000,
SACK=3D11k-23k
!	10000, SACK=3D11k-23k		23000-23999
!*B			26000-26999				10000,
SACK=3D11k-24k
!     ::          ::                ::          ::
!RWND Full:                                     10000, SACK=3D11k-50k
!     10000, SACK=3D11k-50k
!    =20
!     ::          ::                ::          ::
!
!RTO:
!                 10000-10999
!								50000
!Slow-Start                                              =20


All the Lines marked with ! Indicate current best practise behaviour=20
(RFC, no drafts), if I'm not mistaken - left out LimitedTransmit for=20
simplicity's sake.

At point *A (or one ACK later), the sender would have the earliest=20
possibility to detect a lost retransmission, taking into account the=20
usual 3 ACK reordering hold-down... In this example, this happens by=20
coincidence at the same time, that the sender has finished fast=20
retransmission (and would go into fastrecovery, restoring CWND, etc;=20
this is NOT a requirement....; Actually, the cwnd is likely to be in
the order of 100reds of segments, and most of the time, the sender
will have finished the retransmission episode before one RTT is up)

At point *B, the sender could detect with 100% certainty (2*RTT) that=20
one retransmitted segment was lost.

However, current practise (excluding Linux with FACK for now, as=20
that's not in RFCs) is to continue sending SND.NXT, until RWND is=20
full or RTO expires...

Note that the suggested algorithm will never trigger if no=20
retransmitted packet is lost - if would behave exactly the same as=20
currently.

Only when *A or *B marks are detected, the DUPACK detection logic=20
would re-arm (a more complex implementation could re-arm at the first=20
sign of retransmission loss, and dis-arm if a ACK within DupAck=20
distance ACKs the segment, for which it armed before).

Thus, a lost retransmission would be retried close to the earliest=20
possibility, instead of waiting until RTO (a bit like what FACK seems=20
to try to do, but with low reliability as it seems).

Multiple Burst loss events, each lossing a different segment in=20
one cwnd would be handled by SACK.

If there are extended periods of time, where no communication is=20
possible (the same segment doesn't get lost only twice, but multiple=20
times), RTO would eventually fire and, using the RTO backoff=20
algorithm, retry at ever increasing intervals with very low=20
(1-2 segments) rate...


Do you have a test bed, where you can deliberately drop the same
segment twice (or n times) and check the TCP Behaviour for yourself?

Best regards,



Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20
Franz-Klein-Gasse 5
1190 Wien=20


-----Original Message-----
From: Alexander Zimmermann
[mailto:alexander.zimmermann@nets.rwth-aachen.de]=20
Sent: Dienstag, 10. November 2009 13:51
To: Scheffenegger, Richard
Cc: tcpm@ietf.org
Subject: Re: Detect Lost Retransmit with SACK

Hi Richard,

I discussed your example with Arnd (he is a line-of-sight colleague).
Your "algorithms"
may workwhen you have only one bust lost per cwnd. If you have multiple
non-burst
loss (e.g. WLAN), IMHO, is doesn't work.

Am 09.11.2009 um 18:27 schrieb Scheffenegger, Richard:

>=20
> Hi Alexander,
>=20
> Thanks for the welcome :)
>=20
> I fork another thread with the LimitedTransport||FastRecovery / ABC
interaction...
>=20
>=20
> I will try to sketch up an example to demonstrate what problem I'm
trying to address:
>=20
>=20
> Let's assume the cwnd is already open for at least 7 segments, before
the segment with
> sequence number 10000 is the first one to be dropped by the network.
>=20
> Also, let's assume that FastRetransmit runs from the left edge of the
leftmost hole
> (SND.UNA) upwards, and that per ACK only a single segment is sent.
>=20
>=20
>             Triggering    ACK      Left     Right    Left     Right
>             Segment                Edge 1   Edge 1   Edge 2   Edge 2
>=20
>              9000          9000
>             10000  (lost)       *
>             11000  (lost)
>             12000  (lost)
>             13000  (lost)
>             14000  (lost)
>             15000         10000    15000    16000
>             16000         10000    15000    17000
>             17000         10000    15000    18000

Ok, I count 9 segments ;-) Anyway, your example is a little bit strange.
You assume you send 9 segment in a burst. Then your ACK for 9000 will
trigger
the segment 18000. Then the 2 DUPACKs  for 15000 and 16000 will trigger
19000,
20000 respectively (Limited Transmit). The third DUPACK will trigger the
Fast
Retransmit 10000. Since NextSeq() and pipe allow you retransmit 11000,
12000.

At this point you have to wait since the pipe is full (if I calculate
pipe in a rush correctly).
A RTT later you will get 3 Dupacks, even if the 10000 is not lost.

OK, I think you can adjust your example so that it works for the given
case, however in
other scenarios (multiple loss, reordering, packet duplication,...) it
will be much more
complicated.

I suggest writing examples as Ilpo does in
http://tools.ietf.org/html/draft-ietf-tcpm-sack-recovery-entry-00.
It's much more easier to read.

>=20
> 3 ACKs trigger fast retransmit
>=20
>             10000  (lost again)
>             11000         10000    11000    12000    15000    18000
>             12000         10000    11000    13000    15000    18000
>             13000         10000    11000    14000    15000    18000
> -> here we have again 3 ACKs indicating a another loss of one of the
retransmitted
> packets. The leftmost hole did not change, while the overall number of
SACKed
> octets did decrease for 3 consecutive ACKs (4; 3 and 2 segments marked
by SACK).
>=20

Alex

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


From Alexander.Zimmermann@nets.rwth-aachen.de  Tue Nov 10 07:13:21 2009
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From: Alexander Zimmermann <Alexander.Zimmermann@nets.rwth-aachen.de>
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To: "Scheffenegger, Richard" <rs@netapp.com>
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Subject: Re: [tcpm] Detect Lost Retransmit with SACK
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Hi Richard,

thanks for the updated example. However, since your mailer is a little bit too
eager in formatting emails, could you send your example as an attached file?

Alex

Am 10.11.2009 um 15:34 schrieb Scheffenegger, Richard:

> 
> Thanks Alex,
> 
> I will try to give my example in another form: (RWND = 40000; 
> the timing might not be perfectly depicted).
> 
>      ACK         Transmitted       Received    ACK Sent
>      Received    Segment           Segment     (Including SACK Blocks)
> 	
> 
>                      0-  999
>                   1000- 1999
> 						    0-  999	
> 								 1000
> 	 1000		                   1000- 1999
> 			 2000- 2999                    2000
> 	 2000        3000- 3999
> 			 4000- 4999		 2000- 2999
> 	             5000- 5999				 3000
> 	 3000		 6000- 6999		 3000- 3999
> 			 7000- 7999				 4000
> 	 4000		 8000- 8999		 4000- 4999
> 			 9000- 9999				 5000
> 	 5000		10000-10999		 5000- 5999
> 			11000-11999				 6000
> 	 6000		12000-12999		 6000- 6999
> 			13000-13999				 7000
> 	 7000					 7000- 7999
> 			14000-14999				 8000
> 	 8000					 8000- 8999
> 			15000-15999				 9000
> 	 9000					 9000-10000
> 			16000-16999				10000
> 	10000					(dropped)
> 			17000-17999
> 						(dropped)
> 			.			
> 						(dropped)
> 			.			
> 						(dropped)
> 			.
> 						(dropped)
> 			.	
> 						15000-15999
> 								10000,
> SACK=15k-16k
> 	10000, SACK=15k-16k		16000-16999
> 			18000-18999				10000,
> SACK=15k-17k
> 	10000, SACK=15k-17k		17000-17999
> 			19000-19999				10000,
> SACK=15k-18k
> 	10000, SACK=15k-18k		.
>                  10000-10999
> 						18000-18999
> 								10000,
> SACK=15k-19k
> 	10000, SACK=15k-19k		19000-19999
> 			11000-11999				10000,
> SACK=15k-20k
> 	10000, SACK=15k-20k		(dropped)
> 			12000-12999		
> 						11000-11999
> 								10000,
> SACK=11k-12k;15k-20k
> 	10000, SACK=11k-12k;15k-20k	12000-12999
> 			13000-13999				10000,
> SACK=11k-13k;15k-20k
> 	10000, SACK=11k-13k;15k-20k
> 			14000-14999		13000-13999
> 								10000,
> SACK=11k-14k;15k-20k
> 	10000, SACK=11k-14k;15k-20k	14000-14999
> !*A			20000-20999				10000,
> SACK=11k-20k
> !	10000, SACK=11k-20k
> !			21000-21999		20000-20999
> !			22000-22999				10000,
> SACK=11k-21k
> !	10000, SACK=11k-21k
> !			23000-23999		21000-21999
> !			24000-24999				10000,
> SACK=11k-22k
> !	10000, SACK=11k-22k		22000-22999
> !			25000-25999				10000,
> SACK=11k-23k
> !	10000, SACK=11k-23k		23000-23999
> !*B			26000-26999				10000,
> SACK=11k-24k
> !     ::          ::                ::          ::
> !RWND Full:                                     10000, SACK=11k-50k
> !     10000, SACK=11k-50k
> !     
> !     ::          ::                ::          ::
> !
> !RTO:
> !                 10000-10999
> !								50000
> !Slow-Start                                               
> 
> 
> All the Lines marked with ! Indicate current best practise behaviour 
> (RFC, no drafts), if I'm not mistaken - left out LimitedTransmit for 
> simplicity's sake.
> 
> At point *A (or one ACK later), the sender would have the earliest 
> possibility to detect a lost retransmission, taking into account the 
> usual 3 ACK reordering hold-down... In this example, this happens by 
> coincidence at the same time, that the sender has finished fast 
> retransmission (and would go into fastrecovery, restoring CWND, etc; 
> this is NOT a requirement....; Actually, the cwnd is likely to be in
> the order of 100reds of segments, and most of the time, the sender
> will have finished the retransmission episode before one RTT is up)
> 
> At point *B, the sender could detect with 100% certainty (2*RTT) that 
> one retransmitted segment was lost.
> 
> However, current practise (excluding Linux with FACK for now, as 
> that's not in RFCs) is to continue sending SND.NXT, until RWND is 
> full or RTO expires...
> 
> Note that the suggested algorithm will never trigger if no 
> retransmitted packet is lost - if would behave exactly the same as 
> currently.
> 
> Only when *A or *B marks are detected, the DUPACK detection logic 
> would re-arm (a more complex implementation could re-arm at the first 
> sign of retransmission loss, and dis-arm if a ACK within DupAck 
> distance ACKs the segment, for which it armed before).
> 
> Thus, a lost retransmission would be retried close to the earliest 
> possibility, instead of waiting until RTO (a bit like what FACK seems 
> to try to do, but with low reliability as it seems).
> 
> Multiple Burst loss events, each lossing a different segment in 
> one cwnd would be handled by SACK.
> 
> If there are extended periods of time, where no communication is 
> possible (the same segment doesn't get lost only twice, but multiple 
> times), RTO would eventually fire and, using the RTO backoff 
> algorithm, retry at ever increasing intervals with very low 
> (1-2 segments) rate...
> 
> 
> Do you have a test bed, where you can deliberately drop the same
> segment twice (or n times) and check the TCP Behaviour for yourself?
> 
> Best regards,
> 
> 
> 
> Richard Scheffenegger
> Field Escalation Engineer
> NetApp Global Support 
> NetApp
> +43 1 3676811 3146 Office (2143 3146 - internal)
> +43 676 654 3146 Mobile
> www.netapp.com 
> Franz-Klein-Gasse 5
> 1190 Wien 
> 
> 
> -----Original Message-----
> From: Alexander Zimmermann
> [mailto:alexander.zimmermann@nets.rwth-aachen.de] 
> Sent: Dienstag, 10. November 2009 13:51
> To: Scheffenegger, Richard
> Cc: tcpm@ietf.org
> Subject: Re: Detect Lost Retransmit with SACK
> 
> Hi Richard,
> 
> I discussed your example with Arnd (he is a line-of-sight colleague).
> Your "algorithms"
> may workwhen you have only one bust lost per cwnd. If you have multiple
> non-burst
> loss (e.g. WLAN), IMHO, is doesn't work.
> 
> Am 09.11.2009 um 18:27 schrieb Scheffenegger, Richard:
> 
>> 
>> Hi Alexander,
>> 
>> Thanks for the welcome :)
>> 
>> I fork another thread with the LimitedTransport||FastRecovery / ABC
> interaction...
>> 
>> 
>> I will try to sketch up an example to demonstrate what problem I'm
> trying to address:
>> 
>> 
>> Let's assume the cwnd is already open for at least 7 segments, before
> the segment with
>> sequence number 10000 is the first one to be dropped by the network.
>> 
>> Also, let's assume that FastRetransmit runs from the left edge of the
> leftmost hole
>> (SND.UNA) upwards, and that per ACK only a single segment is sent.
>> 
>> 
>>            Triggering    ACK      Left     Right    Left     Right
>>            Segment                Edge 1   Edge 1   Edge 2   Edge 2
>> 
>>             9000          9000
>>            10000  (lost)       *
>>            11000  (lost)
>>            12000  (lost)
>>            13000  (lost)
>>            14000  (lost)
>>            15000         10000    15000    16000
>>            16000         10000    15000    17000
>>            17000         10000    15000    18000
> 
> Ok, I count 9 segments ;-) Anyway, your example is a little bit strange.
> You assume you send 9 segment in a burst. Then your ACK for 9000 will
> trigger
> the segment 18000. Then the 2 DUPACKs  for 15000 and 16000 will trigger
> 19000,
> 20000 respectively (Limited Transmit). The third DUPACK will trigger the
> Fast
> Retransmit 10000. Since NextSeq() and pipe allow you retransmit 11000,
> 12000.
> 
> At this point you have to wait since the pipe is full (if I calculate
> pipe in a rush correctly).
> A RTT later you will get 3 Dupacks, even if the 10000 is not lost.
> 
> OK, I think you can adjust your example so that it works for the given
> case, however in
> other scenarios (multiple loss, reordering, packet duplication,...) it
> will be much more
> complicated.
> 
> I suggest writing examples as Ilpo does in
> http://tools.ietf.org/html/draft-ietf-tcpm-sack-recovery-entry-00.
> It's much more easier to read.
> 
>> 
>> 3 ACKs trigger fast retransmit
>> 
>>            10000  (lost again)
>>            11000         10000    11000    12000    15000    18000
>>            12000         10000    11000    13000    15000    18000
>>            13000         10000    11000    14000    15000    18000
>> -> here we have again 3 ACKs indicating a another loss of one of the
> retransmitted
>> packets. The leftmost hole did not change, while the overall number of
> SACKed
>> octets did decrease for 3 consecutive ACKs (4; 3 and 2 segments marked
> by SACK).
>> 
> 
> Alex
> 
> //
> // Dipl.-Inform. Alexander Zimmermann
> // Department of Computer Science, Informatik 4
> // RWTH Aachen University
> // Ahornstr. 55, 52056 Aachen, Germany
> // phone: (49-241) 80-21422, fax: (49-241) 80-22220
> // email: zimmermann@cs.rwth-aachen.de
> // web: http://www.umic-mesh.net
> //
> 

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


From hannemann@i4.informatik.rwth-aachen.de  Tue Nov 10 08:43:03 2009
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Date: Tue, 10 Nov 2009 17:43:03 +0100
From: Arnd Hannemann <hannemann@i4.informatik.rwth-aachen.de>
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Cc: tcpm@ietf.org, Alexander Zimmermann <Alexander.Zimmermann@nets.rwth-aachen.de>
Subject: Re: [tcpm] Detect Lost Retransmit mit SACK
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Hi Richard,


Scheffenegger, Richard schrieb:
> Hi Arnd,
> 
> Let's think about what would be required to track a lost retransmitted
> segment.
> 
> A) the sender can only know about this one RTT after it send out the
> first 
>    retransmission at the earliest.
> 
>   A1) With TSOpt, they could be used to "remember" when Retransmission
> started; 
>       When an ACKs with TSEcn > (TSOpt + RTT) is seen by the sender, it
> can 
>       re-arm the DUPACK detector.

This is your non-SACK scenario?
Please note that a TCP receiver will NOT echo TSVal
from out-of-order segments. So this won't work.

>   A2) With SACK, tracking SND.NXT and HOLEBYTES (number of octets in all
> current holes) 
>       can be employed to track RTT (they need to be initialized when the
> first 
>       retransmitted segment is sent). When an ACK contains a SACK for >=
> SND.NXT, or the 
>       HOLEBYTES are smaller than when retransmission started, the DUPACK
> detector can 
>       be re-armed.

You should specify what you mean with SND.NXT.
We are in recovery, so we may very well send out new segments.
I don't see why number of holes in sack scoreboard
has anything todo with RTT.

> 
>   A3) If neither TSOpt nor SACK is used, revert to current behavior (RTO
> timeout to
>       detect a lost retransmit).
> 
> B) DUPACK detector would need to be enhanced over what is currently in
> the RFCs (but 
>    I think most of that is already in Drafts, ie. 
>    http://tools.ietf.org/html/draft-jarvinen-tcpm-sack-recovery-entry-01
> ).
> 
>  B1) With SACK enabled, fully duplicate ACKs (ACK and SACK have been
> seen before, and 
>      prev_HOLEBYTES == HOLEBYTES) can be discarded; lost ACKs would
> count for two 
>      (prev_HOLEBYTES < HOLEBYTES - SMSS).

Now B1 is SACK enabled scenario? You try to confuse me, right?
All this makes little sense to me.

> 
>  B2) With TSOpt, if another RTT passes without advancing SND.UNA, 
>      increase DupAcks += DupThresh 
> 
>  When DupThresh # of ACKs are received, reset HighRxt = SND.UNA, DupAcks
> = 0; *)

See above...
non-SACK is not going to work.


> There are two comments I would like to make:
> 
> A) Latency: if the sender first sends out the entire range of holes,
> before
>    reacting to a detectable lost retransmitted packet, the latency
> observed
>    by the application on the receiver side will increase needlessly. It
> might
>    take a number of RTTs under certain circumstances, before all the
> holes
>    have been retransmitted once... 

Show me an example. where a lost retransmission is detectable, while
the sender did not send out retransmissions for every other hole?
Sounds weird.


> 
> B) Complexity: As soon as any congestion event is encountered, TCP won't
> be in
>    fastpath mode any longer; the added computational complexity to
> (reliably, see my
>    check on FACK with Linux 2.6.18 yesterday) trigger another
> retransmission
>    of an already retransmitted segment is neglegible, compared to
> current best
>    practise (waiting for RTO), when you need to move data around as
> quickly (low
>    latency AND high bandwidth) as possible. As mentioned, my background
> is not
>    so much the global internet (with statistically valid implict
> assumptions), but
>    rather High-Speed, short Range, (lossy) LANs, with only very few
> active TCP
>    sessions at any one time. The hosts I deal with might have as few as
> 10-20 TCP
>    sessions, of which only 1-4 are pushing data, but those might run
> over dedicated
>    10GbE ports (overloading L2 switches in the process, and exhibiting
> Burst Loss
>    (correlated packet loss) ).

This probably highly rates from case to case. If you have a huge
bandwidth-delay product even SACK scoreboard handling will have
computational complexity which might outweigh its benefits.
As you have seem to have the equipment at hand, you could do some
cpu load and throughput measurements with 10GbE
and SACK versus non-SACK flows.


> 
> 
> Thus my view might deviate from a "typical" view in certain aspects. I
> might even be 
> in error and would like to hear, why some things cann't work like
> depicted. However, 
> I would place more emphasis on the timely recovery  of any lost (or
> re-lost) segment, 
> over making sure that no single segment is re-transmitted needlessly;
> better retransmit 
> a little (!) too much too soon, instead of waiting for RTO :) I want to
> point out the 
> "little" though, as overwhelming a already lossy network with too much
> retransmissions 
> (ie. Setting RTO = 0/1 tick, with a TCP clock runnig in msec), will only
> cause more loss...
> 

I think it is a good idea to detect lost retransmissions and I think
it is possible with SACK. I already proposed a potential solution ;-)
"Just" store highest seq, for each retransmission. And check incoming SACKs
on those. My main concern is, as already said, that this might get
quiet expensive. But you are free to evaluate this more deeply ;-)
Anyway, maybe we should focus on your other thread...

Best regards,
Arnd

From rs@netapp.com  Tue Nov 10 10:28:09 2009
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Thread-Topic: Detect Lost Retransmit mit SACK
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Arnd Hannemann" <hannemann@i4.informatik.rwth-aachen.de>
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Cc: tcpm@ietf.org, Alexander Zimmermann <Alexander.Zimmermann@nets.rwth-aachen.de>
Subject: Re: [tcpm] Detect Lost Retransmit mit SACK
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See inline

Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20
Franz-Klein-Gasse 5
1190 Wien=20

=20

> -----Original Message-----
> From: Arnd Hannemann [mailto:hannemann@i4.informatik.rwth-aachen.de]=20
> Sent: Dienstag, 10. November 2009 17:43
> To: Scheffenegger, Richard
> Cc: Alexander Zimmermann; tcpm@ietf.org
> Subject: Re: Detect Lost Retransmit mit SACK
>=20
> Hi Richard,
>=20
>=20
> Scheffenegger, Richard schrieb:
::
> >=20
> >   A1) With TSOpt, they could be used to "remember" when=20
> >       Retransmission started;
> >       When an ACKs with TSEcn > (TSOpt + RTT) is seen by=20
> >       the sender, it can
> >       re-arm the DUPACK detector.
>=20
> This is your non-SACK scenario?
> Please note that a TCP receiver will NOT echo TSVal from=20
> out-of-order segments. So this won't work.

Interesting; this doesn't seem to be what certain (most?) stacks are=20
doing - they seem to always reply in the ACK Tsecr the value of
Tsopt/TS.recent=20
triggering the ACK...

Or was your intention to say, that the receiver will not respond with
Tsopt, but rather TS.recent?

If so, than this makes no difference - the sender will detect the=20
first ACK from the receiver, which has the same TS (well, actually TS+1
as multiple segments are likely to carry the same TS) when the
retransmission
was started. This will be the sign for the sender, that at least one RTT
has elapsed (that was the intention for having timestamps in the first
place, right?). If the returned ACK does not cover the first
retransmitted
segment by that time, it's quite reasonable to assume that it got lost
again.

Again, keep in mind that I'm looking on LANs; a TS in 10 ms increments
(100 Hz
TCP clock) is likely to be put into a few hundred or thousand packets...

If there is reason to believe the reordering can occur with delays >>
RTT, then
Yes, this simple detection logic would be fooled and falsely trigger a
retransmission.

However, IF the reordering is >> 2*RTT, then it's very likely that the=20
newly retransmitted segment arrives at the receiver first (over the
faster=20
path), and again this would only serve to keep the application visible
latency=20
as low as possible (between the points in time, where TCP can deliver
some / any=20
new data up the stack...)=20



> >   A2) With SACK, tracking SND.NXT and HOLEBYTES (number of=20
> >       octets in all current holes)
> >       can be employed to track RTT (they need to be=20
> >       initialized when the first
> >       retransmitted segment is sent). When an ACK contains=20
> >       a SACK for >=3D SND.NXT, or the
> >       HOLEBYTES are smaller than when retransmission started, the=20
> >       DUPACK detector can be re-armed.
>=20
> You should specify what you mean with SND.NXT.
> We are in recovery, so we may very well send out new segments.
> I don't see why number of holes in sack scoreboard has=20
> anything todo with RTT.


SND.NXT is the rightmost segment which the sender has never before=20
sent out to the network. Isn't that the meaning of Snd.nxt according=20
to RFC793?

And I was not talking about the # of holes, but rather the amount of
Data (in sum) between FACK (defined by the rightmost offset in the SACK
Scoreboard, i.e. covering the highest ever received segment) and
SND.UNA.

Monitoring Holes will gain nothing (when there is reordering for
example);

But if the any one retransmission makes it though, the amount of
unSACKed=20
data in the sum total of all holes in the scoreboard, will decrease -=20
indicating when at least one RTT has passed since start of
FastRetransmit.



>=20
> >=20
> >   A3) If neither TSOpt nor SACK is used, revert to current behavior=20
> > (RTO timeout to
> >       detect a lost retransmit).
> >=20
> > B) DUPACK detector would need to be enhanced over what is=20
> currently in=20
> > the RFCs (but
> >    I think most of that is already in Drafts, ie.=20
> >   =20
> >=20
> http://tools.ietf.org/html/draft-jarvinen-tcpm-sack-recovery-entry-01
> > ).
> >=20
> >  B1) With SACK enabled, fully duplicate ACKs (ACK and SACK=20
> have been=20
> > seen before, and
> >      prev_HOLEBYTES =3D=3D HOLEBYTES) can be discarded; lost ACKs =
would=20
> > count for two
> >      (prev_HOLEBYTES < HOLEBYTES - SMSS).
>=20
> Now B1 is SACK enabled scenario? You try to confuse me, right?
> All this makes little sense to me.


I agree, it would have made sense to label points B1 and B2 the other
way
around to stay more consistent; that's one of the reasons why I like to=20
discuss this :)

>=20
> >=20
> >  B2) With TSOpt, if another RTT passes without advancing SND.UNA,=20
> >      increase DupAcks +=3D DupThresh
> >=20
> >  When DupThresh # of ACKs are received, reset HighRxt =3D SND.UNA,=20
> > DupAcks =3D 0; *)
>=20
> See above...
> non-SACK is not going to work.


See above - TSOpt is only used for the prime purpose to detect when a=20
RTT has passed. It doesn't matter if Tsecr contains TSopt or
TS.recent...=20
=20
>=20
> > There are two comments I would like to make:
> >=20
> > A) Latency: if the sender first sends out the entire range=20
> of holes,=20
> > before
> >    reacting to a detectable lost retransmitted packet, the latency=20
> > observed
> >    by the application on the receiver side will increase=20
> needlessly.=20
> > It might
> >    take a number of RTTs under certain circumstances,=20
> before all the=20
> > holes
> >    have been retransmitted once...=20
>=20
> Show me an example. where a lost retransmission is=20
> detectable, while the sender did not send out retransmissions=20
> for every other hole?
> Sounds weird.

Can I attach trace files on posts to this list, or shall I sent them=20
to you directly?=20

I assume that on the receiving side, the application is only delivered
data, which has been ACKed (not SACKed). If the sender waits until=20
RTO before retrying to re-send a retransmission (sorry about these
Multiple re-s :) ), the application will at least stall (not get any=20
new data) for RTO+2*RTT - and certain latency sensitive applications
will time out before that. Again: I'm not talking about internet
applications but rather LAN HPC applications with timeouts in the=20
order of a few seconds at most.

Advancing (on the receiver) within that application timeout is
Usually enough to not run into the application timeout.


>=20
>=20
> >=20
> > B) Complexity: As soon as any congestion event is encountered, TCP=20
> > won't be in
> >    fastpath mode any longer; the added computational complexity to=20
> > (reliably, see my
> >    check on FACK with Linux 2.6.18 yesterday) trigger another=20
> > retransmission
> >    of an already retransmitted segment is neglegible, compared to=20
> > current best
> >    practise (waiting for RTO), when you need to move data around as=20
> > quickly (low
> >    latency AND high bandwidth) as possible. As mentioned, my=20
> > background is not
> >    so much the global internet (with statistically valid implict=20
> > assumptions), but
> >    rather High-Speed, short Range, (lossy) LANs, with only very few=20
> > active TCP
> >    sessions at any one time. The hosts I deal with might=20
> have as few=20
> > as 10-20 TCP
> >    sessions, of which only 1-4 are pushing data, but those=20
> might run=20
> > over dedicated
> >    10GbE ports (overloading L2 switches in the process, and=20
> exhibiting=20
> > Burst Loss
> >    (correlated packet loss) ).
>=20
> This probably highly rates from case to case. If you have a=20
> huge bandwidth-delay product even SACK scoreboard handling=20
> will have computational complexity which might outweigh its benefits.
> As you have seem to have the equipment at hand, you could do=20
> some cpu load and throughput measurements with 10GbE and SACK=20
> versus non-SACK flows.

Well, I was doing some test like this:

http://www.ibm.com/developerworks/linux/library/l-tcp-sack/index.html

But the CPU impact (on top of what is required for 10G) was neglegible.

But I have to admit, I didn't had the time yet to test a SACK scoreboard
with every other byte marked mission (which could mean some 0,5 mio
entries).

But it seems that most implementations feature sorted scoreboards of
limited
(16/128) size.

Traversing those is - again in my case of 10G LANs - still many orders
of=20
magnitude faster then waiting for RTO.=20

I personally consider RTOs to be harmful (when there are other means to=20
recover with). :) (perhaps once I have worked through all the issues
raised
here, I submit a draft with "TCP RTO considered harmful" next April? :)


>=20
>=20
> >=20
> >=20
> > Thus my view might deviate from a "typical" view in certain=20
> aspects. I=20
> > might even be in error and would like to hear, why some=20
> things cann't=20
> > work like depicted. However, I would place more emphasis on=20
> the timely=20
> > recovery  of any lost (or
> > re-lost) segment,
> > over making sure that no single segment is re-transmitted=20
> needlessly;=20
> > better retransmit a little (!) too much too soon, instead=20
> of waiting=20
> > for RTO :) I want to point out the "little" though, as=20
> overwhelming a=20
> > already lossy network with too much retransmissions (ie.=20
> Setting RTO =3D=20
> > 0/1 tick, with a TCP clock runnig in msec), will only cause more=20
> > loss...
> >=20
>=20
> I think it is a good idea to detect lost retransmissions and=20
> I think it is possible with SACK. I already proposed a=20
> potential solution ;-) "Just" store highest seq, for each=20
> retransmission. And check incoming SACKs on those. My main=20
> concern is, as already said, that this might get quiet=20
> expensive. But you are free to evaluate this more deeply ;-)=20
> Anyway, maybe we should focus on your other thread...
>=20
> Best regards,
> Arnd
>=20

From ilpo.jarvinen@helsinki.fi  Tue Nov 10 19:07:33 2009
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Cc: k.avrachenkov@sophia.inria.fr, tcpm@ietf.org, jblanton@cs.ohiou.edu
Subject: Re: [tcpm] early retransmit done?
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On Mon, 9 Nov 2009, Mark Allman wrote:

> > We believe we have addressed all comments we received on the early
> > retranmsit ID.  I just went to submit a new version and have missed the
> > cut-off.  I have set a bit to submit when the system opens again on
> > Nov/9.  However, until then I put the version I was going to submit here
> > ...
> > 
> >   http://www.icir.org/mallman/papers/draft-ietf-tcpm-early-rexmt-02a.txt
> > 
> > Comments are very welcome.
> 
> Since TCPM is not meeting this week Lars got this draft approved for
> posting.  So, it is now at the usual place (as "-02" not "-02a).
> 
> Our hit is that it seems finished.  Please yell if there is more to do.

My wish is that it would be slightly more explicit in how to handle a
FIN (only) segment. Through intuition I'd say it is included to ownd and 
oseg but that not mentioned I think. I suppose this flow termination is 
relatively significant case when considering benefits of early rexmit.
The confusion is further reinforced by the example in 3.3 (case 2):

1. 3-way handshake, connect() returns.
2. App sends two "data segments" as per IW=2 using write(), they go 
deep in the stack right away as there is room for that (with IW=1 there 
wouldn't be any reordering possibility so it has to IW=2 or larger!)
3. write() call returns
4. close() is called
5. FIN only segment needs to be sent
...

This third segment does not match to the behavior described in the ID at 
all? ...It would probably be already quite much to address this if just 
this worst case example is converted to have a single data segment and a 
FIN only segment (with persistent reordering between them).

-- 
 i.

From matt.mathis@gmail.com  Tue Nov 10 20:58:41 2009
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From: Matt Mathis <mathis@psc.edu>
To: "Scheffenegger, Richard" <rs@netapp.com>, carsten.wolff@rwth-aachen.de
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Cc: Jeff Semke <semke@netapp.com>, tcpm Extensions WG <tcpm@ietf.org>, Jamshid Mahdavi <jmahdavi@gmail.com>
Subject: Re: [tcpm] Curiosity about draft-mathis-tcp-ratehalving
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We abandoned Rate-Halving because we encountered a problem that we
were not able to overcome at the time. It performs very poorly when
you have bursty applications. Unfortunately the very feature which
causes this problem came to be mandated by 2581 and is now entrenched
in 10 years of other congestion control standards.

This is part of the reason that with large multi core systems, it is
hard to get a single TCP stream to run at really high speeds when
there is much load on the CPUs. Todays TCP implementations all suffer
badly if the sender frequently stalls due to running out of data, even
for fractional RTTs (e.g, waiting for the scheduler or a disk seek.)

The problem comes from setting cwnd to half the flight_size, even if
the flight size was depressed for reasons unrelated to congestion
control. This and other algorithms (e.g. some of the burst suppression
stuff) couple fine grained (less then 1 RTT) events to congestion
control where they potentially have impact lasting for thousands of
RTTs. As far as I am concerned this is wrong and has always been
wrong.

This coupling is an intrinsic property of rate halving. At the time
the draft stalled we were looking for a way to gracefully prevent the
new cwnd from going below old_cwnd/2, without upsetting people if
flight_size was already below cwnd/2 (This condition implies no window
adjustment during recovery).

Now more than 10 years later I think I understand the right answer.
TCP congestion control needs to be reparameterized, with different
control variables. This would require ripping out and replacing most
of the current congestion control code. All of the old problems still
need to be solved, but under the new parameterization the algorithms
would become much simpler and probably better behaved.  This is a lot
more effort than a simple update of rate-halving or some other
document.   The Relentless/TCP-unfriendly stuff touches on it, however
they are really orthogonal problems.

Feel free to re-open the rate-halving draft.  There is no formal
process: take the existing document and edit it.  You might check to
see if there were any changes posted to
http://www.psc.edu/networking/ftp/papers/draft-ratehalving.txt that
don't appear in the last IETF draft.   If the other original authors
(cc'd above) don't choose to participate, their effort should be
mentioned in the acknowledgement section.  I would like to be included
as an author, however I can only invest extremely limited time and
energy.

If you do reopen the document, I would like it to include some version
of the above discussion.

Thanks,
--MM--
-------------------------------------------
Matt Mathis      http://staff.psc.edu/mathis
Work:412.268.3319   Home/Cell:412.654.7529
-------------------------------------------
Evil is defined by mortals who think they know
"The Truth" and use force to apply it to others.



On Mon, Nov 9, 2009 at 7:09 PM, Scheffenegger, Richard <rs@netapp.com> wrot=
e:
>
> Hello Carsten,
>
> This is a very good question indeed.
>
> Keeping the ACK clock running, and avoiding bursts of segments delivered
> all at once in a single burst to a L2 LAN network is something
> where my interests are too... (as todays hosts are increasingly capable
> of oversaturating slightly dated network equipment with full wire speed
> traffic).
>
> Is there a procedure to revive / adopt a dead draft?
>
> I have not spent much time on this issue, but noted the burst behavior
> After fastretransmit.
>
>
> As RateHalving extended the scope of Limited Transmit (with the similar
> goal), shouldn't this be discussed for inclusion in a RFC 3042bis?
>
> Since certain other congestion algorithms have a beta of less than 0.5
> (ie. BIC, CUBIC use 0.8) when selecting a new ssthresh, perhaps the
> effect
> of RateHalfing, but with 2 out of 3 (0.66) policy or 3 out of 4 (0.75)
> at the end of the RTT should be investigated? (To be less conservative
> When reducing the sending rate).
>
> One extension to the draft might be the discussion on how to deal with
> RFC 3465 (ABC) during rate halving; with SACK, ABC could still be
> employed,
> And a new segment triggered only, when 2*MSS has been covered by the new
> SACKs...
>
> Regards,
>
> Richard Scheffenegger
> Field Escalation Engineer
> NetApp Global Support
> NetApp
> +43 1 3676811 3146 Office (2143 3146 - internal)
> +43 676 654 3146 Mobile
> www.netapp.com <BLOCKED::http://www.netapp.com/>
> Franz-Klein-Gasse 5
> 1190 Wien
>
> * =A0 =A0 =A0 To: Matt Mathis <mathis at psc.edu <mailto:mathis@DOMAIN.HI=
DDEN>
>>
> * =A0 =A0 =A0 Subject: [tcpm] Curiosity about draft-mathis-tcp-ratehalvin=
g
> * =A0 =A0 =A0 From: Carsten Wolff <carsten.wolff at rwth-aachen.de
> <mailto:carsten.wolff@DOMAIN.HIDDEN> >
> * =A0 =A0 =A0 Date: Wed, 4 Nov 2009 13:48:51 +0100
> =A0_____
>
> Hello Matt, everyone,
>
> I'm curious about draft-mathis-tcp-ratehalving
> (http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt) and why
> it
> never exceeded version 00 and expired years ago. It seems, several of
> the
> ideas (fack, ratehalving, the concept of rhcwnd) in this paper made it
> into
> the Linux TCP implementation in some form, which means they're part of a
> big
> chunk of today's web servers and other Internet hosts. Why has there
> been no
> interest in taking this draft further?
>
> Cheers,
> Carsten
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From ilpo.jarvinen@helsinki.fi  Wed Nov 11 00:31:58 2009
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Subject: Re: [tcpm] Detect Lost Retransmit with SACK
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I have couple of points to add into this thread but my inbox is a bit of a 
mess at the moment and I really don't know what point into this thread to 
intervene, so I'll just spell them out without quoting anything....

1) You are partially right. Assuming in-order behavior it is possible for 
the sender to fully construct the sequence of receival that too place 
using SACK (+DSACK) infromation (expect for the ambiguous  gaps caused by 
ACK losses). By using this one could run dupThresh-like check for any 
segment regardless of it being retransmission or not, which is what I 
think you are trying to accomplish here. However, with reordering it gets 
rather challenging when there is ambiguity present.

2) And even if reordering is assumed to not be there, keeping track of 
that ordering used at the sending sides and efficiently accessing all 
necessary state in non O(n^2) fashion is, well, a challenging task and 
it is much easier to take a shortcut. The easiest is already mentioned by 
somebody (use only new data SACK blocks), and even then there are per 
segment processing time challenges as was brought up.

3) Linux has implemented that shortcut approach for ages, however, 
unfortunately the (obsolete) versions of Linux you have been using in your 
test contain bugs which prevent lost retransmission detection from working 
efficiently. It was fixed in some 2.6.25 timeframe but that is already so 
far away that it is hard to remember exactly.

4) Even the latest kernels do not perform lost retransmission detection as 
efficiently as I'd want to (I have some local patches to solve that but 
no timeline for submitting them), ie., it cannot keep operating with 1Gbps 
speeds for long periods of time if operating with non-trivial RTTs (see my
recent paper in PFLDnet for the details).

5) Lost retransmission detection in Linux works only with FACK mode, and 
there's automatic switching from FACK to RFC 3517 mode if reordering is 
detected so please be careful to not cause reordering symptoms with your 
test tricks (I'm not trying to imply here that you'd be doing that).


-- 
 i.

From Alexander.Zimmermann@nets.rwth-aachen.de  Wed Nov 11 00:34:48 2009
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To: "Scheffenegger, Richard" <rs@netapp.com>
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Subject: Re: [tcpm] Detect Lost Retransmit mit SACK
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Hi Richard,

it is done. I'm completely lost in the double threaded tread. I really suggest we stop
this ping-pong discussion, since IHMO nobody can follow us....
My suggestion is that you

* start to write a little ID with at least a pseudo code algo
* and a lot of *well indented* examples like Ilpo does in his ID
	=> Basic Case, multiple (burst) loss, ACK loss, reordering, packet duplication,...

With such a document,  it will be much more easier to follow your thoughts.
What do you think?

Alex

Am 10.11.2009 um 19:28 schrieb Scheffenegger, Richard:

> 
> See inline
> 
> Richard Scheffenegger
> Field Escalation Engineer
> NetApp Global Support 
> NetApp
> +43 1 3676811 3146 Office (2143 3146 - internal)
> +43 676 654 3146 Mobile
> www.netapp.com 
> Franz-Klein-Gasse 5
> 1190 Wien 
> 
> 
> 
>> -----Original Message-----
>> From: Arnd Hannemann [mailto:hannemann@i4.informatik.rwth-aachen.de] 
>> Sent: Dienstag, 10. November 2009 17:43
>> To: Scheffenegger, Richard
>> Cc: Alexander Zimmermann; tcpm@ietf.org
>> Subject: Re: Detect Lost Retransmit mit SACK
>> 
>> Hi Richard,
>> 
>> 
>> Scheffenegger, Richard schrieb:
> ::
>>> 
>>>  A1) With TSOpt, they could be used to "remember" when 
>>>      Retransmission started;
>>>      When an ACKs with TSEcn > (TSOpt + RTT) is seen by 
>>>      the sender, it can
>>>      re-arm the DUPACK detector.
>> 
>> This is your non-SACK scenario?
>> Please note that a TCP receiver will NOT echo TSVal from 
>> out-of-order segments. So this won't work.
> 
> Interesting; this doesn't seem to be what certain (most?) stacks are 
> doing - they seem to always reply in the ACK Tsecr the value of
> Tsopt/TS.recent 
> triggering the ACK...
> 
> Or was your intention to say, that the receiver will not respond with
> Tsopt, but rather TS.recent?
> 
> If so, than this makes no difference - the sender will detect the 
> first ACK from the receiver, which has the same TS (well, actually TS+1
> as multiple segments are likely to carry the same TS) when the
> retransmission
> was started. This will be the sign for the sender, that at least one RTT
> has elapsed (that was the intention for having timestamps in the first
> place, right?). If the returned ACK does not cover the first
> retransmitted
> segment by that time, it's quite reasonable to assume that it got lost
> again.
> 
> Again, keep in mind that I'm looking on LANs; a TS in 10 ms increments
> (100 Hz
> TCP clock) is likely to be put into a few hundred or thousand packets...
> 
> If there is reason to believe the reordering can occur with delays >>
> RTT, then
> Yes, this simple detection logic would be fooled and falsely trigger a
> retransmission.
> 
> However, IF the reordering is >> 2*RTT, then it's very likely that the 
> newly retransmitted segment arrives at the receiver first (over the
> faster 
> path), and again this would only serve to keep the application visible
> latency 
> as low as possible (between the points in time, where TCP can deliver
> some / any 
> new data up the stack...) 
> 
> 
> 
>>>  A2) With SACK, tracking SND.NXT and HOLEBYTES (number of 
>>>      octets in all current holes)
>>>      can be employed to track RTT (they need to be 
>>>      initialized when the first
>>>      retransmitted segment is sent). When an ACK contains 
>>>      a SACK for >= SND.NXT, or the
>>>      HOLEBYTES are smaller than when retransmission started, the 
>>>      DUPACK detector can be re-armed.
>> 
>> You should specify what you mean with SND.NXT.
>> We are in recovery, so we may very well send out new segments.
>> I don't see why number of holes in sack scoreboard has 
>> anything todo with RTT.
> 
> 
> SND.NXT is the rightmost segment which the sender has never before 
> sent out to the network. Isn't that the meaning of Snd.nxt according 
> to RFC793?
> 
> And I was not talking about the # of holes, but rather the amount of
> Data (in sum) between FACK (defined by the rightmost offset in the SACK
> Scoreboard, i.e. covering the highest ever received segment) and
> SND.UNA.
> 
> Monitoring Holes will gain nothing (when there is reordering for
> example);
> 
> But if the any one retransmission makes it though, the amount of
> unSACKed 
> data in the sum total of all holes in the scoreboard, will decrease - 
> indicating when at least one RTT has passed since start of
> FastRetransmit.
> 
> 
> 
>> 
>>> 
>>>  A3) If neither TSOpt nor SACK is used, revert to current behavior 
>>> (RTO timeout to
>>>      detect a lost retransmit).
>>> 
>>> B) DUPACK detector would need to be enhanced over what is 
>> currently in 
>>> the RFCs (but
>>>   I think most of that is already in Drafts, ie. 
>>> 
>>> 
>> http://tools.ietf.org/html/draft-jarvinen-tcpm-sack-recovery-entry-01
>>> ).
>>> 
>>> B1) With SACK enabled, fully duplicate ACKs (ACK and SACK 
>> have been 
>>> seen before, and
>>>     prev_HOLEBYTES == HOLEBYTES) can be discarded; lost ACKs would 
>>> count for two
>>>     (prev_HOLEBYTES < HOLEBYTES - SMSS).
>> 
>> Now B1 is SACK enabled scenario? You try to confuse me, right?
>> All this makes little sense to me.
> 
> 
> I agree, it would have made sense to label points B1 and B2 the other
> way
> around to stay more consistent; that's one of the reasons why I like to 
> discuss this :)
> 
>> 
>>> 
>>> B2) With TSOpt, if another RTT passes without advancing SND.UNA, 
>>>     increase DupAcks += DupThresh
>>> 
>>> When DupThresh # of ACKs are received, reset HighRxt = SND.UNA, 
>>> DupAcks = 0; *)
>> 
>> See above...
>> non-SACK is not going to work.
> 
> 
> See above - TSOpt is only used for the prime purpose to detect when a 
> RTT has passed. It doesn't matter if Tsecr contains TSopt or
> TS.recent... 
> 
>> 
>>> There are two comments I would like to make:
>>> 
>>> A) Latency: if the sender first sends out the entire range 
>> of holes, 
>>> before
>>>   reacting to a detectable lost retransmitted packet, the latency 
>>> observed
>>>   by the application on the receiver side will increase 
>> needlessly. 
>>> It might
>>>   take a number of RTTs under certain circumstances, 
>> before all the 
>>> holes
>>>   have been retransmitted once... 
>> 
>> Show me an example. where a lost retransmission is 
>> detectable, while the sender did not send out retransmissions 
>> for every other hole?
>> Sounds weird.
> 
> Can I attach trace files on posts to this list, or shall I sent them 
> to you directly? 
> 
> I assume that on the receiving side, the application is only delivered
> data, which has been ACKed (not SACKed). If the sender waits until 
> RTO before retrying to re-send a retransmission (sorry about these
> Multiple re-s :) ), the application will at least stall (not get any 
> new data) for RTO+2*RTT - and certain latency sensitive applications
> will time out before that. Again: I'm not talking about internet
> applications but rather LAN HPC applications with timeouts in the 
> order of a few seconds at most.
> 
> Advancing (on the receiver) within that application timeout is
> Usually enough to not run into the application timeout.
> 
> 
>> 
>> 
>>> 
>>> B) Complexity: As soon as any congestion event is encountered, TCP 
>>> won't be in
>>>   fastpath mode any longer; the added computational complexity to 
>>> (reliably, see my
>>>   check on FACK with Linux 2.6.18 yesterday) trigger another 
>>> retransmission
>>>   of an already retransmitted segment is neglegible, compared to 
>>> current best
>>>   practise (waiting for RTO), when you need to move data around as 
>>> quickly (low
>>>   latency AND high bandwidth) as possible. As mentioned, my 
>>> background is not
>>>   so much the global internet (with statistically valid implict 
>>> assumptions), but
>>>   rather High-Speed, short Range, (lossy) LANs, with only very few 
>>> active TCP
>>>   sessions at any one time. The hosts I deal with might 
>> have as few 
>>> as 10-20 TCP
>>>   sessions, of which only 1-4 are pushing data, but those 
>> might run 
>>> over dedicated
>>>   10GbE ports (overloading L2 switches in the process, and 
>> exhibiting 
>>> Burst Loss
>>>   (correlated packet loss) ).
>> 
>> This probably highly rates from case to case. If you have a 
>> huge bandwidth-delay product even SACK scoreboard handling 
>> will have computational complexity which might outweigh its benefits.
>> As you have seem to have the equipment at hand, you could do 
>> some cpu load and throughput measurements with 10GbE and SACK 
>> versus non-SACK flows.
> 
> Well, I was doing some test like this:
> 
> http://www.ibm.com/developerworks/linux/library/l-tcp-sack/index.html
> 
> But the CPU impact (on top of what is required for 10G) was neglegible.
> 
> But I have to admit, I didn't had the time yet to test a SACK scoreboard
> with every other byte marked mission (which could mean some 0,5 mio
> entries).
> 
> But it seems that most implementations feature sorted scoreboards of
> limited
> (16/128) size.
> 
> Traversing those is - again in my case of 10G LANs - still many orders
> of 
> magnitude faster then waiting for RTO. 
> 
> I personally consider RTOs to be harmful (when there are other means to 
> recover with). :) (perhaps once I have worked through all the issues
> raised
> here, I submit a draft with "TCP RTO considered harmful" next April? :)
> 
> 
>> 
>> 
>>> 
>>> 
>>> Thus my view might deviate from a "typical" view in certain 
>> aspects. I 
>>> might even be in error and would like to hear, why some 
>> things cann't 
>>> work like depicted. However, I would place more emphasis on 
>> the timely 
>>> recovery  of any lost (or
>>> re-lost) segment,
>>> over making sure that no single segment is re-transmitted 
>> needlessly; 
>>> better retransmit a little (!) too much too soon, instead 
>> of waiting 
>>> for RTO :) I want to point out the "little" though, as 
>> overwhelming a 
>>> already lossy network with too much retransmissions (ie. 
>> Setting RTO = 
>>> 0/1 tick, with a TCP clock runnig in msec), will only cause more 
>>> loss...
>>> 
>> 
>> I think it is a good idea to detect lost retransmissions and 
>> I think it is possible with SACK. I already proposed a 
>> potential solution ;-) "Just" store highest seq, for each 
>> retransmission. And check incoming SACKs on those. My main 
>> concern is, as already said, that this might get quiet 
>> expensive. But you are free to evaluate this more deeply ;-) 
>> Anyway, maybe we should focus on your other thread...
>> 
>> Best regards,
>> Arnd
>> 

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


From hannemann@i4.informatik.rwth-aachen.de  Wed Nov 11 00:45:50 2009
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Date: Wed, 11 Nov 2009 09:45:49 +0100
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Subject: Re: [tcpm] Detect Lost Retransmit mit SACK
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Hi Richard,

Scheffenegger, Richard schrieb:
>>>   A1) With TSOpt, they could be used to "remember" when 
>>>       Retransmission started;
>>>       When an ACKs with TSEcn > (TSOpt + RTT) is seen by 
>>>       the sender, it can
>>>       re-arm the DUPACK detector.
>>>       
>> This is your non-SACK scenario?
>> Please note that a TCP receiver will NOT echo TSVal from 
>> out-of-order segments. So this won't work.
>>     
>
> Interesting; this doesn't seem to be what certain (most?) stacks are 
> doing - they seem to always reply in the ACK Tsecr the value of
> Tsopt/TS.recent 
> triggering the ACK...

Please clarify, is this non-SACK scenario?
The only lost retransmission we could be talking about then,
is the retransmission of SND.UNA.
And the TCP receiver will NOT adjust TS.recent for any
out-of order packets (RFC 1323).
It will continue to echo the timestamp of the last segment which
advanced the window (SND.UNA -1p).


> Or was your intention to say, that the receiver will not respond with
> Tsopt, but rather TS.recent?

Yes, it will not respond with TSVal but with TS.recent.

> If so, than this makes no difference - the sender will detect the 
> first ACK from the receiver, which has the same TS (well, actually TS+1
> as multiple segments are likely to carry the same TS) when the
> retransmission

There won't be an ACK if the retransmission is lost in non-SACK
case, only dupACKs.
(Only retransmission is retransmission of SND.UNA)
TS.Recent will not get updated.


> was started. This will be the sign for the sender, that at least one RTT
> has elapsed (that was the intention for having timestamps in the first
> place, right?). If the returned ACK does not cover the first
> retransmitted
> segment by that time, it's quite reasonable to assume that it got lost
> again
>   

> Again, keep in mind that I'm looking on LANs; a TS in 10 ms increments

If you are searching for something which is not intended for general use,
why not just set MIN_RTO to say 10 ms?

>>>   A2) With SACK, tracking SND.NXT and HOLEBYTES (number of 
>>>       octets in all current holes)
>>>       can be employed to track RTT (they need to be 
>>>       initialized when the first
>>>       retransmitted segment is sent). When an ACK contains 
>>>       a SACK for >= SND.NXT, or the
>>>       HOLEBYTES are smaller than when retransmission started, the 
>>>       DUPACK detector can be re-armed.
>>>       
>> You should specify what you mean with SND.NXT.
>> We are in recovery, so we may very well send out new segments.
>> I don't see why number of holes in sack scoreboard has 
>> anything todo with RTT.
>>     
>
>
> SND.NXT is the rightmost segment which the sender has never before 
> sent out to the network. Isn't that the meaning of Snd.nxt according 
> to RFC793?

Hmm, I would interpret RFC793 differently.
I would call it SND.MAX (see RFC4015) or HighData (RFC3517) the
rightmost segment which the sender has never sent before. But now you
have to clarify how it would be possible to receive a
SACK block > SND.MAX. Even SACK block == SND.MAX will be rarely seen.
But before taking this discussion even further, maybe take some time
and write some preliminary draft of your algorithm.

> And I was not talking about the # of holes, but rather the amount of
> Data (in sum) between FACK (defined by the rightmost offset in the SACK
> Scoreboard, i.e. covering the highest ever received segment) and
> SND.UNA.
>
> Monitoring Holes will gain nothing (when there is reordering for
> example);
>
> But if the any one retransmission makes it though, the amount of
> unSACKed 
> data in the sum total of all holes in the scoreboard, will decrease - 
> indicating when at least one RTT has passed since start of
> FastRetransmit.

This is not true.
Consider a 100 packet in flight, you get 3 dupacks, do  a retransmission.
This is in your words "start of FastRetransmit". After this the sum total
of all holes between SND.UNA and FACK can only increase for one RTT.
Due to Limited Transmit and Fast Recovery, it might be possible that
even the sum total of all holes between SND.UNA and SND.MAX will
increase.
You know when the first Retransmit is lost when a SACK
covers RecoveryPoint (RFC3517). But this is only true for first hole.
To detect lost retransmissions of the other holes, you would
need to store a separate HighData (RFC3617) for every retransmission.
Maybe this is after all what you are trying to express?
Perhaps it would be helpful if you could cook up some pseudo code,
of your proposed algorithm.

>>     
>>> holes
>>>    have been retransmitted once... 
>>>       
>> Show me an example. where a lost retransmission is 
>> detectable, while the sender did not send out retransmissions 
>> for every other hole?
>> Sounds weird.
>>     
>
> Can I attach trace files on posts to this list, or shall I sent them 
> to you directly? 

Never mind, does not actually matter.

>>>       
>> This probably highly rates from case to case. If you have a 
>> huge bandwidth-delay product even SACK scoreboard handling 
>> will have computational complexity which might outweigh its benefits.
>> As you have seem to have the equipment at hand, you could do 
>> some cpu load and throughput measurements with 10GbE and SACK 
>> versus non-SACK flows.
>>     
>
> Well, I was doing some test like this:
>
> http://www.ibm.com/developerworks/linux/library/l-tcp-sack/index.html
>
> But the CPU impact (on top of what is required for 10G) was neglegible.
> But I have to admit, I didn't had the time yet to test a SACK scoreboard
> with every other byte marked mission (which could mean some 0,5 mio
> entries).
>
> But it seems that most implementations feature sorted scoreboards of
> limited
> (16/128) size.
>
> Traversing those is - again in my case of 10G LANs - still many orders
> of 
> magnitude faster then waiting for RTO. 
>
> I personally consider RTOs to be harmful (when there are other means to 
> recover with). :) (perhaps once I have worked through all the issues
> raised
> here, I submit a draft with "TCP RTO considered harmful" next April? :)

Better submit a draft to mitigate the problems with it ;-)

Best regards,
Arnd






From alexander.zimmermann@nets.rwth-aachen.de  Wed Nov 11 05:12:48 2009
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From: Alexander Zimmermann <alexander.zimmermann@nets.rwth-aachen.de>
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To: Matt Mathis <mathis@psc.edu>
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Subject: Re: [tcpm] Curiosity about draft-mathis-tcp-ratehalving
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Hi Matt,

thanks for the quick answer. Carsten and I stumbled upon rate halving during 
our investigation how Linux TCP handles segment reordering. Since we currently 
don't want to skip our primary goal, I want to figure out if we could/should 
reanimate the ID with respect to men power, etc.

Am 11.11.2009 um 05:59 schrieb Matt Mathis:

> We abandoned Rate-Halving because we encountered a problem that we
> were not able to overcome at the time. It performs very poorly when
> you have bursty applications. Unfortunately the very feature which
> causes this problem came to be mandated by 2581 and is now entrenched
> in 10 years of other congestion control standards.

Ok, I see. However, but why is it not possible to do a step by step approach. 
Rate halving reduces TCP burstyness during recovery and IMHO it is a valuable 
feature, more straight forward then artificially inflating/deflating. Fixing 
the poor TCP performance in case of bursty applications can be seen as another 
step.

> 
> This is part of the reason that with large multi core systems, it is
> hard to get a single TCP stream to run at really high speeds when
> there is much load on the CPUs. Todays TCP implementations all suffer
> badly if the sender frequently stalls due to running out of data, even
> for fractional RTTs (e.g, waiting for the scheduler or a disk seek.)

Is this different on a single core? I mean if my TCP doesn't get enough
CPU time on my single core, I cannot send with high speed, too.

> 
> The problem comes from setting cwnd to half the flight_size, even if
> the flight size was depressed for reasons unrelated to congestion
> control.

Ok, but what can we do? As RFC 2861 describes, we can only know
about the current BDP if we are not application (or similarly)
limited. Do you want do move away from AIMD?

> This and other algorithms (e.g. some of the burst suppression
> stuff) couple fine grained (less then 1 RTT) events to congestion
> control where they potentially have impact lasting for thousands of
> RTTs. As far as I am concerned this is wrong and has always been
> wrong.
> 
> This coupling is an intrinsic property of rate halving. At the time
> the draft stalled we were looking for a way to gracefully prevent the
> new cwnd from going below old_cwnd/2, without upsetting people if
> flight_size was already below cwnd/2 (This condition implies no window
> adjustment during recovery).
> 
> Now more than 10 years later I think I understand the right answer.
> TCP congestion control needs to be reparameterized, with different
> control variables.

Which variables/parameter do you have in mind?

> This would require ripping out and replacing most
> of the current congestion control code. All of the old problems still
> need to be solved, but under the new parameterization the algorithms
> would become much simpler and probably better behaved.  This is a lot
> more effort than a simple update of rate-halving or some other
> document.   The Relentless/TCP-unfriendly stuff touches on it, however
> they are really orthogonal problems.
> 
> Feel free to re-open the rate-halving draft.  There is no formal
> process: take the existing document and edit it.  You might check to
> see if there were any changes posted to
> http://www.psc.edu/networking/ftp/papers/draft-ratehalving.txt that
> don't appear in the last IETF draft.   If the other original authors
> (cc'd above) don't choose to participate, their effort should be
> mentioned in the acknowledgement section.  I would like to be included
> as an author, however I can only invest extremely limited time and
> energy.
> 
> If you do reopen the document, I would like it to include some version
> of the above discussion.
> 
> Thanks,
> --MM--
> -------------------------------------------
> Matt Mathis      http://staff.psc.edu/mathis
> Work:412.268.3319   Home/Cell:412.654.7529
> -------------------------------------------
> Evil is defined by mortals who think they know
> "The Truth" and use force to apply it to others.
> 
> 
> 
> On Mon, Nov 9, 2009 at 7:09 PM, Scheffenegger, Richard <rs@netapp.com> wrote:
>> 
>> Hello Carsten,
>> 
>> This is a very good question indeed.
>> 
>> Keeping the ACK clock running, and avoiding bursts of segments delivered
>> all at once in a single burst to a L2 LAN network is something
>> where my interests are too... (as todays hosts are increasingly capable
>> of oversaturating slightly dated network equipment with full wire speed
>> traffic).
>> 
>> Is there a procedure to revive / adopt a dead draft?
>> 
>> I have not spent much time on this issue, but noted the burst behavior
>> After fastretransmit.
>> 
>> 
>> As RateHalving extended the scope of Limited Transmit (with the similar
>> goal), shouldn't this be discussed for inclusion in a RFC 3042bis?
>> 
>> Since certain other congestion algorithms have a beta of less than 0.5
>> (ie. BIC, CUBIC use 0.8) when selecting a new ssthresh, perhaps the
>> effect
>> of RateHalfing, but with 2 out of 3 (0.66) policy or 3 out of 4 (0.75)
>> at the end of the RTT should be investigated? (To be less conservative
>> When reducing the sending rate).
>> 
>> One extension to the draft might be the discussion on how to deal with
>> RFC 3465 (ABC) during rate halving; with SACK, ABC could still be
>> employed,
>> And a new segment triggered only, when 2*MSS has been covered by the new
>> SACKs...
>> 
>> Regards,
>> 
>> Richard Scheffenegger
>> Field Escalation Engineer
>> NetApp Global Support
>> NetApp
>> +43 1 3676811 3146 Office (2143 3146 - internal)
>> +43 676 654 3146 Mobile
>> www.netapp.com <BLOCKED::http://www.netapp.com/>
>> Franz-Klein-Gasse 5
>> 1190 Wien
>> 
>> *       To: Matt Mathis <mathis at psc.edu <mailto:mathis@DOMAIN.HIDDEN>
>>> 
>> *       Subject: [tcpm] Curiosity about draft-mathis-tcp-ratehalving
>> *       From: Carsten Wolff <carsten.wolff at rwth-aachen.de
>> <mailto:carsten.wolff@DOMAIN.HIDDEN> >
>> *       Date: Wed, 4 Nov 2009 13:48:51 +0100
>>  _____
>> 
>> Hello Matt, everyone,
>> 
>> I'm curious about draft-mathis-tcp-ratehalving
>> (http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt) and why
>> it
>> never exceeded version 00 and expired years ago. It seems, several of
>> the
>> ideas (fack, ratehalving, the concept of rhcwnd) in this paper made it
>> into
>> the Linux TCP implementation in some form, which means they're part of a
>> big
>> chunk of today's web servers and other Internet hosts. Why has there
>> been no
>> interest in taking this draft further?
>> 
>> Cheers,
>> Carsten
>> 
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


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From: Alexander Zimmermann <alexander.zimmermann@nets.rwth-aachen.de>
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Subject: Re: [tcpm] early retransmit done?
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Hi Mark,

here my

*** Comments:

* Line 150-153. On Joe's request, you added some lines about ACK compression...
Ah sorry, Delay ACKs ;-). The first new sentence is fine and IMHO sufficient.
The second one confuse me (a litte bit). Why do you write "we assume that
receivers send immediate ACKs when there is a gap in the received sequence
space"? If a TCP would not send a DUPACK directly when OOO segment is received,
Early Rexmit wouldn't work? Maybe one or two sentence more could me "un-puzzle".

*  Line 252-253. 'We call this reduced ACK threshold enabling "Early
Retransimission"'. Maybe it's my limited english knowledge, you are the native
speaker, but IMHO the sentence sounds strange. What about this: "We call this
reduced duplicate ACK threshold "Early Retransmit".

* Line 222-224. In my last review I mentioned that we could probably better
specify the "entry point" of the algorithm. Example:

   Upon the arrival of an ACK, a sender employing byte-based Early
   Retransmit MUST use the following two conditions to determine
   when an Early Retransmit is sent:

* Line 226. Concerning Joe's question where magic *4* comes from. What do you
think about this?

   (2.a) The amount of outstanding data (ownd)---data sent but not yet
         acknowledged---is less or equal than 3*SMSS bytes, where is
         the lower bound of DupThresh from [RFC5681, RFC3517].


Mark, the last two point are really peanuts. If you don't like them, skip them.

*** Nits:
Missing Reference: 'RFC4653' is mentioned on line 419, but not defined

There are 5 instances of too long lines in the document, the longest one
being 2 characters in excess of 72.


*** Misspellings:
Line 91: s/implemention/implementation
Line 253: s/Retransimission/Retransmission (I know that it was Joe's misspelling ;-))
Line 312: s/Retransimission/Retransmission


Am 27.10.2009 um 15:30 schrieb Mark Allman:

> 
> We believe we have addressed all comments we received on the early
> retranmsit ID.  I just went to submit a new version and have missed the
> cut-off.  I have set a bit to submit when the system opens again on
> Nov/9.  However, until then I put the version I was going to submit here
> ...
> 
>  http://www.icir.org/mallman/papers/draft-ietf-tcpm-early-rexmt-02a.txt
> 
> Comments are very welcome.
> 
> allman
> 
> 
> 
> <ATT00001>_______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


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Subject: Re: [tcpm] early retransmit done?
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> * Line 150-153. On Joe's request, you added some lines about ACK compression...
> Ah sorry, Delay ACKs ;-). The first new sentence is fine and IMHO
> sufficient.  The second one confuse me (a litte bit). Why do you write
> "we assume that receivers send immediate ACKs when there is a gap in
> the received sequence space"? If a TCP would not send a DUPACK
> directly when OOO segment is received, Early Rexmit wouldn't work?
> Maybe one or two sentence more could me "un-puzzle".

I don't follow this.  The point here is that delayed ACKs could perhaps
reduce the number of ACKs the sender gets back.  But, a counter-point
(per RFC5681) is that when ACKing out-of-order arrivals the receiver
SHOULD NOT be using delayed ACKs.  I just read the text a few times and
I think it is OK.  But, I might be too close to it and so am open to
suggestions. 

> *  Line 252-253. 'We call this reduced ACK threshold enabling "Early
> Retransimission"'. Maybe it's my limited english knowledge, you are
> the native speaker, but IMHO the sentence sounds strange. What about
> this: "We call this reduced duplicate ACK threshold "Early
> Retransmit".

This language was added for Joe.  I don't think it is either necessary
or harmful.  And, I am not going to get into making excessively small
tweaks to it.

> * Line 222-224. In my last review I mentioned that we could probably better
> specify the "entry point" of the algorithm. Example:
> 
>    Upon the arrival of an ACK, a sender employing byte-based Early
>    Retransmit MUST use the following two conditions to determine
>    when an Early Retransmit is sent:

OK.

> * Line 226. Concerning Joe's question where magic *4* comes from. What do you
> think about this?
> 
>    (2.a) The amount of outstanding data (ownd)---data sent but not yet
>          acknowledged---is less or equal than 3*SMSS bytes, where is
>          the lower bound of DupThresh from [RFC5681, RFC3517].

I didn't address this directly because I think it is clear from the
entire document why the number is what it is.  And, as I understood it
this was a minor comment from Joe.  The above suggested sentence does
not parse for me.

> Missing Reference: 'RFC4653' is mentioned on line 419, but not defined

Added.

> *** Misspellings:
> Line 91: s/implemention/implementation
> Line 253: s/Retransimission/Retransmission (I know that it was Joe's misspelling ;-))
> Line 312: s/Retransimission/Retransmission

Fixed.

I know the chairs are getting ready to forward this.  I can send off
another version just before they do that encompasses this and any other
feedback folks want to send.

Thanks!

allman




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> My wish is that it would be slightly more explicit in how to handle a
> FIN (only) segment.

So, the issue here is that a FIN with no data could be useful for ER?

So, ...

  (1) You send two data packets plus a pure FIN.

  (2) Lose the second data packet.

  (3) Get a cumulative ACK for the first data packet.

  (4) Get a FIN-triggered duplicate ACK.

At this point the sender will think there is one outstanding segment and
set the dupthresh to zero and so not really useful in using the dupack
as a trigger for a retransmit.

I basically don't see anything to be done here short of making the
sender explicitly track that a pure FIN is outstanding.

My inclination here is that this is enough of a corner case that the
document is fine how it is.  (Note in the particular scenario you played
out the TCP implementation may or may not send a payload-less FIN.
Depending on the timing and etc. it could well be tacked on the second
data packet.)  That said, if the WG thinks it is important to address
this corner case at the last minute then we will obviously do so.

allman




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From ilpo.jarvinen@helsinki.fi  Wed Nov 11 17:19:16 2009
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Subject: Re: [tcpm] early retransmit done?
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On Wed, 11 Nov 2009, Mark Allman wrote:

> > My wish is that it would be slightly more explicit in how to handle a
> > FIN (only) segment.
> 
> So, the issue here is that a FIN with no data could be useful for ER?

Yes.

> So, ...
> 
>   (1) You send two data packets plus a pure FIN.
> 
>   (2) Lose the second data packet.
> 
>   (3) Get a cumulative ACK for the first data packet.
> 
>   (4) Get a FIN-triggered duplicate ACK.
> 
> At this point the sender will think there is one outstanding segment and
> set the dupthresh to zero and so not really useful in using the dupack
> as a trigger for a retransmit.

Is it really that well defined that pure FINs are not "segments"? 
Probably this too is implementation dpeending whether outstanding 
segments (already) include FIN or not.

> I basically don't see anything to be done here short of making the
> sender explicitly track that a pure FIN is outstanding.

My guess is that most of the implentations will know this already 
(indirectly if not even directly). Nevertheless, I don't see what is the 
harm if FIN is included to "outstanding" segments/sequences.

> My inclination here is that this is enough of a corner case that the
> document is fine how it is.

...and you still boldly claim in the intro that "A particular case when 
all connections become application limited is as the connection ends"? :-)

> (Note in the particular scenario you played
> out the TCP implementation may or may not send a payload-less FIN.
> Depending on the timing and etc. it could well be tacked on the second
> data packet.)

That's certainly possible yes.

But I would argue that if early retransmit would be made useful here, 
short transfers would greatly benefit from sending it separately because 
of the increased likelyhoods of avoiding RTO at the end which is very 
significant for their latency.

> That said, if the WG thinks it is important to address
> this corner case at the last minute then we will obviously do so.


-- 
 i.

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From: Mark Allman <mallman@icir.org>
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Subject: Re: [tcpm] early retransmit done?
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> > So, ...
> > 
> >   (1) You send two data packets plus a pure FIN.
> > 
> >   (2) Lose the second data packet.
> > 
> >   (3) Get a cumulative ACK for the first data packet.
> > 
> >   (4) Get a FIN-triggered duplicate ACK.
> > 
> > At this point the sender will think there is one outstanding segment and
> > set the dupthresh to zero and so not really useful in using the dupack
> > as a trigger for a retransmit.
> 
> Is it really that well defined that pure FINs are not "segments"?

I didn't mean to say or indicate that.

> > I basically don't see anything to be done here short of making the
> > sender explicitly track that a pure FIN is outstanding.
> 
> My guess is that most of the implentations will know this already
> (indirectly if not even directly). Nevertheless, I don't see what is
> the harm if FIN is included to "outstanding" segments/sequences.

I would be surprised if stacks retain an understanding of whether they
transmitted a FIN with a data packet or without.  There isn't harm, per
se, except for complexity (and perhaps a small bit of extra state).  

> > My inclination here is that this is enough of a corner case that the
> > document is fine how it is.
> 
> ...and you still boldly claim in the intro that "A particular case
> when all connections become application limited is as the connection
> ends"? :-)

Yeah... I don't see that at odds with calling the separate-FIN a corner
case. 

> > (Note in the particular scenario you played out the TCP
> > implementation may or may not send a payload-less FIN.  Depending on
> > the timing and etc. it could well be tacked on the second data
> > packet.)
>
> That's certainly possible yes.
>
> But I would argue that if early retransmit would be made useful here,
> short transfers would greatly benefit from sending it separately
> because of the increased likelyhoods of avoiding RTO at the end which
> is very significant for their latency.

That's not a half bad thought---although it still to me seems like it'd
help only a very small fraction of things.

So, how about a note in section 2.2 that just says an implementation
could also track a dataless FIN as an outstanding segment if explicitly
tracking outstanding segments?  Will that work?  

allman




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From ilpo.jarvinen@helsinki.fi  Thu Nov 12 21:27:49 2009
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Cc: k.avrachenkov@sophia.inria.fr, tcpm@ietf.org, jblanton@cs.ohiou.edu
Subject: Re: [tcpm] early retransmit done?
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On Wed, 11 Nov 2009, Mark Allman wrote:

> 
> > > So, ...
> > > 
> > >   (1) You send two data packets plus a pure FIN.
> > > 
> > >   (2) Lose the second data packet.
> > > 
> > >   (3) Get a cumulative ACK for the first data packet.
> > > 
> > >   (4) Get a FIN-triggered duplicate ACK.
> > > 
> > > At this point the sender will think there is one outstanding segment and
> > > set the dupthresh to zero and so not really useful in using the dupack
> > > as a trigger for a retransmit.
> > 
> > Is it really that well defined that pure FINs are not "segments"?
> 
> I didn't mean to say or indicate that.
>
> > > I basically don't see anything to be done here short of making the
> > > sender explicitly track that a pure FIN is outstanding.
> > 
> > My guess is that most of the implentations will know this already
> > (indirectly if not even directly). Nevertheless, I don't see what is
> > the harm if FIN is included to "outstanding" segments/sequences.
> 
> I would be surprised if stacks retain an understanding of whether they
> transmitted a FIN with a data packet or without.  There isn't harm, per
> se, except for complexity (and perhaps a small bit of extra state).  

I don't perhaps fully follow what is the problem here. If it wasn't sent 
separately, there won't be that dup ACK then either because both FIN and 
final data are either lost or delivered in unison. But then it would have 
been counted as one outstanding segment anyway (if it was counted as two 
for some reason, I'd say the implementation is just broken).

Secondly, if one counts window in segments, I cannot see how it can be 
done robustly without knowing segment boundaries (after all, negative 
in-flight isn't that nice thing) and there is very little need to create 
exception for FIN but treat its boundary the same way. ...Plus I suppose 
the implentation still wants to know that FIN was acked :-). Once segment 
boundaries are known, it is insignificant in context of early rexmit 
whether the last segment is pure-FIN or not..

> > > My inclination here is that this is enough of a corner case that the
> > > document is fine how it is.
> > 
> > ...and you still boldly claim in the intro that "A particular case
> > when all connections become application limited is as the connection
> > ends"? :-)
> 
> Yeah... I don't see that at odds with calling the separate-FIN a corner
> case. 
> 
> > > (Note in the particular scenario you played out the TCP
> > > implementation may or may not send a payload-less FIN.  Depending on
> > > the timing and etc. it could well be tacked on the second data
> > > packet.)
> >
> > That's certainly possible yes.
> >
> > But I would argue that if early retransmit would be made useful here,
> > short transfers would greatly benefit from sending it separately
> > because of the increased likelyhoods of avoiding RTO at the end which
> > is very significant for their latency.
> 
> That's not a half bad thought---although it still to me seems like it'd
> help only a very small fraction of things.
>
> So, how about a note in section 2.2 that just says an implementation
> could also track a dataless FIN as an outstanding segment if explicitly
> tracking outstanding segments?  Will that work?  

A note sounds reasonable to me. 

-- 
 i.

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> > I would be surprised if stacks retain an understanding of whether they
> > transmitted a FIN with a data packet or without.  There isn't harm, per
> > se, except for complexity (and perhaps a small bit of extra state).  
> 
> I don't perhaps fully follow what is the problem here. If it wasn't
> sent separately, there won't be that dup ACK then either because both
> FIN and final data are either lost or delivered in unison. But then it
> would have been counted as one outstanding segment anyway (if it was
> counted as two for some reason, I'd say the implementation is just
> broken).
> 
> Secondly, if one counts window in segments, I cannot see how it can be
> done robustly without knowing segment boundaries (after all, negative
> in-flight isn't that nice thing) and there is very little need to
> create exception for FIN but treat its boundary the same way. ...Plus
> I suppose the implentation still wants to know that FIN was acked
> :-). Once segment boundaries are known, it is insignificant in context
> of early rexmit whether the last segment is pure-FIN or not..

I don't really follow this at all.

If you are tracking segments and include the implicit sequence number
consumed by the FIN then this all just works, I guess.  Is that what
you're saying?

allman




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Thread-Topic: [tcpm] Detect Lost Retransmit mit SACK
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Subject: Re: [tcpm] Detect Lost Retransmit mit SACK
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Hi Arnd,

Sorry for the late response. I did some further investigations=20
lately, and this is what I came up with.

Of course, the majority of work and thoughs have been done by=20
others (most notably, I want to cite [[1]] "Lost Retransmission=20
Detection for TCP Part 2: TCP Using SACK Option" by=20
B. Kim et al (2004), but also by some ideas from [[2]] M. Mathis=20
(dated) RateHalving NetBSD based code found here=20
http://www.psc.edu/networking/projects/rate-halving/=20
(which did something similar but only for a single segment =20
for the SACK recovery generation, if I can tell...)

First, the following prerequisites are required:

1) TCP Session must be SACK-enabled
2) TCP Sender has to have FACK implemented=20
3) TCP Sender has to have a scoreboard implementation to=20
   track the state of the receiver, working with a sorted=20
   list of SACK-Holes rather than positive SACK-Ranges.


The fix might not be mathematically perfect in tracking each single=20
segment, but in it's simplest incarnation, it's about 8 lines of=20
code, adding minimal constant time (!) during ACK-processing.=20

The idea is basically, not to track every sent out segment=20
individually, but rather all holes in the scoreboard at once.

The implicit assumption here is, that retransmissions get sent out
in order (and are received with little to no reordering, or not=20
at all).

When the sender transmits the last segment to "close a hole" in the=20
sender's representation of the scoreboard, represented by a data=20
structure holding start, end and rxmit vector, ie. When rxmit =3D=3D =
end,=20
rxmit will be assinged the current value of snd.max. (In the simple=20
form, a separate variable iskept, but for the purpose of this=20
algorithm, the condition of rxmit <=3D end vs rxmit > end can be used=20
to differentiate between the two uses (current retransmission vector=20
in a hole, vs. segment number by which the hole should be reported=20
closed by the receiver) of rxmit.

So, after the retransmission of the last segment of a hole, it's=20
rxmit will show the expected snd.fack value when that hole is ought=20
to have been properly ACKed by the receiver.

In SACK processing, some precautions have to be made during hole-
splitting (to keep the higher than sackhole.end rxmit vector).=20

Normal SACK processing will remove any fully closed holes before=20
snd.fack becomes greater than sendhole.rxmit, except when one of=20
the segments within that hole is never seen by the receiver. While=20
processing a normal SACK, check the condition of=20
snd.fack > sackhole[0].rxmit;=20

In the worst case (1 segment hole), when some reordering occurred,=20
this will yield one superfluous segment; with larger holes, the=20
intrinsic reordering grace period granted is n-m segments (with n=20
the size of the hole, and m the segment within the hole).

If that condition is met, reset sackhole[0].rxmit to=20
sackhole[0].start, and also update the shortcut lookup pointer=20
(sackhint.nexthole to sackhole[0]).


Extended periods of heavy loss are not dealt with. According to=20
[[1]], these should be counted as additional congestion events,=20
reducing ssthresh and cwnd accordingly; but this method needs a=20
constant stream of at least some new segments to leave the=20
Network).
Thus, RTO + slow start is required for recovery of such severe
Events.

Also, when there is no more data from the socket to transmit,=20
to never exceed snd.max recorded for that hole, it will not=20
work. (Again, RTO to the rescue).

But, it will update the rxmit vector for each fully retransmitted=20
hole - as long as it keeps going. This yields a maximum number or=20
RTO/RTT retries, before "usual" recovery sets in, and should=20
cover the common case.


You could say it's more of a quick hack, but I think the=20
simplicity, intrinsic properties and near no overhead (processing=20
time or variable overhead) are well balanced, but if my assumptions
are incorrect or short sighted, please say so.


The simple version without any tweaks in existing checks of=20
sackhole[]->rxmit (requiring one more tcp_seq var in sackhole[]):

Tcp_sack_doack:

(after sanity checks about the new SACK blocks, BEFORE new snd.fack=20
has been calculated (one additional ACK grace period):
=20
	if (((temp =3D TAILQ_FIRST(&tp->snd_holes)) !=3D NULL) && //###
	     (SEQ_GT(tp->snd_fack, temp->limit))) {           //###
			temp->rxmit =3D temp->start;                //###
			temp->limit =3D tp->snd_max;                //###
			tp->sackhint.nexthole =3D temp;             //###
		}
//###

Tcp_sack_doack:

When splitting a hole in two:

		temp =3D tcp_sackhole_insert(tp, sblkp->end,
				    cur->end, cur);
				if (temp !=3D NULL) {
					if (SEQ_GT(cur->rxmit,
temp->rxmit)) {
						temp->rxmit =3D
cur->rxmit;
=09
tp->sackhint.sack_bytes_rexmit
						    +=3D (temp->rxmit
						    - temp->start);
					}
					temp->limit =3D cur->limit;
//###
					cur->end =3D sblkp->start;
					// RS: SACK+ mod?
					cur->rxmit =3D SEQ_MIN(cur->rxmit,
					    cur->end);
				}


Tcp_output:

(when in Retransmit && SACK mode):

If (sack_rxmit =3D=3D 1) {
		th->th_seq =3D htonl(p->rxmit);
		p->rxmit +=3D len;
		if (SEQ_GEQ(p->rxmit, p->end))
//###
			p->limit =3D tp->snd_max;
//###
		tp->sackhint.sack_bytes_rexmit +=3D len;
}

Tcp_var.h:
struct sackhole {
	tcp_seq start;		/* start seq no. of hole */
	tcp_seq end;		/* end seq no. */
	tcp_seq rxmit;		/* next seq. no in hole to be=20
                                 retransmitted */
	tcp_seq limit;          /* ### for lost retransmission=20
                                 detection - should be rolled=20
                                 into rxmit */
	TAILQ_ENTRY(sackhole) scblink;	/* scoreboard linkage */
};


Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20
Franz-Klein-Gasse 5
1190 Wien=20

=20

> -----Original Message-----
> From: Arnd Hannemann [mailto:hannemann@i4.informatik.rwth-aachen.de]=20
> Sent: Mittwoch, 11. November 2009 09:46
> To: tcpm@ietf.org
> Subject: Re: [tcpm] Detect Lost Retransmit mit SACK
>=20
> Hi Richard,
>=20
> Scheffenegger, Richard schrieb:
> >>>   A1) With TSOpt, they could be used to "remember" when=20
> >>>       Retransmission started;
> >>>       When an ACKs with TSEcn > (TSOpt + RTT) is seen by=20
> >>>       the sender, it can
> >>>       re-arm the DUPACK detector.
> >>>      =20
> >> This is your non-SACK scenario?
> >> Please note that a TCP receiver will NOT echo TSVal from=20
> out-of-order=20
> >> segments. So this won't work.
> >>    =20
> >
> > Interesting; this doesn't seem to be what certain (most?)=20
> stacks are=20
> > doing - they seem to always reply in the ACK Tsecr the value of=20
> > Tsopt/TS.recent triggering the ACK...
>=20
> Please clarify, is this non-SACK scenario?
> The only lost retransmission we could be talking about then,=20
> is the retransmission of SND.UNA.
> And the TCP receiver will NOT adjust TS.recent for any out-of=20
> order packets (RFC 1323).
> It will continue to echo the timestamp of the last segment=20
> which advanced the window (SND.UNA -1p).
>=20
>=20
> > Or was your intention to say, that the receiver will not=20
> respond with=20
> > Tsopt, but rather TS.recent?
>=20
> Yes, it will not respond with TSVal but with TS.recent.
>=20
> > If so, than this makes no difference - the sender will detect the=20
> > first ACK from the receiver, which has the same TS (well, actually=20
> > TS+1 as multiple segments are likely to carry the same TS) when the=20
> > retransmission
>=20
> There won't be an ACK if the retransmission is lost in=20
> non-SACK case, only dupACKs.
> (Only retransmission is retransmission of SND.UNA) TS.Recent=20
> will not get updated.
>=20
>=20
> > was started. This will be the sign for the sender, that at=20
> least one=20
> > RTT has elapsed (that was the intention for having=20
> timestamps in the=20
> > first place, right?). If the returned ACK does not cover the first=20
> > retransmitted segment by that time, it's quite reasonable to assume=20
> > that it got lost again
> >  =20
>=20
> > Again, keep in mind that I'm looking on LANs; a TS in 10 ms=20
> increments
>=20
> If you are searching for something which is not intended for=20
> general use, why not just set MIN_RTO to say 10 ms?
>=20
> >>>   A2) With SACK, tracking SND.NXT and HOLEBYTES (number of=20
> >>>       octets in all current holes)
> >>>       can be employed to track RTT (they need to be=20
> >>>       initialized when the first
> >>>       retransmitted segment is sent). When an ACK contains=20
> >>>       a SACK for >=3D SND.NXT, or the
> >>>       HOLEBYTES are smaller than when retransmission started, the=20
> >>>       DUPACK detector can be re-armed.
> >>>      =20
> >> You should specify what you mean with SND.NXT.
> >> We are in recovery, so we may very well send out new segments.
> >> I don't see why number of holes in sack scoreboard has=20
> anything todo=20
> >> with RTT.
> >>    =20
> >
> >
> > SND.NXT is the rightmost segment which the sender has never before=20
> > sent out to the network. Isn't that the meaning of Snd.nxt=20
> according=20
> > to RFC793?
>=20
> Hmm, I would interpret RFC793 differently.
> I would call it SND.MAX (see RFC4015) or HighData (RFC3517)=20
> the rightmost segment which the sender has never sent before.=20
> But now you have to clarify how it would be possible to=20
> receive a SACK block > SND.MAX. Even SACK block =3D=3D SND.MAX=20
> will be rarely seen.
> But before taking this discussion even further, maybe take=20
> some time and write some preliminary draft of your algorithm.
>=20
> > And I was not talking about the # of holes, but rather the=20
> amount of=20
> > Data (in sum) between FACK (defined by the rightmost offset in the=20
> > SACK Scoreboard, i.e. covering the highest ever received=20
> segment) and=20
> > SND.UNA.
> >
> > Monitoring Holes will gain nothing (when there is reordering for=20
> > example);
> >
> > But if the any one retransmission makes it though, the amount of=20
> > unSACKed data in the sum total of all holes in the scoreboard, will=20
> > decrease - indicating when at least one RTT has passed=20
> since start of=20
> > FastRetransmit.
>=20
> This is not true.
> Consider a 100 packet in flight, you get 3 dupacks, do  a=20
> retransmission.
> This is in your words "start of FastRetransmit". After this=20
> the sum total of all holes between SND.UNA and FACK can only=20
> increase for one RTT.
> Due to Limited Transmit and Fast Recovery, it might be=20
> possible that even the sum total of all holes between SND.UNA=20
> and SND.MAX will increase.
> You know when the first Retransmit is lost when a SACK covers=20
> RecoveryPoint (RFC3517). But this is only true for first hole.
> To detect lost retransmissions of the other holes, you would=20
> need to store a separate HighData (RFC3617) for every retransmission.
> Maybe this is after all what you are trying to express?
> Perhaps it would be helpful if you could cook up some pseudo=20
> code, of your proposed algorithm.
>=20
> >>    =20
> >>> holes
> >>>    have been retransmitted once...=20
> >>>      =20
> >> Show me an example. where a lost retransmission is=20
> detectable, while=20
> >> the sender did not send out retransmissions for every other hole?
> >> Sounds weird.
> >>    =20
> >
> > Can I attach trace files on posts to this list, or shall I=20
> sent them=20
> > to you directly?
>=20
> Never mind, does not actually matter.
>=20
> >>>      =20
> >> This probably highly rates from case to case. If you have a huge=20
> >> bandwidth-delay product even SACK scoreboard handling will have=20
> >> computational complexity which might outweigh its benefits.
> >> As you have seem to have the equipment at hand, you could=20
> do some cpu=20
> >> load and throughput measurements with 10GbE and SACK=20
> versus non-SACK=20
> >> flows.
> >>    =20
> >
> > Well, I was doing some test like this:
> >
> >=20
> http://www.ibm.com/developerworks/linux/library/l-tcp-sack/index.html
> >
> > But the CPU impact (on top of what is required for 10G) was=20
> neglegible.
> > But I have to admit, I didn't had the time yet to test a SACK=20
> > scoreboard with every other byte marked mission (which=20
> could mean some=20
> > 0,5 mio entries).
> >
> > But it seems that most implementations feature sorted=20
> scoreboards of=20
> > limited
> > (16/128) size.
> >
> > Traversing those is - again in my case of 10G LANs - still=20
> many orders=20
> > of magnitude faster then waiting for RTO.
> >
> > I personally consider RTOs to be harmful (when there are=20
> other means=20
> > to recover with). :) (perhaps once I have worked through all the=20
> > issues raised here, I submit a draft with "TCP RTO=20
> considered harmful"=20
> > next April? :)
>=20
> Better submit a draft to mitigate the problems with it ;-)
>=20
> Best regards,
> Arnd
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>=20

From ilpo.jarvinen@helsinki.fi  Tue Nov 17 03:51:28 2009
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Cc: tcpm@ietf.org, k.avrachenkov@sophia.inria.fr, jblanton@cs.ohiou.edu
Subject: Re: [tcpm] early retransmit done?
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On Fri, 13 Nov 2009, Mark Allman wrote:

> 
> > > I would be surprised if stacks retain an understanding of whether they
> > > transmitted a FIN with a data packet or without.  There isn't harm, per
> > > se, except for complexity (and perhaps a small bit of extra state).  
> > 
> > I don't perhaps fully follow what is the problem here. If it wasn't
> > sent separately, there won't be that dup ACK then either because both
> > FIN and final data are either lost or delivered in unison. But then it
> > would have been counted as one outstanding segment anyway (if it was
> > counted as two for some reason, I'd say the implementation is just
> > broken).
> > 
> > Secondly, if one counts window in segments, I cannot see how it can be
> > done robustly without knowing segment boundaries (after all, negative
> > in-flight isn't that nice thing) and there is very little need to
> > create exception for FIN but treat its boundary the same way. ...Plus
> > I suppose the implentation still wants to know that FIN was acked
> > :-). Once segment boundaries are known, it is insignificant in context
> > of early rexmit whether the last segment is pure-FIN or not..
> 
> I don't really follow this at all.
> 
> If you are tracking segments and include the implicit sequence number
> consumed by the FIN then this all just works, I guess.  Is that what
> you're saying?

Yes. ...And there's more, I also tried to say that one has to track that 
FIN sequence number anyway in order to be able to discern that the FIN was 
acked.

-- 
 i.

From wesley.m.eddy@nasa.gov  Tue Nov 17 06:23:12 2009
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Date: Tue, 17 Nov 2009 08:23:06 -0600
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Hello; since TCPM didn't meet in Hiroshima, but we do have
several documents in the process of being finished up, and
a couple others in the process of starting, here's a quick
snapshot of the TCPM working group status and the actions
needed on each document; please let us know if we left
anything out:


Active WG Items
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

TCP Authentication Option
http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt
Milestone Target: Proposed Standard in April 2009
Action: In WGLC until 11/24/09

TCP Authentication Option Crypto
http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt
Milestone Target: Proposed Standard with AO spec
Action: In WGLC until 11/24/09

ICMP Attacks
http://www.ietf.org/id/draft-ietf-tcpm-icmp-attacks-06.txt
Milestone Target: Informational in July 2009
Action: WGLC started 9/1, ended 9/15 ... Fernando Gont to update based on W=
GLC

Early-Retransmit
http://www.ietf.org/id/draft-ietf-tcpm-early-rexmt-02.txt
Milestone Target: Experimental in July 2009
Action: WGLC finished & authors updated (action with chairs to forward to I=
ESG)

1323bis
http://www.ietf.org/id/draft-ietf-tcpm-1323bis-01.txt
Milestone Target: Proposed Standard in July 2009
Action: Revise & WGLC (action with D. Borman)

MSS Option
http://www.ietf.org/id/draft-ietf-tcpm-tcpmss-02.txt
Milestone Target: Proposed Standard in July 2009
Action: WGLC started 8/19, ended 9/8 (extended), being updated based on sev=
eral
        of the comments (action with D. Borman)

Urgent Pointer
http://www.ietf.org/id/draft-ietf-tcpm-urgent-data-01.txt
Milestone Target: Proposed Standard in January 2010
Action: Continue WG reviews; authors have indicated it is ready for WGLC;
        planned to start WGLC after AO WGLC completes

TCP Security
http://www.ietf.org/id/draft-ietf-tcpm-tcp-security-00.txt
Milestone Target: BCP in August 2010
Action: 00-draft / outline has been under active discussion; but not attain=
ed
          very clear consensus
        WG might begin working on recommendations while still looking for a
          possible compromise on outline (action with Fernando Gont)

Long Connectivity Disruptions
http://www.ietf.org/internet-drafts/draft-zimmermann-tcp-lcd-02.txt
(need to submit WG version)
Milestone Target: Experimental in October 2010
Action: need WG to read/discuss
        authors to submit revision (action with Zimmerman)

SACK Entry
http://www.ietf.org/id/draft-ietf-tcpm-sack-recovery-entry-00.txt
Milestone Target: Proposed Standard in October 2010
Action: need WG to read/discuss
        TBD how this relates to 3517


Other Proposals
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Persist
http://tools.ietf.org/id/draft-ananth-tcpm-persist-01.txt
Action: WG Review

Timestamps
http://tools.ietf.org/id/draft-gont-tcpm-tcp-timestamps-01.txt
Action: ???

Parameter Tuning
(no I-D yet)
Action: continue good discussion on mailing list
        presentation given in Stockholm

TCP Option for Transparent Middlebox Discovery
http://www.ietf.org/id/draft-knutsen-tcpm-middlebox-discovery-03.txt
Action: Needs WG review / discussion
        discussion on mailing list has been productive

NewReno Modification
http://www.ietf.org/internet-drafts/draft-nishida-newreno-modification-01.t=
xt
Action: need WG to read/discuss
        presentation given, but WG list not yet polled for adoption

Recent discussions opened on Rate-Halving
http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt
Action: mailing list discussion about the history of this, and whether
        it needs to be revived

--
Wes Eddy
MTI Systems


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Date: Tue, 17 Nov 2009 13:54:09 -0800
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From: Gregory Lebovitz <gregory.ietf@gmail.com>
To: "Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]" <wesley.m.eddy@nasa.gov>, Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] working group status 11/17/09
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should we also list here the current draft on NAT-Traversal for TCP-AO? It's
not a WG item, but it is being done as a possible (likely?) textual addition
to a WG item. comment?

Gregory.

On Tue, Nov 17, 2009 at 6:23 AM, Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE
CORP] <wesley.m.eddy@nasa.gov> wrote:

> Hello; since TCPM didn't meet in Hiroshima, but we do have
> several documents in the process of being finished up, and
> a couple others in the process of starting, here's a quick
> snapshot of the TCPM working group status and the actions
> needed on each document; please let us know if we left
> anything out:
>
>
> Active WG Items
> ===============
>
> TCP Authentication Option
> http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt
> Milestone Target: Proposed Standard in April 2009
> Action: In WGLC until 11/24/09
>
> TCP Authentication Option Crypto
> http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt
> Milestone Target: Proposed Standard with AO spec
> Action: In WGLC until 11/24/09
>
> ICMP Attacks
> http://www.ietf.org/id/draft-ietf-tcpm-icmp-attacks-06.txt
> Milestone Target: Informational in July 2009
> Action: WGLC started 9/1, ended 9/15 ... Fernando Gont to update based on
> WGLC
>
> Early-Retransmit
> http://www.ietf.org/id/draft-ietf-tcpm-early-rexmt-02.txt
> Milestone Target: Experimental in July 2009
> Action: WGLC finished & authors updated (action with chairs to forward to
> IESG)
>
> 1323bis
> http://www.ietf.org/id/draft-ietf-tcpm-1323bis-01.txt
> Milestone Target: Proposed Standard in July 2009
> Action: Revise & WGLC (action with D. Borman)
>
> MSS Option
> http://www.ietf.org/id/draft-ietf-tcpm-tcpmss-02.txt
> Milestone Target: Proposed Standard in July 2009
> Action: WGLC started 8/19, ended 9/8 (extended), being updated based on
> several
>        of the comments (action with D. Borman)
>
> Urgent Pointer
> http://www.ietf.org/id/draft-ietf-tcpm-urgent-data-01.txt
> Milestone Target: Proposed Standard in January 2010
> Action: Continue WG reviews; authors have indicated it is ready for WGLC;
>        planned to start WGLC after AO WGLC completes
>
> TCP Security
> http://www.ietf.org/id/draft-ietf-tcpm-tcp-security-00.txt
> Milestone Target: BCP in August 2010
> Action: 00-draft / outline has been under active discussion; but not
> attained
>          very clear consensus
>        WG might begin working on recommendations while still looking for a
>          possible compromise on outline (action with Fernando Gont)
>
> Long Connectivity Disruptions
> http://www.ietf.org/internet-drafts/draft-zimmermann-tcp-lcd-02.txt
> (need to submit WG version)
> Milestone Target: Experimental in October 2010
> Action: need WG to read/discuss
>        authors to submit revision (action with Zimmerman)
>
> SACK Entry
> http://www.ietf.org/id/draft-ietf-tcpm-sack-recovery-entry-00.txt
> Milestone Target: Proposed Standard in October 2010
> Action: need WG to read/discuss
>        TBD how this relates to 3517
>
>
> Other Proposals
> ===============
>
> Persist
> http://tools.ietf.org/id/draft-ananth-tcpm-persist-01.txt
> Action: WG Review
>
> Timestamps
> http://tools.ietf.org/id/draft-gont-tcpm-tcp-timestamps-01.txt
> Action: ???
>
> Parameter Tuning
> (no I-D yet)
> Action: continue good discussion on mailing list
>        presentation given in Stockholm
>
> TCP Option for Transparent Middlebox Discovery
> http://www.ietf.org/id/draft-knutsen-tcpm-middlebox-discovery-03.txt
> Action: Needs WG review / discussion
>        discussion on mailing list has been productive
>
> NewReno Modification
>
> http://www.ietf.org/internet-drafts/draft-nishida-newreno-modification-01.txt
> Action: need WG to read/discuss
>        presentation given, but WG list not yet polled for adoption
>
> Recent discussions opened on Rate-Halving
> http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt
> Action: mailing list discussion about the history of this, and whether
>        it needs to be revived
>
> --
> Wes Eddy
> MTI Systems
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>



-- 
----
IETF related email from
Gregory M. Lebovitz
Juniper Networks

--001485e29a4c760a250478982aa1
Content-Type: text/html; charset=ISO-8859-1
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should we also list here the current draft on NAT-Traversal for TCP-AO? It&=
#39;s not a WG item, but it is being done as a possible (likely?) textual a=
ddition to a WG item. comment?<div><br></div><div>Gregory.<br><div><br><div=
 class=3D"gmail_quote">
On Tue, Nov 17, 2009 at 6:23 AM, Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE =
CORP] <span dir=3D"ltr">&lt;<a href=3D"mailto:wesley.m.eddy@nasa.gov">wesle=
y.m.eddy@nasa.gov</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hello; since TCPM didn&#39;t meet in Hiroshima, but we do have<br>
several documents in the process of being finished up, and<br>
a couple others in the process of starting, here&#39;s a quick<br>
snapshot of the TCPM working group status and the actions<br>
needed on each document; please let us know if we left<br>
anything out:<br>
<br>
<br>
Active WG Items<br>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<br>
<br>
TCP Authentication Option<br>
<a href=3D"http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt" targ=
et=3D"_blank">http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt</a=
><br>
Milestone Target: Proposed Standard in April 2009<br>
Action: In WGLC until 11/24/09<br>
<br>
TCP Authentication Option Crypto<br>
<a href=3D"http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt" tar=
get=3D"_blank">http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt<=
/a><br>
Milestone Target: Proposed Standard with AO spec<br>
Action: In WGLC until 11/24/09<br>
<br>
ICMP Attacks<br>
<a href=3D"http://www.ietf.org/id/draft-ietf-tcpm-icmp-attacks-06.txt" targ=
et=3D"_blank">http://www.ietf.org/id/draft-ietf-tcpm-icmp-attacks-06.txt</a=
><br>
Milestone Target: Informational in July 2009<br>
Action: WGLC started 9/1, ended 9/15 ... Fernando Gont to update based on W=
GLC<br>
<br>
Early-Retransmit<br>
<a href=3D"http://www.ietf.org/id/draft-ietf-tcpm-early-rexmt-02.txt" targe=
t=3D"_blank">http://www.ietf.org/id/draft-ietf-tcpm-early-rexmt-02.txt</a><=
br>
Milestone Target: Experimental in July 2009<br>
Action: WGLC finished &amp; authors updated (action with chairs to forward =
to IESG)<br>
<br>
1323bis<br>
<a href=3D"http://www.ietf.org/id/draft-ietf-tcpm-1323bis-01.txt" target=3D=
"_blank">http://www.ietf.org/id/draft-ietf-tcpm-1323bis-01.txt</a><br>
Milestone Target: Proposed Standard in July 2009<br>
Action: Revise &amp; WGLC (action with D. Borman)<br>
<br>
MSS Option<br>
<a href=3D"http://www.ietf.org/id/draft-ietf-tcpm-tcpmss-02.txt" target=3D"=
_blank">http://www.ietf.org/id/draft-ietf-tcpm-tcpmss-02.txt</a><br>
Milestone Target: Proposed Standard in July 2009<br>
Action: WGLC started 8/19, ended 9/8 (extended), being updated based on sev=
eral<br>
 =A0 =A0 =A0 =A0of the comments (action with D. Borman)<br>
<br>
Urgent Pointer<br>
<a href=3D"http://www.ietf.org/id/draft-ietf-tcpm-urgent-data-01.txt" targe=
t=3D"_blank">http://www.ietf.org/id/draft-ietf-tcpm-urgent-data-01.txt</a><=
br>
Milestone Target: Proposed Standard in January 2010<br>
Action: Continue WG reviews; authors have indicated it is ready for WGLC;<b=
r>
 =A0 =A0 =A0 =A0planned to start WGLC after AO WGLC completes<br>
<br>
TCP Security<br>
<a href=3D"http://www.ietf.org/id/draft-ietf-tcpm-tcp-security-00.txt" targ=
et=3D"_blank">http://www.ietf.org/id/draft-ietf-tcpm-tcp-security-00.txt</a=
><br>
Milestone Target: BCP in August 2010<br>
Action: 00-draft / outline has been under active discussion; but not attain=
ed<br>
 =A0 =A0 =A0 =A0 =A0very clear consensus<br>
 =A0 =A0 =A0 =A0WG might begin working on recommendations while still looki=
ng for a<br>
 =A0 =A0 =A0 =A0 =A0possible compromise on outline (action with Fernando Go=
nt)<br>
<br>
Long Connectivity Disruptions<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-zimmermann-tcp-lcd-02.=
txt" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-zimmermann=
-tcp-lcd-02.txt</a><br>
(need to submit WG version)<br>
Milestone Target: Experimental in October 2010<br>
Action: need WG to read/discuss<br>
 =A0 =A0 =A0 =A0authors to submit revision (action with Zimmerman)<br>
<br>
SACK Entry<br>
<a href=3D"http://www.ietf.org/id/draft-ietf-tcpm-sack-recovery-entry-00.tx=
t" target=3D"_blank">http://www.ietf.org/id/draft-ietf-tcpm-sack-recovery-e=
ntry-00.txt</a><br>
Milestone Target: Proposed Standard in October 2010<br>
Action: need WG to read/discuss<br>
 =A0 =A0 =A0 =A0TBD how this relates to 3517<br>
<br>
<br>
Other Proposals<br>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<br>
<br>
Persist<br>
<a href=3D"http://tools.ietf.org/id/draft-ananth-tcpm-persist-01.txt" targe=
t=3D"_blank">http://tools.ietf.org/id/draft-ananth-tcpm-persist-01.txt</a><=
br>
Action: WG Review<br>
<br>
Timestamps<br>
<a href=3D"http://tools.ietf.org/id/draft-gont-tcpm-tcp-timestamps-01.txt" =
target=3D"_blank">http://tools.ietf.org/id/draft-gont-tcpm-tcp-timestamps-0=
1.txt</a><br>
Action: ???<br>
<br>
Parameter Tuning<br>
(no I-D yet)<br>
Action: continue good discussion on mailing list<br>
 =A0 =A0 =A0 =A0presentation given in Stockholm<br>
<br>
TCP Option for Transparent Middlebox Discovery<br>
<a href=3D"http://www.ietf.org/id/draft-knutsen-tcpm-middlebox-discovery-03=
.txt" target=3D"_blank">http://www.ietf.org/id/draft-knutsen-tcpm-middlebox=
-discovery-03.txt</a><br>
Action: Needs WG review / discussion<br>
 =A0 =A0 =A0 =A0discussion on mailing list has been productive<br>
<br>
NewReno Modification<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-nishida-newreno-modifi=
cation-01.txt" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-=
nishida-newreno-modification-01.txt</a><br>
Action: need WG to read/discuss<br>
 =A0 =A0 =A0 =A0presentation given, but WG list not yet polled for adoption=
<br>
<br>
Recent discussions opened on Rate-Halving<br>
<a href=3D"http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt" ta=
rget=3D"_blank">http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.tx=
t</a><br>
Action: mailing list discussion about the history of this, and whether<br>
 =A0 =A0 =A0 =A0it needs to be revived<br>
<font color=3D"#888888"><br>
--<br>
Wes Eddy<br>
MTI Systems<br>
<br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
</font></blockquote></div><br><br clear=3D"all"><br>-- <br>----<br>IETF rel=
ated email from<br>Gregory M. Lebovitz<br>Juniper Networks<br>
</div></div>

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Hi, Gregory,

Gregory Lebovitz wrote:
> should we also list here the current draft on NAT-Traversal for TCP-AO?
> It's not a WG item, but it is being done as a possible (likely?) textual
> addition to a WG item. comment?

It might be useful. I know there are a few people who said they might
post their thoughts on this shortly... (i.e., this is a reminder to them ;-)

Joe


> On Tue, Nov 17, 2009 at 6:23 AM, Eddy, Wesley M. (GRC-MS00)[ASRC
> AEROSPACE CORP] <wesley.m.eddy@nasa.gov <mailto:wesley.m.eddy@nasa.gov>>
> wrote:
> 
>     Hello; since TCPM didn't meet in Hiroshima, but we do have
>     several documents in the process of being finished up, and
>     a couple others in the process of starting, here's a quick
>     snapshot of the TCPM working group status and the actions
>     needed on each document; please let us know if we left
>     anything out:
> 
> 
>     Active WG Items
>     ===============
> 
>     TCP Authentication Option
>     http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt
>     Milestone Target: Proposed Standard in April 2009
>     Action: In WGLC until 11/24/09
> 
>     TCP Authentication Option Crypto
>     http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt
>     Milestone Target: Proposed Standard with AO spec
>     Action: In WGLC until 11/24/09
> 
>     ICMP Attacks
>     http://www.ietf.org/id/draft-ietf-tcpm-icmp-attacks-06.txt
>     Milestone Target: Informational in July 2009
>     Action: WGLC started 9/1, ended 9/15 ... Fernando Gont to update
>     based on WGLC
> 
>     Early-Retransmit
>     http://www.ietf.org/id/draft-ietf-tcpm-early-rexmt-02.txt
>     Milestone Target: Experimental in July 2009
>     Action: WGLC finished & authors updated (action with chairs to
>     forward to IESG)
> 
>     1323bis
>     http://www.ietf.org/id/draft-ietf-tcpm-1323bis-01.txt
>     Milestone Target: Proposed Standard in July 2009
>     Action: Revise & WGLC (action with D. Borman)
> 
>     MSS Option
>     http://www.ietf.org/id/draft-ietf-tcpm-tcpmss-02.txt
>     Milestone Target: Proposed Standard in July 2009
>     Action: WGLC started 8/19, ended 9/8 (extended), being updated based
>     on several
>            of the comments (action with D. Borman)
> 
>     Urgent Pointer
>     http://www.ietf.org/id/draft-ietf-tcpm-urgent-data-01.txt
>     Milestone Target: Proposed Standard in January 2010
>     Action: Continue WG reviews; authors have indicated it is ready for
>     WGLC;
>            planned to start WGLC after AO WGLC completes
> 
>     TCP Security
>     http://www.ietf.org/id/draft-ietf-tcpm-tcp-security-00.txt
>     Milestone Target: BCP in August 2010
>     Action: 00-draft / outline has been under active discussion; but not
>     attained
>              very clear consensus
>            WG might begin working on recommendations while still looking
>     for a
>              possible compromise on outline (action with Fernando Gont)
> 
>     Long Connectivity Disruptions
>     http://www.ietf.org/internet-drafts/draft-zimmermann-tcp-lcd-02.txt
>     (need to submit WG version)
>     Milestone Target: Experimental in October 2010
>     Action: need WG to read/discuss
>            authors to submit revision (action with Zimmerman)
> 
>     SACK Entry
>     http://www.ietf.org/id/draft-ietf-tcpm-sack-recovery-entry-00.txt
>     Milestone Target: Proposed Standard in October 2010
>     Action: need WG to read/discuss
>            TBD how this relates to 3517
> 
> 
>     Other Proposals
>     ===============
> 
>     Persist
>     http://tools.ietf.org/id/draft-ananth-tcpm-persist-01.txt
>     Action: WG Review
> 
>     Timestamps
>     http://tools.ietf.org/id/draft-gont-tcpm-tcp-timestamps-01.txt
>     Action: ???
> 
>     Parameter Tuning
>     (no I-D yet)
>     Action: continue good discussion on mailing list
>            presentation given in Stockholm
> 
>     TCP Option for Transparent Middlebox Discovery
>     http://www.ietf.org/id/draft-knutsen-tcpm-middlebox-discovery-03.txt
>     Action: Needs WG review / discussion
>            discussion on mailing list has been productive
> 
>     NewReno Modification
>     http://www.ietf.org/internet-drafts/draft-nishida-newreno-modification-01.txt
>     Action: need WG to read/discuss
>            presentation given, but WG list not yet polled for adoption
> 
>     Recent discussions opened on Rate-Halving
>     http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt
>     Action: mailing list discussion about the history of this, and whether
>            it needs to be revived
> 
>     --
>     Wes Eddy
>     MTI Systems
> 
>     _______________________________________________
>     tcpm mailing list
>     tcpm@ietf.org <mailto:tcpm@ietf.org>
>     https://www.ietf.org/mailman/listinfo/tcpm
> 
> 
> 
> 
> -- 
> ----
> IETF related email from
> Gregory M. Lebovitz
> Juniper Networks
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From: "Dan Wing" <dwing@cisco.com>
To: "'Joe Touch'" <touch@ISI.EDU>, "'tcpm Extensions WG'" <tcpm@ietf.org>
References: <4ADC7C85.4050408@isi.edu>
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> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On 
> Behalf Of Joe Touch
> Sent: Monday, October 19, 2009 7:50 AM
> To: tcpm Extensions WG
> Subject: [tcpm] [Fwd: New Version Notification for 
> draft-touch-tcp-ao-nat-00]
> 
> Hi, all,
> 
> The following provides info on the draft that describes the NAT
> extensions to TCP-AO. They have been submitted as individual for now,
> with the intent of incorporating the result into the TCP-AO 
> doc *if* we
> have time before publication and *if* there is consensus to do so.
> 
> Otherwise, it can published separately as either PS or 
> experimental, if preferred.

Thanks for publishing this document, Joe.  I would prefer to see it
incorporated into the base TCP-AO document, as there are legitimate
uses for TCP-AO for TCP clients behind a NAT44 or NAT64, such as to 
protect long-lived TLS-over-TCP sessions (SIP, XMPP, HTTP, and 
perhaps will be necessary for the forthcoming SmartGrids work).

-d



From dwing@cisco.com  Tue Nov 17 17:15:12 2009
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From: "Dan Wing" <dwing@cisco.com>
To: "'Dan Wing'" <dwing@cisco.com>, "'Joe Touch'" <touch@ISI.EDU>, "'tcpm Extensions WG'" <tcpm@ietf.org>
References: <4ADC7C85.4050408@isi.edu> 
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And to followup -- TCP-AO-NAT is a good idea.  As I have
explained in previous emails to TCPM, I would prefer it
work with a TCP server behind a NAT, as well as the TCP
client behind a NAT, but it's a reasonable trade-off in
complexity as most of the use cases where AO is important
have publicly-accessible TCP servers.

I'm sure some other folks have something to say about
TCP-AO working for a TCP-AO client behind a NAT44 or
behind a NAT64?

-d
 

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: Tuesday, November 17, 2009 3:37 PM
> To: 'Joe Touch'; 'tcpm Extensions WG'
> Subject: RE: [tcpm] [Fwd: New Version Notification for 
> draft-touch-tcp-ao-nat-00]
> 
>  
> 
> > -----Original Message-----
> > From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On 
> > Behalf Of Joe Touch
> > Sent: Monday, October 19, 2009 7:50 AM
> > To: tcpm Extensions WG
> > Subject: [tcpm] [Fwd: New Version Notification for 
> > draft-touch-tcp-ao-nat-00]
> > 
> > Hi, all,
> > 
> > The following provides info on the draft that describes the NAT
> > extensions to TCP-AO. They have been submitted as 
> individual for now,
> > with the intent of incorporating the result into the TCP-AO 
> > doc *if* we
> > have time before publication and *if* there is consensus to do so.
> > 
> > Otherwise, it can published separately as either PS or 
> > experimental, if preferred.
> 
> Thanks for publishing this document, Joe.  I would prefer to see it
> incorporated into the base TCP-AO document, as there are legitimate
> uses for TCP-AO for TCP clients behind a NAT44 or NAT64, such as to 
> protect long-lived TLS-over-TCP sessions (SIP, XMPP, HTTP, and 
> perhaps will be necessary for the forthcoming SmartGrids work).
> 
> -d
> 
> 


From rs@netapp.com  Wed Nov 18 03:18:00 2009
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From: "Scheffenegger, Richard" <rs@netapp.com>
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Hello Group,

Regarding Rate-Halving: It seems to me, that RFC4653 has a "lite"=20
version (Cautious Extended Limited Transmit) of that, and limiting=20
the scope to SACK-enabled streams only.

Also, RR-TCP uses Extended Limited Transmit (but the aggressive=20
variant) to keep the ACK clock going after potential congestion events.=20
But RR-TCP seems to be very much more complex than RFC4653, although=20
I need to investigate the gains/costs a bit further.

I wonder if the mentioned problems wouldn't impact those suggested=20
algorithms just like RH; OTOH, data handling subsystems are getting
Also faster, so that at least starvation during a bulk transfer (in
a dedicated system) should no longer be a hinderance. And worst case=20
operation wouldn't be so much different from NewReno anyway, or not?

Regards,

Richard Scheffenegger
Field Support Engineer
NetApp Global Support
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com
Franz-Klein-Gasse 5
1190 Wien


-----Original Message-----
From: Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]
[mailto:wesley.m.eddy@nasa.gov]=20
Sent: Dienstag, 17. November 2009 15:23
To: tcpm@ietf.org
Subject: [tcpm] working group status 11/17/09

Hello; since TCPM didn't meet in Hiroshima, but we do have several
documents in the process of being finished up, and a couple others in
the process of starting, here's a quick snapshot of the TCPM working
group status and the actions needed on each document; please let us know
if we left anything out:


Active WG Items
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

TCP Authentication Option
http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt
Milestone Target: Proposed Standard in April 2009
Action: In WGLC until 11/24/09

TCP Authentication Option Crypto
http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt
Milestone Target: Proposed Standard with AO spec
Action: In WGLC until 11/24/09

ICMP Attacks
http://www.ietf.org/id/draft-ietf-tcpm-icmp-attacks-06.txt
Milestone Target: Informational in July 2009
Action: WGLC started 9/1, ended 9/15 ... Fernando Gont to update based
on WGLC

Early-Retransmit
http://www.ietf.org/id/draft-ietf-tcpm-early-rexmt-02.txt
Milestone Target: Experimental in July 2009
Action: WGLC finished & authors updated (action with chairs to forward
to IESG)

1323bis
http://www.ietf.org/id/draft-ietf-tcpm-1323bis-01.txt
Milestone Target: Proposed Standard in July 2009
Action: Revise & WGLC (action with D. Borman)

MSS Option
http://www.ietf.org/id/draft-ietf-tcpm-tcpmss-02.txt
Milestone Target: Proposed Standard in July 2009
Action: WGLC started 8/19, ended 9/8 (extended), being updated based on
several
        of the comments (action with D. Borman)

Urgent Pointer
http://www.ietf.org/id/draft-ietf-tcpm-urgent-data-01.txt
Milestone Target: Proposed Standard in January 2010
Action: Continue WG reviews; authors have indicated it is ready for
WGLC;
        planned to start WGLC after AO WGLC completes

TCP Security
http://www.ietf.org/id/draft-ietf-tcpm-tcp-security-00.txt
Milestone Target: BCP in August 2010
Action: 00-draft / outline has been under active discussion; but not
attained
          very clear consensus
        WG might begin working on recommendations while still looking
for a
          possible compromise on outline (action with Fernando Gont)

Long Connectivity Disruptions
http://www.ietf.org/internet-drafts/draft-zimmermann-tcp-lcd-02.txt
(need to submit WG version)
Milestone Target: Experimental in October 2010
Action: need WG to read/discuss
        authors to submit revision (action with Zimmerman)

SACK Entry
http://www.ietf.org/id/draft-ietf-tcpm-sack-recovery-entry-00.txt
Milestone Target: Proposed Standard in October 2010
Action: need WG to read/discuss
        TBD how this relates to 3517


Other Proposals
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Persist
http://tools.ietf.org/id/draft-ananth-tcpm-persist-01.txt
Action: WG Review

Timestamps
http://tools.ietf.org/id/draft-gont-tcpm-tcp-timestamps-01.txt
Action: ???

Parameter Tuning
(no I-D yet)
Action: continue good discussion on mailing list
        presentation given in Stockholm

TCP Option for Transparent Middlebox Discovery
http://www.ietf.org/id/draft-knutsen-tcpm-middlebox-discovery-03.txt
Action: Needs WG review / discussion
        discussion on mailing list has been productive

NewReno Modification
http://www.ietf.org/internet-drafts/draft-nishida-newreno-modification-0
1.txt
Action: need WG to read/discuss
        presentation given, but WG list not yet polled for adoption

Recent discussions opened on Rate-Halving
http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt
Action: mailing list discussion about the history of this, and whether
        it needs to be revived

--
Wes Eddy
MTI Systems

_______________________________________________
tcpm mailing list
tcpm@ietf.org
https://www.ietf.org/mailman/listinfo/tcpm

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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 Maintenance and Minor Extensions Working Group of the IETF.


	Title           : Making TCP more Robust to Long Connectivity Disruptions (TCP-LCD)
	Author(s)       : A. Zimmermann, A. Hannemann
	Filename        : draft-ietf-tcpm-tcp-lcd-00.txt
	Pages           : 25
	Date            : 2009-11-17

Disruptions in end-to-end path connectivity, which last longer than
one retransmission timeout cause suboptimal TCP performance.  The
reason for the performance degradation is that TCP interprets segment
loss induced by long connectivity disruptions as a sign of
congestion, resulting in repeated retransmission timer backoffs.
This leads in turn to a deferred detection of the re-establishment of
the connection since TCP waits until the next retransmission timeout
occurs before attempting the retransmission.

This document proposes a algorithm for making TCP more robust to long
connectivity disruptions (TCP-LCD).  The memo describes how standard
ICMP messages can be exploited during timeout-based loss recovery to
disambiguate true congestion loss from non-congestion loss caused by
connectivity disruptions.  Moreover, a revert strategy of the
retransmission timer is specified that enables a more prompt
detection of whether the connectivity to a previously disconnected
peer node has been restored or not.  TCP-LCD is a TCP sender-only
modification that effectively improves TCP performance in presence of
connectivity disruptions.

Status of this Memo

This Internet-Draft is submitted to IETF in full conformance with the
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From alexander.zimmermann@nets.rwth-aachen.de  Wed Nov 18 08:22:02 2009
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From: Alexander Zimmermann <alexander.zimmermann@nets.rwth-aachen.de>
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Hi Folks,

the first working group version of the connectivity disruption draft is finished. In my opinion the document is feature complete. In particular, we reorganized and heavily extended the discussion section (ambiguity problems, wrapped sequence numbers, packet duplication) and introduced an interoperability issue section (ECN, Tunnels, ICMPv6, ...).

Alex

---

Changes from draft-zimmermann-tcp-lcd-02

   o  Incorporated feedback submitted by Ilpo Jarvinen.
      <http://www.ietf.org/mail-archive/web/tcpm/current/msg04841.html>

   o  Incorporated feedback submitted by Pasi Sarolahti.
      <http://www.ietf.org/mail-archive/web/tcpm/current/msg04870.html>

   o  Incorporated feedback submitted by Joe Touch.
      <http://www.ietf.org/mail-archive/web/tcpm/current/msg04895.html>
      <http://www.ietf.org/mail-archive/web/tcpm/current/msg04900.html>

   o  Extended and reorganized the discussion (Section 5):

      *  Every discussion item got its own title, so that we have a
         better overview.

      *  Extended Retransmission Ambiguity section.  Added also some
         references to the historical retransmission ambiguity problem.

      *  Heavily extended discussion about wrapped sequence numbers (see
         Joe's comments).

      *  Described the influence of packet duplication on the algorithm
         (Thanks to Ilpo).

      *  The section "Protecting Against Misbehaving Routers" is not a
         subsection anymore.  Moreover, the section was renamed to
         "Dissolving Ambiguity Issues" and has now real content.

   o  An interoperability issues section (Section 7) was added.  In
      particular comments to ECN, ICMPv6, and to the two thresholds R1
      and R2 of [RFC1122] (Section 4.2.3.5) were added.

   o  Miscellaneous editorial changes.  In particular, the algorithm has
      a name now: TCP-LCD.


Am 18.11.2009 um 17:00 schrieb <Internet-Drafts@ietf.org>:

> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.
> 
> 
> 	Title           : Making TCP more Robust to Long Connectivity Disruptions (TCP-LCD)
> 	Author(s)       : A. Zimmermann, A. Hannemann
> 	Filename        : draft-ietf-tcpm-tcp-lcd-00.txt
> 	Pages           : 25
> 	Date            : 2009-11-17
> 
> Disruptions in end-to-end path connectivity, which last longer than
> one retransmission timeout cause suboptimal TCP performance.  The
> reason for the performance degradation is that TCP interprets segment
> loss induced by long connectivity disruptions as a sign of
> congestion, resulting in repeated retransmission timer backoffs.
> This leads in turn to a deferred detection of the re-establishment of
> the connection since TCP waits until the next retransmission timeout
> occurs before attempting the retransmission.
> 
> This document proposes a algorithm for making TCP more robust to long
> connectivity disruptions (TCP-LCD).  The memo describes how standard
> ICMP messages can be exploited during timeout-based loss recovery to
> disambiguate true congestion loss from non-congestion loss caused by
> connectivity disruptions.  Moreover, a revert strategy of the
> retransmission timer is specified that enables a more prompt
> detection of whether the connectivity to a previously disconnected
> peer node has been restored or not.  TCP-LCD is a TCP sender-only
> modification that effectively improves TCP performance in presence of
> connectivity disruptions.
> 
> Status of this Memo
> 
> This Internet-Draft is submitted to IETF in full conformance with the
> provisions of BCP 78 and BCP 79.
> 
> Internet-Drafts are working documents of the Internet Engineering
> Task Force (IETF), its areas, and its working groups.  Note that
> other groups may also distribute working documents as Internet-
> Drafts.
> 
> Internet-Drafts are draft documents valid for a maximum of six months
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> 
> The list of current Internet-Drafts can be accessed at
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> 
> The list of Internet-Draft Shadow Directories can be accessed at
> http://www.ietf.org/shadow.html.
> 
> This Internet-Draft will expire on May 21, 2010.
> 
> Copyright Notice
> 
> Copyright (c) 2009 IETF Trust and the persons identified as the
> document authors.  All rights reserved.
> 
> This document is subject to BCP 78 and the IETF Trust's Legal
> Provisions Relating to IETF Documents
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> publication of this document.  Please review these documents
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> include Simplified BSD License text as described in Section 4.e of
> the Trust Legal Provisions and are provided without warranty as
> described in the BSD License.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-tcpm-tcp-lcd-00.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
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> 
> Below is the data which will enable a MIME compliant mail reader
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//
// Dipl.-Inform. Alexander Zimmermann
// Department of Computer Science, Informatik 4
// RWTH Aachen University
// Ahornstr. 55, 52056 Aachen, Germany
// phone: (49-241) 80-21422, fax: (49-241) 80-22220
// email: zimmermann@cs.rwth-aachen.de
// web: http://www.umic-mesh.net
//


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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 Maintenance and Minor Extensions Working Group of the IETF.

	Title		: Early Retransmit for TCP and SCTP
	Author(s)	: M. Allman, K. Avrachenkov, U. Ayesta, E. Blanton, P. Hurtig
	Filename	: draft-ietf-tcpm-early-rexmt-03.txt
	Pages		: 12
	Date		: 2009-11-18
	
This document proposes a new mechanism for TCP and SCTP that can be
    used to recover lost segments when a connection's congestion window
    is small.  The "Early Retransmit" mechanism allows the transport to
    reduce, in certain special circumstances, the number of duplicate
    acknowledgments required to trigger a fast retransmission.  This
    allows the transport to use fast retransmit to recover segment
    losses that would otherwise require a lengthy retransmission
    timeout.

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From wesley.m.eddy@nasa.gov  Mon Nov 23 10:17:37 2009
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From: "Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]" <wesley.m.eddy@nasa.gov>
To: Gregory Lebovitz <gregory.ietf@gmail.com>, Joe Touch <touch@isi.edu>
Date: Mon, 23 Nov 2009 12:17:30 -0600
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draft-touch-tcp-ao-nat should definitely be on the list;
thanks for the reminder :).

So far, there have been a couple of people that said they'd
support this as part of AO, but not very many.  I haven't
seen anything indicating there are people against it yet
either ... it would be good to hear what more people think
about this as part of the AO WGLC that's in-progress.  If
there's reasonable support, it sounds like Joe is willing
to roll it into the AO draft after WGLC.

--
Wes Eddy
MTI Systems


>-----Original Message-----
>From: Gregory Lebovitz [mailto:gregory.ietf@gmail.com]
>Sent: Tuesday, November 17, 2009 4:54 PM
>To: Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]; Joe Touch
>Cc: tcpm@ietf.org
>Subject: Re: [tcpm] working group status 11/17/09
>
>should we also list here the current draft on NAT-Traversal for TCP-AO?
>It's not a WG item, but it is being done as a possible (likely?) textual
>addition to a WG item. comment?
>
>Gregory.
>
>
>On Tue, Nov 17, 2009 at 6:23 AM, Eddy, Wesley M. (GRC-MS00)[ASRC
>AEROSPACE CORP] <wesley.m.eddy@nasa.gov> wrote:
>
>
>	Hello; since TCPM didn't meet in Hiroshima, but we do have
>	several documents in the process of being finished up, and
>	a couple others in the process of starting, here's a quick
>	snapshot of the TCPM working group status and the actions
>	needed on each document; please let us know if we left
>	anything out:
>
>
>	Active WG Items
>	=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>
>	TCP Authentication Option
>	http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt
>	Milestone Target: Proposed Standard in April 2009
>	Action: In WGLC until 11/24/09
>
>	TCP Authentication Option Crypto
>	http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt
>	Milestone Target: Proposed Standard with AO spec
>	Action: In WGLC until 11/24/09
>
>	ICMP Attacks
>	http://www.ietf.org/id/draft-ietf-tcpm-icmp-attacks-06.txt
>	Milestone Target: Informational in July 2009
>	Action: WGLC started 9/1, ended 9/15 ... Fernando Gont to update
>based on WGLC
>
>	Early-Retransmit
>	http://www.ietf.org/id/draft-ietf-tcpm-early-rexmt-02.txt
>	Milestone Target: Experimental in July 2009
>	Action: WGLC finished & authors updated (action with chairs to
>forward to IESG)
>
>	1323bis
>	http://www.ietf.org/id/draft-ietf-tcpm-1323bis-01.txt
>	Milestone Target: Proposed Standard in July 2009
>	Action: Revise & WGLC (action with D. Borman)
>
>	MSS Option
>	http://www.ietf.org/id/draft-ietf-tcpm-tcpmss-02.txt
>	Milestone Target: Proposed Standard in July 2009
>	Action: WGLC started 8/19, ended 9/8 (extended), being updated
>based on several
>	       of the comments (action with D. Borman)
>
>	Urgent Pointer
>	http://www.ietf.org/id/draft-ietf-tcpm-urgent-data-01.txt
>	Milestone Target: Proposed Standard in January 2010
>	Action: Continue WG reviews; authors have indicated it is ready
>for WGLC;
>	       planned to start WGLC after AO WGLC completes
>
>	TCP Security
>	http://www.ietf.org/id/draft-ietf-tcpm-tcp-security-00.txt
>	Milestone Target: BCP in August 2010
>	Action: 00-draft / outline has been under active discussion; but
>not attained
>	         very clear consensus
>	       WG might begin working on recommendations while still
>looking for a
>	         possible compromise on outline (action with Fernando
>Gont)
>
>	Long Connectivity Disruptions
>	http://www.ietf.org/internet-drafts/draft-zimmermann-tcp-lcd-
>02.txt
>	(need to submit WG version)
>	Milestone Target: Experimental in October 2010
>	Action: need WG to read/discuss
>	       authors to submit revision (action with Zimmerman)
>
>	SACK Entry
>	http://www.ietf.org/id/draft-ietf-tcpm-sack-recovery-entry-00.txt
>	Milestone Target: Proposed Standard in October 2010
>	Action: need WG to read/discuss
>	       TBD how this relates to 3517
>
>
>	Other Proposals
>	=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>
>	Persist
>	http://tools.ietf.org/id/draft-ananth-tcpm-persist-01.txt
>	Action: WG Review
>
>	Timestamps
>	http://tools.ietf.org/id/draft-gont-tcpm-tcp-timestamps-01.txt
>	Action: ???
>
>	Parameter Tuning
>	(no I-D yet)
>	Action: continue good discussion on mailing list
>	       presentation given in Stockholm
>
>	TCP Option for Transparent Middlebox Discovery
>	http://www.ietf.org/id/draft-knutsen-tcpm-middlebox-discovery-
>03.txt
>	Action: Needs WG review / discussion
>	       discussion on mailing list has been productive
>
>	NewReno Modification
>	http://www.ietf.org/internet-drafts/draft-nishida-newreno-
>modification-01.txt
>	Action: need WG to read/discuss
>	       presentation given, but WG list not yet polled for adoption
>
>	Recent discussions opened on Rate-Halving
>	http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt
>	Action: mailing list discussion about the history of this, and
>whether
>	       it needs to be revived
>
>	--
>	Wes Eddy
>	MTI Systems
>
>	_______________________________________________
>	tcpm mailing list
>	tcpm@ietf.org
>	https://www.ietf.org/mailman/listinfo/tcpm
>
>
>
>
>
>--
>----
>IETF related email from
>Gregory M. Lebovitz
>Juniper Networks


From lars.eggert@nokia.com  Mon Nov 23 10:50:55 2009
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To: "Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]" <wesley.m.eddy@nasa.gov>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] working group status 11/17/09
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Hi,

without wearing any hats: if it can be added very quickly, i.e., =
O(days), I'd be mildly in favor. Otherwise, let's just ship what we have =
now.

Lars

On 2009-11-23, at 20:17, Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP] =
wrote:

> draft-touch-tcp-ao-nat should definitely be on the list;
> thanks for the reminder :).
>=20
> So far, there have been a couple of people that said they'd
> support this as part of AO, but not very many.  I haven't
> seen anything indicating there are people against it yet
> either ... it would be good to hear what more people think
> about this as part of the AO WGLC that's in-progress.  If
> there's reasonable support, it sounds like Joe is willing
> to roll it into the AO draft after WGLC.
>=20
> --
> Wes Eddy
> MTI Systems
>=20
>=20
>> -----Original Message-----
>> From: Gregory Lebovitz [mailto:gregory.ietf@gmail.com]
>> Sent: Tuesday, November 17, 2009 4:54 PM
>> To: Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]; Joe Touch
>> Cc: tcpm@ietf.org
>> Subject: Re: [tcpm] working group status 11/17/09
>>=20
>> should we also list here the current draft on NAT-Traversal for =
TCP-AO?
>> It's not a WG item, but it is being done as a possible (likely?) =
textual
>> addition to a WG item. comment?
>>=20
>> Gregory.
>>=20
>>=20
>> On Tue, Nov 17, 2009 at 6:23 AM, Eddy, Wesley M. (GRC-MS00)[ASRC
>> AEROSPACE CORP] <wesley.m.eddy@nasa.gov> wrote:
>>=20
>>=20
>> 	Hello; since TCPM didn't meet in Hiroshima, but we do have
>> 	several documents in the process of being finished up, and
>> 	a couple others in the process of starting, here's a quick
>> 	snapshot of the TCPM working group status and the actions
>> 	needed on each document; please let us know if we left
>> 	anything out:
>>=20
>>=20
>> 	Active WG Items
>> 	=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>>=20
>> 	TCP Authentication Option
>> 	http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt
>> 	Milestone Target: Proposed Standard in April 2009
>> 	Action: In WGLC until 11/24/09
>>=20
>> 	TCP Authentication Option Crypto
>> 	http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt
>> 	Milestone Target: Proposed Standard with AO spec
>> 	Action: In WGLC until 11/24/09
>>=20
>> 	ICMP Attacks
>> 	http://www.ietf.org/id/draft-ietf-tcpm-icmp-attacks-06.txt
>> 	Milestone Target: Informational in July 2009
>> 	Action: WGLC started 9/1, ended 9/15 ... Fernando Gont to update
>> based on WGLC
>>=20
>> 	Early-Retransmit
>> 	http://www.ietf.org/id/draft-ietf-tcpm-early-rexmt-02.txt
>> 	Milestone Target: Experimental in July 2009
>> 	Action: WGLC finished & authors updated (action with chairs to
>> forward to IESG)
>>=20
>> 	1323bis
>> 	http://www.ietf.org/id/draft-ietf-tcpm-1323bis-01.txt
>> 	Milestone Target: Proposed Standard in July 2009
>> 	Action: Revise & WGLC (action with D. Borman)
>>=20
>> 	MSS Option
>> 	http://www.ietf.org/id/draft-ietf-tcpm-tcpmss-02.txt
>> 	Milestone Target: Proposed Standard in July 2009
>> 	Action: WGLC started 8/19, ended 9/8 (extended), being updated
>> based on several
>> 	       of the comments (action with D. Borman)
>>=20
>> 	Urgent Pointer
>> 	http://www.ietf.org/id/draft-ietf-tcpm-urgent-data-01.txt
>> 	Milestone Target: Proposed Standard in January 2010
>> 	Action: Continue WG reviews; authors have indicated it is ready
>> for WGLC;
>> 	       planned to start WGLC after AO WGLC completes
>>=20
>> 	TCP Security
>> 	http://www.ietf.org/id/draft-ietf-tcpm-tcp-security-00.txt
>> 	Milestone Target: BCP in August 2010
>> 	Action: 00-draft / outline has been under active discussion; but
>> not attained
>> 	         very clear consensus
>> 	       WG might begin working on recommendations while still
>> looking for a
>> 	         possible compromise on outline (action with Fernando
>> Gont)
>>=20
>> 	Long Connectivity Disruptions
>> 	http://www.ietf.org/internet-drafts/draft-zimmermann-tcp-lcd-
>> 02.txt
>> 	(need to submit WG version)
>> 	Milestone Target: Experimental in October 2010
>> 	Action: need WG to read/discuss
>> 	       authors to submit revision (action with Zimmerman)
>>=20
>> 	SACK Entry
>> 	=
http://www.ietf.org/id/draft-ietf-tcpm-sack-recovery-entry-00.txt
>> 	Milestone Target: Proposed Standard in October 2010
>> 	Action: need WG to read/discuss
>> 	       TBD how this relates to 3517
>>=20
>>=20
>> 	Other Proposals
>> 	=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>>=20
>> 	Persist
>> 	http://tools.ietf.org/id/draft-ananth-tcpm-persist-01.txt
>> 	Action: WG Review
>>=20
>> 	Timestamps
>> 	http://tools.ietf.org/id/draft-gont-tcpm-tcp-timestamps-01.txt
>> 	Action: ???
>>=20
>> 	Parameter Tuning
>> 	(no I-D yet)
>> 	Action: continue good discussion on mailing list
>> 	       presentation given in Stockholm
>>=20
>> 	TCP Option for Transparent Middlebox Discovery
>> 	http://www.ietf.org/id/draft-knutsen-tcpm-middlebox-discovery-
>> 03.txt
>> 	Action: Needs WG review / discussion
>> 	       discussion on mailing list has been productive
>>=20
>> 	NewReno Modification
>> 	http://www.ietf.org/internet-drafts/draft-nishida-newreno-
>> modification-01.txt
>> 	Action: need WG to read/discuss
>> 	       presentation given, but WG list not yet polled for =
adoption
>>=20
>> 	Recent discussions opened on Rate-Halving
>> 	http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt
>> 	Action: mailing list discussion about the history of this, and
>> whether
>> 	       it needs to be revived
>>=20
>> 	--
>> 	Wes Eddy
>> 	MTI Systems
>>=20
>> 	_______________________________________________
>> 	tcpm mailing list
>> 	tcpm@ietf.org
>> 	https://www.ietf.org/mailman/listinfo/tcpm
>>=20
>>=20
>>=20
>>=20
>>=20
>> --
>> ----
>> IETF related email from
>> Gregory M. Lebovitz
>> Juniper Networks
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] working group status 11/17/09
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Lars Eggert wrote:
> Hi,
> 
> without wearing any hats: if it can be added very quickly, i.e.,
> O(days), I'd be mildly in favor. Otherwise, let's just ship what we have
> now.

Once/if I get a go-ahead, that's all it'd take. There isn't much to it.

Joe


> On 2009-11-23, at 20:17, Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP] wrote:
> 
>> draft-touch-tcp-ao-nat should definitely be on the list;
>> thanks for the reminder :).
>>
>> So far, there have been a couple of people that said they'd
>> support this as part of AO, but not very many.  I haven't
>> seen anything indicating there are people against it yet
>> either ... it would be good to hear what more people think
>> about this as part of the AO WGLC that's in-progress.  If
>> there's reasonable support, it sounds like Joe is willing
>> to roll it into the AO draft after WGLC.
>>
>> --
>> Wes Eddy
>> MTI Systems
>>
>>
>>> -----Original Message-----
>>> From: Gregory Lebovitz [mailto:gregory.ietf@gmail.com]
>>> Sent: Tuesday, November 17, 2009 4:54 PM
>>> To: Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]; Joe Touch
>>> Cc: tcpm@ietf.org
>>> Subject: Re: [tcpm] working group status 11/17/09
>>>
>>> should we also list here the current draft on NAT-Traversal for TCP-AO?
>>> It's not a WG item, but it is being done as a possible (likely?) textual
>>> addition to a WG item. comment?
>>>
>>> Gregory.
>>>
>>>
>>> On Tue, Nov 17, 2009 at 6:23 AM, Eddy, Wesley M. (GRC-MS00)[ASRC
>>> AEROSPACE CORP] <wesley.m.eddy@nasa.gov> wrote:
>>>
>>>
>>> 	Hello; since TCPM didn't meet in Hiroshima, but we do have
>>> 	several documents in the process of being finished up, and
>>> 	a couple others in the process of starting, here's a quick
>>> 	snapshot of the TCPM working group status and the actions
>>> 	needed on each document; please let us know if we left
>>> 	anything out:
>>>
>>>
>>> 	Active WG Items
>>> 	===============
>>>
>>> 	TCP Authentication Option
>>> 	http://www.ietf.org/id/draft-ietf-tcpm-tcp-auth-opt-08.txt
>>> 	Milestone Target: Proposed Standard in April 2009
>>> 	Action: In WGLC until 11/24/09
>>>
>>> 	TCP Authentication Option Crypto
>>> 	http://www.ietf.org/id/draft-ietf-tcpm-tcp-ao-crypto-01.txt
>>> 	Milestone Target: Proposed Standard with AO spec
>>> 	Action: In WGLC until 11/24/09
>>>
>>> 	ICMP Attacks
>>> 	http://www.ietf.org/id/draft-ietf-tcpm-icmp-attacks-06.txt
>>> 	Milestone Target: Informational in July 2009
>>> 	Action: WGLC started 9/1, ended 9/15 ... Fernando Gont to update
>>> based on WGLC
>>>
>>> 	Early-Retransmit
>>> 	http://www.ietf.org/id/draft-ietf-tcpm-early-rexmt-02.txt
>>> 	Milestone Target: Experimental in July 2009
>>> 	Action: WGLC finished & authors updated (action with chairs to
>>> forward to IESG)
>>>
>>> 	1323bis
>>> 	http://www.ietf.org/id/draft-ietf-tcpm-1323bis-01.txt
>>> 	Milestone Target: Proposed Standard in July 2009
>>> 	Action: Revise & WGLC (action with D. Borman)
>>>
>>> 	MSS Option
>>> 	http://www.ietf.org/id/draft-ietf-tcpm-tcpmss-02.txt
>>> 	Milestone Target: Proposed Standard in July 2009
>>> 	Action: WGLC started 8/19, ended 9/8 (extended), being updated
>>> based on several
>>> 	       of the comments (action with D. Borman)
>>>
>>> 	Urgent Pointer
>>> 	http://www.ietf.org/id/draft-ietf-tcpm-urgent-data-01.txt
>>> 	Milestone Target: Proposed Standard in January 2010
>>> 	Action: Continue WG reviews; authors have indicated it is ready
>>> for WGLC;
>>> 	       planned to start WGLC after AO WGLC completes
>>>
>>> 	TCP Security
>>> 	http://www.ietf.org/id/draft-ietf-tcpm-tcp-security-00.txt
>>> 	Milestone Target: BCP in August 2010
>>> 	Action: 00-draft / outline has been under active discussion; but
>>> not attained
>>> 	         very clear consensus
>>> 	       WG might begin working on recommendations while still
>>> looking for a
>>> 	         possible compromise on outline (action with Fernando
>>> Gont)
>>>
>>> 	Long Connectivity Disruptions
>>> 	http://www.ietf.org/internet-drafts/draft-zimmermann-tcp-lcd-
>>> 02.txt
>>> 	(need to submit WG version)
>>> 	Milestone Target: Experimental in October 2010
>>> 	Action: need WG to read/discuss
>>> 	       authors to submit revision (action with Zimmerman)
>>>
>>> 	SACK Entry
>>> 	http://www.ietf.org/id/draft-ietf-tcpm-sack-recovery-entry-00.txt
>>> 	Milestone Target: Proposed Standard in October 2010
>>> 	Action: need WG to read/discuss
>>> 	       TBD how this relates to 3517
>>>
>>>
>>> 	Other Proposals
>>> 	===============
>>>
>>> 	Persist
>>> 	http://tools.ietf.org/id/draft-ananth-tcpm-persist-01.txt
>>> 	Action: WG Review
>>>
>>> 	Timestamps
>>> 	http://tools.ietf.org/id/draft-gont-tcpm-tcp-timestamps-01.txt
>>> 	Action: ???
>>>
>>> 	Parameter Tuning
>>> 	(no I-D yet)
>>> 	Action: continue good discussion on mailing list
>>> 	       presentation given in Stockholm
>>>
>>> 	TCP Option for Transparent Middlebox Discovery
>>> 	http://www.ietf.org/id/draft-knutsen-tcpm-middlebox-discovery-
>>> 03.txt
>>> 	Action: Needs WG review / discussion
>>> 	       discussion on mailing list has been productive
>>>
>>> 	NewReno Modification
>>> 	http://www.ietf.org/internet-drafts/draft-nishida-newreno-
>>> modification-01.txt
>>> 	Action: need WG to read/discuss
>>> 	       presentation given, but WG list not yet polled for adoption
>>>
>>> 	Recent discussions opened on Rate-Halving
>>> 	http://tools.ietf.org/id/draft-mathis-tcp-ratehalving-00.txt
>>> 	Action: mailing list discussion about the history of this, and
>>> whether
>>> 	       it needs to be revived
>>>
>>> 	--
>>> 	Wes Eddy
>>> 	MTI Systems
>>>
>>> 	_______________________________________________
>>> 	tcpm mailing list
>>> 	tcpm@ietf.org
>>> 	https://www.ietf.org/mailman/listinfo/tcpm
>>>
>>>
>>>
>>>
>>>
>>> --
>>> ----
>>> IETF related email from
>>> Gregory M. Lebovitz
>>> Juniper Networks
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
> 
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From hkchu@google.com  Mon Nov 23 19:49:07 2009
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Date: Mon, 23 Nov 2009 19:48:56 -0800
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From: Jerry Chu <hkchu@google.com>
To: "Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]" <wesley.m.eddy@nasa.gov>
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Subject: Re: [tcpm] working group status 11/17/09
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<snip>
>
> Parameter Tuning
> (no I-D yet)
> Action: continue good discussion on mailing list
> =A0 =A0 =A0 =A0presentation given in Stockholm

An update on this topic: (my original presentation can be found at
http://www.ietf.org/proceedings/75/slides/tcpm-1.pdf)

After some long off-list discussion between Mark Allman and I on
many gory details, we have agreed on the basic reduction of the
initRTO from 3secs to 1sec, only during 3WHS, only ONCE and
exponentially backed off from there. I hereby request this proposal
to be adopted as a WG item. Please let me know what's needed for
this request to be considered. Mark doesn't seem to think a
separate document is needed. Instead a rev (whatever that means)
to RFC2988 may suffice.

Separately I'd like to propose a moderate increase of the initcwnd
from what is specified in RFC3390 to ~15KB, or 10 segments for
IP/Ethernet.

I understand how to speed up TCP's slow start has been an area of
active research, with many proposals on the table (jump/quick/swift/
fast starts,..., etc). But none of the proposals seems ready to go
forward to become an IETF standard anytime soon. (Some require
router support hence are pretty much out of the question for the
public Internet in the foreseeable future).

In order to better take advantage of today's much faster Internet
sooner, a moderate increase of initcwnd seems a good interim
solution. I will send out more details later. In the mean time if
anyone has an opinion on this please feel free to share.

Jerry

> Wes Eddy
> MTI Systems
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From wwwrun@core3.amsl.com  Tue Nov 24 08:15:57 2009
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The IESG has received a request from the TCP Maintenance and Minor 
Extensions WG (tcpm) to consider the following document:

- 'Early Retransmit for TCP and SCTP '
   <draft-ietf-tcpm-early-rexmt-03.txt> as an Experimental RFC

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send substantive comments to the
ietf@ietf.org mailing lists by 2009-12-08. Exceptionally, 
comments may be sent to iesg@ietf.org instead. In either case, please 
retain the beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-early-rexmt-03.txt


IESG discussion can be tracked via
https://datatracker.ietf.org/public/pidtracker.cgi?command=view_id&dTag=17706&rfc_flag=0


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Hi, Jerry,

Jerry Chu wrote:
...
> Separately I'd like to propose a moderate increase of the initcwnd
> from what is specified in RFC3390 to ~15KB, or 10 segments for
> IP/Ethernet.

The jump from 1 segment to 4KB took quite a bit of analysis, even though
the basis of the change had been widely deployed as an implementation bug.

I suspect it would take similar analysis to ensure that this would be
safe today, including the effect on home gateways, PDAs, etc.

You are also possibly talking about 30 segments, until path MTU
discovery kicks in. So you're going from 4K or 4 segments (whichever is
smaller) to 15K or ?? segments (10? 30?).

Granted this may already be the common case for persistent connections
going active after draining the data, but it's not clear that such
behavior dominates Internet traffic to a point that the current
operational evidence can be used to validate that this is safe.

Joe
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From wesley.m.eddy@nasa.gov  Tue Nov 24 09:55:42 2009
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From: "Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]" <wesley.m.eddy@nasa.gov>
To: Jerry Chu <hkchu@google.com>
Date: Tue, 24 Nov 2009 11:55:33 -0600
Thread-Topic: [tcpm] working group status 11/17/09
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Hi Jerry.  The way we update RFCs is to write more RFCs :).

In the case of the modifications you're talking about, I'm
not sure if we really want to actually update the standards-
track documents or simply write an Experimental document on
parameter tuning.  Given the subject matter, even an
Experimental RFC may take some time and debate to arrive at.

In either case, I think the starting point is to write an
Internet-Draft that describes and discusses the set of
changes in sufficient detail, and then we can poll this
working group on whether or not to adopt one or more work
items for either Experimental or updates to the standards-
track.

Information on writing and submitting Internet-Drafts is on
the IETF webpage here:
http://www.ietf.org/id-info/

--
Wes Eddy
MTI Systems


>-----Original Message-----
>From: Jerry Chu [mailto:hkchu@google.com]
>Sent: Monday, November 23, 2009 10:49 PM
>To: Eddy, Wesley M. (GRC-MS00)[ASRC AEROSPACE CORP]
>Cc: tcpm@ietf.org
>Subject: Re: [tcpm] working group status 11/17/09
>
><snip>
>>
>> Parameter Tuning
>> (no I-D yet)
>> Action: continue good discussion on mailing list
>> =A0 =A0 =A0 =A0presentation given in Stockholm
>
>An update on this topic: (my original presentation can be found at
>http://www.ietf.org/proceedings/75/slides/tcpm-1.pdf)
>
>After some long off-list discussion between Mark Allman and I on
>many gory details, we have agreed on the basic reduction of the
>initRTO from 3secs to 1sec, only during 3WHS, only ONCE and
>exponentially backed off from there. I hereby request this proposal
>to be adopted as a WG item. Please let me know what's needed for
>this request to be considered. Mark doesn't seem to think a
>separate document is needed. Instead a rev (whatever that means)
>to RFC2988 may suffice.
>
>Separately I'd like to propose a moderate increase of the initcwnd
>from what is specified in RFC3390 to ~15KB, or 10 segments for
>IP/Ethernet.
>
>I understand how to speed up TCP's slow start has been an area of
>active research, with many proposals on the table (jump/quick/swift/
>fast starts,..., etc). But none of the proposals seems ready to go
>forward to become an IETF standard anytime soon. (Some require
>router support hence are pretty much out of the question for the
>public Internet in the foreseeable future).
>
>In order to better take advantage of today's much faster Internet
>sooner, a moderate increase of initcwnd seems a good interim
>solution. I will send out more details later. In the mean time if
>anyone has an opinion on this please feel free to share.
>
>Jerry
>
>> Wes Eddy
>> MTI Systems
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>

From hkchu@google.com  Tue Nov 24 18:24:54 2009
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From: Jerry Chu <hkchu@google.com>
To: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] working group status 11/17/09
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Hi Joe,

On Tue, Nov 24, 2009 at 9:43 AM, Joe Touch <touch@isi.edu> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi, Jerry,
>
> Jerry Chu wrote:
> ...
>> Separately I'd like to propose a moderate increase of the initcwnd
>> from what is specified in RFC3390 to ~15KB, or 10 segments for
>> IP/Ethernet.
>
> The jump from 1 segment to 4KB took quite a bit of analysis, even though
> the basis of the change had been widely deployed as an implementation bug.
>
> I suspect it would take similar analysis to ensure that this would be
> safe today, including the effect on home gateways, PDAs, etc.

Understood. (Actually I was involved in the effort to up initcwnd from 1
to 2 in the late 90' when Solaris' SPECWEB numbers blew up on our face
due to the extended delay between Windows clients' delayed acks and
Solaris servers' initcwnd of 1.)

My hope this time is that, rather than relying extensively on simulations, we
can leverage Google's vast network reach and determination to "make the web
faster" (see http://code.google.com/speed/)  to collect data from real
experiments to better understand and devise ways to mitigate the negative
impact, if any, from increasing initcwnd.

>
> You are also possibly talking about 30 segments, until path MTU
> discovery kicks in. So you're going from 4K or 4 segments (whichever is
> smaller) to 15K or ?? segments (10? 30?).

We can work out the exact formula and put a cap on # of segments (e.g., 10).
We would prefer to go higher than 10 but at this moment we still don't
fully understand the negative impact (although small) to warrant the risk.
OTOH we felt 6 or lower is too small to worth the effort. 10 may be a good
balance between benefit vs risk.

>
> Granted this may already be the common case for persistent connections
> going active after draining the data, but it's not clear that such
> behavior dominates Internet traffic to a point that the current
> operational evidence can be used to validate that this is safe.

I think a greater problem is web browser's increasing use of more concurrent
connections (I heard IE8 just up the # of concurrent connections per domain
from 2 to 6), partly to circumvent the small initcwnd each TCP connection
gets. To me this alleviates to some extent my own concern of a larger
initcwnd. But it also speaks to the failure of this community to address the
issue (or at least the perception that initcwnd is too small) sooner to
prevent this bad practice from the browser vendors.

Jerry

>
> Joe
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From touch@ISI.EDU  Tue Nov 24 21:21:11 2009
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Hi, Jerry,

Jerry Chu wrote:
...
>> > The jump from 1 segment to 4KB took quite a bit of analysis, even though
>> > the basis of the change had been widely deployed as an implementation bug.
>> >
>> > I suspect it would take similar analysis to ensure that this would be
>> > safe today, including the effect on home gateways, PDAs, etc.
...
> My hope this time is that, rather than relying extensively on simulations, we
> can leverage Google's vast network reach and determination to "make the web
> faster" (see http://code.google.com/speed/)  to collect data from real
> experiments to better understand and devise ways to mitigate the negative
> impact, if any, from increasing initcwnd.

Experiments are useful. They are also repeatable. To the extent that
real measurements can be used, that's better than simulations.

However, real measurements will probably also show that simultaneous
open never happens, and that more than a few TCP states are never
entered. That's not sufficient to assert safety of a deployed modification.

...
>> > Granted this may already be the common case for persistent connections
>> > going active after draining the data, but it's not clear that such
>> > behavior dominates Internet traffic to a point that the current
>> > operational evidence can be used to validate that this is safe.
> 
> I think a greater problem is web browser's increasing use of more concurrent
> connections (I heard IE8 just up the # of concurrent connections per domain
> from 2 to 6), partly to circumvent the small initcwnd each TCP connection
> gets. To me this alleviates to some extent my own concern of a larger
> initcwnd. But it also speaks to the failure of this community to address the
> issue (or at least the perception that initcwnd is too small) sooner to
> prevent this bad practice from the browser vendors.

Well, it could also be that "bad coding expands to find all leaks".
I.e., just because there's another way of doing something, doesn't make
either way appropriate. We've known about burst-after-idle for a number
of years now, and haven't gotten around to fixing that hole. If we did,
the incentive to using multiple connections to get around small initcwnd
would diminish as well.

There's plenty of useful work to do...

Joe

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From hkchu@google.com  Tue Nov 24 22:05:55 2009
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From: Jerry Chu <hkchu@google.com>
To: Joe Touch <touch@isi.edu>
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On Tue, Nov 24, 2009 at 9:20 PM, Joe Touch <touch@isi.edu> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi, Jerry,
>
> Jerry Chu wrote:
> ...
> >> > The jump from 1 segment to 4KB took quite a bit of analysis, even
> though
> >> > the basis of the change had been widely deployed as an implementation
> bug.
> >> >
> >> > I suspect it would take similar analysis to ensure that this would be
> >> > safe today, including the effect on home gateways, PDAs, etc.
> ...
> > My hope this time is that, rather than relying extensively on
> simulations, we
> > can leverage Google's vast network reach and determination to "make the
> web
> > faster" (see http://code.google.com/speed/)  to collect data from real
> > experiments to better understand and devise ways to mitigate the negative
> > impact, if any, from increasing initcwnd.
>
> Experiments are useful. They are also repeatable. To the extent that
> real measurements can be used, that's better than simulations.
>
> However, real measurements will probably also show that simultaneous
> open never happens, and that more than a few TCP states are never
> entered. That's not sufficient to assert safety of a deployed modification.
>

Correct in general but hopefully protocol correctness won't be an issue
here,
or if still is can most likely be understood or thought up
deterministically.

What's hard in the real, passive measurements that we've conducted is that
the visibility into what really happened in the network is extremely
limited. E.g.,
we could not accurately characterize those cases where an increased initcwnd
is hurting, not by RTT, BDP, or measured b/w, which we typically use to
identify "slow links". My guess is the last hop/access router buffer sizes
may
be the determining factor. But passive measurements just don't give us any
visibility into that.


> ...
> >> > Granted this may already be the common case for persistent connections
> >> > going active after draining the data, but it's not clear that such
> >> > behavior dominates Internet traffic to a point that the current
> >> > operational evidence can be used to validate that this is safe.
> >
> > I think a greater problem is web browser's increasing use of more
> concurrent
> > connections (I heard IE8 just up the # of concurrent connections per
> domain
> > from 2 to 6), partly to circumvent the small initcwnd each TCP connection
> > gets. To me this alleviates to some extent my own concern of a larger
> > initcwnd. But it also speaks to the failure of this community to address
> the
> > issue (or at least the perception that initcwnd is too small) sooner to
> > prevent this bad practice from the browser vendors.
>
> Well, it could also be that "bad coding expands to find all leaks".
> I.e., just because there's another way of doing something, doesn't make
> either way appropriate. We've known about burst-after-idle for a number
> of years now, and haven't gotten around to fixing that hole. If we did,
>

Hmm, I thought all stacks (or at least the two I've worked on) have
slow-start
-after-idle enabled by default...


> the incentive to using multiple connections to get around small initcwnd
> would diminish as well.
>
> There's plenty of useful work to do...
>

Sure there is.

Jerry


>
> Joe
>
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--001636e1f7ae73fc9f04792bd9a9
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On Tue, Nov 24, 2009 at 9:20 PM, Joe Touch <span dir=3D"ltr">&lt;<a href=3D=
"mailto:touch@isi.edu">touch@isi.edu</a>&gt;</span> wrote:<br><div class=3D=
"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"border-left: 1px s=
olid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class=3D"im">-----BEGIN PGP SIGNED MESSAGE-----<br>
Hash: SHA1<br>
<br>
Hi, Jerry,<br>
<br>
Jerry Chu wrote:<br>
...<br>
</div><div class=3D"im">&gt;&gt; &gt; The jump from 1 segment to 4KB took q=
uite a bit of analysis, even though<br>
&gt;&gt; &gt; the basis of the change had been widely deployed as an implem=
entation bug.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; I suspect it would take similar analysis to ensure that this =
would be<br>
&gt;&gt; &gt; safe today, including the effect on home gateways, PDAs, etc.=
<br>
</div>...<br>
<div class=3D"im">&gt; My hope this time is that, rather than relying exten=
sively on simulations, we<br>
&gt; can leverage Google&#39;s vast network reach and determination to &quo=
t;make the web<br>
&gt; faster&quot; (see <a href=3D"http://code.google.com/speed/" target=3D"=
_blank">http://code.google.com/speed/</a>) =A0to collect data from real<br>
&gt; experiments to better understand and devise ways to mitigate the negat=
ive<br>
&gt; impact, if any, from increasing initcwnd.<br>
<br>
</div>Experiments are useful. They are also repeatable. To the extent that<=
br>
real measurements can be used, that&#39;s better than simulations.<br>
<br>
However, real measurements will probably also show that simultaneous<br>
open never happens, and that more than a few TCP states are never<br>
entered. That&#39;s not sufficient to assert safety of a deployed modificat=
ion.<br></blockquote><div><br>Correct in general but hopefully protocol cor=
rectness won&#39;t be an issue here,<br>or if still is can most likely be u=
nderstood or thought up deterministically.<br>
<br>What&#39;s hard in the real, passive measurements that we&#39;ve conduc=
ted is that<br>the visibility into what really happened in the network is e=
xtremely limited. E.g.,<br>we could not accurately characterize those cases=
 where an increased initcwnd<br>
is hurting, not by RTT, BDP, or measured b/w, which we typically use to<br>=
identify &quot;slow links&quot;. My guess is the last hop/access router buf=
fer sizes may<br>be the determining factor. But passive measurements just d=
on&#39;t give us any<br>
visibility into that.<br><br></div><blockquote class=3D"gmail_quote" style=
=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; p=
adding-left: 1ex;">
<br>
...<br>
<div class=3D"im">&gt;&gt; &gt; Granted this may already be the common case=
 for persistent connections<br>
&gt;&gt; &gt; going active after draining the data, but it&#39;s not clear =
that such<br>
&gt;&gt; &gt; behavior dominates Internet traffic to a point that the curre=
nt<br>
&gt;&gt; &gt; operational evidence can be used to validate that this is saf=
e.<br>
&gt;<br>
&gt; I think a greater problem is web browser&#39;s increasing use of more =
concurrent<br>
&gt; connections (I heard IE8 just up the # of concurrent connections per d=
omain<br>
&gt; from 2 to 6), partly to circumvent the small initcwnd each TCP connect=
ion<br>
&gt; gets. To me this alleviates to some extent my own concern of a larger<=
br>
&gt; initcwnd. But it also speaks to the failure of this community to addre=
ss the<br>
&gt; issue (or at least the perception that initcwnd is too small) sooner t=
o<br>
&gt; prevent this bad practice from the browser vendors.<br>
<br>
</div>Well, it could also be that &quot;bad coding expands to find all leak=
s&quot;.<br>
I.e., just because there&#39;s another way of doing something, doesn&#39;t =
make<br>
either way appropriate. We&#39;ve known about burst-after-idle for a number=
<br>
of years now, and haven&#39;t gotten around to fixing that hole. If we did,=
<br></blockquote><div><br>Hmm, I thought all stacks (or at least the two I&=
#39;ve worked on) have slow-start<br>-after-idle enabled by default...<br>
=A0<br></div><blockquote class=3D"gmail_quote" style=3D"border-left: 1px so=
lid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
the incentive to using multiple connections to get around small initcwnd<br=
>
would diminish as well.<br>
<br>
There&#39;s plenty of useful work to do...<br></blockquote><div><br>Sure th=
ere is.<br><br>Jerry<br>=A0<br></div><blockquote class=3D"gmail_quote" styl=
e=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; =
padding-left: 1ex;">

<div class=3D"im"><br>
Joe<br>
<br>
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</blockquote></div><br>

--001636e1f7ae73fc9f04792bd9a9--

From rs@netapp.com  Wed Nov 25 06:36:56 2009
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From: "Scheffenegger, Richard" <rs@netapp.com>
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Hi Jerry,

Have you though about the implications of increasing initcwnd to a=20
level, which exceeds the buffering capacity of your L2 devices?=20
I have observed hosts, which can repeatedly cause heavy packet loss=20
due to droptail in a L2 switch, even with a burst size of as little=20
as 4-6 frames... These hosts are simply capable of wirespeed sending,=20
while the L2 switch is not (or, a common downstream link is slightly=20
saturated).

L2 boxes don't seem to feature that much buffering (ie. 40 frames),=20
thus will enhance the negative effects of any packet burst. (When the=20
L2 buffers are full, and a new session bursts 15+ frames all at once=20
into the L2 box.).=20

IMHO, some basic pacing should be made mandatory when cwnd is mostly=20
empty, to break bursts and deal with slower-than-wirespeed L2 devices=20
in the path...

(Tweaking the IFG, or doing some multiplexing with other streams=20
right at the sender side, might work - but this requires a lower layer
entity to perform that, not TCP all by itself... You might even argue,
that this is an implementation specific tweak, rather than a TCP=20
algrithm all along.

Just my 0.02 EUR.


Richard Scheffenegger
Field Escalation Engineer
NetApp Global Support=20
NetApp
+43 1 3676811 3146 Office (2143 3146 - internal)
+43 676 654 3146 Mobile
www.netapp.com=20
Franz-Klein-Gasse 5
1190 Wien=20

=20

> -----Original Message-----
> From: Jerry Chu [mailto:hkchu@google.com]=20
> Sent: Mittwoch, 25. November 2009 03:25
> To: Joe Touch
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] working group status 11/17/09
>=20
> Hi Joe,
>=20
> On Tue, Nov 24, 2009 at 9:43 AM, Joe Touch <touch@isi.edu> wrote:
> > -----BEGIN PGP SIGNED MESSAGE-----
> > Hash: SHA1
> >
> > Hi, Jerry,
> >
> > Jerry Chu wrote:
> > ...
> >> Separately I'd like to propose a moderate increase of the initcwnd=20
> >> from what is specified in RFC3390 to ~15KB, or 10 segments for=20
> >> IP/Ethernet.
> >
> > The jump from 1 segment to 4KB took quite a bit of analysis, even=20
> > though the basis of the change had been widely deployed as=20
> an implementation bug.
> >
> > I suspect it would take similar analysis to ensure that=20
> this would be=20
> > safe today, including the effect on home gateways, PDAs, etc.
>=20
> Understood. (Actually I was involved in the effort to up=20
> initcwnd from 1 to 2 in the late 90' when Solaris' SPECWEB=20
> numbers blew up on our face due to the extended delay between=20
> Windows clients' delayed acks and Solaris servers' initcwnd of 1.)
>=20
> My hope this time is that, rather than relying extensively on=20
> simulations, we can leverage Google's vast network reach and=20
> determination to "make the web faster" (see=20
> http://code.google.com/speed/)  to collect data from real=20
> experiments to better understand and devise ways to mitigate=20
> the negative impact, if any, from increasing initcwnd.
>=20
> >
> > You are also possibly talking about 30 segments, until path MTU=20
> > discovery kicks in. So you're going from 4K or 4 segments=20
> (whichever=20
> > is
> > smaller) to 15K or ?? segments (10? 30?).
>=20
> We can work out the exact formula and put a cap on # of=20
> segments (e.g., 10).
> We would prefer to go higher than 10 but at this moment we=20
> still don't fully understand the negative impact (although=20
> small) to warrant the risk.
> OTOH we felt 6 or lower is too small to worth the effort. 10=20
> may be a good balance between benefit vs risk.
>=20
> >
> > Granted this may already be the common case for persistent=20
> connections=20
> > going active after draining the data, but it's not clear that such=20
> > behavior dominates Internet traffic to a point that the current=20
> > operational evidence can be used to validate that this is safe.
>=20
> I think a greater problem is web browser's increasing use of=20
> more concurrent connections (I heard IE8 just up the # of=20
> concurrent connections per domain from 2 to 6), partly to=20
> circumvent the small initcwnd each TCP connection gets. To me=20
> this alleviates to some extent my own concern of a larger=20
> initcwnd. But it also speaks to the failure of this community=20
> to address the issue (or at least the perception that=20
> initcwnd is too small) sooner to prevent this bad practice=20
> from the browser vendors.
>=20
> Jerry
>=20
> >
> > Joe
> > -----BEGIN PGP SIGNATURE-----
> > Version: GnuPG v1.4.9 (MingW32)
> >
> > iEYEARECAAYFAksMG0oACgkQE5f5cImnZrv7VwCfWgK4n0yOxFoIwbsBfmKx9pZU
> > 9IgAn3wH6mX26Jn6EPjaUz1J838wT8aw
> > =3DHnjU
> > -----END PGP SIGNATURE-----
> >
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>=20

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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1



Jerry Chu wrote:
...
> Hmm, I thought all stacks (or at least the two I've worked on) have
> slow-start-after-idle enabled by default...

See:

John Heidemann, Katia Obraczka, and Joe Touch, "Modeling the Performance
of HTTP Over Several Transport Protocols" IEEE/ACM Transactions on
Networking, V5, N5, Oct. 1997, pp.616-630.

After idle, the client sends a short GET out fast even with slow-start.

The server receives the GET, thinks that "time since last received"
defines idle, rather than "time since last sent" as it should be. This
is addressed in the paper.

Joe
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Hi Richard,

On Wed, Nov 25, 2009 at 6:36 AM, Scheffenegger, Richard <rs@netapp.com> wro=
te:
>
> Hi Jerry,
>
> Have you though about the implications of increasing initcwnd to a
> level, which exceeds the buffering capacity of your L2 devices?
> I have observed hosts, which can repeatedly cause heavy packet loss
> due to droptail in a L2 switch, even with a burst size of as little
> as 4-6 frames... These hosts are simply capable of wirespeed sending,
> while the L2 switch is not (or, a common downstream link is slightly
> saturated).

Yes, we suspect the limited edge routers/access-links/wireless-gateways
buffer space, rather than congestion collapse to be the main risk a larger
initcwnd may introduce because the former causes pkts to be dropped
hence negatively impacting performance whereas congestion avoidance
remains intact making the latter unlikely to happen, Unfortunately we have
not been able to confirm this due to the limitation of our passive
measurements as I mentioned earlier.

What will be interesting to know is the distribution of the above mentioned
buffer sizes in today's Internet. This will help in answering questions lik=
e
what's the percentage of users that might be affected negatively. If the
percentage is not negligible, how much impact (e.g., just one lost event at
the beginning of a TCP connection?), is there any way to mitigate the
impact,..., etc.

Ultimately we may have to face the choice - does it make sense to slow
down the vast majority of the Internet users just to avoid any negative imp=
act
to a small percentage of users?

>
> L2 boxes don't seem to feature that much buffering (ie. 40 frames),
> thus will enhance the negative effects of any packet burst. (When the
> L2 buffers are full, and a new session bursts 15+ frames all at once
> into the L2 box.).
>
> IMHO, some basic pacing should be made mandatory when cwnd is mostly
> empty, to break bursts and deal with slower-than-wirespeed L2 devices
> in the path...

Yes pacing is one solution to avoid the burst and is required by all the
jump/quick/swift/fast start proposals. Personally I'd prefer to avoid makin=
g
it a requirement for initcwnd of 10 in order to keep the change simple.
The hope is we don't need to go that far unless initcwnd is changed
to a much larger size (e.g., > 16).

Jerry

>
> (Tweaking the IFG, or doing some multiplexing with other streams
> right at the sender side, might work - but this requires a lower layer
> entity to perform that, not TCP all by itself... You might even argue,
> that this is an implementation specific tweak, rather than a TCP
> algrithm all along.
>
> Just my 0.02 EUR.
>
>
> Richard Scheffenegger
> Field Escalation Engineer
> NetApp Global Support
> NetApp
> +43 1 3676811 3146 Office (2143 3146 - internal)
> +43 676 654 3146 Mobile
> www.netapp.com
> Franz-Klein-Gasse 5
> 1190 Wien
>
>
>
>> -----Original Message-----
>> From: Jerry Chu [mailto:hkchu@google.com]
>> Sent: Mittwoch, 25. November 2009 03:25
>> To: Joe Touch
>> Cc: tcpm@ietf.org
>> Subject: Re: [tcpm] working group status 11/17/09
>>
>> Hi Joe,
>>
>> On Tue, Nov 24, 2009 at 9:43 AM, Joe Touch <touch@isi.edu> wrote:
>> > -----BEGIN PGP SIGNED MESSAGE-----
>> > Hash: SHA1
>> >
>> > Hi, Jerry,
>> >
>> > Jerry Chu wrote:
>> > ...
>> >> Separately I'd like to propose a moderate increase of the initcwnd
>> >> from what is specified in RFC3390 to ~15KB, or 10 segments for
>> >> IP/Ethernet.
>> >
>> > The jump from 1 segment to 4KB took quite a bit of analysis, even
>> > though the basis of the change had been widely deployed as
>> an implementation bug.
>> >
>> > I suspect it would take similar analysis to ensure that
>> this would be
>> > safe today, including the effect on home gateways, PDAs, etc.
>>
>> Understood. (Actually I was involved in the effort to up
>> initcwnd from 1 to 2 in the late 90' when Solaris' SPECWEB
>> numbers blew up on our face due to the extended delay between
>> Windows clients' delayed acks and Solaris servers' initcwnd of 1.)
>>
>> My hope this time is that, rather than relying extensively on
>> simulations, we can leverage Google's vast network reach and
>> determination to "make the web faster" (see
>> http://code.google.com/speed/) =A0to collect data from real
>> experiments to better understand and devise ways to mitigate
>> the negative impact, if any, from increasing initcwnd.
>>
>> >
>> > You are also possibly talking about 30 segments, until path MTU
>> > discovery kicks in. So you're going from 4K or 4 segments
>> (whichever
>> > is
>> > smaller) to 15K or ?? segments (10? 30?).
>>
>> We can work out the exact formula and put a cap on # of
>> segments (e.g., 10).
>> We would prefer to go higher than 10 but at this moment we
>> still don't fully understand the negative impact (although
>> small) to warrant the risk.
>> OTOH we felt 6 or lower is too small to worth the effort. 10
>> may be a good balance between benefit vs risk.
>>
>> >
>> > Granted this may already be the common case for persistent
>> connections
>> > going active after draining the data, but it's not clear that such
>> > behavior dominates Internet traffic to a point that the current
>> > operational evidence can be used to validate that this is safe.
>>
>> I think a greater problem is web browser's increasing use of
>> more concurrent connections (I heard IE8 just up the # of
>> concurrent connections per domain from 2 to 6), partly to
>> circumvent the small initcwnd each TCP connection gets. To me
>> this alleviates to some extent my own concern of a larger
>> initcwnd. But it also speaks to the failure of this community
>> to address the issue (or at least the perception that
>> initcwnd is too small) sooner to prevent this bad practice
>> from the browser vendors.
>>
>> Jerry
>>
>> >
>> > Joe
>> > -----BEGIN PGP SIGNATURE-----
>> > Version: GnuPG v1.4.9 (MingW32)
>> >
>> > iEYEARECAAYFAksMG0oACgkQE5f5cImnZrv7VwCfWgK4n0yOxFoIwbsBfmKx9pZU
>> > 9IgAn3wH6mX26Jn6EPjaUz1J838wT8aw
>> > =3DHnjU
>> > -----END PGP SIGNATURE-----
>> >
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>

From root@core3.amsl.com  Thu Nov 26 12:30:01 2009
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Subject: [tcpm] I-D Action:draft-ietf-tcpm-urgent-data-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.


	Title           : On the implementation of the TCP urgent mechanism
	Author(s)       : F. Gont, A. Yourtchenko
	Filename        : draft-ietf-tcpm-urgent-data-02.txt
	Pages           : 13
	Date            : 2009-11-26

This document analyzes how current TCP implementations process TCP
urgent indications, and how the behavior of some widely-deployed
middle-boxes affect how urgent indications are processed by end
systems.  This document updates the relevant specifications such that
they accommodate current practice in processing TCP urgent
indications, provides advice to applications that make use of the
urgent mechanism, and raises awareness about the reliability of TCP
urgent indications in the current Internet.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-urgent-data-02.txt

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